resolve merge conflicts of bddb75c03b7494a7148498a06e72a90fc8198132 to qt-qpr1-dev-plus-aosp
Change-Id: I8c694d9ed1d4b4e74efd8aa643d74adb5033a3db
diff --git a/apex/Android.bp b/apex/Android.bp
index 42a620b..73dc264 100644
--- a/apex/Android.bp
+++ b/apex/Android.bp
@@ -41,6 +41,8 @@
// Use a custom AndroidManifest.xml used for API targeting.
androidManifest: ":com.android.media-androidManifest",
+
+ legacy_android10_support: true,
}
apex {
@@ -76,6 +78,8 @@
// Use a custom AndroidManifest.xml used for API targeting.
androidManifest: ":com.android.media.swcodec-androidManifest",
+
+ legacy_android10_support: true,
}
prebuilt_etc {
diff --git a/apex/ld.config.txt b/apex/ld.config.txt
index af8ec06..bd6af83 100644
--- a/apex/ld.config.txt
+++ b/apex/ld.config.txt
@@ -22,6 +22,12 @@
namespace.default.search.paths = /apex/com.android.media.swcodec/${LIB}
namespace.default.asan.search.paths = /apex/com.android.media.swcodec/${LIB}
+# Below lines are required to be able to access libs in APEXes which are
+# actually symlinks to the files under /system/lib. The symlinks exist for
+# bundled APEXes to reduce space.
+namespace.default.permitted.paths = /system/${LIB}
+namespace.default.asan.permitted.paths = /system/${LIB}
+
namespace.default.links = platform
# TODO: replace the following when apex has a way to auto-generate this list
@@ -44,7 +50,7 @@
namespace.platform.asan.search.paths += /apex/com.android.runtime/${LIB}
# /system/lib/libc.so, etc are symlinks to /apex/com.android.lib/lib/bionic/libc.so, etc.
-# Add /apex/... pat to the permitted paths because linker uses realpath(3)
+# Add /apex/... path to the permitted paths because linker uses realpath(3)
# to check the accessibility of the lib. We could add this to search.paths
# instead but that makes the resolution of bionic libs be dependent on
# the order of /system/lib and /apex/... in search.paths. If /apex/...
@@ -131,3 +137,9 @@
# Add a link for libz.so which is llndk on devices where VNDK is not enforced.
namespace.sphal.link.platform.shared_libs += libz.so
+
+# With VNDK APEX, /system/${LIB}/vndk-sp${VNDK_VER} is a symlink to the following.
+# Add /apex/... path to the permitted paths because linker uses realpath(3)
+# to check the accessibility of the lib.
+namespace.sphal.permitted.paths += /apex/com.android.vndk.${VNDK_APEX_VER}/${LIB}
+namespace.sphal.asan.permitted.paths += /apex/com.android.vndk.${VNDK_APEX_VER}/${LIB}
diff --git a/apex/manifest.json b/apex/manifest.json
index b11187d..3011ee8 100644
--- a/apex/manifest.json
+++ b/apex/manifest.json
@@ -1,4 +1,4 @@
{
"name": "com.android.media",
- "version": 299900000
+ "version": 290000000
}
diff --git a/apex/manifest_codec.json b/apex/manifest_codec.json
index 09c436d..83a5178 100644
--- a/apex/manifest_codec.json
+++ b/apex/manifest_codec.json
@@ -1,4 +1,4 @@
{
"name": "com.android.media.swcodec",
- "version": 299900000
+ "version": 290000000
}
diff --git a/apex/testing/Android.bp b/apex/testing/Android.bp
index 701ced7..376d3e4 100644
--- a/apex/testing/Android.bp
+++ b/apex/testing/Android.bp
@@ -12,18 +12,18 @@
// See the License for the specific language governing permissions and
// limitations under the License.
-apex {
+apex_test {
name: "test_com.android.media",
manifest: "test_manifest.json",
- file_contexts: "com.android.media",
+ file_contexts: ":com.android.media-file_contexts",
defaults: ["com.android.media-defaults"],
installable: false,
}
-apex {
+apex_test {
name: "test_com.android.media.swcodec",
manifest: "test_manifest_codec.json",
- file_contexts: "com.android.media.swcodec",
+ file_contexts: ":com.android.media.swcodec-file_contexts",
defaults: ["com.android.media.swcodec-defaults"],
installable: false,
}
diff --git a/camera/Android.bp b/camera/Android.bp
index 2800595..b288bcf 100644
--- a/camera/Android.bp
+++ b/camera/Android.bp
@@ -86,6 +86,7 @@
"aidl/android/hardware/camera2/ICameraDeviceCallbacks.aidl",
"aidl/android/hardware/camera2/ICameraDeviceUser.aidl",
],
+ path: "aidl",
}
// Extra AIDL files that are used by framework.jar but not libcamera_client
@@ -96,4 +97,5 @@
"aidl/android/hardware/ICamera.aidl",
"aidl/android/hardware/ICameraClient.aidl",
],
+ path: "aidl",
}
diff --git a/camera/cameraserver/Android.bp b/camera/cameraserver/Android.bp
index ecaba3a..320c499 100644
--- a/camera/cameraserver/Android.bp
+++ b/camera/cameraserver/Android.bp
@@ -17,6 +17,10 @@
srcs: ["main_cameraserver.cpp"],
+ header_libs: [
+ "libmedia_headers",
+ ],
+
shared_libs: [
"libcameraservice",
"liblog",
@@ -25,7 +29,6 @@
"libgui",
"libbinder",
"libhidlbase",
- "libhidltransport",
"android.hardware.camera.common@1.0",
"android.hardware.camera.provider@2.4",
"android.hardware.camera.provider@2.5",
diff --git a/camera/include/camera/VendorTagDescriptor.h b/camera/include/camera/VendorTagDescriptor.h
index 6f55890..b2fbf3a 100644
--- a/camera/include/camera/VendorTagDescriptor.h
+++ b/camera/include/camera/VendorTagDescriptor.h
@@ -188,8 +188,8 @@
sp<android::VendorTagDescriptor> *desc /*out*/);
// Parcelable interface
- status_t writeToParcel(Parcel* parcel) const override;
- status_t readFromParcel(const Parcel* parcel) override;
+ status_t writeToParcel(android::Parcel* parcel) const override;
+ status_t readFromParcel(const android::Parcel* parcel) override;
// Returns the number of vendor tags defined.
int getTagCount(metadata_vendor_id_t id) const;
diff --git a/camera/ndk/Android.bp b/camera/ndk/Android.bp
index a2ee65d..d8220eb 100644
--- a/camera/ndk/Android.bp
+++ b/camera/ndk/Android.bp
@@ -107,7 +107,6 @@
],
shared_libs: [
- "libhwbinder",
"libfmq",
"libhidlbase",
"libhardware",
@@ -143,7 +142,6 @@
vendor: true,
srcs: ["ndk_vendor/tests/AImageReaderVendorTest.cpp"],
shared_libs: [
- "libhwbinder",
"libcamera2ndk_vendor",
"libcamera_metadata",
"libmediandk",
diff --git a/camera/ndk/impl/ACameraManager.cpp b/camera/ndk/impl/ACameraManager.cpp
index 9d40fd7..457dea9 100644
--- a/camera/ndk/impl/ACameraManager.cpp
+++ b/camera/ndk/impl/ACameraManager.cpp
@@ -76,6 +76,10 @@
sp<hardware::ICameraService> CameraManagerGlobal::getCameraService() {
Mutex::Autolock _l(mLock);
+ return getCameraServiceLocked();
+}
+
+sp<hardware::ICameraService> CameraManagerGlobal::getCameraServiceLocked() {
if (mCameraService.get() == nullptr) {
if (isCameraServiceDisabled()) {
return mCameraService;
@@ -216,8 +220,12 @@
if (pair.second) {
for (auto& pair : mDeviceStatusMap) {
const String8& cameraId = pair.first;
- int32_t status = pair.second;
-
+ int32_t status = pair.second.status;
+ // Don't send initial callbacks for camera ids which don't support
+ // camera2
+ if (!pair.second.supportsHAL3) {
+ continue;
+ }
sp<AMessage> msg = new AMessage(kWhatSendSingleCallback, mHandler);
ACameraManager_AvailabilityCallback cb = isStatusAvailable(status) ?
callback->onCameraAvailable : callback->onCameraUnavailable;
@@ -236,20 +244,32 @@
mCallbacks.erase(cb);
}
+bool CameraManagerGlobal::supportsCamera2ApiLocked(const String8 &cameraId) {
+ bool camera2Support = false;
+ auto cs = getCameraServiceLocked();
+ binder::Status serviceRet =
+ cs->supportsCameraApi(String16(cameraId),
+ hardware::ICameraService::API_VERSION_2, &camera2Support);
+ if (!serviceRet.isOk()) {
+ ALOGE("%s: supportsCameraApi2Locked() call failed for cameraId %s",
+ __FUNCTION__, cameraId.c_str());
+ return false;
+ }
+ return camera2Support;
+}
+
void CameraManagerGlobal::getCameraIdList(std::vector<String8>* cameraIds) {
// Ensure that we have initialized/refreshed the list of available devices
- auto cs = getCameraService();
Mutex::Autolock _l(mLock);
-
+ // Needed to make sure we're connected to cameraservice
+ getCameraServiceLocked();
for(auto& deviceStatus : mDeviceStatusMap) {
- if (deviceStatus.second == hardware::ICameraServiceListener::STATUS_NOT_PRESENT ||
- deviceStatus.second == hardware::ICameraServiceListener::STATUS_ENUMERATING) {
+ if (deviceStatus.second.status == hardware::ICameraServiceListener::STATUS_NOT_PRESENT ||
+ deviceStatus.second.status ==
+ hardware::ICameraServiceListener::STATUS_ENUMERATING) {
continue;
}
- bool camera2Support = false;
- binder::Status serviceRet = cs->supportsCameraApi(String16(deviceStatus.first),
- hardware::ICameraService::API_VERSION_2, &camera2Support);
- if (!serviceRet.isOk() || !camera2Support) {
+ if (!deviceStatus.second.supportsHAL3) {
continue;
}
cameraIds->push_back(deviceStatus.first);
@@ -377,7 +397,7 @@
bool firstStatus = (mDeviceStatusMap.count(cameraId) == 0);
int32_t oldStatus = firstStatus ?
status : // first status
- mDeviceStatusMap[cameraId];
+ mDeviceStatusMap[cameraId].status;
if (!firstStatus &&
isStatusAvailable(status) == isStatusAvailable(oldStatus)) {
@@ -385,16 +405,19 @@
return;
}
+ bool supportsHAL3 = supportsCamera2ApiLocked(cameraId);
// Iterate through all registered callbacks
- mDeviceStatusMap[cameraId] = status;
- for (auto cb : mCallbacks) {
- sp<AMessage> msg = new AMessage(kWhatSendSingleCallback, mHandler);
- ACameraManager_AvailabilityCallback cbFp = isStatusAvailable(status) ?
- cb.mAvailable : cb.mUnavailable;
- msg->setPointer(kCallbackFpKey, (void *) cbFp);
- msg->setPointer(kContextKey, cb.mContext);
- msg->setString(kCameraIdKey, AString(cameraId));
- msg->post();
+ mDeviceStatusMap[cameraId] = StatusAndHAL3Support(status, supportsHAL3);
+ if (supportsHAL3) {
+ for (auto cb : mCallbacks) {
+ sp<AMessage> msg = new AMessage(kWhatSendSingleCallback, mHandler);
+ ACameraManager_AvailabilityCallback cbFp = isStatusAvailable(status) ?
+ cb.mAvailable : cb.mUnavailable;
+ msg->setPointer(kCallbackFpKey, (void *) cbFp);
+ msg->setPointer(kContextKey, cb.mContext);
+ msg->setString(kCameraIdKey, AString(cameraId));
+ msg->post();
+ }
}
if (status == hardware::ICameraServiceListener::STATUS_NOT_PRESENT) {
mDeviceStatusMap.erase(cameraId);
diff --git a/camera/ndk/impl/ACameraManager.h b/camera/ndk/impl/ACameraManager.h
index 8c1da36..e945ba0 100644
--- a/camera/ndk/impl/ACameraManager.h
+++ b/camera/ndk/impl/ACameraManager.h
@@ -66,9 +66,9 @@
private:
sp<hardware::ICameraService> mCameraService;
- const int kCameraServicePollDelay = 500000; // 0.5s
- const char* kCameraServiceName = "media.camera";
- Mutex mLock;
+ const int kCameraServicePollDelay = 500000; // 0.5s
+ const char* kCameraServiceName = "media.camera";
+ Mutex mLock;
class DeathNotifier : public IBinder::DeathRecipient {
public:
@@ -156,12 +156,14 @@
sp<CallbackHandler> mHandler;
sp<ALooper> mCbLooper; // Looper thread where callbacks actually happen on
+ sp<hardware::ICameraService> getCameraServiceLocked();
void onCameraAccessPrioritiesChanged();
void onStatusChanged(int32_t status, const String8& cameraId);
void onStatusChangedLocked(int32_t status, const String8& cameraId);
// Utils for status
static bool validStatus(int32_t status);
static bool isStatusAvailable(int32_t status);
+ bool supportsCamera2ApiLocked(const String8 &cameraId);
// The sort logic must match the logic in
// libcameraservice/common/CameraProviderManager.cpp::getAPI1CompatibleCameraDeviceIds
@@ -184,8 +186,16 @@
}
};
+ struct StatusAndHAL3Support {
+ int32_t status = hardware::ICameraServiceListener::STATUS_NOT_PRESENT;
+ bool supportsHAL3 = false;
+ StatusAndHAL3Support(int32_t st, bool HAL3support):
+ status(st), supportsHAL3(HAL3support) { };
+ StatusAndHAL3Support() = default;
+ };
+
// Map camera_id -> status
- std::map<String8, int32_t, CameraIdComparator> mDeviceStatusMap;
+ std::map<String8, StatusAndHAL3Support, CameraIdComparator> mDeviceStatusMap;
// For the singleton instance
static Mutex sLock;
diff --git a/cmds/screenrecord/Android.bp b/cmds/screenrecord/Android.bp
index 86476cd..6bdbab1 100644
--- a/cmds/screenrecord/Android.bp
+++ b/cmds/screenrecord/Android.bp
@@ -24,6 +24,10 @@
"Program.cpp",
],
+ header_libs: [
+ "libmediadrm_headers",
+ ],
+
shared_libs: [
"libstagefright",
"libmedia",
diff --git a/cmds/screenrecord/screenrecord.cpp b/cmds/screenrecord/screenrecord.cpp
index 7aa655f..f2a71b3 100644
--- a/cmds/screenrecord/screenrecord.cpp
+++ b/cmds/screenrecord/screenrecord.cpp
@@ -52,7 +52,7 @@
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaMuxer.h>
#include <media/stagefright/PersistentSurface.h>
-#include <media/ICrypto.h>
+#include <mediadrm/ICrypto.h>
#include <media/MediaCodecBuffer.h>
#include "screenrecord.h"
@@ -368,6 +368,7 @@
int64_t startWhenNsec = systemTime(CLOCK_MONOTONIC);
int64_t endWhenNsec = startWhenNsec + seconds_to_nanoseconds(gTimeLimitSec);
DisplayInfo mainDpyInfo;
+ bool firstFrame = true;
assert((rawFp == NULL && muxer != NULL) || (rawFp != NULL && muxer == NULL));
@@ -384,6 +385,11 @@
int64_t ptsUsec;
uint32_t flags;
+ if (firstFrame) {
+ ATRACE_NAME("first_frame");
+ firstFrame = false;
+ }
+
if (systemTime(CLOCK_MONOTONIC) > endWhenNsec) {
if (gVerbose) {
printf("Time limit reached\n");
diff --git a/cmds/stagefright/Android.mk b/cmds/stagefright/Android.mk
index 6eb2e9f..7b447d3 100644
--- a/cmds/stagefright/Android.mk
+++ b/cmds/stagefright/Android.mk
@@ -3,26 +3,27 @@
include $(CLEAR_VARS)
LOCAL_SRC_FILES:= \
+ AudioPlayer.cpp \
stagefright.cpp \
jpeg.cpp \
SineSource.cpp
LOCAL_SHARED_LIBRARIES := \
- libstagefright libmedia libmedia_omx libutils libbinder \
+ libstagefright libmedia libmedia_codeclist libutils libbinder \
libstagefright_foundation libjpeg libui libgui libcutils liblog \
- libhidlbase \
+ libhidlbase libdatasource libaudioclient \
android.hardware.media.omx@1.0 \
LOCAL_C_INCLUDES:= \
frameworks/av/media/libstagefright \
frameworks/av/media/libstagefright/include \
frameworks/native/include/media/openmax \
- external/jpeg \
LOCAL_CFLAGS += -Wno-multichar -Werror -Wall
LOCAL_MODULE_TAGS := optional
+LOCAL_SYSTEM_EXT_MODULE:= true
LOCAL_MODULE:= stagefright
include $(BUILD_EXECUTABLE)
@@ -32,14 +33,16 @@
include $(CLEAR_VARS)
LOCAL_SRC_FILES:= \
+ AudioPlayer.cpp \
SineSource.cpp \
record.cpp
LOCAL_SHARED_LIBRARIES := \
libstagefright libmedia liblog libutils libbinder \
- libstagefright_foundation
+ libstagefright_foundation libdatasource libaudioclient
LOCAL_C_INCLUDES:= \
+ frameworks/av/camera/include \
frameworks/av/media/libstagefright \
frameworks/native/include/media/openmax \
frameworks/native/include/media/hardware
@@ -57,12 +60,12 @@
include $(CLEAR_VARS)
LOCAL_SRC_FILES:= \
- SineSource.cpp \
+ AudioPlayer.cpp \
recordvideo.cpp
LOCAL_SHARED_LIBRARIES := \
libstagefright libmedia liblog libutils libbinder \
- libstagefright_foundation
+ libstagefright_foundation libaudioclient
LOCAL_C_INCLUDES:= \
frameworks/av/media/libstagefright \
@@ -83,12 +86,13 @@
include $(CLEAR_VARS)
LOCAL_SRC_FILES:= \
+ AudioPlayer.cpp \
SineSource.cpp \
audioloop.cpp
LOCAL_SHARED_LIBRARIES := \
libstagefright libmedia liblog libutils libbinder \
- libstagefright_foundation
+ libstagefright_foundation libaudioclient
LOCAL_C_INCLUDES:= \
frameworks/av/media/libstagefright \
@@ -111,7 +115,7 @@
LOCAL_SHARED_LIBRARIES := \
libstagefright liblog libutils libbinder libui libgui \
- libstagefright_foundation libmedia libcutils
+ libstagefright_foundation libmedia libcutils libdatasource
LOCAL_C_INCLUDES:= \
frameworks/av/media/libstagefright \
@@ -133,6 +137,9 @@
codec.cpp \
SimplePlayer.cpp \
+LOCAL_HEADER_LIBRARIES := \
+ libmediadrm_headers \
+
LOCAL_SHARED_LIBRARIES := \
libstagefright liblog libutils libbinder libstagefright_foundation \
libmedia libmedia_omx libaudioclient libui libgui libcutils
@@ -154,22 +161,23 @@
include $(CLEAR_VARS)
LOCAL_SRC_FILES:= \
- filters/argbtorgba.rs \
- filters/nightvision.rs \
- filters/saturation.rs \
+ filters/argbtorgba.rscript \
+ filters/nightvision.rscript \
+ filters/saturation.rscript \
mediafilter.cpp \
+LOCAL_HEADER_LIBRARIES := \
+ libmediadrm_headers \
+
LOCAL_SHARED_LIBRARIES := \
libstagefright \
liblog \
libutils \
libbinder \
libstagefright_foundation \
- libmedia \
libmedia_omx \
libui \
libgui \
- libcutils \
libRScpp \
LOCAL_C_INCLUDES:= \
diff --git a/media/libstagefright/AudioPlayer.cpp b/cmds/stagefright/AudioPlayer.cpp
similarity index 99%
rename from media/libstagefright/AudioPlayer.cpp
rename to cmds/stagefright/AudioPlayer.cpp
index 199b57b..208713d 100644
--- a/media/libstagefright/AudioPlayer.cpp
+++ b/cmds/stagefright/AudioPlayer.cpp
@@ -28,12 +28,13 @@
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/ALookup.h>
#include <media/stagefright/foundation/ALooper.h>
-#include <media/stagefright/AudioPlayer.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/Utils.h>
+#include "AudioPlayer.h"
+
namespace android {
AudioPlayer::AudioPlayer(
diff --git a/media/libstagefright/include/media/stagefright/AudioPlayer.h b/cmds/stagefright/AudioPlayer.h
similarity index 100%
rename from media/libstagefright/include/media/stagefright/AudioPlayer.h
rename to cmds/stagefright/AudioPlayer.h
diff --git a/cmds/stagefright/SimplePlayer.cpp b/cmds/stagefright/SimplePlayer.cpp
index afb7db3..f4b8164 100644
--- a/cmds/stagefright/SimplePlayer.cpp
+++ b/cmds/stagefright/SimplePlayer.cpp
@@ -23,7 +23,7 @@
#include <gui/Surface.h>
#include <media/AudioTrack.h>
-#include <media/ICrypto.h>
+#include <mediadrm/ICrypto.h>
#include <media/IMediaHTTPService.h>
#include <media/MediaCodecBuffer.h>
#include <media/stagefright/foundation/ABuffer.h>
diff --git a/cmds/stagefright/audioloop.cpp b/cmds/stagefright/audioloop.cpp
index d4f2e8d..bd274d8 100644
--- a/cmds/stagefright/audioloop.cpp
+++ b/cmds/stagefright/audioloop.cpp
@@ -29,11 +29,11 @@
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/AMessage.h>
#include <media/stagefright/AMRWriter.h>
-#include <media/stagefright/AudioPlayer.h>
#include <media/stagefright/AudioSource.h>
#include <media/stagefright/MediaCodecSource.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/SimpleDecodingSource.h>
+#include "AudioPlayer.h"
#include "SineSource.h"
using namespace android;
diff --git a/cmds/stagefright/codec.cpp b/cmds/stagefright/codec.cpp
index e5a4337..f2d1c29 100644
--- a/cmds/stagefright/codec.cpp
+++ b/cmds/stagefright/codec.cpp
@@ -23,7 +23,7 @@
#include <binder/IServiceManager.h>
#include <binder/ProcessState.h>
-#include <media/ICrypto.h>
+#include <mediadrm/ICrypto.h>
#include <media/IMediaHTTPService.h>
#include <media/IMediaPlayerService.h>
#include <media/MediaCodecBuffer.h>
diff --git a/cmds/stagefright/filters/argbtorgba.rs b/cmds/stagefright/filters/argbtorgba.rscript
similarity index 100%
rename from cmds/stagefright/filters/argbtorgba.rs
rename to cmds/stagefright/filters/argbtorgba.rscript
diff --git a/cmds/stagefright/filters/nightvision.rs b/cmds/stagefright/filters/nightvision.rscript
similarity index 100%
rename from cmds/stagefright/filters/nightvision.rs
rename to cmds/stagefright/filters/nightvision.rscript
diff --git a/cmds/stagefright/filters/saturation.rs b/cmds/stagefright/filters/saturation.rscript
similarity index 100%
rename from cmds/stagefright/filters/saturation.rs
rename to cmds/stagefright/filters/saturation.rscript
diff --git a/cmds/stagefright/mediafilter.cpp b/cmds/stagefright/mediafilter.cpp
index 2cf6955..66302b0 100644
--- a/cmds/stagefright/mediafilter.cpp
+++ b/cmds/stagefright/mediafilter.cpp
@@ -24,9 +24,9 @@
#include <gui/ISurfaceComposer.h>
#include <gui/SurfaceComposerClient.h>
#include <gui/Surface.h>
-#include <media/ICrypto.h>
#include <media/IMediaHTTPService.h>
#include <media/MediaCodecBuffer.h>
+#include <mediadrm/ICrypto.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/AMessage.h>
diff --git a/cmds/stagefright/record.cpp b/cmds/stagefright/record.cpp
index 95a16f3..37091c4 100644
--- a/cmds/stagefright/record.cpp
+++ b/cmds/stagefright/record.cpp
@@ -17,12 +17,11 @@
#include "SineSource.h"
#include <binder/ProcessState.h>
+#include <datasource/FileSource.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/ALooper.h>
#include <media/stagefright/foundation/AMessage.h>
-#include <media/stagefright/AudioPlayer.h>
#include <media/stagefright/CameraSource.h>
-#include <media/stagefright/FileSource.h>
#include <media/stagefright/MediaBufferGroup.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaCodecSource.h>
@@ -33,6 +32,8 @@
#include <media/stagefright/SimpleDecodingSource.h>
#include <media/MediaPlayerInterface.h>
+#include "AudioPlayer.h"
+
using namespace android;
static const int32_t kAudioBitRate = 12200;
diff --git a/cmds/stagefright/recordvideo.cpp b/cmds/stagefright/recordvideo.cpp
index a63b9b9..01a178e 100644
--- a/cmds/stagefright/recordvideo.cpp
+++ b/cmds/stagefright/recordvideo.cpp
@@ -14,8 +14,6 @@
* limitations under the License.
*/
-#include "SineSource.h"
-
#include <inttypes.h>
#include <sys/types.h>
#include <sys/stat.h>
@@ -25,8 +23,8 @@
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/ALooper.h>
#include <media/stagefright/foundation/AMessage.h>
-#include <media/stagefright/AudioPlayer.h>
#include <media/stagefright/MediaBufferGroup.h>
+#include <media/stagefright/MediaBuffer.h>
#include <media/stagefright/MediaCodecSource.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MetaData.h>
diff --git a/cmds/stagefright/stagefright.cpp b/cmds/stagefright/stagefright.cpp
index bf36be0..02ade94 100644
--- a/cmds/stagefright/stagefright.cpp
+++ b/cmds/stagefright/stagefright.cpp
@@ -31,18 +31,15 @@
#include <binder/IServiceManager.h>
#include <binder/ProcessState.h>
+#include <datasource/DataSourceFactory.h>
#include <media/DataSource.h>
#include <media/MediaSource.h>
-#include <media/ICrypto.h>
#include <media/IMediaHTTPService.h>
#include <media/IMediaPlayerService.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/ALooper.h>
#include <media/stagefright/foundation/AMessage.h>
#include <media/stagefright/foundation/AUtils.h>
-#include "include/NuCachedSource2.h"
-#include <media/stagefright/AudioPlayer.h>
-#include <media/stagefright/DataSourceFactory.h>
#include <media/stagefright/JPEGSource.h>
#include <media/stagefright/InterfaceUtils.h>
#include <media/stagefright/MediaCodec.h>
@@ -69,6 +66,8 @@
#include <android/hardware/media/omx/1.0/IOmx.h>
+#include "AudioPlayer.h"
+
using namespace android;
static long gNumRepetitions;
@@ -305,7 +304,7 @@
seekTimeUs = -1;
if (shouldSeek) {
- seekTimeUs = (rand() * (float)durationUs) / RAND_MAX;
+ seekTimeUs = (rand() * (float)durationUs) / (float)RAND_MAX;
options.setSeekTo(seekTimeUs);
printf("seeking to %" PRId64 " us (%.2f secs)\n",
@@ -1086,7 +1085,7 @@
const char *filename = argv[k];
sp<DataSource> dataSource =
- DataSourceFactory::CreateFromURI(NULL /* httpService */, filename);
+ DataSourceFactory::getInstance()->CreateFromURI(NULL /* httpService */, filename);
if (strncasecmp(filename, "sine:", 5) && dataSource == NULL) {
fprintf(stderr, "Unable to create data source.\n");
diff --git a/cmds/stagefright/stream.cpp b/cmds/stagefright/stream.cpp
index 35bdbc0..22e2ef3 100644
--- a/cmds/stagefright/stream.cpp
+++ b/cmds/stagefright/stream.cpp
@@ -21,6 +21,7 @@
#include <binder/ProcessState.h>
#include <cutils/properties.h> // for property_get
+#include <datasource/DataSourceFactory.h>
#include <media/DataSource.h>
#include <media/IMediaHTTPService.h>
#include <media/IStreamSource.h>
@@ -28,7 +29,6 @@
#include <media/MediaSource.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/AMessage.h>
-#include <media/stagefright/DataSourceFactory.h>
#include <media/stagefright/InterfaceUtils.h>
#include <media/stagefright/MPEG2TSWriter.h>
#include <media/stagefright/MediaExtractor.h>
@@ -164,7 +164,7 @@
: mCurrentBufferIndex(-1),
mCurrentBufferOffset(0) {
sp<DataSource> dataSource =
- DataSourceFactory::CreateFromURI(NULL /* httpService */, filename);
+ DataSourceFactory::getInstance()->CreateFromURI(NULL /* httpService */, filename);
CHECK(dataSource != NULL);
diff --git a/drm/libmediadrm/Android.bp b/drm/libmediadrm/Android.bp
index d6db1d4..52c7438 100644
--- a/drm/libmediadrm/Android.bp
+++ b/drm/libmediadrm/Android.bp
@@ -2,9 +2,16 @@
// libmediadrm
//
-// TODO: change it back to cc_library_shared when MediaPlayer2 switches to
-// using NdkMediaDrm, instead of MediaDrm.java.
-cc_library {
+cc_library_headers {
+ name: "libmediadrm_headers",
+
+ export_include_dirs: [
+ "interface"
+ ],
+
+}
+
+cc_library_shared {
name: "libmediadrm",
srcs: [
@@ -19,14 +26,29 @@
"CryptoHal.cpp",
],
+ local_include_dirs: [
+ "include",
+ "interface"
+ ],
+
+ export_include_dirs: [
+ "include"
+ ],
+
+ header_libs: [
+ "libmedia_headers",
+ ],
+
shared_libs: [
"libbinder",
"libcutils",
"libdl",
"liblog",
+ "libmedia",
"libmediadrmmetrics_lite",
"libmediametrics",
"libmediautils",
+ "libresourcemanagerservice",
"libstagefright_foundation",
"libutils",
"android.hardware.drm@1.0",
@@ -34,7 +56,6 @@
"android.hardware.drm@1.2",
"libhidlallocatorutils",
"libhidlbase",
- "libhidltransport",
],
cflags: [
@@ -52,10 +73,17 @@
"protos/metrics.proto",
],
+ local_include_dirs: [
+ "include"
+ ],
+
proto: {
export_proto_headers: true,
type: "lite",
},
+ header_libs: [
+ "libmedia_headers",
+ ],
shared_libs: [
"android.hardware.drm@1.0",
"android.hardware.drm@1.1",
@@ -83,10 +111,17 @@
"protos/metrics.proto",
],
+ local_include_dirs: [
+ "include"
+ ],
+
proto: {
export_proto_headers: true,
type: "full",
},
+ header_libs: [
+ "libmedia_headers",
+ ],
shared_libs: [
"android.hardware.drm@1.0",
"android.hardware.drm@1.1",
diff --git a/drm/libmediadrm/DrmHal.cpp b/drm/libmediadrm/DrmHal.cpp
index 919f4ee..a2234e6 100644
--- a/drm/libmediadrm/DrmHal.cpp
+++ b/drm/libmediadrm/DrmHal.cpp
@@ -295,38 +295,45 @@
}
}
-
Mutex DrmHal::mLock;
-struct DrmSessionClient : public DrmSessionClientInterface {
- explicit DrmSessionClient(DrmHal* drm) : mDrm(drm) {}
-
- virtual bool reclaimSession(const Vector<uint8_t>& sessionId) {
- sp<DrmHal> drm = mDrm.promote();
- if (drm == NULL) {
- return true;
- }
- status_t err = drm->closeSession(sessionId);
- if (err != OK) {
- return false;
- }
- drm->sendEvent(EventType::SESSION_RECLAIMED,
- toHidlVec(sessionId), hidl_vec<uint8_t>());
+bool DrmHal::DrmSessionClient::reclaimResource() {
+ sp<DrmHal> drm = mDrm.promote();
+ if (drm == NULL) {
return true;
}
+ status_t err = drm->closeSession(mSessionId);
+ if (err != OK) {
+ return false;
+ }
+ drm->sendEvent(EventType::SESSION_RECLAIMED,
+ toHidlVec(mSessionId), hidl_vec<uint8_t>());
+ return true;
+}
-protected:
- virtual ~DrmSessionClient() {}
+String8 DrmHal::DrmSessionClient::getName() {
+ String8 name;
+ sp<DrmHal> drm = mDrm.promote();
+ if (drm == NULL) {
+ name.append("<deleted>");
+ } else if (drm->getPropertyStringInternal(String8("vendor"), name) != OK
+ || name.isEmpty()) {
+ name.append("<Get vendor failed or is empty>");
+ }
+ name.append("[");
+ for (size_t i = 0; i < mSessionId.size(); ++i) {
+ name.appendFormat("%02x", mSessionId[i]);
+ }
+ name.append("]");
+ return name;
+}
-private:
- wp<DrmHal> mDrm;
-
- DISALLOW_EVIL_CONSTRUCTORS(DrmSessionClient);
-};
+DrmHal::DrmSessionClient::~DrmSessionClient() {
+ DrmSessionManager::Instance()->removeSession(mSessionId);
+}
DrmHal::DrmHal()
- : mDrmSessionClient(new DrmSessionClient(this)),
- mFactories(makeDrmFactories()),
+ : mFactories(makeDrmFactories()),
mInitCheck((mFactories.size() == 0) ? ERROR_UNSUPPORTED : NO_INIT) {
}
@@ -335,14 +342,13 @@
auto openSessions = mOpenSessions;
for (size_t i = 0; i < openSessions.size(); i++) {
mLock.unlock();
- closeSession(openSessions[i]);
+ closeSession(openSessions[i]->mSessionId);
mLock.lock();
}
mOpenSessions.clear();
}
DrmHal::~DrmHal() {
- DrmSessionManager::Instance()->removeDrm(mDrmSessionClient);
}
void DrmHal::cleanup() {
@@ -748,9 +754,9 @@
} while (retry);
if (err == OK) {
- DrmSessionManager::Instance()->addSession(getCallingPid(),
- mDrmSessionClient, sessionId);
- mOpenSessions.push(sessionId);
+ sp<DrmSessionClient> client(new DrmSessionClient(this, sessionId));
+ DrmSessionManager::Instance()->addSession(getCallingPid(), client, sessionId);
+ mOpenSessions.push(client);
mMetrics.SetSessionStart(sessionId);
}
@@ -767,7 +773,7 @@
if (status == Status::OK) {
DrmSessionManager::Instance()->removeSession(sessionId);
for (size_t i = 0; i < mOpenSessions.size(); i++) {
- if (mOpenSessions[i] == sessionId) {
+ if (isEqualSessionId(mOpenSessions[i]->mSessionId, sessionId)) {
mOpenSessions.removeAt(i);
break;
}
@@ -895,9 +901,8 @@
status_t DrmHal::provideKeyResponse(Vector<uint8_t> const &sessionId,
Vector<uint8_t> const &response, Vector<uint8_t> &keySetId) {
Mutex::Autolock autoLock(mLock);
- EventTimer<status_t> keyResponseTimer(&mMetrics.mProvideKeyResponseTimeUs);
-
INIT_CHECK();
+ EventTimer<status_t> keyResponseTimer(&mMetrics.mProvideKeyResponseTimeUs);
DrmSessionManager::Instance()->useSession(sessionId);
diff --git a/drm/libmediadrm/DrmSessionManager.cpp b/drm/libmediadrm/DrmSessionManager.cpp
index 375644c..0b927ef 100644
--- a/drm/libmediadrm/DrmSessionManager.cpp
+++ b/drm/libmediadrm/DrmSessionManager.cpp
@@ -21,12 +21,17 @@
#include <binder/IPCThreadState.h>
#include <binder/IProcessInfoService.h>
#include <binder/IServiceManager.h>
-#include <media/stagefright/ProcessInfo.h>
-#include <mediadrm/DrmSessionClientInterface.h>
+#include <cutils/properties.h>
+#include <media/IResourceManagerClient.h>
+#include <media/MediaResource.h>
#include <mediadrm/DrmSessionManager.h>
#include <unistd.h>
#include <utils/String8.h>
+#include <vector>
+
+#include "ResourceManagerService.h"
+
namespace android {
static String8 GetSessionIdString(const Vector<uint8_t> &sessionId) {
@@ -37,6 +42,35 @@
return sessionIdStr;
}
+static std::vector<uint8_t> toStdVec(const Vector<uint8_t> &vector) {
+ const uint8_t *v = vector.array();
+ std::vector<uint8_t> vec(v, v + vector.size());
+ return vec;
+}
+
+static uint64_t toClientId(const sp<IResourceManagerClient>& drm) {
+ return reinterpret_cast<int64_t>(drm.get());
+}
+
+static Vector<MediaResource> toResourceVec(const Vector<uint8_t> &sessionId) {
+ Vector<MediaResource> resources;
+ // use UINT64_MAX to decrement through addition overflow
+ resources.push_back(MediaResource(MediaResource::kDrmSession, toStdVec(sessionId), UINT64_MAX));
+ return resources;
+}
+
+static sp<IResourceManagerService> getResourceManagerService() {
+ if (property_get_bool("persist.device_config.media_native.mediadrmserver", 1)) {
+ return new ResourceManagerService();
+ }
+ sp<IServiceManager> sm = defaultServiceManager();
+ if (sm == NULL) {
+ return NULL;
+ }
+ sp<IBinder> binder = sm->getService(String16("media.resource_manager"));
+ return interface_cast<IResourceManagerService>(binder);
+}
+
bool isEqualSessionId(const Vector<uint8_t> &sessionId1, const Vector<uint8_t> &sessionId2) {
if (sessionId1.size() != sessionId2.size()) {
return false;
@@ -51,189 +85,114 @@
sp<DrmSessionManager> DrmSessionManager::Instance() {
static sp<DrmSessionManager> drmSessionManager = new DrmSessionManager();
+ drmSessionManager->init();
return drmSessionManager;
}
DrmSessionManager::DrmSessionManager()
- : mProcessInfo(new ProcessInfo()),
- mTime(0) {}
+ : DrmSessionManager(getResourceManagerService()) {
+}
-DrmSessionManager::DrmSessionManager(sp<ProcessInfoInterface> processInfo)
- : mProcessInfo(processInfo),
- mTime(0) {}
+DrmSessionManager::DrmSessionManager(const sp<IResourceManagerService> &service)
+ : mService(service),
+ mInitialized(false) {
+ if (mService == NULL) {
+ ALOGE("Failed to init ResourceManagerService");
+ }
+}
-DrmSessionManager::~DrmSessionManager() {}
+DrmSessionManager::~DrmSessionManager() {
+ if (mService != NULL) {
+ IInterface::asBinder(mService)->unlinkToDeath(this);
+ }
+}
-void DrmSessionManager::addSession(
- int pid, const sp<DrmSessionClientInterface>& drm, const Vector<uint8_t> &sessionId) {
- ALOGV("addSession(pid %d, drm %p, sessionId %s)", pid, drm.get(),
+void DrmSessionManager::init() {
+ Mutex::Autolock lock(mLock);
+ if (mInitialized) {
+ return;
+ }
+ mInitialized = true;
+ if (mService != NULL) {
+ IInterface::asBinder(mService)->linkToDeath(this);
+ }
+}
+
+void DrmSessionManager::addSession(int pid,
+ const sp<IResourceManagerClient>& drm, const Vector<uint8_t> &sessionId) {
+ uid_t uid = IPCThreadState::self()->getCallingUid();
+ ALOGV("addSession(pid %d, uid %d, drm %p, sessionId %s)", pid, uid, drm.get(),
GetSessionIdString(sessionId).string());
Mutex::Autolock lock(mLock);
- SessionInfo info;
- info.drm = drm;
- info.sessionId = sessionId;
- info.timeStamp = getTime_l();
- ssize_t index = mSessionMap.indexOfKey(pid);
- if (index < 0) {
- // new pid
- SessionInfos infosForPid;
- infosForPid.push_back(info);
- mSessionMap.add(pid, infosForPid);
- } else {
- mSessionMap.editValueAt(index).push_back(info);
+ if (mService == NULL) {
+ return;
}
+
+ int64_t clientId = toClientId(drm);
+ mSessionMap[toStdVec(sessionId)] = (SessionInfo){pid, uid, clientId};
+ mService->addResource(pid, uid, clientId, drm, toResourceVec(sessionId));
}
void DrmSessionManager::useSession(const Vector<uint8_t> &sessionId) {
ALOGV("useSession(%s)", GetSessionIdString(sessionId).string());
Mutex::Autolock lock(mLock);
- for (size_t i = 0; i < mSessionMap.size(); ++i) {
- SessionInfos& infos = mSessionMap.editValueAt(i);
- for (size_t j = 0; j < infos.size(); ++j) {
- SessionInfo& info = infos.editItemAt(j);
- if (isEqualSessionId(sessionId, info.sessionId)) {
- info.timeStamp = getTime_l();
- return;
- }
- }
+ auto it = mSessionMap.find(toStdVec(sessionId));
+ if (mService == NULL || it == mSessionMap.end()) {
+ return;
}
+
+ auto info = it->second;
+ mService->addResource(info.pid, info.uid, info.clientId, NULL, toResourceVec(sessionId));
}
void DrmSessionManager::removeSession(const Vector<uint8_t> &sessionId) {
ALOGV("removeSession(%s)", GetSessionIdString(sessionId).string());
Mutex::Autolock lock(mLock);
- for (size_t i = 0; i < mSessionMap.size(); ++i) {
- SessionInfos& infos = mSessionMap.editValueAt(i);
- for (size_t j = 0; j < infos.size(); ++j) {
- if (isEqualSessionId(sessionId, infos[j].sessionId)) {
- infos.removeAt(j);
- return;
- }
- }
+ auto it = mSessionMap.find(toStdVec(sessionId));
+ if (mService == NULL || it == mSessionMap.end()) {
+ return;
}
-}
-void DrmSessionManager::removeDrm(const sp<DrmSessionClientInterface>& drm) {
- ALOGV("removeDrm(%p)", drm.get());
-
- Mutex::Autolock lock(mLock);
- bool found = false;
- for (size_t i = 0; i < mSessionMap.size(); ++i) {
- SessionInfos& infos = mSessionMap.editValueAt(i);
- for (size_t j = 0; j < infos.size();) {
- if (infos[j].drm == drm) {
- ALOGV("removed session (%s)", GetSessionIdString(infos[j].sessionId).string());
- j = infos.removeAt(j);
- found = true;
- } else {
- ++j;
- }
- }
- if (found) {
- break;
- }
- }
+ auto info = it->second;
+ mService->removeResource(info.pid, info.clientId, toResourceVec(sessionId));
+ mSessionMap.erase(it);
}
bool DrmSessionManager::reclaimSession(int callingPid) {
ALOGV("reclaimSession(%d)", callingPid);
- sp<DrmSessionClientInterface> drm;
- Vector<uint8_t> sessionId;
- int lowestPriorityPid;
- int lowestPriority;
- {
- Mutex::Autolock lock(mLock);
- int callingPriority;
- if (!mProcessInfo->getPriority(callingPid, &callingPriority)) {
- return false;
- }
- if (!getLowestPriority_l(&lowestPriorityPid, &lowestPriority)) {
- return false;
- }
- if (lowestPriority <= callingPriority) {
- return false;
- }
+ // unlock early because reclaimResource might callback into removeSession
+ mLock.lock();
+ sp<IResourceManagerService> service(mService);
+ mLock.unlock();
- if (!getLeastUsedSession_l(lowestPriorityPid, &drm, &sessionId)) {
- return false;
- }
- }
-
- if (drm == NULL) {
+ if (service == NULL) {
return false;
}
- ALOGV("reclaim session(%s) opened by pid %d",
- GetSessionIdString(sessionId).string(), lowestPriorityPid);
-
- return drm->reclaimSession(sessionId);
+ // cannot update mSessionMap because we do not know which sessionId is reclaimed;
+ // we rely on IResourceManagerClient to removeSession in reclaimResource
+ Vector<uint8_t> dummy;
+ return service->reclaimResource(callingPid, toResourceVec(dummy));
}
-int64_t DrmSessionManager::getTime_l() {
- return mTime++;
+size_t DrmSessionManager::getSessionCount() const {
+ Mutex::Autolock lock(mLock);
+ return mSessionMap.size();
}
-bool DrmSessionManager::getLowestPriority_l(int* lowestPriorityPid, int* lowestPriority) {
- int pid = -1;
- int priority = -1;
- for (size_t i = 0; i < mSessionMap.size(); ++i) {
- if (mSessionMap.valueAt(i).size() == 0) {
- // no opened session by this process.
- continue;
- }
- int tempPid = mSessionMap.keyAt(i);
- int tempPriority;
- if (!mProcessInfo->getPriority(tempPid, &tempPriority)) {
- // shouldn't happen.
- return false;
- }
- if (pid == -1) {
- pid = tempPid;
- priority = tempPriority;
- } else {
- if (tempPriority > priority) {
- pid = tempPid;
- priority = tempPriority;
- }
- }
- }
- if (pid != -1) {
- *lowestPriorityPid = pid;
- *lowestPriority = priority;
- }
- return (pid != -1);
+bool DrmSessionManager::containsSession(const Vector<uint8_t>& sessionId) const {
+ Mutex::Autolock lock(mLock);
+ return mSessionMap.count(toStdVec(sessionId));
}
-bool DrmSessionManager::getLeastUsedSession_l(
- int pid, sp<DrmSessionClientInterface>* drm, Vector<uint8_t>* sessionId) {
- ssize_t index = mSessionMap.indexOfKey(pid);
- if (index < 0) {
- return false;
- }
-
- int leastUsedIndex = -1;
- int64_t minTs = LLONG_MAX;
- const SessionInfos& infos = mSessionMap.valueAt(index);
- for (size_t j = 0; j < infos.size(); ++j) {
- if (leastUsedIndex == -1) {
- leastUsedIndex = j;
- minTs = infos[j].timeStamp;
- } else {
- if (infos[j].timeStamp < minTs) {
- leastUsedIndex = j;
- minTs = infos[j].timeStamp;
- }
- }
- }
- if (leastUsedIndex != -1) {
- *drm = infos[leastUsedIndex].drm;
- *sessionId = infos[leastUsedIndex].sessionId;
- }
- return (leastUsedIndex != -1);
+void DrmSessionManager::binderDied(const wp<IBinder>& /*who*/) {
+ ALOGW("ResourceManagerService died.");
+ Mutex::Autolock lock(mLock);
+ mService.clear();
}
} // namespace android
diff --git a/media/libmedia/include/media/CryptoHal.h b/drm/libmediadrm/include/mediadrm/CryptoHal.h
similarity index 100%
rename from media/libmedia/include/media/CryptoHal.h
rename to drm/libmediadrm/include/mediadrm/CryptoHal.h
diff --git a/media/libmedia/include/media/DrmHal.h b/drm/libmediadrm/include/mediadrm/DrmHal.h
similarity index 93%
rename from media/libmedia/include/media/DrmHal.h
rename to drm/libmediadrm/include/mediadrm/DrmHal.h
index bdf1b30..542d300 100644
--- a/media/libmedia/include/media/DrmHal.h
+++ b/drm/libmediadrm/include/mediadrm/DrmHal.h
@@ -26,8 +26,10 @@
#include <android/hardware/drm/1.2/IDrmPlugin.h>
#include <android/hardware/drm/1.2/IDrmPluginListener.h>
+#include <media/IResourceManagerService.h>
#include <media/MediaAnalyticsItem.h>
#include <mediadrm/DrmMetrics.h>
+#include <mediadrm/DrmSessionManager.h>
#include <mediadrm/IDrm.h>
#include <mediadrm/IDrmClient.h>
#include <utils/threads.h>
@@ -59,6 +61,26 @@
struct DrmHal : public BnDrm,
public IBinder::DeathRecipient,
public IDrmPluginListener_V1_2 {
+
+ struct DrmSessionClient : public BnResourceManagerClient {
+ explicit DrmSessionClient(DrmHal* drm, const Vector<uint8_t>& sessionId)
+ : mSessionId(sessionId),
+ mDrm(drm) {}
+
+ virtual bool reclaimResource();
+ virtual String8 getName();
+
+ const Vector<uint8_t> mSessionId;
+
+ protected:
+ virtual ~DrmSessionClient();
+
+ private:
+ wp<DrmHal> mDrm;
+
+ DISALLOW_EVIL_CONSTRUCTORS(DrmSessionClient);
+ };
+
DrmHal();
virtual ~DrmHal();
@@ -193,8 +215,6 @@
private:
static Mutex mLock;
- sp<DrmSessionClientInterface> mDrmSessionClient;
-
sp<IDrmClient> mListener;
mutable Mutex mEventLock;
mutable Mutex mNotifyLock;
@@ -208,7 +228,7 @@
// Mutable to allow modification within GetPropertyByteArray.
mutable MediaDrmMetrics mMetrics;
- Vector<Vector<uint8_t>> mOpenSessions;
+ Vector<sp<DrmSessionClient>> mOpenSessions;
void closeOpenSessions();
void cleanup();
diff --git a/media/libmedia/include/media/DrmMetrics.h b/drm/libmediadrm/include/mediadrm/DrmMetrics.h
similarity index 100%
rename from media/libmedia/include/media/DrmMetrics.h
rename to drm/libmediadrm/include/mediadrm/DrmMetrics.h
diff --git a/media/libmedia/include/media/DrmPluginPath.h b/drm/libmediadrm/include/mediadrm/DrmPluginPath.h
similarity index 100%
rename from media/libmedia/include/media/DrmPluginPath.h
rename to drm/libmediadrm/include/mediadrm/DrmPluginPath.h
diff --git a/media/libmedia/include/media/DrmSessionClientInterface.h b/drm/libmediadrm/include/mediadrm/DrmSessionClientInterface.h
similarity index 100%
rename from media/libmedia/include/media/DrmSessionClientInterface.h
rename to drm/libmediadrm/include/mediadrm/DrmSessionClientInterface.h
diff --git a/media/libmedia/include/media/DrmSessionManager.h b/drm/libmediadrm/include/mediadrm/DrmSessionManager.h
similarity index 61%
rename from media/libmedia/include/media/DrmSessionManager.h
rename to drm/libmediadrm/include/mediadrm/DrmSessionManager.h
index ba27199..b1ad580 100644
--- a/media/libmedia/include/media/DrmSessionManager.h
+++ b/drm/libmediadrm/include/mediadrm/DrmSessionManager.h
@@ -18,56 +18,61 @@
#define DRM_SESSION_MANAGER_H_
+#include <binder/IBinder.h>
+#include <media/IResourceManagerService.h>
#include <media/stagefright/foundation/ABase.h>
#include <utils/RefBase.h>
#include <utils/KeyedVector.h>
#include <utils/threads.h>
#include <utils/Vector.h>
+#include <map>
+#include <utility>
+#include <vector>
+
namespace android {
class DrmSessionManagerTest;
-struct DrmSessionClientInterface;
-struct ProcessInfoInterface;
+class IResourceManagerClient;
bool isEqualSessionId(const Vector<uint8_t> &sessionId1, const Vector<uint8_t> &sessionId2);
struct SessionInfo {
- sp<DrmSessionClientInterface> drm;
- Vector<uint8_t> sessionId;
- int64_t timeStamp;
+ pid_t pid;
+ uid_t uid;
+ int64_t clientId;
};
-typedef Vector<SessionInfo > SessionInfos;
-typedef KeyedVector<int, SessionInfos > PidSessionInfosMap;
+typedef std::map<std::vector<uint8_t>, SessionInfo> SessionInfoMap;
-struct DrmSessionManager : public RefBase {
+struct DrmSessionManager : public IBinder::DeathRecipient {
static sp<DrmSessionManager> Instance();
DrmSessionManager();
- explicit DrmSessionManager(sp<ProcessInfoInterface> processInfo);
+ explicit DrmSessionManager(const sp<IResourceManagerService> &service);
- void addSession(int pid, const sp<DrmSessionClientInterface>& drm, const Vector<uint8_t>& sessionId);
+ void addSession(int pid, const sp<IResourceManagerClient>& drm, const Vector<uint8_t>& sessionId);
void useSession(const Vector<uint8_t>& sessionId);
void removeSession(const Vector<uint8_t>& sessionId);
- void removeDrm(const sp<DrmSessionClientInterface>& drm);
bool reclaimSession(int callingPid);
+ // sanity check APIs
+ size_t getSessionCount() const;
+ bool containsSession(const Vector<uint8_t>& sessionId) const;
+
+ // implements DeathRecipient
+ virtual void binderDied(const wp<IBinder>& /*who*/);
+
protected:
virtual ~DrmSessionManager();
private:
- friend class DrmSessionManagerTest;
+ void init();
- int64_t getTime_l();
- bool getLowestPriority_l(int* lowestPriorityPid, int* lowestPriority);
- bool getLeastUsedSession_l(
- int pid, sp<DrmSessionClientInterface>* drm, Vector<uint8_t>* sessionId);
-
- sp<ProcessInfoInterface> mProcessInfo;
+ sp<IResourceManagerService> mService;
mutable Mutex mLock;
- PidSessionInfosMap mSessionMap;
- int64_t mTime;
+ SessionInfoMap mSessionMap;
+ bool mInitialized;
DISALLOW_EVIL_CONSTRUCTORS(DrmSessionManager);
};
diff --git a/media/libmedia/include/media/IDrm.h b/drm/libmediadrm/include/mediadrm/IDrm.h
similarity index 100%
rename from media/libmedia/include/media/IDrm.h
rename to drm/libmediadrm/include/mediadrm/IDrm.h
diff --git a/media/libmedia/include/media/IDrmClient.h b/drm/libmediadrm/include/mediadrm/IDrmClient.h
similarity index 100%
rename from media/libmedia/include/media/IDrmClient.h
rename to drm/libmediadrm/include/mediadrm/IDrmClient.h
diff --git a/media/libmedia/include/media/IMediaDrmService.h b/drm/libmediadrm/include/mediadrm/IMediaDrmService.h
similarity index 100%
rename from media/libmedia/include/media/IMediaDrmService.h
rename to drm/libmediadrm/include/mediadrm/IMediaDrmService.h
diff --git a/media/libmedia/include/media/SharedLibrary.h b/drm/libmediadrm/include/mediadrm/SharedLibrary.h
similarity index 100%
rename from media/libmedia/include/media/SharedLibrary.h
rename to drm/libmediadrm/include/mediadrm/SharedLibrary.h
diff --git a/media/libmedia/include/media/ICrypto.h b/drm/libmediadrm/interface/mediadrm/ICrypto.h
similarity index 100%
rename from media/libmedia/include/media/ICrypto.h
rename to drm/libmediadrm/interface/mediadrm/ICrypto.h
diff --git a/drm/libmediadrm/tests/Android.bp b/drm/libmediadrm/tests/Android.bp
index 9e0115e..2e39943 100644
--- a/drm/libmediadrm/tests/Android.bp
+++ b/drm/libmediadrm/tests/Android.bp
@@ -3,8 +3,8 @@
cc_test {
name: "CounterMetric_test",
srcs: ["CounterMetric_test.cpp"],
+ header_libs: ["libmedia_headers"],
shared_libs: ["libmediadrm"],
- include_dirs: ["frameworks/av/include/media"],
cflags: [
"-Werror",
"-Wall",
@@ -14,6 +14,9 @@
cc_test {
name: "DrmMetrics_test",
srcs: ["DrmMetrics_test.cpp"],
+ header_libs: [
+ "libmedia_headers"
+ ],
shared_libs: [
"android.hardware.drm@1.0",
"android.hardware.drm@1.1",
@@ -28,7 +31,7 @@
],
static_libs: ["libgmock"],
include_dirs: [
- "frameworks/av/include/media",
+ "frameworks/av/drm/libmediadrm/include",
],
cflags: [
// Suppress unused parameter and no error options. These cause problems
@@ -40,12 +43,14 @@
cc_test {
name: "EventMetric_test",
srcs: ["EventMetric_test.cpp"],
+ header_libs: [
+ "libmedia_headers"
+ ],
shared_libs: [
"liblog",
"libmediadrm",
"libutils",
],
- include_dirs: ["frameworks/av/include/media"],
cflags: [
"-Werror",
"-Wall",
diff --git a/drm/libmediadrm/tests/CounterMetric_test.cpp b/drm/libmediadrm/tests/CounterMetric_test.cpp
index 6bca0da..c2becb4 100644
--- a/drm/libmediadrm/tests/CounterMetric_test.cpp
+++ b/drm/libmediadrm/tests/CounterMetric_test.cpp
@@ -16,7 +16,7 @@
#include <gtest/gtest.h>
-#include "CounterMetric.h"
+#include <media/CounterMetric.h>
namespace android {
diff --git a/drm/libmediadrm/tests/EventMetric_test.cpp b/drm/libmediadrm/tests/EventMetric_test.cpp
index eb6c4f6..b3c3f62 100644
--- a/drm/libmediadrm/tests/EventMetric_test.cpp
+++ b/drm/libmediadrm/tests/EventMetric_test.cpp
@@ -16,7 +16,7 @@
#include <gtest/gtest.h>
-#include "EventMetric.h"
+#include <media/EventMetric.h>
namespace android {
diff --git a/drm/mediacas/plugins/clearkey/Android.bp b/drm/mediacas/plugins/clearkey/Android.bp
new file mode 100644
index 0000000..0113cb8
--- /dev/null
+++ b/drm/mediacas/plugins/clearkey/Android.bp
@@ -0,0 +1,55 @@
+//
+// Copyright (C) 2017 The Android Open Source Project
+//
+// Licensed under the Apache License, Version 2.0 (the "License");
+// you may not use this file except in compliance with the License.
+// You may obtain a copy of the License at
+//
+// http://www.apache.org/licenses/LICENSE-2.0
+//
+// Unless required by applicable law or agreed to in writing, software
+// distributed under the License is distributed on an "AS IS" BASIS,
+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+// See the License for the specific language governing permissions and
+// limitations under the License.
+//
+
+cc_library_shared {
+ name: "libclearkeycasplugin",
+
+ srcs: [
+ "ClearKeyCasPlugin.cpp",
+ "ClearKeyFetcher.cpp",
+ "ClearKeyLicenseFetcher.cpp",
+ "ClearKeySessionLibrary.cpp",
+ "ecm.cpp",
+ "ecm_generator.cpp",
+ "JsonAssetLoader.cpp",
+ "protos/license_protos.proto",
+ ],
+
+ proprietary: true,
+ relative_install_path: "mediacas",
+
+ shared_libs: [
+ "libutils",
+ "liblog",
+ "libcrypto",
+ "libstagefright_foundation",
+ "libprotobuf-cpp-lite",
+ ],
+
+ header_libs: ["media_plugin_headers"],
+
+ static_libs: ["libjsmn"],
+
+ proto: {
+ type: "full",
+ export_proto_headers: true,
+ },
+
+ include_dirs: [
+ "frameworks/av/include",
+ "frameworks/native/include/media",
+ ],
+}
diff --git a/drm/mediacas/plugins/clearkey/Android.mk b/drm/mediacas/plugins/clearkey/Android.mk
deleted file mode 100644
index 4b139a8..0000000
--- a/drm/mediacas/plugins/clearkey/Android.mk
+++ /dev/null
@@ -1,71 +0,0 @@
-#
-# Copyright (C) 2017 The Android Open Source Project
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-# http://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-#
-LOCAL_PATH:= $(call my-dir)
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES:= \
- ClearKeyCasPlugin.cpp \
- ClearKeyFetcher.cpp \
- ClearKeyLicenseFetcher.cpp \
- ClearKeySessionLibrary.cpp \
- ecm.cpp \
- ecm_generator.cpp \
- JsonAssetLoader.cpp \
- protos/license_protos.proto \
-
-LOCAL_MODULE := libclearkeycasplugin
-
-LOCAL_PROPRIETARY_MODULE := true
-LOCAL_MODULE_RELATIVE_PATH := mediacas
-
-LOCAL_SHARED_LIBRARIES := \
- libutils \
- liblog \
- libcrypto \
- libstagefright_foundation \
- libprotobuf-cpp-lite \
-
-LOCAL_HEADER_LIBRARIES := \
- media_plugin_headers
-
-LOCAL_STATIC_LIBRARIES := \
- libjsmn \
-
-LOCAL_MODULE_CLASS := SHARED_LIBRARIES
-
-LOCAL_PROTOC_OPTIMIZE_TYPE := full
-
-define proto_includes
-$(call local-generated-sources-dir)/proto/$(LOCAL_PATH)
-endef
-
-LOCAL_C_INCLUDES += \
- external/jsmn \
- frameworks/av/include \
- frameworks/native/include/media \
- $(call proto_includes)
-
-LOCAL_EXPORT_C_INCLUDE_DIRS := \
- $(call proto_includes)
-
-LOCAL_MODULE_TAGS := optional
-
-include $(BUILD_SHARED_LIBRARY)
-
-#########################################################################
-# Build unit tests
-
-include $(LOCAL_PATH)/tests/Android.mk
diff --git a/drm/mediacas/plugins/clearkey/ClearKeyCasPlugin.cpp b/drm/mediacas/plugins/clearkey/ClearKeyCasPlugin.cpp
index bf35224..af7c367 100644
--- a/drm/mediacas/plugins/clearkey/ClearKeyCasPlugin.cpp
+++ b/drm/mediacas/plugins/clearkey/ClearKeyCasPlugin.cpp
@@ -97,7 +97,8 @@
///////////////////////////////////////////////////////////////////////////////
ClearKeyCasPlugin::ClearKeyCasPlugin(
void *appData, CasPluginCallback callback)
- : mCallback(callback), mCallbackExt(NULL), mAppData(appData) {
+ : mCallback(callback), mCallbackExt(NULL), mStatusCallback(NULL),
+ mAppData(appData) {
ALOGV("CTOR");
}
@@ -112,6 +113,13 @@
ClearKeySessionLibrary::get()->destroyPlugin(this);
}
+status_t ClearKeyCasPlugin::setStatusCallback(
+ CasPluginStatusCallback callback) {
+ ALOGV("setStatusCallback");
+ mStatusCallback = callback;
+ return OK;
+}
+
status_t ClearKeyCasPlugin::setPrivateData(const CasData &/*data*/) {
ALOGV("setPrivateData");
@@ -135,6 +143,19 @@
return ClearKeySessionLibrary::get()->addSession(this, sessionId);
}
+status_t ClearKeyCasPlugin::openSession(uint32_t intent, uint32_t mode,
+ CasSessionId* sessionId) {
+ ALOGV("openSession with intent=%d, mode=%d", intent, mode);
+ // Echo the received information to the callback.
+ // Clear key plugin doesn't use any event, echo'ing for testing only.
+ if (mStatusCallback != NULL) {
+ mStatusCallback((void*)mAppData, intent, mode);
+ }
+
+ // Clear key plugin doesn't use intent and mode.
+ return ClearKeySessionLibrary::get()->addSession(this, sessionId);
+}
+
status_t ClearKeyCasPlugin::closeSession(const CasSessionId &sessionId) {
ALOGV("closeSession: sessionId=%s", sessionIdToString(sessionId).string());
std::shared_ptr<ClearKeyCasSession> session =
diff --git a/drm/mediacas/plugins/clearkey/ClearKeyCasPlugin.h b/drm/mediacas/plugins/clearkey/ClearKeyCasPlugin.h
index f48d5b1..c6938e6 100644
--- a/drm/mediacas/plugins/clearkey/ClearKeyCasPlugin.h
+++ b/drm/mediacas/plugins/clearkey/ClearKeyCasPlugin.h
@@ -71,11 +71,17 @@
ClearKeyCasPlugin(void *appData, CasPluginCallbackExt callback);
virtual ~ClearKeyCasPlugin();
+ virtual status_t setStatusCallback(
+ CasPluginStatusCallback callback) override;
+
virtual status_t setPrivateData(
const CasData &data) override;
virtual status_t openSession(CasSessionId *sessionId) override;
+ virtual status_t openSession(uint32_t intent, uint32_t mode,
+ CasSessionId *sessionId) override;
+
virtual status_t closeSession(
const CasSessionId &sessionId) override;
@@ -105,6 +111,7 @@
std::unique_ptr<KeyFetcher> mKeyFetcher;
CasPluginCallback mCallback;
CasPluginCallbackExt mCallbackExt;
+ CasPluginStatusCallback mStatusCallback;
void* mAppData;
};
diff --git a/drm/mediacas/plugins/clearkey/ClearKeyFetcher.cpp b/drm/mediacas/plugins/clearkey/ClearKeyFetcher.cpp
index eaa3390..cb69f91 100644
--- a/drm/mediacas/plugins/clearkey/ClearKeyFetcher.cpp
+++ b/drm/mediacas/plugins/clearkey/ClearKeyFetcher.cpp
@@ -89,7 +89,7 @@
// asset_id change. If it sends an EcmContainer with 2 Ecms with different
// asset_ids (old and new) then it might be best to prefetch the Emm.
if ((asset_.id() != 0) && (*asset_id != asset_.id())) {
- ALOGW("Asset_id change from %llu to %" PRIu64, asset_.id(), *asset_id);
+ ALOGW("Asset_id change from %" PRIu64 " to %" PRIu64, asset_.id(), *asset_id);
asset_.Clear();
}
diff --git a/drm/mediacas/plugins/clearkey/ecm.cpp b/drm/mediacas/plugins/clearkey/ecm.cpp
index 9fde13a..b3b5218 100644
--- a/drm/mediacas/plugins/clearkey/ecm.cpp
+++ b/drm/mediacas/plugins/clearkey/ecm.cpp
@@ -17,6 +17,8 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "ecm"
+#include <inttypes.h>
+
#include "ecm.h"
#include "ecm_generator.h"
#include "protos/license_protos.pb.h"
@@ -76,7 +78,7 @@
return status;
}
if (asset.id() != asset_from_emm.id()) {
- ALOGE("Asset_id from Emm (%llu) does not match asset_id from Ecm (%llu).",
+ ALOGE("Asset_id from Emm (%" PRIu64 ") does not match asset_id from Ecm (%" PRIu64 ").",
asset_from_emm.id(), asset.id());
return CLEARKEY_STATUS_INVALID_PARAMETER;
}
diff --git a/drm/mediacas/plugins/clearkey/tests/Android.bp b/drm/mediacas/plugins/clearkey/tests/Android.bp
new file mode 100644
index 0000000..575863c
--- /dev/null
+++ b/drm/mediacas/plugins/clearkey/tests/Android.bp
@@ -0,0 +1,45 @@
+//
+// Copyright (C) 2017 The Android Open Source Project
+//
+// Licensed under the Apache License, Version 2.0 (the "License");
+// you may not use this file except in compliance with the License.
+// You may obtain a copy of the License at
+//
+// http://www.apache.org/licenses/LICENSE-2.0
+//
+// Unless required by applicable law or agreed to in writing, software
+// distributed under the License is distributed on an "AS IS" BASIS,
+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+// See the License for the specific language governing permissions and
+// limitations under the License.
+//
+
+cc_test {
+ name: "ClearKeyFetcherTest",
+
+ srcs: ["ClearKeyFetcherTest.cpp"],
+
+ vendor: true,
+
+ // LOCAL_LDFLAGS is needed here for the test to use the plugin, because
+ // the plugin is not in standard library search path. Without this .so
+ // loading fails at run-time (linking is okay).
+ ldflags: [
+ "-Wl,--rpath,${ORIGIN}/../../../system/vendor/lib/mediacas",
+ "-Wl,--enable-new-dtags",
+ ],
+
+ shared_libs: [
+ "libutils",
+ "libclearkeycasplugin",
+ "libstagefright_foundation",
+ "libprotobuf-cpp-lite",
+ "liblog",
+ ],
+
+ include_dirs: [
+ "frameworks/av/drm/mediacas/plugins/clearkey",
+ "frameworks/av/include",
+ "frameworks/native/include/media",
+ ],
+}
diff --git a/drm/mediacas/plugins/clearkey/tests/Android.mk b/drm/mediacas/plugins/clearkey/tests/Android.mk
deleted file mode 100644
index e1545af..0000000
--- a/drm/mediacas/plugins/clearkey/tests/Android.mk
+++ /dev/null
@@ -1,45 +0,0 @@
-#
-# Copyright (C) 2017 The Android Open Source Project
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-# http://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-#
-LOCAL_PATH:= $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES := \
- ClearKeyFetcherTest.cpp
-
-LOCAL_MODULE := ClearKeyFetcherTest
-LOCAL_VENDOR_MODULE := true
-
-# LOCAL_LDFLAGS is needed here for the test to use the plugin, because
-# the plugin is not in standard library search path. Without this .so
-# loading fails at run-time (linking is okay).
-LOCAL_LDFLAGS := \
- -Wl,--rpath,\$${ORIGIN}/../../../system/vendor/lib/mediacas -Wl,--enable-new-dtags
-
-LOCAL_SHARED_LIBRARIES := \
- libutils libclearkeycasplugin libstagefright_foundation libprotobuf-cpp-lite liblog
-
-LOCAL_C_INCLUDES += \
- $(TOP)/frameworks/av/drm/mediacas/plugins/clearkey \
- $(TOP)/frameworks/av/include \
- $(TOP)/frameworks/native/include/media \
-
-LOCAL_MODULE_TAGS := tests
-
-include $(BUILD_NATIVE_TEST)
-
-
-
diff --git a/drm/mediacas/plugins/mock/Android.bp b/drm/mediacas/plugins/mock/Android.bp
new file mode 100644
index 0000000..e8a3c6f
--- /dev/null
+++ b/drm/mediacas/plugins/mock/Android.bp
@@ -0,0 +1,39 @@
+//
+// Copyright (C) 2017 The Android Open Source Project
+//
+// Licensed under the Apache License, Version 2.0 (the "License");
+// you may not use this file except in compliance with the License.
+// You may obtain a copy of the License at
+//
+// http://www.apache.org/licenses/LICENSE-2.0
+//
+// Unless required by applicable law or agreed to in writing, software
+// distributed under the License is distributed on an "AS IS" BASIS,
+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+// See the License for the specific language governing permissions and
+// limitations under the License.
+//
+
+cc_library_shared {
+ name: "libmockcasplugin",
+
+ srcs: [
+ "MockCasPlugin.cpp",
+ "MockSessionLibrary.cpp",
+ ],
+
+ proprietary: true,
+ relative_install_path: "mediacas",
+
+ shared_libs: [
+ "libutils",
+ "liblog",
+ ],
+
+ header_libs: ["media_plugin_headers"],
+
+ include_dirs: [
+ "frameworks/av/include",
+ "frameworks/native/include/media",
+ ],
+}
diff --git a/drm/mediacas/plugins/mock/Android.mk b/drm/mediacas/plugins/mock/Android.mk
deleted file mode 100644
index a1d61da..0000000
--- a/drm/mediacas/plugins/mock/Android.mk
+++ /dev/null
@@ -1,39 +0,0 @@
-#
-# Copyright (C) 2017 The Android Open Source Project
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-# http://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-#
-LOCAL_PATH:= $(call my-dir)
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES:= \
- MockCasPlugin.cpp \
- MockSessionLibrary.cpp \
-
-LOCAL_MODULE := libmockcasplugin
-
-LOCAL_PROPRIETARY_MODULE := true
-LOCAL_MODULE_RELATIVE_PATH := mediacas
-
-LOCAL_SHARED_LIBRARIES := \
- libutils liblog
-
-LOCAL_HEADER_LIBRARIES := media_plugin_headers
-
-LOCAL_C_INCLUDES += \
- $(TOP)/frameworks/av/include \
- $(TOP)/frameworks/native/include/media \
-
-LOCAL_MODULE_TAGS := optional
-
-include $(BUILD_SHARED_LIBRARY)
diff --git a/drm/mediacas/plugins/mock/MockCasPlugin.cpp b/drm/mediacas/plugins/mock/MockCasPlugin.cpp
index 2964791..f8bab0a 100644
--- a/drm/mediacas/plugins/mock/MockCasPlugin.cpp
+++ b/drm/mediacas/plugins/mock/MockCasPlugin.cpp
@@ -111,6 +111,12 @@
MockSessionLibrary::get()->destroyPlugin(this);
}
+status_t MockCasPlugin::setStatusCallback(
+ CasPluginStatusCallback /*callback*/) {
+ ALOGV("setStatusCallback");
+ return OK;
+}
+
status_t MockCasPlugin::setPrivateData(const CasData& /*data*/) {
ALOGV("setPrivateData");
return OK;
@@ -121,6 +127,13 @@
return MockSessionLibrary::get()->addSession(this, sessionId);
}
+status_t MockCasPlugin::openSession(uint32_t intent, uint32_t mode,
+ CasSessionId* sessionId) {
+ ALOGV("openSession with intent=%d, mode=%d", intent, mode);
+ // Clear key plugin doesn't use intent and mode.
+ return MockSessionLibrary::get()->addSession(this, sessionId);
+}
+
status_t MockCasPlugin::closeSession(const CasSessionId &sessionId) {
ALOGV("closeSession: sessionId=%s", arrayToString(sessionId).string());
Mutex::Autolock lock(mLock);
diff --git a/drm/mediacas/plugins/mock/MockCasPlugin.h b/drm/mediacas/plugins/mock/MockCasPlugin.h
index 74b540c..660fd44 100644
--- a/drm/mediacas/plugins/mock/MockCasPlugin.h
+++ b/drm/mediacas/plugins/mock/MockCasPlugin.h
@@ -65,11 +65,17 @@
MockCasPlugin();
virtual ~MockCasPlugin();
+ virtual status_t setStatusCallback(
+ CasPluginStatusCallback callback) override;
+
virtual status_t setPrivateData(
const CasData &data) override;
virtual status_t openSession(CasSessionId *sessionId) override;
+ virtual status_t openSession(uint32_t intent, uint32_t mode,
+ CasSessionId *sessionId) override;
+
virtual status_t closeSession(
const CasSessionId &sessionId) override;
diff --git a/drm/mediadrm/plugins/clearkey/hidl/Android.bp b/drm/mediadrm/plugins/clearkey/hidl/Android.bp
index e91e918..a153ce2 100644
--- a/drm/mediadrm/plugins/clearkey/hidl/Android.bp
+++ b/drm/mediadrm/plugins/clearkey/hidl/Android.bp
@@ -48,7 +48,6 @@
"libcrypto",
"libhidlbase",
"libhidlmemory",
- "libhidltransport",
"liblog",
"libprotobuf-cpp-lite",
"libutils",
diff --git a/drm/mediadrm/plugins/clearkey/hidl/CryptoPlugin.cpp b/drm/mediadrm/plugins/clearkey/hidl/CryptoPlugin.cpp
index 3ecf6d5..f164f28 100644
--- a/drm/mediadrm/plugins/clearkey/hidl/CryptoPlugin.cpp
+++ b/drm/mediadrm/plugins/clearkey/hidl/CryptoPlugin.cpp
@@ -136,8 +136,6 @@
return Void();
}
- base = static_cast<uint8_t *>(static_cast<void *>(destBase->getPointer()));
-
if (destBuffer.offset + destBuffer.size > destBase->getSize()) {
_hidl_cb(Status_V1_2::ERROR_DRM_FRAME_TOO_LARGE, 0, "invalid buffer size");
return Void();
diff --git a/drm/mediadrm/plugins/clearkey/hidl/DrmPlugin.cpp b/drm/mediadrm/plugins/clearkey/hidl/DrmPlugin.cpp
index aab475e..d74bc53 100644
--- a/drm/mediadrm/plugins/clearkey/hidl/DrmPlugin.cpp
+++ b/drm/mediadrm/plugins/clearkey/hidl/DrmPlugin.cpp
@@ -797,61 +797,37 @@
}
Return<Status> DrmPlugin::releaseSecureStops(const SecureStopRelease& ssRelease) {
- // OpaqueData starts with 4 byte decimal integer string
- const size_t kFourBytesOffset = 4;
- if (ssRelease.opaqueData.size() < kFourBytesOffset) {
- ALOGE("Invalid secureStopRelease length");
+ if (ssRelease.opaqueData.size() == 0) {
return Status::BAD_VALUE;
}
Status status = Status::OK;
std::vector<uint8_t> input = toVector(ssRelease.opaqueData);
- if (input.size() < kSecureStopIdSize + kFourBytesOffset) {
- // The minimum size of SecureStopRelease has to contain
- // a 4 bytes count and one secureStop id
- ALOGE("Total size of secureStops is too short");
- return Status::BAD_VALUE;
- }
-
// The format of opaqueData is shared between the server
// and the drm service. The clearkey implementation consists of:
// count - number of secure stops
// list of fixed length secure stops
size_t countBufferSize = sizeof(uint32_t);
- if (input.size() < countBufferSize) {
- // SafetyNet logging
- android_errorWriteLog(0x534e4554, "144766455");
- return Status::BAD_VALUE;
- }
uint32_t count = 0;
sscanf(reinterpret_cast<char*>(input.data()), "%04" PRIu32, &count);
// Avoid divide by 0 below.
if (count == 0) {
- ALOGE("Invalid 0 secureStop count");
return Status::BAD_VALUE;
}
- // Computes the fixed length secureStop size
- size_t secureStopSize = (input.size() - kFourBytesOffset) / count;
- if (secureStopSize < kSecureStopIdSize) {
- // A valid secureStop contains the id plus data
- ALOGE("Invalid secureStop size");
- return Status::BAD_VALUE;
- }
- uint8_t* buffer = new uint8_t[secureStopSize];
- size_t offset = kFourBytesOffset; // skip the count
+ size_t secureStopSize = (input.size() - countBufferSize) / count;
+ uint8_t buffer[secureStopSize];
+ size_t offset = countBufferSize; // skip the count
for (size_t i = 0; i < count; ++i, offset += secureStopSize) {
memcpy(buffer, input.data() + offset, secureStopSize);
-
- // A secureStop contains id+data, we only use the id for removal
std::vector<uint8_t> id(buffer, buffer + kSecureStopIdSize);
+
status = removeSecureStop(toHidlVec(id));
if (Status::OK != status) break;
}
- delete[] buffer;
return status;
}
diff --git a/drm/mediadrm/plugins/clearkey/hidl/serviceLazy.cpp b/drm/mediadrm/plugins/clearkey/hidl/serviceLazy.cpp
index 99fd883..a510487 100644
--- a/drm/mediadrm/plugins/clearkey/hidl/serviceLazy.cpp
+++ b/drm/mediadrm/plugins/clearkey/hidl/serviceLazy.cpp
@@ -38,7 +38,7 @@
configureRpcThreadpool(8, true /* callerWillJoin */);
// Setup hwbinder service
- LazyServiceRegistrar serviceRegistrar;
+ auto serviceRegistrar = LazyServiceRegistrar::getInstance();
// Setup hwbinder service
CHECK_EQ(serviceRegistrar.registerService(drmFactory, "clearkey"), android::NO_ERROR)
diff --git a/include/camera b/include/camera
deleted file mode 120000
index 00848e3..0000000
--- a/include/camera
+++ /dev/null
@@ -1 +0,0 @@
-../camera/include/camera/
\ No newline at end of file
diff --git a/include/cpustats b/include/cpustats
deleted file mode 120000
index 4a02d41..0000000
--- a/include/cpustats
+++ /dev/null
@@ -1 +0,0 @@
-../media/libcpustats/include/cpustats/
\ No newline at end of file
diff --git a/include/drm/drm_framework_common.h b/include/drm/drm_framework_common.h
index d75f71c..d5f3ba2 100644
--- a/include/drm/drm_framework_common.h
+++ b/include/drm/drm_framework_common.h
@@ -317,14 +317,6 @@
~DecryptHandle() {
delete decryptInfo; decryptInfo = NULL;
}
-
- bool operator<(const DecryptHandle& handle) const {
- return (decryptId < handle.decryptId);
- }
-
- bool operator==(const DecryptHandle& handle) const {
- return (decryptId == handle.decryptId);
- }
};
};
diff --git a/include/media/AVSyncSettings.h b/include/media/AVSyncSettings.h
deleted file mode 120000
index bbe211f..0000000
--- a/include/media/AVSyncSettings.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/AVSyncSettings.h
\ No newline at end of file
diff --git a/include/media/AudioAttributes.h b/include/media/AudioAttributes.h
deleted file mode 120000
index 27ba471..0000000
--- a/include/media/AudioAttributes.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/AudioAttributes.h
\ No newline at end of file
diff --git a/include/media/AudioBufferProvider.h b/include/media/AudioBufferProvider.h
deleted file mode 120000
index c4d6e79..0000000
--- a/include/media/AudioBufferProvider.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/AudioBufferProvider.h
\ No newline at end of file
diff --git a/include/media/AudioClient.h b/include/media/AudioClient.h
deleted file mode 120000
index a0530e4..0000000
--- a/include/media/AudioClient.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/AudioClient.h
\ No newline at end of file
diff --git a/include/media/AudioCommonTypes.h b/include/media/AudioCommonTypes.h
deleted file mode 120000
index ae7c99a..0000000
--- a/include/media/AudioCommonTypes.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/AudioCommonTypes.h
\ No newline at end of file
diff --git a/include/media/AudioEffect.h b/include/media/AudioEffect.h
deleted file mode 120000
index bf52955..0000000
--- a/include/media/AudioEffect.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/AudioEffect.h
\ No newline at end of file
diff --git a/include/media/AudioIoDescriptor.h b/include/media/AudioIoDescriptor.h
deleted file mode 120000
index 68f54c9..0000000
--- a/include/media/AudioIoDescriptor.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/AudioIoDescriptor.h
\ No newline at end of file
diff --git a/include/media/AudioMixer.h b/include/media/AudioMixer.h
deleted file mode 120000
index de839c6..0000000
--- a/include/media/AudioMixer.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/AudioMixer.h
\ No newline at end of file
diff --git a/include/media/AudioParameter.h b/include/media/AudioParameter.h
deleted file mode 120000
index a5889e5..0000000
--- a/include/media/AudioParameter.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/AudioParameter.h
\ No newline at end of file
diff --git a/include/media/AudioPolicy.h b/include/media/AudioPolicy.h
deleted file mode 120000
index dd4cd53..0000000
--- a/include/media/AudioPolicy.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/AudioPolicy.h
\ No newline at end of file
diff --git a/include/media/AudioProductStrategy.h b/include/media/AudioProductStrategy.h
deleted file mode 120000
index 6bfaf11..0000000
--- a/include/media/AudioProductStrategy.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/AudioProductStrategy.h
\ No newline at end of file
diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h
deleted file mode 120000
index 7939dd3..0000000
--- a/include/media/AudioRecord.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/AudioRecord.h
\ No newline at end of file
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
deleted file mode 120000
index 9fad2b7..0000000
--- a/include/media/AudioSystem.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/AudioSystem.h
\ No newline at end of file
diff --git a/include/media/AudioTimestamp.h b/include/media/AudioTimestamp.h
deleted file mode 120000
index b6b9278..0000000
--- a/include/media/AudioTimestamp.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/AudioTimestamp.h
\ No newline at end of file
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
deleted file mode 120000
index 303bfcd..0000000
--- a/include/media/AudioTrack.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/AudioTrack.h
\ No newline at end of file
diff --git a/include/media/AudioVolumeGroup.h b/include/media/AudioVolumeGroup.h
deleted file mode 120000
index d6f1c99..0000000
--- a/include/media/AudioVolumeGroup.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/AudioVolumeGroup.h
\ No newline at end of file
diff --git a/include/media/BufferProviders.h b/include/media/BufferProviders.h
deleted file mode 120000
index 779bb15..0000000
--- a/include/media/BufferProviders.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/BufferProviders.h
\ No newline at end of file
diff --git a/include/media/BufferingSettings.h b/include/media/BufferingSettings.h
deleted file mode 120000
index 409203f..0000000
--- a/include/media/BufferingSettings.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/BufferingSettings.h
\ No newline at end of file
diff --git a/include/media/CharacterEncodingDetector.h b/include/media/CharacterEncodingDetector.h
deleted file mode 120000
index 2b28387..0000000
--- a/include/media/CharacterEncodingDetector.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/CharacterEncodingDetector.h
\ No newline at end of file
diff --git a/include/media/CounterMetric.h b/include/media/CounterMetric.h
deleted file mode 120000
index baba043..0000000
--- a/include/media/CounterMetric.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/CounterMetric.h
\ No newline at end of file
diff --git a/include/media/EventLog.h b/include/media/EventLog.h
deleted file mode 120000
index 9b2c4bf..0000000
--- a/include/media/EventLog.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/utils/include/mediautils/EventLog.h
\ No newline at end of file
diff --git a/include/media/EventMetric.h b/include/media/EventMetric.h
deleted file mode 120000
index 5707d9a..0000000
--- a/include/media/EventMetric.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/EventMetric.h
\ No newline at end of file
diff --git a/include/media/ExtendedAudioBufferProvider.h b/include/media/ExtendedAudioBufferProvider.h
deleted file mode 120000
index d653cc3..0000000
--- a/include/media/ExtendedAudioBufferProvider.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/ExtendedAudioBufferProvider.h
\ No newline at end of file
diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h
deleted file mode 120000
index ef6f5be..0000000
--- a/include/media/IAudioFlinger.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/IAudioFlinger.h
\ No newline at end of file
diff --git a/include/media/IAudioFlingerClient.h b/include/media/IAudioFlingerClient.h
deleted file mode 120000
index dc481e8..0000000
--- a/include/media/IAudioFlingerClient.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/IAudioFlingerClient.h
\ No newline at end of file
diff --git a/include/media/IAudioPolicyService.h b/include/media/IAudioPolicyService.h
deleted file mode 120000
index 08101fc..0000000
--- a/include/media/IAudioPolicyService.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/IAudioPolicyService.h
\ No newline at end of file
diff --git a/include/media/IAudioPolicyServiceClient.h b/include/media/IAudioPolicyServiceClient.h
deleted file mode 120000
index 0d4b3e7..0000000
--- a/include/media/IAudioPolicyServiceClient.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/IAudioPolicyServiceClient.h
\ No newline at end of file
diff --git a/include/media/IAudioTrack.h b/include/media/IAudioTrack.h
deleted file mode 120000
index 7bab1fd..0000000
--- a/include/media/IAudioTrack.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/IAudioTrack.h
\ No newline at end of file
diff --git a/include/media/IDataSource.h b/include/media/IDataSource.h
deleted file mode 120000
index 41cdd8b..0000000
--- a/include/media/IDataSource.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IDataSource.h
\ No newline at end of file
diff --git a/include/media/IEffect.h b/include/media/IEffect.h
deleted file mode 120000
index 2fb8bfb..0000000
--- a/include/media/IEffect.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/IEffect.h
\ No newline at end of file
diff --git a/include/media/IEffectClient.h b/include/media/IEffectClient.h
deleted file mode 120000
index b4e39cf..0000000
--- a/include/media/IEffectClient.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/IEffectClient.h
\ No newline at end of file
diff --git a/include/media/IMediaCodecList.h b/include/media/IMediaCodecList.h
deleted file mode 120000
index 2186312..0000000
--- a/include/media/IMediaCodecList.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IMediaCodecList.h
\ No newline at end of file
diff --git a/include/media/IMediaDeathNotifier.h b/include/media/IMediaDeathNotifier.h
deleted file mode 120000
index ce3b8f0..0000000
--- a/include/media/IMediaDeathNotifier.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IMediaDeathNotifier.h
\ No newline at end of file
diff --git a/include/media/IMediaExtractor.h b/include/media/IMediaExtractor.h
deleted file mode 120000
index 8708c8c..0000000
--- a/include/media/IMediaExtractor.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IMediaExtractor.h
\ No newline at end of file
diff --git a/include/media/IMediaExtractorService.h b/include/media/IMediaExtractorService.h
deleted file mode 120000
index 3ee9f1e..0000000
--- a/include/media/IMediaExtractorService.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IMediaExtractorService.h
\ No newline at end of file
diff --git a/include/media/IMediaHTTPConnection.h b/include/media/IMediaHTTPConnection.h
deleted file mode 120000
index 0970c15..0000000
--- a/include/media/IMediaHTTPConnection.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IMediaHTTPConnection.h
\ No newline at end of file
diff --git a/include/media/IMediaHTTPService.h b/include/media/IMediaHTTPService.h
deleted file mode 120000
index b90c34f..0000000
--- a/include/media/IMediaHTTPService.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IMediaHTTPService.h
\ No newline at end of file
diff --git a/include/media/IMediaLogService.h b/include/media/IMediaLogService.h
deleted file mode 120000
index 245a29d..0000000
--- a/include/media/IMediaLogService.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IMediaLogService.h
\ No newline at end of file
diff --git a/include/media/IMediaMetadataRetriever.h b/include/media/IMediaMetadataRetriever.h
deleted file mode 120000
index 959df1a..0000000
--- a/include/media/IMediaMetadataRetriever.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IMediaMetadataRetriever.h
\ No newline at end of file
diff --git a/include/media/IMediaPlayer.h b/include/media/IMediaPlayer.h
deleted file mode 120000
index 9414d37..0000000
--- a/include/media/IMediaPlayer.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IMediaPlayer.h
\ No newline at end of file
diff --git a/include/media/IMediaPlayerClient.h b/include/media/IMediaPlayerClient.h
deleted file mode 120000
index b6547ce..0000000
--- a/include/media/IMediaPlayerClient.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IMediaPlayerClient.h
\ No newline at end of file
diff --git a/include/media/IMediaPlayerService.h b/include/media/IMediaPlayerService.h
deleted file mode 120000
index 89c96cd..0000000
--- a/include/media/IMediaPlayerService.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IMediaPlayerService.h
\ No newline at end of file
diff --git a/include/media/IMediaRecorder.h b/include/media/IMediaRecorder.h
deleted file mode 120000
index 57d192c..0000000
--- a/include/media/IMediaRecorder.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IMediaRecorder.h
\ No newline at end of file
diff --git a/include/media/IMediaRecorderClient.h b/include/media/IMediaRecorderClient.h
deleted file mode 120000
index 89f4359..0000000
--- a/include/media/IMediaRecorderClient.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IMediaRecorderClient.h
\ No newline at end of file
diff --git a/include/media/IMediaSource.h b/include/media/IMediaSource.h
deleted file mode 120000
index 1330ad3..0000000
--- a/include/media/IMediaSource.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IMediaSource.h
\ No newline at end of file
diff --git a/include/media/IOMX.h b/include/media/IOMX.h
deleted file mode 120000
index 6d5b375..0000000
--- a/include/media/IOMX.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IOMX.h
\ No newline at end of file
diff --git a/include/media/IRemoteDisplay.h b/include/media/IRemoteDisplay.h
deleted file mode 120000
index 4b0cf10..0000000
--- a/include/media/IRemoteDisplay.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IRemoteDisplay.h
\ No newline at end of file
diff --git a/include/media/IRemoteDisplayClient.h b/include/media/IRemoteDisplayClient.h
deleted file mode 120000
index f29a2ee..0000000
--- a/include/media/IRemoteDisplayClient.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IRemoteDisplayClient.h
\ No newline at end of file
diff --git a/include/media/IResourceManagerClient.h b/include/media/IResourceManagerClient.h
deleted file mode 120000
index 100af9b..0000000
--- a/include/media/IResourceManagerClient.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IResourceManagerClient.h
\ No newline at end of file
diff --git a/include/media/IResourceManagerService.h b/include/media/IResourceManagerService.h
deleted file mode 120000
index 9b389c6..0000000
--- a/include/media/IResourceManagerService.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IResourceManagerService.h
\ No newline at end of file
diff --git a/include/media/IStreamSource.h b/include/media/IStreamSource.h
deleted file mode 120000
index 4943af9..0000000
--- a/include/media/IStreamSource.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IStreamSource.h
\ No newline at end of file
diff --git a/include/media/JetPlayer.h b/include/media/JetPlayer.h
deleted file mode 120000
index 5483fda..0000000
--- a/include/media/JetPlayer.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/JetPlayer.h
\ No newline at end of file
diff --git a/include/media/LinearMap.h b/include/media/LinearMap.h
deleted file mode 120000
index 30d4ca8..0000000
--- a/include/media/LinearMap.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/LinearMap.h
\ No newline at end of file
diff --git a/include/media/MediaCodecBuffer.h b/include/media/MediaCodecBuffer.h
deleted file mode 120000
index 8c9aa76..0000000
--- a/include/media/MediaCodecBuffer.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/MediaCodecBuffer.h
\ No newline at end of file
diff --git a/include/media/MediaCodecInfo.h b/include/media/MediaCodecInfo.h
deleted file mode 120000
index ff44ce4..0000000
--- a/include/media/MediaCodecInfo.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/MediaCodecInfo.h
\ No newline at end of file
diff --git a/include/media/MediaMetadataRetrieverInterface.h b/include/media/MediaMetadataRetrieverInterface.h
deleted file mode 120000
index 1c53511..0000000
--- a/include/media/MediaMetadataRetrieverInterface.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/MediaMetadataRetrieverInterface.h
\ No newline at end of file
diff --git a/include/media/MediaProfiles.h b/include/media/MediaProfiles.h
deleted file mode 120000
index 651c6e6..0000000
--- a/include/media/MediaProfiles.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/MediaProfiles.h
\ No newline at end of file
diff --git a/include/media/MediaRecorderBase.h b/include/media/MediaRecorderBase.h
deleted file mode 120000
index e40f992..0000000
--- a/include/media/MediaRecorderBase.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/MediaRecorderBase.h
\ No newline at end of file
diff --git a/include/media/MediaResource.h b/include/media/MediaResource.h
deleted file mode 120000
index 91346aa..0000000
--- a/include/media/MediaResource.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/MediaResource.h
\ No newline at end of file
diff --git a/include/media/MediaResourcePolicy.h b/include/media/MediaResourcePolicy.h
deleted file mode 120000
index 5d165ee..0000000
--- a/include/media/MediaResourcePolicy.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/MediaResourcePolicy.h
\ No newline at end of file
diff --git a/include/media/MemoryLeakTrackUtil.h b/include/media/MemoryLeakTrackUtil.h
deleted file mode 120000
index 504173e..0000000
--- a/include/media/MemoryLeakTrackUtil.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/MemoryLeakTrackUtil.h
\ No newline at end of file
diff --git a/include/media/Metadata.h b/include/media/Metadata.h
deleted file mode 120000
index e421168..0000000
--- a/include/media/Metadata.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/Metadata.h
\ No newline at end of file
diff --git a/include/media/MidiDeviceInfo.h b/include/media/MidiDeviceInfo.h
deleted file mode 120000
index 95da7cf..0000000
--- a/include/media/MidiDeviceInfo.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/MidiDeviceInfo.h
\ No newline at end of file
diff --git a/include/media/MidiIoWrapper.h b/include/media/MidiIoWrapper.h
deleted file mode 120000
index 786ec3d..0000000
--- a/include/media/MidiIoWrapper.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/MidiIoWrapper.h
\ No newline at end of file
diff --git a/include/media/Modulo.h b/include/media/Modulo.h
deleted file mode 120000
index 989c4cb..0000000
--- a/include/media/Modulo.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/Modulo.h
\ No newline at end of file
diff --git a/include/media/OMXBuffer.h b/include/media/OMXBuffer.h
deleted file mode 120000
index 00db207..0000000
--- a/include/media/OMXBuffer.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/OMXBuffer.h
\ No newline at end of file
diff --git a/include/media/OMXFenceParcelable.h b/include/media/OMXFenceParcelable.h
deleted file mode 120000
index c4c1b0a..0000000
--- a/include/media/OMXFenceParcelable.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/OMXFenceParcelable.h
\ No newline at end of file
diff --git a/include/media/PluginLoader.h b/include/media/PluginLoader.h
deleted file mode 120000
index 9101735..0000000
--- a/include/media/PluginLoader.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/PluginLoader.h
\ No newline at end of file
diff --git a/include/media/PluginMetricsReporting.h b/include/media/PluginMetricsReporting.h
deleted file mode 120000
index 7d9a7a0..0000000
--- a/include/media/PluginMetricsReporting.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/PluginMetricsReporting.h
\ No newline at end of file
diff --git a/include/media/RecordBufferConverter.h b/include/media/RecordBufferConverter.h
deleted file mode 120000
index 2d7bc0c..0000000
--- a/include/media/RecordBufferConverter.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/RecordBufferConverter.h
\ No newline at end of file
diff --git a/include/media/RingBuffer.h b/include/media/RingBuffer.h
deleted file mode 120000
index 9af28d5..0000000
--- a/include/media/RingBuffer.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/RingBuffer.h
\ No newline at end of file
diff --git a/include/media/SingleStateQueue.h b/include/media/SingleStateQueue.h
deleted file mode 120000
index 619f6ee..0000000
--- a/include/media/SingleStateQueue.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/SingleStateQueue.h
\ No newline at end of file
diff --git a/include/media/StringArray.h b/include/media/StringArray.h
deleted file mode 120000
index 616ce6c..0000000
--- a/include/media/StringArray.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/StringArray.h
\ No newline at end of file
diff --git a/include/media/TimeCheck.h b/include/media/TimeCheck.h
deleted file mode 120000
index 85e17f9..0000000
--- a/include/media/TimeCheck.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/utils/include/mediautils/TimeCheck.h
\ No newline at end of file
diff --git a/include/media/ToneGenerator.h b/include/media/ToneGenerator.h
deleted file mode 120000
index 33df0e3..0000000
--- a/include/media/ToneGenerator.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/ToneGenerator.h
\ No newline at end of file
diff --git a/include/media/TypeConverter.h b/include/media/TypeConverter.h
deleted file mode 120000
index 837af44..0000000
--- a/include/media/TypeConverter.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/TypeConverter.h
\ No newline at end of file
diff --git a/include/media/Visualizer.h b/include/media/Visualizer.h
deleted file mode 120000
index ed2ec15..0000000
--- a/include/media/Visualizer.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/Visualizer.h
\ No newline at end of file
diff --git a/include/media/convert.h b/include/media/convert.h
deleted file mode 120000
index cb0d00d..0000000
--- a/include/media/convert.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/convert.h
\ No newline at end of file
diff --git a/include/media/mediametadataretriever.h b/include/media/mediametadataretriever.h
deleted file mode 120000
index b401bab..0000000
--- a/include/media/mediametadataretriever.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/mediametadataretriever.h
\ No newline at end of file
diff --git a/include/media/mediaplayer.h b/include/media/mediaplayer.h
deleted file mode 120000
index 06d537b..0000000
--- a/include/media/mediaplayer.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/mediaplayer.h
\ No newline at end of file
diff --git a/include/media/mediarecorder.h b/include/media/mediarecorder.h
deleted file mode 120000
index a24deb3..0000000
--- a/include/media/mediarecorder.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/mediarecorder.h
\ No newline at end of file
diff --git a/include/media/mediascanner.h b/include/media/mediascanner.h
deleted file mode 120000
index 91479e0..0000000
--- a/include/media/mediascanner.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/mediascanner.h
\ No newline at end of file
diff --git a/include/media/nbaio/AudioBufferProviderSource.h b/include/media/nbaio/AudioBufferProviderSource.h
deleted file mode 120000
index 55841e7..0000000
--- a/include/media/nbaio/AudioBufferProviderSource.h
+++ /dev/null
@@ -1 +0,0 @@
-../../../media/libnbaio/include/media/nbaio/AudioBufferProviderSource.h
\ No newline at end of file
diff --git a/include/media/nbaio/AudioStreamInSource.h b/include/media/nbaio/AudioStreamInSource.h
deleted file mode 120000
index f5bcc76..0000000
--- a/include/media/nbaio/AudioStreamInSource.h
+++ /dev/null
@@ -1 +0,0 @@
-../../../media/libnbaio/include/media/nbaio/AudioStreamInSource.h
\ No newline at end of file
diff --git a/include/media/nbaio/LibsndfileSink.h b/include/media/nbaio/LibsndfileSink.h
deleted file mode 120000
index 8a13b6c..0000000
--- a/include/media/nbaio/LibsndfileSink.h
+++ /dev/null
@@ -1 +0,0 @@
-../../../media/libnbaio/include/media/nbaio/LibsndfileSink.h
\ No newline at end of file
diff --git a/include/media/nbaio/LibsndfileSource.h b/include/media/nbaio/LibsndfileSource.h
deleted file mode 120000
index 2750fde..0000000
--- a/include/media/nbaio/LibsndfileSource.h
+++ /dev/null
@@ -1 +0,0 @@
-../../../media/libnbaio/include/media/nbaio/LibsndfileSource.h
\ No newline at end of file
diff --git a/include/media/nbaio/MonoPipe.h b/include/media/nbaio/MonoPipe.h
deleted file mode 120000
index 4ea43be..0000000
--- a/include/media/nbaio/MonoPipe.h
+++ /dev/null
@@ -1 +0,0 @@
-../../../media/libnbaio/include_mono/media/nbaio/MonoPipe.h
\ No newline at end of file
diff --git a/include/media/nbaio/MonoPipeReader.h b/include/media/nbaio/MonoPipeReader.h
deleted file mode 120000
index 30f426c..0000000
--- a/include/media/nbaio/MonoPipeReader.h
+++ /dev/null
@@ -1 +0,0 @@
-../../../media/libnbaio/include_mono/media/nbaio/MonoPipeReader.h
\ No newline at end of file
diff --git a/include/media/nbaio/Pipe.h b/include/media/nbaio/Pipe.h
deleted file mode 120000
index a4bbbc9..0000000
--- a/include/media/nbaio/Pipe.h
+++ /dev/null
@@ -1 +0,0 @@
-../../../media/libnbaio/include/media/nbaio/Pipe.h
\ No newline at end of file
diff --git a/include/media/nbaio/PipeReader.h b/include/media/nbaio/PipeReader.h
deleted file mode 120000
index 64b21cf..0000000
--- a/include/media/nbaio/PipeReader.h
+++ /dev/null
@@ -1 +0,0 @@
-../../../media/libnbaio/include/media/nbaio/PipeReader.h
\ No newline at end of file
diff --git a/include/media/nbaio/SingleStateQueue.h b/include/media/nbaio/SingleStateQueue.h
new file mode 120000
index 0000000..d3e0553
--- /dev/null
+++ b/include/media/nbaio/SingleStateQueue.h
@@ -0,0 +1 @@
+../../../media/libnbaio/include_mono/media/nbaio/SingleStateQueue.h
\ No newline at end of file
diff --git a/include/media/nbaio/SourceAudioBufferProvider.h b/include/media/nbaio/SourceAudioBufferProvider.h
deleted file mode 120000
index 74a3b06..0000000
--- a/include/media/nbaio/SourceAudioBufferProvider.h
+++ /dev/null
@@ -1 +0,0 @@
-../../../media/libnbaio/include/media/nbaio/SourceAudioBufferProvider.h
\ No newline at end of file
diff --git a/include/media/nblog/NBLog.h b/include/media/nblog/NBLog.h
deleted file mode 120000
index 3cc366c..0000000
--- a/include/media/nblog/NBLog.h
+++ /dev/null
@@ -1 +0,0 @@
-../../../media/libnblog/include/media/nblog/NBLog.h
\ No newline at end of file
diff --git a/include/media/nblog/PerformanceAnalysis.h b/include/media/nblog/PerformanceAnalysis.h
deleted file mode 120000
index 6ead3bc..0000000
--- a/include/media/nblog/PerformanceAnalysis.h
+++ /dev/null
@@ -1 +0,0 @@
-../../../media/libnblog/include/media/nblog/PerformanceAnalysis.h
\ No newline at end of file
diff --git a/include/media/nblog/ReportPerformance.h b/include/media/nblog/ReportPerformance.h
deleted file mode 120000
index e9b8e80..0000000
--- a/include/media/nblog/ReportPerformance.h
+++ /dev/null
@@ -1 +0,0 @@
-../../../media/libnblog/include/media/nblog/ReportPerformance.h
\ No newline at end of file
diff --git a/include/mediadrm/CryptoHal.h b/include/mediadrm/CryptoHal.h
deleted file mode 120000
index 92f3137..0000000
--- a/include/mediadrm/CryptoHal.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/CryptoHal.h
\ No newline at end of file
diff --git a/include/mediadrm/DrmHal.h b/include/mediadrm/DrmHal.h
deleted file mode 120000
index 17bb667..0000000
--- a/include/mediadrm/DrmHal.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/DrmHal.h
\ No newline at end of file
diff --git a/include/mediadrm/DrmMetrics.h b/include/mediadrm/DrmMetrics.h
deleted file mode 120000
index abc966b..0000000
--- a/include/mediadrm/DrmMetrics.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/DrmMetrics.h
\ No newline at end of file
diff --git a/include/mediadrm/DrmPluginPath.h b/include/mediadrm/DrmPluginPath.h
deleted file mode 120000
index 9e05194..0000000
--- a/include/mediadrm/DrmPluginPath.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/DrmPluginPath.h
\ No newline at end of file
diff --git a/include/mediadrm/DrmSessionClientInterface.h b/include/mediadrm/DrmSessionClientInterface.h
deleted file mode 120000
index f4e3211..0000000
--- a/include/mediadrm/DrmSessionClientInterface.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/DrmSessionClientInterface.h
\ No newline at end of file
diff --git a/include/mediadrm/DrmSessionManager.h b/include/mediadrm/DrmSessionManager.h
deleted file mode 120000
index f0a47bf..0000000
--- a/include/mediadrm/DrmSessionManager.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/DrmSessionManager.h
\ No newline at end of file
diff --git a/include/mediadrm/ICrypto.h b/include/mediadrm/ICrypto.h
deleted file mode 120000
index b250e07..0000000
--- a/include/mediadrm/ICrypto.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/ICrypto.h
\ No newline at end of file
diff --git a/include/mediadrm/IDrm.h b/include/mediadrm/IDrm.h
deleted file mode 120000
index 841bb1b..0000000
--- a/include/mediadrm/IDrm.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IDrm.h
\ No newline at end of file
diff --git a/include/mediadrm/IDrmClient.h b/include/mediadrm/IDrmClient.h
deleted file mode 120000
index 10aa5c0..0000000
--- a/include/mediadrm/IDrmClient.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IDrmClient.h
\ No newline at end of file
diff --git a/include/mediadrm/IMediaDrmService.h b/include/mediadrm/IMediaDrmService.h
deleted file mode 120000
index f3c260f..0000000
--- a/include/mediadrm/IMediaDrmService.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IMediaDrmService.h
\ No newline at end of file
diff --git a/include/mediadrm/SharedLibrary.h b/include/mediadrm/SharedLibrary.h
deleted file mode 120000
index 9f8f5a4..0000000
--- a/include/mediadrm/SharedLibrary.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/SharedLibrary.h
\ No newline at end of file
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index 5f19f74..1b1f149 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -28,7 +28,7 @@
#include <media/AudioResamplerPublic.h>
#include <media/AudioTimestamp.h>
#include <media/Modulo.h>
-#include <media/SingleStateQueue.h>
+#include <media/nbaio/SingleStateQueue.h>
namespace android {
diff --git a/media/audioserver/Android.mk b/media/audioserver/Android.mk
index 33b36b8..e33804d 100644
--- a/media/audioserver/Android.mk
+++ b/media/audioserver/Android.mk
@@ -9,12 +9,11 @@
libaaudioservice \
libaudioflinger \
libaudiopolicyservice \
+ libaudioprocessing \
libbinder \
libcutils \
liblog \
libhidlbase \
- libhidltransport \
- libhwbinder \
libmedia \
libmedialogservice \
libmediautils \
@@ -34,30 +33,13 @@
frameworks/av/services/audiopolicy/service \
frameworks/av/services/medialog \
frameworks/av/services/oboeservice \
- frameworks/av/services/radio \
frameworks/av/services/soundtrigger \
frameworks/av/media/libaaudio/include \
frameworks/av/media/libaaudio/src \
frameworks/av/media/libaaudio/src/binding \
frameworks/av/media/libmedia \
- $(call include-path-for, audio-utils) \
external/sonic \
-# If AUDIOSERVER_MULTILIB in device.mk is non-empty then it is used to control
-# the LOCAL_MULTILIB for all audioserver exclusive libraries.
-# This is relevant for 64 bit architectures where either or both
-# 32 and 64 bit libraries may be built.
-#
-# AUDIOSERVER_MULTILIB may be set as follows:
-# 32 to build 32 bit audioserver libraries and 32 bit audioserver.
-# 64 to build 64 bit audioserver libraries and 64 bit audioserver.
-# both to build both 32 bit and 64 bit libraries,
-# and use primary target architecture (32 or 64) for audioserver.
-# first to build libraries and audioserver for the primary target architecture only.
-# <empty> to build both 32 and 64 bit libraries and primary target audioserver.
-
-LOCAL_MULTILIB := $(AUDIOSERVER_MULTILIB)
-
LOCAL_MODULE := audioserver
LOCAL_INIT_RC := audioserver.rc
diff --git a/media/audioserver/audioserver.rc b/media/audioserver/audioserver.rc
index dfb1a3f..5484613 100644
--- a/media/audioserver/audioserver.rc
+++ b/media/audioserver/audioserver.rc
@@ -2,14 +2,14 @@
class core
user audioserver
# media gid needed for /dev/fm (radio) and for /data/misc/media (tee)
- group audio camera drmrpc inet media mediadrm net_bt net_bt_admin net_bw_acct wakelock
+ group audio camera drmrpc media mediadrm net_bt net_bt_admin net_bw_acct wakelock
capabilities BLOCK_SUSPEND
ioprio rt 4
writepid /dev/cpuset/foreground/tasks /dev/stune/foreground/tasks
- onrestart restart vendor.audio-hal-2-0
+ onrestart restart vendor.audio-hal
onrestart restart vendor.audio-hal-4-0-msd
- # Keep the original service name for backward compatibility when upgrading
- # O-MR1 devices with framework-only.
+ # Keep the original service names for backward compatibility
+ onrestart restart vendor.audio-hal-2-0
onrestart restart audio-hal-2-0
on property:vts.native_server.on=1
diff --git a/media/bufferpool/1.0/Android.bp b/media/bufferpool/1.0/Android.bp
index c7ea70f..f817c76 100644
--- a/media/bufferpool/1.0/Android.bp
+++ b/media/bufferpool/1.0/Android.bp
@@ -16,8 +16,6 @@
"libcutils",
"libfmq",
"libhidlbase",
- "libhwbinder",
- "libhidltransport",
"liblog",
"libutils",
"android.hardware.media.bufferpool@1.0",
diff --git a/media/bufferpool/2.0/Android.bp b/media/bufferpool/2.0/Android.bp
index e8a69c9..97f114a 100644
--- a/media/bufferpool/2.0/Android.bp
+++ b/media/bufferpool/2.0/Android.bp
@@ -16,8 +16,6 @@
"libcutils",
"libfmq",
"libhidlbase",
- "libhwbinder",
- "libhidltransport",
"liblog",
"libutils",
"android.hardware.media.bufferpool@2.0",
diff --git a/media/codec2/components/avc/C2SoftAvcEnc.cpp b/media/codec2/components/avc/C2SoftAvcEnc.cpp
index b41c271..e3d419c 100644
--- a/media/codec2/components/avc/C2SoftAvcEnc.cpp
+++ b/media/codec2/components/avc/C2SoftAvcEnc.cpp
@@ -40,7 +40,7 @@
namespace {
constexpr char COMPONENT_NAME[] = "c2.android.avc.encoder";
-
+constexpr uint32_t kMinOutBufferSize = 524288;
void ParseGop(
const C2StreamGopTuning::output &gop,
uint32_t *syncInterval, uint32_t *iInterval, uint32_t *maxBframes) {
@@ -440,8 +440,7 @@
mSignalledError(false),
mCodecCtx(nullptr),
mOutBlock(nullptr),
- // TODO: output buffer size
- mOutBufferSize(524288) {
+ mOutBufferSize(kMinOutBufferSize) {
// If dump is enabled, then open create an empty file
GENERATE_FILE_NAMES();
@@ -951,6 +950,9 @@
mStride = width;
+ // Assume worst case output buffer size to be equal to number of bytes in input
+ mOutBufferSize = std::max(width * height * 3 / 2, kMinOutBufferSize);
+
// TODO
mIvVideoColorFormat = IV_YUV_420P;
diff --git a/media/codec2/components/cmds/Android.bp b/media/codec2/components/cmds/Android.bp
index 35f689e..a081e28 100644
--- a/media/codec2/components/cmds/Android.bp
+++ b/media/codec2/components/cmds/Android.bp
@@ -9,10 +9,15 @@
include_dirs: [
],
+ header_libs: [
+ "libmediadrm_headers",
+ ],
+
shared_libs: [
"libbase",
"libbinder",
"libcutils",
+ "libdatasource",
"libgui",
"liblog",
"libstagefright",
diff --git a/media/codec2/components/cmds/codec2.cpp b/media/codec2/components/cmds/codec2.cpp
index f2cf545..38eaf88 100644
--- a/media/codec2/components/cmds/codec2.cpp
+++ b/media/codec2/components/cmds/codec2.cpp
@@ -30,15 +30,15 @@
#include <binder/IServiceManager.h>
#include <binder/ProcessState.h>
+#include <datasource/DataSourceFactory.h>
#include <media/DataSource.h>
-#include <media/ICrypto.h>
+#include <mediadrm/ICrypto.h>
#include <media/IMediaHTTPService.h>
#include <media/MediaSource.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/ALooper.h>
#include <media/stagefright/foundation/AMessage.h>
#include <media/stagefright/foundation/AUtils.h>
-#include <media/stagefright/DataSourceFactory.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaExtractorFactory.h>
@@ -418,7 +418,7 @@
const char *filename = argv[k];
sp<DataSource> dataSource =
- DataSourceFactory::CreateFromURI(nullptr /* httpService */, filename);
+ DataSourceFactory::getInstance()->CreateFromURI(nullptr /* httpService */, filename);
if (strncasecmp(filename, "sine:", 5) && dataSource == nullptr) {
fprintf(stderr, "Unable to create data source.\n");
diff --git a/media/codec2/components/hevc/C2SoftHevcEnc.cpp b/media/codec2/components/hevc/C2SoftHevcEnc.cpp
index b129b1b..19ccbf9 100644
--- a/media/codec2/components/hevc/C2SoftHevcEnc.cpp
+++ b/media/codec2/components/hevc/C2SoftHevcEnc.cpp
@@ -42,6 +42,36 @@
constexpr char COMPONENT_NAME[] = "c2.android.hevc.encoder";
+void ParseGop(
+ const C2StreamGopTuning::output &gop,
+ uint32_t *syncInterval, uint32_t *iInterval, uint32_t *maxBframes) {
+ uint32_t syncInt = 1;
+ uint32_t iInt = 1;
+ for (size_t i = 0; i < gop.flexCount(); ++i) {
+ const C2GopLayerStruct &layer = gop.m.values[i];
+ if (layer.count == UINT32_MAX) {
+ syncInt = 0;
+ } else if (syncInt <= UINT32_MAX / (layer.count + 1)) {
+ syncInt *= (layer.count + 1);
+ }
+ if ((layer.type_ & I_FRAME) == 0) {
+ if (layer.count == UINT32_MAX) {
+ iInt = 0;
+ } else if (iInt <= UINT32_MAX / (layer.count + 1)) {
+ iInt *= (layer.count + 1);
+ }
+ }
+ if (layer.type_ == C2Config::picture_type_t(P_FRAME | B_FRAME) && maxBframes) {
+ *maxBframes = layer.count;
+ }
+ }
+ if (syncInterval) {
+ *syncInterval = syncInt;
+ }
+ if (iInterval) {
+ *iInterval = iInt;
+ }
+}
} // namepsace
class C2SoftHevcEnc::IntfImpl : public SimpleInterface<void>::BaseParams {
@@ -60,13 +90,21 @@
setDerivedInstance(this);
addParameter(
+ DefineParam(mGop, C2_PARAMKEY_GOP)
+ .withDefault(C2StreamGopTuning::output::AllocShared(
+ 0 /* flexCount */, 0u /* stream */))
+ .withFields({C2F(mGop, m.values[0].type_).any(),
+ C2F(mGop, m.values[0].count).any()})
+ .withSetter(GopSetter)
+ .build());
+
+ addParameter(
DefineParam(mActualInputDelay, C2_PARAMKEY_INPUT_DELAY)
.withDefault(new C2PortActualDelayTuning::input(
DEFAULT_B_FRAMES + DEFAULT_RC_LOOKAHEAD))
.withFields({C2F(mActualInputDelay, value).inRange(
0, MAX_B_FRAMES + MAX_RC_LOOKAHEAD)})
- .withSetter(
- Setter<decltype(*mActualInputDelay)>::StrictValueWithNoDeps)
+ .calculatedAs(InputDelaySetter, mGop)
.build());
addParameter(
@@ -172,6 +210,17 @@
.build());
}
+ static C2R InputDelaySetter(
+ bool mayBlock,
+ C2P<C2PortActualDelayTuning::input> &me,
+ const C2P<C2StreamGopTuning::output> &gop) {
+ (void)mayBlock;
+ uint32_t maxBframes = 0;
+ ParseGop(gop.v, nullptr, nullptr, &maxBframes);
+ me.set().value = maxBframes + DEFAULT_RC_LOOKAHEAD;
+ return C2R::Ok();
+ }
+
static C2R BitrateSetter(bool mayBlock,
C2P<C2StreamBitrateInfo::output>& me) {
(void)mayBlock;
@@ -270,6 +319,18 @@
return C2R::Ok();
}
+ static C2R GopSetter(bool mayBlock, C2P<C2StreamGopTuning::output> &me) {
+ (void)mayBlock;
+ for (size_t i = 0; i < me.v.flexCount(); ++i) {
+ const C2GopLayerStruct &layer = me.v.m.values[0];
+ if (layer.type_ == C2Config::picture_type_t(P_FRAME | B_FRAME)
+ && layer.count > MAX_B_FRAMES) {
+ me.set().m.values[i].count = MAX_B_FRAMES;
+ }
+ }
+ return C2R::Ok();
+ }
+
UWORD32 getProfile_l() const {
switch (mProfileLevel->profile) {
case PROFILE_HEVC_MAIN: [[fallthrough]];
@@ -338,6 +399,9 @@
std::shared_ptr<C2StreamQualityTuning::output> getQuality_l() const {
return mQuality;
}
+ std::shared_ptr<C2StreamGopTuning::output> getGop_l() const {
+ return mGop;
+ }
private:
std::shared_ptr<C2StreamUsageTuning::input> mUsage;
@@ -350,6 +414,7 @@
std::shared_ptr<C2StreamQualityTuning::output> mQuality;
std::shared_ptr<C2StreamProfileLevelInfo::output> mProfileLevel;
std::shared_ptr<C2StreamSyncFrameIntervalTuning::output> mSyncFramePeriod;
+ std::shared_ptr<C2StreamGopTuning::output> mGop;
};
static size_t GetCPUCoreCount() {
@@ -449,7 +514,25 @@
ALOGE("HEVC default init failed : 0x%x", err);
return C2_CORRUPTED;
}
-
+ mBframes = 0;
+ if (mGop && mGop->flexCount() > 0) {
+ uint32_t syncInterval = 1;
+ uint32_t iInterval = 1;
+ uint32_t maxBframes = 0;
+ ParseGop(*mGop, &syncInterval, &iInterval, &maxBframes);
+ if (syncInterval > 0) {
+ ALOGD("Updating IDR interval from GOP: old %u new %u", mIDRInterval, syncInterval);
+ mIDRInterval = syncInterval;
+ }
+ if (iInterval > 0) {
+ ALOGD("Updating I interval from GOP: old %u new %u", mIInterval, iInterval);
+ mIInterval = iInterval;
+ }
+ if (mBframes != maxBframes) {
+ ALOGD("Updating max B frames from GOP: old %u new %u", mBframes, maxBframes);
+ mBframes = maxBframes;
+ }
+ }
// update configuration
mEncParams.s_src_prms.i4_width = mSize->width;
mEncParams.s_src_prms.i4_height = mSize->height;
@@ -463,12 +546,20 @@
mBitrate->value << 1;
mEncParams.s_tgt_lyr_prms.as_tgt_params[0].i4_codec_level = mHevcEncLevel;
mEncParams.s_coding_tools_prms.i4_max_i_open_gop_period = mIDRInterval;
- mEncParams.s_coding_tools_prms.i4_max_cra_open_gop_period = mIDRInterval;
+ mEncParams.s_coding_tools_prms.i4_max_cra_open_gop_period = mIInterval;
mIvVideoColorFormat = IV_YUV_420P;
mEncParams.s_multi_thrd_prms.i4_max_num_cores = mNumCores;
mEncParams.s_out_strm_prms.i4_codec_profile = mHevcEncProfile;
mEncParams.s_lap_prms.i4_rc_look_ahead_pics = DEFAULT_RC_LOOKAHEAD;
- mEncParams.s_coding_tools_prms.i4_max_temporal_layers = DEFAULT_B_FRAMES;
+ if (mBframes == 0) {
+ mEncParams.s_coding_tools_prms.i4_max_temporal_layers = 0;
+ } else if (mBframes <= 2) {
+ mEncParams.s_coding_tools_prms.i4_max_temporal_layers = 1;
+ } else if (mBframes <= 6) {
+ mEncParams.s_coding_tools_prms.i4_max_temporal_layers = 2;
+ } else {
+ mEncParams.s_coding_tools_prms.i4_max_temporal_layers = 3;
+ }
switch (mBitrateMode->value) {
case C2Config::BITRATE_IGNORE:
@@ -523,6 +614,7 @@
c2_status_t C2SoftHevcEnc::initEncoder() {
CHECK(!mCodecCtx);
+
{
IntfImpl::Lock lock = mIntf->lock();
mSize = mIntf->getSize_l();
@@ -532,8 +624,10 @@
mHevcEncProfile = mIntf->getProfile_l();
mHevcEncLevel = mIntf->getLevel_l();
mIDRInterval = mIntf->getSyncFramePeriod_l();
+ mIInterval = mIntf->getSyncFramePeriod_l();
mComplexity = mIntf->getComplexity_l();
mQuality = mIntf->getQuality_l();
+ mGop = mIntf->getGop_l();
}
c2_status_t status = initEncParams();
diff --git a/media/codec2/components/hevc/C2SoftHevcEnc.h b/media/codec2/components/hevc/C2SoftHevcEnc.h
index f2c7642..140b4a9 100644
--- a/media/codec2/components/hevc/C2SoftHevcEnc.h
+++ b/media/codec2/components/hevc/C2SoftHevcEnc.h
@@ -67,6 +67,8 @@
ihevce_static_cfg_params_t mEncParams;
size_t mNumCores;
UWORD32 mIDRInterval;
+ UWORD32 mIInterval;
+ UWORD32 mBframes;
IV_COLOR_FORMAT_T mIvVideoColorFormat;
UWORD32 mHevcEncProfile;
UWORD32 mHevcEncLevel;
@@ -85,7 +87,7 @@
std::shared_ptr<C2StreamBitrateModeTuning::output> mBitrateMode;
std::shared_ptr<C2StreamComplexityTuning::output> mComplexity;
std::shared_ptr<C2StreamQualityTuning::output> mQuality;
-
+ std::shared_ptr<C2StreamGopTuning::output> mGop;
#ifdef FILE_DUMP_ENABLE
char mInFile[200];
char mOutFile[200];
diff --git a/media/codec2/components/vpx/C2SoftVpxEnc.cpp b/media/codec2/components/vpx/C2SoftVpxEnc.cpp
index 6dab70b..ebc7a8f 100644
--- a/media/codec2/components/vpx/C2SoftVpxEnc.cpp
+++ b/media/codec2/components/vpx/C2SoftVpxEnc.cpp
@@ -514,7 +514,7 @@
return;
}
vpx_img_wrap(&raw_frame, VPX_IMG_FMT_I420, stride, vstride,
- mStrideAlign, (uint8_t*)rView->data()[0]);
+ mStrideAlign, mConversionBuffer.data());
vpx_img_set_rect(&raw_frame, 0, 0, width, height);
} else {
ALOGE("Conversion buffer is too small: %u x %u for %zu",
diff --git a/media/codec2/hidl/1.0/utils/Android.bp b/media/codec2/hidl/1.0/utils/Android.bp
index f1f1536..a2930a6 100644
--- a/media/codec2/hidl/1.0/utils/Android.bp
+++ b/media/codec2/hidl/1.0/utils/Android.bp
@@ -63,6 +63,7 @@
],
header_libs: [
+ "libbinder_headers",
"libsystem_headers",
"libcodec2_internal", // private
],
@@ -80,8 +81,6 @@
"libcodec2_vndk",
"libcutils",
"libhidlbase",
- "libhidltransport",
- "libhwbinder",
"liblog",
"libstagefright_bufferpool@2.0.1",
"libstagefright_bufferqueue_helper",
diff --git a/media/codec2/hidl/1.0/utils/InputBufferManager.cpp b/media/codec2/hidl/1.0/utils/InputBufferManager.cpp
index 8c0d0a4..a023a05 100644
--- a/media/codec2/hidl/1.0/utils/InputBufferManager.cpp
+++ b/media/codec2/hidl/1.0/utils/InputBufferManager.cpp
@@ -70,7 +70,7 @@
<< ".";
std::lock_guard<std::mutex> lock(mMutex);
- std::set<TrackedBuffer*> &bufferIds =
+ std::set<TrackedBuffer> &bufferIds =
mTrackedBuffersMap[listener][frameIndex];
for (size_t i = 0; i < input.buffers.size(); ++i) {
@@ -79,14 +79,13 @@
<< "Input buffer at index " << i << " is null.";
continue;
}
- TrackedBuffer *bufferId =
- new TrackedBuffer(listener, frameIndex, i, input.buffers[i]);
- mTrackedBufferCache.emplace(bufferId);
- bufferIds.emplace(bufferId);
+ const TrackedBuffer &bufferId =
+ *bufferIds.emplace(listener, frameIndex, i, input.buffers[i]).
+ first;
c2_status_t status = input.buffers[i]->registerOnDestroyNotify(
onBufferDestroyed,
- reinterpret_cast<void*>(bufferId));
+ const_cast<void*>(reinterpret_cast<const void*>(&bufferId)));
if (status != C2_OK) {
LOG(DEBUG) << "InputBufferManager::_registerFrameData -- "
<< "registerOnDestroyNotify() failed "
@@ -120,32 +119,31 @@
auto findListener = mTrackedBuffersMap.find(listener);
if (findListener != mTrackedBuffersMap.end()) {
- std::map<uint64_t, std::set<TrackedBuffer*>> &frameIndex2BufferIds
+ std::map<uint64_t, std::set<TrackedBuffer>> &frameIndex2BufferIds
= findListener->second;
auto findFrameIndex = frameIndex2BufferIds.find(frameIndex);
if (findFrameIndex != frameIndex2BufferIds.end()) {
- std::set<TrackedBuffer*> &bufferIds = findFrameIndex->second;
- for (TrackedBuffer* bufferId : bufferIds) {
- std::shared_ptr<C2Buffer> buffer = bufferId->buffer.lock();
+ std::set<TrackedBuffer> &bufferIds = findFrameIndex->second;
+ for (const TrackedBuffer& bufferId : bufferIds) {
+ std::shared_ptr<C2Buffer> buffer = bufferId.buffer.lock();
if (buffer) {
c2_status_t status = buffer->unregisterOnDestroyNotify(
onBufferDestroyed,
- reinterpret_cast<void*>(bufferId));
+ const_cast<void*>(
+ reinterpret_cast<const void*>(&bufferId)));
if (status != C2_OK) {
LOG(DEBUG) << "InputBufferManager::_unregisterFrameData "
<< "-- unregisterOnDestroyNotify() failed "
<< "(listener @ 0x"
<< std::hex
- << bufferId->listener.unsafe_get()
+ << bufferId.listener.unsafe_get()
<< ", frameIndex = "
- << std::dec << bufferId->frameIndex
- << ", bufferIndex = " << bufferId->bufferIndex
+ << std::dec << bufferId.frameIndex
+ << ", bufferIndex = " << bufferId.bufferIndex
<< ") => status = " << status
<< ".";
}
}
- mTrackedBufferCache.erase(bufferId);
- delete bufferId;
}
frameIndex2BufferIds.erase(findFrameIndex);
@@ -181,32 +179,31 @@
auto findListener = mTrackedBuffersMap.find(listener);
if (findListener != mTrackedBuffersMap.end()) {
- std::map<uint64_t, std::set<TrackedBuffer*>> &frameIndex2BufferIds =
+ std::map<uint64_t, std::set<TrackedBuffer>> &frameIndex2BufferIds =
findListener->second;
for (auto findFrameIndex = frameIndex2BufferIds.begin();
findFrameIndex != frameIndex2BufferIds.end();
++findFrameIndex) {
- std::set<TrackedBuffer*> &bufferIds = findFrameIndex->second;
- for (TrackedBuffer* bufferId : bufferIds) {
- std::shared_ptr<C2Buffer> buffer = bufferId->buffer.lock();
+ std::set<TrackedBuffer> &bufferIds = findFrameIndex->second;
+ for (const TrackedBuffer& bufferId : bufferIds) {
+ std::shared_ptr<C2Buffer> buffer = bufferId.buffer.lock();
if (buffer) {
c2_status_t status = buffer->unregisterOnDestroyNotify(
onBufferDestroyed,
- reinterpret_cast<void*>(bufferId));
+ const_cast<void*>(
+ reinterpret_cast<const void*>(&bufferId)));
if (status != C2_OK) {
LOG(DEBUG) << "InputBufferManager::_unregisterFrameData "
<< "-- unregisterOnDestroyNotify() failed "
<< "(listener @ 0x"
<< std::hex
- << bufferId->listener.unsafe_get()
+ << bufferId.listener.unsafe_get()
<< ", frameIndex = "
- << std::dec << bufferId->frameIndex
- << ", bufferIndex = " << bufferId->bufferIndex
+ << std::dec << bufferId.frameIndex
+ << ", bufferIndex = " << bufferId.bufferIndex
<< ") => status = " << status
<< ".";
}
- mTrackedBufferCache.erase(bufferId);
- delete bufferId;
}
}
}
@@ -239,59 +236,50 @@
<< std::dec << ".";
return;
}
-
- std::lock_guard<std::mutex> lock(mMutex);
- TrackedBuffer *bufferId = reinterpret_cast<TrackedBuffer*>(arg);
-
- if (mTrackedBufferCache.find(bufferId) == mTrackedBufferCache.end()) {
- LOG(VERBOSE) << "InputBufferManager::_onBufferDestroyed -- called with "
- << "unregistered buffer: "
- << "buf @ 0x" << std::hex << buf
- << ", arg @ 0x" << std::hex << arg
- << std::dec << ".";
- return;
- }
-
+ TrackedBuffer id(*reinterpret_cast<TrackedBuffer*>(arg));
LOG(VERBOSE) << "InputBufferManager::_onBufferDestroyed -- called with "
<< "buf @ 0x" << std::hex << buf
<< ", arg @ 0x" << std::hex << arg
<< std::dec << " -- "
- << "listener @ 0x" << std::hex << bufferId->listener.unsafe_get()
- << ", frameIndex = " << std::dec << bufferId->frameIndex
- << ", bufferIndex = " << bufferId->bufferIndex
+ << "listener @ 0x" << std::hex << id.listener.unsafe_get()
+ << ", frameIndex = " << std::dec << id.frameIndex
+ << ", bufferIndex = " << id.bufferIndex
<< ".";
- auto findListener = mTrackedBuffersMap.find(bufferId->listener);
- if (findListener == mTrackedBuffersMap.end()) {
- LOG(VERBOSE) << "InputBufferManager::_onBufferDestroyed -- "
- << "received invalid listener: "
- << "listener @ 0x" << std::hex << bufferId->listener.unsafe_get()
- << " (frameIndex = " << std::dec << bufferId->frameIndex
- << ", bufferIndex = " << bufferId->bufferIndex
- << ").";
- return;
- }
- std::map<uint64_t, std::set<TrackedBuffer*>> &frameIndex2BufferIds
- = findListener->second;
- auto findFrameIndex = frameIndex2BufferIds.find(bufferId->frameIndex);
- if (findFrameIndex == frameIndex2BufferIds.end()) {
+ std::lock_guard<std::mutex> lock(mMutex);
+
+ auto findListener = mTrackedBuffersMap.find(id.listener);
+ if (findListener == mTrackedBuffersMap.end()) {
LOG(DEBUG) << "InputBufferManager::_onBufferDestroyed -- "
- << "received invalid frame index: "
- << "frameIndex = " << bufferId->frameIndex
- << " (listener @ 0x" << std::hex << bufferId->listener.unsafe_get()
- << ", bufferIndex = " << std::dec << bufferId->bufferIndex
+ << "received invalid listener: "
+ << "listener @ 0x" << std::hex << id.listener.unsafe_get()
+ << " (frameIndex = " << std::dec << id.frameIndex
+ << ", bufferIndex = " << id.bufferIndex
<< ").";
return;
}
- std::set<TrackedBuffer*> &bufferIds = findFrameIndex->second;
- auto findBufferId = bufferIds.find(bufferId);
+ std::map<uint64_t, std::set<TrackedBuffer>> &frameIndex2BufferIds
+ = findListener->second;
+ auto findFrameIndex = frameIndex2BufferIds.find(id.frameIndex);
+ if (findFrameIndex == frameIndex2BufferIds.end()) {
+ LOG(DEBUG) << "InputBufferManager::_onBufferDestroyed -- "
+ << "received invalid frame index: "
+ << "frameIndex = " << id.frameIndex
+ << " (listener @ 0x" << std::hex << id.listener.unsafe_get()
+ << ", bufferIndex = " << std::dec << id.bufferIndex
+ << ").";
+ return;
+ }
+
+ std::set<TrackedBuffer> &bufferIds = findFrameIndex->second;
+ auto findBufferId = bufferIds.find(id);
if (findBufferId == bufferIds.end()) {
LOG(DEBUG) << "InputBufferManager::_onBufferDestroyed -- "
<< "received invalid buffer index: "
- << "bufferIndex = " << bufferId->bufferIndex
- << " (frameIndex = " << bufferId->frameIndex
- << ", listener @ 0x" << std::hex << bufferId->listener.unsafe_get()
+ << "bufferIndex = " << id.bufferIndex
+ << " (frameIndex = " << id.frameIndex
+ << ", listener @ 0x" << std::hex << id.listener.unsafe_get()
<< std::dec << ").";
return;
}
@@ -304,13 +292,10 @@
}
}
- DeathNotifications &deathNotifications = mDeathNotifications[bufferId->listener];
- deathNotifications.indices[bufferId->frameIndex].emplace_back(bufferId->bufferIndex);
+ DeathNotifications &deathNotifications = mDeathNotifications[id.listener];
+ deathNotifications.indices[id.frameIndex].emplace_back(id.bufferIndex);
++deathNotifications.count;
mOnBufferDestroyed.notify_one();
-
- mTrackedBufferCache.erase(bufferId);
- delete bufferId;
}
// Notify the clients about buffer destructions.
diff --git a/media/codec2/hidl/1.0/utils/include/codec2/hidl/1.0/InputBufferManager.h b/media/codec2/hidl/1.0/utils/include/codec2/hidl/1.0/InputBufferManager.h
index 42fa557..b6857d5 100644
--- a/media/codec2/hidl/1.0/utils/include/codec2/hidl/1.0/InputBufferManager.h
+++ b/media/codec2/hidl/1.0/utils/include/codec2/hidl/1.0/InputBufferManager.h
@@ -196,9 +196,13 @@
frameIndex(frameIndex),
bufferIndex(bufferIndex),
buffer(buffer) {}
+ TrackedBuffer(const TrackedBuffer&) = default;
+ bool operator<(const TrackedBuffer& other) const {
+ return bufferIndex < other.bufferIndex;
+ }
};
- // Map: listener -> frameIndex -> set<TrackedBuffer*>.
+ // Map: listener -> frameIndex -> set<TrackedBuffer>.
// Essentially, this is used to store triples (listener, frameIndex,
// bufferIndex) that's searchable by listener and (listener, frameIndex).
// However, the value of the innermost map is TrackedBuffer, which also
@@ -206,7 +210,7 @@
// because onBufferDestroyed() needs to know listener and frameIndex too.
typedef std::map<wp<IComponentListener>,
std::map<uint64_t,
- std::set<TrackedBuffer*>>> TrackedBuffersMap;
+ std::set<TrackedBuffer>>> TrackedBuffersMap;
// Storage for pending (unsent) death notifications for one listener.
// Each pair in member named "indices" are (frameIndex, bufferIndex) from
@@ -243,16 +247,6 @@
// Mutex for the management of all input buffers.
std::mutex mMutex;
- // Cache for all TrackedBuffers.
- //
- // Whenever registerOnDestroyNotify() is called, an argument of type
- // TrackedBuffer is created and stored into this cache.
- // Whenever unregisterOnDestroyNotify() or onBufferDestroyed() is called,
- // the TrackedBuffer is removed from this cache.
- //
- // mTrackedBuffersMap stores references to TrackedBuffers inside this cache.
- std::set<TrackedBuffer*> mTrackedBufferCache;
-
// Tracked input buffers.
TrackedBuffersMap mTrackedBuffersMap;
diff --git a/media/codec2/hidl/client/Android.bp b/media/codec2/hidl/client/Android.bp
index e184223..89c1c4a 100644
--- a/media/codec2/hidl/client/Android.bp
+++ b/media/codec2/hidl/client/Android.bp
@@ -17,7 +17,6 @@
"libcutils",
"libgui",
"libhidlbase",
- "libhidltransport",
"liblog",
"libstagefright_bufferpool@2.0.1",
"libui",
diff --git a/media/codec2/hidl/services/Android.bp b/media/codec2/hidl/services/Android.bp
index 216525e..0403a1f 100644
--- a/media/codec2/hidl/services/Android.bp
+++ b/media/codec2/hidl/services/Android.bp
@@ -17,8 +17,6 @@
"libcodec2_hidl@1.0",
"libcodec2_vndk",
"libhidlbase",
- "libhidltransport",
- "libhwbinder",
"liblog",
"libstagefright_omx",
"libstagefright_xmlparser",
diff --git a/media/codec2/hidl/services/vendor.cpp b/media/codec2/hidl/services/vendor.cpp
index ef2f98e..a4e079d 100644
--- a/media/codec2/hidl/services/vendor.cpp
+++ b/media/codec2/hidl/services/vendor.cpp
@@ -22,7 +22,9 @@
#include <binder/ProcessState.h>
#include <minijail.h>
+#include <util/C2InterfaceHelper.h>
#include <C2Component.h>
+#include <C2Config.h>
// OmxStore is added for visibility by dumpstate.
#include <media/stagefright/omx/1.0/OmxStore.h>
@@ -35,11 +37,14 @@
static constexpr char kExtSeccompPolicyPath[] =
"/vendor/etc/seccomp_policy/codec2.vendor.ext.policy";
-class DummyC2Store : public C2ComponentStore {
+class StoreImpl : public C2ComponentStore {
public:
- DummyC2Store() = default;
+ StoreImpl()
+ : mReflectorHelper(std::make_shared<C2ReflectorHelper>()),
+ mInterface(mReflectorHelper) {
+ }
- virtual ~DummyC2Store() override = default;
+ virtual ~StoreImpl() override = default;
virtual C2String getName() const override {
return "default";
@@ -69,31 +74,69 @@
}
virtual c2_status_t query_sm(
- const std::vector<C2Param*>& /* stackParams */,
- const std::vector<C2Param::Index>& /* heapParamIndices */,
- std::vector<std::unique_ptr<C2Param>>* const /* heapParams */) const override {
- return C2_OMITTED;
+ const std::vector<C2Param*>& stackParams,
+ const std::vector<C2Param::Index>& heapParamIndices,
+ std::vector<std::unique_ptr<C2Param>>* const heapParams) const override {
+ return mInterface.query(stackParams, heapParamIndices, C2_MAY_BLOCK, heapParams);
}
virtual c2_status_t config_sm(
- const std::vector<C2Param*>& /* params */,
- std::vector<std::unique_ptr<C2SettingResult>>* const /* failures */) override {
- return C2_OMITTED;
+ const std::vector<C2Param*>& params,
+ std::vector<std::unique_ptr<C2SettingResult>>* const failures) override {
+ return mInterface.config(params, C2_MAY_BLOCK, failures);
}
virtual std::shared_ptr<C2ParamReflector> getParamReflector() const override {
- return nullptr;
+ return mReflectorHelper;
}
virtual c2_status_t querySupportedParams_nb(
- std::vector<std::shared_ptr<C2ParamDescriptor>>* const /* params */) const override {
- return C2_OMITTED;
+ std::vector<std::shared_ptr<C2ParamDescriptor>>* const params) const override {
+ return mInterface.querySupportedParams(params);
}
virtual c2_status_t querySupportedValues_sm(
- std::vector<C2FieldSupportedValuesQuery>& /* fields */) const override {
- return C2_OMITTED;
+ std::vector<C2FieldSupportedValuesQuery>& fields) const override {
+ return mInterface.querySupportedValues(fields, C2_MAY_BLOCK);
}
+
+private:
+ class Interface : public C2InterfaceHelper {
+ public:
+ Interface(const std::shared_ptr<C2ReflectorHelper> &helper)
+ : C2InterfaceHelper(helper) {
+ setDerivedInstance(this);
+
+ addParameter(
+ DefineParam(mIonUsageInfo, "ion-usage")
+ .withDefault(new C2StoreIonUsageInfo())
+ .withFields({
+ C2F(mIonUsageInfo, usage).flags(
+ {C2MemoryUsage::CPU_READ | C2MemoryUsage::CPU_WRITE}),
+ C2F(mIonUsageInfo, capacity).inRange(0, UINT32_MAX, 1024),
+ C2F(mIonUsageInfo, heapMask).any(),
+ C2F(mIonUsageInfo, allocFlags).flags({}),
+ C2F(mIonUsageInfo, minAlignment).equalTo(0)
+ })
+ .withSetter(SetIonUsage)
+ .build());
+ }
+
+ virtual ~Interface() = default;
+
+ private:
+ static C2R SetIonUsage(bool /* mayBlock */, C2P<C2StoreIonUsageInfo> &me) {
+ // Vendor's TODO: put appropriate mapping logic
+ me.set().heapMask = ~0;
+ me.set().allocFlags = 0;
+ me.set().minAlignment = 0;
+ return C2R::Ok();
+ }
+
+ std::shared_ptr<C2StoreIonUsageInfo> mIonUsageInfo;
+ };
+ std::shared_ptr<C2ReflectorHelper> mReflectorHelper;
+ Interface mInterface;
};
int main(int /* argc */, char** /* argv */) {
@@ -120,7 +163,7 @@
// /* implementation of C2ComponentStore */);
ALOGD("Instantiating Codec2's dummy IComponentStore service...");
store = new utils::ComponentStore(
- std::make_shared<DummyC2Store>());
+ std::make_shared<StoreImpl>());
if (store == nullptr) {
ALOGE("Cannot create Codec2's IComponentStore service.");
diff --git a/media/codec2/sfplugin/Android.bp b/media/codec2/sfplugin/Android.bp
index 9c84c71..ec576c9 100644
--- a/media/codec2/sfplugin/Android.bp
+++ b/media/codec2/sfplugin/Android.bp
@@ -22,6 +22,8 @@
header_libs: [
"libcodec2_internal",
+ "libmediadrm_headers",
+ "media_ndk_headers",
],
shared_libs: [
@@ -39,7 +41,7 @@
"libhidlallocatorutils",
"libhidlbase",
"liblog",
- "libmedia",
+ "libmedia_codeclist",
"libmedia_omx",
"libsfplugin_ccodec_utils",
"libstagefright_bufferqueue_helper",
diff --git a/media/codec2/sfplugin/CCodec.cpp b/media/codec2/sfplugin/CCodec.cpp
index 4a31953..78ddd6d 100644
--- a/media/codec2/sfplugin/CCodec.cpp
+++ b/media/codec2/sfplugin/CCodec.cpp
@@ -1286,7 +1286,8 @@
{
Mutexed<Config>::Locked config(mConfig);
inputFormat = config->mInputFormat;
- outputFormat = config->mOutputFormat;
+ // start triggers format dup
+ outputFormat = config->mOutputFormat = config->mOutputFormat->dup();
if (config->mInputSurface) {
err2 = config->mInputSurface->start();
}
@@ -1295,6 +1296,8 @@
mCallback->onError(err2, ACTION_CODE_FATAL);
return;
}
+ // We're not starting after flush.
+ (void)mSentConfigAfterResume.test_and_set();
err2 = mChannel->start(inputFormat, outputFormat);
if (err2 != OK) {
mCallback->onError(err2, ACTION_CODE_FATAL);
@@ -1523,18 +1526,26 @@
}
void CCodec::signalResume() {
- auto setResuming = [this] {
+ std::shared_ptr<Codec2Client::Component> comp;
+ auto setResuming = [this, &comp] {
Mutexed<State>::Locked state(mState);
if (state->get() != FLUSHED) {
return UNKNOWN_ERROR;
}
state->set(RESUMING);
+ comp = state->comp;
return OK;
};
if (tryAndReportOnError(setResuming) != OK) {
return;
}
+ mSentConfigAfterResume.clear();
+ {
+ Mutexed<Config>::Locked config(mConfig);
+ config->queryConfiguration(comp);
+ }
+
(void)mChannel->start(nullptr, nullptr);
{
@@ -1730,7 +1741,7 @@
// handle configuration changes in work done
Mutexed<Config>::Locked config(mConfig);
- bool changed = false;
+ bool changed = !mSentConfigAfterResume.test_and_set();
Config::Watcher<C2StreamInitDataInfo::output> initData =
config->watch<C2StreamInitDataInfo::output>();
if (!work->worklets.empty()
@@ -1762,7 +1773,9 @@
++stream;
}
- changed = config->updateConfiguration(updates, config->mOutputDomain);
+ if (config->updateConfiguration(updates, config->mOutputDomain)) {
+ changed = true;
+ }
// copy standard infos to graphic buffers if not already present (otherwise, we
// may overwrite the actual intermediate value with a final value)
diff --git a/media/codec2/sfplugin/CCodec.h b/media/codec2/sfplugin/CCodec.h
index b0b3c4f..a580d1d 100644
--- a/media/codec2/sfplugin/CCodec.h
+++ b/media/codec2/sfplugin/CCodec.h
@@ -17,6 +17,7 @@
#ifndef C_CODEC_H_
#define C_CODEC_H_
+#include <atomic>
#include <chrono>
#include <list>
#include <memory>
@@ -175,6 +176,7 @@
typedef CCodecConfig Config;
Mutexed<Config> mConfig;
Mutexed<std::list<std::unique_ptr<C2Work>>> mWorkDoneQueue;
+ std::atomic_flag mSentConfigAfterResume;
friend class CCodecCallbackImpl;
diff --git a/media/codec2/sfplugin/CCodecBufferChannel.cpp b/media/codec2/sfplugin/CCodecBufferChannel.cpp
index 2efb987..a4c30fa 100644
--- a/media/codec2/sfplugin/CCodecBufferChannel.cpp
+++ b/media/codec2/sfplugin/CCodecBufferChannel.cpp
@@ -29,6 +29,7 @@
#include <android/hardware/cas/native/1.0/IDescrambler.h>
#include <android-base/stringprintf.h>
#include <binder/MemoryDealer.h>
+#include <cutils/properties.h>
#include <gui/Surface.h>
#include <media/openmax/OMX_Core.h>
#include <media/stagefright/foundation/ABuffer.h>
@@ -1072,7 +1073,7 @@
} else {
output->buffers.reset(new LinearOutputBuffers(mName));
}
- output->buffers->setFormat(outputFormat->dup());
+ output->buffers->setFormat(outputFormat);
// Try to set output surface to created block pool if given.
@@ -1276,6 +1277,24 @@
std::unique_ptr<C2Work> work,
const sp<AMessage> &outputFormat,
const C2StreamInitDataInfo::output *initData) {
+ if (outputFormat != nullptr) {
+ Mutexed<Output>::Locked output(mOutput);
+ ALOGD("[%s] onWorkDone: output format changed to %s",
+ mName, outputFormat->debugString().c_str());
+ output->buffers->setFormat(outputFormat);
+
+ AString mediaType;
+ if (outputFormat->findString(KEY_MIME, &mediaType)
+ && mediaType == MIMETYPE_AUDIO_RAW) {
+ int32_t channelCount;
+ int32_t sampleRate;
+ if (outputFormat->findInt32(KEY_CHANNEL_COUNT, &channelCount)
+ && outputFormat->findInt32(KEY_SAMPLE_RATE, &sampleRate)) {
+ output->buffers->updateSkipCutBuffer(sampleRate, channelCount);
+ }
+ }
+ }
+
if ((work->input.ordinal.frameIndex - mFirstValidFrameIndex.load()).peek() < 0) {
// Discard frames from previous generation.
ALOGD("[%s] Discard frames from previous generation.", mName);
@@ -1453,24 +1472,6 @@
}
}
- if (outputFormat != nullptr) {
- Mutexed<Output>::Locked output(mOutput);
- ALOGD("[%s] onWorkDone: output format changed to %s",
- mName, outputFormat->debugString().c_str());
- output->buffers->setFormat(outputFormat);
-
- AString mediaType;
- if (outputFormat->findString(KEY_MIME, &mediaType)
- && mediaType == MIMETYPE_AUDIO_RAW) {
- int32_t channelCount;
- int32_t sampleRate;
- if (outputFormat->findInt32(KEY_CHANNEL_COUNT, &channelCount)
- && outputFormat->findInt32(KEY_SAMPLE_RATE, &sampleRate)) {
- output->buffers->updateSkipCutBuffer(sampleRate, channelCount);
- }
- }
- }
-
int32_t flags = 0;
if (worklet->output.flags & C2FrameData::FLAG_END_OF_STREAM) {
flags |= MediaCodec::BUFFER_FLAG_EOS;
diff --git a/media/codec2/sfplugin/CCodecBufferChannel.h b/media/codec2/sfplugin/CCodecBufferChannel.h
index ee3455d..c0fa138 100644
--- a/media/codec2/sfplugin/CCodecBufferChannel.h
+++ b/media/codec2/sfplugin/CCodecBufferChannel.h
@@ -29,7 +29,6 @@
#include <codec2/hidl/client.h>
#include <media/stagefright/foundation/Mutexed.h>
#include <media/stagefright/CodecBase.h>
-#include <media/ICrypto.h>
#include "CCodecBuffers.h"
#include "InputSurfaceWrapper.h"
diff --git a/media/codec2/sfplugin/CCodecBuffers.cpp b/media/codec2/sfplugin/CCodecBuffers.cpp
index 26c702d..ed8b832 100644
--- a/media/codec2/sfplugin/CCodecBuffers.cpp
+++ b/media/codec2/sfplugin/CCodecBuffers.cpp
@@ -878,9 +878,10 @@
switch (c2buffer->data().type()) {
case C2BufferData::LINEAR: {
uint32_t size = kLinearBufferSize;
- const C2ConstLinearBlock &block = c2buffer->data().linearBlocks().front();
- if (block.size() < kMaxLinearBufferSize / 2) {
- size = block.size() * 2;
+ const std::vector<C2ConstLinearBlock> &linear_blocks = c2buffer->data().linearBlocks();
+ const uint32_t block_size = linear_blocks.front().size();
+ if (block_size < kMaxLinearBufferSize / 2) {
+ size = block_size * 2;
} else {
size = kMaxLinearBufferSize;
}
diff --git a/media/codec2/sfplugin/Codec2Buffer.h b/media/codec2/sfplugin/Codec2Buffer.h
index 36dcab9..6f87101 100644
--- a/media/codec2/sfplugin/Codec2Buffer.h
+++ b/media/codec2/sfplugin/Codec2Buffer.h
@@ -25,7 +25,7 @@
#include <media/hardware/VideoAPI.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/MediaCodecBuffer.h>
-#include <media/ICrypto.h>
+#include <mediadrm/ICrypto.h>
namespace android {
diff --git a/media/codec2/sfplugin/tests/Android.bp b/media/codec2/sfplugin/tests/Android.bp
index be7f55c..b6eb2b4 100644
--- a/media/codec2/sfplugin/tests/Android.bp
+++ b/media/codec2/sfplugin/tests/Android.bp
@@ -33,6 +33,10 @@
"frameworks/av/media/codec2/sfplugin",
],
+ header_libs: [
+ "libmediadrm_headers",
+ ],
+
shared_libs: [
"libbinder",
"libcodec2",
diff --git a/media/codec2/sfplugin/tests/MediaCodec_sanity_test.cpp b/media/codec2/sfplugin/tests/MediaCodec_sanity_test.cpp
index ba3687b..6deede0 100644
--- a/media/codec2/sfplugin/tests/MediaCodec_sanity_test.cpp
+++ b/media/codec2/sfplugin/tests/MediaCodec_sanity_test.cpp
@@ -21,7 +21,7 @@
#include <binder/ProcessState.h>
#include <gtest/gtest.h>
#include <gui/Surface.h>
-#include <media/ICrypto.h>
+#include <mediadrm/ICrypto.h>
#include <media/MediaCodecBuffer.h>
#include <media/hardware/VideoAPI.h>
#include <media/stagefright/MediaCodec.h>
diff --git a/media/codec2/vndk/util/C2InterfaceUtils.cpp b/media/codec2/vndk/util/C2InterfaceUtils.cpp
index 61ec911..0c1729b 100644
--- a/media/codec2/vndk/util/C2InterfaceUtils.cpp
+++ b/media/codec2/vndk/util/C2InterfaceUtils.cpp
@@ -216,9 +216,14 @@
if (limit.contains(minMask) && contains(minMask)) {
values[0] = minMask;
// keep only flags that are covered by limit
- std::remove_if(values.begin(), values.end(), [&limit, minMask](const C2Value::Primitive &v) -> bool {
- T value = v.ref<ValueType>() | minMask;
- return value == minMask || !limit.contains(value); });
+ values.erase(std::remove_if(values.begin(), values.end(),
+ [&limit, minMask](
+ const C2Value::Primitive &v) -> bool {
+ T value = v.ref<ValueType>() | minMask;
+ return value == minMask ||
+ !limit.contains(value);
+ }),
+ values.end());
// we also need to do it vice versa
for (const C2Value::Primitive &v : _mValues) {
T value = v.ref<ValueType>() | minMask;
@@ -264,24 +269,33 @@
template<typename T>
C2SupportedValueSet<T> C2SupportedValueSet<T>::limitedTo(const C2SupportedValueSet<T> &limit) const {
std::vector<C2Value::Primitive> values = _mValues; // make a copy
- std::remove_if(values.begin(), values.end(), [&limit](const C2Value::Primitive &v) -> bool {
- return !limit.contains(v.ref<ValueType>()); });
+ values.erase(std::remove_if(values.begin(), values.end(),
+ [&limit](const C2Value::Primitive &v) -> bool {
+ return !limit.contains(v.ref<ValueType>());
+ }),
+ values.end());
return C2SupportedValueSet(std::move(values));
}
template<typename T>
C2SupportedValueSet<T> C2SupportedValueSet<T>::limitedTo(const C2SupportedRange<T> &limit) const {
std::vector<C2Value::Primitive> values = _mValues; // make a copy
- std::remove_if(values.begin(), values.end(), [&limit](const C2Value::Primitive &v) -> bool {
- return !limit.contains(v.ref<ValueType>()); });
+ values.erase(std::remove_if(values.begin(), values.end(),
+ [&limit](const C2Value::Primitive &v) -> bool {
+ return !limit.contains(v.ref<ValueType>());
+ }),
+ values.end());
return C2SupportedValueSet(std::move(values));
}
template<typename T>
C2SupportedValueSet<T> C2SupportedValueSet<T>::limitedTo(const C2SupportedFlags<T> &limit) const {
std::vector<C2Value::Primitive> values = _mValues; // make a copy
- std::remove_if(values.begin(), values.end(), [&limit](const C2Value::Primitive &v) -> bool {
- return !limit.contains(v.ref<ValueType>()); });
+ values.erase(std::remove_if(values.begin(), values.end(),
+ [&limit](const C2Value::Primitive &v) -> bool {
+ return !limit.contains(v.ref<ValueType>());
+ }),
+ values.end());
return C2SupportedValueSet(std::move(values));
}
diff --git a/media/extractors/aac/Android.bp b/media/extractors/aac/Android.bp
index a58167a..53b394f 100644
--- a/media/extractors/aac/Android.bp
+++ b/media/extractors/aac/Android.bp
@@ -1,4 +1,4 @@
-cc_library_shared {
+cc_library {
srcs: ["AACExtractor.cpp"],
diff --git a/media/extractors/amr/Android.bp b/media/extractors/amr/Android.bp
index 4bd933d..cd76062 100644
--- a/media/extractors/amr/Android.bp
+++ b/media/extractors/amr/Android.bp
@@ -1,4 +1,4 @@
-cc_library_shared {
+cc_library {
srcs: ["AMRExtractor.cpp"],
diff --git a/media/extractors/flac/Android.bp b/media/extractors/flac/Android.bp
index 3a3d051..c669b34 100644
--- a/media/extractors/flac/Android.bp
+++ b/media/extractors/flac/Android.bp
@@ -1,4 +1,4 @@
-cc_library_shared {
+cc_library {
srcs: ["FLACExtractor.cpp"],
diff --git a/media/extractors/flac/FLACExtractor.h b/media/extractors/flac/FLACExtractor.h
index 5a73d20..223d359 100644
--- a/media/extractors/flac/FLACExtractor.h
+++ b/media/extractors/flac/FLACExtractor.h
@@ -17,7 +17,6 @@
#ifndef FLAC_EXTRACTOR_H_
#define FLAC_EXTRACTOR_H_
-#include <media/DataSourceBase.h>
#include <media/MediaExtractorPluginApi.h>
#include <media/MediaExtractorPluginHelper.h>
#include <media/NdkMediaFormat.h>
diff --git a/media/extractors/midi/Android.bp b/media/extractors/midi/Android.bp
index 7d42e70..40c91e7 100644
--- a/media/extractors/midi/Android.bp
+++ b/media/extractors/midi/Android.bp
@@ -1,4 +1,4 @@
-cc_library_shared {
+cc_library {
srcs: ["MidiExtractor.cpp"],
@@ -6,6 +6,10 @@
"frameworks/av/media/libstagefright/include",
],
+ header_libs: [
+ "libmedia_headers",
+ ],
+
shared_libs: [
"liblog",
"libmediandk",
diff --git a/media/extractors/midi/MidiExtractor.h b/media/extractors/midi/MidiExtractor.h
index 2e78086..b486fc6 100644
--- a/media/extractors/midi/MidiExtractor.h
+++ b/media/extractors/midi/MidiExtractor.h
@@ -17,7 +17,6 @@
#ifndef MIDI_EXTRACTOR_H_
#define MIDI_EXTRACTOR_H_
-#include <media/DataSourceBase.h>
#include <media/MediaExtractorPluginApi.h>
#include <media/MediaExtractorPluginHelper.h>
#include <media/stagefright/MediaBufferBase.h>
diff --git a/media/extractors/mkv/Android.bp b/media/extractors/mkv/Android.bp
index 1744d3d..650d79d 100644
--- a/media/extractors/mkv/Android.bp
+++ b/media/extractors/mkv/Android.bp
@@ -1,4 +1,4 @@
-cc_library_shared {
+cc_library {
srcs: ["MatroskaExtractor.cpp"],
diff --git a/media/extractors/mp3/Android.bp b/media/extractors/mp3/Android.bp
index 4e2f248..6f02b0f 100644
--- a/media/extractors/mp3/Android.bp
+++ b/media/extractors/mp3/Android.bp
@@ -1,4 +1,4 @@
-cc_library_shared {
+cc_library {
srcs: [
"MP3Extractor.cpp",
diff --git a/media/extractors/mp4/Android.bp b/media/extractors/mp4/Android.bp
index 1b308aa..d9f11fc 100644
--- a/media/extractors/mp4/Android.bp
+++ b/media/extractors/mp4/Android.bp
@@ -35,7 +35,7 @@
compile_multilib: "first",
}
-cc_library_shared {
+cc_library {
name: "libmp4extractor",
diff --git a/media/extractors/mp4/SampleIterator.cpp b/media/extractors/mp4/SampleIterator.cpp
index 2890b26..85fbf97 100644
--- a/media/extractors/mp4/SampleIterator.cpp
+++ b/media/extractors/mp4/SampleIterator.cpp
@@ -22,7 +22,6 @@
#include <arpa/inet.h>
-#include <media/DataSourceBase.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/ByteUtils.h>
@@ -355,7 +354,7 @@
if (offset > 0) {
*time += offset;
} else {
- *time -= (offset == INT64_MIN ? INT64_MAX : (-offset));
+ *time -= (offset == INT32_MIN ? INT64_MAX : (-offset));
}
*duration = mTTSDuration;
diff --git a/media/extractors/mpeg2/Android.bp b/media/extractors/mpeg2/Android.bp
index 0f0c72c..14bd644 100644
--- a/media/extractors/mpeg2/Android.bp
+++ b/media/extractors/mpeg2/Android.bp
@@ -1,4 +1,4 @@
-cc_library_shared {
+cc_library {
srcs: [
"ExtractorBundle.cpp",
@@ -16,6 +16,7 @@
"android.hardware.cas.native@1.0",
"android.hidl.token@1.0-utils",
"android.hidl.allocator@1.0",
+ "libcrypto",
"libhidlmemory",
"libhidlbase",
"liblog",
@@ -23,13 +24,13 @@
],
header_libs: [
+ "libaudioclient_headers",
"libbase_headers",
"libstagefright_headers",
"libmedia_headers",
],
static_libs: [
- "libcrypto",
"libstagefright_foundation_without_imemory",
"libstagefright_mpeg2support",
"libutils",
diff --git a/media/extractors/mpeg2/MPEG2PSExtractor.cpp b/media/extractors/mpeg2/MPEG2PSExtractor.cpp
index 92ba039..002a855 100644
--- a/media/extractors/mpeg2/MPEG2PSExtractor.cpp
+++ b/media/extractors/mpeg2/MPEG2PSExtractor.cpp
@@ -23,7 +23,6 @@
#include "mpeg2ts/AnotherPacketSource.h"
#include "mpeg2ts/ESQueue.h"
-#include <media/DataSourceBase.h>
#include <media/stagefright/foundation/ABitReader.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/ADebug.h>
diff --git a/media/extractors/ogg/Android.bp b/media/extractors/ogg/Android.bp
index 604ec59..e661b5d 100644
--- a/media/extractors/ogg/Android.bp
+++ b/media/extractors/ogg/Android.bp
@@ -1,4 +1,4 @@
-cc_library_shared {
+cc_library {
srcs: ["OggExtractor.cpp"],
diff --git a/media/extractors/tests/Android.bp b/media/extractors/tests/Android.bp
new file mode 100644
index 0000000..059c308
--- /dev/null
+++ b/media/extractors/tests/Android.bp
@@ -0,0 +1,96 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+cc_test {
+ name: "ExtractorUnitTest",
+ gtest: true,
+
+ srcs: ["ExtractorUnitTest.cpp"],
+
+ static_libs: [
+ "libaacextractor",
+ "libamrextractor",
+ "libmp3extractor",
+ "libwavextractor",
+ "liboggextractor",
+ "libflacextractor",
+ "libmidiextractor",
+ "libmkvextractor",
+ "libmpeg2extractor",
+ "libmp4extractor",
+ "libaudioutils",
+ "libdatasource",
+
+ "libstagefright",
+ "libstagefright_id3",
+ "libstagefright_flacdec",
+ "libstagefright_esds",
+ "libstagefright_mpeg2support",
+ "libstagefright_mpeg2extractor",
+ "libstagefright_foundation",
+ "libstagefright_metadatautils",
+
+ "libmedia_midiiowrapper",
+ "libsonivox",
+ "libvorbisidec",
+ "libwebm",
+ "libFLAC",
+ ],
+
+ shared_libs: [
+ "android.hardware.cas@1.0",
+ "android.hardware.cas.native@1.0",
+ "android.hidl.token@1.0-utils",
+ "android.hidl.allocator@1.0",
+ "libbinder",
+ "libbinder_ndk",
+ "libutils",
+ "liblog",
+ "libcutils",
+ "libmediandk",
+ "libmedia",
+ "libcrypto",
+ "libhidlmemory",
+ "libhidlbase",
+ ],
+
+ include_dirs: [
+ "frameworks/av/media/extractors/",
+ "frameworks/av/media/libstagefright/",
+ ],
+
+ compile_multilib: "first",
+
+ cflags: [
+ "-Werror",
+ "-Wall",
+ ],
+
+ ldflags: [
+ "-Wl",
+ "-Bsymbolic",
+ // to ignore duplicate symbol: GETEXTRACTORDEF
+ "-z muldefs",
+ ],
+
+ sanitize: {
+ cfi: true,
+ misc_undefined: [
+ "unsigned-integer-overflow",
+ "signed-integer-overflow",
+ ],
+ },
+}
diff --git a/media/extractors/tests/AndroidTest.xml b/media/extractors/tests/AndroidTest.xml
new file mode 100644
index 0000000..6bb2c8a
--- /dev/null
+++ b/media/extractors/tests/AndroidTest.xml
@@ -0,0 +1,31 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!-- Copyright (C) 2020 The Android Open Source Project
+
+ Licensed under the Apache License, Version 2.0 (the "License");
+ you may not use this file except in compliance with the License.
+ You may obtain a copy of the License at
+
+ http://www.apache.org/licenses/LICENSE-2.0
+
+ Unless required by applicable law or agreed to in writing, software
+ distributed under the License is distributed on an "AS IS" BASIS,
+ WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ See the License for the specific language governing permissions and
+ limitations under the License.
+-->
+<configuration description="Test module config for extractor unit tests">
+ <option name="test-suite-tag" value="ExtractorUnitTest" />
+ <target_preparer class="com.android.tradefed.targetprep.PushFilePreparer">
+ <option name="cleanup" value="true" />
+ <option name="push" value="ExtractorUnitTest->/data/local/tmp/ExtractorUnitTest" />
+ <option name="push-file"
+ key="https://storage.googleapis.com/android_media/frameworks/av/media/extractors/tests/extractor.zip?unzip=true"
+ value="/data/local/tmp/ExtractorUnitTestRes/" />
+ </target_preparer>
+
+ <test class="com.android.tradefed.testtype.GTest" >
+ <option name="native-test-device-path" value="/data/local/tmp" />
+ <option name="module-name" value="ExtractorUnitTest" />
+ <option name="native-test-flag" value="-P /data/local/tmp/ExtractorUnitTestRes/" />
+ </test>
+</configuration>
diff --git a/media/extractors/tests/ExtractorUnitTest.cpp b/media/extractors/tests/ExtractorUnitTest.cpp
new file mode 100644
index 0000000..64eda75
--- /dev/null
+++ b/media/extractors/tests/ExtractorUnitTest.cpp
@@ -0,0 +1,529 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "ExtractorUnitTest"
+#include <utils/Log.h>
+
+#include <datasource/FileSource.h>
+#include <media/stagefright/MediaBufferGroup.h>
+#include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/MetaDataUtils.h>
+
+#include "aac/AACExtractor.h"
+#include "amr/AMRExtractor.h"
+#include "flac/FLACExtractor.h"
+#include "midi/MidiExtractor.h"
+#include "mkv/MatroskaExtractor.h"
+#include "mp3/MP3Extractor.h"
+#include "mp4/MPEG4Extractor.h"
+#include "mp4/SampleTable.h"
+#include "mpeg2/MPEG2PSExtractor.h"
+#include "mpeg2/MPEG2TSExtractor.h"
+#include "ogg/OggExtractor.h"
+#include "wav/WAVExtractor.h"
+
+#include "ExtractorUnitTestEnvironment.h"
+
+using namespace android;
+
+#define OUTPUT_DUMP_FILE "/data/local/tmp/extractorOutput"
+
+constexpr int32_t kMaxCount = 10;
+constexpr int32_t kOpusSeekPreRollUs = 80000; // 80 ms;
+
+static ExtractorUnitTestEnvironment *gEnv = nullptr;
+
+class ExtractorUnitTest : public ::testing::TestWithParam<pair<string, string>> {
+ public:
+ ExtractorUnitTest() : mInputFp(nullptr), mDataSource(nullptr), mExtractor(nullptr) {}
+
+ ~ExtractorUnitTest() {
+ if (mInputFp) {
+ fclose(mInputFp);
+ mInputFp = nullptr;
+ }
+ if (mDataSource) {
+ mDataSource.clear();
+ mDataSource = nullptr;
+ }
+ if (mExtractor) {
+ delete mExtractor;
+ mExtractor = nullptr;
+ }
+ }
+
+ virtual void SetUp() override {
+ mExtractorName = unknown_comp;
+ mDisableTest = false;
+
+ static const std::map<std::string, standardExtractors> mapExtractor = {
+ {"aac", AAC}, {"amr", AMR}, {"mp3", MP3}, {"ogg", OGG},
+ {"wav", WAV}, {"mkv", MKV}, {"flac", FLAC}, {"midi", MIDI},
+ {"mpeg4", MPEG4}, {"mpeg2ts", MPEG2TS}, {"mpeg2ps", MPEG2PS}};
+ // Find the component type
+ string writerFormat = GetParam().first;
+ if (mapExtractor.find(writerFormat) != mapExtractor.end()) {
+ mExtractorName = mapExtractor.at(writerFormat);
+ }
+ if (mExtractorName == standardExtractors::unknown_comp) {
+ cout << "[ WARN ] Test Skipped. Invalid extractor\n";
+ mDisableTest = true;
+ }
+ }
+
+ int32_t setDataSource(string inputFileName);
+
+ int32_t createExtractor();
+
+ enum standardExtractors {
+ AAC,
+ AMR,
+ FLAC,
+ MIDI,
+ MKV,
+ MP3,
+ MPEG4,
+ MPEG2PS,
+ MPEG2TS,
+ OGG,
+ WAV,
+ unknown_comp,
+ };
+
+ bool mDisableTest;
+ standardExtractors mExtractorName;
+
+ FILE *mInputFp;
+ sp<DataSource> mDataSource;
+ MediaExtractorPluginHelper *mExtractor;
+};
+
+int32_t ExtractorUnitTest::setDataSource(string inputFileName) {
+ mInputFp = fopen(inputFileName.c_str(), "rb");
+ if (!mInputFp) {
+ ALOGE("Unable to open input file for reading");
+ return -1;
+ }
+ struct stat buf;
+ stat(inputFileName.c_str(), &buf);
+ int32_t fd = fileno(mInputFp);
+ mDataSource = new FileSource(dup(fd), 0, buf.st_size);
+ if (!mDataSource) return -1;
+ return 0;
+}
+
+int32_t ExtractorUnitTest::createExtractor() {
+ switch (mExtractorName) {
+ case AAC:
+ mExtractor = new AACExtractor(new DataSourceHelper(mDataSource->wrap()), 0);
+ break;
+ case AMR:
+ mExtractor = new AMRExtractor(new DataSourceHelper(mDataSource->wrap()));
+ break;
+ case MP3:
+ mExtractor = new MP3Extractor(new DataSourceHelper(mDataSource->wrap()), nullptr);
+ break;
+ case OGG:
+ mExtractor = new OggExtractor(new DataSourceHelper(mDataSource->wrap()));
+ break;
+ case WAV:
+ mExtractor = new WAVExtractor(new DataSourceHelper(mDataSource->wrap()));
+ break;
+ case MKV:
+ mExtractor = new MatroskaExtractor(new DataSourceHelper(mDataSource->wrap()));
+ break;
+ case FLAC:
+ mExtractor = new FLACExtractor(new DataSourceHelper(mDataSource->wrap()));
+ break;
+ case MPEG4:
+ mExtractor = new MPEG4Extractor(new DataSourceHelper(mDataSource->wrap()));
+ break;
+ case MPEG2TS:
+ mExtractor = new MPEG2TSExtractor(new DataSourceHelper(mDataSource->wrap()));
+ break;
+ case MPEG2PS:
+ mExtractor = new MPEG2PSExtractor(new DataSourceHelper(mDataSource->wrap()));
+ break;
+ case MIDI:
+ mExtractor = new MidiExtractor(mDataSource->wrap());
+ break;
+ default:
+ return -1;
+ }
+ if (!mExtractor) return -1;
+ return 0;
+}
+
+void getSeekablePoints(vector<int64_t> &seekablePoints, MediaTrackHelper *track) {
+ int32_t status = 0;
+ if (!seekablePoints.empty()) {
+ seekablePoints.clear();
+ }
+ int64_t timeStamp;
+ while (status != AMEDIA_ERROR_END_OF_STREAM) {
+ MediaBufferHelper *buffer = nullptr;
+ status = track->read(&buffer);
+ if (buffer) {
+ AMediaFormat *metaData = buffer->meta_data();
+ int32_t isSync = 0;
+ AMediaFormat_getInt32(metaData, AMEDIAFORMAT_KEY_IS_SYNC_FRAME, &isSync);
+ if (isSync) {
+ AMediaFormat_getInt64(metaData, AMEDIAFORMAT_KEY_TIME_US, &timeStamp);
+ seekablePoints.push_back(timeStamp);
+ }
+ buffer->release();
+ }
+ }
+}
+
+TEST_P(ExtractorUnitTest, CreateExtractorTest) {
+ if (mDisableTest) return;
+
+ ALOGV("Checks if a valid extractor is created for a given input file");
+ string inputFileName = gEnv->getRes() + GetParam().second;
+
+ ASSERT_EQ(setDataSource(inputFileName), 0)
+ << "SetDataSource failed for" << GetParam().first << "extractor";
+
+ ASSERT_EQ(createExtractor(), 0)
+ << "Extractor creation failed for" << GetParam().first << "extractor";
+
+ // A valid extractor instace should return success for following calls
+ ASSERT_GT(mExtractor->countTracks(), 0);
+
+ AMediaFormat *format = AMediaFormat_new();
+ ASSERT_NE(format, nullptr) << "AMediaFormat_new returned null AMediaformat";
+
+ ASSERT_EQ(mExtractor->getMetaData(format), AMEDIA_OK);
+ AMediaFormat_delete(format);
+}
+
+TEST_P(ExtractorUnitTest, ExtractorTest) {
+ if (mDisableTest) return;
+
+ ALOGV("Validates %s Extractor for a given input file", GetParam().first.c_str());
+ string inputFileName = gEnv->getRes() + GetParam().second;
+
+ int32_t status = setDataSource(inputFileName);
+ ASSERT_EQ(status, 0) << "SetDataSource failed for" << GetParam().first << "extractor";
+
+ status = createExtractor();
+ ASSERT_EQ(status, 0) << "Extractor creation failed for" << GetParam().first << "extractor";
+
+ int32_t numTracks = mExtractor->countTracks();
+ ASSERT_GT(numTracks, 0) << "Extractor didn't find any track for the given clip";
+
+ for (int32_t idx = 0; idx < numTracks; idx++) {
+ MediaTrackHelper *track = mExtractor->getTrack(idx);
+ ASSERT_NE(track, nullptr) << "Failed to get track for index " << idx;
+
+ CMediaTrack *cTrack = wrap(track);
+ ASSERT_NE(cTrack, nullptr) << "Failed to get track wrapper for index " << idx;
+
+ MediaBufferGroup *bufferGroup = new MediaBufferGroup();
+ status = cTrack->start(track, bufferGroup->wrap());
+ ASSERT_EQ(OK, (media_status_t)status) << "Failed to start the track";
+
+ FILE *outFp = fopen((OUTPUT_DUMP_FILE + to_string(idx)).c_str(), "wb");
+ if (!outFp) {
+ ALOGW("Unable to open output file for dumping extracted stream");
+ }
+
+ while (status != AMEDIA_ERROR_END_OF_STREAM) {
+ MediaBufferHelper *buffer = nullptr;
+ status = track->read(&buffer);
+ ALOGV("track->read Status = %d buffer %p", status, buffer);
+ if (buffer) {
+ ALOGV("buffer->data %p buffer->size() %zu buffer->range_length() %zu",
+ buffer->data(), buffer->size(), buffer->range_length());
+ if (outFp) fwrite(buffer->data(), 1, buffer->range_length(), outFp);
+ buffer->release();
+ }
+ }
+ if (outFp) fclose(outFp);
+ status = cTrack->stop(track);
+ ASSERT_EQ(OK, status) << "Failed to stop the track";
+ delete bufferGroup;
+ delete track;
+ }
+}
+
+TEST_P(ExtractorUnitTest, MetaDataComparisonTest) {
+ if (mDisableTest) return;
+
+ ALOGV("Validates Extractor's meta data for a given input file");
+ string inputFileName = gEnv->getRes() + GetParam().second;
+
+ int32_t status = setDataSource(inputFileName);
+ ASSERT_EQ(status, 0) << "SetDataSource failed for" << GetParam().first << "extractor";
+
+ status = createExtractor();
+ ASSERT_EQ(status, 0) << "Extractor creation failed for" << GetParam().first << "extractor";
+
+ int32_t numTracks = mExtractor->countTracks();
+ ASSERT_GT(numTracks, 0) << "Extractor didn't find any track for the given clip";
+
+ AMediaFormat *extractorFormat = AMediaFormat_new();
+ ASSERT_NE(extractorFormat, nullptr) << "AMediaFormat_new returned null AMediaformat";
+ AMediaFormat *trackFormat = AMediaFormat_new();
+ ASSERT_NE(trackFormat, nullptr) << "AMediaFormat_new returned null AMediaformat";
+
+ for (int32_t idx = 0; idx < numTracks; idx++) {
+ MediaTrackHelper *track = mExtractor->getTrack(idx);
+ ASSERT_NE(track, nullptr) << "Failed to get track for index " << idx;
+
+ CMediaTrack *cTrack = wrap(track);
+ ASSERT_NE(cTrack, nullptr) << "Failed to get track wrapper for index " << idx;
+
+ MediaBufferGroup *bufferGroup = new MediaBufferGroup();
+ status = cTrack->start(track, bufferGroup->wrap());
+ ASSERT_EQ(OK, (media_status_t)status) << "Failed to start the track";
+
+ status = mExtractor->getTrackMetaData(extractorFormat, idx, 1);
+ ASSERT_EQ(OK, (media_status_t)status) << "Failed to get trackMetaData";
+
+ status = track->getFormat(trackFormat);
+ ASSERT_EQ(OK, (media_status_t)status) << "Failed to get track meta data";
+
+ const char *extractorMime, *trackMime;
+ AMediaFormat_getString(extractorFormat, AMEDIAFORMAT_KEY_MIME, &extractorMime);
+ AMediaFormat_getString(trackFormat, AMEDIAFORMAT_KEY_MIME, &trackMime);
+ ASSERT_TRUE(!strcmp(extractorMime, trackMime))
+ << "Extractor's format doesn't match track format";
+
+ if (!strncmp(extractorMime, "audio/", 6)) {
+ int32_t exSampleRate, exChannelCount;
+ int32_t trackSampleRate, trackChannelCount;
+ ASSERT_TRUE(AMediaFormat_getInt32(extractorFormat, AMEDIAFORMAT_KEY_CHANNEL_COUNT,
+ &exChannelCount));
+ ASSERT_TRUE(AMediaFormat_getInt32(extractorFormat, AMEDIAFORMAT_KEY_SAMPLE_RATE,
+ &exSampleRate));
+ ASSERT_TRUE(AMediaFormat_getInt32(trackFormat, AMEDIAFORMAT_KEY_CHANNEL_COUNT,
+ &trackChannelCount));
+ ASSERT_TRUE(AMediaFormat_getInt32(trackFormat, AMEDIAFORMAT_KEY_SAMPLE_RATE,
+ &trackSampleRate));
+ ASSERT_EQ(exChannelCount, trackChannelCount) << "ChannelCount not as expected";
+ ASSERT_EQ(exSampleRate, trackSampleRate) << "SampleRate not as expected";
+ } else {
+ int32_t exWidth, exHeight;
+ int32_t trackWidth, trackHeight;
+ ASSERT_TRUE(AMediaFormat_getInt32(extractorFormat, AMEDIAFORMAT_KEY_WIDTH, &exWidth));
+ ASSERT_TRUE(AMediaFormat_getInt32(extractorFormat, AMEDIAFORMAT_KEY_HEIGHT, &exHeight));
+ ASSERT_TRUE(AMediaFormat_getInt32(trackFormat, AMEDIAFORMAT_KEY_WIDTH, &trackWidth));
+ ASSERT_TRUE(AMediaFormat_getInt32(trackFormat, AMEDIAFORMAT_KEY_HEIGHT, &trackHeight));
+ ASSERT_EQ(exWidth, trackWidth) << "Width not as expected";
+ ASSERT_EQ(exHeight, trackHeight) << "Height not as expected";
+ }
+ status = cTrack->stop(track);
+ ASSERT_EQ(OK, status) << "Failed to stop the track";
+ delete bufferGroup;
+ delete track;
+ }
+ AMediaFormat_delete(trackFormat);
+ AMediaFormat_delete(extractorFormat);
+}
+
+TEST_P(ExtractorUnitTest, MultipleStartStopTest) {
+ if (mDisableTest) return;
+
+ ALOGV("Test %s extractor for multiple start and stop calls", GetParam().first.c_str());
+ string inputFileName = gEnv->getRes() + GetParam().second;
+
+ int32_t status = setDataSource(inputFileName);
+ ASSERT_EQ(status, 0) << "SetDataSource failed for" << GetParam().first << "extractor";
+
+ status = createExtractor();
+ ASSERT_EQ(status, 0) << "Extractor creation failed for" << GetParam().first << "extractor";
+
+ int32_t numTracks = mExtractor->countTracks();
+ ASSERT_GT(numTracks, 0) << "Extractor didn't find any track for the given clip";
+
+ // start/stop the tracks multiple times
+ for (int32_t count = 0; count < kMaxCount; count++) {
+ for (int32_t idx = 0; idx < numTracks; idx++) {
+ MediaTrackHelper *track = mExtractor->getTrack(idx);
+ ASSERT_NE(track, nullptr) << "Failed to get track for index " << idx;
+
+ CMediaTrack *cTrack = wrap(track);
+ ASSERT_NE(cTrack, nullptr) << "Failed to get track wrapper for index " << idx;
+
+ MediaBufferGroup *bufferGroup = new MediaBufferGroup();
+ status = cTrack->start(track, bufferGroup->wrap());
+ ASSERT_EQ(OK, (media_status_t)status) << "Failed to start the track";
+ MediaBufferHelper *buffer = nullptr;
+ status = track->read(&buffer);
+ if (buffer) {
+ ALOGV("buffer->data %p buffer->size() %zu buffer->range_length() %zu",
+ buffer->data(), buffer->size(), buffer->range_length());
+ buffer->release();
+ }
+ status = cTrack->stop(track);
+ ASSERT_EQ(OK, status) << "Failed to stop the track";
+ delete bufferGroup;
+ delete track;
+ }
+ }
+}
+
+TEST_P(ExtractorUnitTest, SeekTest) {
+ // Flac, Midi and Wav extractor can give samples from any pts and mark the given sample as
+ // sync frame. So, this seek test is not applicable to these extractors
+ if (mDisableTest || mExtractorName == FLAC || mExtractorName == WAV || mExtractorName == MIDI) {
+ return;
+ }
+
+ ALOGV("Validates %s Extractor behaviour for different seek modes", GetParam().first.c_str());
+ string inputFileName = gEnv->getRes() + GetParam().second;
+
+ int32_t status = setDataSource(inputFileName);
+ ASSERT_EQ(status, 0) << "SetDataSource failed for" << GetParam().first << "extractor";
+
+ status = createExtractor();
+ ASSERT_EQ(status, 0) << "Extractor creation failed for" << GetParam().first << "extractor";
+
+ int32_t numTracks = mExtractor->countTracks();
+ ASSERT_GT(numTracks, 0) << "Extractor didn't find any track for the given clip";
+
+ uint32_t seekFlag = mExtractor->flags();
+ if (!(seekFlag & MediaExtractorPluginHelper::CAN_SEEK)) {
+ cout << "[ WARN ] Test Skipped. " << GetParam().first
+ << " Extractor doesn't support seek\n";
+ return;
+ }
+
+ vector<int64_t> seekablePoints;
+ for (int32_t idx = 0; idx < numTracks; idx++) {
+ MediaTrackHelper *track = mExtractor->getTrack(idx);
+ ASSERT_NE(track, nullptr) << "Failed to get track for index " << idx;
+
+ CMediaTrack *cTrack = wrap(track);
+ ASSERT_NE(cTrack, nullptr) << "Failed to get track wrapper for index " << idx;
+
+ // Get all the seekable points of a given input
+ MediaBufferGroup *bufferGroup = new MediaBufferGroup();
+ status = cTrack->start(track, bufferGroup->wrap());
+ ASSERT_EQ(OK, (media_status_t)status) << "Failed to start the track";
+ getSeekablePoints(seekablePoints, track);
+ ASSERT_GT(seekablePoints.size(), 0)
+ << "Failed to get seekable points for " << GetParam().first << " extractor";
+
+ AMediaFormat *trackFormat = AMediaFormat_new();
+ ASSERT_NE(trackFormat, nullptr) << "AMediaFormat_new returned null format";
+ status = track->getFormat(trackFormat);
+ ASSERT_EQ(OK, (media_status_t)status) << "Failed to get track meta data";
+
+ bool isOpus = false;
+ const char *mime;
+ AMediaFormat_getString(trackFormat, AMEDIAFORMAT_KEY_MIME, &mime);
+ if (!strcmp(mime, "audio/opus")) isOpus = true;
+ AMediaFormat_delete(trackFormat);
+
+ int32_t seekIdx = 0;
+ size_t seekablePointsSize = seekablePoints.size();
+ for (int32_t mode = CMediaTrackReadOptions::SEEK_PREVIOUS_SYNC;
+ mode <= CMediaTrackReadOptions::SEEK_CLOSEST; mode++) {
+ for (int32_t seekCount = 0; seekCount < kMaxCount; seekCount++) {
+ seekIdx = rand() % seekablePointsSize + 1;
+ if (seekIdx >= seekablePointsSize) seekIdx = seekablePointsSize - 1;
+
+ int64_t seekToTimeStamp = seekablePoints[seekIdx];
+ if (seekablePointsSize > 1) {
+ int64_t prevTimeStamp = seekablePoints[seekIdx - 1];
+ seekToTimeStamp = seekToTimeStamp - ((seekToTimeStamp - prevTimeStamp) >> 3);
+ }
+
+ // Opus has a seekPreRollUs. TimeStamp returned by the
+ // extractor is calculated based on (seekPts - seekPreRollUs).
+ // So we add the preRoll value to the timeStamp we want to seek to.
+ if (isOpus) {
+ seekToTimeStamp += kOpusSeekPreRollUs;
+ }
+
+ MediaTrackHelper::ReadOptions *options = new MediaTrackHelper::ReadOptions(
+ mode | CMediaTrackReadOptions::SEEK, seekToTimeStamp);
+ ASSERT_NE(options, nullptr) << "Cannot create read option";
+
+ MediaBufferHelper *buffer = nullptr;
+ status = track->read(&buffer, options);
+ if (status == AMEDIA_ERROR_END_OF_STREAM) {
+ delete options;
+ continue;
+ }
+ if (buffer) {
+ AMediaFormat *metaData = buffer->meta_data();
+ int64_t timeStamp;
+ AMediaFormat_getInt64(metaData, AMEDIAFORMAT_KEY_TIME_US, &timeStamp);
+ buffer->release();
+
+ // CMediaTrackReadOptions::SEEK is 8. Using mask 0111b to get true modes
+ switch (mode & 0x7) {
+ case CMediaTrackReadOptions::SEEK_PREVIOUS_SYNC:
+ if (seekablePointsSize == 1) {
+ EXPECT_EQ(timeStamp, seekablePoints[seekIdx]);
+ } else {
+ EXPECT_EQ(timeStamp, seekablePoints[seekIdx - 1]);
+ }
+ break;
+ case CMediaTrackReadOptions::SEEK_NEXT_SYNC:
+ case CMediaTrackReadOptions::SEEK_CLOSEST_SYNC:
+ case CMediaTrackReadOptions::SEEK_CLOSEST:
+ EXPECT_EQ(timeStamp, seekablePoints[seekIdx]);
+ break;
+ default:
+ break;
+ }
+ }
+ delete options;
+ }
+ }
+ status = cTrack->stop(track);
+ ASSERT_EQ(OK, status) << "Failed to stop the track";
+ delete bufferGroup;
+ delete track;
+ }
+ seekablePoints.clear();
+}
+
+INSTANTIATE_TEST_SUITE_P(ExtractorUnitTestAll, ExtractorUnitTest,
+ ::testing::Values(make_pair("aac", "loudsoftaac.aac"),
+ make_pair("amr", "testamr.amr"),
+ make_pair("amr", "amrwb.wav"),
+ make_pair("ogg", "john_cage.ogg"),
+ make_pair("wav", "monotestgsm.wav"),
+ make_pair("mpeg2ts", "segment000001.ts"),
+ make_pair("flac", "sinesweepflac.flac"),
+ make_pair("ogg", "testopus.opus"),
+ make_pair("midi", "midi_a.mid"),
+ make_pair("mkv", "sinesweepvorbis.mkv"),
+ make_pair("mpeg4", "sinesweepoggmp4.mp4"),
+ make_pair("mp3", "sinesweepmp3lame.mp3"),
+ make_pair("mkv", "swirl_144x136_vp9.webm"),
+ make_pair("mkv", "swirl_144x136_vp8.webm"),
+ make_pair("mpeg2ps", "swirl_144x136_mpeg2.mpg"),
+ make_pair("mpeg4", "swirl_132x130_mpeg4.mp4")));
+
+int main(int argc, char **argv) {
+ gEnv = new ExtractorUnitTestEnvironment();
+ ::testing::AddGlobalTestEnvironment(gEnv);
+ ::testing::InitGoogleTest(&argc, argv);
+ int status = gEnv->initFromOptions(argc, argv);
+ if (status == 0) {
+ status = RUN_ALL_TESTS();
+ ALOGV("Test result = %d\n", status);
+ }
+ return status;
+}
diff --git a/media/extractors/tests/ExtractorUnitTestEnvironment.h b/media/extractors/tests/ExtractorUnitTestEnvironment.h
new file mode 100644
index 0000000..fce8fc2
--- /dev/null
+++ b/media/extractors/tests/ExtractorUnitTestEnvironment.h
@@ -0,0 +1,72 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __EXTRACTOR_UNIT_TEST_ENVIRONMENT_H__
+#define __EXTRACTOR_UNIT_TEST_ENVIRONMENT_H__
+
+#include <gtest/gtest.h>
+
+#include <getopt.h>
+
+using namespace std;
+
+class ExtractorUnitTestEnvironment : public ::testing::Environment {
+ public:
+ ExtractorUnitTestEnvironment() : res("/data/local/tmp/") {}
+
+ // Parses the command line arguments
+ int initFromOptions(int argc, char **argv);
+
+ void setRes(const char *_res) { res = _res; }
+
+ const string getRes() const { return res; }
+
+ private:
+ string res;
+};
+
+int ExtractorUnitTestEnvironment::initFromOptions(int argc, char **argv) {
+ static struct option options[] = {{"res", required_argument, 0, 'P'}, {0, 0, 0, 0}};
+
+ while (true) {
+ int index = 0;
+ int c = getopt_long(argc, argv, "P:", options, &index);
+ if (c == -1) {
+ break;
+ }
+
+ switch (c) {
+ case 'P':
+ setRes(optarg);
+ break;
+ default:
+ break;
+ }
+ }
+
+ if (optind < argc) {
+ fprintf(stderr,
+ "unrecognized option: %s\n\n"
+ "usage: %s <gtest options> <test options>\n\n"
+ "test options are:\n\n"
+ "-P, --path: Resource files directory location\n",
+ argv[optind ?: 1], argv[0]);
+ return 2;
+ }
+ return 0;
+}
+
+#endif // __EXTRACTOR_UNIT_TEST_ENVIRONMENT_H__
diff --git a/media/extractors/tests/README.md b/media/extractors/tests/README.md
new file mode 100644
index 0000000..69538b6
--- /dev/null
+++ b/media/extractors/tests/README.md
@@ -0,0 +1,39 @@
+## Media Testing ##
+---
+#### Extractor :
+The Extractor Test Suite validates the extractors available in the device.
+
+Run the following steps to build the test suite:
+```
+m ExtractorUnitTest
+```
+
+The 32-bit binaries will be created in the following path : ${OUT}/data/nativetest/
+
+The 64-bit binaries will be created in the following path : ${OUT}/data/nativetest64/
+
+To test 64-bit binary push binaries from nativetest64.
+```
+adb push ${OUT}/data/nativetest64/ExtractorUnitTest/ExtractorUnitTest /data/local/tmp/
+```
+
+To test 32-bit binary push binaries from nativetest.
+```
+adb push ${OUT}/data/nativetest/ExtractorUnitTest/ExtractorUnitTest /data/local/tmp/
+```
+
+The resource file for the tests is taken from [here](https://storage.googleapis.com/android_media/frameworks/av/media/extractors/tests/extractor.zip). Download, unzip and push these files into device for testing.
+
+```
+adb push extractor /data/local/tmp/
+```
+
+usage: ExtractorUnitTest -P \<path_to_folder\>
+```
+adb shell /data/local/tmp/ExtractorUnitTest -P /data/local/tmp/extractor/
+```
+Alternatively, the test can also be run using atest command.
+
+```
+atest ExtractorUnitTest -- --enable-module-dynamic-download=true
+```
diff --git a/media/extractors/wav/Android.bp b/media/extractors/wav/Android.bp
index 7e89271..51e3c31 100644
--- a/media/extractors/wav/Android.bp
+++ b/media/extractors/wav/Android.bp
@@ -1,4 +1,4 @@
-cc_library_shared {
+cc_library {
srcs: ["WAVExtractor.cpp"],
diff --git a/media/libaaudio/Android.bp b/media/libaaudio/Android.bp
index 16958f9..140052f 100644
--- a/media/libaaudio/Android.bp
+++ b/media/libaaudio/Android.bp
@@ -24,7 +24,7 @@
ndk_library {
name: "libaaudio",
// deliberately includes symbols from AAudioTesting.h
- symbol_file: "libaaudio.map.txt",
+ symbol_file: "src/libaaudio.map.txt",
first_version: "26",
unversioned_until: "current",
}
@@ -32,6 +32,5 @@
cc_library_headers {
name: "libaaudio_headers",
export_include_dirs: ["include"],
- version_script: "libaaudio.map.txt",
}
diff --git a/media/libaaudio/examples/input_monitor/Android.bp b/media/libaaudio/examples/input_monitor/Android.bp
index 5d399b5..d8c5843 100644
--- a/media/libaaudio/examples/input_monitor/Android.bp
+++ b/media/libaaudio/examples/input_monitor/Android.bp
@@ -5,7 +5,6 @@
cflags: ["-Wall", "-Werror"],
shared_libs: ["libaaudio"],
header_libs: ["libaaudio_example_utils"],
- pack_relocations: false,
}
cc_test {
@@ -15,5 +14,4 @@
cflags: ["-Wall", "-Werror"],
shared_libs: ["libaaudio"],
header_libs: ["libaaudio_example_utils"],
- pack_relocations: false,
}
diff --git a/media/libaaudio/examples/loopback/Android.bp b/media/libaaudio/examples/loopback/Android.bp
index 53e5020..5b7d956 100644
--- a/media/libaaudio/examples/loopback/Android.bp
+++ b/media/libaaudio/examples/loopback/Android.bp
@@ -9,5 +9,4 @@
"libaudioutils",
],
header_libs: ["libaaudio_example_utils"],
- pack_relocations: false,
}
diff --git a/media/libaaudio/examples/write_sine/Android.bp b/media/libaaudio/examples/write_sine/Android.bp
index cc80861..aa25e67 100644
--- a/media/libaaudio/examples/write_sine/Android.bp
+++ b/media/libaaudio/examples/write_sine/Android.bp
@@ -4,7 +4,6 @@
cflags: ["-Wall", "-Werror"],
shared_libs: ["libaaudio"],
header_libs: ["libaaudio_example_utils"],
- pack_relocations: false,
}
cc_test {
@@ -13,5 +12,4 @@
cflags: ["-Wall", "-Werror"],
shared_libs: ["libaaudio"],
header_libs: ["libaaudio_example_utils"],
- pack_relocations: false,
}
diff --git a/media/libaaudio/include/aaudio/AAudio.h b/media/libaaudio/include/aaudio/AAudio.h
index ee5d089..8173e3c 100644
--- a/media/libaaudio/include/aaudio/AAudio.h
+++ b/media/libaaudio/include/aaudio/AAudio.h
@@ -472,6 +472,8 @@
* This is intended for developers to use when debugging.
* It is not for display to users.
*
+ * Available since API level 26.
+ *
* @return pointer to a text representation of an AAudio result code.
*/
AAUDIO_API const char * AAudio_convertResultToText(aaudio_result_t returnCode) __INTRODUCED_IN(26);
@@ -482,6 +484,8 @@
* This is intended for developers to use when debugging.
* It is not for display to users.
*
+ * Available since API level 26.
+ *
* @return pointer to a text representation of an AAudio state.
*/
AAUDIO_API const char * AAudio_convertStreamStateToText(aaudio_stream_state_t state)
@@ -502,6 +506,8 @@
* chosen by the device when it is opened.
*
* AAudioStreamBuilder_delete() must be called when you are done using the builder.
+ *
+ * Available since API level 26.
*/
AAUDIO_API aaudio_result_t AAudio_createStreamBuilder(AAudioStreamBuilder** builder)
__INTRODUCED_IN(26);
@@ -513,6 +519,8 @@
* The default, if you do not call this function, is {@link #AAUDIO_UNSPECIFIED},
* in which case the primary device will be used.
*
+ * Available since API level 26.
+ *
* @param builder reference provided by AAudio_createStreamBuilder()
* @param deviceId device identifier or {@link #AAUDIO_UNSPECIFIED}
*/
@@ -530,6 +538,8 @@
* If an exact value is specified then an opened stream will use that value.
* If a stream cannot be opened with the specified value then the open will fail.
*
+ * Available since API level 26.
+ *
* @param builder reference provided by AAudio_createStreamBuilder()
* @param sampleRate frames per second. Common rates include 44100 and 48000 Hz.
*/
@@ -547,6 +557,8 @@
* If an exact value is specified then an opened stream will use that value.
* If a stream cannot be opened with the specified value then the open will fail.
*
+ * Available since API level 26.
+ *
* @param builder reference provided by AAudio_createStreamBuilder()
* @param channelCount Number of channels desired.
*/
@@ -556,6 +568,8 @@
/**
* Identical to AAudioStreamBuilder_setChannelCount().
*
+ * Available since API level 26.
+ *
* @param builder reference provided by AAudio_createStreamBuilder()
* @param samplesPerFrame Number of samples in a frame.
*/
@@ -573,6 +587,8 @@
* If an exact value is specified then an opened stream will use that value.
* If a stream cannot be opened with the specified value then the open will fail.
*
+ * Available since API level 26.
+ *
* @param builder reference provided by AAudio_createStreamBuilder()
* @param format common formats are {@link #AAUDIO_FORMAT_PCM_FLOAT} and
* {@link #AAUDIO_FORMAT_PCM_I16}.
@@ -588,6 +604,8 @@
* The requested sharing mode may not be available.
* The application can query for the actual mode after the stream is opened.
*
+ * Available since API level 26.
+ *
* @param builder reference provided by AAudio_createStreamBuilder()
* @param sharingMode {@link #AAUDIO_SHARING_MODE_SHARED} or {@link #AAUDIO_SHARING_MODE_EXCLUSIVE}
*/
@@ -599,6 +617,8 @@
*
* The default, if you do not call this function, is {@link #AAUDIO_DIRECTION_OUTPUT}.
*
+ * Available since API level 26.
+ *
* @param builder reference provided by AAudio_createStreamBuilder()
* @param direction {@link #AAUDIO_DIRECTION_OUTPUT} or {@link #AAUDIO_DIRECTION_INPUT}
*/
@@ -611,6 +631,8 @@
*
* The default, if you do not call this function, is {@link #AAUDIO_UNSPECIFIED}.
*
+ * Available since API level 26.
+ *
* @param builder reference provided by AAudio_createStreamBuilder()
* @param numFrames the desired buffer capacity in frames or {@link #AAUDIO_UNSPECIFIED}
*/
@@ -629,6 +651,8 @@
* You can call AAudioStream_getPerformanceMode()
* to find out the final mode for the stream.
*
+ * Available since API level 26.
+ *
* @param builder reference provided by AAudio_createStreamBuilder()
* @param mode the desired performance mode, eg. {@link #AAUDIO_PERFORMANCE_MODE_LOW_LATENCY}
*/
@@ -644,7 +668,7 @@
*
* The default, if you do not call this function, is {@link #AAUDIO_USAGE_MEDIA}.
*
- * Added in API level 28.
+ * Available since API level 28.
*
* @param builder reference provided by AAudio_createStreamBuilder()
* @param usage the desired usage, eg. {@link #AAUDIO_USAGE_GAME}
@@ -661,7 +685,7 @@
*
* The default, if you do not call this function, is {@link #AAUDIO_CONTENT_TYPE_MUSIC}.
*
- * Added in API level 28.
+ * Available since API level 28.
*
* @param builder reference provided by AAudio_createStreamBuilder()
* @param contentType the type of audio data, eg. {@link #AAUDIO_CONTENT_TYPE_SPEECH}
@@ -681,7 +705,7 @@
* That is because VOICE_RECOGNITION is the preset with the lowest latency
* on many platforms.
*
- * Added in API level 28.
+ * Available since API level 28.
*
* @param builder reference provided by AAudio_createStreamBuilder()
* @param inputPreset the desired configuration for recording
@@ -697,7 +721,7 @@
* Note that an application can also set its global policy, in which case the most restrictive
* policy is always applied. See {@link android.media.AudioAttributes#setAllowedCapturePolicy(int)}
*
- * Added in API level 29.
+ * Available since API level 29.
*
* @param builder reference provided by AAudio_createStreamBuilder()
* @param inputPreset the desired level of opt-out from being captured.
@@ -727,7 +751,7 @@
*
* Allocated session IDs will always be positive and nonzero.
*
- * Added in API level 28.
+ * Available since API level 28.
*
* @param builder reference provided by AAudio_createStreamBuilder()
* @param sessionId an allocated sessionID or {@link #AAUDIO_SESSION_ID_ALLOCATE}
@@ -826,6 +850,8 @@
*
* Note that the AAudio callbacks will never be called simultaneously from multiple threads.
*
+ * Available since API level 26.
+ *
* @param builder reference provided by AAudio_createStreamBuilder()
* @param callback pointer to a function that will process audio data.
* @param userData pointer to an application data structure that will be passed
@@ -854,6 +880,8 @@
* If you do call this function then the requested size should be less than
* half the buffer capacity, to allow double buffering.
*
+ * Available since API level 26.
+ *
* @param builder reference provided by AAudio_createStreamBuilder()
* @param numFrames the desired buffer size in frames or {@link #AAUDIO_UNSPECIFIED}
*/
@@ -905,6 +933,8 @@
*
* Note that the AAudio callbacks will never be called simultaneously from multiple threads.
*
+ * Available since API level 26.
+ *
* @param builder reference provided by AAudio_createStreamBuilder()
* @param callback pointer to a function that will be called if an error occurs.
* @param userData pointer to an application data structure that will be passed
@@ -919,6 +949,8 @@
* AAudioStream_close() must be called when finished with the stream to recover
* the memory and to free the associated resources.
*
+ * Available since API level 26.
+ *
* @param builder reference provided by AAudio_createStreamBuilder()
* @param stream pointer to a variable to receive the new stream reference
* @return {@link #AAUDIO_OK} or a negative error.
@@ -929,6 +961,8 @@
/**
* Delete the resources associated with the StreamBuilder.
*
+ * Available since API level 26.
+ *
* @param builder reference provided by AAudio_createStreamBuilder()
* @return {@link #AAUDIO_OK} or a negative error.
*/
@@ -942,6 +976,8 @@
/**
* Free the resources associated with a stream created by AAudioStreamBuilder_openStream()
*
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return {@link #AAUDIO_OK} or a negative error.
*/
@@ -954,6 +990,8 @@
* After this call the state will be in {@link #AAUDIO_STREAM_STATE_STARTING} or
* {@link #AAUDIO_STREAM_STATE_STARTED}.
*
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return {@link #AAUDIO_OK} or a negative error.
*/
@@ -969,6 +1007,8 @@
* This will return {@link #AAUDIO_ERROR_UNIMPLEMENTED} for input streams.
* For input streams use AAudioStream_requestStop().
*
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return {@link #AAUDIO_OK} or a negative error.
*/
@@ -984,6 +1024,8 @@
*
* This will return {@link #AAUDIO_ERROR_UNIMPLEMENTED} for input streams.
*
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return {@link #AAUDIO_OK} or a negative error.
*/
@@ -995,6 +1037,8 @@
* After this call the state will be in {@link #AAUDIO_STREAM_STATE_STOPPING} or
* {@link #AAUDIO_STREAM_STATE_STOPPED}.
*
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return {@link #AAUDIO_OK} or a negative error.
*/
@@ -1008,6 +1052,8 @@
* call AAudioStream_waitForStateChange() with currentState
* set to {@link #AAUDIO_STREAM_STATE_UNKNOWN} and a zero timeout.
*
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
*/
AAUDIO_API aaudio_stream_state_t AAudioStream_getState(AAudioStream* stream) __INTRODUCED_IN(26);
@@ -1028,6 +1074,8 @@
* }
* </code></pre>
*
+ * Available since API level 26.
+ *
* @param stream A reference provided by AAudioStreamBuilder_openStream()
* @param inputState The state we want to avoid.
* @param nextState Pointer to a variable that will be set to the new state.
@@ -1056,6 +1104,8 @@
*
* If the call times out then zero or a partial frame count will be returned.
*
+ * Available since API level 26.
+ *
* @param stream A stream created using AAudioStreamBuilder_openStream().
* @param buffer The address of the first sample.
* @param numFrames Number of frames to read. Only complete frames will be written.
@@ -1079,6 +1129,8 @@
*
* If the call times out then zero or a partial frame count will be returned.
*
+ * Available since API level 26.
+ *
* @param stream A stream created using AAudioStreamBuilder_openStream().
* @param buffer The address of the first sample.
* @param numFrames Number of frames to write. Only complete frames will be written.
@@ -1104,6 +1156,8 @@
* You can check the return value or call AAudioStream_getBufferSizeInFrames()
* to see what the actual final size is.
*
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @param numFrames requested number of frames that can be filled without blocking
* @return actual buffer size in frames or a negative error
@@ -1114,6 +1168,8 @@
/**
* Query the maximum number of frames that can be filled without blocking.
*
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return buffer size in frames.
*/
@@ -1129,6 +1185,8 @@
* For some endpoints, the burst size can vary dynamically.
* But these tend to be devices with high latency.
*
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return burst size
*/
@@ -1137,6 +1195,8 @@
/**
* Query maximum buffer capacity in frames.
*
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return buffer capacity in frames
*/
@@ -1158,6 +1218,8 @@
* {@link #AAUDIO_UNSPECIFIED} indicates that the callback buffer size for this stream
* may vary from one dataProc callback to the next.
*
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return callback buffer size in frames or {@link #AAUDIO_UNSPECIFIED}
*/
@@ -1175,12 +1237,16 @@
* Note that some INPUT devices may not support this function.
* In that case a 0 will always be returned.
*
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return the underrun or overrun count
*/
AAUDIO_API int32_t AAudioStream_getXRunCount(AAudioStream* stream) __INTRODUCED_IN(26);
/**
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return actual sample rate
*/
@@ -1190,6 +1256,8 @@
* A stream has one or more channels of data.
* A frame will contain one sample for each channel.
*
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return actual number of channels
*/
@@ -1198,18 +1266,24 @@
/**
* Identical to AAudioStream_getChannelCount().
*
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return actual number of samples frame
*/
AAUDIO_API int32_t AAudioStream_getSamplesPerFrame(AAudioStream* stream) __INTRODUCED_IN(26);
/**
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return actual device ID
*/
AAUDIO_API int32_t AAudioStream_getDeviceId(AAudioStream* stream) __INTRODUCED_IN(26);
/**
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return actual data format
*/
@@ -1217,6 +1291,9 @@
/**
* Provide actual sharing mode.
+ *
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return actual sharing mode
*/
@@ -1226,12 +1303,16 @@
/**
* Get the performance mode used by the stream.
*
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
*/
AAUDIO_API aaudio_performance_mode_t AAudioStream_getPerformanceMode(AAudioStream* stream)
__INTRODUCED_IN(26);
/**
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return direction
*/
@@ -1245,6 +1326,8 @@
*
* The frame position is monotonically increasing.
*
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return frames written
*/
@@ -1258,6 +1341,8 @@
*
* The frame position is monotonically increasing.
*
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return frames read
*/
@@ -1281,7 +1366,7 @@
*
* The sessionID for a stream should not change once the stream has been opened.
*
- * Added in API level 28.
+ * Available since API level 28.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return session ID or {@link #AAUDIO_SESSION_ID_NONE}
@@ -1304,6 +1389,8 @@
*
* The position and time passed back are monotonically increasing.
*
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @param clockid CLOCK_MONOTONIC or CLOCK_BOOTTIME
* @param framePosition pointer to a variable to receive the position
@@ -1316,7 +1403,7 @@
/**
* Return the use case for the stream.
*
- * Added in API level 28.
+ * Available since API level 28.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return frames read
@@ -1326,7 +1413,7 @@
/**
* Return the content type for the stream.
*
- * Added in API level 28.
+ * Available since API level 28.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return content type, for example {@link #AAUDIO_CONTENT_TYPE_MUSIC}
@@ -1337,7 +1424,7 @@
/**
* Return the input preset for the stream.
*
- * Added in API level 28.
+ * Available since API level 28.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return input preset, for example {@link #AAUDIO_INPUT_PRESET_CAMCORDER}
@@ -1349,7 +1436,7 @@
* Return the policy that determines whether the audio may or may not be captured
* by other apps or the system.
*
- * Added in API level 29.
+ * Available since API level 29.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return the allowed capture policy, for example {@link #AAUDIO_ALLOW_CAPTURE_BY_ALL}
diff --git a/media/libaaudio/src/Android.bp b/media/libaaudio/src/Android.bp
index 4090286..25246b3 100644
--- a/media/libaaudio/src/Android.bp
+++ b/media/libaaudio/src/Android.bp
@@ -10,14 +10,81 @@
"legacy",
"utility",
],
- export_include_dirs: ["."],
- header_libs: ["libaaudio_headers"],
+ header_libs: [
+ "libaaudio_headers",
+ ],
export_header_lib_headers: ["libaaudio_headers"],
+ version_script: "libaaudio.map.txt",
srcs: [
+ "core/AAudioAudio.cpp",
+ ],
+
+ cflags: [
+ "-Wno-unused-parameter",
+ "-Wall",
+ "-Werror",
+
+ // By default, all symbols are hidden.
+ // "-fvisibility=hidden",
+ // AAUDIO_API is used to explicitly export a function or a variable as a visible symbol.
+ "-DAAUDIO_API=__attribute__((visibility(\"default\")))",
+ ],
+
+ shared_libs: [
+ "libaaudio_internal",
+ "libaudioclient",
+ "libaudioutils",
+ "liblog",
+ "libcutils",
+ "libutils",
+ "libbinder",
+ ],
+
+ stubs: {
+ symbol_file: "libaaudio.map.txt",
+ versions: ["28"],
+ },
+}
+
+cc_library {
+ name: "libaaudio_internal",
+
+ local_include_dirs: [
+ "binding",
+ "client",
+ "core",
+ "fifo",
+ "legacy",
+ "utility",
+ ],
+
+ export_include_dirs: ["."],
+ header_libs: [
+ "libaaudio_headers",
+ "libmedia_headers"
+ ],
+ export_header_lib_headers: ["libaaudio_headers"],
+
+ shared_libs: [
+ "libaudioclient",
+ "libaudioutils",
+ "liblog",
+ "libcutils",
+ "libutils",
+ "libbinder",
+ ],
+
+ cflags: [
+ "-Wno-unused-parameter",
+ "-Wall",
+ "-Werror",
+ ],
+
+ srcs: [
+ "core/AudioGlobal.cpp",
"core/AudioStream.cpp",
"core/AudioStreamBuilder.cpp",
- "core/AAudioAudio.cpp",
"core/AAudioStreamParameters.cpp",
"legacy/AudioStreamLegacy.cpp",
"legacy/AudioStreamRecord.cpp",
@@ -54,25 +121,4 @@
"flowgraph/SourceI16.cpp",
"flowgraph/SourceI24.cpp",
],
-
- cflags: [
- "-Wno-unused-parameter",
- "-Wall",
- "-Werror",
-
- // By default, all symbols are hidden.
- // "-fvisibility=hidden",
- // AAUDIO_API is used to explicitly export a function or a variable as a visible symbol.
- "-DAAUDIO_API=__attribute__((visibility(\"default\")))",
- ],
-
- shared_libs: [
- "libaudioclient",
- "libaudioutils",
- "liblog",
- "libcutils",
- "libutils",
- "libbinder",
- "libaudiomanager",
- ],
}
diff --git a/media/libaaudio/src/client/AudioStreamInternal.cpp b/media/libaaudio/src/client/AudioStreamInternal.cpp
index 52eadd4..fb276c2 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternal.cpp
@@ -36,6 +36,7 @@
#include "binding/AAudioStreamConfiguration.h"
#include "binding/IAAudioService.h"
#include "binding/AAudioServiceMessage.h"
+#include "core/AudioGlobal.h"
#include "core/AudioStreamBuilder.h"
#include "fifo/FifoBuffer.h"
#include "utility/AudioClock.h"
diff --git a/media/libaaudio/src/core/AAudioAudio.cpp b/media/libaaudio/src/core/AAudioAudio.cpp
index 44d5122..8040e6a 100644
--- a/media/libaaudio/src/core/AAudioAudio.cpp
+++ b/media/libaaudio/src/core/AAudioAudio.cpp
@@ -27,6 +27,7 @@
#include <aaudio/AAudioTesting.h>
#include "AudioClock.h"
+#include "AudioGlobal.h"
#include "AudioStreamBuilder.h"
#include "AudioStream.h"
#include "binding/AAudioCommon.h"
@@ -45,63 +46,14 @@
return AAUDIO_ERROR_NULL; \
}
-#define AAUDIO_CASE_ENUM(name) case name: return #name
-
AAUDIO_API const char * AAudio_convertResultToText(aaudio_result_t returnCode) {
- switch (returnCode) {
- AAUDIO_CASE_ENUM(AAUDIO_OK);
- AAUDIO_CASE_ENUM(AAUDIO_ERROR_DISCONNECTED);
- AAUDIO_CASE_ENUM(AAUDIO_ERROR_ILLEGAL_ARGUMENT);
- // reserved
- AAUDIO_CASE_ENUM(AAUDIO_ERROR_INTERNAL);
- AAUDIO_CASE_ENUM(AAUDIO_ERROR_INVALID_STATE);
- // reserved
- // reserved
- AAUDIO_CASE_ENUM(AAUDIO_ERROR_INVALID_HANDLE);
- // reserved
- AAUDIO_CASE_ENUM(AAUDIO_ERROR_UNIMPLEMENTED);
- AAUDIO_CASE_ENUM(AAUDIO_ERROR_UNAVAILABLE);
- AAUDIO_CASE_ENUM(AAUDIO_ERROR_NO_FREE_HANDLES);
- AAUDIO_CASE_ENUM(AAUDIO_ERROR_NO_MEMORY);
- AAUDIO_CASE_ENUM(AAUDIO_ERROR_NULL);
- AAUDIO_CASE_ENUM(AAUDIO_ERROR_TIMEOUT);
- AAUDIO_CASE_ENUM(AAUDIO_ERROR_WOULD_BLOCK);
- AAUDIO_CASE_ENUM(AAUDIO_ERROR_INVALID_FORMAT);
- AAUDIO_CASE_ENUM(AAUDIO_ERROR_OUT_OF_RANGE);
- AAUDIO_CASE_ENUM(AAUDIO_ERROR_NO_SERVICE);
- AAUDIO_CASE_ENUM(AAUDIO_ERROR_INVALID_RATE);
- }
- return "Unrecognized AAudio error.";
+ return AudioGlobal_convertResultToText(returnCode);
}
AAUDIO_API const char * AAudio_convertStreamStateToText(aaudio_stream_state_t state) {
- switch (state) {
- AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_UNINITIALIZED);
- AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_UNKNOWN);
- AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_OPEN);
- AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_STARTING);
- AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_STARTED);
- AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_PAUSING);
- AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_PAUSED);
- AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_FLUSHING);
- AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_FLUSHED);
- AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_STOPPING);
- AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_STOPPED);
- AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_DISCONNECTED);
- AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_CLOSING);
- AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_CLOSED);
- }
- return "Unrecognized AAudio state.";
+ return AudioGlobal_convertStreamStateToText(state);
}
-#undef AAUDIO_CASE_ENUM
-
-
-/******************************************
- * Static globals.
- */
-static aaudio_policy_t s_MMapPolicy = AAUDIO_UNSPECIFIED;
-
static AudioStream *convertAAudioStreamToAudioStream(AAudioStream* stream)
{
return (AudioStream*) stream;
@@ -543,23 +495,11 @@
}
AAUDIO_API aaudio_policy_t AAudio_getMMapPolicy() {
- return s_MMapPolicy;
+ return AudioGlobal_getMMapPolicy();
}
AAUDIO_API aaudio_result_t AAudio_setMMapPolicy(aaudio_policy_t policy) {
- aaudio_result_t result = AAUDIO_OK;
- switch(policy) {
- case AAUDIO_UNSPECIFIED:
- case AAUDIO_POLICY_NEVER:
- case AAUDIO_POLICY_AUTO:
- case AAUDIO_POLICY_ALWAYS:
- s_MMapPolicy = policy;
- break;
- default:
- result = AAUDIO_ERROR_ILLEGAL_ARGUMENT;
- break;
- }
- return result;
+ return AudioGlobal_setMMapPolicy(policy);
}
AAUDIO_API bool AAudioStream_isMMapUsed(AAudioStream* stream)
diff --git a/media/libaaudio/src/core/AudioGlobal.cpp b/media/libaaudio/src/core/AudioGlobal.cpp
new file mode 100644
index 0000000..e6d9a0d
--- /dev/null
+++ b/media/libaaudio/src/core/AudioGlobal.cpp
@@ -0,0 +1,99 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+#include <aaudio/AAudio.h>
+#include <aaudio/AAudioTesting.h>
+
+#include "AudioGlobal.h"
+
+/******************************************
+ * Static globals.
+ */
+namespace aaudio {
+
+static aaudio_policy_t g_MMapPolicy = AAUDIO_UNSPECIFIED;
+
+aaudio_policy_t AudioGlobal_getMMapPolicy() {
+ return g_MMapPolicy;
+}
+
+aaudio_result_t AudioGlobal_setMMapPolicy(aaudio_policy_t policy) {
+ aaudio_result_t result = AAUDIO_OK;
+ switch(policy) {
+ case AAUDIO_UNSPECIFIED:
+ case AAUDIO_POLICY_NEVER:
+ case AAUDIO_POLICY_AUTO:
+ case AAUDIO_POLICY_ALWAYS:
+ g_MMapPolicy = policy;
+ break;
+ default:
+ result = AAUDIO_ERROR_ILLEGAL_ARGUMENT;
+ break;
+ }
+ return result;
+}
+
+#define AAUDIO_CASE_ENUM(name) case name: return #name
+
+const char* AudioGlobal_convertResultToText(aaudio_result_t returnCode) {
+ switch (returnCode) {
+ AAUDIO_CASE_ENUM(AAUDIO_OK);
+ AAUDIO_CASE_ENUM(AAUDIO_ERROR_DISCONNECTED);
+ AAUDIO_CASE_ENUM(AAUDIO_ERROR_ILLEGAL_ARGUMENT);
+ // reserved
+ AAUDIO_CASE_ENUM(AAUDIO_ERROR_INTERNAL);
+ AAUDIO_CASE_ENUM(AAUDIO_ERROR_INVALID_STATE);
+ // reserved
+ // reserved
+ AAUDIO_CASE_ENUM(AAUDIO_ERROR_INVALID_HANDLE);
+ // reserved
+ AAUDIO_CASE_ENUM(AAUDIO_ERROR_UNIMPLEMENTED);
+ AAUDIO_CASE_ENUM(AAUDIO_ERROR_UNAVAILABLE);
+ AAUDIO_CASE_ENUM(AAUDIO_ERROR_NO_FREE_HANDLES);
+ AAUDIO_CASE_ENUM(AAUDIO_ERROR_NO_MEMORY);
+ AAUDIO_CASE_ENUM(AAUDIO_ERROR_NULL);
+ AAUDIO_CASE_ENUM(AAUDIO_ERROR_TIMEOUT);
+ AAUDIO_CASE_ENUM(AAUDIO_ERROR_WOULD_BLOCK);
+ AAUDIO_CASE_ENUM(AAUDIO_ERROR_INVALID_FORMAT);
+ AAUDIO_CASE_ENUM(AAUDIO_ERROR_OUT_OF_RANGE);
+ AAUDIO_CASE_ENUM(AAUDIO_ERROR_NO_SERVICE);
+ AAUDIO_CASE_ENUM(AAUDIO_ERROR_INVALID_RATE);
+ }
+ return "Unrecognized AAudio error.";
+}
+
+const char* AudioGlobal_convertStreamStateToText(aaudio_stream_state_t state) {
+ switch (state) {
+ AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_UNINITIALIZED);
+ AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_UNKNOWN);
+ AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_OPEN);
+ AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_STARTING);
+ AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_STARTED);
+ AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_PAUSING);
+ AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_PAUSED);
+ AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_FLUSHING);
+ AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_FLUSHED);
+ AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_STOPPING);
+ AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_STOPPED);
+ AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_DISCONNECTED);
+ AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_CLOSING);
+ AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_CLOSED);
+ }
+ return "Unrecognized AAudio state.";
+}
+
+#undef AAUDIO_CASE_ENUM
+
+} // namespace aaudio
diff --git a/media/libaaudio/src/core/AudioGlobal.h b/media/libaaudio/src/core/AudioGlobal.h
new file mode 100644
index 0000000..312cef2
--- /dev/null
+++ b/media/libaaudio/src/core/AudioGlobal.h
@@ -0,0 +1,34 @@
+/*
+ * Copyright 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+#ifndef AAUDIO_AUDIOGLOBAL_H
+#define AAUDIO_AUDIOGLOBAL_H
+
+#include <aaudio/AAudio.h>
+#include <aaudio/AAudioTesting.h>
+
+
+namespace aaudio {
+
+aaudio_policy_t AudioGlobal_getMMapPolicy();
+aaudio_result_t AudioGlobal_setMMapPolicy(aaudio_policy_t policy);
+
+const char* AudioGlobal_convertResultToText(aaudio_result_t returnCode);
+const char* AudioGlobal_convertStreamStateToText(aaudio_stream_state_t state);
+
+}
+
+#endif // AAUDIO_AUDIOGLOBAL_H
+
diff --git a/media/libaaudio/src/core/AudioStream.cpp b/media/libaaudio/src/core/AudioStream.cpp
index f5c97d8..6a8db22 100644
--- a/media/libaaudio/src/core/AudioStream.cpp
+++ b/media/libaaudio/src/core/AudioStream.cpp
@@ -25,8 +25,9 @@
#include "AudioStreamBuilder.h"
#include "AudioStream.h"
#include "AudioClock.h"
+#include "AudioGlobal.h"
-using namespace aaudio;
+namespace aaudio {
// Sequential number assigned to streams solely for debugging purposes.
@@ -51,7 +52,7 @@
|| getState() == AAUDIO_STREAM_STATE_UNINITIALIZED
|| getState() == AAUDIO_STREAM_STATE_DISCONNECTED),
"~AudioStream() - still in use, state = %s",
- AAudio_convertStreamStateToText(getState()));
+ AudioGlobal_convertStreamStateToText(getState()));
mPlayerBase->clearParentReference(); // remove reference to this AudioStream
}
@@ -155,7 +156,7 @@
case AAUDIO_STREAM_STATE_CLOSED:
default:
ALOGW("safePause() stream not running, state = %s",
- AAudio_convertStreamStateToText(getState()));
+ AudioGlobal_convertStreamStateToText(getState()));
return AAUDIO_ERROR_INVALID_STATE;
}
@@ -240,7 +241,7 @@
case AAUDIO_STREAM_STATE_CLOSED:
default:
ALOGW("%s() stream not running, state = %s", __func__,
- AAudio_convertStreamStateToText(getState()));
+ AudioGlobal_convertStreamStateToText(getState()));
return AAUDIO_ERROR_INVALID_STATE;
}
@@ -488,3 +489,5 @@
void AudioStream::MyPlayerBase::destroy() {
unregisterWithAudioManager();
}
+
+} // namespace aaudio
diff --git a/media/libaaudio/src/core/AudioStreamBuilder.cpp b/media/libaaudio/src/core/AudioStreamBuilder.cpp
index 08f4958..44f45b3 100644
--- a/media/libaaudio/src/core/AudioStreamBuilder.cpp
+++ b/media/libaaudio/src/core/AudioStreamBuilder.cpp
@@ -27,6 +27,7 @@
#include "binding/AAudioBinderClient.h"
#include "client/AudioStreamInternalCapture.h"
#include "client/AudioStreamInternalPlay.h"
+#include "core/AudioGlobal.h"
#include "core/AudioStream.h"
#include "core/AudioStreamBuilder.h"
#include "legacy/AudioStreamRecord.h"
@@ -112,7 +113,7 @@
}
// The API setting is the highest priority.
- aaudio_policy_t mmapPolicy = AAudio_getMMapPolicy();
+ aaudio_policy_t mmapPolicy = AudioGlobal_getMMapPolicy();
// If not specified then get from a system property.
if (mmapPolicy == AAUDIO_UNSPECIFIED) {
mmapPolicy = AAudioProperty_getMMapPolicy();
diff --git a/media/libaaudio/libaaudio.map.txt b/media/libaaudio/src/libaaudio.map.txt
similarity index 100%
rename from media/libaaudio/libaaudio.map.txt
rename to media/libaaudio/src/libaaudio.map.txt
diff --git a/media/libaaudio/src/utility/AAudioUtilities.cpp b/media/libaaudio/src/utility/AAudioUtilities.cpp
index 96ed56a..cdd02c0 100644
--- a/media/libaaudio/src/utility/AAudioUtilities.cpp
+++ b/media/libaaudio/src/utility/AAudioUtilities.cpp
@@ -24,6 +24,7 @@
#include <utils/Errors.h>
#include "aaudio/AAudio.h"
+#include "core/AudioGlobal.h"
#include <aaudio/AAudioTesting.h>
#include <math.h>
#include <system/audio-base.h>
@@ -355,7 +356,7 @@
case AAUDIO_STREAM_STATE_DISCONNECTED:
default:
ALOGE("can only flush stream when PAUSED, OPEN or STOPPED, state = %s",
- AAudio_convertStreamStateToText(state));
+ aaudio::AudioGlobal_convertStreamStateToText(state));
result = AAUDIO_ERROR_INVALID_STATE;
break;
}
diff --git a/media/libaaudio/tests/Android.bp b/media/libaaudio/tests/Android.bp
index 6101e99..19cd0a0 100644
--- a/media/libaaudio/tests/Android.bp
+++ b/media/libaaudio/tests/Android.bp
@@ -11,7 +11,7 @@
defaults: ["libaaudio_tests_defaults"],
srcs: ["test_marshalling.cpp"],
shared_libs: [
- "libaaudio",
+ "libaaudio_internal",
"libbinder",
"libcutils",
"libutils",
@@ -23,7 +23,7 @@
defaults: ["libaaudio_tests_defaults"],
srcs: ["test_clock_model.cpp"],
shared_libs: [
- "libaaudio",
+ "libaaudio_internal",
"libaudioutils",
"libcutils",
"libutils",
@@ -34,7 +34,7 @@
name: "test_block_adapter",
defaults: ["libaaudio_tests_defaults"],
srcs: ["test_block_adapter.cpp"],
- shared_libs: ["libaaudio"],
+ shared_libs: ["libaaudio_internal"],
}
cc_test {
@@ -170,7 +170,7 @@
name: "test_atomic_fifo",
defaults: ["libaaudio_tests_defaults"],
srcs: ["test_atomic_fifo.cpp"],
- shared_libs: ["libaaudio"],
+ shared_libs: ["libaaudio_internal"],
}
cc_test {
@@ -178,7 +178,7 @@
defaults: ["libaaudio_tests_defaults"],
srcs: ["test_flowgraph.cpp"],
shared_libs: [
- "libaaudio",
+ "libaaudio_internal",
"libbinder",
"libcutils",
"libutils",
diff --git a/media/libaudioclient/Android.bp b/media/libaudioclient/Android.bp
index 03bd6ce..d1812e6 100644
--- a/media/libaudioclient/Android.bp
+++ b/media/libaudioclient/Android.bp
@@ -1,7 +1,15 @@
cc_library_headers {
name: "libaudioclient_headers",
vendor_available: true,
- export_include_dirs: ["include"],
+ header_libs: [
+ "libaudiofoundation_headers",
+ ],
+ export_include_dirs: [
+ "include",
+ ],
+ export_header_lib_headers: [
+ "libaudiofoundation_headers",
+ ],
}
cc_library_shared {
@@ -13,6 +21,7 @@
"AudioVolumeGroup.cpp",
],
shared_libs: [
+ "libaudiofoundation",
"libaudioutils",
"libbinder",
"libcutils",
@@ -42,7 +51,7 @@
// AIDL files for audioclient interfaces
// The headers for these interfaces will be available to any modules that
// include libaudioclient, at the path "aidl/package/path/BnFoo.h"
- "aidl/android/media/IAudioRecord.aidl",
+ ":libaudioclient_aidl_private",
":libaudioclient_aidl",
"AudioEffect.cpp",
@@ -63,6 +72,7 @@
"TrackPlayerBase.cpp",
],
shared_libs: [
+ "libaudiofoundation",
"libaudioutils",
"libaudiopolicy",
"libaudiomanager",
@@ -84,6 +94,7 @@
header_libs: [
"libaudioclient_headers",
"libbase_headers",
+ "libmedia_headers",
],
export_header_lib_headers: ["libaudioclient_headers"],
@@ -110,4 +121,15 @@
srcs: [
"aidl/android/media/IPlayer.aidl",
],
+ path: "aidl",
+}
+
+// Used to strip the "aidl/" from the path, so the build system can predict the
+// output filename.
+filegroup {
+ name: "libaudioclient_aidl_private",
+ srcs: [
+ "aidl/android/media/IAudioRecord.aidl",
+ ],
+ path: "aidl",
}
diff --git a/media/libaudioclient/AudioAttributes.cpp b/media/libaudioclient/AudioAttributes.cpp
index 1ee6930..ff4ba06 100644
--- a/media/libaudioclient/AudioAttributes.cpp
+++ b/media/libaudioclient/AudioAttributes.cpp
@@ -57,7 +57,7 @@
parcel->writeInt32(0);
} else {
parcel->writeInt32(1);
- parcel->writeUtf8AsUtf16(mAttributes.tags);
+ parcel->writeUtf8AsUtf16(std::string(mAttributes.tags));
}
parcel->writeInt32(static_cast<int32_t>(mStreamType));
parcel->writeUint32(static_cast<uint32_t>(mGroupId));
diff --git a/media/libaudioclient/AudioEffect.cpp b/media/libaudioclient/AudioEffect.cpp
index cf11936..28190ea 100644
--- a/media/libaudioclient/AudioEffect.cpp
+++ b/media/libaudioclient/AudioEffect.cpp
@@ -48,12 +48,13 @@
effect_callback_t cbf,
void* user,
audio_session_t sessionId,
- audio_io_handle_t io
+ audio_io_handle_t io,
+ const AudioDeviceTypeAddr& device
)
: mStatus(NO_INIT), mOpPackageName(opPackageName)
{
AutoMutex lock(mConstructLock);
- mStatus = set(type, uuid, priority, cbf, user, sessionId, io);
+ mStatus = set(type, uuid, priority, cbf, user, sessionId, io, device);
}
AudioEffect::AudioEffect(const char *typeStr,
@@ -63,7 +64,8 @@
effect_callback_t cbf,
void* user,
audio_session_t sessionId,
- audio_io_handle_t io
+ audio_io_handle_t io,
+ const AudioDeviceTypeAddr& device
)
: mStatus(NO_INIT), mOpPackageName(opPackageName)
{
@@ -87,7 +89,7 @@
}
AutoMutex lock(mConstructLock);
- mStatus = set(pType, pUuid, priority, cbf, user, sessionId, io);
+ mStatus = set(pType, pUuid, priority, cbf, user, sessionId, io, device);
}
status_t AudioEffect::set(const effect_uuid_t *type,
@@ -96,7 +98,8 @@
effect_callback_t cbf,
void* user,
audio_session_t sessionId,
- audio_io_handle_t io)
+ audio_io_handle_t io,
+ const AudioDeviceTypeAddr& device)
{
sp<IEffect> iEffect;
sp<IMemory> cblk;
@@ -109,6 +112,10 @@
return INVALID_OPERATION;
}
+ if (sessionId == AUDIO_SESSION_DEVICE && io != AUDIO_IO_HANDLE_NONE) {
+ ALOGW("IO handle should not be specified for device effect");
+ return BAD_VALUE;
+ }
const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
if (audioFlinger == 0) {
ALOGE("set(): Could not get audioflinger");
@@ -133,7 +140,7 @@
mClientPid = IPCThreadState::self()->getCallingPid();
iEffect = audioFlinger->createEffect((effect_descriptor_t *)&mDescriptor,
- mIEffectClient, priority, io, mSessionId, mOpPackageName, mClientPid,
+ mIEffectClient, priority, io, mSessionId, device, mOpPackageName, mClientPid,
&mStatus, &mId, &enabled);
if (iEffect == 0 || (mStatus != NO_ERROR && mStatus != ALREADY_EXISTS)) {
@@ -167,7 +174,7 @@
ALOGV("set() %p OK effect: %s id: %d status %d enabled %d pid %d", this, mDescriptor.name, mId,
mStatus, mEnabled, mClientPid);
- if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) {
+ if (!audio_is_global_session(mSessionId)) {
AudioSystem::acquireAudioSessionId(mSessionId, mClientPid);
}
@@ -180,7 +187,7 @@
ALOGV("Destructor %p", this);
if (mStatus == NO_ERROR || mStatus == ALREADY_EXISTS) {
- if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) {
+ if (!audio_is_global_session(mSessionId)) {
AudioSystem::releaseAudioSessionId(mSessionId, mClientPid);
}
if (mIEffect != NULL) {
diff --git a/media/libaudioclient/AudioPolicy.cpp b/media/libaudioclient/AudioPolicy.cpp
index 3cdf095..06fc23c 100644
--- a/media/libaudioclient/AudioPolicy.cpp
+++ b/media/libaudioclient/AudioPolicy.cpp
@@ -22,22 +22,6 @@
namespace android {
//
-// AudioDeviceTypeAddr implementation
-//
-status_t AudioDeviceTypeAddr::readFromParcel(Parcel *parcel) {
- mType = (audio_devices_t) parcel->readInt32();
- mAddress = parcel->readString8();
- return NO_ERROR;
-}
-
-status_t AudioDeviceTypeAddr::writeToParcel(Parcel *parcel) const {
- parcel->writeInt32((int32_t) mType);
- parcel->writeString8(mAddress);
- return NO_ERROR;
-}
-
-
-//
// AudioMixMatchCriterion implementation
//
AudioMixMatchCriterion::AudioMixMatchCriterion(audio_usage_t usage,
diff --git a/media/libaudioclient/AudioSystem.cpp b/media/libaudioclient/AudioSystem.cpp
index 02dc516..480930b 100644
--- a/media/libaudioclient/AudioSystem.cpp
+++ b/media/libaudioclient/AudioSystem.cpp
@@ -1392,6 +1392,12 @@
return af->getMicrophones(microphones);
}
+status_t AudioSystem::setAudioHalPids(const std::vector<pid_t>& pids) {
+ const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
+ if (af == nullptr) return PERMISSION_DENIED;
+ return af->setAudioHalPids(pids);
+}
+
status_t AudioSystem::getSurroundFormats(unsigned int *numSurroundFormats,
audio_format_t *surroundFormats,
bool *surroundFormatsEnabled,
@@ -1485,7 +1491,14 @@
}
}
}
- ALOGE("invalid attributes %s when converting to stream", toString(attr).c_str());
+ switch (attr.usage) {
+ case AUDIO_USAGE_VIRTUAL_SOURCE:
+ // virtual source is not expected to have an associated product strategy
+ break;
+ default:
+ ALOGE("invalid attributes %s when converting to stream", toString(attr).c_str());
+ break;
+ }
return AUDIO_STREAM_MUSIC;
}
@@ -1519,6 +1532,35 @@
return aps->setRttEnabled(enabled);
}
+status_t AudioSystem::setPreferredDeviceForStrategy(product_strategy_t strategy,
+ const AudioDeviceTypeAddr &device)
+{
+ const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
+ if (aps == 0) {
+ return PERMISSION_DENIED;
+ }
+ return aps->setPreferredDeviceForStrategy(strategy, device);
+}
+
+status_t AudioSystem::removePreferredDeviceForStrategy(product_strategy_t strategy)
+{
+ const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
+ if (aps == 0) {
+ return PERMISSION_DENIED;
+ }
+ return aps->removePreferredDeviceForStrategy(strategy);
+}
+
+status_t AudioSystem::getPreferredDeviceForStrategy(product_strategy_t strategy,
+ AudioDeviceTypeAddr &device)
+{
+ const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
+ if (aps == 0) {
+ return PERMISSION_DENIED;
+ }
+ return aps->getPreferredDeviceForStrategy(strategy, device);
+}
+
// ---------------------------------------------------------------------------
int AudioSystem::AudioPolicyServiceClient::addAudioPortCallback(
diff --git a/media/libaudioclient/IAudioFlinger.cpp b/media/libaudioclient/IAudioFlinger.cpp
index efa0512..53d46f1 100644
--- a/media/libaudioclient/IAudioFlinger.cpp
+++ b/media/libaudioclient/IAudioFlinger.cpp
@@ -24,8 +24,8 @@
#include <binder/IPCThreadState.h>
#include <binder/Parcel.h>
-#include <media/TimeCheck.h>
#include <mediautils/ServiceUtilities.h>
+#include <mediautils/TimeCheck.h>
#include "IAudioFlinger.h"
namespace android {
@@ -90,10 +90,12 @@
SET_MASTER_BALANCE,
GET_MASTER_BALANCE,
SET_EFFECT_SUSPENDED,
+ SET_AUDIO_HAL_PIDS
};
#define MAX_ITEMS_PER_LIST 1024
+
class BpAudioFlinger : public BpInterface<IAudioFlinger>
{
public:
@@ -392,20 +394,18 @@
virtual status_t openOutput(audio_module_handle_t module,
audio_io_handle_t *output,
audio_config_t *config,
- audio_devices_t *devices,
- const String8& address,
+ const sp<DeviceDescriptorBase>& device,
uint32_t *latencyMs,
audio_output_flags_t flags)
{
- if (output == NULL || config == NULL || devices == NULL || latencyMs == NULL) {
+ if (output == nullptr || config == nullptr || device == nullptr || latencyMs == nullptr) {
return BAD_VALUE;
}
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
data.writeInt32(module);
data.write(config, sizeof(audio_config_t));
- data.writeInt32(*devices);
- data.writeString8(address);
+ data.writeParcelable(*device);
data.writeInt32((int32_t) flags);
status_t status = remote()->transact(OPEN_OUTPUT, data, &reply);
if (status != NO_ERROR) {
@@ -420,7 +420,6 @@
*output = (audio_io_handle_t)reply.readInt32();
ALOGV("openOutput() returned output, %d", *output);
reply.read(config, sizeof(audio_config_t));
- *devices = (audio_devices_t)reply.readInt32();
*latencyMs = reply.readInt32();
return NO_ERROR;
}
@@ -659,6 +658,7 @@
int32_t priority,
audio_io_handle_t output,
audio_session_t sessionId,
+ const AudioDeviceTypeAddr& device,
const String16& opPackageName,
pid_t pid,
status_t *status,
@@ -667,12 +667,11 @@
{
Parcel data, reply;
sp<IEffect> effect;
-
if (pDesc == NULL) {
if (status != NULL) {
*status = BAD_VALUE;
}
- return effect;
+ return nullptr;
}
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
@@ -681,6 +680,12 @@
data.writeInt32(priority);
data.writeInt32((int32_t) output);
data.writeInt32(sessionId);
+ if (data.writeParcelable(device) != NO_ERROR) {
+ if (status != NULL) {
+ *status = NO_INIT;
+ }
+ return nullptr;
+ }
data.writeString16(opPackageName);
data.writeInt32((int32_t) pid);
@@ -903,6 +908,20 @@
status = reply.readParcelableVector(microphones);
return status;
}
+ virtual status_t setAudioHalPids(const std::vector<pid_t>& pids)
+ {
+ Parcel data, reply;
+ data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
+ data.writeInt32(pids.size());
+ for (auto pid : pids) {
+ data.writeInt32(pid);
+ }
+ status_t status = remote()->transact(SET_AUDIO_HAL_PIDS, data, &reply);
+ if (status != NO_ERROR) {
+ return status;
+ }
+ return static_cast <status_t> (reply.readInt32());
+ }
};
IMPLEMENT_META_INTERFACE(AudioFlinger, "android.media.IAudioFlinger");
@@ -958,7 +977,8 @@
case SET_MODE:
case SET_MIC_MUTE:
case SET_LOW_RAM_DEVICE:
- case SYSTEM_READY: {
+ case SYSTEM_READY:
+ case SET_AUDIO_HAL_PIDS: {
if (!isServiceUid(IPCThreadState::self()->getCallingUid())) {
ALOGW("%s: transaction %d received from PID %d unauthorized UID %d",
__func__, code, IPCThreadState::self()->getCallingPid(),
@@ -1200,19 +1220,21 @@
if (data.read(&config, sizeof(audio_config_t)) != NO_ERROR) {
ALOGE("b/23905951");
}
- audio_devices_t devices = (audio_devices_t)data.readInt32();
- String8 address(data.readString8());
+ sp<DeviceDescriptorBase> device = new DeviceDescriptorBase(AUDIO_DEVICE_NONE);
+ status_t status = NO_ERROR;
+ if ((status = data.readParcelable(device.get())) != NO_ERROR) {
+ reply->writeInt32((int32_t)status);
+ return NO_ERROR;
+ }
audio_output_flags_t flags = (audio_output_flags_t) data.readInt32();
uint32_t latencyMs = 0;
audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
- status_t status = openOutput(module, &output, &config,
- &devices, address, &latencyMs, flags);
+ status = openOutput(module, &output, &config, device, &latencyMs, flags);
ALOGV("OPEN_OUTPUT output, %d", output);
reply->writeInt32((int32_t)status);
if (status == NO_ERROR) {
reply->writeInt32((int32_t)output);
reply->write(&config, sizeof(audio_config_t));
- reply->writeInt32(devices);
reply->writeInt32(latencyMs);
}
return NO_ERROR;
@@ -1366,14 +1388,18 @@
int32_t priority = data.readInt32();
audio_io_handle_t output = (audio_io_handle_t) data.readInt32();
audio_session_t sessionId = (audio_session_t) data.readInt32();
+ AudioDeviceTypeAddr device;
+ status_t status = NO_ERROR;
+ if ((status = data.readParcelable(&device)) != NO_ERROR) {
+ return status;
+ }
const String16 opPackageName = data.readString16();
pid_t pid = (pid_t)data.readInt32();
- status_t status = NO_ERROR;
int id = 0;
int enabled = 0;
- sp<IEffect> effect = createEffect(&desc, client, priority, output, sessionId,
+ sp<IEffect> effect = createEffect(&desc, client, priority, output, sessionId, device,
opPackageName, pid, &status, &id, &enabled);
reply->writeInt32(status);
reply->writeInt32(id);
@@ -1547,6 +1573,31 @@
}
return NO_ERROR;
}
+ case SET_AUDIO_HAL_PIDS: {
+ CHECK_INTERFACE(IAudioFlinger, data, reply);
+ std::vector<pid_t> pids;
+ int32_t size;
+ status_t status = data.readInt32(&size);
+ if (status != NO_ERROR) {
+ return status;
+ }
+ if (size < 0) {
+ return BAD_VALUE;
+ }
+ if (size > MAX_ITEMS_PER_LIST) {
+ size = MAX_ITEMS_PER_LIST;
+ }
+ for (int32_t i = 0; i < size; i++) {
+ int32_t pid;
+ status = data.readInt32(&pid);
+ if (status != NO_ERROR) {
+ return status;
+ }
+ pids.push_back(pid);
+ }
+ reply->writeInt32(setAudioHalPids(pids));
+ return NO_ERROR;
+ }
default:
return BBinder::onTransact(code, data, reply, flags);
}
diff --git a/media/libaudioclient/IAudioPolicyService.cpp b/media/libaudioclient/IAudioPolicyService.cpp
index 2fb9491..52eb9a4 100644
--- a/media/libaudioclient/IAudioPolicyService.cpp
+++ b/media/libaudioclient/IAudioPolicyService.cpp
@@ -26,8 +26,8 @@
#include <binder/Parcel.h>
#include <media/AudioEffect.h>
#include <media/IAudioPolicyService.h>
-#include <media/TimeCheck.h>
#include <mediautils/ServiceUtilities.h>
+#include <mediautils/TimeCheck.h>
#include <system/audio.h>
namespace android {
@@ -104,7 +104,10 @@
GET_VOLUME_GROUP_FOR_ATTRIBUTES,
SET_ALLOWED_CAPTURE_POLICY,
MOVE_EFFECTS_TO_IO,
- SET_RTT_ENABLED
+ SET_RTT_ENABLED,
+ SET_PREFERRED_DEVICE_FOR_PRODUCT_STRATEGY,
+ REMOVE_PREFERRED_DEVICE_FOR_PRODUCT_STRATEGY,
+ GET_PREFERRED_DEVICE_FOR_PRODUCT_STRATEGY,
};
#define MAX_ITEMS_PER_LIST 1024
@@ -1284,6 +1287,55 @@
}
return static_cast<status_t>(reply.readInt32());
}
+
+ virtual status_t setPreferredDeviceForStrategy(product_strategy_t strategy,
+ const AudioDeviceTypeAddr &device)
+ {
+ Parcel data, reply;
+ data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
+ data.writeUint32(static_cast<uint32_t>(strategy));
+ status_t status = device.writeToParcel(&data);
+ if (status != NO_ERROR) {
+ return BAD_VALUE;
+ }
+ status = remote()->transact(SET_PREFERRED_DEVICE_FOR_PRODUCT_STRATEGY,
+ data, &reply);
+ if (status != NO_ERROR) {
+ return status;
+ }
+ return static_cast<status_t>(reply.readInt32());
+ }
+
+ virtual status_t removePreferredDeviceForStrategy(product_strategy_t strategy)
+ {
+ Parcel data, reply;
+ data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
+ data.writeUint32(static_cast<uint32_t>(strategy));
+ status_t status = remote()->transact(REMOVE_PREFERRED_DEVICE_FOR_PRODUCT_STRATEGY,
+ data, &reply);
+ if (status != NO_ERROR) {
+ return status;
+ }
+ return static_cast<status_t>(reply.readInt32());
+ }
+
+ virtual status_t getPreferredDeviceForStrategy(product_strategy_t strategy,
+ AudioDeviceTypeAddr &device)
+ {
+ Parcel data, reply;
+ data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
+ data.writeUint32(static_cast<uint32_t>(strategy));
+ status_t status = remote()->transact(GET_PREFERRED_DEVICE_FOR_PRODUCT_STRATEGY,
+ data, &reply);
+ if (status != NO_ERROR) {
+ return status;
+ }
+ status = device.readFromParcel(&reply);
+ if (status != NO_ERROR) {
+ return status;
+ }
+ return static_cast<status_t>(reply.readInt32());
+ }
};
IMPLEMENT_META_INTERFACE(AudioPolicyService, "android.media.IAudioPolicyService");
@@ -1347,6 +1399,9 @@
case LIST_AUDIO_VOLUME_GROUPS:
case GET_VOLUME_GROUP_FOR_ATTRIBUTES:
case SET_RTT_ENABLED:
+ case SET_PREFERRED_DEVICE_FOR_PRODUCT_STRATEGY:
+ case REMOVE_PREFERRED_DEVICE_FOR_PRODUCT_STRATEGY:
+ case GET_PREFERRED_DEVICE_FOR_PRODUCT_STRATEGY:
case SET_ALLOWED_CAPTURE_POLICY: {
if (!isServiceUid(IPCThreadState::self()->getCallingUid())) {
ALOGW("%s: transaction %d received from PID %d unauthorized UID %d",
@@ -2370,6 +2425,40 @@
return NO_ERROR;
}
+ case SET_PREFERRED_DEVICE_FOR_PRODUCT_STRATEGY: {
+ CHECK_INTERFACE(IAudioPolicyService, data, reply);
+ product_strategy_t strategy = (product_strategy_t) data.readUint32();
+ AudioDeviceTypeAddr device;
+ status_t status = device.readFromParcel((Parcel*)&data);
+ if (status != NO_ERROR) {
+ return status;
+ }
+ status = setPreferredDeviceForStrategy(strategy, device);
+ reply->writeInt32(status);
+ return NO_ERROR;
+ }
+
+ case REMOVE_PREFERRED_DEVICE_FOR_PRODUCT_STRATEGY: {
+ CHECK_INTERFACE(IAudioPolicyService, data, reply);
+ product_strategy_t strategy = (product_strategy_t) data.readUint32();
+ status_t status = removePreferredDeviceForStrategy(strategy);
+ reply->writeInt32(status);
+ return NO_ERROR;
+ }
+
+ case GET_PREFERRED_DEVICE_FOR_PRODUCT_STRATEGY: {
+ CHECK_INTERFACE(IAudioPolicyService, data, reply);
+ product_strategy_t strategy = (product_strategy_t) data.readUint32();
+ AudioDeviceTypeAddr device;
+ status_t status = getPreferredDeviceForStrategy(strategy, device);
+ status_t marshall_status = device.writeToParcel(reply);
+ if (marshall_status != NO_ERROR) {
+ return marshall_status;
+ }
+ reply->writeInt32(status);
+ return NO_ERROR;
+ }
+
default:
return BBinder::onTransact(code, data, reply, flags);
}
diff --git a/media/libaudioclient/include/media/AudioEffect.h b/media/libaudioclient/include/media/AudioEffect.h
index 6bd4137..f17d737 100644
--- a/media/libaudioclient/include/media/AudioEffect.h
+++ b/media/libaudioclient/include/media/AudioEffect.h
@@ -362,17 +362,21 @@
* (AudioTrack or MediaPLayer) within the same audio session.
* io: HAL audio output or input stream to which this effect must be attached. Leave at 0 for
* automatic output selection by AudioFlinger.
+ * device: An audio device descriptor. Only used when "sessionID" is AUDIO_SESSION_DEVICE.
+ * Specifies the audio device type and address the effect must be attached to.
+ * If "sessionID" is AUDIO_SESSION_DEVICE then "io" must be AUDIO_IO_HANDLE_NONE.
*/
AudioEffect(const effect_uuid_t *type,
const String16& opPackageName,
const effect_uuid_t *uuid = NULL,
- int32_t priority = 0,
- effect_callback_t cbf = NULL,
- void* user = NULL,
- audio_session_t sessionId = AUDIO_SESSION_OUTPUT_MIX,
- audio_io_handle_t io = AUDIO_IO_HANDLE_NONE
- );
+ int32_t priority = 0,
+ effect_callback_t cbf = NULL,
+ void* user = NULL,
+ audio_session_t sessionId = AUDIO_SESSION_OUTPUT_MIX,
+ audio_io_handle_t io = AUDIO_IO_HANDLE_NONE,
+ const AudioDeviceTypeAddr& device = {}
+ );
/* Constructor.
* Same as above but with type and uuid specified by character strings
@@ -384,7 +388,8 @@
effect_callback_t cbf = NULL,
void* user = NULL,
audio_session_t sessionId = AUDIO_SESSION_OUTPUT_MIX,
- audio_io_handle_t io = AUDIO_IO_HANDLE_NONE
+ audio_io_handle_t io = AUDIO_IO_HANDLE_NONE,
+ const AudioDeviceTypeAddr& device = {}
);
/* Terminates the AudioEffect and unregisters it from AudioFlinger.
@@ -406,7 +411,8 @@
effect_callback_t cbf = NULL,
void* user = NULL,
audio_session_t sessionId = AUDIO_SESSION_OUTPUT_MIX,
- audio_io_handle_t io = AUDIO_IO_HANDLE_NONE
+ audio_io_handle_t io = AUDIO_IO_HANDLE_NONE,
+ const AudioDeviceTypeAddr& device = {}
);
/* Result of constructing the AudioEffect. This must be checked
diff --git a/media/libaudioclient/include/media/AudioMixer.h b/media/libaudioclient/include/media/AudioMixer.h
index 783eef3..3f7cd48 100644
--- a/media/libaudioclient/include/media/AudioMixer.h
+++ b/media/libaudioclient/include/media/AudioMixer.h
@@ -18,87 +18,38 @@
#ifndef ANDROID_AUDIO_MIXER_H
#define ANDROID_AUDIO_MIXER_H
-#include <map>
#include <pthread.h>
-#include <sstream>
#include <stdint.h>
#include <sys/types.h>
-#include <unordered_map>
-#include <vector>
#include <android/os/IExternalVibratorService.h>
-#include <media/AudioBufferProvider.h>
-#include <media/AudioResampler.h>
-#include <media/AudioResamplerPublic.h>
+#include <media/AudioMixerBase.h>
#include <media/BufferProviders.h>
-#include <system/audio.h>
-#include <utils/Compat.h>
#include <utils/threads.h>
// FIXME This is actually unity gain, which might not be max in future, expressed in U.12
-#define MAX_GAIN_INT AudioMixer::UNITY_GAIN_INT
-
-// This must match frameworks/av/services/audioflinger/Configuration.h
-#define FLOAT_AUX
+#define MAX_GAIN_INT AudioMixerBase::UNITY_GAIN_INT
namespace android {
-namespace NBLog {
-class Writer;
-} // namespace NBLog
-
// ----------------------------------------------------------------------------
-class AudioMixer
+// AudioMixer extends AudioMixerBase by adding support for down- and up-mixing
+// and time stretch that are implemented via Effects HAL, and adding support
+// for haptic channels which depends on Vibrator service. This is the version
+// that is used by Audioflinger.
+
+class AudioMixer : public AudioMixerBase
{
public:
- // Do not change these unless underlying code changes.
- // This mixer has a hard-coded upper limit of 8 channels for output.
- static constexpr uint32_t MAX_NUM_CHANNELS = FCC_8;
- static constexpr uint32_t MAX_NUM_VOLUMES = FCC_2; // stereo volume only
// maximum number of channels supported for the content
static const uint32_t MAX_NUM_CHANNELS_TO_DOWNMIX = AUDIO_CHANNEL_COUNT_MAX;
- static const uint16_t UNITY_GAIN_INT = 0x1000;
- static const CONSTEXPR float UNITY_GAIN_FLOAT = 1.0f;
-
- enum { // names
- // setParameter targets
- TRACK = 0x3000,
- RESAMPLE = 0x3001,
- RAMP_VOLUME = 0x3002, // ramp to new volume
- VOLUME = 0x3003, // don't ramp
- TIMESTRETCH = 0x3004,
-
- // set Parameter names
- // for target TRACK
- CHANNEL_MASK = 0x4000,
- FORMAT = 0x4001,
- MAIN_BUFFER = 0x4002,
- AUX_BUFFER = 0x4003,
- DOWNMIX_TYPE = 0X4004,
- MIXER_FORMAT = 0x4005, // AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
- MIXER_CHANNEL_MASK = 0x4006, // Channel mask for mixer output
+ enum { // extension of AudioMixerBase parameters
+ DOWNMIX_TYPE = 0x4004,
// for haptic
HAPTIC_ENABLED = 0x4007, // Set haptic data from this track should be played or not.
HAPTIC_INTENSITY = 0x4008, // Set the intensity to play haptic data.
- // for target RESAMPLE
- SAMPLE_RATE = 0x4100, // Configure sample rate conversion on this track name;
- // parameter 'value' is the new sample rate in Hz.
- // Only creates a sample rate converter the first time that
- // the track sample rate is different from the mix sample rate.
- // If the new sample rate is the same as the mix sample rate,
- // and a sample rate converter already exists,
- // then the sample rate converter remains present but is a no-op.
- RESET = 0x4101, // Reset sample rate converter without changing sample rate.
- // This clears out the resampler's input buffer.
- REMOVE = 0x4102, // Remove the sample rate converter on this track name;
- // the track is restored to the mix sample rate.
- // for target RAMP_VOLUME and VOLUME (8 channels max)
- // FIXME use float for these 3 to improve the dynamic range
- VOLUME0 = 0x4200,
- VOLUME1 = 0x4201,
- AUXLEVEL = 0x4210,
// for target TIMESTRETCH
PLAYBACK_RATE = 0x4300, // Configure timestretch on this track name;
// parameter 'value' is a pointer to the new playback rate.
@@ -131,142 +82,23 @@
}
AudioMixer(size_t frameCount, uint32_t sampleRate)
- : mSampleRate(sampleRate)
- , mFrameCount(frameCount) {
+ : AudioMixerBase(frameCount, sampleRate) {
pthread_once(&sOnceControl, &sInitRoutine);
}
- // Create a new track in the mixer.
- //
- // \param name a unique user-provided integer associated with the track.
- // If name already exists, the function will abort.
- // \param channelMask output channel mask.
- // \param format PCM format
- // \param sessionId Session id for the track. Tracks with the same
- // session id will be submixed together.
- //
- // \return OK on success.
- // BAD_VALUE if the format does not satisfy isValidFormat()
- // or the channelMask does not satisfy isValidChannelMask().
- status_t create(
- int name, audio_channel_mask_t channelMask, audio_format_t format, int sessionId);
+ bool isValidChannelMask(audio_channel_mask_t channelMask) const override;
- bool exists(int name) const {
- return mTracks.count(name) > 0;
- }
-
- // Free an allocated track by name.
- void destroy(int name);
-
- // Enable or disable an allocated track by name
- void enable(int name);
- void disable(int name);
-
- void setParameter(int name, int target, int param, void *value);
-
- void setBufferProvider(int name, AudioBufferProvider* bufferProvider);
-
- void process() {
- for (const auto &pair : mTracks) {
- // Clear contracted buffer before processing if contracted channels are saved
- const std::shared_ptr<Track> &t = pair.second;
- if (t->mKeepContractedChannels) {
- t->clearContractedBuffer();
- }
- }
- (this->*mHook)();
- processHapticData();
- }
-
- size_t getUnreleasedFrames(int name) const;
-
- std::string trackNames() const {
- std::stringstream ss;
- for (const auto &pair : mTracks) {
- ss << pair.first << " ";
- }
- return ss.str();
- }
-
- void setNBLogWriter(NBLog::Writer *logWriter) {
- mNBLogWriter = logWriter;
- }
-
- static inline bool isValidFormat(audio_format_t format) {
- switch (format) {
- case AUDIO_FORMAT_PCM_8_BIT:
- case AUDIO_FORMAT_PCM_16_BIT:
- case AUDIO_FORMAT_PCM_24_BIT_PACKED:
- case AUDIO_FORMAT_PCM_32_BIT:
- case AUDIO_FORMAT_PCM_FLOAT:
- return true;
- default:
- return false;
- }
- }
-
- static inline bool isValidChannelMask(audio_channel_mask_t channelMask) {
- return audio_channel_mask_is_valid(channelMask); // the RemixBufferProvider is flexible.
- }
+ void setParameter(int name, int target, int param, void *value) override;
+ void setBufferProvider(int name, AudioBufferProvider* bufferProvider);
private:
- /* For multi-format functions (calls template functions
- * in AudioMixerOps.h). The template parameters are as follows:
- *
- * MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
- * USEFLOATVOL (set to true if float volume is used)
- * ADJUSTVOL (set to true if volume ramp parameters needs adjustment afterwards)
- * TO: int32_t (Q4.27) or float
- * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
- * TA: int32_t (Q4.27)
- */
-
- enum {
- // FIXME this representation permits up to 8 channels
- NEEDS_CHANNEL_COUNT__MASK = 0x00000007,
- };
-
- enum {
- NEEDS_CHANNEL_1 = 0x00000000, // mono
- NEEDS_CHANNEL_2 = 0x00000001, // stereo
-
- // sample format is not explicitly specified, and is assumed to be AUDIO_FORMAT_PCM_16_BIT
-
- NEEDS_MUTE = 0x00000100,
- NEEDS_RESAMPLE = 0x00001000,
- NEEDS_AUX = 0x00010000,
- };
-
- // hook types
- enum {
- PROCESSTYPE_NORESAMPLEONETRACK, // others set elsewhere
- };
-
- enum {
- TRACKTYPE_NOP,
- TRACKTYPE_RESAMPLE,
- TRACKTYPE_NORESAMPLE,
- TRACKTYPE_NORESAMPLEMONO,
- };
-
- // process hook functionality
- using process_hook_t = void(AudioMixer::*)();
-
- struct Track;
- using hook_t = void(Track::*)(int32_t* output, size_t numOutFrames, int32_t* temp, int32_t* aux);
-
- struct Track {
- Track()
- : bufferProvider(nullptr)
- {
- // TODO: move additional initialization here.
- }
+ struct Track : public TrackBase {
+ Track() : TrackBase() {}
~Track()
{
- // bufferProvider, mInputBufferProvider need not be deleted.
- mResampler.reset(nullptr);
+ // mInputBufferProvider need not be deleted.
// Ensure the order of destruction of buffer providers as they
// release the upstream provider in the destructor.
mTimestretchBufferProvider.reset(nullptr);
@@ -277,13 +109,12 @@
mAdjustChannelsBufferProvider.reset(nullptr);
}
- bool needsRamp() { return (volumeInc[0] | volumeInc[1] | auxInc) != 0; }
- bool setResampler(uint32_t trackSampleRate, uint32_t devSampleRate);
- bool doesResample() const { return mResampler.get() != nullptr; }
- void resetResampler() { if (mResampler.get() != nullptr) mResampler->reset(); }
- void adjustVolumeRamp(bool aux, bool useFloat = false);
- size_t getUnreleasedFrames() const { return mResampler.get() != nullptr ?
- mResampler->getUnreleasedFrames() : 0; };
+ uint32_t getOutputChannelCount() override {
+ return mDownmixerBufferProvider.get() != nullptr ? mMixerChannelCount : channelCount;
+ }
+ uint32_t getMixerChannelCount() override {
+ return mMixerChannelCount + mMixerHapticChannelCount;
+ }
status_t prepareForDownmix();
void unprepareForDownmix();
@@ -297,51 +128,9 @@
bool setPlaybackRate(const AudioPlaybackRate &playbackRate);
void reconfigureBufferProviders();
- static hook_t getTrackHook(int trackType, uint32_t channelCount,
- audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
-
- void track__nop(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
-
- template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL,
- typename TO, typename TI, typename TA>
- void volumeMix(TO *out, size_t outFrames, const TI *in, TA *aux, bool ramp);
-
- uint32_t needs;
-
- // TODO: Eventually remove legacy integer volume settings
- union {
- int16_t volume[MAX_NUM_VOLUMES]; // U4.12 fixed point (top bit should be zero)
- int32_t volumeRL;
- };
-
- int32_t prevVolume[MAX_NUM_VOLUMES];
- int32_t volumeInc[MAX_NUM_VOLUMES];
- int32_t auxInc;
- int32_t prevAuxLevel;
- int16_t auxLevel; // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance
-
- uint16_t frameCount;
-
- uint8_t channelCount; // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK)
- uint8_t unused_padding; // formerly format, was always 16
- uint16_t enabled; // actually bool
- audio_channel_mask_t channelMask;
-
- // actual buffer provider used by the track hooks, see DownmixerBufferProvider below
- // for how the Track buffer provider is wrapped by another one when dowmixing is required
- AudioBufferProvider* bufferProvider;
-
- mutable AudioBufferProvider::Buffer buffer; // 8 bytes
-
- hook_t hook;
- const void *mIn; // current location in buffer
-
- std::unique_ptr<AudioResampler> mResampler;
- uint32_t sampleRate;
- int32_t* mainBuffer;
- int32_t* auxBuffer;
-
/* Buffer providers are constructed to translate the track input data as needed.
+ * See DownmixerBufferProvider below for how the Track buffer provider
+ * is wrapped by another one when dowmixing is required.
*
* TODO: perhaps make a single PlaybackConverterProvider class to move
* all pre-mixer track buffer conversions outside the AudioMixer class.
@@ -363,7 +152,7 @@
* the downmixer requirements to the mixer engine input requirements.
* 7) mTimestretchBufferProvider: Adds timestretching for playback rate
*/
- AudioBufferProvider* mInputBufferProvider; // externally provided buffer provider.
+ AudioBufferProvider* mInputBufferProvider; // externally provided buffer provider.
// TODO: combine mAdjustChannelsBufferProvider and
// mContractChannelsNonDestructiveBufferProvider
std::unique_ptr<PassthruBufferProvider> mAdjustChannelsBufferProvider;
@@ -373,27 +162,10 @@
std::unique_ptr<PassthruBufferProvider> mPostDownmixReformatBufferProvider;
std::unique_ptr<PassthruBufferProvider> mTimestretchBufferProvider;
- int32_t sessionId;
-
- audio_format_t mMixerFormat; // output mix format: AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
- audio_format_t mFormat; // input track format
- audio_format_t mMixerInFormat; // mix internal format AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
- // each track must be converted to this format.
audio_format_t mDownmixRequiresFormat; // required downmixer format
// AUDIO_FORMAT_PCM_16_BIT if 16 bit necessary
// AUDIO_FORMAT_INVALID if no required format
- float mVolume[MAX_NUM_VOLUMES]; // floating point set volume
- float mPrevVolume[MAX_NUM_VOLUMES]; // floating point previous volume
- float mVolumeInc[MAX_NUM_VOLUMES]; // floating point volume increment
-
- float mAuxLevel; // floating point set aux level
- float mPrevAuxLevel; // floating point prev aux level
- float mAuxInc; // floating point aux increment
-
- audio_channel_mask_t mMixerChannelMask;
- uint32_t mMixerChannelCount;
-
AudioPlaybackRate mPlaybackRate;
// Haptic
@@ -440,76 +212,23 @@
return 0.0f;
}
}
-
- private:
- // hooks
- void track__genericResample(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
- void track__16BitsStereo(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
- void track__16BitsMono(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
-
- void volumeRampStereo(int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
- void volumeStereo(int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
-
- // multi-format track hooks
- template <int MIXTYPE, typename TO, typename TI, typename TA>
- void track__Resample(TO* out, size_t frameCount, TO* temp __unused, TA* aux);
- template <int MIXTYPE, typename TO, typename TI, typename TA>
- void track__NoResample(TO* out, size_t frameCount, TO* temp __unused, TA* aux);
};
- // TODO: remove BLOCKSIZE unit of processing - it isn't needed anymore.
- static constexpr int BLOCKSIZE = 16;
-
- bool setChannelMasks(int name,
- audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask);
-
- // Called when track info changes and a new process hook should be determined.
- void invalidate() {
- mHook = &AudioMixer::process__validate;
+ inline std::shared_ptr<Track> getTrack(int name) {
+ return std::static_pointer_cast<Track>(mTracks[name]);
}
- void process__validate();
- void process__nop();
- void process__genericNoResampling();
- void process__genericResampling();
- void process__oneTrack16BitsStereoNoResampling();
+ std::shared_ptr<TrackBase> preCreateTrack() override;
+ status_t postCreateTrack(TrackBase *track) override;
- template <int MIXTYPE, typename TO, typename TI, typename TA>
- void process__noResampleOneTrack();
+ void preProcess() override;
+ void postProcess() override;
- void processHapticData();
-
- static process_hook_t getProcessHook(int processType, uint32_t channelCount,
- audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
-
- static void convertMixerFormat(void *out, audio_format_t mixerOutFormat,
- void *in, audio_format_t mixerInFormat, size_t sampleCount);
+ bool setChannelMasks(int name,
+ audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask) override;
static void sInitRoutine();
- // initialization constants
- const uint32_t mSampleRate;
- const size_t mFrameCount;
-
- NBLog::Writer *mNBLogWriter = nullptr; // associated NBLog::Writer
-
- process_hook_t mHook = &AudioMixer::process__nop; // one of process__*, never nullptr
-
- // the size of the type (int32_t) should be the largest of all types supported
- // by the mixer.
- std::unique_ptr<int32_t[]> mOutputTemp;
- std::unique_ptr<int32_t[]> mResampleTemp;
-
- // track names grouped by main buffer, in no particular order of main buffer.
- // however names for a particular main buffer are in order (by construction).
- std::unordered_map<void * /* mainBuffer */, std::vector<int /* name */>> mGroups;
-
- // track names that are enabled, in increasing order (by construction).
- std::vector<int /* name */> mEnabled;
-
- // track smart pointers, by name, in increasing order of name.
- std::map<int /* name */, std::shared_ptr<Track>> mTracks;
-
static pthread_once_t sOnceControl; // initialized in constructor by first new
};
diff --git a/media/libaudioclient/include/media/AudioPolicy.h b/media/libaudioclient/include/media/AudioPolicy.h
index ef39fd1..0ab1c9d 100644
--- a/media/libaudioclient/include/media/AudioPolicy.h
+++ b/media/libaudioclient/include/media/AudioPolicy.h
@@ -18,9 +18,10 @@
#ifndef ANDROID_AUDIO_POLICY_H
#define ANDROID_AUDIO_POLICY_H
+#include <binder/Parcel.h>
+#include <media/AudioDeviceTypeAddr.h>
#include <system/audio.h>
#include <system/audio_policy.h>
-#include <binder/Parcel.h>
#include <utils/String8.h>
#include <utils/Vector.h>
@@ -60,19 +61,6 @@
#define MAX_MIXES_PER_POLICY 10
#define MAX_CRITERIA_PER_MIX 20
-class AudioDeviceTypeAddr {
-public:
- AudioDeviceTypeAddr() {}
- AudioDeviceTypeAddr(audio_devices_t type, String8 address) :
- mType(type), mAddress(address) {}
-
- status_t readFromParcel(Parcel *parcel);
- status_t writeToParcel(Parcel *parcel) const;
-
- audio_devices_t mType;
- String8 mAddress;
-};
-
class AudioMixMatchCriterion {
public:
AudioMixMatchCriterion() {}
diff --git a/media/libaudioclient/include/media/AudioSystem.h b/media/libaudioclient/include/media/AudioSystem.h
index 09e80b2..a86297d 100644
--- a/media/libaudioclient/include/media/AudioSystem.h
+++ b/media/libaudioclient/include/media/AudioSystem.h
@@ -19,6 +19,7 @@
#include <sys/types.h>
+#include <media/AudioDeviceTypeAddr.h>
#include <media/AudioPolicy.h>
#include <media/AudioProductStrategy.h>
#include <media/AudioVolumeGroup.h>
@@ -396,6 +397,23 @@
static status_t setRttEnabled(bool enabled);
+ /**
+ * Send audio HAL server process pids to native audioserver process for use
+ * when generating audio HAL servers tombstones
+ */
+ static status_t setAudioHalPids(const std::vector<pid_t>& pids);
+
+ static status_t setPreferredDeviceForStrategy(product_strategy_t strategy,
+ const AudioDeviceTypeAddr &device);
+
+ static status_t removePreferredDeviceForStrategy(product_strategy_t strategy);
+
+ static status_t getPreferredDeviceForStrategy(product_strategy_t strategy,
+ AudioDeviceTypeAddr &device);
+
+ static status_t getDeviceForStrategy(product_strategy_t strategy,
+ AudioDeviceTypeAddr &device);
+
// ----------------------------------------------------------------------------
class AudioVolumeGroupCallback : public RefBase
diff --git a/media/libmedia/include/media/ExtendedAudioBufferProvider.h b/media/libaudioclient/include/media/ExtendedAudioBufferProvider.h
similarity index 100%
rename from media/libmedia/include/media/ExtendedAudioBufferProvider.h
rename to media/libaudioclient/include/media/ExtendedAudioBufferProvider.h
diff --git a/media/libaudioclient/include/media/IAudioFlinger.h b/media/libaudioclient/include/media/IAudioFlinger.h
index 8ec8931..1c35ff0 100644
--- a/media/libaudioclient/include/media/IAudioFlinger.h
+++ b/media/libaudioclient/include/media/IAudioFlinger.h
@@ -27,6 +27,7 @@
#include <binder/Parcel.h>
#include <binder/Parcelable.h>
#include <media/AudioClient.h>
+#include <media/DeviceDescriptorBase.h>
#include <media/IAudioTrack.h>
#include <media/IAudioFlingerClient.h>
#include <system/audio.h>
@@ -404,8 +405,7 @@
virtual status_t openOutput(audio_module_handle_t module,
audio_io_handle_t *output,
audio_config_t *config,
- audio_devices_t *devices,
- const String8& address,
+ const sp<DeviceDescriptorBase>& device,
uint32_t *latencyMs,
audio_output_flags_t flags) = 0;
virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1,
@@ -453,6 +453,7 @@
// AudioFlinger doesn't take over handle reference from client
audio_io_handle_t output,
audio_session_t sessionId,
+ const AudioDeviceTypeAddr& device,
const String16& callingPackage,
pid_t pid,
status_t *status,
@@ -511,6 +512,8 @@
/* List available microphones and their characteristics */
virtual status_t getMicrophones(std::vector<media::MicrophoneInfo> *microphones) = 0;
+
+ virtual status_t setAudioHalPids(const std::vector<pid_t>& pids) = 0;
};
diff --git a/media/libaudioclient/include/media/IAudioPolicyService.h b/media/libaudioclient/include/media/IAudioPolicyService.h
index 32275cf..9b91d6d 100644
--- a/media/libaudioclient/include/media/IAudioPolicyService.h
+++ b/media/libaudioclient/include/media/IAudioPolicyService.h
@@ -23,6 +23,7 @@
#include <utils/RefBase.h>
#include <utils/Errors.h>
#include <binder/IInterface.h>
+#include <media/AudioDeviceTypeAddr.h>
#include <media/AudioSystem.h>
#include <media/AudioPolicy.h>
#include <media/IAudioPolicyServiceClient.h>
@@ -222,6 +223,14 @@
volume_group_t &volumeGroup) = 0;
virtual status_t setRttEnabled(bool enabled) = 0;
+
+ virtual status_t setPreferredDeviceForStrategy(product_strategy_t strategy,
+ const AudioDeviceTypeAddr &device) = 0;
+
+ virtual status_t removePreferredDeviceForStrategy(product_strategy_t strategy) = 0;
+
+ virtual status_t getPreferredDeviceForStrategy(product_strategy_t strategy,
+ AudioDeviceTypeAddr &device) = 0;
};
diff --git a/media/libaudioclient/tests/Android.bp b/media/libaudioclient/tests/Android.bp
index 52bb2fb..d509be6 100644
--- a/media/libaudioclient/tests/Android.bp
+++ b/media/libaudioclient/tests/Android.bp
@@ -11,6 +11,9 @@
defaults: ["libaudioclient_tests_defaults"],
srcs: ["test_create_audiotrack.cpp",
"test_create_utils.cpp"],
+ header_libs: [
+ "libmedia_headers",
+ ],
shared_libs: [
"libaudioclient",
"libbinder",
@@ -25,6 +28,9 @@
defaults: ["libaudioclient_tests_defaults"],
srcs: ["test_create_audiorecord.cpp",
"test_create_utils.cpp"],
+ header_libs: [
+ "libmedia_headers",
+ ],
shared_libs: [
"libaudioclient",
"libbinder",
diff --git a/media/libaudiofoundation/Android.bp b/media/libaudiofoundation/Android.bp
new file mode 100644
index 0000000..93bc4d9
--- /dev/null
+++ b/media/libaudiofoundation/Android.bp
@@ -0,0 +1,50 @@
+cc_library_headers {
+ name: "libaudiofoundation_headers",
+ vendor_available: true,
+ export_include_dirs: ["include"],
+ header_libs: [
+ "libaudio_system_headers",
+ "libmedia_helper_headers",
+ ],
+ export_header_lib_headers: [
+ "libaudio_system_headers",
+ "libmedia_helper_headers",
+ ],
+}
+
+cc_library {
+ name: "libaudiofoundation",
+ vendor_available: true,
+ double_loadable: true,
+
+ srcs: [
+ "AudioContainers.cpp",
+ "AudioDeviceTypeAddr.cpp",
+ "AudioGain.cpp",
+ "AudioPort.cpp",
+ "AudioProfile.cpp",
+ "DeviceDescriptorBase.cpp",
+ ],
+
+ shared_libs: [
+ "libaudioutils",
+ "libbase",
+ "libbinder",
+ "liblog",
+ "libmedia_helper",
+ "libutils",
+ ],
+
+ header_libs: [
+ "libaudiofoundation_headers",
+ ],
+
+ export_header_lib_headers: [
+ "libaudiofoundation_headers",
+ ],
+
+ cflags: [
+ "-Werror",
+ "-Wall",
+ ],
+}
diff --git a/media/libaudiofoundation/AudioContainers.cpp b/media/libaudiofoundation/AudioContainers.cpp
new file mode 100644
index 0000000..31257d5
--- /dev/null
+++ b/media/libaudiofoundation/AudioContainers.cpp
@@ -0,0 +1,110 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <sstream>
+#include <string>
+
+#include <media/AudioContainers.h>
+
+namespace android {
+
+const DeviceTypeSet& getAudioDeviceOutAllSet() {
+ static const DeviceTypeSet audioDeviceOutAllSet = DeviceTypeSet(
+ std::begin(AUDIO_DEVICE_OUT_ALL_ARRAY),
+ std::end(AUDIO_DEVICE_OUT_ALL_ARRAY));
+ return audioDeviceOutAllSet;
+}
+
+const DeviceTypeSet& getAudioDeviceOutAllA2dpSet() {
+ static const DeviceTypeSet audioDeviceOutAllA2dpSet = DeviceTypeSet(
+ std::begin(AUDIO_DEVICE_OUT_ALL_A2DP_ARRAY),
+ std::end(AUDIO_DEVICE_OUT_ALL_A2DP_ARRAY));
+ return audioDeviceOutAllA2dpSet;
+}
+
+const DeviceTypeSet& getAudioDeviceOutAllScoSet() {
+ static const DeviceTypeSet audioDeviceOutAllScoSet = DeviceTypeSet(
+ std::begin(AUDIO_DEVICE_OUT_ALL_SCO_ARRAY),
+ std::end(AUDIO_DEVICE_OUT_ALL_SCO_ARRAY));
+ return audioDeviceOutAllScoSet;
+}
+
+const DeviceTypeSet& getAudioDeviceOutAllUsbSet() {
+ static const DeviceTypeSet audioDeviceOutAllUsbSet = DeviceTypeSet(
+ std::begin(AUDIO_DEVICE_OUT_ALL_USB_ARRAY),
+ std::end(AUDIO_DEVICE_OUT_ALL_USB_ARRAY));
+ return audioDeviceOutAllUsbSet;
+}
+
+const DeviceTypeSet& getAudioDeviceInAllSet() {
+ static const DeviceTypeSet audioDeviceInAllSet = DeviceTypeSet(
+ std::begin(AUDIO_DEVICE_IN_ALL_ARRAY),
+ std::end(AUDIO_DEVICE_IN_ALL_ARRAY));
+ return audioDeviceInAllSet;
+}
+
+const DeviceTypeSet& getAudioDeviceInAllUsbSet() {
+ static const DeviceTypeSet audioDeviceInAllUsbSet = DeviceTypeSet(
+ std::begin(AUDIO_DEVICE_IN_ALL_USB_ARRAY),
+ std::end(AUDIO_DEVICE_IN_ALL_USB_ARRAY));
+ return audioDeviceInAllUsbSet;
+}
+
+bool deviceTypesToString(const DeviceTypeSet &deviceTypes, std::string &str) {
+ if (deviceTypes.empty()) {
+ str = "Empty device types";
+ return true;
+ }
+ bool ret = true;
+ for (auto it = deviceTypes.begin(); it != deviceTypes.end();) {
+ std::string deviceTypeStr;
+ ret = audio_is_output_device(*it) ?
+ OutputDeviceConverter::toString(*it, deviceTypeStr) :
+ InputDeviceConverter::toString(*it, deviceTypeStr);
+ if (!ret) {
+ break;
+ }
+ str.append(deviceTypeStr);
+ if (++it != deviceTypes.end()) {
+ str.append(" , ");
+ }
+ }
+ if (!ret) {
+ str = "Unknown values";
+ }
+ return ret;
+}
+
+std::string dumpDeviceTypes(const DeviceTypeSet &deviceTypes) {
+ std::string ret;
+ for (auto it = deviceTypes.begin(); it != deviceTypes.end();) {
+ std::stringstream ss;
+ ss << "0x" << std::hex << (*it);
+ ret.append(ss.str());
+ if (++it != deviceTypes.end()) {
+ ret.append(" , ");
+ }
+ }
+ return ret;
+}
+
+std::string toString(const DeviceTypeSet& deviceTypes) {
+ std::string ret;
+ deviceTypesToString(deviceTypes, ret);
+ return ret;
+}
+
+} // namespace android
diff --git a/media/libaudiofoundation/AudioDeviceTypeAddr.cpp b/media/libaudiofoundation/AudioDeviceTypeAddr.cpp
new file mode 100644
index 0000000..b44043a
--- /dev/null
+++ b/media/libaudiofoundation/AudioDeviceTypeAddr.cpp
@@ -0,0 +1,67 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <media/AudioDeviceTypeAddr.h>
+
+namespace android {
+
+const char* AudioDeviceTypeAddr::getAddress() const {
+ return mAddress.c_str();
+}
+
+bool AudioDeviceTypeAddr::equals(const AudioDeviceTypeAddr& other) const {
+ return mType == other.mType && mAddress == other.mAddress;
+}
+
+bool AudioDeviceTypeAddr::operator<(const AudioDeviceTypeAddr& other) const {
+ if (mType < other.mType) return true;
+ if (mType > other.mType) return false;
+
+ if (mAddress < other.mAddress) return true;
+ // if (mAddress > other.mAddress) return false;
+
+ return false;
+}
+
+void AudioDeviceTypeAddr::reset() {
+ mType = AUDIO_DEVICE_NONE;
+ mAddress = "";
+}
+
+status_t AudioDeviceTypeAddr::readFromParcel(const Parcel *parcel) {
+ status_t status;
+ if ((status = parcel->readUint32(&mType)) != NO_ERROR) return status;
+ status = parcel->readUtf8FromUtf16(&mAddress);
+ return status;
+}
+
+status_t AudioDeviceTypeAddr::writeToParcel(Parcel *parcel) const {
+ status_t status;
+ if ((status = parcel->writeUint32(mType)) != NO_ERROR) return status;
+ status = parcel->writeUtf8AsUtf16(mAddress);
+ return status;
+}
+
+
+DeviceTypeSet getAudioDeviceTypes(const AudioDeviceTypeAddrVector& deviceTypeAddrs) {
+ DeviceTypeSet deviceTypes;
+ for (const auto& deviceTypeAddr : deviceTypeAddrs) {
+ deviceTypes.insert(deviceTypeAddr.mType);
+ }
+ return deviceTypes;
+}
+
+}
\ No newline at end of file
diff --git a/media/libaudiofoundation/AudioGain.cpp b/media/libaudiofoundation/AudioGain.cpp
new file mode 100644
index 0000000..0d28335
--- /dev/null
+++ b/media/libaudiofoundation/AudioGain.cpp
@@ -0,0 +1,199 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "AudioGain"
+//#define LOG_NDEBUG 0
+
+//#define VERY_VERBOSE_LOGGING
+#ifdef VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+#else
+#define ALOGVV(a...) do { } while(0)
+#endif
+
+#include <algorithm>
+
+#include <android-base/stringprintf.h>
+#include <media/AudioGain.h>
+#include <utils/Log.h>
+
+#include <math.h>
+
+namespace android {
+
+AudioGain::AudioGain(int index, bool useInChannelMask)
+{
+ mIndex = index;
+ mUseInChannelMask = useInChannelMask;
+ memset(&mGain, 0, sizeof(struct audio_gain));
+}
+
+void AudioGain::getDefaultConfig(struct audio_gain_config *config)
+{
+ config->index = mIndex;
+ config->mode = mGain.mode;
+ config->channel_mask = mGain.channel_mask;
+ if ((mGain.mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
+ config->values[0] = mGain.default_value;
+ } else {
+ uint32_t numValues;
+ if (mUseInChannelMask) {
+ numValues = audio_channel_count_from_in_mask(mGain.channel_mask);
+ } else {
+ numValues = audio_channel_count_from_out_mask(mGain.channel_mask);
+ }
+ for (size_t i = 0; i < numValues; i++) {
+ config->values[i] = mGain.default_value;
+ }
+ }
+ if ((mGain.mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
+ config->ramp_duration_ms = mGain.min_ramp_ms;
+ }
+}
+
+status_t AudioGain::checkConfig(const struct audio_gain_config *config)
+{
+ if ((config->mode & ~mGain.mode) != 0) {
+ return BAD_VALUE;
+ }
+ if ((config->mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
+ if ((config->values[0] < mGain.min_value) ||
+ (config->values[0] > mGain.max_value)) {
+ return BAD_VALUE;
+ }
+ } else {
+ if ((config->channel_mask & ~mGain.channel_mask) != 0) {
+ return BAD_VALUE;
+ }
+ uint32_t numValues;
+ if (mUseInChannelMask) {
+ numValues = audio_channel_count_from_in_mask(config->channel_mask);
+ } else {
+ numValues = audio_channel_count_from_out_mask(config->channel_mask);
+ }
+ for (size_t i = 0; i < numValues; i++) {
+ if ((config->values[i] < mGain.min_value) ||
+ (config->values[i] > mGain.max_value)) {
+ return BAD_VALUE;
+ }
+ }
+ }
+ if ((config->mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
+ if ((config->ramp_duration_ms < mGain.min_ramp_ms) ||
+ (config->ramp_duration_ms > mGain.max_ramp_ms)) {
+ return BAD_VALUE;
+ }
+ }
+ return NO_ERROR;
+}
+
+void AudioGain::dump(std::string *dst, int spaces, int index) const
+{
+ dst->append(base::StringPrintf("%*sGain %d:\n", spaces, "", index+1));
+ dst->append(base::StringPrintf("%*s- mode: %08x\n", spaces, "", mGain.mode));
+ dst->append(base::StringPrintf("%*s- channel_mask: %08x\n", spaces, "", mGain.channel_mask));
+ dst->append(base::StringPrintf("%*s- min_value: %d mB\n", spaces, "", mGain.min_value));
+ dst->append(base::StringPrintf("%*s- max_value: %d mB\n", spaces, "", mGain.max_value));
+ dst->append(base::StringPrintf("%*s- default_value: %d mB\n", spaces, "", mGain.default_value));
+ dst->append(base::StringPrintf("%*s- step_value: %d mB\n", spaces, "", mGain.step_value));
+ dst->append(base::StringPrintf("%*s- min_ramp_ms: %d ms\n", spaces, "", mGain.min_ramp_ms));
+ dst->append(base::StringPrintf("%*s- max_ramp_ms: %d ms\n", spaces, "", mGain.max_ramp_ms));
+}
+
+bool AudioGain::equals(const sp<AudioGain>& other) const
+{
+ return other != nullptr &&
+ mUseInChannelMask == other->mUseInChannelMask &&
+ mUseForVolume == other->mUseForVolume &&
+ // Compare audio gain
+ mGain.mode == other->mGain.mode &&
+ mGain.channel_mask == other->mGain.channel_mask &&
+ mGain.min_value == other->mGain.min_value &&
+ mGain.max_value == other->mGain.max_value &&
+ mGain.default_value == other->mGain.default_value &&
+ mGain.step_value == other->mGain.step_value &&
+ mGain.min_ramp_ms == other->mGain.min_ramp_ms &&
+ mGain.max_ramp_ms == other->mGain.max_ramp_ms;
+}
+
+status_t AudioGain::writeToParcel(android::Parcel *parcel) const
+{
+ status_t status = NO_ERROR;
+ if ((status = parcel->writeInt32(mIndex)) != NO_ERROR) return status;
+ if ((status = parcel->writeBool(mUseInChannelMask)) != NO_ERROR) return status;
+ if ((status = parcel->writeBool(mUseForVolume)) != NO_ERROR) return status;
+ if ((status = parcel->writeUint32(mGain.mode)) != NO_ERROR) return status;
+ if ((status = parcel->writeUint32(mGain.channel_mask)) != NO_ERROR) return status;
+ if ((status = parcel->writeInt32(mGain.min_value)) != NO_ERROR) return status;
+ if ((status = parcel->writeInt32(mGain.max_value)) != NO_ERROR) return status;
+ if ((status = parcel->writeInt32(mGain.default_value)) != NO_ERROR) return status;
+ if ((status = parcel->writeUint32(mGain.step_value)) != NO_ERROR) return status;
+ if ((status = parcel->writeUint32(mGain.min_ramp_ms)) != NO_ERROR) return status;
+ status = parcel->writeUint32(mGain.max_ramp_ms);
+ return status;
+}
+
+status_t AudioGain::readFromParcel(const android::Parcel *parcel)
+{
+ status_t status = NO_ERROR;
+ if ((status = parcel->readInt32(&mIndex)) != NO_ERROR) return status;
+ if ((status = parcel->readBool(&mUseInChannelMask)) != NO_ERROR) return status;
+ if ((status = parcel->readBool(&mUseForVolume)) != NO_ERROR) return status;
+ if ((status = parcel->readUint32(&mGain.mode)) != NO_ERROR) return status;
+ if ((status = parcel->readUint32(&mGain.channel_mask)) != NO_ERROR) return status;
+ if ((status = parcel->readInt32(&mGain.min_value)) != NO_ERROR) return status;
+ if ((status = parcel->readInt32(&mGain.max_value)) != NO_ERROR) return status;
+ if ((status = parcel->readInt32(&mGain.default_value)) != NO_ERROR) return status;
+ if ((status = parcel->readUint32(&mGain.step_value)) != NO_ERROR) return status;
+ if ((status = parcel->readUint32(&mGain.min_ramp_ms)) != NO_ERROR) return status;
+ status = parcel->readUint32(&mGain.max_ramp_ms);
+ return status;
+}
+
+bool AudioGains::equals(const AudioGains &other) const
+{
+ return std::equal(begin(), end(), other.begin(), other.end(),
+ [](const sp<AudioGain>& left, const sp<AudioGain>& right) {
+ return left->equals(right);
+ });
+}
+
+status_t AudioGains::writeToParcel(android::Parcel *parcel) const {
+ status_t status = NO_ERROR;
+ if ((status = parcel->writeVectorSize(*this)) != NO_ERROR) return status;
+ for (const auto &audioGain : *this) {
+ if ((status = parcel->writeParcelable(*audioGain)) != NO_ERROR) {
+ break;
+ }
+ }
+ return status;
+}
+
+status_t AudioGains::readFromParcel(const android::Parcel *parcel) {
+ status_t status = NO_ERROR;
+ this->clear();
+ if ((status = parcel->resizeOutVector(this)) != NO_ERROR) return status;
+ for (size_t i = 0; i < this->size(); i++) {
+ this->at(i) = new AudioGain(0, false);
+ if ((status = parcel->readParcelable(this->at(i).get())) != NO_ERROR) {
+ this->clear();
+ break;
+ }
+ }
+ return status;
+}
+
+} // namespace android
diff --git a/media/libaudiofoundation/AudioPort.cpp b/media/libaudiofoundation/AudioPort.cpp
new file mode 100644
index 0000000..f988690
--- /dev/null
+++ b/media/libaudiofoundation/AudioPort.cpp
@@ -0,0 +1,287 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+#define LOG_TAG "AudioPort"
+
+#include <algorithm>
+
+#include <android-base/stringprintf.h>
+#include <media/AudioPort.h>
+#include <utils/Log.h>
+
+namespace android {
+
+void AudioPort::importAudioPort(const sp<AudioPort>& port, bool force __unused)
+{
+ for (const auto& profileToImport : port->mProfiles) {
+ // Import only valid port, i.e. valid format, non empty rates and channels masks
+ if (!profileToImport->isValid()) {
+ continue;
+ }
+ if (std::find_if(mProfiles.begin(), mProfiles.end(),
+ [profileToImport](const auto &profile) {
+ return *profile == *profileToImport; }) == mProfiles.end()) {
+ addAudioProfile(profileToImport);
+ }
+ }
+}
+
+void AudioPort::toAudioPort(struct audio_port *port) const {
+ // TODO: update this function once audio_port structure reflects the new profile definition.
+ // For compatibility reason: flatening the AudioProfile into audio_port structure.
+ FormatSet flatenedFormats;
+ SampleRateSet flatenedRates;
+ ChannelMaskSet flatenedChannels;
+ for (const auto& profile : mProfiles) {
+ if (profile->isValid()) {
+ audio_format_t formatToExport = profile->getFormat();
+ const SampleRateSet &ratesToExport = profile->getSampleRates();
+ const ChannelMaskSet &channelsToExport = profile->getChannels();
+
+ flatenedFormats.insert(formatToExport);
+ flatenedRates.insert(ratesToExport.begin(), ratesToExport.end());
+ flatenedChannels.insert(channelsToExport.begin(), channelsToExport.end());
+
+ if (flatenedRates.size() > AUDIO_PORT_MAX_SAMPLING_RATES ||
+ flatenedChannels.size() > AUDIO_PORT_MAX_CHANNEL_MASKS ||
+ flatenedFormats.size() > AUDIO_PORT_MAX_FORMATS) {
+ ALOGE("%s: bailing out: cannot export profiles to port config", __func__);
+ return;
+ }
+ }
+ }
+ port->role = mRole;
+ port->type = mType;
+ strlcpy(port->name, mName.c_str(), AUDIO_PORT_MAX_NAME_LEN);
+ port->num_sample_rates = flatenedRates.size();
+ port->num_channel_masks = flatenedChannels.size();
+ port->num_formats = flatenedFormats.size();
+ std::copy(flatenedRates.begin(), flatenedRates.end(), port->sample_rates);
+ std::copy(flatenedChannels.begin(), flatenedChannels.end(), port->channel_masks);
+ std::copy(flatenedFormats.begin(), flatenedFormats.end(), port->formats);
+
+ ALOGV("AudioPort::toAudioPort() num gains %zu", mGains.size());
+
+ port->num_gains = std::min(mGains.size(), (size_t) AUDIO_PORT_MAX_GAINS);
+ for (size_t i = 0; i < port->num_gains; i++) {
+ port->gains[i] = mGains[i]->getGain();
+ }
+}
+
+void AudioPort::dump(std::string *dst, int spaces, bool verbose) const {
+ if (!mName.empty()) {
+ dst->append(base::StringPrintf("%*s- name: %s\n", spaces, "", mName.c_str()));
+ }
+ if (verbose) {
+ std::string profilesStr;
+ mProfiles.dump(&profilesStr, spaces);
+ dst->append(profilesStr);
+
+ if (mGains.size() != 0) {
+ dst->append(base::StringPrintf("%*s- gains:\n", spaces, ""));
+ for (size_t i = 0; i < mGains.size(); i++) {
+ std::string gainStr;
+ mGains[i]->dump(&gainStr, spaces + 2, i);
+ dst->append(gainStr);
+ }
+ }
+ }
+}
+
+void AudioPort::log(const char* indent) const
+{
+ ALOGI("%s Port[nm:%s, type:%d, role:%d]", indent, mName.c_str(), mType, mRole);
+}
+
+bool AudioPort::equals(const sp<AudioPort> &other) const
+{
+ return other != nullptr &&
+ mGains.equals(other->getGains()) &&
+ mName.compare(other->getName()) == 0 &&
+ mType == other->getType() &&
+ mRole == other->getRole() &&
+ mProfiles.equals(other->getAudioProfiles());
+}
+
+status_t AudioPort::writeToParcel(Parcel *parcel) const
+{
+ status_t status = NO_ERROR;
+ if ((status = parcel->writeUtf8AsUtf16(mName)) != NO_ERROR) return status;
+ if ((status = parcel->writeUint32(mType)) != NO_ERROR) return status;
+ if ((status = parcel->writeUint32(mRole)) != NO_ERROR) return status;
+ if ((status = parcel->writeParcelable(mProfiles)) != NO_ERROR) return status;
+ if ((status = parcel->writeParcelable(mGains)) != NO_ERROR) return status;
+ return status;
+}
+
+status_t AudioPort::readFromParcel(const Parcel *parcel)
+{
+ status_t status = NO_ERROR;
+ if ((status = parcel->readUtf8FromUtf16(&mName)) != NO_ERROR) return status;
+ static_assert(sizeof(mType) == sizeof(uint32_t));
+ if ((status = parcel->readUint32(reinterpret_cast<uint32_t*>(&mType))) != NO_ERROR) {
+ return status;
+ }
+ static_assert(sizeof(mRole) == sizeof(uint32_t));
+ if ((status = parcel->readUint32(reinterpret_cast<uint32_t*>(&mRole))) != NO_ERROR) {
+ return status;
+ }
+ mProfiles.clear();
+ if ((status = parcel->readParcelable(&mProfiles)) != NO_ERROR) return status;
+ mGains.clear();
+ if ((status = parcel->readParcelable(&mGains)) != NO_ERROR) return status;
+ return status;
+}
+
+// --- AudioPortConfig class implementation
+
+status_t AudioPortConfig::applyAudioPortConfig(
+ const struct audio_port_config *config,
+ struct audio_port_config *backupConfig __unused)
+{
+ if (config->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
+ mSamplingRate = config->sample_rate;
+ }
+ if (config->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
+ mChannelMask = config->channel_mask;
+ }
+ if (config->config_mask & AUDIO_PORT_CONFIG_FORMAT) {
+ mFormat = config->format;
+ }
+ if (config->config_mask & AUDIO_PORT_CONFIG_GAIN) {
+ mGain = config->gain;
+ }
+
+ return NO_ERROR;
+}
+
+namespace {
+
+template<typename T>
+void updateField(
+ const T& portConfigField, T audio_port_config::*port_config_field,
+ struct audio_port_config *dstConfig, const struct audio_port_config *srcConfig,
+ unsigned int configMask, T defaultValue)
+{
+ if (dstConfig->config_mask & configMask) {
+ if ((srcConfig != nullptr) && (srcConfig->config_mask & configMask)) {
+ dstConfig->*port_config_field = srcConfig->*port_config_field;
+ } else {
+ dstConfig->*port_config_field = portConfigField;
+ }
+ } else {
+ dstConfig->*port_config_field = defaultValue;
+ }
+}
+
+} // namespace
+
+void AudioPortConfig::toAudioPortConfig(
+ struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig) const
+{
+ updateField(mSamplingRate, &audio_port_config::sample_rate,
+ dstConfig, srcConfig, AUDIO_PORT_CONFIG_SAMPLE_RATE, 0u);
+ updateField(mChannelMask, &audio_port_config::channel_mask,
+ dstConfig, srcConfig, AUDIO_PORT_CONFIG_CHANNEL_MASK,
+ (audio_channel_mask_t)AUDIO_CHANNEL_NONE);
+ updateField(mFormat, &audio_port_config::format,
+ dstConfig, srcConfig, AUDIO_PORT_CONFIG_FORMAT, AUDIO_FORMAT_INVALID);
+ dstConfig->id = mId;
+
+ sp<AudioPort> audioport = getAudioPort();
+ if ((dstConfig->config_mask & AUDIO_PORT_CONFIG_GAIN) && audioport != NULL) {
+ dstConfig->gain = mGain;
+ if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_GAIN)
+ && audioport->checkGain(&srcConfig->gain, srcConfig->gain.index) == OK) {
+ dstConfig->gain = srcConfig->gain;
+ }
+ } else {
+ dstConfig->gain.index = -1;
+ }
+ if (dstConfig->gain.index != -1) {
+ dstConfig->config_mask |= AUDIO_PORT_CONFIG_GAIN;
+ } else {
+ dstConfig->config_mask &= ~AUDIO_PORT_CONFIG_GAIN;
+ }
+}
+
+bool AudioPortConfig::hasGainController(bool canUseForVolume) const
+{
+ sp<AudioPort> audioport = getAudioPort();
+ if (!audioport) {
+ return false;
+ }
+ return canUseForVolume ? audioport->getGains().canUseForVolume()
+ : audioport->getGains().size() > 0;
+}
+
+bool AudioPortConfig::equals(const sp<AudioPortConfig> &other) const
+{
+ return other != nullptr &&
+ mSamplingRate == other->getSamplingRate() &&
+ mFormat == other->getFormat() &&
+ mChannelMask == other->getChannelMask() &&
+ // Compare audio gain config
+ mGain.index == other->mGain.index &&
+ mGain.mode == other->mGain.mode &&
+ mGain.channel_mask == other->mGain.channel_mask &&
+ std::equal(std::begin(mGain.values), std::end(mGain.values),
+ std::begin(other->mGain.values)) &&
+ mGain.ramp_duration_ms == other->mGain.ramp_duration_ms;
+}
+
+status_t AudioPortConfig::writeToParcel(Parcel *parcel) const
+{
+ status_t status = NO_ERROR;
+ if ((status = parcel->writeUint32(mSamplingRate)) != NO_ERROR) return status;
+ if ((status = parcel->writeUint32(mFormat)) != NO_ERROR) return status;
+ if ((status = parcel->writeUint32(mChannelMask)) != NO_ERROR) return status;
+ if ((status = parcel->writeInt32(mId)) != NO_ERROR) return status;
+ // Write mGain to parcel.
+ if ((status = parcel->writeInt32(mGain.index)) != NO_ERROR) return status;
+ if ((status = parcel->writeUint32(mGain.mode)) != NO_ERROR) return status;
+ if ((status = parcel->writeUint32(mGain.channel_mask)) != NO_ERROR) return status;
+ if ((status = parcel->writeUint32(mGain.ramp_duration_ms)) != NO_ERROR) return status;
+ std::vector<int> values(std::begin(mGain.values), std::end(mGain.values));
+ if ((status = parcel->writeInt32Vector(values)) != NO_ERROR) return status;
+ return status;
+}
+
+status_t AudioPortConfig::readFromParcel(const Parcel *parcel)
+{
+ status_t status = NO_ERROR;
+ if ((status = parcel->readUint32(&mSamplingRate)) != NO_ERROR) return status;
+ static_assert(sizeof(mFormat) == sizeof(uint32_t));
+ if ((status = parcel->readUint32(reinterpret_cast<uint32_t*>(&mFormat))) != NO_ERROR) {
+ return status;
+ }
+ if ((status = parcel->readUint32(&mChannelMask)) != NO_ERROR) return status;
+ if ((status = parcel->readInt32(&mId)) != NO_ERROR) return status;
+ // Read mGain from parcel.
+ if ((status = parcel->readInt32(&mGain.index)) != NO_ERROR) return status;
+ if ((status = parcel->readUint32(&mGain.mode)) != NO_ERROR) return status;
+ if ((status = parcel->readUint32(&mGain.channel_mask)) != NO_ERROR) return status;
+ if ((status = parcel->readUint32(&mGain.ramp_duration_ms)) != NO_ERROR) return status;
+ std::vector<int> values;
+ if ((status = parcel->readInt32Vector(&values)) != NO_ERROR) return status;
+ if (values.size() != std::size(mGain.values)) {
+ return BAD_VALUE;
+ }
+ std::copy(values.begin(), values.end(), mGain.values);
+ return status;
+}
+
+} // namespace android
diff --git a/media/libaudiofoundation/AudioProfile.cpp b/media/libaudiofoundation/AudioProfile.cpp
new file mode 100644
index 0000000..91be346
--- /dev/null
+++ b/media/libaudiofoundation/AudioProfile.cpp
@@ -0,0 +1,307 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <set>
+
+#define LOG_TAG "AudioProfile"
+//#define LOG_NDEBUG 0
+
+#include <android-base/stringprintf.h>
+#include <media/AudioContainers.h>
+#include <media/AudioProfile.h>
+#include <media/TypeConverter.h>
+#include <utils/Errors.h>
+
+namespace android {
+
+bool operator == (const AudioProfile &left, const AudioProfile &right)
+{
+ return (left.getFormat() == right.getFormat()) &&
+ (left.getChannels() == right.getChannels()) &&
+ (left.getSampleRates() == right.getSampleRates());
+}
+
+// static
+sp<AudioProfile> AudioProfile::createFullDynamic(audio_format_t dynamicFormat)
+{
+ AudioProfile* dynamicProfile = new AudioProfile(dynamicFormat,
+ ChannelMaskSet(), SampleRateSet());
+ dynamicProfile->setDynamicFormat(true);
+ dynamicProfile->setDynamicChannels(true);
+ dynamicProfile->setDynamicRate(true);
+ return dynamicProfile;
+}
+
+AudioProfile::AudioProfile(audio_format_t format,
+ audio_channel_mask_t channelMasks,
+ uint32_t samplingRate) :
+ mName(""),
+ mFormat(format)
+{
+ mChannelMasks.insert(channelMasks);
+ mSamplingRates.insert(samplingRate);
+}
+
+AudioProfile::AudioProfile(audio_format_t format,
+ const ChannelMaskSet &channelMasks,
+ const SampleRateSet &samplingRateCollection) :
+ mName(""),
+ mFormat(format),
+ mChannelMasks(channelMasks),
+ mSamplingRates(samplingRateCollection) {}
+
+void AudioProfile::setChannels(const ChannelMaskSet &channelMasks)
+{
+ if (mIsDynamicChannels) {
+ mChannelMasks = channelMasks;
+ }
+}
+
+void AudioProfile::setSampleRates(const SampleRateSet &sampleRates)
+{
+ if (mIsDynamicRate) {
+ mSamplingRates = sampleRates;
+ }
+}
+
+void AudioProfile::clear()
+{
+ if (mIsDynamicChannels) {
+ mChannelMasks.clear();
+ }
+ if (mIsDynamicRate) {
+ mSamplingRates.clear();
+ }
+}
+
+void AudioProfile::dump(std::string *dst, int spaces) const
+{
+ dst->append(base::StringPrintf("%s%s%s\n", mIsDynamicFormat ? "[dynamic format]" : "",
+ mIsDynamicChannels ? "[dynamic channels]" : "",
+ mIsDynamicRate ? "[dynamic rates]" : ""));
+ if (mName.length() != 0) {
+ dst->append(base::StringPrintf("%*s- name: %s\n", spaces, "", mName.c_str()));
+ }
+ std::string formatLiteral;
+ if (FormatConverter::toString(mFormat, formatLiteral)) {
+ dst->append(base::StringPrintf("%*s- format: %s\n", spaces, "", formatLiteral.c_str()));
+ }
+ if (!mSamplingRates.empty()) {
+ dst->append(base::StringPrintf("%*s- sampling rates:", spaces, ""));
+ for (auto it = mSamplingRates.begin(); it != mSamplingRates.end();) {
+ dst->append(base::StringPrintf("%d", *it));
+ dst->append(++it == mSamplingRates.end() ? "" : ", ");
+ }
+ dst->append("\n");
+ }
+
+ if (!mChannelMasks.empty()) {
+ dst->append(base::StringPrintf("%*s- channel masks:", spaces, ""));
+ for (auto it = mChannelMasks.begin(); it != mChannelMasks.end();) {
+ dst->append(base::StringPrintf("0x%04x", *it));
+ dst->append(++it == mChannelMasks.end() ? "" : ", ");
+ }
+ dst->append("\n");
+ }
+}
+
+bool AudioProfile::equals(const sp<AudioProfile>& other) const
+{
+ return other != nullptr &&
+ mName.compare(other->mName) == 0 &&
+ mFormat == other->getFormat() &&
+ mChannelMasks == other->getChannels() &&
+ mSamplingRates == other->getSampleRates() &&
+ mIsDynamicFormat == other->isDynamicFormat() &&
+ mIsDynamicChannels == other->isDynamicChannels() &&
+ mIsDynamicRate == other->isDynamicRate();
+}
+
+status_t AudioProfile::writeToParcel(Parcel *parcel) const
+{
+ status_t status = NO_ERROR;
+ if ((status = parcel->writeUtf8AsUtf16(mName)) != NO_ERROR) return status;
+ if ((status = parcel->writeUint32(mFormat)) != NO_ERROR) return status;
+ std::vector<int> values(mChannelMasks.begin(), mChannelMasks.end());
+ if ((status = parcel->writeInt32Vector(values)) != NO_ERROR) return status;
+ values.clear();
+ values.assign(mSamplingRates.begin(), mSamplingRates.end());
+ if ((status = parcel->writeInt32Vector(values)) != NO_ERROR) return status;
+ if ((status = parcel->writeBool(mIsDynamicFormat)) != NO_ERROR) return status;
+ if ((status = parcel->writeBool(mIsDynamicChannels)) != NO_ERROR) return status;
+ if ((status = parcel->writeBool(mIsDynamicRate)) != NO_ERROR) return status;
+ return status;
+}
+
+status_t AudioProfile::readFromParcel(const Parcel *parcel)
+{
+ status_t status = NO_ERROR;
+ if ((status = parcel->readUtf8FromUtf16(&mName)) != NO_ERROR) return status;
+ static_assert(sizeof(mFormat) == sizeof(uint32_t));
+ if ((status = parcel->readUint32(reinterpret_cast<uint32_t*>(&mFormat))) != NO_ERROR) {
+ return status;
+ }
+ std::vector<int> values;
+ if ((status = parcel->readInt32Vector(&values)) != NO_ERROR) return status;
+ mChannelMasks.clear();
+ mChannelMasks.insert(values.begin(), values.end());
+ values.clear();
+ if ((status = parcel->readInt32Vector(&values)) != NO_ERROR) return status;
+ mSamplingRates.clear();
+ mSamplingRates.insert(values.begin(), values.end());
+ if ((status = parcel->readBool(&mIsDynamicFormat)) != NO_ERROR) return status;
+ if ((status = parcel->readBool(&mIsDynamicChannels)) != NO_ERROR) return status;
+ if ((status = parcel->readBool(&mIsDynamicRate)) != NO_ERROR) return status;
+ return status;
+}
+
+ssize_t AudioProfileVector::add(const sp<AudioProfile> &profile)
+{
+ ssize_t index = size();
+ push_back(profile);
+ return index;
+}
+
+void AudioProfileVector::clearProfiles()
+{
+ for (auto it = begin(); it != end();) {
+ if ((*it)->isDynamicFormat() && (*it)->hasValidFormat()) {
+ it = erase(it);
+ } else {
+ (*it)->clear();
+ ++it;
+ }
+ }
+}
+
+sp<AudioProfile> AudioProfileVector::getFirstValidProfile() const
+{
+ for (const auto &profile : *this) {
+ if (profile->isValid()) {
+ return profile;
+ }
+ }
+ return nullptr;
+}
+
+sp<AudioProfile> AudioProfileVector::getFirstValidProfileFor(audio_format_t format) const
+{
+ for (const auto &profile : *this) {
+ if (profile->isValid() && profile->getFormat() == format) {
+ return profile;
+ }
+ }
+ return nullptr;
+}
+
+FormatVector AudioProfileVector::getSupportedFormats() const
+{
+ FormatVector supportedFormats;
+ for (const auto &profile : *this) {
+ if (profile->hasValidFormat()) {
+ supportedFormats.push_back(profile->getFormat());
+ }
+ }
+ return supportedFormats;
+}
+
+bool AudioProfileVector::hasDynamicChannelsFor(audio_format_t format) const
+{
+ for (const auto &profile : *this) {
+ if (profile->getFormat() == format && profile->isDynamicChannels()) {
+ return true;
+ }
+ }
+ return false;
+}
+
+bool AudioProfileVector::hasDynamicFormat() const
+{
+ for (const auto &profile : *this) {
+ if (profile->isDynamicFormat()) {
+ return true;
+ }
+ }
+ return false;
+}
+
+bool AudioProfileVector::hasDynamicProfile() const
+{
+ for (const auto &profile : *this) {
+ if (profile->isDynamic()) {
+ return true;
+ }
+ }
+ return false;
+}
+
+bool AudioProfileVector::hasDynamicRateFor(audio_format_t format) const
+{
+ for (const auto &profile : *this) {
+ if (profile->getFormat() == format && profile->isDynamicRate()) {
+ return true;
+ }
+ }
+ return false;
+}
+
+void AudioProfileVector::dump(std::string *dst, int spaces) const
+{
+ dst->append(base::StringPrintf("%*s- Profiles:\n", spaces, ""));
+ for (size_t i = 0; i < size(); i++) {
+ dst->append(base::StringPrintf("%*sProfile %zu:", spaces + 4, "", i));
+ std::string profileStr;
+ at(i)->dump(&profileStr, spaces + 8);
+ dst->append(profileStr);
+ }
+}
+
+status_t AudioProfileVector::writeToParcel(Parcel *parcel) const
+{
+ status_t status = NO_ERROR;
+ if ((status = parcel->writeVectorSize(*this)) != NO_ERROR) return status;
+ for (const auto &audioProfile : *this) {
+ if ((status = parcel->writeParcelable(*audioProfile)) != NO_ERROR) {
+ break;
+ }
+ }
+ return status;
+}
+
+status_t AudioProfileVector::readFromParcel(const Parcel *parcel)
+{
+ status_t status = NO_ERROR;
+ this->clear();
+ if ((status = parcel->resizeOutVector(this)) != NO_ERROR) return status;
+ for (size_t i = 0; i < this->size(); ++i) {
+ this->at(i) = new AudioProfile(AUDIO_FORMAT_DEFAULT, AUDIO_CHANNEL_NONE, 0 /*sampleRate*/);
+ if ((status = parcel->readParcelable(this->at(i).get())) != NO_ERROR) {
+ this->clear();
+ break;
+ }
+ }
+ return status;
+}
+
+bool AudioProfileVector::equals(const AudioProfileVector& other) const
+{
+ return std::equal(begin(), end(), other.begin(), other.end(),
+ [](const sp<AudioProfile>& left, const sp<AudioProfile>& right) {
+ return left->equals(right);
+ });
+}
+
+} // namespace android
diff --git a/media/libaudiofoundation/DeviceDescriptorBase.cpp b/media/libaudiofoundation/DeviceDescriptorBase.cpp
new file mode 100644
index 0000000..ef7576e
--- /dev/null
+++ b/media/libaudiofoundation/DeviceDescriptorBase.cpp
@@ -0,0 +1,171 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "DeviceDescriptorBase"
+//#define LOG_NDEBUG 0
+
+#include <android-base/stringprintf.h>
+#include <audio_utils/string.h>
+#include <media/DeviceDescriptorBase.h>
+#include <media/TypeConverter.h>
+
+namespace android {
+
+DeviceDescriptorBase::DeviceDescriptorBase(audio_devices_t type) :
+ DeviceDescriptorBase(type, "")
+{
+}
+
+DeviceDescriptorBase::DeviceDescriptorBase(audio_devices_t type, const std::string& address) :
+ DeviceDescriptorBase(AudioDeviceTypeAddr(type, address))
+{
+}
+
+DeviceDescriptorBase::DeviceDescriptorBase(const AudioDeviceTypeAddr &deviceTypeAddr) :
+ AudioPort("", AUDIO_PORT_TYPE_DEVICE,
+ audio_is_output_device(deviceTypeAddr.mType) ? AUDIO_PORT_ROLE_SINK :
+ AUDIO_PORT_ROLE_SOURCE),
+ mDeviceTypeAddr(deviceTypeAddr)
+{
+ if (mDeviceTypeAddr.mAddress.empty() && audio_is_remote_submix_device(mDeviceTypeAddr.mType)) {
+ mDeviceTypeAddr.mAddress = "0";
+ }
+}
+
+void DeviceDescriptorBase::toAudioPortConfig(struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig) const
+{
+ dstConfig->config_mask = AUDIO_PORT_CONFIG_GAIN;
+ if (mSamplingRate != 0) {
+ dstConfig->config_mask |= AUDIO_PORT_CONFIG_SAMPLE_RATE;
+ }
+ if (mChannelMask != AUDIO_CHANNEL_NONE) {
+ dstConfig->config_mask |= AUDIO_PORT_CONFIG_CHANNEL_MASK;
+ }
+ if (mFormat != AUDIO_FORMAT_INVALID) {
+ dstConfig->config_mask |= AUDIO_PORT_CONFIG_FORMAT;
+ }
+
+ if (srcConfig != NULL) {
+ dstConfig->config_mask |= srcConfig->config_mask;
+ }
+
+ AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
+
+ dstConfig->role = audio_is_output_device(mDeviceTypeAddr.mType) ?
+ AUDIO_PORT_ROLE_SINK : AUDIO_PORT_ROLE_SOURCE;
+ dstConfig->type = AUDIO_PORT_TYPE_DEVICE;
+ dstConfig->ext.device.type = mDeviceTypeAddr.mType;
+
+ (void)audio_utils_strlcpy_zerofill(dstConfig->ext.device.address, mDeviceTypeAddr.getAddress());
+}
+
+void DeviceDescriptorBase::toAudioPort(struct audio_port *port) const
+{
+ ALOGV("DeviceDescriptorBase::toAudioPort() handle %d type %08x", mId, mDeviceTypeAddr.mType);
+ AudioPort::toAudioPort(port);
+ toAudioPortConfig(&port->active_config);
+ port->id = mId;
+ port->ext.device.type = mDeviceTypeAddr.mType;
+ (void)audio_utils_strlcpy_zerofill(port->ext.device.address, mDeviceTypeAddr.getAddress());
+}
+
+void DeviceDescriptorBase::dump(std::string *dst, int spaces, int index,
+ const char* extraInfo, bool verbose) const
+{
+ dst->append(base::StringPrintf("%*sDevice %d:\n", spaces, "", index + 1));
+ if (mId != 0) {
+ dst->append(base::StringPrintf("%*s- id: %2d\n", spaces, "", mId));
+ }
+
+ if (extraInfo != nullptr) {
+ dst->append(extraInfo);
+ }
+
+ dst->append(base::StringPrintf("%*s- type: %-48s\n",
+ spaces, "", ::android::toString(mDeviceTypeAddr.mType).c_str()));
+
+ if (mDeviceTypeAddr.mAddress.size() != 0) {
+ dst->append(base::StringPrintf(
+ "%*s- address: %-32s\n", spaces, "", mDeviceTypeAddr.getAddress()));
+ }
+ AudioPort::dump(dst, spaces, verbose);
+}
+
+std::string DeviceDescriptorBase::toString() const
+{
+ std::stringstream sstream;
+ sstream << "type:0x" << std::hex << type() << ",@:" << mDeviceTypeAddr.mAddress;
+ return sstream.str();
+}
+
+void DeviceDescriptorBase::log() const
+{
+ ALOGI("Device id:%d type:0x%08X:%s, addr:%s", mId, mDeviceTypeAddr.mType,
+ ::android::toString(mDeviceTypeAddr.mType).c_str(),
+ mDeviceTypeAddr.getAddress());
+
+ AudioPort::log(" ");
+}
+
+bool DeviceDescriptorBase::equals(const sp<DeviceDescriptorBase> &other) const
+{
+ return other != nullptr &&
+ static_cast<const AudioPort*>(this)->equals(other) &&
+ static_cast<const AudioPortConfig*>(this)->equals(other) &&
+ mDeviceTypeAddr.equals(other->mDeviceTypeAddr);
+}
+
+status_t DeviceDescriptorBase::writeToParcel(Parcel *parcel) const
+{
+ status_t status = NO_ERROR;
+ if ((status = AudioPort::writeToParcel(parcel)) != NO_ERROR) return status;
+ if ((status = AudioPortConfig::writeToParcel(parcel)) != NO_ERROR) return status;
+ if ((status = parcel->writeParcelable(mDeviceTypeAddr)) != NO_ERROR) return status;
+ return status;
+}
+
+status_t DeviceDescriptorBase::readFromParcel(const Parcel *parcel)
+{
+ status_t status = NO_ERROR;
+ if ((status = AudioPort::readFromParcel(parcel)) != NO_ERROR) return status;
+ if ((status = AudioPortConfig::readFromParcel(parcel)) != NO_ERROR) return status;
+ if ((status = parcel->readParcelable(&mDeviceTypeAddr)) != NO_ERROR) return status;
+ return status;
+}
+
+std::string toString(const DeviceDescriptorBaseVector& devices)
+{
+ std::string ret;
+ for (const auto& device : devices) {
+ if (device != *devices.begin()) {
+ ret += ";";
+ }
+ ret += device->toString();
+ }
+ return ret;
+}
+
+AudioDeviceTypeAddrVector deviceTypeAddrsFromDescriptors(const DeviceDescriptorBaseVector& devices)
+{
+ AudioDeviceTypeAddrVector deviceTypeAddrs;
+ for (const auto& device : devices) {
+ deviceTypeAddrs.push_back(device->getDeviceTypeAddr());
+ }
+ return deviceTypeAddrs;
+}
+
+} // namespace android
diff --git a/media/libaudiofoundation/TEST_MAPPING b/media/libaudiofoundation/TEST_MAPPING
new file mode 100644
index 0000000..f6d249a
--- /dev/null
+++ b/media/libaudiofoundation/TEST_MAPPING
@@ -0,0 +1,7 @@
+{
+ "presubmit": [
+ {
+ "name": "audiofoundation_parcelable_test"
+ }
+ ]
+}
diff --git a/media/libaudiofoundation/include/media/AudioContainers.h b/media/libaudiofoundation/include/media/AudioContainers.h
new file mode 100644
index 0000000..72fda49
--- /dev/null
+++ b/media/libaudiofoundation/include/media/AudioContainers.h
@@ -0,0 +1,134 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#pragma once
+
+#include <algorithm>
+#include <functional>
+#include <iterator>
+#include <set>
+#include <vector>
+
+#include <media/TypeConverter.h>
+#include <system/audio.h>
+
+namespace android {
+
+using ChannelMaskSet = std::set<audio_channel_mask_t>;
+using DeviceTypeSet = std::set<audio_devices_t>;
+using FormatSet = std::set<audio_format_t>;
+using SampleRateSet = std::set<uint32_t>;
+
+using FormatVector = std::vector<audio_format_t>;
+
+const DeviceTypeSet& getAudioDeviceOutAllSet();
+const DeviceTypeSet& getAudioDeviceOutAllA2dpSet();
+const DeviceTypeSet& getAudioDeviceOutAllScoSet();
+const DeviceTypeSet& getAudioDeviceOutAllUsbSet();
+const DeviceTypeSet& getAudioDeviceInAllSet();
+const DeviceTypeSet& getAudioDeviceInAllUsbSet();
+
+template<typename T>
+static std::vector<T> Intersection(const std::set<T>& a, const std::set<T>& b) {
+ std::vector<T> intersection;
+ std::set_intersection(a.begin(), a.end(),
+ b.begin(), b.end(),
+ std::back_inserter(intersection));
+ return intersection;
+}
+
+static inline ChannelMaskSet asInMask(const ChannelMaskSet& channelMasks) {
+ ChannelMaskSet inMaskSet;
+ for (const auto &channel : channelMasks) {
+ if (audio_channel_mask_out_to_in(channel) != AUDIO_CHANNEL_INVALID) {
+ inMaskSet.insert(audio_channel_mask_out_to_in(channel));
+ }
+ }
+ return inMaskSet;
+}
+
+static inline ChannelMaskSet asOutMask(const ChannelMaskSet& channelMasks) {
+ ChannelMaskSet outMaskSet;
+ for (const auto &channel : channelMasks) {
+ if (audio_channel_mask_in_to_out(channel) != AUDIO_CHANNEL_INVALID) {
+ outMaskSet.insert(audio_channel_mask_in_to_out(channel));
+ }
+ }
+ return outMaskSet;
+}
+
+static inline bool isSingleDeviceType(const DeviceTypeSet& deviceTypes,
+ audio_devices_t deviceType) {
+ return deviceTypes.size() == 1 && *(deviceTypes.begin()) == deviceType;
+}
+
+typedef bool (*DeviceTypeUnaryPredicate)(audio_devices_t);
+static inline bool isSingleDeviceType(const DeviceTypeSet& deviceTypes,
+ DeviceTypeUnaryPredicate p) {
+ return deviceTypes.size() == 1 && p(*(deviceTypes.begin()));
+}
+
+static inline bool areAllOfSameDeviceType(const DeviceTypeSet& deviceTypes,
+ std::function<bool(audio_devices_t)> p) {
+ return std::all_of(deviceTypes.begin(), deviceTypes.end(), p);
+}
+
+static inline void resetDeviceTypes(DeviceTypeSet& deviceTypes, audio_devices_t typeToAdd) {
+ deviceTypes.clear();
+ deviceTypes.insert(typeToAdd);
+}
+
+// FIXME: This is temporary helper function. Remove this when getting rid of all
+// bit mask usages of audio device types.
+static inline audio_devices_t deviceTypesToBitMask(const DeviceTypeSet& deviceTypes) {
+ audio_devices_t types = AUDIO_DEVICE_NONE;
+ for (auto deviceType : deviceTypes) {
+ types |= deviceType;
+ }
+ return types;
+}
+
+// FIXME: This is temporary helper function. Remove this when getting rid of all
+// bit mask usages of audio device types.
+static inline DeviceTypeSet deviceTypesFromBitMask(audio_devices_t types) {
+ DeviceTypeSet deviceTypes;
+ if ((types & AUDIO_DEVICE_BIT_IN) == 0) {
+ for (auto deviceType : AUDIO_DEVICE_OUT_ALL_ARRAY) {
+ if ((types & deviceType) == deviceType) {
+ deviceTypes.insert(deviceType);
+ }
+ }
+ } else {
+ for (auto deviceType : AUDIO_DEVICE_IN_ALL_ARRAY) {
+ if ((types & deviceType) == deviceType) {
+ deviceTypes.insert(deviceType);
+ }
+ }
+ }
+ return deviceTypes;
+}
+
+bool deviceTypesToString(const DeviceTypeSet& deviceTypes, std::string &str);
+
+std::string dumpDeviceTypes(const DeviceTypeSet& deviceTypes);
+
+/**
+ * Return human readable string for device types.
+ */
+std::string toString(const DeviceTypeSet& deviceTypes);
+
+
+} // namespace android
\ No newline at end of file
diff --git a/media/libaudiofoundation/include/media/AudioDeviceTypeAddr.h b/media/libaudiofoundation/include/media/AudioDeviceTypeAddr.h
new file mode 100644
index 0000000..60ea78e
--- /dev/null
+++ b/media/libaudiofoundation/include/media/AudioDeviceTypeAddr.h
@@ -0,0 +1,61 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#pragma once
+
+#include <string>
+#include <vector>
+
+#include <binder/Parcelable.h>
+#include <binder/Parcel.h>
+#include <media/AudioContainers.h>
+#include <system/audio.h>
+#include <utils/Errors.h>
+
+namespace android {
+
+struct AudioDeviceTypeAddr : public Parcelable {
+ AudioDeviceTypeAddr() = default;
+
+ AudioDeviceTypeAddr(audio_devices_t type, const std::string& address) :
+ mType(type), mAddress(address) {}
+
+ const char* getAddress() const;
+
+ bool equals(const AudioDeviceTypeAddr& other) const;
+
+ AudioDeviceTypeAddr& operator= (const AudioDeviceTypeAddr&) = default;
+
+ bool operator<(const AudioDeviceTypeAddr& other) const;
+
+ void reset();
+
+ status_t readFromParcel(const Parcel *parcel) override;
+
+ status_t writeToParcel(Parcel *parcel) const override;
+
+ audio_devices_t mType = AUDIO_DEVICE_NONE;
+ std::string mAddress;
+};
+
+using AudioDeviceTypeAddrVector = std::vector<AudioDeviceTypeAddr>;
+
+/**
+ * Return a collection of audio device types from a collection of AudioDeviceTypeAddr
+ */
+DeviceTypeSet getAudioDeviceTypes(const AudioDeviceTypeAddrVector& deviceTypeAddrs);
+
+}
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioGain.h b/media/libaudiofoundation/include/media/AudioGain.h
similarity index 82%
rename from services/audiopolicy/common/managerdefinitions/include/AudioGain.h
rename to media/libaudiofoundation/include/media/AudioGain.h
index 4af93e1..859f1e7 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioGain.h
+++ b/media/libaudiofoundation/include/media/AudioGain.h
@@ -16,15 +16,17 @@
#pragma once
+#include <binder/Parcel.h>
+#include <binder/Parcelable.h>
#include <utils/Errors.h>
#include <utils/RefBase.h>
-#include <utils/String8.h>
#include <system/audio.h>
+#include <string>
#include <vector>
namespace android {
-class AudioGain: public RefBase
+class AudioGain: public RefBase, public Parcelable
{
public:
AudioGain(int index, bool useInChannelMask);
@@ -55,7 +57,7 @@
int getMaxRampInMs() const { return mGain.max_ramp_ms; }
// TODO: remove dump from here (split serialization)
- void dump(String8 *dst, int spaces, int index) const;
+ void dump(std::string *dst, int spaces, int index) const;
void getDefaultConfig(struct audio_gain_config *config);
status_t checkConfig(const struct audio_gain_config *config);
@@ -65,6 +67,11 @@
const struct audio_gain &getGain() const { return mGain; }
+ bool equals(const sp<AudioGain>& other) const;
+
+ status_t writeToParcel(Parcel* parcel) const override;
+ status_t readFromParcel(const Parcel* parcel) override;
+
private:
int mIndex;
struct audio_gain mGain;
@@ -72,7 +79,7 @@
bool mUseForVolume = false;
};
-class AudioGains : public std::vector<sp<AudioGain> >
+class AudioGains : public std::vector<sp<AudioGain> >, public Parcelable
{
public:
bool canUseForVolume() const
@@ -90,6 +97,11 @@
push_back(gain);
return 0;
}
+
+ bool equals(const AudioGains& other) const;
+
+ status_t writeToParcel(Parcel* parcel) const override;
+ status_t readFromParcel(const Parcel* parcel) override;
};
} // namespace android
diff --git a/media/libaudiofoundation/include/media/AudioPort.h b/media/libaudiofoundation/include/media/AudioPort.h
new file mode 100644
index 0000000..3c013cb
--- /dev/null
+++ b/media/libaudiofoundation/include/media/AudioPort.h
@@ -0,0 +1,131 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#pragma once
+
+#include <string>
+
+#include <binder/Parcel.h>
+#include <binder/Parcelable.h>
+#include <media/AudioGain.h>
+#include <media/AudioProfile.h>
+#include <utils/Errors.h>
+#include <utils/RefBase.h>
+#include <system/audio.h>
+#include <cutils/config_utils.h>
+
+namespace android {
+
+class AudioPort : public virtual RefBase, public virtual Parcelable
+{
+public:
+ AudioPort(const std::string& name, audio_port_type_t type, audio_port_role_t role) :
+ mName(name), mType(type), mRole(role) {}
+
+ virtual ~AudioPort() = default;
+
+ void setName(const std::string &name) { mName = name; }
+ const std::string &getName() const { return mName; }
+
+ audio_port_type_t getType() const { return mType; }
+ audio_port_role_t getRole() const { return mRole; }
+
+ void setGains(const AudioGains &gains) { mGains = gains; }
+ const AudioGains &getGains() const { return mGains; }
+
+ virtual void toAudioPort(struct audio_port *port) const;
+
+ virtual void addAudioProfile(const sp<AudioProfile> &profile) {
+ mProfiles.add(profile);
+ }
+ virtual void clearAudioProfiles() {
+ mProfiles.clearProfiles();
+ }
+
+ bool hasValidAudioProfile() const { return mProfiles.hasValidProfile(); }
+
+ bool hasDynamicAudioProfile() const { return mProfiles.hasDynamicProfile(); }
+
+ void setAudioProfiles(const AudioProfileVector &profiles) { mProfiles = profiles; }
+ AudioProfileVector &getAudioProfiles() { return mProfiles; }
+
+ virtual void importAudioPort(const sp<AudioPort>& port, bool force = false);
+
+ status_t checkGain(const struct audio_gain_config *gainConfig, int index) const {
+ if (index < 0 || (size_t)index >= mGains.size()) {
+ return BAD_VALUE;
+ }
+ return mGains[index]->checkConfig(gainConfig);
+ }
+
+ bool useInputChannelMask() const
+ {
+ return ((mType == AUDIO_PORT_TYPE_DEVICE) && (mRole == AUDIO_PORT_ROLE_SOURCE)) ||
+ ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SINK));
+ }
+
+ void dump(std::string *dst, int spaces, bool verbose = true) const;
+
+ void log(const char* indent) const;
+
+ bool equals(const sp<AudioPort>& other) const;
+
+ status_t writeToParcel(Parcel* parcel) const override;
+ status_t readFromParcel(const Parcel* parcel) override;
+
+ AudioGains mGains; // gain controllers
+protected:
+ std::string mName;
+ audio_port_type_t mType;
+ audio_port_role_t mRole;
+ AudioProfileVector mProfiles; // AudioProfiles supported by this port (format, Rates, Channels)
+};
+
+
+class AudioPortConfig : public virtual RefBase, public virtual Parcelable
+{
+public:
+ virtual ~AudioPortConfig() = default;
+
+ virtual sp<AudioPort> getAudioPort() const = 0;
+
+ virtual status_t applyAudioPortConfig(const struct audio_port_config *config,
+ struct audio_port_config *backupConfig = NULL);
+
+ virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig = NULL) const;
+
+ unsigned int getSamplingRate() const { return mSamplingRate; }
+ audio_format_t getFormat() const { return mFormat; }
+ audio_channel_mask_t getChannelMask() const { return mChannelMask; }
+ audio_port_handle_t getId() const { return mId; }
+
+ bool hasGainController(bool canUseForVolume = false) const;
+
+ bool equals(const sp<AudioPortConfig>& other) const;
+
+ status_t writeToParcel(Parcel* parcel) const override;
+ status_t readFromParcel(const Parcel* parcel) override;
+
+protected:
+ unsigned int mSamplingRate = 0u;
+ audio_format_t mFormat = AUDIO_FORMAT_INVALID;
+ audio_channel_mask_t mChannelMask = AUDIO_CHANNEL_NONE;
+ audio_port_handle_t mId = AUDIO_PORT_HANDLE_NONE;
+ struct audio_gain_config mGain = { .index = -1 };
+};
+
+} // namespace android
diff --git a/media/libaudiofoundation/include/media/AudioProfile.h b/media/libaudiofoundation/include/media/AudioProfile.h
new file mode 100644
index 0000000..730138a
--- /dev/null
+++ b/media/libaudiofoundation/include/media/AudioProfile.h
@@ -0,0 +1,118 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#pragma once
+
+#include <string>
+#include <vector>
+
+#include <binder/Parcel.h>
+#include <binder/Parcelable.h>
+#include <media/AudioContainers.h>
+#include <system/audio.h>
+#include <utils/RefBase.h>
+
+namespace android {
+
+class AudioProfile final : public RefBase, public Parcelable
+{
+public:
+ static sp<AudioProfile> createFullDynamic(audio_format_t dynamicFormat = AUDIO_FORMAT_DEFAULT);
+
+ AudioProfile(audio_format_t format, audio_channel_mask_t channelMasks, uint32_t samplingRate);
+ AudioProfile(audio_format_t format,
+ const ChannelMaskSet &channelMasks,
+ const SampleRateSet &samplingRateCollection);
+
+ audio_format_t getFormat() const { return mFormat; }
+ const ChannelMaskSet &getChannels() const { return mChannelMasks; }
+ const SampleRateSet &getSampleRates() const { return mSamplingRates; }
+ void setChannels(const ChannelMaskSet &channelMasks);
+ void setSampleRates(const SampleRateSet &sampleRates);
+
+ void clear();
+ bool isValid() const { return hasValidFormat() && hasValidRates() && hasValidChannels(); }
+ bool supportsChannels(audio_channel_mask_t channels) const
+ {
+ return mChannelMasks.count(channels) != 0;
+ }
+ bool supportsRate(uint32_t rate) const { return mSamplingRates.count(rate) != 0; }
+
+ bool hasValidFormat() const { return mFormat != AUDIO_FORMAT_DEFAULT; }
+ bool hasValidRates() const { return !mSamplingRates.empty(); }
+ bool hasValidChannels() const { return !mChannelMasks.empty(); }
+
+ void setDynamicChannels(bool dynamic) { mIsDynamicChannels = dynamic; }
+ bool isDynamicChannels() const { return mIsDynamicChannels; }
+
+ void setDynamicRate(bool dynamic) { mIsDynamicRate = dynamic; }
+ bool isDynamicRate() const { return mIsDynamicRate; }
+
+ void setDynamicFormat(bool dynamic) { mIsDynamicFormat = dynamic; }
+ bool isDynamicFormat() const { return mIsDynamicFormat; }
+
+ bool isDynamic() { return mIsDynamicFormat || mIsDynamicChannels || mIsDynamicRate; }
+
+ void dump(std::string *dst, int spaces) const;
+
+ bool equals(const sp<AudioProfile>& other) const;
+
+ status_t writeToParcel(Parcel* parcel) const override;
+ status_t readFromParcel(const Parcel* parcel) override;
+
+private:
+ std::string mName;
+ audio_format_t mFormat; // The format for an audio profile should only be set when initialized.
+ ChannelMaskSet mChannelMasks;
+ SampleRateSet mSamplingRates;
+
+ bool mIsDynamicFormat = false;
+ bool mIsDynamicChannels = false;
+ bool mIsDynamicRate = false;
+};
+
+class AudioProfileVector : public std::vector<sp<AudioProfile>>, public Parcelable
+{
+public:
+ virtual ~AudioProfileVector() = default;
+
+ virtual ssize_t add(const sp<AudioProfile> &profile);
+
+ // If the profile is dynamic format and has valid format, it will be removed when doing
+ // clearProfiles(). Otherwise, AudioProfile::clear() will be called.
+ virtual void clearProfiles();
+
+ sp<AudioProfile> getFirstValidProfile() const;
+ sp<AudioProfile> getFirstValidProfileFor(audio_format_t format) const;
+ bool hasValidProfile() const { return getFirstValidProfile() != 0; }
+
+ FormatVector getSupportedFormats() const;
+ bool hasDynamicChannelsFor(audio_format_t format) const;
+ bool hasDynamicFormat() const;
+ bool hasDynamicProfile() const;
+ bool hasDynamicRateFor(audio_format_t format) const;
+
+ virtual void dump(std::string *dst, int spaces) const;
+
+ bool equals(const AudioProfileVector& other) const;
+
+ status_t writeToParcel(Parcel* parcel) const override;
+ status_t readFromParcel(const Parcel* parcel) override;
+};
+
+bool operator == (const AudioProfile &left, const AudioProfile &right);
+
+} // namespace android
diff --git a/media/libaudiofoundation/include/media/DeviceDescriptorBase.h b/media/libaudiofoundation/include/media/DeviceDescriptorBase.h
new file mode 100644
index 0000000..4c03667
--- /dev/null
+++ b/media/libaudiofoundation/include/media/DeviceDescriptorBase.h
@@ -0,0 +1,85 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#pragma once
+
+#include <vector>
+
+#include <binder/Parcel.h>
+#include <binder/Parcelable.h>
+#include <media/AudioContainers.h>
+#include <media/AudioPort.h>
+#include <media/AudioDeviceTypeAddr.h>
+#include <utils/Errors.h>
+#include <cutils/config_utils.h>
+#include <system/audio.h>
+#include <system/audio_policy.h>
+
+namespace android {
+
+class DeviceDescriptorBase : public AudioPort, public AudioPortConfig
+{
+public:
+ // Note that empty name refers by convention to a generic device.
+ explicit DeviceDescriptorBase(audio_devices_t type);
+ DeviceDescriptorBase(audio_devices_t type, const std::string& address);
+ explicit DeviceDescriptorBase(const AudioDeviceTypeAddr& deviceTypeAddr);
+
+ virtual ~DeviceDescriptorBase() {}
+
+ audio_devices_t type() const { return mDeviceTypeAddr.mType; }
+ std::string address() const { return mDeviceTypeAddr.mAddress; }
+ void setAddress(const std::string &address) { mDeviceTypeAddr.mAddress = address; }
+ const AudioDeviceTypeAddr& getDeviceTypeAddr() const { return mDeviceTypeAddr; }
+
+ // AudioPortConfig
+ virtual sp<AudioPort> getAudioPort() const {
+ return static_cast<AudioPort*>(const_cast<DeviceDescriptorBase*>(this));
+ }
+ virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig = NULL) const;
+
+ // AudioPort
+ virtual void toAudioPort(struct audio_port *port) const;
+
+ void dump(std::string *dst, int spaces, int index,
+ const char* extraInfo = nullptr, bool verbose = true) const;
+ void log() const;
+ std::string toString() const;
+
+ bool equals(const sp<DeviceDescriptorBase>& other) const;
+
+ status_t writeToParcel(Parcel* parcel) const override;
+ status_t readFromParcel(const Parcel* parcel) override;
+
+protected:
+ AudioDeviceTypeAddr mDeviceTypeAddr;
+};
+
+using DeviceDescriptorBaseVector = std::vector<sp<DeviceDescriptorBase>>;
+
+/**
+ * Return human readable string for collection of DeviceDescriptorBase.
+ * For a DeviceDescriptorBase, it contains port id, audio device type and address.
+ */
+std::string toString(const DeviceDescriptorBaseVector& devices);
+
+/**
+ * Return a set of device types and addresses from collection of DeviceDescriptorBase.
+ */
+AudioDeviceTypeAddrVector deviceTypeAddrsFromDescriptors(const DeviceDescriptorBaseVector& devices);
+
+} // namespace android
diff --git a/media/libaudiofoundation/tests/Android.bp b/media/libaudiofoundation/tests/Android.bp
new file mode 100644
index 0000000..f258b14
--- /dev/null
+++ b/media/libaudiofoundation/tests/Android.bp
@@ -0,0 +1,25 @@
+cc_test {
+ name: "audiofoundation_parcelable_test",
+
+ shared_libs: [
+ "libaudiofoundation",
+ "libbinder",
+ "liblog",
+ "libutils",
+ ],
+
+ header_libs: [
+ "libaudio_system_headers",
+ ],
+
+ srcs: [
+ "audiofoundation_parcelable_test.cpp",
+ ],
+
+ cflags: [
+ "-Werror",
+ "-Wall",
+ ],
+
+ test_suites: ["device-tests"],
+}
diff --git a/media/libaudiofoundation/tests/audiofoundation_parcelable_test.cpp b/media/libaudiofoundation/tests/audiofoundation_parcelable_test.cpp
new file mode 100644
index 0000000..5baa072
--- /dev/null
+++ b/media/libaudiofoundation/tests/audiofoundation_parcelable_test.cpp
@@ -0,0 +1,142 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "audiofoundation_parcelable_test"
+
+#include <gtest/gtest.h>
+
+#include <binder/IServiceManager.h>
+#include <binder/Parcelable.h>
+#include <binder/ProcessState.h>
+#include <media/AudioGain.h>
+#include <media/AudioPort.h>
+#include <media/AudioProfile.h>
+#include <media/DeviceDescriptorBase.h>
+#include <utils/Log.h>
+#include <utils/String16.h>
+
+namespace android {
+
+static const audio_port_config TEST_AUDIO_PORT_CONFIG = {
+ .id = 0,
+ .role = AUDIO_PORT_ROLE_SINK,
+ .type = AUDIO_PORT_TYPE_DEVICE,
+ .config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE | AUDIO_PORT_CONFIG_CHANNEL_MASK |
+ AUDIO_PORT_CONFIG_FORMAT | AUDIO_PORT_CONFIG_GAIN,
+ .sample_rate = 48000,
+ .channel_mask = AUDIO_CHANNEL_OUT_STEREO,
+ .format = AUDIO_FORMAT_PCM_16_BIT,
+ .gain = {
+ .index = 0,
+ .mode = AUDIO_GAIN_MODE_JOINT,
+ .channel_mask = AUDIO_CHANNEL_OUT_STEREO,
+ }
+};
+
+class AudioPortConfigTestStub : public AudioPortConfig {
+public:
+ sp<AudioPort> getAudioPort() const override { return nullptr; }
+};
+
+AudioGains getAudioGainsForTest() {
+ AudioGains audioGains;
+ sp<AudioGain> audioGain = new AudioGain(0 /*index*/, false /*useInChannelMask*/);
+ audioGain->setMode(AUDIO_GAIN_MODE_JOINT);
+ audioGain->setChannelMask(AUDIO_CHANNEL_OUT_STEREO);
+ audioGain->setMinValueInMb(-3200);
+ audioGain->setMaxValueInMb(600);
+ audioGain->setDefaultValueInMb(0);
+ audioGain->setStepValueInMb(100);
+ audioGain->setMinRampInMs(100);
+ audioGain->setMaxRampInMs(500);
+ audioGains.push_back(audioGain);
+ return audioGains;
+}
+
+AudioProfileVector getAudioProfileVectorForTest() {
+ AudioProfileVector audioProfiles;
+ sp<AudioProfile> audioProfile = AudioProfile::createFullDynamic();
+ audioProfile->setChannels({AUDIO_CHANNEL_OUT_MONO, AUDIO_CHANNEL_OUT_STEREO});
+ audioProfile->setSampleRates({48000});
+ audioProfiles.add(audioProfile);
+ return audioProfiles;
+}
+
+TEST(AudioFoundationParcelableTest, ParcelingAudioGain) {
+ Parcel data;
+ AudioGains audioGains = getAudioGainsForTest();
+
+ ASSERT_EQ(data.writeParcelable(audioGains), NO_ERROR);
+ data.setDataPosition(0);
+ AudioGains audioGainsFromParcel;
+ ASSERT_EQ(data.readParcelable(&audioGainsFromParcel), NO_ERROR);
+ ASSERT_TRUE(audioGainsFromParcel.equals(audioGains));
+}
+
+TEST(AudioFoundationParcelableTest, ParcelingAudioProfileVector) {
+ Parcel data;
+ AudioProfileVector audioProfiles = getAudioProfileVectorForTest();
+
+ ASSERT_EQ(data.writeParcelable(audioProfiles), NO_ERROR);
+ data.setDataPosition(0);
+ AudioProfileVector audioProfilesFromParcel;
+ ASSERT_EQ(data.readParcelable(&audioProfilesFromParcel), NO_ERROR);
+ ASSERT_TRUE(audioProfilesFromParcel.equals(audioProfiles));
+}
+
+TEST(AudioFoundationParcelableTest, ParcelingAudioPort) {
+ Parcel data;
+ sp<AudioPort> audioPort = new AudioPort(
+ "AudioPortName", AUDIO_PORT_TYPE_DEVICE, AUDIO_PORT_ROLE_SINK);
+ audioPort->setGains(getAudioGainsForTest());
+ audioPort->setAudioProfiles(getAudioProfileVectorForTest());
+
+ ASSERT_EQ(data.writeParcelable(*audioPort), NO_ERROR);
+ data.setDataPosition(0);
+ sp<AudioPort> audioPortFromParcel = new AudioPort(
+ "", AUDIO_PORT_TYPE_NONE, AUDIO_PORT_ROLE_NONE);
+ ASSERT_EQ(data.readParcelable(audioPortFromParcel.get()), NO_ERROR);
+ ASSERT_TRUE(audioPortFromParcel->equals(audioPort));
+}
+
+TEST(AudioFoundationParcelableTest, ParcelingAudioPortConfig) {
+ Parcel data;
+ sp<AudioPortConfig> audioPortConfig = new AudioPortConfigTestStub();
+ audioPortConfig->applyAudioPortConfig(&TEST_AUDIO_PORT_CONFIG);
+
+ ASSERT_EQ(data.writeParcelable(*audioPortConfig), NO_ERROR);
+ data.setDataPosition(0);
+ sp<AudioPortConfig> audioPortConfigFromParcel = new AudioPortConfigTestStub();
+ ASSERT_EQ(data.readParcelable(audioPortConfigFromParcel.get()), NO_ERROR);
+ ASSERT_TRUE(audioPortConfigFromParcel->equals(audioPortConfig));
+}
+
+TEST(AudioFoundationParcelableTest, ParcelingDeviceDescriptorBase) {
+ Parcel data;
+ sp<DeviceDescriptorBase> desc = new DeviceDescriptorBase(AUDIO_DEVICE_OUT_SPEAKER);
+ desc->setGains(getAudioGainsForTest());
+ desc->setAudioProfiles(getAudioProfileVectorForTest());
+ desc->applyAudioPortConfig(&TEST_AUDIO_PORT_CONFIG);
+ desc->setAddress("DeviceDescriptorBaseTestAddress");
+
+ ASSERT_EQ(data.writeParcelable(*desc), NO_ERROR);
+ data.setDataPosition(0);
+ sp<DeviceDescriptorBase> descFromParcel = new DeviceDescriptorBase(AUDIO_DEVICE_NONE);
+ ASSERT_EQ(data.readParcelable(descFromParcel.get()), NO_ERROR);
+ ASSERT_TRUE(descFromParcel->equals(desc));
+}
+
+} // namespace android
diff --git a/media/libaudiohal/Android.bp b/media/libaudiohal/Android.bp
index 584c2c0..1709d1e 100644
--- a/media/libaudiohal/Android.bp
+++ b/media/libaudiohal/Android.bp
@@ -4,6 +4,7 @@
srcs: [
"DevicesFactoryHalInterface.cpp",
"EffectsFactoryHalInterface.cpp",
+ "FactoryHalHidl.cpp",
],
cflags: [
@@ -12,21 +13,23 @@
"-Werror",
],
- shared_libs: [
- "android.hardware.audio.effect@2.0",
- "android.hardware.audio.effect@4.0",
- "android.hardware.audio.effect@5.0",
- "android.hardware.audio@2.0",
- "android.hardware.audio@4.0",
- "android.hardware.audio@5.0",
+ required: [
"libaudiohal@2.0",
"libaudiohal@4.0",
"libaudiohal@5.0",
+ "libaudiohal@6.0",
+ ],
+
+ shared_libs: [
+ "libdl",
+ "libhidlbase",
+ "liblog",
"libutils",
],
header_libs: [
- "libaudiohal_headers"
+ "libaudiohal_headers",
+ "libbase_headers",
]
}
@@ -57,4 +60,10 @@
name: "libaudiohal_headers",
export_include_dirs: ["include"],
+
+ // This is needed because the stream interface includes media/MicrophoneInfo.h
+ // which is not in any library but has a dependency on headers from libbinder.
+ header_libs: ["libbinder_headers"],
+
+ export_header_lib_headers: ["libbinder_headers"],
}
diff --git a/media/libaudiohal/DevicesFactoryHalInterface.cpp b/media/libaudiohal/DevicesFactoryHalInterface.cpp
index f86009c..325a547 100644
--- a/media/libaudiohal/DevicesFactoryHalInterface.cpp
+++ b/media/libaudiohal/DevicesFactoryHalInterface.cpp
@@ -14,26 +14,15 @@
* limitations under the License.
*/
-#include <android/hardware/audio/2.0/IDevicesFactory.h>
-#include <android/hardware/audio/4.0/IDevicesFactory.h>
-#include <android/hardware/audio/5.0/IDevicesFactory.h>
-
-#include <libaudiohal/FactoryHalHidl.h>
+#include <media/audiohal/DevicesFactoryHalInterface.h>
+#include <media/audiohal/FactoryHalHidl.h>
namespace android {
// static
sp<DevicesFactoryHalInterface> DevicesFactoryHalInterface::create() {
- if (hardware::audio::V5_0::IDevicesFactory::getService() != nullptr) {
- return V5_0::createDevicesFactoryHal();
- }
- if (hardware::audio::V4_0::IDevicesFactory::getService() != nullptr) {
- return V4_0::createDevicesFactoryHal();
- }
- if (hardware::audio::V2_0::IDevicesFactory::getService() != nullptr) {
- return V2_0::createDevicesFactoryHal();
- }
- return nullptr;
+ return createPreferredImpl<DevicesFactoryHalInterface>(
+ "android.hardware.audio", "IDevicesFactory");
}
} // namespace android
diff --git a/media/libaudiohal/EffectsFactoryHalInterface.cpp b/media/libaudiohal/EffectsFactoryHalInterface.cpp
index bd3ef61..bc3b4c1 100644
--- a/media/libaudiohal/EffectsFactoryHalInterface.cpp
+++ b/media/libaudiohal/EffectsFactoryHalInterface.cpp
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2016 The Android Open Source Project
+ * Copyright (C) 2017 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
@@ -14,26 +14,15 @@
* limitations under the License.
*/
-#include <android/hardware/audio/effect/2.0/IEffectsFactory.h>
-#include <android/hardware/audio/effect/4.0/IEffectsFactory.h>
-#include <android/hardware/audio/effect/5.0/IEffectsFactory.h>
-
-#include <libaudiohal/FactoryHalHidl.h>
+#include <media/audiohal/EffectsFactoryHalInterface.h>
+#include <media/audiohal/FactoryHalHidl.h>
namespace android {
// static
sp<EffectsFactoryHalInterface> EffectsFactoryHalInterface::create() {
- if (hardware::audio::effect::V5_0::IEffectsFactory::getService() != nullptr) {
- return effect::V5_0::createEffectsFactoryHal();
- }
- if (hardware::audio::effect::V4_0::IEffectsFactory::getService() != nullptr) {
- return effect::V4_0::createEffectsFactoryHal();
- }
- if (hardware::audio::effect::V2_0::IEffectsFactory::getService() != nullptr) {
- return effect::V2_0::createEffectsFactoryHal();
- }
- return nullptr;
+ return createPreferredImpl<EffectsFactoryHalInterface>(
+ "android.hardware.audio.effect", "IEffectsFactory");
}
// static
diff --git a/media/libaudiohal/FactoryHalHidl.cpp b/media/libaudiohal/FactoryHalHidl.cpp
new file mode 100644
index 0000000..5985ef0
--- /dev/null
+++ b/media/libaudiohal/FactoryHalHidl.cpp
@@ -0,0 +1,107 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "FactoryHalHidl"
+
+#include <media/audiohal/FactoryHalHidl.h>
+
+#include <dlfcn.h>
+
+#include <android/hidl/manager/1.0/IServiceManager.h>
+#include <hidl/ServiceManagement.h>
+#include <hidl/Status.h>
+#include <utils/Log.h>
+
+namespace android::detail {
+
+namespace {
+/** Supported HAL versions, in order of preference.
+ */
+const char* sAudioHALVersions[] = {
+ "6.0",
+ "5.0",
+ "4.0",
+ "2.0",
+ nullptr
+};
+
+bool createHalService(const std::string& version, const std::string& interface,
+ void** rawInterface) {
+ const std::string libName = "libaudiohal@" + version + ".so";
+ const std::string factoryFunctionName = "create" + interface;
+ constexpr int dlMode = RTLD_LAZY;
+ void* handle = nullptr;
+ dlerror(); // clear
+ handle = dlopen(libName.c_str(), dlMode);
+ if (handle == nullptr) {
+ const char* error = dlerror();
+ ALOGE("Failed to dlopen %s: %s", libName.c_str(),
+ error != nullptr ? error : "unknown error");
+ return false;
+ }
+ void* (*factoryFunction)();
+ *(void **)(&factoryFunction) = dlsym(handle, factoryFunctionName.c_str());
+ if (!factoryFunction) {
+ const char* error = dlerror();
+ ALOGE("Factory function %s not found in library %s: %s",
+ factoryFunctionName.c_str(), libName.c_str(),
+ error != nullptr ? error : "unknown error");
+ dlclose(handle);
+ return false;
+ }
+ *rawInterface = (*factoryFunction)();
+ ALOGW_IF(!*rawInterface, "Factory function %s from %s returned nullptr",
+ factoryFunctionName.c_str(), libName.c_str());
+ return true;
+}
+
+bool hasHalService(const std::string& package, const std::string& version,
+ const std::string& interface) {
+ using ::android::hidl::manager::V1_0::IServiceManager;
+ sp<IServiceManager> sm = ::android::hardware::defaultServiceManager();
+ if (!sm) {
+ ALOGE("Failed to obtain HIDL ServiceManager");
+ return false;
+ }
+ // Since audio HAL doesn't support multiple clients, avoid instantiating
+ // the interface right away. Instead, query the transport type for it.
+ using ::android::hardware::Return;
+ using Transport = IServiceManager::Transport;
+ const std::string fqName = package + "@" + version + "::" + interface;
+ const std::string instance = "default";
+ Return<Transport> transport = sm->getTransport(fqName, instance);
+ if (!transport.isOk()) {
+ ALOGE("Failed to obtain transport type for %s/%s: %s",
+ fqName.c_str(), instance.c_str(), transport.description().c_str());
+ return false;
+ }
+ return transport != Transport::EMPTY;
+}
+
+} // namespace
+
+void* createPreferredImpl(const std::string& package, const std::string& interface) {
+ for (auto version = detail::sAudioHALVersions; version != nullptr; ++version) {
+ void* rawInterface = nullptr;
+ if (hasHalService(package, *version, interface)
+ && createHalService(*version, interface, &rawInterface)) {
+ return rawInterface;
+ }
+ }
+ return nullptr;
+}
+
+} // namespace android::detail
diff --git a/media/libaudiohal/impl/Android.bp b/media/libaudiohal/impl/Android.bp
index 88533da..967fba1 100644
--- a/media/libaudiohal/impl/Android.bp
+++ b/media/libaudiohal/impl/Android.bp
@@ -16,17 +16,17 @@
"StreamHalHidl.cpp",
],
- export_include_dirs: ["include"],
-
cflags: [
"-Wall",
"-Wextra",
"-Werror",
+ "-fvisibility=hidden",
],
shared_libs: [
"android.hardware.audio.common-util",
"android.hidl.allocator@1.0",
"android.hidl.memory@1.0",
+ "libaudiofoundation",
"libaudiohal_deathhandler",
"libaudioutils",
"libbase",
@@ -36,8 +36,6 @@
"libhardware",
"libhidlbase",
"libhidlmemory",
- "libhidltransport",
- "libhwbinder",
"liblog",
"libmedia_helper",
"libmediautils",
@@ -45,6 +43,7 @@
],
header_libs: [
"android.hardware.audio.common.util@all-versions",
+ "libaudioclient_headers",
"libaudiohal_headers"
],
@@ -100,3 +99,20 @@
"-include common/all-versions/VersionMacro.h",
]
}
+
+cc_library_shared {
+ name: "libaudiohal@6.0",
+ defaults: ["libaudiohal_default"],
+ shared_libs: [
+ "android.hardware.audio.common@6.0",
+ "android.hardware.audio.common@6.0-util",
+ "android.hardware.audio.effect@6.0",
+ "android.hardware.audio@6.0",
+ ],
+ cflags: [
+ "-DMAJOR_VERSION=6",
+ "-DMINOR_VERSION=0",
+ "-include common/all-versions/VersionMacro.h",
+ ]
+}
+
diff --git a/media/libaudiohal/impl/ConversionHelperHidl.cpp b/media/libaudiohal/impl/ConversionHelperHidl.cpp
index 9f8a520..f29b0f3 100644
--- a/media/libaudiohal/impl/ConversionHelperHidl.cpp
+++ b/media/libaudiohal/impl/ConversionHelperHidl.cpp
@@ -17,6 +17,7 @@
#include <string.h>
#define LOG_TAG "HalHidl"
+#include <media/AudioContainers.h>
#include <media/AudioParameter.h>
#include <utils/Log.h>
@@ -109,26 +110,22 @@
char halAddress[AUDIO_DEVICE_MAX_ADDRESS_LEN];
memset(halAddress, 0, sizeof(halAddress));
audio_devices_t halDevice = static_cast<audio_devices_t>(address.device);
- const bool isInput = (halDevice & AUDIO_DEVICE_BIT_IN) != 0;
- if (isInput) halDevice &= ~AUDIO_DEVICE_BIT_IN;
- if ((!isInput && (halDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != 0) ||
- (isInput && (halDevice & AUDIO_DEVICE_IN_BLUETOOTH_A2DP) != 0)) {
+ if (getAudioDeviceOutAllA2dpSet().count(halDevice) > 0 ||
+ halDevice == AUDIO_DEVICE_IN_BLUETOOTH_A2DP) {
snprintf(halAddress, sizeof(halAddress), "%02X:%02X:%02X:%02X:%02X:%02X",
address.address.mac[0], address.address.mac[1], address.address.mac[2],
address.address.mac[3], address.address.mac[4], address.address.mac[5]);
- } else if ((!isInput && (halDevice & AUDIO_DEVICE_OUT_IP) != 0) ||
- (isInput && (halDevice & AUDIO_DEVICE_IN_IP) != 0)) {
+ } else if (halDevice == AUDIO_DEVICE_OUT_IP || halDevice == AUDIO_DEVICE_IN_IP) {
snprintf(halAddress, sizeof(halAddress), "%d.%d.%d.%d", address.address.ipv4[0],
address.address.ipv4[1], address.address.ipv4[2], address.address.ipv4[3]);
- } else if ((!isInput && (halDevice & AUDIO_DEVICE_OUT_ALL_USB) != 0) ||
- (isInput && (halDevice & AUDIO_DEVICE_IN_ALL_USB) != 0)) {
+ } else if (getAudioDeviceOutAllUsbSet().count(halDevice) > 0 ||
+ getAudioDeviceInAllUsbSet().count(halDevice) > 0) {
snprintf(halAddress, sizeof(halAddress), "card=%d;device=%d", address.address.alsa.card,
address.address.alsa.device);
- } else if ((!isInput && (halDevice & AUDIO_DEVICE_OUT_BUS) != 0) ||
- (isInput && (halDevice & AUDIO_DEVICE_IN_BUS) != 0)) {
+ } else if (halDevice == AUDIO_DEVICE_OUT_BUS || halDevice == AUDIO_DEVICE_IN_BUS) {
snprintf(halAddress, sizeof(halAddress), "%s", address.busAddress.c_str());
- } else if ((!isInput && (halDevice & AUDIO_DEVICE_OUT_REMOTE_SUBMIX)) != 0 ||
- (isInput && (halDevice & AUDIO_DEVICE_IN_REMOTE_SUBMIX) != 0)) {
+ } else if (halDevice == AUDIO_DEVICE_OUT_REMOTE_SUBMIX ||
+ halDevice == AUDIO_DEVICE_IN_REMOTE_SUBMIX) {
snprintf(halAddress, sizeof(halAddress), "%s", address.rSubmixAddress.c_str());
} else {
snprintf(halAddress, sizeof(halAddress), "%s", address.busAddress.c_str());
diff --git a/media/libaudiohal/impl/DeviceHalHidl.cpp b/media/libaudiohal/impl/DeviceHalHidl.cpp
index b25f82e..f529cd1 100644
--- a/media/libaudiohal/impl/DeviceHalHidl.cpp
+++ b/media/libaudiohal/impl/DeviceHalHidl.cpp
@@ -22,11 +22,13 @@
#include PATH(android/hardware/audio/FILE_VERSION/IPrimaryDevice.h)
#include <cutils/native_handle.h>
#include <hwbinder/IPCThreadState.h>
+#include <media/AudioContainers.h>
#include <utils/Log.h>
#include <common/all-versions/VersionUtils.h>
#include "DeviceHalHidl.h"
+#include "EffectHalHidl.h"
#include "HidlUtils.h"
#include "StreamHalHidl.h"
#include "VersionUtils.h"
@@ -42,6 +44,8 @@
using namespace ::android::hardware::audio::common::CPP_VERSION;
using namespace ::android::hardware::audio::CPP_VERSION;
+using EffectHalHidl = ::android::effect::CPP_VERSION::EffectHalHidl;
+
namespace {
status_t deviceAddressFromHal(
@@ -51,42 +55,32 @@
if (halAddress == nullptr || strnlen(halAddress, AUDIO_DEVICE_MAX_ADDRESS_LEN) == 0) {
return OK;
}
- const bool isInput = (device & AUDIO_DEVICE_BIT_IN) != 0;
- if (isInput) device &= ~AUDIO_DEVICE_BIT_IN;
- if ((!isInput && (device & AUDIO_DEVICE_OUT_ALL_A2DP) != 0)
- || (isInput && (device & AUDIO_DEVICE_IN_BLUETOOTH_A2DP) != 0)) {
+ if (getAudioDeviceOutAllA2dpSet().count(device) > 0
+ || device == AUDIO_DEVICE_IN_BLUETOOTH_A2DP) {
int status = sscanf(halAddress,
"%hhX:%hhX:%hhX:%hhX:%hhX:%hhX",
&address->address.mac[0], &address->address.mac[1], &address->address.mac[2],
&address->address.mac[3], &address->address.mac[4], &address->address.mac[5]);
return status == 6 ? OK : BAD_VALUE;
- } else if ((!isInput && (device & AUDIO_DEVICE_OUT_IP) != 0)
- || (isInput && (device & AUDIO_DEVICE_IN_IP) != 0)) {
+ } else if (device == AUDIO_DEVICE_OUT_IP || device == AUDIO_DEVICE_IN_IP) {
int status = sscanf(halAddress,
"%hhu.%hhu.%hhu.%hhu",
&address->address.ipv4[0], &address->address.ipv4[1],
&address->address.ipv4[2], &address->address.ipv4[3]);
return status == 4 ? OK : BAD_VALUE;
- } else if ((!isInput && (device & AUDIO_DEVICE_OUT_ALL_USB)) != 0
- || (isInput && (device & AUDIO_DEVICE_IN_ALL_USB)) != 0) {
+ } else if (getAudioDeviceOutAllUsbSet().count(device) > 0
+ || getAudioDeviceInAllUsbSet().count(device) > 0) {
int status = sscanf(halAddress,
"card=%d;device=%d",
&address->address.alsa.card, &address->address.alsa.device);
return status == 2 ? OK : BAD_VALUE;
- } else if ((!isInput && (device & AUDIO_DEVICE_OUT_BUS) != 0)
- || (isInput && (device & AUDIO_DEVICE_IN_BUS) != 0)) {
- if (halAddress != NULL) {
- address->busAddress = halAddress;
- return OK;
- }
- return BAD_VALUE;
- } else if ((!isInput && (device & AUDIO_DEVICE_OUT_REMOTE_SUBMIX)) != 0
- || (isInput && (device & AUDIO_DEVICE_IN_REMOTE_SUBMIX) != 0)) {
- if (halAddress != NULL) {
- address->rSubmixAddress = halAddress;
- return OK;
- }
- return BAD_VALUE;
+ } else if (device == AUDIO_DEVICE_OUT_BUS || device == AUDIO_DEVICE_IN_BUS) {
+ address->busAddress = halAddress;
+ return OK;
+ } else if (device == AUDIO_DEVICE_OUT_REMOTE_SUBMIX
+ || device == AUDIO_DEVICE_IN_REMOTE_SUBMIX) {
+ address->rSubmixAddress = halAddress;
+ return OK;
}
return OK;
}
@@ -100,8 +94,12 @@
DeviceHalHidl::~DeviceHalHidl() {
if (mDevice != 0) {
+#if MAJOR_VERSION <= 5
mDevice.clear();
hardware::IPCThreadState::self()->flushCommands();
+#elif MAJOR_VERSION >= 6
+ mDevice->close();
+#endif
}
}
@@ -229,14 +227,14 @@
status_t DeviceHalHidl::openOutputStream(
audio_io_handle_t handle,
- audio_devices_t devices,
+ audio_devices_t deviceType,
audio_output_flags_t flags,
struct audio_config *config,
const char *address,
sp<StreamOutHalInterface> *outStream) {
if (mDevice == 0) return NO_INIT;
DeviceAddress hidlDevice;
- status_t status = deviceAddressFromHal(devices, address, &hidlDevice);
+ status_t status = deviceAddressFromHal(deviceType, address, &hidlDevice);
if (status != OK) return status;
AudioConfig hidlConfig;
HidlUtils::audioConfigFromHal(*config, &hidlConfig);
@@ -390,6 +388,36 @@
}
#endif
+#if MAJOR_VERSION >= 6
+status_t DeviceHalHidl::addDeviceEffect(
+ audio_port_handle_t device, sp<EffectHalInterface> effect) {
+ if (mDevice == 0) return NO_INIT;
+ return processReturn("addDeviceEffect", mDevice->addDeviceEffect(
+ static_cast<AudioPortHandle>(device),
+ static_cast<EffectHalHidl*>(effect.get())->effectId()));
+}
+#else
+status_t DeviceHalHidl::addDeviceEffect(
+ audio_port_handle_t device __unused, sp<EffectHalInterface> effect __unused) {
+ return INVALID_OPERATION;
+}
+#endif
+
+#if MAJOR_VERSION >= 6
+status_t DeviceHalHidl::removeDeviceEffect(
+ audio_port_handle_t device, sp<EffectHalInterface> effect) {
+ if (mDevice == 0) return NO_INIT;
+ return processReturn("removeDeviceEffect", mDevice->removeDeviceEffect(
+ static_cast<AudioPortHandle>(device),
+ static_cast<EffectHalHidl*>(effect.get())->effectId()));
+}
+#else
+status_t DeviceHalHidl::removeDeviceEffect(
+ audio_port_handle_t device __unused, sp<EffectHalInterface> effect __unused) {
+ return INVALID_OPERATION;
+}
+#endif
+
status_t DeviceHalHidl::dump(int fd) {
if (mDevice == 0) return NO_INIT;
native_handle_t* hidlHandle = native_handle_create(1, 0);
diff --git a/media/libaudiohal/impl/DeviceHalHidl.h b/media/libaudiohal/impl/DeviceHalHidl.h
index f7d465f..d342d4a 100644
--- a/media/libaudiohal/impl/DeviceHalHidl.h
+++ b/media/libaudiohal/impl/DeviceHalHidl.h
@@ -113,6 +113,9 @@
// List microphones
virtual status_t getMicrophones(std::vector<media::MicrophoneInfo> *microphones);
+ status_t addDeviceEffect(audio_port_handle_t device, sp<EffectHalInterface> effect) override;
+ status_t removeDeviceEffect(audio_port_handle_t device, sp<EffectHalInterface> effect) override;
+
virtual status_t dump(int fd);
private:
diff --git a/media/libaudiohal/impl/DeviceHalLocal.cpp b/media/libaudiohal/impl/DeviceHalLocal.cpp
index ee68252..8021d92 100644
--- a/media/libaudiohal/impl/DeviceHalLocal.cpp
+++ b/media/libaudiohal/impl/DeviceHalLocal.cpp
@@ -104,7 +104,7 @@
status_t DeviceHalLocal::openOutputStream(
audio_io_handle_t handle,
- audio_devices_t devices,
+ audio_devices_t deviceType,
audio_output_flags_t flags,
struct audio_config *config,
const char *address,
@@ -112,11 +112,11 @@
audio_stream_out_t *halStream;
ALOGV("open_output_stream handle: %d devices: %x flags: %#x"
"srate: %d format %#x channels %x address %s",
- handle, devices, flags,
+ handle, deviceType, flags,
config->sample_rate, config->format, config->channel_mask,
address);
int openResut = mDev->open_output_stream(
- mDev, handle, devices, flags, config, &halStream, address);
+ mDev, handle, deviceType, flags, config, &halStream, address);
if (openResut == OK) {
*outStream = new StreamOutHalLocal(halStream, this);
}
@@ -206,6 +206,17 @@
}
#endif
+// Local HAL implementation does not support effects
+status_t DeviceHalLocal::addDeviceEffect(
+ audio_port_handle_t device __unused, sp<EffectHalInterface> effect __unused) {
+ return INVALID_OPERATION;
+}
+
+status_t DeviceHalLocal::removeDeviceEffect(
+ audio_port_handle_t device __unused, sp<EffectHalInterface> effect __unused) {
+ return INVALID_OPERATION;
+}
+
status_t DeviceHalLocal::dump(int fd) {
return mDev->dump(mDev, fd);
}
diff --git a/media/libaudiohal/impl/DeviceHalLocal.h b/media/libaudiohal/impl/DeviceHalLocal.h
index 36db72e..d85e2a7 100644
--- a/media/libaudiohal/impl/DeviceHalLocal.h
+++ b/media/libaudiohal/impl/DeviceHalLocal.h
@@ -106,6 +106,9 @@
// List microphones
virtual status_t getMicrophones(std::vector<media::MicrophoneInfo> *microphones);
+ status_t addDeviceEffect(audio_port_handle_t device, sp<EffectHalInterface> effect) override;
+ status_t removeDeviceEffect(audio_port_handle_t device, sp<EffectHalInterface> effect) override;
+
virtual status_t dump(int fd);
void closeOutputStream(struct audio_stream_out *stream_out);
diff --git a/media/libaudiohal/impl/DevicesFactoryHalHidl.cpp b/media/libaudiohal/impl/DevicesFactoryHalHidl.cpp
index 5e01e42..e6e9688 100644
--- a/media/libaudiohal/impl/DevicesFactoryHalHidl.cpp
+++ b/media/libaudiohal/impl/DevicesFactoryHalHidl.cpp
@@ -20,6 +20,7 @@
#define LOG_TAG "DevicesFactoryHalHidl"
//#define LOG_NDEBUG 0
+#include <android/hidl/manager/1.0/IServiceManager.h>
#include PATH(android/hardware/audio/FILE_VERSION/IDevice.h)
#include <media/audiohal/hidl/HalDeathHandler.h>
#include <utils/Log.h>
@@ -28,6 +29,8 @@
#include "DeviceHalHidl.h"
#include "DevicesFactoryHalHidl.h"
+#include <set>
+
using ::android::hardware::audio::CPP_VERSION::IDevice;
using ::android::hardware::audio::CPP_VERSION::Result;
using ::android::hardware::Return;
@@ -35,13 +38,10 @@
namespace android {
namespace CPP_VERSION {
-DevicesFactoryHalHidl::DevicesFactoryHalHidl() {
- sp<IDevicesFactory> defaultFactory{IDevicesFactory::getService()};
- if (!defaultFactory) {
- ALOGE("Failed to obtain IDevicesFactory/default service, terminating process.");
- exit(1);
- }
- mDeviceFactories.push_back(defaultFactory);
+DevicesFactoryHalHidl::DevicesFactoryHalHidl(sp<IDevicesFactory> devicesFactory) {
+ ALOG_ASSERT(devicesFactory != nullptr, "Provided IDevicesFactory service is NULL");
+
+ mDeviceFactories.push_back(devicesFactory);
if (MAJOR_VERSION >= 4) {
// The MSD factory is optional and only available starting at HAL 4.0
sp<IDevicesFactory> msdFactory{IDevicesFactory::getService(AUDIO_HAL_SERVICE_NAME_MSD)};
@@ -111,5 +111,29 @@
return BAD_VALUE;
}
+status_t DevicesFactoryHalHidl::getHalPids(std::vector<pid_t> *pids) {
+ std::set<pid_t> pidsSet;
+
+ for (const auto& factory : mDeviceFactories) {
+ using ::android::hidl::base::V1_0::DebugInfo;
+ using android::hidl::manager::V1_0::IServiceManager;
+
+ DebugInfo debugInfo;
+ auto ret = factory->getDebugInfo([&] (const auto &info) {
+ debugInfo = info;
+ });
+ if (!ret.isOk()) {
+ return INVALID_OPERATION;
+ }
+ if (debugInfo.pid == (int)IServiceManager::PidConstant::NO_PID) {
+ continue;
+ }
+ pidsSet.insert(debugInfo.pid);
+ }
+
+ *pids = {pidsSet.begin(), pidsSet.end()};
+ return NO_ERROR;
+}
+
} // namespace CPP_VERSION
} // namespace android
diff --git a/media/libaudiohal/impl/DevicesFactoryHalHidl.h b/media/libaudiohal/impl/DevicesFactoryHalHidl.h
index 27e0649..52185c8 100644
--- a/media/libaudiohal/impl/DevicesFactoryHalHidl.h
+++ b/media/libaudiohal/impl/DevicesFactoryHalHidl.h
@@ -32,18 +32,17 @@
class DevicesFactoryHalHidl : public DevicesFactoryHalInterface
{
public:
+ DevicesFactoryHalHidl(sp<IDevicesFactory> devicesFactory);
+
// Opens a device with the specified name. To close the device, it is
// necessary to release references to the returned object.
virtual status_t openDevice(const char *name, sp<DeviceHalInterface> *device);
+ status_t getHalPids(std::vector<pid_t> *pids) override;
+
private:
- friend class DevicesFactoryHalHybrid;
-
std::vector<sp<IDevicesFactory>> mDeviceFactories;
- // Can not be constructed directly by clients.
- DevicesFactoryHalHidl();
-
virtual ~DevicesFactoryHalHidl() = default;
};
diff --git a/media/libaudiohal/impl/DevicesFactoryHalHybrid.cpp b/media/libaudiohal/impl/DevicesFactoryHalHybrid.cpp
index f337a8b..52f150a 100644
--- a/media/libaudiohal/impl/DevicesFactoryHalHybrid.cpp
+++ b/media/libaudiohal/impl/DevicesFactoryHalHybrid.cpp
@@ -17,16 +17,16 @@
#define LOG_TAG "DevicesFactoryHalHybrid"
//#define LOG_NDEBUG 0
+#include "DevicesFactoryHalHidl.h"
#include "DevicesFactoryHalHybrid.h"
#include "DevicesFactoryHalLocal.h"
-#include "DevicesFactoryHalHidl.h"
namespace android {
namespace CPP_VERSION {
-DevicesFactoryHalHybrid::DevicesFactoryHalHybrid()
+DevicesFactoryHalHybrid::DevicesFactoryHalHybrid(sp<IDevicesFactory> hidlFactory)
: mLocalFactory(new DevicesFactoryHalLocal()),
- mHidlFactory(new DevicesFactoryHalHidl()) {
+ mHidlFactory(new DevicesFactoryHalHidl(hidlFactory)) {
}
status_t DevicesFactoryHalHybrid::openDevice(const char *name, sp<DeviceHalInterface> *device) {
@@ -37,5 +37,18 @@
return mLocalFactory->openDevice(name, device);
}
+status_t DevicesFactoryHalHybrid::getHalPids(std::vector<pid_t> *pids) {
+ if (mHidlFactory != 0) {
+ return mHidlFactory->getHalPids(pids);
+ }
+ return INVALID_OPERATION;
+}
+
} // namespace CPP_VERSION
+
+extern "C" __attribute__((visibility("default"))) void* createIDevicesFactory() {
+ auto service = hardware::audio::CPP_VERSION::IDevicesFactory::getService();
+ return service ? new CPP_VERSION::DevicesFactoryHalHybrid(service) : nullptr;
+}
+
} // namespace android
diff --git a/media/libaudiohal/impl/DevicesFactoryHalHybrid.h b/media/libaudiohal/impl/DevicesFactoryHalHybrid.h
index 5ac0d0d..2189b36 100644
--- a/media/libaudiohal/impl/DevicesFactoryHalHybrid.h
+++ b/media/libaudiohal/impl/DevicesFactoryHalHybrid.h
@@ -17,31 +17,32 @@
#ifndef ANDROID_HARDWARE_DEVICES_FACTORY_HAL_HYBRID_H
#define ANDROID_HARDWARE_DEVICES_FACTORY_HAL_HYBRID_H
+#include PATH(android/hardware/audio/FILE_VERSION/IDevicesFactory.h)
#include <media/audiohal/DevicesFactoryHalInterface.h>
#include <utils/Errors.h>
#include <utils/RefBase.h>
+using ::android::hardware::audio::CPP_VERSION::IDevicesFactory;
+
namespace android {
namespace CPP_VERSION {
class DevicesFactoryHalHybrid : public DevicesFactoryHalInterface
{
public:
- DevicesFactoryHalHybrid();
+ DevicesFactoryHalHybrid(sp<IDevicesFactory> hidlFactory);
// Opens a device with the specified name. To close the device, it is
// necessary to release references to the returned object.
virtual status_t openDevice(const char *name, sp<DeviceHalInterface> *device);
+ status_t getHalPids(std::vector<pid_t> *pids) override;
+
private:
sp<DevicesFactoryHalInterface> mLocalFactory;
sp<DevicesFactoryHalInterface> mHidlFactory;
};
-sp<DevicesFactoryHalInterface> createDevicesFactoryHal() {
- return new DevicesFactoryHalHybrid();
-}
-
} // namespace CPP_VERSION
} // namespace android
diff --git a/media/libaudiohal/impl/DevicesFactoryHalLocal.h b/media/libaudiohal/impl/DevicesFactoryHalLocal.h
index 5d108dd..2b011f4 100644
--- a/media/libaudiohal/impl/DevicesFactoryHalLocal.h
+++ b/media/libaudiohal/impl/DevicesFactoryHalLocal.h
@@ -33,6 +33,10 @@
// necessary to release references to the returned object.
virtual status_t openDevice(const char *name, sp<DeviceHalInterface> *device);
+ status_t getHalPids(std::vector<pid_t> *pids __unused) override {
+ return INVALID_OPERATION;
+ }
+
private:
friend class DevicesFactoryHalHybrid;
diff --git a/media/libaudiohal/impl/EffectsFactoryHalHidl.cpp b/media/libaudiohal/impl/EffectsFactoryHalHidl.cpp
index 7fd6bde..9192a31 100644
--- a/media/libaudiohal/impl/EffectsFactoryHalHidl.cpp
+++ b/media/libaudiohal/impl/EffectsFactoryHalHidl.cpp
@@ -19,10 +19,10 @@
#include <cutils/native_handle.h>
-#include "EffectsFactoryHalHidl.h"
#include "ConversionHelperHidl.h"
#include "EffectBufferHalHidl.h"
#include "EffectHalHidl.h"
+#include "EffectsFactoryHalHidl.h"
#include "HidlUtils.h"
using ::android::hardware::audio::common::CPP_VERSION::implementation::HidlUtils;
@@ -35,12 +35,10 @@
using namespace ::android::hardware::audio::common::CPP_VERSION;
using namespace ::android::hardware::audio::effect::CPP_VERSION;
-EffectsFactoryHalHidl::EffectsFactoryHalHidl() : ConversionHelperHidl("EffectsFactory") {
- mEffectsFactory = IEffectsFactory::getService();
- if (mEffectsFactory == 0) {
- ALOGE("Failed to obtain IEffectsFactory service, terminating process.");
- exit(1);
- }
+EffectsFactoryHalHidl::EffectsFactoryHalHidl(sp<IEffectsFactory> effectsFactory)
+ : ConversionHelperHidl("EffectsFactory") {
+ ALOG_ASSERT(effectsFactory != nullptr, "Provided IDevicesFactory service is NULL");
+ mEffectsFactory = effectsFactory;
}
status_t EffectsFactoryHalHidl::queryAllDescriptors() {
@@ -106,12 +104,26 @@
status_t EffectsFactoryHalHidl::createEffect(
const effect_uuid_t *pEffectUuid, int32_t sessionId, int32_t ioId,
- sp<EffectHalInterface> *effect) {
+ int32_t deviceId __unused, sp<EffectHalInterface> *effect) {
if (mEffectsFactory == 0) return NO_INIT;
Uuid hidlUuid;
HidlUtils::uuidFromHal(*pEffectUuid, &hidlUuid);
Result retval = Result::NOT_INITIALIZED;
- Return<void> ret = mEffectsFactory->createEffect(
+ Return<void> ret;
+#if MAJOR_VERSION >= 6
+ ret = mEffectsFactory->createEffect(
+ hidlUuid, sessionId, ioId, deviceId,
+ [&](Result r, const sp<IEffect>& result, uint64_t effectId) {
+ retval = r;
+ if (retval == Result::OK) {
+ *effect = new EffectHalHidl(result, effectId);
+ }
+ });
+#else
+ if (sessionId == AUDIO_SESSION_DEVICE && ioId == AUDIO_IO_HANDLE_NONE) {
+ return INVALID_OPERATION;
+ }
+ ret = mEffectsFactory->createEffect(
hidlUuid, sessionId, ioId,
[&](Result r, const sp<IEffect>& result, uint64_t effectId) {
retval = r;
@@ -119,6 +131,7 @@
*effect = new EffectHalHidl(result, effectId);
}
});
+#endif
if (ret.isOk()) {
if (retval == Result::OK) return OK;
else if (retval == Result::INVALID_ARGUMENTS) return NAME_NOT_FOUND;
@@ -147,4 +160,10 @@
} // namespace CPP_VERSION
} // namespace effect
+
+extern "C" __attribute__((visibility("default"))) void* createIEffectsFactory() {
+ auto service = hardware::audio::effect::CPP_VERSION::IEffectsFactory::getService();
+ return service ? new effect::CPP_VERSION::EffectsFactoryHalHidl(service) : nullptr;
+}
+
} // namespace android
diff --git a/media/libaudiohal/impl/EffectsFactoryHalHidl.h b/media/libaudiohal/impl/EffectsFactoryHalHidl.h
index 01178ff..dece1bb 100644
--- a/media/libaudiohal/impl/EffectsFactoryHalHidl.h
+++ b/media/libaudiohal/impl/EffectsFactoryHalHidl.h
@@ -18,7 +18,6 @@
#define ANDROID_HARDWARE_EFFECTS_FACTORY_HAL_HIDL_H
#include PATH(android/hardware/audio/effect/FILE_VERSION/IEffectsFactory.h)
-#include PATH(android/hardware/audio/effect/FILE_VERSION/types.h)
#include <media/audiohal/EffectsFactoryHalInterface.h>
#include "ConversionHelperHidl.h"
@@ -34,7 +33,7 @@
class EffectsFactoryHalHidl : public EffectsFactoryHalInterface, public ConversionHelperHidl
{
public:
- EffectsFactoryHalHidl();
+ EffectsFactoryHalHidl(sp<IEffectsFactory> effectsFactory);
// Returns the number of different effects in all loaded libraries.
virtual status_t queryNumberEffects(uint32_t *pNumEffects);
@@ -50,7 +49,7 @@
// To release the effect engine, it is necessary to release references
// to the returned effect object.
virtual status_t createEffect(const effect_uuid_t *pEffectUuid,
- int32_t sessionId, int32_t ioId,
+ int32_t sessionId, int32_t ioId, int32_t deviceId,
sp<EffectHalInterface> *effect);
virtual status_t dumpEffects(int fd);
@@ -66,10 +65,6 @@
status_t queryAllDescriptors();
};
-sp<EffectsFactoryHalInterface> createEffectsFactoryHal() {
- return new EffectsFactoryHalHidl();
-}
-
} // namespace CPP_VERSION
} // namespace effect
} // namespace android
diff --git a/media/libaudiohal/impl/include/libaudiohal/FactoryHalHidl.h b/media/libaudiohal/impl/include/libaudiohal/FactoryHalHidl.h
deleted file mode 100644
index c7319d0..0000000
--- a/media/libaudiohal/impl/include/libaudiohal/FactoryHalHidl.h
+++ /dev/null
@@ -1,56 +0,0 @@
-/*
- * Copyright (C) 2018 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_HARDWARE_FACTORY_HAL_HIDL_H
-#define ANDROID_HARDWARE_FACTORY_HAL_HIDL_H
-
-/** @file Library entry points to create the HAL factories. */
-
-#include <media/audiohal/DevicesFactoryHalInterface.h>
-#include <media/audiohal/EffectsFactoryHalInterface.h>
-#include <utils/StrongPointer.h>
-
-namespace android {
-
-namespace effect {
-namespace V2_0 {
-sp<EffectsFactoryHalInterface> createEffectsFactoryHal();
-} // namespace V2_0
-
-namespace V4_0 {
-sp<EffectsFactoryHalInterface> createEffectsFactoryHal();
-} // namespace V4_0
-
-namespace V5_0 {
-sp<EffectsFactoryHalInterface> createEffectsFactoryHal();
-} // namespace V5_0
-} // namespace effect
-
-namespace V2_0 {
-sp<DevicesFactoryHalInterface> createDevicesFactoryHal();
-} // namespace V2_0
-
-namespace V4_0 {
-sp<DevicesFactoryHalInterface> createDevicesFactoryHal();
-} // namespace V4_0
-
-namespace V5_0 {
-sp<DevicesFactoryHalInterface> createDevicesFactoryHal();
-} // namespace V5_0
-
-} // namespace android
-
-#endif // ANDROID_HARDWARE_FACTORY_HAL_HIDL_H
diff --git a/media/libaudiohal/include/media/audiohal/DeviceHalInterface.h b/media/libaudiohal/include/media/audiohal/DeviceHalInterface.h
index e565237..1e04b21 100644
--- a/media/libaudiohal/include/media/audiohal/DeviceHalInterface.h
+++ b/media/libaudiohal/include/media/audiohal/DeviceHalInterface.h
@@ -17,6 +17,7 @@
#ifndef ANDROID_HARDWARE_DEVICE_HAL_INTERFACE_H
#define ANDROID_HARDWARE_DEVICE_HAL_INTERFACE_H
+#include <media/audiohal/EffectHalInterface.h>
#include <media/MicrophoneInfo.h>
#include <system/audio.h>
#include <utils/Errors.h>
@@ -69,7 +70,7 @@
// by releasing all references to the returned object.
virtual status_t openOutputStream(
audio_io_handle_t handle,
- audio_devices_t devices,
+ audio_devices_t deviceType,
audio_output_flags_t flags,
struct audio_config *config,
const char *address,
@@ -111,6 +112,11 @@
// List microphones
virtual status_t getMicrophones(std::vector<media::MicrophoneInfo> *microphones) = 0;
+ virtual status_t addDeviceEffect(
+ audio_port_handle_t device, sp<EffectHalInterface> effect) = 0;
+ virtual status_t removeDeviceEffect(
+ audio_port_handle_t device, sp<EffectHalInterface> effect) = 0;
+
virtual status_t dump(int fd) = 0;
protected:
diff --git a/media/libaudiohal/include/media/audiohal/DevicesFactoryHalInterface.h b/media/libaudiohal/include/media/audiohal/DevicesFactoryHalInterface.h
index 14af384..e9ac1ce 100644
--- a/media/libaudiohal/include/media/audiohal/DevicesFactoryHalInterface.h
+++ b/media/libaudiohal/include/media/audiohal/DevicesFactoryHalInterface.h
@@ -20,6 +20,7 @@
#include <media/audiohal/DeviceHalInterface.h>
#include <utils/Errors.h>
#include <utils/RefBase.h>
+#include <vector>
namespace android {
@@ -30,6 +31,8 @@
// necessary to release references to the returned object.
virtual status_t openDevice(const char *name, sp<DeviceHalInterface> *device) = 0;
+ virtual status_t getHalPids(std::vector<pid_t> *pids) = 0;
+
static sp<DevicesFactoryHalInterface> create();
protected:
diff --git a/media/libaudiohal/include/media/audiohal/EffectsFactoryHalInterface.h b/media/libaudiohal/include/media/audiohal/EffectsFactoryHalInterface.h
index 316a46c..3a76f9f 100644
--- a/media/libaudiohal/include/media/audiohal/EffectsFactoryHalInterface.h
+++ b/media/libaudiohal/include/media/audiohal/EffectsFactoryHalInterface.h
@@ -41,7 +41,7 @@
// To release the effect engine, it is necessary to release references
// to the returned effect object.
virtual status_t createEffect(const effect_uuid_t *pEffectUuid,
- int32_t sessionId, int32_t ioId,
+ int32_t sessionId, int32_t ioId, int32_t deviceId,
sp<EffectHalInterface> *effect) = 0;
virtual status_t dumpEffects(int fd) = 0;
diff --git a/media/libaudiohal/include/media/audiohal/FactoryHalHidl.h b/media/libaudiohal/include/media/audiohal/FactoryHalHidl.h
new file mode 100644
index 0000000..d353ed0
--- /dev/null
+++ b/media/libaudiohal/include/media/audiohal/FactoryHalHidl.h
@@ -0,0 +1,40 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_HARDWARE_FACTORY_HAL_HIDL_H
+#define ANDROID_HARDWARE_FACTORY_HAL_HIDL_H
+
+#include <string>
+
+#include <utils/StrongPointer.h>
+
+namespace android {
+
+namespace detail {
+
+void* createPreferredImpl(const std::string& package, const std::string& interface);
+
+} // namespace detail
+
+/** @Return the preferred available implementation or nullptr if none are available. */
+template <class Interface>
+static sp<Interface> createPreferredImpl(const std::string& package, const std::string& interface) {
+ return sp<Interface>{static_cast<Interface*>(detail::createPreferredImpl(package, interface))};
+}
+
+} // namespace android
+
+#endif // ANDROID_HARDWARE_FACTORY_HAL_HIDL_H
diff --git a/media/libaudioprocessing/Android.bp b/media/libaudioprocessing/Android.bp
index cb78063..e756ada 100644
--- a/media/libaudioprocessing/Android.bp
+++ b/media/libaudioprocessing/Android.bp
@@ -3,20 +3,13 @@
export_include_dirs: ["include"],
+ header_libs: ["libaudioclient_headers"],
+
shared_libs: [
- "libaudiohal",
"libaudioutils",
"libcutils",
"liblog",
- "libnbaio",
- "libnblog",
- "libsonic",
"libutils",
- "libvibrator",
- ],
-
- header_libs: [
- "libbase_headers",
],
cflags: [
@@ -26,6 +19,25 @@
// uncomment to disable NEON on architectures that actually do support NEON, for benchmarking
// "-DUSE_NEON=false",
],
+
+ arch: {
+ x86: {
+ avx2: {
+ cflags: [
+ "-mavx2",
+ "-mfma",
+ ],
+ },
+ },
+ x86_64: {
+ avx2: {
+ cflags: [
+ "-mavx2",
+ "-mfma",
+ ],
+ },
+ },
+ },
}
cc_library_shared {
@@ -33,18 +45,32 @@
defaults: ["libaudioprocessing_defaults"],
srcs: [
+ "AudioMixer.cpp",
"BufferProviders.cpp",
"RecordBufferConverter.cpp",
],
- whole_static_libs: ["libaudioprocessing_arm"],
+
+ header_libs: [
+ "libbase_headers",
+ "libmedia_headers"
+ ],
+
+ shared_libs: [
+ "libaudiohal",
+ "libsonic",
+ "libvibrator",
+ ],
+
+ whole_static_libs: ["libaudioprocessing_base"],
}
cc_library_static {
- name: "libaudioprocessing_arm",
+ name: "libaudioprocessing_base",
defaults: ["libaudioprocessing_defaults"],
+ vendor_available: true,
srcs: [
- "AudioMixer.cpp",
+ "AudioMixerBase.cpp",
"AudioResampler.cpp",
"AudioResamplerCubic.cpp",
"AudioResamplerSinc.cpp",
diff --git a/media/libaudioprocessing/AudioMixer.cpp b/media/libaudioprocessing/AudioMixer.cpp
index f7cc096..c0b11a4 100644
--- a/media/libaudioprocessing/AudioMixer.cpp
+++ b/media/libaudioprocessing/AudioMixer.cpp
@@ -18,6 +18,7 @@
#define LOG_TAG "AudioMixer"
//#define LOG_NDEBUG 0
+#include <sstream>
#include <stdint.h>
#include <string.h>
#include <stdlib.h>
@@ -27,9 +28,6 @@
#include <utils/Errors.h>
#include <utils/Log.h>
-#include <cutils/compiler.h>
-#include <utils/Debug.h>
-
#include <system/audio.h>
#include <audio_utils/primitives.h>
@@ -58,138 +56,15 @@
#define ALOGVV(a...) do { } while (0)
#endif
-#ifndef ARRAY_SIZE
-#define ARRAY_SIZE(x) (sizeof(x)/sizeof((x)[0]))
-#endif
-
-// Set kUseNewMixer to true to use the new mixer engine always. Otherwise the
-// original code will be used for stereo sinks, the new mixer for multichannel.
-static constexpr bool kUseNewMixer = true;
-
-// Set kUseFloat to true to allow floating input into the mixer engine.
-// If kUseNewMixer is false, this is ignored or may be overridden internally
-// because of downmix/upmix support.
-static constexpr bool kUseFloat = true;
-
-#ifdef FLOAT_AUX
-using TYPE_AUX = float;
-static_assert(kUseNewMixer && kUseFloat,
- "kUseNewMixer and kUseFloat must be true for FLOAT_AUX option");
-#else
-using TYPE_AUX = int32_t; // q4.27
-#endif
-
// Set to default copy buffer size in frames for input processing.
-static const size_t kCopyBufferFrameCount = 256;
+static constexpr size_t kCopyBufferFrameCount = 256;
namespace android {
// ----------------------------------------------------------------------------
-static inline audio_format_t selectMixerInFormat(audio_format_t inputFormat __unused) {
- return kUseFloat && kUseNewMixer ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
-}
-
-status_t AudioMixer::create(
- int name, audio_channel_mask_t channelMask, audio_format_t format, int sessionId)
-{
- LOG_ALWAYS_FATAL_IF(exists(name), "name %d already exists", name);
-
- if (!isValidChannelMask(channelMask)) {
- ALOGE("%s invalid channelMask: %#x", __func__, channelMask);
- return BAD_VALUE;
- }
- if (!isValidFormat(format)) {
- ALOGE("%s invalid format: %#x", __func__, format);
- return BAD_VALUE;
- }
-
- auto t = std::make_shared<Track>();
- {
- // TODO: move initialization to the Track constructor.
- // assume default parameters for the track, except where noted below
- t->needs = 0;
-
- // Integer volume.
- // Currently integer volume is kept for the legacy integer mixer.
- // Will be removed when the legacy mixer path is removed.
- t->volume[0] = 0;
- t->volume[1] = 0;
- t->prevVolume[0] = 0 << 16;
- t->prevVolume[1] = 0 << 16;
- t->volumeInc[0] = 0;
- t->volumeInc[1] = 0;
- t->auxLevel = 0;
- t->auxInc = 0;
- t->prevAuxLevel = 0;
-
- // Floating point volume.
- t->mVolume[0] = 0.f;
- t->mVolume[1] = 0.f;
- t->mPrevVolume[0] = 0.f;
- t->mPrevVolume[1] = 0.f;
- t->mVolumeInc[0] = 0.;
- t->mVolumeInc[1] = 0.;
- t->mAuxLevel = 0.;
- t->mAuxInc = 0.;
- t->mPrevAuxLevel = 0.;
-
- // no initialization needed
- // t->frameCount
- t->mHapticChannelMask = channelMask & AUDIO_CHANNEL_HAPTIC_ALL;
- t->mHapticChannelCount = audio_channel_count_from_out_mask(t->mHapticChannelMask);
- channelMask &= ~AUDIO_CHANNEL_HAPTIC_ALL;
- t->channelCount = audio_channel_count_from_out_mask(channelMask);
- t->enabled = false;
- ALOGV_IF(audio_channel_mask_get_bits(channelMask) != AUDIO_CHANNEL_OUT_STEREO,
- "Non-stereo channel mask: %d\n", channelMask);
- t->channelMask = channelMask;
- t->sessionId = sessionId;
- // setBufferProvider(name, AudioBufferProvider *) is required before enable(name)
- t->bufferProvider = NULL;
- t->buffer.raw = NULL;
- // no initialization needed
- // t->buffer.frameCount
- t->hook = NULL;
- t->mIn = NULL;
- t->sampleRate = mSampleRate;
- // setParameter(name, TRACK, MAIN_BUFFER, mixBuffer) is required before enable(name)
- t->mainBuffer = NULL;
- t->auxBuffer = NULL;
- t->mInputBufferProvider = NULL;
- t->mMixerFormat = AUDIO_FORMAT_PCM_16_BIT;
- t->mFormat = format;
- t->mMixerInFormat = selectMixerInFormat(format);
- t->mDownmixRequiresFormat = AUDIO_FORMAT_INVALID; // no format required
- t->mMixerChannelMask = audio_channel_mask_from_representation_and_bits(
- AUDIO_CHANNEL_REPRESENTATION_POSITION, AUDIO_CHANNEL_OUT_STEREO);
- t->mMixerChannelCount = audio_channel_count_from_out_mask(t->mMixerChannelMask);
- t->mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT;
- // haptic
- t->mHapticPlaybackEnabled = false;
- t->mHapticIntensity = HAPTIC_SCALE_NONE;
- t->mMixerHapticChannelMask = AUDIO_CHANNEL_NONE;
- t->mMixerHapticChannelCount = 0;
- t->mAdjustInChannelCount = t->channelCount + t->mHapticChannelCount;
- t->mAdjustOutChannelCount = t->channelCount + t->mMixerHapticChannelCount;
- t->mAdjustNonDestructiveInChannelCount = t->mAdjustOutChannelCount;
- t->mAdjustNonDestructiveOutChannelCount = t->channelCount;
- t->mKeepContractedChannels = false;
- // Check the downmixing (or upmixing) requirements.
- status_t status = t->prepareForDownmix();
- if (status != OK) {
- ALOGE("AudioMixer::getTrackName invalid channelMask (%#x)", channelMask);
- return BAD_VALUE;
- }
- // prepareForDownmix() may change mDownmixRequiresFormat
- ALOGVV("mMixerFormat:%#x mMixerInFormat:%#x\n", t->mMixerFormat, t->mMixerInFormat);
- t->prepareForReformat();
- t->prepareForAdjustChannelsNonDestructive(mFrameCount);
- t->prepareForAdjustChannels();
-
- mTracks[name] = t;
- return OK;
- }
+bool AudioMixer::isValidChannelMask(audio_channel_mask_t channelMask) const {
+ return audio_channel_mask_is_valid(channelMask); // the RemixBufferProvider is flexible.
}
// Called when channel masks have changed for a track name
@@ -198,7 +73,7 @@
bool AudioMixer::setChannelMasks(int name,
audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask) {
LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
- const std::shared_ptr<Track> &track = mTracks[name];
+ const std::shared_ptr<Track> &track = getTrack(name);
if (trackChannelMask == (track->channelMask | track->mHapticChannelMask)
&& mixerChannelMask == (track->mMixerChannelMask | track->mMixerHapticChannelMask)) {
@@ -255,14 +130,8 @@
track->prepareForAdjustChannelsNonDestructive(mFrameCount);
track->prepareForAdjustChannels();
- if (track->mResampler.get() != nullptr) {
- // resampler channels may have changed.
- const uint32_t resetToSampleRate = track->sampleRate;
- track->mResampler.reset(nullptr);
- track->sampleRate = mSampleRate; // without resampler, track rate is device sample rate.
- // recreate the resampler with updated format, channels, saved sampleRate.
- track->setResampler(resetToSampleRate /*trackSampleRate*/, mSampleRate /*devSampleRate*/);
- }
+ // Resampler channels may have changed.
+ track->recreateResampler(mSampleRate);
return true;
}
@@ -477,171 +346,10 @@
}
}
-void AudioMixer::destroy(int name)
-{
- LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
- ALOGV("deleteTrackName(%d)", name);
-
- if (mTracks[name]->enabled) {
- invalidate();
- }
- mTracks.erase(name); // deallocate track
-}
-
-void AudioMixer::enable(int name)
-{
- LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
- const std::shared_ptr<Track> &track = mTracks[name];
-
- if (!track->enabled) {
- track->enabled = true;
- ALOGV("enable(%d)", name);
- invalidate();
- }
-}
-
-void AudioMixer::disable(int name)
-{
- LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
- const std::shared_ptr<Track> &track = mTracks[name];
-
- if (track->enabled) {
- track->enabled = false;
- ALOGV("disable(%d)", name);
- invalidate();
- }
-}
-
-/* Sets the volume ramp variables for the AudioMixer.
- *
- * The volume ramp variables are used to transition from the previous
- * volume to the set volume. ramp controls the duration of the transition.
- * Its value is typically one state framecount period, but may also be 0,
- * meaning "immediate."
- *
- * FIXME: 1) Volume ramp is enabled only if there is a nonzero integer increment
- * even if there is a nonzero floating point increment (in that case, the volume
- * change is immediate). This restriction should be changed when the legacy mixer
- * is removed (see #2).
- * FIXME: 2) Integer volume variables are used for Legacy mixing and should be removed
- * when no longer needed.
- *
- * @param newVolume set volume target in floating point [0.0, 1.0].
- * @param ramp number of frames to increment over. if ramp is 0, the volume
- * should be set immediately. Currently ramp should not exceed 65535 (frames).
- * @param pIntSetVolume pointer to the U4.12 integer target volume, set on return.
- * @param pIntPrevVolume pointer to the U4.28 integer previous volume, set on return.
- * @param pIntVolumeInc pointer to the U4.28 increment per output audio frame, set on return.
- * @param pSetVolume pointer to the float target volume, set on return.
- * @param pPrevVolume pointer to the float previous volume, set on return.
- * @param pVolumeInc pointer to the float increment per output audio frame, set on return.
- * @return true if the volume has changed, false if volume is same.
- */
-static inline bool setVolumeRampVariables(float newVolume, int32_t ramp,
- int16_t *pIntSetVolume, int32_t *pIntPrevVolume, int32_t *pIntVolumeInc,
- float *pSetVolume, float *pPrevVolume, float *pVolumeInc) {
- // check floating point volume to see if it is identical to the previously
- // set volume.
- // We do not use a tolerance here (and reject changes too small)
- // as it may be confusing to use a different value than the one set.
- // If the resulting volume is too small to ramp, it is a direct set of the volume.
- if (newVolume == *pSetVolume) {
- return false;
- }
- if (newVolume < 0) {
- newVolume = 0; // should not have negative volumes
- } else {
- switch (fpclassify(newVolume)) {
- case FP_SUBNORMAL:
- case FP_NAN:
- newVolume = 0;
- break;
- case FP_ZERO:
- break; // zero volume is fine
- case FP_INFINITE:
- // Infinite volume could be handled consistently since
- // floating point math saturates at infinities,
- // but we limit volume to unity gain float.
- // ramp = 0; break;
- //
- newVolume = AudioMixer::UNITY_GAIN_FLOAT;
- break;
- case FP_NORMAL:
- default:
- // Floating point does not have problems with overflow wrap
- // that integer has. However, we limit the volume to
- // unity gain here.
- // TODO: Revisit the volume limitation and perhaps parameterize.
- if (newVolume > AudioMixer::UNITY_GAIN_FLOAT) {
- newVolume = AudioMixer::UNITY_GAIN_FLOAT;
- }
- break;
- }
- }
-
- // set floating point volume ramp
- if (ramp != 0) {
- // when the ramp completes, *pPrevVolume is set to *pSetVolume, so there
- // is no computational mismatch; hence equality is checked here.
- ALOGD_IF(*pPrevVolume != *pSetVolume, "previous float ramp hasn't finished,"
- " prev:%f set_to:%f", *pPrevVolume, *pSetVolume);
- const float inc = (newVolume - *pPrevVolume) / ramp; // could be inf, nan, subnormal
- // could be inf, cannot be nan, subnormal
- const float maxv = std::max(newVolume, *pPrevVolume);
-
- if (isnormal(inc) // inc must be a normal number (no subnormals, infinite, nan)
- && maxv + inc != maxv) { // inc must make forward progress
- *pVolumeInc = inc;
- // ramp is set now.
- // Note: if newVolume is 0, then near the end of the ramp,
- // it may be possible that the ramped volume may be subnormal or
- // temporarily negative by a small amount or subnormal due to floating
- // point inaccuracies.
- } else {
- ramp = 0; // ramp not allowed
- }
- }
-
- // compute and check integer volume, no need to check negative values
- // The integer volume is limited to "unity_gain" to avoid wrapping and other
- // audio artifacts, so it never reaches the range limit of U4.28.
- // We safely use signed 16 and 32 bit integers here.
- const float scaledVolume = newVolume * AudioMixer::UNITY_GAIN_INT; // not neg, subnormal, nan
- const int32_t intVolume = (scaledVolume >= (float)AudioMixer::UNITY_GAIN_INT) ?
- AudioMixer::UNITY_GAIN_INT : (int32_t)scaledVolume;
-
- // set integer volume ramp
- if (ramp != 0) {
- // integer volume is U4.12 (to use 16 bit multiplies), but ramping uses U4.28.
- // when the ramp completes, *pIntPrevVolume is set to *pIntSetVolume << 16, so there
- // is no computational mismatch; hence equality is checked here.
- ALOGD_IF(*pIntPrevVolume != *pIntSetVolume << 16, "previous int ramp hasn't finished,"
- " prev:%d set_to:%d", *pIntPrevVolume, *pIntSetVolume << 16);
- const int32_t inc = ((intVolume << 16) - *pIntPrevVolume) / ramp;
-
- if (inc != 0) { // inc must make forward progress
- *pIntVolumeInc = inc;
- } else {
- ramp = 0; // ramp not allowed
- }
- }
-
- // if no ramp, or ramp not allowed, then clear float and integer increments
- if (ramp == 0) {
- *pVolumeInc = 0;
- *pPrevVolume = newVolume;
- *pIntVolumeInc = 0;
- *pIntPrevVolume = intVolume << 16;
- }
- *pSetVolume = newVolume;
- *pIntSetVolume = intVolume;
- return true;
-}
-
void AudioMixer::setParameter(int name, int target, int param, void *value)
{
LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
- const std::shared_ptr<Track> &track = mTracks[name];
+ const std::shared_ptr<Track> &track = getTrack(name);
int valueInt = static_cast<int>(reinterpret_cast<uintptr_t>(value));
int32_t *valueBuf = reinterpret_cast<int32_t*>(value);
@@ -670,11 +378,7 @@
}
break;
case AUX_BUFFER:
- if (track->auxBuffer != valueBuf) {
- track->auxBuffer = valueBuf;
- ALOGV("setParameter(TRACK, AUX_BUFFER, %p)", valueBuf);
- invalidate();
- }
+ AudioMixerBase::setParameter(name, target, param, value);
break;
case FORMAT: {
audio_format_t format = static_cast<audio_format_t>(valueInt);
@@ -730,127 +434,38 @@
break;
case RESAMPLE:
- switch (param) {
- case SAMPLE_RATE:
- ALOG_ASSERT(valueInt > 0, "bad sample rate %d", valueInt);
- if (track->setResampler(uint32_t(valueInt), mSampleRate)) {
- ALOGV("setParameter(RESAMPLE, SAMPLE_RATE, %u)",
- uint32_t(valueInt));
- invalidate();
- }
- break;
- case RESET:
- track->resetResampler();
- invalidate();
- break;
- case REMOVE:
- track->mResampler.reset(nullptr);
- track->sampleRate = mSampleRate;
- invalidate();
- break;
- default:
- LOG_ALWAYS_FATAL("setParameter resample: bad param %d", param);
- }
- break;
-
case RAMP_VOLUME:
case VOLUME:
+ AudioMixerBase::setParameter(name, target, param, value);
+ break;
+ case TIMESTRETCH:
switch (param) {
- case AUXLEVEL:
- if (setVolumeRampVariables(*reinterpret_cast<float*>(value),
- target == RAMP_VOLUME ? mFrameCount : 0,
- &track->auxLevel, &track->prevAuxLevel, &track->auxInc,
- &track->mAuxLevel, &track->mPrevAuxLevel, &track->mAuxInc)) {
- ALOGV("setParameter(%s, AUXLEVEL: %04x)",
- target == VOLUME ? "VOLUME" : "RAMP_VOLUME", track->auxLevel);
- invalidate();
+ case PLAYBACK_RATE: {
+ const AudioPlaybackRate *playbackRate =
+ reinterpret_cast<AudioPlaybackRate*>(value);
+ ALOGW_IF(!isAudioPlaybackRateValid(*playbackRate),
+ "bad parameters speed %f, pitch %f",
+ playbackRate->mSpeed, playbackRate->mPitch);
+ if (track->setPlaybackRate(*playbackRate)) {
+ ALOGV("setParameter(TIMESTRETCH, PLAYBACK_RATE, STRETCH_MODE, FALLBACK_MODE "
+ "%f %f %d %d",
+ playbackRate->mSpeed,
+ playbackRate->mPitch,
+ playbackRate->mStretchMode,
+ playbackRate->mFallbackMode);
+ // invalidate(); (should not require reconfigure)
}
- break;
+ } break;
default:
- if ((unsigned)param >= VOLUME0 && (unsigned)param < VOLUME0 + MAX_NUM_VOLUMES) {
- if (setVolumeRampVariables(*reinterpret_cast<float*>(value),
- target == RAMP_VOLUME ? mFrameCount : 0,
- &track->volume[param - VOLUME0],
- &track->prevVolume[param - VOLUME0],
- &track->volumeInc[param - VOLUME0],
- &track->mVolume[param - VOLUME0],
- &track->mPrevVolume[param - VOLUME0],
- &track->mVolumeInc[param - VOLUME0])) {
- ALOGV("setParameter(%s, VOLUME%d: %04x)",
- target == VOLUME ? "VOLUME" : "RAMP_VOLUME", param - VOLUME0,
- track->volume[param - VOLUME0]);
- invalidate();
- }
- } else {
- LOG_ALWAYS_FATAL("setParameter volume: bad param %d", param);
- }
+ LOG_ALWAYS_FATAL("setParameter timestretch: bad param %d", param);
}
break;
- case TIMESTRETCH:
- switch (param) {
- case PLAYBACK_RATE: {
- const AudioPlaybackRate *playbackRate =
- reinterpret_cast<AudioPlaybackRate*>(value);
- ALOGW_IF(!isAudioPlaybackRateValid(*playbackRate),
- "bad parameters speed %f, pitch %f",
- playbackRate->mSpeed, playbackRate->mPitch);
- if (track->setPlaybackRate(*playbackRate)) {
- ALOGV("setParameter(TIMESTRETCH, PLAYBACK_RATE, STRETCH_MODE, FALLBACK_MODE "
- "%f %f %d %d",
- playbackRate->mSpeed,
- playbackRate->mPitch,
- playbackRate->mStretchMode,
- playbackRate->mFallbackMode);
- // invalidate(); (should not require reconfigure)
- }
- } break;
- default:
- LOG_ALWAYS_FATAL("setParameter timestretch: bad param %d", param);
- }
- break;
default:
LOG_ALWAYS_FATAL("setParameter: bad target %d", target);
}
}
-bool AudioMixer::Track::setResampler(uint32_t trackSampleRate, uint32_t devSampleRate)
-{
- if (trackSampleRate != devSampleRate || mResampler.get() != nullptr) {
- if (sampleRate != trackSampleRate) {
- sampleRate = trackSampleRate;
- if (mResampler.get() == nullptr) {
- ALOGV("Creating resampler from track %d Hz to device %d Hz",
- trackSampleRate, devSampleRate);
- AudioResampler::src_quality quality;
- // force lowest quality level resampler if use case isn't music or video
- // FIXME this is flawed for dynamic sample rates, as we choose the resampler
- // quality level based on the initial ratio, but that could change later.
- // Should have a way to distinguish tracks with static ratios vs. dynamic ratios.
- if (isMusicRate(trackSampleRate)) {
- quality = AudioResampler::DEFAULT_QUALITY;
- } else {
- quality = AudioResampler::DYN_LOW_QUALITY;
- }
-
- // TODO: Remove MONO_HACK. Resampler sees #channels after the downmixer
- // but if none exists, it is the channel count (1 for mono).
- const int resamplerChannelCount = mDownmixerBufferProvider.get() != nullptr
- ? mMixerChannelCount : channelCount;
- ALOGVV("Creating resampler:"
- " format(%#x) channels(%d) devSampleRate(%u) quality(%d)\n",
- mMixerInFormat, resamplerChannelCount, devSampleRate, quality);
- mResampler.reset(AudioResampler::create(
- mMixerInFormat,
- resamplerChannelCount,
- devSampleRate, quality));
- }
- return true;
- }
- }
- return false;
-}
-
bool AudioMixer::Track::setPlaybackRate(const AudioPlaybackRate &playbackRate)
{
if ((mTimestretchBufferProvider.get() == nullptr &&
@@ -863,8 +478,7 @@
if (mTimestretchBufferProvider.get() == nullptr) {
// TODO: Remove MONO_HACK. Resampler sees #channels after the downmixer
// but if none exists, it is the channel count (1 for mono).
- const int timestretchChannelCount = mDownmixerBufferProvider.get() != nullptr
- ? mMixerChannelCount : channelCount;
+ const int timestretchChannelCount = getOutputChannelCount();
mTimestretchBufferProvider.reset(new TimestretchBufferProvider(timestretchChannelCount,
mMixerInFormat, sampleRate, playbackRate));
reconfigureBufferProviders();
@@ -875,84 +489,10 @@
return true;
}
-/* Checks to see if the volume ramp has completed and clears the increment
- * variables appropriately.
- *
- * FIXME: There is code to handle int/float ramp variable switchover should it not
- * complete within a mixer buffer processing call, but it is preferred to avoid switchover
- * due to precision issues. The switchover code is included for legacy code purposes
- * and can be removed once the integer volume is removed.
- *
- * It is not sufficient to clear only the volumeInc integer variable because
- * if one channel requires ramping, all channels are ramped.
- *
- * There is a bit of duplicated code here, but it keeps backward compatibility.
- */
-inline void AudioMixer::Track::adjustVolumeRamp(bool aux, bool useFloat)
-{
- if (useFloat) {
- for (uint32_t i = 0; i < MAX_NUM_VOLUMES; i++) {
- if ((mVolumeInc[i] > 0 && mPrevVolume[i] + mVolumeInc[i] >= mVolume[i]) ||
- (mVolumeInc[i] < 0 && mPrevVolume[i] + mVolumeInc[i] <= mVolume[i])) {
- volumeInc[i] = 0;
- prevVolume[i] = volume[i] << 16;
- mVolumeInc[i] = 0.;
- mPrevVolume[i] = mVolume[i];
- } else {
- //ALOGV("ramp: %f %f %f", mVolume[i], mPrevVolume[i], mVolumeInc[i]);
- prevVolume[i] = u4_28_from_float(mPrevVolume[i]);
- }
- }
- } else {
- for (uint32_t i = 0; i < MAX_NUM_VOLUMES; i++) {
- if (((volumeInc[i]>0) && (((prevVolume[i]+volumeInc[i])>>16) >= volume[i])) ||
- ((volumeInc[i]<0) && (((prevVolume[i]+volumeInc[i])>>16) <= volume[i]))) {
- volumeInc[i] = 0;
- prevVolume[i] = volume[i] << 16;
- mVolumeInc[i] = 0.;
- mPrevVolume[i] = mVolume[i];
- } else {
- //ALOGV("ramp: %d %d %d", volume[i] << 16, prevVolume[i], volumeInc[i]);
- mPrevVolume[i] = float_from_u4_28(prevVolume[i]);
- }
- }
- }
-
- if (aux) {
-#ifdef FLOAT_AUX
- if (useFloat) {
- if ((mAuxInc > 0.f && mPrevAuxLevel + mAuxInc >= mAuxLevel) ||
- (mAuxInc < 0.f && mPrevAuxLevel + mAuxInc <= mAuxLevel)) {
- auxInc = 0;
- prevAuxLevel = auxLevel << 16;
- mAuxInc = 0.f;
- mPrevAuxLevel = mAuxLevel;
- }
- } else
-#endif
- if ((auxInc > 0 && ((prevAuxLevel + auxInc) >> 16) >= auxLevel) ||
- (auxInc < 0 && ((prevAuxLevel + auxInc) >> 16) <= auxLevel)) {
- auxInc = 0;
- prevAuxLevel = auxLevel << 16;
- mAuxInc = 0.f;
- mPrevAuxLevel = mAuxLevel;
- }
- }
-}
-
-size_t AudioMixer::getUnreleasedFrames(int name) const
-{
- const auto it = mTracks.find(name);
- if (it != mTracks.end()) {
- return it->second->getUnreleasedFrames();
- }
- return 0;
-}
-
void AudioMixer::setBufferProvider(int name, AudioBufferProvider* bufferProvider)
{
LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
- const std::shared_ptr<Track> &track = mTracks[name];
+ const std::shared_ptr<Track> &track = getTrack(name);
if (track->mInputBufferProvider == bufferProvider) {
return; // don't reset any buffer providers if identical.
@@ -976,679 +516,6 @@
track->reconfigureBufferProviders();
}
-void AudioMixer::process__validate()
-{
- // TODO: fix all16BitsStereNoResample logic to
- // either properly handle muted tracks (it should ignore them)
- // or remove altogether as an obsolete optimization.
- bool all16BitsStereoNoResample = true;
- bool resampling = false;
- bool volumeRamp = false;
-
- mEnabled.clear();
- mGroups.clear();
- for (const auto &pair : mTracks) {
- const int name = pair.first;
- const std::shared_ptr<Track> &t = pair.second;
- if (!t->enabled) continue;
-
- mEnabled.emplace_back(name); // we add to mEnabled in order of name.
- mGroups[t->mainBuffer].emplace_back(name); // mGroups also in order of name.
-
- uint32_t n = 0;
- // FIXME can overflow (mask is only 3 bits)
- n |= NEEDS_CHANNEL_1 + t->channelCount - 1;
- if (t->doesResample()) {
- n |= NEEDS_RESAMPLE;
- }
- if (t->auxLevel != 0 && t->auxBuffer != NULL) {
- n |= NEEDS_AUX;
- }
-
- if (t->volumeInc[0]|t->volumeInc[1]) {
- volumeRamp = true;
- } else if (!t->doesResample() && t->volumeRL == 0) {
- n |= NEEDS_MUTE;
- }
- t->needs = n;
-
- if (n & NEEDS_MUTE) {
- t->hook = &Track::track__nop;
- } else {
- if (n & NEEDS_AUX) {
- all16BitsStereoNoResample = false;
- }
- if (n & NEEDS_RESAMPLE) {
- all16BitsStereoNoResample = false;
- resampling = true;
- t->hook = Track::getTrackHook(TRACKTYPE_RESAMPLE, t->mMixerChannelCount,
- t->mMixerInFormat, t->mMixerFormat);
- ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
- "Track %d needs downmix + resample", name);
- } else {
- if ((n & NEEDS_CHANNEL_COUNT__MASK) == NEEDS_CHANNEL_1){
- t->hook = Track::getTrackHook(
- (t->mMixerChannelMask == AUDIO_CHANNEL_OUT_STEREO // TODO: MONO_HACK
- && t->channelMask == AUDIO_CHANNEL_OUT_MONO)
- ? TRACKTYPE_NORESAMPLEMONO : TRACKTYPE_NORESAMPLE,
- t->mMixerChannelCount,
- t->mMixerInFormat, t->mMixerFormat);
- all16BitsStereoNoResample = false;
- }
- if ((n & NEEDS_CHANNEL_COUNT__MASK) >= NEEDS_CHANNEL_2){
- t->hook = Track::getTrackHook(TRACKTYPE_NORESAMPLE, t->mMixerChannelCount,
- t->mMixerInFormat, t->mMixerFormat);
- ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
- "Track %d needs downmix", name);
- }
- }
- }
- }
-
- // select the processing hooks
- mHook = &AudioMixer::process__nop;
- if (mEnabled.size() > 0) {
- if (resampling) {
- if (mOutputTemp.get() == nullptr) {
- mOutputTemp.reset(new int32_t[MAX_NUM_CHANNELS * mFrameCount]);
- }
- if (mResampleTemp.get() == nullptr) {
- mResampleTemp.reset(new int32_t[MAX_NUM_CHANNELS * mFrameCount]);
- }
- mHook = &AudioMixer::process__genericResampling;
- } else {
- // we keep temp arrays around.
- mHook = &AudioMixer::process__genericNoResampling;
- if (all16BitsStereoNoResample && !volumeRamp) {
- if (mEnabled.size() == 1) {
- const std::shared_ptr<Track> &t = mTracks[mEnabled[0]];
- if ((t->needs & NEEDS_MUTE) == 0) {
- // The check prevents a muted track from acquiring a process hook.
- //
- // This is dangerous if the track is MONO as that requires
- // special case handling due to implicit channel duplication.
- // Stereo or Multichannel should actually be fine here.
- mHook = getProcessHook(PROCESSTYPE_NORESAMPLEONETRACK,
- t->mMixerChannelCount, t->mMixerInFormat, t->mMixerFormat);
- }
- }
- }
- }
- }
-
- ALOGV("mixer configuration change: %zu "
- "all16BitsStereoNoResample=%d, resampling=%d, volumeRamp=%d",
- mEnabled.size(), all16BitsStereoNoResample, resampling, volumeRamp);
-
- process();
-
- // Now that the volume ramp has been done, set optimal state and
- // track hooks for subsequent mixer process
- if (mEnabled.size() > 0) {
- bool allMuted = true;
-
- for (const int name : mEnabled) {
- const std::shared_ptr<Track> &t = mTracks[name];
- if (!t->doesResample() && t->volumeRL == 0) {
- t->needs |= NEEDS_MUTE;
- t->hook = &Track::track__nop;
- } else {
- allMuted = false;
- }
- }
- if (allMuted) {
- mHook = &AudioMixer::process__nop;
- } else if (all16BitsStereoNoResample) {
- if (mEnabled.size() == 1) {
- //const int i = 31 - __builtin_clz(enabledTracks);
- const std::shared_ptr<Track> &t = mTracks[mEnabled[0]];
- // Muted single tracks handled by allMuted above.
- mHook = getProcessHook(PROCESSTYPE_NORESAMPLEONETRACK,
- t->mMixerChannelCount, t->mMixerInFormat, t->mMixerFormat);
- }
- }
- }
-}
-
-void AudioMixer::Track::track__genericResample(
- int32_t* out, size_t outFrameCount, int32_t* temp, int32_t* aux)
-{
- ALOGVV("track__genericResample\n");
- mResampler->setSampleRate(sampleRate);
-
- // ramp gain - resample to temp buffer and scale/mix in 2nd step
- if (aux != NULL) {
- // always resample with unity gain when sending to auxiliary buffer to be able
- // to apply send level after resampling
- mResampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
- memset(temp, 0, outFrameCount * mMixerChannelCount * sizeof(int32_t));
- mResampler->resample(temp, outFrameCount, bufferProvider);
- if (CC_UNLIKELY(volumeInc[0]|volumeInc[1]|auxInc)) {
- volumeRampStereo(out, outFrameCount, temp, aux);
- } else {
- volumeStereo(out, outFrameCount, temp, aux);
- }
- } else {
- if (CC_UNLIKELY(volumeInc[0]|volumeInc[1])) {
- mResampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
- memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t));
- mResampler->resample(temp, outFrameCount, bufferProvider);
- volumeRampStereo(out, outFrameCount, temp, aux);
- }
-
- // constant gain
- else {
- mResampler->setVolume(mVolume[0], mVolume[1]);
- mResampler->resample(out, outFrameCount, bufferProvider);
- }
- }
-}
-
-void AudioMixer::Track::track__nop(int32_t* out __unused,
- size_t outFrameCount __unused, int32_t* temp __unused, int32_t* aux __unused)
-{
-}
-
-void AudioMixer::Track::volumeRampStereo(
- int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
-{
- int32_t vl = prevVolume[0];
- int32_t vr = prevVolume[1];
- const int32_t vlInc = volumeInc[0];
- const int32_t vrInc = volumeInc[1];
-
- //ALOGD("[0] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
- // t, vlInc/65536.0f, vl/65536.0f, volume[0],
- // (vl + vlInc*frameCount)/65536.0f, frameCount);
-
- // ramp volume
- if (CC_UNLIKELY(aux != NULL)) {
- int32_t va = prevAuxLevel;
- const int32_t vaInc = auxInc;
- int32_t l;
- int32_t r;
-
- do {
- l = (*temp++ >> 12);
- r = (*temp++ >> 12);
- *out++ += (vl >> 16) * l;
- *out++ += (vr >> 16) * r;
- *aux++ += (va >> 17) * (l + r);
- vl += vlInc;
- vr += vrInc;
- va += vaInc;
- } while (--frameCount);
- prevAuxLevel = va;
- } else {
- do {
- *out++ += (vl >> 16) * (*temp++ >> 12);
- *out++ += (vr >> 16) * (*temp++ >> 12);
- vl += vlInc;
- vr += vrInc;
- } while (--frameCount);
- }
- prevVolume[0] = vl;
- prevVolume[1] = vr;
- adjustVolumeRamp(aux != NULL);
-}
-
-void AudioMixer::Track::volumeStereo(
- int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
-{
- const int16_t vl = volume[0];
- const int16_t vr = volume[1];
-
- if (CC_UNLIKELY(aux != NULL)) {
- const int16_t va = auxLevel;
- do {
- int16_t l = (int16_t)(*temp++ >> 12);
- int16_t r = (int16_t)(*temp++ >> 12);
- out[0] = mulAdd(l, vl, out[0]);
- int16_t a = (int16_t)(((int32_t)l + r) >> 1);
- out[1] = mulAdd(r, vr, out[1]);
- out += 2;
- aux[0] = mulAdd(a, va, aux[0]);
- aux++;
- } while (--frameCount);
- } else {
- do {
- int16_t l = (int16_t)(*temp++ >> 12);
- int16_t r = (int16_t)(*temp++ >> 12);
- out[0] = mulAdd(l, vl, out[0]);
- out[1] = mulAdd(r, vr, out[1]);
- out += 2;
- } while (--frameCount);
- }
-}
-
-void AudioMixer::Track::track__16BitsStereo(
- int32_t* out, size_t frameCount, int32_t* temp __unused, int32_t* aux)
-{
- ALOGVV("track__16BitsStereo\n");
- const int16_t *in = static_cast<const int16_t *>(mIn);
-
- if (CC_UNLIKELY(aux != NULL)) {
- int32_t l;
- int32_t r;
- // ramp gain
- if (CC_UNLIKELY(volumeInc[0]|volumeInc[1]|auxInc)) {
- int32_t vl = prevVolume[0];
- int32_t vr = prevVolume[1];
- int32_t va = prevAuxLevel;
- const int32_t vlInc = volumeInc[0];
- const int32_t vrInc = volumeInc[1];
- const int32_t vaInc = auxInc;
- // ALOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
- // t, vlInc/65536.0f, vl/65536.0f, volume[0],
- // (vl + vlInc*frameCount)/65536.0f, frameCount);
-
- do {
- l = (int32_t)*in++;
- r = (int32_t)*in++;
- *out++ += (vl >> 16) * l;
- *out++ += (vr >> 16) * r;
- *aux++ += (va >> 17) * (l + r);
- vl += vlInc;
- vr += vrInc;
- va += vaInc;
- } while (--frameCount);
-
- prevVolume[0] = vl;
- prevVolume[1] = vr;
- prevAuxLevel = va;
- adjustVolumeRamp(true);
- }
-
- // constant gain
- else {
- const uint32_t vrl = volumeRL;
- const int16_t va = (int16_t)auxLevel;
- do {
- uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
- int16_t a = (int16_t)(((int32_t)in[0] + in[1]) >> 1);
- in += 2;
- out[0] = mulAddRL(1, rl, vrl, out[0]);
- out[1] = mulAddRL(0, rl, vrl, out[1]);
- out += 2;
- aux[0] = mulAdd(a, va, aux[0]);
- aux++;
- } while (--frameCount);
- }
- } else {
- // ramp gain
- if (CC_UNLIKELY(volumeInc[0]|volumeInc[1])) {
- int32_t vl = prevVolume[0];
- int32_t vr = prevVolume[1];
- const int32_t vlInc = volumeInc[0];
- const int32_t vrInc = volumeInc[1];
-
- // ALOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
- // t, vlInc/65536.0f, vl/65536.0f, volume[0],
- // (vl + vlInc*frameCount)/65536.0f, frameCount);
-
- do {
- *out++ += (vl >> 16) * (int32_t) *in++;
- *out++ += (vr >> 16) * (int32_t) *in++;
- vl += vlInc;
- vr += vrInc;
- } while (--frameCount);
-
- prevVolume[0] = vl;
- prevVolume[1] = vr;
- adjustVolumeRamp(false);
- }
-
- // constant gain
- else {
- const uint32_t vrl = volumeRL;
- do {
- uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
- in += 2;
- out[0] = mulAddRL(1, rl, vrl, out[0]);
- out[1] = mulAddRL(0, rl, vrl, out[1]);
- out += 2;
- } while (--frameCount);
- }
- }
- mIn = in;
-}
-
-void AudioMixer::Track::track__16BitsMono(
- int32_t* out, size_t frameCount, int32_t* temp __unused, int32_t* aux)
-{
- ALOGVV("track__16BitsMono\n");
- const int16_t *in = static_cast<int16_t const *>(mIn);
-
- if (CC_UNLIKELY(aux != NULL)) {
- // ramp gain
- if (CC_UNLIKELY(volumeInc[0]|volumeInc[1]|auxInc)) {
- int32_t vl = prevVolume[0];
- int32_t vr = prevVolume[1];
- int32_t va = prevAuxLevel;
- const int32_t vlInc = volumeInc[0];
- const int32_t vrInc = volumeInc[1];
- const int32_t vaInc = auxInc;
-
- // ALOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
- // t, vlInc/65536.0f, vl/65536.0f, volume[0],
- // (vl + vlInc*frameCount)/65536.0f, frameCount);
-
- do {
- int32_t l = *in++;
- *out++ += (vl >> 16) * l;
- *out++ += (vr >> 16) * l;
- *aux++ += (va >> 16) * l;
- vl += vlInc;
- vr += vrInc;
- va += vaInc;
- } while (--frameCount);
-
- prevVolume[0] = vl;
- prevVolume[1] = vr;
- prevAuxLevel = va;
- adjustVolumeRamp(true);
- }
- // constant gain
- else {
- const int16_t vl = volume[0];
- const int16_t vr = volume[1];
- const int16_t va = (int16_t)auxLevel;
- do {
- int16_t l = *in++;
- out[0] = mulAdd(l, vl, out[0]);
- out[1] = mulAdd(l, vr, out[1]);
- out += 2;
- aux[0] = mulAdd(l, va, aux[0]);
- aux++;
- } while (--frameCount);
- }
- } else {
- // ramp gain
- if (CC_UNLIKELY(volumeInc[0]|volumeInc[1])) {
- int32_t vl = prevVolume[0];
- int32_t vr = prevVolume[1];
- const int32_t vlInc = volumeInc[0];
- const int32_t vrInc = volumeInc[1];
-
- // ALOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
- // t, vlInc/65536.0f, vl/65536.0f, volume[0],
- // (vl + vlInc*frameCount)/65536.0f, frameCount);
-
- do {
- int32_t l = *in++;
- *out++ += (vl >> 16) * l;
- *out++ += (vr >> 16) * l;
- vl += vlInc;
- vr += vrInc;
- } while (--frameCount);
-
- prevVolume[0] = vl;
- prevVolume[1] = vr;
- adjustVolumeRamp(false);
- }
- // constant gain
- else {
- const int16_t vl = volume[0];
- const int16_t vr = volume[1];
- do {
- int16_t l = *in++;
- out[0] = mulAdd(l, vl, out[0]);
- out[1] = mulAdd(l, vr, out[1]);
- out += 2;
- } while (--frameCount);
- }
- }
- mIn = in;
-}
-
-// no-op case
-void AudioMixer::process__nop()
-{
- ALOGVV("process__nop\n");
-
- for (const auto &pair : mGroups) {
- // process by group of tracks with same output buffer to
- // avoid multiple memset() on same buffer
- const auto &group = pair.second;
-
- const std::shared_ptr<Track> &t = mTracks[group[0]];
- memset(t->mainBuffer, 0,
- mFrameCount * audio_bytes_per_frame(
- t->mMixerChannelCount + t->mMixerHapticChannelCount, t->mMixerFormat));
-
- // now consume data
- for (const int name : group) {
- const std::shared_ptr<Track> &t = mTracks[name];
- size_t outFrames = mFrameCount;
- while (outFrames) {
- t->buffer.frameCount = outFrames;
- t->bufferProvider->getNextBuffer(&t->buffer);
- if (t->buffer.raw == NULL) break;
- outFrames -= t->buffer.frameCount;
- t->bufferProvider->releaseBuffer(&t->buffer);
- }
- }
- }
-}
-
-// generic code without resampling
-void AudioMixer::process__genericNoResampling()
-{
- ALOGVV("process__genericNoResampling\n");
- int32_t outTemp[BLOCKSIZE * MAX_NUM_CHANNELS] __attribute__((aligned(32)));
-
- for (const auto &pair : mGroups) {
- // process by group of tracks with same output main buffer to
- // avoid multiple memset() on same buffer
- const auto &group = pair.second;
-
- // acquire buffer
- for (const int name : group) {
- const std::shared_ptr<Track> &t = mTracks[name];
- t->buffer.frameCount = mFrameCount;
- t->bufferProvider->getNextBuffer(&t->buffer);
- t->frameCount = t->buffer.frameCount;
- t->mIn = t->buffer.raw;
- }
-
- int32_t *out = (int *)pair.first;
- size_t numFrames = 0;
- do {
- const size_t frameCount = std::min((size_t)BLOCKSIZE, mFrameCount - numFrames);
- memset(outTemp, 0, sizeof(outTemp));
- for (const int name : group) {
- const std::shared_ptr<Track> &t = mTracks[name];
- int32_t *aux = NULL;
- if (CC_UNLIKELY(t->needs & NEEDS_AUX)) {
- aux = t->auxBuffer + numFrames;
- }
- for (int outFrames = frameCount; outFrames > 0; ) {
- // t->in == nullptr can happen if the track was flushed just after having
- // been enabled for mixing.
- if (t->mIn == nullptr) {
- break;
- }
- size_t inFrames = (t->frameCount > outFrames)?outFrames:t->frameCount;
- if (inFrames > 0) {
- (t.get()->*t->hook)(
- outTemp + (frameCount - outFrames) * t->mMixerChannelCount,
- inFrames, mResampleTemp.get() /* naked ptr */, aux);
- t->frameCount -= inFrames;
- outFrames -= inFrames;
- if (CC_UNLIKELY(aux != NULL)) {
- aux += inFrames;
- }
- }
- if (t->frameCount == 0 && outFrames) {
- t->bufferProvider->releaseBuffer(&t->buffer);
- t->buffer.frameCount = (mFrameCount - numFrames) -
- (frameCount - outFrames);
- t->bufferProvider->getNextBuffer(&t->buffer);
- t->mIn = t->buffer.raw;
- if (t->mIn == nullptr) {
- break;
- }
- t->frameCount = t->buffer.frameCount;
- }
- }
- }
-
- const std::shared_ptr<Track> &t1 = mTracks[group[0]];
- convertMixerFormat(out, t1->mMixerFormat, outTemp, t1->mMixerInFormat,
- frameCount * t1->mMixerChannelCount);
- // TODO: fix ugly casting due to choice of out pointer type
- out = reinterpret_cast<int32_t*>((uint8_t*)out
- + frameCount * t1->mMixerChannelCount
- * audio_bytes_per_sample(t1->mMixerFormat));
- numFrames += frameCount;
- } while (numFrames < mFrameCount);
-
- // release each track's buffer
- for (const int name : group) {
- const std::shared_ptr<Track> &t = mTracks[name];
- t->bufferProvider->releaseBuffer(&t->buffer);
- }
- }
-}
-
-// generic code with resampling
-void AudioMixer::process__genericResampling()
-{
- ALOGVV("process__genericResampling\n");
- int32_t * const outTemp = mOutputTemp.get(); // naked ptr
- size_t numFrames = mFrameCount;
-
- for (const auto &pair : mGroups) {
- const auto &group = pair.second;
- const std::shared_ptr<Track> &t1 = mTracks[group[0]];
-
- // clear temp buffer
- memset(outTemp, 0, sizeof(*outTemp) * t1->mMixerChannelCount * mFrameCount);
- for (const int name : group) {
- const std::shared_ptr<Track> &t = mTracks[name];
- int32_t *aux = NULL;
- if (CC_UNLIKELY(t->needs & NEEDS_AUX)) {
- aux = t->auxBuffer;
- }
-
- // this is a little goofy, on the resampling case we don't
- // acquire/release the buffers because it's done by
- // the resampler.
- if (t->needs & NEEDS_RESAMPLE) {
- (t.get()->*t->hook)(outTemp, numFrames, mResampleTemp.get() /* naked ptr */, aux);
- } else {
-
- size_t outFrames = 0;
-
- while (outFrames < numFrames) {
- t->buffer.frameCount = numFrames - outFrames;
- t->bufferProvider->getNextBuffer(&t->buffer);
- t->mIn = t->buffer.raw;
- // t->mIn == nullptr can happen if the track was flushed just after having
- // been enabled for mixing.
- if (t->mIn == nullptr) break;
-
- (t.get()->*t->hook)(
- outTemp + outFrames * t->mMixerChannelCount, t->buffer.frameCount,
- mResampleTemp.get() /* naked ptr */,
- aux != nullptr ? aux + outFrames : nullptr);
- outFrames += t->buffer.frameCount;
-
- t->bufferProvider->releaseBuffer(&t->buffer);
- }
- }
- }
- convertMixerFormat(t1->mainBuffer, t1->mMixerFormat,
- outTemp, t1->mMixerInFormat, numFrames * t1->mMixerChannelCount);
- }
-}
-
-// one track, 16 bits stereo without resampling is the most common case
-void AudioMixer::process__oneTrack16BitsStereoNoResampling()
-{
- ALOGVV("process__oneTrack16BitsStereoNoResampling\n");
- LOG_ALWAYS_FATAL_IF(mEnabled.size() != 0,
- "%zu != 1 tracks enabled", mEnabled.size());
- const int name = mEnabled[0];
- const std::shared_ptr<Track> &t = mTracks[name];
-
- AudioBufferProvider::Buffer& b(t->buffer);
-
- int32_t* out = t->mainBuffer;
- float *fout = reinterpret_cast<float*>(out);
- size_t numFrames = mFrameCount;
-
- const int16_t vl = t->volume[0];
- const int16_t vr = t->volume[1];
- const uint32_t vrl = t->volumeRL;
- while (numFrames) {
- b.frameCount = numFrames;
- t->bufferProvider->getNextBuffer(&b);
- const int16_t *in = b.i16;
-
- // in == NULL can happen if the track was flushed just after having
- // been enabled for mixing.
- if (in == NULL || (((uintptr_t)in) & 3)) {
- if ( AUDIO_FORMAT_PCM_FLOAT == t->mMixerFormat ) {
- memset((char*)fout, 0, numFrames
- * t->mMixerChannelCount * audio_bytes_per_sample(t->mMixerFormat));
- } else {
- memset((char*)out, 0, numFrames
- * t->mMixerChannelCount * audio_bytes_per_sample(t->mMixerFormat));
- }
- ALOGE_IF((((uintptr_t)in) & 3),
- "process__oneTrack16BitsStereoNoResampling: misaligned buffer"
- " %p track %d, channels %d, needs %08x, volume %08x vfl %f vfr %f",
- in, name, t->channelCount, t->needs, vrl, t->mVolume[0], t->mVolume[1]);
- return;
- }
- size_t outFrames = b.frameCount;
-
- switch (t->mMixerFormat) {
- case AUDIO_FORMAT_PCM_FLOAT:
- do {
- uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
- in += 2;
- int32_t l = mulRL(1, rl, vrl);
- int32_t r = mulRL(0, rl, vrl);
- *fout++ = float_from_q4_27(l);
- *fout++ = float_from_q4_27(r);
- // Note: In case of later int16_t sink output,
- // conversion and clamping is done by memcpy_to_i16_from_float().
- } while (--outFrames);
- break;
- case AUDIO_FORMAT_PCM_16_BIT:
- if (CC_UNLIKELY(uint32_t(vl) > UNITY_GAIN_INT || uint32_t(vr) > UNITY_GAIN_INT)) {
- // volume is boosted, so we might need to clamp even though
- // we process only one track.
- do {
- uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
- in += 2;
- int32_t l = mulRL(1, rl, vrl) >> 12;
- int32_t r = mulRL(0, rl, vrl) >> 12;
- // clamping...
- l = clamp16(l);
- r = clamp16(r);
- *out++ = (r<<16) | (l & 0xFFFF);
- } while (--outFrames);
- } else {
- do {
- uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
- in += 2;
- int32_t l = mulRL(1, rl, vrl) >> 12;
- int32_t r = mulRL(0, rl, vrl) >> 12;
- *out++ = (r<<16) | (l & 0xFFFF);
- } while (--outFrames);
- }
- break;
- default:
- LOG_ALWAYS_FATAL("bad mixer format: %d", t->mMixerFormat);
- }
- numFrames -= b.frameCount;
- t->bufferProvider->releaseBuffer(&b);
- }
-}
-
/*static*/ pthread_once_t AudioMixer::sOnceControl = PTHREAD_ONCE_INIT;
/*static*/ void AudioMixer::sInitRoutine()
@@ -1656,211 +523,71 @@
DownmixerBufferProvider::init(); // for the downmixer
}
-/* TODO: consider whether this level of optimization is necessary.
- * Perhaps just stick with a single for loop.
- */
-
-// Needs to derive a compile time constant (constexpr). Could be targeted to go
-// to a MONOVOL mixtype based on MAX_NUM_VOLUMES, but that's an unnecessary complication.
-#define MIXTYPE_MONOVOL(mixtype) ((mixtype) == MIXTYPE_MULTI ? MIXTYPE_MULTI_MONOVOL : \
- (mixtype) == MIXTYPE_MULTI_SAVEONLY ? MIXTYPE_MULTI_SAVEONLY_MONOVOL : (mixtype))
-
-/* MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
- * TO: int32_t (Q4.27) or float
- * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
- * TA: int32_t (Q4.27) or float
- */
-template <int MIXTYPE,
- typename TO, typename TI, typename TV, typename TA, typename TAV>
-static void volumeRampMulti(uint32_t channels, TO* out, size_t frameCount,
- const TI* in, TA* aux, TV *vol, const TV *volinc, TAV *vola, TAV volainc)
+std::shared_ptr<AudioMixerBase::TrackBase> AudioMixer::preCreateTrack()
{
- switch (channels) {
- case 1:
- volumeRampMulti<MIXTYPE, 1>(out, frameCount, in, aux, vol, volinc, vola, volainc);
- break;
- case 2:
- volumeRampMulti<MIXTYPE, 2>(out, frameCount, in, aux, vol, volinc, vola, volainc);
- break;
- case 3:
- volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 3>(out,
- frameCount, in, aux, vol, volinc, vola, volainc);
- break;
- case 4:
- volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 4>(out,
- frameCount, in, aux, vol, volinc, vola, volainc);
- break;
- case 5:
- volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 5>(out,
- frameCount, in, aux, vol, volinc, vola, volainc);
- break;
- case 6:
- volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 6>(out,
- frameCount, in, aux, vol, volinc, vola, volainc);
- break;
- case 7:
- volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 7>(out,
- frameCount, in, aux, vol, volinc, vola, volainc);
- break;
- case 8:
- volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 8>(out,
- frameCount, in, aux, vol, volinc, vola, volainc);
- break;
- }
+ return std::make_shared<Track>();
}
-/* MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
- * TO: int32_t (Q4.27) or float
- * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
- * TA: int32_t (Q4.27) or float
- */
-template <int MIXTYPE,
- typename TO, typename TI, typename TV, typename TA, typename TAV>
-static void volumeMulti(uint32_t channels, TO* out, size_t frameCount,
- const TI* in, TA* aux, const TV *vol, TAV vola)
+status_t AudioMixer::postCreateTrack(TrackBase *track)
{
- switch (channels) {
- case 1:
- volumeMulti<MIXTYPE, 1>(out, frameCount, in, aux, vol, vola);
- break;
- case 2:
- volumeMulti<MIXTYPE, 2>(out, frameCount, in, aux, vol, vola);
- break;
- case 3:
- volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 3>(out, frameCount, in, aux, vol, vola);
- break;
- case 4:
- volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 4>(out, frameCount, in, aux, vol, vola);
- break;
- case 5:
- volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 5>(out, frameCount, in, aux, vol, vola);
- break;
- case 6:
- volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 6>(out, frameCount, in, aux, vol, vola);
- break;
- case 7:
- volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 7>(out, frameCount, in, aux, vol, vola);
- break;
- case 8:
- volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 8>(out, frameCount, in, aux, vol, vola);
- break;
+ Track* t = static_cast<Track*>(track);
+
+ audio_channel_mask_t channelMask = t->channelMask;
+ t->mHapticChannelMask = channelMask & AUDIO_CHANNEL_HAPTIC_ALL;
+ t->mHapticChannelCount = audio_channel_count_from_out_mask(t->mHapticChannelMask);
+ channelMask &= ~AUDIO_CHANNEL_HAPTIC_ALL;
+ t->channelCount = audio_channel_count_from_out_mask(channelMask);
+ ALOGV_IF(audio_channel_mask_get_bits(channelMask) != AUDIO_CHANNEL_OUT_STEREO,
+ "Non-stereo channel mask: %d\n", channelMask);
+ t->channelMask = channelMask;
+ t->mInputBufferProvider = NULL;
+ t->mDownmixRequiresFormat = AUDIO_FORMAT_INVALID; // no format required
+ t->mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT;
+ // haptic
+ t->mHapticPlaybackEnabled = false;
+ t->mHapticIntensity = HAPTIC_SCALE_NONE;
+ t->mMixerHapticChannelMask = AUDIO_CHANNEL_NONE;
+ t->mMixerHapticChannelCount = 0;
+ t->mAdjustInChannelCount = t->channelCount + t->mHapticChannelCount;
+ t->mAdjustOutChannelCount = t->channelCount + t->mMixerHapticChannelCount;
+ t->mAdjustNonDestructiveInChannelCount = t->mAdjustOutChannelCount;
+ t->mAdjustNonDestructiveOutChannelCount = t->channelCount;
+ t->mKeepContractedChannels = false;
+ // Check the downmixing (or upmixing) requirements.
+ status_t status = t->prepareForDownmix();
+ if (status != OK) {
+ ALOGE("AudioMixer::getTrackName invalid channelMask (%#x)", channelMask);
+ return BAD_VALUE;
}
+ // prepareForDownmix() may change mDownmixRequiresFormat
+ ALOGVV("mMixerFormat:%#x mMixerInFormat:%#x\n", t->mMixerFormat, t->mMixerInFormat);
+ t->prepareForReformat();
+ t->prepareForAdjustChannelsNonDestructive(mFrameCount);
+ t->prepareForAdjustChannels();
+ return OK;
}
-/* MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
- * USEFLOATVOL (set to true if float volume is used)
- * ADJUSTVOL (set to true if volume ramp parameters needs adjustment afterwards)
- * TO: int32_t (Q4.27) or float
- * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
- * TA: int32_t (Q4.27) or float
- */
-template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL,
- typename TO, typename TI, typename TA>
-void AudioMixer::Track::volumeMix(TO *out, size_t outFrames,
- const TI *in, TA *aux, bool ramp)
+void AudioMixer::preProcess()
{
- if (USEFLOATVOL) {
- if (ramp) {
- volumeRampMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
- mPrevVolume, mVolumeInc,
-#ifdef FLOAT_AUX
- &mPrevAuxLevel, mAuxInc
-#else
- &prevAuxLevel, auxInc
-#endif
- );
- if (ADJUSTVOL) {
- adjustVolumeRamp(aux != NULL, true);
- }
- } else {
- volumeMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
- mVolume,
-#ifdef FLOAT_AUX
- mAuxLevel
-#else
- auxLevel
-#endif
- );
- }
- } else {
- if (ramp) {
- volumeRampMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
- prevVolume, volumeInc, &prevAuxLevel, auxInc);
- if (ADJUSTVOL) {
- adjustVolumeRamp(aux != NULL);
- }
- } else {
- volumeMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
- volume, auxLevel);
+ for (const auto &pair : mTracks) {
+ // Clear contracted buffer before processing if contracted channels are saved
+ const std::shared_ptr<TrackBase> &tb = pair.second;
+ Track *t = static_cast<Track*>(tb.get());
+ if (t->mKeepContractedChannels) {
+ t->clearContractedBuffer();
}
}
}
-/* This process hook is called when there is a single track without
- * aux buffer, volume ramp, or resampling.
- * TODO: Update the hook selection: this can properly handle aux and ramp.
- *
- * MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
- * TO: int32_t (Q4.27) or float
- * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
- * TA: int32_t (Q4.27)
- */
-template <int MIXTYPE, typename TO, typename TI, typename TA>
-void AudioMixer::process__noResampleOneTrack()
+void AudioMixer::postProcess()
{
- ALOGVV("process__noResampleOneTrack\n");
- LOG_ALWAYS_FATAL_IF(mEnabled.size() != 1,
- "%zu != 1 tracks enabled", mEnabled.size());
- const std::shared_ptr<Track> &t = mTracks[mEnabled[0]];
- const uint32_t channels = t->mMixerChannelCount;
- TO* out = reinterpret_cast<TO*>(t->mainBuffer);
- TA* aux = reinterpret_cast<TA*>(t->auxBuffer);
- const bool ramp = t->needsRamp();
-
- for (size_t numFrames = mFrameCount; numFrames > 0; ) {
- AudioBufferProvider::Buffer& b(t->buffer);
- // get input buffer
- b.frameCount = numFrames;
- t->bufferProvider->getNextBuffer(&b);
- const TI *in = reinterpret_cast<TI*>(b.raw);
-
- // in == NULL can happen if the track was flushed just after having
- // been enabled for mixing.
- if (in == NULL || (((uintptr_t)in) & 3)) {
- memset(out, 0, numFrames
- * channels * audio_bytes_per_sample(t->mMixerFormat));
- ALOGE_IF((((uintptr_t)in) & 3), "process__noResampleOneTrack: bus error: "
- "buffer %p track %p, channels %d, needs %#x",
- in, &t, t->channelCount, t->needs);
- return;
- }
-
- const size_t outFrames = b.frameCount;
- t->volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL */, false /* ADJUSTVOL */> (
- out, outFrames, in, aux, ramp);
-
- out += outFrames * channels;
- if (aux != NULL) {
- aux += outFrames;
- }
- numFrames -= b.frameCount;
-
- // release buffer
- t->bufferProvider->releaseBuffer(&b);
- }
- if (ramp) {
- t->adjustVolumeRamp(aux != NULL, is_same<TI, float>::value);
- }
-}
-
-void AudioMixer::processHapticData()
-{
+ // Process haptic data.
// Need to keep consistent with VibrationEffect.scale(int, float, int)
for (const auto &pair : mGroups) {
// process by group of tracks with same output main buffer.
const auto &group = pair.second;
for (const int name : group) {
- const std::shared_ptr<Track> &t = mTracks[name];
+ const std::shared_ptr<Track> &t = getTrack(name);
if (t->mHapticPlaybackEnabled) {
size_t sampleCount = mFrameCount * t->mMixerHapticChannelCount;
float gamma = t->getHapticScaleGamma();
@@ -1887,225 +614,5 @@
}
}
-/* This track hook is called to do resampling then mixing,
- * pulling from the track's upstream AudioBufferProvider.
- *
- * MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
- * TO: int32_t (Q4.27) or float
- * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
- * TA: int32_t (Q4.27) or float
- */
-template <int MIXTYPE, typename TO, typename TI, typename TA>
-void AudioMixer::Track::track__Resample(TO* out, size_t outFrameCount, TO* temp, TA* aux)
-{
- ALOGVV("track__Resample\n");
- mResampler->setSampleRate(sampleRate);
- const bool ramp = needsRamp();
- if (ramp || aux != NULL) {
- // if ramp: resample with unity gain to temp buffer and scale/mix in 2nd step.
- // if aux != NULL: resample with unity gain to temp buffer then apply send level.
-
- mResampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
- memset(temp, 0, outFrameCount * mMixerChannelCount * sizeof(TO));
- mResampler->resample((int32_t*)temp, outFrameCount, bufferProvider);
-
- volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL */, true /* ADJUSTVOL */>(
- out, outFrameCount, temp, aux, ramp);
-
- } else { // constant volume gain
- mResampler->setVolume(mVolume[0], mVolume[1]);
- mResampler->resample((int32_t*)out, outFrameCount, bufferProvider);
- }
-}
-
-/* This track hook is called to mix a track, when no resampling is required.
- * The input buffer should be present in in.
- *
- * MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
- * TO: int32_t (Q4.27) or float
- * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
- * TA: int32_t (Q4.27) or float
- */
-template <int MIXTYPE, typename TO, typename TI, typename TA>
-void AudioMixer::Track::track__NoResample(TO* out, size_t frameCount, TO* temp __unused, TA* aux)
-{
- ALOGVV("track__NoResample\n");
- const TI *in = static_cast<const TI *>(mIn);
-
- volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL */, true /* ADJUSTVOL */>(
- out, frameCount, in, aux, needsRamp());
-
- // MIXTYPE_MONOEXPAND reads a single input channel and expands to NCHAN output channels.
- // MIXTYPE_MULTI reads NCHAN input channels and places to NCHAN output channels.
- in += (MIXTYPE == MIXTYPE_MONOEXPAND) ? frameCount : frameCount * mMixerChannelCount;
- mIn = in;
-}
-
-/* The Mixer engine generates either int32_t (Q4_27) or float data.
- * We use this function to convert the engine buffers
- * to the desired mixer output format, either int16_t (Q.15) or float.
- */
-/* static */
-void AudioMixer::convertMixerFormat(void *out, audio_format_t mixerOutFormat,
- void *in, audio_format_t mixerInFormat, size_t sampleCount)
-{
- switch (mixerInFormat) {
- case AUDIO_FORMAT_PCM_FLOAT:
- switch (mixerOutFormat) {
- case AUDIO_FORMAT_PCM_FLOAT:
- memcpy(out, in, sampleCount * sizeof(float)); // MEMCPY. TODO optimize out
- break;
- case AUDIO_FORMAT_PCM_16_BIT:
- memcpy_to_i16_from_float((int16_t*)out, (float*)in, sampleCount);
- break;
- default:
- LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
- break;
- }
- break;
- case AUDIO_FORMAT_PCM_16_BIT:
- switch (mixerOutFormat) {
- case AUDIO_FORMAT_PCM_FLOAT:
- memcpy_to_float_from_q4_27((float*)out, (const int32_t*)in, sampleCount);
- break;
- case AUDIO_FORMAT_PCM_16_BIT:
- memcpy_to_i16_from_q4_27((int16_t*)out, (const int32_t*)in, sampleCount);
- break;
- default:
- LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
- break;
- }
- break;
- default:
- LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
- break;
- }
-}
-
-/* Returns the proper track hook to use for mixing the track into the output buffer.
- */
-/* static */
-AudioMixer::hook_t AudioMixer::Track::getTrackHook(int trackType, uint32_t channelCount,
- audio_format_t mixerInFormat, audio_format_t mixerOutFormat __unused)
-{
- if (!kUseNewMixer && channelCount == FCC_2 && mixerInFormat == AUDIO_FORMAT_PCM_16_BIT) {
- switch (trackType) {
- case TRACKTYPE_NOP:
- return &Track::track__nop;
- case TRACKTYPE_RESAMPLE:
- return &Track::track__genericResample;
- case TRACKTYPE_NORESAMPLEMONO:
- return &Track::track__16BitsMono;
- case TRACKTYPE_NORESAMPLE:
- return &Track::track__16BitsStereo;
- default:
- LOG_ALWAYS_FATAL("bad trackType: %d", trackType);
- break;
- }
- }
- LOG_ALWAYS_FATAL_IF(channelCount > MAX_NUM_CHANNELS);
- switch (trackType) {
- case TRACKTYPE_NOP:
- return &Track::track__nop;
- case TRACKTYPE_RESAMPLE:
- switch (mixerInFormat) {
- case AUDIO_FORMAT_PCM_FLOAT:
- return (AudioMixer::hook_t) &Track::track__Resample<
- MIXTYPE_MULTI, float /*TO*/, float /*TI*/, TYPE_AUX>;
- case AUDIO_FORMAT_PCM_16_BIT:
- return (AudioMixer::hook_t) &Track::track__Resample<
- MIXTYPE_MULTI, int32_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
- default:
- LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
- break;
- }
- break;
- case TRACKTYPE_NORESAMPLEMONO:
- switch (mixerInFormat) {
- case AUDIO_FORMAT_PCM_FLOAT:
- return (AudioMixer::hook_t) &Track::track__NoResample<
- MIXTYPE_MONOEXPAND, float /*TO*/, float /*TI*/, TYPE_AUX>;
- case AUDIO_FORMAT_PCM_16_BIT:
- return (AudioMixer::hook_t) &Track::track__NoResample<
- MIXTYPE_MONOEXPAND, int32_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
- default:
- LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
- break;
- }
- break;
- case TRACKTYPE_NORESAMPLE:
- switch (mixerInFormat) {
- case AUDIO_FORMAT_PCM_FLOAT:
- return (AudioMixer::hook_t) &Track::track__NoResample<
- MIXTYPE_MULTI, float /*TO*/, float /*TI*/, TYPE_AUX>;
- case AUDIO_FORMAT_PCM_16_BIT:
- return (AudioMixer::hook_t) &Track::track__NoResample<
- MIXTYPE_MULTI, int32_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
- default:
- LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
- break;
- }
- break;
- default:
- LOG_ALWAYS_FATAL("bad trackType: %d", trackType);
- break;
- }
- return NULL;
-}
-
-/* Returns the proper process hook for mixing tracks. Currently works only for
- * PROCESSTYPE_NORESAMPLEONETRACK, a mix involving one track, no resampling.
- *
- * TODO: Due to the special mixing considerations of duplicating to
- * a stereo output track, the input track cannot be MONO. This should be
- * prevented by the caller.
- */
-/* static */
-AudioMixer::process_hook_t AudioMixer::getProcessHook(
- int processType, uint32_t channelCount,
- audio_format_t mixerInFormat, audio_format_t mixerOutFormat)
-{
- if (processType != PROCESSTYPE_NORESAMPLEONETRACK) { // Only NORESAMPLEONETRACK
- LOG_ALWAYS_FATAL("bad processType: %d", processType);
- return NULL;
- }
- if (!kUseNewMixer && channelCount == FCC_2 && mixerInFormat == AUDIO_FORMAT_PCM_16_BIT) {
- return &AudioMixer::process__oneTrack16BitsStereoNoResampling;
- }
- LOG_ALWAYS_FATAL_IF(channelCount > MAX_NUM_CHANNELS);
- switch (mixerInFormat) {
- case AUDIO_FORMAT_PCM_FLOAT:
- switch (mixerOutFormat) {
- case AUDIO_FORMAT_PCM_FLOAT:
- return &AudioMixer::process__noResampleOneTrack<
- MIXTYPE_MULTI_SAVEONLY, float /*TO*/, float /*TI*/, TYPE_AUX>;
- case AUDIO_FORMAT_PCM_16_BIT:
- return &AudioMixer::process__noResampleOneTrack<
- MIXTYPE_MULTI_SAVEONLY, int16_t /*TO*/, float /*TI*/, TYPE_AUX>;
- default:
- LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
- break;
- }
- break;
- case AUDIO_FORMAT_PCM_16_BIT:
- switch (mixerOutFormat) {
- case AUDIO_FORMAT_PCM_FLOAT:
- return &AudioMixer::process__noResampleOneTrack<
- MIXTYPE_MULTI_SAVEONLY, float /*TO*/, int16_t /*TI*/, TYPE_AUX>;
- case AUDIO_FORMAT_PCM_16_BIT:
- return &AudioMixer::process__noResampleOneTrack<
- MIXTYPE_MULTI_SAVEONLY, int16_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
- default:
- LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
- break;
- }
- break;
- default:
- LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
- break;
- }
- return NULL;
-}
-
// ----------------------------------------------------------------------------
} // namespace android
diff --git a/media/libaudioprocessing/AudioMixerBase.cpp b/media/libaudioprocessing/AudioMixerBase.cpp
new file mode 100644
index 0000000..75c077d
--- /dev/null
+++ b/media/libaudioprocessing/AudioMixerBase.cpp
@@ -0,0 +1,1692 @@
+/*
+**
+** Copyright 2019, The Android Open Source Project
+**
+** Licensed under the Apache License, Version 2.0 (the "License");
+** you may not use this file except in compliance with the License.
+** You may obtain a copy of the License at
+**
+** http://www.apache.org/licenses/LICENSE-2.0
+**
+** Unless required by applicable law or agreed to in writing, software
+** distributed under the License is distributed on an "AS IS" BASIS,
+** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+** See the License for the specific language governing permissions and
+** limitations under the License.
+*/
+
+#define LOG_TAG "AudioMixer"
+//#define LOG_NDEBUG 0
+
+#include <sstream>
+#include <string.h>
+
+#include <audio_utils/primitives.h>
+#include <cutils/compiler.h>
+#include <media/AudioMixerBase.h>
+#include <utils/Log.h>
+
+#include "AudioMixerOps.h"
+
+// The FCC_2 macro refers to the Fixed Channel Count of 2 for the legacy integer mixer.
+#ifndef FCC_2
+#define FCC_2 2
+#endif
+
+// Look for MONO_HACK for any Mono hack involving legacy mono channel to
+// stereo channel conversion.
+
+/* VERY_VERY_VERBOSE_LOGGING will show exactly which process hook and track hook is
+ * being used. This is a considerable amount of log spam, so don't enable unless you
+ * are verifying the hook based code.
+ */
+//#define VERY_VERY_VERBOSE_LOGGING
+#ifdef VERY_VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+//define ALOGVV printf // for test-mixer.cpp
+#else
+#define ALOGVV(a...) do { } while (0)
+#endif
+
+// TODO: remove BLOCKSIZE unit of processing - it isn't needed anymore.
+static constexpr int BLOCKSIZE = 16;
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+bool AudioMixerBase::isValidFormat(audio_format_t format) const
+{
+ switch (format) {
+ case AUDIO_FORMAT_PCM_8_BIT:
+ case AUDIO_FORMAT_PCM_16_BIT:
+ case AUDIO_FORMAT_PCM_24_BIT_PACKED:
+ case AUDIO_FORMAT_PCM_32_BIT:
+ case AUDIO_FORMAT_PCM_FLOAT:
+ return true;
+ default:
+ return false;
+ }
+}
+
+bool AudioMixerBase::isValidChannelMask(audio_channel_mask_t channelMask) const
+{
+ return audio_channel_count_from_out_mask(channelMask) <= MAX_NUM_CHANNELS;
+}
+
+std::shared_ptr<AudioMixerBase::TrackBase> AudioMixerBase::preCreateTrack()
+{
+ return std::make_shared<TrackBase>();
+}
+
+status_t AudioMixerBase::create(
+ int name, audio_channel_mask_t channelMask, audio_format_t format, int sessionId)
+{
+ LOG_ALWAYS_FATAL_IF(exists(name), "name %d already exists", name);
+
+ if (!isValidChannelMask(channelMask)) {
+ ALOGE("%s invalid channelMask: %#x", __func__, channelMask);
+ return BAD_VALUE;
+ }
+ if (!isValidFormat(format)) {
+ ALOGE("%s invalid format: %#x", __func__, format);
+ return BAD_VALUE;
+ }
+
+ auto t = preCreateTrack();
+ {
+ // TODO: move initialization to the Track constructor.
+ // assume default parameters for the track, except where noted below
+ t->needs = 0;
+
+ // Integer volume.
+ // Currently integer volume is kept for the legacy integer mixer.
+ // Will be removed when the legacy mixer path is removed.
+ t->volume[0] = 0;
+ t->volume[1] = 0;
+ t->prevVolume[0] = 0 << 16;
+ t->prevVolume[1] = 0 << 16;
+ t->volumeInc[0] = 0;
+ t->volumeInc[1] = 0;
+ t->auxLevel = 0;
+ t->auxInc = 0;
+ t->prevAuxLevel = 0;
+
+ // Floating point volume.
+ t->mVolume[0] = 0.f;
+ t->mVolume[1] = 0.f;
+ t->mPrevVolume[0] = 0.f;
+ t->mPrevVolume[1] = 0.f;
+ t->mVolumeInc[0] = 0.;
+ t->mVolumeInc[1] = 0.;
+ t->mAuxLevel = 0.;
+ t->mAuxInc = 0.;
+ t->mPrevAuxLevel = 0.;
+
+ // no initialization needed
+ // t->frameCount
+ t->channelCount = audio_channel_count_from_out_mask(channelMask);
+ t->enabled = false;
+ ALOGV_IF(audio_channel_mask_get_bits(channelMask) != AUDIO_CHANNEL_OUT_STEREO,
+ "Non-stereo channel mask: %d\n", channelMask);
+ t->channelMask = channelMask;
+ t->sessionId = sessionId;
+ // setBufferProvider(name, AudioBufferProvider *) is required before enable(name)
+ t->bufferProvider = NULL;
+ t->buffer.raw = NULL;
+ // no initialization needed
+ // t->buffer.frameCount
+ t->hook = NULL;
+ t->mIn = NULL;
+ t->sampleRate = mSampleRate;
+ // setParameter(name, TRACK, MAIN_BUFFER, mixBuffer) is required before enable(name)
+ t->mainBuffer = NULL;
+ t->auxBuffer = NULL;
+ t->mMixerFormat = AUDIO_FORMAT_PCM_16_BIT;
+ t->mFormat = format;
+ t->mMixerInFormat = kUseFloat && kUseNewMixer ?
+ AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
+ t->mMixerChannelMask = audio_channel_mask_from_representation_and_bits(
+ AUDIO_CHANNEL_REPRESENTATION_POSITION, AUDIO_CHANNEL_OUT_STEREO);
+ t->mMixerChannelCount = audio_channel_count_from_out_mask(t->mMixerChannelMask);
+ status_t status = postCreateTrack(t.get());
+ if (status != OK) return status;
+ mTracks[name] = t;
+ return OK;
+ }
+}
+
+// Called when channel masks have changed for a track name
+bool AudioMixerBase::setChannelMasks(int name,
+ audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask)
+{
+ LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
+ const std::shared_ptr<TrackBase> &track = mTracks[name];
+
+ if (trackChannelMask == track->channelMask && mixerChannelMask == track->mMixerChannelMask) {
+ return false; // no need to change
+ }
+ // always recompute for both channel masks even if only one has changed.
+ const uint32_t trackChannelCount = audio_channel_count_from_out_mask(trackChannelMask);
+ const uint32_t mixerChannelCount = audio_channel_count_from_out_mask(mixerChannelMask);
+
+ ALOG_ASSERT(trackChannelCount && mixerChannelCount);
+ track->channelMask = trackChannelMask;
+ track->channelCount = trackChannelCount;
+ track->mMixerChannelMask = mixerChannelMask;
+ track->mMixerChannelCount = mixerChannelCount;
+
+ // Resampler channels may have changed.
+ track->recreateResampler(mSampleRate);
+ return true;
+}
+
+void AudioMixerBase::destroy(int name)
+{
+ LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
+ ALOGV("deleteTrackName(%d)", name);
+
+ if (mTracks[name]->enabled) {
+ invalidate();
+ }
+ mTracks.erase(name); // deallocate track
+}
+
+void AudioMixerBase::enable(int name)
+{
+ LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
+ const std::shared_ptr<TrackBase> &track = mTracks[name];
+
+ if (!track->enabled) {
+ track->enabled = true;
+ ALOGV("enable(%d)", name);
+ invalidate();
+ }
+}
+
+void AudioMixerBase::disable(int name)
+{
+ LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
+ const std::shared_ptr<TrackBase> &track = mTracks[name];
+
+ if (track->enabled) {
+ track->enabled = false;
+ ALOGV("disable(%d)", name);
+ invalidate();
+ }
+}
+
+/* Sets the volume ramp variables for the AudioMixer.
+ *
+ * The volume ramp variables are used to transition from the previous
+ * volume to the set volume. ramp controls the duration of the transition.
+ * Its value is typically one state framecount period, but may also be 0,
+ * meaning "immediate."
+ *
+ * FIXME: 1) Volume ramp is enabled only if there is a nonzero integer increment
+ * even if there is a nonzero floating point increment (in that case, the volume
+ * change is immediate). This restriction should be changed when the legacy mixer
+ * is removed (see #2).
+ * FIXME: 2) Integer volume variables are used for Legacy mixing and should be removed
+ * when no longer needed.
+ *
+ * @param newVolume set volume target in floating point [0.0, 1.0].
+ * @param ramp number of frames to increment over. if ramp is 0, the volume
+ * should be set immediately. Currently ramp should not exceed 65535 (frames).
+ * @param pIntSetVolume pointer to the U4.12 integer target volume, set on return.
+ * @param pIntPrevVolume pointer to the U4.28 integer previous volume, set on return.
+ * @param pIntVolumeInc pointer to the U4.28 increment per output audio frame, set on return.
+ * @param pSetVolume pointer to the float target volume, set on return.
+ * @param pPrevVolume pointer to the float previous volume, set on return.
+ * @param pVolumeInc pointer to the float increment per output audio frame, set on return.
+ * @return true if the volume has changed, false if volume is same.
+ */
+static inline bool setVolumeRampVariables(float newVolume, int32_t ramp,
+ int16_t *pIntSetVolume, int32_t *pIntPrevVolume, int32_t *pIntVolumeInc,
+ float *pSetVolume, float *pPrevVolume, float *pVolumeInc) {
+ // check floating point volume to see if it is identical to the previously
+ // set volume.
+ // We do not use a tolerance here (and reject changes too small)
+ // as it may be confusing to use a different value than the one set.
+ // If the resulting volume is too small to ramp, it is a direct set of the volume.
+ if (newVolume == *pSetVolume) {
+ return false;
+ }
+ if (newVolume < 0) {
+ newVolume = 0; // should not have negative volumes
+ } else {
+ switch (fpclassify(newVolume)) {
+ case FP_SUBNORMAL:
+ case FP_NAN:
+ newVolume = 0;
+ break;
+ case FP_ZERO:
+ break; // zero volume is fine
+ case FP_INFINITE:
+ // Infinite volume could be handled consistently since
+ // floating point math saturates at infinities,
+ // but we limit volume to unity gain float.
+ // ramp = 0; break;
+ //
+ newVolume = AudioMixerBase::UNITY_GAIN_FLOAT;
+ break;
+ case FP_NORMAL:
+ default:
+ // Floating point does not have problems with overflow wrap
+ // that integer has. However, we limit the volume to
+ // unity gain here.
+ // TODO: Revisit the volume limitation and perhaps parameterize.
+ if (newVolume > AudioMixerBase::UNITY_GAIN_FLOAT) {
+ newVolume = AudioMixerBase::UNITY_GAIN_FLOAT;
+ }
+ break;
+ }
+ }
+
+ // set floating point volume ramp
+ if (ramp != 0) {
+ // when the ramp completes, *pPrevVolume is set to *pSetVolume, so there
+ // is no computational mismatch; hence equality is checked here.
+ ALOGD_IF(*pPrevVolume != *pSetVolume, "previous float ramp hasn't finished,"
+ " prev:%f set_to:%f", *pPrevVolume, *pSetVolume);
+ const float inc = (newVolume - *pPrevVolume) / ramp; // could be inf, nan, subnormal
+ // could be inf, cannot be nan, subnormal
+ const float maxv = std::max(newVolume, *pPrevVolume);
+
+ if (isnormal(inc) // inc must be a normal number (no subnormals, infinite, nan)
+ && maxv + inc != maxv) { // inc must make forward progress
+ *pVolumeInc = inc;
+ // ramp is set now.
+ // Note: if newVolume is 0, then near the end of the ramp,
+ // it may be possible that the ramped volume may be subnormal or
+ // temporarily negative by a small amount or subnormal due to floating
+ // point inaccuracies.
+ } else {
+ ramp = 0; // ramp not allowed
+ }
+ }
+
+ // compute and check integer volume, no need to check negative values
+ // The integer volume is limited to "unity_gain" to avoid wrapping and other
+ // audio artifacts, so it never reaches the range limit of U4.28.
+ // We safely use signed 16 and 32 bit integers here.
+ const float scaledVolume = newVolume * AudioMixerBase::UNITY_GAIN_INT; // not neg, subnormal, nan
+ const int32_t intVolume = (scaledVolume >= (float)AudioMixerBase::UNITY_GAIN_INT) ?
+ AudioMixerBase::UNITY_GAIN_INT : (int32_t)scaledVolume;
+
+ // set integer volume ramp
+ if (ramp != 0) {
+ // integer volume is U4.12 (to use 16 bit multiplies), but ramping uses U4.28.
+ // when the ramp completes, *pIntPrevVolume is set to *pIntSetVolume << 16, so there
+ // is no computational mismatch; hence equality is checked here.
+ ALOGD_IF(*pIntPrevVolume != *pIntSetVolume << 16, "previous int ramp hasn't finished,"
+ " prev:%d set_to:%d", *pIntPrevVolume, *pIntSetVolume << 16);
+ const int32_t inc = ((intVolume << 16) - *pIntPrevVolume) / ramp;
+
+ if (inc != 0) { // inc must make forward progress
+ *pIntVolumeInc = inc;
+ } else {
+ ramp = 0; // ramp not allowed
+ }
+ }
+
+ // if no ramp, or ramp not allowed, then clear float and integer increments
+ if (ramp == 0) {
+ *pVolumeInc = 0;
+ *pPrevVolume = newVolume;
+ *pIntVolumeInc = 0;
+ *pIntPrevVolume = intVolume << 16;
+ }
+ *pSetVolume = newVolume;
+ *pIntSetVolume = intVolume;
+ return true;
+}
+
+void AudioMixerBase::setParameter(int name, int target, int param, void *value)
+{
+ LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
+ const std::shared_ptr<TrackBase> &track = mTracks[name];
+
+ int valueInt = static_cast<int>(reinterpret_cast<uintptr_t>(value));
+ int32_t *valueBuf = reinterpret_cast<int32_t*>(value);
+
+ switch (target) {
+
+ case TRACK:
+ switch (param) {
+ case CHANNEL_MASK: {
+ const audio_channel_mask_t trackChannelMask =
+ static_cast<audio_channel_mask_t>(valueInt);
+ if (setChannelMasks(name, trackChannelMask, track->mMixerChannelMask)) {
+ ALOGV("setParameter(TRACK, CHANNEL_MASK, %x)", trackChannelMask);
+ invalidate();
+ }
+ } break;
+ case MAIN_BUFFER:
+ if (track->mainBuffer != valueBuf) {
+ track->mainBuffer = valueBuf;
+ ALOGV("setParameter(TRACK, MAIN_BUFFER, %p)", valueBuf);
+ invalidate();
+ }
+ break;
+ case AUX_BUFFER:
+ if (track->auxBuffer != valueBuf) {
+ track->auxBuffer = valueBuf;
+ ALOGV("setParameter(TRACK, AUX_BUFFER, %p)", valueBuf);
+ invalidate();
+ }
+ break;
+ case FORMAT: {
+ audio_format_t format = static_cast<audio_format_t>(valueInt);
+ if (track->mFormat != format) {
+ ALOG_ASSERT(audio_is_linear_pcm(format), "Invalid format %#x", format);
+ track->mFormat = format;
+ ALOGV("setParameter(TRACK, FORMAT, %#x)", format);
+ invalidate();
+ }
+ } break;
+ case MIXER_FORMAT: {
+ audio_format_t format = static_cast<audio_format_t>(valueInt);
+ if (track->mMixerFormat != format) {
+ track->mMixerFormat = format;
+ ALOGV("setParameter(TRACK, MIXER_FORMAT, %#x)", format);
+ }
+ } break;
+ case MIXER_CHANNEL_MASK: {
+ const audio_channel_mask_t mixerChannelMask =
+ static_cast<audio_channel_mask_t>(valueInt);
+ if (setChannelMasks(name, track->channelMask, mixerChannelMask)) {
+ ALOGV("setParameter(TRACK, MIXER_CHANNEL_MASK, %#x)", mixerChannelMask);
+ invalidate();
+ }
+ } break;
+ default:
+ LOG_ALWAYS_FATAL("setParameter track: bad param %d", param);
+ }
+ break;
+
+ case RESAMPLE:
+ switch (param) {
+ case SAMPLE_RATE:
+ ALOG_ASSERT(valueInt > 0, "bad sample rate %d", valueInt);
+ if (track->setResampler(uint32_t(valueInt), mSampleRate)) {
+ ALOGV("setParameter(RESAMPLE, SAMPLE_RATE, %u)",
+ uint32_t(valueInt));
+ invalidate();
+ }
+ break;
+ case RESET:
+ track->resetResampler();
+ invalidate();
+ break;
+ case REMOVE:
+ track->mResampler.reset(nullptr);
+ track->sampleRate = mSampleRate;
+ invalidate();
+ break;
+ default:
+ LOG_ALWAYS_FATAL("setParameter resample: bad param %d", param);
+ }
+ break;
+
+ case RAMP_VOLUME:
+ case VOLUME:
+ switch (param) {
+ case AUXLEVEL:
+ if (setVolumeRampVariables(*reinterpret_cast<float*>(value),
+ target == RAMP_VOLUME ? mFrameCount : 0,
+ &track->auxLevel, &track->prevAuxLevel, &track->auxInc,
+ &track->mAuxLevel, &track->mPrevAuxLevel, &track->mAuxInc)) {
+ ALOGV("setParameter(%s, AUXLEVEL: %04x)",
+ target == VOLUME ? "VOLUME" : "RAMP_VOLUME", track->auxLevel);
+ invalidate();
+ }
+ break;
+ default:
+ if ((unsigned)param >= VOLUME0 && (unsigned)param < VOLUME0 + MAX_NUM_VOLUMES) {
+ if (setVolumeRampVariables(*reinterpret_cast<float*>(value),
+ target == RAMP_VOLUME ? mFrameCount : 0,
+ &track->volume[param - VOLUME0],
+ &track->prevVolume[param - VOLUME0],
+ &track->volumeInc[param - VOLUME0],
+ &track->mVolume[param - VOLUME0],
+ &track->mPrevVolume[param - VOLUME0],
+ &track->mVolumeInc[param - VOLUME0])) {
+ ALOGV("setParameter(%s, VOLUME%d: %04x)",
+ target == VOLUME ? "VOLUME" : "RAMP_VOLUME", param - VOLUME0,
+ track->volume[param - VOLUME0]);
+ invalidate();
+ }
+ } else {
+ LOG_ALWAYS_FATAL("setParameter volume: bad param %d", param);
+ }
+ }
+ break;
+
+ default:
+ LOG_ALWAYS_FATAL("setParameter: bad target %d", target);
+ }
+}
+
+bool AudioMixerBase::TrackBase::setResampler(uint32_t trackSampleRate, uint32_t devSampleRate)
+{
+ if (trackSampleRate != devSampleRate || mResampler.get() != nullptr) {
+ if (sampleRate != trackSampleRate) {
+ sampleRate = trackSampleRate;
+ if (mResampler.get() == nullptr) {
+ ALOGV("Creating resampler from track %d Hz to device %d Hz",
+ trackSampleRate, devSampleRate);
+ AudioResampler::src_quality quality;
+ // force lowest quality level resampler if use case isn't music or video
+ // FIXME this is flawed for dynamic sample rates, as we choose the resampler
+ // quality level based on the initial ratio, but that could change later.
+ // Should have a way to distinguish tracks with static ratios vs. dynamic ratios.
+ if (isMusicRate(trackSampleRate)) {
+ quality = AudioResampler::DEFAULT_QUALITY;
+ } else {
+ quality = AudioResampler::DYN_LOW_QUALITY;
+ }
+
+ // TODO: Remove MONO_HACK. Resampler sees #channels after the downmixer
+ // but if none exists, it is the channel count (1 for mono).
+ const int resamplerChannelCount = getOutputChannelCount();
+ ALOGVV("Creating resampler:"
+ " format(%#x) channels(%d) devSampleRate(%u) quality(%d)\n",
+ mMixerInFormat, resamplerChannelCount, devSampleRate, quality);
+ mResampler.reset(AudioResampler::create(
+ mMixerInFormat,
+ resamplerChannelCount,
+ devSampleRate, quality));
+ }
+ return true;
+ }
+ }
+ return false;
+}
+
+/* Checks to see if the volume ramp has completed and clears the increment
+ * variables appropriately.
+ *
+ * FIXME: There is code to handle int/float ramp variable switchover should it not
+ * complete within a mixer buffer processing call, but it is preferred to avoid switchover
+ * due to precision issues. The switchover code is included for legacy code purposes
+ * and can be removed once the integer volume is removed.
+ *
+ * It is not sufficient to clear only the volumeInc integer variable because
+ * if one channel requires ramping, all channels are ramped.
+ *
+ * There is a bit of duplicated code here, but it keeps backward compatibility.
+ */
+void AudioMixerBase::TrackBase::adjustVolumeRamp(bool aux, bool useFloat)
+{
+ if (useFloat) {
+ for (uint32_t i = 0; i < MAX_NUM_VOLUMES; i++) {
+ if ((mVolumeInc[i] > 0 && mPrevVolume[i] + mVolumeInc[i] >= mVolume[i]) ||
+ (mVolumeInc[i] < 0 && mPrevVolume[i] + mVolumeInc[i] <= mVolume[i])) {
+ volumeInc[i] = 0;
+ prevVolume[i] = volume[i] << 16;
+ mVolumeInc[i] = 0.;
+ mPrevVolume[i] = mVolume[i];
+ } else {
+ //ALOGV("ramp: %f %f %f", mVolume[i], mPrevVolume[i], mVolumeInc[i]);
+ prevVolume[i] = u4_28_from_float(mPrevVolume[i]);
+ }
+ }
+ } else {
+ for (uint32_t i = 0; i < MAX_NUM_VOLUMES; i++) {
+ if (((volumeInc[i]>0) && (((prevVolume[i]+volumeInc[i])>>16) >= volume[i])) ||
+ ((volumeInc[i]<0) && (((prevVolume[i]+volumeInc[i])>>16) <= volume[i]))) {
+ volumeInc[i] = 0;
+ prevVolume[i] = volume[i] << 16;
+ mVolumeInc[i] = 0.;
+ mPrevVolume[i] = mVolume[i];
+ } else {
+ //ALOGV("ramp: %d %d %d", volume[i] << 16, prevVolume[i], volumeInc[i]);
+ mPrevVolume[i] = float_from_u4_28(prevVolume[i]);
+ }
+ }
+ }
+
+ if (aux) {
+#ifdef FLOAT_AUX
+ if (useFloat) {
+ if ((mAuxInc > 0.f && mPrevAuxLevel + mAuxInc >= mAuxLevel) ||
+ (mAuxInc < 0.f && mPrevAuxLevel + mAuxInc <= mAuxLevel)) {
+ auxInc = 0;
+ prevAuxLevel = auxLevel << 16;
+ mAuxInc = 0.f;
+ mPrevAuxLevel = mAuxLevel;
+ }
+ } else
+#endif
+ if ((auxInc > 0 && ((prevAuxLevel + auxInc) >> 16) >= auxLevel) ||
+ (auxInc < 0 && ((prevAuxLevel + auxInc) >> 16) <= auxLevel)) {
+ auxInc = 0;
+ prevAuxLevel = auxLevel << 16;
+ mAuxInc = 0.f;
+ mPrevAuxLevel = mAuxLevel;
+ }
+ }
+}
+
+void AudioMixerBase::TrackBase::recreateResampler(uint32_t devSampleRate)
+{
+ if (mResampler.get() != nullptr) {
+ const uint32_t resetToSampleRate = sampleRate;
+ mResampler.reset(nullptr);
+ sampleRate = devSampleRate; // without resampler, track rate is device sample rate.
+ // recreate the resampler with updated format, channels, saved sampleRate.
+ setResampler(resetToSampleRate /*trackSampleRate*/, devSampleRate);
+ }
+}
+
+size_t AudioMixerBase::getUnreleasedFrames(int name) const
+{
+ const auto it = mTracks.find(name);
+ if (it != mTracks.end()) {
+ return it->second->getUnreleasedFrames();
+ }
+ return 0;
+}
+
+std::string AudioMixerBase::trackNames() const
+{
+ std::stringstream ss;
+ for (const auto &pair : mTracks) {
+ ss << pair.first << " ";
+ }
+ return ss.str();
+}
+
+void AudioMixerBase::process__validate()
+{
+ // TODO: fix all16BitsStereNoResample logic to
+ // either properly handle muted tracks (it should ignore them)
+ // or remove altogether as an obsolete optimization.
+ bool all16BitsStereoNoResample = true;
+ bool resampling = false;
+ bool volumeRamp = false;
+
+ mEnabled.clear();
+ mGroups.clear();
+ for (const auto &pair : mTracks) {
+ const int name = pair.first;
+ const std::shared_ptr<TrackBase> &t = pair.second;
+ if (!t->enabled) continue;
+
+ mEnabled.emplace_back(name); // we add to mEnabled in order of name.
+ mGroups[t->mainBuffer].emplace_back(name); // mGroups also in order of name.
+
+ uint32_t n = 0;
+ // FIXME can overflow (mask is only 3 bits)
+ n |= NEEDS_CHANNEL_1 + t->channelCount - 1;
+ if (t->doesResample()) {
+ n |= NEEDS_RESAMPLE;
+ }
+ if (t->auxLevel != 0 && t->auxBuffer != NULL) {
+ n |= NEEDS_AUX;
+ }
+
+ if (t->volumeInc[0]|t->volumeInc[1]) {
+ volumeRamp = true;
+ } else if (!t->doesResample() && t->volumeRL == 0) {
+ n |= NEEDS_MUTE;
+ }
+ t->needs = n;
+
+ if (n & NEEDS_MUTE) {
+ t->hook = &TrackBase::track__nop;
+ } else {
+ if (n & NEEDS_AUX) {
+ all16BitsStereoNoResample = false;
+ }
+ if (n & NEEDS_RESAMPLE) {
+ all16BitsStereoNoResample = false;
+ resampling = true;
+ t->hook = TrackBase::getTrackHook(TRACKTYPE_RESAMPLE, t->mMixerChannelCount,
+ t->mMixerInFormat, t->mMixerFormat);
+ ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
+ "Track %d needs downmix + resample", name);
+ } else {
+ if ((n & NEEDS_CHANNEL_COUNT__MASK) == NEEDS_CHANNEL_1){
+ t->hook = TrackBase::getTrackHook(
+ (t->mMixerChannelMask == AUDIO_CHANNEL_OUT_STEREO // TODO: MONO_HACK
+ && t->channelMask == AUDIO_CHANNEL_OUT_MONO)
+ ? TRACKTYPE_NORESAMPLEMONO : TRACKTYPE_NORESAMPLE,
+ t->mMixerChannelCount,
+ t->mMixerInFormat, t->mMixerFormat);
+ all16BitsStereoNoResample = false;
+ }
+ if ((n & NEEDS_CHANNEL_COUNT__MASK) >= NEEDS_CHANNEL_2){
+ t->hook = TrackBase::getTrackHook(TRACKTYPE_NORESAMPLE, t->mMixerChannelCount,
+ t->mMixerInFormat, t->mMixerFormat);
+ ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
+ "Track %d needs downmix", name);
+ }
+ }
+ }
+ }
+
+ // select the processing hooks
+ mHook = &AudioMixerBase::process__nop;
+ if (mEnabled.size() > 0) {
+ if (resampling) {
+ if (mOutputTemp.get() == nullptr) {
+ mOutputTemp.reset(new int32_t[MAX_NUM_CHANNELS * mFrameCount]);
+ }
+ if (mResampleTemp.get() == nullptr) {
+ mResampleTemp.reset(new int32_t[MAX_NUM_CHANNELS * mFrameCount]);
+ }
+ mHook = &AudioMixerBase::process__genericResampling;
+ } else {
+ // we keep temp arrays around.
+ mHook = &AudioMixerBase::process__genericNoResampling;
+ if (all16BitsStereoNoResample && !volumeRamp) {
+ if (mEnabled.size() == 1) {
+ const std::shared_ptr<TrackBase> &t = mTracks[mEnabled[0]];
+ if ((t->needs & NEEDS_MUTE) == 0) {
+ // The check prevents a muted track from acquiring a process hook.
+ //
+ // This is dangerous if the track is MONO as that requires
+ // special case handling due to implicit channel duplication.
+ // Stereo or Multichannel should actually be fine here.
+ mHook = getProcessHook(PROCESSTYPE_NORESAMPLEONETRACK,
+ t->mMixerChannelCount, t->mMixerInFormat, t->mMixerFormat);
+ }
+ }
+ }
+ }
+ }
+
+ ALOGV("mixer configuration change: %zu "
+ "all16BitsStereoNoResample=%d, resampling=%d, volumeRamp=%d",
+ mEnabled.size(), all16BitsStereoNoResample, resampling, volumeRamp);
+
+ process();
+
+ // Now that the volume ramp has been done, set optimal state and
+ // track hooks for subsequent mixer process
+ if (mEnabled.size() > 0) {
+ bool allMuted = true;
+
+ for (const int name : mEnabled) {
+ const std::shared_ptr<TrackBase> &t = mTracks[name];
+ if (!t->doesResample() && t->volumeRL == 0) {
+ t->needs |= NEEDS_MUTE;
+ t->hook = &TrackBase::track__nop;
+ } else {
+ allMuted = false;
+ }
+ }
+ if (allMuted) {
+ mHook = &AudioMixerBase::process__nop;
+ } else if (all16BitsStereoNoResample) {
+ if (mEnabled.size() == 1) {
+ //const int i = 31 - __builtin_clz(enabledTracks);
+ const std::shared_ptr<TrackBase> &t = mTracks[mEnabled[0]];
+ // Muted single tracks handled by allMuted above.
+ mHook = getProcessHook(PROCESSTYPE_NORESAMPLEONETRACK,
+ t->mMixerChannelCount, t->mMixerInFormat, t->mMixerFormat);
+ }
+ }
+ }
+}
+
+void AudioMixerBase::TrackBase::track__genericResample(
+ int32_t* out, size_t outFrameCount, int32_t* temp, int32_t* aux)
+{
+ ALOGVV("track__genericResample\n");
+ mResampler->setSampleRate(sampleRate);
+
+ // ramp gain - resample to temp buffer and scale/mix in 2nd step
+ if (aux != NULL) {
+ // always resample with unity gain when sending to auxiliary buffer to be able
+ // to apply send level after resampling
+ mResampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
+ memset(temp, 0, outFrameCount * mMixerChannelCount * sizeof(int32_t));
+ mResampler->resample(temp, outFrameCount, bufferProvider);
+ if (CC_UNLIKELY(volumeInc[0]|volumeInc[1]|auxInc)) {
+ volumeRampStereo(out, outFrameCount, temp, aux);
+ } else {
+ volumeStereo(out, outFrameCount, temp, aux);
+ }
+ } else {
+ if (CC_UNLIKELY(volumeInc[0]|volumeInc[1])) {
+ mResampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
+ memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t));
+ mResampler->resample(temp, outFrameCount, bufferProvider);
+ volumeRampStereo(out, outFrameCount, temp, aux);
+ }
+
+ // constant gain
+ else {
+ mResampler->setVolume(mVolume[0], mVolume[1]);
+ mResampler->resample(out, outFrameCount, bufferProvider);
+ }
+ }
+}
+
+void AudioMixerBase::TrackBase::track__nop(int32_t* out __unused,
+ size_t outFrameCount __unused, int32_t* temp __unused, int32_t* aux __unused)
+{
+}
+
+void AudioMixerBase::TrackBase::volumeRampStereo(
+ int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
+{
+ int32_t vl = prevVolume[0];
+ int32_t vr = prevVolume[1];
+ const int32_t vlInc = volumeInc[0];
+ const int32_t vrInc = volumeInc[1];
+
+ //ALOGD("[0] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
+ // t, vlInc/65536.0f, vl/65536.0f, volume[0],
+ // (vl + vlInc*frameCount)/65536.0f, frameCount);
+
+ // ramp volume
+ if (CC_UNLIKELY(aux != NULL)) {
+ int32_t va = prevAuxLevel;
+ const int32_t vaInc = auxInc;
+ int32_t l;
+ int32_t r;
+
+ do {
+ l = (*temp++ >> 12);
+ r = (*temp++ >> 12);
+ *out++ += (vl >> 16) * l;
+ *out++ += (vr >> 16) * r;
+ *aux++ += (va >> 17) * (l + r);
+ vl += vlInc;
+ vr += vrInc;
+ va += vaInc;
+ } while (--frameCount);
+ prevAuxLevel = va;
+ } else {
+ do {
+ *out++ += (vl >> 16) * (*temp++ >> 12);
+ *out++ += (vr >> 16) * (*temp++ >> 12);
+ vl += vlInc;
+ vr += vrInc;
+ } while (--frameCount);
+ }
+ prevVolume[0] = vl;
+ prevVolume[1] = vr;
+ adjustVolumeRamp(aux != NULL);
+}
+
+void AudioMixerBase::TrackBase::volumeStereo(
+ int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
+{
+ const int16_t vl = volume[0];
+ const int16_t vr = volume[1];
+
+ if (CC_UNLIKELY(aux != NULL)) {
+ const int16_t va = auxLevel;
+ do {
+ int16_t l = (int16_t)(*temp++ >> 12);
+ int16_t r = (int16_t)(*temp++ >> 12);
+ out[0] = mulAdd(l, vl, out[0]);
+ int16_t a = (int16_t)(((int32_t)l + r) >> 1);
+ out[1] = mulAdd(r, vr, out[1]);
+ out += 2;
+ aux[0] = mulAdd(a, va, aux[0]);
+ aux++;
+ } while (--frameCount);
+ } else {
+ do {
+ int16_t l = (int16_t)(*temp++ >> 12);
+ int16_t r = (int16_t)(*temp++ >> 12);
+ out[0] = mulAdd(l, vl, out[0]);
+ out[1] = mulAdd(r, vr, out[1]);
+ out += 2;
+ } while (--frameCount);
+ }
+}
+
+void AudioMixerBase::TrackBase::track__16BitsStereo(
+ int32_t* out, size_t frameCount, int32_t* temp __unused, int32_t* aux)
+{
+ ALOGVV("track__16BitsStereo\n");
+ const int16_t *in = static_cast<const int16_t *>(mIn);
+
+ if (CC_UNLIKELY(aux != NULL)) {
+ int32_t l;
+ int32_t r;
+ // ramp gain
+ if (CC_UNLIKELY(volumeInc[0]|volumeInc[1]|auxInc)) {
+ int32_t vl = prevVolume[0];
+ int32_t vr = prevVolume[1];
+ int32_t va = prevAuxLevel;
+ const int32_t vlInc = volumeInc[0];
+ const int32_t vrInc = volumeInc[1];
+ const int32_t vaInc = auxInc;
+ // ALOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
+ // t, vlInc/65536.0f, vl/65536.0f, volume[0],
+ // (vl + vlInc*frameCount)/65536.0f, frameCount);
+
+ do {
+ l = (int32_t)*in++;
+ r = (int32_t)*in++;
+ *out++ += (vl >> 16) * l;
+ *out++ += (vr >> 16) * r;
+ *aux++ += (va >> 17) * (l + r);
+ vl += vlInc;
+ vr += vrInc;
+ va += vaInc;
+ } while (--frameCount);
+
+ prevVolume[0] = vl;
+ prevVolume[1] = vr;
+ prevAuxLevel = va;
+ adjustVolumeRamp(true);
+ }
+
+ // constant gain
+ else {
+ const uint32_t vrl = volumeRL;
+ const int16_t va = (int16_t)auxLevel;
+ do {
+ uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
+ int16_t a = (int16_t)(((int32_t)in[0] + in[1]) >> 1);
+ in += 2;
+ out[0] = mulAddRL(1, rl, vrl, out[0]);
+ out[1] = mulAddRL(0, rl, vrl, out[1]);
+ out += 2;
+ aux[0] = mulAdd(a, va, aux[0]);
+ aux++;
+ } while (--frameCount);
+ }
+ } else {
+ // ramp gain
+ if (CC_UNLIKELY(volumeInc[0]|volumeInc[1])) {
+ int32_t vl = prevVolume[0];
+ int32_t vr = prevVolume[1];
+ const int32_t vlInc = volumeInc[0];
+ const int32_t vrInc = volumeInc[1];
+
+ // ALOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
+ // t, vlInc/65536.0f, vl/65536.0f, volume[0],
+ // (vl + vlInc*frameCount)/65536.0f, frameCount);
+
+ do {
+ *out++ += (vl >> 16) * (int32_t) *in++;
+ *out++ += (vr >> 16) * (int32_t) *in++;
+ vl += vlInc;
+ vr += vrInc;
+ } while (--frameCount);
+
+ prevVolume[0] = vl;
+ prevVolume[1] = vr;
+ adjustVolumeRamp(false);
+ }
+
+ // constant gain
+ else {
+ const uint32_t vrl = volumeRL;
+ do {
+ uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
+ in += 2;
+ out[0] = mulAddRL(1, rl, vrl, out[0]);
+ out[1] = mulAddRL(0, rl, vrl, out[1]);
+ out += 2;
+ } while (--frameCount);
+ }
+ }
+ mIn = in;
+}
+
+void AudioMixerBase::TrackBase::track__16BitsMono(
+ int32_t* out, size_t frameCount, int32_t* temp __unused, int32_t* aux)
+{
+ ALOGVV("track__16BitsMono\n");
+ const int16_t *in = static_cast<int16_t const *>(mIn);
+
+ if (CC_UNLIKELY(aux != NULL)) {
+ // ramp gain
+ if (CC_UNLIKELY(volumeInc[0]|volumeInc[1]|auxInc)) {
+ int32_t vl = prevVolume[0];
+ int32_t vr = prevVolume[1];
+ int32_t va = prevAuxLevel;
+ const int32_t vlInc = volumeInc[0];
+ const int32_t vrInc = volumeInc[1];
+ const int32_t vaInc = auxInc;
+
+ // ALOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
+ // t, vlInc/65536.0f, vl/65536.0f, volume[0],
+ // (vl + vlInc*frameCount)/65536.0f, frameCount);
+
+ do {
+ int32_t l = *in++;
+ *out++ += (vl >> 16) * l;
+ *out++ += (vr >> 16) * l;
+ *aux++ += (va >> 16) * l;
+ vl += vlInc;
+ vr += vrInc;
+ va += vaInc;
+ } while (--frameCount);
+
+ prevVolume[0] = vl;
+ prevVolume[1] = vr;
+ prevAuxLevel = va;
+ adjustVolumeRamp(true);
+ }
+ // constant gain
+ else {
+ const int16_t vl = volume[0];
+ const int16_t vr = volume[1];
+ const int16_t va = (int16_t)auxLevel;
+ do {
+ int16_t l = *in++;
+ out[0] = mulAdd(l, vl, out[0]);
+ out[1] = mulAdd(l, vr, out[1]);
+ out += 2;
+ aux[0] = mulAdd(l, va, aux[0]);
+ aux++;
+ } while (--frameCount);
+ }
+ } else {
+ // ramp gain
+ if (CC_UNLIKELY(volumeInc[0]|volumeInc[1])) {
+ int32_t vl = prevVolume[0];
+ int32_t vr = prevVolume[1];
+ const int32_t vlInc = volumeInc[0];
+ const int32_t vrInc = volumeInc[1];
+
+ // ALOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
+ // t, vlInc/65536.0f, vl/65536.0f, volume[0],
+ // (vl + vlInc*frameCount)/65536.0f, frameCount);
+
+ do {
+ int32_t l = *in++;
+ *out++ += (vl >> 16) * l;
+ *out++ += (vr >> 16) * l;
+ vl += vlInc;
+ vr += vrInc;
+ } while (--frameCount);
+
+ prevVolume[0] = vl;
+ prevVolume[1] = vr;
+ adjustVolumeRamp(false);
+ }
+ // constant gain
+ else {
+ const int16_t vl = volume[0];
+ const int16_t vr = volume[1];
+ do {
+ int16_t l = *in++;
+ out[0] = mulAdd(l, vl, out[0]);
+ out[1] = mulAdd(l, vr, out[1]);
+ out += 2;
+ } while (--frameCount);
+ }
+ }
+ mIn = in;
+}
+
+// no-op case
+void AudioMixerBase::process__nop()
+{
+ ALOGVV("process__nop\n");
+
+ for (const auto &pair : mGroups) {
+ // process by group of tracks with same output buffer to
+ // avoid multiple memset() on same buffer
+ const auto &group = pair.second;
+
+ const std::shared_ptr<TrackBase> &t = mTracks[group[0]];
+ memset(t->mainBuffer, 0,
+ mFrameCount * audio_bytes_per_frame(t->getMixerChannelCount(), t->mMixerFormat));
+
+ // now consume data
+ for (const int name : group) {
+ const std::shared_ptr<TrackBase> &t = mTracks[name];
+ size_t outFrames = mFrameCount;
+ while (outFrames) {
+ t->buffer.frameCount = outFrames;
+ t->bufferProvider->getNextBuffer(&t->buffer);
+ if (t->buffer.raw == NULL) break;
+ outFrames -= t->buffer.frameCount;
+ t->bufferProvider->releaseBuffer(&t->buffer);
+ }
+ }
+ }
+}
+
+// generic code without resampling
+void AudioMixerBase::process__genericNoResampling()
+{
+ ALOGVV("process__genericNoResampling\n");
+ int32_t outTemp[BLOCKSIZE * MAX_NUM_CHANNELS] __attribute__((aligned(32)));
+
+ for (const auto &pair : mGroups) {
+ // process by group of tracks with same output main buffer to
+ // avoid multiple memset() on same buffer
+ const auto &group = pair.second;
+
+ // acquire buffer
+ for (const int name : group) {
+ const std::shared_ptr<TrackBase> &t = mTracks[name];
+ t->buffer.frameCount = mFrameCount;
+ t->bufferProvider->getNextBuffer(&t->buffer);
+ t->frameCount = t->buffer.frameCount;
+ t->mIn = t->buffer.raw;
+ }
+
+ int32_t *out = (int *)pair.first;
+ size_t numFrames = 0;
+ do {
+ const size_t frameCount = std::min((size_t)BLOCKSIZE, mFrameCount - numFrames);
+ memset(outTemp, 0, sizeof(outTemp));
+ for (const int name : group) {
+ const std::shared_ptr<TrackBase> &t = mTracks[name];
+ int32_t *aux = NULL;
+ if (CC_UNLIKELY(t->needs & NEEDS_AUX)) {
+ aux = t->auxBuffer + numFrames;
+ }
+ for (int outFrames = frameCount; outFrames > 0; ) {
+ // t->in == nullptr can happen if the track was flushed just after having
+ // been enabled for mixing.
+ if (t->mIn == nullptr) {
+ break;
+ }
+ size_t inFrames = (t->frameCount > outFrames)?outFrames:t->frameCount;
+ if (inFrames > 0) {
+ (t.get()->*t->hook)(
+ outTemp + (frameCount - outFrames) * t->mMixerChannelCount,
+ inFrames, mResampleTemp.get() /* naked ptr */, aux);
+ t->frameCount -= inFrames;
+ outFrames -= inFrames;
+ if (CC_UNLIKELY(aux != NULL)) {
+ aux += inFrames;
+ }
+ }
+ if (t->frameCount == 0 && outFrames) {
+ t->bufferProvider->releaseBuffer(&t->buffer);
+ t->buffer.frameCount = (mFrameCount - numFrames) -
+ (frameCount - outFrames);
+ t->bufferProvider->getNextBuffer(&t->buffer);
+ t->mIn = t->buffer.raw;
+ if (t->mIn == nullptr) {
+ break;
+ }
+ t->frameCount = t->buffer.frameCount;
+ }
+ }
+ }
+
+ const std::shared_ptr<TrackBase> &t1 = mTracks[group[0]];
+ convertMixerFormat(out, t1->mMixerFormat, outTemp, t1->mMixerInFormat,
+ frameCount * t1->mMixerChannelCount);
+ // TODO: fix ugly casting due to choice of out pointer type
+ out = reinterpret_cast<int32_t*>((uint8_t*)out
+ + frameCount * t1->mMixerChannelCount
+ * audio_bytes_per_sample(t1->mMixerFormat));
+ numFrames += frameCount;
+ } while (numFrames < mFrameCount);
+
+ // release each track's buffer
+ for (const int name : group) {
+ const std::shared_ptr<TrackBase> &t = mTracks[name];
+ t->bufferProvider->releaseBuffer(&t->buffer);
+ }
+ }
+}
+
+// generic code with resampling
+void AudioMixerBase::process__genericResampling()
+{
+ ALOGVV("process__genericResampling\n");
+ int32_t * const outTemp = mOutputTemp.get(); // naked ptr
+ size_t numFrames = mFrameCount;
+
+ for (const auto &pair : mGroups) {
+ const auto &group = pair.second;
+ const std::shared_ptr<TrackBase> &t1 = mTracks[group[0]];
+
+ // clear temp buffer
+ memset(outTemp, 0, sizeof(*outTemp) * t1->mMixerChannelCount * mFrameCount);
+ for (const int name : group) {
+ const std::shared_ptr<TrackBase> &t = mTracks[name];
+ int32_t *aux = NULL;
+ if (CC_UNLIKELY(t->needs & NEEDS_AUX)) {
+ aux = t->auxBuffer;
+ }
+
+ // this is a little goofy, on the resampling case we don't
+ // acquire/release the buffers because it's done by
+ // the resampler.
+ if (t->needs & NEEDS_RESAMPLE) {
+ (t.get()->*t->hook)(outTemp, numFrames, mResampleTemp.get() /* naked ptr */, aux);
+ } else {
+
+ size_t outFrames = 0;
+
+ while (outFrames < numFrames) {
+ t->buffer.frameCount = numFrames - outFrames;
+ t->bufferProvider->getNextBuffer(&t->buffer);
+ t->mIn = t->buffer.raw;
+ // t->mIn == nullptr can happen if the track was flushed just after having
+ // been enabled for mixing.
+ if (t->mIn == nullptr) break;
+
+ (t.get()->*t->hook)(
+ outTemp + outFrames * t->mMixerChannelCount, t->buffer.frameCount,
+ mResampleTemp.get() /* naked ptr */,
+ aux != nullptr ? aux + outFrames : nullptr);
+ outFrames += t->buffer.frameCount;
+
+ t->bufferProvider->releaseBuffer(&t->buffer);
+ }
+ }
+ }
+ convertMixerFormat(t1->mainBuffer, t1->mMixerFormat,
+ outTemp, t1->mMixerInFormat, numFrames * t1->mMixerChannelCount);
+ }
+}
+
+// one track, 16 bits stereo without resampling is the most common case
+void AudioMixerBase::process__oneTrack16BitsStereoNoResampling()
+{
+ ALOGVV("process__oneTrack16BitsStereoNoResampling\n");
+ LOG_ALWAYS_FATAL_IF(mEnabled.size() != 0,
+ "%zu != 1 tracks enabled", mEnabled.size());
+ const int name = mEnabled[0];
+ const std::shared_ptr<TrackBase> &t = mTracks[name];
+
+ AudioBufferProvider::Buffer& b(t->buffer);
+
+ int32_t* out = t->mainBuffer;
+ float *fout = reinterpret_cast<float*>(out);
+ size_t numFrames = mFrameCount;
+
+ const int16_t vl = t->volume[0];
+ const int16_t vr = t->volume[1];
+ const uint32_t vrl = t->volumeRL;
+ while (numFrames) {
+ b.frameCount = numFrames;
+ t->bufferProvider->getNextBuffer(&b);
+ const int16_t *in = b.i16;
+
+ // in == NULL can happen if the track was flushed just after having
+ // been enabled for mixing.
+ if (in == NULL || (((uintptr_t)in) & 3)) {
+ if ( AUDIO_FORMAT_PCM_FLOAT == t->mMixerFormat ) {
+ memset((char*)fout, 0, numFrames
+ * t->mMixerChannelCount * audio_bytes_per_sample(t->mMixerFormat));
+ } else {
+ memset((char*)out, 0, numFrames
+ * t->mMixerChannelCount * audio_bytes_per_sample(t->mMixerFormat));
+ }
+ ALOGE_IF((((uintptr_t)in) & 3),
+ "process__oneTrack16BitsStereoNoResampling: misaligned buffer"
+ " %p track %d, channels %d, needs %08x, volume %08x vfl %f vfr %f",
+ in, name, t->channelCount, t->needs, vrl, t->mVolume[0], t->mVolume[1]);
+ return;
+ }
+ size_t outFrames = b.frameCount;
+
+ switch (t->mMixerFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ do {
+ uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
+ in += 2;
+ int32_t l = mulRL(1, rl, vrl);
+ int32_t r = mulRL(0, rl, vrl);
+ *fout++ = float_from_q4_27(l);
+ *fout++ = float_from_q4_27(r);
+ // Note: In case of later int16_t sink output,
+ // conversion and clamping is done by memcpy_to_i16_from_float().
+ } while (--outFrames);
+ break;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ if (CC_UNLIKELY(uint32_t(vl) > UNITY_GAIN_INT || uint32_t(vr) > UNITY_GAIN_INT)) {
+ // volume is boosted, so we might need to clamp even though
+ // we process only one track.
+ do {
+ uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
+ in += 2;
+ int32_t l = mulRL(1, rl, vrl) >> 12;
+ int32_t r = mulRL(0, rl, vrl) >> 12;
+ // clamping...
+ l = clamp16(l);
+ r = clamp16(r);
+ *out++ = (r<<16) | (l & 0xFFFF);
+ } while (--outFrames);
+ } else {
+ do {
+ uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
+ in += 2;
+ int32_t l = mulRL(1, rl, vrl) >> 12;
+ int32_t r = mulRL(0, rl, vrl) >> 12;
+ *out++ = (r<<16) | (l & 0xFFFF);
+ } while (--outFrames);
+ }
+ break;
+ default:
+ LOG_ALWAYS_FATAL("bad mixer format: %d", t->mMixerFormat);
+ }
+ numFrames -= b.frameCount;
+ t->bufferProvider->releaseBuffer(&b);
+ }
+}
+
+/* TODO: consider whether this level of optimization is necessary.
+ * Perhaps just stick with a single for loop.
+ */
+
+// Needs to derive a compile time constant (constexpr). Could be targeted to go
+// to a MONOVOL mixtype based on MAX_NUM_VOLUMES, but that's an unnecessary complication.
+#define MIXTYPE_MONOVOL(mixtype) ((mixtype) == MIXTYPE_MULTI ? MIXTYPE_MULTI_MONOVOL : \
+ (mixtype) == MIXTYPE_MULTI_SAVEONLY ? MIXTYPE_MULTI_SAVEONLY_MONOVOL : (mixtype))
+
+/* MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
+ * TO: int32_t (Q4.27) or float
+ * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
+ * TA: int32_t (Q4.27) or float
+ */
+template <int MIXTYPE,
+ typename TO, typename TI, typename TV, typename TA, typename TAV>
+static void volumeRampMulti(uint32_t channels, TO* out, size_t frameCount,
+ const TI* in, TA* aux, TV *vol, const TV *volinc, TAV *vola, TAV volainc)
+{
+ switch (channels) {
+ case 1:
+ volumeRampMulti<MIXTYPE, 1>(out, frameCount, in, aux, vol, volinc, vola, volainc);
+ break;
+ case 2:
+ volumeRampMulti<MIXTYPE, 2>(out, frameCount, in, aux, vol, volinc, vola, volainc);
+ break;
+ case 3:
+ volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 3>(out,
+ frameCount, in, aux, vol, volinc, vola, volainc);
+ break;
+ case 4:
+ volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 4>(out,
+ frameCount, in, aux, vol, volinc, vola, volainc);
+ break;
+ case 5:
+ volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 5>(out,
+ frameCount, in, aux, vol, volinc, vola, volainc);
+ break;
+ case 6:
+ volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 6>(out,
+ frameCount, in, aux, vol, volinc, vola, volainc);
+ break;
+ case 7:
+ volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 7>(out,
+ frameCount, in, aux, vol, volinc, vola, volainc);
+ break;
+ case 8:
+ volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 8>(out,
+ frameCount, in, aux, vol, volinc, vola, volainc);
+ break;
+ }
+}
+
+/* MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
+ * TO: int32_t (Q4.27) or float
+ * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
+ * TA: int32_t (Q4.27) or float
+ */
+template <int MIXTYPE,
+ typename TO, typename TI, typename TV, typename TA, typename TAV>
+static void volumeMulti(uint32_t channels, TO* out, size_t frameCount,
+ const TI* in, TA* aux, const TV *vol, TAV vola)
+{
+ switch (channels) {
+ case 1:
+ volumeMulti<MIXTYPE, 1>(out, frameCount, in, aux, vol, vola);
+ break;
+ case 2:
+ volumeMulti<MIXTYPE, 2>(out, frameCount, in, aux, vol, vola);
+ break;
+ case 3:
+ volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 3>(out, frameCount, in, aux, vol, vola);
+ break;
+ case 4:
+ volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 4>(out, frameCount, in, aux, vol, vola);
+ break;
+ case 5:
+ volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 5>(out, frameCount, in, aux, vol, vola);
+ break;
+ case 6:
+ volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 6>(out, frameCount, in, aux, vol, vola);
+ break;
+ case 7:
+ volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 7>(out, frameCount, in, aux, vol, vola);
+ break;
+ case 8:
+ volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 8>(out, frameCount, in, aux, vol, vola);
+ break;
+ }
+}
+
+/* MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
+ * USEFLOATVOL (set to true if float volume is used)
+ * ADJUSTVOL (set to true if volume ramp parameters needs adjustment afterwards)
+ * TO: int32_t (Q4.27) or float
+ * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
+ * TA: int32_t (Q4.27) or float
+ */
+template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL,
+ typename TO, typename TI, typename TA>
+void AudioMixerBase::TrackBase::volumeMix(TO *out, size_t outFrames,
+ const TI *in, TA *aux, bool ramp)
+{
+ if (USEFLOATVOL) {
+ if (ramp) {
+ volumeRampMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
+ mPrevVolume, mVolumeInc,
+#ifdef FLOAT_AUX
+ &mPrevAuxLevel, mAuxInc
+#else
+ &prevAuxLevel, auxInc
+#endif
+ );
+ if (ADJUSTVOL) {
+ adjustVolumeRamp(aux != NULL, true);
+ }
+ } else {
+ volumeMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
+ mVolume,
+#ifdef FLOAT_AUX
+ mAuxLevel
+#else
+ auxLevel
+#endif
+ );
+ }
+ } else {
+ if (ramp) {
+ volumeRampMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
+ prevVolume, volumeInc, &prevAuxLevel, auxInc);
+ if (ADJUSTVOL) {
+ adjustVolumeRamp(aux != NULL);
+ }
+ } else {
+ volumeMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
+ volume, auxLevel);
+ }
+ }
+}
+
+/* This process hook is called when there is a single track without
+ * aux buffer, volume ramp, or resampling.
+ * TODO: Update the hook selection: this can properly handle aux and ramp.
+ *
+ * MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
+ * TO: int32_t (Q4.27) or float
+ * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
+ * TA: int32_t (Q4.27)
+ */
+template <int MIXTYPE, typename TO, typename TI, typename TA>
+void AudioMixerBase::process__noResampleOneTrack()
+{
+ ALOGVV("process__noResampleOneTrack\n");
+ LOG_ALWAYS_FATAL_IF(mEnabled.size() != 1,
+ "%zu != 1 tracks enabled", mEnabled.size());
+ const std::shared_ptr<TrackBase> &t = mTracks[mEnabled[0]];
+ const uint32_t channels = t->mMixerChannelCount;
+ TO* out = reinterpret_cast<TO*>(t->mainBuffer);
+ TA* aux = reinterpret_cast<TA*>(t->auxBuffer);
+ const bool ramp = t->needsRamp();
+
+ for (size_t numFrames = mFrameCount; numFrames > 0; ) {
+ AudioBufferProvider::Buffer& b(t->buffer);
+ // get input buffer
+ b.frameCount = numFrames;
+ t->bufferProvider->getNextBuffer(&b);
+ const TI *in = reinterpret_cast<TI*>(b.raw);
+
+ // in == NULL can happen if the track was flushed just after having
+ // been enabled for mixing.
+ if (in == NULL || (((uintptr_t)in) & 3)) {
+ memset(out, 0, numFrames
+ * channels * audio_bytes_per_sample(t->mMixerFormat));
+ ALOGE_IF((((uintptr_t)in) & 3), "process__noResampleOneTrack: bus error: "
+ "buffer %p track %p, channels %d, needs %#x",
+ in, &t, t->channelCount, t->needs);
+ return;
+ }
+
+ const size_t outFrames = b.frameCount;
+ t->volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL */, false /* ADJUSTVOL */> (
+ out, outFrames, in, aux, ramp);
+
+ out += outFrames * channels;
+ if (aux != NULL) {
+ aux += outFrames;
+ }
+ numFrames -= b.frameCount;
+
+ // release buffer
+ t->bufferProvider->releaseBuffer(&b);
+ }
+ if (ramp) {
+ t->adjustVolumeRamp(aux != NULL, is_same<TI, float>::value);
+ }
+}
+
+/* This track hook is called to do resampling then mixing,
+ * pulling from the track's upstream AudioBufferProvider.
+ *
+ * MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
+ * TO: int32_t (Q4.27) or float
+ * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
+ * TA: int32_t (Q4.27) or float
+ */
+template <int MIXTYPE, typename TO, typename TI, typename TA>
+void AudioMixerBase::TrackBase::track__Resample(TO* out, size_t outFrameCount, TO* temp, TA* aux)
+{
+ ALOGVV("track__Resample\n");
+ mResampler->setSampleRate(sampleRate);
+ const bool ramp = needsRamp();
+ if (ramp || aux != NULL) {
+ // if ramp: resample with unity gain to temp buffer and scale/mix in 2nd step.
+ // if aux != NULL: resample with unity gain to temp buffer then apply send level.
+
+ mResampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
+ memset(temp, 0, outFrameCount * mMixerChannelCount * sizeof(TO));
+ mResampler->resample((int32_t*)temp, outFrameCount, bufferProvider);
+
+ volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL */, true /* ADJUSTVOL */>(
+ out, outFrameCount, temp, aux, ramp);
+
+ } else { // constant volume gain
+ mResampler->setVolume(mVolume[0], mVolume[1]);
+ mResampler->resample((int32_t*)out, outFrameCount, bufferProvider);
+ }
+}
+
+/* This track hook is called to mix a track, when no resampling is required.
+ * The input buffer should be present in in.
+ *
+ * MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
+ * TO: int32_t (Q4.27) or float
+ * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
+ * TA: int32_t (Q4.27) or float
+ */
+template <int MIXTYPE, typename TO, typename TI, typename TA>
+void AudioMixerBase::TrackBase::track__NoResample(
+ TO* out, size_t frameCount, TO* temp __unused, TA* aux)
+{
+ ALOGVV("track__NoResample\n");
+ const TI *in = static_cast<const TI *>(mIn);
+
+ volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL */, true /* ADJUSTVOL */>(
+ out, frameCount, in, aux, needsRamp());
+
+ // MIXTYPE_MONOEXPAND reads a single input channel and expands to NCHAN output channels.
+ // MIXTYPE_MULTI reads NCHAN input channels and places to NCHAN output channels.
+ in += (MIXTYPE == MIXTYPE_MONOEXPAND) ? frameCount : frameCount * mMixerChannelCount;
+ mIn = in;
+}
+
+/* The Mixer engine generates either int32_t (Q4_27) or float data.
+ * We use this function to convert the engine buffers
+ * to the desired mixer output format, either int16_t (Q.15) or float.
+ */
+/* static */
+void AudioMixerBase::convertMixerFormat(void *out, audio_format_t mixerOutFormat,
+ void *in, audio_format_t mixerInFormat, size_t sampleCount)
+{
+ switch (mixerInFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ switch (mixerOutFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ memcpy(out, in, sampleCount * sizeof(float)); // MEMCPY. TODO optimize out
+ break;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ memcpy_to_i16_from_float((int16_t*)out, (float*)in, sampleCount);
+ break;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
+ break;
+ }
+ break;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ switch (mixerOutFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ memcpy_to_float_from_q4_27((float*)out, (const int32_t*)in, sampleCount);
+ break;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ memcpy_to_i16_from_q4_27((int16_t*)out, (const int32_t*)in, sampleCount);
+ break;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
+ break;
+ }
+ break;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
+ break;
+ }
+}
+
+/* Returns the proper track hook to use for mixing the track into the output buffer.
+ */
+/* static */
+AudioMixerBase::hook_t AudioMixerBase::TrackBase::getTrackHook(int trackType, uint32_t channelCount,
+ audio_format_t mixerInFormat, audio_format_t mixerOutFormat __unused)
+{
+ if (!kUseNewMixer && channelCount == FCC_2 && mixerInFormat == AUDIO_FORMAT_PCM_16_BIT) {
+ switch (trackType) {
+ case TRACKTYPE_NOP:
+ return &TrackBase::track__nop;
+ case TRACKTYPE_RESAMPLE:
+ return &TrackBase::track__genericResample;
+ case TRACKTYPE_NORESAMPLEMONO:
+ return &TrackBase::track__16BitsMono;
+ case TRACKTYPE_NORESAMPLE:
+ return &TrackBase::track__16BitsStereo;
+ default:
+ LOG_ALWAYS_FATAL("bad trackType: %d", trackType);
+ break;
+ }
+ }
+ LOG_ALWAYS_FATAL_IF(channelCount > MAX_NUM_CHANNELS);
+ switch (trackType) {
+ case TRACKTYPE_NOP:
+ return &TrackBase::track__nop;
+ case TRACKTYPE_RESAMPLE:
+ switch (mixerInFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ return (AudioMixerBase::hook_t) &TrackBase::track__Resample<
+ MIXTYPE_MULTI, float /*TO*/, float /*TI*/, TYPE_AUX>;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ return (AudioMixerBase::hook_t) &TrackBase::track__Resample<
+ MIXTYPE_MULTI, int32_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
+ break;
+ }
+ break;
+ case TRACKTYPE_NORESAMPLEMONO:
+ switch (mixerInFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ return (AudioMixerBase::hook_t) &TrackBase::track__NoResample<
+ MIXTYPE_MONOEXPAND, float /*TO*/, float /*TI*/, TYPE_AUX>;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ return (AudioMixerBase::hook_t) &TrackBase::track__NoResample<
+ MIXTYPE_MONOEXPAND, int32_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
+ break;
+ }
+ break;
+ case TRACKTYPE_NORESAMPLE:
+ switch (mixerInFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ return (AudioMixerBase::hook_t) &TrackBase::track__NoResample<
+ MIXTYPE_MULTI, float /*TO*/, float /*TI*/, TYPE_AUX>;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ return (AudioMixerBase::hook_t) &TrackBase::track__NoResample<
+ MIXTYPE_MULTI, int32_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
+ break;
+ }
+ break;
+ default:
+ LOG_ALWAYS_FATAL("bad trackType: %d", trackType);
+ break;
+ }
+ return NULL;
+}
+
+/* Returns the proper process hook for mixing tracks. Currently works only for
+ * PROCESSTYPE_NORESAMPLEONETRACK, a mix involving one track, no resampling.
+ *
+ * TODO: Due to the special mixing considerations of duplicating to
+ * a stereo output track, the input track cannot be MONO. This should be
+ * prevented by the caller.
+ */
+/* static */
+AudioMixerBase::process_hook_t AudioMixerBase::getProcessHook(
+ int processType, uint32_t channelCount,
+ audio_format_t mixerInFormat, audio_format_t mixerOutFormat)
+{
+ if (processType != PROCESSTYPE_NORESAMPLEONETRACK) { // Only NORESAMPLEONETRACK
+ LOG_ALWAYS_FATAL("bad processType: %d", processType);
+ return NULL;
+ }
+ if (!kUseNewMixer && channelCount == FCC_2 && mixerInFormat == AUDIO_FORMAT_PCM_16_BIT) {
+ return &AudioMixerBase::process__oneTrack16BitsStereoNoResampling;
+ }
+ LOG_ALWAYS_FATAL_IF(channelCount > MAX_NUM_CHANNELS);
+ switch (mixerInFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ switch (mixerOutFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ return &AudioMixerBase::process__noResampleOneTrack<
+ MIXTYPE_MULTI_SAVEONLY, float /*TO*/, float /*TI*/, TYPE_AUX>;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ return &AudioMixerBase::process__noResampleOneTrack<
+ MIXTYPE_MULTI_SAVEONLY, int16_t /*TO*/, float /*TI*/, TYPE_AUX>;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
+ break;
+ }
+ break;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ switch (mixerOutFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ return &AudioMixerBase::process__noResampleOneTrack<
+ MIXTYPE_MULTI_SAVEONLY, float /*TO*/, int16_t /*TI*/, TYPE_AUX>;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ return &AudioMixerBase::process__noResampleOneTrack<
+ MIXTYPE_MULTI_SAVEONLY, int16_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
+ break;
+ }
+ break;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
+ break;
+ }
+ return NULL;
+}
+
+// ----------------------------------------------------------------------------
+} // namespace android
diff --git a/media/libaudioprocessing/AudioResamplerFirOps.h b/media/libaudioprocessing/AudioResamplerFirOps.h
index 2e4cee3..a3f5ff5 100644
--- a/media/libaudioprocessing/AudioResamplerFirOps.h
+++ b/media/libaudioprocessing/AudioResamplerFirOps.h
@@ -36,13 +36,20 @@
#include <arm_neon.h>
#endif
-#if defined(__SSSE3__) // Should be supported in x86 ABI for both 32 & 64-bit.
+#if defined(__AVX2__) // Should be supported in x86 ABI for both 32 & 64-bit.
+#define USE_AVX2 (true) // Inference AVX2/FMA Intrinsics
#define USE_SSE (true)
+#include <immintrin.h>
+#elif defined(__SSSE3__) // Should be supported in x86 ABI for both 32 & 64-bit.
+#define USE_SSE (true) // Inference SSE Intrinsics
+#define USE_AVX2 (false)
#include <tmmintrin.h>
#else
#define USE_SSE (false)
+#define USE_AVX2(false)
#endif
+
template<typename T, typename U>
struct is_same
{
diff --git a/media/libaudioprocessing/AudioResamplerFirProcessSSE.h b/media/libaudioprocessing/AudioResamplerFirProcessSSE.h
index 30233b5..1c16bc4 100644
--- a/media/libaudioprocessing/AudioResamplerFirProcessSSE.h
+++ b/media/libaudioprocessing/AudioResamplerFirProcessSSE.h
@@ -80,11 +80,16 @@
posCoef1 = _mm_sub_ps(posCoef1, posCoef);
negCoef = _mm_sub_ps(negCoef, negCoef1);
+
+ #if USE_AVX2
+ posCoef = _mm_fmadd_ps(posCoef1, interp, posCoef);
+ negCoef = _mm_fmadd_ps(negCoef, interp, negCoef1);
+ #else
posCoef1 = _mm_mul_ps(posCoef1, interp);
negCoef = _mm_mul_ps(negCoef, interp);
-
posCoef = _mm_add_ps(posCoef1, posCoef);
negCoef = _mm_add_ps(negCoef, negCoef1);
+ #endif //USE_AVX2
}
switch (CHANNELS) {
case 1: {
@@ -94,11 +99,17 @@
sN += 4;
posSamp = _mm_shuffle_ps(posSamp, posSamp, 0x1B);
+
+ #if USE_AVX2
+ accL = _mm_fmadd_ps(posSamp, posCoef, accL);
+ accL = _mm_fmadd_ps(negSamp, negCoef, accL);
+ #else
posSamp = _mm_mul_ps(posSamp, posCoef);
negSamp = _mm_mul_ps(negSamp, negCoef);
-
accL = _mm_add_ps(accL, posSamp);
accL = _mm_add_ps(accL, negSamp);
+ #endif
+
} break;
case 2: {
__m128 posSamp0 = _mm_loadu_ps(sP);
@@ -114,15 +125,23 @@
__m128 negSampL = _mm_shuffle_ps(negSamp0, negSamp1, 0x88);
__m128 negSampR = _mm_shuffle_ps(negSamp0, negSamp1, 0xDD);
- posSampL = _mm_mul_ps(posSampL, posCoef);
- posSampR = _mm_mul_ps(posSampR, posCoef);
- negSampL = _mm_mul_ps(negSampL, negCoef);
- negSampR = _mm_mul_ps(negSampR, negCoef);
+ #if USE_AVX2
+ accL = _mm_fmadd_ps(posSampL, posCoef, accL);
+ accR = _mm_fmadd_ps(posSampR, posCoef, accR);
+ accL = _mm_fmadd_ps(negSampL, negCoef, accL);
+ accR = _mm_fmadd_ps(negSampR, negCoef, accR);
+ #else
+ posSampL = _mm_mul_ps(posSampL, posCoef);
+ posSampR = _mm_mul_ps(posSampR, posCoef);
+ negSampL = _mm_mul_ps(negSampL, negCoef);
+ negSampR = _mm_mul_ps(negSampR, negCoef);
- accL = _mm_add_ps(accL, posSampL);
- accR = _mm_add_ps(accR, posSampR);
- accL = _mm_add_ps(accL, negSampL);
- accR = _mm_add_ps(accR, negSampR);
+ accL = _mm_add_ps(accL, posSampL);
+ accR = _mm_add_ps(accR, posSampR);
+ accL = _mm_add_ps(accL, negSampL);
+ accR = _mm_add_ps(accR, negSampR);
+ #endif
+
} break;
}
} while (count -= 4);
@@ -144,9 +163,13 @@
outAccum = _mm_hadd_ps(accL, accR);
outAccum = _mm_hadd_ps(outAccum, outAccum);
}
-
+ #if USE_AVX2
+ outSamp = _mm_fmadd_ps(outAccum, vLR,outSamp);
+ #else
outAccum = _mm_mul_ps(outAccum, vLR);
outSamp = _mm_add_ps(outSamp, outAccum);
+ #endif
+
_mm_storel_pi(reinterpret_cast<__m64*>(out), outSamp);
}
diff --git a/media/libaudioprocessing/BufferProviders.cpp b/media/libaudioprocessing/BufferProviders.cpp
index 21d25e1..6d31c12 100644
--- a/media/libaudioprocessing/BufferProviders.cpp
+++ b/media/libaudioprocessing/BufferProviders.cpp
@@ -164,6 +164,7 @@
if (mEffectsFactory->createEffect(&sDwnmFxDesc.uuid,
sessionId,
SESSION_ID_INVALID_AND_IGNORED,
+ AUDIO_PORT_HANDLE_NONE,
&mDownmixInterface) != 0) {
ALOGE("DownmixerBufferProvider() error creating downmixer effect");
mDownmixInterface.clear();
diff --git a/media/libaudioprocessing/include/media/AudioMixerBase.h b/media/libaudioprocessing/include/media/AudioMixerBase.h
new file mode 100644
index 0000000..805b6d0
--- /dev/null
+++ b/media/libaudioprocessing/include/media/AudioMixerBase.h
@@ -0,0 +1,359 @@
+/*
+**
+** Copyright 2019, The Android Open Source Project
+**
+** Licensed under the Apache License, Version 2.0 (the "License");
+** you may not use this file except in compliance with the License.
+** You may obtain a copy of the License at
+**
+** http://www.apache.org/licenses/LICENSE-2.0
+**
+** Unless required by applicable law or agreed to in writing, software
+** distributed under the License is distributed on an "AS IS" BASIS,
+** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+** See the License for the specific language governing permissions and
+** limitations under the License.
+*/
+
+#ifndef ANDROID_AUDIO_MIXER_BASE_H
+#define ANDROID_AUDIO_MIXER_BASE_H
+
+#include <map>
+#include <memory>
+#include <string>
+#include <unordered_map>
+#include <vector>
+
+#include <media/AudioBufferProvider.h>
+#include <media/AudioResampler.h>
+#include <media/AudioResamplerPublic.h>
+#include <system/audio.h>
+#include <utils/Compat.h>
+
+// This must match frameworks/av/services/audioflinger/Configuration.h
+// when used with the Audio Framework.
+#define FLOAT_AUX
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+// AudioMixerBase is functional on its own if only mixing and resampling
+// is needed.
+
+class AudioMixerBase
+{
+public:
+ // Do not change these unless underlying code changes.
+ // This mixer has a hard-coded upper limit of 8 channels for output.
+ static constexpr uint32_t MAX_NUM_CHANNELS = FCC_8;
+ static constexpr uint32_t MAX_NUM_VOLUMES = FCC_2; // stereo volume only
+
+ static const uint16_t UNITY_GAIN_INT = 0x1000;
+ static const CONSTEXPR float UNITY_GAIN_FLOAT = 1.0f;
+
+ enum { // names
+ // setParameter targets
+ TRACK = 0x3000,
+ RESAMPLE = 0x3001,
+ RAMP_VOLUME = 0x3002, // ramp to new volume
+ VOLUME = 0x3003, // don't ramp
+ TIMESTRETCH = 0x3004,
+
+ // set Parameter names
+ // for target TRACK
+ CHANNEL_MASK = 0x4000,
+ FORMAT = 0x4001,
+ MAIN_BUFFER = 0x4002,
+ AUX_BUFFER = 0x4003,
+ // 0x4004 reserved
+ MIXER_FORMAT = 0x4005, // AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
+ MIXER_CHANNEL_MASK = 0x4006, // Channel mask for mixer output
+ // for target RESAMPLE
+ SAMPLE_RATE = 0x4100, // Configure sample rate conversion on this track name;
+ // parameter 'value' is the new sample rate in Hz.
+ // Only creates a sample rate converter the first time that
+ // the track sample rate is different from the mix sample rate.
+ // If the new sample rate is the same as the mix sample rate,
+ // and a sample rate converter already exists,
+ // then the sample rate converter remains present but is a no-op.
+ RESET = 0x4101, // Reset sample rate converter without changing sample rate.
+ // This clears out the resampler's input buffer.
+ REMOVE = 0x4102, // Remove the sample rate converter on this track name;
+ // the track is restored to the mix sample rate.
+ // for target RAMP_VOLUME and VOLUME (8 channels max)
+ // FIXME use float for these 3 to improve the dynamic range
+ VOLUME0 = 0x4200,
+ VOLUME1 = 0x4201,
+ AUXLEVEL = 0x4210,
+ };
+
+ AudioMixerBase(size_t frameCount, uint32_t sampleRate)
+ : mSampleRate(sampleRate)
+ , mFrameCount(frameCount) {
+ }
+
+ virtual ~AudioMixerBase() {}
+
+ virtual bool isValidFormat(audio_format_t format) const;
+ virtual bool isValidChannelMask(audio_channel_mask_t channelMask) const;
+
+ // Create a new track in the mixer.
+ //
+ // \param name a unique user-provided integer associated with the track.
+ // If name already exists, the function will abort.
+ // \param channelMask output channel mask.
+ // \param format PCM format
+ // \param sessionId Session id for the track. Tracks with the same
+ // session id will be submixed together.
+ //
+ // \return OK on success.
+ // BAD_VALUE if the format does not satisfy isValidFormat()
+ // or the channelMask does not satisfy isValidChannelMask().
+ status_t create(
+ int name, audio_channel_mask_t channelMask, audio_format_t format, int sessionId);
+
+ bool exists(int name) const {
+ return mTracks.count(name) > 0;
+ }
+
+ // Free an allocated track by name.
+ void destroy(int name);
+
+ // Enable or disable an allocated track by name
+ void enable(int name);
+ void disable(int name);
+
+ virtual void setParameter(int name, int target, int param, void *value);
+
+ void process() {
+ preProcess();
+ (this->*mHook)();
+ postProcess();
+ }
+
+ size_t getUnreleasedFrames(int name) const;
+
+ std::string trackNames() const;
+
+ protected:
+ // Set kUseNewMixer to true to use the new mixer engine always. Otherwise the
+ // original code will be used for stereo sinks, the new mixer for everything else.
+ static constexpr bool kUseNewMixer = true;
+
+ // Set kUseFloat to true to allow floating input into the mixer engine.
+ // If kUseNewMixer is false, this is ignored or may be overridden internally
+ static constexpr bool kUseFloat = true;
+
+#ifdef FLOAT_AUX
+ using TYPE_AUX = float;
+ static_assert(kUseNewMixer && kUseFloat,
+ "kUseNewMixer and kUseFloat must be true for FLOAT_AUX option");
+#else
+ using TYPE_AUX = int32_t; // q4.27
+#endif
+
+ /* For multi-format functions (calls template functions
+ * in AudioMixerOps.h). The template parameters are as follows:
+ *
+ * MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
+ * USEFLOATVOL (set to true if float volume is used)
+ * ADJUSTVOL (set to true if volume ramp parameters needs adjustment afterwards)
+ * TO: int32_t (Q4.27) or float
+ * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
+ * TA: int32_t (Q4.27)
+ */
+
+ enum {
+ // FIXME this representation permits up to 8 channels
+ NEEDS_CHANNEL_COUNT__MASK = 0x00000007,
+ };
+
+ enum {
+ NEEDS_CHANNEL_1 = 0x00000000, // mono
+ NEEDS_CHANNEL_2 = 0x00000001, // stereo
+
+ // sample format is not explicitly specified, and is assumed to be AUDIO_FORMAT_PCM_16_BIT
+
+ NEEDS_MUTE = 0x00000100,
+ NEEDS_RESAMPLE = 0x00001000,
+ NEEDS_AUX = 0x00010000,
+ };
+
+ // hook types
+ enum {
+ PROCESSTYPE_NORESAMPLEONETRACK, // others set elsewhere
+ };
+
+ enum {
+ TRACKTYPE_NOP,
+ TRACKTYPE_RESAMPLE,
+ TRACKTYPE_NORESAMPLE,
+ TRACKTYPE_NORESAMPLEMONO,
+ };
+
+ // process hook functionality
+ using process_hook_t = void(AudioMixerBase::*)();
+
+ struct TrackBase;
+ using hook_t = void(TrackBase::*)(
+ int32_t* output, size_t numOutFrames, int32_t* temp, int32_t* aux);
+
+ struct TrackBase {
+ TrackBase()
+ : bufferProvider(nullptr)
+ {
+ // TODO: move additional initialization here.
+ }
+ virtual ~TrackBase() {}
+
+ virtual uint32_t getOutputChannelCount() { return channelCount; }
+ virtual uint32_t getMixerChannelCount() { return mMixerChannelCount; }
+
+ bool needsRamp() { return (volumeInc[0] | volumeInc[1] | auxInc) != 0; }
+ bool setResampler(uint32_t trackSampleRate, uint32_t devSampleRate);
+ bool doesResample() const { return mResampler.get() != nullptr; }
+ void recreateResampler(uint32_t devSampleRate);
+ void resetResampler() { if (mResampler.get() != nullptr) mResampler->reset(); }
+ void adjustVolumeRamp(bool aux, bool useFloat = false);
+ size_t getUnreleasedFrames() const { return mResampler.get() != nullptr ?
+ mResampler->getUnreleasedFrames() : 0; };
+
+ static hook_t getTrackHook(int trackType, uint32_t channelCount,
+ audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
+
+ void track__nop(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
+
+ template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL,
+ typename TO, typename TI, typename TA>
+ void volumeMix(TO *out, size_t outFrames, const TI *in, TA *aux, bool ramp);
+
+ uint32_t needs;
+
+ // TODO: Eventually remove legacy integer volume settings
+ union {
+ int16_t volume[MAX_NUM_VOLUMES]; // U4.12 fixed point (top bit should be zero)
+ int32_t volumeRL;
+ };
+
+ int32_t prevVolume[MAX_NUM_VOLUMES];
+ int32_t volumeInc[MAX_NUM_VOLUMES];
+ int32_t auxInc;
+ int32_t prevAuxLevel;
+ int16_t auxLevel; // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance
+
+ uint16_t frameCount;
+
+ uint8_t channelCount; // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK)
+ uint8_t unused_padding; // formerly format, was always 16
+ uint16_t enabled; // actually bool
+ audio_channel_mask_t channelMask;
+
+ // actual buffer provider used by the track hooks
+ AudioBufferProvider* bufferProvider;
+
+ mutable AudioBufferProvider::Buffer buffer; // 8 bytes
+
+ hook_t hook;
+ const void *mIn; // current location in buffer
+
+ std::unique_ptr<AudioResampler> mResampler;
+ uint32_t sampleRate;
+ int32_t* mainBuffer;
+ int32_t* auxBuffer;
+
+ int32_t sessionId;
+
+ audio_format_t mMixerFormat; // output mix format: AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
+ audio_format_t mFormat; // input track format
+ audio_format_t mMixerInFormat; // mix internal format AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
+ // each track must be converted to this format.
+
+ float mVolume[MAX_NUM_VOLUMES]; // floating point set volume
+ float mPrevVolume[MAX_NUM_VOLUMES]; // floating point previous volume
+ float mVolumeInc[MAX_NUM_VOLUMES]; // floating point volume increment
+
+ float mAuxLevel; // floating point set aux level
+ float mPrevAuxLevel; // floating point prev aux level
+ float mAuxInc; // floating point aux increment
+
+ audio_channel_mask_t mMixerChannelMask;
+ uint32_t mMixerChannelCount;
+
+ protected:
+
+ // hooks
+ void track__genericResample(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
+ void track__16BitsStereo(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
+ void track__16BitsMono(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
+
+ void volumeRampStereo(int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
+ void volumeStereo(int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
+
+ // multi-format track hooks
+ template <int MIXTYPE, typename TO, typename TI, typename TA>
+ void track__Resample(TO* out, size_t frameCount, TO* temp __unused, TA* aux);
+ template <int MIXTYPE, typename TO, typename TI, typename TA>
+ void track__NoResample(TO* out, size_t frameCount, TO* temp __unused, TA* aux);
+ };
+
+ // preCreateTrack must create an instance of a proper TrackBase descendant.
+ // postCreateTrack is called after filling out fields of TrackBase. It can
+ // abort track creation by returning non-OK status. See the implementation
+ // of create() for details.
+ virtual std::shared_ptr<TrackBase> preCreateTrack();
+ virtual status_t postCreateTrack(TrackBase *track __unused) { return OK; }
+
+ // preProcess is called before the process hook, postProcess after,
+ // see the implementation of process() method.
+ virtual void preProcess() {}
+ virtual void postProcess() {}
+
+ virtual bool setChannelMasks(int name,
+ audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask);
+
+ // Called when track info changes and a new process hook should be determined.
+ void invalidate() {
+ mHook = &AudioMixerBase::process__validate;
+ }
+
+ void process__validate();
+ void process__nop();
+ void process__genericNoResampling();
+ void process__genericResampling();
+ void process__oneTrack16BitsStereoNoResampling();
+
+ template <int MIXTYPE, typename TO, typename TI, typename TA>
+ void process__noResampleOneTrack();
+
+ static process_hook_t getProcessHook(int processType, uint32_t channelCount,
+ audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
+
+ static void convertMixerFormat(void *out, audio_format_t mixerOutFormat,
+ void *in, audio_format_t mixerInFormat, size_t sampleCount);
+
+ // initialization constants
+ const uint32_t mSampleRate;
+ const size_t mFrameCount;
+
+ process_hook_t mHook = &AudioMixerBase::process__nop; // one of process__*, never nullptr
+
+ // the size of the type (int32_t) should be the largest of all types supported
+ // by the mixer.
+ std::unique_ptr<int32_t[]> mOutputTemp;
+ std::unique_ptr<int32_t[]> mResampleTemp;
+
+ // track names grouped by main buffer, in no particular order of main buffer.
+ // however names for a particular main buffer are in order (by construction).
+ std::unordered_map<void * /* mainBuffer */, std::vector<int /* name */>> mGroups;
+
+ // track names that are enabled, in increasing order (by construction).
+ std::vector<int /* name */> mEnabled;
+
+ // track smart pointers, by name, in increasing order of name.
+ std::map<int /* name */, std::shared_ptr<TrackBase>> mTracks;
+};
+
+} // namespace android
+
+#endif // ANDROID_AUDIO_MIXER_BASE_H
diff --git a/media/libmedia/include/media/RecordBufferConverter.h b/media/libaudioprocessing/include/media/RecordBufferConverter.h
similarity index 100%
rename from media/libmedia/include/media/RecordBufferConverter.h
rename to media/libaudioprocessing/include/media/RecordBufferConverter.h
diff --git a/media/libaudioprocessing/tests/Android.bp b/media/libaudioprocessing/tests/Android.bp
index d990111..20c2c2c 100644
--- a/media/libaudioprocessing/tests/Android.bp
+++ b/media/libaudioprocessing/tests/Android.bp
@@ -3,8 +3,13 @@
cc_defaults {
name: "libaudioprocessing_test_defaults",
- header_libs: ["libbase_headers"],
+ header_libs: [
+ "libbase_headers",
+ "libmedia_headers",
+ ],
+
shared_libs: [
+ "libaudioclient",
"libaudioprocessing",
"libaudioutils",
"libcutils",
diff --git a/media/libaudioprocessing/tests/fuzzer/Android.bp b/media/libaudioprocessing/tests/fuzzer/Android.bp
new file mode 100644
index 0000000..1df47b7
--- /dev/null
+++ b/media/libaudioprocessing/tests/fuzzer/Android.bp
@@ -0,0 +1,10 @@
+cc_fuzz {
+ name: "libaudioprocessing_resampler_fuzzer",
+ srcs: [
+ "libaudioprocessing_resampler_fuzzer.cpp",
+ ],
+ defaults: ["libaudioprocessing_test_defaults"],
+ static_libs: [
+ "libsndfile",
+ ],
+}
diff --git a/media/libaudioprocessing/tests/fuzzer/libaudioprocessing_resampler_fuzzer.cpp b/media/libaudioprocessing/tests/fuzzer/libaudioprocessing_resampler_fuzzer.cpp
new file mode 100644
index 0000000..938c610
--- /dev/null
+++ b/media/libaudioprocessing/tests/fuzzer/libaudioprocessing_resampler_fuzzer.cpp
@@ -0,0 +1,188 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <android-base/macros.h>
+#include <audio_utils/primitives.h>
+#include <audio_utils/sndfile.h>
+#include <errno.h>
+#include <fcntl.h>
+#include <inttypes.h>
+#include <math.h>
+#include <media/AudioBufferProvider.h>
+#include <media/AudioResampler.h>
+#include <stddef.h>
+#include <stdint.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <sys/mman.h>
+#include <sys/stat.h>
+#include <time.h>
+#include <unistd.h>
+#include <utils/Vector.h>
+
+#include <memory>
+
+using namespace android;
+
+const int MAX_FRAMES = 10;
+const int MIN_FREQ = 1e3;
+const int MAX_FREQ = 100e3;
+
+const AudioResampler::src_quality qualities[] = {
+ AudioResampler::DEFAULT_QUALITY,
+ AudioResampler::LOW_QUALITY,
+ AudioResampler::MED_QUALITY,
+ AudioResampler::HIGH_QUALITY,
+ AudioResampler::VERY_HIGH_QUALITY,
+ AudioResampler::DYN_LOW_QUALITY,
+ AudioResampler::DYN_MED_QUALITY,
+ AudioResampler::DYN_HIGH_QUALITY,
+};
+
+class Provider : public AudioBufferProvider {
+ const void* mAddr; // base address
+ const size_t mNumFrames; // total frames
+ const size_t mFrameSize; // size of each frame in bytes
+ size_t mNextFrame; // index of next frame to provide
+ size_t mUnrel; // number of frames not yet released
+ public:
+ Provider(const void* addr, size_t frames, size_t frameSize)
+ : mAddr(addr),
+ mNumFrames(frames),
+ mFrameSize(frameSize),
+ mNextFrame(0),
+ mUnrel(0) {}
+ status_t getNextBuffer(Buffer* buffer) override {
+ if (buffer->frameCount > mNumFrames - mNextFrame) {
+ buffer->frameCount = mNumFrames - mNextFrame;
+ }
+ mUnrel = buffer->frameCount;
+ if (buffer->frameCount > 0) {
+ buffer->raw = (char*)mAddr + mFrameSize * mNextFrame;
+ return NO_ERROR;
+ } else {
+ buffer->raw = nullptr;
+ return NOT_ENOUGH_DATA;
+ }
+ }
+ virtual void releaseBuffer(Buffer* buffer) {
+ if (buffer->frameCount > mUnrel) {
+ mNextFrame += mUnrel;
+ mUnrel = 0;
+ } else {
+ mNextFrame += buffer->frameCount;
+ mUnrel -= buffer->frameCount;
+ }
+ buffer->frameCount = 0;
+ buffer->raw = nullptr;
+ }
+ void reset() { mNextFrame = 0; }
+};
+
+audio_format_t chooseFormat(AudioResampler::src_quality quality,
+ uint8_t input_byte) {
+ switch (quality) {
+ case AudioResampler::DYN_LOW_QUALITY:
+ case AudioResampler::DYN_MED_QUALITY:
+ case AudioResampler::DYN_HIGH_QUALITY:
+ if (input_byte % 2) {
+ return AUDIO_FORMAT_PCM_FLOAT;
+ }
+ FALLTHROUGH_INTENDED;
+ default:
+ return AUDIO_FORMAT_PCM_16_BIT;
+ }
+}
+
+int parseValue(const uint8_t* src, int index, void* dst, size_t size) {
+ memcpy(dst, &src[index], size);
+ return size;
+}
+
+bool validFreq(int freq) { return freq > MIN_FREQ && freq < MAX_FREQ; }
+
+extern "C" int LLVMFuzzerTestOneInput(const uint8_t* data, size_t size) {
+ int input_freq = 0;
+ int output_freq = 0;
+ int input_channels = 0;
+
+ float left_volume = 0;
+ float right_volume = 0;
+
+ size_t metadata_size = 2 + 3 * sizeof(int) + 2 * sizeof(float);
+ if (size < metadata_size) {
+ // not enough data to set options
+ return 0;
+ }
+
+ AudioResampler::src_quality quality = qualities[data[0] % 8];
+ audio_format_t format = chooseFormat(quality, data[1]);
+
+ int index = 2;
+
+ index += parseValue(data, index, &input_freq, sizeof(int));
+ index += parseValue(data, index, &output_freq, sizeof(int));
+ index += parseValue(data, index, &input_channels, sizeof(int));
+
+ index += parseValue(data, index, &left_volume, sizeof(float));
+ index += parseValue(data, index, &right_volume, sizeof(float));
+
+ if (!validFreq(input_freq) || !validFreq(output_freq)) {
+ // sampling frequencies must be reasonable
+ return 0;
+ }
+
+ if (input_channels < 1 ||
+ input_channels > (quality < AudioResampler::DYN_LOW_QUALITY ? 2 : 8)) {
+ // invalid number of input channels
+ return 0;
+ }
+
+ size_t single_channel_size =
+ format == AUDIO_FORMAT_PCM_FLOAT ? sizeof(float) : sizeof(int16_t);
+ size_t input_frame_size = single_channel_size * input_channels;
+ size_t input_size = size - metadata_size;
+ uint8_t input_data[input_size];
+ memcpy(input_data, &data[metadata_size], input_size);
+
+ size_t input_frames = input_size / input_frame_size;
+ if (input_frames > MAX_FRAMES) {
+ return 0;
+ }
+
+ Provider provider(input_data, input_frames, input_frame_size);
+
+ std::unique_ptr<AudioResampler> resampler(
+ AudioResampler::create(format, input_channels, output_freq, quality));
+
+ resampler->setSampleRate(input_freq);
+ resampler->setVolume(left_volume, right_volume);
+
+ // output is at least stereo samples
+ int output_channels = input_channels > 2 ? input_channels : 2;
+ size_t output_frame_size = output_channels * sizeof(int32_t);
+ size_t output_frames = (input_frames * output_freq) / input_freq;
+ size_t output_size = output_frames * output_frame_size;
+
+ uint8_t output_data[output_size];
+ for (size_t i = 0; i < output_frames; i++) {
+ memset(output_data, 0, output_size);
+ resampler->resample((int*)output_data, i, &provider);
+ }
+
+ return 0;
+}
diff --git a/media/libcpustats/Android.bp b/media/libcpustats/Android.bp
index 8fcd8a4..6e8ca1d 100644
--- a/media/libcpustats/Android.bp
+++ b/media/libcpustats/Android.bp
@@ -6,6 +6,14 @@
"ThreadCpuUsage.cpp",
],
+ local_include_dirs: [
+ "include",
+ ],
+
+ export_include_dirs: [
+ "include",
+ ],
+
cflags: [
"-Werror",
"-Wall",
diff --git a/media/libdatasource/Android.bp b/media/libdatasource/Android.bp
new file mode 100644
index 0000000..f191c21
--- /dev/null
+++ b/media/libdatasource/Android.bp
@@ -0,0 +1,63 @@
+cc_library {
+ name: "libdatasource",
+
+ srcs: [
+ "DataSourceFactory.cpp",
+ "DataURISource.cpp",
+ "FileSource.cpp",
+ "HTTPBase.cpp",
+ "MediaHTTP.cpp",
+ "NuCachedSource2.cpp",
+ ],
+
+ aidl: {
+ local_include_dirs: ["aidl"],
+ export_aidl_headers: true,
+ },
+
+ header_libs: [
+ "libstagefright_headers",
+ "media_ndk_headers",
+ "libmedia_headers",
+ ],
+
+ export_header_lib_headers: [
+ "libstagefright_headers",
+ "media_ndk_headers",
+ ],
+
+ shared_libs: [
+ "liblog",
+ "libcutils",
+ "libutils",
+ "libstagefright_foundation",
+ "libdl",
+ ],
+
+ static_libs: [
+ "libc_malloc_debug_backtrace", // for memory heap analysis
+ "libmedia_midiiowrapper",
+ ],
+
+ local_include_dirs: [
+ "include",
+ ],
+
+ export_include_dirs: [
+ "include",
+ ],
+
+ cflags: [
+ "-Werror",
+ "-Wno-error=deprecated-declarations",
+ "-Wall",
+ ],
+
+ sanitize: {
+ misc_undefined: [
+ "unsigned-integer-overflow",
+ "signed-integer-overflow",
+ ],
+ cfi: true,
+ },
+}
diff --git a/media/libstagefright/DataSourceFactory.cpp b/media/libdatasource/DataSourceFactory.cpp
similarity index 72%
rename from media/libstagefright/DataSourceFactory.cpp
rename to media/libdatasource/DataSourceFactory.cpp
index 54bf0cc..bb6a08c 100644
--- a/media/libstagefright/DataSourceFactory.cpp
+++ b/media/libdatasource/DataSourceFactory.cpp
@@ -16,20 +16,33 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "DataSource"
-#include "include/HTTPBase.h"
-#include "include/NuCachedSource2.h"
+#include <datasource/DataSourceFactory.h>
+#include <datasource/DataURISource.h>
+#include <datasource/HTTPBase.h>
+#include <datasource/FileSource.h>
+#include <datasource/MediaHTTP.h>
+#include <datasource/NuCachedSource2.h>
#include <media/MediaHTTPConnection.h>
#include <media/MediaHTTPService.h>
-#include <media/stagefright/DataSourceFactory.h>
-#include <media/stagefright/DataURISource.h>
-#include <media/stagefright/FileSource.h>
-#include <media/stagefright/MediaHTTP.h>
#include <utils/String8.h>
namespace android {
// static
+sp<DataSourceFactory> DataSourceFactory::sInstance;
+// static
+Mutex DataSourceFactory::sInstanceLock;
+
+// static
+sp<DataSourceFactory> DataSourceFactory::getInstance() {
+ Mutex::Autolock l(sInstanceLock);
+ if (!sInstance) {
+ sInstance = new DataSourceFactory();
+ }
+ return sInstance;
+}
+
sp<DataSource> DataSourceFactory::CreateFromURI(
const sp<MediaHTTPService> &httpService,
const char *uri,
@@ -42,20 +55,16 @@
sp<DataSource> source;
if (!strncasecmp("file://", uri, 7)) {
- source = new FileSource(uri + 7);
+ source = CreateFileSource(uri + 7);
} else if (!strncasecmp("http://", uri, 7) || !strncasecmp("https://", uri, 8)) {
if (httpService == NULL) {
ALOGE("Invalid http service!");
return NULL;
}
- if (httpSource == NULL) {
- sp<MediaHTTPConnection> conn = httpService->makeHTTPConnection();
- if (conn == NULL) {
- ALOGE("Failed to make http connection from http service!");
- return NULL;
- }
- httpSource = new MediaHTTP(conn);
+ sp<HTTPBase> mediaHTTP = httpSource;
+ if (mediaHTTP == NULL) {
+ mediaHTTP = static_cast<HTTPBase *>(CreateMediaHTTP(httpService).get());
}
String8 cacheConfig;
@@ -69,24 +78,24 @@
&disconnectAtHighwatermark);
}
- if (httpSource->connect(uri, &nonCacheSpecificHeaders) != OK) {
+ if (mediaHTTP->connect(uri, &nonCacheSpecificHeaders) != OK) {
ALOGE("Failed to connect http source!");
return NULL;
}
if (contentType != NULL) {
- *contentType = httpSource->getMIMEType();
+ *contentType = mediaHTTP->getMIMEType();
}
source = NuCachedSource2::Create(
- httpSource,
+ mediaHTTP,
cacheConfig.isEmpty() ? NULL : cacheConfig.string(),
disconnectAtHighwatermark);
} else if (!strncasecmp("data:", uri, 5)) {
source = DataURISource::Create(uri);
} else {
// Assume it's a filename.
- source = new FileSource(uri);
+ source = CreateFileSource(uri);
}
if (source == NULL || source->initCheck() != OK) {
@@ -108,10 +117,15 @@
sp<MediaHTTPConnection> conn = httpService->makeHTTPConnection();
if (conn == NULL) {
+ ALOGE("Failed to make http connection from http service!");
return NULL;
} else {
return new MediaHTTP(conn);
}
}
+sp<DataSource> DataSourceFactory::CreateFileSource(const char *uri) {
+ return new FileSource(uri);
+}
+
} // namespace android
diff --git a/media/libstagefright/DataURISource.cpp b/media/libdatasource/DataURISource.cpp
similarity index 98%
rename from media/libstagefright/DataURISource.cpp
rename to media/libdatasource/DataURISource.cpp
index b975b38..216f3d0 100644
--- a/media/libstagefright/DataURISource.cpp
+++ b/media/libdatasource/DataURISource.cpp
@@ -13,7 +13,7 @@
* See the License for the specific language governing permissions and
* limitations under the License.
*/
-#include <media/stagefright/DataURISource.h>
+#include <datasource/DataURISource.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/AString.h>
diff --git a/media/libstagefright/ClearFileSource.cpp b/media/libdatasource/FileSource.cpp
similarity index 85%
rename from media/libstagefright/ClearFileSource.cpp
rename to media/libdatasource/FileSource.cpp
index e3a2cb7..bbf7dda 100644
--- a/media/libstagefright/ClearFileSource.cpp
+++ b/media/libdatasource/FileSource.cpp
@@ -15,12 +15,12 @@
*/
//#define LOG_NDEBUG 0
-#define LOG_TAG "ClearFileSource"
+#define LOG_TAG "FileSource"
#include <utils/Log.h>
+#include <datasource/FileSource.h>
#include <media/stagefright/foundation/ADebug.h>
-#include <media/stagefright/ClearFileSource.h>
-#include <media/stagefright/Utils.h>
+#include <media/stagefright/FoundationUtils.h>
#include <sys/types.h>
#include <unistd.h>
#include <sys/types.h>
@@ -29,7 +29,7 @@
namespace android {
-ClearFileSource::ClearFileSource(const char *filename)
+FileSource::FileSource(const char *filename)
: mFd(-1),
mOffset(0),
mLength(-1),
@@ -48,7 +48,7 @@
}
}
-ClearFileSource::ClearFileSource(int fd, int64_t offset, int64_t length)
+FileSource::FileSource(int fd, int64_t offset, int64_t length)
: mFd(fd),
mOffset(offset),
mLength(length),
@@ -89,18 +89,18 @@
}
-ClearFileSource::~ClearFileSource() {
+FileSource::~FileSource() {
if (mFd >= 0) {
::close(mFd);
mFd = -1;
}
}
-status_t ClearFileSource::initCheck() const {
+status_t FileSource::initCheck() const {
return mFd >= 0 ? OK : NO_INIT;
}
-ssize_t ClearFileSource::readAt(off64_t offset, void *data, size_t size) {
+ssize_t FileSource::readAt(off64_t offset, void *data, size_t size) {
if (mFd < 0) {
return NO_INIT;
}
@@ -118,7 +118,7 @@
return readAt_l(offset, data, size);
}
-ssize_t ClearFileSource::readAt_l(off64_t offset, void *data, size_t size) {
+ssize_t FileSource::readAt_l(off64_t offset, void *data, size_t size) {
off64_t result = lseek64(mFd, offset + mOffset, SEEK_SET);
if (result == -1) {
ALOGE("seek to %lld failed", (long long)(offset + mOffset));
@@ -128,7 +128,7 @@
return ::read(mFd, data, size);
}
-status_t ClearFileSource::getSize(off64_t *size) {
+status_t FileSource::getSize(off64_t *size) {
Mutex::Autolock autoLock(mLock);
if (mFd < 0) {
diff --git a/media/libstagefright/HTTPBase.cpp b/media/libdatasource/HTTPBase.cpp
similarity index 98%
rename from media/libstagefright/HTTPBase.cpp
rename to media/libdatasource/HTTPBase.cpp
index d118e8c..ef29c48 100644
--- a/media/libstagefright/HTTPBase.cpp
+++ b/media/libdatasource/HTTPBase.cpp
@@ -18,7 +18,7 @@
#define LOG_TAG "HTTPBase"
#include <utils/Log.h>
-#include "include/HTTPBase.h"
+#include <datasource/HTTPBase.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/ALooper.h>
diff --git a/media/libstagefright/http/ClearMediaHTTP.cpp b/media/libdatasource/MediaHTTP.cpp
similarity index 82%
rename from media/libstagefright/http/ClearMediaHTTP.cpp
rename to media/libdatasource/MediaHTTP.cpp
index 9557c8a..58c1ce8 100644
--- a/media/libstagefright/http/ClearMediaHTTP.cpp
+++ b/media/libdatasource/MediaHTTP.cpp
@@ -15,30 +15,30 @@
*/
//#define LOG_NDEBUG 0
-#define LOG_TAG "ClearMediaHTTP"
+#define LOG_TAG "MediaHTTP"
#include <utils/Log.h>
-#include <media/stagefright/ClearMediaHTTP.h>
+#include <datasource/MediaHTTP.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/ALooper.h>
-#include <media/stagefright/Utils.h>
+#include <media/stagefright/FoundationUtils.h>
#include <media/MediaHTTPConnection.h>
namespace android {
-ClearMediaHTTP::ClearMediaHTTP(const sp<MediaHTTPConnection> &conn)
+MediaHTTP::MediaHTTP(const sp<MediaHTTPConnection> &conn)
: mInitCheck((conn != NULL) ? OK : NO_INIT),
mHTTPConnection(conn),
mCachedSizeValid(false),
mCachedSize(0ll) {
}
-ClearMediaHTTP::~ClearMediaHTTP() {
+MediaHTTP::~MediaHTTP() {
}
-status_t ClearMediaHTTP::connect(
+status_t MediaHTTP::connect(
const char *uri,
const KeyedVector<String8, String8> *headers,
off64_t /* offset */) {
@@ -68,18 +68,18 @@
if (success) {
AString sanitized = uriDebugString(mLastURI);
- mName = String8::format("ClearMediaHTTP(%s)", sanitized.c_str());
+ mName = String8::format("MediaHTTP(%s)", sanitized.c_str());
}
return success ? OK : UNKNOWN_ERROR;
}
-void ClearMediaHTTP::close() {
+void MediaHTTP::close() {
disconnect();
}
-void ClearMediaHTTP::disconnect() {
- mName = String8("ClearMediaHTTP(<disconnected>)");
+void MediaHTTP::disconnect() {
+ mName = String8("MediaHTTP(<disconnected>)");
if (mInitCheck != OK) {
return;
}
@@ -87,11 +87,11 @@
mHTTPConnection->disconnect();
}
-status_t ClearMediaHTTP::initCheck() const {
+status_t MediaHTTP::initCheck() const {
return mInitCheck;
}
-ssize_t ClearMediaHTTP::readAt(off64_t offset, void *data, size_t size) {
+ssize_t MediaHTTP::readAt(off64_t offset, void *data, size_t size) {
if (mInitCheck != OK) {
return mInitCheck;
}
@@ -127,7 +127,7 @@
return numBytesRead;
}
-status_t ClearMediaHTTP::getSize(off64_t *size) {
+status_t MediaHTTP::getSize(off64_t *size) {
if (mInitCheck != OK) {
return mInitCheck;
}
@@ -145,16 +145,16 @@
return *size < 0 ? *size : static_cast<status_t>(OK);
}
-uint32_t ClearMediaHTTP::flags() {
+uint32_t MediaHTTP::flags() {
return kWantsPrefetching | kIsHTTPBasedSource;
}
-status_t ClearMediaHTTP::reconnectAtOffset(off64_t offset) {
+status_t MediaHTTP::reconnectAtOffset(off64_t offset) {
return connect(mLastURI.c_str(), &mLastHeaders, offset);
}
-String8 ClearMediaHTTP::getUri() {
+String8 MediaHTTP::getUri() {
if (mInitCheck != OK) {
return String8::empty();
}
@@ -166,7 +166,7 @@
return String8(mLastURI.c_str());
}
-String8 ClearMediaHTTP::getMIMEType() const {
+String8 MediaHTTP::getMIMEType() const {
if (mInitCheck != OK) {
return String8("application/octet-stream");
}
diff --git a/media/libstagefright/NuCachedSource2.cpp b/media/libdatasource/NuCachedSource2.cpp
similarity index 98%
rename from media/libstagefright/NuCachedSource2.cpp
rename to media/libdatasource/NuCachedSource2.cpp
index 522c81d..6d63ffb 100644
--- a/media/libstagefright/NuCachedSource2.cpp
+++ b/media/libdatasource/NuCachedSource2.cpp
@@ -20,8 +20,8 @@
#define LOG_TAG "NuCachedSource2"
#include <utils/Log.h>
-#include "include/NuCachedSource2.h"
-#include "include/HTTPBase.h"
+#include <datasource/NuCachedSource2.h>
+#include <datasource/HTTPBase.h>
#include <cutils/properties.h>
#include <media/stagefright/foundation/ADebug.h>
@@ -689,10 +689,6 @@
restartPrefetcherIfNecessary_l(true /* ignore low water threshold */);
}
-sp<DecryptHandle> NuCachedSource2::DrmInitialization(const char* mime) {
- return mSource->DrmInitialization(mime);
-}
-
String8 NuCachedSource2::getUri() {
return mSource->getUri();
}
diff --git a/media/libstagefright/include/media/stagefright/DataSourceFactory.h b/media/libdatasource/include/datasource/DataSourceFactory.h
similarity index 66%
rename from media/libstagefright/include/media/stagefright/DataSourceFactory.h
rename to media/libdatasource/include/datasource/DataSourceFactory.h
index 2a1d491..194abe2 100644
--- a/media/libstagefright/include/media/stagefright/DataSourceFactory.h
+++ b/media/libdatasource/include/datasource/DataSourceFactory.h
@@ -18,7 +18,9 @@
#define DATA_SOURCE_FACTORY_H_
+#include <media/DataSource.h>
#include <sys/types.h>
+#include <utils/KeyedVector.h>
#include <utils/RefBase.h>
namespace android {
@@ -27,17 +29,27 @@
class String8;
struct HTTPBase;
-class DataSourceFactory {
+class DataSourceFactory : public RefBase {
public:
- static sp<DataSource> CreateFromURI(
+ static sp<DataSourceFactory> getInstance();
+ sp<DataSource> CreateFromURI(
const sp<MediaHTTPService> &httpService,
const char *uri,
const KeyedVector<String8, String8> *headers = NULL,
String8 *contentType = NULL,
HTTPBase *httpSource = NULL);
- static sp<DataSource> CreateMediaHTTP(const sp<MediaHTTPService> &httpService);
- static sp<DataSource> CreateFromFd(int fd, int64_t offset, int64_t length);
+ virtual sp<DataSource> CreateMediaHTTP(const sp<MediaHTTPService> &httpService);
+ sp<DataSource> CreateFromFd(int fd, int64_t offset, int64_t length);
+
+protected:
+ virtual sp<DataSource> CreateFileSource(const char *uri);
+ DataSourceFactory() {};
+ virtual ~DataSourceFactory() {};
+
+private:
+ static sp<DataSourceFactory> sInstance;
+ static Mutex sInstanceLock;
};
} // namespace android
diff --git a/media/libstagefright/include/media/stagefright/DataURISource.h b/media/libdatasource/include/datasource/DataURISource.h
similarity index 100%
rename from media/libstagefright/include/media/stagefright/DataURISource.h
rename to media/libdatasource/include/datasource/DataURISource.h
diff --git a/media/libstagefright/include/media/stagefright/ClearFileSource.h b/media/libdatasource/include/datasource/FileSource.h
similarity index 74%
rename from media/libstagefright/include/media/stagefright/ClearFileSource.h
rename to media/libdatasource/include/datasource/FileSource.h
index be83748..dee0c33 100644
--- a/media/libstagefright/include/media/stagefright/ClearFileSource.h
+++ b/media/libdatasource/include/datasource/FileSource.h
@@ -14,9 +14,9 @@
* limitations under the License.
*/
-#ifndef CLEAR_FILE_SOURCE_H_
+#ifndef FILE_SOURCE_H_
-#define CLEAR_FILE_SOURCE_H_
+#define FILE_SOURCE_H_
#include <stdio.h>
@@ -26,11 +26,11 @@
namespace android {
-class ClearFileSource : public DataSource {
+class FileSource : public DataSource {
public:
- ClearFileSource(const char *filename);
- // ClearFileSource takes ownership and will close the fd
- ClearFileSource(int fd, int64_t offset, int64_t length);
+ FileSource(const char *filename);
+ // FileSource takes ownership and will close the fd
+ FileSource(int fd, int64_t offset, int64_t length);
virtual status_t initCheck() const;
@@ -47,7 +47,7 @@
}
protected:
- virtual ~ClearFileSource();
+ virtual ~FileSource();
virtual ssize_t readAt_l(off64_t offset, void *data, size_t size);
int mFd;
@@ -58,11 +58,11 @@
private:
String8 mName;
- ClearFileSource(const ClearFileSource &);
- ClearFileSource &operator=(const ClearFileSource &);
+ FileSource(const FileSource &);
+ FileSource &operator=(const FileSource &);
};
} // namespace android
-#endif // CLEAR_FILE_SOURCE_H_
+#endif // FILE_SOURCE_H_
diff --git a/media/libstagefright/include/HTTPBase.h b/media/libdatasource/include/datasource/HTTPBase.h
similarity index 100%
rename from media/libstagefright/include/HTTPBase.h
rename to media/libdatasource/include/datasource/HTTPBase.h
diff --git a/media/libstagefright/include/media/stagefright/ClearMediaHTTP.h b/media/libdatasource/include/datasource/MediaHTTP.h
similarity index 83%
rename from media/libstagefright/include/media/stagefright/ClearMediaHTTP.h
rename to media/libdatasource/include/datasource/MediaHTTP.h
index 72907a9..a8d203b 100644
--- a/media/libstagefright/include/media/stagefright/ClearMediaHTTP.h
+++ b/media/libdatasource/include/datasource/MediaHTTP.h
@@ -14,20 +14,20 @@
* limitations under the License.
*/
-#ifndef CLEAR_MEDIA_HTTP_H_
+#ifndef MEDIA_HTTP_H_
-#define CLEAR_MEDIA_HTTP_H_
+#define MEDIA_HTTP_H_
#include <media/stagefright/foundation/AString.h>
-#include "include/HTTPBase.h"
+#include "HTTPBase.h"
namespace android {
struct MediaHTTPConnection;
-struct ClearMediaHTTP : public HTTPBase {
- ClearMediaHTTP(const sp<MediaHTTPConnection> &conn);
+struct MediaHTTP : public HTTPBase {
+ MediaHTTP(const sp<MediaHTTPConnection> &conn);
virtual status_t connect(
const char *uri,
@@ -49,7 +49,7 @@
virtual status_t reconnectAtOffset(off64_t offset);
protected:
- virtual ~ClearMediaHTTP();
+ virtual ~MediaHTTP();
virtual String8 getUri();
virtual String8 getMIMEType() const;
@@ -65,9 +65,9 @@
bool mCachedSizeValid;
off64_t mCachedSize;
- DISALLOW_EVIL_CONSTRUCTORS(ClearMediaHTTP);
+ DISALLOW_EVIL_CONSTRUCTORS(MediaHTTP);
};
} // namespace android
-#endif // CLEAR_MEDIA_HTTP_H_
+#endif // MEDIA_HTTP_H_
diff --git a/media/libstagefright/include/NuCachedSource2.h b/media/libdatasource/include/datasource/NuCachedSource2.h
similarity index 98%
rename from media/libstagefright/include/NuCachedSource2.h
rename to media/libdatasource/include/datasource/NuCachedSource2.h
index 596efb8..4c253ad 100644
--- a/media/libstagefright/include/NuCachedSource2.h
+++ b/media/libdatasource/include/datasource/NuCachedSource2.h
@@ -44,7 +44,6 @@
virtual status_t getSize(off64_t *size);
virtual uint32_t flags();
- virtual sp<DecryptHandle> DrmInitialization(const char* mime);
virtual String8 getUri();
virtual String8 getMIMEType() const;
diff --git a/media/libeffects/config/Android.bp b/media/libeffects/config/Android.bp
index 5fa9da9..8476f82 100644
--- a/media/libeffects/config/Android.bp
+++ b/media/libeffects/config/Android.bp
@@ -13,6 +13,8 @@
shared_libs: [
"liblog",
"libtinyxml2",
+ "libutils",
+ "libmedia_helper",
],
header_libs: ["libaudio_system_headers"],
diff --git a/media/libeffects/config/include/media/EffectsConfig.h b/media/libeffects/config/include/media/EffectsConfig.h
index fa0415b..ef10e0d 100644
--- a/media/libeffects/config/include/media/EffectsConfig.h
+++ b/media/libeffects/config/include/media/EffectsConfig.h
@@ -76,6 +76,10 @@
using OutputStream = Stream<audio_stream_type_t>;
using InputStream = Stream<audio_source_t>;
+struct DeviceEffects : Stream<audio_devices_t> {
+ std::string address;
+};
+
/** Parsed configuration.
* Intended to be a transient structure only used for deserialization.
* Note: Everything is copied in the configuration from the xml dom.
@@ -89,6 +93,7 @@
Effects effects;
std::vector<OutputStream> postprocess;
std::vector<InputStream> preprocess;
+ std::vector<DeviceEffects> deviceprocess;
};
/** Result of `parse(const char*)` */
diff --git a/media/libeffects/config/src/EffectsConfig.cpp b/media/libeffects/config/src/EffectsConfig.cpp
index f39eb0c..85fbf11 100644
--- a/media/libeffects/config/src/EffectsConfig.cpp
+++ b/media/libeffects/config/src/EffectsConfig.cpp
@@ -26,6 +26,7 @@
#include <log/log.h>
#include <media/EffectsConfig.h>
+#include <media/TypeConverter.h>
using namespace tinyxml2;
@@ -100,6 +101,7 @@
{AUDIO_STREAM_ENFORCED_AUDIBLE, "enforced_audible"},
{AUDIO_STREAM_DTMF, "dtmf"},
{AUDIO_STREAM_TTS, "tts"},
+ {AUDIO_STREAM_ASSISTANT, "assistant"},
};
/** All input stream types which support effects.
@@ -116,6 +118,8 @@
{AUDIO_SOURCE_VOICE_COMMUNICATION, "voice_communication"},
{AUDIO_SOURCE_UNPROCESSED, "unprocessed"},
{AUDIO_SOURCE_VOICE_PERFORMANCE, "voice_performance"},
+ {AUDIO_SOURCE_ECHO_REFERENCE, "echo_reference"},
+ {AUDIO_SOURCE_FM_TUNER, "fm_tuner"},
};
/** Find the stream type enum corresponding to the stream type name or return false */
@@ -131,6 +135,11 @@
return false;
}
+template <>
+bool stringToStreamType(const char *streamName, audio_devices_t* type) {
+ return deviceFromString(streamName, *type);
+}
+
/** Parse a library xml note and push the result in libraries or return false on failure. */
bool parseLibrary(const XMLElement& xmlLibrary, Libraries* libraries) {
const char* name = xmlLibrary.Attribute("name");
@@ -218,7 +227,7 @@
return true;
}
-/** Parse an stream from an xml element describing it.
+/** Parse an <Output|Input>stream or a device from an xml element describing it.
* @return true and pushes the stream in streams on success,
* false on failure. */
template <class Stream>
@@ -230,14 +239,14 @@
}
Stream stream;
if (!stringToStreamType(streamType, &stream.type)) {
- ALOGE("Invalid stream type %s: %s", streamType, dump(xmlStream));
+ ALOGE("Invalid <stream|device> type %s: %s", streamType, dump(xmlStream));
return false;
}
for (auto& xmlApply : getChildren(xmlStream, "apply")) {
const char* effectName = xmlApply.get().Attribute("effect");
if (effectName == nullptr) {
- ALOGE("stream/apply must have reference an effect: %s", dump(xmlApply));
+ ALOGE("<stream|device>/apply must have reference an effect: %s", dump(xmlApply));
return false;
}
auto* effect = findByName(effectName, effects);
@@ -251,6 +260,21 @@
return true;
}
+bool parseDeviceEffects(
+ const XMLElement& xmlDevice, Effects& effects, std::vector<DeviceEffects>* deviceEffects) {
+
+ const char* address = xmlDevice.Attribute("address");
+ if (address == nullptr) {
+ ALOGE("device must have an address: %s", dump(xmlDevice));
+ return false;
+ }
+ if (!parseStream(xmlDevice, effects, deviceEffects)) {
+ return false;
+ }
+ deviceEffects->back().address = address;
+ return true;
+}
+
/** Internal version of the public parse(const char* path) where path always exist. */
ParsingResult parseWithPath(std::string&& path) {
XMLDocument doc;
@@ -295,6 +319,14 @@
registerFailure(parseStream(xmlStream, config->effects, &config->postprocess));
}
}
+
+ // Parse device effect chains
+ for (auto& xmlDeviceEffects : getChildren(xmlConfig, "deviceEffects")) {
+ for (auto& xmlDevice : getChildren(xmlDeviceEffects, "devicePort")) {
+ registerFailure(
+ parseDeviceEffects(xmlDevice, config->effects, &config->deviceprocess));
+ }
+ }
}
return {std::move(config), nbSkippedElements, std::move(path)};
}
diff --git a/media/libeffects/data/audio_effects.xml b/media/libeffects/data/audio_effects.xml
index 3f85052..2e5f529 100644
--- a/media/libeffects/data/audio_effects.xml
+++ b/media/libeffects/data/audio_effects.xml
@@ -99,4 +99,31 @@
</postprocess>
-->
+ <!-- Device pre/post processor configurations.
+ The device pre/post processor configuration is described in a deviceEffects element and
+ consists in a list of elements each describing pre/post proecessor settings for a given
+ device or "devicePort".
+ Each devicePort element has a "type" attribute corresponding to the device type (e.g.
+ speaker, bus), an "address" attribute corresponding to the device address and contains a
+ list of "apply" elements indicating one effect to apply.
+ If the device is a source, only pre processing effects are expected, if the
+ device is a sink, only post processing effects are expected.
+ The effect to apply is designated by its name in the "effects" elements.
+ The effect will be enabled by default and the audio framework will automatically add
+ and activate the effect if the given port is involved in an audio patch.
+ If the patch is "HW", the effect must be HW accelerated.
+
+ <deviceEffects>
+ <devicePort type="AUDIO_DEVICE_OUT_BUS" address="BUS00_USAGE_MAIN">
+ <apply effect="equalizer"/>
+ </devicePort>
+ <devicePort type="AUDIO_DEVICE_OUT_BUS" address="BUS04_USAGE_VOICE">
+ <apply effect="volume"/>
+ </devicePort>
+ <devicePort type="AUDIO_DEVICE_IN_BUILTIN_MIC" address="bottom">
+ <apply effect="agc"/>
+ </devicePort>
+ </deviceEffects>
+ -->
+
</audio_effects_conf>
diff --git a/media/libeffects/factory/EffectsFactory.c b/media/libeffects/factory/EffectsFactory.c
index c1ce513..dcdf634 100644
--- a/media/libeffects/factory/EffectsFactory.c
+++ b/media/libeffects/factory/EffectsFactory.c
@@ -254,7 +254,8 @@
return ret;
}
-int EffectCreate(const effect_uuid_t *uuid, int32_t sessionId, int32_t ioId, effect_handle_t *pHandle)
+int doEffectCreate(const effect_uuid_t *uuid, int32_t sessionId, int32_t ioId, int32_t deviceId,
+ effect_handle_t *pHandle)
{
list_elem_t *e = gLibraryList;
lib_entry_t *l = NULL;
@@ -268,9 +269,9 @@
}
ALOGV("EffectCreate() UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
- uuid->timeLow, uuid->timeMid, uuid->timeHiAndVersion,
- uuid->clockSeq, uuid->node[0], uuid->node[1],uuid->node[2],
- uuid->node[3],uuid->node[4],uuid->node[5]);
+ uuid->timeLow, uuid->timeMid, uuid->timeHiAndVersion,
+ uuid->clockSeq, uuid->node[0], uuid->node[1], uuid->node[2],
+ uuid->node[3], uuid->node[4], uuid->node[5]);
ret = init();
@@ -282,17 +283,29 @@
pthread_mutex_lock(&gLibLock);
ret = findEffect(NULL, uuid, &l, &d);
- if (ret < 0){
+ if (ret < 0) {
// Sub effects are not associated with the library->effects,
// so, findEffect will fail. Search for the effect in gSubEffectList.
ret = findSubEffect(uuid, &l, &d);
- if (ret < 0 ) {
+ if (ret < 0) {
goto exit;
}
}
// create effect in library
- ret = l->desc->create_effect(uuid, sessionId, ioId, &itfe);
+ if (sessionId == AUDIO_SESSION_DEVICE) {
+ if (l->desc->version >= EFFECT_LIBRARY_API_VERSION_3_1) {
+ ALOGI("EffectCreate() create_effect_3_1");
+ ret = l->desc->create_effect_3_1(uuid, sessionId, ioId, deviceId, &itfe);
+ } else {
+ ALOGE("EffectCreate() cannot create device effect on library with API version < 3.1");
+ ret = -ENOSYS;
+ }
+ } else {
+ ALOGI("EffectCreate() create_effect");
+ ret = l->desc->create_effect(uuid, sessionId, ioId, &itfe);
+ }
+
if (ret != 0) {
ALOGW("EffectCreate() library %s: could not create fx %s, error %d", l->name, d->name, ret);
goto exit;
@@ -324,6 +337,16 @@
return ret;
}
+int EffectCreate(const effect_uuid_t *uuid, int32_t sessionId, int32_t ioId,
+ effect_handle_t *pHandle) {
+ return doEffectCreate(uuid, sessionId, ioId, AUDIO_PORT_HANDLE_NONE, pHandle);
+}
+
+int EffectCreateOnDevice(const effect_uuid_t *uuid, int32_t deviceId, int32_t ioId,
+ effect_handle_t *pHandle) {
+ return doEffectCreate(uuid, AUDIO_SESSION_DEVICE, ioId, deviceId, pHandle);
+}
+
int EffectRelease(effect_handle_t handle)
{
effect_entry_t *fx;
diff --git a/media/libeffects/factory/EffectsFactory.h b/media/libeffects/factory/EffectsFactory.h
index 29dbc9c..1936343 100644
--- a/media/libeffects/factory/EffectsFactory.h
+++ b/media/libeffects/factory/EffectsFactory.h
@@ -27,6 +27,8 @@
extern "C" {
#endif
+#define EFFECT_LIBRARY_API_VERSION_CURRENT EFFECT_LIBRARY_API_VERSION_3_1
+
#define PROPERTY_IGNORE_EFFECTS "ro.audio.ignore_effects"
typedef struct list_elem_s {
diff --git a/media/libeffects/factory/EffectsXmlConfigLoader.cpp b/media/libeffects/factory/EffectsXmlConfigLoader.cpp
index 052a88b..505be7c 100644
--- a/media/libeffects/factory/EffectsXmlConfigLoader.cpp
+++ b/media/libeffects/factory/EffectsXmlConfigLoader.cpp
@@ -94,7 +94,7 @@
}
uint32_t majorVersion = EFFECT_API_VERSION_MAJOR(description->version);
- uint32_t expectedMajorVersion = EFFECT_API_VERSION_MAJOR(EFFECT_LIBRARY_API_VERSION);
+ uint32_t expectedMajorVersion = EFFECT_API_VERSION_MAJOR(EFFECT_LIBRARY_API_VERSION_CURRENT);
if (majorVersion != expectedMajorVersion) {
ALOGE("Unsupported major version %#08x, expected %#08x for library %s",
majorVersion, expectedMajorVersion, path);
diff --git a/media/libeffects/factory/include/media/EffectsFactoryApi.h b/media/libeffects/factory/include/media/EffectsFactoryApi.h
index a5a12eb..8f7239e 100644
--- a/media/libeffects/factory/include/media/EffectsFactoryApi.h
+++ b/media/libeffects/factory/include/media/EffectsFactoryApi.h
@@ -119,6 +119,36 @@
////////////////////////////////////////////////////////////////////////////////
//
+// Function: EffectCreateOnDevice
+//
+// Description: Same as EffectCreate but uesed when creating an effect attached to a
+// particular audio device instance
+//
+// Input:
+// pEffectUuid: pointer to the effect uuid.
+// deviceId: identifies the sink or source device this effect is directed to in
+// audio HAL. Must be specified if sessionId is AUDIO_SESSION_DEVICE.
+// deviceId is the audio_port_handle_t used for the device when the audio
+// patch is created at the audio HAL.//
+// ioId: identifies the output or input stream this effect is directed to at audio HAL.
+// For future use especially with tunneled HW accelerated effects
+// Input/Output:
+// pHandle: address where to return the effect handle.
+//
+// Output:
+// returned value: 0 successful operation.
+// -ENODEV factory failed to initialize
+// -EINVAL invalid pEffectUuid or pHandle
+// -ENOENT no effect with this uuid found
+// *pHandle: updated with the effect handle.
+//
+////////////////////////////////////////////////////////////////////////////////
+ANDROID_API
+int EffectCreateOnDevice(const effect_uuid_t *pEffectUuid, int32_t deviceId, int32_t ioId,
+ effect_handle_t *pHandle);
+
+////////////////////////////////////////////////////////////////////////////////
+//
// Function: EffectRelease
//
// Description: Releases the effect engine whose handle is given as argument.
diff --git a/media/libeffects/lvm/lib/Android.bp b/media/libeffects/lvm/lib/Android.bp
index d150f18..1f2a5e1 100644
--- a/media/libeffects/lvm/lib/Android.bp
+++ b/media/libeffects/lvm/lib/Android.bp
@@ -10,107 +10,107 @@
vendor: true,
srcs: [
- "StereoWidening/src/LVCS_BypassMix.c",
- "StereoWidening/src/LVCS_Control.c",
- "StereoWidening/src/LVCS_Equaliser.c",
- "StereoWidening/src/LVCS_Init.c",
- "StereoWidening/src/LVCS_Process.c",
- "StereoWidening/src/LVCS_ReverbGenerator.c",
- "StereoWidening/src/LVCS_StereoEnhancer.c",
- "StereoWidening/src/LVCS_Tables.c",
- "Bass/src/LVDBE_Control.c",
- "Bass/src/LVDBE_Init.c",
- "Bass/src/LVDBE_Process.c",
- "Bass/src/LVDBE_Tables.c",
- "Bundle/src/LVM_API_Specials.c",
- "Bundle/src/LVM_Buffers.c",
- "Bundle/src/LVM_Init.c",
- "Bundle/src/LVM_Process.c",
- "Bundle/src/LVM_Tables.c",
- "Bundle/src/LVM_Control.c",
- "SpectrumAnalyzer/src/LVPSA_Control.c",
- "SpectrumAnalyzer/src/LVPSA_Init.c",
- "SpectrumAnalyzer/src/LVPSA_Memory.c",
- "SpectrumAnalyzer/src/LVPSA_Process.c",
- "SpectrumAnalyzer/src/LVPSA_QPD_Init.c",
- "SpectrumAnalyzer/src/LVPSA_QPD_Process.c",
- "SpectrumAnalyzer/src/LVPSA_Tables.c",
- "Eq/src/LVEQNB_CalcCoef.c",
- "Eq/src/LVEQNB_Control.c",
- "Eq/src/LVEQNB_Init.c",
- "Eq/src/LVEQNB_Process.c",
- "Eq/src/LVEQNB_Tables.c",
- "Common/src/InstAlloc.c",
- "Common/src/DC_2I_D16_TRC_WRA_01.c",
- "Common/src/DC_2I_D16_TRC_WRA_01_Init.c",
- "Common/src/FO_2I_D16F32C15_LShx_TRC_WRA_01.c",
- "Common/src/FO_2I_D16F32Css_LShx_TRC_WRA_01_Init.c",
- "Common/src/FO_1I_D16F16C15_TRC_WRA_01.c",
- "Common/src/FO_1I_D16F16Css_TRC_WRA_01_Init.c",
- "Common/src/BP_1I_D16F32C30_TRC_WRA_01.c",
- "Common/src/BP_1I_D16F16C14_TRC_WRA_01.c",
- "Common/src/BP_1I_D32F32C30_TRC_WRA_02.c",
- "Common/src/BP_1I_D16F16Css_TRC_WRA_01_Init.c",
- "Common/src/BP_1I_D16F32Cll_TRC_WRA_01_Init.c",
- "Common/src/BP_1I_D32F32Cll_TRC_WRA_02_Init.c",
- "Common/src/BQ_2I_D32F32Cll_TRC_WRA_01_Init.c",
- "Common/src/BQ_2I_D32F32C30_TRC_WRA_01.c",
- "Common/src/BQ_2I_D16F32C15_TRC_WRA_01.c",
- "Common/src/BQ_2I_D16F32C14_TRC_WRA_01.c",
- "Common/src/BQ_2I_D16F32C13_TRC_WRA_01.c",
- "Common/src/BQ_2I_D16F32Css_TRC_WRA_01_init.c",
- "Common/src/BQ_2I_D16F16C15_TRC_WRA_01.c",
- "Common/src/BQ_2I_D16F16C14_TRC_WRA_01.c",
- "Common/src/BQ_2I_D16F16Css_TRC_WRA_01_Init.c",
- "Common/src/BQ_1I_D16F16C15_TRC_WRA_01.c",
- "Common/src/BQ_1I_D16F16Css_TRC_WRA_01_Init.c",
- "Common/src/BQ_1I_D16F32C14_TRC_WRA_01.c",
- "Common/src/BQ_1I_D16F32Css_TRC_WRA_01_init.c",
- "Common/src/PK_2I_D32F32C30G11_TRC_WRA_01.c",
- "Common/src/PK_2I_D32F32C14G11_TRC_WRA_01.c",
- "Common/src/PK_2I_D32F32CssGss_TRC_WRA_01_Init.c",
- "Common/src/PK_2I_D32F32CllGss_TRC_WRA_01_Init.c",
- "Common/src/Int16LShiftToInt32_16x32.c",
- "Common/src/From2iToMono_16.c",
- "Common/src/Copy_16.c",
- "Common/src/MonoTo2I_16.c",
- "Common/src/MonoTo2I_32.c",
- "Common/src/LoadConst_16.c",
- "Common/src/LoadConst_32.c",
- "Common/src/dB_to_Lin32.c",
- "Common/src/Shift_Sat_v16xv16.c",
- "Common/src/Shift_Sat_v32xv32.c",
- "Common/src/Abs_32.c",
- "Common/src/Int32RShiftToInt16_Sat_32x16.c",
- "Common/src/From2iToMono_32.c",
- "Common/src/mult3s_16x16.c",
- "Common/src/Mult3s_32x16.c",
- "Common/src/NonLinComp_D16.c",
- "Common/src/DelayMix_16x16.c",
- "Common/src/MSTo2i_Sat_16x16.c",
- "Common/src/From2iToMS_16x16.c",
- "Common/src/Mac3s_Sat_16x16.c",
- "Common/src/Mac3s_Sat_32x16.c",
- "Common/src/Add2_Sat_16x16.c",
- "Common/src/Add2_Sat_32x32.c",
- "Common/src/LVC_MixSoft_1St_2i_D16C31_SAT.c",
- "Common/src/LVC_MixSoft_1St_D16C31_SAT.c",
- "Common/src/LVC_Mixer_VarSlope_SetTimeConstant.c",
- "Common/src/LVC_Mixer_SetTimeConstant.c",
- "Common/src/LVC_Mixer_SetTarget.c",
- "Common/src/LVC_Mixer_GetTarget.c",
- "Common/src/LVC_Mixer_Init.c",
- "Common/src/LVC_Core_MixHard_1St_2i_D16C31_SAT.c",
- "Common/src/LVC_Core_MixSoft_1St_2i_D16C31_WRA.c",
- "Common/src/LVC_Core_MixInSoft_D16C31_SAT.c",
- "Common/src/LVC_Mixer_GetCurrent.c",
- "Common/src/LVC_MixSoft_2St_D16C31_SAT.c",
- "Common/src/LVC_Core_MixSoft_1St_D16C31_WRA.c",
- "Common/src/LVC_Core_MixHard_2St_D16C31_SAT.c",
- "Common/src/LVC_MixInSoft_D16C31_SAT.c",
- "Common/src/AGC_MIX_VOL_2St1Mon_D32_WRA.c",
- "Common/src/LVM_Timer.c",
- "Common/src/LVM_Timer_Init.c",
+ "StereoWidening/src/LVCS_BypassMix.cpp",
+ "StereoWidening/src/LVCS_Control.cpp",
+ "StereoWidening/src/LVCS_Equaliser.cpp",
+ "StereoWidening/src/LVCS_Init.cpp",
+ "StereoWidening/src/LVCS_Process.cpp",
+ "StereoWidening/src/LVCS_ReverbGenerator.cpp",
+ "StereoWidening/src/LVCS_StereoEnhancer.cpp",
+ "StereoWidening/src/LVCS_Tables.cpp",
+ "Bass/src/LVDBE_Control.cpp",
+ "Bass/src/LVDBE_Init.cpp",
+ "Bass/src/LVDBE_Process.cpp",
+ "Bass/src/LVDBE_Tables.cpp",
+ "Bundle/src/LVM_API_Specials.cpp",
+ "Bundle/src/LVM_Buffers.cpp",
+ "Bundle/src/LVM_Init.cpp",
+ "Bundle/src/LVM_Process.cpp",
+ "Bundle/src/LVM_Tables.cpp",
+ "Bundle/src/LVM_Control.cpp",
+ "SpectrumAnalyzer/src/LVPSA_Control.cpp",
+ "SpectrumAnalyzer/src/LVPSA_Init.cpp",
+ "SpectrumAnalyzer/src/LVPSA_Memory.cpp",
+ "SpectrumAnalyzer/src/LVPSA_Process.cpp",
+ "SpectrumAnalyzer/src/LVPSA_QPD_Init.cpp",
+ "SpectrumAnalyzer/src/LVPSA_QPD_Process.cpp",
+ "SpectrumAnalyzer/src/LVPSA_Tables.cpp",
+ "Eq/src/LVEQNB_CalcCoef.cpp",
+ "Eq/src/LVEQNB_Control.cpp",
+ "Eq/src/LVEQNB_Init.cpp",
+ "Eq/src/LVEQNB_Process.cpp",
+ "Eq/src/LVEQNB_Tables.cpp",
+ "Common/src/InstAlloc.cpp",
+ "Common/src/DC_2I_D16_TRC_WRA_01.cpp",
+ "Common/src/DC_2I_D16_TRC_WRA_01_Init.cpp",
+ "Common/src/FO_2I_D16F32C15_LShx_TRC_WRA_01.cpp",
+ "Common/src/FO_2I_D16F32Css_LShx_TRC_WRA_01_Init.cpp",
+ "Common/src/FO_1I_D16F16C15_TRC_WRA_01.cpp",
+ "Common/src/FO_1I_D16F16Css_TRC_WRA_01_Init.cpp",
+ "Common/src/BP_1I_D16F32C30_TRC_WRA_01.cpp",
+ "Common/src/BP_1I_D16F16C14_TRC_WRA_01.cpp",
+ "Common/src/BP_1I_D32F32C30_TRC_WRA_02.cpp",
+ "Common/src/BP_1I_D16F16Css_TRC_WRA_01_Init.cpp",
+ "Common/src/BP_1I_D16F32Cll_TRC_WRA_01_Init.cpp",
+ "Common/src/BP_1I_D32F32Cll_TRC_WRA_02_Init.cpp",
+ "Common/src/BQ_2I_D32F32Cll_TRC_WRA_01_Init.cpp",
+ "Common/src/BQ_2I_D32F32C30_TRC_WRA_01.cpp",
+ "Common/src/BQ_2I_D16F32C15_TRC_WRA_01.cpp",
+ "Common/src/BQ_2I_D16F32C14_TRC_WRA_01.cpp",
+ "Common/src/BQ_2I_D16F32C13_TRC_WRA_01.cpp",
+ "Common/src/BQ_2I_D16F32Css_TRC_WRA_01_init.cpp",
+ "Common/src/BQ_2I_D16F16C15_TRC_WRA_01.cpp",
+ "Common/src/BQ_2I_D16F16C14_TRC_WRA_01.cpp",
+ "Common/src/BQ_2I_D16F16Css_TRC_WRA_01_Init.cpp",
+ "Common/src/BQ_1I_D16F16C15_TRC_WRA_01.cpp",
+ "Common/src/BQ_1I_D16F16Css_TRC_WRA_01_Init.cpp",
+ "Common/src/BQ_1I_D16F32C14_TRC_WRA_01.cpp",
+ "Common/src/BQ_1I_D16F32Css_TRC_WRA_01_init.cpp",
+ "Common/src/PK_2I_D32F32C30G11_TRC_WRA_01.cpp",
+ "Common/src/PK_2I_D32F32C14G11_TRC_WRA_01.cpp",
+ "Common/src/PK_2I_D32F32CssGss_TRC_WRA_01_Init.cpp",
+ "Common/src/PK_2I_D32F32CllGss_TRC_WRA_01_Init.cpp",
+ "Common/src/Int16LShiftToInt32_16x32.cpp",
+ "Common/src/From2iToMono_16.cpp",
+ "Common/src/Copy_16.cpp",
+ "Common/src/MonoTo2I_16.cpp",
+ "Common/src/MonoTo2I_32.cpp",
+ "Common/src/LoadConst_16.cpp",
+ "Common/src/LoadConst_32.cpp",
+ "Common/src/dB_to_Lin32.cpp",
+ "Common/src/Shift_Sat_v16xv16.cpp",
+ "Common/src/Shift_Sat_v32xv32.cpp",
+ "Common/src/Abs_32.cpp",
+ "Common/src/Int32RShiftToInt16_Sat_32x16.cpp",
+ "Common/src/From2iToMono_32.cpp",
+ "Common/src/mult3s_16x16.cpp",
+ "Common/src/Mult3s_32x16.cpp",
+ "Common/src/NonLinComp_D16.cpp",
+ "Common/src/DelayMix_16x16.cpp",
+ "Common/src/MSTo2i_Sat_16x16.cpp",
+ "Common/src/From2iToMS_16x16.cpp",
+ "Common/src/Mac3s_Sat_16x16.cpp",
+ "Common/src/Mac3s_Sat_32x16.cpp",
+ "Common/src/Add2_Sat_16x16.cpp",
+ "Common/src/Add2_Sat_32x32.cpp",
+ "Common/src/LVC_MixSoft_1St_2i_D16C31_SAT.cpp",
+ "Common/src/LVC_MixSoft_1St_D16C31_SAT.cpp",
+ "Common/src/LVC_Mixer_VarSlope_SetTimeConstant.cpp",
+ "Common/src/LVC_Mixer_SetTimeConstant.cpp",
+ "Common/src/LVC_Mixer_SetTarget.cpp",
+ "Common/src/LVC_Mixer_GetTarget.cpp",
+ "Common/src/LVC_Mixer_Init.cpp",
+ "Common/src/LVC_Core_MixHard_1St_2i_D16C31_SAT.cpp",
+ "Common/src/LVC_Core_MixSoft_1St_2i_D16C31_WRA.cpp",
+ "Common/src/LVC_Core_MixInSoft_D16C31_SAT.cpp",
+ "Common/src/LVC_Mixer_GetCurrent.cpp",
+ "Common/src/LVC_MixSoft_2St_D16C31_SAT.cpp",
+ "Common/src/LVC_Core_MixSoft_1St_D16C31_WRA.cpp",
+ "Common/src/LVC_Core_MixHard_2St_D16C31_SAT.cpp",
+ "Common/src/LVC_MixInSoft_D16C31_SAT.cpp",
+ "Common/src/AGC_MIX_VOL_2St1Mon_D32_WRA.cpp",
+ "Common/src/LVM_Timer.cpp",
+ "Common/src/LVM_Timer_Init.cpp",
],
local_include_dirs: [
@@ -135,10 +135,8 @@
header_libs: [
"libhardware_headers"
],
- cflags: [
+ cppflags: [
"-fvisibility=hidden",
- "-DBUILD_FLOAT",
- "-DHIGHER_FS",
"-DSUPPORT_MC",
"-Wall",
@@ -159,42 +157,42 @@
vendor: true,
srcs: [
- "Reverb/src/LVREV_ApplyNewSettings.c",
- "Reverb/src/LVREV_ClearAudioBuffers.c",
- "Reverb/src/LVREV_GetControlParameters.c",
- "Reverb/src/LVREV_GetInstanceHandle.c",
- "Reverb/src/LVREV_GetMemoryTable.c",
- "Reverb/src/LVREV_Process.c",
- "Reverb/src/LVREV_SetControlParameters.c",
- "Reverb/src/LVREV_Tables.c",
- "Common/src/Abs_32.c",
- "Common/src/InstAlloc.c",
- "Common/src/LoadConst_16.c",
- "Common/src/LoadConst_32.c",
- "Common/src/From2iToMono_32.c",
- "Common/src/Mult3s_32x16.c",
- "Common/src/FO_1I_D32F32C31_TRC_WRA_01.c",
- "Common/src/FO_1I_D32F32Cll_TRC_WRA_01_Init.c",
- "Common/src/DelayAllPass_Sat_32x16To32.c",
- "Common/src/Copy_16.c",
- "Common/src/Mac3s_Sat_32x16.c",
- "Common/src/DelayWrite_32.c",
- "Common/src/Shift_Sat_v32xv32.c",
- "Common/src/Add2_Sat_32x32.c",
- "Common/src/JoinTo2i_32x32.c",
- "Common/src/MonoTo2I_32.c",
- "Common/src/LVM_FO_HPF.c",
- "Common/src/LVM_FO_LPF.c",
- "Common/src/LVM_Polynomial.c",
- "Common/src/LVM_Power10.c",
- "Common/src/LVM_GetOmega.c",
- "Common/src/MixSoft_2St_D32C31_SAT.c",
- "Common/src/MixSoft_1St_D32C31_WRA.c",
- "Common/src/MixInSoft_D32C31_SAT.c",
- "Common/src/LVM_Mixer_TimeConstant.c",
- "Common/src/Core_MixHard_2St_D32C31_SAT.c",
- "Common/src/Core_MixSoft_1St_D32C31_WRA.c",
- "Common/src/Core_MixInSoft_D32C31_SAT.c",
+ "Reverb/src/LVREV_ApplyNewSettings.cpp",
+ "Reverb/src/LVREV_ClearAudioBuffers.cpp",
+ "Reverb/src/LVREV_GetControlParameters.cpp",
+ "Reverb/src/LVREV_GetInstanceHandle.cpp",
+ "Reverb/src/LVREV_GetMemoryTable.cpp",
+ "Reverb/src/LVREV_Process.cpp",
+ "Reverb/src/LVREV_SetControlParameters.cpp",
+ "Reverb/src/LVREV_Tables.cpp",
+ "Common/src/Abs_32.cpp",
+ "Common/src/InstAlloc.cpp",
+ "Common/src/LoadConst_16.cpp",
+ "Common/src/LoadConst_32.cpp",
+ "Common/src/From2iToMono_32.cpp",
+ "Common/src/Mult3s_32x16.cpp",
+ "Common/src/FO_1I_D32F32C31_TRC_WRA_01.cpp",
+ "Common/src/FO_1I_D32F32Cll_TRC_WRA_01_Init.cpp",
+ "Common/src/DelayAllPass_Sat_32x16To32.cpp",
+ "Common/src/Copy_16.cpp",
+ "Common/src/Mac3s_Sat_32x16.cpp",
+ "Common/src/DelayWrite_32.cpp",
+ "Common/src/Shift_Sat_v32xv32.cpp",
+ "Common/src/Add2_Sat_32x32.cpp",
+ "Common/src/JoinTo2i_32x32.cpp",
+ "Common/src/MonoTo2I_32.cpp",
+ "Common/src/LVM_FO_HPF.cpp",
+ "Common/src/LVM_FO_LPF.cpp",
+ "Common/src/LVM_Polynomial.cpp",
+ "Common/src/LVM_Power10.cpp",
+ "Common/src/LVM_GetOmega.cpp",
+ "Common/src/MixSoft_2St_D32C31_SAT.cpp",
+ "Common/src/MixSoft_1St_D32C31_WRA.cpp",
+ "Common/src/MixInSoft_D32C31_SAT.cpp",
+ "Common/src/LVM_Mixer_TimeConstant.cpp",
+ "Common/src/Core_MixHard_2St_D32C31_SAT.cpp",
+ "Common/src/Core_MixSoft_1St_D32C31_WRA.cpp",
+ "Common/src/Core_MixInSoft_D32C31_SAT.cpp",
],
local_include_dirs: [
@@ -206,10 +204,8 @@
"Common/lib",
],
- cflags: [
+ cppflags: [
"-fvisibility=hidden",
- "-DBUILD_FLOAT",
- "-DHIGHER_FS",
"-Wall",
"-Werror",
diff --git a/media/libeffects/lvm/lib/Bass/lib/LVDBE.h b/media/libeffects/lvm/lib/Bass/lib/LVDBE.h
index cc066b0..948d79c 100644
--- a/media/libeffects/lvm/lib/Bass/lib/LVDBE.h
+++ b/media/libeffects/lvm/lib/Bass/lib/LVDBE.h
@@ -55,11 +55,6 @@
#ifndef __LVDBE_H__
#define __LVDBE_H__
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
-
/****************************************************************************************/
/* */
/* Includes */
@@ -68,7 +63,6 @@
#include "LVM_Types.h"
-
/****************************************************************************************/
/* */
/* Definitions */
@@ -85,7 +79,6 @@
#define LVDBE_EFFECT_12DB 12
#define LVDBE_EFFECT_15DB 15
-
/****************************************************************************************/
/* */
/* Types */
@@ -95,7 +88,6 @@
/* Instance handle */
typedef void *LVDBE_Handle_t;
-
/* Operating modes */
typedef enum
{
@@ -104,7 +96,6 @@
LVDBE_MODE_MAX = LVM_MAXINT_32
} LVDBE_Mode_en;
-
/* High pass filter */
typedef enum
{
@@ -113,7 +104,6 @@
LVDBE_HPF_MAX = LVM_MAXINT_32
} LVDBE_FilterSelect_en;
-
/* Volume control */
typedef enum
{
@@ -122,7 +112,6 @@
LVDBE_VOLUME_MAX = LVM_MAXINT_32
} LVDBE_Volume_en;
-
/* Memory Types */
typedef enum
{
@@ -134,7 +123,6 @@
} LVDBE_MemoryTypes_en;
-
/* Function return status */
typedef enum
{
@@ -146,7 +134,6 @@
LVDBE_STATUS_MAX = LVM_MAXINT_32
} LVDBE_ReturnStatus_en;
-
/****************************************************************************************/
/* */
/* Linked enumerated type and capability definitions */
@@ -185,7 +172,6 @@
LVDBE_CENTRE_MAX = LVM_MAXINT_32
} LVDBE_CentreFreq_en;
-
/*
* Supported sample rates in samples per second
*/
@@ -198,12 +184,10 @@
#define LVDBE_CAP_FS_32000 64
#define LVDBE_CAP_FS_44100 128
#define LVDBE_CAP_FS_48000 256
-#if defined(BUILD_FLOAT) && defined(HIGHER_FS)
#define LVDBE_CAP_FS_88200 512
#define LVDBE_CAP_FS_96000 1024
#define LVDBE_CAP_FS_176400 2048
#define LVDBE_CAP_FS_192000 4096
-#endif
typedef enum
{
@@ -216,16 +200,13 @@
LVDBE_FS_32000 = 6,
LVDBE_FS_44100 = 7,
LVDBE_FS_48000 = 8,
-#if defined(BUILD_FLOAT) && defined(HIGHER_FS)
LVDBE_FS_88200 = 9,
LVDBE_FS_96000 = 10,
LVDBE_FS_176400 = 11,
LVDBE_FS_192000 = 12,
-#endif
LVDBE_FS_MAX = LVM_MAXINT_32
} LVDBE_Fs_en;
-
/****************************************************************************************/
/* */
/* Structures */
@@ -241,14 +222,12 @@
void *pBaseAddress; /* Pointer to the region base address */
} LVDBE_MemoryRegion_t;
-
/* Memory table containing the region definitions */
typedef struct
{
LVDBE_MemoryRegion_t Region[LVDBE_NR_MEMORY_REGIONS]; /* One definition for each region */
} LVDBE_MemTab_t;
-
/* Parameter structure */
typedef struct
{
@@ -266,7 +245,6 @@
} LVDBE_Params_t;
-
/* Capability structure */
typedef struct
{
@@ -275,7 +253,6 @@
LVM_UINT16 MaxBlockSize; /* Maximum block size in sample pairs */
} LVDBE_Capabilities_t;
-
/****************************************************************************************/
/* */
/* Function Prototypes */
@@ -317,7 +294,6 @@
LVDBE_MemTab_t *pMemoryTable,
LVDBE_Capabilities_t *pCapabilities);
-
/****************************************************************************************/
/* */
/* FUNCTION: LVDBE_Init */
@@ -355,7 +331,6 @@
LVDBE_MemTab_t *pMemoryTable,
LVDBE_Capabilities_t *pCapabilities);
-
/****************************************************************************************/
/* */
/* FUNCTION: LVDBE_GetParameters */
@@ -379,7 +354,6 @@
LVDBE_ReturnStatus_en LVDBE_GetParameters(LVDBE_Handle_t hInstance,
LVDBE_Params_t *pParams);
-
/****************************************************************************************/
/* */
/* FUNCTION: LVDBE_GetCapabilities */
@@ -403,7 +377,6 @@
LVDBE_ReturnStatus_en LVDBE_GetCapabilities(LVDBE_Handle_t hInstance,
LVDBE_Capabilities_t *pCapabilities);
-
/****************************************************************************************/
/* */
/* FUNCTION: LVDBE_Control */
@@ -444,7 +417,6 @@
LVDBE_ReturnStatus_en LVDBE_Control(LVDBE_Handle_t hInstance,
LVDBE_Params_t *pParams);
-
/****************************************************************************************/
/* */
/* FUNCTION: LVDBE_Process */
@@ -465,20 +437,9 @@
/* NOTES: */
/* */
/****************************************************************************************/
-#ifdef BUILD_FLOAT
LVDBE_ReturnStatus_en LVDBE_Process(LVDBE_Handle_t hInstance,
const LVM_FLOAT *pInData,
LVM_FLOAT *pOutData,
LVM_UINT16 NumSamples);
-#else
-LVDBE_ReturnStatus_en LVDBE_Process(LVDBE_Handle_t hInstance,
- const LVM_INT16 *pInData,
- LVM_INT16 *pOutData,
- LVM_UINT16 NumSamples);
-#endif
-
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
#endif /* __LVDBE_H__ */
diff --git a/media/libeffects/lvm/lib/Bass/src/LVDBE_Coeffs.h b/media/libeffects/lvm/lib/Bass/src/LVDBE_Coeffs.h
index 8f058e8..b364dae 100644
--- a/media/libeffects/lvm/lib/Bass/src/LVDBE_Coeffs.h
+++ b/media/libeffects/lvm/lib/Bass/src/LVDBE_Coeffs.h
@@ -18,8 +18,6 @@
#ifndef __LVDBE_COEFFS_H__
#define __LVDBE_COEFFS_H__
-
-#ifndef BUILD_FLOAT
/************************************************************************************/
/* */
/* General */
@@ -28,504 +26,6 @@
#define LVDBE_SCALESHIFT 10 /* As a power of 2 */
-
-/************************************************************************************/
-/* */
-/* High Pass Filter coefficients */
-/* */
-/************************************************************************************/
-
- /* Coefficients for centre frequency 55Hz */
-#define HPF_Fs8000_Fc55_A0 1029556328 /* Floating point value 0.958849 */
-#define HPF_Fs8000_Fc55_A1 (-2059112655) /* Floating point value -1.917698 */
-#define HPF_Fs8000_Fc55_A2 1029556328 /* Floating point value 0.958849 */
-#define HPF_Fs8000_Fc55_B1 (-2081986375) /* Floating point value -1.939001 */
-#define HPF_Fs8000_Fc55_B2 1010183914 /* Floating point value 0.940807 */
-#define HPF_Fs11025_Fc55_A0 1038210831 /* Floating point value 0.966909 */
-#define HPF_Fs11025_Fc55_A1 (-2076421662) /* Floating point value -1.933818 */
-#define HPF_Fs11025_Fc55_A2 1038210831 /* Floating point value 0.966909 */
-#define HPF_Fs11025_Fc55_B1 (-2099950710) /* Floating point value -1.955732 */
-#define HPF_Fs11025_Fc55_B2 1027238450 /* Floating point value 0.956690 */
-#define HPF_Fs12000_Fc55_A0 1040079943 /* Floating point value 0.968650 */
-#define HPF_Fs12000_Fc55_A1 (-2080159885) /* Floating point value -1.937300 */
-#define HPF_Fs12000_Fc55_A2 1040079943 /* Floating point value 0.968650 */
-#define HPF_Fs12000_Fc55_B1 (-2103811702) /* Floating point value -1.959327 */
-#define HPF_Fs12000_Fc55_B2 1030940477 /* Floating point value 0.960138 */
-#define HPF_Fs16000_Fc55_A0 1045381988 /* Floating point value 0.973588 */
-#define HPF_Fs16000_Fc55_A1 (-2090763976) /* Floating point value -1.947176 */
-#define HPF_Fs16000_Fc55_A2 1045381988 /* Floating point value 0.973588 */
-#define HPF_Fs16000_Fc55_B1 (-2114727793) /* Floating point value -1.969494 */
-#define HPF_Fs16000_Fc55_B2 1041478147 /* Floating point value 0.969952 */
-#define HPF_Fs22050_Fc55_A0 1049766523 /* Floating point value 0.977671 */
-#define HPF_Fs22050_Fc55_A1 (-2099533046) /* Floating point value -1.955343 */
-#define HPF_Fs22050_Fc55_A2 1049766523 /* Floating point value 0.977671 */
-#define HPF_Fs22050_Fc55_B1 (-2123714381) /* Floating point value -1.977863 */
-#define HPF_Fs22050_Fc55_B2 1050232780 /* Floating point value 0.978105 */
-#define HPF_Fs24000_Fc55_A0 1050711051 /* Floating point value 0.978551 */
-#define HPF_Fs24000_Fc55_A1 (-2101422103) /* Floating point value -1.957102 */
-#define HPF_Fs24000_Fc55_A2 1050711051 /* Floating point value 0.978551 */
-#define HPF_Fs24000_Fc55_B1 (-2125645498) /* Floating point value -1.979662 */
-#define HPF_Fs24000_Fc55_B2 1052123526 /* Floating point value 0.979866 */
-#define HPF_Fs32000_Fc55_A0 1053385759 /* Floating point value 0.981042 */
-#define HPF_Fs32000_Fc55_A1 (-2106771519) /* Floating point value -1.962084 */
-#define HPF_Fs32000_Fc55_A2 1053385759 /* Floating point value 0.981042 */
-#define HPF_Fs32000_Fc55_B1 (-2131104794) /* Floating point value -1.984746 */
-#define HPF_Fs32000_Fc55_B2 1057486949 /* Floating point value 0.984861 */
-#define HPF_Fs44100_Fc55_A0 1055592498 /* Floating point value 0.983097 */
-#define HPF_Fs44100_Fc55_A1 (-2111184995) /* Floating point value -1.966194 */
-#define HPF_Fs44100_Fc55_A2 1055592498 /* Floating point value 0.983097 */
-#define HPF_Fs44100_Fc55_B1 (-2135598658) /* Floating point value -1.988931 */
-#define HPF_Fs44100_Fc55_B2 1061922249 /* Floating point value 0.988992 */
-#define HPF_Fs48000_Fc55_A0 1056067276 /* Floating point value 0.983539 */
-#define HPF_Fs48000_Fc55_A1 (-2112134551) /* Floating point value -1.967079 */
-#define HPF_Fs48000_Fc55_A2 1056067276 /* Floating point value 0.983539 */
-#define HPF_Fs48000_Fc55_B1 (-2136564296) /* Floating point value -1.989831 */
-#define HPF_Fs48000_Fc55_B2 1062877714 /* Floating point value 0.989882 */
-
- /* Coefficients for centre frequency 66Hz */
-#define HPF_Fs8000_Fc66_A0 1023293271 /* Floating point value 0.953016 */
-#define HPF_Fs8000_Fc66_A1 (-2046586542) /* Floating point value -1.906032 */
-#define HPF_Fs8000_Fc66_A2 1023293271 /* Floating point value 0.953016 */
-#define HPF_Fs8000_Fc66_B1 (-2068896860) /* Floating point value -1.926810 */
-#define HPF_Fs8000_Fc66_B2 997931110 /* Floating point value 0.929396 */
-#define HPF_Fs11025_Fc66_A0 1033624228 /* Floating point value 0.962638 */
-#define HPF_Fs11025_Fc66_A1 (-2067248455) /* Floating point value -1.925275 */
-#define HPF_Fs11025_Fc66_A2 1033624228 /* Floating point value 0.962638 */
-#define HPF_Fs11025_Fc66_B1 (-2090448000) /* Floating point value -1.946881 */
-#define HPF_Fs11025_Fc66_B2 1018182305 /* Floating point value 0.948256 */
-#define HPF_Fs12000_Fc66_A0 1035857662 /* Floating point value 0.964718 */
-#define HPF_Fs12000_Fc66_A1 (-2071715325) /* Floating point value -1.929435 */
-#define HPF_Fs12000_Fc66_A2 1035857662 /* Floating point value 0.964718 */
-#define HPF_Fs12000_Fc66_B1 (-2095080333) /* Floating point value -1.951196 */
-#define HPF_Fs12000_Fc66_B2 1022587158 /* Floating point value 0.952359 */
-#define HPF_Fs16000_Fc66_A0 1042197528 /* Floating point value 0.970622 */
-#define HPF_Fs16000_Fc66_A1 (-2084395056) /* Floating point value -1.941244 */
-#define HPF_Fs16000_Fc66_A2 1042197528 /* Floating point value 0.970622 */
-#define HPF_Fs16000_Fc66_B1 (-2108177912) /* Floating point value -1.963394 */
-#define HPF_Fs16000_Fc66_B2 1035142690 /* Floating point value 0.964052 */
-#define HPF_Fs22050_Fc66_A0 1047445145 /* Floating point value 0.975509 */
-#define HPF_Fs22050_Fc66_A1 (-2094890289) /* Floating point value -1.951019 */
-#define HPF_Fs22050_Fc66_A2 1047445145 /* Floating point value 0.975509 */
-#define HPF_Fs22050_Fc66_B1 (-2118961025) /* Floating point value -1.973436 */
-#define HPF_Fs22050_Fc66_B2 1045593102 /* Floating point value 0.973784 */
-#define HPF_Fs24000_Fc66_A0 1048576175 /* Floating point value 0.976563 */
-#define HPF_Fs24000_Fc66_A1 (-2097152349) /* Floating point value -1.953125 */
-#define HPF_Fs24000_Fc66_A2 1048576175 /* Floating point value 0.976563 */
-#define HPF_Fs24000_Fc66_B1 (-2121278255) /* Floating point value -1.975594 */
-#define HPF_Fs24000_Fc66_B2 1047852379 /* Floating point value 0.975889 */
-#define HPF_Fs32000_Fc66_A0 1051780119 /* Floating point value 0.979547 */
-#define HPF_Fs32000_Fc66_A1 (-2103560237) /* Floating point value -1.959093 */
-#define HPF_Fs32000_Fc66_A2 1051780119 /* Floating point value 0.979547 */
-#define HPF_Fs32000_Fc66_B1 (-2127829187) /* Floating point value -1.981695 */
-#define HPF_Fs32000_Fc66_B2 1054265623 /* Floating point value 0.981861 */
-#define HPF_Fs44100_Fc66_A0 1054424722 /* Floating point value 0.982010 */
-#define HPF_Fs44100_Fc66_A1 (-2108849444) /* Floating point value -1.964019 */
-#define HPF_Fs44100_Fc66_A2 1054424722 /* Floating point value 0.982010 */
-#define HPF_Fs44100_Fc66_B1 (-2133221723) /* Floating point value -1.986718 */
-#define HPF_Fs44100_Fc66_B2 1059573993 /* Floating point value 0.986805 */
-#define HPF_Fs48000_Fc66_A0 1054993851 /* Floating point value 0.982540 */
-#define HPF_Fs48000_Fc66_A1 (-2109987702) /* Floating point value -1.965079 */
-#define HPF_Fs48000_Fc66_A2 1054993851 /* Floating point value 0.982540 */
-#define HPF_Fs48000_Fc66_B1 (-2134380475) /* Floating point value -1.987797 */
-#define HPF_Fs48000_Fc66_B2 1060718118 /* Floating point value 0.987871 */
-
- /* Coefficients for centre frequency 78Hz */
-#define HPF_Fs8000_Fc78_A0 1016504203 /* Floating point value 0.946693 */
-#define HPF_Fs8000_Fc78_A1 (-2033008405) /* Floating point value -1.893387 */
-#define HPF_Fs8000_Fc78_A2 1016504203 /* Floating point value 0.946693 */
-#define HPF_Fs8000_Fc78_B1 (-2054623390) /* Floating point value -1.913517 */
-#define HPF_Fs8000_Fc78_B2 984733853 /* Floating point value 0.917105 */
-#define HPF_Fs11025_Fc78_A0 1028643741 /* Floating point value 0.957999 */
-#define HPF_Fs11025_Fc78_A1 (-2057287482) /* Floating point value -1.915998 */
-#define HPF_Fs11025_Fc78_A2 1028643741 /* Floating point value 0.957999 */
-#define HPF_Fs11025_Fc78_B1 (-2080083769) /* Floating point value -1.937229 */
-#define HPF_Fs11025_Fc78_B2 1008393904 /* Floating point value 0.939140 */
-#define HPF_Fs12000_Fc78_A0 1031271067 /* Floating point value 0.960446 */
-#define HPF_Fs12000_Fc78_A1 (-2062542133) /* Floating point value -1.920892 */
-#define HPF_Fs12000_Fc78_A2 1031271067 /* Floating point value 0.960446 */
-#define HPF_Fs12000_Fc78_B1 (-2085557048) /* Floating point value -1.942326 */
-#define HPF_Fs12000_Fc78_B2 1013551620 /* Floating point value 0.943944 */
-#define HPF_Fs16000_Fc78_A0 1038734628 /* Floating point value 0.967397 */
-#define HPF_Fs16000_Fc78_A1 (-2077469256) /* Floating point value -1.934794 */
-#define HPF_Fs16000_Fc78_A2 1038734628 /* Floating point value 0.967397 */
-#define HPF_Fs16000_Fc78_B1 (-2101033380) /* Floating point value -1.956740 */
-#define HPF_Fs16000_Fc78_B2 1028275228 /* Floating point value 0.957656 */
-#define HPF_Fs22050_Fc78_A0 1044918584 /* Floating point value 0.973156 */
-#define HPF_Fs22050_Fc78_A1 (-2089837169) /* Floating point value -1.946313 */
-#define HPF_Fs22050_Fc78_A2 1044918584 /* Floating point value 0.973156 */
-#define HPF_Fs22050_Fc78_B1 (-2113775854) /* Floating point value -1.968607 */
-#define HPF_Fs22050_Fc78_B2 1040555007 /* Floating point value 0.969092 */
-#define HPF_Fs24000_Fc78_A0 1046252164 /* Floating point value 0.974398 */
-#define HPF_Fs24000_Fc78_A1 (-2092504328) /* Floating point value -1.948797 */
-#define HPF_Fs24000_Fc78_A2 1046252164 /* Floating point value 0.974398 */
-#define HPF_Fs24000_Fc78_B1 (-2116514229) /* Floating point value -1.971157 */
-#define HPF_Fs24000_Fc78_B2 1043212719 /* Floating point value 0.971568 */
-#define HPF_Fs32000_Fc78_A0 1050031301 /* Floating point value 0.977918 */
-#define HPF_Fs32000_Fc78_A1 (-2100062603) /* Floating point value -1.955836 */
-#define HPF_Fs32000_Fc78_A2 1050031301 /* Floating point value 0.977918 */
-#define HPF_Fs32000_Fc78_B1 (-2124255900) /* Floating point value -1.978367 */
-#define HPF_Fs32000_Fc78_B2 1050762639 /* Floating point value 0.978599 */
-#define HPF_Fs44100_Fc78_A0 1053152258 /* Floating point value 0.980824 */
-#define HPF_Fs44100_Fc78_A1 (-2106304516) /* Floating point value -1.961649 */
-#define HPF_Fs44100_Fc78_A2 1053152258 /* Floating point value 0.980824 */
-#define HPF_Fs44100_Fc78_B1 (-2130628742) /* Floating point value -1.984303 */
-#define HPF_Fs44100_Fc78_B2 1057018180 /* Floating point value 0.984425 */
-#define HPF_Fs48000_Fc78_A0 1053824087 /* Floating point value 0.981450 */
-#define HPF_Fs48000_Fc78_A1 (-2107648173) /* Floating point value -1.962900 */
-#define HPF_Fs48000_Fc78_A2 1053824087 /* Floating point value 0.981450 */
-#define HPF_Fs48000_Fc78_B1 (-2131998154) /* Floating point value -1.985578 */
-#define HPF_Fs48000_Fc78_B2 1058367200 /* Floating point value 0.985681 */
-
- /* Coefficients for centre frequency 90Hz */
-#define HPF_Fs8000_Fc90_A0 1009760053 /* Floating point value 0.940412 */
-#define HPF_Fs8000_Fc90_A1 (-2019520105) /* Floating point value -1.880825 */
-#define HPF_Fs8000_Fc90_A2 1009760053 /* Floating point value 0.940412 */
-#define HPF_Fs8000_Fc90_B1 (-2040357139) /* Floating point value -1.900231 */
-#define HPF_Fs8000_Fc90_B2 971711129 /* Floating point value 0.904977 */
-#define HPF_Fs11025_Fc90_A0 1023687217 /* Floating point value 0.953383 */
-#define HPF_Fs11025_Fc90_A1 (-2047374434) /* Floating point value -1.906766 */
-#define HPF_Fs11025_Fc90_A2 1023687217 /* Floating point value 0.953383 */
-#define HPF_Fs11025_Fc90_B1 (-2069722397) /* Floating point value -1.927579 */
-#define HPF_Fs11025_Fc90_B2 998699604 /* Floating point value 0.930111 */
-#define HPF_Fs12000_Fc90_A0 1026704754 /* Floating point value 0.956193 */
-#define HPF_Fs12000_Fc90_A1 (-2053409508) /* Floating point value -1.912387 */
-#define HPF_Fs12000_Fc90_A2 1026704754 /* Floating point value 0.956193 */
-#define HPF_Fs12000_Fc90_B1 (-2076035996) /* Floating point value -1.933459 */
-#define HPF_Fs12000_Fc90_B2 1004595918 /* Floating point value 0.935603 */
-#define HPF_Fs16000_Fc90_A0 1035283225 /* Floating point value 0.964183 */
-#define HPF_Fs16000_Fc90_A1 (-2070566451) /* Floating point value -1.928365 */
-#define HPF_Fs16000_Fc90_A2 1035283225 /* Floating point value 0.964183 */
-#define HPF_Fs16000_Fc90_B1 (-2093889811) /* Floating point value -1.950087 */
-#define HPF_Fs16000_Fc90_B2 1021453326 /* Floating point value 0.951303 */
-#define HPF_Fs22050_Fc90_A0 1042398116 /* Floating point value 0.970809 */
-#define HPF_Fs22050_Fc90_A1 (-2084796232) /* Floating point value -1.941618 */
-#define HPF_Fs22050_Fc90_A2 1042398116 /* Floating point value 0.970809 */
-#define HPF_Fs22050_Fc90_B1 (-2108591057) /* Floating point value -1.963778 */
-#define HPF_Fs22050_Fc90_B2 1035541188 /* Floating point value 0.964423 */
-#define HPF_Fs24000_Fc90_A0 1043933302 /* Floating point value 0.972239 */
-#define HPF_Fs24000_Fc90_A1 (-2087866604) /* Floating point value -1.944477 */
-#define HPF_Fs24000_Fc90_A2 1043933302 /* Floating point value 0.972239 */
-#define HPF_Fs24000_Fc90_B1 (-2111750495) /* Floating point value -1.966721 */
-#define HPF_Fs24000_Fc90_B2 1038593601 /* Floating point value 0.967266 */
-#define HPF_Fs32000_Fc90_A0 1048285391 /* Floating point value 0.976292 */
-#define HPF_Fs32000_Fc90_A1 (-2096570783) /* Floating point value -1.952584 */
-#define HPF_Fs32000_Fc90_A2 1048285391 /* Floating point value 0.976292 */
-#define HPF_Fs32000_Fc90_B1 (-2120682737) /* Floating point value -1.975040 */
-#define HPF_Fs32000_Fc90_B2 1047271295 /* Floating point value 0.975347 */
-#define HPF_Fs44100_Fc90_A0 1051881330 /* Floating point value 0.979641 */
-#define HPF_Fs44100_Fc90_A1 (-2103762660) /* Floating point value -1.959282 */
-#define HPF_Fs44100_Fc90_A2 1051881330 /* Floating point value 0.979641 */
-#define HPF_Fs44100_Fc90_B1 (-2128035809) /* Floating point value -1.981888 */
-#define HPF_Fs44100_Fc90_B2 1054468533 /* Floating point value 0.982050 */
-#define HPF_Fs48000_Fc90_A0 1052655619 /* Floating point value 0.980362 */
-#define HPF_Fs48000_Fc90_A1 (-2105311238) /* Floating point value -1.960724 */
-#define HPF_Fs48000_Fc90_A2 1052655619 /* Floating point value 0.980362 */
-#define HPF_Fs48000_Fc90_B1 (-2129615871) /* Floating point value -1.983359 */
-#define HPF_Fs48000_Fc90_B2 1056021492 /* Floating point value 0.983497 */
-
-
-/************************************************************************************/
-/* */
-/* Band Pass Filter coefficients */
-/* */
-/************************************************************************************/
-
- /* Coefficients for centre frequency 55Hz */
-#define BPF_Fs8000_Fc55_A0 9875247 /* Floating point value 0.009197 */
-#define BPF_Fs8000_Fc55_A1 0 /* Floating point value 0.000000 */
-#define BPF_Fs8000_Fc55_A2 (-9875247) /* Floating point value -0.009197 */
-#define BPF_Fs8000_Fc55_B1 (-2125519830) /* Floating point value -1.979545 */
-#define BPF_Fs8000_Fc55_B2 1053762629 /* Floating point value 0.981393 */
-#define BPF_Fs11025_Fc55_A0 7183952 /* Floating point value 0.006691 */
-#define BPF_Fs11025_Fc55_A1 0 /* Floating point value 0.000000 */
-#define BPF_Fs11025_Fc55_A2 (-7183952) /* Floating point value -0.006691 */
-#define BPF_Fs11025_Fc55_B1 (-2131901658) /* Floating point value -1.985488 */
-#define BPF_Fs11025_Fc55_B2 1059207548 /* Floating point value 0.986464 */
-#define BPF_Fs12000_Fc55_A0 6603871 /* Floating point value 0.006150 */
-#define BPF_Fs12000_Fc55_A1 0 /* Floating point value 0.000000 */
-#define BPF_Fs12000_Fc55_A2 (-6603871) /* Floating point value -0.006150 */
-#define BPF_Fs12000_Fc55_B1 (-2133238092) /* Floating point value -1.986733 */
-#define BPF_Fs12000_Fc55_B2 1060381143 /* Floating point value 0.987557 */
-#define BPF_Fs16000_Fc55_A0 4960591 /* Floating point value 0.004620 */
-#define BPF_Fs16000_Fc55_A1 0 /* Floating point value 0.000000 */
-#define BPF_Fs16000_Fc55_A2 (-4960591) /* Floating point value -0.004620 */
-#define BPF_Fs16000_Fc55_B1 (-2136949052) /* Floating point value -1.990189 */
-#define BPF_Fs16000_Fc55_B2 1063705760 /* Floating point value 0.990653 */
-#define BPF_Fs22050_Fc55_A0 3604131 /* Floating point value 0.003357 */
-#define BPF_Fs22050_Fc55_A1 0 /* Floating point value 0.000000 */
-#define BPF_Fs22050_Fc55_A2 (-3604131) /* Floating point value -0.003357 */
-#define BPF_Fs22050_Fc55_B1 (-2139929085) /* Floating point value -1.992964 */
-#define BPF_Fs22050_Fc55_B2 1066450095 /* Floating point value 0.993209 */
-#define BPF_Fs24000_Fc55_A0 3312207 /* Floating point value 0.003085 */
-#define BPF_Fs24000_Fc55_A1 0 /* Floating point value 0.000000 */
-#define BPF_Fs24000_Fc55_A2 (-3312207) /* Floating point value -0.003085 */
-#define BPF_Fs24000_Fc55_B1 (-2140560606) /* Floating point value -1.993552 */
-#define BPF_Fs24000_Fc55_B2 1067040703 /* Floating point value 0.993759 */
-#define BPF_Fs32000_Fc55_A0 2486091 /* Floating point value 0.002315 */
-#define BPF_Fs32000_Fc55_A1 0 /* Floating point value 0.000000 */
-#define BPF_Fs32000_Fc55_A2 (-2486091) /* Floating point value -0.002315 */
-#define BPF_Fs32000_Fc55_B1 (-2142328962) /* Floating point value -1.995199 */
-#define BPF_Fs32000_Fc55_B2 1068712067 /* Floating point value 0.995316 */
-#define BPF_Fs44100_Fc55_A0 1805125 /* Floating point value 0.001681 */
-#define BPF_Fs44100_Fc55_A1 0 /* Floating point value 0.000000 */
-#define BPF_Fs44100_Fc55_A2 (-1805125) /* Floating point value -0.001681 */
-#define BPF_Fs44100_Fc55_B1 (-2143765772) /* Floating point value -1.996537 */
-#define BPF_Fs44100_Fc55_B2 1070089770 /* Floating point value 0.996599 */
-#define BPF_Fs48000_Fc55_A0 1658687 /* Floating point value 0.001545 */
-#define BPF_Fs48000_Fc55_A1 0 /* Floating point value 0.000000 */
-#define BPF_Fs48000_Fc55_A2 (-1658687) /* Floating point value -0.001545 */
-#define BPF_Fs48000_Fc55_B1 (-2144072292) /* Floating point value -1.996823 */
-#define BPF_Fs48000_Fc55_B2 1070386036 /* Floating point value 0.996875 */
-
- /* Coefficients for centre frequency 66Hz */
-#define BPF_Fs8000_Fc66_A0 13580189 /* Floating point value 0.012648 */
-#define BPF_Fs8000_Fc66_A1 0 /* Floating point value 0.000000 */
-#define BPF_Fs8000_Fc66_A2 (-13580189) /* Floating point value -0.012648 */
-#define BPF_Fs8000_Fc66_B1 (-2117161175) /* Floating point value -1.971760 */
-#define BPF_Fs8000_Fc66_B2 1046266945 /* Floating point value 0.974412 */
-#define BPF_Fs11025_Fc66_A0 9888559 /* Floating point value 0.009209 */
-#define BPF_Fs11025_Fc66_A1 0 /* Floating point value 0.000000 */
-#define BPF_Fs11025_Fc66_A2 (-9888559) /* Floating point value -0.009209 */
-#define BPF_Fs11025_Fc66_B1 (-2125972738) /* Floating point value -1.979966 */
-#define BPF_Fs11025_Fc66_B2 1053735698 /* Floating point value 0.981368 */
-#define BPF_Fs12000_Fc66_A0 9091954 /* Floating point value 0.008468 */
-#define BPF_Fs12000_Fc66_A1 0 /* Floating point value 0.000000 */
-#define BPF_Fs12000_Fc66_A2 (-9091954) /* Floating point value -0.008468 */
-#define BPF_Fs12000_Fc66_B1 (-2127818004) /* Floating point value -1.981685 */
-#define BPF_Fs12000_Fc66_B2 1055347356 /* Floating point value 0.982869 */
-#define BPF_Fs16000_Fc66_A0 6833525 /* Floating point value 0.006364 */
-#define BPF_Fs16000_Fc66_A1 0 /* Floating point value 0.000000 */
-#define BPF_Fs16000_Fc66_A2 (-6833525) /* Floating point value -0.006364 */
-#define BPF_Fs16000_Fc66_B1 (-2132941739) /* Floating point value -1.986457 */
-#define BPF_Fs16000_Fc66_B2 1059916517 /* Floating point value 0.987124 */
-#define BPF_Fs22050_Fc66_A0 4967309 /* Floating point value 0.004626 */
-#define BPF_Fs22050_Fc66_A1 0 /* Floating point value 0.000000 */
-#define BPF_Fs22050_Fc66_A2 (-4967309) /* Floating point value -0.004626 */
-#define BPF_Fs22050_Fc66_B1 (-2137056003) /* Floating point value -1.990288 */
-#define BPF_Fs22050_Fc66_B2 1063692170 /* Floating point value 0.990641 */
-#define BPF_Fs24000_Fc66_A0 4565445 /* Floating point value 0.004252 */
-#define BPF_Fs24000_Fc66_A1 0 /* Floating point value 0.000000 */
-#define BPF_Fs24000_Fc66_A2 (-4565445) /* Floating point value -0.004252 */
-#define BPF_Fs24000_Fc66_B1 (-2137927842) /* Floating point value -1.991100 */
-#define BPF_Fs24000_Fc66_B2 1064505202 /* Floating point value 0.991398 */
-#define BPF_Fs32000_Fc66_A0 3427761 /* Floating point value 0.003192 */
-#define BPF_Fs32000_Fc66_A1 0 /* Floating point value 0.000000 */
-#define BPF_Fs32000_Fc66_A2 (-3427761) /* Floating point value -0.003192 */
-#define BPF_Fs32000_Fc66_B1 (-2140369007) /* Floating point value -1.993374 */
-#define BPF_Fs32000_Fc66_B2 1066806920 /* Floating point value 0.993541 */
-#define BPF_Fs44100_Fc66_A0 2489466 /* Floating point value 0.002318 */
-#define BPF_Fs44100_Fc66_A1 0 /* Floating point value 0.000000 */
-#define BPF_Fs44100_Fc66_A2 (-2489466) /* Floating point value -0.002318 */
-#define BPF_Fs44100_Fc66_B1 (-2142352342) /* Floating point value -1.995221 */
-#define BPF_Fs44100_Fc66_B2 1068705240 /* Floating point value 0.995309 */
-#define BPF_Fs48000_Fc66_A0 2287632 /* Floating point value 0.002131 */
-#define BPF_Fs48000_Fc66_A1 0 /* Floating point value 0.000000 */
-#define BPF_Fs48000_Fc66_A2 (-2287632) /* Floating point value -0.002131 */
-#define BPF_Fs48000_Fc66_B1 (-2142775436) /* Floating point value -1.995615 */
-#define BPF_Fs48000_Fc66_B2 1069113581 /* Floating point value 0.995690 */
-
- /* Coefficients for centre frequency 78Hz */
-#define BPF_Fs8000_Fc78_A0 19941180 /* Floating point value 0.018572 */
-#define BPF_Fs8000_Fc78_A1 0 /* Floating point value 0.000000 */
-#define BPF_Fs8000_Fc78_A2 (-19941180) /* Floating point value -0.018572 */
-#define BPF_Fs8000_Fc78_B1 (-2103186749) /* Floating point value -1.958745 */
-#define BPF_Fs8000_Fc78_B2 1033397648 /* Floating point value 0.962427 */
-#define BPF_Fs11025_Fc78_A0 14543934 /* Floating point value 0.013545 */
-#define BPF_Fs11025_Fc78_A1 0 /* Floating point value 0.000000 */
-#define BPF_Fs11025_Fc78_A2 (-14543934) /* Floating point value -0.013545 */
-#define BPF_Fs11025_Fc78_B1 (-2115966638) /* Floating point value -1.970647 */
-#define BPF_Fs11025_Fc78_B2 1044317135 /* Floating point value 0.972596 */
-#define BPF_Fs12000_Fc78_A0 13376999 /* Floating point value 0.012458 */
-#define BPF_Fs12000_Fc78_A1 0 /* Floating point value 0.000000 */
-#define BPF_Fs12000_Fc78_A2 (-13376999) /* Floating point value -0.012458 */
-#define BPF_Fs12000_Fc78_B1 (-2118651708) /* Floating point value -1.973148 */
-#define BPF_Fs12000_Fc78_B2 1046678029 /* Floating point value 0.974795 */
-#define BPF_Fs16000_Fc78_A0 10064222 /* Floating point value 0.009373 */
-#define BPF_Fs16000_Fc78_A1 0 /* Floating point value 0.000000 */
-#define BPF_Fs16000_Fc78_A2 (-10064222) /* Floating point value -0.009373 */
-#define BPF_Fs16000_Fc78_B1 (-2126124342) /* Floating point value -1.980108 */
-#define BPF_Fs16000_Fc78_B2 1053380304 /* Floating point value 0.981037 */
-#define BPF_Fs22050_Fc78_A0 7321780 /* Floating point value 0.006819 */
-#define BPF_Fs22050_Fc78_A1 0 /* Floating point value 0.000000 */
-#define BPF_Fs22050_Fc78_A2 (-7321780) /* Floating point value -0.006819 */
-#define BPF_Fs22050_Fc78_B1 (-2132143771) /* Floating point value -1.985714 */
-#define BPF_Fs22050_Fc78_B2 1058928700 /* Floating point value 0.986204 */
-#define BPF_Fs24000_Fc78_A0 6730640 /* Floating point value 0.006268 */
-#define BPF_Fs24000_Fc78_A1 0 /* Floating point value 0.000000 */
-#define BPF_Fs24000_Fc78_A2 (-6730640) /* Floating point value -0.006268 */
-#define BPF_Fs24000_Fc78_B1 (-2133421607) /* Floating point value -1.986904 */
-#define BPF_Fs24000_Fc78_B2 1060124669 /* Floating point value 0.987318 */
-#define BPF_Fs32000_Fc78_A0 5055965 /* Floating point value 0.004709 */
-#define BPF_Fs32000_Fc78_A1 0 /* Floating point value 0.000000 */
-#define BPF_Fs32000_Fc78_A2 (-5055965) /* Floating point value -0.004709 */
-#define BPF_Fs32000_Fc78_B1 (-2137003977) /* Floating point value -1.990240 */
-#define BPF_Fs32000_Fc78_B2 1063512802 /* Floating point value 0.990473 */
-#define BPF_Fs44100_Fc78_A0 3673516 /* Floating point value 0.003421 */
-#define BPF_Fs44100_Fc78_A1 0 /* Floating point value 0.000000 */
-#define BPF_Fs44100_Fc78_A2 (-3673516) /* Floating point value -0.003421 */
-#define BPF_Fs44100_Fc78_B1 (-2139919394) /* Floating point value -1.992955 */
-#define BPF_Fs44100_Fc78_B2 1066309718 /* Floating point value 0.993078 */
-#define BPF_Fs48000_Fc78_A0 3375990 /* Floating point value 0.003144 */
-#define BPF_Fs48000_Fc78_A1 0 /* Floating point value 0.000000 */
-#define BPF_Fs48000_Fc78_A2 (-3375990) /* Floating point value -0.003144 */
-#define BPF_Fs48000_Fc78_B1 (-2140541906) /* Floating point value -1.993535 */
-#define BPF_Fs48000_Fc78_B2 1066911660 /* Floating point value 0.993639 */
-
- /* Coefficients for centre frequency 90Hz */
-#define BPF_Fs8000_Fc90_A0 24438548 /* Floating point value 0.022760 */
-#define BPF_Fs8000_Fc90_A1 0 /* Floating point value 0.000000 */
-#define BPF_Fs8000_Fc90_A2 (-24438548) /* Floating point value -0.022760 */
-#define BPF_Fs8000_Fc90_B1 (-2092801347) /* Floating point value -1.949073 */
-#define BPF_Fs8000_Fc90_B2 1024298757 /* Floating point value 0.953953 */
-#define BPF_Fs11025_Fc90_A0 17844385 /* Floating point value 0.016619 */
-#define BPF_Fs11025_Fc90_A1 0 /* Floating point value 0.000000 */
-#define BPF_Fs11025_Fc90_A2 (-17844385) /* Floating point value -0.016619 */
-#define BPF_Fs11025_Fc90_B1 (-2108604921) /* Floating point value -1.963791 */
-#define BPF_Fs11025_Fc90_B2 1037639797 /* Floating point value 0.966377 */
-#define BPF_Fs12000_Fc90_A0 16416707 /* Floating point value 0.015289 */
-#define BPF_Fs12000_Fc90_A1 0 /* Floating point value 0.000000 */
-#define BPF_Fs12000_Fc90_A2 (-16416707) /* Floating point value -0.015289 */
-#define BPF_Fs12000_Fc90_B1 (-2111922936) /* Floating point value -1.966882 */
-#define BPF_Fs12000_Fc90_B2 1040528216 /* Floating point value 0.969067 */
-#define BPF_Fs16000_Fc90_A0 12359883 /* Floating point value 0.011511 */
-#define BPF_Fs16000_Fc90_A1 0 /* Floating point value 0.000000 */
-#define BPF_Fs16000_Fc90_A2 (-12359883) /* Floating point value -0.011511 */
-#define BPF_Fs16000_Fc90_B1 (-2121152162) /* Floating point value -1.975477 */
-#define BPF_Fs16000_Fc90_B2 1048735817 /* Floating point value 0.976711 */
-#define BPF_Fs22050_Fc90_A0 8997173 /* Floating point value 0.008379 */
-#define BPF_Fs22050_Fc90_A1 0 /* Floating point value 0.000000 */
-#define BPF_Fs22050_Fc90_A2 (-8997173) /* Floating point value -0.008379 */
-#define BPF_Fs22050_Fc90_B1 (-2128580762) /* Floating point value -1.982395 */
-#define BPF_Fs22050_Fc90_B2 1055539113 /* Floating point value 0.983047 */
-#define BPF_Fs24000_Fc90_A0 8271818 /* Floating point value 0.007704 */
-#define BPF_Fs24000_Fc90_A1 0 /* Floating point value 0.000000 */
-#define BPF_Fs24000_Fc90_A2 (-8271818) /* Floating point value -0.007704 */
-#define BPF_Fs24000_Fc90_B1 (-2130157013) /* Floating point value -1.983863 */
-#define BPF_Fs24000_Fc90_B2 1057006621 /* Floating point value 0.984414 */
-#define BPF_Fs32000_Fc90_A0 6215918 /* Floating point value 0.005789 */
-#define BPF_Fs32000_Fc90_A1 0 /* Floating point value 0.000000 */
-#define BPF_Fs32000_Fc90_A2 (-6215918) /* Floating point value -0.005789 */
-#define BPF_Fs32000_Fc90_B1 (-2134574521) /* Floating point value -1.987977 */
-#define BPF_Fs32000_Fc90_B2 1061166033 /* Floating point value 0.988288 */
-#define BPF_Fs44100_Fc90_A0 4517651 /* Floating point value 0.004207 */
-#define BPF_Fs44100_Fc90_A1 0 /* Floating point value 0.000000 */
-#define BPF_Fs44100_Fc90_A2 (-4517651) /* Floating point value -0.004207 */
-#define BPF_Fs44100_Fc90_B1 (-2138167926) /* Floating point value -1.991324 */
-#define BPF_Fs44100_Fc90_B2 1064601898 /* Floating point value 0.991488 */
-#define BPF_Fs48000_Fc90_A0 4152024 /* Floating point value 0.003867 */
-#define BPF_Fs48000_Fc90_A1 0 /* Floating point value 0.000000 */
-#define BPF_Fs48000_Fc90_A2 (-4152024) /* Floating point value -0.003867 */
-#define BPF_Fs48000_Fc90_B1 (-2138935002) /* Floating point value -1.992038 */
-#define BPF_Fs48000_Fc90_B2 1065341620 /* Floating point value 0.992177 */
-
-
-/************************************************************************************/
-/* */
-/* Automatic Gain Control time constants and gain settings */
-/* */
-/************************************************************************************/
-
-/* AGC Time constants */
-#define AGC_ATTACK_Fs8000 27571 /* Floating point value 0.841395 */
-#define AGC_ATTACK_Fs11025 28909 /* Floating point value 0.882223 */
-#define AGC_ATTACK_Fs12000 29205 /* Floating point value 0.891251 */
-#define AGC_ATTACK_Fs16000 30057 /* Floating point value 0.917276 */
-#define AGC_ATTACK_Fs22050 30778 /* Floating point value 0.939267 */
-#define AGC_ATTACK_Fs24000 30935 /* Floating point value 0.944061 */
-#define AGC_ATTACK_Fs32000 31383 /* Floating point value 0.957745 */
-#define AGC_ATTACK_Fs44100 31757 /* Floating point value 0.969158 */
-#define AGC_ATTACK_Fs48000 31838 /* Floating point value 0.971628 */
-#define DECAY_SHIFT 10 /* As a power of 2 */
-#define AGC_DECAY_Fs8000 44 /* Floating point value 0.000042 */
-#define AGC_DECAY_Fs11025 32 /* Floating point value 0.000030 */
-#define AGC_DECAY_Fs12000 29 /* Floating point value 0.000028 */
-#define AGC_DECAY_Fs16000 22 /* Floating point value 0.000021 */
-#define AGC_DECAY_Fs22050 16 /* Floating point value 0.000015 */
-#define AGC_DECAY_Fs24000 15 /* Floating point value 0.000014 */
-#define AGC_DECAY_Fs32000 11 /* Floating point value 0.000010 */
-#define AGC_DECAY_Fs44100 8 /* Floating point value 0.000008 */
-#define AGC_DECAY_Fs48000 7 /* Floating point value 0.000007 */
-
-/* AGC Gain settings */
-#define AGC_GAIN_SCALE 31 /* As a power of 2 */
-#define AGC_GAIN_SHIFT 4 /* As a power of 2 */
-#define AGC_TARGETLEVEL 33170337 /* Floating point value -0.100000dB */
-#define AGC_HPFGAIN_0dB 110739704 /* Floating point value 0.412538 */
-#define AGC_GAIN_0dB 0 /* Floating point value 0.000000 */
-#define AGC_HPFGAIN_1dB 157006071 /* Floating point value 0.584893 */
-#define AGC_GAIN_1dB 32754079 /* Floating point value 0.122018 */
-#define AGC_HPFGAIN_2dB 208917788 /* Floating point value 0.778279 */
-#define AGC_GAIN_2dB 69504761 /* Floating point value 0.258925 */
-#define AGC_HPFGAIN_3dB 267163693 /* Floating point value 0.995262 */
-#define AGC_GAIN_3dB 110739704 /* Floating point value 0.412538 */
-#define AGC_HPFGAIN_4dB 332516674 /* Floating point value 1.238721 */
-#define AGC_GAIN_4dB 157006071 /* Floating point value 0.584893 */
-#define AGC_HPFGAIN_5dB 405843924 /* Floating point value 1.511886 */
-#define AGC_GAIN_5dB 208917788 /* Floating point value 0.778279 */
-#define AGC_HPFGAIN_6dB 488118451 /* Floating point value 1.818383 */
-#define AGC_GAIN_6dB 267163693 /* Floating point value 0.995262 */
-#define AGC_HPFGAIN_7dB 580431990 /* Floating point value 2.162278 */
-#define AGC_GAIN_7dB 332516674 /* Floating point value 1.238721 */
-#define AGC_HPFGAIN_8dB 684009483 /* Floating point value 2.548134 */
-#define AGC_GAIN_8dB 405843924 /* Floating point value 1.511886 */
-#define AGC_HPFGAIN_9dB 800225343 /* Floating point value 2.981072 */
-#define AGC_GAIN_9dB 488118451 /* Floating point value 1.818383 */
-#define AGC_HPFGAIN_10dB 930621681 /* Floating point value 3.466836 */
-#define AGC_GAIN_10dB 580431990 /* Floating point value 2.162278 */
-#define AGC_HPFGAIN_11dB 1076928780 /* Floating point value 4.011872 */
-#define AGC_GAIN_11dB 684009483 /* Floating point value 2.548134 */
-#define AGC_HPFGAIN_12dB 1241088045 /* Floating point value 4.623413 */
-#define AGC_GAIN_12dB 800225343 /* Floating point value 2.981072 */
-#define AGC_HPFGAIN_13dB 1425277769 /* Floating point value 5.309573 */
-#define AGC_GAIN_13dB 930621681 /* Floating point value 3.466836 */
-#define AGC_HPFGAIN_14dB 1631942039 /* Floating point value 6.079458 */
-#define AGC_GAIN_14dB 1076928780 /* Floating point value 4.011872 */
-#define AGC_HPFGAIN_15dB 1863823163 /* Floating point value 6.943282 */
-#define AGC_GAIN_15dB 1241088045 /* Floating point value 4.623413 */
-
-
-/************************************************************************************/
-/* */
-/* Volume control */
-/* */
-/************************************************************************************/
-
-/* Volume control gain */
-#define VOLUME_MAX 0 /* In dBs */
-#define VOLUME_SHIFT 0 /* In dBs */
-
-/* Volume control time constants */
-#define VOL_TC_SHIFT 21 /* As a power of 2 */
-#define VOL_TC_Fs8000 25889 /* Floating point value 0.024690 */
-#define VOL_TC_Fs11025 18850 /* Floating point value 0.017977 */
-#define VOL_TC_Fs12000 17331 /* Floating point value 0.016529 */
-#define VOL_TC_Fs16000 13026 /* Floating point value 0.012422 */
-#define VOL_TC_Fs22050 9468 /* Floating point value 0.009029 */
-#define VOL_TC_Fs24000 8702 /* Floating point value 0.008299 */
-#define VOL_TC_Fs32000 6533 /* Floating point value 0.006231 */
-#define VOL_TC_Fs44100 4745 /* Floating point value 0.004525 */
-#define VOL_TC_Fs48000 4360 /* Floating point value 0.004158 */
-#define MIX_TC_Fs8000 29365 /* Floating point value 0.896151 */
-#define MIX_TC_Fs11025 30230 /* Floating point value 0.922548 */
-#define MIX_TC_Fs12000 30422 /* Floating point value 0.928415 */
-#define MIX_TC_Fs16000 30978 /* Floating point value 0.945387 */
-#define MIX_TC_Fs22050 31451 /* Floating point value 0.959804 */
-#define MIX_TC_Fs24000 31554 /* Floating point value 0.962956 */
-#define MIX_TC_Fs32000 31850 /* Floating point value 0.971973 */
-#define MIX_TC_Fs44100 32097 /* Floating point value 0.979515 */
-#define MIX_TC_Fs48000 32150 /* Floating point value 0.981150 */
-
-#else /*BUILD_FLOAT*/
-
-/************************************************************************************/
-/* */
-/* General */
-/* */
-/************************************************************************************/
-
-#define LVDBE_SCALESHIFT 10 /* As a power of 2 */
-
-
/************************************************************************************/
/* */
/* High Pass Filter coefficients */
@@ -579,7 +79,6 @@
#define HPF_Fs48000_Fc55_B1 (-1.989831f)
#define HPF_Fs48000_Fc55_B2 0.989882f
-#ifdef HIGHER_FS
#define HPF_Fs88200_Fc55_A0 0.985818f
#define HPF_Fs88200_Fc55_A1 (-1.971636f)
#define HPF_Fs88200_Fc55_A2 0.985818f
@@ -603,8 +102,6 @@
#define HPF_Fs192000_Fc55_A2 0.987294f
#define HPF_Fs192000_Fc55_B1 (-1.997458f)
#define HPF_Fs192000_Fc55_B2 0.997461f
-#endif
-
/* Coefficients for centre frequency 66Hz */
#define HPF_Fs8000_Fc66_A0 0.953016f
@@ -653,7 +150,6 @@
#define HPF_Fs48000_Fc66_B1 (-1.987797f)
#define HPF_Fs48000_Fc66_B2 0.987871f
-#ifdef HIGHER_FS
#define HPF_Fs88200_Fc66_A0 0.985273f
#define HPF_Fs88200_Fc66_A1 (-1.970546f)
#define HPF_Fs88200_Fc66_A2 0.985273f
@@ -677,7 +173,6 @@
#define HPF_Fs192000_Fc66_A2 0.987043f
#define HPF_Fs192000_Fc66_B1 (-1.996949f)
#define HPF_Fs192000_Fc66_B2 0.996954f
-#endif
/* Coefficients for centre frequency 78Hz */
#define HPF_Fs8000_Fc78_A0 0.946693f
@@ -726,7 +221,6 @@
#define HPF_Fs48000_Fc78_B1 (-1.985578f)
#define HPF_Fs48000_Fc78_B2 0.985681f
-#ifdef HIGHER_FS
#define HPF_Fs88200_Fc78_A0 0.984678f
#define HPF_Fs88200_Fc78_A1 (-1.969356f)
#define HPF_Fs88200_Fc78_A2 0.984678f
@@ -750,7 +244,6 @@
#define HPF_Fs192000_Fc78_A2 0.986769f
#define HPF_Fs192000_Fc78_B1 (-1.996394f)
#define HPF_Fs192000_Fc78_B2 0.996401f
-#endif
/* Coefficients for centre frequency 90Hz */
#define HPF_Fs8000_Fc90_A0 0.940412f
@@ -799,7 +292,6 @@
#define HPF_Fs48000_Fc90_B1 (-1.983359f)
#define HPF_Fs48000_Fc90_B2 0.983497f
-#ifdef HIGHER_FS
#define HPF_Fs88200_Fc90_A0 0.984084f
#define HPF_Fs88200_Fc90_A1 (-1.968168f)
#define HPF_Fs88200_Fc90_A2 0.984084f
@@ -823,7 +315,6 @@
#define HPF_Fs192000_Fc90_A2 0.986496f
#define HPF_Fs192000_Fc90_B1 (-1.995840f)
#define HPF_Fs192000_Fc90_B2 0.995848f
-#endif
/************************************************************************************/
/* */
@@ -878,7 +369,6 @@
#define BPF_Fs48000_Fc55_B1 (-1.996823f)
#define BPF_Fs48000_Fc55_B2 0.996875f
-#ifdef HIGHER_FS
#define BPF_Fs88200_Fc55_A0 0.000831f
#define BPF_Fs88200_Fc55_A1 0.000000f
#define BPF_Fs88200_Fc55_A2 (-0.000831f)
@@ -902,7 +392,6 @@
#define BPF_Fs192000_Fc55_A2 (-0.000381f)
#define BPF_Fs192000_Fc55_B1 (-1.999234f)
#define BPF_Fs192000_Fc55_B2 0.999238f
-#endif
/* Coefficients for centre frequency 66Hz */
#define BPF_Fs8000_Fc66_A0 0.012648f
@@ -951,7 +440,6 @@
#define BPF_Fs48000_Fc66_B1 (-1.995615f)
#define BPF_Fs48000_Fc66_B2 0.995690f
-#ifdef HIGHER_FS
#define BPF_Fs88200_Fc66_A0 0.001146f
#define BPF_Fs88200_Fc66_A1 0.000000f
#define BPF_Fs88200_Fc66_A2 (-0.001146f)
@@ -975,7 +463,6 @@
#define BPF_Fs192000_Fc66_A2 (-0.000528f)
#define BPF_Fs192000_Fc66_B1 (-1.998939f)
#define BPF_Fs192000_Fc66_B2 0.998945f
-#endif
/* Coefficients for centre frequency 78Hz */
#define BPF_Fs8000_Fc78_A0 0.018572f
@@ -1024,7 +511,6 @@
#define BPF_Fs48000_Fc78_B1 (-1.993535f)
#define BPF_Fs48000_Fc78_B2 0.993639f
-#ifdef HIGHER_FS
#define BPF_Fs88200_Fc78_A0 0.001693f
#define BPF_Fs88200_Fc78_A1 0.000000f
#define BPF_Fs88200_Fc78_A2 (-0.001693f)
@@ -1048,7 +534,6 @@
#define BPF_Fs192000_Fc78_A2 (-0.000778f)
#define BPF_Fs192000_Fc78_B1 (-1.998437f)
#define BPF_Fs192000_Fc78_B2 0.998444f
-#endif
/* Coefficients for centre frequency 90Hz */
#define BPF_Fs8000_Fc90_A0 0.022760f
@@ -1097,7 +582,6 @@
#define BPF_Fs48000_Fc90_B1 (-1.992038f)
#define BPF_Fs48000_Fc90_B2 0.992177f
-#ifdef HIGHER_FS
#define BPF_Fs88200_Fc90_A0 0.002083f
#define BPF_Fs88200_Fc90_A1 0.000000f
#define BPF_Fs88200_Fc90_A2 (-0.002083f)
@@ -1121,7 +605,6 @@
#define BPF_Fs192000_Fc90_A2 (-0.000958f)
#define BPF_Fs192000_Fc90_B1 (-1.998075f)
#define BPF_Fs192000_Fc90_B2 0.998085f
-#endif
/************************************************************************************/
/* */
@@ -1140,12 +623,10 @@
#define AGC_ATTACK_Fs44100 0.969158f
#define AGC_ATTACK_Fs48000 0.971628f
-#ifdef HIGHER_FS
#define AGC_ATTACK_Fs88200 0.984458f
#define AGC_ATTACK_Fs96000 0.985712f
#define AGC_ATTACK_Fs176400 0.992199f
#define AGC_ATTACK_Fs192000 0.992830f
-#endif
#define DECAY_SHIFT 10
@@ -1159,12 +640,10 @@
#define AGC_DECAY_Fs44100 0.000008f
#define AGC_DECAY_Fs48000 0.000007f
-#ifdef HIGHER_FS
#define AGC_DECAY_Fs88200 0.0000038f
#define AGC_DECAY_FS96000 0.0000035f
#define AGC_DECAY_Fs176400 0.00000188f
#define AGC_DECAY_FS192000 0.00000175f
-#endif
/* AGC Gain settings */
#define AGC_GAIN_SCALE 31 /* As a power of 2 */
@@ -1224,12 +703,10 @@
#define VOL_TC_Fs32000 0.006231f
#define VOL_TC_Fs44100 0.004525f
#define VOL_TC_Fs48000 0.004158f
-#ifdef HIGHER_FS
#define VOL_TC_Fs88200 0.002263f
#define VOL_TC_Fs96000 0.002079f
#define VOL_TC_Fs176400 0.001131f
#define VOL_TC_Fs192000 0.001039f
-#endif
#define MIX_TC_Fs8000 29365 /* Floating point value 0.896151 */
#define MIX_TC_Fs11025 30230 /* Floating point value 0.922548 */
#define MIX_TC_Fs12000 30422 /* Floating point value 0.928415 */
@@ -1239,14 +716,11 @@
#define MIX_TC_Fs32000 31850 /* Floating point value 0.971973 */
#define MIX_TC_Fs44100 32097 /* Floating point value 0.979515 */
#define MIX_TC_Fs48000 32150 /* Floating point value 0.981150 */
-#ifdef HIGHER_FS
/* Floating point value 0.989704 */
#define MIX_TC_Fs88200 32430
#define MIX_TC_Fs96000 32456 /* Floating point value 0.990530 */
/* Floating point value 0.994838 */
#define MIX_TC_Fs176400 32598
#define MIX_TC_Fs192000 32611 /* Floating point value 0.992524 */
-#endif
-#endif /*BUILD_FLOAT*/
#endif
diff --git a/media/libeffects/lvm/lib/Bass/src/LVDBE_Control.c b/media/libeffects/lvm/lib/Bass/src/LVDBE_Control.cpp
similarity index 82%
rename from media/libeffects/lvm/lib/Bass/src/LVDBE_Control.c
rename to media/libeffects/lvm/lib/Bass/src/LVDBE_Control.cpp
index 0ba2c86..53feae8 100644
--- a/media/libeffects/lvm/lib/Bass/src/LVDBE_Control.c
+++ b/media/libeffects/lvm/lib/Bass/src/LVDBE_Control.cpp
@@ -58,7 +58,6 @@
return(LVDBE_SUCCESS);
}
-
/************************************************************************************/
/* */
/* FUNCTION: LVDBE_GetCapabilities */
@@ -89,7 +88,6 @@
return(LVDBE_SUCCESS);
}
-
/************************************************************************************/
/* */
/* FUNCTION: LVDBE_SetFilters */
@@ -107,72 +105,33 @@
LVDBE_Params_t *pParams)
{
-#if defined(BUILD_FLOAT) && defined(HIGHER_FS)
/*
* Calculate the table offsets
*/
LVM_UINT16 Offset = (LVM_UINT16)((LVM_UINT16)pParams->SampleRate + \
(LVM_UINT16)(pParams->CentreFrequency * (1+LVDBE_FS_192000)));
-#else
- /*
- * Calculate the table offsets
- */
- LVM_UINT16 Offset = (LVM_UINT16)((LVM_UINT16)pParams->SampleRate + \
- (LVM_UINT16)(pParams->CentreFrequency * (1+LVDBE_FS_48000)));
-#endif
/*
* Setup the high pass filter
*/
-#ifndef BUILD_FLOAT
- LoadConst_16(0, /* Clear the history, value 0 */
- (void *)&pInstance->pData->HPFTaps, /* Destination Cast to void: \
- no dereferencing in function*/
- sizeof(pInstance->pData->HPFTaps)/sizeof(LVM_INT16)); /* Number of words */
-#else
LoadConst_Float(0, /* Clear the history, value 0 */
- (void *)&pInstance->pData->HPFTaps, /* Destination Cast to void: \
- no dereferencing in function*/
+ (LVM_FLOAT *)&pInstance->pData->HPFTaps, /* Destination */
sizeof(pInstance->pData->HPFTaps) / sizeof(LVM_FLOAT)); /* Number of words */
-#endif
-#ifndef BUILD_FLOAT
- BQ_2I_D32F32Cll_TRC_WRA_01_Init(&pInstance->pCoef->HPFInstance, /* Initialise the filter */
- &pInstance->pData->HPFTaps,
- (BQ_C32_Coefs_t *)&LVDBE_HPF_Table[Offset]);
-#else
BQ_2I_D32F32Cll_TRC_WRA_01_Init(&pInstance->pCoef->HPFInstance, /* Initialise the filter */
&pInstance->pData->HPFTaps,
(BQ_FLOAT_Coefs_t *)&LVDBE_HPF_Table[Offset]);
-#endif
-
/*
* Setup the band pass filter
*/
-#ifndef BUILD_FLOAT
- LoadConst_16(0, /* Clear the history, value 0 */
- (void *)&pInstance->pData->BPFTaps, /* Destination Cast to void: \
- no dereferencing in function*/
- sizeof(pInstance->pData->BPFTaps)/sizeof(LVM_INT16)); /* Number of words */
-#else
LoadConst_Float(0, /* Clear the history, value 0 */
- (void *)&pInstance->pData->BPFTaps, /* Destination Cast to void: \
- no dereferencing in function*/
+ (LVM_FLOAT *)&pInstance->pData->BPFTaps, /* Destination */
sizeof(pInstance->pData->BPFTaps) / sizeof(LVM_FLOAT)); /* Number of words */
-#endif
-#ifndef BUILD_FLOAT
- BP_1I_D32F32Cll_TRC_WRA_02_Init(&pInstance->pCoef->BPFInstance, /* Initialise the filter */
- &pInstance->pData->BPFTaps,
- (BP_C32_Coefs_t *)&LVDBE_BPF_Table[Offset]);
-#else
BP_1I_D32F32Cll_TRC_WRA_02_Init(&pInstance->pCoef->BPFInstance, /* Initialise the filter */
&pInstance->pData->BPFTaps,
(BP_FLOAT_Coefs_t *)&LVDBE_BPF_Table[Offset]);
-#endif
}
-
-
/************************************************************************************/
/* */
/* FUNCTION: LVDBE_SetAGC */
@@ -196,7 +155,6 @@
pInstance->pData->AGCInstance.AGC_Attack = LVDBE_AGC_ATTACK_Table[(LVM_UINT16)pParams->SampleRate]; /* Attack multiplier */
pInstance->pData->AGCInstance.AGC_Decay = LVDBE_AGC_DECAY_Table[(LVM_UINT16)pParams->SampleRate]; /* Decay multipler */
-
/*
* Get the boost gain
*/
@@ -208,14 +166,10 @@
{
pInstance->pData->AGCInstance.AGC_MaxGain = LVDBE_AGC_GAIN_Table[(LVM_UINT16)pParams->EffectLevel]; /* High pass filter off */
}
-#ifndef BUILD_FLOAT
- pInstance->pData->AGCInstance.AGC_GainShift = AGC_GAIN_SHIFT;
-#endif
pInstance->pData->AGCInstance.AGC_Target = AGC_TARGETLEVEL;
}
-
/************************************************************************************/
/* */
/* FUNCTION: LVDBE_SetVolume */
@@ -247,9 +201,7 @@
LVM_UINT16 dBOffset; /* Table offset */
LVM_INT16 Volume = 0; /* Required volume in dBs */
-#ifdef BUILD_FLOAT
LVM_FLOAT dBShifts_fac;
-#endif
/*
* Apply the volume if enabled
*/
@@ -268,68 +220,41 @@
}
}
-
/*
* Calculate the required gain and shifts
*/
dBOffset = (LVM_UINT16)(6 + Volume % 6); /* Get the dBs 0-5 */
dBShifts = (LVM_UINT16)(Volume / -6); /* Get the 6dB shifts */
-#ifdef BUILD_FLOAT
dBShifts_fac = (LVM_FLOAT)(1 << dBShifts);
-#endif
/*
* When DBE is enabled use AGC volume
*/
-#ifndef BUILD_FLOAT
- pInstance->pData->AGCInstance.Target = ((LVM_INT32)LVDBE_VolumeTable[dBOffset] << 16);
- pInstance->pData->AGCInstance.Target = pInstance->pData->AGCInstance.Target >> dBShifts;
-#else
pInstance->pData->AGCInstance.Target = (LVDBE_VolumeTable[dBOffset]);
pInstance->pData->AGCInstance.Target = pInstance->pData->AGCInstance.Target / dBShifts_fac;
-#endif
pInstance->pData->AGCInstance.VolumeTC = LVDBE_VolumeTCTable[(LVM_UINT16)pParams->SampleRate]; /* Volume update time constant */
-#ifndef BUILD_FLOAT
- pInstance->pData->AGCInstance.VolumeShift = VOLUME_SHIFT+1;
-#endif
/*
* When DBE is disabled use the bypass volume control
*/
if(dBShifts > 0)
{
-#ifndef BUILD_FLOAT
- LVC_Mixer_SetTarget(&pInstance->pData->BypassVolume.MixerStream[0],(((LVM_INT32)LVDBE_VolumeTable[dBOffset]) >> dBShifts));
-#else
LVC_Mixer_SetTarget(&pInstance->pData->BypassVolume.MixerStream[0],
LVDBE_VolumeTable[dBOffset] / dBShifts_fac);
-#endif
}
else
{
-#ifndef BUILD_FLOAT
- LVC_Mixer_SetTarget(&pInstance->pData->BypassVolume.MixerStream[0],(LVM_INT32)LVDBE_VolumeTable[dBOffset]);
-#else
LVC_Mixer_SetTarget(&pInstance->pData->BypassVolume.MixerStream[0],
LVDBE_VolumeTable[dBOffset]);
-#endif
}
pInstance->pData->BypassVolume.MixerStream[0].CallbackSet = 1;
-#ifndef BUILD_FLOAT
LVC_Mixer_VarSlope_SetTimeConstant(&pInstance->pData->BypassVolume.MixerStream[0],
LVDBE_MIXER_TC,
(LVM_Fs_en)pInstance->Params.SampleRate,
2);
-#else
- LVC_Mixer_VarSlope_SetTimeConstant(&pInstance->pData->BypassVolume.MixerStream[0],
- LVDBE_MIXER_TC,
- (LVM_Fs_en)pInstance->Params.SampleRate,
- 2);
-#endif
}
-
/****************************************************************************************/
/* */
/* FUNCTION: LVDBE_Control */
@@ -372,12 +297,7 @@
{
LVDBE_Instance_t *pInstance =(LVDBE_Instance_t *)hInstance;
-#ifndef BUILD_FLOAT
- LVMixer3_2St_st *pBypassMixer_Instance = &pInstance->pData->BypassMixer;
-#else
LVMixer3_2St_FLOAT_st *pBypassMixer_Instance = &pInstance->pData->BypassMixer;
-#endif
-
/*
* Update the filters
@@ -389,7 +309,6 @@
pParams); /* New parameters */
}
-
/*
* Update the AGC is the effect level has changed
*/
@@ -399,24 +318,14 @@
{
LVDBE_SetAGC(pInstance, /* Instance pointer */
pParams); /* New parameters */
-#ifndef BUILD_FLOAT
- LVC_Mixer_SetTimeConstant(&pBypassMixer_Instance->MixerStream[0],
- LVDBE_BYPASS_MIXER_TC,(LVM_Fs_en)pParams->SampleRate,2);
-
- LVC_Mixer_SetTimeConstant(&pBypassMixer_Instance->MixerStream[1],
- LVDBE_BYPASS_MIXER_TC,(LVM_Fs_en)pParams->SampleRate,2);
-#else
LVC_Mixer_SetTimeConstant(&pBypassMixer_Instance->MixerStream[0],
LVDBE_BYPASS_MIXER_TC,(LVM_Fs_en)pParams->SampleRate, 2);
LVC_Mixer_SetTimeConstant(&pBypassMixer_Instance->MixerStream[1],
LVDBE_BYPASS_MIXER_TC,(LVM_Fs_en)pParams->SampleRate, 2);
-#endif
-
}
-
/*
* Update the Volume if the volume demand has changed
*/
@@ -431,23 +340,13 @@
if (pInstance->Params.OperatingMode==LVDBE_ON && pParams->OperatingMode==LVDBE_OFF)
{
-#ifndef BUILD_FLOAT
- LVC_Mixer_SetTarget(&pInstance->pData->BypassMixer.MixerStream[0],0);
- LVC_Mixer_SetTarget(&pInstance->pData->BypassMixer.MixerStream[1],0x00007FFF);
-#else
LVC_Mixer_SetTarget(&pInstance->pData->BypassMixer.MixerStream[0], 0);
LVC_Mixer_SetTarget(&pInstance->pData->BypassMixer.MixerStream[1], 1.0f);
-#endif
}
if (pInstance->Params.OperatingMode==LVDBE_OFF && pParams->OperatingMode==LVDBE_ON)
{
-#ifndef BUILD_FLOAT
- LVC_Mixer_SetTarget(&pInstance->pData->BypassMixer.MixerStream[0],0x00007FFF);
- LVC_Mixer_SetTarget(&pInstance->pData->BypassMixer.MixerStream[1],0);
-#else
LVC_Mixer_SetTarget(&pInstance->pData->BypassMixer.MixerStream[0], 1.0f);
LVC_Mixer_SetTarget(&pInstance->pData->BypassMixer.MixerStream[1], 0);
-#endif
}
/*
@@ -455,6 +354,5 @@
*/
pInstance->Params = *pParams;
-
return(LVDBE_SUCCESS);
}
diff --git a/media/libeffects/lvm/lib/Bass/src/LVDBE_Init.c b/media/libeffects/lvm/lib/Bass/src/LVDBE_Init.cpp
similarity index 90%
rename from media/libeffects/lvm/lib/Bass/src/LVDBE_Init.c
rename to media/libeffects/lvm/lib/Bass/src/LVDBE_Init.cpp
index 2946734..ad77696 100644
--- a/media/libeffects/lvm/lib/Bass/src/LVDBE_Init.c
+++ b/media/libeffects/lvm/lib/Bass/src/LVDBE_Init.cpp
@@ -63,7 +63,6 @@
LVM_UINT32 ScratchSize;
LVDBE_Instance_t *pInstance = (LVDBE_Instance_t *)hInstance;
-
/*
* Fill in the memory table
*/
@@ -80,11 +79,7 @@
/*
* Data memory
*/
-#ifdef BUILD_FLOAT
pMemoryTable->Region[LVDBE_MEMREGION_PERSISTENT_DATA].Size = sizeof(LVDBE_Data_FLOAT_t);
-#else
- pMemoryTable->Region[LVDBE_MEMREGION_PERSISTENT_DATA].Size = sizeof(LVDBE_Data_t);
-#endif
pMemoryTable->Region[LVDBE_MEMREGION_PERSISTENT_DATA].Alignment = LVDBE_PERSISTENT_DATA_ALIGN;
pMemoryTable->Region[LVDBE_MEMREGION_PERSISTENT_DATA].Type = LVDBE_PERSISTENT_DATA;
pMemoryTable->Region[LVDBE_MEMREGION_PERSISTENT_DATA].pBaseAddress = LVM_NULL;
@@ -92,11 +87,7 @@
/*
* Coef memory
*/
-#ifdef BUILD_FLOAT
pMemoryTable->Region[LVDBE_MEMREGION_PERSISTENT_COEF].Size = sizeof(LVDBE_Coef_FLOAT_t);
-#else
- pMemoryTable->Region[LVDBE_MEMREGION_PERSISTENT_COEF].Size = sizeof(LVDBE_Coef_t);
-#endif
pMemoryTable->Region[LVDBE_MEMREGION_PERSISTENT_COEF].Alignment = LVDBE_PERSISTENT_COEF_ALIGN;
pMemoryTable->Region[LVDBE_MEMREGION_PERSISTENT_COEF].Type = LVDBE_PERSISTENT_COEF;
pMemoryTable->Region[LVDBE_MEMREGION_PERSISTENT_COEF].pBaseAddress = LVM_NULL;
@@ -104,12 +95,8 @@
/*
* Scratch memory
*/
-#ifdef BUILD_FLOAT
ScratchSize = (LVM_UINT32)(LVDBE_SCRATCHBUFFERS_INPLACE*sizeof(LVM_FLOAT) * \
pCapabilities->MaxBlockSize);
-#else /*BUILD_FLOAT*/
- ScratchSize = (LVM_UINT32)(LVDBE_SCRATCHBUFFERS_INPLACE*sizeof(LVM_INT16)*pCapabilities->MaxBlockSize);
-#endif
pMemoryTable->Region[LVDBE_MEMREGION_SCRATCH].Size = ScratchSize;
pMemoryTable->Region[LVDBE_MEMREGION_SCRATCH].Alignment = LVDBE_SCRATCH_ALIGN;
pMemoryTable->Region[LVDBE_MEMREGION_SCRATCH].Type = LVDBE_SCRATCH;
@@ -124,7 +111,6 @@
return(LVDBE_SUCCESS);
}
-
/****************************************************************************************/
/* */
/* FUNCTION: LVDBE_Init */
@@ -164,18 +150,11 @@
{
LVDBE_Instance_t *pInstance;
-#ifdef BUILD_FLOAT
LVMixer3_1St_FLOAT_st *pMixer_Instance;
LVMixer3_2St_FLOAT_st *pBypassMixer_Instance;
LVM_FLOAT MixGain;
-#else
- LVMixer3_1St_st *pMixer_Instance;
- LVMixer3_2St_st *pBypassMixer_Instance;
- LVM_INT32 MixGain;
-#endif
LVM_INT16 i;
-
/*
* Set the instance handle if not already initialised
*/
@@ -185,7 +164,6 @@
}
pInstance =(LVDBE_Instance_t *)*phInstance;
-
/*
* Check the memory table for NULL pointers and incorrectly aligned data
*/
@@ -203,19 +181,16 @@
}
}
-
/*
* Save the memory table in the instance structure
*/
pInstance->Capabilities = *pCapabilities;
-
/*
* Save the memory table in the instance structure
*/
pInstance->MemoryTable = *pMemoryTable;
-
/*
* Set the default instance parameters
*/
@@ -228,13 +203,13 @@
pInstance->Params.VolumeControl = LVDBE_VOLUME_OFF;
pInstance->Params.VolumedB = 0;
-
/*
* Set pointer to data and coef memory
*/
- pInstance->pData = pMemoryTable->Region[LVDBE_MEMREGION_PERSISTENT_DATA].pBaseAddress;
- pInstance->pCoef = pMemoryTable->Region[LVDBE_MEMREGION_PERSISTENT_COEF].pBaseAddress;
-
+ pInstance->pData =
+ (LVDBE_Data_FLOAT_t *)pMemoryTable->Region[LVDBE_MEMREGION_PERSISTENT_DATA].pBaseAddress;
+ pInstance->pCoef =
+ (LVDBE_Coef_FLOAT_t *)pMemoryTable->Region[LVDBE_MEMREGION_PERSISTENT_COEF].pBaseAddress;
/*
* Initialise the filters
@@ -242,7 +217,6 @@
LVDBE_SetFilters(pInstance, /* Set the filter taps and coefficients */
&pInstance->Params);
-
/*
* Initialise the AGC
*/
@@ -254,11 +228,7 @@
// initialize the mixer with some fixes values since otherwise LVDBE_SetVolume ends up
// reading uninitialized data
pMixer_Instance = &pInstance->pData->BypassVolume;
-#ifndef BUILD_FLOAT
- LVC_Mixer_Init(&pMixer_Instance->MixerStream[0],0x00007FFF,0x00007FFF);
-#else
LVC_Mixer_Init(&pMixer_Instance->MixerStream[0], 1.0, 1.0);
-#endif
/*
* Initialise the volume
@@ -268,13 +238,8 @@
pInstance->pData->AGCInstance.Volume = pInstance->pData->AGCInstance.Target;
/* Initialise as the target */
-#ifndef BUILD_FLOAT
- MixGain = LVC_Mixer_GetTarget(&pMixer_Instance->MixerStream[0]);
- LVC_Mixer_Init(&pMixer_Instance->MixerStream[0],MixGain,MixGain);
-#else
MixGain = LVC_Mixer_GetTarget(&pMixer_Instance->MixerStream[0]);
LVC_Mixer_Init(&pMixer_Instance->MixerStream[0], MixGain, MixGain);
-#endif
/* Configure the mixer process path */
pMixer_Instance->MixerStream[0].CallbackParam = 0;
@@ -307,15 +272,9 @@
pBypassMixer_Instance->MixerStream[1].pCallbackHandle = LVM_NULL;
pBypassMixer_Instance->MixerStream[1].pCallBack = LVM_NULL;
pBypassMixer_Instance->MixerStream[1].CallbackSet=0;
-#ifndef BUILD_FLOAT
- LVC_Mixer_Init(&pBypassMixer_Instance->MixerStream[1],0x00007FFF,0x00007FFF);
- LVC_Mixer_SetTimeConstant(&pBypassMixer_Instance->MixerStream[1],
- LVDBE_BYPASS_MIXER_TC,(LVM_Fs_en)pInstance->Params.SampleRate,2);
-#else
LVC_Mixer_Init(&pBypassMixer_Instance->MixerStream[1], 1.0, 1.0);
LVC_Mixer_SetTimeConstant(&pBypassMixer_Instance->MixerStream[1],
LVDBE_BYPASS_MIXER_TC,(LVM_Fs_en)pInstance->Params.SampleRate, 2);
-#endif
return(LVDBE_SUCCESS);
}
diff --git a/media/libeffects/lvm/lib/Bass/src/LVDBE_Private.h b/media/libeffects/lvm/lib/Bass/src/LVDBE_Private.h
index 4225a30..f3faaed 100644
--- a/media/libeffects/lvm/lib/Bass/src/LVDBE_Private.h
+++ b/media/libeffects/lvm/lib/Bass/src/LVDBE_Private.h
@@ -27,11 +27,6 @@
#ifndef __LVDBE_PRIVATE_H__
#define __LVDBE_PRIVATE_H__
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
-
/****************************************************************************************/
/* */
/* Includes */
@@ -43,7 +38,6 @@
#include "LVC_Mixer.h"
#include "AGC.h"
-
/****************************************************************************************/
/* */
/* Defines */
@@ -74,7 +68,6 @@
#define LVDBE_MIXER_TC 5 /* Mixer time */
#define LVDBE_BYPASS_MIXER_TC 100 /* Bypass mixer time */
-
/****************************************************************************************/
/* */
/* Structures */
@@ -82,29 +75,6 @@
/****************************************************************************************/
/* Data structure */
-#ifndef BUILD_FLOAT
-typedef struct
-{
- /* AGC parameters */
- AGC_MIX_VOL_2St1Mon_D32_t AGCInstance; /* AGC instance parameters */
-
- /* Process variables */
- Biquad_2I_Order2_Taps_t HPFTaps; /* High pass filter taps */
- Biquad_1I_Order2_Taps_t BPFTaps; /* Band pass filter taps */
- LVMixer3_1St_st BypassVolume; /* Bypass volume scaler */
- LVMixer3_2St_st BypassMixer; /* Bypass Mixer for Click Removal */
-
-} LVDBE_Data_t;
-
-/* Coefs structure */
-typedef struct
-{
- /* Process variables */
- Biquad_Instance_t HPFInstance; /* High pass filter instance */
- Biquad_Instance_t BPFInstance; /* Band pass filter instance */
-
-} LVDBE_Coef_t;
-#else
/* Data structure */
typedef struct
{
@@ -126,7 +96,6 @@
Biquad_FLOAT_Instance_t HPFInstance; /* High pass filter instance */
Biquad_FLOAT_Instance_t BPFInstance; /* Band pass filter instance */
} LVDBE_Coef_FLOAT_t;
-#endif
/* Instance structure */
typedef struct
{
@@ -136,16 +105,10 @@
LVDBE_Capabilities_t Capabilities; /* Instance capabilities */
/* Data and coefficient pointers */
-#ifndef BUILD_FLOAT
- LVDBE_Data_t *pData; /* Instance data */
- LVDBE_Coef_t *pCoef; /* Instance coefficients */
-#else
LVDBE_Data_FLOAT_t *pData; /* Instance data */
LVDBE_Coef_FLOAT_t *pCoef; /* Instance coefficients */
-#endif
} LVDBE_Instance_t;
-
/****************************************************************************************/
/* */
/* Function prototypes */
@@ -155,17 +118,10 @@
void LVDBE_SetAGC(LVDBE_Instance_t *pInstance,
LVDBE_Params_t *pParams);
-
void LVDBE_SetVolume(LVDBE_Instance_t *pInstance,
LVDBE_Params_t *pParams);
-
void LVDBE_SetFilters(LVDBE_Instance_t *pInstance,
LVDBE_Params_t *pParams);
-
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
-
#endif /* __LVDBE_PRIVATE_H__ */
diff --git a/media/libeffects/lvm/lib/Bass/src/LVDBE_Process.c b/media/libeffects/lvm/lib/Bass/src/LVDBE_Process.cpp
similarity index 72%
rename from media/libeffects/lvm/lib/Bass/src/LVDBE_Process.c
rename to media/libeffects/lvm/lib/Bass/src/LVDBE_Process.cpp
index c4d3403..b4a71c7 100644
--- a/media/libeffects/lvm/lib/Bass/src/LVDBE_Process.c
+++ b/media/libeffects/lvm/lib/Bass/src/LVDBE_Process.cpp
@@ -73,119 +73,6 @@
/* overall end to end gain is odB. */
/* */
/********************************************************************************************/
-#ifndef BUILD_FLOAT
-LVDBE_ReturnStatus_en LVDBE_Process(LVDBE_Handle_t hInstance,
- const LVM_INT16 *pInData, LVM_INT16 *pOutData, LVM_UINT16 NumSamples) {
-
- LVDBE_Instance_t *pInstance = (LVDBE_Instance_t *) hInstance;
- LVM_INT32 *pScratch =
- (LVM_INT32 *) pInstance->MemoryTable.Region[LVDBE_MEMREGION_SCRATCH].pBaseAddress;
- LVM_INT32 *pMono;
- LVM_INT16 *pInput = (LVM_INT16 *) pInData;
-
- /* Scratch for Volume Control starts at offset of 2*NumSamples short values from pScratch */
- LVM_INT16 *pScratchVol = (LVM_INT16 *) (&pScratch[NumSamples]);
-
- /* Scratch for Mono path starts at offset of 2*NumSamples 32-bit values from pScratch */
- pMono = &pScratch[2 * NumSamples];
-
- /*
- * Check the number of samples is not too large
- */
- if (NumSamples > pInstance->Capabilities.MaxBlockSize) {
- return (LVDBE_TOOMANYSAMPLES);
- }
-
- /*
- * Check if the algorithm is enabled
- */
- /* DBE path is processed when DBE is ON or during On/Off transitions */
- if ((pInstance->Params.OperatingMode == LVDBE_ON)
- || (LVC_Mixer_GetCurrent(
- &pInstance->pData->BypassMixer.MixerStream[0])
- != LVC_Mixer_GetTarget(
- &pInstance->pData->BypassMixer.MixerStream[0]))) {
-
- /*
- * Convert 16-bit samples to 32-bit and scale
- * (For a 16-bit implementation apply headroom loss here)
- */
- Int16LShiftToInt32_16x32(pInput, /* Source 16-bit data */
- pScratch, /* Dest. 32-bit data */
- (LVM_INT16) (2 * NumSamples), /* Left and right */
- LVDBE_SCALESHIFT); /* Shift scale */
-
- /*
- * Apply the high pass filter if selected
- */
- if (pInstance->Params.HPFSelect == LVDBE_HPF_ON) {
- BQ_2I_D32F32C30_TRC_WRA_01(&pInstance->pCoef->HPFInstance,/* Filter instance */
- (LVM_INT32 *) pScratch, /* Source */
- (LVM_INT32 *) pScratch, /* Destination */
- (LVM_INT16) NumSamples); /* Number of samples */
- }
-
- /*
- * Create the mono stream
- */
- From2iToMono_32(pScratch, /* Stereo source */
- pMono, /* Mono destination */
- (LVM_INT16) NumSamples); /* Number of samples */
-
- /*
- * Apply the band pass filter
- */
- BP_1I_D32F32C30_TRC_WRA_02(&pInstance->pCoef->BPFInstance, /* Filter instance */
- (LVM_INT32 *) pMono, /* Source */
- (LVM_INT32 *) pMono, /* Destination */
- (LVM_INT16) NumSamples); /* Number of samples */
-
- /*
- * Apply the AGC and mix
- */
- AGC_MIX_VOL_2St1Mon_D32_WRA(&pInstance->pData->AGCInstance, /* Instance pointer */
- pScratch, /* Stereo source */
- pMono, /* Mono band pass source */
- pScratch, /* Stereo destination */
- NumSamples); /* Number of samples */
-
- /*
- * Convert 32-bit samples to 16-bit and saturate
- * (Not required for 16-bit implemenations)
- */
- Int32RShiftToInt16_Sat_32x16(pScratch, /* Source 32-bit data */
- (LVM_INT16 *) pScratch, /* Dest. 16-bit data */
- (LVM_INT16) (2 * NumSamples), /* Left and right */
- LVDBE_SCALESHIFT); /* Shift scale */
-
- }
-
- /* Bypass Volume path is processed when DBE is OFF or during On/Off transitions */
- if ((pInstance->Params.OperatingMode == LVDBE_OFF)
- || (LVC_Mixer_GetCurrent(
- &pInstance->pData->BypassMixer.MixerStream[1])
- != LVC_Mixer_GetTarget(
- &pInstance->pData->BypassMixer.MixerStream[1]))) {
-
- /*
- * The algorithm is disabled but volume management is required to compensate for
- * headroom and volume (if enabled)
- */
- LVC_MixSoft_1St_D16C31_SAT(&pInstance->pData->BypassVolume, pInData,
- pScratchVol, (LVM_INT16) (2 * NumSamples)); /* Left and right */
-
- }
-
- /*
- * Mix DBE processed path and bypass volume path
- */
- LVC_MixSoft_2St_D16C31_SAT(&pInstance->pData->BypassMixer,
- (LVM_INT16 *) pScratch, pScratchVol, pOutData,
- (LVM_INT16) (2 * NumSamples));
-
- return (LVDBE_SUCCESS);
-}
-#else /*BUILD_FLOAT*/
LVDBE_ReturnStatus_en LVDBE_Process(LVDBE_Handle_t hInstance,
const LVM_FLOAT *pInData,
LVM_FLOAT *pOutData,
@@ -362,4 +249,3 @@
#endif
return LVDBE_SUCCESS;
}
-#endif
diff --git a/media/libeffects/lvm/lib/Bass/src/LVDBE_Tables.c b/media/libeffects/lvm/lib/Bass/src/LVDBE_Tables.cpp
similarity index 93%
rename from media/libeffects/lvm/lib/Bass/src/LVDBE_Tables.c
rename to media/libeffects/lvm/lib/Bass/src/LVDBE_Tables.cpp
index a2ce404..728575c 100644
--- a/media/libeffects/lvm/lib/Bass/src/LVDBE_Tables.c
+++ b/media/libeffects/lvm/lib/Bass/src/LVDBE_Tables.cpp
@@ -15,7 +15,6 @@
* limitations under the License.
*/
-
/************************************************************************************/
/* */
/* Includes */
@@ -24,9 +23,9 @@
#include "LVDBE.h"
#include "LVDBE_Coeffs.h" /* Filter coefficients */
+#include "LVDBE_Tables.h"
#include "BIQUAD.h"
-
/************************************************************************************/
/* */
/* Coefficients constant table */
@@ -36,11 +35,7 @@
/*
* High Pass Filter Coefficient table
*/
-#ifndef BUILD_FLOAT
-const BQ_C32_Coefs_t LVDBE_HPF_Table[] = {
-#else /*BUILD_FLOAT*/
const BQ_FLOAT_Coefs_t LVDBE_HPF_Table[] = {
-#endif /*BUILD_FLOAT*/
/* Coefficients for 55Hz centre frequency */
{HPF_Fs8000_Fc55_A2, /* 8kS/s coefficients */
HPF_Fs8000_Fc55_A1,
@@ -87,7 +82,6 @@
HPF_Fs48000_Fc55_A0,
-HPF_Fs48000_Fc55_B2,
-HPF_Fs48000_Fc55_B1},
-#ifdef HIGHER_FS
{HPF_Fs88200_Fc55_A2, /* 88kS/s coefficients */
HPF_Fs88200_Fc55_A1,
HPF_Fs88200_Fc55_A0,
@@ -108,7 +102,6 @@
HPF_Fs192000_Fc55_A0,
-HPF_Fs192000_Fc55_B2,
-HPF_Fs192000_Fc55_B1},
-#endif
/* Coefficients for 66Hz centre frequency */
{HPF_Fs8000_Fc66_A2, /* 8kS/s coefficients */
@@ -156,7 +149,6 @@
HPF_Fs48000_Fc66_A0,
-HPF_Fs48000_Fc66_B2,
-HPF_Fs48000_Fc66_B1},
-#ifdef HIGHER_FS
{HPF_Fs88200_Fc66_A2, /* 88kS/s coefficients */
HPF_Fs88200_Fc66_A1,
HPF_Fs88200_Fc66_A0,
@@ -177,8 +169,6 @@
HPF_Fs192000_Fc66_A0,
-HPF_Fs192000_Fc66_B2,
-HPF_Fs192000_Fc66_B1},
-#endif
-
/* Coefficients for 78Hz centre frequency */
{HPF_Fs8000_Fc78_A2, /* 8kS/s coefficients */
@@ -226,7 +216,6 @@
HPF_Fs48000_Fc78_A0,
-HPF_Fs48000_Fc78_B2,
-HPF_Fs48000_Fc78_B1},
-#ifdef HIGHER_FS
{HPF_Fs88200_Fc78_A2, /* 88kS/s coefficients */
HPF_Fs88200_Fc78_A1,
HPF_Fs88200_Fc78_A0,
@@ -247,8 +236,6 @@
HPF_Fs192000_Fc78_A0,
-HPF_Fs192000_Fc78_B2,
-HPF_Fs192000_Fc78_B1},
-#endif
-
/* Coefficients for 90Hz centre frequency */
{HPF_Fs8000_Fc90_A2, /* 8kS/s coefficients */
@@ -297,7 +284,6 @@
-HPF_Fs48000_Fc90_B2,
-HPF_Fs48000_Fc90_B1}
-#ifdef HIGHER_FS
,
{HPF_Fs88200_Fc90_A2, /* 88kS/s coefficients */
HPF_Fs88200_Fc90_A1,
@@ -319,18 +305,13 @@
HPF_Fs192000_Fc90_A0,
-HPF_Fs192000_Fc90_B2,
-HPF_Fs192000_Fc90_B1}
-#endif
};
/*
* Band Pass Filter coefficient table
*/
-#ifndef BUILD_FLOAT
-const BP_C32_Coefs_t LVDBE_BPF_Table[] = {
-#else /*BUILD_FLOAT*/
const BP_FLOAT_Coefs_t LVDBE_BPF_Table[] = {
-#endif /*BUILD_FLOAT*/
/* Coefficients for 55Hz centre frequency */
{BPF_Fs8000_Fc55_A0, /* 8kS/s coefficients */
-BPF_Fs8000_Fc55_B2,
@@ -359,7 +340,6 @@
{BPF_Fs48000_Fc55_A0, /* 48kS/s coefficients */
-BPF_Fs48000_Fc55_B2,
-BPF_Fs48000_Fc55_B1},
-#ifdef HIGHER_FS
{BPF_Fs88200_Fc55_A0, /* 88kS/s coefficients */
-BPF_Fs88200_Fc55_B2,
-BPF_Fs88200_Fc55_B1},
@@ -372,7 +352,6 @@
{BPF_Fs192000_Fc55_A0, /* 192kS/s coefficients */
-BPF_Fs192000_Fc55_B2,
-BPF_Fs192000_Fc55_B1},
-#endif
/* Coefficients for 66Hz centre frequency */
{BPF_Fs8000_Fc66_A0, /* 8kS/s coefficients */
@@ -402,7 +381,6 @@
{BPF_Fs48000_Fc66_A0, /* 48kS/s coefficients */
-BPF_Fs48000_Fc66_B2,
-BPF_Fs48000_Fc66_B1},
-#ifdef HIGHER_FS
{BPF_Fs88200_Fc66_A0, /* 88kS/s coefficients */
-BPF_Fs88200_Fc66_B2,
-BPF_Fs88200_Fc66_B1},
@@ -415,7 +393,6 @@
{BPF_Fs192000_Fc66_A0, /* 192kS/s coefficients */
-BPF_Fs192000_Fc66_B2,
-BPF_Fs192000_Fc66_B1},
-#endif
/* Coefficients for 78Hz centre frequency */
{BPF_Fs8000_Fc78_A0, /* 8kS/s coefficients */
@@ -445,7 +422,6 @@
{BPF_Fs48000_Fc78_A0, /* 48kS/s coefficients */
-BPF_Fs48000_Fc78_B2,
-BPF_Fs48000_Fc78_B1},
-#ifdef HIGHER_FS
{BPF_Fs88200_Fc66_A0, /* 88kS/s coefficients */
-BPF_Fs88200_Fc66_B2,
-BPF_Fs88200_Fc66_B1},
@@ -458,7 +434,6 @@
{BPF_Fs192000_Fc78_A0, /* 192kS/s coefficients */
-BPF_Fs192000_Fc78_B2,
-BPF_Fs192000_Fc78_B1},
-#endif
/* Coefficients for 90Hz centre frequency */
{BPF_Fs8000_Fc90_A0, /* 8kS/s coefficients */
@@ -488,7 +463,6 @@
{BPF_Fs48000_Fc90_A0, /* 48kS/s coefficients */
-BPF_Fs48000_Fc90_B2,
-BPF_Fs48000_Fc90_B1}
-#ifdef HIGHER_FS
,
{BPF_Fs88200_Fc90_A0, /* 88kS/s coefficients */
-BPF_Fs88200_Fc90_B2,
@@ -502,12 +476,9 @@
{BPF_Fs192000_Fc90_A0, /* 192kS/s coefficients */
-BPF_Fs192000_Fc90_B2,
-BPF_Fs192000_Fc90_B1}
-#endif
-
};
-
/************************************************************************************/
/* */
/* AGC constant tables */
@@ -515,11 +486,7 @@
/************************************************************************************/
/* Attack time (signal too large) */
-#ifndef BUILD_FLOAT
-const LVM_INT16 LVDBE_AGC_ATTACK_Table[] = {
-#else /*BUILD_FLOAT*/
const LVM_FLOAT LVDBE_AGC_ATTACK_Table[] = {
-#endif /*BUILD_FLOAT*/
AGC_ATTACK_Fs8000,
AGC_ATTACK_Fs11025,
AGC_ATTACK_Fs12000,
@@ -529,21 +496,15 @@
AGC_ATTACK_Fs32000,
AGC_ATTACK_Fs44100,
AGC_ATTACK_Fs48000
-#ifdef HIGHER_FS
,AGC_ATTACK_Fs88200
,AGC_ATTACK_Fs96000
,AGC_ATTACK_Fs176400
,AGC_ATTACK_Fs192000
-#endif
};
/* Decay time (signal too small) */
-#ifndef BUILD_FLOAT
-const LVM_INT16 LVDBE_AGC_DECAY_Table[] = {
-#else /*BUILD_FLOAT*/
const LVM_FLOAT LVDBE_AGC_DECAY_Table[] = {
-#endif /*BUILD_FLOAT*/
AGC_DECAY_Fs8000,
AGC_DECAY_Fs11025,
AGC_DECAY_Fs12000,
@@ -553,21 +514,15 @@
AGC_DECAY_Fs32000,
AGC_DECAY_Fs44100,
AGC_DECAY_Fs48000
-#ifdef HIGHER_FS
,AGC_DECAY_Fs88200
,AGC_DECAY_FS96000
,AGC_DECAY_Fs176400
,AGC_DECAY_FS192000
-#endif
};
/* Gain for use without the high pass filter */
-#ifndef BUILD_FLOAT
-const LVM_INT32 LVDBE_AGC_GAIN_Table[] = {
-#else /*BUILD_FLOAT*/
const LVM_FLOAT LVDBE_AGC_GAIN_Table[] = {
-#endif /*BUILD_FLOAT*/
AGC_GAIN_0dB,
AGC_GAIN_1dB,
AGC_GAIN_2dB,
@@ -586,11 +541,7 @@
AGC_GAIN_15dB};
/* Gain for use with the high pass filter */
-#ifndef BUILD_FLOAT
-const LVM_INT32 LVDBE_AGC_HPFGAIN_Table[] = {
-#else /*BUILD_FLOAT*/
const LVM_FLOAT LVDBE_AGC_HPFGAIN_Table[] = {
-#endif /*BUILD_FLOAT*/
AGC_HPFGAIN_0dB,
AGC_HPFGAIN_1dB,
AGC_HPFGAIN_2dB,
@@ -608,7 +559,6 @@
AGC_HPFGAIN_14dB,
AGC_HPFGAIN_15dB};
-
/************************************************************************************/
/* */
/* Volume control gain and time constant tables */
@@ -616,16 +566,6 @@
/************************************************************************************/
/* dB to linear conversion table */
-#ifndef BUILD_FLOAT
-const LVM_INT16 LVDBE_VolumeTable[] = {
- 0x4000, /* -6dB */
- 0x47FB, /* -5dB */
- 0x50C3, /* -4dB */
- 0x5A9E, /* -3dB */
- 0x65AD, /* -2dB */
- 0x7215, /* -1dB */
- 0x7FFF}; /* 0dB */
-#else /*BUILD_FLOAT*/
const LVM_FLOAT LVDBE_VolumeTable[] = {
0.500000f, /* -6dB */
0.562341f, /* -5dB */
@@ -634,13 +574,8 @@
0.794328f, /* -2dB */
0.891251f, /* -1dB */
1.000000f}; /* 0dB */
-#endif /*BUILD_FLOAT*/
-#ifndef BUILD_FLOAT
-const LVM_INT16 LVDBE_VolumeTCTable[] = {
-#else /*BUILD_FLOAT*/
const LVM_FLOAT LVDBE_VolumeTCTable[] = {
-#endif /*BUILD_FLOAT*/
VOL_TC_Fs8000,
VOL_TC_Fs11025,
VOL_TC_Fs12000,
@@ -650,16 +585,12 @@
VOL_TC_Fs32000,
VOL_TC_Fs44100,
VOL_TC_Fs48000
-#ifdef HIGHER_FS
,VOL_TC_Fs88200
,VOL_TC_Fs96000
,VOL_TC_Fs176400
,VOL_TC_Fs192000
-#endif
};
-
-
const LVM_INT16 LVDBE_MixerTCTable[] = {
MIX_TC_Fs8000,
@@ -671,11 +602,9 @@
MIX_TC_Fs32000,
MIX_TC_Fs44100,
MIX_TC_Fs48000
-#ifdef HIGHER_FS
,MIX_TC_Fs88200
,MIX_TC_Fs96000
,MIX_TC_Fs176400
,MIX_TC_Fs192000
-#endif
};
diff --git a/media/libeffects/lvm/lib/Bass/src/LVDBE_Tables.h b/media/libeffects/lvm/lib/Bass/src/LVDBE_Tables.h
index ca46e37..6eabdd2 100644
--- a/media/libeffects/lvm/lib/Bass/src/LVDBE_Tables.h
+++ b/media/libeffects/lvm/lib/Bass/src/LVDBE_Tables.h
@@ -15,7 +15,6 @@
* limitations under the License.
*/
-
/************************************************************************************/
/* */
/* Includes */
@@ -24,61 +23,9 @@
#ifndef __LVBDE_TABLES_H__
#define __LVBDE_TABLES_H__
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
#include "BIQUAD.h"
#include "LVM_Types.h"
-#ifndef BUILD_FLOAT
-/************************************************************************************/
-/* */
-/* Coefficients constant table */
-/* */
-/************************************************************************************/
-
-/*
- * High Pass Filter Coefficient table
- */
-extern const BQ_C32_Coefs_t LVDBE_HPF_Table[];
-
-/*
- * Band Pass Filter coefficient table
- */
-extern const BP_C32_Coefs_t LVDBE_BPF_Table[];
-
-/************************************************************************************/
-/* */
-/* AGC constant tables */
-/* */
-/************************************************************************************/
-
-/* Attack time (signal too large) */
-extern const LVM_INT16 LVDBE_AGC_ATTACK_Table[];
-
-/* Decay time (signal too small) */
-extern const LVM_INT16 LVDBE_AGC_DECAY_Table[];
-
-/* Gain for use without the high pass filter */
-extern const LVM_INT32 LVDBE_AGC_GAIN_Table[];
-
-/* Gain for use with the high pass filter */
-extern const LVM_INT32 LVDBE_AGC_HPFGAIN_Table[];
-
-/************************************************************************************/
-/* */
-/* Volume control gain and time constant tables */
-/* */
-/************************************************************************************/
-
-/* dB to linear conversion table */
-extern const LVM_INT16 LVDBE_VolumeTable[];
-
-extern const LVM_INT16 LVDBE_VolumeTCTable[];
-
-#else /*BUILD_FLOAT*/
-
/************************************************************************************/
/* */
/* Coefficients constant table */
@@ -123,13 +70,6 @@
extern const LVM_FLOAT LVDBE_VolumeTable[];
extern const LVM_FLOAT LVDBE_VolumeTCTable[];
-#endif /*BUILD_FLOAT*/
-
extern const LVM_INT16 LVDBE_MixerTCTable[];
-
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
-
#endif /* __LVBDE_TABLES_H__ */
diff --git a/media/libeffects/lvm/lib/Bundle/lib/LVM.h b/media/libeffects/lvm/lib/Bundle/lib/LVM.h
index 5082a53..e4e8450 100644
--- a/media/libeffects/lvm/lib/Bundle/lib/LVM.h
+++ b/media/libeffects/lvm/lib/Bundle/lib/LVM.h
@@ -53,11 +53,6 @@
#ifndef __LVM_H__
#define __LVM_H__
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
-
/****************************************************************************************/
/* */
/* Includes */
@@ -66,7 +61,6 @@
#include "LVM_Types.h"
-
/****************************************************************************************/
/* */
/* Definitions */
@@ -108,7 +102,6 @@
/* Instance handle */
typedef void *LVM_Handle_t;
-
/* Status return values */
typedef enum
{
@@ -123,7 +116,6 @@
LVM_RETURNSTATUS_DUMMY = LVM_MAXENUM
} LVM_ReturnStatus_en;
-
/* Buffer Management mode */
typedef enum
{
@@ -227,7 +219,6 @@
LVM_CHAR *pPlatform; /* Pointer to the library platform type */
} LVM_VersionInfo_st;
-
/****************************************************************************************/
/* */
/* Structures */
@@ -248,7 +239,6 @@
LVM_UINT16 QFactor; /* Band quality factor (x100) */
} LVM_EQNB_BandDef_t;
-
/* Headroom band definition */
typedef struct
{
@@ -257,7 +247,6 @@
LVM_INT16 Headroom_Offset; /* Headroom = biggest band gain - Headroom_Offset */
} LVM_HeadroomBandDef_t;
-
/* Control Parameter structure */
typedef struct
{
@@ -303,7 +292,6 @@
} LVM_ControlParams_t;
-
/* Instance Parameter structure */
typedef struct
{
@@ -333,7 +321,6 @@
/* */
/****************************************************************************************/
-
/****************************************************************************************/
/* */
/* FUNCTION: LVM_GetVersionInfo */
@@ -354,7 +341,6 @@
/****************************************************************************************/
LVM_ReturnStatus_en LVM_GetVersionInfo(LVM_VersionInfo_st *pVersion);
-
/****************************************************************************************/
/* */
/* FUNCTION: LVM_GetMemoryTable */
@@ -391,7 +377,6 @@
LVM_MemTab_t *pMemoryTable,
LVM_InstParams_t *pInstParams);
-
/****************************************************************************************/
/* */
/* FUNCTION: LVM_GetInstanceHandle */
@@ -418,7 +403,6 @@
LVM_MemTab_t *pMemoryTable,
LVM_InstParams_t *pInstParams);
-
/****************************************************************************************/
/* */
/* FUNCTION: LVM_ClearAudioBuffers */
@@ -439,7 +423,6 @@
/****************************************************************************************/
LVM_ReturnStatus_en LVM_ClearAudioBuffers(LVM_Handle_t hInstance);
-
/****************************************************************************************/
/* */
/* FUNCTION: LVM_GetControlParameters */
@@ -463,7 +446,6 @@
LVM_ReturnStatus_en LVM_GetControlParameters(LVM_Handle_t hInstance,
LVM_ControlParams_t *pParams);
-
/****************************************************************************************/
/* */
/* FUNCTION: LVM_SetControlParameters */
@@ -487,7 +469,6 @@
LVM_ReturnStatus_en LVM_SetControlParameters(LVM_Handle_t hInstance,
LVM_ControlParams_t *pParams);
-
/****************************************************************************************/
/* */
/* FUNCTION: LVM_Process */
@@ -518,20 +499,11 @@
/* STEREO the number of sample pairs in the block */
/* */
/****************************************************************************************/
-#ifdef BUILD_FLOAT
LVM_ReturnStatus_en LVM_Process(LVM_Handle_t hInstance,
const LVM_FLOAT *pInData,
LVM_FLOAT *pOutData,
LVM_UINT16 NumSamples,
LVM_UINT32 AudioTime);
-#else
-LVM_ReturnStatus_en LVM_Process(LVM_Handle_t hInstance,
- const LVM_INT16 *pInData,
- LVM_INT16 *pOutData,
- LVM_UINT16 NumSamples,
- LVM_UINT32 AudioTime);
-#endif
-
/****************************************************************************************/
/* */
@@ -555,7 +527,6 @@
LVM_ReturnStatus_en LVM_SetHeadroomParams( LVM_Handle_t hInstance,
LVM_HeadroomParams_t *pHeadroomParams);
-
/****************************************************************************************/
/* */
/* FUNCTION: LVM_GetHeadroomParams */
@@ -578,7 +549,6 @@
LVM_ReturnStatus_en LVM_GetHeadroomParams( LVM_Handle_t hInstance,
LVM_HeadroomParams_t *pHeadroomParams);
-
/****************************************************************************************/
/* */
/* FUNCTION: LVM_GetSpectrum */
@@ -632,10 +602,5 @@
LVM_ReturnStatus_en LVM_SetVolumeNoSmoothing( LVM_Handle_t hInstance,
LVM_ControlParams_t *pParams);
-
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
-
#endif /* __LVM_H__ */
diff --git a/media/libeffects/lvm/lib/Bundle/src/LVM_API_Specials.c b/media/libeffects/lvm/lib/Bundle/src/LVM_API_Specials.cpp
similarity index 98%
rename from media/libeffects/lvm/lib/Bundle/src/LVM_API_Specials.c
rename to media/libeffects/lvm/lib/Bundle/src/LVM_API_Specials.cpp
index 07b7f0e..e241cdd 100644
--- a/media/libeffects/lvm/lib/Bundle/src/LVM_API_Specials.c
+++ b/media/libeffects/lvm/lib/Bundle/src/LVM_API_Specials.cpp
@@ -15,7 +15,6 @@
* limitations under the License.
*/
-
/****************************************************************************************/
/* */
/* Includes */
@@ -60,7 +59,6 @@
pLVPSA_Handle_t *hPSAInstance;
LVPSA_RETURN LVPSA_Status;
-
if(pInstance == LVM_NULL)
{
return LVM_NULLADDRESS;
@@ -72,7 +70,7 @@
return LVM_SUCCESS;
}
- hPSAInstance = pInstance->hPSAInstance;
+ hPSAInstance = (pLVPSA_Handle_t *)pInstance->hPSAInstance;
if((pCurrentPeaks == LVM_NULL) ||
(pPastPeaks == LVM_NULL))
@@ -80,7 +78,6 @@
return LVM_NULLADDRESS;
}
-
/*
* Update new parameters if necessary
*/
@@ -115,7 +112,6 @@
return(LVM_SUCCESS);
}
-
/****************************************************************************************/
/* */
/* FUNCTION: LVM_SetVolumeNoSmoothing */
diff --git a/media/libeffects/lvm/lib/Bundle/src/LVM_Buffers.c b/media/libeffects/lvm/lib/Bundle/src/LVM_Buffers.cpp
similarity index 61%
rename from media/libeffects/lvm/lib/Bundle/src/LVM_Buffers.c
rename to media/libeffects/lvm/lib/Bundle/src/LVM_Buffers.cpp
index bdca5e3..3aeddbb 100644
--- a/media/libeffects/lvm/lib/Bundle/src/LVM_Buffers.c
+++ b/media/libeffects/lvm/lib/Bundle/src/LVM_Buffers.cpp
@@ -15,7 +15,6 @@
* limitations under the License.
*/
-
/****************************************************************************************/
/* */
/* Includes */
@@ -50,7 +49,6 @@
/* NOTES: */
/* */
/****************************************************************************************/
-#ifdef BUILD_FLOAT
void LVM_BufferManagedIn(LVM_Handle_t hInstance,
const LVM_FLOAT *pInData,
LVM_FLOAT **pToProcess,
@@ -94,7 +92,6 @@
pBuffer->SamplesToOutput = 0; /* Samples to output is same as
number read for inplace processing */
-
/*
* Calculate the number of samples to process this call and update the buffer state
*/
@@ -131,7 +128,6 @@
}
*pNumSamples = (LVM_UINT16)SampleCount; /* Set the number of samples to process this call */
-
/*
* Copy samples from the delay buffer as required
*/
@@ -147,7 +143,6 @@
pDest += NumChannels * pBuffer->InDelaySamples; /* Update the destination pointer */
}
-
/*
* Copy the rest of the samples for this call from the input buffer
*/
@@ -165,7 +160,6 @@
pBuffer->SamplesToOutput = (LVM_INT16)(pBuffer->SamplesToOutput + NumSamples);
}
-
/*
* Update the sample count and input pointer
*/
@@ -173,7 +167,6 @@
pInstance->SamplesToProcess = (LVM_INT16)(pInstance->SamplesToProcess - SampleCount);
pInstance->pInputSamples = pStart; /* Update input sample pointer */
-
/*
* Save samples to the delay buffer if any left unprocessed
*/
@@ -190,7 +183,6 @@
(LVM_INT16)(NumChannels * NumSamples)); /* Number of input samples */
}
-
/*
* Update the delay sample count
*/
@@ -198,147 +190,6 @@
pInstance->SamplesToProcess = 0; /* All Samples used */
}
}
-#else
-void LVM_BufferManagedIn(LVM_Handle_t hInstance,
- const LVM_INT16 *pInData,
- LVM_INT16 **pToProcess,
- LVM_INT16 **pProcessed,
- LVM_UINT16 *pNumSamples)
-{
-
- LVM_INT16 SampleCount; /* Number of samples to be processed this call */
- LVM_INT16 NumSamples; /* Number of samples in scratch buffer */
- LVM_INT16 *pStart;
- LVM_Instance_t *pInstance = (LVM_Instance_t *)hInstance;
- LVM_Buffer_t *pBuffer;
- LVM_INT16 *pDest;
- LVM_INT16 NumChannels = 2;
-
- /*
- * Set the processing address pointers
- */
- pBuffer = pInstance->pBufferManagement;
- pDest = pBuffer->pScratch;
- *pToProcess = pBuffer->pScratch;
- *pProcessed = pBuffer->pScratch;
-
- /*
- * Check if it is the first call of a block
- */
- if (pInstance->SamplesToProcess == 0)
- {
- /*
- * First call for a new block of samples
- */
- pInstance->SamplesToProcess = (LVM_INT16)(*pNumSamples + pBuffer->InDelaySamples);
- pInstance->pInputSamples = (LVM_INT16 *)pInData;
- pBuffer->BufferState = LVM_FIRSTCALL;
- }
- pStart = pInstance->pInputSamples; /* Pointer to the input samples */
- pBuffer->SamplesToOutput = 0; /* Samples to output is same as number read for inplace processing */
-
-
- /*
- * Calculate the number of samples to process this call and update the buffer state
- */
- if (pInstance->SamplesToProcess > pInstance->InternalBlockSize)
- {
- /*
- * Process the maximum bock size of samples.
- */
- SampleCount = pInstance->InternalBlockSize;
- NumSamples = pInstance->InternalBlockSize;
- }
- else
- {
- /*
- * Last call for the block, so calculate how many frames and samples to process
- */
- LVM_INT16 NumFrames;
-
- NumSamples = pInstance->SamplesToProcess;
- NumFrames = (LVM_INT16)(NumSamples >> MIN_INTERNAL_BLOCKSHIFT);
- SampleCount = (LVM_INT16)(NumFrames << MIN_INTERNAL_BLOCKSHIFT);
-
- /*
- * Update the buffer state
- */
- if (pBuffer->BufferState == LVM_FIRSTCALL)
- {
- pBuffer->BufferState = LVM_FIRSTLASTCALL;
- }
- else
- {
- pBuffer->BufferState = LVM_LASTCALL;
- }
- }
- *pNumSamples = (LVM_UINT16)SampleCount; /* Set the number of samples to process this call */
-
-
- /*
- * Copy samples from the delay buffer as required
- */
- if (((pBuffer->BufferState == LVM_FIRSTCALL) ||
- (pBuffer->BufferState == LVM_FIRSTLASTCALL)) &&
- (pBuffer->InDelaySamples != 0))
- {
- Copy_16(&pBuffer->InDelayBuffer[0], /* Source */
- pDest, /* Destination */
- (LVM_INT16)(NumChannels*pBuffer->InDelaySamples)); /* Number of delay samples, left and right */
- NumSamples = (LVM_INT16)(NumSamples - pBuffer->InDelaySamples); /* Update sample count */
- pDest += NumChannels * pBuffer->InDelaySamples; /* Update the destination pointer */
- }
-
-
- /*
- * Copy the rest of the samples for this call from the input buffer
- */
- if (NumSamples > 0)
- {
- Copy_16(pStart, /* Source */
- pDest, /* Destination */
- (LVM_INT16)(NumChannels*NumSamples)); /* Number of input samples */
- pStart += NumChannels * NumSamples; /* Update the input pointer */
-
- /*
- * Update the input data pointer and samples to output
- */
- pBuffer->SamplesToOutput = (LVM_INT16)(pBuffer->SamplesToOutput + NumSamples); /* Update samples to output */
- }
-
-
- /*
- * Update the sample count and input pointer
- */
- pInstance->SamplesToProcess = (LVM_INT16)(pInstance->SamplesToProcess - SampleCount); /* Update the count of samples */
- pInstance->pInputSamples = pStart; /* Update input sample pointer */
-
-
- /*
- * Save samples to the delay buffer if any left unprocessed
- */
- if ((pBuffer->BufferState == LVM_FIRSTLASTCALL) ||
- (pBuffer->BufferState == LVM_LASTCALL))
- {
- NumSamples = pInstance->SamplesToProcess;
- pStart = pBuffer->pScratch; /* Start of the buffer */
- pStart += NumChannels*SampleCount; /* Offset by the number of processed samples */
- if (NumSamples != 0)
- {
- Copy_16(pStart, /* Source */
- &pBuffer->InDelayBuffer[0], /* Destination */
- (LVM_INT16)(NumChannels*NumSamples)); /* Number of input samples */
- }
-
-
- /*
- * Update the delay sample count
- */
- pBuffer->InDelaySamples = NumSamples; /* Number of delay sample pairs */
- pInstance->SamplesToProcess = 0; /* All Samples used */
- }
-}
-#endif
/****************************************************************************************/
/* */
@@ -362,7 +213,6 @@
/* NOTES: */
/* */
/****************************************************************************************/
-#ifdef BUILD_FLOAT
void LVM_BufferUnmanagedIn(LVM_Handle_t hInstance,
LVM_FLOAT **pToProcess,
LVM_FLOAT **pProcessed,
@@ -371,7 +221,6 @@
LVM_Instance_t *pInstance = (LVM_Instance_t *)hInstance;
-
/*
* Check if this is the first call of a block
*/
@@ -382,7 +231,6 @@
pInstance->pInputSamples = *pToProcess; /* Get the I/O pointers */
pInstance->pOutputSamples = *pProcessed;
-
/*
* Set te block size to process
*/
@@ -402,46 +250,6 @@
*pToProcess = pInstance->pInputSamples;
*pProcessed = pInstance->pOutputSamples;
}
-#else
-void LVM_BufferUnmanagedIn(LVM_Handle_t hInstance,
- LVM_INT16 **pToProcess,
- LVM_INT16 **pProcessed,
- LVM_UINT16 *pNumSamples)
-{
-
- LVM_Instance_t *pInstance = (LVM_Instance_t *)hInstance;
-
-
- /*
- * Check if this is the first call of a block
- */
- if (pInstance->SamplesToProcess == 0)
- {
- pInstance->SamplesToProcess = (LVM_INT16)*pNumSamples; /* Get the number of samples on first call */
- pInstance->pInputSamples = *pToProcess; /* Get the I/O pointers */
- pInstance->pOutputSamples = *pProcessed;
-
-
- /*
- * Set te block size to process
- */
- if (pInstance->SamplesToProcess > pInstance->InternalBlockSize)
- {
- *pNumSamples = (LVM_UINT16)pInstance->InternalBlockSize;
- }
- else
- {
- *pNumSamples = (LVM_UINT16)pInstance->SamplesToProcess;
- }
- }
-
- /*
- * Set the process pointers
- */
- *pToProcess = pInstance->pInputSamples;
- *pProcessed = pInstance->pOutputSamples;
-}
-#endif
/****************************************************************************************/
/* */
@@ -467,146 +275,6 @@
/* */
/****************************************************************************************/
-#ifndef BUILD_FLOAT
-void LVM_BufferOptimisedIn(LVM_Handle_t hInstance,
- const LVM_INT16 *pInData,
- LVM_INT16 **pToProcess,
- LVM_INT16 **pProcessed,
- LVM_UINT16 *pNumSamples)
-{
-
- LVM_Instance_t *pInstance = (LVM_Instance_t *)hInstance;
- LVM_Buffer_t *pBuffer = pInstance->pBufferManagement;
- LVM_INT16 *pDest;
- LVM_INT16 SampleCount;
- LVM_INT16 NumSamples;
- LVM_INT16 NumFrames;
-
- /*
- * Check if it is the first call for this block
- */
- if (pInstance->SamplesToProcess == 0)
- {
- /*
- * First call for a new block of samples
- */
- pBuffer->BufferState = LVM_FIRSTCALL;
- pInstance->pInputSamples = (LVM_INT16 *)pInData;
- pInstance->SamplesToProcess = (LVM_INT16)*pNumSamples;
- pBuffer->SamplesToOutput = (LVM_INT16)*pNumSamples;
- pDest = *pProcessed; /* The start of the output buffer */
-
-
- /*
- * Copy the already processed samples to the output buffer
- */
- if (pBuffer->OutDelaySamples != 0)
- {
- Copy_16(&pBuffer->OutDelayBuffer[0], /* Source */
- pDest, /* Destination */
- (LVM_INT16)(2*pBuffer->OutDelaySamples)); /* Number of delay samples */
- pDest += 2 * pBuffer->OutDelaySamples; /* Update the output pointer */
- pBuffer->SamplesToOutput = (LVM_INT16)(pBuffer->SamplesToOutput - pBuffer->OutDelaySamples); /* Update the numbr of samples to output */
- }
- *pToProcess = pDest; /* Set the address to start processing */
- *pProcessed = pDest; /* Process in the output buffer, now inplace */
-
- /*
- * Copy the input delay buffer (unprocessed) samples to the output buffer
- */
- if (pBuffer->InDelaySamples != 0)
- {
- Copy_16(&pBuffer->InDelayBuffer[0], /* Source */
- pDest, /* Destination */
- (LVM_INT16)(2*pBuffer->InDelaySamples)); /* Number of delay samples */
- pDest += 2 * pBuffer->InDelaySamples; /* Update the output pointer */
- }
-
-
- /*
- * Calculate how many input samples to process and copy
- */
- NumSamples = (LVM_INT16)(*pNumSamples - pBuffer->OutDelaySamples); /* Number that will fit in the output buffer */
- if (NumSamples >= pInstance->InternalBlockSize)
- {
- NumSamples = pInstance->InternalBlockSize;
- }
- NumFrames = (LVM_INT16)(NumSamples >> MIN_INTERNAL_BLOCKSHIFT);
- SampleCount = (LVM_INT16)(NumFrames << MIN_INTERNAL_BLOCKSHIFT);
- *pNumSamples = (LVM_UINT16)SampleCount; /* The number of samples to process */
- pBuffer->SamplesToOutput = (LVM_INT16)(pBuffer->SamplesToOutput - SampleCount); /* Update the number of samples to output */
- SampleCount = (LVM_INT16)(SampleCount - pBuffer->InDelaySamples); /* The number of samples to copy from the input */
-
-
- /*
- * Copy samples from the input buffer and update counts and pointers
- */
- Copy_16(pInstance->pInputSamples, /* Source */
- pDest, /* Destination */
- (LVM_INT16)(2*SampleCount)); /* Number of input samples */
- pInstance->pInputSamples += 2 * SampleCount; /* Update the input pointer */
- pInstance->pOutputSamples = pDest + (2 * SampleCount); /* Update the output pointer */
- pInstance->SamplesToProcess = (LVM_INT16)(pInstance->SamplesToProcess - SampleCount); /* Samples left in the input buffer */
- }
- else
- {
- /*
- * Second or subsequent call in optimised mode
- */
- if (pBuffer->SamplesToOutput >= MIN_INTERNAL_BLOCKSIZE)
- {
- /*
- * More samples can be processed directly in the output buffer
- */
- *pToProcess = pInstance->pOutputSamples; /* Set the address to start processing */
- *pProcessed = pInstance->pOutputSamples; /* Process in the output buffer, now inplace */
- NumSamples = pBuffer->SamplesToOutput; /* Number that will fit in the output buffer */
- if (NumSamples >= pInstance->InternalBlockSize)
- {
- NumSamples = pInstance->InternalBlockSize;
- }
- NumFrames = (LVM_INT16)(NumSamples >> MIN_INTERNAL_BLOCKSHIFT);
- SampleCount = (LVM_INT16)(NumFrames << MIN_INTERNAL_BLOCKSHIFT);
- *pNumSamples = (LVM_UINT16)SampleCount; /* The number of samples to process */
-
-
- /*
- * Copy samples from the input buffer and update counts and pointers
- */
- Copy_16(pInstance->pInputSamples, /* Source */
- pInstance->pOutputSamples, /* Destination */
- (LVM_INT16)(2*SampleCount)); /* Number of input samples */
- pInstance->pInputSamples += 2 * SampleCount; /* Update the input pointer */
- pInstance->pOutputSamples += 2 * SampleCount; /* Update the output pointer */
- pInstance->SamplesToProcess = (LVM_INT16)(pInstance->SamplesToProcess - SampleCount); /* Samples left in the input buffer */
- pBuffer->SamplesToOutput = (LVM_INT16)(pBuffer->SamplesToOutput - SampleCount); /* Number that will fit in the output buffer */
- }
- else
- {
- /*
- * The remaining samples can not be processed in the output buffer
- */
- pBuffer->BufferState = LVM_LASTCALL; /* Indicate this is the last bock to process */
- *pToProcess = pBuffer->pScratch; /* Set the address to start processing */
- *pProcessed = pBuffer->pScratch; /* Process in the output buffer, now inplace */
- NumSamples = pInstance->SamplesToProcess; /* Number left to be processed */
- NumFrames = (LVM_INT16)(NumSamples >> MIN_INTERNAL_BLOCKSHIFT);
- SampleCount = (LVM_INT16)(NumFrames << MIN_INTERNAL_BLOCKSHIFT);
- *pNumSamples = (LVM_UINT16)SampleCount; /* The number of samples to process */
-
-
- /*
- * Copy samples from the input buffer and update counts and pointers
- */
- Copy_16(pInstance->pInputSamples, /* Source */
- pBuffer->pScratch, /* Destination */
- (LVM_INT16)(2*SampleCount)); /* Number of input samples */
- pInstance->pInputSamples += 2 * SampleCount; /* Update the input pointer */
- pInstance->SamplesToProcess = (LVM_INT16)(pInstance->SamplesToProcess - SampleCount); /* Samples left in the input buffer */
- }
- }
-}
-#endif
/****************************************************************************************/
/* */
/* FUNCTION: LVM_BufferIn */
@@ -661,7 +329,6 @@
/* NOTES: */
/* */
/****************************************************************************************/
-#ifdef BUILD_FLOAT
void LVM_BufferIn(LVM_Handle_t hInstance,
const LVM_FLOAT *pInData,
LVM_FLOAT **pToProcess,
@@ -671,7 +338,6 @@
LVM_Instance_t *pInstance = (LVM_Instance_t *)hInstance;
-
/*
* Check which mode, managed or unmanaged
*/
@@ -691,37 +357,6 @@
pNumSamples);
}
}
-#else
-void LVM_BufferIn(LVM_Handle_t hInstance,
- const LVM_INT16 *pInData,
- LVM_INT16 **pToProcess,
- LVM_INT16 **pProcessed,
- LVM_UINT16 *pNumSamples)
-{
-
- LVM_Instance_t *pInstance = (LVM_Instance_t *)hInstance;
-
-
- /*
- * Check which mode, managed or unmanaged
- */
- if (pInstance->InstParams.BufferMode == LVM_MANAGED_BUFFERS)
- {
- LVM_BufferManagedIn(hInstance,
- pInData,
- pToProcess,
- pProcessed,
- pNumSamples);
- }
- else
- {
- LVM_BufferUnmanagedIn(hInstance,
- pToProcess,
- pProcessed,
- pNumSamples);
- }
-}
-#endif
/****************************************************************************************/
/* */
/* FUNCTION: LVM_BufferManagedOut */
@@ -742,7 +377,6 @@
/* NOTES: */
/* */
/****************************************************************************************/
-#ifdef BUILD_FLOAT
void LVM_BufferManagedOut(LVM_Handle_t hInstance,
LVM_FLOAT *pOutData,
LVM_UINT16 *pNumSamples)
@@ -777,7 +411,6 @@
}
pDest = pInstance->pOutputSamples; /* Set the output address */
-
/*
* If the number of samples is non-zero then there are still samples to send to
* the output buffer
@@ -859,7 +492,6 @@
}
}
-
/*
* Copy the processed results to the output
*/
@@ -920,7 +552,6 @@
}
}
-
/*
* Copy the remaining processed data to the output delay buffer
*/
@@ -950,157 +581,6 @@
/* This will terminate the loop when all samples processed */
*pNumSamples = (LVM_UINT16)pInstance->SamplesToProcess;
}
-#else
-void LVM_BufferManagedOut(LVM_Handle_t hInstance,
- LVM_INT16 *pOutData,
- LVM_UINT16 *pNumSamples)
-{
-
- LVM_Instance_t *pInstance = (LVM_Instance_t *)hInstance;
- LVM_Buffer_t *pBuffer = pInstance->pBufferManagement;
- LVM_INT16 SampleCount = (LVM_INT16)*pNumSamples;
- LVM_INT16 NumSamples;
- LVM_INT16 *pStart;
- LVM_INT16 *pDest;
-
-
- /*
- * Set the pointers
- */
- NumSamples = pBuffer->SamplesToOutput;
- pStart = pBuffer->pScratch;
-
-
- /*
- * check if it is the first call of a block
- */
- if ((pBuffer->BufferState == LVM_FIRSTCALL) ||
- (pBuffer->BufferState == LVM_FIRSTLASTCALL))
- {
- /* First call for a new block */
- pInstance->pOutputSamples = pOutData; /* Initialise the destination */
- }
- pDest = pInstance->pOutputSamples; /* Set the output address */
-
-
- /*
- * If the number of samples is non-zero then there are still samples to send to
- * the output buffer
- */
- if ((NumSamples != 0) &&
- (pBuffer->OutDelaySamples != 0))
- {
- /*
- * Copy the delayed output buffer samples to the output
- */
- if (pBuffer->OutDelaySamples <= NumSamples)
- {
- /*
- * Copy all output delay samples to the output
- */
- Copy_16(&pBuffer->OutDelayBuffer[0], /* Source */
- pDest, /* Detsination */
- (LVM_INT16)(2*pBuffer->OutDelaySamples)); /* Number of delay samples */
-
- /*
- * Update the pointer and sample counts
- */
- pDest += 2*pBuffer->OutDelaySamples; /* Output sample pointer */
- NumSamples = (LVM_INT16)(NumSamples - pBuffer->OutDelaySamples); /* Samples left to send */
- pBuffer->OutDelaySamples = 0; /* No samples left in the buffer */
-
- }
- else
- {
- /*
- * Copy only some of the ouput delay samples to the output
- */
- Copy_16(&pBuffer->OutDelayBuffer[0], /* Source */
- pDest, /* Detsination */
- (LVM_INT16)(2*NumSamples)); /* Number of delay samples */
-
- /*
- * Update the pointer and sample counts
- */
- pDest += 2*NumSamples; /* Output sample pointer */
- pBuffer->OutDelaySamples = (LVM_INT16)(pBuffer->OutDelaySamples - NumSamples); /* No samples left in the buffer */
-
-
- /*
- * Realign the delay buffer data to avoid using circular buffer management
- */
- Copy_16(&pBuffer->OutDelayBuffer[2*NumSamples], /* Source */
- &pBuffer->OutDelayBuffer[0], /* Destination */
- (LVM_INT16)(2*pBuffer->OutDelaySamples)); /* Number of samples to move */
- NumSamples = 0; /* Samples left to send */
- }
- }
-
-
- /*
- * Copy the processed results to the output
- */
- if ((NumSamples != 0) &&
- (SampleCount != 0))
- {
- if (SampleCount <= NumSamples)
- {
- /*
- * Copy all processed samples to the output
- */
- Copy_16(pStart, /* Source */
- pDest, /* Detsination */
- (LVM_INT16)(2*SampleCount)); /* Number of processed samples */
-
- /*
- * Update the pointer and sample counts
- */
- pDest += 2 * SampleCount; /* Output sample pointer */
- NumSamples = (LVM_INT16)(NumSamples - SampleCount); /* Samples left to send */
- SampleCount = 0; /* No samples left in the buffer */
- }
- else
- {
- /*
- * Copy only some processed samples to the output
- */
- Copy_16(pStart, /* Source */
- pDest, /* Destination */
- (LVM_INT16)(2*NumSamples)); /* Number of processed samples */
-
-
- /*
- * Update the pointers and sample counts
- */
- pStart += 2 * NumSamples; /* Processed sample pointer */
- pDest += 2 * NumSamples; /* Output sample pointer */
- SampleCount = (LVM_INT16)(SampleCount - NumSamples); /* Processed samples left */
- NumSamples = 0; /* Clear the sample count */
- }
- }
-
-
- /*
- * Copy the remaining processed data to the output delay buffer
- */
- if (SampleCount != 0)
- {
- Copy_16(pStart, /* Source */
- &pBuffer->OutDelayBuffer[2*pBuffer->OutDelaySamples], /* Destination */
- (LVM_INT16)(2*SampleCount)); /* Number of processed samples */
- pBuffer->OutDelaySamples = (LVM_INT16)(pBuffer->OutDelaySamples + SampleCount); /* Update the buffer count */
- }
-
-
- /*
- * pointers, counts and set default buffer processing
- */
- pBuffer->SamplesToOutput = NumSamples; /* Samples left to send */
- pInstance->pOutputSamples = pDest; /* Output sample pointer */
- pBuffer->BufferState = LVM_MAXBLOCKCALL; /* Set for the default call block size */
- *pNumSamples = (LVM_UINT16)pInstance->SamplesToProcess; /* This will terminate the loop when all samples processed */
-}
-#endif
/****************************************************************************************/
/* */
@@ -1139,7 +619,6 @@
LVM_INT16 NumChannels = 2;
#endif
-
/*
* Update sample counts
*/
@@ -1164,7 +643,6 @@
}
}
-
/****************************************************************************************/
/* */
/* FUNCTION: LVM_BufferOptimisedOut */
@@ -1184,73 +662,6 @@
/* */
/****************************************************************************************/
-#ifndef BUILD_FLOAT
-void LVM_BufferOptimisedOut(LVM_Handle_t hInstance,
- LVM_UINT16 *pNumSamples)
-{
-
- LVM_Instance_t *pInstance = (LVM_Instance_t *)hInstance;
- LVM_Buffer_t *pBuffer = pInstance->pBufferManagement;
-
- /*
- * Check if it is the last block to process
- */
- if (pBuffer->BufferState == LVM_LASTCALL)
- {
- LVM_INT16 *pSrc = pBuffer->pScratch;
-
- /*
- * Copy the unprocessed samples to the input delay buffer
- */
- if (pInstance->SamplesToProcess != 0)
- {
- Copy_16(pInstance->pInputSamples, /* Source */
- &pBuffer->InDelayBuffer[0], /* Destination */
- (LVM_INT16)(2*pInstance->SamplesToProcess)); /* Number of input samples */
- pBuffer->InDelaySamples = pInstance->SamplesToProcess;
- pInstance->SamplesToProcess = 0;
- }
- else
- {
- pBuffer->InDelaySamples = 0;
- }
-
-
- /*
- * Fill the last empty spaces in the output buffer
- */
- if (pBuffer->SamplesToOutput != 0)
- {
- Copy_16(pSrc, /* Source */
- pInstance->pOutputSamples, /* Destination */
- (LVM_INT16)(2*pBuffer->SamplesToOutput)); /* Number of input samples */
- *pNumSamples = (LVM_UINT16)(*pNumSamples - pBuffer->SamplesToOutput);
- pSrc += 2 * pBuffer->SamplesToOutput; /* Update scratch pointer */
- pBuffer->SamplesToOutput = 0; /* No more samples in this block */
- }
-
-
- /*
- * Save any remaining processed samples in the output delay buffer
- */
- if (*pNumSamples != 0)
- {
- Copy_16(pSrc, /* Source */
- &pBuffer->OutDelayBuffer[0], /* Destination */
- (LVM_INT16)(2**pNumSamples)); /* Number of input samples */
-
- pBuffer->OutDelaySamples = (LVM_INT16)*pNumSamples;
-
- *pNumSamples = 0; /* No more samples in this block */
- }
- else
- {
- pBuffer->OutDelaySamples = 0;
- }
- }
-}
-#endif
-
/****************************************************************************************/
/* */
/* FUNCTION: LVM_BufferOut */
@@ -1287,7 +698,6 @@
/* NOTES: */
/* */
/****************************************************************************************/
-#ifdef BUILD_FLOAT
void LVM_BufferOut(LVM_Handle_t hInstance,
LVM_FLOAT *pOutData,
LVM_UINT16 *pNumSamples)
@@ -1295,7 +705,6 @@
LVM_Instance_t *pInstance = (LVM_Instance_t *)hInstance;
-
/*
* Check which mode, managed or unmanaged
*/
@@ -1311,28 +720,3 @@
pNumSamples);
}
}
-#else
-void LVM_BufferOut(LVM_Handle_t hInstance,
- LVM_INT16 *pOutData,
- LVM_UINT16 *pNumSamples)
-{
-
- LVM_Instance_t *pInstance = (LVM_Instance_t *)hInstance;
-
-
- /*
- * Check which mode, managed or unmanaged
- */
- if (pInstance->InstParams.BufferMode == LVM_MANAGED_BUFFERS)
- {
- LVM_BufferManagedOut(hInstance,
- pOutData,
- pNumSamples);
- }
- else
- {
- LVM_BufferUnmanagedOut(hInstance,
- pNumSamples);
- }
-}
-#endif
diff --git a/media/libeffects/lvm/lib/Bundle/src/LVM_Coeffs.h b/media/libeffects/lvm/lib/Bundle/src/LVM_Coeffs.h
index bab4049..812f8e5 100644
--- a/media/libeffects/lvm/lib/Bundle/src/LVM_Coeffs.h
+++ b/media/libeffects/lvm/lib/Bundle/src/LVM_Coeffs.h
@@ -18,7 +18,6 @@
#ifndef __LVM_COEFFS_H__
#define __LVM_COEFFS_H__
-
/************************************************************************************/
/* */
/* High Pass Shelving Filter coefficients */
@@ -29,7 +28,6 @@
#define TrebleBoostMinRate 4
#define TrebleBoostSteps 15
-#ifdef BUILD_FLOAT
/* Coefficients for sample rate 22050Hz */
/* Gain = 1.000000 dB */
#define HPF_Fs22050_Gain1_A0 1.038434
@@ -486,7 +484,6 @@
#define HPF_Fs48000_Gain15_B1 (-0.267949)
#define HPF_Fs48000_Gain15_B2 0.000000
-#ifdef HIGHER_FS
/* Coefficients for sample rate 88200 */
/* Gain = 1.000000 dB */
#define HPF_Fs88200_Gain1_A0 1.094374f
@@ -856,547 +853,3 @@
#define HPF_Fs192000_Gain15_B2 0.000000
#endif
-
-#else
-/* Coefficients for sample rate 22050Hz */
- /* Gain = 1.000000 dB */
-#define HPF_Fs22050_Gain1_A0 5383 /* Floating point value 0.164291 */
-#define HPF_Fs22050_Gain1_A1 16859 /* Floating point value 0.514492 */
-#define HPF_Fs22050_Gain1_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs22050_Gain1_B1 12125 /* Floating point value 0.370033 */
-#define HPF_Fs22050_Gain1_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs22050_Gain1_Shift 1 /* Shift value */
- /* Gain = 2.000000 dB */
-#define HPF_Fs22050_Gain2_A0 4683 /* Floating point value 0.142925 */
-#define HPF_Fs22050_Gain2_A1 17559 /* Floating point value 0.535858 */
-#define HPF_Fs22050_Gain2_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs22050_Gain2_B1 12125 /* Floating point value 0.370033 */
-#define HPF_Fs22050_Gain2_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs22050_Gain2_Shift 1 /* Shift value */
- /* Gain = 3.000000 dB */
-#define HPF_Fs22050_Gain3_A0 3898 /* Floating point value 0.118953 */
-#define HPF_Fs22050_Gain3_A1 18345 /* Floating point value 0.559830 */
-#define HPF_Fs22050_Gain3_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs22050_Gain3_B1 12125 /* Floating point value 0.370033 */
-#define HPF_Fs22050_Gain3_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs22050_Gain3_Shift 1 /* Shift value */
- /* Gain = 4.000000 dB */
-#define HPF_Fs22050_Gain4_A0 3016 /* Floating point value 0.092055 */
-#define HPF_Fs22050_Gain4_A1 19226 /* Floating point value 0.586728 */
-#define HPF_Fs22050_Gain4_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs22050_Gain4_B1 12125 /* Floating point value 0.370033 */
-#define HPF_Fs22050_Gain4_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs22050_Gain4_Shift 1 /* Shift value */
- /* Gain = 5.000000 dB */
-#define HPF_Fs22050_Gain5_A0 2028 /* Floating point value 0.061876 */
-#define HPF_Fs22050_Gain5_A1 20215 /* Floating point value 0.616907 */
-#define HPF_Fs22050_Gain5_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs22050_Gain5_B1 12125 /* Floating point value 0.370033 */
-#define HPF_Fs22050_Gain5_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs22050_Gain5_Shift 1 /* Shift value */
- /* Gain = 6.000000 dB */
-#define HPF_Fs22050_Gain6_A0 918 /* Floating point value 0.028013 */
-#define HPF_Fs22050_Gain6_A1 21324 /* Floating point value 0.650770 */
-#define HPF_Fs22050_Gain6_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs22050_Gain6_B1 12125 /* Floating point value 0.370033 */
-#define HPF_Fs22050_Gain6_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs22050_Gain6_Shift 1 /* Shift value */
- /* Gain = 7.000000 dB */
-#define HPF_Fs22050_Gain7_A0 (-164) /* Floating point value -0.005002 */
-#define HPF_Fs22050_Gain7_A1 11311 /* Floating point value 0.345199 */
-#define HPF_Fs22050_Gain7_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs22050_Gain7_B1 12125 /* Floating point value 0.370033 */
-#define HPF_Fs22050_Gain7_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs22050_Gain7_Shift 2 /* Shift value */
- /* Gain = 8.000000 dB */
-#define HPF_Fs22050_Gain8_A0 (-864) /* Floating point value -0.026368 */
-#define HPF_Fs22050_Gain8_A1 12012 /* Floating point value 0.366565 */
-#define HPF_Fs22050_Gain8_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs22050_Gain8_B1 12125 /* Floating point value 0.370033 */
-#define HPF_Fs22050_Gain8_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs22050_Gain8_Shift 2 /* Shift value */
- /* Gain = 9.000000 dB */
-#define HPF_Fs22050_Gain9_A0 (-1650) /* Floating point value -0.050340 */
-#define HPF_Fs22050_Gain9_A1 12797 /* Floating point value 0.390537 */
-#define HPF_Fs22050_Gain9_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs22050_Gain9_B1 12125 /* Floating point value 0.370033 */
-#define HPF_Fs22050_Gain9_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs22050_Gain9_Shift 2 /* Shift value */
- /* Gain = 10.000000 dB */
-#define HPF_Fs22050_Gain10_A0 (-2531) /* Floating point value -0.077238 */
-#define HPF_Fs22050_Gain10_A1 13679 /* Floating point value 0.417435 */
-#define HPF_Fs22050_Gain10_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs22050_Gain10_B1 12125 /* Floating point value 0.370033 */
-#define HPF_Fs22050_Gain10_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs22050_Gain10_Shift 2 /* Shift value */
- /* Gain = 11.000000 dB */
-#define HPF_Fs22050_Gain11_A0 (-3520) /* Floating point value -0.107417 */
-#define HPF_Fs22050_Gain11_A1 14667 /* Floating point value 0.447615 */
-#define HPF_Fs22050_Gain11_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs22050_Gain11_B1 12125 /* Floating point value 0.370033 */
-#define HPF_Fs22050_Gain11_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs22050_Gain11_Shift 2 /* Shift value */
- /* Gain = 12.000000 dB */
-#define HPF_Fs22050_Gain12_A0 (-4629) /* Floating point value -0.141279 */
-#define HPF_Fs22050_Gain12_A1 15777 /* Floating point value 0.481477 */
-#define HPF_Fs22050_Gain12_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs22050_Gain12_B1 12125 /* Floating point value 0.370033 */
-#define HPF_Fs22050_Gain12_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs22050_Gain12_Shift 2 /* Shift value */
- /* Gain = 13.000000 dB */
-#define HPF_Fs22050_Gain13_A0 (-2944) /* Floating point value -0.089849 */
-#define HPF_Fs22050_Gain13_A1 8531 /* Floating point value 0.260352 */
-#define HPF_Fs22050_Gain13_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs22050_Gain13_B1 12125 /* Floating point value 0.370033 */
-#define HPF_Fs22050_Gain13_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs22050_Gain13_Shift 3 /* Shift value */
- /* Gain = 14.000000 dB */
-#define HPF_Fs22050_Gain14_A0 (-3644) /* Floating point value -0.111215 */
-#define HPF_Fs22050_Gain14_A1 9231 /* Floating point value 0.281718 */
-#define HPF_Fs22050_Gain14_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs22050_Gain14_B1 12125 /* Floating point value 0.370033 */
-#define HPF_Fs22050_Gain14_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs22050_Gain14_Shift 3 /* Shift value */
- /* Gain = 15.000000 dB */
-#define HPF_Fs22050_Gain15_A0 (-4430) /* Floating point value -0.135187 */
-#define HPF_Fs22050_Gain15_A1 10017 /* Floating point value 0.305690 */
-#define HPF_Fs22050_Gain15_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs22050_Gain15_B1 12125 /* Floating point value 0.370033 */
-#define HPF_Fs22050_Gain15_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs22050_Gain15_Shift 3 /* Shift value */
-
-
-/* Coefficients for sample rate 24000Hz */
- /* Gain = 1.000000 dB */
-#define HPF_Fs24000_Gain1_A0 3625 /* Floating point value 0.110628 */
-#define HPF_Fs24000_Gain1_A1 16960 /* Floating point value 0.517578 */
-#define HPF_Fs24000_Gain1_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs24000_Gain1_B1 8780 /* Floating point value 0.267949 */
-#define HPF_Fs24000_Gain1_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs24000_Gain1_Shift 1 /* Shift value */
- /* Gain = 2.000000 dB */
-#define HPF_Fs24000_Gain2_A0 2811 /* Floating point value 0.085800 */
-#define HPF_Fs24000_Gain2_A1 17774 /* Floating point value 0.542406 */
-#define HPF_Fs24000_Gain2_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs24000_Gain2_B1 8780 /* Floating point value 0.267949 */
-#define HPF_Fs24000_Gain2_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs24000_Gain2_Shift 1 /* Shift value */
- /* Gain = 3.000000 dB */
-#define HPF_Fs24000_Gain3_A0 1899 /* Floating point value 0.057943 */
-#define HPF_Fs24000_Gain3_A1 18686 /* Floating point value 0.570263 */
-#define HPF_Fs24000_Gain3_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs24000_Gain3_B1 8780 /* Floating point value 0.267949 */
-#define HPF_Fs24000_Gain3_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs24000_Gain3_Shift 1 /* Shift value */
- /* Gain = 4.000000 dB */
-#define HPF_Fs24000_Gain4_A0 874 /* Floating point value 0.026687 */
-#define HPF_Fs24000_Gain4_A1 19711 /* Floating point value 0.601519 */
-#define HPF_Fs24000_Gain4_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs24000_Gain4_B1 8780 /* Floating point value 0.267949 */
-#define HPF_Fs24000_Gain4_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs24000_Gain4_Shift 1 /* Shift value */
- /* Gain = 5.000000 dB */
-#define HPF_Fs24000_Gain5_A0 (-275) /* Floating point value -0.008383 */
-#define HPF_Fs24000_Gain5_A1 20860 /* Floating point value 0.636589 */
-#define HPF_Fs24000_Gain5_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs24000_Gain5_B1 8780 /* Floating point value 0.267949 */
-#define HPF_Fs24000_Gain5_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs24000_Gain5_Shift 1 /* Shift value */
- /* Gain = 6.000000 dB */
-#define HPF_Fs24000_Gain6_A0 (-1564) /* Floating point value -0.047733 */
-#define HPF_Fs24000_Gain6_A1 22149 /* Floating point value 0.675938 */
-#define HPF_Fs24000_Gain6_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs24000_Gain6_B1 8780 /* Floating point value 0.267949 */
-#define HPF_Fs24000_Gain6_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs24000_Gain6_Shift 1 /* Shift value */
- /* Gain = 7.000000 dB */
-#define HPF_Fs24000_Gain7_A0 (-1509) /* Floating point value -0.046051 */
-#define HPF_Fs24000_Gain7_A1 11826 /* Floating point value 0.360899 */
-#define HPF_Fs24000_Gain7_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs24000_Gain7_B1 8780 /* Floating point value 0.267949 */
-#define HPF_Fs24000_Gain7_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs24000_Gain7_Shift 2 /* Shift value */
- /* Gain = 8.000000 dB */
-#define HPF_Fs24000_Gain8_A0 (-2323) /* Floating point value -0.070878 */
-#define HPF_Fs24000_Gain8_A1 12640 /* Floating point value 0.385727 */
-#define HPF_Fs24000_Gain8_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs24000_Gain8_B1 8780 /* Floating point value 0.267949 */
-#define HPF_Fs24000_Gain8_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs24000_Gain8_Shift 2 /* Shift value */
- /* Gain = 9.000000 dB */
-#define HPF_Fs24000_Gain9_A0 (-3235) /* Floating point value -0.098736 */
-#define HPF_Fs24000_Gain9_A1 13552 /* Floating point value 0.413584 */
-#define HPF_Fs24000_Gain9_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs24000_Gain9_B1 8780 /* Floating point value 0.267949 */
-#define HPF_Fs24000_Gain9_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs24000_Gain9_Shift 2 /* Shift value */
- /* Gain = 10.000000 dB */
-#define HPF_Fs24000_Gain10_A0 (-4260) /* Floating point value -0.129992 */
-#define HPF_Fs24000_Gain10_A1 14577 /* Floating point value 0.444841 */
-#define HPF_Fs24000_Gain10_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs24000_Gain10_B1 8780 /* Floating point value 0.267949 */
-#define HPF_Fs24000_Gain10_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs24000_Gain10_Shift 2 /* Shift value */
- /* Gain = 11.000000 dB */
-#define HPF_Fs24000_Gain11_A0 (-5409) /* Floating point value -0.165062 */
-#define HPF_Fs24000_Gain11_A1 15726 /* Floating point value 0.479911 */
-#define HPF_Fs24000_Gain11_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs24000_Gain11_B1 8780 /* Floating point value 0.267949 */
-#define HPF_Fs24000_Gain11_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs24000_Gain11_Shift 2 /* Shift value */
- /* Gain = 12.000000 dB */
-#define HPF_Fs24000_Gain12_A0 (-6698) /* Floating point value -0.204411 */
-#define HPF_Fs24000_Gain12_A1 17015 /* Floating point value 0.519260 */
-#define HPF_Fs24000_Gain12_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs24000_Gain12_B1 8780 /* Floating point value 0.267949 */
-#define HPF_Fs24000_Gain12_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs24000_Gain12_Shift 2 /* Shift value */
- /* Gain = 13.000000 dB */
-#define HPF_Fs24000_Gain13_A0 (-4082) /* Floating point value -0.124576 */
-#define HPF_Fs24000_Gain13_A1 9253 /* Floating point value 0.282374 */
-#define HPF_Fs24000_Gain13_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs24000_Gain13_B1 8780 /* Floating point value 0.267949 */
-#define HPF_Fs24000_Gain13_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs24000_Gain13_Shift 3 /* Shift value */
- /* Gain = 14.000000 dB */
-#define HPF_Fs24000_Gain14_A0 (-4896) /* Floating point value -0.149404 */
-#define HPF_Fs24000_Gain14_A1 10066 /* Floating point value 0.307202 */
-#define HPF_Fs24000_Gain14_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs24000_Gain14_B1 8780 /* Floating point value 0.267949 */
-#define HPF_Fs24000_Gain14_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs24000_Gain14_Shift 3 /* Shift value */
- /* Gain = 15.000000 dB */
-#define HPF_Fs24000_Gain15_A0 (-5808) /* Floating point value -0.177261 */
-#define HPF_Fs24000_Gain15_A1 10979 /* Floating point value 0.335059 */
-#define HPF_Fs24000_Gain15_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs24000_Gain15_B1 8780 /* Floating point value 0.267949 */
-#define HPF_Fs24000_Gain15_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs24000_Gain15_Shift 3 /* Shift value */
-
-
-/* Coefficients for sample rate 32000Hz */
- /* Gain = 1.000000 dB */
-#define HPF_Fs32000_Gain1_A0 17225 /* Floating point value 0.525677 */
-#define HPF_Fs32000_Gain1_A1 (-990) /* Floating point value -0.030227 */
-#define HPF_Fs32000_Gain1_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs32000_Gain1_B1 0 /* Floating point value -0.000000 */
-#define HPF_Fs32000_Gain1_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs32000_Gain1_Shift 1 /* Shift value */
- /* Gain = 2.000000 dB */
-#define HPF_Fs32000_Gain2_A0 18337 /* Floating point value 0.559593 */
-#define HPF_Fs32000_Gain2_A1 (-2102) /* Floating point value -0.064142 */
-#define HPF_Fs32000_Gain2_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs32000_Gain2_B1 0 /* Floating point value -0.000000 */
-#define HPF_Fs32000_Gain2_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs32000_Gain2_Shift 1 /* Shift value */
- /* Gain = 3.000000 dB */
-#define HPF_Fs32000_Gain3_A0 19584 /* Floating point value 0.597646 */
-#define HPF_Fs32000_Gain3_A1 (-3349) /* Floating point value -0.102196 */
-#define HPF_Fs32000_Gain3_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs32000_Gain3_B1 0 /* Floating point value -0.000000 */
-#define HPF_Fs32000_Gain3_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs32000_Gain3_Shift 1 /* Shift value */
- /* Gain = 4.000000 dB */
-#define HPF_Fs32000_Gain4_A0 20983 /* Floating point value 0.640343 */
-#define HPF_Fs32000_Gain4_A1 (-4748) /* Floating point value -0.144893 */
-#define HPF_Fs32000_Gain4_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs32000_Gain4_B1 0 /* Floating point value -0.000000 */
-#define HPF_Fs32000_Gain4_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs32000_Gain4_Shift 1 /* Shift value */
- /* Gain = 5.000000 dB */
-#define HPF_Fs32000_Gain5_A0 22553 /* Floating point value 0.688250 */
-#define HPF_Fs32000_Gain5_A1 (-6318) /* Floating point value -0.192799 */
-#define HPF_Fs32000_Gain5_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs32000_Gain5_B1 0 /* Floating point value -0.000000 */
-#define HPF_Fs32000_Gain5_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs32000_Gain5_Shift 1 /* Shift value */
- /* Gain = 6.000000 dB */
-#define HPF_Fs32000_Gain6_A0 24314 /* Floating point value 0.742002 */
-#define HPF_Fs32000_Gain6_A1 (-8079) /* Floating point value -0.246551 */
-#define HPF_Fs32000_Gain6_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs32000_Gain6_B1 0 /* Floating point value -0.000000 */
-#define HPF_Fs32000_Gain6_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs32000_Gain6_Shift 1 /* Shift value */
- /* Gain = 7.000000 dB */
-#define HPF_Fs32000_Gain7_A0 13176 /* Floating point value 0.402109 */
-#define HPF_Fs32000_Gain7_A1 (-5040) /* Floating point value -0.153795 */
-#define HPF_Fs32000_Gain7_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs32000_Gain7_B1 0 /* Floating point value -0.000000 */
-#define HPF_Fs32000_Gain7_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs32000_Gain7_Shift 2 /* Shift value */
- /* Gain = 8.000000 dB */
-#define HPF_Fs32000_Gain8_A0 14288 /* Floating point value 0.436024 */
-#define HPF_Fs32000_Gain8_A1 (-6151) /* Floating point value -0.187711 */
-#define HPF_Fs32000_Gain8_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs32000_Gain8_B1 0 /* Floating point value -0.000000 */
-#define HPF_Fs32000_Gain8_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs32000_Gain8_Shift 2 /* Shift value */
- /* Gain = 9.000000 dB */
-#define HPF_Fs32000_Gain9_A0 15535 /* Floating point value 0.474078 */
-#define HPF_Fs32000_Gain9_A1 (-7398) /* Floating point value -0.225764 */
-#define HPF_Fs32000_Gain9_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs32000_Gain9_B1 0 /* Floating point value -0.000000 */
-#define HPF_Fs32000_Gain9_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs32000_Gain9_Shift 2 /* Shift value */
- /* Gain = 10.000000 dB */
-#define HPF_Fs32000_Gain10_A0 16934 /* Floating point value 0.516774 */
-#define HPF_Fs32000_Gain10_A1 (-8797) /* Floating point value -0.268461 */
-#define HPF_Fs32000_Gain10_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs32000_Gain10_B1 0 /* Floating point value -0.000000 */
-#define HPF_Fs32000_Gain10_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs32000_Gain10_Shift 2 /* Shift value */
- /* Gain = 11.000000 dB */
-#define HPF_Fs32000_Gain11_A0 18503 /* Floating point value 0.564681 */
-#define HPF_Fs32000_Gain11_A1 (-10367) /* Floating point value -0.316368 */
-#define HPF_Fs32000_Gain11_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs32000_Gain11_B1 0 /* Floating point value -0.000000 */
-#define HPF_Fs32000_Gain11_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs32000_Gain11_Shift 2 /* Shift value */
- /* Gain = 12.000000 dB */
-#define HPF_Fs32000_Gain12_A0 20265 /* Floating point value 0.618433 */
-#define HPF_Fs32000_Gain12_A1 (-12128) /* Floating point value -0.370120 */
-#define HPF_Fs32000_Gain12_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs32000_Gain12_B1 0 /* Floating point value -0.000000 */
-#define HPF_Fs32000_Gain12_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs32000_Gain12_Shift 2 /* Shift value */
- /* Gain = 13.000000 dB */
-#define HPF_Fs32000_Gain13_A0 11147 /* Floating point value 0.340178 */
-#define HPF_Fs32000_Gain13_A1 (-7069) /* Floating point value -0.215726 */
-#define HPF_Fs32000_Gain13_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs32000_Gain13_B1 0 /* Floating point value -0.000000 */
-#define HPF_Fs32000_Gain13_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs32000_Gain13_Shift 3 /* Shift value */
- /* Gain = 14.000000 dB */
-#define HPF_Fs32000_Gain14_A0 12258 /* Floating point value 0.374093 */
-#define HPF_Fs32000_Gain14_A1 (-8180) /* Floating point value -0.249642 */
-#define HPF_Fs32000_Gain14_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs32000_Gain14_B1 0 /* Floating point value -0.000000 */
-#define HPF_Fs32000_Gain14_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs32000_Gain14_Shift 3 /* Shift value */
- /* Gain = 15.000000 dB */
-#define HPF_Fs32000_Gain15_A0 13505 /* Floating point value 0.412147 */
-#define HPF_Fs32000_Gain15_A1 (-9427) /* Floating point value -0.287695 */
-#define HPF_Fs32000_Gain15_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs32000_Gain15_B1 0 /* Floating point value -0.000000 */
-#define HPF_Fs32000_Gain15_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs32000_Gain15_Shift 3 /* Shift value */
-
-
-/* Coefficients for sample rate 44100Hz */
- /* Gain = 1.000000 dB */
-#define HPF_Fs44100_Gain1_A0 17442 /* Floating point value 0.532294 */
-#define HPF_Fs44100_Gain1_A1 (-4761) /* Floating point value -0.145294 */
-#define HPF_Fs44100_Gain1_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs44100_Gain1_B1 (-7173) /* Floating point value -0.218894 */
-#define HPF_Fs44100_Gain1_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs44100_Gain1_Shift 1 /* Shift value */
- /* Gain = 2.000000 dB */
-#define HPF_Fs44100_Gain2_A0 18797 /* Floating point value 0.573633 */
-#define HPF_Fs44100_Gain2_A1 (-6116) /* Floating point value -0.186634 */
-#define HPF_Fs44100_Gain2_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs44100_Gain2_B1 (-7173) /* Floating point value -0.218894 */
-#define HPF_Fs44100_Gain2_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs44100_Gain2_Shift 1 /* Shift value */
- /* Gain = 3.000000 dB */
-#define HPF_Fs44100_Gain3_A0 20317 /* Floating point value 0.620016 */
-#define HPF_Fs44100_Gain3_A1 (-7635) /* Floating point value -0.233017 */
-#define HPF_Fs44100_Gain3_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs44100_Gain3_B1 (-7173) /* Floating point value -0.218894 */
-#define HPF_Fs44100_Gain3_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs44100_Gain3_Shift 1 /* Shift value */
- /* Gain = 4.000000 dB */
-#define HPF_Fs44100_Gain4_A0 22022 /* Floating point value 0.672059 */
-#define HPF_Fs44100_Gain4_A1 (-9341) /* Floating point value -0.285060 */
-#define HPF_Fs44100_Gain4_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs44100_Gain4_B1 (-7173) /* Floating point value -0.218894 */
-#define HPF_Fs44100_Gain4_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs44100_Gain4_Shift 1 /* Shift value */
- /* Gain = 5.000000 dB */
-#define HPF_Fs44100_Gain5_A0 23935 /* Floating point value 0.730452 */
-#define HPF_Fs44100_Gain5_A1 (-11254) /* Floating point value -0.343453 */
-#define HPF_Fs44100_Gain5_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs44100_Gain5_B1 (-7173) /* Floating point value -0.218894 */
-#define HPF_Fs44100_Gain5_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs44100_Gain5_Shift 1 /* Shift value */
- /* Gain = 6.000000 dB */
-#define HPF_Fs44100_Gain6_A0 26082 /* Floating point value 0.795970 */
-#define HPF_Fs44100_Gain6_A1 (-13401) /* Floating point value -0.408971 */
-#define HPF_Fs44100_Gain6_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs44100_Gain6_B1 (-7173) /* Floating point value -0.218894 */
-#define HPF_Fs44100_Gain6_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs44100_Gain6_Shift 1 /* Shift value */
- /* Gain = 7.000000 dB */
-#define HPF_Fs44100_Gain7_A0 14279 /* Floating point value 0.435774 */
-#define HPF_Fs44100_Gain7_A1 (-7924) /* Floating point value -0.241815 */
-#define HPF_Fs44100_Gain7_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs44100_Gain7_B1 (-7173) /* Floating point value -0.218894 */
-#define HPF_Fs44100_Gain7_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs44100_Gain7_Shift 2 /* Shift value */
- /* Gain = 8.000000 dB */
-#define HPF_Fs44100_Gain8_A0 15634 /* Floating point value 0.477113 */
-#define HPF_Fs44100_Gain8_A1 (-9278) /* Floating point value -0.283154 */
-#define HPF_Fs44100_Gain8_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs44100_Gain8_B1 (-7173) /* Floating point value -0.218894 */
-#define HPF_Fs44100_Gain8_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs44100_Gain8_Shift 2 /* Shift value */
- /* Gain = 9.000000 dB */
-#define HPF_Fs44100_Gain9_A0 17154 /* Floating point value 0.523496 */
-#define HPF_Fs44100_Gain9_A1 (-10798) /* Floating point value -0.329537 */
-#define HPF_Fs44100_Gain9_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs44100_Gain9_B1 (-7173) /* Floating point value -0.218894 */
-#define HPF_Fs44100_Gain9_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs44100_Gain9_Shift 2 /* Shift value */
- /* Gain = 10.000000 dB */
-#define HPF_Fs44100_Gain10_A0 18859 /* Floating point value 0.575539 */
-#define HPF_Fs44100_Gain10_A1 (-12504) /* Floating point value -0.381580 */
-#define HPF_Fs44100_Gain10_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs44100_Gain10_B1 (-7173) /* Floating point value -0.218894 */
-#define HPF_Fs44100_Gain10_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs44100_Gain10_Shift 2 /* Shift value */
- /* Gain = 11.000000 dB */
-#define HPF_Fs44100_Gain11_A0 20773 /* Floating point value 0.633932 */
-#define HPF_Fs44100_Gain11_A1 (-14417) /* Floating point value -0.439973 */
-#define HPF_Fs44100_Gain11_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs44100_Gain11_B1 (-7173) /* Floating point value -0.218894 */
-#define HPF_Fs44100_Gain11_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs44100_Gain11_Shift 2 /* Shift value */
- /* Gain = 12.000000 dB */
-#define HPF_Fs44100_Gain12_A0 22920 /* Floating point value 0.699450 */
-#define HPF_Fs44100_Gain12_A1 (-16564) /* Floating point value -0.505491 */
-#define HPF_Fs44100_Gain12_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs44100_Gain12_B1 (-7173) /* Floating point value -0.218894 */
-#define HPF_Fs44100_Gain12_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs44100_Gain12_Shift 2 /* Shift value */
- /* Gain = 13.000000 dB */
-#define HPF_Fs44100_Gain13_A0 12694 /* Floating point value 0.387399 */
-#define HPF_Fs44100_Gain13_A1 (-9509) /* Floating point value -0.290189 */
-#define HPF_Fs44100_Gain13_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs44100_Gain13_B1 (-7173) /* Floating point value -0.218894 */
-#define HPF_Fs44100_Gain13_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs44100_Gain13_Shift 3 /* Shift value */
- /* Gain = 14.000000 dB */
-#define HPF_Fs44100_Gain14_A0 14049 /* Floating point value 0.428738 */
-#define HPF_Fs44100_Gain14_A1 (-10864) /* Floating point value -0.331528 */
-#define HPF_Fs44100_Gain14_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs44100_Gain14_B1 (-7173) /* Floating point value -0.218894 */
-#define HPF_Fs44100_Gain14_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs44100_Gain14_Shift 3 /* Shift value */
- /* Gain = 15.000000 dB */
-#define HPF_Fs44100_Gain15_A0 15569 /* Floating point value 0.475121 */
-#define HPF_Fs44100_Gain15_A1 (-12383) /* Floating point value -0.377912 */
-#define HPF_Fs44100_Gain15_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs44100_Gain15_B1 (-7173) /* Floating point value -0.218894 */
-#define HPF_Fs44100_Gain15_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs44100_Gain15_Shift 3 /* Shift value */
-
-
-/* Coefficients for sample rate 48000Hz */
- /* Gain = 1.000000 dB */
-#define HPF_Fs48000_Gain1_A0 17491 /* Floating point value 0.533777 */
-#define HPF_Fs48000_Gain1_A1 (-5606) /* Floating point value -0.171082 */
-#define HPF_Fs48000_Gain1_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs48000_Gain1_B1 (-8780) /* Floating point value -0.267949 */
-#define HPF_Fs48000_Gain1_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs48000_Gain1_Shift 1 /* Shift value */
- /* Gain = 2.000000 dB */
-#define HPF_Fs48000_Gain2_A0 18900 /* Floating point value 0.576779 */
-#define HPF_Fs48000_Gain2_A1 (-7015) /* Floating point value -0.214085 */
-#define HPF_Fs48000_Gain2_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs48000_Gain2_B1 (-8780) /* Floating point value -0.267949 */
-#define HPF_Fs48000_Gain2_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs48000_Gain2_Shift 1 /* Shift value */
- /* Gain = 3.000000 dB */
-#define HPF_Fs48000_Gain3_A0 20481 /* Floating point value 0.625029 */
-#define HPF_Fs48000_Gain3_A1 (-8596) /* Floating point value -0.262335 */
-#define HPF_Fs48000_Gain3_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs48000_Gain3_B1 (-8780) /* Floating point value -0.267949 */
-#define HPF_Fs48000_Gain3_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs48000_Gain3_Shift 1 /* Shift value */
- /* Gain = 4.000000 dB */
-#define HPF_Fs48000_Gain4_A0 22255 /* Floating point value 0.679167 */
-#define HPF_Fs48000_Gain4_A1 (-10370) /* Floating point value -0.316472 */
-#define HPF_Fs48000_Gain4_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs48000_Gain4_B1 (-8780) /* Floating point value -0.267949 */
-#define HPF_Fs48000_Gain4_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs48000_Gain4_Shift 1 /* Shift value */
- /* Gain = 5.000000 dB */
-#define HPF_Fs48000_Gain5_A0 24245 /* Floating point value 0.739910 */
-#define HPF_Fs48000_Gain5_A1 (-12361) /* Floating point value -0.377215 */
-#define HPF_Fs48000_Gain5_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs48000_Gain5_B1 (-8780) /* Floating point value -0.267949 */
-#define HPF_Fs48000_Gain5_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs48000_Gain5_Shift 1 /* Shift value */
- /* Gain = 6.000000 dB */
-#define HPF_Fs48000_Gain6_A0 26479 /* Floating point value 0.808065 */
-#define HPF_Fs48000_Gain6_A1 (-14594) /* Floating point value -0.445370 */
-#define HPF_Fs48000_Gain6_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs48000_Gain6_B1 (-8780) /* Floating point value -0.267949 */
-#define HPF_Fs48000_Gain6_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs48000_Gain6_Shift 1 /* Shift value */
- /* Gain = 7.000000 dB */
-#define HPF_Fs48000_Gain7_A0 14527 /* Floating point value 0.443318 */
-#define HPF_Fs48000_Gain7_A1 (-8570) /* Floating point value -0.261540 */
-#define HPF_Fs48000_Gain7_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs48000_Gain7_B1 (-8780) /* Floating point value -0.267949 */
-#define HPF_Fs48000_Gain7_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs48000_Gain7_Shift 2 /* Shift value */
- /* Gain = 8.000000 dB */
-#define HPF_Fs48000_Gain8_A0 15936 /* Floating point value 0.486321 */
-#define HPF_Fs48000_Gain8_A1 (-9979) /* Floating point value -0.304543 */
-#define HPF_Fs48000_Gain8_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs48000_Gain8_B1 (-8780) /* Floating point value -0.267949 */
-#define HPF_Fs48000_Gain8_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs48000_Gain8_Shift 2 /* Shift value */
- /* Gain = 9.000000 dB */
-#define HPF_Fs48000_Gain9_A0 17517 /* Floating point value 0.534571 */
-#define HPF_Fs48000_Gain9_A1 (-11560) /* Floating point value -0.352793 */
-#define HPF_Fs48000_Gain9_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs48000_Gain9_B1 (-8780) /* Floating point value -0.267949 */
-#define HPF_Fs48000_Gain9_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs48000_Gain9_Shift 2 /* Shift value */
- /* Gain = 10.000000 dB */
-#define HPF_Fs48000_Gain10_A0 19291 /* Floating point value 0.588708 */
-#define HPF_Fs48000_Gain10_A1 (-13334) /* Floating point value -0.406930 */
-#define HPF_Fs48000_Gain10_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs48000_Gain10_B1 (-8780) /* Floating point value -0.267949 */
-#define HPF_Fs48000_Gain10_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs48000_Gain10_Shift 2 /* Shift value */
- /* Gain = 11.000000 dB */
-#define HPF_Fs48000_Gain11_A0 21281 /* Floating point value 0.649452 */
-#define HPF_Fs48000_Gain11_A1 (-15325) /* Floating point value -0.467674 */
-#define HPF_Fs48000_Gain11_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs48000_Gain11_B1 (-8780) /* Floating point value -0.267949 */
-#define HPF_Fs48000_Gain11_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs48000_Gain11_Shift 2 /* Shift value */
- /* Gain = 12.000000 dB */
-#define HPF_Fs48000_Gain12_A0 23515 /* Floating point value 0.717607 */
-#define HPF_Fs48000_Gain12_A1 (-17558) /* Floating point value -0.535829 */
-#define HPF_Fs48000_Gain12_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs48000_Gain12_B1 (-8780) /* Floating point value -0.267949 */
-#define HPF_Fs48000_Gain12_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs48000_Gain12_Shift 2 /* Shift value */
- /* Gain = 13.000000 dB */
-#define HPF_Fs48000_Gain13_A0 13041 /* Floating point value 0.397982 */
-#define HPF_Fs48000_Gain13_A1 (-10056) /* Floating point value -0.306877 */
-#define HPF_Fs48000_Gain13_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs48000_Gain13_B1 (-8780) /* Floating point value -0.267949 */
-#define HPF_Fs48000_Gain13_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs48000_Gain13_Shift 3 /* Shift value */
- /* Gain = 14.000000 dB */
-#define HPF_Fs48000_Gain14_A0 14450 /* Floating point value 0.440984 */
-#define HPF_Fs48000_Gain14_A1 (-11465) /* Floating point value -0.349880 */
-#define HPF_Fs48000_Gain14_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs48000_Gain14_B1 (-8780) /* Floating point value -0.267949 */
-#define HPF_Fs48000_Gain14_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs48000_Gain14_Shift 3 /* Shift value */
- /* Gain = 15.000000 dB */
-#define HPF_Fs48000_Gain15_A0 16031 /* Floating point value 0.489234 */
-#define HPF_Fs48000_Gain15_A1 (-13046) /* Floating point value -0.398130 */
-#define HPF_Fs48000_Gain15_A2 0 /* Floating point value 0.000000 */
-#define HPF_Fs48000_Gain15_B1 (-8780) /* Floating point value -0.267949 */
-#define HPF_Fs48000_Gain15_B2 0 /* Floating point value 0.000000 */
-#define HPF_Fs48000_Gain15_Shift 3 /* Shift value */
-
-
-#endif
-#endif
diff --git a/media/libeffects/lvm/lib/Bundle/src/LVM_Control.c b/media/libeffects/lvm/lib/Bundle/src/LVM_Control.cpp
similarity index 89%
rename from media/libeffects/lvm/lib/Bundle/src/LVM_Control.c
rename to media/libeffects/lvm/lib/Bundle/src/LVM_Control.cpp
index 1b27cb4..ff2c90a 100644
--- a/media/libeffects/lvm/lib/Bundle/src/LVM_Control.c
+++ b/media/libeffects/lvm/lib/Bundle/src/LVM_Control.cpp
@@ -15,7 +15,6 @@
* limitations under the License.
*/
-
/****************************************************************************************/
/* */
/* Includes */
@@ -56,7 +55,6 @@
{
LVM_Instance_t *pInstance =(LVM_Instance_t *)hInstance;
-
if ((pParams == LVM_NULL) || (hInstance == LVM_NULL))
{
return (LVM_NULLADDRESS);
@@ -67,17 +65,11 @@
if(
/* General parameters */
((pParams->OperatingMode != LVM_MODE_OFF) && (pParams->OperatingMode != LVM_MODE_ON)) ||
-#if defined(BUILD_FLOAT) && defined(HIGHER_FS)
((pParams->SampleRate != LVM_FS_8000) && (pParams->SampleRate != LVM_FS_11025) && (pParams->SampleRate != LVM_FS_12000) &&
(pParams->SampleRate != LVM_FS_16000) && (pParams->SampleRate != LVM_FS_22050) && (pParams->SampleRate != LVM_FS_24000) &&
(pParams->SampleRate != LVM_FS_32000) && (pParams->SampleRate != LVM_FS_44100) && (pParams->SampleRate != LVM_FS_48000) &&
(pParams->SampleRate != LVM_FS_88200) && (pParams->SampleRate != LVM_FS_96000) &&
(pParams->SampleRate != LVM_FS_176400) && (pParams->SampleRate != LVM_FS_192000)) ||
-#else
- ((pParams->SampleRate != LVM_FS_8000) && (pParams->SampleRate != LVM_FS_11025) && (pParams->SampleRate != LVM_FS_12000) &&
- (pParams->SampleRate != LVM_FS_16000) && (pParams->SampleRate != LVM_FS_22050) && (pParams->SampleRate != LVM_FS_24000) &&
- (pParams->SampleRate != LVM_FS_32000) && (pParams->SampleRate != LVM_FS_44100) && (pParams->SampleRate != LVM_FS_48000)) ||
-#endif
#ifdef SUPPORT_MC
((pParams->SourceFormat != LVM_STEREO) &&
(pParams->SourceFormat != LVM_MONOINSTEREO) &&
@@ -198,13 +190,12 @@
/*
* PSA parameters
*/
- if( (pParams->PSA_PeakDecayRate > LVPSA_SPEED_HIGH) ||
+ if (((LVPSA_LevelDetectSpeed_en)pParams->PSA_PeakDecayRate > LVPSA_SPEED_HIGH) ||
(pParams->PSA_Enable > LVM_PSA_ON))
{
return (LVM_OUTOFRANGE);
}
-
/*
* Set the flag to indicate there are new parameters to use
*
@@ -218,7 +209,6 @@
return(LVM_SUCCESS);
}
-
/****************************************************************************************/
/* */
/* FUNCTION: LVM_GetControlParameters */
@@ -245,7 +235,6 @@
{
LVM_Instance_t *pInstance =(LVM_Instance_t *)hInstance;
-
/*
* Check pointer
*/
@@ -272,7 +261,6 @@
return(LVM_SUCCESS);
}
-
/****************************************************************************************/
/* */
/* FUNCTION: LVM_SetTrebleBoost */
@@ -289,11 +277,7 @@
void LVM_SetTrebleBoost(LVM_Instance_t *pInstance,
LVM_ControlParams_t *pParams)
{
-#ifdef BUILD_FLOAT
extern FO_FLOAT_LShx_Coefs_t LVM_TrebleBoostCoefs[];
-#else
- extern FO_C16_LShx_Coefs_t LVM_TrebleBoostCoefs[];
-#endif
LVM_INT16 Offset;
LVM_INT16 EffectLevel = 0;
@@ -324,7 +308,6 @@
* Load the coefficients and enabled the treble boost
*/
Offset = (LVM_INT16)(EffectLevel - 1 + TrebleBoostSteps * (pParams->SampleRate - TrebleBoostMinRate));
-#ifdef BUILD_FLOAT
FO_2I_D16F32Css_LShx_TRC_WRA_01_Init(&pInstance->pTE_State->TrebleBoost_State,
&pInstance->pTE_Taps->TrebleBoost_Taps,
&LVM_TrebleBoostCoefs[Offset]);
@@ -333,23 +316,10 @@
* Clear the taps
*/
LoadConst_Float((LVM_FLOAT)0, /* Value */
- (void *)&pInstance->pTE_Taps->TrebleBoost_Taps, /* Destination.\
+ (LVM_FLOAT *)&pInstance->pTE_Taps->TrebleBoost_Taps, /* Destination.\
Cast to void: no dereferencing in function */
(LVM_UINT16)(sizeof(pInstance->pTE_Taps->TrebleBoost_Taps) / \
sizeof(LVM_FLOAT))); /* Number of words */
-#else
- FO_2I_D16F32Css_LShx_TRC_WRA_01_Init(&pInstance->pTE_State->TrebleBoost_State,
- &pInstance->pTE_Taps->TrebleBoost_Taps,
- &LVM_TrebleBoostCoefs[Offset]);
-
- /*
- * Clear the taps
- */
- LoadConst_16((LVM_INT16)0, /* Value */
- (void *)&pInstance->pTE_Taps->TrebleBoost_Taps, /* Destination.\
- Cast to void: no dereferencing in function */
- (LVM_UINT16)(sizeof(pInstance->pTE_Taps->TrebleBoost_Taps)/sizeof(LVM_INT16))); /* Number of words */
-#endif
}
}
else
@@ -363,7 +333,6 @@
return;
}
-
/************************************************************************************/
/* */
/* FUNCTION: LVM_SetVolume */
@@ -383,9 +352,7 @@
LVM_UINT16 dBShifts; /* 6dB shifts */
LVM_UINT16 dBOffset; /* Table offset */
LVM_INT16 Volume = 0; /* Required volume in dBs */
-#ifdef BUILD_FLOAT
LVM_FLOAT Temp;
-#endif
/*
* Limit the gain to the maximum allowed
@@ -439,56 +406,36 @@
dBOffset = (LVM_UINT16)((-Volume) % 6); /* Get the dBs 0-5 */
dBShifts = (LVM_UINT16)(Volume / -6); /* Get the 6dB shifts */
-
/*
* Set the parameters
*/
if(dBShifts == 0)
{
-#ifdef BUILD_FLOAT
LVC_Mixer_SetTarget(&pInstance->VC_Volume.MixerStream[0],
(LVM_FLOAT)LVM_VolumeTable[dBOffset]);
-#else
- LVC_Mixer_SetTarget(&pInstance->VC_Volume.MixerStream[0],
- (LVM_INT32)LVM_VolumeTable[dBOffset]);
-#endif
}
else
{
-#ifdef BUILD_FLOAT
Temp = LVM_VolumeTable[dBOffset];
while(dBShifts) {
Temp = Temp / 2.0f;
dBShifts--;
}
LVC_Mixer_SetTarget(&pInstance->VC_Volume.MixerStream[0], Temp);
-#else
- LVC_Mixer_SetTarget(&pInstance->VC_Volume.MixerStream[0],
- (((LVM_INT32)LVM_VolumeTable[dBOffset])>>dBShifts));
-#endif
}
pInstance->VC_Volume.MixerStream[0].CallbackSet = 1;
if(pInstance->NoSmoothVolume == LVM_TRUE)
{
-#ifdef BUILD_FLOAT
LVC_Mixer_SetTimeConstant(&pInstance->VC_Volume.MixerStream[0], 0,
pInstance->Params.SampleRate, 2);
-#else
- LVC_Mixer_SetTimeConstant(&pInstance->VC_Volume.MixerStream[0],0,pInstance->Params.SampleRate,2);
-#endif
}
else
{
-#ifdef BUILD_FLOAT
LVC_Mixer_VarSlope_SetTimeConstant(&pInstance->VC_Volume.MixerStream[0],
LVM_VC_MIXER_TIME, pInstance->Params.SampleRate, 2);
-#else
- LVC_Mixer_VarSlope_SetTimeConstant(&pInstance->VC_Volume.MixerStream[0],LVM_VC_MIXER_TIME,pInstance->Params.SampleRate,2);
-#endif
}
}
-
/************************************************************************************/
/* */
/* FUNCTION: LVM_SetHeadroom */
@@ -513,8 +460,8 @@
LVM_INT16 Headroom = 0;
LVM_INT16 MaxGain = 0;
-
- if ((pParams->EQNB_OperatingMode == LVEQNB_ON) && (pInstance->HeadroomParams.Headroom_OperatingMode == LVM_HEADROOM_ON))
+ if (((LVEQNB_Mode_en)pParams->EQNB_OperatingMode == LVEQNB_ON)
+ && (pInstance->HeadroomParams.Headroom_OperatingMode == LVM_HEADROOM_ON))
{
/* Find typical headroom value */
for(jj = 0; jj < pInstance->HeadroomParams.NHeadroomBands; jj++)
@@ -545,7 +492,6 @@
}
-
/****************************************************************************************/
/* */
/* FUNCTION: LVM_ApplyNewSettings */
@@ -570,7 +516,6 @@
LVM_ControlParams_t LocalParams;
LVM_INT16 Count = 5;
-
/*
* Copy the new parameters but make sure they didn't change while copying
*/
@@ -627,13 +572,8 @@
/* Configure Mixer module for gradual changes to volume*/
if(LocalParams.VC_Balance < 0)
{
-#ifdef BUILD_FLOAT
LVM_FLOAT Target_Float;
-#else
- LVM_INT32 Target;
-#endif
/* Drop in right channel volume*/
-#ifdef BUILD_FLOAT
Target_Float = LVM_MAXFLOAT;
LVC_Mixer_SetTarget(&pInstance->VC_BalanceMix.MixerStream[0], Target_Float);
LVC_Mixer_VarSlope_SetTimeConstant(&pInstance->VC_BalanceMix.MixerStream[0],
@@ -643,25 +583,11 @@
LVC_Mixer_SetTarget(&pInstance->VC_BalanceMix.MixerStream[1], Target_Float);
LVC_Mixer_VarSlope_SetTimeConstant(&pInstance->VC_BalanceMix.MixerStream[1],
LVM_VC_MIXER_TIME, LocalParams.SampleRate, 1);
-#else
- Target = LVM_MAXINT_16;
- LVC_Mixer_SetTarget(&pInstance->VC_BalanceMix.MixerStream[0],Target);
- LVC_Mixer_VarSlope_SetTimeConstant(&pInstance->VC_BalanceMix.MixerStream[0],LVM_VC_MIXER_TIME,LocalParams.SampleRate,1);
-
- Target = dB_to_Lin32((LVM_INT16)(LocalParams.VC_Balance<<4));
- LVC_Mixer_SetTarget(&pInstance->VC_BalanceMix.MixerStream[1],Target);
- LVC_Mixer_VarSlope_SetTimeConstant(&pInstance->VC_BalanceMix.MixerStream[1],LVM_VC_MIXER_TIME,LocalParams.SampleRate,1);
-#endif
}
else if(LocalParams.VC_Balance >0)
{
-#ifdef BUILD_FLOAT
LVM_FLOAT Target_Float;
-#else
- LVM_INT32 Target;
-#endif
/* Drop in left channel volume*/
-#ifdef BUILD_FLOAT
Target_Float = dB_to_LinFloat((LVM_INT16)((-LocalParams.VC_Balance) << 4));
LVC_Mixer_SetTarget(&pInstance->VC_BalanceMix.MixerStream[0], Target_Float);
LVC_Mixer_VarSlope_SetTimeConstant(&pInstance->VC_BalanceMix.MixerStream[0],
@@ -671,30 +597,12 @@
LVC_Mixer_SetTarget(&pInstance->VC_BalanceMix.MixerStream[1], Target_Float);
LVC_Mixer_VarSlope_SetTimeConstant(&pInstance->VC_BalanceMix.MixerStream[1],
LVM_VC_MIXER_TIME, LocalParams.SampleRate, 1);
-#else
- Target = dB_to_Lin32((LVM_INT16)((-LocalParams.VC_Balance)<<4));
- LVC_Mixer_SetTarget(&pInstance->VC_BalanceMix.MixerStream[0],Target);
- LVC_Mixer_VarSlope_SetTimeConstant(&pInstance->VC_BalanceMix.MixerStream[0],LVM_VC_MIXER_TIME,LocalParams.SampleRate,1);
-
- Target = LVM_MAXINT_16;
- LVC_Mixer_SetTarget(&pInstance->VC_BalanceMix.MixerStream[1],Target);
- LVC_Mixer_VarSlope_SetTimeConstant(&pInstance->VC_BalanceMix.MixerStream[1],LVM_VC_MIXER_TIME,LocalParams.SampleRate,1);
-#endif
}
else
{
-#ifdef BUILD_FLOAT
LVM_FLOAT Target_Float;
-#else
- LVM_INT32 Target;
-#endif
/* No drop*/
-#ifdef BUILD_FLOAT
Target_Float = LVM_MAXFLOAT;
-#else
- Target = LVM_MAXINT_16;
-#endif
-#ifdef BUILD_FLOAT
LVC_Mixer_SetTarget(&pInstance->VC_BalanceMix.MixerStream[0],Target_Float);
LVC_Mixer_VarSlope_SetTimeConstant(&pInstance->VC_BalanceMix.MixerStream[0],
LVM_VC_MIXER_TIME,LocalParams.SampleRate, 1);
@@ -702,13 +610,6 @@
LVC_Mixer_SetTarget(&pInstance->VC_BalanceMix.MixerStream[1],Target_Float);
LVC_Mixer_VarSlope_SetTimeConstant(&pInstance->VC_BalanceMix.MixerStream[1],
LVM_VC_MIXER_TIME,LocalParams.SampleRate, 1);
-#else
- LVC_Mixer_SetTarget(&pInstance->VC_BalanceMix.MixerStream[0],Target);
- LVC_Mixer_VarSlope_SetTimeConstant(&pInstance->VC_BalanceMix.MixerStream[0],LVM_VC_MIXER_TIME,LocalParams.SampleRate,1);
-
- LVC_Mixer_SetTarget(&pInstance->VC_BalanceMix.MixerStream[1],Target);
- LVC_Mixer_VarSlope_SetTimeConstant(&pInstance->VC_BalanceMix.MixerStream[1],LVM_VC_MIXER_TIME,LocalParams.SampleRate,1);
-#endif
}
}
/*
@@ -717,8 +618,7 @@
{
LVDBE_ReturnStatus_en DBE_Status;
LVDBE_Params_t DBE_Params;
- LVDBE_Handle_t *hDBEInstance = pInstance->hDBEInstance;
-
+ LVDBE_Handle_t *hDBEInstance = (LVDBE_Handle_t *)pInstance->hDBEInstance;
/*
* Set the new parameters
@@ -748,7 +648,6 @@
DBE_Status = LVDBE_Control(hDBEInstance,
&DBE_Params);
-
/*
* Quit if the changes were not accepted
*/
@@ -757,7 +656,6 @@
return((LVM_ReturnStatus_en)DBE_Status);
}
-
/*
* Set the control flag
*/
@@ -770,8 +668,7 @@
{
LVEQNB_ReturnStatus_en EQNB_Status;
LVEQNB_Params_t EQNB_Params;
- LVEQNB_Handle_t *hEQNBInstance = pInstance->hEQNBInstance;
-
+ LVEQNB_Handle_t *hEQNBInstance = (LVEQNB_Handle_t *)pInstance->hEQNBInstance;
/*
* Set the new parameters
@@ -829,7 +726,6 @@
EQNB_Status = LVEQNB_Control(hEQNBInstance,
&EQNB_Params);
-
/*
* Quit if the changes were not accepted
*/
@@ -840,14 +736,13 @@
}
-
/*
* Update concert sound
*/
{
LVCS_ReturnStatus_en CS_Status;
LVCS_Params_t CS_Params;
- LVCS_Handle_t *hCSInstance = pInstance->hCSInstance;
+ LVCS_Handle_t *hCSInstance = (LVCS_Handle_t *)pInstance->hCSInstance;
LVM_Mode_en CompressorMode=LVM_MODE_ON;
/*
@@ -898,8 +793,8 @@
/*
* Set the control flag
*/
- if ((LocalParams.OperatingMode == LVM_MODE_ON) &&
- (LocalParams.VirtualizerOperatingMode != LVCS_OFF))
+ if (((LVM_Mode_en)LocalParams.OperatingMode == LVM_MODE_ON) &&
+ ((LVCS_Modes_en)LocalParams.VirtualizerOperatingMode != LVCS_OFF))
{
pInstance->CS_Active = LVM_TRUE;
}
@@ -916,7 +811,6 @@
CS_Status = LVCS_Control(hCSInstance,
&CS_Params);
-
/*
* Quit if the changes were not accepted
*/
@@ -933,8 +827,7 @@
{
LVPSA_RETURN PSA_Status;
LVPSA_ControlParams_t PSA_Params;
- pLVPSA_Handle_t *hPSAInstance = pInstance->hPSAInstance;
-
+ pLVPSA_Handle_t *hPSAInstance = (pLVPSA_Handle_t *)pInstance->hPSAInstance;
/*
* Set the new parameters
@@ -972,11 +865,9 @@
pInstance->NoSmoothVolume = LVM_FALSE;
pInstance->Params = LocalParams;
-
return(LVM_SUCCESS);
}
-
/****************************************************************************************/
/* */
/* FUNCTION: LVM_SetHeadroomParams */
@@ -1070,14 +961,12 @@
pHeadroomParams->NHeadroomBands = pInstance->NewHeadroomParams.NHeadroomBands;
-
/* Copy settings in memory */
for(ii = 0; ii < pInstance->NewHeadroomParams.NHeadroomBands; ii++)
{
pInstance->pHeadroom_UserDefs[ii] = pInstance->pHeadroom_BandDefs[ii];
}
-
pHeadroomParams->pHeadroomDefinition = pInstance->pHeadroom_UserDefs;
pHeadroomParams->Headroom_OperatingMode = pInstance->NewHeadroomParams.Headroom_OperatingMode;
return(LVM_SUCCESS);
@@ -1156,30 +1045,17 @@
short CallBackParam)
{
LVM_Instance_t *pInstance =(LVM_Instance_t *)pBundleHandle;
-#ifdef BUILD_FLOAT
LVM_FLOAT Target;
-#else
- LVM_INT32 Target;
-#endif
(void) pGeneralPurpose;
(void) CallBackParam;
/* When volume mixer has reached 0 dB target then stop it to avoid
unnecessary processing. */
-#ifdef BUILD_FLOAT
Target = LVC_Mixer_GetTarget(&pInstance->VC_Volume.MixerStream[0]);
if(Target == 1.0f)
{
pInstance->VC_Active = LVM_FALSE;
}
-#else
- Target = LVC_Mixer_GetTarget(&pInstance->VC_Volume.MixerStream[0]);
-
- if(Target == 0x7FFF)
- {
- pInstance->VC_Active = LVM_FALSE;
- }
-#endif
return 1;
}
diff --git a/media/libeffects/lvm/lib/Bundle/src/LVM_Init.c b/media/libeffects/lvm/lib/Bundle/src/LVM_Init.cpp
similarity index 91%
rename from media/libeffects/lvm/lib/Bundle/src/LVM_Init.c
rename to media/libeffects/lvm/lib/Bundle/src/LVM_Init.cpp
index c57498e..5620529 100644
--- a/media/libeffects/lvm/lib/Bundle/src/LVM_Init.c
+++ b/media/libeffects/lvm/lib/Bundle/src/LVM_Init.cpp
@@ -139,7 +139,6 @@
INST_ALLOC AllocMem[LVM_NR_MEMORY_REGIONS];
LVM_INT16 i;
-
/*
* Check parameters
*/
@@ -148,7 +147,6 @@
return LVM_NULLADDRESS;
}
-
/*
* Return memory table if the instance has already been created
*/
@@ -227,20 +225,15 @@
InstAlloc_AddMember(&AllocMem[LVM_MEMREGION_PERSISTENT_SLOW_DATA],
sizeof(LVM_Instance_t));
-
/*
* Set the algorithm and bundle scratch requirements
*/
AlgScratchSize = 0;
if (pInstParams->BufferMode == LVM_MANAGED_BUFFERS)
{
-#ifdef BUILD_FLOAT
BundleScratchSize = 3 * LVM_MAX_CHANNELS \
* (MIN_INTERNAL_BLOCKSIZE + InternalBlockSize) \
* sizeof(LVM_FLOAT);
-#else
- BundleScratchSize = 6 * (MIN_INTERNAL_BLOCKSIZE + InternalBlockSize) * sizeof(LVM_INT16);
-#endif
InstAlloc_AddMember(&AllocMem[LVM_MEMREGION_TEMPORARY_FAST], /* Scratch buffer */
BundleScratchSize);
InstAlloc_AddMember(&AllocMem[LVM_MEMREGION_PERSISTENT_SLOW_DATA],
@@ -293,7 +286,6 @@
}
-
/*
* Dynamic Bass Enhancement requirements
*/
@@ -304,7 +296,6 @@
/*
* Set the capabilities
*/
-#if defined(BUILD_FLOAT) && defined(HIGHER_FS)
DBE_Capabilities.SampleRate = LVDBE_CAP_FS_8000 | LVDBE_CAP_FS_11025 |
LVDBE_CAP_FS_12000 | LVDBE_CAP_FS_16000 |
LVDBE_CAP_FS_22050 | LVDBE_CAP_FS_24000 |
@@ -312,9 +303,6 @@
LVDBE_CAP_FS_48000 | LVDBE_CAP_FS_88200 |
LVDBE_CAP_FS_96000 | LVDBE_CAP_FS_176400 |
LVDBE_CAP_FS_192000;
-#else
- DBE_Capabilities.SampleRate = LVDBE_CAP_FS_8000 | LVDBE_CAP_FS_11025 | LVDBE_CAP_FS_12000 | LVDBE_CAP_FS_16000 | LVDBE_CAP_FS_22050 | LVDBE_CAP_FS_24000 | LVDBE_CAP_FS_32000 | LVDBE_CAP_FS_44100 | LVDBE_CAP_FS_48000;
-#endif
DBE_Capabilities.CentreFrequency = LVDBE_CAP_CENTRE_55Hz | LVDBE_CAP_CENTRE_55Hz | LVDBE_CAP_CENTRE_66Hz | LVDBE_CAP_CENTRE_78Hz | LVDBE_CAP_CENTRE_90Hz;
DBE_Capabilities.MaxBlockSize = InternalBlockSize;
@@ -336,7 +324,6 @@
}
-
/*
* N-Band equaliser requirements
*/
@@ -347,7 +334,6 @@
/*
* Set the capabilities
*/
-#if defined(BUILD_FLOAT) && defined(HIGHER_FS)
EQNB_Capabilities.SampleRate = LVEQNB_CAP_FS_8000 | LVEQNB_CAP_FS_11025 |
LVEQNB_CAP_FS_12000 | LVEQNB_CAP_FS_16000 |
LVEQNB_CAP_FS_22050 | LVEQNB_CAP_FS_24000 |
@@ -355,9 +341,6 @@
LVEQNB_CAP_FS_48000 | LVEQNB_CAP_FS_88200 |
LVEQNB_CAP_FS_96000 | LVEQNB_CAP_FS_176400 |
LVEQNB_CAP_FS_192000;
-#else
- EQNB_Capabilities.SampleRate = LVEQNB_CAP_FS_8000 | LVEQNB_CAP_FS_11025 | LVEQNB_CAP_FS_12000 | LVEQNB_CAP_FS_16000 | LVEQNB_CAP_FS_22050 | LVEQNB_CAP_FS_24000 | LVEQNB_CAP_FS_32000 | LVEQNB_CAP_FS_44100 | LVEQNB_CAP_FS_48000;
-#endif
EQNB_Capabilities.SourceFormat = LVEQNB_CAP_STEREO | LVEQNB_CAP_MONOINSTEREO;
EQNB_Capabilities.MaxBlockSize = InternalBlockSize;
EQNB_Capabilities.MaxBands = pInstParams->EQNB_NumBands;
@@ -388,7 +371,6 @@
InstAlloc_AddMember(&AllocMem[LVM_MEMREGION_PERSISTENT_FAST_DATA],
(LVM_HEADROOM_MAX_NBANDS * sizeof(LVM_HeadroomBandDef_t)));
-
/*
* Spectrum Analyzer memory requirements
*/
@@ -441,13 +423,8 @@
PSA_MemTab.Region[LVM_PERSISTENT_FAST_COEF].Size);
/* Fast Temporary */
-#ifdef BUILD_FLOAT
InstAlloc_AddMember(&AllocMem[LVM_TEMPORARY_FAST],
MAX_INTERNAL_BLOCKSIZE * sizeof(LVM_FLOAT));
-#else
- InstAlloc_AddMember(&AllocMem[LVM_TEMPORARY_FAST],
- MAX_INTERNAL_BLOCKSIZE * sizeof(LVM_INT16));
-#endif
if (PSA_MemTab.Region[LVM_TEMPORARY_FAST].Size > AlgScratchSize)
{
@@ -493,7 +470,6 @@
}
-
/****************************************************************************************/
/* */
/* FUNCTION: LVM_GetInstanceHandle */
@@ -529,7 +505,6 @@
LVM_UINT16 InternalBlockSize;
LVM_INT32 BundleScratchSize;
-
/*
* Check valid points have been given
*/
@@ -592,7 +567,6 @@
pMemoryTable->Region[i].pBaseAddress);
}
-
/*
* Set the instance handle
*/
@@ -600,14 +574,12 @@
sizeof(LVM_Instance_t));
pInstance =(LVM_Instance_t *)*phInstance;
-
/*
* Save the memory table, parameters and capabilities
*/
pInstance->MemoryTable = *pMemoryTable;
pInstance->InstParams = *pInstParams;
-
/*
* Set the bundle scratch memory and initialse the buffer management
*/
@@ -624,7 +596,6 @@
}
pInstance->InternalBlockSize = (LVM_INT16)InternalBlockSize;
-
/*
* Common settings for managed and unmanaged buffers
*/
@@ -634,33 +605,25 @@
/*
* Managed buffers required
*/
- pInstance->pBufferManagement = InstAlloc_AddMember(&AllocMem[LVM_MEMREGION_PERSISTENT_SLOW_DATA],
+ pInstance->pBufferManagement = (LVM_Buffer_t *)
+ InstAlloc_AddMember(&AllocMem[LVM_MEMREGION_PERSISTENT_SLOW_DATA],
sizeof(LVM_Buffer_t));
-#ifdef BUILD_FLOAT
BundleScratchSize = (LVM_INT32)
(3 * LVM_MAX_CHANNELS \
* (MIN_INTERNAL_BLOCKSIZE + InternalBlockSize) \
* sizeof(LVM_FLOAT));
-#else
- BundleScratchSize = (LVM_INT32)(6 * (MIN_INTERNAL_BLOCKSIZE + InternalBlockSize) * sizeof(LVM_INT16));
-#endif
- pInstance->pBufferManagement->pScratch = InstAlloc_AddMember(&AllocMem[LVM_MEMREGION_TEMPORARY_FAST], /* Scratch 1 buffer */
- (LVM_UINT32)BundleScratchSize);
-#ifdef BUILD_FLOAT
+ pInstance->pBufferManagement->pScratch = (LVM_FLOAT *)
+ InstAlloc_AddMember(
+ &AllocMem[LVM_MEMREGION_TEMPORARY_FAST], /* Scratch 1 buffer */
+ (LVM_UINT32)BundleScratchSize);
LoadConst_Float(0, /* Clear the input delay buffer */
(LVM_FLOAT *)&pInstance->pBufferManagement->InDelayBuffer,
(LVM_INT16)(LVM_MAX_CHANNELS * MIN_INTERNAL_BLOCKSIZE));
-#else
- LoadConst_16(0, /* Clear the input delay buffer */
- (LVM_INT16 *)&pInstance->pBufferManagement->InDelayBuffer,
- (LVM_INT16)(2 * MIN_INTERNAL_BLOCKSIZE));
-#endif
pInstance->pBufferManagement->InDelaySamples = MIN_INTERNAL_BLOCKSIZE; /* Set the number of delay samples */
pInstance->pBufferManagement->OutDelaySamples = 0; /* No samples in the output buffer */
pInstance->pBufferManagement->BufferState = LVM_FIRSTCALL; /* Set the state ready for the first call */
}
-
/*
* Set default parameters
*/
@@ -676,7 +639,6 @@
*/
pInstance->CallBack = LVM_AlgoCallBack;
-
/*
* DC removal filter
*/
@@ -698,7 +660,6 @@
pInstance->Params.TE_EffectLevel = 0;
pInstance->TE_Active = LVM_FALSE;
-
/*
* Set the volume control and initialise Current to Target
*/
@@ -710,26 +671,14 @@
/* In managed buffering, start with low signal level as delay in buffer management causes a click*/
if (pInstParams->BufferMode == LVM_MANAGED_BUFFERS)
{
-#ifdef BUILD_FLOAT
LVC_Mixer_Init(&pInstance->VC_Volume.MixerStream[0], 0, 0);
-#else
- LVC_Mixer_Init(&pInstance->VC_Volume.MixerStream[0],0,0);
-#endif
}
else
{
-#ifdef BUILD_FLOAT
LVC_Mixer_Init(&pInstance->VC_Volume.MixerStream[0], LVM_MAXFLOAT, LVM_MAXFLOAT);
-#else
- LVC_Mixer_Init(&pInstance->VC_Volume.MixerStream[0],LVM_MAXINT_16,LVM_MAXINT_16);
-#endif
}
-#ifdef BUILD_FLOAT
LVC_Mixer_SetTimeConstant(&pInstance->VC_Volume.MixerStream[0],0,LVM_FS_8000,2);
-#else
- LVC_Mixer_SetTimeConstant(&pInstance->VC_Volume.MixerStream[0], 0, LVM_FS_8000, 2);
-#endif
pInstance->VC_VolumedB = 0;
pInstance->VC_AVLFixedVolume = 0;
@@ -739,32 +688,25 @@
pInstance->VC_BalanceMix.MixerStream[0].CallbackSet = 0;
pInstance->VC_BalanceMix.MixerStream[0].pCallbackHandle = pInstance;
pInstance->VC_BalanceMix.MixerStream[0].pCallBack = LVM_VCCallBack;
-#ifdef BUILD_FLOAT
LVC_Mixer_Init(&pInstance->VC_BalanceMix.MixerStream[0], LVM_MAXFLOAT, LVM_MAXFLOAT);
-#else
- LVC_Mixer_Init(&pInstance->VC_BalanceMix.MixerStream[0],LVM_MAXINT_16,LVM_MAXINT_16);
-#endif
LVC_Mixer_VarSlope_SetTimeConstant(&pInstance->VC_BalanceMix.MixerStream[0],LVM_VC_MIXER_TIME,LVM_FS_8000,2);
pInstance->VC_BalanceMix.MixerStream[1].CallbackParam = 0;
pInstance->VC_BalanceMix.MixerStream[1].CallbackSet = 0;
pInstance->VC_BalanceMix.MixerStream[1].pCallbackHandle = pInstance;
pInstance->VC_BalanceMix.MixerStream[1].pCallBack = LVM_VCCallBack;
-#ifdef BUILD_FLOAT
LVC_Mixer_Init(&pInstance->VC_BalanceMix.MixerStream[1], LVM_MAXFLOAT, LVM_MAXFLOAT);
-#else
- LVC_Mixer_Init(&pInstance->VC_BalanceMix.MixerStream[1],LVM_MAXINT_16,LVM_MAXINT_16);
-#endif
LVC_Mixer_VarSlope_SetTimeConstant(&pInstance->VC_BalanceMix.MixerStream[1],LVM_VC_MIXER_TIME,LVM_FS_8000,2);
/*
* Set the default EQNB pre-gain and pointer to the band definitions
*/
- pInstance->pEQNB_BandDefs = InstAlloc_AddMember(&AllocMem[LVM_MEMREGION_PERSISTENT_FAST_DATA],
- (pInstParams->EQNB_NumBands * sizeof(LVM_EQNB_BandDef_t)));
- pInstance->pEQNB_UserDefs = InstAlloc_AddMember(&AllocMem[LVM_MEMREGION_PERSISTENT_FAST_DATA],
- (pInstParams->EQNB_NumBands * sizeof(LVM_EQNB_BandDef_t)));
-
+ pInstance->pEQNB_BandDefs =
+ (LVM_EQNB_BandDef_t *)InstAlloc_AddMember(&AllocMem[LVM_MEMREGION_PERSISTENT_FAST_DATA],
+ (pInstParams->EQNB_NumBands * sizeof(LVM_EQNB_BandDef_t)));
+ pInstance->pEQNB_UserDefs =
+ (LVM_EQNB_BandDef_t *)InstAlloc_AddMember(&AllocMem[LVM_MEMREGION_PERSISTENT_FAST_DATA],
+ (pInstParams->EQNB_NumBands * sizeof(LVM_EQNB_BandDef_t)));
/*
* Initialise the Concert Sound module
@@ -790,7 +732,6 @@
CS_Capabilities.CallBack = pInstance->CallBack;
CS_Capabilities.pBundleInstance = (void*)pInstance;
-
/*
* Get the memory requirements and then set the address pointers, forcing alignment
*/
@@ -826,7 +767,6 @@
LVDBE_Capabilities_t DBE_Capabilities; /* Initial capabilities */
LVDBE_ReturnStatus_en LVDBE_Status; /* Function call status */
-
/*
* Set the initialisation parameters
*/
@@ -837,12 +777,9 @@
pInstance->DBE_Active = LVM_FALSE;
-
-
/*
* Set the initialisation capabilities
*/
-#if defined(BUILD_FLOAT) && defined(HIGHER_FS)
DBE_Capabilities.SampleRate = LVDBE_CAP_FS_8000 | LVDBE_CAP_FS_11025 |
LVDBE_CAP_FS_12000 | LVDBE_CAP_FS_16000 |
LVDBE_CAP_FS_22050 | LVDBE_CAP_FS_24000 |
@@ -850,13 +787,9 @@
LVDBE_CAP_FS_48000 | LVDBE_CAP_FS_88200 |
LVDBE_CAP_FS_96000 | LVDBE_CAP_FS_176400 |
LVDBE_CAP_FS_192000;
-#else
- DBE_Capabilities.SampleRate = LVDBE_CAP_FS_8000 | LVDBE_CAP_FS_11025 | LVDBE_CAP_FS_12000 | LVDBE_CAP_FS_16000 | LVDBE_CAP_FS_22050 | LVDBE_CAP_FS_24000 | LVDBE_CAP_FS_32000 | LVDBE_CAP_FS_44100 | LVDBE_CAP_FS_48000;
-#endif
DBE_Capabilities.CentreFrequency = LVDBE_CAP_CENTRE_55Hz | LVDBE_CAP_CENTRE_55Hz | LVDBE_CAP_CENTRE_66Hz | LVDBE_CAP_CENTRE_78Hz | LVDBE_CAP_CENTRE_90Hz;
DBE_Capabilities.MaxBlockSize = (LVM_UINT16)InternalBlockSize;
-
/*
* Get the memory requirements and then set the address pointers
*/
@@ -871,7 +804,6 @@
DBE_MemTab.Region[LVDBE_MEMREGION_SCRATCH].pBaseAddress = (void *)InstAlloc_AddMember(&AllocMem[LVM_MEMREGION_TEMPORARY_FAST],
0);
-
/*
* Initialise the Dynamic Bass Enhancement instance and save the instance handle
*/
@@ -883,7 +815,6 @@
pInstance->hDBEInstance = hDBEInstance; /* Save the instance handle */
}
-
/*
* Initialise the N-Band Equaliser module
*/
@@ -893,7 +824,6 @@
LVEQNB_Capabilities_t EQNB_Capabilities; /* Initial capabilities */
LVEQNB_ReturnStatus_en LVEQNB_Status; /* Function call status */
-
/*
* Set the initialisation parameters
*/
@@ -902,11 +832,9 @@
pInstance->Params.pEQNB_BandDefinition = LVM_NULL;
pInstance->EQNB_Active = LVM_FALSE;
-
/*
* Set the initialisation capabilities
*/
-#if defined(BUILD_FLOAT) && defined(HIGHER_FS)
EQNB_Capabilities.SampleRate = LVEQNB_CAP_FS_8000 | LVEQNB_CAP_FS_11025 |
LVEQNB_CAP_FS_12000 | LVEQNB_CAP_FS_16000 |
LVEQNB_CAP_FS_22050 | LVEQNB_CAP_FS_24000 |
@@ -914,16 +842,12 @@
LVEQNB_CAP_FS_48000 | LVEQNB_CAP_FS_88200 |
LVEQNB_CAP_FS_96000 | LVEQNB_CAP_FS_176400 |
LVEQNB_CAP_FS_192000;
-#else
- EQNB_Capabilities.SampleRate = LVEQNB_CAP_FS_8000 | LVEQNB_CAP_FS_11025 | LVEQNB_CAP_FS_12000 | LVEQNB_CAP_FS_16000 | LVEQNB_CAP_FS_22050 | LVEQNB_CAP_FS_24000 | LVEQNB_CAP_FS_32000 | LVEQNB_CAP_FS_44100 | LVEQNB_CAP_FS_48000;
-#endif
EQNB_Capabilities.MaxBlockSize = (LVM_UINT16)InternalBlockSize;
EQNB_Capabilities.MaxBands = pInstParams->EQNB_NumBands;
EQNB_Capabilities.SourceFormat = LVEQNB_CAP_STEREO | LVEQNB_CAP_MONOINSTEREO;
EQNB_Capabilities.CallBack = pInstance->CallBack;
EQNB_Capabilities.pBundleInstance = (void*)pInstance;
-
/*
* Get the memory requirements and then set the address pointers, forcing alignment
*/
@@ -938,7 +862,6 @@
EQNB_MemTab.Region[LVEQNB_MEMREGION_SCRATCH].pBaseAddress = (void *)InstAlloc_AddMember(&AllocMem[LVM_MEMREGION_TEMPORARY_FAST],
0);
-
/*
* Initialise the Dynamic Bass Enhancement instance and save the instance handle
*/
@@ -954,10 +877,12 @@
* Headroom management memory allocation
*/
{
- pInstance->pHeadroom_BandDefs = InstAlloc_AddMember(&AllocMem[LVM_MEMREGION_PERSISTENT_FAST_DATA],
- (LVM_HEADROOM_MAX_NBANDS * sizeof(LVM_HeadroomBandDef_t)));
- pInstance->pHeadroom_UserDefs = InstAlloc_AddMember(&AllocMem[LVM_MEMREGION_PERSISTENT_FAST_DATA],
- (LVM_HEADROOM_MAX_NBANDS * sizeof(LVM_HeadroomBandDef_t)));
+ pInstance->pHeadroom_BandDefs = (LVM_HeadroomBandDef_t *)
+ InstAlloc_AddMember(&AllocMem[LVM_MEMREGION_PERSISTENT_FAST_DATA],
+ (LVM_HEADROOM_MAX_NBANDS * sizeof(LVM_HeadroomBandDef_t)));
+ pInstance->pHeadroom_UserDefs = (LVM_HeadroomBandDef_t *)
+ InstAlloc_AddMember(&AllocMem[LVM_MEMREGION_PERSISTENT_FAST_DATA],
+ (LVM_HEADROOM_MAX_NBANDS * sizeof(LVM_HeadroomBandDef_t)));
/* Headroom management parameters initialisation */
pInstance->NewHeadroomParams.NHeadroomBands = 2;
@@ -973,7 +898,6 @@
pInstance->Headroom =0;
}
-
/*
* Initialise the PSA module
*/
@@ -1010,28 +934,20 @@
PSA_MemTab.Region[LVM_PERSISTENT_SLOW_DATA].pBaseAddress = (void *)InstAlloc_AddMember(&AllocMem[LVM_MEMREGION_PERSISTENT_SLOW_DATA],
PSA_MemTab.Region[LVM_PERSISTENT_SLOW_DATA].Size);
-
/* Fast Data */
PSA_MemTab.Region[LVM_PERSISTENT_FAST_DATA].pBaseAddress = (void *)InstAlloc_AddMember(&AllocMem[LVM_MEMREGION_PERSISTENT_FAST_DATA],
PSA_MemTab.Region[LVM_PERSISTENT_FAST_DATA].Size);
-
/* Fast Coef */
PSA_MemTab.Region[LVM_PERSISTENT_FAST_COEF].pBaseAddress = (void *)InstAlloc_AddMember(&AllocMem[LVM_MEMREGION_PERSISTENT_FAST_COEF],
PSA_MemTab.Region[LVM_PERSISTENT_FAST_COEF].Size);
/* Fast Temporary */
-#ifdef BUILD_FLOAT
- pInstance->pPSAInput = InstAlloc_AddMember(&AllocMem[LVM_TEMPORARY_FAST],
+ pInstance->pPSAInput = (LVM_FLOAT *)InstAlloc_AddMember(&AllocMem[LVM_TEMPORARY_FAST],
(LVM_UINT32) MAX_INTERNAL_BLOCKSIZE * \
sizeof(LVM_FLOAT));
-#else
- pInstance->pPSAInput = InstAlloc_AddMember(&AllocMem[LVM_TEMPORARY_FAST],
- (LVM_UINT32) MAX_INTERNAL_BLOCKSIZE * sizeof(LVM_INT16));
-#endif
PSA_MemTab.Region[LVM_TEMPORARY_FAST].pBaseAddress = (void *)InstAlloc_AddMember(&AllocMem[LVM_MEMREGION_TEMPORARY_FAST],0);
-
/*Initialise PSA instance and save the instance handle*/
pInstance->PSA_ControlParams.Fs = LVM_FS_48000;
pInstance->PSA_ControlParams.LevelDetectionSpeed = LVPSA_SPEED_MEDIUM;
@@ -1066,7 +982,6 @@
*/
pInstance->NewParams = pInstance->Params;
-
/*
* Create configuration number
*/
@@ -1093,7 +1008,6 @@
return(Status);
}
-
/****************************************************************************************/
/* */
/* FUNCTION: LVM_ClearAudioBuffers */
@@ -1121,7 +1035,6 @@
LVM_Instance_t *pInstance = (LVM_Instance_t *)hInstance; /* Pointer to Instance */
LVM_HeadroomParams_t HeadroomParams;
-
if(hInstance == LVM_NULL){
return LVM_NULLADDRESS;
}
@@ -1159,5 +1072,3 @@
return LVM_SUCCESS;
}
-
-
diff --git a/media/libeffects/lvm/lib/Bundle/src/LVM_Private.h b/media/libeffects/lvm/lib/Bundle/src/LVM_Private.h
index cdd3134..ddaac99 100644
--- a/media/libeffects/lvm/lib/Bundle/src/LVM_Private.h
+++ b/media/libeffects/lvm/lib/Bundle/src/LVM_Private.h
@@ -27,11 +27,6 @@
#ifndef __LVM_PRIVATE_H__
#define __LVM_PRIVATE_H__
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
-
/************************************************************************************/
/* */
/* Includes */
@@ -47,7 +42,6 @@
#include "LVEQNB_Private.h" /* N-Band equaliser */
#include "LVPSA_Private.h" /* Parametric Spectrum Analyzer */
-
/************************************************************************************/
/* */
/* Defines */
@@ -113,7 +107,6 @@
#define LVM_TE_MASK 32
#define LVM_PSA_MASK 2048
-
/************************************************************************************/
/* */
/* Structures */
@@ -129,16 +122,13 @@
void *pBaseAddress; /* Pointer to the region base address */
} LVM_IntMemoryRegion_t;
-
/* Memory table containing the region definitions */
typedef struct
{
LVM_IntMemoryRegion_t Region[LVM_NR_MEMORY_REGIONS]; /* One definition for each region */
} LVM_IntMemTab_t;
-
/* Buffer Management */
-#ifdef BUILD_FLOAT
typedef struct
{
LVM_FLOAT *pScratch; /* Bundle scratch buffer */
@@ -161,39 +151,17 @@
left and right */
LVM_INT16 SamplesToOutput; /* Samples to write to the output */
} LVM_Buffer_t;
-#else
-typedef struct
-{
- LVM_INT16 *pScratch; /* Bundle scratch buffer */
-
- LVM_INT16 BufferState; /* Buffer status */
- LVM_INT16 InDelayBuffer[6*MIN_INTERNAL_BLOCKSIZE]; /* Input buffer delay line, left and right */
- LVM_INT16 InDelaySamples; /* Number of samples in the input delay buffer */
-
- LVM_INT16 OutDelayBuffer[2*MIN_INTERNAL_BLOCKSIZE]; /* Output buffer delay line */
- LVM_INT16 OutDelaySamples; /* Number of samples in the output delay buffer, left and right */
- LVM_INT16 SamplesToOutput; /* Samples to write to the output */
-} LVM_Buffer_t;
-#endif
/* Filter taps */
typedef struct
{
-#ifdef BUILD_FLOAT
Biquad_2I_Order1_FLOAT_Taps_t TrebleBoost_Taps; /* Treble boost Taps */
-#else
- Biquad_2I_Order1_Taps_t TrebleBoost_Taps; /* Treble boost Taps */
-#endif
} LVM_TE_Data_t;
/* Coefficients */
typedef struct
{
-#ifdef BUILD_FLOAT
Biquad_FLOAT_Instance_t TrebleBoost_State; /* State for the treble boost filter */
-#else
- Biquad_Instance_t TrebleBoost_State; /* State for the treble boost filter */
-#endif
} LVM_TE_Coefs_t;
typedef struct
@@ -211,24 +179,15 @@
LVM_INT16 InternalBlockSize; /* Maximum internal block size */
LVM_Buffer_t *pBufferManagement; /* Buffer management variables */
LVM_INT16 SamplesToProcess; /* Input samples left to process */
-#ifdef BUILD_FLOAT
LVM_FLOAT *pInputSamples; /* External input sample pointer */
LVM_FLOAT *pOutputSamples; /* External output sample pointer */
-#else
- LVM_INT16 *pInputSamples; /* External input sample pointer */
- LVM_INT16 *pOutputSamples; /* External output sample pointer */
-#endif
/* Configuration number */
LVM_INT32 ConfigurationNumber;
LVM_INT32 BlickSizeMultiple;
/* DC removal */
-#ifdef BUILD_FLOAT
Biquad_FLOAT_Instance_t DC_RemovalInstance; /* DC removal filter instance */
-#else
- Biquad_Instance_t DC_RemovalInstance; /* DC removal filter instance */
-#endif
/* Concert Sound */
LVCS_Handle_t hCSInstance; /* Concert Sound instance handle */
@@ -248,16 +207,8 @@
LVM_INT16 DBE_Active; /* Control flag */
/* Volume Control */
-#ifdef BUILD_FLOAT
LVMixer3_1St_FLOAT_st VC_Volume; /* Volume scaler */
-#else
- LVMixer3_1St_st VC_Volume; /* Volume scaler */
-#endif
-#ifdef BUILD_FLOAT
LVMixer3_2St_FLOAT_st VC_BalanceMix; /* VC balance mixer */
-#else
- LVMixer3_2St_st VC_BalanceMix; /* VC balance mixer */
-#endif
LVM_INT16 VC_VolumedB; /* Gain in dB */
LVM_INT16 VC_Active; /* Control flag */
LVM_INT16 VC_AVLFixedVolume; /* AVL fixed volume */
@@ -281,11 +232,7 @@
LVPSA_ControlParams_t PSA_ControlParams; /* Spectrum Analyzer control parameters */
LVM_INT16 PSA_GainOffset; /* Tone control flag */
LVM_Callback CallBack;
-#ifdef BUILD_FLOAT
LVM_FLOAT *pPSAInput; /* PSA input pointer */
-#else
- LVM_INT16 *pPSAInput; /* PSA input pointer */
-#endif
LVM_INT16 NoSmoothVolume; /* Enable or disable smooth volume changes*/
@@ -296,7 +243,6 @@
} LVM_Instance_t;
-
/************************************************************************************/
/* */
/* Function Prototypes */
@@ -317,36 +263,18 @@
void LVM_SetHeadroom( LVM_Instance_t *pInstance,
LVM_ControlParams_t *pParams);
-#ifdef BUILD_FLOAT
void LVM_BufferIn( LVM_Handle_t hInstance,
const LVM_FLOAT *pInData,
LVM_FLOAT **pToProcess,
LVM_FLOAT **pProcessed,
LVM_UINT16 *pNumSamples);
-#else
-void LVM_BufferIn( LVM_Handle_t hInstance,
- const LVM_INT16 *pInData,
- LVM_INT16 **pToProcess,
- LVM_INT16 **pProcessed,
- LVM_UINT16 *pNumSamples);
-#endif
-#ifdef BUILD_FLOAT
void LVM_BufferOut( LVM_Handle_t hInstance,
LVM_FLOAT *pOutData,
LVM_UINT16 *pNumSamples);
-#else
-void LVM_BufferOut( LVM_Handle_t hInstance,
- LVM_INT16 *pOutData,
- LVM_UINT16 *pNumSamples);
-#endif
LVM_INT32 LVM_AlgoCallBack( void *pBundleHandle,
void *pData,
LVM_INT16 callbackId);
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
-
#endif /* __LVM_PRIVATE_H__ */
diff --git a/media/libeffects/lvm/lib/Bundle/src/LVM_Process.c b/media/libeffects/lvm/lib/Bundle/src/LVM_Process.cpp
similarity index 64%
rename from media/libeffects/lvm/lib/Bundle/src/LVM_Process.c
rename to media/libeffects/lvm/lib/Bundle/src/LVM_Process.cpp
index bc666a9..dc86cfd 100644
--- a/media/libeffects/lvm/lib/Bundle/src/LVM_Process.c
+++ b/media/libeffects/lvm/lib/Bundle/src/LVM_Process.cpp
@@ -15,7 +15,6 @@
* limitations under the License.
*/
-
/****************************************************************************************/
/* */
/* Includes */
@@ -52,7 +51,6 @@
/* NOTES: */
/* */
/****************************************************************************************/
-#ifdef BUILD_FLOAT
LVM_ReturnStatus_en LVM_Process(LVM_Handle_t hInstance,
const LVM_FLOAT *pInData,
LVM_FLOAT *pOutData,
@@ -80,7 +78,6 @@
return(LVM_SUCCESS);
}
-
/*
* Check valid points have been given
*/
@@ -111,7 +108,6 @@
}
}
-
/*
* Update new parameters if necessary
*/
@@ -130,7 +126,6 @@
}
}
-
/*
* Convert from Mono if necessary
*/
@@ -147,7 +142,6 @@
#endif
}
-
/*
* Process the data with managed buffers
*/
@@ -333,226 +327,3 @@
return(LVM_SUCCESS);
}
-#else
-LVM_ReturnStatus_en LVM_Process(LVM_Handle_t hInstance,
- const LVM_INT16 *pInData,
- LVM_INT16 *pOutData,
- LVM_UINT16 NumSamples,
- LVM_UINT32 AudioTime)
-{
-
- LVM_Instance_t *pInstance = (LVM_Instance_t *)hInstance;
- LVM_UINT16 SampleCount = NumSamples;
- LVM_INT16 *pInput = (LVM_INT16 *)pInData;
- LVM_INT16 *pToProcess = (LVM_INT16 *)pInData;
- LVM_INT16 *pProcessed = pOutData;
- LVM_ReturnStatus_en Status;
-
- /*
- * Check if the number of samples is zero
- */
- if (NumSamples == 0)
- {
- return(LVM_SUCCESS);
- }
-
-
- /*
- * Check valid points have been given
- */
- if ((hInstance == LVM_NULL) || (pInData == LVM_NULL) || (pOutData == LVM_NULL))
- {
- return (LVM_NULLADDRESS);
- }
-
- /*
- * For unmanaged mode only
- */
- if(pInstance->InstParams.BufferMode == LVM_UNMANAGED_BUFFERS)
- {
- /*
- * Check if the number of samples is a good multiple (unmanaged mode only)
- */
- if((NumSamples % pInstance->BlickSizeMultiple) != 0)
- {
- return(LVM_INVALIDNUMSAMPLES);
- }
-
- /*
- * Check the buffer alignment
- */
- if((((uintptr_t)pInData % 4) != 0) || (((uintptr_t)pOutData % 4) != 0))
- {
- return(LVM_ALIGNMENTERROR);
- }
- }
-
-
- /*
- * Update new parameters if necessary
- */
- if (pInstance->ControlPending == LVM_TRUE)
- {
- Status = LVM_ApplyNewSettings(hInstance);
-
- if(Status != LVM_SUCCESS)
- {
- return Status;
- }
- }
-
-
- /*
- * Convert from Mono if necessary
- */
- if (pInstance->Params.SourceFormat == LVM_MONO)
- {
- MonoTo2I_16(pInData, /* Source */
- pOutData, /* Destination */
- (LVM_INT16)NumSamples); /* Number of input samples */
- pInput = pOutData;
- pToProcess = pOutData;
- }
-
-
- /*
- * Process the data with managed buffers
- */
- while (SampleCount != 0)
- {
- /*
- * Manage the input buffer and frame processing
- */
- LVM_BufferIn(hInstance,
- pInput,
- &pToProcess,
- &pProcessed,
- &SampleCount);
-
- /*
- * Only process data when SampleCount is none zero, a zero count can occur when
- * the BufferIn routine is working in managed mode.
- */
- if (SampleCount != 0)
- {
-
- /*
- * Apply ConcertSound if required
- */
- if (pInstance->CS_Active == LVM_TRUE)
- {
- (void)LVCS_Process(pInstance->hCSInstance, /* Concert Sound instance handle */
- pToProcess,
- pProcessed,
- SampleCount);
- pToProcess = pProcessed;
- }
-
- /*
- * Apply volume if required
- */
- if (pInstance->VC_Active!=0)
- {
- LVC_MixSoft_1St_D16C31_SAT(&pInstance->VC_Volume,
- pToProcess,
- pProcessed,
- (LVM_INT16)(2*SampleCount)); /* Left and right*/
- pToProcess = pProcessed;
- }
-
- /*
- * Call N-Band equaliser if enabled
- */
- if (pInstance->EQNB_Active == LVM_TRUE)
- {
- LVEQNB_Process(pInstance->hEQNBInstance, /* N-Band equaliser instance handle */
- pToProcess,
- pProcessed,
- SampleCount);
- pToProcess = pProcessed;
- }
-
- /*
- * Call bass enhancement if enabled
- */
- if (pInstance->DBE_Active == LVM_TRUE)
- {
- LVDBE_Process(pInstance->hDBEInstance, /* Dynamic Bass Enhancement instance handle */
- pToProcess,
- pProcessed,
- SampleCount);
- pToProcess = pProcessed;
- }
-
- /*
- * Bypass mode or everything off, so copy the input to the output
- */
- if (pToProcess != pProcessed)
- {
- Copy_16(pToProcess, /* Source */
- pProcessed, /* Destination */
- (LVM_INT16)(2*SampleCount)); /* Left and right */
- }
-
- /*
- * Apply treble boost if required
- */
- if (pInstance->TE_Active == LVM_TRUE)
- {
- /*
- * Apply the filter
- */
- FO_2I_D16F32C15_LShx_TRC_WRA_01(&pInstance->pTE_State->TrebleBoost_State,
- pProcessed,
- pProcessed,
- (LVM_INT16)SampleCount);
-
- }
-
- /*
- * Volume balance
- */
- LVC_MixSoft_1St_2i_D16C31_SAT(&pInstance->VC_BalanceMix,
- pProcessed,
- pProcessed,
- SampleCount);
-
- /*
- * Perform Parametric Spectum Analysis
- */
- if ((pInstance->Params.PSA_Enable == LVM_PSA_ON)&&(pInstance->InstParams.PSA_Included==LVM_PSA_ON))
- {
- From2iToMono_16(pProcessed,
- pInstance->pPSAInput,
- (LVM_INT16) (SampleCount));
-
- LVPSA_Process(pInstance->hPSAInstance,
- pInstance->pPSAInput,
- (LVM_UINT16) (SampleCount),
- AudioTime);
- }
-
-
- /*
- * DC removal
- */
- DC_2I_D16_TRC_WRA_01(&pInstance->DC_RemovalInstance,
- pProcessed,
- pProcessed,
- (LVM_INT16)SampleCount);
-
-
- }
-
- /*
- * Manage the output buffer
- */
- LVM_BufferOut(hInstance,
- pOutData,
- &SampleCount);
-
- }
-
- return(LVM_SUCCESS);
-}
-#endif
diff --git a/media/libeffects/lvm/lib/Bundle/src/LVM_Tables.c b/media/libeffects/lvm/lib/Bundle/src/LVM_Tables.cpp
similarity index 61%
rename from media/libeffects/lvm/lib/Bundle/src/LVM_Tables.c
rename to media/libeffects/lvm/lib/Bundle/src/LVM_Tables.cpp
index a5356d2..66392e2 100644
--- a/media/libeffects/lvm/lib/Bundle/src/LVM_Tables.c
+++ b/media/libeffects/lvm/lib/Bundle/src/LVM_Tables.cpp
@@ -29,7 +29,6 @@
/* Treble Boost Filter Coefficients */
/* */
/************************************************************************************/
-#ifdef BUILD_FLOAT
FO_FLOAT_LShx_Coefs_t LVM_TrebleBoostCoefs[] = {
@@ -267,7 +266,6 @@
{HPF_Fs48000_Gain15_A1, /* Gain setting 15 */
HPF_Fs48000_Gain15_A0,
-HPF_Fs48000_Gain15_B1}
-#ifdef HIGHER_FS
,
/* 88kHz Sampling rate */
{HPF_Fs88200_Gain1_A1, /* Gain Setting 1 */
@@ -456,322 +454,7 @@
{HPF_Fs192000_Gain15_A1, /* Gain setting 15 */
HPF_Fs192000_Gain15_A0,
-HPF_Fs192000_Gain15_B1}
-#endif
};
-#else
-FO_C16_LShx_Coefs_t LVM_TrebleBoostCoefs[] = {
-
- /* 22kHz sampling rate */
- {HPF_Fs22050_Gain1_A1, /* Gain setting 1 */
- HPF_Fs22050_Gain1_A0,
- -HPF_Fs22050_Gain1_B1,
- HPF_Fs22050_Gain1_Shift},
- {HPF_Fs22050_Gain2_A1, /* Gain setting 2 */
- HPF_Fs22050_Gain2_A0,
- -HPF_Fs22050_Gain2_B1,
- HPF_Fs22050_Gain2_Shift},
- {HPF_Fs22050_Gain3_A1, /* Gain setting 3 */
- HPF_Fs22050_Gain3_A0,
- -HPF_Fs22050_Gain3_B1,
- HPF_Fs22050_Gain3_Shift},
- {HPF_Fs22050_Gain4_A1, /* Gain setting 4 */
- HPF_Fs22050_Gain4_A0,
- -HPF_Fs22050_Gain4_B1,
- HPF_Fs22050_Gain4_Shift},
- {HPF_Fs22050_Gain5_A1, /* Gain setting 5 */
- HPF_Fs22050_Gain5_A0,
- -HPF_Fs22050_Gain5_B1,
- HPF_Fs22050_Gain5_Shift},
- {HPF_Fs22050_Gain6_A1, /* Gain setting 6 */
- HPF_Fs22050_Gain6_A0,
- -HPF_Fs22050_Gain6_B1,
- HPF_Fs22050_Gain6_Shift},
- {HPF_Fs22050_Gain7_A1, /* Gain setting 7 */
- HPF_Fs22050_Gain7_A0,
- -HPF_Fs22050_Gain7_B1,
- HPF_Fs22050_Gain7_Shift},
- {HPF_Fs22050_Gain8_A1, /* Gain setting 8 */
- HPF_Fs22050_Gain8_A0,
- -HPF_Fs22050_Gain8_B1,
- HPF_Fs22050_Gain8_Shift},
- {HPF_Fs22050_Gain9_A1, /* Gain setting 9 */
- HPF_Fs22050_Gain9_A0,
- -HPF_Fs22050_Gain9_B1,
- HPF_Fs22050_Gain9_Shift},
- {HPF_Fs22050_Gain10_A1, /* Gain setting 10 */
- HPF_Fs22050_Gain10_A0,
- -HPF_Fs22050_Gain10_B1,
- HPF_Fs22050_Gain10_Shift},
- {HPF_Fs22050_Gain11_A1, /* Gain setting 11 */
- HPF_Fs22050_Gain11_A0,
- -HPF_Fs22050_Gain11_B1,
- HPF_Fs22050_Gain11_Shift},
- {HPF_Fs22050_Gain12_A1, /* Gain setting 12 */
- HPF_Fs22050_Gain12_A0,
- -HPF_Fs22050_Gain12_B1,
- HPF_Fs22050_Gain12_Shift},
- {HPF_Fs22050_Gain13_A1, /* Gain setting 13 */
- HPF_Fs22050_Gain13_A0,
- -HPF_Fs22050_Gain13_B1,
- HPF_Fs22050_Gain13_Shift},
- {HPF_Fs22050_Gain14_A1, /* Gain setting 14 */
- HPF_Fs22050_Gain14_A0,
- -HPF_Fs22050_Gain14_B1,
- HPF_Fs22050_Gain14_Shift},
- {HPF_Fs22050_Gain15_A1, /* Gain setting 15 */
- HPF_Fs22050_Gain15_A0,
- -HPF_Fs22050_Gain15_B1,
- HPF_Fs22050_Gain15_Shift},
-
- /* 24kHz sampling rate */
- {HPF_Fs24000_Gain1_A1, /* Gain setting 1 */
- HPF_Fs24000_Gain1_A0,
- -HPF_Fs24000_Gain1_B1,
- HPF_Fs24000_Gain1_Shift},
- {HPF_Fs24000_Gain2_A1, /* Gain setting 2 */
- HPF_Fs24000_Gain2_A0,
- -HPF_Fs24000_Gain2_B1,
- HPF_Fs24000_Gain2_Shift},
- {HPF_Fs24000_Gain3_A1, /* Gain setting 3 */
- HPF_Fs24000_Gain3_A0,
- -HPF_Fs24000_Gain3_B1,
- HPF_Fs24000_Gain3_Shift},
- {HPF_Fs24000_Gain4_A1, /* Gain setting 4 */
- HPF_Fs24000_Gain4_A0,
- -HPF_Fs24000_Gain4_B1,
- HPF_Fs24000_Gain4_Shift},
- {HPF_Fs24000_Gain5_A1, /* Gain setting 5 */
- HPF_Fs24000_Gain5_A0,
- -HPF_Fs24000_Gain5_B1,
- HPF_Fs24000_Gain5_Shift},
- {HPF_Fs24000_Gain6_A1, /* Gain setting 6 */
- HPF_Fs24000_Gain6_A0,
- -HPF_Fs24000_Gain6_B1,
- HPF_Fs24000_Gain6_Shift},
- {HPF_Fs24000_Gain7_A1, /* Gain setting 7 */
- HPF_Fs24000_Gain7_A0,
- -HPF_Fs24000_Gain7_B1,
- HPF_Fs24000_Gain7_Shift},
- {HPF_Fs24000_Gain8_A1, /* Gain setting 8 */
- HPF_Fs24000_Gain8_A0,
- -HPF_Fs24000_Gain8_B1,
- HPF_Fs24000_Gain8_Shift},
- {HPF_Fs24000_Gain9_A1, /* Gain setting 9 */
- HPF_Fs24000_Gain9_A0,
- -HPF_Fs24000_Gain9_B1,
- HPF_Fs24000_Gain9_Shift},
- {HPF_Fs24000_Gain10_A1, /* Gain setting 10 */
- HPF_Fs24000_Gain10_A0,
- -HPF_Fs24000_Gain10_B1,
- HPF_Fs24000_Gain10_Shift},
- {HPF_Fs24000_Gain11_A1, /* Gain setting 11 */
- HPF_Fs24000_Gain11_A0,
- -HPF_Fs24000_Gain11_B1,
- HPF_Fs24000_Gain11_Shift},
- {HPF_Fs24000_Gain12_A1, /* Gain setting 12 */
- HPF_Fs24000_Gain12_A0,
- -HPF_Fs24000_Gain12_B1,
- HPF_Fs24000_Gain12_Shift},
- {HPF_Fs24000_Gain13_A1, /* Gain setting 13 */
- HPF_Fs24000_Gain13_A0,
- -HPF_Fs24000_Gain13_B1,
- HPF_Fs24000_Gain13_Shift},
- {HPF_Fs24000_Gain14_A1, /* Gain setting 14 */
- HPF_Fs24000_Gain14_A0,
- -HPF_Fs24000_Gain14_B1,
- HPF_Fs24000_Gain14_Shift},
- {HPF_Fs24000_Gain15_A1, /* Gain setting 15 */
- HPF_Fs24000_Gain15_A0,
- -HPF_Fs24000_Gain15_B1,
- HPF_Fs24000_Gain15_Shift},
-
- /* 32kHz sampling rate */
- {HPF_Fs32000_Gain1_A1, /* Gain setting 1 */
- HPF_Fs32000_Gain1_A0,
- -HPF_Fs32000_Gain1_B1,
- HPF_Fs32000_Gain1_Shift},
- {HPF_Fs32000_Gain2_A1, /* Gain setting 2 */
- HPF_Fs32000_Gain2_A0,
- -HPF_Fs32000_Gain2_B1,
- HPF_Fs32000_Gain2_Shift},
- {HPF_Fs32000_Gain3_A1, /* Gain setting 3 */
- HPF_Fs32000_Gain3_A0,
- -HPF_Fs32000_Gain3_B1,
- HPF_Fs32000_Gain3_Shift},
- {HPF_Fs32000_Gain4_A1, /* Gain setting 4 */
- HPF_Fs32000_Gain4_A0,
- -HPF_Fs32000_Gain4_B1,
- HPF_Fs32000_Gain4_Shift},
- {HPF_Fs32000_Gain5_A1, /* Gain setting 5 */
- HPF_Fs32000_Gain5_A0,
- -HPF_Fs32000_Gain5_B1,
- HPF_Fs32000_Gain5_Shift},
- {HPF_Fs32000_Gain6_A1, /* Gain setting 6 */
- HPF_Fs32000_Gain6_A0,
- -HPF_Fs32000_Gain6_B1,
- HPF_Fs32000_Gain6_Shift},
- {HPF_Fs32000_Gain7_A1, /* Gain setting 7 */
- HPF_Fs32000_Gain7_A0,
- -HPF_Fs32000_Gain7_B1,
- HPF_Fs32000_Gain7_Shift},
- {HPF_Fs32000_Gain8_A1, /* Gain setting 8 */
- HPF_Fs32000_Gain8_A0,
- -HPF_Fs32000_Gain8_B1,
- HPF_Fs32000_Gain8_Shift},
- {HPF_Fs32000_Gain9_A1, /* Gain setting 9 */
- HPF_Fs32000_Gain9_A0,
- -HPF_Fs32000_Gain9_B1,
- HPF_Fs32000_Gain9_Shift},
- {HPF_Fs32000_Gain10_A1, /* Gain setting 10 */
- HPF_Fs32000_Gain10_A0,
- -HPF_Fs32000_Gain10_B1,
- HPF_Fs32000_Gain10_Shift},
- {HPF_Fs32000_Gain11_A1, /* Gain setting 11 */
- HPF_Fs32000_Gain11_A0,
- -HPF_Fs32000_Gain11_B1,
- HPF_Fs32000_Gain11_Shift},
- {HPF_Fs32000_Gain12_A1, /* Gain setting 12 */
- HPF_Fs32000_Gain12_A0,
- -HPF_Fs32000_Gain12_B1,
- HPF_Fs32000_Gain12_Shift},
- {HPF_Fs32000_Gain13_A1, /* Gain setting 13 */
- HPF_Fs32000_Gain13_A0,
- -HPF_Fs32000_Gain13_B1,
- HPF_Fs32000_Gain13_Shift},
- {HPF_Fs32000_Gain14_A1, /* Gain setting 14 */
- HPF_Fs32000_Gain14_A0,
- -HPF_Fs32000_Gain14_B1,
- HPF_Fs32000_Gain14_Shift},
- {HPF_Fs32000_Gain15_A1, /* Gain setting 15 */
- HPF_Fs32000_Gain15_A0,
- -HPF_Fs32000_Gain15_B1,
- HPF_Fs32000_Gain15_Shift},
-
- /* 44kHz sampling rate */
- {HPF_Fs44100_Gain1_A1, /* Gain setting 1 */
- HPF_Fs44100_Gain1_A0,
- -HPF_Fs44100_Gain1_B1,
- HPF_Fs44100_Gain1_Shift},
- {HPF_Fs44100_Gain2_A1, /* Gain setting 2 */
- HPF_Fs44100_Gain2_A0,
- -HPF_Fs44100_Gain2_B1,
- HPF_Fs44100_Gain2_Shift},
- {HPF_Fs44100_Gain3_A1, /* Gain setting 3 */
- HPF_Fs44100_Gain3_A0,
- -HPF_Fs44100_Gain3_B1,
- HPF_Fs44100_Gain3_Shift},
- {HPF_Fs44100_Gain4_A1, /* Gain setting 4 */
- HPF_Fs44100_Gain4_A0,
- -HPF_Fs44100_Gain4_B1,
- HPF_Fs44100_Gain4_Shift},
- {HPF_Fs44100_Gain5_A1, /* Gain setting 5 */
- HPF_Fs44100_Gain5_A0,
- -HPF_Fs44100_Gain5_B1,
- HPF_Fs44100_Gain5_Shift},
- {HPF_Fs44100_Gain6_A1, /* Gain setting 6 */
- HPF_Fs44100_Gain6_A0,
- -HPF_Fs44100_Gain6_B1,
- HPF_Fs44100_Gain6_Shift},
- {HPF_Fs44100_Gain7_A1, /* Gain setting 7 */
- HPF_Fs44100_Gain7_A0,
- -HPF_Fs44100_Gain7_B1,
- HPF_Fs44100_Gain7_Shift},
- {HPF_Fs44100_Gain8_A1, /* Gain setting 8 */
- HPF_Fs44100_Gain8_A0,
- -HPF_Fs44100_Gain8_B1,
- HPF_Fs44100_Gain8_Shift},
- {HPF_Fs44100_Gain9_A1, /* Gain setting 9 */
- HPF_Fs44100_Gain9_A0,
- -HPF_Fs44100_Gain9_B1,
- HPF_Fs44100_Gain9_Shift},
- {HPF_Fs44100_Gain10_A1, /* Gain setting 10 */
- HPF_Fs44100_Gain10_A0,
- -HPF_Fs44100_Gain10_B1,
- HPF_Fs44100_Gain10_Shift},
- {HPF_Fs44100_Gain11_A1, /* Gain setting 11 */
- HPF_Fs44100_Gain11_A0,
- -HPF_Fs44100_Gain11_B1,
- HPF_Fs44100_Gain11_Shift},
- {HPF_Fs44100_Gain12_A1, /* Gain setting 12 */
- HPF_Fs44100_Gain12_A0,
- -HPF_Fs44100_Gain12_B1,
- HPF_Fs44100_Gain12_Shift},
- {HPF_Fs44100_Gain13_A1, /* Gain setting 13 */
- HPF_Fs44100_Gain13_A0,
- -HPF_Fs44100_Gain13_B1,
- HPF_Fs44100_Gain13_Shift},
- {HPF_Fs44100_Gain14_A1, /* Gain setting 14 */
- HPF_Fs44100_Gain14_A0,
- -HPF_Fs44100_Gain14_B1,
- HPF_Fs44100_Gain14_Shift},
- {HPF_Fs44100_Gain15_A1, /* Gain setting 15 */
- HPF_Fs44100_Gain15_A0,
- -HPF_Fs44100_Gain15_B1,
- HPF_Fs44100_Gain15_Shift},
-
- /* 48kHz sampling rate */
- {HPF_Fs48000_Gain1_A1, /* Gain setting 1 */
- HPF_Fs48000_Gain1_A0,
- -HPF_Fs48000_Gain1_B1,
- HPF_Fs48000_Gain1_Shift},
- {HPF_Fs48000_Gain2_A1, /* Gain setting 2 */
- HPF_Fs48000_Gain2_A0,
- -HPF_Fs48000_Gain2_B1,
- HPF_Fs48000_Gain2_Shift},
- {HPF_Fs48000_Gain3_A1, /* Gain setting 3 */
- HPF_Fs48000_Gain3_A0,
- -HPF_Fs48000_Gain3_B1,
- HPF_Fs48000_Gain3_Shift},
- {HPF_Fs48000_Gain4_A1, /* Gain setting 4 */
- HPF_Fs48000_Gain4_A0,
- -HPF_Fs48000_Gain4_B1,
- HPF_Fs48000_Gain4_Shift},
- {HPF_Fs48000_Gain5_A1, /* Gain setting 5 */
- HPF_Fs48000_Gain5_A0,
- -HPF_Fs48000_Gain5_B1,
- HPF_Fs48000_Gain5_Shift},
- {HPF_Fs48000_Gain6_A1, /* Gain setting 6 */
- HPF_Fs48000_Gain6_A0,
- -HPF_Fs48000_Gain6_B1,
- HPF_Fs48000_Gain6_Shift},
- {HPF_Fs48000_Gain7_A1, /* Gain setting 7 */
- HPF_Fs48000_Gain7_A0,
- -HPF_Fs48000_Gain7_B1,
- HPF_Fs48000_Gain7_Shift},
- {HPF_Fs48000_Gain8_A1, /* Gain setting 8 */
- HPF_Fs48000_Gain8_A0,
- -HPF_Fs48000_Gain8_B1,
- HPF_Fs48000_Gain8_Shift},
- {HPF_Fs48000_Gain9_A1, /* Gain setting 9 */
- HPF_Fs48000_Gain9_A0,
- -HPF_Fs48000_Gain9_B1,
- HPF_Fs48000_Gain9_Shift},
- {HPF_Fs48000_Gain10_A1, /* Gain setting 10 */
- HPF_Fs48000_Gain10_A0,
- -HPF_Fs48000_Gain10_B1,
- HPF_Fs48000_Gain10_Shift},
- {HPF_Fs48000_Gain11_A1, /* Gain setting 11 */
- HPF_Fs48000_Gain11_A0,
- -HPF_Fs48000_Gain11_B1,
- HPF_Fs48000_Gain11_Shift},
- {HPF_Fs48000_Gain12_A1, /* Gain setting 12 */
- HPF_Fs48000_Gain12_A0,
- -HPF_Fs48000_Gain12_B1,
- HPF_Fs48000_Gain12_Shift},
- {HPF_Fs48000_Gain13_A1, /* Gain setting 13 */
- HPF_Fs48000_Gain13_A0,
- -HPF_Fs48000_Gain13_B1,
- HPF_Fs48000_Gain13_Shift},
- {HPF_Fs48000_Gain14_A1, /* Gain setting 14 */
- HPF_Fs48000_Gain14_A0,
- -HPF_Fs48000_Gain14_B1,
- HPF_Fs48000_Gain14_Shift},
- {HPF_Fs48000_Gain15_A1, /* Gain setting 15 */
- HPF_Fs48000_Gain15_A0,
- -HPF_Fs48000_Gain15_B1,
- HPF_Fs48000_Gain15_Shift}
- };
-#endif
/************************************************************************************/
/* */
@@ -780,7 +463,6 @@
/************************************************************************************/
/* dB to linear conversion table */
-#ifdef BUILD_FLOAT
const LVM_FLOAT LVM_VolumeTable[] = {
1.000f, /* 0dB */
0.891f, /* -1dB */
@@ -789,16 +471,6 @@
0.631f, /* -4dB */
0.562f, /* -5dB */
0.501f}; /* -6dB */
-#else
-const LVM_INT16 LVM_VolumeTable[] = {
- 0x7FFF, /* 0dB */
- 0x7215, /* -1dB */
- 0x65AD, /* -2dB */
- 0x5A9E, /* -3dB */
- 0x50C3, /* -4dB */
- 0x47FB, /* -5dB */
- 0x4000}; /* -6dB */
-#endif
/************************************************************************************/
/* */
@@ -816,7 +488,6 @@
#define LVM_MIX_TC_Fs44100 32734 /* Floating point value 0.998962402 */
#define LVM_MIX_TC_Fs48000 32737 /* Floating point value 0.999053955 */
-
const LVM_INT16 LVM_MixerTCTable[] = {
LVM_MIX_TC_Fs8000,
LVM_MIX_TC_Fs11025,
diff --git a/media/libeffects/lvm/lib/Bundle/src/LVM_Tables.h b/media/libeffects/lvm/lib/Bundle/src/LVM_Tables.h
index 4cf7119..fc82194 100644
--- a/media/libeffects/lvm/lib/Bundle/src/LVM_Tables.h
+++ b/media/libeffects/lvm/lib/Bundle/src/LVM_Tables.h
@@ -18,10 +18,6 @@
#ifndef __LVM_TABLES_H__
#define __LVM_TABLES_H__
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
/************************************************************************************/
/* */
/* Includes */
@@ -37,30 +33,16 @@
/* */
/************************************************************************************/
-#ifdef BUILD_FLOAT
extern FO_FLOAT_LShx_Coefs_t LVM_TrebleBoostCoefs[];
-#else
-extern FO_C16_LShx_Coefs_t LVM_TrebleBoostCoefs[];
-#endif
/************************************************************************************/
/* */
/* Volume control gain and time constant tables */
/* */
/************************************************************************************/
-#ifdef BUILD_FLOAT
extern const LVM_FLOAT LVM_VolumeTable[];
-#else
-extern const LVM_INT16 LVM_VolumeTable[];
-#endif
extern const LVM_INT16 LVM_MixerTCTable[];
-
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
-
#endif /* __LVM_TABLES_H__ */
-
diff --git a/media/libeffects/lvm/lib/Common/lib/AGC.h b/media/libeffects/lvm/lib/Common/lib/AGC.h
index 06e742e..bef7fa1 100644
--- a/media/libeffects/lvm/lib/Common/lib/AGC.h
+++ b/media/libeffects/lvm/lib/Common/lib/AGC.h
@@ -18,11 +18,6 @@
#ifndef __AGC_H__
#define __AGC_H__
-
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
/**********************************************************************************/
/* */
/* Includes */
@@ -31,28 +26,11 @@
#include "LVM_Types.h"
-
/**********************************************************************************/
/* */
/* Types */
/* */
/**********************************************************************************/
-#ifndef BUILD_FLOAT
-typedef struct
-{
- LVM_INT32 AGC_Gain; /* The current AGC gain */
- LVM_INT32 AGC_MaxGain; /* The maximum AGC gain */
- LVM_INT32 Volume; /* The current volume setting */
- LVM_INT32 Target; /* The target volume setting */
- LVM_INT32 AGC_Target; /* AGC target level */
- LVM_INT16 AGC_Attack; /* AGC attack scaler */
- LVM_INT16 AGC_Decay; /* AGC decay scaler */
- LVM_INT16 AGC_GainShift; /* The gain shift */
- LVM_INT16 VolumeShift; /* Volume shift scaling */
- LVM_INT16 VolumeTC; /* Volume update time constant */
-
-} AGC_MIX_VOL_2St1Mon_D32_t;
-#else
typedef struct
{
LVM_FLOAT AGC_Gain; /* The current AGC gain */
@@ -65,14 +43,12 @@
LVM_FLOAT VolumeTC; /* Volume update time constant */
} AGC_MIX_VOL_2St1Mon_FLOAT_t;
-#endif
/**********************************************************************************/
/* */
/* Function Prototypes */
/* */
/**********************************************************************************/
-#ifdef BUILD_FLOAT
void AGC_MIX_VOL_2St1Mon_D32_WRA(AGC_MIX_VOL_2St1Mon_FLOAT_t *pInstance, /* Instance pointer */
const LVM_FLOAT *pStSrc, /* Stereo source */
const LVM_FLOAT *pMonoSrc, /* Mono source */
@@ -87,26 +63,5 @@
LVM_UINT16 NrChannels); /* Number of channels */
#endif
-#else
-void AGC_MIX_VOL_2St1Mon_D32_WRA(AGC_MIX_VOL_2St1Mon_D32_t *pInstance, /* Instance pointer */
- const LVM_INT32 *pStSrc, /* Stereo source */
- const LVM_INT32 *pMonoSrc, /* Mono source */
- LVM_INT32 *pDst, /* Stereo destination */
- LVM_UINT16 n); /* Number of samples */
-#endif
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
-
-
#endif /* __AGC_H__ */
-
-
-
-
-
-
-
-
-
diff --git a/media/libeffects/lvm/lib/Common/lib/BIQUAD.h b/media/libeffects/lvm/lib/Common/lib/BIQUAD.h
index 01539b2..c050cd0 100644
--- a/media/libeffects/lvm/lib/Common/lib/BIQUAD.h
+++ b/media/libeffects/lvm/lib/Common/lib/BIQUAD.h
@@ -18,16 +18,10 @@
#ifndef _BIQUAD_H_
#define _BIQUAD_H_
-
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
#include "LVM_Types.h"
/**********************************************************************************
INSTANCE MEMORY TYPE DEFINITION
***********************************************************************************/
-#ifdef BUILD_FLOAT
typedef struct
{
#ifdef SUPPORT_MC
@@ -42,19 +36,11 @@
LVM_FLOAT Storage[6];
#endif
} Biquad_FLOAT_Instance_t;
-#else
-typedef struct
-{
- LVM_INT32 Storage[6];
-
-} Biquad_Instance_t;
-#endif
/**********************************************************************************
COEFFICIENT TYPE DEFINITIONS
***********************************************************************************/
/*** Biquad coefficients **********************************************************/
-#ifdef BUILD_FLOAT
typedef struct
{
LVM_FLOAT A2; /* a2 */
@@ -63,93 +49,31 @@
LVM_FLOAT B2; /* -b2! */
LVM_FLOAT B1; /* -b1! */
} BQ_FLOAT_Coefs_t;
-#else
-typedef struct
-{
- LVM_INT16 A2; /* a2 */
- LVM_INT16 A1; /* a1 */
- LVM_INT16 A0; /* a0 */
- LVM_INT16 B2; /* -b2! */
- LVM_INT16 B1; /* -b1! */
-} BQ_C16_Coefs_t;
-
-typedef struct
-{
- LVM_INT32 A2; /* a2 */
- LVM_INT32 A1; /* a1 */
- LVM_INT32 A0; /* a0 */
- LVM_INT32 B2; /* -b2! */
- LVM_INT32 B1; /* -b1! */
-} BQ_C32_Coefs_t;
-#endif
/*** First order coefficients *****************************************************/
-#ifdef BUILD_FLOAT
typedef struct
{
LVM_FLOAT A1; /* a1 */
LVM_FLOAT A0; /* a0 */
LVM_FLOAT B1; /* -b1! */
} FO_FLOAT_Coefs_t;
-#else
-typedef struct
-{
- LVM_INT16 A1; /* a1 */
- LVM_INT16 A0; /* a0 */
- LVM_INT16 B1; /* -b1! */
-} FO_C16_Coefs_t;
-
-typedef struct
-{
- LVM_INT32 A1; /* a1 */
- LVM_INT32 A0; /* a0 */
- LVM_INT32 B1; /* -b1! */
-} FO_C32_Coefs_t;
-#endif
/*** First order coefficients with Shift*****************************************************/
-#ifdef BUILD_FLOAT
typedef struct
{
LVM_FLOAT A1; /* a1 */
LVM_FLOAT A0; /* a0 */
LVM_FLOAT B1; /* -b1! */
} FO_FLOAT_LShx_Coefs_t;
-#else
-typedef struct
-{
- LVM_INT16 A1; /* a1 */
- LVM_INT16 A0; /* a0 */
- LVM_INT16 B1; /* -b1! */
- LVM_INT16 Shift; /* Shift */
-} FO_C16_LShx_Coefs_t;
-#endif
/*** Band pass coefficients *******************************************************/
-#ifdef BUILD_FLOAT
typedef struct
{
LVM_FLOAT A0; /* a0 */
LVM_FLOAT B2; /* -b2! */
LVM_FLOAT B1; /* -b1! */
} BP_FLOAT_Coefs_t;
-#else
-typedef struct
-{
- LVM_INT16 A0; /* a0 */
- LVM_INT16 B2; /* -b2! */
- LVM_INT16 B1; /* -b1! */
-} BP_C16_Coefs_t;
-
-typedef struct
-{
- LVM_INT32 A0; /* a0 */
- LVM_INT32 B2; /* -b2! */
- LVM_INT32 B1; /* -b1! */
-} BP_C32_Coefs_t;
-#endif
/*** Peaking coefficients *********************************************************/
-#ifdef BUILD_FLOAT
typedef struct
{
LVM_FLOAT A0; /* a0 */
@@ -157,30 +81,12 @@
LVM_FLOAT B1; /* -b1! */
LVM_FLOAT G; /* Gain */
} PK_FLOAT_Coefs_t;
-#else
-typedef struct
-{
- LVM_INT16 A0; /* a0 */
- LVM_INT16 B2; /* -b2! */
- LVM_INT16 B1; /* -b1! */
- LVM_INT16 G; /* Gain */
-} PK_C16_Coefs_t;
-
-typedef struct
-{
- LVM_INT32 A0; /* a0 */
- LVM_INT32 B2; /* -b2! */
- LVM_INT32 B1; /* -b1! */
- LVM_INT16 G; /* Gain */
-} PK_C32_Coefs_t;
-#endif
/**********************************************************************************
TAPS TYPE DEFINITIONS
***********************************************************************************/
/*** Types used for first order and shelving filter *******************************/
-#ifdef BUILD_FLOAT
typedef struct
{
LVM_FLOAT Storage[ (1 * 2) ]; /* One channel, two taps of size LVM_INT32 */
@@ -195,20 +101,8 @@
LVM_FLOAT Storage[ (2 * 2) ]; /* Two channels, two taps of size LVM_FLOAT */
#endif
} Biquad_2I_Order1_FLOAT_Taps_t;
-#else
-typedef struct
-{
- LVM_INT32 Storage[ (1*2) ]; /* One channel, two taps of size LVM_INT32 */
-} Biquad_1I_Order1_Taps_t;
-
-typedef struct
-{
- LVM_INT32 Storage[ (2*2) ]; /* Two channels, two taps of size LVM_INT32 */
-} Biquad_2I_Order1_Taps_t;
-#endif
/*** Types used for biquad, band pass and peaking filter **************************/
-#ifdef BUILD_FLOAT
typedef struct
{
LVM_FLOAT Storage[ (1 * 4) ]; /* One channel, four taps of size LVM_FLOAT */
@@ -223,17 +117,6 @@
LVM_FLOAT Storage[ (2 * 4) ]; /* Two channels, four taps of size LVM_FLOAT */
#endif
} Biquad_2I_Order2_FLOAT_Taps_t;
-#else
-typedef struct
-{
- LVM_INT32 Storage[ (1*4) ]; /* One channel, four taps of size LVM_INT32 */
-} Biquad_1I_Order2_Taps_t;
-
-typedef struct
-{
- LVM_INT32 Storage[ (2*4) ]; /* Two channels, four taps of size LVM_INT32 */
-} Biquad_2I_Order2_Taps_t;
-#endif
/* The names of the functions are changed to satisfy QAC rules: Name should be Unique withing 16 characters*/
#define BQ_2I_D32F32Cll_TRC_WRA_01_Init Init_BQ_2I_D32F32Cll_TRC_WRA_01
#define BP_1I_D32F32C30_TRC_WRA_02 TWO_BP_1I_D32F32C30_TRC_WRA_02
@@ -244,140 +127,57 @@
/*** 16 bit data path *************************************************************/
-
-#ifdef BUILD_FLOAT
void BQ_2I_D16F32Css_TRC_WRA_01_Init ( Biquad_FLOAT_Instance_t *pInstance,
Biquad_2I_Order2_FLOAT_Taps_t *pTaps,
BQ_FLOAT_Coefs_t *pCoef);
-#else
-void BQ_2I_D16F32Css_TRC_WRA_01_Init ( Biquad_Instance_t *pInstance,
- Biquad_2I_Order2_Taps_t *pTaps,
- BQ_C16_Coefs_t *pCoef);
-#endif
-#ifdef BUILD_FLOAT
void BQ_2I_D16F32C15_TRC_WRA_01 ( Biquad_FLOAT_Instance_t *pInstance,
LVM_FLOAT *pDataIn,
LVM_FLOAT *pDataOut,
LVM_INT16 NrSamples);
-#else
-void BQ_2I_D16F32C15_TRC_WRA_01 ( Biquad_Instance_t *pInstance,
- LVM_INT16 *pDataIn,
- LVM_INT16 *pDataOut,
- LVM_INT16 NrSamples);
-#endif
-#ifdef BUILD_FLOAT
void BQ_2I_D16F32C14_TRC_WRA_01 ( Biquad_FLOAT_Instance_t *pInstance,
LVM_FLOAT *pDataIn,
LVM_FLOAT *pDataOut,
LVM_INT16 NrSamples);
-#else
-void BQ_2I_D16F32C14_TRC_WRA_01 ( Biquad_Instance_t *pInstance,
- LVM_INT16 *pDataIn,
- LVM_INT16 *pDataOut,
- LVM_INT16 NrSamples);
-#endif
-
-#ifdef BUILD_FLOAT
void BQ_2I_D16F32C13_TRC_WRA_01 ( Biquad_FLOAT_Instance_t *pInstance,
LVM_FLOAT *pDataIn,
LVM_FLOAT *pDataOut,
LVM_INT16 NrSamples);
-#else
-void BQ_2I_D16F32C13_TRC_WRA_01 ( Biquad_Instance_t *pInstance,
- LVM_INT16 *pDataIn,
- LVM_INT16 *pDataOut,
- LVM_INT16 NrSamples);
-#endif
-
-#ifdef BUILD_FLOAT
void BQ_2I_D16F16Css_TRC_WRA_01_Init ( Biquad_FLOAT_Instance_t *pInstance,
Biquad_2I_Order2_FLOAT_Taps_t *pTaps,
BQ_FLOAT_Coefs_t *pCoef);
-#else
-void BQ_2I_D16F16Css_TRC_WRA_01_Init ( Biquad_Instance_t *pInstance,
- Biquad_2I_Order2_Taps_t *pTaps,
- BQ_C16_Coefs_t *pCoef);
-#endif
-
-#ifdef BUILD_FLOAT
void BQ_2I_D16F16C15_TRC_WRA_01( Biquad_FLOAT_Instance_t *pInstance,
LVM_FLOAT *pDataIn,
LVM_FLOAT *pDataOut,
LVM_INT16 NrSamples);
-#else
-void BQ_2I_D16F16C15_TRC_WRA_01( Biquad_Instance_t *pInstance,
- LVM_INT16 *pDataIn,
- LVM_INT16 *pDataOut,
- LVM_INT16 NrSamples);
-#endif
-
-#ifdef BUILD_FLOAT
void BQ_2I_D16F16C14_TRC_WRA_01( Biquad_FLOAT_Instance_t *pInstance,
LVM_FLOAT *pDataIn,
LVM_FLOAT *pDataOut,
LVM_INT16 NrSamples);
-#else
-void BQ_2I_D16F16C14_TRC_WRA_01( Biquad_Instance_t *pInstance,
- LVM_INT16 *pDataIn,
- LVM_INT16 *pDataOut,
- LVM_INT16 NrSamples);
-#endif
-#ifdef BUILD_FLOAT
void BQ_1I_D16F16Css_TRC_WRA_01_Init ( Biquad_FLOAT_Instance_t *pInstance,
Biquad_1I_Order2_FLOAT_Taps_t *pTaps,
BQ_FLOAT_Coefs_t *pCoef);
-#else
-void BQ_1I_D16F16Css_TRC_WRA_01_Init ( Biquad_Instance_t *pInstance,
- Biquad_1I_Order2_Taps_t *pTaps,
- BQ_C16_Coefs_t *pCoef);
-#endif
-
-#ifdef BUILD_FLOAT
void BQ_1I_D16F16C15_TRC_WRA_01 ( Biquad_FLOAT_Instance_t *pInstance,
LVM_FLOAT *pDataIn,
LVM_FLOAT *pDataOut,
LVM_INT16 NrSamples);
-#else
-void BQ_1I_D16F16C15_TRC_WRA_01( Biquad_Instance_t *pInstance,
- LVM_INT16 *pDataIn,
- LVM_INT16 *pDataOut,
- LVM_INT16 NrSamples);
-#endif
-
-#ifdef BUILD_FLOAT
void BQ_1I_D16F32Css_TRC_WRA_01_Init ( Biquad_FLOAT_Instance_t *pInstance,
Biquad_1I_Order2_FLOAT_Taps_t *pTaps,
BQ_FLOAT_Coefs_t *pCoef);
-#else
-void BQ_1I_D16F32Css_TRC_WRA_01_Init ( Biquad_Instance_t *pInstance,
- Biquad_1I_Order2_Taps_t *pTaps,
- BQ_C16_Coefs_t *pCoef);
-#endif
-
-#ifdef BUILD_FLOAT
void BQ_1I_D16F32C14_TRC_WRA_01 ( Biquad_FLOAT_Instance_t *pInstance,
LVM_FLOAT *pDataIn,
LVM_FLOAT *pDataOut,
LVM_INT16 NrSamples);
-#else
-void BQ_1I_D16F32C14_TRC_WRA_01 ( Biquad_Instance_t *pInstance,
- LVM_INT16 *pDataIn,
- LVM_INT16 *pDataOut,
- LVM_INT16 NrSamples);
-
-#endif
/*** 32 bit data path *************************************************************/
-#ifdef BUILD_FLOAT
void BQ_2I_D32F32Cll_TRC_WRA_01_Init ( Biquad_FLOAT_Instance_t *pInstance,
Biquad_2I_Order2_FLOAT_Taps_t *pTaps,
BQ_FLOAT_Coefs_t *pCoef);
@@ -392,67 +192,30 @@
LVM_INT16 NrFrames,
LVM_INT16 NrChannels);
#endif
-#else
-void BQ_2I_D32F32Cll_TRC_WRA_01_Init ( Biquad_Instance_t *pInstance,
- Biquad_2I_Order2_Taps_t *pTaps,
- BQ_C32_Coefs_t *pCoef);
-
-void BQ_2I_D32F32C30_TRC_WRA_01 ( Biquad_Instance_t *pInstance,
- LVM_INT32 *pDataIn,
- LVM_INT32 *pDataOut,
- LVM_INT16 NrSamples);
-#endif
/**********************************************************************************
FUNCTION PROTOTYPES: FIRST ORDER FILTERS
***********************************************************************************/
/*** 16 bit data path *************************************************************/
-#ifdef BUILD_FLOAT
void FO_1I_D16F16Css_TRC_WRA_01_Init( Biquad_FLOAT_Instance_t *pInstance,
Biquad_1I_Order1_FLOAT_Taps_t *pTaps,
FO_FLOAT_Coefs_t *pCoef);
-#else
-void FO_1I_D16F16Css_TRC_WRA_01_Init( Biquad_Instance_t *pInstance,
- Biquad_1I_Order1_Taps_t *pTaps,
- FO_C16_Coefs_t *pCoef);
-#endif
-#ifdef BUILD_FLOAT
void FO_1I_D16F16C15_TRC_WRA_01( Biquad_FLOAT_Instance_t *pInstance,
LVM_FLOAT *pDataIn,
LVM_FLOAT *pDataOut,
LVM_INT16 NrSamples);
-#else
-void FO_1I_D16F16C15_TRC_WRA_01( Biquad_Instance_t *pInstance,
- LVM_INT16 *pDataIn,
- LVM_INT16 *pDataOut,
- LVM_INT16 NrSamples);
-#endif
-#ifdef BUILD_FLOAT
void FO_2I_D16F32Css_LShx_TRC_WRA_01_Init(Biquad_FLOAT_Instance_t *pInstance,
Biquad_2I_Order1_FLOAT_Taps_t *pTaps,
FO_FLOAT_LShx_Coefs_t *pCoef);
-#else
-void FO_2I_D16F32Css_LShx_TRC_WRA_01_Init(Biquad_Instance_t *pInstance,
- Biquad_2I_Order1_Taps_t *pTaps,
- FO_C16_LShx_Coefs_t *pCoef);
-#endif
-#ifdef BUILD_FLOAT
void FO_2I_D16F32C15_LShx_TRC_WRA_01(Biquad_FLOAT_Instance_t *pInstance,
LVM_FLOAT *pDataIn,
LVM_FLOAT *pDataOut,
LVM_INT16 NrSamples);
-#else
-void FO_2I_D16F32C15_LShx_TRC_WRA_01(Biquad_Instance_t *pInstance,
- LVM_INT16 *pDataIn,
- LVM_INT16 *pDataOut,
- LVM_INT16 NrSamples);
-#endif
/*** 32 bit data path *************************************************************/
-#ifdef BUILD_FLOAT
void FO_1I_D32F32Cll_TRC_WRA_01_Init( Biquad_FLOAT_Instance_t *pInstance,
Biquad_1I_Order1_FLOAT_Taps_t *pTaps,
FO_FLOAT_Coefs_t *pCoef);
@@ -467,22 +230,11 @@
LVM_INT16 NrFrames,
LVM_INT16 NrChannels);
#endif
-#else
-void FO_1I_D32F32Cll_TRC_WRA_01_Init( Biquad_Instance_t *pInstance,
- Biquad_1I_Order1_Taps_t *pTaps,
- FO_C32_Coefs_t *pCoef);
-
-void FO_1I_D32F32C31_TRC_WRA_01( Biquad_Instance_t *pInstance,
- LVM_INT32 *pDataIn,
- LVM_INT32 *pDataOut,
- LVM_INT16 NrSamples);
-#endif
/**********************************************************************************
FUNCTION PROTOTYPES: BAND PASS FILTERS
***********************************************************************************/
/*** 16 bit data path *************************************************************/
-#ifdef BUILD_FLOAT
void BP_1I_D16F16Css_TRC_WRA_01_Init( Biquad_FLOAT_Instance_t *pInstance,
Biquad_1I_Order2_FLOAT_Taps_t *pTaps,
BP_FLOAT_Coefs_t *pCoef);
@@ -497,27 +249,7 @@
LVM_FLOAT *pDataIn,
LVM_FLOAT *pDataOut,
LVM_INT16 NrSamples);
-#else
-void BP_1I_D16F16Css_TRC_WRA_01_Init ( Biquad_Instance_t *pInstance,
- Biquad_1I_Order2_Taps_t *pTaps,
- BP_C16_Coefs_t *pCoef);
-
-void BP_1I_D16F16C14_TRC_WRA_01 ( Biquad_Instance_t *pInstance,
- LVM_INT16 *pDataIn,
- LVM_INT16 *pDataOut,
- LVM_INT16 NrSamples);
-
-void BP_1I_D16F32Cll_TRC_WRA_01_Init ( Biquad_Instance_t *pInstance,
- Biquad_1I_Order2_Taps_t *pTaps,
- BP_C32_Coefs_t *pCoef);
-
-void BP_1I_D16F32C30_TRC_WRA_01 ( Biquad_Instance_t *pInstance,
- LVM_INT16 *pDataIn,
- LVM_INT16 *pDataOut,
- LVM_INT16 NrSamples);
-#endif
/*** 32 bit data path *************************************************************/
-#ifdef BUILD_FLOAT
void BP_1I_D32F32Cll_TRC_WRA_02_Init ( Biquad_FLOAT_Instance_t *pInstance,
Biquad_1I_Order2_FLOAT_Taps_t *pTaps,
BP_FLOAT_Coefs_t *pCoef);
@@ -525,37 +257,11 @@
LVM_FLOAT *pDataIn,
LVM_FLOAT *pDataOut,
LVM_INT16 NrSamples);
-#else
-void BP_1I_D32F32Cll_TRC_WRA_02_Init ( Biquad_Instance_t *pInstance,
- Biquad_1I_Order2_Taps_t *pTaps,
- BP_C32_Coefs_t *pCoef);
-
-void BP_1I_D32F32C30_TRC_WRA_02( Biquad_Instance_t *pInstance,
- LVM_INT32 *pDataIn,
- LVM_INT32 *pDataOut,
- LVM_INT16 NrSamples);
-#endif
/*** 32 bit data path STEREO ******************************************************/
-#ifndef BUILD_FLOAT
-void PK_2I_D32F32CllGss_TRC_WRA_01_Init ( Biquad_Instance_t *pInstance,
- Biquad_2I_Order2_Taps_t *pTaps,
- PK_C32_Coefs_t *pCoef);
-void PK_2I_D32F32C30G11_TRC_WRA_01 ( Biquad_Instance_t *pInstance,
- LVM_INT32 *pDataIn,
- LVM_INT32 *pDataOut,
- LVM_INT16 NrSamples);
-#endif
-#ifdef BUILD_FLOAT
void PK_2I_D32F32CssGss_TRC_WRA_01_Init ( Biquad_FLOAT_Instance_t *pInstance,
Biquad_2I_Order2_FLOAT_Taps_t *pTaps,
PK_FLOAT_Coefs_t *pCoef);
-#else
-void PK_2I_D32F32CssGss_TRC_WRA_01_Init ( Biquad_Instance_t *pInstance,
- Biquad_2I_Order2_Taps_t *pTaps,
- PK_C16_Coefs_t *pCoef);
-#endif
-#ifdef BUILD_FLOAT
void PK_2I_D32F32C14G11_TRC_WRA_01( Biquad_FLOAT_Instance_t *pInstance,
LVM_FLOAT *pDataIn,
LVM_FLOAT *pDataOut,
@@ -567,19 +273,12 @@
LVM_INT16 NrFrames,
LVM_INT16 NrChannels);
#endif
-#else
-void PK_2I_D32F32C14G11_TRC_WRA_01 ( Biquad_Instance_t *pInstance,
- LVM_INT32 *pDataIn,
- LVM_INT32 *pDataOut,
- LVM_INT16 NrSamples);
-#endif
/**********************************************************************************
FUNCTION PROTOTYPES: DC REMOVAL FILTERS
***********************************************************************************/
/*** 16 bit data path STEREO ******************************************************/
-#ifdef BUILD_FLOAT
#ifdef SUPPORT_MC
void DC_Mc_D16_TRC_WRA_01_Init ( Biquad_FLOAT_Instance_t *pInstance);
@@ -596,18 +295,6 @@
LVM_FLOAT *pDataOut,
LVM_INT16 NrSamples);
#endif
-#else
-void DC_2I_D16_TRC_WRA_01_Init ( Biquad_Instance_t *pInstance);
-
-void DC_2I_D16_TRC_WRA_01 ( Biquad_Instance_t *pInstance,
- LVM_INT16 *pDataIn,
- LVM_INT16 *pDataOut,
- LVM_INT16 NrSamples);
-#endif
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
-
/**********************************************************************************/
diff --git a/media/libeffects/lvm/lib/Common/lib/CompLim.h b/media/libeffects/lvm/lib/Common/lib/CompLim.h
index 498faa3..5b7cb1b 100644
--- a/media/libeffects/lvm/lib/Common/lib/CompLim.h
+++ b/media/libeffects/lvm/lib/Common/lib/CompLim.h
@@ -18,11 +18,6 @@
#ifndef _COMP_LIM_H
#define _COMP_LIM_H
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
-
/************************************************************************************/
/* */
/* Includes */
@@ -31,7 +26,6 @@
#include "LVM_Types.h"
-
/************************************************************************************/
/* */
/* Structures */
@@ -57,31 +51,17 @@
LVM_INT32 CompIntSlow; /* Compressor slow integrator current value */
LVM_INT32 CompIntFast; /* Compressor fast integrator current value */
-
} CompLim_Instance_t;
-
/************************************************************************************/
/* */
/* Function Prototypes */
/* */
/************************************************************************************/
-#ifdef BUILD_FLOAT
void NonLinComp_Float(LVM_FLOAT Gain,
LVM_FLOAT *pDataIn,
LVM_FLOAT *pDataOut,
LVM_INT32 BlockLength);
-#else
-void NonLinComp_D16(LVM_INT16 Gain,
- LVM_INT16 *pSterBfIn,
- LVM_INT16 *pSterBfOut,
- LVM_INT32 BlockLength);
-#endif
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
#endif /* #ifndef _COMP_LIM_H */
-
-
diff --git a/media/libeffects/lvm/lib/Common/lib/Filter.h b/media/libeffects/lvm/lib/Common/lib/Filter.h
index 0c8955d..1eeb321 100644
--- a/media/libeffects/lvm/lib/Common/lib/Filter.h
+++ b/media/libeffects/lvm/lib/Common/lib/Filter.h
@@ -18,39 +18,27 @@
#ifndef _FILTER_H_
#define _FILTER_H_
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
/**********************************************************************************
INCLUDES
***********************************************************************************/
#include "LVM_Types.h"
#include "BIQUAD.h"
-
/**********************************************************************************
DEFINES
***********************************************************************************/
#define FILTER_LOSS 32730 /* -0.01dB loss to avoid wrapping due to band ripple */
-#ifdef BUILD_FLOAT
#define FILTER_LOSS_FLOAT 0.998849f
-#endif
/**********************************************************************************
FUNCTION PROTOTYPES
***********************************************************************************/
-#ifdef BUILD_FLOAT
LVM_FLOAT LVM_Power10( LVM_FLOAT X);
LVM_FLOAT LVM_Polynomial(LVM_UINT16 N,
LVM_FLOAT *pCoefficients,
LVM_FLOAT X);
-#ifdef HIGHER_FS
LVM_FLOAT LVM_GetOmega(LVM_UINT32 Fc,
-#else
-LVM_FLOAT LVM_GetOmega(LVM_UINT16 Fc,
-#endif
LVM_Fs_en SampleRate);
LVM_FLOAT LVM_FO_LPF( LVM_FLOAT w,
@@ -58,26 +46,7 @@
LVM_FLOAT LVM_FO_HPF( LVM_FLOAT w,
FO_FLOAT_Coefs_t *pCoeffs);
-#else
-LVM_INT32 LVM_Polynomial(LVM_UINT16 N,
- LVM_INT32 *pCoefficients,
- LVM_INT32 X);
-
-LVM_INT32 LVM_Power10( LVM_INT32 X);
-
-LVM_INT32 LVM_FO_LPF( LVM_INT32 w,
- FO_C32_Coefs_t *pCoeffs);
-
-LVM_INT32 LVM_FO_HPF( LVM_INT32 w,
- FO_C32_Coefs_t *pCoeffs);
-
-LVM_INT32 LVM_GetOmega(LVM_UINT16 Fc,
- LVM_Fs_en SampleRate);
-#endif
/**********************************************************************************/
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
#endif /** _FILTER_H_ **/
diff --git a/media/libeffects/lvm/lib/Common/lib/InstAlloc.h b/media/libeffects/lvm/lib/Common/lib/InstAlloc.h
index 7f725f4..bae84e7 100644
--- a/media/libeffects/lvm/lib/Common/lib/InstAlloc.h
+++ b/media/libeffects/lvm/lib/Common/lib/InstAlloc.h
@@ -18,10 +18,6 @@
#ifndef __INSTALLOC_H__
#define __INSTALLOC_H__
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
#include "LVM_Types.h"
/*######################################################################################*/
/* Type declarations */
@@ -32,7 +28,6 @@
uintptr_t pNextMember; /* Pointer to the next instance member to be allocated */
} INST_ALLOC;
-
/*######################################################################################*/
/* Function prototypes */
/*######################################################################################*/
@@ -48,7 +43,6 @@
void InstAlloc_Init( INST_ALLOC *pms, void *StartAddr );
-
/****************************************************************************************
* Name : InstAlloc_AddMember()
* Input : pms - Pointer to the INST_ALLOC instance
@@ -85,8 +79,4 @@
void InstAlloc_InitAll_NULL( INST_ALLOC *pms);
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
-
#endif /* __JBS_INSTALLOC_H__ */
diff --git a/media/libeffects/lvm/lib/Common/lib/LVM_Common.h b/media/libeffects/lvm/lib/Common/lib/LVM_Common.h
index ceccd7b..49f16ad 100644
--- a/media/libeffects/lvm/lib/Common/lib/LVM_Common.h
+++ b/media/libeffects/lvm/lib/Common/lib/LVM_Common.h
@@ -23,15 +23,9 @@
/* */
/****************************************************************************************/
-
#ifndef __LVM_COMMON_H__
#define __LVM_COMMON_H__
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
-
/****************************************************************************************/
/* */
/* Includes */
@@ -39,7 +33,6 @@
/****************************************************************************************/
#include "LVM_Types.h"
-
/****************************************************************************************/
/* */
/* Definitions */
@@ -53,9 +46,5 @@
#define ALGORITHM_VC_ID 0x0500
#define ALGORITHM_TE_ID 0x0600
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
-
#endif /* __LVM_COMMON_H__ */
diff --git a/media/libeffects/lvm/lib/Common/lib/LVM_Macros.h b/media/libeffects/lvm/lib/Common/lib/LVM_Macros.h
index 97d13a5..1a15125 100644
--- a/media/libeffects/lvm/lib/Common/lib/LVM_Macros.h
+++ b/media/libeffects/lvm/lib/Common/lib/LVM_Macros.h
@@ -18,10 +18,6 @@
#ifndef _LVM_MACROS_H_
#define _LVM_MACROS_H_
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
/**********************************************************************************
MUL32x32INTO32(A,B,C,ShiftR)
C = (A * B) >> ShiftR
@@ -32,7 +28,6 @@
of overflow is undefined.
***********************************************************************************/
-#ifndef MUL32x32INTO32
#define MUL32x32INTO32(A,B,C,ShiftR) \
{LVM_INT32 MUL32x32INTO32_temp,MUL32x32INTO32_temp2,MUL32x32INTO32_mask,MUL32x32INTO32_HH,MUL32x32INTO32_HL,MUL32x32INTO32_LH,MUL32x32INTO32_LL;\
LVM_INT32 shiftValue;\
@@ -58,7 +53,6 @@
}\
(C) = MUL32x32INTO32_temp2;\
}
-#endif
/**********************************************************************************
MUL32x16INTO32(A,B,C,ShiftR)
@@ -71,7 +65,6 @@
of overflow is undefined.
***********************************************************************************/
-#ifndef MUL32x16INTO32
#define MUL32x16INTO32(A,B,C,ShiftR) \
{LVM_INT32 MUL32x16INTO32_mask,MUL32x16INTO32_HH,MUL32x16INTO32_LL;\
LVM_INT32 shiftValue;\
@@ -91,7 +84,6 @@
else {\
(C)=MUL32x16INTO32_HH>>(shiftValue-16);}\
}
-#endif
/**********************************************************************************
ADD2_SAT_32x32(A,B,C)
@@ -99,7 +91,6 @@
A,B and C are 32 bit SIGNED numbers.
***********************************************************************************/
-#ifndef ADD2_SAT_32x32
#define ADD2_SAT_32x32(A,B,C) \
{(C)=(A)+(B);\
if ((((C) ^ (A)) & ((C) ^ (B))) >> 31)\
@@ -110,12 +101,6 @@
(C)=0x7FFFFFFFl;\
}\
}
-#endif
-
-
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
#endif /* _LVM_MACROS_H_ */
diff --git a/media/libeffects/lvm/lib/Common/lib/LVM_Timer.h b/media/libeffects/lvm/lib/Common/lib/LVM_Timer.h
index a76354d..dbf9e6a 100644
--- a/media/libeffects/lvm/lib/Common/lib/LVM_Timer.h
+++ b/media/libeffects/lvm/lib/Common/lib/LVM_Timer.h
@@ -33,11 +33,6 @@
/* The timer currently does not suport changes in sampling rate while timing. */
/****************************************************************************************/
-
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
/****************************************************************************************/
/* TYPE DEFINITIONS */
/****************************************************************************************/
@@ -74,17 +69,11 @@
void LVM_Timer_Init ( LVM_Timer_Instance_t *pInstance,
LVM_Timer_Params_t *pParams );
-
void LVM_Timer ( LVM_Timer_Instance_t *pInstance,
LVM_INT16 BlockSize );
-
/****************************************************************************************/
/* END OF HEADER */
/****************************************************************************************/
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
-
#endif /* __LVM_TIMER_H__ */
diff --git a/media/libeffects/lvm/lib/Common/lib/LVM_Types.h b/media/libeffects/lvm/lib/Common/lib/LVM_Types.h
index fbfdd4d..8b687f6 100644
--- a/media/libeffects/lvm/lib/Common/lib/LVM_Types.h
+++ b/media/libeffects/lvm/lib/Common/lib/LVM_Types.h
@@ -25,10 +25,6 @@
#ifndef LVM_TYPES_H
#define LVM_TYPES_H
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
#include <stdint.h>
/****************************************************************************************/
@@ -96,32 +92,15 @@
typedef uint32_t LVM_UINT32; /* Unsigned 32-bit word */
typedef int64_t LVM_INT64; /* Signed 64-bit word */
-#ifdef BUILD_FLOAT
-
#define LVM_MAXFLOAT 1.f
typedef float LVM_FLOAT; /* single precision floating point */
-// If NATIVE_FLOAT_BUFFER is defined, we expose effects as floating point format;
-// otherwise we expose as integer 16 bit and translate to float for the effect libraries.
-// Hence, NATIVE_FLOAT_BUFFER should only be enabled under BUILD_FLOAT compilation.
-
-#define NATIVE_FLOAT_BUFFER
-
-#endif // BUILD_FLOAT
-
// Select whether we expose int16_t or float buffers.
-#ifdef NATIVE_FLOAT_BUFFER
#define EFFECT_BUFFER_FORMAT AUDIO_FORMAT_PCM_FLOAT
typedef float effect_buffer_t;
-#else // NATIVE_FLOAT_BUFFER
-
-#define EFFECT_BUFFER_FORMAT AUDIO_FORMAT_PCM_16_BIT
-typedef int16_t effect_buffer_t;
-
-#endif // NATIVE_FLOAT_BUFFER
#ifdef SUPPORT_MC
#define LVM_MAX_CHANNELS 8 // FCC_8
@@ -143,7 +122,6 @@
LVM_MODE_DUMMY = LVM_MAXENUM
} LVM_Mode_en;
-
/* Format */
typedef enum
{
@@ -156,7 +134,6 @@
LVM_SOURCE_DUMMY = LVM_MAXENUM
} LVM_Format_en;
-
/* LVM sampling rates */
typedef enum
{
@@ -169,17 +146,14 @@
LVM_FS_32000 = 6,
LVM_FS_44100 = 7,
LVM_FS_48000 = 8,
-#ifdef HIGHER_FS
LVM_FS_88200 = 9,
LVM_FS_96000 = 10,
LVM_FS_176400 = 11,
LVM_FS_192000 = 12,
-#endif
LVM_FS_INVALID = LVM_MAXENUM-1,
LVM_FS_DUMMY = LVM_MAXENUM
} LVM_Fs_en;
-
/* Memory Types */
typedef enum
{
@@ -190,7 +164,6 @@
LVM_MEMORYTYPE_DUMMY = LVM_MAXENUM
} LVM_MemoryTypes_en;
-
/* Memory region definition */
typedef struct
{
@@ -199,14 +172,12 @@
void *pBaseAddress; /* Pointer to the region base address */
} LVM_MemoryRegion_st;
-
/* Memory table containing the region definitions */
typedef struct
{
LVM_MemoryRegion_st Region[LVM_NR_MEMORY_REGIONS]; /* One definition for each region */
} LVM_MemoryTable_st;
-
/****************************************************************************************/
/* */
/* Standard Function Prototypes */
@@ -216,15 +187,10 @@
void *pGeneralPurpose, /* General purpose pointer (e.g. to a data structure needed in the callback) */
LVM_INT16 GeneralPurpose ); /* General purpose variable (e.g. to be used as callback ID) */
-
/****************************************************************************************/
/* */
/* End of file */
/* */
/****************************************************************************************/
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
-
#endif /* LVM_TYPES_H */
diff --git a/media/libeffects/lvm/lib/Common/lib/Mixer.h b/media/libeffects/lvm/lib/Common/lib/Mixer.h
index 07c53cd..b2e0195 100644
--- a/media/libeffects/lvm/lib/Common/lib/Mixer.h
+++ b/media/libeffects/lvm/lib/Common/lib/Mixer.h
@@ -18,19 +18,12 @@
#ifndef __MIXER_H__
#define __MIXER_H__
-
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
-
#include "LVM_Types.h"
/**********************************************************************************
INSTANCE MEMORY TYPE DEFINITION
***********************************************************************************/
-#ifdef BUILD_FLOAT /* BUILD_FLOAT*/
typedef struct
{
LVM_FLOAT Alpha; /* Time constant. Set by calling application. \
@@ -66,52 +59,11 @@
void *pGeneralPurpose2;
LVM_Callback pCallBack2;
} Mix_2St_Cll_FLOAT_t;
-#else
-typedef struct
-{
- LVM_INT32 Alpha; /* Time constant. Set by calling application. Can be changed at any time */
- LVM_INT32 Target; /* Target value. Set by calling application. Can be changed at any time */
- LVM_INT32 Current; /* Current value. Set by the mixer function. */
- LVM_INT16 CallbackSet; /* Boolean. Should be set by calling application each time the target value is updated */
- LVM_INT16 CallbackParam; /* Parameter that will be used in the calback function */
- void *pCallbackHandle; /* Pointer to the instance of the callback function */
- void *pGeneralPurpose; /* Pointer for general purpose usage */
- LVM_Callback pCallBack; /* Pointer to the callback function */
-} Mix_1St_Cll_t;
-
-typedef struct
-{
- LVM_INT32 Alpha1;
- LVM_INT32 Target1;
- LVM_INT32 Current1;
- LVM_INT16 CallbackSet1;
- LVM_INT16 CallbackParam1;
- void *pCallbackHandle1;
- void *pGeneralPurpose1;
- LVM_Callback pCallBack1;
-
- LVM_INT32 Alpha2; /* Warning the address of this location is passed as a pointer to Mix_1St_Cll_t in some functions */
- LVM_INT32 Target2;
- LVM_INT32 Current2;
- LVM_INT16 CallbackSet2;
- LVM_INT16 CallbackParam2;
- void *pCallbackHandle2;
- void *pGeneralPurpose2;
- LVM_Callback pCallBack2;
-
-} Mix_2St_Cll_t;
-
-#endif
/*** General functions ************************************************************/
-#ifdef BUILD_FLOAT
LVM_FLOAT LVM_Mixer_TimeConstant(LVM_UINT32 tc,
-#ifdef HIGHER_FS
LVM_UINT32 Fs,
-#else
- LVM_UINT16 Fs,
-#endif
LVM_UINT16 NumChannels);
void MixSoft_1St_D32C31_WRA( Mix_1St_Cll_FLOAT_t *pInstance,
@@ -129,34 +81,10 @@
const LVM_FLOAT *src,
LVM_FLOAT *dst,
LVM_INT16 n);
-#else
-LVM_UINT32 LVM_Mixer_TimeConstant(LVM_UINT32 tc,
- LVM_UINT16 Fs,
- LVM_UINT16 NumChannels);
-
-
-void MixSoft_1St_D32C31_WRA( Mix_1St_Cll_t *pInstance,
- const LVM_INT32 *src,
- LVM_INT32 *dst,
- LVM_INT16 n);
-
-void MixSoft_2St_D32C31_SAT( Mix_2St_Cll_t *pInstance,
- const LVM_INT32 *src1,
- const LVM_INT32 *src2,
- LVM_INT32 *dst,
- LVM_INT16 n);
-
-void MixInSoft_D32C31_SAT( Mix_1St_Cll_t *pInstance,
- const LVM_INT32 *src,
- LVM_INT32 *dst,
- LVM_INT16 n);
-
-#endif
/**********************************************************************************
FUNCTION PROTOTYPES (LOW LEVEL SUBFUNCTIONS)
***********************************************************************************/
-#ifdef BUILD_FLOAT
void Core_MixSoft_1St_D32C31_WRA( Mix_1St_Cll_FLOAT_t *pInstance,
const LVM_FLOAT *src,
LVM_FLOAT *dst,
@@ -170,27 +98,6 @@
const LVM_FLOAT *src,
LVM_FLOAT *dst,
LVM_INT16 n);
-#else
-void Core_MixSoft_1St_D32C31_WRA( Mix_1St_Cll_t *pInstance,
- const LVM_INT32 *src,
- LVM_INT32 *dst,
- LVM_INT16 n);
-
-void Core_MixHard_2St_D32C31_SAT( Mix_2St_Cll_t *pInstance,
- const LVM_INT32 *src1,
- const LVM_INT32 *src2,
- LVM_INT32 *dst,
- LVM_INT16 n);
-
-void Core_MixInSoft_D32C31_SAT( Mix_1St_Cll_t *pInstance,
- const LVM_INT32 *src,
- LVM_INT32 *dst,
- LVM_INT16 n);
-#endif
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
-
/**********************************************************************************/
diff --git a/media/libeffects/lvm/lib/Common/lib/ScalarArithmetic.h b/media/libeffects/lvm/lib/Common/lib/ScalarArithmetic.h
index cdb3837..ae54419 100644
--- a/media/libeffects/lvm/lib/Common/lib/ScalarArithmetic.h
+++ b/media/libeffects/lvm/lib/Common/lib/ScalarArithmetic.h
@@ -18,11 +18,6 @@
#ifndef __SCALARARITHMETIC_H__
#define __SCALARARITHMETIC_H__
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
-
/*######################################################################################*/
/* Include files */
/*######################################################################################*/
@@ -35,11 +30,7 @@
/* Absolute value including the corner case for the extreme negative value */
-#ifdef BUILD_FLOAT
LVM_FLOAT Abs_Float(LVM_FLOAT input);
-#else
-LVM_INT32 Abs_32(LVM_INT32 input);
-#endif
/****************************************************************************************
* Name : dB_to_Lin32()
@@ -53,16 +44,7 @@
* (15->01) = decimal part
* Returns : Lin value format 1.16.15
****************************************************************************************/
-#ifdef BUILD_FLOAT
LVM_FLOAT dB_to_LinFloat(LVM_INT16 db_fix);
-#else
-LVM_INT32 dB_to_Lin32(LVM_INT16 db_fix);
-#endif
-
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
#endif /* __SCALARARITHMETIC_H__ */
-
diff --git a/media/libeffects/lvm/lib/Common/lib/VectorArithmetic.h b/media/libeffects/lvm/lib/Common/lib/VectorArithmetic.h
index 7468a90..2af1eeb 100644
--- a/media/libeffects/lvm/lib/Common/lib/VectorArithmetic.h
+++ b/media/libeffects/lvm/lib/Common/lib/VectorArithmetic.h
@@ -18,32 +18,16 @@
#ifndef _VECTOR_ARITHMETIC_H_
#define _VECTOR_ARITHMETIC_H_
-
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
#include "LVM_Types.h"
/**********************************************************************************
VARIOUS FUNCTIONS
***********************************************************************************/
-#ifdef BUILD_FLOAT
void LoadConst_Float( const LVM_FLOAT val,
LVM_FLOAT *dst,
LVM_INT16 n );
-#else
-void LoadConst_16( const LVM_INT16 val,
- LVM_INT16 *dst,
- LVM_INT16 n );
-void LoadConst_32( const LVM_INT32 val,
- LVM_INT32 *dst,
- LVM_INT16 n );
-#endif
-
-#ifdef BUILD_FLOAT
void Copy_Float( const LVM_FLOAT *src,
LVM_FLOAT *dst,
LVM_INT16 n );
@@ -57,11 +41,6 @@
LVM_INT16 NrFrames,
LVM_INT32 NrChannels);
#endif
-#else
-void Copy_16( const LVM_INT16 *src,
- LVM_INT16 *dst,
- LVM_INT16 n );
-#endif
/*********************************************************************************
* note: In Mult3s_16x16() saturation of result is not taken care when *
@@ -71,17 +50,10 @@
* This is the only case which will give wrong result. *
* For more information refer to Vector_Arithmetic.doc in /doc folder *
*********************************************************************************/
-#ifdef BUILD_FLOAT
void Mult3s_Float( const LVM_FLOAT *src,
const LVM_FLOAT val,
LVM_FLOAT *dst,
LVM_INT16 n);
-#else
-void Mult3s_16x16( const LVM_INT16 *src,
- const LVM_INT16 val,
- LVM_INT16 *dst,
- LVM_INT16 n);
-#endif
/*********************************************************************************
* note: In Mult3s_32x16() saturation of result is not taken care when *
@@ -95,55 +67,24 @@
const LVM_INT16 val,
LVM_INT32 *dst,
LVM_INT16 n);
-#ifdef BUILD_FLOAT
void DelayMix_Float(const LVM_FLOAT *src, /* Source 1, to be delayed */
LVM_FLOAT *delay, /* Delay buffer */
LVM_INT16 size, /* Delay size */
LVM_FLOAT *dst, /* Source/destination */
LVM_INT16 *pOffset, /* Delay offset */
LVM_INT16 n) ; /* Number of stereo samples */
-#else
-void DelayMix_16x16( const LVM_INT16 *src,
- LVM_INT16 *delay,
- LVM_INT16 size,
- LVM_INT16 *dst,
- LVM_INT16 *pOffset,
- LVM_INT16 n);
-#endif
void DelayWrite_32( const LVM_INT32 *src, /* Source 1, to be delayed */
LVM_INT32 *delay, /* Delay buffer */
LVM_UINT16 size, /* Delay size */
LVM_UINT16 *pOffset, /* Delay offset */
LVM_INT16 n);
-#ifdef BUILD_FLOAT
void Add2_Sat_Float( const LVM_FLOAT *src,
LVM_FLOAT *dst,
LVM_INT16 n );
-#else
-void Add2_Sat_16x16( const LVM_INT16 *src,
- LVM_INT16 *dst,
- LVM_INT16 n );
-
-void Add2_Sat_32x32( const LVM_INT32 *src,
- LVM_INT32 *dst,
- LVM_INT16 n );
-#endif
-#ifdef BUILD_FLOAT
void Mac3s_Sat_Float( const LVM_FLOAT *src,
const LVM_FLOAT val,
LVM_FLOAT *dst,
LVM_INT16 n);
-#else
-void Mac3s_Sat_16x16( const LVM_INT16 *src,
- const LVM_INT16 val,
- LVM_INT16 *dst,
- LVM_INT16 n);
-
-void Mac3s_Sat_32x16( const LVM_INT32 *src,
- const LVM_INT16 val,
- LVM_INT32 *dst,
- LVM_INT16 n);
-#endif
void DelayAllPass_Sat_32x16To32( LVM_INT32 *delay, /* Delay buffer */
LVM_UINT16 size, /* Delay size */
LVM_INT16 coeff, /* All pass filter coefficient */
@@ -155,39 +96,16 @@
/**********************************************************************************
SHIFT FUNCTIONS
***********************************************************************************/
-#ifdef BUILD_FLOAT
void Shift_Sat_Float (const LVM_INT16 val,
const LVM_FLOAT *src,
LVM_FLOAT *dst,
LVM_INT16 n);
-#else
-void Shift_Sat_v16xv16 ( const LVM_INT16 val,
- const LVM_INT16 *src,
- LVM_INT16 *dst,
- LVM_INT16 n);
-
-void Shift_Sat_v32xv32 ( const LVM_INT16 val,
- const LVM_INT32 *src,
- LVM_INT32 *dst,
- LVM_INT16 n);
-#endif
/**********************************************************************************
AUDIO FORMAT CONVERSION FUNCTIONS
***********************************************************************************/
-#ifdef BUILD_FLOAT
void MonoTo2I_Float( const LVM_FLOAT *src,
LVM_FLOAT *dst,
LVM_INT16 n);
-#else
-void MonoTo2I_16( const LVM_INT16 *src,
- LVM_INT16 *dst,
- LVM_INT16 n);
-
-void MonoTo2I_32( const LVM_INT32 *src,
- LVM_INT32 *dst,
- LVM_INT16 n);
-#endif
-#ifdef BUILD_FLOAT
void From2iToMono_Float( const LVM_FLOAT *src,
LVM_FLOAT *dst,
LVM_INT16 n);
@@ -197,47 +115,18 @@
LVM_INT16 NrFrames,
LVM_INT16 NrChannels);
#endif
-#else
-void From2iToMono_32( const LVM_INT32 *src,
- LVM_INT32 *dst,
- LVM_INT16 n);
-#endif
-#ifdef BUILD_FLOAT
void MSTo2i_Sat_Float( const LVM_FLOAT *srcM,
const LVM_FLOAT *srcS,
LVM_FLOAT *dst,
LVM_INT16 n );
-#else
-void MSTo2i_Sat_16x16( const LVM_INT16 *srcM,
- const LVM_INT16 *srcS,
- LVM_INT16 *dst,
- LVM_INT16 n );
-#endif
-#ifdef BUILD_FLOAT
void From2iToMS_Float( const LVM_FLOAT *src,
LVM_FLOAT *dstM,
LVM_FLOAT *dstS,
LVM_INT16 n );
-#else
-void From2iToMS_16x16( const LVM_INT16 *src,
- LVM_INT16 *dstM,
- LVM_INT16 *dstS,
- LVM_INT16 n );
-#endif
-#ifdef BUILD_FLOAT
void JoinTo2i_Float( const LVM_FLOAT *srcL,
const LVM_FLOAT *srcR,
LVM_FLOAT *dst,
LVM_INT16 n );
-#else
-void From2iToMono_16( const LVM_INT16 *src,
- LVM_INT16 *dst,
- LVM_INT16 n);
-void JoinTo2i_32x32( const LVM_INT32 *srcL,
- const LVM_INT32 *srcR,
- LVM_INT32 *dst,
- LVM_INT16 n );
-#endif
/**********************************************************************************
DATA TYPE CONVERSION FUNCTIONS
@@ -253,11 +142,6 @@
LVM_INT16 n,
LVM_INT16 shift );
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
-
-
/**********************************************************************************/
#endif /* _VECTOR_ARITHMETIC_H_ */
diff --git a/media/libeffects/lvm/lib/Common/src/AGC_MIX_VOL_2St1Mon_D32_WRA.c b/media/libeffects/lvm/lib/Common/src/AGC_MIX_VOL_2St1Mon_D32_WRA.cpp
similarity index 74%
rename from media/libeffects/lvm/lib/Common/src/AGC_MIX_VOL_2St1Mon_D32_WRA.c
rename to media/libeffects/lvm/lib/Common/src/AGC_MIX_VOL_2St1Mon_D32_WRA.cpp
index 5c8655f..e18aa78 100644
--- a/media/libeffects/lvm/lib/Common/src/AGC_MIX_VOL_2St1Mon_D32_WRA.c
+++ b/media/libeffects/lvm/lib/Common/src/AGC_MIX_VOL_2St1Mon_D32_WRA.cpp
@@ -24,7 +24,6 @@
#include "AGC.h"
#include "ScalarArithmetic.h"
-
/****************************************************************************************/
/* */
/* Defines */
@@ -33,10 +32,8 @@
#define VOL_TC_SHIFT 21 /* As a power of 2 */
#define DECAY_SHIFT 10 /* As a power of 2 */
-#ifdef BUILD_FLOAT
#define VOL_TC_FLOAT 2.0f /* As a power of 2 */
#define DECAY_FAC_FLOAT 64.0f /* As a power of 2 */
-#endif
/****************************************************************************************/
/* */
@@ -72,131 +69,6 @@
/* NOTES: */
/* */
/****************************************************************************************/
-#ifndef BUILD_FLOAT
-void AGC_MIX_VOL_2St1Mon_D32_WRA(AGC_MIX_VOL_2St1Mon_D32_t *pInstance, /* Instance pointer */
- const LVM_INT32 *pStSrc, /* Stereo source */
- const LVM_INT32 *pMonoSrc, /* Mono source */
- LVM_INT32 *pDst, /* Stereo destination */
- LVM_UINT16 NumSamples) /* Number of samples */
-{
-
- /*
- * General variables
- */
- LVM_UINT16 i; /* Sample index */
- LVM_INT32 Left; /* Left sample */
- LVM_INT32 Right; /* Right sample */
- LVM_INT32 Mono; /* Mono sample */
- LVM_INT32 AbsPeak; /* Absolute peak signal */
- LVM_INT32 HighWord; /* High word in intermediate calculations */
- LVM_INT32 LowWord; /* Low word in intermediate calculations */
- LVM_INT16 AGC_Mult; /* Short AGC gain */
- LVM_INT16 Vol_Mult; /* Short volume */
-
-
- /*
- * Instance control variables
- */
- LVM_INT32 AGC_Gain = pInstance->AGC_Gain; /* Get the current AGC gain */
- LVM_INT32 AGC_MaxGain = pInstance->AGC_MaxGain; /* Get maximum AGC gain */
- LVM_INT16 AGC_GainShift = pInstance->AGC_GainShift; /* Get the AGC shift */
- LVM_INT16 AGC_Attack = pInstance->AGC_Attack; /* Attack scaler */
- LVM_INT16 AGC_Decay = pInstance->AGC_Decay; /* Decay scaler */
- LVM_INT32 AGC_Target = pInstance->AGC_Target; /* Get the target level */
- LVM_INT32 Vol_Current = pInstance->Volume; /* Actual volume setting */
- LVM_INT32 Vol_Target = pInstance->Target; /* Target volume setting */
- LVM_INT16 Vol_Shift = pInstance->VolumeShift; /* Volume shift scaling */
- LVM_INT16 Vol_TC = pInstance->VolumeTC; /* Time constant */
-
-
- /*
- * Process on a sample by sample basis
- */
- for (i=0;i<NumSamples;i++) /* For each sample */
- {
-
- /*
- * Get the short scalers
- */
- AGC_Mult = (LVM_INT16)(AGC_Gain >> 16); /* Get the short AGC gain */
- Vol_Mult = (LVM_INT16)(Vol_Current >> 16); /* Get the short volume gain */
-
-
- /*
- * Get the input samples
- */
- Left = *pStSrc++; /* Get the left sample */
- Right = *pStSrc++; /* Get the right sample */
- Mono = *pMonoSrc++; /* Get the mono sample */
-
-
- /*
- * Apply the AGC gain to the mono input and mix with the stereo signal
- */
- HighWord = (AGC_Mult * (Mono >> 16)); /* signed long (Mono) by unsigned short (AGC_Mult) multiply */
- LowWord = (AGC_Mult * (Mono & 0xffff));
- Mono = (HighWord + (LowWord >> 16)) << (AGC_GainShift);
- Left += Mono; /* Mix in the mono signal */
- Right += Mono;
-
-
- /*
- * Apply the volume and write to the output stream
- */
- HighWord = (Vol_Mult * (Left >> 16)); /* signed long (Left) by unsigned short (Vol_Mult) multiply */
- LowWord = (Vol_Mult * (Left & 0xffff));
- Left = (HighWord + (LowWord >> 16)) << (Vol_Shift);
- HighWord = (Vol_Mult * (Right >> 16)); /* signed long (Right) by unsigned short (Vol_Mult) multiply */
- LowWord = (Vol_Mult * (Right & 0xffff));
- Right = (HighWord + (LowWord >> 16)) << (Vol_Shift);
- *pDst++ = Left; /* Save the results */
- *pDst++ = Right;
-
-
- /*
- * Update the AGC gain
- */
- AbsPeak = (Abs_32(Left)>Abs_32(Right)) ? Abs_32(Left) : Abs_32(Right); /* Get the absolute peak */
- if (AbsPeak > AGC_Target)
- {
- /*
- * The signal is too large so decrease the gain
- */
- HighWord = (AGC_Attack * (AGC_Gain >> 16)); /* signed long (AGC_Gain) by unsigned short (AGC_Attack) multiply */
- LowWord = (AGC_Attack * (AGC_Gain & 0xffff));
- AGC_Gain = (HighWord + (LowWord >> 16)) << 1;
- }
- else
- {
- /*
- * The signal is too small so increase the gain
- */
- if (AGC_Gain > AGC_MaxGain)
- {
- AGC_Gain -= (AGC_Decay << DECAY_SHIFT);
- }
- else
- {
- AGC_Gain += (AGC_Decay << DECAY_SHIFT);
- }
- }
-
- /*
- * Update the gain
- */
- Vol_Current += Vol_TC * ((Vol_Target - Vol_Current) >> VOL_TC_SHIFT);
- }
-
-
- /*
- * Update the parameters
- */
- pInstance->Volume = Vol_Current; /* Actual volume setting */
- pInstance->AGC_Gain = AGC_Gain;
-
- return;
-}
-#else
void AGC_MIX_VOL_2St1Mon_D32_WRA(AGC_MIX_VOL_2St1Mon_FLOAT_t *pInstance, /* Instance pointer */
const LVM_FLOAT *pStSrc, /* Stereo source */
const LVM_FLOAT *pMonoSrc, /* Mono source */
@@ -215,7 +87,6 @@
LVM_FLOAT AGC_Mult; /* Short AGC gain */
LVM_FLOAT Vol_Mult; /* Short volume */
-
/*
* Instance control variables
*/
@@ -228,7 +99,6 @@
LVM_FLOAT Vol_Target = pInstance->Target; /* Target volume setting */
LVM_FLOAT Vol_TC = pInstance->VolumeTC; /* Time constant */
-
/*
* Process on a sample by sample basis
*/
@@ -241,7 +111,6 @@
AGC_Mult = (LVM_FLOAT)(AGC_Gain); /* Get the short AGC gain */
Vol_Mult = (LVM_FLOAT)(Vol_Current); /* Get the short volume gain */
-
/*
* Get the input samples
*/
@@ -249,7 +118,6 @@
Right = *pStSrc++; /* Get the right sample */
Mono = *pMonoSrc++; /* Get the mono sample */
-
/*
* Apply the AGC gain to the mono input and mix with the stereo signal
*/
@@ -296,7 +164,6 @@
Vol_Current += (Vol_Target - Vol_Current) * ((LVM_FLOAT)Vol_TC / VOL_TC_FLOAT);
}
-
/*
* Update the parameters
*/
@@ -360,7 +227,6 @@
LVM_FLOAT AGC_Mult; /* Short AGC gain */
LVM_FLOAT Vol_Mult; /* Short volume */
-
/*
* Instance control variables
*/
@@ -374,7 +240,6 @@
LVM_FLOAT Vol_Target = pInstance->Target; /* Target volume setting */
LVM_FLOAT Vol_TC = pInstance->VolumeTC; /* Time constant */
-
/*
* Process on a sample by sample basis
*/
@@ -441,7 +306,6 @@
Vol_Current += (Vol_Target - Vol_Current) * ((LVM_FLOAT)Vol_TC / VOL_TC_FLOAT);
}
-
/*
* Update the parameters
*/
@@ -451,4 +315,3 @@
return;
}
#endif /*SUPPORT_MC*/
-#endif /*BUILD_FLOAT*/
diff --git a/media/libeffects/lvm/lib/Common/src/Abs_32.c b/media/libeffects/lvm/lib/Common/src/Abs_32.cpp
similarity index 98%
rename from media/libeffects/lvm/lib/Common/src/Abs_32.c
rename to media/libeffects/lvm/lib/Common/src/Abs_32.cpp
index 84fabd8..e013809 100644
--- a/media/libeffects/lvm/lib/Common/src/Abs_32.c
+++ b/media/libeffects/lvm/lib/Common/src/Abs_32.cpp
@@ -47,7 +47,6 @@
}
return input;
}
-#ifdef BUILD_FLOAT
LVM_FLOAT Abs_Float(LVM_FLOAT input)
{
if(input < 0)
@@ -57,4 +56,3 @@
}
return input;
}
-#endif
diff --git a/media/libeffects/lvm/lib/Common/src/Add2_Sat_16x16.c b/media/libeffects/lvm/lib/Common/src/Add2_Sat_16x16.cpp
similarity index 100%
rename from media/libeffects/lvm/lib/Common/src/Add2_Sat_16x16.c
rename to media/libeffects/lvm/lib/Common/src/Add2_Sat_16x16.cpp
diff --git a/media/libeffects/lvm/lib/Common/src/Add2_Sat_32x32.c b/media/libeffects/lvm/lib/Common/src/Add2_Sat_32x32.cpp
similarity index 98%
rename from media/libeffects/lvm/lib/Common/src/Add2_Sat_32x32.c
rename to media/libeffects/lvm/lib/Common/src/Add2_Sat_32x32.cpp
index 66d6adb..a48e668 100644
--- a/media/libeffects/lvm/lib/Common/src/Add2_Sat_32x32.c
+++ b/media/libeffects/lvm/lib/Common/src/Add2_Sat_32x32.cpp
@@ -21,7 +21,6 @@
#include "VectorArithmetic.h"
-
/**********************************************************************************
FUNCTION ADD2_SAT_32X32
***********************************************************************************/
@@ -57,7 +56,6 @@
return;
}
-#ifdef BUILD_FLOAT
void Add2_Sat_Float( const LVM_FLOAT *src,
LVM_FLOAT *dst,
LVM_INT16 n )
@@ -85,5 +83,4 @@
}
return;
}
-#endif
/**********************************************************************************/
diff --git a/media/libeffects/lvm/lib/Common/src/BP_1I_D16F16C14_TRC_WRA_01.c b/media/libeffects/lvm/lib/Common/src/BP_1I_D16F16C14_TRC_WRA_01.cpp
similarity index 60%
rename from media/libeffects/lvm/lib/Common/src/BP_1I_D16F16C14_TRC_WRA_01.c
rename to media/libeffects/lvm/lib/Common/src/BP_1I_D16F16C14_TRC_WRA_01.cpp
index 88f9986..1a5e07f 100644
--- a/media/libeffects/lvm/lib/Common/src/BP_1I_D16F16C14_TRC_WRA_01.c
+++ b/media/libeffects/lvm/lib/Common/src/BP_1I_D16F16C14_TRC_WRA_01.cpp
@@ -19,7 +19,6 @@
#include "BP_1I_D16F16Css_TRC_WRA_01_Private.h"
#include "LVM_Macros.h"
-
/**************************************************************************
ASSUMPTIONS:
COEFS-
@@ -33,13 +32,11 @@
pBiquadState->pDelays[2] is y(n-1)L in Q0 format
pBiquadState->pDelays[3] is y(n-2)L in Q0 format
***************************************************************************/
-#ifdef BUILD_FLOAT
void BP_1I_D16F16C14_TRC_WRA_01 ( Biquad_FLOAT_Instance_t *pInstance,
LVM_FLOAT *pDataIn,
LVM_FLOAT *pDataOut,
LVM_INT16 NrSamples)
-
{
LVM_FLOAT ynL;
LVM_INT16 ii;
@@ -48,7 +45,6 @@
for (ii = NrSamples; ii != 0; ii--)
{
-
/**************************************************************************
PROCESSING OF THE LEFT CHANNEL
***************************************************************************/
@@ -77,50 +73,4 @@
}
}
-#else
-void BP_1I_D16F16C14_TRC_WRA_01 ( Biquad_Instance_t *pInstance,
- LVM_INT16 *pDataIn,
- LVM_INT16 *pDataOut,
- LVM_INT16 NrSamples)
-
-
- {
- LVM_INT32 ynL;
- LVM_INT16 ii;
- PFilter_State pBiquadState = (PFilter_State) pInstance;
-
- for (ii = NrSamples; ii != 0; ii--)
- {
-
-
- /**************************************************************************
- PROCESSING OF THE LEFT CHANNEL
- ***************************************************************************/
- // ynL= (A0 (Q14) * (x(n)L (Q0) - x(n-2)L (Q0) ) ) in Q14
- ynL=(LVM_INT32)pBiquadState->coefs[0]* ((*pDataIn)-pBiquadState->pDelays[1]);
-
- // ynL+= ((-B2 (Q14) * y(n-2)L (Q0) ) ) in Q14
- ynL+=(LVM_INT32)pBiquadState->coefs[1]*pBiquadState->pDelays[3];
-
- // ynL+= ((-B1 (Q30) * y(n-1)L (Q0) ) ) in Q14
- ynL+=(LVM_INT32)pBiquadState->coefs[2]*pBiquadState->pDelays[2];
-
- ynL=(LVM_INT16)(ynL>>14); // ynL in Q0
- /**************************************************************************
- UPDATING THE DELAYS
- ***************************************************************************/
- pBiquadState->pDelays[3]=pBiquadState->pDelays[2]; // y(n-2)L=y(n-1)L
- pBiquadState->pDelays[1]=pBiquadState->pDelays[0]; // x(n-2)L=x(n-1)L
- pBiquadState->pDelays[2]=ynL; // Update y(n-1)L in Q0
- pBiquadState->pDelays[0]=(*pDataIn++); // Update x(n-1)L in Q0
-
- /**************************************************************************
- WRITING THE OUTPUT
- ***************************************************************************/
- *pDataOut++=(LVM_INT16)ynL; // Write Left output in Q0
-
- }
-
- }
-#endif
diff --git a/media/libeffects/lvm/lib/Common/src/BP_1I_D16F16Css_TRC_WRA_01_Init.c b/media/libeffects/lvm/lib/Common/src/BP_1I_D16F16Css_TRC_WRA_01_Init.cpp
similarity index 85%
rename from media/libeffects/lvm/lib/Common/src/BP_1I_D16F16Css_TRC_WRA_01_Init.c
rename to media/libeffects/lvm/lib/Common/src/BP_1I_D16F16Css_TRC_WRA_01_Init.cpp
index 27ab57a..60b6c16 100644
--- a/media/libeffects/lvm/lib/Common/src/BP_1I_D16F16Css_TRC_WRA_01_Init.c
+++ b/media/libeffects/lvm/lib/Common/src/BP_1I_D16F16Css_TRC_WRA_01_Init.cpp
@@ -19,7 +19,6 @@
#include "BIQUAD.h"
#include "BP_1I_D16F16Css_TRC_WRA_01_Private.h"
-
/*-------------------------------------------------------------------------*/
/* FUNCTION: */
/* BP_1I_D16F16Css_TRC_WRA_01_Init */
@@ -38,7 +37,6 @@
/* RETURNS: */
/* void return code */
/*-------------------------------------------------------------------------*/
-#ifdef BUILD_FLOAT
void BP_1I_D16F16Css_TRC_WRA_01_Init ( Biquad_FLOAT_Instance_t *pInstance,
Biquad_1I_Order2_FLOAT_Taps_t *pTaps,
BP_FLOAT_Coefs_t *pCoef)
@@ -50,19 +48,6 @@
pBiquadState->coefs[1] = pCoef->B2;
pBiquadState->coefs[2] = pCoef->B1;
}
-#else
-void BP_1I_D16F16Css_TRC_WRA_01_Init ( Biquad_Instance_t *pInstance,
- Biquad_1I_Order2_Taps_t *pTaps,
- BP_C16_Coefs_t *pCoef)
-{
- PFilter_State pBiquadState = (PFilter_State) pInstance;
- pBiquadState->pDelays =(LVM_INT32 *) pTaps;
-
- pBiquadState->coefs[0]=pCoef->A0;
- pBiquadState->coefs[1]=pCoef->B2;
- pBiquadState->coefs[2]=pCoef->B1;
- }
-#endif
/*-------------------------------------------------------------------------*/
/* End Of File: BP_1I_D16F16Css_TRC_WRA_01_Init.c */
diff --git a/media/libeffects/lvm/lib/Common/src/BP_1I_D16F16Css_TRC_WRA_01_Private.h b/media/libeffects/lvm/lib/Common/src/BP_1I_D16F16Css_TRC_WRA_01_Private.h
index e194f92..8a000b6 100644
--- a/media/libeffects/lvm/lib/Common/src/BP_1I_D16F16Css_TRC_WRA_01_Private.h
+++ b/media/libeffects/lvm/lib/Common/src/BP_1I_D16F16Css_TRC_WRA_01_Private.h
@@ -27,7 +27,6 @@
typedef Filter_State * PFilter_State ;
-#ifdef BUILD_FLOAT
typedef struct _Filter_State_FLOAT
{
@@ -35,5 +34,4 @@
LVM_FLOAT coefs[3]; /* pointer to the filter coefficients */
}Filter_State_FLOAT;
typedef Filter_State_FLOAT * PFilter_State_FLOAT ;
-#endif
#endif /*_BP_1I_D16F16CSS_TRC_WRA_01_PRIVATE_H_*/
diff --git a/media/libeffects/lvm/lib/Common/src/BP_1I_D16F32C30_TRC_WRA_01.c b/media/libeffects/lvm/lib/Common/src/BP_1I_D16F32C30_TRC_WRA_01.c
deleted file mode 100644
index 3abdd43..0000000
--- a/media/libeffects/lvm/lib/Common/src/BP_1I_D16F32C30_TRC_WRA_01.c
+++ /dev/null
@@ -1,124 +0,0 @@
-/*
- * Copyright (C) 2004-2010 NXP Software
- * Copyright (C) 2010 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include "BIQUAD.h"
-#include "BP_1I_D16F32Cll_TRC_WRA_01_Private.h"
-#include "LVM_Macros.h"
-
-
-/**************************************************************************
- ASSUMPTIONS:
- COEFS-
- pBiquadState->coefs[0] is A0,
- pBiquadState->coefs[1] is -B2,
- pBiquadState->coefs[2] is -B1, these are in Q30 format
-
- DELAYS-
- pBiquadState->pDelays[0] is x(n-1)L in Q0 format
- pBiquadState->pDelays[1] is x(n-2)L in Q0 format
- pBiquadState->pDelays[2] is y(n-1)L in Q16 format
- pBiquadState->pDelays[3] is y(n-2)L in Q16 format
-***************************************************************************/
-#ifdef BUILD_FLOAT
-void BP_1I_D16F32C30_TRC_WRA_01 ( Biquad_FLOAT_Instance_t *pInstance,
- LVM_FLOAT *pDataIn,
- LVM_FLOAT *pDataOut,
- LVM_INT16 NrSamples)
-{
- LVM_FLOAT ynL,templ;
- LVM_INT16 ii;
- PFilter_State_FLOAT pBiquadState = (PFilter_State_FLOAT)pInstance;
-
- for (ii = NrSamples; ii != 0; ii--)
- {
- /**************************************************************************
- PROCESSING OF THE LEFT CHANNEL
- ***************************************************************************/
- // ynL= (A0 * (x(n)L - x(n-2)L ))
- templ = (LVM_FLOAT) *pDataIn - pBiquadState->pDelays[1];
- ynL = pBiquadState->coefs[0] * templ;
-
- // ynL+= ((-B2 * y(n-2)L ) )
- templ = pBiquadState->coefs[1] * pBiquadState->pDelays[3];
- ynL += templ;
-
- // ynL+= ((-B1 * y(n-1)L ))
- templ = pBiquadState->coefs[2] * pBiquadState->pDelays[2];
- ynL += templ;
-
- /**************************************************************************
- UPDATING THE DELAYS
- ***************************************************************************/
- pBiquadState->pDelays[3] = pBiquadState->pDelays[2]; // y(n-2)L=y(n-1)L
- pBiquadState->pDelays[1] = pBiquadState->pDelays[0]; // x(n-2)L=x(n-1)L
- pBiquadState->pDelays[2] = ynL; // Update y(n-1)L in Q16
- pBiquadState->pDelays[0] = (*pDataIn++); // Update x(n-1)L in Q0
-
- /**************************************************************************
- WRITING THE OUTPUT
- ***************************************************************************/
- *pDataOut++ = (ynL); // Write Left output
- }
-}
-#else
-void BP_1I_D16F32C30_TRC_WRA_01 ( Biquad_Instance_t *pInstance,
- LVM_INT16 *pDataIn,
- LVM_INT16 *pDataOut,
- LVM_INT16 NrSamples)
-
-
- {
- LVM_INT32 ynL,templ;
- LVM_INT16 ii;
- PFilter_State pBiquadState = (PFilter_State) pInstance;
-
- for (ii = NrSamples; ii != 0; ii--)
- {
-
-
- /**************************************************************************
- PROCESSING OF THE LEFT CHANNEL
- ***************************************************************************/
- // ynL= (A0 (Q30) * (x(n)L (Q0) - x(n-2)L (Q0) ) >>14) in Q16
- templ= (LVM_INT32) *pDataIn-pBiquadState->pDelays[1];
- MUL32x32INTO32(pBiquadState->coefs[0],templ,ynL,14)
-
- // ynL+= ((-B2 (Q30) * y(n-2)L (Q16) ) >>30) in Q16
- MUL32x32INTO32(pBiquadState->coefs[1],pBiquadState->pDelays[3],templ,30)
- ynL+=templ;
-
- // ynL+= ((-B1 (Q30) * y(n-1)L (Q16) ) >>30) in Q16
- MUL32x32INTO32(pBiquadState->coefs[2],pBiquadState->pDelays[2],templ,30)
- ynL+=templ;
-
- /**************************************************************************
- UPDATING THE DELAYS
- ***************************************************************************/
- pBiquadState->pDelays[3]=pBiquadState->pDelays[2]; // y(n-2)L=y(n-1)L
- pBiquadState->pDelays[1]=pBiquadState->pDelays[0]; // x(n-2)L=x(n-1)L
- pBiquadState->pDelays[2]=ynL; // Update y(n-1)L in Q16
- pBiquadState->pDelays[0]=(*pDataIn++); // Update x(n-1)L in Q0
-
- /**************************************************************************
- WRITING THE OUTPUT
- ***************************************************************************/
- *pDataOut++=(LVM_INT16)(ynL>>16); // Write Left output in Q0
-
- }
-
- }
-#endif
diff --git a/media/libeffects/lvm/lib/Common/src/BP_1I_D16F32C30_TRC_WRA_01.cpp b/media/libeffects/lvm/lib/Common/src/BP_1I_D16F32C30_TRC_WRA_01.cpp
new file mode 100644
index 0000000..c844d03
--- /dev/null
+++ b/media/libeffects/lvm/lib/Common/src/BP_1I_D16F32C30_TRC_WRA_01.cpp
@@ -0,0 +1,74 @@
+/*
+ * Copyright (C) 2004-2010 NXP Software
+ * Copyright (C) 2010 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include "BIQUAD.h"
+#include "BP_1I_D16F32Cll_TRC_WRA_01_Private.h"
+#include "LVM_Macros.h"
+
+/**************************************************************************
+ ASSUMPTIONS:
+ COEFS-
+ pBiquadState->coefs[0] is A0,
+ pBiquadState->coefs[1] is -B2,
+ pBiquadState->coefs[2] is -B1, these are in Q30 format
+
+ DELAYS-
+ pBiquadState->pDelays[0] is x(n-1)L in Q0 format
+ pBiquadState->pDelays[1] is x(n-2)L in Q0 format
+ pBiquadState->pDelays[2] is y(n-1)L in Q16 format
+ pBiquadState->pDelays[3] is y(n-2)L in Q16 format
+***************************************************************************/
+void BP_1I_D16F32C30_TRC_WRA_01 ( Biquad_FLOAT_Instance_t *pInstance,
+ LVM_FLOAT *pDataIn,
+ LVM_FLOAT *pDataOut,
+ LVM_INT16 NrSamples)
+{
+ LVM_FLOAT ynL,templ;
+ LVM_INT16 ii;
+ PFilter_State_FLOAT pBiquadState = (PFilter_State_FLOAT)pInstance;
+
+ for (ii = NrSamples; ii != 0; ii--)
+ {
+ /**************************************************************************
+ PROCESSING OF THE LEFT CHANNEL
+ ***************************************************************************/
+ // ynL= (A0 * (x(n)L - x(n-2)L ))
+ templ = (LVM_FLOAT) *pDataIn - pBiquadState->pDelays[1];
+ ynL = pBiquadState->coefs[0] * templ;
+
+ // ynL+= ((-B2 * y(n-2)L ) )
+ templ = pBiquadState->coefs[1] * pBiquadState->pDelays[3];
+ ynL += templ;
+
+ // ynL+= ((-B1 * y(n-1)L ))
+ templ = pBiquadState->coefs[2] * pBiquadState->pDelays[2];
+ ynL += templ;
+
+ /**************************************************************************
+ UPDATING THE DELAYS
+ ***************************************************************************/
+ pBiquadState->pDelays[3] = pBiquadState->pDelays[2]; // y(n-2)L=y(n-1)L
+ pBiquadState->pDelays[1] = pBiquadState->pDelays[0]; // x(n-2)L=x(n-1)L
+ pBiquadState->pDelays[2] = ynL; // Update y(n-1)L in Q16
+ pBiquadState->pDelays[0] = (*pDataIn++); // Update x(n-1)L in Q0
+
+ /**************************************************************************
+ WRITING THE OUTPUT
+ ***************************************************************************/
+ *pDataOut++ = (ynL); // Write Left output
+ }
+}
diff --git a/media/libeffects/lvm/lib/Common/src/BP_1I_D16F32Cll_TRC_WRA_01_Init.c b/media/libeffects/lvm/lib/Common/src/BP_1I_D16F32Cll_TRC_WRA_01_Init.cpp
similarity index 88%
rename from media/libeffects/lvm/lib/Common/src/BP_1I_D16F32Cll_TRC_WRA_01_Init.c
rename to media/libeffects/lvm/lib/Common/src/BP_1I_D16F32Cll_TRC_WRA_01_Init.cpp
index d6e047a..eb15032 100644
--- a/media/libeffects/lvm/lib/Common/src/BP_1I_D16F32Cll_TRC_WRA_01_Init.c
+++ b/media/libeffects/lvm/lib/Common/src/BP_1I_D16F32Cll_TRC_WRA_01_Init.cpp
@@ -19,7 +19,6 @@
#include "BIQUAD.h"
#include "BP_1I_D16F32Cll_TRC_WRA_01_Private.h"
-
/*-------------------------------------------------------------------------*/
/* FUNCTION: */
/* BP_1I_D16F32Cll_TRC_WRA_01_Init */
@@ -48,7 +47,6 @@
/* RETURNS: */
/* void return code */
/*-------------------------------------------------------------------------*/
-#ifdef BUILD_FLOAT
void BP_1I_D16F32Cll_TRC_WRA_01_Init ( Biquad_FLOAT_Instance_t *pInstance,
Biquad_1I_Order2_FLOAT_Taps_t *pTaps,
BP_FLOAT_Coefs_t *pCoef)
@@ -56,24 +54,10 @@
PFilter_State_FLOAT pBiquadState = (PFilter_State_FLOAT) pInstance;
pBiquadState->pDelays =(LVM_FLOAT *) pTaps;
-
pBiquadState->coefs[0] = pCoef->A0;
pBiquadState->coefs[1] = pCoef->B2;
pBiquadState->coefs[2] = pCoef->B1;
}
-#else
-void BP_1I_D16F32Cll_TRC_WRA_01_Init ( Biquad_Instance_t *pInstance,
- Biquad_1I_Order2_Taps_t *pTaps,
- BP_C32_Coefs_t *pCoef)
-{
- PFilter_State pBiquadState = (PFilter_State) pInstance;
- pBiquadState->pDelays =(LVM_INT32 *) pTaps;
-
- pBiquadState->coefs[0] = pCoef->A0;
- pBiquadState->coefs[1] = pCoef->B2;
- pBiquadState->coefs[2] = pCoef->B1;
-}
-#endif
/*-------------------------------------------------------------------------*/
/* End Of File: BP_1I_D16F32Cll_TRC_WRA_01_Init.c */
diff --git a/media/libeffects/lvm/lib/Common/src/BP_1I_D16F32Cll_TRC_WRA_01_Private.h b/media/libeffects/lvm/lib/Common/src/BP_1I_D16F32Cll_TRC_WRA_01_Private.h
index aa9e669..6d754e2 100644
--- a/media/libeffects/lvm/lib/Common/src/BP_1I_D16F32Cll_TRC_WRA_01_Private.h
+++ b/media/libeffects/lvm/lib/Common/src/BP_1I_D16F32Cll_TRC_WRA_01_Private.h
@@ -26,12 +26,10 @@
}Filter_State;
typedef Filter_State * PFilter_State ;
-#ifdef BUILD_FLOAT
typedef struct _Filter_State_FLOAT
{
LVM_FLOAT * pDelays; /* pointer to the delayed samples (data of 32 bits) */
LVM_FLOAT coefs[3]; /* pointer to the filter coefficients */
}Filter_State_Float;
typedef Filter_State_Float * PFilter_State_FLOAT ;
-#endif
#endif /*_BP_1I_D16F32CLL_TRC_WRA_01_PRIVATE_H_*/
diff --git a/media/libeffects/lvm/lib/Common/src/BP_1I_D32F32C30_TRC_WRA_02.c b/media/libeffects/lvm/lib/Common/src/BP_1I_D32F32C30_TRC_WRA_02.cpp
similarity index 60%
rename from media/libeffects/lvm/lib/Common/src/BP_1I_D32F32C30_TRC_WRA_02.c
rename to media/libeffects/lvm/lib/Common/src/BP_1I_D32F32C30_TRC_WRA_02.cpp
index abdb2f7..d0ba206 100644
--- a/media/libeffects/lvm/lib/Common/src/BP_1I_D32F32C30_TRC_WRA_02.c
+++ b/media/libeffects/lvm/lib/Common/src/BP_1I_D32F32C30_TRC_WRA_02.cpp
@@ -19,7 +19,6 @@
#include "BP_1I_D32F32Cll_TRC_WRA_02_Private.h"
#include "LVM_Macros.h"
-
/**************************************************************************
ASSUMPTIONS:
COEFS-
@@ -33,7 +32,6 @@
pBiquadState->pDelays[2] is y(n-1)L in Q0 format
pBiquadState->pDelays[3] is y(n-2)L in Q0 format
***************************************************************************/
-#ifdef BUILD_FLOAT
void BP_1I_D32F32C30_TRC_WRA_02 ( Biquad_FLOAT_Instance_t *pInstance,
LVM_FLOAT *pDataIn,
LVM_FLOAT *pDataOut,
@@ -46,7 +44,6 @@
for (ii = NrSamples; ii != 0; ii--)
{
-
/**************************************************************************
PROCESSING OF THE LEFT CHANNEL
***************************************************************************/
@@ -78,49 +75,3 @@
}
}
-#else
-void BP_1I_D32F32C30_TRC_WRA_02 ( Biquad_Instance_t *pInstance,
- LVM_INT32 *pDataIn,
- LVM_INT32 *pDataOut,
- LVM_INT16 NrSamples)
- {
- LVM_INT32 ynL,templ;
- LVM_INT16 ii;
- PFilter_State pBiquadState = (PFilter_State) pInstance;
-
- for (ii = NrSamples; ii != 0; ii--)
- {
-
-
- /**************************************************************************
- PROCESSING OF THE LEFT CHANNEL
- ***************************************************************************/
- // ynL= (A0 (Q30) * (x(n)L (Q0) - x(n-2)L (Q0) ) >>30) in Q0
- templ=(*pDataIn)-pBiquadState->pDelays[1];
- MUL32x32INTO32(pBiquadState->coefs[0],templ,ynL,30)
-
- // ynL+= ((-B2 (Q30) * y(n-2)L (Q0) ) >>30) in Q0
- MUL32x32INTO32(pBiquadState->coefs[1],pBiquadState->pDelays[3],templ,30)
- ynL+=templ;
-
- // ynL+= ((-B1 (Q30) * y(n-1)L (Q0) ) >>30) in Q0
- MUL32x32INTO32(pBiquadState->coefs[2],pBiquadState->pDelays[2],templ,30)
- ynL+=templ;
-
- /**************************************************************************
- UPDATING THE DELAYS
- ***************************************************************************/
- pBiquadState->pDelays[3]=pBiquadState->pDelays[2]; // y(n-2)L=y(n-1)L
- pBiquadState->pDelays[1]=pBiquadState->pDelays[0]; // x(n-2)L=x(n-1)L
- pBiquadState->pDelays[2]=ynL; // Update y(n-1)L in Q0
- pBiquadState->pDelays[0]=(*pDataIn++); // Update x(n-1)L in Q0
-
- /**************************************************************************
- WRITING THE OUTPUT
- ***************************************************************************/
- *pDataOut++=ynL; // Write Left output in Q0
-
- }
-
- }
-#endif
diff --git a/media/libeffects/lvm/lib/Common/src/BP_1I_D32F32Cll_TRC_WRA_02_Init.c b/media/libeffects/lvm/lib/Common/src/BP_1I_D32F32Cll_TRC_WRA_02_Init.cpp
similarity index 85%
rename from media/libeffects/lvm/lib/Common/src/BP_1I_D32F32Cll_TRC_WRA_02_Init.c
rename to media/libeffects/lvm/lib/Common/src/BP_1I_D32F32Cll_TRC_WRA_02_Init.cpp
index 5590c32..6f7d0b5 100644
--- a/media/libeffects/lvm/lib/Common/src/BP_1I_D32F32Cll_TRC_WRA_02_Init.c
+++ b/media/libeffects/lvm/lib/Common/src/BP_1I_D32F32Cll_TRC_WRA_02_Init.cpp
@@ -37,7 +37,6 @@
/* RETURNS: */
/* void return code */
/*-------------------------------------------------------------------------*/
-#ifdef BUILD_FLOAT
void BP_1I_D32F32Cll_TRC_WRA_02_Init ( Biquad_FLOAT_Instance_t *pInstance,
Biquad_1I_Order2_FLOAT_Taps_t *pTaps,
BP_FLOAT_Coefs_t *pCoef)
@@ -51,21 +50,6 @@
pBiquadState->coefs[2] = pCoef->B1;
}
-#else
-void BP_1I_D32F32Cll_TRC_WRA_02_Init ( Biquad_Instance_t *pInstance,
- Biquad_1I_Order2_Taps_t *pTaps,
- BP_C32_Coefs_t *pCoef)
-{
- PFilter_State pBiquadState = (PFilter_State) pInstance;
- pBiquadState->pDelays =(LVM_INT32 *) pTaps;
-
- pBiquadState->coefs[0]=pCoef->A0;
-
- pBiquadState->coefs[1]=pCoef->B2;
-
- pBiquadState->coefs[2]=pCoef->B1;
-}
-#endif
/*-------------------------------------------------------------------------*/
/* End Of File: BP_1I_D32F32Cll_TRC_WRA_02_Init.c */
diff --git a/media/libeffects/lvm/lib/Common/src/BP_1I_D32F32Cll_TRC_WRA_02_Private.h b/media/libeffects/lvm/lib/Common/src/BP_1I_D32F32Cll_TRC_WRA_02_Private.h
index 80c3920..9f1c66a 100644
--- a/media/libeffects/lvm/lib/Common/src/BP_1I_D32F32Cll_TRC_WRA_02_Private.h
+++ b/media/libeffects/lvm/lib/Common/src/BP_1I_D32F32Cll_TRC_WRA_02_Private.h
@@ -26,13 +26,11 @@
}Filter_State;
typedef Filter_State * PFilter_State ;
-#ifdef BUILD_FLOAT
typedef struct _Filter_State_FLOAT
{
LVM_FLOAT * pDelays; /* pointer to the delayed samples (data of 32 bits) */
LVM_FLOAT coefs[3]; /* pointer to the filter coefficients */
}Filter_State_Float;
typedef Filter_State_Float* PFilter_State_FLOAT ;
-#endif
#endif /*_BP_1I_D32F32CLL_TRC_WRA_02_PRIVATE_H_*/
diff --git a/media/libeffects/lvm/lib/Common/src/BQ_1I_D16F16C15_TRC_WRA_01.c b/media/libeffects/lvm/lib/Common/src/BQ_1I_D16F16C15_TRC_WRA_01.cpp
similarity index 60%
rename from media/libeffects/lvm/lib/Common/src/BQ_1I_D16F16C15_TRC_WRA_01.c
rename to media/libeffects/lvm/lib/Common/src/BQ_1I_D16F16C15_TRC_WRA_01.cpp
index ee9bf7a..9aecc40 100644
--- a/media/libeffects/lvm/lib/Common/src/BQ_1I_D16F16C15_TRC_WRA_01.c
+++ b/media/libeffects/lvm/lib/Common/src/BQ_1I_D16F16C15_TRC_WRA_01.cpp
@@ -32,7 +32,6 @@
pBiquadState->pDelays[2] is y(n-1)L in Q0 format
pBiquadState->pDelays[3] is y(n-2)L in Q0 format
***************************************************************************/
-#ifdef BUILD_FLOAT
void BQ_1I_D16F16C15_TRC_WRA_01 ( Biquad_FLOAT_Instance_t *pInstance,
LVM_FLOAT *pDataIn,
LVM_FLOAT *pDataOut,
@@ -45,7 +44,6 @@
for (ii = NrSamples; ii != 0; ii--)
{
-
/**************************************************************************
PROCESSING OF THE LEFT CHANNEL
***************************************************************************/
@@ -77,58 +75,6 @@
***************************************************************************/
*pDataOut++ = (LVM_FLOAT)ynL; // Write Left output in Q0
-
}
}
-#else
-void BQ_1I_D16F16C15_TRC_WRA_01 ( Biquad_Instance_t *pInstance,
- LVM_INT16 *pDataIn,
- LVM_INT16 *pDataOut,
- LVM_INT16 NrSamples)
- {
- LVM_INT32 ynL;
- LVM_INT16 ii;
- PFilter_State pBiquadState = (PFilter_State) pInstance;
-
- for (ii = NrSamples; ii != 0; ii--)
- {
-
-
- /**************************************************************************
- PROCESSING OF THE LEFT CHANNEL
- ***************************************************************************/
- // ynL=A2 (Q15) * x(n-2)L (Q0) in Q15
- ynL=(LVM_INT32)pBiquadState->coefs[0]* pBiquadState->pDelays[1];
-
- // ynL+=A1 (Q15) * x(n-1)L (Q0) in Q15
- ynL+=(LVM_INT32)pBiquadState->coefs[1]* pBiquadState->pDelays[0];
-
- // ynL+=A0 (Q15) * x(n)L (Q0) in Q15
- ynL+=(LVM_INT32)pBiquadState->coefs[2]* (*pDataIn);
-
- // ynL+= (-B2 (Q15) * y(n-2)L (Q0) ) in Q15
- ynL+=(LVM_INT32)pBiquadState->coefs[3]*pBiquadState->pDelays[3];
-
- // ynL+= (-B1 (Q15) * y(n-1)L (Q0) ) in Q15
- ynL+=(LVM_INT32)pBiquadState->coefs[4]*pBiquadState->pDelays[2];
-
- ynL=ynL>>15; // ynL in Q0 format
- /**************************************************************************
- UPDATING THE DELAYS
- ***************************************************************************/
- pBiquadState->pDelays[3]=pBiquadState->pDelays[2]; // y(n-2)L=y(n-1)L
- pBiquadState->pDelays[1]=pBiquadState->pDelays[0]; // x(n-2)L=x(n-1)L
- pBiquadState->pDelays[2]=ynL; // Update y(n-1)L in Q0
- pBiquadState->pDelays[0]=(*pDataIn++); // Update x(n-1)L in Q0
-
- /**************************************************************************
- WRITING THE OUTPUT
- ***************************************************************************/
- *pDataOut++=(LVM_INT16)ynL; // Write Left output in Q0
-
-
- }
-
- }
-#endif
diff --git a/media/libeffects/lvm/lib/Common/src/BQ_1I_D16F16Css_TRC_WRA_01_Init.c b/media/libeffects/lvm/lib/Common/src/BQ_1I_D16F16Css_TRC_WRA_01_Init.cpp
similarity index 82%
rename from media/libeffects/lvm/lib/Common/src/BQ_1I_D16F16Css_TRC_WRA_01_Init.c
rename to media/libeffects/lvm/lib/Common/src/BQ_1I_D16F16Css_TRC_WRA_01_Init.cpp
index 3d5befa..f0b5d06 100644
--- a/media/libeffects/lvm/lib/Common/src/BQ_1I_D16F16Css_TRC_WRA_01_Init.c
+++ b/media/libeffects/lvm/lib/Common/src/BQ_1I_D16F16Css_TRC_WRA_01_Init.cpp
@@ -37,7 +37,6 @@
/* RETURNS: */
/* void return code */
/*-------------------------------------------------------------------------*/
-#ifdef BUILD_FLOAT
void BQ_1I_D16F16Css_TRC_WRA_01_Init ( Biquad_FLOAT_Instance_t *pInstance,
Biquad_1I_Order2_FLOAT_Taps_t *pTaps,
BQ_FLOAT_Coefs_t *pCoef)
@@ -56,27 +55,6 @@
temp = pCoef->B1;
pBiquadState->coefs[4] = temp;
}
-#else
-void BQ_1I_D16F16Css_TRC_WRA_01_Init ( Biquad_Instance_t *pInstance,
- Biquad_1I_Order2_Taps_t *pTaps,
- BQ_C16_Coefs_t *pCoef)
-{
- LVM_INT16 temp;
- PFilter_State pBiquadState = (PFilter_State) pInstance;
- pBiquadState->pDelays =(LVM_INT32 *) pTaps ;
-
- temp=pCoef->A2;
- pBiquadState->coefs[0]=temp;
- temp=pCoef->A1;
- pBiquadState->coefs[1]=temp;
- temp=pCoef->A0;
- pBiquadState->coefs[2]=temp;
- temp=pCoef->B2;
- pBiquadState->coefs[3]=temp;
- temp=pCoef->B1;
- pBiquadState->coefs[4]=temp;
-}
-#endif
/*-------------------------------------------------------------------------*/
/* End Of File: BQ_1I_D16F16Css_TRC_WRA_01_Init.c */
diff --git a/media/libeffects/lvm/lib/Common/src/BQ_1I_D16F16Css_TRC_WRA_01_Private.h b/media/libeffects/lvm/lib/Common/src/BQ_1I_D16F16Css_TRC_WRA_01_Private.h
index 811da8b..fad345d 100644
--- a/media/libeffects/lvm/lib/Common/src/BQ_1I_D16F16Css_TRC_WRA_01_Private.h
+++ b/media/libeffects/lvm/lib/Common/src/BQ_1I_D16F16Css_TRC_WRA_01_Private.h
@@ -27,7 +27,6 @@
typedef Filter_State * PFilter_State ;
-#ifdef BUILD_FLOAT
typedef struct _Filter_State_FLOAT
{
LVM_FLOAT * pDelays; /* pointer to the delayed samples (data of 32 bits) */
@@ -35,5 +34,4 @@
}Filter_State_FLOAT;
typedef Filter_State_FLOAT * PFilter_State_FLOAT ;
-#endif
#endif /*_BQ_1I_D16F16CSS_TRC_WRA_01_PRIVATE_H_ */
diff --git a/media/libeffects/lvm/lib/Common/src/BQ_1I_D16F32C14_TRC_WRA_01.c b/media/libeffects/lvm/lib/Common/src/BQ_1I_D16F32C14_TRC_WRA_01.c
deleted file mode 100644
index c74a137..0000000
--- a/media/libeffects/lvm/lib/Common/src/BQ_1I_D16F32C14_TRC_WRA_01.c
+++ /dev/null
@@ -1,131 +0,0 @@
-/*
- * Copyright (C) 2004-2010 NXP Software
- * Copyright (C) 2010 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include "BIQUAD.h"
-#include "BQ_1I_D16F32Css_TRC_WRA_01_Private.h"
-#include "LVM_Macros.h"
-
-/**************************************************************************
- ASSUMPTIONS:
- COEFS-
- pBiquadState->coefs[0] is A2, pBiquadState->coefs[1] is A1
- pBiquadState->coefs[2] is A0, pBiquadState->coefs[3] is -B2
- pBiquadState->coefs[4] is -B1, these are in Q14 format
-
- DELAYS-
- pBiquadState->pDelays[0] is x(n-1)L in Q0 format
- pBiquadState->pDelays[1] is x(n-2)L in Q0 format
- pBiquadState->pDelays[2] is y(n-1)L in Q16 format
- pBiquadState->pDelays[3] is y(n-2)L in Q16 format
-***************************************************************************/
-#ifdef BUILD_FLOAT
-void BQ_1I_D16F32C14_TRC_WRA_01 ( Biquad_FLOAT_Instance_t *pInstance,
- LVM_FLOAT *pDataIn,
- LVM_FLOAT *pDataOut,
- LVM_INT16 NrSamples)
- {
- LVM_FLOAT ynL;
- LVM_INT16 ii;
- PFilter_State_FLOAT pBiquadState = (PFilter_State_FLOAT) pInstance;
-
- for (ii = NrSamples; ii != 0; ii--)
- {
-
-
- /**************************************************************************
- PROCESSING OF THE LEFT CHANNEL
- ***************************************************************************/
- // ynL=A2 * x(n-2)L
- ynL = (LVM_FLOAT)pBiquadState->coefs[0] * pBiquadState->pDelays[1];
-
- // ynL+=A1 * x(n-1)L
- ynL += (LVM_FLOAT)pBiquadState->coefs[1] * pBiquadState->pDelays[0];
-
- // ynL+=A0 * x(n)L
- ynL += (LVM_FLOAT)pBiquadState->coefs[2] * (*pDataIn);
-
- // ynL+= ( (-B2 * y(n-2)L )
- ynL += pBiquadState->pDelays[3] * pBiquadState->coefs[3];
-
- // ynL+= -B1 * y(n-1)L
- ynL += pBiquadState->pDelays[2] * pBiquadState->coefs[4];
-
- /**************************************************************************
- UPDATING THE DELAYS
- ***************************************************************************/
- pBiquadState->pDelays[3] = pBiquadState->pDelays[2]; // y(n-2)L=y(n-1)L
- pBiquadState->pDelays[1] = pBiquadState->pDelays[0]; // x(n-2)L=x(n-1)L
- pBiquadState->pDelays[2] = ynL; // Update y(n-1)L
- pBiquadState->pDelays[0] = (*pDataIn++); // Update x(n-1)L
-
- /**************************************************************************
- WRITING THE OUTPUT
- ***************************************************************************/
- *pDataOut++ = (LVM_FLOAT)(ynL); // Write Left output
-
- }
- }
-#else
-void BQ_1I_D16F32C14_TRC_WRA_01 ( Biquad_Instance_t *pInstance,
- LVM_INT16 *pDataIn,
- LVM_INT16 *pDataOut,
- LVM_INT16 NrSamples)
- {
- LVM_INT32 ynL,templ;
- LVM_INT16 ii;
- PFilter_State pBiquadState = (PFilter_State) pInstance;
-
- for (ii = NrSamples; ii != 0; ii--)
- {
-
-
- /**************************************************************************
- PROCESSING OF THE LEFT CHANNEL
- ***************************************************************************/
- // ynL=A2 (Q14) * x(n-2)L (Q0) in Q14
- ynL=(LVM_INT32)pBiquadState->coefs[0]* pBiquadState->pDelays[1];
-
- // ynL+=A1 (Q14) * x(n-1)L (Q0) in Q14
- ynL+=(LVM_INT32)pBiquadState->coefs[1]* pBiquadState->pDelays[0];
-
- // ynL+=A0 (Q14) * x(n)L (Q0) in Q14
- ynL+=(LVM_INT32)pBiquadState->coefs[2]* (*pDataIn);
-
- // ynL+= ( (-B2 (Q14) * y(n-2)L (Q16) )>>16) in Q14
- MUL32x16INTO32(pBiquadState->pDelays[3],pBiquadState->coefs[3],templ,16)
- ynL+=templ;
-
- // ynL+= ( (-B1 (Q14) * y(n-1)L (Q16) )>>16) in Q14
- MUL32x16INTO32(pBiquadState->pDelays[2],pBiquadState->coefs[4],templ,16)
- ynL+=templ;
-
- /**************************************************************************
- UPDATING THE DELAYS
- ***************************************************************************/
- pBiquadState->pDelays[3]=pBiquadState->pDelays[2]; // y(n-2)L=y(n-1)L
- pBiquadState->pDelays[1]=pBiquadState->pDelays[0]; // x(n-2)L=x(n-1)L
- pBiquadState->pDelays[2]=ynL<<2; // Update y(n-1)L in Q16
- pBiquadState->pDelays[0]=(*pDataIn++); // Update x(n-1)L in Q0
-
- /**************************************************************************
- WRITING THE OUTPUT
- ***************************************************************************/
- *pDataOut++=(LVM_INT16)(ynL>>14); // Write Left output in Q0
-
- }
- }
-#endif
diff --git a/media/libeffects/lvm/lib/Common/src/BQ_1I_D16F32C14_TRC_WRA_01.cpp b/media/libeffects/lvm/lib/Common/src/BQ_1I_D16F32C14_TRC_WRA_01.cpp
new file mode 100644
index 0000000..043bc5f
--- /dev/null
+++ b/media/libeffects/lvm/lib/Common/src/BQ_1I_D16F32C14_TRC_WRA_01.cpp
@@ -0,0 +1,79 @@
+/*
+ * Copyright (C) 2004-2010 NXP Software
+ * Copyright (C) 2010 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include "BIQUAD.h"
+#include "BQ_1I_D16F32Css_TRC_WRA_01_Private.h"
+#include "LVM_Macros.h"
+
+/**************************************************************************
+ ASSUMPTIONS:
+ COEFS-
+ pBiquadState->coefs[0] is A2, pBiquadState->coefs[1] is A1
+ pBiquadState->coefs[2] is A0, pBiquadState->coefs[3] is -B2
+ pBiquadState->coefs[4] is -B1, these are in Q14 format
+
+ DELAYS-
+ pBiquadState->pDelays[0] is x(n-1)L in Q0 format
+ pBiquadState->pDelays[1] is x(n-2)L in Q0 format
+ pBiquadState->pDelays[2] is y(n-1)L in Q16 format
+ pBiquadState->pDelays[3] is y(n-2)L in Q16 format
+***************************************************************************/
+void BQ_1I_D16F32C14_TRC_WRA_01 ( Biquad_FLOAT_Instance_t *pInstance,
+ LVM_FLOAT *pDataIn,
+ LVM_FLOAT *pDataOut,
+ LVM_INT16 NrSamples)
+ {
+ LVM_FLOAT ynL;
+ LVM_INT16 ii;
+ PFilter_State_FLOAT pBiquadState = (PFilter_State_FLOAT) pInstance;
+
+ for (ii = NrSamples; ii != 0; ii--)
+ {
+
+ /**************************************************************************
+ PROCESSING OF THE LEFT CHANNEL
+ ***************************************************************************/
+ // ynL=A2 * x(n-2)L
+ ynL = (LVM_FLOAT)pBiquadState->coefs[0] * pBiquadState->pDelays[1];
+
+ // ynL+=A1 * x(n-1)L
+ ynL += (LVM_FLOAT)pBiquadState->coefs[1] * pBiquadState->pDelays[0];
+
+ // ynL+=A0 * x(n)L
+ ynL += (LVM_FLOAT)pBiquadState->coefs[2] * (*pDataIn);
+
+ // ynL+= ( (-B2 * y(n-2)L )
+ ynL += pBiquadState->pDelays[3] * pBiquadState->coefs[3];
+
+ // ynL+= -B1 * y(n-1)L
+ ynL += pBiquadState->pDelays[2] * pBiquadState->coefs[4];
+
+ /**************************************************************************
+ UPDATING THE DELAYS
+ ***************************************************************************/
+ pBiquadState->pDelays[3] = pBiquadState->pDelays[2]; // y(n-2)L=y(n-1)L
+ pBiquadState->pDelays[1] = pBiquadState->pDelays[0]; // x(n-2)L=x(n-1)L
+ pBiquadState->pDelays[2] = ynL; // Update y(n-1)L
+ pBiquadState->pDelays[0] = (*pDataIn++); // Update x(n-1)L
+
+ /**************************************************************************
+ WRITING THE OUTPUT
+ ***************************************************************************/
+ *pDataOut++ = (LVM_FLOAT)(ynL); // Write Left output
+
+ }
+ }
diff --git a/media/libeffects/lvm/lib/Common/src/BQ_1I_D16F32Css_TRC_WRA_01_Private.h b/media/libeffects/lvm/lib/Common/src/BQ_1I_D16F32Css_TRC_WRA_01_Private.h
index 9812274..6a61d9a 100644
--- a/media/libeffects/lvm/lib/Common/src/BQ_1I_D16F32Css_TRC_WRA_01_Private.h
+++ b/media/libeffects/lvm/lib/Common/src/BQ_1I_D16F32Css_TRC_WRA_01_Private.h
@@ -27,7 +27,6 @@
typedef Filter_State * PFilter_State ;
-#ifdef BUILD_FLOAT
typedef struct _Filter_State_FLOAT
{
LVM_FLOAT * pDelays; /* pointer to the delayed samples (data of 32 bits) */
@@ -35,5 +34,4 @@
}Filter_State_FLOAT;
typedef Filter_State_FLOAT * PFilter_State_FLOAT ;
-#endif
#endif /*_BQ_1I_D16F32CSS_TRC_WRA_01_PRIVATE_H_*/
diff --git a/media/libeffects/lvm/lib/Common/src/BQ_1I_D16F32Css_TRC_WRA_01_init.c b/media/libeffects/lvm/lib/Common/src/BQ_1I_D16F32Css_TRC_WRA_01_init.cpp
similarity index 82%
rename from media/libeffects/lvm/lib/Common/src/BQ_1I_D16F32Css_TRC_WRA_01_init.c
rename to media/libeffects/lvm/lib/Common/src/BQ_1I_D16F32Css_TRC_WRA_01_init.cpp
index feae20d..2b80691 100644
--- a/media/libeffects/lvm/lib/Common/src/BQ_1I_D16F32Css_TRC_WRA_01_init.c
+++ b/media/libeffects/lvm/lib/Common/src/BQ_1I_D16F32Css_TRC_WRA_01_init.cpp
@@ -19,7 +19,6 @@
#include "BIQUAD.h"
#include "BQ_1I_D16F32Css_TRC_WRA_01_Private.h"
-
/*-------------------------------------------------------------------------*/
/* FUNCTION: */
/* BQ_1I_D16F32Css_TRC_WRA_01_Init */
@@ -38,7 +37,6 @@
/* RETURNS: */
/* void return code */
/*-------------------------------------------------------------------------*/
-#ifdef BUILD_FLOAT
void BQ_1I_D16F32Css_TRC_WRA_01_Init ( Biquad_FLOAT_Instance_t *pInstance,
Biquad_1I_Order2_FLOAT_Taps_t *pTaps,
BQ_FLOAT_Coefs_t *pCoef)
@@ -58,27 +56,6 @@
temp = pCoef->B1;
pBiquadState->coefs[4] = temp;
}
-#else
-void BQ_1I_D16F32Css_TRC_WRA_01_Init ( Biquad_Instance_t *pInstance,
- Biquad_1I_Order2_Taps_t *pTaps,
- BQ_C16_Coefs_t *pCoef)
-{
- LVM_INT16 temp;
- PFilter_State pBiquadState = (PFilter_State) pInstance;
- pBiquadState->pDelays =(LVM_INT32 *) pTaps ;
-
- temp=pCoef->A2;
- pBiquadState->coefs[0]=temp;
- temp=pCoef->A1;
- pBiquadState->coefs[1]=temp;
- temp=pCoef->A0;
- pBiquadState->coefs[2]=temp;
- temp=pCoef->B2;
- pBiquadState->coefs[3]=temp;
- temp=pCoef->B1;
- pBiquadState->coefs[4]=temp;
-}
-#endif
/*-------------------------------------------------------------------------*/
/* End Of File: BQ_1I_D16F32Css_TRC_WRA_01_Init */
diff --git a/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F16C14_TRC_WRA_01.c b/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F16C14_TRC_WRA_01.c
deleted file mode 100644
index 9b0fde3..0000000
--- a/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F16C14_TRC_WRA_01.c
+++ /dev/null
@@ -1,189 +0,0 @@
-/*
- * Copyright (C) 2004-2010 NXP Software
- * Copyright (C) 2010 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include "BIQUAD.h"
-#include "BQ_2I_D16F16Css_TRC_WRA_01_Private.h"
-#include "LVM_Macros.h"
-
-
-/**************************************************************************
- ASSUMPTIONS:
- COEFS-
- pBiquadState->coefs[0] is A2, pBiquadState->coefs[1] is A1
- pBiquadState->coefs[2] is A0, pBiquadState->coefs[3] is -B2
- pBiquadState->coefs[4] is -B1, these are in Q14 format
-
- DELAYS-
- pBiquadState->pDelays[0] is x(n-1)L in Q0 format
- pBiquadState->pDelays[1] is x(n-1)R in Q0 format
- pBiquadState->pDelays[2] is x(n-2)L in Q0 format
- pBiquadState->pDelays[3] is x(n-2)R in Q0 format
- pBiquadState->pDelays[4] is y(n-1)L in Q0 format
- pBiquadState->pDelays[5] is y(n-1)R in Q0 format
- pBiquadState->pDelays[6] is y(n-2)L in Q0 format
- pBiquadState->pDelays[7] is y(n-2)R in Q0 format
-***************************************************************************/
-#ifdef BUILD_FLOAT
-void BQ_2I_D16F16C14_TRC_WRA_01 ( Biquad_FLOAT_Instance_t *pInstance,
- LVM_FLOAT *pDataIn,
- LVM_FLOAT *pDataOut,
- LVM_INT16 NrSamples)
- {
- LVM_FLOAT ynL,ynR;
- LVM_INT16 ii;
- PFilter_State_FLOAT pBiquadState = (PFilter_State_FLOAT) pInstance;
-
- for (ii = NrSamples; ii != 0; ii--)
- {
-
- /**************************************************************************
- PROCESSING OF THE LEFT CHANNEL
- ***************************************************************************/
- // ynL=A2 * x(n-2)L
- ynL = (LVM_FLOAT)pBiquadState->coefs[0] * pBiquadState->pDelays[2];
-
- // ynL+=A1 * x(n-1)L
- ynL += (LVM_FLOAT)pBiquadState->coefs[1] * pBiquadState->pDelays[0];
-
- // ynL+=A0 * x(n)L
- ynL += (LVM_FLOAT)pBiquadState->coefs[2] * (*pDataIn);
-
- // ynL+= ( -B2 * y(n-2)L )
- ynL += (LVM_FLOAT)pBiquadState->coefs[3] * pBiquadState->pDelays[6];
-
- // ynL+=( -B1 * y(n-1)L )
- ynL += (LVM_FLOAT)pBiquadState->coefs[4] * pBiquadState->pDelays[4];
-
-
-
- /**************************************************************************
- PROCESSING OF THE RIGHT CHANNEL
- ***************************************************************************/
- // ynR=A2 * x(n-2)R
- ynR = (LVM_FLOAT)pBiquadState->coefs[0] * pBiquadState->pDelays[3];
-
- // ynR+=A1 * x(n-1)R
- ynR += (LVM_FLOAT)pBiquadState->coefs[1] * pBiquadState->pDelays[1];
-
- // ynR+=A0 * x(n)R
- ynR += (LVM_FLOAT)pBiquadState->coefs[2] * (*(pDataIn+1));
-
- // ynR+= ( -B2 * y(n-2)R )
- ynR += (LVM_FLOAT)pBiquadState->coefs[3] * pBiquadState->pDelays[7];
-
- // ynR+=( -B1 * y(n-1)R )
- ynR += (LVM_FLOAT)pBiquadState->coefs[4] * pBiquadState->pDelays[5];
-
-
- /**************************************************************************
- UPDATING THE DELAYS
- ***************************************************************************/
- pBiquadState->pDelays[7] = pBiquadState->pDelays[5]; // y(n-2)R=y(n-1)R
- pBiquadState->pDelays[6] = pBiquadState->pDelays[4]; // y(n-2)L=y(n-1)L
- pBiquadState->pDelays[3] = pBiquadState->pDelays[1]; // x(n-2)R=x(n-1)R
- pBiquadState->pDelays[2] = pBiquadState->pDelays[0]; // x(n-2)L=x(n-1)L
- pBiquadState->pDelays[5] = ynR; // Update y(n-1)R
- pBiquadState->pDelays[4] = ynL; // Update y(n-1)L
- pBiquadState->pDelays[0] = (*pDataIn++); // Update x(n-1)L
- pBiquadState->pDelays[1] = (*pDataIn++); // Update x(n-1)R
-
- /**************************************************************************
- WRITING THE OUTPUT
- ***************************************************************************/
- *pDataOut++ = (LVM_FLOAT)ynL; // Write Left output
- *pDataOut++ = (LVM_FLOAT)ynR; // Write Right ouput
-
-
- }
-
- }
-#else
-void BQ_2I_D16F16C14_TRC_WRA_01 ( Biquad_Instance_t *pInstance,
- LVM_INT16 *pDataIn,
- LVM_INT16 *pDataOut,
- LVM_INT16 NrSamples)
- {
- LVM_INT32 ynL,ynR;
- LVM_INT16 ii;
- PFilter_State pBiquadState = (PFilter_State) pInstance;
-
- for (ii = NrSamples; ii != 0; ii--)
- {
-
- /**************************************************************************
- PROCESSING OF THE LEFT CHANNEL
- ***************************************************************************/
- // ynL=A2 (Q14) * x(n-2)L (Q0) in Q14
- ynL=(LVM_INT32)pBiquadState->coefs[0]* pBiquadState->pDelays[2];
-
- // ynL+=A1 (Q14) * x(n-1)L (Q0) in Q14
- ynL+=(LVM_INT32)pBiquadState->coefs[1]* pBiquadState->pDelays[0];
-
- // ynL+=A0 (Q14) * x(n)L (Q0) in Q14
- ynL+=(LVM_INT32)pBiquadState->coefs[2]* (*pDataIn);
-
- // ynL+= ( -B2 (Q14) * y(n-2)L (Q0) ) in Q14
- ynL+=(LVM_INT32)pBiquadState->coefs[3]*pBiquadState->pDelays[6];
-
- // ynL+=( -B1 (Q14) * y(n-1)L (Q0) ) in Q14
- ynL+=(LVM_INT32)pBiquadState->coefs[4]*pBiquadState->pDelays[4];
-
- ynL=ynL>>14; // ynL in Q0 format
-
- /**************************************************************************
- PROCESSING OF THE RIGHT CHANNEL
- ***************************************************************************/
- // ynR=A2 (Q14) * x(n-2)R (Q0) in Q14
- ynR=(LVM_INT32)pBiquadState->coefs[0]*pBiquadState->pDelays[3];
-
- // ynR+=A1 (Q14) * x(n-1)R (Q0) in Q14
- ynR+=(LVM_INT32)pBiquadState->coefs[1]*pBiquadState->pDelays[1];
-
- // ynR+=A0 (Q14) * x(n)R (Q0) in Q14
- ynR+=(LVM_INT32)pBiquadState->coefs[2]*(*(pDataIn+1));
-
- // ynR+= ( -B2 (Q14) * y(n-2)R (Q0) ) in Q14
- ynR+=(LVM_INT32)pBiquadState->coefs[3]*pBiquadState->pDelays[7];
-
- // ynR+=( -B1 (Q14) * y(n-1)R (Q0) ) in Q14
- ynR+=(LVM_INT32)pBiquadState->coefs[4]*pBiquadState->pDelays[5];
-
- ynR=ynR>>14; // ynL in Q0 format
- /**************************************************************************
- UPDATING THE DELAYS
- ***************************************************************************/
- pBiquadState->pDelays[7]=pBiquadState->pDelays[5]; // y(n-2)R=y(n-1)R
- pBiquadState->pDelays[6]=pBiquadState->pDelays[4]; // y(n-2)L=y(n-1)L
- pBiquadState->pDelays[3]=pBiquadState->pDelays[1]; // x(n-2)R=x(n-1)R
- pBiquadState->pDelays[2]=pBiquadState->pDelays[0]; // x(n-2)L=x(n-1)L
- pBiquadState->pDelays[5]=ynR; // Update y(n-1)R in Q0
- pBiquadState->pDelays[4]=ynL; // Update y(n-1)L in Q0
- pBiquadState->pDelays[0]=(*pDataIn++); // Update x(n-1)L in Q0
- pBiquadState->pDelays[1]=(*pDataIn++); // Update x(n-1)R in Q0
-
- /**************************************************************************
- WRITING THE OUTPUT
- ***************************************************************************/
- *pDataOut++=(LVM_INT16)ynL; // Write Left output in Q0
- *pDataOut++=(LVM_INT16)ynR; // Write Right ouput in Q0
-
-
- }
-
- }
-
-#endif
\ No newline at end of file
diff --git a/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F16C14_TRC_WRA_01.cpp b/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F16C14_TRC_WRA_01.cpp
new file mode 100644
index 0000000..51cd918
--- /dev/null
+++ b/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F16C14_TRC_WRA_01.cpp
@@ -0,0 +1,107 @@
+/*
+ * Copyright (C) 2004-2010 NXP Software
+ * Copyright (C) 2010 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include "BIQUAD.h"
+#include "BQ_2I_D16F16Css_TRC_WRA_01_Private.h"
+#include "LVM_Macros.h"
+
+/**************************************************************************
+ ASSUMPTIONS:
+ COEFS-
+ pBiquadState->coefs[0] is A2, pBiquadState->coefs[1] is A1
+ pBiquadState->coefs[2] is A0, pBiquadState->coefs[3] is -B2
+ pBiquadState->coefs[4] is -B1, these are in Q14 format
+
+ DELAYS-
+ pBiquadState->pDelays[0] is x(n-1)L in Q0 format
+ pBiquadState->pDelays[1] is x(n-1)R in Q0 format
+ pBiquadState->pDelays[2] is x(n-2)L in Q0 format
+ pBiquadState->pDelays[3] is x(n-2)R in Q0 format
+ pBiquadState->pDelays[4] is y(n-1)L in Q0 format
+ pBiquadState->pDelays[5] is y(n-1)R in Q0 format
+ pBiquadState->pDelays[6] is y(n-2)L in Q0 format
+ pBiquadState->pDelays[7] is y(n-2)R in Q0 format
+***************************************************************************/
+void BQ_2I_D16F16C14_TRC_WRA_01 ( Biquad_FLOAT_Instance_t *pInstance,
+ LVM_FLOAT *pDataIn,
+ LVM_FLOAT *pDataOut,
+ LVM_INT16 NrSamples)
+ {
+ LVM_FLOAT ynL,ynR;
+ LVM_INT16 ii;
+ PFilter_State_FLOAT pBiquadState = (PFilter_State_FLOAT) pInstance;
+
+ for (ii = NrSamples; ii != 0; ii--)
+ {
+
+ /**************************************************************************
+ PROCESSING OF THE LEFT CHANNEL
+ ***************************************************************************/
+ // ynL=A2 * x(n-2)L
+ ynL = (LVM_FLOAT)pBiquadState->coefs[0] * pBiquadState->pDelays[2];
+
+ // ynL+=A1 * x(n-1)L
+ ynL += (LVM_FLOAT)pBiquadState->coefs[1] * pBiquadState->pDelays[0];
+
+ // ynL+=A0 * x(n)L
+ ynL += (LVM_FLOAT)pBiquadState->coefs[2] * (*pDataIn);
+
+ // ynL+= ( -B2 * y(n-2)L )
+ ynL += (LVM_FLOAT)pBiquadState->coefs[3] * pBiquadState->pDelays[6];
+
+ // ynL+=( -B1 * y(n-1)L )
+ ynL += (LVM_FLOAT)pBiquadState->coefs[4] * pBiquadState->pDelays[4];
+
+ /**************************************************************************
+ PROCESSING OF THE RIGHT CHANNEL
+ ***************************************************************************/
+ // ynR=A2 * x(n-2)R
+ ynR = (LVM_FLOAT)pBiquadState->coefs[0] * pBiquadState->pDelays[3];
+
+ // ynR+=A1 * x(n-1)R
+ ynR += (LVM_FLOAT)pBiquadState->coefs[1] * pBiquadState->pDelays[1];
+
+ // ynR+=A0 * x(n)R
+ ynR += (LVM_FLOAT)pBiquadState->coefs[2] * (*(pDataIn+1));
+
+ // ynR+= ( -B2 * y(n-2)R )
+ ynR += (LVM_FLOAT)pBiquadState->coefs[3] * pBiquadState->pDelays[7];
+
+ // ynR+=( -B1 * y(n-1)R )
+ ynR += (LVM_FLOAT)pBiquadState->coefs[4] * pBiquadState->pDelays[5];
+
+ /**************************************************************************
+ UPDATING THE DELAYS
+ ***************************************************************************/
+ pBiquadState->pDelays[7] = pBiquadState->pDelays[5]; // y(n-2)R=y(n-1)R
+ pBiquadState->pDelays[6] = pBiquadState->pDelays[4]; // y(n-2)L=y(n-1)L
+ pBiquadState->pDelays[3] = pBiquadState->pDelays[1]; // x(n-2)R=x(n-1)R
+ pBiquadState->pDelays[2] = pBiquadState->pDelays[0]; // x(n-2)L=x(n-1)L
+ pBiquadState->pDelays[5] = ynR; // Update y(n-1)R
+ pBiquadState->pDelays[4] = ynL; // Update y(n-1)L
+ pBiquadState->pDelays[0] = (*pDataIn++); // Update x(n-1)L
+ pBiquadState->pDelays[1] = (*pDataIn++); // Update x(n-1)R
+
+ /**************************************************************************
+ WRITING THE OUTPUT
+ ***************************************************************************/
+ *pDataOut++ = (LVM_FLOAT)ynL; // Write Left output
+ *pDataOut++ = (LVM_FLOAT)ynR; // Write Right ouput
+
+ }
+
+ }
diff --git a/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F16C15_TRC_WRA_01.c b/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F16C15_TRC_WRA_01.c
deleted file mode 100644
index f24db8f..0000000
--- a/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F16C15_TRC_WRA_01.c
+++ /dev/null
@@ -1,188 +0,0 @@
-/*
- * Copyright (C) 2004-2010 NXP Software
- * Copyright (C) 2010 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include "BIQUAD.h"
-#include "BQ_2I_D16F16Css_TRC_WRA_01_Private.h"
-#include "LVM_Macros.h"
-
-
-/**************************************************************************
- ASSUMPTIONS:
- COEFS-
- pBiquadState->coefs[0] is A2, pBiquadState->coefs[1] is A1
- pBiquadState->coefs[2] is A0, pBiquadState->coefs[3] is -B2
- pBiquadState->coefs[4] is -B1, these are in Q15 format
-
- DELAYS-
- pBiquadState->pDelays[0] is x(n-1)L in Q0 format
- pBiquadState->pDelays[1] is x(n-1)R in Q0 format
- pBiquadState->pDelays[2] is x(n-2)L in Q0 format
- pBiquadState->pDelays[3] is x(n-2)R in Q0 format
- pBiquadState->pDelays[4] is y(n-1)L in Q0 format
- pBiquadState->pDelays[5] is y(n-1)R in Q0 format
- pBiquadState->pDelays[6] is y(n-2)L in Q0 format
- pBiquadState->pDelays[7] is y(n-2)R in Q0 format
-***************************************************************************/
-#ifdef BUILD_FLOAT
-void BQ_2I_D16F16C15_TRC_WRA_01 ( Biquad_FLOAT_Instance_t *pInstance,
- LVM_FLOAT *pDataIn,
- LVM_FLOAT *pDataOut,
- LVM_INT16 NrSamples)
- {
- LVM_FLOAT ynL,ynR;
- LVM_INT16 ii;
- PFilter_State_FLOAT pBiquadState = (PFilter_State_FLOAT) pInstance;
-
- for (ii = NrSamples; ii != 0; ii--)
- {
-
-
- /**************************************************************************
- PROCESSING OF THE LEFT CHANNEL
- ***************************************************************************/
- // ynL=A2 * x(n-2)L
- ynL = (LVM_FLOAT)pBiquadState->coefs[0] * pBiquadState->pDelays[2];
-
- // ynL+=A1 * x(n-1)L
- ynL += (LVM_FLOAT)pBiquadState->coefs[1] * pBiquadState->pDelays[0];
-
- // ynL+=A0 * x(n)L
- ynL += (LVM_FLOAT)pBiquadState->coefs[2] * (*pDataIn);
-
- // ynL+= ( -B2 * y(n-2)L
- ynL += (LVM_FLOAT)pBiquadState->coefs[3] * pBiquadState->pDelays[6];
-
- // ynL+=( -B1 * y(n-1)L
- ynL += (LVM_FLOAT)pBiquadState->coefs[4] * pBiquadState->pDelays[4];
-
-
-
- /**************************************************************************
- PROCESSING OF THE RIGHT CHANNEL
- ***************************************************************************/
- // ynR=A2 * x(n-2)R
- ynR = (LVM_FLOAT)pBiquadState->coefs[0] * pBiquadState->pDelays[3];
-
- // ynR+=A1 * x(n-1)R
- ynR += (LVM_FLOAT)pBiquadState->coefs[1] * pBiquadState->pDelays[1];
-
- // ynR+=A0 * x(n)R
- ynR += (LVM_FLOAT)pBiquadState->coefs[2] * (*(pDataIn+1));
-
- // ynR+= ( -B2 * y(n-2)R )
- ynR += (LVM_FLOAT)pBiquadState->coefs[3] * pBiquadState->pDelays[7];
-
- // ynR+=( -B1 * y(n-1)R )
- ynR += (LVM_FLOAT)pBiquadState->coefs[4] * pBiquadState->pDelays[5];
-
-
- /**************************************************************************
- UPDATING THE DELAYS
- ***************************************************************************/
- pBiquadState->pDelays[7] = pBiquadState->pDelays[5]; // y(n-2)R=y(n-1)R
- pBiquadState->pDelays[6] = pBiquadState->pDelays[4]; // y(n-2)L=y(n-1)L
- pBiquadState->pDelays[3] = pBiquadState->pDelays[1]; // x(n-2)R=x(n-1)R
- pBiquadState->pDelays[2] = pBiquadState->pDelays[0]; // x(n-2)L=x(n-1)L
- pBiquadState->pDelays[5] = ynR; // Update y(n-1)R
- pBiquadState->pDelays[4] = ynL; // Update y(n-1)L
- pBiquadState->pDelays[0] = (*pDataIn++); // Update x(n-1)L
- pBiquadState->pDelays[1] = (*pDataIn++); // Update x(n-1)R
-
- /**************************************************************************
- WRITING THE OUTPUT
- ***************************************************************************/
- *pDataOut++ = (LVM_FLOAT)ynL; // Write Left output
- *pDataOut++ = (LVM_FLOAT)ynR; // Write Right ouput
-
- }
-
- }
-#else
-void BQ_2I_D16F16C15_TRC_WRA_01 ( Biquad_Instance_t *pInstance,
- LVM_INT16 *pDataIn,
- LVM_INT16 *pDataOut,
- LVM_INT16 NrSamples)
- {
- LVM_INT32 ynL,ynR;
- LVM_INT16 ii;
- PFilter_State pBiquadState = (PFilter_State) pInstance;
-
- for (ii = NrSamples; ii != 0; ii--)
- {
-
-
- /**************************************************************************
- PROCESSING OF THE LEFT CHANNEL
- ***************************************************************************/
- // ynL=A2 (Q15) * x(n-2)L (Q0) in Q15
- ynL=(LVM_INT32)pBiquadState->coefs[0]* pBiquadState->pDelays[2];
-
- // ynL+=A1 (Q15) * x(n-1)L (Q0) in Q15
- ynL+=(LVM_INT32)pBiquadState->coefs[1]* pBiquadState->pDelays[0];
-
- // ynL+=A0 (Q15) * x(n)L (Q0) in Q15
- ynL+=(LVM_INT32)pBiquadState->coefs[2]* (*pDataIn);
-
- // ynL+= ( -B2 (Q15) * y(n-2)L (Q0) ) in Q15
- ynL+=(LVM_INT32)pBiquadState->coefs[3]*pBiquadState->pDelays[6];
-
- // ynL+=( -B1 (Q15) * y(n-1)L (Q0) ) in Q15
- ynL+=(LVM_INT32)pBiquadState->coefs[4]*pBiquadState->pDelays[4];
-
- ynL=ynL>>15; // ynL in Q0 format
-
- /**************************************************************************
- PROCESSING OF THE RIGHT CHANNEL
- ***************************************************************************/
- // ynR=A2 (Q15) * x(n-2)R (Q0) in Q15
- ynR=(LVM_INT32)pBiquadState->coefs[0]*pBiquadState->pDelays[3];
-
- // ynR+=A1 (Q15) * x(n-1)R (Q0) in Q15
- ynR+=(LVM_INT32)pBiquadState->coefs[1]*pBiquadState->pDelays[1];
-
- // ynR+=A0 (Q15) * x(n)R (Q0) in Q15
- ynR+=(LVM_INT32)pBiquadState->coefs[2]*(*(pDataIn+1));
-
- // ynR+= ( -B2 (Q15) * y(n-2)R (Q0) ) in Q15
- ynR+=(LVM_INT32)pBiquadState->coefs[3]*pBiquadState->pDelays[7];
-
- // ynR+=( -B1 (Q15) * y(n-1)R (Q0) ) in Q15
- ynR+=(LVM_INT32)pBiquadState->coefs[4]*pBiquadState->pDelays[5];
-
- ynR=ynR>>15; // ynL in Q0 format
- /**************************************************************************
- UPDATING THE DELAYS
- ***************************************************************************/
- pBiquadState->pDelays[7]=pBiquadState->pDelays[5]; // y(n-2)R=y(n-1)R
- pBiquadState->pDelays[6]=pBiquadState->pDelays[4]; // y(n-2)L=y(n-1)L
- pBiquadState->pDelays[3]=pBiquadState->pDelays[1]; // x(n-2)R=x(n-1)R
- pBiquadState->pDelays[2]=pBiquadState->pDelays[0]; // x(n-2)L=x(n-1)L
- pBiquadState->pDelays[5]=ynR; // Update y(n-1)R in Q0
- pBiquadState->pDelays[4]=ynL; // Update y(n-1)L in Q0
- pBiquadState->pDelays[0]=(*pDataIn++); // Update x(n-1)L in Q0
- pBiquadState->pDelays[1]=(*pDataIn++); // Update x(n-1)R in Q0
-
- /**************************************************************************
- WRITING THE OUTPUT
- ***************************************************************************/
- *pDataOut++=(LVM_INT16)ynL; // Write Left output in Q0
- *pDataOut++=(LVM_INT16)ynR; // Write Right ouput in Q0
-
- }
-
- }
-#endif
diff --git a/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F16C15_TRC_WRA_01.cpp b/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F16C15_TRC_WRA_01.cpp
new file mode 100644
index 0000000..8f74749
--- /dev/null
+++ b/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F16C15_TRC_WRA_01.cpp
@@ -0,0 +1,107 @@
+/*
+ * Copyright (C) 2004-2010 NXP Software
+ * Copyright (C) 2010 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include "BIQUAD.h"
+#include "BQ_2I_D16F16Css_TRC_WRA_01_Private.h"
+#include "LVM_Macros.h"
+
+/**************************************************************************
+ ASSUMPTIONS:
+ COEFS-
+ pBiquadState->coefs[0] is A2, pBiquadState->coefs[1] is A1
+ pBiquadState->coefs[2] is A0, pBiquadState->coefs[3] is -B2
+ pBiquadState->coefs[4] is -B1, these are in Q15 format
+
+ DELAYS-
+ pBiquadState->pDelays[0] is x(n-1)L in Q0 format
+ pBiquadState->pDelays[1] is x(n-1)R in Q0 format
+ pBiquadState->pDelays[2] is x(n-2)L in Q0 format
+ pBiquadState->pDelays[3] is x(n-2)R in Q0 format
+ pBiquadState->pDelays[4] is y(n-1)L in Q0 format
+ pBiquadState->pDelays[5] is y(n-1)R in Q0 format
+ pBiquadState->pDelays[6] is y(n-2)L in Q0 format
+ pBiquadState->pDelays[7] is y(n-2)R in Q0 format
+***************************************************************************/
+void BQ_2I_D16F16C15_TRC_WRA_01 ( Biquad_FLOAT_Instance_t *pInstance,
+ LVM_FLOAT *pDataIn,
+ LVM_FLOAT *pDataOut,
+ LVM_INT16 NrSamples)
+ {
+ LVM_FLOAT ynL,ynR;
+ LVM_INT16 ii;
+ PFilter_State_FLOAT pBiquadState = (PFilter_State_FLOAT) pInstance;
+
+ for (ii = NrSamples; ii != 0; ii--)
+ {
+
+ /**************************************************************************
+ PROCESSING OF THE LEFT CHANNEL
+ ***************************************************************************/
+ // ynL=A2 * x(n-2)L
+ ynL = (LVM_FLOAT)pBiquadState->coefs[0] * pBiquadState->pDelays[2];
+
+ // ynL+=A1 * x(n-1)L
+ ynL += (LVM_FLOAT)pBiquadState->coefs[1] * pBiquadState->pDelays[0];
+
+ // ynL+=A0 * x(n)L
+ ynL += (LVM_FLOAT)pBiquadState->coefs[2] * (*pDataIn);
+
+ // ynL+= ( -B2 * y(n-2)L
+ ynL += (LVM_FLOAT)pBiquadState->coefs[3] * pBiquadState->pDelays[6];
+
+ // ynL+=( -B1 * y(n-1)L
+ ynL += (LVM_FLOAT)pBiquadState->coefs[4] * pBiquadState->pDelays[4];
+
+ /**************************************************************************
+ PROCESSING OF THE RIGHT CHANNEL
+ ***************************************************************************/
+ // ynR=A2 * x(n-2)R
+ ynR = (LVM_FLOAT)pBiquadState->coefs[0] * pBiquadState->pDelays[3];
+
+ // ynR+=A1 * x(n-1)R
+ ynR += (LVM_FLOAT)pBiquadState->coefs[1] * pBiquadState->pDelays[1];
+
+ // ynR+=A0 * x(n)R
+ ynR += (LVM_FLOAT)pBiquadState->coefs[2] * (*(pDataIn+1));
+
+ // ynR+= ( -B2 * y(n-2)R )
+ ynR += (LVM_FLOAT)pBiquadState->coefs[3] * pBiquadState->pDelays[7];
+
+ // ynR+=( -B1 * y(n-1)R )
+ ynR += (LVM_FLOAT)pBiquadState->coefs[4] * pBiquadState->pDelays[5];
+
+ /**************************************************************************
+ UPDATING THE DELAYS
+ ***************************************************************************/
+ pBiquadState->pDelays[7] = pBiquadState->pDelays[5]; // y(n-2)R=y(n-1)R
+ pBiquadState->pDelays[6] = pBiquadState->pDelays[4]; // y(n-2)L=y(n-1)L
+ pBiquadState->pDelays[3] = pBiquadState->pDelays[1]; // x(n-2)R=x(n-1)R
+ pBiquadState->pDelays[2] = pBiquadState->pDelays[0]; // x(n-2)L=x(n-1)L
+ pBiquadState->pDelays[5] = ynR; // Update y(n-1)R
+ pBiquadState->pDelays[4] = ynL; // Update y(n-1)L
+ pBiquadState->pDelays[0] = (*pDataIn++); // Update x(n-1)L
+ pBiquadState->pDelays[1] = (*pDataIn++); // Update x(n-1)R
+
+ /**************************************************************************
+ WRITING THE OUTPUT
+ ***************************************************************************/
+ *pDataOut++ = (LVM_FLOAT)ynL; // Write Left output
+ *pDataOut++ = (LVM_FLOAT)ynR; // Write Right ouput
+
+ }
+
+ }
diff --git a/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F16Css_TRC_WRA_01_Init.c b/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F16Css_TRC_WRA_01_Init.cpp
similarity index 82%
rename from media/libeffects/lvm/lib/Common/src/BQ_2I_D16F16Css_TRC_WRA_01_Init.c
rename to media/libeffects/lvm/lib/Common/src/BQ_2I_D16F16Css_TRC_WRA_01_Init.cpp
index 39e1bda..987cbcf 100644
--- a/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F16Css_TRC_WRA_01_Init.c
+++ b/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F16Css_TRC_WRA_01_Init.cpp
@@ -19,7 +19,6 @@
#include "BIQUAD.h"
#include "BQ_2I_D16F16Css_TRC_WRA_01_Private.h"
-
/*-------------------------------------------------------------------------*/
/* FUNCTION: */
/* BQ_2I_D16F16Css_TRC_WRA_01_Init */
@@ -38,7 +37,6 @@
/* RETURNS: */
/* void return code */
/*-------------------------------------------------------------------------*/
-#ifdef BUILD_FLOAT
void BQ_2I_D16F16Css_TRC_WRA_01_Init ( Biquad_FLOAT_Instance_t *pInstance,
Biquad_2I_Order2_FLOAT_Taps_t *pTaps,
BQ_FLOAT_Coefs_t *pCoef)
@@ -58,27 +56,6 @@
temp = pCoef->B1;
pBiquadState->coefs[4] = temp;
}
-#else
-void BQ_2I_D16F16Css_TRC_WRA_01_Init ( Biquad_Instance_t *pInstance,
- Biquad_2I_Order2_Taps_t *pTaps,
- BQ_C16_Coefs_t *pCoef)
-{
- LVM_INT16 temp;
- PFilter_State pBiquadState = (PFilter_State) pInstance;
- pBiquadState->pDelays =(LVM_INT32 *) pTaps ;
-
- temp=pCoef->A2;
- pBiquadState->coefs[0]=temp;
- temp=pCoef->A1;
- pBiquadState->coefs[1]=temp;
- temp=pCoef->A0;
- pBiquadState->coefs[2]=temp;
- temp=pCoef->B2;
- pBiquadState->coefs[3]=temp;
- temp=pCoef->B1;
- pBiquadState->coefs[4]=temp;
-}
-#endif
/*-------------------------------------------------------------------------*/
/* End Of File: BQ_2I_D16F16Css_TRC_WRA_01_Init.c */
diff --git a/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F16Css_TRC_WRA_01_Private.h b/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F16Css_TRC_WRA_01_Private.h
index 0691b8c..5a9a0e9 100644
--- a/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F16Css_TRC_WRA_01_Private.h
+++ b/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F16Css_TRC_WRA_01_Private.h
@@ -28,7 +28,6 @@
typedef Filter_State * PFilter_State ;
-#ifdef BUILD_FLOAT
typedef struct _Filter_State_FLOAT
{
LVM_FLOAT * pDelays; /* pointer to the delayed samples (data of 32 bits) */
@@ -36,6 +35,5 @@
}Filter_State_FLOAT;
typedef Filter_State_FLOAT * PFilter_State_FLOAT ;
-#endif
#endif /* _BQ_2I_D16F16CSS_TRC_WRA_01_PRIVATE_H_ */
diff --git a/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F32C13_TRC_WRA_01.c b/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F32C13_TRC_WRA_01.c
deleted file mode 100644
index 61c07c7..0000000
--- a/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F32C13_TRC_WRA_01.c
+++ /dev/null
@@ -1,190 +0,0 @@
-/*
- * Copyright (C) 2004-2010 NXP Software
- * Copyright (C) 2010 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include "BIQUAD.h"
-#include "BQ_2I_D16F32Css_TRC_WRA_01_Private.h"
-#include "LVM_Macros.h"
-
-
-/**************************************************************************
- ASSUMPTIONS:
- COEFS-
- pBiquadState->coefs[0] is A2, pBiquadState->coefs[1] is A1
- pBiquadState->coefs[2] is A0, pBiquadState->coefs[3] is -B2
- pBiquadState->coefs[4] is -B1, these are in Q13 format
-
- DELAYS-
- pBiquadState->pDelays[0] is x(n-1)L in Q0 format
- pBiquadState->pDelays[1] is x(n-1)R in Q0 format
- pBiquadState->pDelays[2] is x(n-2)L in Q0 format
- pBiquadState->pDelays[3] is x(n-2)R in Q0 format
- pBiquadState->pDelays[4] is y(n-1)L in Q16 format
- pBiquadState->pDelays[5] is y(n-1)R in Q16 format
- pBiquadState->pDelays[6] is y(n-2)L in Q16 format
- pBiquadState->pDelays[7] is y(n-2)R in Q16 format
-***************************************************************************/
-#ifdef BUILD_FLOAT
-void BQ_2I_D16F32C13_TRC_WRA_01 ( Biquad_FLOAT_Instance_t *pInstance,
- LVM_FLOAT *pDataIn,
- LVM_FLOAT *pDataOut,
- LVM_INT16 NrSamples)
- {
- LVM_FLOAT ynL,ynR;
- LVM_INT16 ii;
- PFilter_State_FLOAT pBiquadState = (PFilter_State_FLOAT) pInstance;
-
- for (ii = NrSamples; ii != 0; ii--)
- {
-
- /**************************************************************************
- PROCESSING OF THE LEFT CHANNEL
- ***************************************************************************/
- /* ynL=A2 * x(n-2)L */
- ynL = (LVM_FLOAT)pBiquadState->coefs[0] * pBiquadState->pDelays[2];
-
- /* ynL+=A1* x(n-1)L */
- ynL += (LVM_FLOAT)pBiquadState->coefs[1] * pBiquadState->pDelays[0];
-
- /* ynL+=A0* x(n)L */
- ynL += (LVM_FLOAT)pBiquadState->coefs[2] * (*pDataIn);
-
- /* ynL+=-B2*y(n-2)L */
- ynL += pBiquadState->pDelays[6] * pBiquadState->coefs[3];
-
- /* ynL+=-B1*y(n-1)L */
- ynL += pBiquadState->pDelays[4] * pBiquadState->coefs[4];
-
- /**************************************************************************
- PROCESSING OF THE RIGHT CHANNEL
- ***************************************************************************/
- /* ynR=A2 * x(n-2)R */
- ynR = (LVM_FLOAT)pBiquadState->coefs[0] * pBiquadState->pDelays[3];
-
- /* ynR+=A1* x(n-1)R */
- ynR += (LVM_FLOAT)pBiquadState->coefs[1] * pBiquadState->pDelays[1];
-
- /* ynR+=A0* x(n)R */
- ynR += (LVM_FLOAT)pBiquadState->coefs[2] * (*(pDataIn+1));
-
- /* ynR+=-B2 * y(n-2)R */
- ynR += pBiquadState->pDelays[7] * pBiquadState->coefs[3];
-
- /* ynR+=-B1 * y(n-1)R */
- ynR += pBiquadState->pDelays[5] * pBiquadState->coefs[4];
-
- /**************************************************************************
- UPDATING THE DELAYS
- ***************************************************************************/
- pBiquadState->pDelays[7] = pBiquadState->pDelays[5]; /* y(n-2)R=y(n-1)R*/
- pBiquadState->pDelays[6] = pBiquadState->pDelays[4]; /* y(n-2)L=y(n-1)L*/
- pBiquadState->pDelays[3] = pBiquadState->pDelays[1]; /* x(n-2)R=x(n-1)R*/
- pBiquadState->pDelays[2] = pBiquadState->pDelays[0]; /* x(n-2)L=x(n-1)L*/
- pBiquadState->pDelays[5] = ynR; /* Update y(n-1)R */
- pBiquadState->pDelays[4] = ynL; /* Update y(n-1)L */
- pBiquadState->pDelays[0] = (*pDataIn); /* Update x(n-1)L */
- pDataIn++;
- pBiquadState->pDelays[1] = (*pDataIn); /* Update x(n-1)R */
- pDataIn++;
-
- /**************************************************************************
- WRITING THE OUTPUT
- ***************************************************************************/
- *pDataOut = (LVM_FLOAT)(ynL); /* Write Left output */
- pDataOut++;
- *pDataOut = (LVM_FLOAT)(ynR); /* Write Right ouput */
- pDataOut++;
- }
- }
-#else
-void BQ_2I_D16F32C13_TRC_WRA_01 ( Biquad_Instance_t *pInstance,
- LVM_INT16 *pDataIn,
- LVM_INT16 *pDataOut,
- LVM_INT16 NrSamples)
- {
- LVM_INT32 ynL,ynR,templ;
- LVM_INT16 ii;
- PFilter_State pBiquadState = (PFilter_State) pInstance;
-
- for (ii = NrSamples; ii != 0; ii--)
- {
-
-
- /**************************************************************************
- PROCESSING OF THE LEFT CHANNEL
- ***************************************************************************/
- /* ynL=A2 (Q13) * x(n-2)L (Q0) in Q13*/
- ynL=(LVM_INT32)pBiquadState->coefs[0]* pBiquadState->pDelays[2];
-
- /* ynL+=A1 (Q13) * x(n-1)L (Q0) in Q13*/
- ynL+=(LVM_INT32)pBiquadState->coefs[1]* pBiquadState->pDelays[0];
-
- /* ynL+=A0 (Q13) * x(n)L (Q0) in Q13*/
- ynL+=(LVM_INT32)pBiquadState->coefs[2]* (*pDataIn);
-
- /* ynL+= ( (-B2 (Q13) * y(n-2)L (Q16) )>>16) in Q13 */
- MUL32x16INTO32(pBiquadState->pDelays[6],pBiquadState->coefs[3],templ,16)
- ynL+=templ;
-
- /* ynL+=( (-B1 (Q13) * y(n-1)L (Q16) )>>16) in Q13 */
- MUL32x16INTO32(pBiquadState->pDelays[4],pBiquadState->coefs[4],templ,16)
- ynL+=templ;
-
- /**************************************************************************
- PROCESSING OF THE RIGHT CHANNEL
- ***************************************************************************/
- /* ynR=A2 (Q13) * x(n-2)R (Q0) in Q13*/
- ynR=(LVM_INT32)pBiquadState->coefs[0]*pBiquadState->pDelays[3];
-
- /* ynR+=A1 (Q13) * x(n-1)R (Q0) in Q13*/
- ynR+=(LVM_INT32)pBiquadState->coefs[1]*pBiquadState->pDelays[1];
-
- /* ynR+=A0 (Q13) * x(n)R (Q0) in Q13*/
- ynR+=(LVM_INT32)pBiquadState->coefs[2]*(*(pDataIn+1));
-
- /* ynR+= ( (-B2 (Q13) * y(n-2)R (Q16) )>>16) in Q13*/
- MUL32x16INTO32(pBiquadState->pDelays[7],pBiquadState->coefs[3],templ,16)
- ynR+=templ;
-
- /* ynR+=( (-B1 (Q13) * y(n-1)R (Q16) )>>16) in Q13 */
- MUL32x16INTO32(pBiquadState->pDelays[5],pBiquadState->coefs[4],templ,16)
- ynR+=templ;
-
- /**************************************************************************
- UPDATING THE DELAYS
- ***************************************************************************/
- pBiquadState->pDelays[7]=pBiquadState->pDelays[5]; /* y(n-2)R=y(n-1)R*/
- pBiquadState->pDelays[6]=pBiquadState->pDelays[4]; /* y(n-2)L=y(n-1)L*/
- pBiquadState->pDelays[3]=pBiquadState->pDelays[1]; /* x(n-2)R=x(n-1)R*/
- pBiquadState->pDelays[2]=pBiquadState->pDelays[0]; /* x(n-2)L=x(n-1)L*/
- pBiquadState->pDelays[5]=ynR<<3; /* Update y(n-1)R in Q16*/
- pBiquadState->pDelays[4]=ynL<<3; /* Update y(n-1)L in Q16*/
- pBiquadState->pDelays[0]=(*pDataIn); /* Update x(n-1)L in Q0*/
- pDataIn++;
- pBiquadState->pDelays[1]=(*pDataIn); /* Update x(n-1)R in Q0*/
- pDataIn++;
-
- /**************************************************************************
- WRITING THE OUTPUT
- ***************************************************************************/
- *pDataOut=(LVM_INT16)(ynL>>13); /* Write Left output in Q0*/
- pDataOut++;
- *pDataOut=(LVM_INT16)(ynR>>13); /* Write Right ouput in Q0*/
- pDataOut++;
- }
-
- }
-#endif
diff --git a/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F32C13_TRC_WRA_01.cpp b/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F32C13_TRC_WRA_01.cpp
new file mode 100644
index 0000000..331c97f
--- /dev/null
+++ b/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F32C13_TRC_WRA_01.cpp
@@ -0,0 +1,109 @@
+/*
+ * Copyright (C) 2004-2010 NXP Software
+ * Copyright (C) 2010 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include "BIQUAD.h"
+#include "BQ_2I_D16F32Css_TRC_WRA_01_Private.h"
+#include "LVM_Macros.h"
+
+/**************************************************************************
+ ASSUMPTIONS:
+ COEFS-
+ pBiquadState->coefs[0] is A2, pBiquadState->coefs[1] is A1
+ pBiquadState->coefs[2] is A0, pBiquadState->coefs[3] is -B2
+ pBiquadState->coefs[4] is -B1, these are in Q13 format
+
+ DELAYS-
+ pBiquadState->pDelays[0] is x(n-1)L in Q0 format
+ pBiquadState->pDelays[1] is x(n-1)R in Q0 format
+ pBiquadState->pDelays[2] is x(n-2)L in Q0 format
+ pBiquadState->pDelays[3] is x(n-2)R in Q0 format
+ pBiquadState->pDelays[4] is y(n-1)L in Q16 format
+ pBiquadState->pDelays[5] is y(n-1)R in Q16 format
+ pBiquadState->pDelays[6] is y(n-2)L in Q16 format
+ pBiquadState->pDelays[7] is y(n-2)R in Q16 format
+***************************************************************************/
+void BQ_2I_D16F32C13_TRC_WRA_01 ( Biquad_FLOAT_Instance_t *pInstance,
+ LVM_FLOAT *pDataIn,
+ LVM_FLOAT *pDataOut,
+ LVM_INT16 NrSamples)
+ {
+ LVM_FLOAT ynL,ynR;
+ LVM_INT16 ii;
+ PFilter_State_FLOAT pBiquadState = (PFilter_State_FLOAT) pInstance;
+
+ for (ii = NrSamples; ii != 0; ii--)
+ {
+
+ /**************************************************************************
+ PROCESSING OF THE LEFT CHANNEL
+ ***************************************************************************/
+ /* ynL=A2 * x(n-2)L */
+ ynL = (LVM_FLOAT)pBiquadState->coefs[0] * pBiquadState->pDelays[2];
+
+ /* ynL+=A1* x(n-1)L */
+ ynL += (LVM_FLOAT)pBiquadState->coefs[1] * pBiquadState->pDelays[0];
+
+ /* ynL+=A0* x(n)L */
+ ynL += (LVM_FLOAT)pBiquadState->coefs[2] * (*pDataIn);
+
+ /* ynL+=-B2*y(n-2)L */
+ ynL += pBiquadState->pDelays[6] * pBiquadState->coefs[3];
+
+ /* ynL+=-B1*y(n-1)L */
+ ynL += pBiquadState->pDelays[4] * pBiquadState->coefs[4];
+
+ /**************************************************************************
+ PROCESSING OF THE RIGHT CHANNEL
+ ***************************************************************************/
+ /* ynR=A2 * x(n-2)R */
+ ynR = (LVM_FLOAT)pBiquadState->coefs[0] * pBiquadState->pDelays[3];
+
+ /* ynR+=A1* x(n-1)R */
+ ynR += (LVM_FLOAT)pBiquadState->coefs[1] * pBiquadState->pDelays[1];
+
+ /* ynR+=A0* x(n)R */
+ ynR += (LVM_FLOAT)pBiquadState->coefs[2] * (*(pDataIn+1));
+
+ /* ynR+=-B2 * y(n-2)R */
+ ynR += pBiquadState->pDelays[7] * pBiquadState->coefs[3];
+
+ /* ynR+=-B1 * y(n-1)R */
+ ynR += pBiquadState->pDelays[5] * pBiquadState->coefs[4];
+
+ /**************************************************************************
+ UPDATING THE DELAYS
+ ***************************************************************************/
+ pBiquadState->pDelays[7] = pBiquadState->pDelays[5]; /* y(n-2)R=y(n-1)R*/
+ pBiquadState->pDelays[6] = pBiquadState->pDelays[4]; /* y(n-2)L=y(n-1)L*/
+ pBiquadState->pDelays[3] = pBiquadState->pDelays[1]; /* x(n-2)R=x(n-1)R*/
+ pBiquadState->pDelays[2] = pBiquadState->pDelays[0]; /* x(n-2)L=x(n-1)L*/
+ pBiquadState->pDelays[5] = ynR; /* Update y(n-1)R */
+ pBiquadState->pDelays[4] = ynL; /* Update y(n-1)L */
+ pBiquadState->pDelays[0] = (*pDataIn); /* Update x(n-1)L */
+ pDataIn++;
+ pBiquadState->pDelays[1] = (*pDataIn); /* Update x(n-1)R */
+ pDataIn++;
+
+ /**************************************************************************
+ WRITING THE OUTPUT
+ ***************************************************************************/
+ *pDataOut = (LVM_FLOAT)(ynL); /* Write Left output */
+ pDataOut++;
+ *pDataOut = (LVM_FLOAT)(ynR); /* Write Right ouput */
+ pDataOut++;
+ }
+ }
diff --git a/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F32C14_TRC_WRA_01.c b/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F32C14_TRC_WRA_01.c
deleted file mode 100644
index cf19e06..0000000
--- a/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F32C14_TRC_WRA_01.c
+++ /dev/null
@@ -1,192 +0,0 @@
-/*
- * Copyright (C) 2004-2010 NXP Software
- * Copyright (C) 2010 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include "BIQUAD.h"
-#include "BQ_2I_D16F32Css_TRC_WRA_01_Private.h"
-#include "LVM_Macros.h"
-
-/**************************************************************************
- ASSUMPTIONS:
- COEFS-
- pBiquadState->coefs[0] is A2, pBiquadState->coefs[1] is A1
- pBiquadState->coefs[2] is A0, pBiquadState->coefs[3] is -B2
- pBiquadState->coefs[4] is -B1, these are in Q14 format
-
- DELAYS-
- pBiquadState->pDelays[0] is x(n-1)L in Q0 format
- pBiquadState->pDelays[1] is x(n-1)R in Q0 format
- pBiquadState->pDelays[2] is x(n-2)L in Q0 format
- pBiquadState->pDelays[3] is x(n-2)R in Q0 format
- pBiquadState->pDelays[4] is y(n-1)L in Q16 format
- pBiquadState->pDelays[5] is y(n-1)R in Q16 format
- pBiquadState->pDelays[6] is y(n-2)L in Q16 format
- pBiquadState->pDelays[7] is y(n-2)R in Q16 format
-***************************************************************************/
-#ifdef BUILD_FLOAT
-void BQ_2I_D16F32C14_TRC_WRA_01 ( Biquad_FLOAT_Instance_t *pInstance,
- LVM_FLOAT *pDataIn,
- LVM_FLOAT *pDataOut,
- LVM_INT16 NrSamples)
- {
- LVM_FLOAT ynL,ynR;
- LVM_INT16 ii;
- PFilter_State_FLOAT pBiquadState = (PFilter_State_FLOAT) pInstance;
-
- for (ii = NrSamples; ii != 0; ii--)
- {
-
-
- /**************************************************************************
- PROCESSING OF THE LEFT CHANNEL
- ***************************************************************************/
- /* ynL=A2 * x(n-2)L */
- ynL = (LVM_FLOAT)pBiquadState->coefs[0] * pBiquadState->pDelays[2];
-
- /* ynL+=A1 * x(n-1)L */
- ynL += (LVM_FLOAT)pBiquadState->coefs[1] * pBiquadState->pDelays[0];
-
- /* ynL+=A0 * x(n)L */
- ynL += (LVM_FLOAT)pBiquadState->coefs[2] * (*pDataIn);
-
- /* ynL+= ( (-B2 * y(n-2)L ))*/
- ynL += pBiquadState->pDelays[6] * pBiquadState->coefs[3];
-
-
- /* ynL+=( (-B1 * y(n-1)L )) */
- ynL += pBiquadState->pDelays[4] * pBiquadState->coefs[4];
-
- /**************************************************************************
- PROCESSING OF THE RIGHT CHANNEL
- ***************************************************************************/
- /* ynR=A2 * x(n-2)R */
- ynR = (LVM_FLOAT)pBiquadState->coefs[0] * pBiquadState->pDelays[3];
-
- /* ynR+=A1 * x(n-1)R */
- ynR += (LVM_FLOAT)pBiquadState->coefs[1] * pBiquadState->pDelays[1];
-
- /* ynR+=A0 * x(n)R */
- ynR += (LVM_FLOAT)pBiquadState->coefs[2] * (*(pDataIn+1));
-
- /* ynR+= ( (-B2 * y(n-2)R ))*/
- ynR += pBiquadState->pDelays[7] * pBiquadState->coefs[3];
-
- /* ynR+=( (-B1 * y(n-1)R )) */
- ynR += pBiquadState->pDelays[5] * pBiquadState->coefs[4];
-
- /**************************************************************************
- UPDATING THE DELAYS
- ***************************************************************************/
- pBiquadState->pDelays[7] = pBiquadState->pDelays[5]; /* y(n-2)R=y(n-1)R*/
- pBiquadState->pDelays[6] = pBiquadState->pDelays[4]; /* y(n-2)L=y(n-1)L*/
- pBiquadState->pDelays[3] = pBiquadState->pDelays[1]; /* x(n-2)R=x(n-1)R*/
- pBiquadState->pDelays[2] = pBiquadState->pDelays[0]; /* x(n-2)L=x(n-1)L*/
- pBiquadState->pDelays[5] = ynR; /* Update y(n-1)R */
- pBiquadState->pDelays[4] = ynL; /* Update y(n-1)L */
- pBiquadState->pDelays[0] = (*pDataIn); /* Update x(n-1)L */
- pDataIn++;
- pBiquadState->pDelays[1] = (*pDataIn); /* Update x(n-1)R */
- pDataIn++;
-
- /**************************************************************************
- WRITING THE OUTPUT
- ***************************************************************************/
- *pDataOut = (LVM_FLOAT)(ynL); /* Write Left output */
- pDataOut++;
- *pDataOut = (LVM_FLOAT)(ynR); /* Write Right ouput */
- pDataOut++;
- }
-
- }
-#else
-void BQ_2I_D16F32C14_TRC_WRA_01 ( Biquad_Instance_t *pInstance,
- LVM_INT16 *pDataIn,
- LVM_INT16 *pDataOut,
- LVM_INT16 NrSamples)
- {
- LVM_INT32 ynL,ynR,templ;
- LVM_INT16 ii;
- PFilter_State pBiquadState = (PFilter_State) pInstance;
-
- for (ii = NrSamples; ii != 0; ii--)
- {
-
-
- /**************************************************************************
- PROCESSING OF THE LEFT CHANNEL
- ***************************************************************************/
- /* ynL=A2 (Q14) * x(n-2)L (Q0) in Q14*/
- ynL=(LVM_INT32)pBiquadState->coefs[0]* pBiquadState->pDelays[2];
-
- /* ynL+=A1 (Q14) * x(n-1)L (Q0) in Q14*/
- ynL+=(LVM_INT32)pBiquadState->coefs[1]* pBiquadState->pDelays[0];
-
- /* ynL+=A0 (Q14) * x(n)L (Q0) in Q14*/
- ynL+=(LVM_INT32)pBiquadState->coefs[2]* (*pDataIn);
-
- /* ynL+= ( (-B2 (Q14) * y(n-2)L (Q16) )>>16) in Q14 */
- MUL32x16INTO32(pBiquadState->pDelays[6],pBiquadState->coefs[3],templ,16)
- ynL+=templ;
-
- /* ynL+=( (-B1 (Q14) * y(n-1)L (Q16) )>>16) in Q14 */
- MUL32x16INTO32(pBiquadState->pDelays[4],pBiquadState->coefs[4],templ,16)
- ynL+=templ;
-
- /**************************************************************************
- PROCESSING OF THE RIGHT CHANNEL
- ***************************************************************************/
- /* ynR=A2 (Q14) * x(n-2)R (Q0) in Q14*/
- ynR=(LVM_INT32)pBiquadState->coefs[0]*pBiquadState->pDelays[3];
-
- /* ynR+=A1 (Q14) * x(n-1)R (Q0) in Q14*/
- ynR+=(LVM_INT32)pBiquadState->coefs[1]*pBiquadState->pDelays[1];
-
- /* ynR+=A0 (Q14) * x(n)R (Q0) in Q14*/
- ynR+=(LVM_INT32)pBiquadState->coefs[2]*(*(pDataIn+1));
-
- /* ynR+= ( (-B2 (Q14) * y(n-2)R (Q16) )>>16) in Q14*/
- MUL32x16INTO32(pBiquadState->pDelays[7],pBiquadState->coefs[3],templ,16)
- ynR+=templ;
-
- /* ynR+=( (-B1 (Q14) * y(n-1)R (Q16) )>>16) in Q14 */
- MUL32x16INTO32(pBiquadState->pDelays[5],pBiquadState->coefs[4],templ,16)
- ynR+=templ;
-
- /**************************************************************************
- UPDATING THE DELAYS
- ***************************************************************************/
- pBiquadState->pDelays[7]=pBiquadState->pDelays[5]; /* y(n-2)R=y(n-1)R*/
- pBiquadState->pDelays[6]=pBiquadState->pDelays[4]; /* y(n-2)L=y(n-1)L*/
- pBiquadState->pDelays[3]=pBiquadState->pDelays[1]; /* x(n-2)R=x(n-1)R*/
- pBiquadState->pDelays[2]=pBiquadState->pDelays[0]; /* x(n-2)L=x(n-1)L*/
- pBiquadState->pDelays[5]=ynR<<2; /* Update y(n-1)R in Q16*/
- pBiquadState->pDelays[4]=ynL<<2; /* Update y(n-1)L in Q16*/
- pBiquadState->pDelays[0]=(*pDataIn); /* Update x(n-1)L in Q0*/
- pDataIn++;
- pBiquadState->pDelays[1]=(*pDataIn); /* Update x(n-1)R in Q0*/
- pDataIn++;
-
- /**************************************************************************
- WRITING THE OUTPUT
- ***************************************************************************/
- *pDataOut=(LVM_INT16)(ynL>>14); /* Write Left output in Q0*/
- pDataOut++;
- *pDataOut=(LVM_INT16)(ynR>>14); /* Write Right ouput in Q0*/
- pDataOut++;
- }
-
- }
-#endif
diff --git a/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F32C14_TRC_WRA_01.cpp b/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F32C14_TRC_WRA_01.cpp
new file mode 100644
index 0000000..3a396df
--- /dev/null
+++ b/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F32C14_TRC_WRA_01.cpp
@@ -0,0 +1,110 @@
+/*
+ * Copyright (C) 2004-2010 NXP Software
+ * Copyright (C) 2010 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include "BIQUAD.h"
+#include "BQ_2I_D16F32Css_TRC_WRA_01_Private.h"
+#include "LVM_Macros.h"
+
+/**************************************************************************
+ ASSUMPTIONS:
+ COEFS-
+ pBiquadState->coefs[0] is A2, pBiquadState->coefs[1] is A1
+ pBiquadState->coefs[2] is A0, pBiquadState->coefs[3] is -B2
+ pBiquadState->coefs[4] is -B1, these are in Q14 format
+
+ DELAYS-
+ pBiquadState->pDelays[0] is x(n-1)L in Q0 format
+ pBiquadState->pDelays[1] is x(n-1)R in Q0 format
+ pBiquadState->pDelays[2] is x(n-2)L in Q0 format
+ pBiquadState->pDelays[3] is x(n-2)R in Q0 format
+ pBiquadState->pDelays[4] is y(n-1)L in Q16 format
+ pBiquadState->pDelays[5] is y(n-1)R in Q16 format
+ pBiquadState->pDelays[6] is y(n-2)L in Q16 format
+ pBiquadState->pDelays[7] is y(n-2)R in Q16 format
+***************************************************************************/
+void BQ_2I_D16F32C14_TRC_WRA_01 ( Biquad_FLOAT_Instance_t *pInstance,
+ LVM_FLOAT *pDataIn,
+ LVM_FLOAT *pDataOut,
+ LVM_INT16 NrSamples)
+ {
+ LVM_FLOAT ynL,ynR;
+ LVM_INT16 ii;
+ PFilter_State_FLOAT pBiquadState = (PFilter_State_FLOAT) pInstance;
+
+ for (ii = NrSamples; ii != 0; ii--)
+ {
+
+ /**************************************************************************
+ PROCESSING OF THE LEFT CHANNEL
+ ***************************************************************************/
+ /* ynL=A2 * x(n-2)L */
+ ynL = (LVM_FLOAT)pBiquadState->coefs[0] * pBiquadState->pDelays[2];
+
+ /* ynL+=A1 * x(n-1)L */
+ ynL += (LVM_FLOAT)pBiquadState->coefs[1] * pBiquadState->pDelays[0];
+
+ /* ynL+=A0 * x(n)L */
+ ynL += (LVM_FLOAT)pBiquadState->coefs[2] * (*pDataIn);
+
+ /* ynL+= ( (-B2 * y(n-2)L ))*/
+ ynL += pBiquadState->pDelays[6] * pBiquadState->coefs[3];
+
+ /* ynL+=( (-B1 * y(n-1)L )) */
+ ynL += pBiquadState->pDelays[4] * pBiquadState->coefs[4];
+
+ /**************************************************************************
+ PROCESSING OF THE RIGHT CHANNEL
+ ***************************************************************************/
+ /* ynR=A2 * x(n-2)R */
+ ynR = (LVM_FLOAT)pBiquadState->coefs[0] * pBiquadState->pDelays[3];
+
+ /* ynR+=A1 * x(n-1)R */
+ ynR += (LVM_FLOAT)pBiquadState->coefs[1] * pBiquadState->pDelays[1];
+
+ /* ynR+=A0 * x(n)R */
+ ynR += (LVM_FLOAT)pBiquadState->coefs[2] * (*(pDataIn+1));
+
+ /* ynR+= ( (-B2 * y(n-2)R ))*/
+ ynR += pBiquadState->pDelays[7] * pBiquadState->coefs[3];
+
+ /* ynR+=( (-B1 * y(n-1)R )) */
+ ynR += pBiquadState->pDelays[5] * pBiquadState->coefs[4];
+
+ /**************************************************************************
+ UPDATING THE DELAYS
+ ***************************************************************************/
+ pBiquadState->pDelays[7] = pBiquadState->pDelays[5]; /* y(n-2)R=y(n-1)R*/
+ pBiquadState->pDelays[6] = pBiquadState->pDelays[4]; /* y(n-2)L=y(n-1)L*/
+ pBiquadState->pDelays[3] = pBiquadState->pDelays[1]; /* x(n-2)R=x(n-1)R*/
+ pBiquadState->pDelays[2] = pBiquadState->pDelays[0]; /* x(n-2)L=x(n-1)L*/
+ pBiquadState->pDelays[5] = ynR; /* Update y(n-1)R */
+ pBiquadState->pDelays[4] = ynL; /* Update y(n-1)L */
+ pBiquadState->pDelays[0] = (*pDataIn); /* Update x(n-1)L */
+ pDataIn++;
+ pBiquadState->pDelays[1] = (*pDataIn); /* Update x(n-1)R */
+ pDataIn++;
+
+ /**************************************************************************
+ WRITING THE OUTPUT
+ ***************************************************************************/
+ *pDataOut = (LVM_FLOAT)(ynL); /* Write Left output */
+ pDataOut++;
+ *pDataOut = (LVM_FLOAT)(ynR); /* Write Right ouput */
+ pDataOut++;
+ }
+
+ }
diff --git a/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F32C15_TRC_WRA_01.c b/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F32C15_TRC_WRA_01.c
deleted file mode 100644
index 2611b19..0000000
--- a/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F32C15_TRC_WRA_01.c
+++ /dev/null
@@ -1,194 +0,0 @@
-/*
- * Copyright (C) 2004-2010 NXP Software
- * Copyright (C) 2010 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include "BIQUAD.h"
-#include "BQ_2I_D16F32Css_TRC_WRA_01_Private.h"
-#include "LVM_Macros.h"
-
-/**************************************************************************
- ASSUMPTIONS:
- COEFS-
- pBiquadState->coefs[0] is A2, pBiquadState->coefs[1] is A1
- pBiquadState->coefs[2] is A0, pBiquadState->coefs[3] is -B2
- pBiquadState->coefs[4] is -B1, these are in Q15 format
-
- DELAYS-
- pBiquadState->pDelays[0] is x(n-1)L in Q0 format
- pBiquadState->pDelays[1] is x(n-1)R in Q0 format
- pBiquadState->pDelays[2] is x(n-2)L in Q0 format
- pBiquadState->pDelays[3] is x(n-2)R in Q0 format
- pBiquadState->pDelays[4] is y(n-1)L in Q16 format
- pBiquadState->pDelays[5] is y(n-1)R in Q16 format
- pBiquadState->pDelays[6] is y(n-2)L in Q16 format
- pBiquadState->pDelays[7] is y(n-2)R in Q16 format
-***************************************************************************/
-#ifdef BUILD_FLOAT
-void BQ_2I_D16F32C15_TRC_WRA_01 ( Biquad_FLOAT_Instance_t *pInstance,
- LVM_FLOAT *pDataIn,
- LVM_FLOAT *pDataOut,
- LVM_INT16 NrSamples)
- {
- LVM_FLOAT ynL,ynR;
- LVM_INT16 ii;
- PFilter_State_FLOAT pBiquadState = (PFilter_State_FLOAT) pInstance;
-
- for (ii = NrSamples; ii != 0; ii--)
- {
-
-
- /**************************************************************************
- PROCESSING OF THE LEFT CHANNEL
- ***************************************************************************/
- /* ynL=A2 * x(n-2)L */
- ynL = (LVM_FLOAT)pBiquadState->coefs[0] * pBiquadState->pDelays[2];
-
- /* ynL+=A1 * x(n-1)L */
- ynL += (LVM_FLOAT)pBiquadState->coefs[1] * pBiquadState->pDelays[0];
-
- /* ynL+=A0 * x(n)L */
- ynL += (LVM_FLOAT)pBiquadState->coefs[2] * (*pDataIn);
-
- /* ynL+= ( (-B2 * y(n-2)L ) */
- ynL += pBiquadState->pDelays[6] * pBiquadState->coefs[3];
-
-
- /* ynL+=( (-B1 * y(n-1)L )) */
- ynL += pBiquadState->pDelays[4] * pBiquadState->coefs[4];
-
-
- /**************************************************************************
- PROCESSING OF THE RIGHT CHANNEL
- ***************************************************************************/
- /* ynR=A2 * x(n-2)R */
- ynR = (LVM_FLOAT)pBiquadState->coefs[0] * pBiquadState->pDelays[3];
-
- /* ynR+=A1 * x(n-1)R */
- ynR += (LVM_FLOAT)pBiquadState->coefs[1] * pBiquadState->pDelays[1];
-
- /* ynR+=A0 * x(n)R */
- ynR += (LVM_FLOAT)pBiquadState->coefs[2] * (*(pDataIn+1));
-
- /* ynR+= ( (-B2 * y(n-2)R ) */
- ynR += pBiquadState->pDelays[7] * pBiquadState->coefs[3];
-
-
- /* ynR+=( (-B1 * y(n-1)R )) in Q15 */
- ynR += pBiquadState->pDelays[5] * pBiquadState->coefs[4];
-
- /**************************************************************************
- UPDATING THE DELAYS
- ***************************************************************************/
- pBiquadState->pDelays[7] = pBiquadState->pDelays[5]; /* y(n-2)R=y(n-1)R*/
- pBiquadState->pDelays[6] = pBiquadState->pDelays[4]; /* y(n-2)L=y(n-1)L*/
- pBiquadState->pDelays[3] = pBiquadState->pDelays[1]; /* x(n-2)R=x(n-1)R*/
- pBiquadState->pDelays[2] = pBiquadState->pDelays[0]; /* x(n-2)L=x(n-1)L*/
- pBiquadState->pDelays[5] = ynR; /* Update y(n-1)R*/
- pBiquadState->pDelays[4] = ynL; /* Update y(n-1)L*/
- pBiquadState->pDelays[0] = (*pDataIn); /* Update x(n-1)L*/
- pDataIn++;
- pBiquadState->pDelays[1] = (*pDataIn); /* Update x(n-1)R*/
- pDataIn++;
-
- /**************************************************************************
- WRITING THE OUTPUT
- ***************************************************************************/
- *pDataOut = (LVM_FLOAT)(ynL); /* Write Left output*/
- pDataOut++;
- *pDataOut = (LVM_FLOAT)(ynR); /* Write Right ouput*/
- pDataOut++;
- }
-
- }
-#else
-void BQ_2I_D16F32C15_TRC_WRA_01 ( Biquad_Instance_t *pInstance,
- LVM_INT16 *pDataIn,
- LVM_INT16 *pDataOut,
- LVM_INT16 NrSamples)
- {
- LVM_INT32 ynL,ynR,templ;
- LVM_INT16 ii;
- PFilter_State pBiquadState = (PFilter_State) pInstance;
-
- for (ii = NrSamples; ii != 0; ii--)
- {
-
-
- /**************************************************************************
- PROCESSING OF THE LEFT CHANNEL
- ***************************************************************************/
- /* ynL=A2 (Q15) * x(n-2)L (Q0) in Q15*/
- ynL=(LVM_INT32)pBiquadState->coefs[0]* pBiquadState->pDelays[2];
-
- /* ynL+=A1 (Q15) * x(n-1)L (Q0) in Q15*/
- ynL+=(LVM_INT32)pBiquadState->coefs[1]* pBiquadState->pDelays[0];
-
- /* ynL+=A0 (Q15) * x(n)L (Q0) in Q15*/
- ynL+=(LVM_INT32)pBiquadState->coefs[2]* (*pDataIn);
-
- /* ynL+= ( (-B2 (Q15) * y(n-2)L (Q16) )>>16) in Q15 */
- MUL32x16INTO32(pBiquadState->pDelays[6],pBiquadState->coefs[3],templ,16)
- ynL+=templ;
-
- /* ynL+=( (-B1 (Q15) * y(n-1)L (Q16) )>>16) in Q15 */
- MUL32x16INTO32(pBiquadState->pDelays[4],pBiquadState->coefs[4],templ,16)
- ynL+=templ;
-
- /**************************************************************************
- PROCESSING OF THE RIGHT CHANNEL
- ***************************************************************************/
- /* ynR=A2 (Q15) * x(n-2)R (Q0) in Q15*/
- ynR=(LVM_INT32)pBiquadState->coefs[0]*pBiquadState->pDelays[3];
-
- /* ynR+=A1 (Q15) * x(n-1)R (Q0) in Q15*/
- ynR+=(LVM_INT32)pBiquadState->coefs[1]*pBiquadState->pDelays[1];
-
- /* ynR+=A0 (Q15) * x(n)R (Q0) in Q15*/
- ynR+=(LVM_INT32)pBiquadState->coefs[2]*(*(pDataIn+1));
-
- /* ynR+= ( (-B2 (Q15) * y(n-2)R (Q16) )>>16) in Q15 */
- MUL32x16INTO32(pBiquadState->pDelays[7],pBiquadState->coefs[3],templ,16)
- ynR+=templ;
-
- /* ynR+=( (-B1 (Q15) * y(n-1)R (Q16) )>>16) in Q15 */
- MUL32x16INTO32(pBiquadState->pDelays[5],pBiquadState->coefs[4],templ,16)
- ynR+=templ;
-
- /**************************************************************************
- UPDATING THE DELAYS
- ***************************************************************************/
- pBiquadState->pDelays[7]=pBiquadState->pDelays[5]; /* y(n-2)R=y(n-1)R*/
- pBiquadState->pDelays[6]=pBiquadState->pDelays[4]; /* y(n-2)L=y(n-1)L*/
- pBiquadState->pDelays[3]=pBiquadState->pDelays[1]; /* x(n-2)R=x(n-1)R*/
- pBiquadState->pDelays[2]=pBiquadState->pDelays[0]; /* x(n-2)L=x(n-1)L*/
- pBiquadState->pDelays[5]=ynR<<1; /* Update y(n-1)R in Q16*/
- pBiquadState->pDelays[4]=ynL<<1; /* Update y(n-1)L in Q16*/
- pBiquadState->pDelays[0]=(*pDataIn); /* Update x(n-1)L in Q0*/
- pDataIn++;
- pBiquadState->pDelays[1]=(*pDataIn); /* Update x(n-1)R in Q0*/
- pDataIn++;
-
- /**************************************************************************
- WRITING THE OUTPUT
- ***************************************************************************/
- *pDataOut=(LVM_INT16)(ynL>>15); /* Write Left output in Q0*/
- pDataOut++;
- *pDataOut=(LVM_INT16)(ynR>>15); /* Write Right ouput in Q0*/
- pDataOut++;
- }
-
- }
-#endif
diff --git a/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F32C15_TRC_WRA_01.cpp b/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F32C15_TRC_WRA_01.cpp
new file mode 100644
index 0000000..1cbff1a
--- /dev/null
+++ b/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F32C15_TRC_WRA_01.cpp
@@ -0,0 +1,110 @@
+/*
+ * Copyright (C) 2004-2010 NXP Software
+ * Copyright (C) 2010 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include "BIQUAD.h"
+#include "BQ_2I_D16F32Css_TRC_WRA_01_Private.h"
+#include "LVM_Macros.h"
+
+/**************************************************************************
+ ASSUMPTIONS:
+ COEFS-
+ pBiquadState->coefs[0] is A2, pBiquadState->coefs[1] is A1
+ pBiquadState->coefs[2] is A0, pBiquadState->coefs[3] is -B2
+ pBiquadState->coefs[4] is -B1, these are in Q15 format
+
+ DELAYS-
+ pBiquadState->pDelays[0] is x(n-1)L in Q0 format
+ pBiquadState->pDelays[1] is x(n-1)R in Q0 format
+ pBiquadState->pDelays[2] is x(n-2)L in Q0 format
+ pBiquadState->pDelays[3] is x(n-2)R in Q0 format
+ pBiquadState->pDelays[4] is y(n-1)L in Q16 format
+ pBiquadState->pDelays[5] is y(n-1)R in Q16 format
+ pBiquadState->pDelays[6] is y(n-2)L in Q16 format
+ pBiquadState->pDelays[7] is y(n-2)R in Q16 format
+***************************************************************************/
+void BQ_2I_D16F32C15_TRC_WRA_01 ( Biquad_FLOAT_Instance_t *pInstance,
+ LVM_FLOAT *pDataIn,
+ LVM_FLOAT *pDataOut,
+ LVM_INT16 NrSamples)
+ {
+ LVM_FLOAT ynL,ynR;
+ LVM_INT16 ii;
+ PFilter_State_FLOAT pBiquadState = (PFilter_State_FLOAT) pInstance;
+
+ for (ii = NrSamples; ii != 0; ii--)
+ {
+
+ /**************************************************************************
+ PROCESSING OF THE LEFT CHANNEL
+ ***************************************************************************/
+ /* ynL=A2 * x(n-2)L */
+ ynL = (LVM_FLOAT)pBiquadState->coefs[0] * pBiquadState->pDelays[2];
+
+ /* ynL+=A1 * x(n-1)L */
+ ynL += (LVM_FLOAT)pBiquadState->coefs[1] * pBiquadState->pDelays[0];
+
+ /* ynL+=A0 * x(n)L */
+ ynL += (LVM_FLOAT)pBiquadState->coefs[2] * (*pDataIn);
+
+ /* ynL+= ( (-B2 * y(n-2)L ) */
+ ynL += pBiquadState->pDelays[6] * pBiquadState->coefs[3];
+
+ /* ynL+=( (-B1 * y(n-1)L )) */
+ ynL += pBiquadState->pDelays[4] * pBiquadState->coefs[4];
+
+ /**************************************************************************
+ PROCESSING OF THE RIGHT CHANNEL
+ ***************************************************************************/
+ /* ynR=A2 * x(n-2)R */
+ ynR = (LVM_FLOAT)pBiquadState->coefs[0] * pBiquadState->pDelays[3];
+
+ /* ynR+=A1 * x(n-1)R */
+ ynR += (LVM_FLOAT)pBiquadState->coefs[1] * pBiquadState->pDelays[1];
+
+ /* ynR+=A0 * x(n)R */
+ ynR += (LVM_FLOAT)pBiquadState->coefs[2] * (*(pDataIn+1));
+
+ /* ynR+= ( (-B2 * y(n-2)R ) */
+ ynR += pBiquadState->pDelays[7] * pBiquadState->coefs[3];
+
+ /* ynR+=( (-B1 * y(n-1)R )) in Q15 */
+ ynR += pBiquadState->pDelays[5] * pBiquadState->coefs[4];
+
+ /**************************************************************************
+ UPDATING THE DELAYS
+ ***************************************************************************/
+ pBiquadState->pDelays[7] = pBiquadState->pDelays[5]; /* y(n-2)R=y(n-1)R*/
+ pBiquadState->pDelays[6] = pBiquadState->pDelays[4]; /* y(n-2)L=y(n-1)L*/
+ pBiquadState->pDelays[3] = pBiquadState->pDelays[1]; /* x(n-2)R=x(n-1)R*/
+ pBiquadState->pDelays[2] = pBiquadState->pDelays[0]; /* x(n-2)L=x(n-1)L*/
+ pBiquadState->pDelays[5] = ynR; /* Update y(n-1)R*/
+ pBiquadState->pDelays[4] = ynL; /* Update y(n-1)L*/
+ pBiquadState->pDelays[0] = (*pDataIn); /* Update x(n-1)L*/
+ pDataIn++;
+ pBiquadState->pDelays[1] = (*pDataIn); /* Update x(n-1)R*/
+ pDataIn++;
+
+ /**************************************************************************
+ WRITING THE OUTPUT
+ ***************************************************************************/
+ *pDataOut = (LVM_FLOAT)(ynL); /* Write Left output*/
+ pDataOut++;
+ *pDataOut = (LVM_FLOAT)(ynR); /* Write Right ouput*/
+ pDataOut++;
+ }
+
+ }
diff --git a/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F32Css_TRC_WRA_01_Private.h b/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F32Css_TRC_WRA_01_Private.h
index c0319c9..314388a 100644
--- a/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F32Css_TRC_WRA_01_Private.h
+++ b/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F32Css_TRC_WRA_01_Private.h
@@ -28,7 +28,6 @@
typedef Filter_State * PFilter_State ;
-#ifdef BUILD_FLOAT
typedef struct _Filter_State_FLOAT
{
LVM_FLOAT * pDelays; /* pointer to the delayed samples \
@@ -36,6 +35,5 @@
LVM_FLOAT coefs[5]; /* pointer to the filter coefficients */
}Filter_State_FLOAT;
typedef Filter_State_FLOAT * PFilter_State_FLOAT ;
-#endif
#endif /* _BQ_2I_D16F32CSS_TRC_WRA_01_PRIVATE_H_ */
diff --git a/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F32Css_TRC_WRA_01_init.c b/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F32Css_TRC_WRA_01_init.cpp
similarity index 81%
rename from media/libeffects/lvm/lib/Common/src/BQ_2I_D16F32Css_TRC_WRA_01_init.c
rename to media/libeffects/lvm/lib/Common/src/BQ_2I_D16F32Css_TRC_WRA_01_init.cpp
index 4d9bbfe..058541a 100644
--- a/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F32Css_TRC_WRA_01_init.c
+++ b/media/libeffects/lvm/lib/Common/src/BQ_2I_D16F32Css_TRC_WRA_01_init.cpp
@@ -18,7 +18,6 @@
#include "BIQUAD.h"
#include "BQ_2I_D16F32Css_TRC_WRA_01_Private.h"
-
/*-------------------------------------------------------------------------*/
/* FUNCTION: */
/* BQ_2I_D16F32Css_TRC_WRA_01_Init */
@@ -37,7 +36,6 @@
/* RETURNS: */
/* void return code */
/*-------------------------------------------------------------------------*/
-#ifdef BUILD_FLOAT
void BQ_2I_D16F32Css_TRC_WRA_01_Init ( Biquad_FLOAT_Instance_t *pInstance,
Biquad_2I_Order2_FLOAT_Taps_t *pTaps,
BQ_FLOAT_Coefs_t *pCoef)
@@ -56,27 +54,6 @@
temp = pCoef->B1;
pBiquadState->coefs[4] = temp;
}
-#else
-void BQ_2I_D16F32Css_TRC_WRA_01_Init ( Biquad_Instance_t *pInstance,
- Biquad_2I_Order2_Taps_t *pTaps,
- BQ_C16_Coefs_t *pCoef)
-{
- LVM_INT16 temp;
- PFilter_State pBiquadState = (PFilter_State) pInstance;
- pBiquadState->pDelays =(LVM_INT32 *) pTaps ;
-
- temp=pCoef->A2;
- pBiquadState->coefs[0]=temp;
- temp=pCoef->A1;
- pBiquadState->coefs[1]=temp;
- temp=pCoef->A0;
- pBiquadState->coefs[2]=temp;
- temp=pCoef->B2;
- pBiquadState->coefs[3]=temp;
- temp=pCoef->B1;
- pBiquadState->coefs[4]=temp;
-}
-#endif
/*-------------------------------------------------------------------------*/
/* End Of File: BQ_2I_D16F32Css_TRC_WRA_01_Init */
diff --git a/media/libeffects/lvm/lib/Common/src/BQ_2I_D32F32C30_TRC_WRA_01.c b/media/libeffects/lvm/lib/Common/src/BQ_2I_D32F32C30_TRC_WRA_01.cpp
similarity index 68%
rename from media/libeffects/lvm/lib/Common/src/BQ_2I_D32F32C30_TRC_WRA_01.c
rename to media/libeffects/lvm/lib/Common/src/BQ_2I_D32F32C30_TRC_WRA_01.cpp
index d63365c..78d1ba1 100644
--- a/media/libeffects/lvm/lib/Common/src/BQ_2I_D32F32C30_TRC_WRA_01.c
+++ b/media/libeffects/lvm/lib/Common/src/BQ_2I_D32F32C30_TRC_WRA_01.cpp
@@ -36,13 +36,11 @@
pBiquadState->pDelays[6] is y(n-2)L in Q0 format
pBiquadState->pDelays[7] is y(n-2)R in Q0 format
***************************************************************************/
-#ifdef BUILD_FLOAT
void BQ_2I_D32F32C30_TRC_WRA_01 ( Biquad_FLOAT_Instance_t *pInstance,
LVM_FLOAT *pDataIn,
LVM_FLOAT *pDataOut,
LVM_INT16 NrSamples)
-
{
LVM_FLOAT ynL,ynR,templ,tempd;
LVM_INT16 ii;
@@ -51,7 +49,6 @@
for (ii = NrSamples; ii != 0; ii--)
{
-
/**************************************************************************
PROCESSING OF THE LEFT CHANNEL
***************************************************************************/
@@ -119,7 +116,6 @@
*pDataOut = (LVM_FLOAT)ynR; /* Write Right ouput */
pDataOut++;
-
}
}
@@ -151,7 +147,6 @@
LVM_INT16 NrFrames,
LVM_INT16 NrChannels)
-
{
LVM_FLOAT yn, temp;
LVM_INT16 ii, jj;
@@ -204,91 +199,3 @@
}
#endif /*SUPPORT_MC*/
-#else
-void BQ_2I_D32F32C30_TRC_WRA_01 ( Biquad_Instance_t *pInstance,
- LVM_INT32 *pDataIn,
- LVM_INT32 *pDataOut,
- LVM_INT16 NrSamples)
-
-
- {
- LVM_INT32 ynL,ynR,templ,tempd;
- LVM_INT16 ii;
- PFilter_State pBiquadState = (PFilter_State) pInstance;
-
- for (ii = NrSamples; ii != 0; ii--)
- {
-
-
- /**************************************************************************
- PROCESSING OF THE LEFT CHANNEL
- ***************************************************************************/
- /* ynL= ( A2 (Q30) * x(n-2)L (Q0) ) >>30 in Q0*/
- MUL32x32INTO32(pBiquadState->coefs[0],pBiquadState->pDelays[2],ynL,30)
-
- /* ynL+= ( A1 (Q30) * x(n-1)L (Q0) ) >> 30 in Q0*/
- MUL32x32INTO32(pBiquadState->coefs[1],pBiquadState->pDelays[0],templ,30)
- ynL+=templ;
-
- /* ynL+= ( A0 (Q30) * x(n)L (Q0) ) >> 30 in Q0*/
- MUL32x32INTO32(pBiquadState->coefs[2],*pDataIn,templ,30)
- ynL+=templ;
-
- /* ynL+= (-B2 (Q30) * y(n-2)L (Q0) ) >> 30 in Q0*/
- MUL32x32INTO32(pBiquadState->coefs[3],pBiquadState->pDelays[6],templ,30)
- ynL+=templ;
-
- /* ynL+= (-B1 (Q30) * y(n-1)L (Q0) ) >> 30 in Q0 */
- MUL32x32INTO32(pBiquadState->coefs[4],pBiquadState->pDelays[4],templ,30)
- ynL+=templ;
-
- /**************************************************************************
- PROCESSING OF THE RIGHT CHANNEL
- ***************************************************************************/
- /* ynR= ( A2 (Q30) * x(n-2)R (Q0) ) >> 30 in Q0*/
- MUL32x32INTO32(pBiquadState->coefs[0],pBiquadState->pDelays[3],ynR,30)
-
- /* ynR+= ( A1 (Q30) * x(n-1)R (Q0) ) >> 30 in Q0*/
- MUL32x32INTO32(pBiquadState->coefs[1],pBiquadState->pDelays[1],templ,30)
- ynR+=templ;
-
- /* ynR+= ( A0 (Q30) * x(n)R (Q0) ) >> 30 in Q0*/
- tempd=*(pDataIn+1);
- MUL32x32INTO32(pBiquadState->coefs[2],tempd,templ,30)
- ynR+=templ;
-
- /* ynR+= (-B2 (Q30) * y(n-2)R (Q0) ) >> 30 in Q0*/
- MUL32x32INTO32(pBiquadState->coefs[3],pBiquadState->pDelays[7],templ,30)
- ynR+=templ;
-
- /* ynR+= (-B1 (Q30) * y(n-1)R (Q0) ) >> 30 in Q0 */
- MUL32x32INTO32(pBiquadState->coefs[4],pBiquadState->pDelays[5],templ,30)
- ynR+=templ;
-
- /**************************************************************************
- UPDATING THE DELAYS
- ***************************************************************************/
- pBiquadState->pDelays[7]=pBiquadState->pDelays[5]; /* y(n-2)R=y(n-1)R*/
- pBiquadState->pDelays[6]=pBiquadState->pDelays[4]; /* y(n-2)L=y(n-1)L*/
- pBiquadState->pDelays[3]=pBiquadState->pDelays[1]; /* x(n-2)R=x(n-1)R*/
- pBiquadState->pDelays[2]=pBiquadState->pDelays[0]; /* x(n-2)L=x(n-1)L*/
- pBiquadState->pDelays[5]=(LVM_INT32)ynR; /* Update y(n-1)R in Q0*/
- pBiquadState->pDelays[4]=(LVM_INT32)ynL; /* Update y(n-1)L in Q0*/
- pBiquadState->pDelays[0]=(*pDataIn); /* Update x(n-1)L in Q0*/
- pDataIn++;
- pBiquadState->pDelays[1]=(*pDataIn); /* Update x(n-1)R in Q0*/
- pDataIn++;
-
- /**************************************************************************
- WRITING THE OUTPUT
- ***************************************************************************/
- *pDataOut=(LVM_INT32)ynL; /* Write Left output in Q0*/
- pDataOut++;
- *pDataOut=(LVM_INT32)ynR; /* Write Right ouput in Q0*/
- pDataOut++;
-
-
- }
-
- }
-#endif /*BUILD_FLOAT*/
diff --git a/media/libeffects/lvm/lib/Common/src/BQ_2I_D32F32Cll_TRC_WRA_01_Init.c b/media/libeffects/lvm/lib/Common/src/BQ_2I_D32F32Cll_TRC_WRA_01_Init.cpp
similarity index 82%
rename from media/libeffects/lvm/lib/Common/src/BQ_2I_D32F32Cll_TRC_WRA_01_Init.c
rename to media/libeffects/lvm/lib/Common/src/BQ_2I_D32F32Cll_TRC_WRA_01_Init.cpp
index fff05ed..492a9e0 100644
--- a/media/libeffects/lvm/lib/Common/src/BQ_2I_D32F32Cll_TRC_WRA_01_Init.c
+++ b/media/libeffects/lvm/lib/Common/src/BQ_2I_D32F32Cll_TRC_WRA_01_Init.cpp
@@ -37,7 +37,6 @@
/* RETURNS: */
/* void return code */
/*-------------------------------------------------------------------------*/
-#ifdef BUILD_FLOAT
void BQ_2I_D32F32Cll_TRC_WRA_01_Init ( Biquad_FLOAT_Instance_t *pInstance,
Biquad_2I_Order2_FLOAT_Taps_t *pTaps,
BQ_FLOAT_Coefs_t *pCoef)
@@ -56,27 +55,6 @@
temp = pCoef->B1;
pBiquadState->coefs[4] = temp;
}
-#else
-void BQ_2I_D32F32Cll_TRC_WRA_01_Init ( Biquad_Instance_t *pInstance,
- Biquad_2I_Order2_Taps_t *pTaps,
- BQ_C32_Coefs_t *pCoef)
-{
- LVM_INT32 temp;
- PFilter_State pBiquadState = (PFilter_State) pInstance;
- pBiquadState->pDelays =(LVM_INT32 *) pTaps ;
-
- temp=pCoef->A2;
- pBiquadState->coefs[0]=temp;
- temp=pCoef->A1;
- pBiquadState->coefs[1]=temp;
- temp=pCoef->A0;
- pBiquadState->coefs[2]=temp;
- temp=pCoef->B2;
- pBiquadState->coefs[3]=temp;
- temp=pCoef->B1;
- pBiquadState->coefs[4]=temp;
-}
-#endif
/*-------------------------------------------------------------------------*/
/* End Of File: BQ_2I_D32F32C32_TRC_WRA_01_Init.c */
diff --git a/media/libeffects/lvm/lib/Common/src/BQ_2I_D32F32Cll_TRC_WRA_01_Private.h b/media/libeffects/lvm/lib/Common/src/BQ_2I_D32F32Cll_TRC_WRA_01_Private.h
index c0f0dcc..7eb6474 100644
--- a/media/libeffects/lvm/lib/Common/src/BQ_2I_D32F32Cll_TRC_WRA_01_Private.h
+++ b/media/libeffects/lvm/lib/Common/src/BQ_2I_D32F32Cll_TRC_WRA_01_Private.h
@@ -18,7 +18,6 @@
#ifndef _BQ_2I_D32F32CLL_TRC_WRA_01_PRIVATE_H_
#define _BQ_2I_D32F32CLL_TRC_WRA_01_PRIVATE_H_
-
/* The internal state variables are implemented in a (for the user) hidden structure */
/* In this (private) file, the internal structure is declared fro private use. */
typedef struct _Filter_State_
@@ -29,7 +28,6 @@
typedef Filter_State * PFilter_State ;
-#ifdef BUILD_FLOAT
typedef struct _Filter_State_FLOAT
{
LVM_FLOAT * pDelays; /* pointer to the delayed samples \
@@ -37,6 +35,5 @@
LVM_FLOAT coefs[5]; /* pointer to the filter coefficients */
}Filter_State_FLOAT;
typedef Filter_State_FLOAT * PFilter_State_FLOAT ;
-#endif
#endif /* _BQ_2I_D32F32CLL_TRC_WRA_01_PRIVATE_H_*/
diff --git a/media/libeffects/lvm/lib/Common/src/Copy_16.c b/media/libeffects/lvm/lib/Common/src/Copy_16.cpp
similarity index 98%
rename from media/libeffects/lvm/lib/Common/src/Copy_16.c
rename to media/libeffects/lvm/lib/Common/src/Copy_16.cpp
index 3858450..7cb642f 100644
--- a/media/libeffects/lvm/lib/Common/src/Copy_16.c
+++ b/media/libeffects/lvm/lib/Common/src/Copy_16.cpp
@@ -54,7 +54,6 @@
return;
}
-#ifdef BUILD_FLOAT
void Copy_Float( const LVM_FLOAT *src,
LVM_FLOAT *dst,
LVM_INT16 n )
@@ -144,5 +143,4 @@
}
}
#endif
-#endif
/**********************************************************************************/
diff --git a/media/libeffects/lvm/lib/Common/src/Core_MixHard_2St_D32C31_SAT.c b/media/libeffects/lvm/lib/Common/src/Core_MixHard_2St_D32C31_SAT.cpp
similarity index 67%
rename from media/libeffects/lvm/lib/Common/src/Core_MixHard_2St_D32C31_SAT.c
rename to media/libeffects/lvm/lib/Common/src/Core_MixHard_2St_D32C31_SAT.cpp
index ea98041..5e77335 100644
--- a/media/libeffects/lvm/lib/Common/src/Core_MixHard_2St_D32C31_SAT.c
+++ b/media/libeffects/lvm/lib/Common/src/Core_MixHard_2St_D32C31_SAT.cpp
@@ -25,7 +25,6 @@
/**********************************************************************************
FUNCTION CORE_MIXHARD_2ST_D32C31_SAT
***********************************************************************************/
-#ifdef BUILD_FLOAT
void Core_MixHard_2St_D32C31_SAT( Mix_2St_Cll_FLOAT_t *pInstance,
const LVM_FLOAT *src1,
const LVM_FLOAT *src2,
@@ -55,35 +54,4 @@
*dst++ = Temp2;
}
}
-#else
-void Core_MixHard_2St_D32C31_SAT( Mix_2St_Cll_t *pInstance,
- const LVM_INT32 *src1,
- const LVM_INT32 *src2,
- LVM_INT32 *dst,
- LVM_INT16 n)
-{
- LVM_INT32 Temp1,Temp2,Temp3;
- LVM_INT16 ii;
- LVM_INT16 Current1Short;
- LVM_INT16 Current2Short;
-
- Current1Short = (LVM_INT16)(pInstance->Current1 >> 16);
- Current2Short = (LVM_INT16)(pInstance->Current2 >> 16);
-
- for (ii = n; ii != 0; ii--){
- Temp1=*src1++;
- MUL32x16INTO32(Temp1,Current1Short,Temp3,15)
- Temp2=*src2++;
- MUL32x16INTO32(Temp2,Current2Short,Temp1,15)
- Temp2=(Temp1>>1)+(Temp3>>1);
- if (Temp2 > 0x3FFFFFFF)
- Temp2 = 0x7FFFFFFF;
- else if (Temp2 < - 0x40000000)
- Temp2 = 0x80000000;
- else
- Temp2=(Temp2<<1);
- *dst++ = Temp2;
- }
-}
-#endif
/**********************************************************************************/
diff --git a/media/libeffects/lvm/lib/Common/src/Core_MixInSoft_D32C31_SAT.c b/media/libeffects/lvm/lib/Common/src/Core_MixInSoft_D32C31_SAT.c
deleted file mode 100644
index 2814f19..0000000
--- a/media/libeffects/lvm/lib/Common/src/Core_MixInSoft_D32C31_SAT.c
+++ /dev/null
@@ -1,157 +0,0 @@
-/*
- * Copyright (C) 2004-2010 NXP Software
- * Copyright (C) 2010 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-/**********************************************************************************
- INCLUDE FILES
-***********************************************************************************/
-
-#include "Mixer_private.h"
-#include "LVM_Macros.h"
-
-/**********************************************************************************
- FUNCTION CORE_MIXSOFT_1ST_D32C31_WRA
-***********************************************************************************/
-
-#ifdef BUILD_FLOAT /* BUILD_FLOAT */
-void Core_MixInSoft_D32C31_SAT( Mix_1St_Cll_FLOAT_t *pInstance,
- const LVM_FLOAT *src,
- LVM_FLOAT *dst,
- LVM_INT16 n)
-{
- LVM_FLOAT Temp1,Temp2,Temp3;
- LVM_INT16 OutLoop;
- LVM_INT16 InLoop;
- LVM_FLOAT TargetTimesOneMinAlpha;
- LVM_FLOAT CurrentTimesAlpha;
- LVM_INT16 ii,jj;
-
-
- InLoop = (LVM_INT16)(n >> 2); /* Process per 4 samples */
- OutLoop = (LVM_INT16)(n - (InLoop << 2));
-
- TargetTimesOneMinAlpha = ((1.0f -pInstance->Alpha) * pInstance->Target);
- if (pInstance->Target >= pInstance->Current){
- TargetTimesOneMinAlpha +=(LVM_FLOAT)(2.0f / 2147483647.0f); /* Ceil*/
- }
-
- if (OutLoop){
-
- CurrentTimesAlpha = pInstance->Current * pInstance->Alpha;
- pInstance->Current = TargetTimesOneMinAlpha + CurrentTimesAlpha;
-
- for (ii = OutLoop; ii != 0; ii--){
- Temp1 = *src++;
- Temp2 = *dst;
-
- Temp3 = Temp1 * (pInstance->Current);
- Temp1 = Temp2 + Temp3;
-
- if (Temp1 > 1.0f)
- Temp1 = 1.0f;
- else if (Temp1 < -1.0f)
- Temp1 = -1.0f;
-
- *dst++ = Temp1;
- }
- }
-
- for (ii = InLoop; ii != 0; ii--){
-
- CurrentTimesAlpha = pInstance->Current * pInstance->Alpha;
- pInstance->Current = TargetTimesOneMinAlpha + CurrentTimesAlpha;
-
- for (jj = 4; jj!=0 ; jj--){
- Temp1 = *src++;
- Temp2 = *dst;
-
- Temp3 = Temp1 * (pInstance->Current);
- Temp1 = Temp2 + Temp3;
-
- if (Temp1 > 1.0f)
- Temp1 = 1.0f;
- else if (Temp1 < -1.0f)
- Temp1 = -1.0f;
- *dst++ = Temp1;
- }
- }
-}
-#else
-void Core_MixInSoft_D32C31_SAT( Mix_1St_Cll_t *pInstance,
- const LVM_INT32 *src,
- LVM_INT32 *dst,
- LVM_INT16 n)
-{
- LVM_INT32 Temp1,Temp2,Temp3;
- LVM_INT16 OutLoop;
- LVM_INT16 InLoop;
- LVM_INT32 TargetTimesOneMinAlpha;
- LVM_INT32 CurrentTimesAlpha;
- LVM_INT16 ii,jj;
- LVM_INT16 CurrentShort;
-
- InLoop = (LVM_INT16)(n >> 2); /* Process per 4 samples */
- OutLoop = (LVM_INT16)(n - (InLoop << 2));
-
- MUL32x32INTO32((0x7FFFFFFF-pInstance->Alpha),pInstance->Target,TargetTimesOneMinAlpha,31); /* Q31 * Q0 in Q0 */
- if (pInstance->Target >= pInstance->Current){
- TargetTimesOneMinAlpha +=2; /* Ceil*/
- }
-
- if (OutLoop){
- MUL32x32INTO32(pInstance->Current,pInstance->Alpha,CurrentTimesAlpha,31); /* Q0 * Q31 in Q0 */
- pInstance->Current = TargetTimesOneMinAlpha + CurrentTimesAlpha; /* Q0 + Q0 into Q0*/
- CurrentShort = (LVM_INT16)(pInstance->Current>>16); /* From Q31 to Q15*/
-
- for (ii = OutLoop; ii != 0; ii--){
- Temp1=*src++;
- Temp2=*dst;
- MUL32x16INTO32(Temp1,CurrentShort,Temp3,15)
- Temp1=(Temp2>>1)+(Temp3>>1);
-
- if (Temp1 > 0x3FFFFFFF)
- Temp1 = 0x7FFFFFFF;
- else if (Temp1 < - 0x40000000)
- Temp1 = 0x80000000;
- else
- Temp1=(Temp1<<1);
- *dst++ = Temp1;
- }
- }
-
- for (ii = InLoop; ii != 0; ii--){
- MUL32x32INTO32(pInstance->Current,pInstance->Alpha,CurrentTimesAlpha,31); /* Q0 * Q31 in Q0 */
- pInstance->Current = TargetTimesOneMinAlpha + CurrentTimesAlpha; /* Q0 + Q0 into Q0*/
- CurrentShort = (LVM_INT16)(pInstance->Current>>16); /* From Q31 to Q15*/
-
- for (jj = 4; jj!=0 ; jj--){
- Temp1=*src++;
- Temp2=*dst;
- MUL32x16INTO32(Temp1,CurrentShort,Temp3,15)
- Temp1=(Temp2>>1)+(Temp3>>1);
-
- if (Temp1 > 0x3FFFFFFF)
- Temp1 = 0x7FFFFFFF;
- else if (Temp1 < - 0x40000000)
- Temp1 = 0x80000000;
- else
- Temp1=(Temp1<<1);
- *dst++ = Temp1;
- }
- }
-}
-#endif
-/**********************************************************************************/
diff --git a/media/libeffects/lvm/lib/Common/src/Core_MixInSoft_D32C31_SAT.cpp b/media/libeffects/lvm/lib/Common/src/Core_MixInSoft_D32C31_SAT.cpp
new file mode 100644
index 0000000..8f5c0ae
--- /dev/null
+++ b/media/libeffects/lvm/lib/Common/src/Core_MixInSoft_D32C31_SAT.cpp
@@ -0,0 +1,90 @@
+/*
+ * Copyright (C) 2004-2010 NXP Software
+ * Copyright (C) 2010 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+/**********************************************************************************
+ INCLUDE FILES
+***********************************************************************************/
+
+#include "Mixer_private.h"
+#include "LVM_Macros.h"
+
+/**********************************************************************************
+ FUNCTION CORE_MIXSOFT_1ST_D32C31_WRA
+***********************************************************************************/
+
+void Core_MixInSoft_D32C31_SAT( Mix_1St_Cll_FLOAT_t *pInstance,
+ const LVM_FLOAT *src,
+ LVM_FLOAT *dst,
+ LVM_INT16 n)
+{
+ LVM_FLOAT Temp1,Temp2,Temp3;
+ LVM_INT16 OutLoop;
+ LVM_INT16 InLoop;
+ LVM_FLOAT TargetTimesOneMinAlpha;
+ LVM_FLOAT CurrentTimesAlpha;
+ LVM_INT16 ii,jj;
+
+ InLoop = (LVM_INT16)(n >> 2); /* Process per 4 samples */
+ OutLoop = (LVM_INT16)(n - (InLoop << 2));
+
+ TargetTimesOneMinAlpha = ((1.0f -pInstance->Alpha) * pInstance->Target);
+ if (pInstance->Target >= pInstance->Current){
+ TargetTimesOneMinAlpha +=(LVM_FLOAT)(2.0f / 2147483647.0f); /* Ceil*/
+ }
+
+ if (OutLoop){
+
+ CurrentTimesAlpha = pInstance->Current * pInstance->Alpha;
+ pInstance->Current = TargetTimesOneMinAlpha + CurrentTimesAlpha;
+
+ for (ii = OutLoop; ii != 0; ii--){
+ Temp1 = *src++;
+ Temp2 = *dst;
+
+ Temp3 = Temp1 * (pInstance->Current);
+ Temp1 = Temp2 + Temp3;
+
+ if (Temp1 > 1.0f)
+ Temp1 = 1.0f;
+ else if (Temp1 < -1.0f)
+ Temp1 = -1.0f;
+
+ *dst++ = Temp1;
+ }
+ }
+
+ for (ii = InLoop; ii != 0; ii--){
+
+ CurrentTimesAlpha = pInstance->Current * pInstance->Alpha;
+ pInstance->Current = TargetTimesOneMinAlpha + CurrentTimesAlpha;
+
+ for (jj = 4; jj!=0 ; jj--){
+ Temp1 = *src++;
+ Temp2 = *dst;
+
+ Temp3 = Temp1 * (pInstance->Current);
+ Temp1 = Temp2 + Temp3;
+
+ if (Temp1 > 1.0f)
+ Temp1 = 1.0f;
+ else if (Temp1 < -1.0f)
+ Temp1 = -1.0f;
+ *dst++ = Temp1;
+ }
+ }
+}
+/**********************************************************************************/
diff --git a/media/libeffects/lvm/lib/Common/src/Core_MixSoft_1St_D32C31_WRA.c b/media/libeffects/lvm/lib/Common/src/Core_MixSoft_1St_D32C31_WRA.c
deleted file mode 100644
index 814ccee..0000000
--- a/media/libeffects/lvm/lib/Common/src/Core_MixSoft_1St_D32C31_WRA.c
+++ /dev/null
@@ -1,174 +0,0 @@
-/*
- * Copyright (C) 2004-2010 NXP Software
- * Copyright (C) 2010 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-/**********************************************************************************
- INCLUDE FILES
-***********************************************************************************/
-
-#include "Mixer_private.h"
-#include "LVM_Macros.h"
-
-/**********************************************************************************
- FUNCTION CORE_MIXSOFT_1ST_D32C31_WRA
-***********************************************************************************/
-#ifdef BUILD_FLOAT
-void Core_MixSoft_1St_D32C31_WRA( Mix_1St_Cll_FLOAT_t *pInstance,
- const LVM_FLOAT *src,
- LVM_FLOAT *dst,
- LVM_INT16 n)
-{
- LVM_FLOAT Temp1,Temp2;
- LVM_INT16 OutLoop;
- LVM_INT16 InLoop;
- LVM_FLOAT TargetTimesOneMinAlpha;
- LVM_FLOAT CurrentTimesAlpha;
-
- LVM_INT16 ii;
-
- InLoop = (LVM_INT16)(n >> 2); /* Process per 4 samples */
- OutLoop = (LVM_INT16)(n - (InLoop << 2));
-
- TargetTimesOneMinAlpha = (1.0f - pInstance->Alpha) * pInstance->Target; /* float * float in float */
- if (pInstance->Target >= pInstance->Current)
- {
- TargetTimesOneMinAlpha += (LVM_FLOAT)(2.0f / 2147483647.0f); /* Ceil*/
- }
-
- if (OutLoop != 0)
- {
- CurrentTimesAlpha = (pInstance->Current * pInstance->Alpha);
- pInstance->Current = TargetTimesOneMinAlpha + CurrentTimesAlpha;
-
- for (ii = OutLoop; ii != 0; ii--)
- {
- Temp1 = *src;
- src++;
-
- Temp2 = Temp1 * (pInstance->Current);
- *dst = Temp2;
- dst++;
- }
- }
-
- for (ii = InLoop; ii != 0; ii--)
- {
- CurrentTimesAlpha = pInstance->Current * pInstance->Alpha;
- pInstance->Current = TargetTimesOneMinAlpha + CurrentTimesAlpha;
-
- Temp1 = *src;
- src++;
-
- Temp2 = Temp1 * (pInstance->Current);
- *dst = Temp2;
- dst++;
-
- Temp1 = *src;
- src++;
-
- Temp2 = Temp1 * (pInstance->Current);
- *dst = Temp2;
- dst++;
-
- Temp1 = *src;
- src++;
-
- Temp2 = Temp1 * (pInstance->Current);
- *dst = Temp2;
- dst++;
-
- Temp1 = *src;
- src++;
- Temp2 = Temp1 * (pInstance->Current);
- *dst = Temp2;
- dst++;
- }
-}
-#else
-void Core_MixSoft_1St_D32C31_WRA( Mix_1St_Cll_t *pInstance,
- const LVM_INT32 *src,
- LVM_INT32 *dst,
- LVM_INT16 n)
-{
- LVM_INT32 Temp1,Temp2;
- LVM_INT16 OutLoop;
- LVM_INT16 InLoop;
- LVM_INT32 TargetTimesOneMinAlpha;
- LVM_INT32 CurrentTimesAlpha;
- LVM_INT16 CurrentShort;
- LVM_INT16 ii;
-
- InLoop = (LVM_INT16)(n >> 2); /* Process per 4 samples */
- OutLoop = (LVM_INT16)(n - (InLoop << 2));
-
- MUL32x32INTO32((0x7FFFFFFF-pInstance->Alpha),pInstance->Target,TargetTimesOneMinAlpha,31) /* Q31 * Q31 in Q31 */
- if (pInstance->Target >= pInstance->Current)
- {
- TargetTimesOneMinAlpha +=2; /* Ceil*/
- }
-
- if (OutLoop!=0)
- {
- MUL32x32INTO32(pInstance->Current,pInstance->Alpha,CurrentTimesAlpha,31) /* Q31 * Q31 in Q31 */
- pInstance->Current = TargetTimesOneMinAlpha + CurrentTimesAlpha; /* Q31 + Q31 into Q31*/
- CurrentShort = (LVM_INT16)(pInstance->Current>>16); /* From Q31 to Q15*/
-
- for (ii = OutLoop; ii != 0; ii--)
- {
- Temp1=*src;
- src++;
-
- MUL32x16INTO32(Temp1,CurrentShort,Temp2,15)
- *dst = Temp2;
- dst++;
- }
- }
-
- for (ii = InLoop; ii != 0; ii--)
- {
- MUL32x32INTO32(pInstance->Current,pInstance->Alpha,CurrentTimesAlpha,31) /* Q31 * Q31 in Q31 */
- pInstance->Current = TargetTimesOneMinAlpha + CurrentTimesAlpha; /* Q31 + Q31 into Q31*/
- CurrentShort = (LVM_INT16)(pInstance->Current>>16); /* From Q31 to Q15*/
- Temp1=*src;
- src++;
-
- MUL32x16INTO32(Temp1,CurrentShort,Temp2,15)
- *dst = Temp2;
- dst++;
-
- Temp1=*src;
- src++;
-
- MUL32x16INTO32(Temp1,CurrentShort,Temp2,15)
- *dst = Temp2;
- dst++;
-
- Temp1=*src;
- src++;
-
- MUL32x16INTO32(Temp1,CurrentShort,Temp2,15)
- *dst = Temp2;
- dst++;
-
- Temp1=*src;
- src++;
- MUL32x16INTO32(Temp1,CurrentShort,Temp2,15)
- *dst = Temp2;
- dst++;
- }
-}
-#endif
-/**********************************************************************************/
diff --git a/media/libeffects/lvm/lib/Common/src/Core_MixSoft_1St_D32C31_WRA.cpp b/media/libeffects/lvm/lib/Common/src/Core_MixSoft_1St_D32C31_WRA.cpp
new file mode 100644
index 0000000..6ff7853
--- /dev/null
+++ b/media/libeffects/lvm/lib/Common/src/Core_MixSoft_1St_D32C31_WRA.cpp
@@ -0,0 +1,99 @@
+/*
+ * Copyright (C) 2004-2010 NXP Software
+ * Copyright (C) 2010 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+/**********************************************************************************
+ INCLUDE FILES
+***********************************************************************************/
+
+#include "Mixer_private.h"
+#include "LVM_Macros.h"
+
+/**********************************************************************************
+ FUNCTION CORE_MIXSOFT_1ST_D32C31_WRA
+***********************************************************************************/
+void Core_MixSoft_1St_D32C31_WRA( Mix_1St_Cll_FLOAT_t *pInstance,
+ const LVM_FLOAT *src,
+ LVM_FLOAT *dst,
+ LVM_INT16 n)
+{
+ LVM_FLOAT Temp1,Temp2;
+ LVM_INT16 OutLoop;
+ LVM_INT16 InLoop;
+ LVM_FLOAT TargetTimesOneMinAlpha;
+ LVM_FLOAT CurrentTimesAlpha;
+
+ LVM_INT16 ii;
+
+ InLoop = (LVM_INT16)(n >> 2); /* Process per 4 samples */
+ OutLoop = (LVM_INT16)(n - (InLoop << 2));
+
+ TargetTimesOneMinAlpha = (1.0f - pInstance->Alpha) * pInstance->Target; /* float * float in float */
+ if (pInstance->Target >= pInstance->Current)
+ {
+ TargetTimesOneMinAlpha += (LVM_FLOAT)(2.0f / 2147483647.0f); /* Ceil*/
+ }
+
+ if (OutLoop != 0)
+ {
+ CurrentTimesAlpha = (pInstance->Current * pInstance->Alpha);
+ pInstance->Current = TargetTimesOneMinAlpha + CurrentTimesAlpha;
+
+ for (ii = OutLoop; ii != 0; ii--)
+ {
+ Temp1 = *src;
+ src++;
+
+ Temp2 = Temp1 * (pInstance->Current);
+ *dst = Temp2;
+ dst++;
+ }
+ }
+
+ for (ii = InLoop; ii != 0; ii--)
+ {
+ CurrentTimesAlpha = pInstance->Current * pInstance->Alpha;
+ pInstance->Current = TargetTimesOneMinAlpha + CurrentTimesAlpha;
+
+ Temp1 = *src;
+ src++;
+
+ Temp2 = Temp1 * (pInstance->Current);
+ *dst = Temp2;
+ dst++;
+
+ Temp1 = *src;
+ src++;
+
+ Temp2 = Temp1 * (pInstance->Current);
+ *dst = Temp2;
+ dst++;
+
+ Temp1 = *src;
+ src++;
+
+ Temp2 = Temp1 * (pInstance->Current);
+ *dst = Temp2;
+ dst++;
+
+ Temp1 = *src;
+ src++;
+ Temp2 = Temp1 * (pInstance->Current);
+ *dst = Temp2;
+ dst++;
+ }
+}
+/**********************************************************************************/
diff --git a/media/libeffects/lvm/lib/Common/src/DC_2I_D16_TRC_WRA_01.c b/media/libeffects/lvm/lib/Common/src/DC_2I_D16_TRC_WRA_01.cpp
similarity index 71%
rename from media/libeffects/lvm/lib/Common/src/DC_2I_D16_TRC_WRA_01.c
rename to media/libeffects/lvm/lib/Common/src/DC_2I_D16_TRC_WRA_01.cpp
index 13fac5e..a7ce4d3 100644
--- a/media/libeffects/lvm/lib/Common/src/DC_2I_D16_TRC_WRA_01.c
+++ b/media/libeffects/lvm/lib/Common/src/DC_2I_D16_TRC_WRA_01.cpp
@@ -18,7 +18,6 @@
#include "BIQUAD.h"
#include "DC_2I_D16_TRC_WRA_01_Private.h"
#include "LVM_Macros.h"
-#ifdef BUILD_FLOAT
void DC_2I_D16_TRC_WRA_01( Biquad_FLOAT_Instance_t *pInstance,
LVM_FLOAT *pDataIn,
LVM_FLOAT *pDataOut,
@@ -45,7 +44,6 @@
else {
LeftDC += DC_FLOAT_STEP; }
-
/* Subtract DC an saturate */
Diff =* (pDataIn++) - (RightDC);
if (Diff > 1.0f) {
@@ -62,7 +60,6 @@
pBiquadState->LeftDC = LeftDC;
pBiquadState->RightDC = RightDC;
-
}
#ifdef SUPPORT_MC
/*
@@ -116,50 +113,3 @@
}
#endif
-#else
-void DC_2I_D16_TRC_WRA_01( Biquad_Instance_t *pInstance,
- LVM_INT16 *pDataIn,
- LVM_INT16 *pDataOut,
- LVM_INT16 NrSamples)
- {
- LVM_INT32 LeftDC,RightDC;
- LVM_INT32 Diff;
- LVM_INT32 j;
- PFilter_State pBiquadState = (PFilter_State) pInstance;
-
- LeftDC = pBiquadState->LeftDC;
- RightDC = pBiquadState->RightDC;
- for(j=NrSamples-1;j>=0;j--)
- {
- /* Subtract DC an saturate */
- Diff=*(pDataIn++)-(LeftDC>>16);
- if (Diff > 32767) {
- Diff = 32767; }
- else if (Diff < -32768) {
- Diff = -32768; }
- *(pDataOut++)=(LVM_INT16)Diff;
- if (Diff < 0) {
- LeftDC -= DC_D16_STEP; }
- else {
- LeftDC += DC_D16_STEP; }
-
-
- /* Subtract DC an saturate */
- Diff=*(pDataIn++)-(RightDC>>16);
- if (Diff > 32767) {
- Diff = 32767; }
- else if (Diff < -32768) {
- Diff = -32768; }
- *(pDataOut++)=(LVM_INT16)Diff;
- if (Diff < 0) {
- RightDC -= DC_D16_STEP; }
- else {
- RightDC += DC_D16_STEP; }
-
- }
- pBiquadState->LeftDC = LeftDC;
- pBiquadState->RightDC = RightDC;
-
-
- }
-#endif
diff --git a/media/libeffects/lvm/lib/Common/src/DC_2I_D16_TRC_WRA_01_Init.c b/media/libeffects/lvm/lib/Common/src/DC_2I_D16_TRC_WRA_01_Init.cpp
similarity index 83%
rename from media/libeffects/lvm/lib/Common/src/DC_2I_D16_TRC_WRA_01_Init.c
rename to media/libeffects/lvm/lib/Common/src/DC_2I_D16_TRC_WRA_01_Init.cpp
index 0f941a0..beee112 100644
--- a/media/libeffects/lvm/lib/Common/src/DC_2I_D16_TRC_WRA_01_Init.c
+++ b/media/libeffects/lvm/lib/Common/src/DC_2I_D16_TRC_WRA_01_Init.cpp
@@ -17,7 +17,6 @@
#include "BIQUAD.h"
#include "DC_2I_D16_TRC_WRA_01_Private.h"
-#ifdef BUILD_FLOAT
void DC_2I_D16_TRC_WRA_01_Init(Biquad_FLOAT_Instance_t *pInstance)
{
PFilter_FLOAT_State pBiquadState = (PFilter_FLOAT_State) pInstance;
@@ -35,11 +34,3 @@
}
}
#endif
-#else
-void DC_2I_D16_TRC_WRA_01_Init(Biquad_Instance_t *pInstance)
-{
- PFilter_State pBiquadState = (PFilter_State) pInstance;
- pBiquadState->LeftDC = 0;
- pBiquadState->RightDC = 0;
-}
-#endif
diff --git a/media/libeffects/lvm/lib/Common/src/DC_2I_D16_TRC_WRA_01_Private.h b/media/libeffects/lvm/lib/Common/src/DC_2I_D16_TRC_WRA_01_Private.h
index db3a6d3..4170b3c 100644
--- a/media/libeffects/lvm/lib/Common/src/DC_2I_D16_TRC_WRA_01_Private.h
+++ b/media/libeffects/lvm/lib/Common/src/DC_2I_D16_TRC_WRA_01_Private.h
@@ -18,16 +18,10 @@
#ifndef _DC_2I_D16_TRC_WRA_01_PRIVATE_H_
#define _DC_2I_D16_TRC_WRA_01_PRIVATE_H_
-#ifdef BUILD_FLOAT
-#define DC_FLOAT_STEP 0.0000002384f;
-#else
-#define DC_D16_STEP 0x200;
-#endif
-
+#define DC_FLOAT_STEP 0.0000002384f
/* The internal state variables are implemented in a (for the user) hidden structure */
/* In this (private) file, the internal structure is declared fro private use.*/
-#ifdef BUILD_FLOAT
typedef struct _Filter_FLOAT_State_
{
LVM_FLOAT LeftDC; /* LeftDC */
@@ -41,13 +35,4 @@
} Filter_FLOAT_State_Mc;
typedef Filter_FLOAT_State_Mc * PFilter_FLOAT_State_Mc ;
#endif
-#else
-typedef struct _Filter_State_
-{
- LVM_INT32 LeftDC; /* LeftDC */
- LVM_INT32 RightDC; /* RightDC */
-}Filter_State;
-
-typedef Filter_State * PFilter_State ;
-#endif
#endif /* _DC_2I_D16_TRC_WRA_01_PRIVATE_H_ */
diff --git a/media/libeffects/lvm/lib/Common/src/DelayAllPass_Sat_32x16To32.c b/media/libeffects/lvm/lib/Common/src/DelayAllPass_Sat_32x16To32.cpp
similarity index 99%
rename from media/libeffects/lvm/lib/Common/src/DelayAllPass_Sat_32x16To32.c
rename to media/libeffects/lvm/lib/Common/src/DelayAllPass_Sat_32x16To32.cpp
index b04e98e..771fae2 100644
--- a/media/libeffects/lvm/lib/Common/src/DelayAllPass_Sat_32x16To32.c
+++ b/media/libeffects/lvm/lib/Common/src/DelayAllPass_Sat_32x16To32.cpp
@@ -63,7 +63,6 @@
*dst = c;
dst++;
-
MUL32x16INTO32(c, -coeff, temp, 15)
a = temp;
b = delay[AllPassOffset];
diff --git a/media/libeffects/lvm/lib/Common/src/DelayMix_16x16.c b/media/libeffects/lvm/lib/Common/src/DelayMix_16x16.cpp
similarity index 98%
rename from media/libeffects/lvm/lib/Common/src/DelayMix_16x16.c
rename to media/libeffects/lvm/lib/Common/src/DelayMix_16x16.cpp
index f502716..52d263f 100644
--- a/media/libeffects/lvm/lib/Common/src/DelayMix_16x16.c
+++ b/media/libeffects/lvm/lib/Common/src/DelayMix_16x16.cpp
@@ -47,7 +47,6 @@
Offset++;
src++;
-
/* Right channel */
temp = (LVM_INT16)((LVM_UINT32)((LVM_INT32)(*dst) - (LVM_INT32)delay[Offset]) >> 1);
*dst = temp;
@@ -69,7 +68,6 @@
return;
}
-#ifdef BUILD_FLOAT
void DelayMix_Float(const LVM_FLOAT *src, /* Source 1, to be delayed */
LVM_FLOAT *delay, /* Delay buffer */
LVM_INT16 size, /* Delay size */
@@ -92,7 +90,6 @@
Offset++;
src++;
-
/* Right channel */
temp = (LVM_FLOAT)((LVM_FLOAT)(*dst - (LVM_FLOAT)delay[Offset]) / 2.0f);
*dst = temp;
@@ -114,5 +111,4 @@
return;
}
-#endif
/**********************************************************************************/
diff --git a/media/libeffects/lvm/lib/Common/src/DelayWrite_32.c b/media/libeffects/lvm/lib/Common/src/DelayWrite_32.cpp
similarity index 100%
rename from media/libeffects/lvm/lib/Common/src/DelayWrite_32.c
rename to media/libeffects/lvm/lib/Common/src/DelayWrite_32.cpp
diff --git a/media/libeffects/lvm/lib/Common/src/FO_1I_D16F16C15_TRC_WRA_01.c b/media/libeffects/lvm/lib/Common/src/FO_1I_D16F16C15_TRC_WRA_01.cpp
similarity index 60%
rename from media/libeffects/lvm/lib/Common/src/FO_1I_D16F16C15_TRC_WRA_01.c
rename to media/libeffects/lvm/lib/Common/src/FO_1I_D16F16C15_TRC_WRA_01.cpp
index 039c88c..bef0d62 100644
--- a/media/libeffects/lvm/lib/Common/src/FO_1I_D16F16C15_TRC_WRA_01.c
+++ b/media/libeffects/lvm/lib/Common/src/FO_1I_D16F16C15_TRC_WRA_01.cpp
@@ -31,7 +31,6 @@
pBiquadState->pDelays[1] is y(n-1)L in Q0 format
***************************************************************************/
-#ifdef BUILD_FLOAT
void FO_1I_D16F16C15_TRC_WRA_01( Biquad_FLOAT_Instance_t *pInstance,
LVM_FLOAT *pDataIn,
LVM_FLOAT *pDataOut,
@@ -70,45 +69,3 @@
}
}
-#else
-void FO_1I_D16F16C15_TRC_WRA_01( Biquad_Instance_t *pInstance,
- LVM_INT16 *pDataIn,
- LVM_INT16 *pDataOut,
- LVM_INT16 NrSamples)
- {
- LVM_INT32 ynL;
- LVM_INT16 ii;
- PFilter_State pBiquadState = (PFilter_State) pInstance;
-
- for (ii = NrSamples; ii != 0; ii--)
- {
-
- /**************************************************************************
- PROCESSING OF THE LEFT CHANNEL
- ***************************************************************************/
- // ynL=A1 (Q15) * x(n-1)L (Q0) in Q15
- ynL=(LVM_INT32)pBiquadState->coefs[0]* pBiquadState->pDelays[0];
-
- // ynL+=A0 (Q15) * x(n)L (Q0) in Q15
- ynL+=(LVM_INT32)pBiquadState->coefs[1]* (*pDataIn);
-
- // ynL+= (-B1 (Q15) * y(n-1)L (Q0) ) in Q15
- ynL+=(LVM_INT32)pBiquadState->coefs[2]*pBiquadState->pDelays[1];
-
-
- ynL=(LVM_INT16)(ynL>>15); // ynL in Q0 format
- /**************************************************************************
- UPDATING THE DELAYS
- ***************************************************************************/
- pBiquadState->pDelays[1]=ynL; // Update y(n-1)L in Q0
- pBiquadState->pDelays[0]=(*pDataIn++); // Update x(n-1)L in Q0
-
- /**************************************************************************
- WRITING THE OUTPUT
- ***************************************************************************/
- *pDataOut++=(LVM_INT16)ynL; // Write Left output in Q0
-
- }
-
- }
-#endif
diff --git a/media/libeffects/lvm/lib/Common/src/FO_1I_D16F16Css_TRC_WRA_01_Init.c b/media/libeffects/lvm/lib/Common/src/FO_1I_D16F16Css_TRC_WRA_01_Init.cpp
similarity index 83%
rename from media/libeffects/lvm/lib/Common/src/FO_1I_D16F16Css_TRC_WRA_01_Init.c
rename to media/libeffects/lvm/lib/Common/src/FO_1I_D16F16Css_TRC_WRA_01_Init.cpp
index b21b8a4..161225e 100644
--- a/media/libeffects/lvm/lib/Common/src/FO_1I_D16F16Css_TRC_WRA_01_Init.c
+++ b/media/libeffects/lvm/lib/Common/src/FO_1I_D16F16Css_TRC_WRA_01_Init.cpp
@@ -19,7 +19,6 @@
#include "BIQUAD.h"
#include "FO_1I_D16F16Css_TRC_WRA_01_Private.h"
-
/*-------------------------------------------------------------------------*/
/* FUNCTION: */
/* FO_1I_D16F16Css_TRC_WRA_01_Init */
@@ -38,7 +37,6 @@
/* RETURNS: */
/* void return code */
/*-------------------------------------------------------------------------*/
-#ifdef BUILD_FLOAT
void FO_1I_D16F16Css_TRC_WRA_01_Init( Biquad_FLOAT_Instance_t *pInstance,
Biquad_1I_Order1_FLOAT_Taps_t *pTaps,
FO_FLOAT_Coefs_t *pCoef)
@@ -53,23 +51,6 @@
temp = pCoef->B1;
pBiquadState->coefs[2] = temp;
}
-#else
-void FO_1I_D16F16Css_TRC_WRA_01_Init( Biquad_Instance_t *pInstance,
- Biquad_1I_Order1_Taps_t *pTaps,
- FO_C16_Coefs_t *pCoef)
-{
- LVM_INT16 temp;
- PFilter_State pBiquadState = (PFilter_State) pInstance;
- pBiquadState->pDelays =(LVM_INT32 *) pTaps;
-
- temp=pCoef->A1;
- pBiquadState->coefs[0]=temp;
- temp=pCoef->A0;
- pBiquadState->coefs[1]=temp;
- temp=pCoef->B1;
- pBiquadState->coefs[2]=temp;
-}
-#endif
/*------------------------------------------------*/
/* End Of File: FO_1I_D16F16Css_TRC_WRA_01_Init.c */
diff --git a/media/libeffects/lvm/lib/Common/src/FO_1I_D16F16Css_TRC_WRA_01_Private.h b/media/libeffects/lvm/lib/Common/src/FO_1I_D16F16Css_TRC_WRA_01_Private.h
index 6fdb039..34f3df9 100644
--- a/media/libeffects/lvm/lib/Common/src/FO_1I_D16F16Css_TRC_WRA_01_Private.h
+++ b/media/libeffects/lvm/lib/Common/src/FO_1I_D16F16Css_TRC_WRA_01_Private.h
@@ -28,7 +28,6 @@
typedef Filter_State * PFilter_State ;
-#ifdef BUILD_FLOAT
typedef struct _Filter_State_FLOAT
{
LVM_FLOAT * pDelays; /* pointer to the delayed samples \
@@ -37,5 +36,4 @@
}Filter_State_FLOAT;
typedef Filter_State_FLOAT * PFilter_State_FLOAT ;
-#endif
#endif /* _FO_1I_D16F16CSS_TRC_WRA_01_PRIVATE_H_ */
diff --git a/media/libeffects/lvm/lib/Common/src/FO_1I_D32F32C31_TRC_WRA_01.c b/media/libeffects/lvm/lib/Common/src/FO_1I_D32F32C31_TRC_WRA_01.cpp
similarity index 60%
rename from media/libeffects/lvm/lib/Common/src/FO_1I_D32F32C31_TRC_WRA_01.c
rename to media/libeffects/lvm/lib/Common/src/FO_1I_D32F32C31_TRC_WRA_01.cpp
index 416e8eb..e3efad7 100644
--- a/media/libeffects/lvm/lib/Common/src/FO_1I_D32F32C31_TRC_WRA_01.c
+++ b/media/libeffects/lvm/lib/Common/src/FO_1I_D32F32C31_TRC_WRA_01.cpp
@@ -19,7 +19,6 @@
#include "FO_1I_D32F32Cll_TRC_WRA_01_Private.h"
#include "LVM_Macros.h"
-
/**************************************************************************
ASSUMPTIONS:
COEFS-
@@ -31,7 +30,6 @@
pBiquadState->pDelays[0] is x(n-1)L in Q0 format
pBiquadState->pDelays[1] is y(n-1)L in Q0 format
***************************************************************************/
-#ifdef BUILD_FLOAT
void FO_1I_D32F32C31_TRC_WRA_01( Biquad_FLOAT_Instance_t *pInstance,
LVM_FLOAT *pDataIn,
LVM_FLOAT *pDataOut,
@@ -71,44 +69,3 @@
}
}
-#else
-void FO_1I_D32F32C31_TRC_WRA_01( Biquad_Instance_t *pInstance,
- LVM_INT32 *pDataIn,
- LVM_INT32 *pDataOut,
- LVM_INT16 NrSamples)
- {
- LVM_INT32 ynL,templ;
- LVM_INT16 ii;
- PFilter_State pBiquadState = (PFilter_State) pInstance;
-
- for (ii = NrSamples; ii != 0; ii--)
- {
-
- /**************************************************************************
- PROCESSING OF THE LEFT CHANNEL
- ***************************************************************************/
- // ynL=A1 (Q31) * x(n-1)L (Q0) >>31 in Q0
- MUL32x32INTO32(pBiquadState->coefs[0],pBiquadState->pDelays[0],ynL,31)
-
- // ynL+=A0 (Q31) * x(n)L (Q0) >> 31 in Q0
- MUL32x32INTO32(pBiquadState->coefs[1],*pDataIn,templ,31)
- ynL+=templ;
-
- // ynL+= (-B1 (Q31) * y(n-1)L (Q0) ) >> 31 in Q0
- MUL32x32INTO32(pBiquadState->coefs[2],pBiquadState->pDelays[1],templ,31)
- ynL+=templ;
-
- /**************************************************************************
- UPDATING THE DELAYS
- ***************************************************************************/
- pBiquadState->pDelays[1]=ynL; // Update y(n-1)L in Q0
- pBiquadState->pDelays[0]=(*pDataIn++); // Update x(n-1)L in Q0
-
- /**************************************************************************
- WRITING THE OUTPUT
- ***************************************************************************/
- *pDataOut++=(LVM_INT32)ynL; // Write Left output in Q0
- }
-
- }
-#endif
diff --git a/media/libeffects/lvm/lib/Common/src/FO_1I_D32F32Cll_TRC_WRA_01_Init.c b/media/libeffects/lvm/lib/Common/src/FO_1I_D32F32Cll_TRC_WRA_01_Init.cpp
similarity index 83%
rename from media/libeffects/lvm/lib/Common/src/FO_1I_D32F32Cll_TRC_WRA_01_Init.c
rename to media/libeffects/lvm/lib/Common/src/FO_1I_D32F32Cll_TRC_WRA_01_Init.cpp
index f33d24d..bb5295c 100644
--- a/media/libeffects/lvm/lib/Common/src/FO_1I_D32F32Cll_TRC_WRA_01_Init.c
+++ b/media/libeffects/lvm/lib/Common/src/FO_1I_D32F32Cll_TRC_WRA_01_Init.cpp
@@ -18,7 +18,6 @@
#include "BIQUAD.h"
#include "FO_1I_D32F32Cll_TRC_WRA_01_Private.h"
-
/*-------------------------------------------------------------------------*/
/* FUNCTION: */
/* FO_1I_D32F32Cll_TRC_WRA_01_Init */
@@ -37,7 +36,6 @@
/* RETURNS: */
/* void return code */
/*-------------------------------------------------------------------------*/
-#ifdef BUILD_FLOAT
void FO_1I_D32F32Cll_TRC_WRA_01_Init( Biquad_FLOAT_Instance_t *pInstance,
Biquad_1I_Order1_FLOAT_Taps_t *pTaps,
FO_FLOAT_Coefs_t *pCoef)
@@ -53,23 +51,6 @@
temp = pCoef->B1;
pBiquadState->coefs[2] = temp;
}
-#else
-void FO_1I_D32F32Cll_TRC_WRA_01_Init( Biquad_Instance_t *pInstance,
- Biquad_1I_Order1_Taps_t *pTaps,
- FO_C32_Coefs_t *pCoef)
-{
- LVM_INT32 temp;
- PFilter_State pBiquadState = (PFilter_State) pInstance;
- pBiquadState->pDelays = (LVM_INT32 *) pTaps;
-
- temp=pCoef->A1;
- pBiquadState->coefs[0]=temp;
- temp=pCoef->A0;
- pBiquadState->coefs[1]=temp;
- temp=pCoef->B1;
- pBiquadState->coefs[2]=temp;
-}
-#endif
/*------------------------------------------------*/
/* End Of File: FO_1I_D32F32Cll_TRC_WRA_01_Init.c */
diff --git a/media/libeffects/lvm/lib/Common/src/FO_1I_D32F32Cll_TRC_WRA_01_Private.h b/media/libeffects/lvm/lib/Common/src/FO_1I_D32F32Cll_TRC_WRA_01_Private.h
index fdb528b..67d1384 100644
--- a/media/libeffects/lvm/lib/Common/src/FO_1I_D32F32Cll_TRC_WRA_01_Private.h
+++ b/media/libeffects/lvm/lib/Common/src/FO_1I_D32F32Cll_TRC_WRA_01_Private.h
@@ -18,7 +18,6 @@
#ifndef _FO_1I_D32F32CLL_TRC_WRA_01_PRIVATE_H_
#define _FO_1I_D32F32CLL_TRC_WRA_01_PRIVATE_H_
-
/* The internal state variables are implemented in a (for the user) hidden structure */
/* In this (private) file, the internal structure is declared fro private use. */
typedef struct _Filter_State_
@@ -29,7 +28,6 @@
typedef Filter_State * PFilter_State ;
-#ifdef BUILD_FLOAT
typedef struct _Filter_State_FLOAT_
{
LVM_FLOAT * pDelays; /* pointer to the delayed samples (data of 32 bits) */
@@ -37,5 +35,4 @@
}Filter_State_FLOAT;
typedef Filter_State_FLOAT * PFilter_State_FLOAT ;
-#endif
#endif /* _FO_1I_D32F32CLL_TRC_WRA_01_PRIVATE_H_ */
diff --git a/media/libeffects/lvm/lib/Common/src/FO_2I_D16F32C15_LShx_TRC_WRA_01.c b/media/libeffects/lvm/lib/Common/src/FO_2I_D16F32C15_LShx_TRC_WRA_01.cpp
similarity index 67%
rename from media/libeffects/lvm/lib/Common/src/FO_2I_D16F32C15_LShx_TRC_WRA_01.c
rename to media/libeffects/lvm/lib/Common/src/FO_2I_D16F32C15_LShx_TRC_WRA_01.cpp
index 2a50f18..6ca819a 100644
--- a/media/libeffects/lvm/lib/Common/src/FO_2I_D16F32C15_LShx_TRC_WRA_01.c
+++ b/media/libeffects/lvm/lib/Common/src/FO_2I_D16F32C15_LShx_TRC_WRA_01.cpp
@@ -32,7 +32,6 @@
pBiquadState->pDelays[2] is x(n-1)R in Q15 format
pBiquadState->pDelays[3] is y(n-1)R in Q30 format
***************************************************************************/
-#ifdef BUILD_FLOAT
void FO_2I_D16F32C15_LShx_TRC_WRA_01(Biquad_FLOAT_Instance_t *pInstance,
LVM_FLOAT *pDataIn,
LVM_FLOAT *pDataOut,
@@ -59,13 +58,11 @@
// ynR =A1 * x(n-1)R
ynR = (LVM_FLOAT)pBiquadState->coefs[0] * pBiquadState->pDelays[2];
-
// ynL+=A0 * x(n)L
ynL += (LVM_FLOAT)pBiquadState->coefs[1] * (*pDataIn);
// ynR+=A0 * x(n)L
ynR += (LVM_FLOAT)pBiquadState->coefs[1] * (*(pDataIn+1));
-
// ynL += (-B1 * y(n-1)L )
Temp = pBiquadState->pDelays[1] * pBiquadState->coefs[2];
ynL += Temp;
@@ -73,7 +70,6 @@
Temp = pBiquadState->pDelays[3] * pBiquadState->coefs[2];
ynR += Temp;
-
/**************************************************************************
UPDATING THE DELAYS
***************************************************************************/
@@ -157,9 +153,6 @@
LVM_FLOAT A1 = pCoefs[0];
LVM_FLOAT B1 = pCoefs[2];
-
-
-
for (ii = NrFrames; ii != 0; ii--)
{
@@ -178,7 +171,6 @@
Temp = B1 * pDelays[1];
yn += Temp;
-
/**************************************************************************
UPDATING THE DELAYS
***************************************************************************/
@@ -204,97 +196,3 @@
}
}
#endif
-#else
-void FO_2I_D16F32C15_LShx_TRC_WRA_01(Biquad_Instance_t *pInstance,
- LVM_INT16 *pDataIn,
- LVM_INT16 *pDataOut,
- LVM_INT16 NrSamples)
- {
- LVM_INT32 ynL,ynR;
- LVM_INT32 Temp;
- LVM_INT32 NegSatValue;
- LVM_INT16 ii;
- LVM_INT16 Shift;
- PFilter_State pBiquadState = (PFilter_State) pInstance;
-
- NegSatValue = LVM_MAXINT_16 +1;
- NegSatValue = -NegSatValue;
-
- Shift = pBiquadState->Shift;
-
-
- for (ii = NrSamples; ii != 0; ii--)
- {
-
- /**************************************************************************
- PROCESSING OF THE LEFT CHANNEL
- ***************************************************************************/
-
- // ynL =A1 (Q15) * x(n-1)L (Q15) in Q30
- ynL=(LVM_INT32)pBiquadState->coefs[0]* pBiquadState->pDelays[0];
- // ynR =A1 (Q15) * x(n-1)R (Q15) in Q30
- ynR=(LVM_INT32)pBiquadState->coefs[0]* pBiquadState->pDelays[2];
-
-
- // ynL+=A0 (Q15) * x(n)L (Q15) in Q30
- ynL+=(LVM_INT32)pBiquadState->coefs[1]* (*pDataIn);
- // ynR+=A0 (Q15) * x(n)L (Q15) in Q30
- ynR+=(LVM_INT32)pBiquadState->coefs[1]* (*(pDataIn+1));
-
-
- // ynL += (-B1 (Q15) * y(n-1)L (Q30) ) in Q30
- MUL32x16INTO32(pBiquadState->pDelays[1],pBiquadState->coefs[2],Temp,15);
- ynL +=Temp;
- // ynR += (-B1 (Q15) * y(n-1)R (Q30) ) in Q30
- MUL32x16INTO32(pBiquadState->pDelays[3],pBiquadState->coefs[2],Temp,15);
- ynR +=Temp;
-
-
- /**************************************************************************
- UPDATING THE DELAYS
- ***************************************************************************/
- pBiquadState->pDelays[1]=ynL; // Update y(n-1)L in Q30
- pBiquadState->pDelays[0]=(*pDataIn++); // Update x(n-1)L in Q15
-
- pBiquadState->pDelays[3]=ynR; // Update y(n-1)R in Q30
- pBiquadState->pDelays[2]=(*pDataIn++); // Update x(n-1)R in Q15
-
- /**************************************************************************
- WRITING THE OUTPUT
- ***************************************************************************/
- /*Apply shift: Instead of left shift on 16-bit result, right shift of (15-shift) is applied
- for better SNR*/
- ynL = ynL>>(15-Shift);
- ynR = ynR>>(15-Shift);
-
- /*Saturate results*/
- if(ynL > LVM_MAXINT_16)
- {
- ynL = LVM_MAXINT_16;
- }
- else
- {
- if(ynL < NegSatValue)
- {
- ynL = NegSatValue;
- }
- }
-
- if(ynR > LVM_MAXINT_16)
- {
- ynR = LVM_MAXINT_16;
- }
- else
- {
- if(ynR < NegSatValue)
- {
- ynR = NegSatValue;
- }
- }
-
- *pDataOut++=(LVM_INT16)ynL;
- *pDataOut++=(LVM_INT16)ynR;
- }
-
- }
-#endif
diff --git a/media/libeffects/lvm/lib/Common/src/FO_2I_D16F32Css_LShx_TRC_WRA_01_Init.c b/media/libeffects/lvm/lib/Common/src/FO_2I_D16F32Css_LShx_TRC_WRA_01_Init.cpp
similarity index 82%
rename from media/libeffects/lvm/lib/Common/src/FO_2I_D16F32Css_LShx_TRC_WRA_01_Init.c
rename to media/libeffects/lvm/lib/Common/src/FO_2I_D16F32Css_LShx_TRC_WRA_01_Init.cpp
index 33ca6cf..b81b976 100644
--- a/media/libeffects/lvm/lib/Common/src/FO_2I_D16F32Css_LShx_TRC_WRA_01_Init.c
+++ b/media/libeffects/lvm/lib/Common/src/FO_2I_D16F32Css_LShx_TRC_WRA_01_Init.cpp
@@ -37,7 +37,6 @@
/* RETURNS: */
/* void return code */
/*-------------------------------------------------------------------------*/
-#ifdef BUILD_FLOAT
void FO_2I_D16F32Css_LShx_TRC_WRA_01_Init(Biquad_FLOAT_Instance_t *pInstance,
Biquad_2I_Order1_FLOAT_Taps_t *pTaps,
FO_FLOAT_LShx_Coefs_t *pCoef)
@@ -53,26 +52,6 @@
temp = pCoef->B1;
pBiquadState->coefs[2] = temp;
}
-#else
-void FO_2I_D16F32Css_LShx_TRC_WRA_01_Init(Biquad_Instance_t *pInstance,
- Biquad_2I_Order1_Taps_t *pTaps,
- FO_C16_LShx_Coefs_t *pCoef)
-{
- LVM_INT16 temp;
- PFilter_State pBiquadState = (PFilter_State) pInstance;
- pBiquadState->pDelays =(LVM_INT32 *) pTaps ;
-
- temp=pCoef->A1;
- pBiquadState->coefs[0]=temp;
- temp=pCoef->A0;
- pBiquadState->coefs[1]=temp;
- temp=pCoef->B1;
- pBiquadState->coefs[2]=temp;
-
- temp=pCoef->Shift;
- pBiquadState->Shift = temp;
-}
-#endif
/*-------------------------------------------------------------------------*/
/* End Of File: FO_2I_D16F32Css_LShx_TRC_WRA_01_Init.c */
diff --git a/media/libeffects/lvm/lib/Common/src/FO_2I_D16F32Css_LShx_TRC_WRA_01_Private.h b/media/libeffects/lvm/lib/Common/src/FO_2I_D16F32Css_LShx_TRC_WRA_01_Private.h
index 368bfce..5022500 100644
--- a/media/libeffects/lvm/lib/Common/src/FO_2I_D16F32Css_LShx_TRC_WRA_01_Private.h
+++ b/media/libeffects/lvm/lib/Common/src/FO_2I_D16F32Css_LShx_TRC_WRA_01_Private.h
@@ -20,7 +20,6 @@
/* The internal state variables are implemented in a (for the user) hidden structure */
/* In this (private) file, the internal structure is declared fro private use. */
-#ifdef BUILD_FLOAT
typedef struct _Filter_State_
{
LVM_FLOAT *pDelays; /* pointer to the delayed samples (data of 32 bits) */
@@ -28,14 +27,4 @@
}Filter_Float_State;
typedef Filter_Float_State * PFilter_Float_State ;
-#else
-typedef struct _Filter_State_
-{
- LVM_INT32 *pDelays; /* pointer to the delayed samples (data of 32 bits) */
- LVM_INT16 coefs[3]; /* pointer to the filter coefficients */
- LVM_INT16 Shift; /* Shift value*/
-}Filter_State;
-
-typedef Filter_State * PFilter_State ;
-#endif
#endif /* _FO_2I_D16F32CSS_LSHX_TRC_WRA_01_PRIVATE_H_ */
diff --git a/media/libeffects/lvm/lib/Common/src/Filters.h b/media/libeffects/lvm/lib/Common/src/Filters.h
index b1fde0c..b5db8f4 100644
--- a/media/libeffects/lvm/lib/Common/src/Filters.h
+++ b/media/libeffects/lvm/lib/Common/src/Filters.h
@@ -18,10 +18,6 @@
#ifndef FILTERS_H
#define FILTERS_H
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
#include "LVM_Types.h"
/************************************************************************************/
@@ -34,17 +30,6 @@
* Biquad with coefficients A0, A1, A2, B1 and B2 coefficients
*/
/* Single precision (16-bit) Biquad section coefficients */
-#ifndef BUILD_FLOAT
-typedef struct
-{
- LVM_INT16 A0;
- LVM_INT16 A1;
- LVM_INT16 A2;
- LVM_INT16 B1;
- LVM_INT16 B2;
- LVM_UINT16 Scale;
-} BiquadA012B12CoefsSP_t;
-#else
typedef struct
{
LVM_FLOAT A0;
@@ -54,20 +39,10 @@
LVM_FLOAT B2;
LVM_UINT16 Scale;
} BiquadA012B12CoefsSP_t;
-#endif
/*
* Biquad with coefficients A0, A1 and B1 coefficients
*/
/* Single precision (16-bit) Biquad section coefficients */
-#ifndef BUILD_FLOAT
-typedef struct
-{
- LVM_INT16 A0;
- LVM_INT16 A1;
- LVM_INT16 B1;
- LVM_UINT16 Scale;
-} BiquadA01B1CoefsSP_t;
-#else
typedef struct
{
LVM_FLOAT A0;
@@ -75,10 +50,6 @@
LVM_FLOAT B1;
LVM_UINT16 Scale;
} BiquadA01B1CoefsSP_t;
-#endif
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
#endif /* FILTERS_H */
diff --git a/media/libeffects/lvm/lib/Common/src/From2iToMS_16x16.c b/media/libeffects/lvm/lib/Common/src/From2iToMS_16x16.cpp
similarity index 98%
rename from media/libeffects/lvm/lib/Common/src/From2iToMS_16x16.c
rename to media/libeffects/lvm/lib/Common/src/From2iToMS_16x16.cpp
index 2c6e6c3..c3f6648 100644
--- a/media/libeffects/lvm/lib/Common/src/From2iToMS_16x16.c
+++ b/media/libeffects/lvm/lib/Common/src/From2iToMS_16x16.cpp
@@ -53,7 +53,6 @@
return;
}
-#ifdef BUILD_FLOAT
void From2iToMS_Float( const LVM_FLOAT *src,
LVM_FLOAT *dstM,
LVM_FLOAT *dstS,
@@ -82,5 +81,4 @@
return;
}
-#endif
/**********************************************************************************/
diff --git a/media/libeffects/lvm/lib/Common/src/From2iToMono_16.c b/media/libeffects/lvm/lib/Common/src/From2iToMono_16.cpp
similarity index 100%
rename from media/libeffects/lvm/lib/Common/src/From2iToMono_16.c
rename to media/libeffects/lvm/lib/Common/src/From2iToMono_16.cpp
diff --git a/media/libeffects/lvm/lib/Common/src/From2iToMono_32.c b/media/libeffects/lvm/lib/Common/src/From2iToMono_32.cpp
similarity index 98%
rename from media/libeffects/lvm/lib/Common/src/From2iToMono_32.c
rename to media/libeffects/lvm/lib/Common/src/From2iToMono_32.cpp
index d02af88..a8688b4 100644
--- a/media/libeffects/lvm/lib/Common/src/From2iToMono_32.c
+++ b/media/libeffects/lvm/lib/Common/src/From2iToMono_32.cpp
@@ -46,7 +46,6 @@
return;
}
-#ifdef BUILD_FLOAT
void From2iToMono_Float( const LVM_FLOAT *src,
LVM_FLOAT *dst,
LVM_INT16 n)
@@ -110,5 +109,4 @@
}
#endif
-#endif
/**********************************************************************************/
diff --git a/media/libeffects/lvm/lib/Common/src/InstAlloc.c b/media/libeffects/lvm/lib/Common/src/InstAlloc.cpp
similarity index 99%
rename from media/libeffects/lvm/lib/Common/src/InstAlloc.c
rename to media/libeffects/lvm/lib/Common/src/InstAlloc.cpp
index a89a5c3..a039bf5 100644
--- a/media/libeffects/lvm/lib/Common/src/InstAlloc.c
+++ b/media/libeffects/lvm/lib/Common/src/InstAlloc.cpp
@@ -33,7 +33,6 @@
pms->pNextMember = (((uintptr_t)StartAddr + 3) & (uintptr_t)~3);
}
-
/****************************************************************************************
* Name : InstAlloc_AddMember()
* Input : pms - Pointer to the INST_ALLOC instance
@@ -59,7 +58,6 @@
return(NewMemberAddress);
}
-
/****************************************************************************************
* Name : InstAlloc_GetTotal()
* Input : pms - Pointer to the INST_ALLOC instance
@@ -80,7 +78,6 @@
}
}
-
void InstAlloc_InitAll( INST_ALLOC *pms,
LVM_MemoryTable_st *pMemoryTable)
{
@@ -91,19 +88,16 @@
pms[0].TotalSize = 3;
pms[0].pNextMember = ((StartAddr + 3) & (uintptr_t)~3);
-
StartAddr = (uintptr_t)pMemoryTable->Region[LVM_PERSISTENT_FAST_DATA].pBaseAddress;
pms[1].TotalSize = 3;
pms[1].pNextMember = ((StartAddr + 3) & (uintptr_t)~3);
-
StartAddr = (uintptr_t)pMemoryTable->Region[LVM_PERSISTENT_FAST_COEF].pBaseAddress;
pms[2].TotalSize = 3;
pms[2].pNextMember = ((StartAddr + 3) & (uintptr_t)~3);
-
StartAddr = (uintptr_t)pMemoryTable->Region[LVM_TEMPORARY_FAST].pBaseAddress;
pms[3].TotalSize = 3;
@@ -125,7 +119,6 @@
pms[0].TotalSize = 3;
pms[0].pNextMember = 0;
-
pms[1].TotalSize = 3;
pms[1].pNextMember = 0;
@@ -137,7 +130,6 @@
}
-
void* InstAlloc_AddMemberAll( INST_ALLOC *pms,
LVM_UINT32 Size[],
LVM_MemoryTable_st *pMemoryTable)
@@ -172,7 +164,6 @@
return(NewMemberAddress);
}
-
void* InstAlloc_AddMemberAllRet( INST_ALLOC *pms,
LVM_UINT32 Size[],
void **ptr)
diff --git a/media/libeffects/lvm/lib/Common/src/Int16LShiftToInt32_16x32.c b/media/libeffects/lvm/lib/Common/src/Int16LShiftToInt32_16x32.cpp
similarity index 100%
rename from media/libeffects/lvm/lib/Common/src/Int16LShiftToInt32_16x32.c
rename to media/libeffects/lvm/lib/Common/src/Int16LShiftToInt32_16x32.cpp
diff --git a/media/libeffects/lvm/lib/Common/src/Int32RShiftToInt16_Sat_32x16.c b/media/libeffects/lvm/lib/Common/src/Int32RShiftToInt16_Sat_32x16.cpp
similarity index 100%
rename from media/libeffects/lvm/lib/Common/src/Int32RShiftToInt16_Sat_32x16.c
rename to media/libeffects/lvm/lib/Common/src/Int32RShiftToInt16_Sat_32x16.cpp
diff --git a/media/libeffects/lvm/lib/Common/src/JoinTo2i_32x32.c b/media/libeffects/lvm/lib/Common/src/JoinTo2i_32x32.cpp
similarity index 98%
rename from media/libeffects/lvm/lib/Common/src/JoinTo2i_32x32.c
rename to media/libeffects/lvm/lib/Common/src/JoinTo2i_32x32.cpp
index ebc477e..05df656 100644
--- a/media/libeffects/lvm/lib/Common/src/JoinTo2i_32x32.c
+++ b/media/libeffects/lvm/lib/Common/src/JoinTo2i_32x32.cpp
@@ -49,7 +49,6 @@
return;
}
-#ifdef BUILD_FLOAT
void JoinTo2i_Float( const LVM_FLOAT *srcL,
const LVM_FLOAT *srcR,
LVM_FLOAT *dst,
@@ -74,6 +73,5 @@
return;
}
-#endif
/**********************************************************************************/
diff --git a/media/libeffects/lvm/lib/Common/src/LVC_Core_MixHard_1St_2i_D16C31_SAT.c b/media/libeffects/lvm/lib/Common/src/LVC_Core_MixHard_1St_2i_D16C31_SAT.cpp
similarity index 70%
rename from media/libeffects/lvm/lib/Common/src/LVC_Core_MixHard_1St_2i_D16C31_SAT.c
rename to media/libeffects/lvm/lib/Common/src/LVC_Core_MixHard_1St_2i_D16C31_SAT.cpp
index db76cd1..14d61bd 100644
--- a/media/libeffects/lvm/lib/Common/src/LVC_Core_MixHard_1St_2i_D16C31_SAT.c
+++ b/media/libeffects/lvm/lib/Common/src/LVC_Core_MixHard_1St_2i_D16C31_SAT.cpp
@@ -23,11 +23,9 @@
#include "LVM_Macros.h"
#include "ScalarArithmetic.h"
-
/**********************************************************************************
FUNCTION LVC_Core_MixHard_1St_2i_D16C31_SAT
***********************************************************************************/
-#ifdef BUILD_FLOAT
void LVC_Core_MixHard_1St_2i_D16C31_SAT( LVMixer3_FLOAT_st *ptrInstance1,
LVMixer3_FLOAT_st *ptrInstance2,
const LVM_FLOAT *src,
@@ -57,7 +55,6 @@
*dst++ = (LVM_FLOAT)Temp;
}
-
}
#ifdef SUPPORT_MC
void LVC_Core_MixHard_1St_MC_float_SAT (Mix_Private_FLOAT_st **ptrInstance,
@@ -84,44 +81,4 @@
}
}
#endif
-#else
-void LVC_Core_MixHard_1St_2i_D16C31_SAT( LVMixer3_st *ptrInstance1,
- LVMixer3_st *ptrInstance2,
- const LVM_INT16 *src,
- LVM_INT16 *dst,
- LVM_INT16 n)
-{
- LVM_INT32 Temp;
- LVM_INT16 ii;
- LVM_INT16 Current1Short;
- LVM_INT16 Current2Short;
- Mix_Private_st *pInstance1=(Mix_Private_st *)(ptrInstance1->PrivateParams);
- Mix_Private_st *pInstance2=(Mix_Private_st *)(ptrInstance2->PrivateParams);
-
-
- Current1Short = (LVM_INT16)(pInstance1->Current >> 16);
- Current2Short = (LVM_INT16)(pInstance2->Current >> 16);
-
- for (ii = n; ii != 0; ii--)
- {
- Temp = ((LVM_INT32)*(src++) * (LVM_INT32)Current1Short)>>15;
- if (Temp > 0x00007FFF)
- *dst++ = 0x7FFF;
- else if (Temp < -0x00008000)
- *dst++ = - 0x8000;
- else
- *dst++ = (LVM_INT16)Temp;
-
- Temp = ((LVM_INT32)*(src++) * (LVM_INT32)Current2Short)>>15;
- if (Temp > 0x00007FFF)
- *dst++ = 0x7FFF;
- else if (Temp < -0x00008000)
- *dst++ = - 0x8000;
- else
- *dst++ = (LVM_INT16)Temp;
- }
-
-
-}
-#endif
/**********************************************************************************/
diff --git a/media/libeffects/lvm/lib/Common/src/LVC_Core_MixHard_2St_D16C31_SAT.c b/media/libeffects/lvm/lib/Common/src/LVC_Core_MixHard_2St_D16C31_SAT.cpp
similarity index 65%
rename from media/libeffects/lvm/lib/Common/src/LVC_Core_MixHard_2St_D16C31_SAT.c
rename to media/libeffects/lvm/lib/Common/src/LVC_Core_MixHard_2St_D16C31_SAT.cpp
index ec0baaf..841fa1e 100644
--- a/media/libeffects/lvm/lib/Common/src/LVC_Core_MixHard_2St_D16C31_SAT.c
+++ b/media/libeffects/lvm/lib/Common/src/LVC_Core_MixHard_2St_D16C31_SAT.cpp
@@ -24,7 +24,6 @@
/**********************************************************************************
FUNCTION LVCore_MIXHARD_2ST_D16C31_SAT
***********************************************************************************/
-#ifdef BUILD_FLOAT
void LVC_Core_MixHard_2St_D16C31_SAT( LVMixer3_FLOAT_st *ptrInstance1,
LVMixer3_FLOAT_st *ptrInstance2,
const LVM_FLOAT *src1,
@@ -39,7 +38,6 @@
Mix_Private_FLOAT_st *pInstance1 = (Mix_Private_FLOAT_st *)(ptrInstance1->PrivateParams);
Mix_Private_FLOAT_st *pInstance2 = (Mix_Private_FLOAT_st *)(ptrInstance2->PrivateParams);
-
Current1 = (pInstance1->Current);
Current2 = (pInstance2->Current);
@@ -54,35 +52,4 @@
*dst++ = Temp;
}
}
-#else
-void LVC_Core_MixHard_2St_D16C31_SAT( LVMixer3_st *ptrInstance1,
- LVMixer3_st *ptrInstance2,
- const LVM_INT16 *src1,
- const LVM_INT16 *src2,
- LVM_INT16 *dst,
- LVM_INT16 n)
-{
- LVM_INT32 Temp;
- LVM_INT16 ii;
- LVM_INT16 Current1Short;
- LVM_INT16 Current2Short;
- Mix_Private_st *pInstance1=(Mix_Private_st *)(ptrInstance1->PrivateParams);
- Mix_Private_st *pInstance2=(Mix_Private_st *)(ptrInstance2->PrivateParams);
-
-
- Current1Short = (LVM_INT16)(pInstance1->Current >> 16);
- Current2Short = (LVM_INT16)(pInstance2->Current >> 16);
-
- for (ii = n; ii != 0; ii--){
- Temp = (((LVM_INT32)*(src1++) * (LVM_INT32)Current1Short)>>15) +
- (((LVM_INT32)*(src2++) * (LVM_INT32)Current2Short)>>15);
- if (Temp > 0x00007FFF)
- *dst++ = 0x7FFF;
- else if (Temp < -0x00008000)
- *dst++ = - 0x8000;
- else
- *dst++ = (LVM_INT16)Temp;
- }
-}
-#endif
/**********************************************************************************/
diff --git a/media/libeffects/lvm/lib/Common/src/LVC_Core_MixInSoft_D16C31_SAT.c b/media/libeffects/lvm/lib/Common/src/LVC_Core_MixInSoft_D16C31_SAT.cpp
similarity index 67%
rename from media/libeffects/lvm/lib/Common/src/LVC_Core_MixInSoft_D16C31_SAT.c
rename to media/libeffects/lvm/lib/Common/src/LVC_Core_MixInSoft_D16C31_SAT.cpp
index 419c7c5..318138d 100644
--- a/media/libeffects/lvm/lib/Common/src/LVC_Core_MixInSoft_D16C31_SAT.c
+++ b/media/libeffects/lvm/lib/Common/src/LVC_Core_MixInSoft_D16C31_SAT.cpp
@@ -25,7 +25,6 @@
/**********************************************************************************
FUNCTION LVCore_MIXSOFT_1ST_D16C31_WRA
***********************************************************************************/
-#ifdef BUILD_FLOAT
void LVC_Core_MixInSoft_D16C31_SAT(LVMixer3_FLOAT_st *ptrInstance,
const LVM_FLOAT *src,
LVM_FLOAT *dst,
@@ -247,103 +246,4 @@
}
#endif
-#else
-void LVC_Core_MixInSoft_D16C31_SAT( LVMixer3_st *ptrInstance,
- const LVM_INT16 *src,
- LVM_INT16 *dst,
- LVM_INT16 n)
-{
-
- LVM_INT16 OutLoop;
- LVM_INT16 InLoop;
- LVM_INT16 CurrentShort;
- LVM_INT32 ii,jj;
- Mix_Private_st *pInstance=(Mix_Private_st *)(ptrInstance->PrivateParams);
- LVM_INT32 Delta=pInstance->Delta;
- LVM_INT32 Current=pInstance->Current;
- LVM_INT32 Target=pInstance->Target;
- LVM_INT32 Temp;
-
- InLoop = (LVM_INT16)(n >> 2); /* Process per 4 samples */
- OutLoop = (LVM_INT16)(n - (InLoop << 2));
-
- if(Current<Target){
- if (OutLoop){
- ADD2_SAT_32x32(Current,Delta,Temp); /* Q31 + Q31 into Q31*/
- Current=Temp;
- if (Current > Target)
- Current = Target;
-
- CurrentShort = (LVM_INT16)(Current>>16); /* From Q31 to Q15*/
-
- for (ii = OutLoop; ii != 0; ii--){
- Temp = ((LVM_INT32)*dst) + (((LVM_INT32)*(src++) * CurrentShort)>>15); /* Q15 + Q15*Q15>>15 into Q15 */
- if (Temp > 0x00007FFF)
- *dst++ = 0x7FFF;
- else if (Temp < -0x00008000)
- *dst++ = - 0x8000;
- else
- *dst++ = (LVM_INT16)Temp;
- }
- }
-
- for (ii = InLoop; ii != 0; ii--){
- ADD2_SAT_32x32(Current,Delta,Temp); /* Q31 + Q31 into Q31*/
- Current=Temp;
- if (Current > Target)
- Current = Target;
-
- CurrentShort = (LVM_INT16)(Current>>16); /* From Q31 to Q15*/
-
- for (jj = 4; jj!=0 ; jj--){
- Temp = ((LVM_INT32)*dst) + (((LVM_INT32)*(src++) * CurrentShort)>>15); /* Q15 + Q15*Q15>>15 into Q15 */
- if (Temp > 0x00007FFF)
- *dst++ = 0x7FFF;
- else if (Temp < -0x00008000)
- *dst++ = - 0x8000;
- else
- *dst++ = (LVM_INT16)Temp;
- }
- }
- }
- else{
- if (OutLoop){
- Current -= Delta; /* Q31 + Q31 into Q31*/
- if (Current < Target)
- Current = Target;
-
- CurrentShort = (LVM_INT16)(Current>>16); /* From Q31 to Q15*/
-
- for (ii = OutLoop; ii != 0; ii--){
- Temp = ((LVM_INT32)*dst) + (((LVM_INT32)*(src++) * CurrentShort)>>15); /* Q15 + Q15*Q15>>15 into Q15 */
- if (Temp > 0x00007FFF)
- *dst++ = 0x7FFF;
- else if (Temp < -0x00008000)
- *dst++ = - 0x8000;
- else
- *dst++ = (LVM_INT16)Temp;
- }
- }
-
- for (ii = InLoop; ii != 0; ii--){
- Current -= Delta; /* Q31 + Q31 into Q31*/
- if (Current < Target)
- Current = Target;
-
- CurrentShort = (LVM_INT16)(Current>>16); /* From Q31 to Q15*/
-
- for (jj = 4; jj!=0 ; jj--){
- Temp = ((LVM_INT32)*dst) + (((LVM_INT32)*(src++) * CurrentShort)>>15); /* Q15 + Q15*Q15>>15 into Q15 */
- if (Temp > 0x00007FFF)
- *dst++ = 0x7FFF;
- else if (Temp < -0x00008000)
- *dst++ = - 0x8000;
- else
- *dst++ = (LVM_INT16)Temp;
- }
- }
- }
- pInstance->Current=Current;
-}
-#endif
/**********************************************************************************/
diff --git a/media/libeffects/lvm/lib/Common/src/LVC_Core_MixSoft_1St_2i_D16C31_WRA.c b/media/libeffects/lvm/lib/Common/src/LVC_Core_MixSoft_1St_2i_D16C31_WRA.c
deleted file mode 100644
index 56b5dae..0000000
--- a/media/libeffects/lvm/lib/Common/src/LVC_Core_MixSoft_1St_2i_D16C31_WRA.c
+++ /dev/null
@@ -1,309 +0,0 @@
-/*
- * Copyright (C) 2004-2010 NXP Software
- * Copyright (C) 2010 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-/**********************************************************************************
- INCLUDE FILES
-***********************************************************************************/
-
-#include "LVC_Mixer_Private.h"
-#include "ScalarArithmetic.h"
-#include "LVM_Macros.h"
-
-/**********************************************************************************
- FUNCTION LVC_Core_MixSoft_1St_2i_D16C31_WRA
-***********************************************************************************/
-#ifdef BUILD_FLOAT
-static LVM_FLOAT ADD2_SAT_FLOAT(LVM_FLOAT a,
- LVM_FLOAT b,
- LVM_FLOAT c)
-{
- LVM_FLOAT temp;
- temp = a + b ;
- if (temp < -1.0f)
- c = -1.0f;
- else if (temp > 1.0f)
- c = 1.0f;
- else
- c = temp;
- return c;
-}
-void LVC_Core_MixSoft_1St_2i_D16C31_WRA( LVMixer3_FLOAT_st *ptrInstance1,
- LVMixer3_FLOAT_st *ptrInstance2,
- const LVM_FLOAT *src,
- LVM_FLOAT *dst,
- LVM_INT16 n)
-{
- LVM_INT16 OutLoop;
- LVM_INT16 InLoop;
- LVM_INT32 ii;
- Mix_Private_FLOAT_st *pInstanceL = (Mix_Private_FLOAT_st *)(ptrInstance1->PrivateParams);
- Mix_Private_FLOAT_st *pInstanceR = (Mix_Private_FLOAT_st *)(ptrInstance2->PrivateParams);
-
- LVM_FLOAT DeltaL = pInstanceL->Delta;
- LVM_FLOAT CurrentL = pInstanceL->Current;
- LVM_FLOAT TargetL = pInstanceL->Target;
-
- LVM_FLOAT DeltaR = pInstanceR->Delta;
- LVM_FLOAT CurrentR = pInstanceR->Current;
- LVM_FLOAT TargetR = pInstanceR->Target;
-
- LVM_FLOAT Temp = 0;
-
- InLoop = (LVM_INT16)(n >> 2); /* Process per 4 samples */
- OutLoop = (LVM_INT16)(n - (InLoop << 2));
-
- if (OutLoop)
- {
- if(CurrentL < TargetL)
- {
- ADD2_SAT_FLOAT(CurrentL, DeltaL, Temp);
- CurrentL = Temp;
- if (CurrentL > TargetL)
- CurrentL = TargetL;
- }
- else
- {
- CurrentL -= DeltaL;
- if (CurrentL < TargetL)
- CurrentL = TargetL;
- }
-
- if(CurrentR < TargetR)
- {
- ADD2_SAT_FLOAT(CurrentR, DeltaR, Temp);
- CurrentR = Temp;
- if (CurrentR > TargetR)
- CurrentR = TargetR;
- }
- else
- {
- CurrentR -= DeltaR;
- if (CurrentR < TargetR)
- CurrentR = TargetR;
- }
-
- for (ii = OutLoop * 2; ii != 0; ii -= 2)
- {
- *(dst++) = (LVM_FLOAT)(((LVM_FLOAT)*(src++) * (LVM_FLOAT)CurrentL));
- *(dst++) = (LVM_FLOAT)(((LVM_FLOAT)*(src++) * (LVM_FLOAT)CurrentR));
- }
- }
-
- for (ii = InLoop * 2; ii != 0; ii-=2)
- {
- if(CurrentL < TargetL)
- {
- ADD2_SAT_FLOAT(CurrentL, DeltaL, Temp);
- CurrentL = Temp;
- if (CurrentL > TargetL)
- CurrentL = TargetL;
- }
- else
- {
- CurrentL -= DeltaL;
- if (CurrentL < TargetL)
- CurrentL = TargetL;
- }
-
- if(CurrentR < TargetR)
- {
- ADD2_SAT_FLOAT(CurrentR, DeltaR, Temp);
- CurrentR = Temp;
- if (CurrentR > TargetR)
- CurrentR = TargetR;
- }
- else
- {
- CurrentR -= DeltaR;
- if (CurrentR < TargetR)
- CurrentR = TargetR;
- }
-
- *(dst++) = (LVM_FLOAT)(((LVM_FLOAT)*(src++) * (LVM_FLOAT)CurrentL));
- *(dst++) = (LVM_FLOAT)(((LVM_FLOAT)*(src++) * (LVM_FLOAT)CurrentR));
- *(dst++) = (LVM_FLOAT)(((LVM_FLOAT)*(src++) * (LVM_FLOAT)CurrentL));
- *(dst++) = (LVM_FLOAT)(((LVM_FLOAT)*(src++) * (LVM_FLOAT)CurrentR));
- *(dst++) = (LVM_FLOAT)(((LVM_FLOAT)*(src++) * (LVM_FLOAT)CurrentL));
- *(dst++) = (LVM_FLOAT)(((LVM_FLOAT)*(src++) * (LVM_FLOAT)CurrentR));
- *(dst++) = (LVM_FLOAT)(((LVM_FLOAT)*(src++) * (LVM_FLOAT)CurrentL));
- *(dst++) = (LVM_FLOAT)(((LVM_FLOAT)*(src++) * (LVM_FLOAT)CurrentR));
- }
- pInstanceL->Current = CurrentL;
- pInstanceR->Current = CurrentR;
-
-}
-#ifdef SUPPORT_MC
-void LVC_Core_MixSoft_1St_MC_float_WRA (Mix_Private_FLOAT_st **ptrInstance,
- const LVM_FLOAT *src,
- LVM_FLOAT *dst,
- LVM_INT16 NrFrames,
- LVM_INT16 NrChannels)
-{
- LVM_INT32 ii, ch;
- LVM_FLOAT Temp =0.0f;
- LVM_FLOAT tempCurrent[NrChannels];
- for (ch = 0; ch < NrChannels; ch++)
- {
- tempCurrent[ch] = ptrInstance[ch]->Current;
- }
- for (ii = NrFrames; ii > 0; ii--)
- {
- for (ch = 0; ch < NrChannels; ch++)
- {
- Mix_Private_FLOAT_st *pInstance = ptrInstance[ch];
- const LVM_FLOAT Delta = pInstance->Delta;
- LVM_FLOAT Current = tempCurrent[ch];
- const LVM_FLOAT Target = pInstance->Target;
- if (Current < Target)
- {
- ADD2_SAT_FLOAT(Current, Delta, Temp);
- Current = Temp;
- if (Current > Target)
- Current = Target;
- }
- else
- {
- Current -= Delta;
- if (Current < Target)
- Current = Target;
- }
- *dst++ = *src++ * Current;
- tempCurrent[ch] = Current;
- }
- }
- for (ch = 0; ch < NrChannels; ch++)
- {
- ptrInstance[ch]->Current = tempCurrent[ch];
- }
-}
-#endif
-#else
-void LVC_Core_MixSoft_1St_2i_D16C31_WRA( LVMixer3_st *ptrInstance1,
- LVMixer3_st *ptrInstance2,
- const LVM_INT16 *src,
- LVM_INT16 *dst,
- LVM_INT16 n)
-{
- LVM_INT16 OutLoop;
- LVM_INT16 InLoop;
- LVM_INT16 CurrentShortL;
- LVM_INT16 CurrentShortR;
- LVM_INT32 ii;
- Mix_Private_st *pInstanceL=(Mix_Private_st *)(ptrInstance1->PrivateParams);
- Mix_Private_st *pInstanceR=(Mix_Private_st *)(ptrInstance2->PrivateParams);
-
- LVM_INT32 DeltaL=pInstanceL->Delta;
- LVM_INT32 CurrentL=pInstanceL->Current;
- LVM_INT32 TargetL=pInstanceL->Target;
-
- LVM_INT32 DeltaR=pInstanceR->Delta;
- LVM_INT32 CurrentR=pInstanceR->Current;
- LVM_INT32 TargetR=pInstanceR->Target;
-
- LVM_INT32 Temp;
-
- InLoop = (LVM_INT16)(n >> 2); /* Process per 4 samples */
- OutLoop = (LVM_INT16)(n - (InLoop << 2));
-
- if (OutLoop)
- {
- if(CurrentL<TargetL)
- {
- ADD2_SAT_32x32(CurrentL,DeltaL,Temp); /* Q31 + Q31 into Q31*/
- CurrentL=Temp;
- if (CurrentL > TargetL)
- CurrentL = TargetL;
- }
- else
- {
- CurrentL -= DeltaL; /* Q31 + Q31 into Q31*/
- if (CurrentL < TargetL)
- CurrentL = TargetL;
- }
-
- if(CurrentR<TargetR)
- {
- ADD2_SAT_32x32(CurrentR,DeltaR,Temp); /* Q31 + Q31 into Q31*/
- CurrentR=Temp;
- if (CurrentR > TargetR)
- CurrentR = TargetR;
- }
- else
- {
- CurrentR -= DeltaR; /* Q31 + Q31 into Q31*/
- if (CurrentR < TargetR)
- CurrentR = TargetR;
- }
-
- CurrentShortL = (LVM_INT16)(CurrentL>>16); /* From Q31 to Q15*/
- CurrentShortR = (LVM_INT16)(CurrentR>>16); /* From Q31 to Q15*/
-
- for (ii = OutLoop*2; ii != 0; ii-=2)
- {
- *(dst++) = (LVM_INT16)(((LVM_INT32)*(src++) * (LVM_INT32)CurrentShortL)>>15); /* Q15*Q15>>15 into Q15 */
- *(dst++) = (LVM_INT16)(((LVM_INT32)*(src++) * (LVM_INT32)CurrentShortR)>>15); /* Q15*Q15>>15 into Q15 */
- }
- }
-
- for (ii = InLoop*2; ii != 0; ii-=2)
- {
- if(CurrentL<TargetL)
- {
- ADD2_SAT_32x32(CurrentL,DeltaL,Temp); /* Q31 + Q31 into Q31*/
- CurrentL=Temp;
- if (CurrentL > TargetL)
- CurrentL = TargetL;
- }
- else
- {
- CurrentL -= DeltaL; /* Q31 + Q31 into Q31*/
- if (CurrentL < TargetL)
- CurrentL = TargetL;
- }
-
- if(CurrentR<TargetR)
- {
- ADD2_SAT_32x32(CurrentR,DeltaR,Temp); /* Q31 + Q31 into Q31*/
- CurrentR=Temp;
- if (CurrentR > TargetR)
- CurrentR = TargetR;
- }
- else
- {
- CurrentR -= DeltaR; /* Q31 + Q31 into Q31*/
- if (CurrentR < TargetR)
- CurrentR = TargetR;
- }
-
- CurrentShortL = (LVM_INT16)(CurrentL>>16); /* From Q31 to Q15*/
- CurrentShortR = (LVM_INT16)(CurrentR>>16); /* From Q31 to Q15*/
-
- *(dst++) = (LVM_INT16)(((LVM_INT32)*(src++) * (LVM_INT32)CurrentShortL)>>15); /* Q15*Q15>>15 into Q15 */
- *(dst++) = (LVM_INT16)(((LVM_INT32)*(src++) * (LVM_INT32)CurrentShortR)>>15); /* Q15*Q15>>15 into Q15 */
- *(dst++) = (LVM_INT16)(((LVM_INT32)*(src++) * (LVM_INT32)CurrentShortL)>>15);
- *(dst++) = (LVM_INT16)(((LVM_INT32)*(src++) * (LVM_INT32)CurrentShortR)>>15);
- *(dst++) = (LVM_INT16)(((LVM_INT32)*(src++) * (LVM_INT32)CurrentShortL)>>15);
- *(dst++) = (LVM_INT16)(((LVM_INT32)*(src++) * (LVM_INT32)CurrentShortR)>>15);
- *(dst++) = (LVM_INT16)(((LVM_INT32)*(src++) * (LVM_INT32)CurrentShortL)>>15);
- *(dst++) = (LVM_INT16)(((LVM_INT32)*(src++) * (LVM_INT32)CurrentShortR)>>15);
- }
- pInstanceL->Current=CurrentL;
- pInstanceR->Current=CurrentR;
-
-}
-#endif
-/**********************************************************************************/
diff --git a/media/libeffects/lvm/lib/Common/src/LVC_Core_MixSoft_1St_2i_D16C31_WRA.cpp b/media/libeffects/lvm/lib/Common/src/LVC_Core_MixSoft_1St_2i_D16C31_WRA.cpp
new file mode 100644
index 0000000..1f4b08a
--- /dev/null
+++ b/media/libeffects/lvm/lib/Common/src/LVC_Core_MixSoft_1St_2i_D16C31_WRA.cpp
@@ -0,0 +1,193 @@
+/*
+ * Copyright (C) 2004-2010 NXP Software
+ * Copyright (C) 2010 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+/**********************************************************************************
+ INCLUDE FILES
+***********************************************************************************/
+
+#include "LVC_Mixer_Private.h"
+#include "ScalarArithmetic.h"
+#include "LVM_Macros.h"
+
+/**********************************************************************************
+ FUNCTION LVC_Core_MixSoft_1St_2i_D16C31_WRA
+***********************************************************************************/
+static LVM_FLOAT ADD2_SAT_FLOAT(LVM_FLOAT a,
+ LVM_FLOAT b,
+ LVM_FLOAT c)
+{
+ LVM_FLOAT temp;
+ temp = a + b ;
+ if (temp < -1.0f)
+ c = -1.0f;
+ else if (temp > 1.0f)
+ c = 1.0f;
+ else
+ c = temp;
+ return c;
+}
+void LVC_Core_MixSoft_1St_2i_D16C31_WRA( LVMixer3_FLOAT_st *ptrInstance1,
+ LVMixer3_FLOAT_st *ptrInstance2,
+ const LVM_FLOAT *src,
+ LVM_FLOAT *dst,
+ LVM_INT16 n)
+{
+ LVM_INT16 OutLoop;
+ LVM_INT16 InLoop;
+ LVM_INT32 ii;
+ Mix_Private_FLOAT_st *pInstanceL = (Mix_Private_FLOAT_st *)(ptrInstance1->PrivateParams);
+ Mix_Private_FLOAT_st *pInstanceR = (Mix_Private_FLOAT_st *)(ptrInstance2->PrivateParams);
+
+ LVM_FLOAT DeltaL = pInstanceL->Delta;
+ LVM_FLOAT CurrentL = pInstanceL->Current;
+ LVM_FLOAT TargetL = pInstanceL->Target;
+
+ LVM_FLOAT DeltaR = pInstanceR->Delta;
+ LVM_FLOAT CurrentR = pInstanceR->Current;
+ LVM_FLOAT TargetR = pInstanceR->Target;
+
+ LVM_FLOAT Temp = 0;
+
+ InLoop = (LVM_INT16)(n >> 2); /* Process per 4 samples */
+ OutLoop = (LVM_INT16)(n - (InLoop << 2));
+
+ if (OutLoop)
+ {
+ if(CurrentL < TargetL)
+ {
+ ADD2_SAT_FLOAT(CurrentL, DeltaL, Temp);
+ CurrentL = Temp;
+ if (CurrentL > TargetL)
+ CurrentL = TargetL;
+ }
+ else
+ {
+ CurrentL -= DeltaL;
+ if (CurrentL < TargetL)
+ CurrentL = TargetL;
+ }
+
+ if(CurrentR < TargetR)
+ {
+ ADD2_SAT_FLOAT(CurrentR, DeltaR, Temp);
+ CurrentR = Temp;
+ if (CurrentR > TargetR)
+ CurrentR = TargetR;
+ }
+ else
+ {
+ CurrentR -= DeltaR;
+ if (CurrentR < TargetR)
+ CurrentR = TargetR;
+ }
+
+ for (ii = OutLoop * 2; ii != 0; ii -= 2)
+ {
+ *(dst++) = (LVM_FLOAT)(((LVM_FLOAT)*(src++) * (LVM_FLOAT)CurrentL));
+ *(dst++) = (LVM_FLOAT)(((LVM_FLOAT)*(src++) * (LVM_FLOAT)CurrentR));
+ }
+ }
+
+ for (ii = InLoop * 2; ii != 0; ii-=2)
+ {
+ if(CurrentL < TargetL)
+ {
+ ADD2_SAT_FLOAT(CurrentL, DeltaL, Temp);
+ CurrentL = Temp;
+ if (CurrentL > TargetL)
+ CurrentL = TargetL;
+ }
+ else
+ {
+ CurrentL -= DeltaL;
+ if (CurrentL < TargetL)
+ CurrentL = TargetL;
+ }
+
+ if(CurrentR < TargetR)
+ {
+ ADD2_SAT_FLOAT(CurrentR, DeltaR, Temp);
+ CurrentR = Temp;
+ if (CurrentR > TargetR)
+ CurrentR = TargetR;
+ }
+ else
+ {
+ CurrentR -= DeltaR;
+ if (CurrentR < TargetR)
+ CurrentR = TargetR;
+ }
+
+ *(dst++) = (LVM_FLOAT)(((LVM_FLOAT)*(src++) * (LVM_FLOAT)CurrentL));
+ *(dst++) = (LVM_FLOAT)(((LVM_FLOAT)*(src++) * (LVM_FLOAT)CurrentR));
+ *(dst++) = (LVM_FLOAT)(((LVM_FLOAT)*(src++) * (LVM_FLOAT)CurrentL));
+ *(dst++) = (LVM_FLOAT)(((LVM_FLOAT)*(src++) * (LVM_FLOAT)CurrentR));
+ *(dst++) = (LVM_FLOAT)(((LVM_FLOAT)*(src++) * (LVM_FLOAT)CurrentL));
+ *(dst++) = (LVM_FLOAT)(((LVM_FLOAT)*(src++) * (LVM_FLOAT)CurrentR));
+ *(dst++) = (LVM_FLOAT)(((LVM_FLOAT)*(src++) * (LVM_FLOAT)CurrentL));
+ *(dst++) = (LVM_FLOAT)(((LVM_FLOAT)*(src++) * (LVM_FLOAT)CurrentR));
+ }
+ pInstanceL->Current = CurrentL;
+ pInstanceR->Current = CurrentR;
+
+}
+#ifdef SUPPORT_MC
+void LVC_Core_MixSoft_1St_MC_float_WRA (Mix_Private_FLOAT_st **ptrInstance,
+ const LVM_FLOAT *src,
+ LVM_FLOAT *dst,
+ LVM_INT16 NrFrames,
+ LVM_INT16 NrChannels)
+{
+ LVM_INT32 ii, ch;
+ LVM_FLOAT Temp =0.0f;
+ LVM_FLOAT tempCurrent[NrChannels];
+ for (ch = 0; ch < NrChannels; ch++)
+ {
+ tempCurrent[ch] = ptrInstance[ch]->Current;
+ }
+ for (ii = NrFrames; ii > 0; ii--)
+ {
+ for (ch = 0; ch < NrChannels; ch++)
+ {
+ Mix_Private_FLOAT_st *pInstance = ptrInstance[ch];
+ const LVM_FLOAT Delta = pInstance->Delta;
+ LVM_FLOAT Current = tempCurrent[ch];
+ const LVM_FLOAT Target = pInstance->Target;
+ if (Current < Target)
+ {
+ ADD2_SAT_FLOAT(Current, Delta, Temp);
+ Current = Temp;
+ if (Current > Target)
+ Current = Target;
+ }
+ else
+ {
+ Current -= Delta;
+ if (Current < Target)
+ Current = Target;
+ }
+ *dst++ = *src++ * Current;
+ tempCurrent[ch] = Current;
+ }
+ }
+ for (ch = 0; ch < NrChannels; ch++)
+ {
+ ptrInstance[ch]->Current = tempCurrent[ch];
+ }
+}
+#endif
+/**********************************************************************************/
diff --git a/media/libeffects/lvm/lib/Common/src/LVC_Core_MixSoft_1St_D16C31_WRA.c b/media/libeffects/lvm/lib/Common/src/LVC_Core_MixSoft_1St_D16C31_WRA.cpp
similarity index 67%
rename from media/libeffects/lvm/lib/Common/src/LVC_Core_MixSoft_1St_D16C31_WRA.c
rename to media/libeffects/lvm/lib/Common/src/LVC_Core_MixSoft_1St_D16C31_WRA.cpp
index 5bfdad8..5d8aadc 100644
--- a/media/libeffects/lvm/lib/Common/src/LVC_Core_MixSoft_1St_D16C31_WRA.c
+++ b/media/libeffects/lvm/lib/Common/src/LVC_Core_MixSoft_1St_D16C31_WRA.cpp
@@ -26,7 +26,6 @@
/**********************************************************************************
FUNCTION LVCore_MIXSOFT_1ST_D16C31_WRA
***********************************************************************************/
-#ifdef BUILD_FLOAT
void LVC_Core_MixSoft_1St_D16C31_WRA(LVMixer3_FLOAT_st *ptrInstance,
const LVM_FLOAT *src,
LVM_FLOAT *dst,
@@ -106,7 +105,6 @@
pInstance->Current=Current;
}
-
#ifdef SUPPORT_MC
/*
* FUNCTION: LVC_Core_MixSoft_Mc_D16C31_WRA
@@ -218,80 +216,4 @@
}
#endif
-#else
-void LVC_Core_MixSoft_1St_D16C31_WRA( LVMixer3_st *ptrInstance,
- const LVM_INT16 *src,
- LVM_INT16 *dst,
- LVM_INT16 n)
-{
- LVM_INT16 OutLoop;
- LVM_INT16 InLoop;
- LVM_INT16 CurrentShort;
- LVM_INT32 ii;
- Mix_Private_st *pInstance=(Mix_Private_st *)(ptrInstance->PrivateParams);
- LVM_INT32 Delta=pInstance->Delta;
- LVM_INT32 Current=pInstance->Current;
- LVM_INT32 Target=pInstance->Target;
- LVM_INT32 Temp;
-
- InLoop = (LVM_INT16)(n >> 2); /* Process per 4 samples */
- OutLoop = (LVM_INT16)(n - (InLoop << 2));
-
- if(Current<Target){
- if (OutLoop){
- ADD2_SAT_32x32(Current,Delta,Temp); /* Q31 + Q31 into Q31*/
- Current=Temp;
- if (Current > Target)
- Current = Target;
-
- CurrentShort = (LVM_INT16)(Current>>16); /* From Q31 to Q15*/
-
- for (ii = OutLoop; ii != 0; ii--){
- *(dst++) = (LVM_INT16)(((LVM_INT32)*(src++) * (LVM_INT32)CurrentShort)>>15); /* Q15*Q15>>15 into Q15 */
- }
- }
-
- for (ii = InLoop; ii != 0; ii--){
- ADD2_SAT_32x32(Current,Delta,Temp); /* Q31 + Q31 into Q31*/
- Current=Temp;
- if (Current > Target)
- Current = Target;
-
- CurrentShort = (LVM_INT16)(Current>>16); /* From Q31 to Q15*/
-
- *(dst++) = (LVM_INT16)(((LVM_INT32)*(src++) * (LVM_INT32)CurrentShort)>>15); /* Q15*Q15>>15 into Q15 */
- *(dst++) = (LVM_INT16)(((LVM_INT32)*(src++) * (LVM_INT32)CurrentShort)>>15);
- *(dst++) = (LVM_INT16)(((LVM_INT32)*(src++) * (LVM_INT32)CurrentShort)>>15);
- *(dst++) = (LVM_INT16)(((LVM_INT32)*(src++) * (LVM_INT32)CurrentShort)>>15);
- }
- }
- else{
- if (OutLoop){
- Current -= Delta; /* Q31 + Q31 into Q31*/
- if (Current < Target)
- Current = Target;
-
- CurrentShort = (LVM_INT16)(Current>>16); /* From Q31 to Q15*/
-
- for (ii = OutLoop; ii != 0; ii--){
- *(dst++) = (LVM_INT16)(((LVM_INT32)*(src++) * (LVM_INT32)CurrentShort)>>15); /* Q15*Q15>>15 into Q15 */
- }
- }
-
- for (ii = InLoop; ii != 0; ii--){
- Current -= Delta; /* Q31 + Q31 into Q31*/
- if (Current < Target)
- Current = Target;
-
- CurrentShort = (LVM_INT16)(Current>>16); /* From Q31 to Q15*/
-
- *(dst++) = (LVM_INT16)(((LVM_INT32)*(src++) * (LVM_INT32)CurrentShort)>>15); /* Q15*Q15>>15 into Q15 */
- *(dst++) = (LVM_INT16)(((LVM_INT32)*(src++) * (LVM_INT32)CurrentShort)>>15);
- *(dst++) = (LVM_INT16)(((LVM_INT32)*(src++) * (LVM_INT32)CurrentShort)>>15);
- *(dst++) = (LVM_INT16)(((LVM_INT32)*(src++) * (LVM_INT32)CurrentShort)>>15);
- }
- }
- pInstance->Current=Current;
-}
-#endif
/**********************************************************************************/
diff --git a/media/libeffects/lvm/lib/Common/src/LVC_MixInSoft_D16C31_SAT.c b/media/libeffects/lvm/lib/Common/src/LVC_MixInSoft_D16C31_SAT.cpp
similarity index 70%
rename from media/libeffects/lvm/lib/Common/src/LVC_MixInSoft_D16C31_SAT.c
rename to media/libeffects/lvm/lib/Common/src/LVC_MixInSoft_D16C31_SAT.cpp
index 65956f7..2bec3be 100644
--- a/media/libeffects/lvm/lib/Common/src/LVC_MixInSoft_D16C31_SAT.c
+++ b/media/libeffects/lvm/lib/Common/src/LVC_MixInSoft_D16C31_SAT.cpp
@@ -33,7 +33,6 @@
/**********************************************************************************
FUNCTION MIXINSOFT_D16C31_SAT
***********************************************************************************/
-#ifdef BUILD_FLOAT
void LVC_MixInSoft_D16C31_SAT(LVMixer3_1St_FLOAT_st *ptrInstance,
const LVM_FLOAT *src,
LVM_FLOAT *dst,
@@ -84,7 +83,6 @@
}
}
-
/******************************************************************************
CALL BACK
*******************************************************************************/
@@ -107,8 +105,6 @@
}
-
-
#ifdef SUPPORT_MC
/*
* FUNCTION: LVC_MixInSoft_Mc_D16C31_SAT
@@ -185,7 +181,6 @@
}
}
-
/******************************************************************************
CALL BACK
*******************************************************************************/
@@ -209,81 +204,4 @@
}
#endif
-
-#else
-void LVC_MixInSoft_D16C31_SAT( LVMixer3_1St_st *ptrInstance,
- LVM_INT16 *src,
- LVM_INT16 *dst,
- LVM_INT16 n)
-{
- char HardMixing = TRUE;
- LVM_INT32 TargetGain;
- Mix_Private_st *pInstance=(Mix_Private_st *)(ptrInstance->MixerStream[0].PrivateParams);
-
- if(n<=0) return;
-
- /******************************************************************************
- SOFT MIXING
- *******************************************************************************/
- if (pInstance->Current != pInstance->Target)
- {
- if(pInstance->Delta == 0x7FFFFFFF){
- pInstance->Current = pInstance->Target;
- TargetGain=pInstance->Target>>(16-pInstance->Shift); // TargetGain in Q16.15 format
- LVC_Mixer_SetTarget(&(ptrInstance->MixerStream[0]),TargetGain);
- }else if (Abs_32(pInstance->Current-pInstance->Target) < pInstance->Delta){
- pInstance->Current = pInstance->Target; /* Difference is not significant anymore. Make them equal. */
- TargetGain=pInstance->Target>>(16-pInstance->Shift); // TargetGain in Q16.15 format
- LVC_Mixer_SetTarget(&(ptrInstance->MixerStream[0]),TargetGain);
- }else{
- /* Soft mixing has to be applied */
- HardMixing = FALSE;
- if(pInstance->Shift!=0){
- Shift_Sat_v16xv16 ((LVM_INT16)pInstance->Shift,src,src,n);
- LVC_Core_MixInSoft_D16C31_SAT( &(ptrInstance->MixerStream[0]), src, dst, n);
- }
- else
- LVC_Core_MixInSoft_D16C31_SAT( &(ptrInstance->MixerStream[0]), src, dst, n);
- }
- }
-
- /******************************************************************************
- HARD MIXING
- *******************************************************************************/
-
- if (HardMixing){
- if (pInstance->Target != 0){ /* Nothing to do in case Target = 0 */
- if ((pInstance->Target>>16) == 0x7FFF){
- if(pInstance->Shift!=0)
- Shift_Sat_v16xv16 ((LVM_INT16)pInstance->Shift,src,src,n);
- Add2_Sat_16x16( src, dst, n );
- }
- else{
- if(pInstance->Shift!=0)
- Shift_Sat_v16xv16 ((LVM_INT16)pInstance->Shift,src,src,n);
- Mac3s_Sat_16x16(src,(LVM_INT16)(pInstance->Target>>16),dst,n);
- pInstance->Current = pInstance->Target; /* In case the LVCore function would have changed the Current value */
- }
- }
- }
-
-
- /******************************************************************************
- CALL BACK
- *******************************************************************************/
-
- if (ptrInstance->MixerStream[0].CallbackSet){
- if (Abs_32(pInstance->Current-pInstance->Target) < pInstance->Delta){
- pInstance->Current = pInstance->Target; /* Difference is not significant anymore. Make them equal. */
- TargetGain=pInstance->Target>>(16-pInstance->Shift); // TargetGain in Q16.15 format
- LVC_Mixer_SetTarget(ptrInstance->MixerStream,TargetGain);
- ptrInstance->MixerStream[0].CallbackSet = FALSE;
- if (ptrInstance->MixerStream[0].pCallBack != 0){
- (*ptrInstance->MixerStream[0].pCallBack) ( ptrInstance->MixerStream[0].pCallbackHandle, ptrInstance->MixerStream[0].pGeneralPurpose,ptrInstance->MixerStream[0].CallbackParam );
- }
- }
- }
-
-}
-#endif
/**********************************************************************************/
diff --git a/media/libeffects/lvm/lib/Common/src/LVC_MixSoft_1St_2i_D16C31_SAT.c b/media/libeffects/lvm/lib/Common/src/LVC_MixSoft_1St_2i_D16C31_SAT.cpp
similarity index 74%
rename from media/libeffects/lvm/lib/Common/src/LVC_MixSoft_1St_2i_D16C31_SAT.c
rename to media/libeffects/lvm/lib/Common/src/LVC_MixSoft_1St_2i_D16C31_SAT.cpp
index a4682d3..3153ada 100644
--- a/media/libeffects/lvm/lib/Common/src/LVC_MixSoft_1St_2i_D16C31_SAT.c
+++ b/media/libeffects/lvm/lib/Common/src/LVC_MixSoft_1St_2i_D16C31_SAT.cpp
@@ -37,7 +37,6 @@
/**********************************************************************************
FUNCTION LVC_MixSoft_1St_2i_D16C31_SAT
***********************************************************************************/
-#ifdef BUILD_FLOAT
#ifdef SUPPORT_MC
/* This threshold is used to decide on the processing to be applied on
* front center and back center channels
@@ -363,120 +362,4 @@
}
}
}
-#else
-void LVC_MixSoft_1St_2i_D16C31_SAT( LVMixer3_2St_st *ptrInstance,
- const LVM_INT16 *src,
- LVM_INT16 *dst,
- LVM_INT16 n)
-{
- char HardMixing = TRUE;
- LVM_INT32 TargetGain;
- Mix_Private_st *pInstance1=(Mix_Private_st *)(ptrInstance->MixerStream[0].PrivateParams);
- Mix_Private_st *pInstance2=(Mix_Private_st *)(ptrInstance->MixerStream[1].PrivateParams);
-
- if(n<=0) return;
-
- /******************************************************************************
- SOFT MIXING
- *******************************************************************************/
- if ((pInstance1->Current != pInstance1->Target)||(pInstance2->Current != pInstance2->Target))
- {
- if(pInstance1->Delta == 0x7FFFFFFF)
- {
- pInstance1->Current = pInstance1->Target;
- TargetGain=pInstance1->Target>>16; // TargetGain in Q16.15 format, no integer part
- LVC_Mixer_SetTarget(&(ptrInstance->MixerStream[0]),TargetGain);
- }
- else if (Abs_32(pInstance1->Current-pInstance1->Target) < pInstance1->Delta)
- {
- pInstance1->Current = pInstance1->Target; /* Difference is not significant anymore. Make them equal. */
- TargetGain=pInstance1->Target>>16; // TargetGain in Q16.15 format, no integer part
- LVC_Mixer_SetTarget(&(ptrInstance->MixerStream[0]),TargetGain);
- }
- else
- {
- /* Soft mixing has to be applied */
- HardMixing = FALSE;
- }
-
- if(HardMixing == TRUE)
- {
- if(pInstance2->Delta == 0x7FFFFFFF)
- {
- pInstance2->Current = pInstance2->Target;
- TargetGain=pInstance2->Target>>16; // TargetGain in Q16.15 format, no integer part
- LVC_Mixer_SetTarget(&(ptrInstance->MixerStream[1]),TargetGain);
- }
- else if (Abs_32(pInstance2->Current-pInstance2->Target) < pInstance2->Delta)
- {
- pInstance2->Current = pInstance2->Target; /* Difference is not significant anymore. Make them equal. */
- TargetGain=pInstance2->Target>>16; // TargetGain in Q16.15 format, no integer part
- LVC_Mixer_SetTarget(&(ptrInstance->MixerStream[1]),TargetGain);
- }
- else
- {
- /* Soft mixing has to be applied */
- HardMixing = FALSE;
- }
- }
-
- if(HardMixing == FALSE)
- {
- LVC_Core_MixSoft_1St_2i_D16C31_WRA( &(ptrInstance->MixerStream[0]),&(ptrInstance->MixerStream[1]), src, dst, n);
- }
- }
-
- /******************************************************************************
- HARD MIXING
- *******************************************************************************/
-
- if (HardMixing)
- {
- if (((pInstance1->Target>>16) == 0x7FFF)&&((pInstance2->Target>>16) == 0x7FFF))
- {
- if(src!=dst)
- {
- Copy_16(src, dst, n);
- }
- }
- else
- {
- LVC_Core_MixHard_1St_2i_D16C31_SAT(&(ptrInstance->MixerStream[0]),&(ptrInstance->MixerStream[1]), src, dst, n);
- }
- }
-
- /******************************************************************************
- CALL BACK
- *******************************************************************************/
-
- if (ptrInstance->MixerStream[0].CallbackSet)
- {
- if (Abs_32(pInstance1->Current-pInstance1->Target) < pInstance1->Delta)
- {
- pInstance1->Current = pInstance1->Target; /* Difference is not significant anymore. Make them equal. */
- TargetGain=pInstance1->Target>>(16-pInstance1->Shift); // TargetGain in Q16.15 format
- LVC_Mixer_SetTarget(&ptrInstance->MixerStream[0],TargetGain);
- ptrInstance->MixerStream[0].CallbackSet = FALSE;
- if (ptrInstance->MixerStream[0].pCallBack != 0)
- {
- (*ptrInstance->MixerStream[0].pCallBack) ( ptrInstance->MixerStream[0].pCallbackHandle, ptrInstance->MixerStream[0].pGeneralPurpose,ptrInstance->MixerStream[0].CallbackParam );
- }
- }
- }
- if (ptrInstance->MixerStream[1].CallbackSet)
- {
- if (Abs_32(pInstance2->Current-pInstance2->Target) < pInstance2->Delta)
- {
- pInstance2->Current = pInstance2->Target; /* Difference is not significant anymore. Make them equal. */
- TargetGain=pInstance2->Target>>(16-pInstance2->Shift); // TargetGain in Q16.15 format
- LVC_Mixer_SetTarget(&ptrInstance->MixerStream[1],TargetGain);
- ptrInstance->MixerStream[1].CallbackSet = FALSE;
- if (ptrInstance->MixerStream[1].pCallBack != 0)
- {
- (*ptrInstance->MixerStream[1].pCallBack) ( ptrInstance->MixerStream[1].pCallbackHandle, ptrInstance->MixerStream[1].pGeneralPurpose,ptrInstance->MixerStream[1].CallbackParam );
- }
- }
- }
-}
-#endif
/**********************************************************************************/
diff --git a/media/libeffects/lvm/lib/Common/src/LVC_MixSoft_1St_D16C31_SAT.c b/media/libeffects/lvm/lib/Common/src/LVC_MixSoft_1St_D16C31_SAT.cpp
similarity index 70%
rename from media/libeffects/lvm/lib/Common/src/LVC_MixSoft_1St_D16C31_SAT.c
rename to media/libeffects/lvm/lib/Common/src/LVC_MixSoft_1St_D16C31_SAT.cpp
index 0678ae0..4d229da 100644
--- a/media/libeffects/lvm/lib/Common/src/LVC_MixSoft_1St_D16C31_SAT.c
+++ b/media/libeffects/lvm/lib/Common/src/LVC_MixSoft_1St_D16C31_SAT.cpp
@@ -33,7 +33,6 @@
/**********************************************************************************
FUNCTION LVMixer3_MIXSOFT_1ST_D16C31_SAT
***********************************************************************************/
-#ifdef BUILD_FLOAT
void LVC_MixSoft_1St_D16C31_SAT( LVMixer3_1St_FLOAT_st *ptrInstance,
const LVM_FLOAT *src,
LVM_FLOAT *dst,
@@ -198,79 +197,4 @@
#endif
-#else
-void LVC_MixSoft_1St_D16C31_SAT( LVMixer3_1St_st *ptrInstance,
- const LVM_INT16 *src,
- LVM_INT16 *dst,
- LVM_INT16 n)
-{
- char HardMixing = TRUE;
- LVM_INT32 TargetGain;
- Mix_Private_st *pInstance=(Mix_Private_st *)(ptrInstance->MixerStream[0].PrivateParams);
-
- if(n<=0) return;
-
- /******************************************************************************
- SOFT MIXING
- *******************************************************************************/
- if (pInstance->Current != pInstance->Target)
- {
- if(pInstance->Delta == 0x7FFFFFFF){
- pInstance->Current = pInstance->Target;
- TargetGain=pInstance->Target>>(16-pInstance->Shift); // TargetGain in Q16.15 format
- LVC_Mixer_SetTarget(&(ptrInstance->MixerStream[0]),TargetGain);
- }else if (Abs_32(pInstance->Current-pInstance->Target) < pInstance->Delta){
- pInstance->Current = pInstance->Target; /* Difference is not significant anymore. Make them equal. */
- TargetGain=pInstance->Target>>(16-pInstance->Shift); // TargetGain in Q16.15 format
- LVC_Mixer_SetTarget(&(ptrInstance->MixerStream[0]),TargetGain);
- }else{
- /* Soft mixing has to be applied */
- HardMixing = FALSE;
- if(pInstance->Shift!=0){
- Shift_Sat_v16xv16 ((LVM_INT16)pInstance->Shift,src,dst,n);
- LVC_Core_MixSoft_1St_D16C31_WRA( &(ptrInstance->MixerStream[0]), dst, dst, n);
- }
- else
- LVC_Core_MixSoft_1St_D16C31_WRA( &(ptrInstance->MixerStream[0]), src, dst, n);
- }
- }
-
- /******************************************************************************
- HARD MIXING
- *******************************************************************************/
-
- if (HardMixing){
- if (pInstance->Target == 0)
- LoadConst_16(0, dst, n);
- else if(pInstance->Shift!=0){
- Shift_Sat_v16xv16 ((LVM_INT16)pInstance->Shift,src,dst,n);
- if ((pInstance->Target>>16) != 0x7FFF)
- Mult3s_16x16( dst, (LVM_INT16)(pInstance->Target>>16), dst, n );
- }
- else {
- if ((pInstance->Target>>16) != 0x7FFF)
- Mult3s_16x16( src, (LVM_INT16)(pInstance->Target>>16), dst, n );
- else if(src!=dst)
- Copy_16(src, dst, n);
- }
-
- }
-
- /******************************************************************************
- CALL BACK
- *******************************************************************************/
-
- if (ptrInstance->MixerStream[0].CallbackSet){
- if (Abs_32(pInstance->Current-pInstance->Target) < pInstance->Delta){
- pInstance->Current = pInstance->Target; /* Difference is not significant anymore. Make them equal. */
- TargetGain=pInstance->Target>>(16-pInstance->Shift); // TargetGain in Q16.15 format
- LVC_Mixer_SetTarget(ptrInstance->MixerStream,TargetGain);
- ptrInstance->MixerStream[0].CallbackSet = FALSE;
- if (ptrInstance->MixerStream[0].pCallBack != 0){
- (*ptrInstance->MixerStream[0].pCallBack) ( ptrInstance->MixerStream[0].pCallbackHandle, ptrInstance->MixerStream[0].pGeneralPurpose,ptrInstance->MixerStream[0].CallbackParam );
- }
- }
- }
-}
-#endif/*BUILD_FLOAT*/
/**********************************************************************************/
diff --git a/media/libeffects/lvm/lib/Common/src/LVC_MixSoft_2St_D16C31_SAT.c b/media/libeffects/lvm/lib/Common/src/LVC_MixSoft_2St_D16C31_SAT.cpp
similarity index 73%
rename from media/libeffects/lvm/lib/Common/src/LVC_MixSoft_2St_D16C31_SAT.c
rename to media/libeffects/lvm/lib/Common/src/LVC_MixSoft_2St_D16C31_SAT.cpp
index 8a89de1..54ab79d 100644
--- a/media/libeffects/lvm/lib/Common/src/LVC_MixSoft_2St_D16C31_SAT.c
+++ b/media/libeffects/lvm/lib/Common/src/LVC_MixSoft_2St_D16C31_SAT.cpp
@@ -25,7 +25,6 @@
/**********************************************************************************
FUNCTION LVC_MixSoft_2St_D16C31_SAT.c
***********************************************************************************/
-#ifdef BUILD_FLOAT
void LVC_MixSoft_2St_D16C31_SAT(LVMixer3_2St_FLOAT_st *ptrInstance,
const LVM_FLOAT *src1,
const LVM_FLOAT *src2,
@@ -131,46 +130,4 @@
}
#endif
-#else
-void LVC_MixSoft_2St_D16C31_SAT( LVMixer3_2St_st *ptrInstance,
- const LVM_INT16 *src1,
- LVM_INT16 *src2,
- LVM_INT16 *dst,
- LVM_INT16 n)
-{
- Mix_Private_st *pInstance1=(Mix_Private_st *)(ptrInstance->MixerStream[0].PrivateParams);
- Mix_Private_st *pInstance2=(Mix_Private_st *)(ptrInstance->MixerStream[1].PrivateParams);
-
- if(n<=0) return;
-
- /******************************************************************************
- SOFT MIXING
- *******************************************************************************/
- if ((pInstance1->Current == pInstance1->Target)&&(pInstance1->Current == 0)){
- LVC_MixSoft_1St_D16C31_SAT( (LVMixer3_1St_st *)(&ptrInstance->MixerStream[1]), src2, dst, n);
- }
- else if ((pInstance2->Current == pInstance2->Target)&&(pInstance2->Current == 0)){
- LVC_MixSoft_1St_D16C31_SAT( (LVMixer3_1St_st *)(&ptrInstance->MixerStream[0]), src1, dst, n);
- }
- else if ((pInstance1->Current != pInstance1->Target) || (pInstance2->Current != pInstance2->Target))
- {
- LVC_MixSoft_1St_D16C31_SAT((LVMixer3_1St_st *)(&ptrInstance->MixerStream[0]), src1, dst, n);
- LVC_MixInSoft_D16C31_SAT( (LVMixer3_1St_st *)(&ptrInstance->MixerStream[1]), src2, dst, n);
- }
- else{
- /******************************************************************************
- HARD MIXING
- *******************************************************************************/
- if(pInstance2->Shift!=0)
- Shift_Sat_v16xv16 ((LVM_INT16)pInstance2->Shift,src2,src2,n);
- if(pInstance1->Shift!=0)
- {
- Shift_Sat_v16xv16 ((LVM_INT16)pInstance1->Shift,src1,dst,n);
- LVC_Core_MixHard_2St_D16C31_SAT( &ptrInstance->MixerStream[0], &ptrInstance->MixerStream[1], dst, src2, dst, n);
- }
- else
- LVC_Core_MixHard_2St_D16C31_SAT( &ptrInstance->MixerStream[0], &ptrInstance->MixerStream[1], src1, src2, dst, n);
- }
-}
-#endif /*BUILD_FLOAT*/
/**********************************************************************************/
diff --git a/media/libeffects/lvm/lib/Common/src/LVC_Mixer.h b/media/libeffects/lvm/lib/Common/src/LVC_Mixer.h
index 199d529..ce42d2e 100644
--- a/media/libeffects/lvm/lib/Common/src/LVC_Mixer.h
+++ b/media/libeffects/lvm/lib/Common/src/LVC_Mixer.h
@@ -18,12 +18,6 @@
#ifndef __LVC_MIXER_H__
#define __LVC_MIXER_H__
-
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
-
#include "LVM_Types.h"
/**********************************************************************************
@@ -31,7 +25,6 @@
***********************************************************************************/
/* LVMixer3_st structure stores Instance parameters for one audio stream */
-#ifdef BUILD_FLOAT
typedef struct
{
LVM_FLOAT PrivateParams[3]; /* Private Instance params for \
@@ -43,45 +36,14 @@
void *pGeneralPurpose; /* Pointer for general purpose usage */
LVM_Callback pCallBack; /* Pointer to the callback function */
} LVMixer3_FLOAT_st;
-#else
-typedef struct
-{
- LVM_INT32 PrivateParams[4]; /* Private Instance params for Audio Stream */
- LVM_INT16 CallbackSet; /* Boolean. Should be set by calling application each time the target value is updated */
- LVM_INT16 CallbackParam; /* Parameter that will be used in the calback function */
- void *pCallbackHandle; /* Pointer to the instance of the callback function */
- void *pGeneralPurpose; /* Pointer for general purpose usage */
- LVM_Callback pCallBack; /* Pointer to the callback function */
-} LVMixer3_st;
-#endif
-#ifdef BUILD_FLOAT
typedef struct
{
LVMixer3_FLOAT_st MixerStream[1]; /* Instance Params for one Audio Stream */
} LVMixer3_1St_FLOAT_st;
-#else
-typedef struct
-{
- LVMixer3_st MixerStream[1]; /* Instance Params for one Audio Stream */
-} LVMixer3_1St_st;
-#endif
-#ifdef BUILD_FLOAT
typedef struct
{
LVMixer3_FLOAT_st MixerStream[2]; /* Instance Params for two Audio Streams */
} LVMixer3_2St_FLOAT_st;
-#else
-typedef struct
-{
- LVMixer3_st MixerStream[2]; /* Instance Params for two Audio Streams */
-} LVMixer3_2St_st;
-#endif
-#ifndef BUILD_FLOAT
-typedef struct
-{
- LVMixer3_st MixerStream[3]; /* Instance Params for three Audio Streams */
-} LVMixer3_3St_st;
-#endif
/**********************************************************************************
FUNCTION PROTOTYPES (HIGH LEVEL FUNCTIONS)
***********************************************************************************/
@@ -92,7 +54,6 @@
#define LVMixer3_MixSoft_2St_D16C31_SAT LVMixer3_2St_D16C31_SAT_MixSoft
#define LVMixer3_MixSoft_3St_D16C31_SAT LVMixer3_3St_D16C31_SAT_MixSoft
-
/*** General functions ************************************************************/
/**********************************************************************************/
@@ -101,62 +62,28 @@
/* then the calculation will give an incorrect value for alpha, see the mixer */
/* documentation for further details. */
/* ********************************************************************************/
-#ifdef BUILD_FLOAT
void LVC_Mixer_SetTarget( LVMixer3_FLOAT_st *pStream,
LVM_FLOAT TargetGain);
-#else
-void LVC_Mixer_SetTarget( LVMixer3_st *pStream,
- LVM_INT32 TargetGain);
-#endif
-#ifdef BUILD_FLOAT
LVM_FLOAT LVC_Mixer_GetTarget( LVMixer3_FLOAT_st *pStream);
-#else
-LVM_INT32 LVC_Mixer_GetTarget( LVMixer3_st *pStream);
-#endif
-#ifdef BUILD_FLOAT
LVM_FLOAT LVC_Mixer_GetCurrent( LVMixer3_FLOAT_st *pStream);
-#else
-LVM_INT32 LVC_Mixer_GetCurrent( LVMixer3_st *pStream);
-#endif
-#ifdef BUILD_FLOAT
void LVC_Mixer_Init( LVMixer3_FLOAT_st *pStream,
LVM_FLOAT TargetGain,
LVM_FLOAT CurrentGain);
-#else
-void LVC_Mixer_Init( LVMixer3_st *pStream,
- LVM_INT32 TargetGain,
- LVM_INT32 CurrentGain);
-#endif
-#ifdef BUILD_FLOAT
void LVC_Mixer_SetTimeConstant( LVMixer3_FLOAT_st *pStream,
LVM_INT32 Tc_millisec,
LVM_Fs_en Fs,
LVM_INT16 NumChannels);
-#else
-void LVC_Mixer_SetTimeConstant( LVMixer3_st *pStream,
- LVM_INT32 Tc_millisec,
- LVM_Fs_en Fs,
- LVM_INT16 NumChannels);
-#endif
-#ifdef BUILD_FLOAT
void LVC_Mixer_VarSlope_SetTimeConstant( LVMixer3_FLOAT_st *pStream,
LVM_INT32 Tc_millisec,
LVM_Fs_en Fs,
LVM_INT16 NumChannels);
-#else
-void LVC_Mixer_VarSlope_SetTimeConstant( LVMixer3_st *pStream,
- LVM_INT32 Tc_millisec,
- LVM_Fs_en Fs,
- LVM_INT16 NumChannels);
-#endif
/*** 16 bit functions *************************************************************/
-#ifdef BUILD_FLOAT
void LVC_MixSoft_1St_D16C31_SAT(LVMixer3_1St_FLOAT_st *pInstance,
const LVM_FLOAT *src,
LVM_FLOAT *dst,
@@ -169,14 +96,6 @@
LVM_INT16 NrChannels);
#endif
-#else
-void LVC_MixSoft_1St_D16C31_SAT( LVMixer3_1St_st *pInstance,
- const LVM_INT16 *src,
- LVM_INT16 *dst,
- LVM_INT16 n);
-#endif
-
-#ifdef BUILD_FLOAT
void LVC_MixInSoft_D16C31_SAT(LVMixer3_1St_FLOAT_st *pInstance,
const LVM_FLOAT *src,
LVM_FLOAT *dst,
@@ -189,14 +108,6 @@
LVM_INT16 NrChannels);
#endif
-#else
-void LVC_MixInSoft_D16C31_SAT( LVMixer3_1St_st *pInstance,
- LVM_INT16 *src,
- LVM_INT16 *dst,
- LVM_INT16 n);
-#endif
-
-#ifdef BUILD_FLOAT
void LVC_MixSoft_2St_D16C31_SAT(LVMixer3_2St_FLOAT_st *pInstance,
const LVM_FLOAT *src1,
const LVM_FLOAT *src2,
@@ -210,20 +121,12 @@
LVM_INT16 NrFrames,
LVM_INT16 NrChannels);
#endif
-#else
-void LVC_MixSoft_2St_D16C31_SAT( LVMixer3_2St_st *pInstance,
- const LVM_INT16 *src1,
- LVM_INT16 *src2,
- LVM_INT16 *dst, /* dst cannot be equal to src2 */
- LVM_INT16 n);
-#endif
/**********************************************************************************/
/* For applying different gains to Left and right chennals */
/* MixerStream[0] applies to Left channel */
/* MixerStream[1] applies to Right channel */
/* Gain values should not be more that 1.0 */
/**********************************************************************************/
-#ifdef BUILD_FLOAT
#ifdef SUPPORT_MC
void LVC_MixSoft_1St_MC_float_SAT(LVMixer3_2St_FLOAT_st *pInstance,
const LVM_FLOAT *src,
@@ -236,15 +139,6 @@
const LVM_FLOAT *src,
LVM_FLOAT *dst, /* dst can be equal to src */
LVM_INT16 n); /* Number of stereo samples */
-#else
-void LVC_MixSoft_1St_2i_D16C31_SAT( LVMixer3_2St_st *pInstance,
- const LVM_INT16 *src,
- LVM_INT16 *dst, /* dst can be equal to src */
- LVM_INT16 n); /* Number of stereo samples */
-#endif
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
/**********************************************************************************/
diff --git a/media/libeffects/lvm/lib/Common/src/LVC_Mixer_GetCurrent.c b/media/libeffects/lvm/lib/Common/src/LVC_Mixer_GetCurrent.cpp
similarity index 85%
rename from media/libeffects/lvm/lib/Common/src/LVC_Mixer_GetCurrent.c
rename to media/libeffects/lvm/lib/Common/src/LVC_Mixer_GetCurrent.cpp
index 5990412..d0b50e6 100644
--- a/media/libeffects/lvm/lib/Common/src/LVC_Mixer_GetCurrent.c
+++ b/media/libeffects/lvm/lib/Common/src/LVC_Mixer_GetCurrent.cpp
@@ -19,7 +19,6 @@
#include "LVM_Macros.h"
#include "LVC_Mixer_Private.h"
-
/************************************************************************/
/* FUNCTION: */
/* LVMixer3_GetCurrent */
@@ -31,7 +30,6 @@
/* CurrentGain - CurrentGain value in Q 16.15 format */
/* */
/************************************************************************/
-#ifdef BUILD_FLOAT
LVM_FLOAT LVC_Mixer_GetCurrent( LVMixer3_FLOAT_st *pStream)
{
LVM_FLOAT CurrentGain;
@@ -39,12 +37,3 @@
CurrentGain = pInstance->Current; // CurrentGain
return CurrentGain;
}
-#else
-LVM_INT32 LVC_Mixer_GetCurrent( LVMixer3_st *pStream)
-{
- LVM_INT32 CurrentGain;
- Mix_Private_st *pInstance=(Mix_Private_st *)pStream->PrivateParams;
- CurrentGain=pInstance->Current>>(16-pInstance->Shift); // CurrentGain in Q16.15 format
- return CurrentGain;
-}
-#endif
diff --git a/media/libeffects/lvm/lib/Common/src/LVC_Mixer_GetTarget.c b/media/libeffects/lvm/lib/Common/src/LVC_Mixer_GetTarget.cpp
similarity index 85%
rename from media/libeffects/lvm/lib/Common/src/LVC_Mixer_GetTarget.c
rename to media/libeffects/lvm/lib/Common/src/LVC_Mixer_GetTarget.cpp
index 507eefa..3ae5ba4 100644
--- a/media/libeffects/lvm/lib/Common/src/LVC_Mixer_GetTarget.c
+++ b/media/libeffects/lvm/lib/Common/src/LVC_Mixer_GetTarget.cpp
@@ -30,7 +30,6 @@
/* TargetGain - TargetGain value in Q 16.15 format */
/* */
/************************************************************************/
-#ifdef BUILD_FLOAT
LVM_FLOAT LVC_Mixer_GetTarget( LVMixer3_FLOAT_st *pStream)
{
LVM_FLOAT TargetGain;
@@ -39,14 +38,3 @@
TargetGain = pInstance->Target; // TargetGain
return TargetGain;
}
-#else
-LVM_INT32 LVC_Mixer_GetTarget( LVMixer3_st *pStream)
-{
- LVM_INT32 TargetGain;
- Mix_Private_st *pInstance=(Mix_Private_st *)pStream->PrivateParams;
-
- TargetGain=pInstance->Target>>(16-pInstance->Shift); // TargetGain in Q16.15 format
-
- return TargetGain;
-}
-#endif
diff --git a/media/libeffects/lvm/lib/Common/src/LVC_Mixer_Init.c b/media/libeffects/lvm/lib/Common/src/LVC_Mixer_Init.cpp
similarity index 76%
rename from media/libeffects/lvm/lib/Common/src/LVC_Mixer_Init.c
rename to media/libeffects/lvm/lib/Common/src/LVC_Mixer_Init.cpp
index 737e26b..c9fd344 100644
--- a/media/libeffects/lvm/lib/Common/src/LVC_Mixer_Init.c
+++ b/media/libeffects/lvm/lib/Common/src/LVC_Mixer_Init.cpp
@@ -44,7 +44,6 @@
/* void */
/* */
/************************************************************************/
-#ifdef BUILD_FLOAT
void LVC_Mixer_Init( LVMixer3_FLOAT_st *pStream,
LVM_FLOAT TargetGain,
LVM_FLOAT CurrentGain)
@@ -56,24 +55,3 @@
pInstance->Target = TargetGain; // Update fractional gain Target
pInstance->Current = CurrentGain; // Update fractional gain Current
}
-#else
-void LVC_Mixer_Init( LVMixer3_st *pStream,
- LVM_INT32 TargetGain,
- LVM_INT32 CurrentGain)
-{
- LVM_INT16 Shift=0;
- LVM_INT32 MaxGain=TargetGain; // MaxGain is in Q16.15 format
- Mix_Private_st *pInstance=(Mix_Private_st *)pStream->PrivateParams;
- if(CurrentGain>MaxGain)
- MaxGain=CurrentGain; // MaxGain=max(CurrentGain,TargetGain)
-
- MaxGain=MaxGain>>15; // MaxGain in Q31.0 format i.e Integer part only
- while(MaxGain>0){ // Update Shift required to provide integer gain
- Shift++;
- MaxGain=MaxGain>>1;
- }
- pInstance->Target=TargetGain<<(16-Shift); // Update fractional gain Target
- pInstance->Current=CurrentGain<<(16-Shift); // Update fractional gain Current
- pInstance->Shift=Shift; // Update Shift
-}
-#endif
diff --git a/media/libeffects/lvm/lib/Common/src/LVC_Mixer_Private.h b/media/libeffects/lvm/lib/Common/src/LVC_Mixer_Private.h
index 453a6a5..123d22b 100644
--- a/media/libeffects/lvm/lib/Common/src/LVC_Mixer_Private.h
+++ b/media/libeffects/lvm/lib/Common/src/LVC_Mixer_Private.h
@@ -26,7 +26,6 @@
#include "VectorArithmetic.h"
/* Instance parameter structure */
-#ifdef BUILD_FLOAT
typedef struct
{
/* General */
@@ -34,16 +33,6 @@
LVM_FLOAT Current; /*number specifying value of Current Gain */
LVM_FLOAT Delta; /*number specifying value of Delta Gain */
} Mix_Private_FLOAT_st;
-#else
-typedef struct
-{
- /* General */
- LVM_INT32 Target; /* 32 bit number specifying fractional value of Target Gain */
- LVM_INT32 Current; /* 32 bit number specifying fractional valude of Current Gain */
- LVM_INT32 Shift; /* Left Shift for Integer part of Gain */
- LVM_INT32 Delta; /* 32 bit number specifying the fractional value of Delta Gain */
-} Mix_Private_st;
-#endif
/**********************************************************************************
DEFINITIONS
@@ -57,7 +46,6 @@
***********************************************************************************/
/*** 16 bit functions *************************************************************/
-#ifdef BUILD_FLOAT
void LVC_Core_MixInSoft_D16C31_SAT( LVMixer3_FLOAT_st *ptrInstance,
const LVM_FLOAT *src,
LVM_FLOAT *dst,
@@ -69,13 +57,6 @@
LVM_INT16 NrFrames,
LVM_INT16 NrChannels);
#endif
-#else
-void LVC_Core_MixInSoft_D16C31_SAT( LVMixer3_st *pInstance,
- const LVM_INT16 *src,
- LVM_INT16 *dst,
- LVM_INT16 n);
-#endif
-#ifdef BUILD_FLOAT
void LVC_Core_MixSoft_1St_D16C31_WRA( LVMixer3_FLOAT_st *ptrInstance,
const LVM_FLOAT *src,
LVM_FLOAT *dst,
@@ -87,27 +68,12 @@
LVM_INT16 NrFrames,
LVM_INT16 NrChannels);
#endif
-#else
-void LVC_Core_MixSoft_1St_D16C31_WRA( LVMixer3_st *pInstance,
- const LVM_INT16 *src,
- LVM_INT16 *dst,
- LVM_INT16 n);
-#endif
-#ifdef BUILD_FLOAT
void LVC_Core_MixHard_2St_D16C31_SAT( LVMixer3_FLOAT_st *pInstance1,
LVMixer3_FLOAT_st *pInstance2,
const LVM_FLOAT *src1,
const LVM_FLOAT *src2,
LVM_FLOAT *dst,
LVM_INT16 n);
-#else
-void LVC_Core_MixHard_2St_D16C31_SAT( LVMixer3_st *pInstance1,
- LVMixer3_st *pInstance2,
- const LVM_INT16 *src1,
- const LVM_INT16 *src2,
- LVM_INT16 *dst,
- LVM_INT16 n);
-#endif
/**********************************************************************************/
/* For applying different gains to Left and right chennals */
@@ -115,7 +81,6 @@
/* ptrInstance2 applies to Right channel */
/* Gain values should not be more that 1.0 */
/**********************************************************************************/
-#ifdef BUILD_FLOAT
#ifdef SUPPORT_MC
void LVC_Core_MixSoft_1St_MC_float_WRA(Mix_Private_FLOAT_st **ptrInstance,
const LVM_FLOAT *src,
@@ -128,13 +93,6 @@
const LVM_FLOAT *src,
LVM_FLOAT *dst,
LVM_INT16 n);
-#else
-void LVC_Core_MixSoft_1St_2i_D16C31_WRA( LVMixer3_st *ptrInstance1,
- LVMixer3_st *ptrInstance2,
- const LVM_INT16 *src,
- LVM_INT16 *dst, /* dst can be equal to src */
- LVM_INT16 n); /* Number of stereo samples */
-#endif
/**********************************************************************************/
/* For applying different gains to Left and right chennals */
@@ -142,7 +100,6 @@
/* ptrInstance2 applies to Right channel */
/* Gain values should not be more that 1.0 */
/**********************************************************************************/
-#ifdef BUILD_FLOAT
#ifdef SUPPORT_MC
void LVC_Core_MixHard_1St_MC_float_SAT(Mix_Private_FLOAT_st **ptrInstance,
const LVM_FLOAT *src,
@@ -155,43 +112,9 @@
const LVM_FLOAT *src,
LVM_FLOAT *dst,
LVM_INT16 n);
-#else
-void LVC_Core_MixHard_1St_2i_D16C31_SAT( LVMixer3_st *ptrInstance1,
- LVMixer3_st *ptrInstance2,
- const LVM_INT16 *src,
- LVM_INT16 *dst, /* dst can be equal to src */
- LVM_INT16 n); /* Number of stereo samples */
-#endif
/*** 32 bit functions *************************************************************/
-#ifndef BUILD_FLOAT
-void LVC_Core_MixInSoft_D32C31_SAT( LVMixer3_st *pInstance,
- const LVM_INT32 *src,
- LVM_INT32 *dst,
- LVM_INT16 n);
-
-void LVC_Core_MixSoft_1St_D32C31_WRA( LVMixer3_st *pInstance,
- const LVM_INT32 *src,
- LVM_INT32 *dst,
- LVM_INT16 n);
-
-void LVC_Core_MixHard_2St_D32C31_SAT( LVMixer3_st *pInstance1,
- LVMixer3_st *pInstance2,
- const LVM_INT32 *src1,
- const LVM_INT32 *src2,
- LVM_INT32 *dst,
- LVM_INT16 n);
-#endif
/**********************************************************************************/
#endif //#ifndef __LVC_MIXER_PRIVATE_H__
-
-
-
-
-
-
-
-
-
diff --git a/media/libeffects/lvm/lib/Common/src/LVC_Mixer_SetTarget.c b/media/libeffects/lvm/lib/Common/src/LVC_Mixer_SetTarget.cpp
similarity index 71%
rename from media/libeffects/lvm/lib/Common/src/LVC_Mixer_SetTarget.c
rename to media/libeffects/lvm/lib/Common/src/LVC_Mixer_SetTarget.cpp
index 577179d..47b0cec 100644
--- a/media/libeffects/lvm/lib/Common/src/LVC_Mixer_SetTarget.c
+++ b/media/libeffects/lvm/lib/Common/src/LVC_Mixer_SetTarget.cpp
@@ -43,32 +43,9 @@
/* void */
/* */
/************************************************************************/
-#ifdef BUILD_FLOAT
void LVC_Mixer_SetTarget(LVMixer3_FLOAT_st *pStream,
LVM_FLOAT TargetGain)
{
Mix_Private_FLOAT_st *pInstance = (Mix_Private_FLOAT_st *)pStream->PrivateParams;
pInstance->Target = TargetGain; // Update gain Target
}
-#else
-void LVC_Mixer_SetTarget(LVMixer3_st *pStream,
- LVM_INT32 TargetGain)
-{
- LVM_INT32 Shift=0;
- LVM_INT32 CurrentGain;
- LVM_INT32 MaxGain=TargetGain; // MaxGain is in Q16.15 format
- Mix_Private_st *pInstance=(Mix_Private_st *)pStream->PrivateParams;
- CurrentGain=pInstance->Current>>(16-pInstance->Shift); // CurrentGain in Q16.15 format
- if(CurrentGain>MaxGain)
- MaxGain=CurrentGain; // MaxGain=max(CurrentGain,TargetGain)
-
- MaxGain=MaxGain>>15; // MaxGain in Q31.0 format i.e Integer part only
- while(MaxGain>0){ // Update Shift required to provide integer gain
- Shift++;
- MaxGain=MaxGain>>1;
- }
- pInstance->Target=TargetGain<<(16-Shift); // Update fractional gain Target
- pInstance->Current=CurrentGain<<(16-Shift); // Update fractional gain Current
- pInstance->Shift=Shift; // Update Shift
-}
-#endif
diff --git a/media/libeffects/lvm/lib/Common/src/LVC_Mixer_SetTimeConstant.c b/media/libeffects/lvm/lib/Common/src/LVC_Mixer_SetTimeConstant.cpp
similarity index 71%
rename from media/libeffects/lvm/lib/Common/src/LVC_Mixer_SetTimeConstant.c
rename to media/libeffects/lvm/lib/Common/src/LVC_Mixer_SetTimeConstant.cpp
index 9d3ee88..1a8da7a 100644
--- a/media/libeffects/lvm/lib/Common/src/LVC_Mixer_SetTimeConstant.c
+++ b/media/libeffects/lvm/lib/Common/src/LVC_Mixer_SetTimeConstant.cpp
@@ -44,13 +44,11 @@
/* RETURNS: */
/* void */
/************************************************************************/
-#ifdef BUILD_FLOAT
void LVC_Mixer_SetTimeConstant(LVMixer3_FLOAT_st *pStream,
LVM_INT32 Tc_millisec,
LVM_Fs_en Fs,
LVM_INT16 NumChannels)
{
-#ifdef HIGHER_FS
LVM_FLOAT DeltaTable[13] = {0.500000f,/*8000*/
0.362812f,/*11025*/
0.333333f,/*12000*/
@@ -64,17 +62,6 @@
0.041667f,/*96000*/
0.022676f,/*176400*/
0.020833f};/*192000*/
-#else
- LVM_FLOAT DeltaTable[9] = {0.500000f,/*8000*/
- 0.362812f,/*11025*/
- 0.333333f,/*12000*/
- 0.250000f,/*16000*/
- 0.181406f,/*22050*/
- 0.166666f,/*24000*/
- 0.125000f,/*32000*/
- 0.090703f,/*44100*/
- 0.083333f};/*48000*/
-#endif
Mix_Private_FLOAT_st *pInstance = (Mix_Private_FLOAT_st *)pStream->PrivateParams;
LVM_FLOAT Delta = DeltaTable[Fs];
@@ -90,33 +77,3 @@
assign minimum value to Delta */
pInstance->Delta = Delta; // Delta=(2147483647*4*1000)/(NumChannels*SampleRate*Tc_millisec)
}
-#else
-void LVC_Mixer_SetTimeConstant(LVMixer3_st *pStream,
- LVM_INT32 Tc_millisec,
- LVM_Fs_en Fs,
- LVM_INT16 NumChannels)
-{
- LVM_INT32 DeltaTable[9]={1073741824,
- 779132389,
- 715827882,
- 536870912,
- 389566194,
- 357913941,
- 268435456,
- 194783097,
- 178956971};
- Mix_Private_st *pInstance=(Mix_Private_st *)pStream->PrivateParams;
- LVM_INT32 Delta=DeltaTable[Fs];
- Delta=Delta>>(NumChannels-1);
-
- if(Tc_millisec==0)
- Delta=0x7FFFFFFF;
- else
- Delta=Delta/Tc_millisec;
-
- if(Delta==0)
- Delta=1; // If Time Constant is so large that Delta is 0, assign minimum value to Delta
-
- pInstance->Delta=Delta; // Delta=(2147483647*4*1000)/(NumChannels*SampleRate*Tc_millisec) in Q 0.31 format
-}
-#endif
diff --git a/media/libeffects/lvm/lib/Common/src/LVC_Mixer_VarSlope_SetTimeConstant.c b/media/libeffects/lvm/lib/Common/src/LVC_Mixer_VarSlope_SetTimeConstant.cpp
similarity index 68%
rename from media/libeffects/lvm/lib/Common/src/LVC_Mixer_VarSlope_SetTimeConstant.c
rename to media/libeffects/lvm/lib/Common/src/LVC_Mixer_VarSlope_SetTimeConstant.cpp
index 0e0acf1..f335a1e 100644
--- a/media/libeffects/lvm/lib/Common/src/LVC_Mixer_VarSlope_SetTimeConstant.c
+++ b/media/libeffects/lvm/lib/Common/src/LVC_Mixer_VarSlope_SetTimeConstant.cpp
@@ -19,7 +19,6 @@
#include "LVM_Macros.h"
#include "LVC_Mixer_Private.h"
-
/************************************************************************/
/* FUNCTION: */
/* LVMixer3_VarSlope_SetTimeConstant */
@@ -45,13 +44,11 @@
/* RETURNS: */
/* void */
/************************************************************************/
-#ifdef BUILD_FLOAT
void LVC_Mixer_VarSlope_SetTimeConstant( LVMixer3_FLOAT_st *pStream,
LVM_INT32 Tc_millisec,
LVM_Fs_en Fs,
LVM_INT16 NumChannels)
{
-#ifdef HIGHER_FS
LVM_FLOAT DeltaTable[13] = {0.500000f,/*8000*/
0.362812f,/*11025*/
0.333333f,/*12000*/
@@ -65,17 +62,6 @@
0.041666f,/*96000*/
0.022676f,/*176400*/
0.020833f};/*192000*/
-#else
- LVM_FLOAT DeltaTable[9] = {0.500000f,/*8000*/
- 0.362812f,/*11025*/
- 0.333333f,/*12000*/
- 0.250000f,/*16000*/
- 0.181406f,/*22050*/
- 0.166666f,/*24000*/
- 0.125000f,/*32000*/
- 0.090703f,/*44100*/
- 0.083333f};/*48000*/
-#endif
LVM_FLOAT Tc_millisec_float;
Mix_Private_FLOAT_st *pInstance = (Mix_Private_FLOAT_st *)pStream->PrivateParams;
LVM_FLOAT Delta = DeltaTable[Fs];
@@ -112,52 +98,3 @@
pInstance->Delta = Delta; // Delta=(2147483647*4*1000)/(NumChannels*SampleRate*Tc_millisec)
}
-#else
-void LVC_Mixer_VarSlope_SetTimeConstant( LVMixer3_st *pStream,
- LVM_INT32 Tc_millisec,
- LVM_Fs_en Fs,
- LVM_INT16 NumChannels)
-{
- LVM_INT32 DeltaTable[9]={1073741824,
- 779132389,
- 715827882,
- 536870912,
- 389566194,
- 357913941,
- 268435456,
- 194783097,
- 178956971};
- Mix_Private_st *pInstance=(Mix_Private_st *)pStream->PrivateParams;
- LVM_INT32 Delta=DeltaTable[Fs];
-
- LVM_INT32 Current;
- LVM_INT32 Target;
-
- Delta=Delta>>(NumChannels-1);
-
- /* Get gain values */
- Current = LVC_Mixer_GetCurrent( pStream );
- Target = LVC_Mixer_GetTarget( pStream );
-
- if (Current != Target)
- {
- Tc_millisec = Tc_millisec * 32767 / (Current - Target);
- if (Tc_millisec<0) Tc_millisec = -Tc_millisec;
-
- if(Tc_millisec==0)
- Delta=0x7FFFFFFF;
- else
- Delta=Delta/Tc_millisec;
-
- if(Delta==0)
- Delta=1; // If Time Constant is so large that Delta is 0, assign minimum value to Delta
- }
- else
- {
- Delta =1; // Minimum value for proper call-backs (setting it to zero has some problems, to be corrected)
- }
-
-
- pInstance->Delta=Delta; // Delta=(2147483647*4*1000)/(NumChannels*SampleRate*Tc_millisec) in Q 0.31 format
-}
-#endif
diff --git a/media/libeffects/lvm/lib/Common/src/LVM_FO_HPF.c b/media/libeffects/lvm/lib/Common/src/LVM_FO_HPF.cpp
similarity index 81%
rename from media/libeffects/lvm/lib/Common/src/LVM_FO_HPF.c
rename to media/libeffects/lvm/lib/Common/src/LVM_FO_HPF.cpp
index 9094622..2497d29 100644
--- a/media/libeffects/lvm/lib/Common/src/LVM_FO_HPF.c
+++ b/media/libeffects/lvm/lib/Common/src/LVM_FO_HPF.cpp
@@ -21,7 +21,6 @@
#include "BIQUAD.h"
#include "Filter.h"
-
/*-------------------------------------------------------------------------*/
/* FUNCTION: */
/* void LVM_FO_LPF( LVM_INT32 w , */
@@ -68,7 +67,6 @@
/* RETURNS: */
/* */
/*-------------------------------------------------------------------------*/
-#ifdef BUILD_FLOAT
LVM_FLOAT LVM_FO_HPF( LVM_FLOAT w,
FO_FLOAT_Coefs_t *pCoeffs)
{
@@ -97,33 +95,3 @@
return 1;
}
-#else
-LVM_INT32 LVM_FO_HPF( LVM_INT32 w,
- FO_C32_Coefs_t *pCoeffs)
-{
- LVM_INT32 Y,Coefficients[13]={ -8388571,
- 33547744,
- -66816791,
- 173375308,
- -388437573,
- 752975383,
- -1103016663,
- 1121848567,
- -688078159,
- 194669577,
- 8,
- 0,
- 0};
- Y=LVM_Polynomial( (LVM_UINT16)9,
- Coefficients,
- w);
- pCoeffs->B1=-Y; /* Store -B1 in filter structure instead of B1!*/
- /* A0=(1-B1)/2= B1/2 - 0.5*/
- Y=Y>>1; /* A0=Y=B1/2*/
- Y=Y-0x40000000; /* A0=Y=(B1/2 - 0.5)*/
- MUL32x16INTO32(Y, FILTER_LOSS ,pCoeffs->A0 ,15) /* Apply loss to avoid overflow*/
- pCoeffs->A1=-pCoeffs->A0; /* Store A1=-A0*/
-
- return 1;
-}
-#endif
diff --git a/media/libeffects/lvm/lib/Common/src/LVM_FO_LPF.c b/media/libeffects/lvm/lib/Common/src/LVM_FO_LPF.cpp
similarity index 82%
rename from media/libeffects/lvm/lib/Common/src/LVM_FO_LPF.c
rename to media/libeffects/lvm/lib/Common/src/LVM_FO_LPF.cpp
index 9fe67f8..7bc6046 100644
--- a/media/libeffects/lvm/lib/Common/src/LVM_FO_LPF.c
+++ b/media/libeffects/lvm/lib/Common/src/LVM_FO_LPF.cpp
@@ -21,7 +21,6 @@
#include "BIQUAD.h"
#include "Filter.h"
-
/*-------------------------------------------------------------------------*/
/* FUNCTION: */
/* void LVM_FO_LPF( LVM_INT32 w , */
@@ -68,7 +67,6 @@
/* RETURNS: */
/* */
/*-------------------------------------------------------------------------*/
-#ifdef BUILD_FLOAT
LVM_FLOAT LVM_FO_LPF( LVM_FLOAT w,
FO_FLOAT_Coefs_t *pCoeffs)
{
@@ -94,30 +92,3 @@
return 1;
}
-#else
-LVM_INT32 LVM_FO_LPF( LVM_INT32 w,
- FO_C32_Coefs_t *pCoeffs)
-{
- LVM_INT32 Y,Coefficients[13]={ -8388571,
- 33547744,
- -66816791,
- 173375308,
- -388437573,
- 752975383,
- -1103016663,
- 1121848567,
- -688078159,
- 194669577,
- 8};
- Y=LVM_Polynomial( (LVM_UINT16)9,
- Coefficients,
- w);
- pCoeffs->B1=-Y; // Store -B1 in filter structure instead of B1!
- // A0=(1+B1)/2= B1/2 + 0.5
- Y=Y>>1; // A0=Y=B1/2
- Y=Y+0x40000000; // A0=Y=(B1/2 + 0.5)
- MUL32x16INTO32(Y, FILTER_LOSS ,pCoeffs->A0 ,15) // Apply loss to avoid overflow
- pCoeffs->A1=pCoeffs->A0;
- return 1;
-}
-#endif
diff --git a/media/libeffects/lvm/lib/Common/src/LVM_GetOmega.c b/media/libeffects/lvm/lib/Common/src/LVM_GetOmega.cpp
similarity index 79%
rename from media/libeffects/lvm/lib/Common/src/LVM_GetOmega.c
rename to media/libeffects/lvm/lib/Common/src/LVM_GetOmega.cpp
index 6307e68..2a7cca2 100644
--- a/media/libeffects/lvm/lib/Common/src/LVM_GetOmega.c
+++ b/media/libeffects/lvm/lib/Common/src/LVM_GetOmega.cpp
@@ -32,16 +32,6 @@
#define LVVDL_2PiByFs_SHIFT1 12 /* Qformat shift for 8kHz, 11.025kHz and 12kHz i.e. 12=41-29 */
#define LVVDL_2PiByFs_SHIFT2 13 /* Qformat shift for 16kHz, 22.050kHz and 24kHz i.e. 13=42-29 */
#define LVVDL_2PiByFs_SHIFT3 14 /* Qformat shift for 32kHz, 44.1kHz and 48kHz i.e. 14=43-29 */
-#ifndef BUILD_FLOAT
-const LVM_INT32 LVVDL_2PiOnFsTable[] = {LVVDL_2PiBy_8000 , /* 8kHz in Q41, 16kHz in Q42, 32kHz in Q43 */
- LVVDL_2PiBy_11025, /* 11025 Hz in Q41, 22050Hz in Q42, 44100 Hz in Q43*/
- LVVDL_2PiBy_12000}; /* 12kHz in Q41, 24kHz in Q42, 48kHz in Q43 */
-#endif
-
-const LVM_INT32 LVVDL_2PiOnFsShiftTable[]={LVVDL_2PiByFs_SHIFT1 , /* 8kHz, 11025Hz, 12kHz */
- LVVDL_2PiByFs_SHIFT2, /* 16kHz, 22050Hz, 24kHz*/
- LVVDL_2PiByFs_SHIFT3}; /* 32kHz, 44100Hz, 48kHz */
-#ifdef BUILD_FLOAT
#define LVVDL_2PiBy_8000_f 0.000785398f
#define LVVDL_2PiBy_11025_f 0.000569903f
#define LVVDL_2PiBy_12000_f 0.000523599f
@@ -52,12 +42,10 @@
#define LVVDL_2PiBy_44100_f 0.000142476f
#define LVVDL_2PiBy_48000_f 0.000130900f
-#ifdef HIGHER_FS
#define LVVDL_2PiBy_88200_f 0.000071238f
#define LVVDL_2PiBy_96000_f 0.000065450f
#define LVVDL_2PiBy_176400_f 0.000035619f
#define LVVDL_2PiBy_192000_f 0.000032725f
-#endif
const LVM_FLOAT LVVDL_2PiOnFsTable[] = {LVVDL_2PiBy_8000_f,
LVVDL_2PiBy_11025_f,
LVVDL_2PiBy_12000_f,
@@ -67,14 +55,11 @@
LVVDL_2PiBy_32000_f,
LVVDL_2PiBy_44100_f,
LVVDL_2PiBy_48000_f
-#ifdef HIGHER_FS
,LVVDL_2PiBy_88200_f
,LVVDL_2PiBy_96000_f
,LVVDL_2PiBy_176400_f
,LVVDL_2PiBy_192000_f
-#endif
};
-#endif
/*-------------------------------------------------------------------------*/
/* FUNCTION: */
/* LVM_GetOmega */
@@ -92,25 +77,10 @@
/* RETURNS: */
/* w=2*pi*Fc/Fs in Q2.29 format */
/*-------------------------------------------------------------------------*/
-#ifdef BUILD_FLOAT
-#ifdef HIGHER_FS
LVM_FLOAT LVM_GetOmega(LVM_UINT32 Fc,
LVM_Fs_en Fs)
-#else
-LVM_FLOAT LVM_GetOmega(LVM_UINT16 Fc,
- LVM_Fs_en Fs)
-#endif
{
LVM_FLOAT w;
w = (LVM_FLOAT)Fc * LVVDL_2PiOnFsTable[Fs];
return w;
}
-#else
-LVM_INT32 LVM_GetOmega(LVM_UINT16 Fc,
- LVM_Fs_en Fs)
-{
- LVM_INT32 w;
- MUL32x32INTO32((LVM_INT32)Fc,LVVDL_2PiOnFsTable[Fs%3],w,LVVDL_2PiOnFsShiftTable[Fs/3])
- return w;
-}
-#endif
diff --git a/media/libeffects/lvm/lib/Common/src/LVM_Mixer_FilterCoeffs.h b/media/libeffects/lvm/lib/Common/src/LVM_Mixer_FilterCoeffs.h
index f1e45fa..244f09d 100644
--- a/media/libeffects/lvm/lib/Common/src/LVM_Mixer_FilterCoeffs.h
+++ b/media/libeffects/lvm/lib/Common/src/LVM_Mixer_FilterCoeffs.h
@@ -27,7 +27,6 @@
#ifndef __LVM_MIXER_FILTER_COEFFS_H__
#define __LVM_MIXER_FILTER_COEFFS_H__
-
/************************************************************************************/
/* */
/* Alpha Time Constant table */
@@ -87,7 +86,6 @@
#define ALPHA_49 0 /* Floating point Alpha = 0.000000 */
#define ALPHA_50 0 /* Floating point Alpha = 0.000000 */
-#ifdef BUILD_FLOAT /* BUILD_FLOAT */
#define ALPHA_Float_0 0.999999f
#define ALPHA_Float_1 0.999998f
#define ALPHA_Float_2 0.999997f
@@ -139,6 +137,5 @@
#define ALPHA_Float_48 0.000000f
#define ALPHA_Float_49 0.000000f
#define ALPHA_Float_50 0.000000f
-#endif
#endif
diff --git a/media/libeffects/lvm/lib/Common/src/LVM_Mixer_TimeConstant.c b/media/libeffects/lvm/lib/Common/src/LVM_Mixer_TimeConstant.cpp
similarity index 68%
rename from media/libeffects/lvm/lib/Common/src/LVM_Mixer_TimeConstant.c
rename to media/libeffects/lvm/lib/Common/src/LVM_Mixer_TimeConstant.cpp
index 18b2782..73da2cf 100644
--- a/media/libeffects/lvm/lib/Common/src/LVM_Mixer_TimeConstant.c
+++ b/media/libeffects/lvm/lib/Common/src/LVM_Mixer_TimeConstant.cpp
@@ -20,7 +20,6 @@
#include "Mixer.h"
#include "LVM_Mixer_FilterCoeffs.h"
-
/************************************************************************/
/* FUNCTION: */
/* LVM_Mix_GetTimeConstant */
@@ -57,13 +56,8 @@
/* Alpha - the filter coefficient Q31 format */
/* */
/************************************************************************/
-#ifdef BUILD_FLOAT
LVM_FLOAT LVM_Mixer_TimeConstant(LVM_UINT32 tc,
-#ifdef HIGHER_FS
LVM_UINT32 Fs,
-#else
- LVM_UINT16 Fs,
-#endif
LVM_UINT16 NumChannels)
{
@@ -160,101 +154,3 @@
return ProductFloat;
}
-#else
-LVM_UINT32 LVM_Mixer_TimeConstant(LVM_UINT32 tc,
- LVM_UINT16 Fs,
- LVM_UINT16 NumChannels)
-{
-
- LVM_UINT32 Product;
- LVM_INT16 Interpolate;
- LVM_UINT16 Shift;
- LVM_INT32 Diff;
- LVM_UINT32 Table[] = {ALPHA_0, /* Log spaced look-up table */
- ALPHA_1,
- ALPHA_2,
- ALPHA_3,
- ALPHA_4,
- ALPHA_5,
- ALPHA_6,
- ALPHA_7,
- ALPHA_8,
- ALPHA_9,
- ALPHA_10,
- ALPHA_11,
- ALPHA_12,
- ALPHA_13,
- ALPHA_14,
- ALPHA_15,
- ALPHA_16,
- ALPHA_17,
- ALPHA_18,
- ALPHA_19,
- ALPHA_20,
- ALPHA_21,
- ALPHA_22,
- ALPHA_23,
- ALPHA_24,
- ALPHA_25,
- ALPHA_26,
- ALPHA_27,
- ALPHA_28,
- ALPHA_29,
- ALPHA_30,
- ALPHA_31,
- ALPHA_32,
- ALPHA_33,
- ALPHA_34,
- ALPHA_35,
- ALPHA_36,
- ALPHA_37,
- ALPHA_38,
- ALPHA_39,
- ALPHA_40,
- ALPHA_41,
- ALPHA_42,
- ALPHA_43,
- ALPHA_44,
- ALPHA_45,
- ALPHA_46,
- ALPHA_47,
- ALPHA_48,
- ALPHA_49,
- ALPHA_50};
-
-
- /* Calculate the product of the time constant and the sample rate */
- Product = ((tc >> 16) * (LVM_UINT32)Fs) << 13; /* Stereo value */
- Product = Product + (((tc & 0x0000FFFF) * (LVM_UINT32)Fs) >> 3);
-
- if (NumChannels == 1)
- {
- Product = Product >> 1; /* Mono value */
- }
-
- /* Normalize to get the table index and interpolation factor */
- for (Shift=0; Shift<((Alpha_TableSize-1)/2); Shift++)
- {
- if ((Product & 0x80000000)!=0)
- {
- break;
- }
-
- Product = Product << 1;
- }
- Shift = (LVM_UINT16)((Shift << 1));
-
- if ((Product & 0x40000000)==0)
- {
- Shift++;
- }
-
- Interpolate = (LVM_INT16)((Product >> 15) & 0x00007FFF);
-
- Diff = (LVM_INT32)(Table[Shift] - Table[Shift+1]);
- MUL32x16INTO32(Diff,Interpolate,Diff,15)
- Product = Table[Shift+1] + (LVM_UINT32)Diff;
-
- return Product;
-}
-#endif
diff --git a/media/libeffects/lvm/lib/Common/src/LVM_Polynomial.c b/media/libeffects/lvm/lib/Common/src/LVM_Polynomial.cpp
similarity index 76%
rename from media/libeffects/lvm/lib/Common/src/LVM_Polynomial.c
rename to media/libeffects/lvm/lib/Common/src/LVM_Polynomial.cpp
index cd57767..2c3e9ec 100644
--- a/media/libeffects/lvm/lib/Common/src/LVM_Polynomial.c
+++ b/media/libeffects/lvm/lib/Common/src/LVM_Polynomial.cpp
@@ -40,7 +40,6 @@
/* RETURNS: */
/* The result of the polynomial expansion in Q1.31 format */
/*-------------------------------------------------------------------------*/
-#ifdef BUILD_FLOAT
LVM_FLOAT LVM_Polynomial(LVM_UINT16 N,
LVM_FLOAT *pCoefficients,
LVM_FLOAT X)
@@ -62,7 +61,6 @@
sign *= Temp;
}
-
}
else
{
@@ -81,57 +79,3 @@
}
return Y;
}
-#else
-LVM_INT32 LVM_Polynomial(LVM_UINT16 N,
- LVM_INT32 *pCoefficients,
- LVM_INT32 X)
-{
- LVM_INT32 i;
- LVM_INT32 Y,A,XTemp,Temp,sign;
-
- Y=*pCoefficients; /* Y=A0*/
- pCoefficients++;
-
- if((LVM_UINT32)X==0x80000000)
- {
- Temp=-1;
- sign=Temp;
- for(i=1;i<=N;i++)
- {
- Y+=((*pCoefficients)*sign);
- pCoefficients++;
- sign*=Temp;
- }
-
-
- }
- else
- {
- XTemp=X;
- for(i=N-1;i>=0;i--)
- {
- A=*pCoefficients;
- pCoefficients++;
-
- MUL32x32INTO32(A,XTemp,Temp,31)
- Y+=Temp;
-
- MUL32x32INTO32(XTemp,X,Temp,31)
- XTemp=Temp;
- }
- }
- A=*pCoefficients;
- pCoefficients++;
-
- if(A<0)
- {
- A=Abs_32(A);
- Y=Y>>A;
- }
- else
- {
- Y = Y<<A;
- }
- return Y;
-}
-#endif
diff --git a/media/libeffects/lvm/lib/Common/src/LVM_Power10.c b/media/libeffects/lvm/lib/Common/src/LVM_Power10.cpp
similarity index 82%
rename from media/libeffects/lvm/lib/Common/src/LVM_Power10.c
rename to media/libeffects/lvm/lib/Common/src/LVM_Power10.cpp
index 8785594..ae8e9d1 100644
--- a/media/libeffects/lvm/lib/Common/src/LVM_Power10.c
+++ b/media/libeffects/lvm/lib/Common/src/LVM_Power10.cpp
@@ -20,7 +20,6 @@
#include "ScalarArithmetic.h"
#include "Filter.h"
-
/*-------------------------------------------------------------------------*/
/* FUNCTION: */
/* LVM_Power10 */
@@ -54,7 +53,6 @@
/* RETURNS: */
/* The result of the 10x expansion in Q8.24 format */
/*-------------------------------------------------------------------------*/
-#ifdef BUILD_FLOAT
LVM_FLOAT LVM_Power10(LVM_FLOAT X)
{
LVM_FLOAT Y,Coefficients[13]={0.999906f,
@@ -75,25 +73,3 @@
X);
return Y;
}
-#else
-LVM_INT32 LVM_Power10(LVM_INT32 X)
-{
- LVM_INT32 Y,Coefficients[13]={ 16775636,
- 77258249,
- 178024032,
- 273199333,
- 312906284,
- 288662365,
- 228913700,
- 149470921,
- 71094558,
- 37565524,
- 31223618,
- 12619311,
- 0};
- Y=LVM_Polynomial((LVM_UINT16)11,
- Coefficients,
- X);
- return Y;
-}
-#endif
diff --git a/media/libeffects/lvm/lib/Common/src/LVM_Timer.c b/media/libeffects/lvm/lib/Common/src/LVM_Timer.cpp
similarity index 100%
rename from media/libeffects/lvm/lib/Common/src/LVM_Timer.c
rename to media/libeffects/lvm/lib/Common/src/LVM_Timer.cpp
diff --git a/media/libeffects/lvm/lib/Common/src/LVM_Timer_Init.c b/media/libeffects/lvm/lib/Common/src/LVM_Timer_Init.cpp
similarity index 96%
rename from media/libeffects/lvm/lib/Common/src/LVM_Timer_Init.c
rename to media/libeffects/lvm/lib/Common/src/LVM_Timer_Init.cpp
index a935cfe..3015057 100644
--- a/media/libeffects/lvm/lib/Common/src/LVM_Timer_Init.c
+++ b/media/libeffects/lvm/lib/Common/src/LVM_Timer_Init.cpp
@@ -40,7 +40,7 @@
pInstancePr = (LVM_Timer_Instance_Private_t *)pInstance;
pInstancePr->CallBackParam = pParams->CallBackParam;
- pInstancePr->pCallBackParams = pParams->pCallBackParams;
+ pInstancePr->pCallBackParams = (LVM_INT32 *)pParams->pCallBackParams;
pInstancePr->pCallbackInstance = pParams->pCallbackInstance;
pInstancePr->pCallBack = pParams->pCallBack;
pInstancePr->TimerArmed = 1;
diff --git a/media/libeffects/lvm/lib/Common/src/LVM_Timer_Private.h b/media/libeffects/lvm/lib/Common/src/LVM_Timer_Private.h
index 480944f..a372b82 100644
--- a/media/libeffects/lvm/lib/Common/src/LVM_Timer_Private.h
+++ b/media/libeffects/lvm/lib/Common/src/LVM_Timer_Private.h
@@ -18,12 +18,6 @@
#ifndef LVM_TIMER_PRIVATE_H
#define LVM_TIMER_PRIVATE_H
-
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
-
#include "LVM_Types.h"
/****************************************************************************************/
@@ -45,8 +39,4 @@
/* END OF HEADER */
/****************************************************************************************/
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
-
#endif /* LVM_TIMER_PRIVATE_H */
diff --git a/media/libeffects/lvm/lib/Common/src/LoadConst_16.c b/media/libeffects/lvm/lib/Common/src/LoadConst_16.cpp
similarity index 100%
rename from media/libeffects/lvm/lib/Common/src/LoadConst_16.c
rename to media/libeffects/lvm/lib/Common/src/LoadConst_16.cpp
diff --git a/media/libeffects/lvm/lib/Common/src/LoadConst_32.c b/media/libeffects/lvm/lib/Common/src/LoadConst_32.cpp
similarity index 83%
rename from media/libeffects/lvm/lib/Common/src/LoadConst_32.c
rename to media/libeffects/lvm/lib/Common/src/LoadConst_32.cpp
index 9e14c3b..c789756 100644
--- a/media/libeffects/lvm/lib/Common/src/LoadConst_32.c
+++ b/media/libeffects/lvm/lib/Common/src/LoadConst_32.cpp
@@ -24,7 +24,6 @@
/**********************************************************************************
FUNCTION LoadConst_32
***********************************************************************************/
-#ifdef BUILD_FLOAT
void LoadConst_Float(const LVM_FLOAT val,
LVM_FLOAT *dst,
LVM_INT16 n )
@@ -39,21 +38,5 @@
return;
}
-#else
-void LoadConst_32(const LVM_INT32 val,
- LVM_INT32 *dst,
- LVM_INT16 n )
-{
- LVM_INT16 ii;
-
- for (ii = n; ii != 0; ii--)
- {
- *dst = val;
- dst++;
- }
-
- return;
-}
-#endif
/**********************************************************************************/
diff --git a/media/libeffects/lvm/lib/Common/src/MSTo2i_Sat_16x16.c b/media/libeffects/lvm/lib/Common/src/MSTo2i_Sat_16x16.cpp
similarity index 98%
rename from media/libeffects/lvm/lib/Common/src/MSTo2i_Sat_16x16.c
rename to media/libeffects/lvm/lib/Common/src/MSTo2i_Sat_16x16.cpp
index 02c906a..1ea765a 100644
--- a/media/libeffects/lvm/lib/Common/src/MSTo2i_Sat_16x16.c
+++ b/media/libeffects/lvm/lib/Common/src/MSTo2i_Sat_16x16.cpp
@@ -33,7 +33,6 @@
LVM_INT32 temp,mVal,sVal;
LVM_INT16 ii;
-
for (ii = n; ii != 0; ii--)
{
mVal=(LVM_INT32)*srcM;
@@ -77,7 +76,6 @@
return;
}
-#ifdef BUILD_FLOAT
void MSTo2i_Sat_Float(const LVM_FLOAT *srcM,
const LVM_FLOAT *srcS,
LVM_FLOAT *dst,
@@ -86,7 +84,6 @@
LVM_FLOAT temp,mVal,sVal;
LVM_INT16 ii;
-
for (ii = n; ii != 0; ii--)
{
mVal = (LVM_FLOAT)*srcM;
@@ -130,5 +127,4 @@
return;
}
-#endif
/**********************************************************************************/
diff --git a/media/libeffects/lvm/lib/Common/src/Mac3s_Sat_16x16.c b/media/libeffects/lvm/lib/Common/src/Mac3s_Sat_16x16.cpp
similarity index 99%
rename from media/libeffects/lvm/lib/Common/src/Mac3s_Sat_16x16.c
rename to media/libeffects/lvm/lib/Common/src/Mac3s_Sat_16x16.cpp
index ef04ae8..6584251 100644
--- a/media/libeffects/lvm/lib/Common/src/Mac3s_Sat_16x16.c
+++ b/media/libeffects/lvm/lib/Common/src/Mac3s_Sat_16x16.cpp
@@ -44,7 +44,6 @@
LVM_INT16 srcval;
LVM_INT32 Temp,dInVal;
-
for (ii = n; ii != 0; ii--)
{
srcval=*src;
@@ -77,5 +76,3 @@
/**********************************************************************************/
-
-
diff --git a/media/libeffects/lvm/lib/Common/src/Mac3s_Sat_32x16.c b/media/libeffects/lvm/lib/Common/src/Mac3s_Sat_32x16.cpp
similarity index 98%
rename from media/libeffects/lvm/lib/Common/src/Mac3s_Sat_32x16.c
rename to media/libeffects/lvm/lib/Common/src/Mac3s_Sat_32x16.cpp
index 17fd833..5d5564f 100644
--- a/media/libeffects/lvm/lib/Common/src/Mac3s_Sat_32x16.c
+++ b/media/libeffects/lvm/lib/Common/src/Mac3s_Sat_32x16.cpp
@@ -34,7 +34,6 @@
LVM_INT16 ii;
LVM_INT32 srcval,temp, dInVal, dOutVal;
-
for (ii = n; ii != 0; ii--)
{
srcval=*src;
@@ -45,7 +44,6 @@
dInVal = *dst;
dOutVal = temp + dInVal;
-
if ((((dOutVal ^ temp) & (dOutVal ^ dInVal)) >> 31)!=0) /* overflow / underflow */
{
if(temp<0)
@@ -64,7 +62,6 @@
return;
}
-#ifdef BUILD_FLOAT
void Mac3s_Sat_Float(const LVM_FLOAT *src,
const LVM_FLOAT val,
LVM_FLOAT *dst,
@@ -101,8 +98,5 @@
return;
}
-#endif
/**********************************************************************************/
-
-
diff --git a/media/libeffects/lvm/lib/Common/src/MixInSoft_D32C31_SAT.c b/media/libeffects/lvm/lib/Common/src/MixInSoft_D32C31_SAT.cpp
similarity index 62%
rename from media/libeffects/lvm/lib/Common/src/MixInSoft_D32C31_SAT.c
rename to media/libeffects/lvm/lib/Common/src/MixInSoft_D32C31_SAT.cpp
index 16e367b..7c7b36f 100644
--- a/media/libeffects/lvm/lib/Common/src/MixInSoft_D32C31_SAT.c
+++ b/media/libeffects/lvm/lib/Common/src/MixInSoft_D32C31_SAT.cpp
@@ -32,7 +32,6 @@
/**********************************************************************************
FUNCTION MIXINSOFT_D32C31_SAT
***********************************************************************************/
-#ifdef BUILD_FLOAT
void MixInSoft_D32C31_SAT( Mix_1St_Cll_FLOAT_t *pInstance,
const LVM_FLOAT *src,
LVM_FLOAT *dst,
@@ -96,64 +95,4 @@
}
}
}
-#else
-void MixInSoft_D32C31_SAT( Mix_1St_Cll_t *pInstance,
- const LVM_INT32 *src,
- LVM_INT32 *dst,
- LVM_INT16 n)
-{
- char HardMixing = TRUE;
-
- if(n<=0) return;
-
- /******************************************************************************
- SOFT MIXING
- *******************************************************************************/
- if (pInstance->Current != pInstance->Target)
- {
- if(pInstance->Alpha == 0){
- pInstance->Current = pInstance->Target;
- }else if ((pInstance->Current-pInstance->Target <POINT_ZERO_ONE_DB)&&
- (pInstance->Current-pInstance->Target > -POINT_ZERO_ONE_DB)){
- pInstance->Current = pInstance->Target; /* Difference is not significant anymore. Make them equal. */
- }else{
- /* Soft mixing has to be applied */
- HardMixing = FALSE;
- Core_MixInSoft_D32C31_SAT( pInstance, src, dst, n);
- }
- }
-
- /******************************************************************************
- HARD MIXING
- *******************************************************************************/
-
- if (HardMixing){
- if (pInstance->Target != 0){ /* Nothing to do in case Target = 0 */
- if ((pInstance->Target>>16) == 0x7FFF)
- Add2_Sat_32x32( src, dst, n );
- else{
- Core_MixInSoft_D32C31_SAT( pInstance, src, dst, n);
- pInstance->Current = pInstance->Target; /* In case the core function would have changed the Current value */
- }
- }
- }
-
- /******************************************************************************
- CALL BACK
- *******************************************************************************/
- /* Call back before the hard mixing, because in this case, hard mixing makes
- use of the core soft mix function which can change the Current value! */
-
- if (pInstance->CallbackSet){
- if ((pInstance->Current-pInstance->Target <POINT_ZERO_ONE_DB)&&
- (pInstance->Current-pInstance->Target > -POINT_ZERO_ONE_DB)){
- pInstance->Current = pInstance->Target; /* Difference is not significant anymore. Make them equal. */
- pInstance->CallbackSet = FALSE;
- if (pInstance->pCallBack != 0){
- (*pInstance->pCallBack) ( pInstance->pCallbackHandle, pInstance->pGeneralPurpose,pInstance->CallbackParam );
- }
- }
- }
-}
-#endif
/**********************************************************************************/
diff --git a/media/libeffects/lvm/lib/Common/src/MixSoft_1St_D32C31_WRA.c b/media/libeffects/lvm/lib/Common/src/MixSoft_1St_D32C31_WRA.cpp
similarity index 62%
rename from media/libeffects/lvm/lib/Common/src/MixSoft_1St_D32C31_WRA.c
rename to media/libeffects/lvm/lib/Common/src/MixSoft_1St_D32C31_WRA.cpp
index 869293b..d3325ec 100644
--- a/media/libeffects/lvm/lib/Common/src/MixSoft_1St_D32C31_WRA.c
+++ b/media/libeffects/lvm/lib/Common/src/MixSoft_1St_D32C31_WRA.cpp
@@ -29,12 +29,9 @@
#define TRUE 1
#define FALSE 0
-
-
/**********************************************************************************
FUNCTION MIXSOFT_1ST_D32C31_WRA
***********************************************************************************/
-#ifdef BUILD_FLOAT
void MixSoft_1St_D32C31_WRA( Mix_1St_Cll_FLOAT_t *pInstance,
const LVM_FLOAT *src,
LVM_FLOAT *dst,
@@ -95,62 +92,4 @@
}
}
}
-#else
-void MixSoft_1St_D32C31_WRA( Mix_1St_Cll_t *pInstance,
- const LVM_INT32 *src,
- LVM_INT32 *dst,
- LVM_INT16 n)
-{
- char HardMixing = TRUE;
-
- if(n<=0) return;
-
- /******************************************************************************
- SOFT MIXING
- *******************************************************************************/
- if (pInstance->Current != pInstance->Target)
- {
- if(pInstance->Alpha == 0){
- pInstance->Current = pInstance->Target;
- }else if ((pInstance->Current-pInstance->Target <POINT_ZERO_ONE_DB)&&
- (pInstance->Current-pInstance->Target > -POINT_ZERO_ONE_DB)){
- pInstance->Current = pInstance->Target; /* Difference is not significant anymore. Make them equal. */
- }else{
- /* Soft mixing has to be applied */
- HardMixing = FALSE;
- Core_MixSoft_1St_D32C31_WRA( pInstance, src, dst, n);
- }
- }
-
- /******************************************************************************
- HARD MIXING
- *******************************************************************************/
-
- if (HardMixing){
- if (pInstance->Target == 0)
- LoadConst_32(0, dst, n);
- else if ((pInstance->Target>>16) == 0x7FFF){
- if (src != dst)
- Copy_16((LVM_INT16*)src, (LVM_INT16*)dst, (LVM_INT16)(n * 2));
- }
- else
- Mult3s_32x16( src, (LVM_INT16)(pInstance->Current>>16), dst, n );
- }
-
- /******************************************************************************
- CALL BACK
- *******************************************************************************/
-
- if (pInstance->CallbackSet){
- if ((pInstance->Current-pInstance->Target <POINT_ZERO_ONE_DB)&&
- (pInstance->Current-pInstance->Target > -POINT_ZERO_ONE_DB)){
- pInstance->Current = pInstance->Target; /* Difference is not significant anymore. Make them equal. */
- pInstance->CallbackSet = FALSE;
- if (pInstance->pCallBack != 0){
- (*pInstance->pCallBack) ( pInstance->pCallbackHandle, pInstance->pGeneralPurpose,pInstance->CallbackParam );
- }
- }
- }
-}
-#endif
/**********************************************************************************/
diff --git a/media/libeffects/lvm/lib/Common/src/MixSoft_2St_D32C31_SAT.c b/media/libeffects/lvm/lib/Common/src/MixSoft_2St_D32C31_SAT.cpp
similarity index 62%
rename from media/libeffects/lvm/lib/Common/src/MixSoft_2St_D32C31_SAT.c
rename to media/libeffects/lvm/lib/Common/src/MixSoft_2St_D32C31_SAT.cpp
index 6fc1b92..b002738 100644
--- a/media/libeffects/lvm/lib/Common/src/MixSoft_2St_D32C31_SAT.c
+++ b/media/libeffects/lvm/lib/Common/src/MixSoft_2St_D32C31_SAT.cpp
@@ -22,11 +22,9 @@
#include "Mixer_private.h"
#include "VectorArithmetic.h"
-
/**********************************************************************************
FUNCTION MIXSOFT_2ST_D32C31_SAT
***********************************************************************************/
-#ifdef BUILD_FLOAT
void MixSoft_2St_D32C31_SAT( Mix_2St_Cll_FLOAT_t *pInstance,
const LVM_FLOAT *src1,
const LVM_FLOAT *src2,
@@ -42,7 +40,7 @@
if ((pInstance->Current1 != pInstance->Target1) || (pInstance->Current2 != pInstance->Target2))
{
MixSoft_1St_D32C31_WRA((Mix_1St_Cll_FLOAT_t*)pInstance, src1, dst, n);
- MixInSoft_D32C31_SAT((void *)&pInstance->Alpha2, /* Cast to void: \
+ MixInSoft_D32C31_SAT((Mix_1St_Cll_FLOAT_t *)&pInstance->Alpha2, /* Cast to void: \
no dereferencing in function*/
src2, dst, n);
}
@@ -54,7 +52,8 @@
else
{
if (pInstance->Current1 == 0)
- MixSoft_1St_D32C31_WRA((void *) &pInstance->Alpha2, /* Cast to void: no \
+ MixSoft_1St_D32C31_WRA(
+ (Mix_1St_Cll_FLOAT_t *) &pInstance->Alpha2, /* Cast to void: no \
dereferencing in function*/
src2, dst, n);
else if (pInstance->Current2 == 0)
@@ -63,41 +62,5 @@
Core_MixHard_2St_D32C31_SAT(pInstance, src1, src2, dst, n);
}
}
-#else
-void MixSoft_2St_D32C31_SAT( Mix_2St_Cll_t *pInstance,
- const LVM_INT32 *src1,
- const LVM_INT32 *src2,
- LVM_INT32 *dst,
- LVM_INT16 n)
-{
-
- if(n<=0) return;
-
- /******************************************************************************
- SOFT MIXING
- *******************************************************************************/
- if ((pInstance->Current1 != pInstance->Target1) || (pInstance->Current2 != pInstance->Target2))
- {
- MixSoft_1St_D32C31_WRA( (Mix_1St_Cll_t*) pInstance, src1, dst, n);
- MixInSoft_D32C31_SAT( (void *) &pInstance->Alpha2, /* Cast to void: no dereferencing in function*/
- src2, dst, n);
- }
-
- /******************************************************************************
- HARD MIXING
- *******************************************************************************/
-
- else
- {
- if (pInstance->Current1 == 0)
- MixSoft_1St_D32C31_WRA( (void *) &pInstance->Alpha2, /* Cast to void: no dereferencing in function*/
- src2, dst, n);
- else if (pInstance->Current2 == 0)
- MixSoft_1St_D32C31_WRA( (Mix_1St_Cll_t*) pInstance, src1, dst, n);
- else
- Core_MixHard_2St_D32C31_SAT( pInstance, src1, src2, dst, n);
- }
-}
-#endif
/**********************************************************************************/
diff --git a/media/libeffects/lvm/lib/Common/src/Mixer_private.h b/media/libeffects/lvm/lib/Common/src/Mixer_private.h
index 00d55ed..1d653bb 100644
--- a/media/libeffects/lvm/lib/Common/src/Mixer_private.h
+++ b/media/libeffects/lvm/lib/Common/src/Mixer_private.h
@@ -26,10 +26,8 @@
#define POINT_ZERO_ONE_DB 2473805 /* 0.01 dB on a full scale signal = (10^(0.01/20) -1) * 2^31 */
-#ifdef BUILD_FLOAT
#define POINT_ZERO_ONE_DB_FLOAT 0.001152 /* 0.01 dB on a full scale \
signal = (10^(0.01/20) -1) * 2^31 */
-#endif
/**********************************************************************************
DEFINITIONS
***********************************************************************************/
diff --git a/media/libeffects/lvm/lib/Common/src/MonoTo2I_16.c b/media/libeffects/lvm/lib/Common/src/MonoTo2I_16.cpp
similarity index 100%
rename from media/libeffects/lvm/lib/Common/src/MonoTo2I_16.c
rename to media/libeffects/lvm/lib/Common/src/MonoTo2I_16.cpp
diff --git a/media/libeffects/lvm/lib/Common/src/MonoTo2I_32.c b/media/libeffects/lvm/lib/Common/src/MonoTo2I_32.cpp
similarity index 98%
rename from media/libeffects/lvm/lib/Common/src/MonoTo2I_32.c
rename to media/libeffects/lvm/lib/Common/src/MonoTo2I_32.cpp
index 796a15c..603d1fc 100644
--- a/media/libeffects/lvm/lib/Common/src/MonoTo2I_32.c
+++ b/media/libeffects/lvm/lib/Common/src/MonoTo2I_32.cpp
@@ -45,7 +45,6 @@
return;
}
-#ifdef BUILD_FLOAT
void MonoTo2I_Float( const LVM_FLOAT *src,
LVM_FLOAT *dst,
LVM_INT16 n)
@@ -66,5 +65,4 @@
return;
}
-#endif
/**********************************************************************************/
diff --git a/media/libeffects/lvm/lib/Common/src/Mult3s_32x16.c b/media/libeffects/lvm/lib/Common/src/Mult3s_32x16.cpp
similarity index 98%
rename from media/libeffects/lvm/lib/Common/src/Mult3s_32x16.c
rename to media/libeffects/lvm/lib/Common/src/Mult3s_32x16.cpp
index c758560..370c39a 100644
--- a/media/libeffects/lvm/lib/Common/src/Mult3s_32x16.c
+++ b/media/libeffects/lvm/lib/Common/src/Mult3s_32x16.cpp
@@ -47,7 +47,6 @@
return;
}
-#ifdef BUILD_FLOAT
void Mult3s_Float( const LVM_FLOAT *src,
const LVM_FLOAT val,
LVM_FLOAT *dst,
@@ -65,5 +64,4 @@
}
return;
}
-#endif
/**********************************************************************************/
diff --git a/media/libeffects/lvm/lib/Common/src/NonLinComp_D16.c b/media/libeffects/lvm/lib/Common/src/NonLinComp_D16.cpp
similarity index 99%
rename from media/libeffects/lvm/lib/Common/src/NonLinComp_D16.c
rename to media/libeffects/lvm/lib/Common/src/NonLinComp_D16.cpp
index 5156edc..36d1149 100644
--- a/media/libeffects/lvm/lib/Common/src/NonLinComp_D16.c
+++ b/media/libeffects/lvm/lib/Common/src/NonLinComp_D16.cpp
@@ -71,7 +71,6 @@
LVM_INT32 SampleNo; /* Sample index */
LVM_INT16 Temp;
-
/*
* Process a block of samples
*/
@@ -84,7 +83,6 @@
Sample = *pDataIn;
pDataIn++;
-
/*
* Apply the compander, this compresses the signal at the expense of
* harmonic distortion. The amount of compression is control by the
@@ -103,18 +101,15 @@
}
}
-
/*
* Save the output
*/
*pDataOut = Sample;
pDataOut++;
-
}
}
-#ifdef BUILD_FLOAT
void NonLinComp_Float(LVM_FLOAT Gain,
LVM_FLOAT *pDataIn,
LVM_FLOAT *pDataOut,
@@ -125,7 +120,6 @@
LVM_INT32 SampleNo; /* Sample index */
LVM_FLOAT Temp;
-
/*
* Process a block of samples
*/
@@ -137,7 +131,6 @@
Sample = *pDataIn;
pDataIn++;
-
/*
* Apply the compander, this compresses the signal at the expense of
* harmonic distortion. The amount of compression is control by the
@@ -156,7 +149,6 @@
}
}
-
/*
* Save the output
*/
@@ -164,4 +156,3 @@
pDataOut++;
}
}
-#endif
diff --git a/media/libeffects/lvm/lib/Common/src/PK_2I_D32F32C14G11_TRC_WRA_01.c b/media/libeffects/lvm/lib/Common/src/PK_2I_D32F32C14G11_TRC_WRA_01.cpp
similarity index 68%
rename from media/libeffects/lvm/lib/Common/src/PK_2I_D32F32C14G11_TRC_WRA_01.c
rename to media/libeffects/lvm/lib/Common/src/PK_2I_D32F32C14G11_TRC_WRA_01.cpp
index 6c8b2db..3f62f99 100644
--- a/media/libeffects/lvm/lib/Common/src/PK_2I_D32F32C14G11_TRC_WRA_01.c
+++ b/media/libeffects/lvm/lib/Common/src/PK_2I_D32F32C14G11_TRC_WRA_01.cpp
@@ -27,7 +27,6 @@
pBiquadState->coefs[2] is -B1, these are in Q14 format
pBiquadState->coefs[3] is Gain, in Q11 format
-
DELAYS-
pBiquadState->pDelays[0] is x(n-1)L in Q0 format
pBiquadState->pDelays[1] is x(n-1)R in Q0 format
@@ -38,7 +37,6 @@
pBiquadState->pDelays[6] is y(n-2)L in Q0 format
pBiquadState->pDelays[7] is y(n-2)R in Q0 format
***************************************************************************/
-#ifdef BUILD_FLOAT
void PK_2I_D32F32C14G11_TRC_WRA_01 ( Biquad_FLOAT_Instance_t *pInstance,
LVM_FLOAT *pDataIn,
LVM_FLOAT *pDataOut,
@@ -51,7 +49,6 @@
for (ii = NrSamples; ii != 0; ii--)
{
-
/**************************************************************************
PROCESSING OF THE LEFT CHANNEL
***************************************************************************/
@@ -193,85 +190,3 @@
}
#endif
-#else
-void PK_2I_D32F32C14G11_TRC_WRA_01 ( Biquad_Instance_t *pInstance,
- LVM_INT32 *pDataIn,
- LVM_INT32 *pDataOut,
- LVM_INT16 NrSamples)
- {
- LVM_INT32 ynL,ynR,ynLO,ynRO,templ;
- LVM_INT16 ii;
- PFilter_State pBiquadState = (PFilter_State) pInstance;
-
- for (ii = NrSamples; ii != 0; ii--)
- {
-
-
- /**************************************************************************
- PROCESSING OF THE LEFT CHANNEL
- ***************************************************************************/
- /* ynL= (A0 (Q14) * (x(n)L (Q0) - x(n-2)L (Q0) ) >>14) in Q0*/
- templ=(*pDataIn)-pBiquadState->pDelays[2];
- MUL32x16INTO32(templ,pBiquadState->coefs[0],ynL,14)
-
- /* ynL+= ((-B2 (Q14) * y(n-2)L (Q0) ) >>14) in Q0*/
- MUL32x16INTO32(pBiquadState->pDelays[6],pBiquadState->coefs[1],templ,14)
- ynL+=templ;
-
- /* ynL+= ((-B1 (Q14) * y(n-1)L (Q0) ) >>14) in Q0 */
- MUL32x16INTO32(pBiquadState->pDelays[4],pBiquadState->coefs[2],templ,14)
- ynL+=templ;
-
- /* ynLO= ((Gain (Q11) * ynL (Q0))>>11) in Q0*/
- MUL32x16INTO32(ynL,pBiquadState->coefs[3],ynLO,11)
-
- /* ynLO=( ynLO(Q0) + x(n)L (Q0) ) in Q0*/
- ynLO+= (*pDataIn);
-
- /**************************************************************************
- PROCESSING OF THE RIGHT CHANNEL
- ***************************************************************************/
- /* ynR= (A0 (Q14) * (x(n)R (Q0) - x(n-2)R (Q0) ) >>14) in Q0*/
- templ=(*(pDataIn+1))-pBiquadState->pDelays[3];
- MUL32x16INTO32(templ,pBiquadState->coefs[0],ynR,14)
-
- /* ynR+= ((-B2 (Q14) * y(n-2)R (Q0) ) >>14) in Q0*/
- MUL32x16INTO32(pBiquadState->pDelays[7],pBiquadState->coefs[1],templ,14)
- ynR+=templ;
-
- /* ynR+= ((-B1 (Q14) * y(n-1)R (Q0) ) >>14) in Q0 */
- MUL32x16INTO32(pBiquadState->pDelays[5],pBiquadState->coefs[2],templ,14)
- ynR+=templ;
-
- /* ynRO= ((Gain (Q11) * ynR (Q0))>>11) in Q0*/
- MUL32x16INTO32(ynR,pBiquadState->coefs[3],ynRO,11)
-
- /* ynRO=( ynRO(Q0) + x(n)R (Q0) ) in Q0*/
- ynRO+= (*(pDataIn+1));
-
- /**************************************************************************
- UPDATING THE DELAYS
- ***************************************************************************/
- pBiquadState->pDelays[7]=pBiquadState->pDelays[5]; /* y(n-2)R=y(n-1)R*/
- pBiquadState->pDelays[6]=pBiquadState->pDelays[4]; /* y(n-2)L=y(n-1)L*/
- pBiquadState->pDelays[3]=pBiquadState->pDelays[1]; /* x(n-2)R=x(n-1)R*/
- pBiquadState->pDelays[2]=pBiquadState->pDelays[0]; /* x(n-2)L=x(n-1)L*/
- pBiquadState->pDelays[5]=ynR; /* Update y(n-1)R in Q0*/
- pBiquadState->pDelays[4]=ynL; /* Update y(n-1)L in Q0*/
- pBiquadState->pDelays[0]=(*pDataIn); /* Update x(n-1)L in Q0*/
- pDataIn++;
- pBiquadState->pDelays[1]=(*pDataIn); /* Update x(n-1)R in Q0*/
- pDataIn++;
-
- /**************************************************************************
- WRITING THE OUTPUT
- ***************************************************************************/
- *pDataOut=ynLO; /* Write Left output in Q0*/
- pDataOut++;
- *pDataOut=ynRO; /* Write Right ouput in Q0*/
- pDataOut++;
-
- }
-
- }
-#endif
diff --git a/media/libeffects/lvm/lib/Common/src/PK_2I_D32F32C30G11_TRC_WRA_01.c b/media/libeffects/lvm/lib/Common/src/PK_2I_D32F32C30G11_TRC_WRA_01.c
deleted file mode 100644
index f705cbf..0000000
--- a/media/libeffects/lvm/lib/Common/src/PK_2I_D32F32C30G11_TRC_WRA_01.c
+++ /dev/null
@@ -1,120 +0,0 @@
-/*
- * Copyright (C) 2004-2010 NXP Software
- * Copyright (C) 2010 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include "BIQUAD.h"
-#include "PK_2I_D32F32CllGss_TRC_WRA_01_Private.h"
-#include "LVM_Macros.h"
-
-/**************************************************************************
- ASSUMPTIONS:
- COEFS-
- pBiquadState->coefs[0] is A0,
- pBiquadState->coefs[1] is -B2,
- pBiquadState->coefs[2] is -B1, these are in Q30 format
- pBiquadState->coefs[3] is Gain, in Q11 format
-
-
- DELAYS-
- pBiquadState->pDelays[0] is x(n-1)L in Q0 format
- pBiquadState->pDelays[1] is x(n-1)R in Q0 format
- pBiquadState->pDelays[2] is x(n-2)L in Q0 format
- pBiquadState->pDelays[3] is x(n-2)R in Q0 format
- pBiquadState->pDelays[4] is y(n-1)L in Q0 format
- pBiquadState->pDelays[5] is y(n-1)R in Q0 format
- pBiquadState->pDelays[6] is y(n-2)L in Q0 format
- pBiquadState->pDelays[7] is y(n-2)R in Q0 format
-***************************************************************************/
-#ifndef BUILD_FLOAT
-void PK_2I_D32F32C30G11_TRC_WRA_01 ( Biquad_Instance_t *pInstance,
- LVM_INT32 *pDataIn,
- LVM_INT32 *pDataOut,
- LVM_INT16 NrSamples)
- {
- LVM_INT32 ynL,ynR,ynLO,ynRO,templ;
- LVM_INT16 ii;
- PFilter_State pBiquadState = (PFilter_State) pInstance;
-
- for (ii = NrSamples; ii != 0; ii--)
- {
-
-
- /**************************************************************************
- PROCESSING OF THE LEFT CHANNEL
- ***************************************************************************/
- /* ynL= (A0 (Q30) * (x(n)L (Q0) - x(n-2)L (Q0) ) >>30) in Q0*/
- templ=(*pDataIn)-pBiquadState->pDelays[2];
- MUL32x32INTO32(templ,pBiquadState->coefs[0],ynL,30)
-
- /* ynL+= ((-B2 (Q30) * y(n-2)L (Q0) ) >>30) in Q0*/
- MUL32x32INTO32(pBiquadState->pDelays[6],pBiquadState->coefs[1],templ,30)
- ynL+=templ;
-
- /* ynL+= ((-B1 (Q30) * y(n-1)L (Q0) ) >>30) in Q0 */
- MUL32x32INTO32(pBiquadState->pDelays[4],pBiquadState->coefs[2],templ,30)
- ynL+=templ;
-
- /* ynLO= ((Gain (Q11) * ynL (Q0))>>11) in Q0*/
- MUL32x16INTO32(ynL,pBiquadState->coefs[3],ynLO,11)
- /* ynLO=( ynLO(Q0) + x(n)L (Q0) ) in Q0*/
- ynLO+= (*pDataIn);
-
- /**************************************************************************
- PROCESSING OF THE RIGHT CHANNEL
- ***************************************************************************/
- /* ynR= (A0 (Q30) * (x(n)R (Q0) - x(n-2)R (Q0) ) >>30) in Q0*/
- templ=(*(pDataIn+1))-pBiquadState->pDelays[3];
- MUL32x32INTO32(templ,pBiquadState->coefs[0],ynR,30)
-
- /* ynR+= ((-B2 (Q30) * y(n-2)R (Q0) ) >>30) in Q0*/
- MUL32x32INTO32(pBiquadState->pDelays[7],pBiquadState->coefs[1],templ,30)
- ynR+=templ;
-
- /* ynR+= ((-B1 (Q30) * y(n-1)R (Q0) ) >>30) in Q0 */
- MUL32x32INTO32(pBiquadState->pDelays[5],pBiquadState->coefs[2],templ,30)
- ynR+=templ;
-
- /* ynRO= ((Gain (Q11) * ynR (Q0))>>11) in Q0*/
- MUL32x16INTO32(ynR,pBiquadState->coefs[3],ynRO,11)
-
- /* ynRO=( ynRO(Q0) + x(n)R (Q0) ) in Q0*/
- ynRO+= (*(pDataIn+1));
-
- /**************************************************************************
- UPDATING THE DELAYS
- ***************************************************************************/
- pBiquadState->pDelays[7]=pBiquadState->pDelays[5]; /* y(n-2)R=y(n-1)R*/
- pBiquadState->pDelays[6]=pBiquadState->pDelays[4]; /* y(n-2)L=y(n-1)L*/
- pBiquadState->pDelays[3]=pBiquadState->pDelays[1]; /* x(n-2)R=x(n-1)R*/
- pBiquadState->pDelays[2]=pBiquadState->pDelays[0]; /* x(n-2)L=x(n-1)L*/
- pBiquadState->pDelays[5]=ynR; /* Update y(n-1)R in Q0*/
- pBiquadState->pDelays[4]=ynL; /* Update y(n-1)L in Q0*/
- pBiquadState->pDelays[0]=(*pDataIn); /* Update x(n-1)L in Q0*/
- pDataIn++;
- pBiquadState->pDelays[1]=(*pDataIn); /* Update x(n-1)R in Q0*/
- pDataIn++;
-
- /**************************************************************************
- WRITING THE OUTPUT
- ***************************************************************************/
- *pDataOut=ynLO; /* Write Left output in Q0*/
- pDataOut++;
- *pDataOut=ynRO; /* Write Right ouput in Q0*/
- pDataOut++;
- }
-
- }
-#endif
diff --git a/media/libeffects/lvm/lib/Common/src/PK_2I_D32F32C30G11_TRC_WRA_01.cpp b/media/libeffects/lvm/lib/Common/src/PK_2I_D32F32C30G11_TRC_WRA_01.cpp
new file mode 100644
index 0000000..41de1de
--- /dev/null
+++ b/media/libeffects/lvm/lib/Common/src/PK_2I_D32F32C30G11_TRC_WRA_01.cpp
@@ -0,0 +1,39 @@
+/*
+ * Copyright (C) 2004-2010 NXP Software
+ * Copyright (C) 2010 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include "BIQUAD.h"
+#include "PK_2I_D32F32CllGss_TRC_WRA_01_Private.h"
+#include "LVM_Macros.h"
+
+/**************************************************************************
+ ASSUMPTIONS:
+ COEFS-
+ pBiquadState->coefs[0] is A0,
+ pBiquadState->coefs[1] is -B2,
+ pBiquadState->coefs[2] is -B1, these are in Q30 format
+ pBiquadState->coefs[3] is Gain, in Q11 format
+
+ DELAYS-
+ pBiquadState->pDelays[0] is x(n-1)L in Q0 format
+ pBiquadState->pDelays[1] is x(n-1)R in Q0 format
+ pBiquadState->pDelays[2] is x(n-2)L in Q0 format
+ pBiquadState->pDelays[3] is x(n-2)R in Q0 format
+ pBiquadState->pDelays[4] is y(n-1)L in Q0 format
+ pBiquadState->pDelays[5] is y(n-1)R in Q0 format
+ pBiquadState->pDelays[6] is y(n-2)L in Q0 format
+ pBiquadState->pDelays[7] is y(n-2)R in Q0 format
+***************************************************************************/
diff --git a/media/libeffects/lvm/lib/Common/src/PK_2I_D32F32CllGss_TRC_WRA_01_Init.c b/media/libeffects/lvm/lib/Common/src/PK_2I_D32F32CllGss_TRC_WRA_01_Init.c
deleted file mode 100644
index 65475a3..0000000
--- a/media/libeffects/lvm/lib/Common/src/PK_2I_D32F32CllGss_TRC_WRA_01_Init.c
+++ /dev/null
@@ -1,38 +0,0 @@
-/*
- * Copyright (C) 2004-2010 NXP Software
- * Copyright (C) 2010 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include "BIQUAD.h"
-#include "PK_2I_D32F32CllGss_TRC_WRA_01_Private.h"
-
-#ifndef BUILD_FLOAT
-void PK_2I_D32F32CllGss_TRC_WRA_01_Init(Biquad_Instance_t *pInstance,
- Biquad_2I_Order2_Taps_t *pTaps,
- PK_C32_Coefs_t *pCoef)
-{
- PFilter_State pBiquadState = (PFilter_State) pInstance;
- pBiquadState->pDelays =(LVM_INT32 *) pTaps;
-
- pBiquadState->coefs[0]=pCoef->A0;
-
- pBiquadState->coefs[1]=pCoef->B2;
-
- pBiquadState->coefs[2]=pCoef->B1;
-
- pBiquadState->coefs[3]=pCoef->G;
-
-}
-#endif
diff --git a/media/libstagefright/include/media/stagefright/NdkUtils.h b/media/libeffects/lvm/lib/Common/src/PK_2I_D32F32CllGss_TRC_WRA_01_Init.cpp
similarity index 62%
copy from media/libstagefright/include/media/stagefright/NdkUtils.h
copy to media/libeffects/lvm/lib/Common/src/PK_2I_D32F32CllGss_TRC_WRA_01_Init.cpp
index a68884a..714aa52 100644
--- a/media/libstagefright/include/media/stagefright/NdkUtils.h
+++ b/media/libeffects/lvm/lib/Common/src/PK_2I_D32F32CllGss_TRC_WRA_01_Init.cpp
@@ -1,5 +1,6 @@
/*
- * Copyright (C) 2018 The Android Open Source Project
+ * Copyright (C) 2004-2010 NXP Software
+ * Copyright (C) 2010 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
@@ -14,18 +15,6 @@
* limitations under the License.
*/
-#ifndef NDK_UTILS_H_
+#include "BIQUAD.h"
+#include "PK_2I_D32F32CllGss_TRC_WRA_01_Private.h"
-#define NDK_UTILS_H_
-
-#include <media/stagefright/MetaData.h>
-#include <media/NdkWrapper.h>
-
-namespace android {
-
-sp<MetaData> convertMediaFormatWrapperToMetaData(
- const sp<AMediaFormatWrapper> &fmt);
-
-} // namespace android
-
-#endif // NDK_UTILS_H_
diff --git a/media/libeffects/lvm/lib/Common/src/PK_2I_D32F32CssGss_TRC_WRA_01_Init.c b/media/libeffects/lvm/lib/Common/src/PK_2I_D32F32CssGss_TRC_WRA_01_Init.cpp
similarity index 70%
rename from media/libeffects/lvm/lib/Common/src/PK_2I_D32F32CssGss_TRC_WRA_01_Init.c
rename to media/libeffects/lvm/lib/Common/src/PK_2I_D32F32CssGss_TRC_WRA_01_Init.cpp
index a36330e..f6c05da 100644
--- a/media/libeffects/lvm/lib/Common/src/PK_2I_D32F32CssGss_TRC_WRA_01_Init.c
+++ b/media/libeffects/lvm/lib/Common/src/PK_2I_D32F32CssGss_TRC_WRA_01_Init.cpp
@@ -17,7 +17,6 @@
#include "BIQUAD.h"
#include "PK_2I_D32F32CssGss_TRC_WRA_01_Private.h"
-#ifdef BUILD_FLOAT
void PK_2I_D32F32CssGss_TRC_WRA_01_Init(Biquad_FLOAT_Instance_t *pInstance,
Biquad_2I_Order2_FLOAT_Taps_t *pTaps,
PK_FLOAT_Coefs_t *pCoef)
@@ -33,21 +32,3 @@
pBiquadState->coefs[3] = pCoef->G;
}
-#else
-void PK_2I_D32F32CssGss_TRC_WRA_01_Init(Biquad_Instance_t *pInstance,
- Biquad_2I_Order2_Taps_t *pTaps,
- PK_C16_Coefs_t *pCoef)
-{
- PFilter_State pBiquadState = (PFilter_State) pInstance;
- pBiquadState->pDelays =(LVM_INT32 *) pTaps;
-
- pBiquadState->coefs[0]=pCoef->A0;
-
- pBiquadState->coefs[1]=pCoef->B2;
-
- pBiquadState->coefs[2]=pCoef->B1;
-
- pBiquadState->coefs[3]=pCoef->G;
-
-}
-#endif
diff --git a/media/libeffects/lvm/lib/Common/src/PK_2I_D32F32CssGss_TRC_WRA_01_Private.h b/media/libeffects/lvm/lib/Common/src/PK_2I_D32F32CssGss_TRC_WRA_01_Private.h
index 1e32062..cc924c4 100644
--- a/media/libeffects/lvm/lib/Common/src/PK_2I_D32F32CssGss_TRC_WRA_01_Private.h
+++ b/media/libeffects/lvm/lib/Common/src/PK_2I_D32F32CssGss_TRC_WRA_01_Private.h
@@ -18,11 +18,9 @@
#ifndef _PK_2I_D32F32CSSGSS_TRC_WRA_01_PRIVATE_H_
#define _PK_2I_D32F32CSSGSS_TRC_WRA_01_PRIVATE_H_
-
/* The internal state variables are implemented in a (for the user) hidden structure */
/* In this (private) file, the internal structure is declared fro private use. */
-#ifdef BUILD_FLOAT
typedef struct _Filter_State_Float_
{
LVM_FLOAT * pDelays; /* pointer to the delayed samples (data of 32 bits) */
@@ -30,7 +28,6 @@
}Filter_State_Float;
typedef Filter_State_Float * PFilter_State_Float ;
-#endif
typedef struct _Filter_State_
{
LVM_INT32 * pDelays; /* pointer to the delayed samples (data of 32 bits) */
diff --git a/media/libeffects/lvm/lib/Common/src/LoadConst_32.c b/media/libeffects/lvm/lib/Common/src/Shift_Sat_v16xv16.cpp
similarity index 68%
copy from media/libeffects/lvm/lib/Common/src/LoadConst_32.c
copy to media/libeffects/lvm/lib/Common/src/Shift_Sat_v16xv16.cpp
index 9e14c3b..668a4b6 100644
--- a/media/libeffects/lvm/lib/Common/src/LoadConst_32.c
+++ b/media/libeffects/lvm/lib/Common/src/Shift_Sat_v16xv16.cpp
@@ -22,38 +22,6 @@
#include "VectorArithmetic.h"
/**********************************************************************************
- FUNCTION LoadConst_32
+ FUNCTION Shift_Sat_v16xv16
***********************************************************************************/
-#ifdef BUILD_FLOAT
-void LoadConst_Float(const LVM_FLOAT val,
- LVM_FLOAT *dst,
- LVM_INT16 n )
-{
- LVM_INT16 ii;
-
- for (ii = n; ii != 0; ii--)
- {
- *dst = val;
- dst++;
- }
-
- return;
-}
-#else
-void LoadConst_32(const LVM_INT32 val,
- LVM_INT32 *dst,
- LVM_INT16 n )
-{
- LVM_INT16 ii;
-
- for (ii = n; ii != 0; ii--)
- {
- *dst = val;
- dst++;
- }
-
- return;
-}
-#endif
-
/**********************************************************************************/
diff --git a/media/libeffects/lvm/lib/Common/src/Shift_Sat_v32xv32.c b/media/libeffects/lvm/lib/Common/src/Shift_Sat_v32xv32.c
deleted file mode 100644
index fac9de7..0000000
--- a/media/libeffects/lvm/lib/Common/src/Shift_Sat_v32xv32.c
+++ /dev/null
@@ -1,138 +0,0 @@
-/*
- * Copyright (C) 2004-2010 NXP Software
- * Copyright (C) 2010 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-/**********************************************************************************
- INCLUDE FILES
-***********************************************************************************/
-
-#include "VectorArithmetic.h"
-
-/**********************************************************************************
- FUNCTION Shift_Sat_v32xv32
-***********************************************************************************/
-#ifdef BUILD_FLOAT
-void Shift_Sat_Float (const LVM_INT16 val,
- const LVM_FLOAT *src,
- LVM_FLOAT *dst,
- LVM_INT16 n)
-{
- LVM_FLOAT temp;
- LVM_INT32 ii,ij;
- LVM_INT16 RShift;
-
- if(val > 0)
- {
- for (ii = n; ii != 0; ii--)
- {
- temp = (LVM_FLOAT)*src;
- src++;
- for(ij = 0; ij < val; ij++)
- {
- temp = temp * 2;
- }
-
- if(temp > 1.0)
- temp = 1.0;
- if(temp < -1.0)
- temp = -1.0;
-
- *dst = (LVM_FLOAT)temp;
- dst++;
- }
- }
- else if(val < 0)
- {
- RShift=(LVM_INT16)(-val);
-
- for (ii = n; ii != 0; ii--)
- {
- temp = (LVM_FLOAT)*src;
- src++;
- for(ij = 0; ij < RShift; ij++)
- {
- temp = temp / 2;
- }
- *dst = (LVM_FLOAT)temp;
- dst++;
- }
- }
- else
- {
- if(src != dst)
- {
- Copy_Float(src, dst, n);
- }
- }
- return;
-}
-#else
-void Shift_Sat_v32xv32 (const LVM_INT16 val,
- const LVM_INT32 *src,
- LVM_INT32 *dst,
- LVM_INT16 n)
-{
- LVM_INT32 ii;
- LVM_INT16 RShift;
-
- if(val>0)
- {
- LVM_INT32 a,b;
-
- for (ii = n; ii != 0; ii--)
- {
- a=*src;
- src++;
-
- b=(a<<val);
-
- if( (b>>val) != a ) /* if overflow occured, right shift will show difference*/
- {
- if(a<0)
- {
- b=0x80000000l;
- }
- else
- {
- b=0x7FFFFFFFl;
- }
- }
-
- *dst = b;
- dst++;
- }
- }
- else if(val<0)
- {
- RShift=(LVM_INT16)(-val);
- for (ii = n; ii != 0; ii--)
- {
- *dst = (*src >> RShift);
- dst++;
- src++;
- }
- }
- else
- {
- if(src!=dst)
- {
- Copy_16((LVM_INT16 *)src,(LVM_INT16 *)dst,(LVM_INT16)(n<<1));
- }
- }
- return;
-}
-#endif
-/**********************************************************************************/
diff --git a/media/libeffects/lvm/lib/Common/src/Shift_Sat_v16xv16.c b/media/libeffects/lvm/lib/Common/src/Shift_Sat_v32xv32.cpp
similarity index 66%
rename from media/libeffects/lvm/lib/Common/src/Shift_Sat_v16xv16.c
rename to media/libeffects/lvm/lib/Common/src/Shift_Sat_v32xv32.cpp
index 28fea65..97a04c1 100644
--- a/media/libeffects/lvm/lib/Common/src/Shift_Sat_v16xv16.c
+++ b/media/libeffects/lvm/lib/Common/src/Shift_Sat_v32xv32.cpp
@@ -22,60 +22,60 @@
#include "VectorArithmetic.h"
/**********************************************************************************
- FUNCTION Shift_Sat_v16xv16
+ FUNCTION Shift_Sat_v32xv32
***********************************************************************************/
-#ifndef BUILD_FLOAT
-void Shift_Sat_v16xv16 (const LVM_INT16 val,
- const LVM_INT16 *src,
- LVM_INT16 *dst,
- LVM_INT16 n)
+void Shift_Sat_Float (const LVM_INT16 val,
+ const LVM_FLOAT *src,
+ LVM_FLOAT *dst,
+ LVM_INT16 n)
{
- LVM_INT32 temp;
- LVM_INT32 ii;
+ LVM_FLOAT temp;
+ LVM_INT32 ii,ij;
LVM_INT16 RShift;
- if(val>0)
+
+ if(val > 0)
{
for (ii = n; ii != 0; ii--)
{
- temp = (LVM_INT32)*src;
+ temp = (LVM_FLOAT)*src;
src++;
+ for(ij = 0; ij < val; ij++)
+ {
+ temp = temp * 2;
+ }
- temp = temp << val;
+ if(temp > 1.0)
+ temp = 1.0;
+ if(temp < -1.0)
+ temp = -1.0;
- if (temp > 0x00007FFF)
- {
- *dst = 0x7FFF;
- }
- else if (temp < -0x00008000)
- {
- *dst = - 0x8000;
- }
- else
- {
- *dst = (LVM_INT16)temp;
- }
+ *dst = (LVM_FLOAT)temp;
dst++;
}
}
- else if(val<0)
+ else if(val < 0)
{
RShift=(LVM_INT16)(-val);
for (ii = n; ii != 0; ii--)
{
- *dst = (LVM_INT16)(*src >> RShift);
- dst++;
+ temp = (LVM_FLOAT)*src;
src++;
+ for(ij = 0; ij < RShift; ij++)
+ {
+ temp = temp / 2;
+ }
+ *dst = (LVM_FLOAT)temp;
+ dst++;
}
}
else
{
- if(src!=dst)
+ if(src != dst)
{
- Copy_16(src,dst,n);
+ Copy_Float(src, dst, n);
}
}
return;
}
-#endif
/**********************************************************************************/
diff --git a/media/libeffects/lvm/lib/Common/src/dB_to_Lin32.c b/media/libeffects/lvm/lib/Common/src/dB_to_Lin32.cpp
similarity index 71%
rename from media/libeffects/lvm/lib/Common/src/dB_to_Lin32.c
rename to media/libeffects/lvm/lib/Common/src/dB_to_Lin32.cpp
index 9a726f2..4da2013 100644
--- a/media/libeffects/lvm/lib/Common/src/dB_to_Lin32.c
+++ b/media/libeffects/lvm/lib/Common/src/dB_to_Lin32.cpp
@@ -29,10 +29,7 @@
/*######################################################################################*/
#include "ScalarArithmetic.h"
-#ifdef BUILD_FLOAT
#include <math.h>
-#endif
-
/****************************************************************************************
* Name : dB_to_Lin32()
@@ -67,7 +64,6 @@
#define SECOND_COEF 38836
#define MAX_VALUE 1536 /* 96 * 16 */
-#ifdef BUILD_FLOAT
LVM_FLOAT dB_to_LinFloat(LVM_INT16 db_fix)
{
LVM_FLOAT dB_Float;
@@ -78,47 +74,3 @@
return LinFloat;
}
-#else
-LVM_INT32 dB_to_Lin32(LVM_INT16 db_fix)
-{
- LVM_INT32 Lin_val_32;
- LVM_INT16 Shift;
- LVM_INT32 Remain;
-
-
- /*
- * Check sign of the input
- */
- if (db_fix<0)
- {
- if (db_fix > -MAX_VALUE)
- {
- Shift = (LVM_INT16)((((LVM_UINT32)(-db_fix) >> 4) * FOUR_OVER_SIX) >> 17); /* Number of 6dB steps in Q11.4 format */
- Remain = -db_fix - (Shift * SIX_DB);
- Remain = (0x7FFFFFFF - (Remain * FIRST_COEF_NEG)) + (Remain * Remain * SECOND_COEF);
- Lin_val_32 = (LVM_INT32)((LVM_UINT32)Remain >> (16 + Shift));
- }
- else
- {
- Lin_val_32 = 0;
- }
- }
- else
- {
- if (db_fix < MAX_VALUE)
- {
- Shift = (LVM_INT16)((((LVM_UINT32)db_fix >> 4) * FOUR_OVER_SIX) >> 17); /* Number of 6dB steps in Q11.4 format */
- Remain = db_fix - (Shift * SIX_DB);
- Remain = 0x3FFFFFFF + (Remain * FIRST_COEF_POS) + (Remain * Remain * SECOND_COEF);
- Lin_val_32 = (LVM_INT32)((LVM_UINT32)Remain >> (15 - Shift));
- }
- else
- {
- Lin_val_32 = 0x7FFFFFFF;
- }
- }
-
-
- return Lin_val_32; /* format 1.16.15 */
-}
-#endif
diff --git a/media/libeffects/lvm/lib/Common/src/mult3s_16x16.c b/media/libeffects/lvm/lib/Common/src/mult3s_16x16.cpp
similarity index 100%
rename from media/libeffects/lvm/lib/Common/src/mult3s_16x16.c
rename to media/libeffects/lvm/lib/Common/src/mult3s_16x16.cpp
diff --git a/media/libeffects/lvm/lib/Eq/lib/LVEQNB.h b/media/libeffects/lvm/lib/Eq/lib/LVEQNB.h
index 804f1bf..c5ddf77 100644
--- a/media/libeffects/lvm/lib/Eq/lib/LVEQNB.h
+++ b/media/libeffects/lvm/lib/Eq/lib/LVEQNB.h
@@ -68,15 +68,9 @@
/* */
/****************************************************************************************/
-
#ifndef __LVEQNB_H__
#define __LVEQNB_H__
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
-
/****************************************************************************************/
/* */
/* Includes */
@@ -112,7 +106,6 @@
/* Instance handle */
typedef void *LVEQNB_Handle_t;
-
/* Operating modes */
typedef enum
{
@@ -121,7 +114,6 @@
LVEQNB_MODE_MAX = LVM_MAXINT_32
} LVEQNB_Mode_en;
-
/* Filter mode control */
typedef enum
{
@@ -130,7 +122,6 @@
LVEQNB_FILTER_DUMMY = LVM_MAXINT_32
} LVEQNB_FilterMode_en;
-
/* Memory Types */
typedef enum
{
@@ -141,7 +132,6 @@
LVEQNB_MEMORY_MAX = LVM_MAXINT_32
} LVEQNB_MemoryTypes_en;
-
/* Function return status */
typedef enum
{
@@ -152,7 +142,6 @@
LVEQNB_STATUS_MAX = LVM_MAXINT_32
} LVEQNB_ReturnStatus_en;
-
/****************************************************************************************/
/* */
/* Linked enumerated type and capability definitions */
@@ -190,7 +179,6 @@
LVEQNB_SOURCE_MAX = LVM_MAXINT_32
} LVEQNB_SourceFormat_en;
-
/*
* Supported sample rates in samples per second
*/
@@ -203,12 +191,10 @@
#define LVEQNB_CAP_FS_32000 64
#define LVEQNB_CAP_FS_44100 128
#define LVEQNB_CAP_FS_48000 256
-#if defined(BUILD_FLOAT) && defined(HIGHER_FS)
#define LVEQNB_CAP_FS_88200 512
#define LVEQNB_CAP_FS_96000 1024
#define LVEQNB_CAP_FS_176400 2048
#define LVEQNB_CAP_FS_192000 4096
-#endif
typedef enum
{
@@ -221,16 +207,13 @@
LVEQNB_FS_32000 = 6,
LVEQNB_FS_44100 = 7,
LVEQNB_FS_48000 = 8,
-#ifdef HIGHER_FS
LVEQNB_FS_88200 = 9,
LVEQNB_FS_96000 = 10,
LVEQNB_FS_176400 = 11,
LVEQNB_FS_192000 = 12,
-#endif
LVEQNB_FS_MAX = LVM_MAXINT_32
} LVEQNB_Fs_en;
-
/****************************************************************************************/
/* */
/* Structures */
@@ -246,14 +229,12 @@
void *pBaseAddress; /* Pointer to the region base address */
} LVEQNB_MemoryRegion_t;
-
/* Memory table containing the region definitions */
typedef struct
{
LVEQNB_MemoryRegion_t Region[LVEQNB_NR_MEMORY_REGIONS]; /* One definition for each region */
} LVEQNB_MemTab_t;
-
/* Equaliser band definition */
typedef struct
{
@@ -262,7 +243,6 @@
LVM_UINT16 QFactor; /* Band quality factor */
} LVEQNB_BandDef_t;
-
/* Parameter structure */
typedef struct
{
@@ -279,7 +259,6 @@
#endif
} LVEQNB_Params_t;
-
/* Capability structure */
typedef struct
{
@@ -296,7 +275,6 @@
} LVEQNB_Capabilities_t;
-
/****************************************************************************************/
/* */
/* Function Prototypes */
@@ -339,7 +317,6 @@
LVEQNB_MemTab_t *pMemoryTable,
LVEQNB_Capabilities_t *pCapabilities);
-
/****************************************************************************************/
/* */
/* FUNCTION: LVEQNB_Init */
@@ -379,7 +356,6 @@
LVEQNB_MemTab_t *pMemoryTable,
LVEQNB_Capabilities_t *pCapabilities);
-
/****************************************************************************************/
/* */
/* FUNCTION: LVEQNB_GetParameters */
@@ -404,7 +380,6 @@
LVEQNB_ReturnStatus_en LVEQNB_GetParameters(LVEQNB_Handle_t hInstance,
LVEQNB_Params_t *pParams);
-
/****************************************************************************************/
/* */
/* FUNCTION: LVEQNB_GetCapabilities */
@@ -429,7 +404,6 @@
LVEQNB_ReturnStatus_en LVEQNB_GetCapabilities(LVEQNB_Handle_t hInstance,
LVEQNB_Capabilities_t *pCapabilities);
-
/****************************************************************************************/
/* */
/* FUNCTION: LVEQNB_Control */
@@ -455,7 +429,6 @@
LVEQNB_ReturnStatus_en LVEQNB_Control(LVEQNB_Handle_t hInstance,
LVEQNB_Params_t *pParams);
-
/****************************************************************************************/
/* */
/* FUNCTION: LVEQNB_Process */
@@ -478,23 +451,10 @@
/* NOTES: */
/* */
/****************************************************************************************/
-#ifdef BUILD_FLOAT
LVEQNB_ReturnStatus_en LVEQNB_Process(LVEQNB_Handle_t hInstance,
const LVM_FLOAT *pInData,
LVM_FLOAT *pOutData,
LVM_UINT16 NumSamples);
-#else
-LVEQNB_ReturnStatus_en LVEQNB_Process(LVEQNB_Handle_t hInstance,
- const LVM_INT16 *pInData,
- LVM_INT16 *pOutData,
- LVM_UINT16 NumSamples);
-#endif
-
-
-
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
#endif /* __LVEQNB__ */
diff --git a/media/libeffects/lvm/lib/Eq/src/LVEQNB_CalcCoef.c b/media/libeffects/lvm/lib/Eq/src/LVEQNB_CalcCoef.cpp
similarity index 62%
rename from media/libeffects/lvm/lib/Eq/src/LVEQNB_CalcCoef.c
rename to media/libeffects/lvm/lib/Eq/src/LVEQNB_CalcCoef.cpp
index ff52b7f..c3c0fad 100644
--- a/media/libeffects/lvm/lib/Eq/src/LVEQNB_CalcCoef.c
+++ b/media/libeffects/lvm/lib/Eq/src/LVEQNB_CalcCoef.cpp
@@ -22,9 +22,7 @@
/****************************************************************************************/
#include "LVEQNB_Private.h"
-#ifdef BUILD_FLOAT
#include <math.h>
-#endif
/****************************************************************************************/
/* */
@@ -78,101 +76,6 @@
/* */
/****************************************************************************************/
-
-#ifndef BUILD_FLOAT
-LVEQNB_ReturnStatus_en LVEQNB_DoublePrecCoefs(LVM_UINT16 Fs,
- LVEQNB_BandDef_t *pFilterDefinition,
- PK_C32_Coefs_t *pCoefficients)
-{
-
- extern LVM_INT16 LVEQNB_GainTable[];
- extern LVM_INT16 LVEQNB_TwoPiOnFsTable[];
- extern LVM_INT16 LVEQNB_DTable[];
- extern LVM_INT16 LVEQNB_DPCosCoef[];
-
- /*
- * Get the filter definition
- */
- LVM_INT16 Gain = pFilterDefinition->Gain;
- LVM_UINT16 Frequency = pFilterDefinition->Frequency;
- LVM_UINT16 QFactor = pFilterDefinition->QFactor;
-
- /*
- * Intermediate variables and temporary values
- */
- LVM_INT32 T0;
- LVM_INT16 D;
- LVM_INT32 A0;
- LVM_INT32 B1;
- LVM_INT32 B2;
- LVM_INT32 Dt0;
- LVM_INT32 B2_Den;
- LVM_INT32 B2_Num;
- LVM_INT32 CosErr;
- LVM_INT16 coef;
- LVM_INT32 factor;
- LVM_INT16 t0;
- LVM_INT16 i;
-
- /*
- * Calculating the intermediate values
- */
- T0 = (LVM_INT32)Frequency * LVEQNB_TwoPiOnFsTable[Fs]; /* T0 = 2 * Pi * Fc / Fs */
- if (Gain >= 0)
- {
- D = LVEQNB_DTable[15]; /* D = 1 if GaindB >= 0 */
- }
- else
- {
- D = LVEQNB_DTable[Gain+15]; /* D = 1 / (1 + G) if GaindB < 0 */
- }
-
- /*
- * Calculate the B2 coefficient
- */
- Dt0 = D * (T0 >> 10);
- B2_Den = ((LVM_INT32)QFactor << 19) + (Dt0 >> 2);
- B2_Num = (Dt0 >> 3) - ((LVM_INT32)QFactor << 18);
- B2 = (B2_Num / (B2_Den >> 16)) << 15;
-
- /*
- * Calculate the cosine error by a polynomial expansion using the equation:
- *
- * CosErr += coef(n) * t0^n For n = 0 to 4
- */
- T0 = (T0 >> 6) * 0x7f53; /* Scale to 1.0 in 16-bit for range 0 to fs/50 */
- t0 = (LVM_INT16)(T0 >> 16);
- factor = 0x7fff; /* Initialise to 1.0 for the a0 coefficient */
- CosErr = 0; /* Initialise the error to zero */
- for (i=1; i<5; i++)
- {
- coef = LVEQNB_DPCosCoef[i]; /* Get the nth coefficient */
- CosErr += (factor * coef) >> 5; /* The nth partial sum */
- factor = (factor * t0) >> 15; /* Calculate t0^n */
- }
- CosErr = CosErr << (LVEQNB_DPCosCoef[0]); /* Correct the scaling */
-
- /*
- * Calculate the B1 and A0 coefficients
- */
- B1 = (0x40000000 - B2); /* B1 = (0.5 - b2/2) */
- A0 = ((B1 >> 16) * (CosErr >> 10)) >> 6; /* Temporary storage for (0.5 - b2/2) * coserr(t0) */
- B1 -= A0; /* B1 = (0.5 - b2/2) * (1 - coserr(t0)) */
- A0 = (0x40000000 + B2) >> 1; /* A0 = (0.5 + b2) */
-
- /*
- * Write coeff into the data structure
- */
- pCoefficients->A0 = A0;
- pCoefficients->B1 = B1;
- pCoefficients->B2 = B2;
- pCoefficients->G = LVEQNB_GainTable[Gain+15];
-
- return(LVEQNB_SUCCESS);
-
-}
-#endif
-
/****************************************************************************************/
/* */
/* FUNCTION: LVEQNB_SinglePrecCoefs */
@@ -208,7 +111,6 @@
/* */
/****************************************************************************************/
-#ifdef BUILD_FLOAT
LVEQNB_ReturnStatus_en LVEQNB_SinglePrecCoefs(LVM_UINT16 Fs,
LVEQNB_BandDef_t *pFilterDefinition,
PK_FLOAT_Coefs_t *pCoefficients)
@@ -218,7 +120,6 @@
extern LVM_FLOAT LVEQNB_TwoPiOnFsTable[];
extern LVM_FLOAT LVEQNB_DTable[];
-
/*
* Get the filter definition
*/
@@ -227,7 +128,6 @@
/* As mentioned in effectbundle.h */
LVM_FLOAT QFactor = (LVM_FLOAT)pFilterDefinition->QFactor / 100.0f;
-
/*
* Intermediate variables and temporary values
*/
@@ -268,95 +168,3 @@
return(LVEQNB_SUCCESS);
}
-#else
-LVEQNB_ReturnStatus_en LVEQNB_SinglePrecCoefs(LVM_UINT16 Fs,
- LVEQNB_BandDef_t *pFilterDefinition,
- PK_C16_Coefs_t *pCoefficients)
-{
-
- extern LVM_INT16 LVEQNB_GainTable[];
- extern LVM_INT16 LVEQNB_TwoPiOnFsTable[];
- extern LVM_INT16 LVEQNB_DTable[];
- extern LVM_INT16 LVEQNB_CosCoef[];
-
-
- /*
- * Get the filter definition
- */
- LVM_INT16 Gain = pFilterDefinition->Gain;
- LVM_UINT16 Frequency = pFilterDefinition->Frequency;
- LVM_UINT16 QFactor = pFilterDefinition->QFactor;
-
-
- /*
- * Intermediate variables and temporary values
- */
- LVM_INT32 T0;
- LVM_INT16 D;
- LVM_INT32 A0;
- LVM_INT32 B1;
- LVM_INT32 B2;
- LVM_INT32 Dt0;
- LVM_INT32 B2_Den;
- LVM_INT32 B2_Num;
- LVM_INT32 COS_T0;
- LVM_INT16 coef;
- LVM_INT32 factor;
- LVM_INT16 t0;
- LVM_INT16 i;
-
- /*
- * Calculating the intermediate values
- */
- T0 = (LVM_INT32)Frequency * LVEQNB_TwoPiOnFsTable[Fs]; /* T0 = 2 * Pi * Fc / Fs */
- if (Gain >= 0)
- {
- D = LVEQNB_DTable[15]; /* D = 1 if GaindB >= 0 */
- }
- else
- {
- D = LVEQNB_DTable[Gain+15]; /* D = 1 / (1 + G) if GaindB < 0 */
- }
-
- /*
- * Calculate the B2 coefficient
- */
- Dt0 = D * (T0 >> 10);
- B2_Den = ((LVM_INT32)QFactor << 19) + (Dt0 >> 2);
- B2_Num = (Dt0 >> 3) - ((LVM_INT32)QFactor << 18);
- B2 = (B2_Num / (B2_Den >> 16)) << 15;
-
- /*
- * Calculate the cosine by a polynomial expansion using the equation:
- *
- * Cos += coef(n) * t0^n For n = 0 to 6
- */
- T0 = (T0 >> 10) * 20859; /* Scale to 1.0 in 16-bit for range 0 to fs/2 */
- t0 = (LVM_INT16)(T0 >> 16);
- factor = 0x7fff; /* Initialise to 1.0 for the a0 coefficient */
- COS_T0 = 0; /* Initialise the error to zero */
- for (i=1; i<7; i++)
- {
- coef = LVEQNB_CosCoef[i]; /* Get the nth coefficient */
- COS_T0 += (factor * coef) >> 5; /* The nth partial sum */
- factor = (factor * t0) >> 15; /* Calculate t0^n */
- }
- COS_T0 = COS_T0 << (LVEQNB_CosCoef[0]+6); /* Correct the scaling */
-
-
- B1 = ((0x40000000 - B2) >> 16) * (COS_T0 >> 16); /* B1 = (0.5 - b2/2) * cos(t0) */
- A0 = (0x40000000 + B2) >> 1; /* A0 = (0.5 + b2/2) */
-
- /*
- * Write coeff into the data structure
- */
- pCoefficients->A0 = (LVM_INT16)(A0>>16);
- pCoefficients->B1 = (LVM_INT16)(B1>>15);
- pCoefficients->B2 = (LVM_INT16)(B2>>16);
- pCoefficients->G = LVEQNB_GainTable[Gain+15];
-
-
- return(LVEQNB_SUCCESS);
-
-}
-#endif
\ No newline at end of file
diff --git a/media/libeffects/lvm/lib/Eq/src/LVEQNB_Coeffs.h b/media/libeffects/lvm/lib/Eq/src/LVEQNB_Coeffs.h
index 755141e..6329181 100644
--- a/media/libeffects/lvm/lib/Eq/src/LVEQNB_Coeffs.h
+++ b/media/libeffects/lvm/lib/Eq/src/LVEQNB_Coeffs.h
@@ -15,17 +15,14 @@
* limitations under the License.
*/
-
#ifndef __LVEQNB_COEFFS_H__
#define __LVEQNB_COEFFS_H__
-
/************************************************************************************/
/* */
/* Gain table for (10^(Gain/20) - 1) */
/* */
/************************************************************************************/
-#ifdef BUILD_FLOAT
#define LVEQNB_Gain_Neg15_dB (-0.822172f)
#define LVEQNB_Gain_Neg14_dB (-0.800474f)
#define LVEQNB_Gain_Neg13_dB (-0.776128f)
@@ -57,47 +54,12 @@
#define LVEQNB_Gain_13_dB 3.466836f
#define LVEQNB_Gain_14_dB 4.011872f
#define LVEQNB_Gain_15_dB 4.623413f
-#else
-#define LVEQNB_GAINSHIFT 11 /* As a power of 2 */
-#define LVEQNB_Gain_Neg15_dB (-1684) /* Floating point value -0.822172 */
-#define LVEQNB_Gain_Neg14_dB (-1639) /* Floating point value -0.800474 */
-#define LVEQNB_Gain_Neg13_dB (-1590) /* Floating point value -0.776128 */
-#define LVEQNB_Gain_Neg12_dB (-1534) /* Floating point value -0.748811 */
-#define LVEQNB_Gain_Neg11_dB (-1471) /* Floating point value -0.718162 */
-#define LVEQNB_Gain_Neg10_dB (-1400) /* Floating point value -0.683772 */
-#define LVEQNB_Gain_Neg9_dB (-1321) /* Floating point value -0.645187 */
-#define LVEQNB_Gain_Neg8_dB (-1233) /* Floating point value -0.601893 */
-#define LVEQNB_Gain_Neg7_dB (-1133) /* Floating point value -0.553316 */
-#define LVEQNB_Gain_Neg6_dB (-1022) /* Floating point value -0.498813 */
-#define LVEQNB_Gain_Neg5_dB (-896) /* Floating point value -0.437659 */
-#define LVEQNB_Gain_Neg4_dB (-756) /* Floating point value -0.369043 */
-#define LVEQNB_Gain_Neg3_dB (-598) /* Floating point value -0.292054 */
-#define LVEQNB_Gain_Neg2_dB (-421) /* Floating point value -0.205672 */
-#define LVEQNB_Gain_Neg1_dB (-223) /* Floating point value -0.108749 */
-#define LVEQNB_Gain_0_dB 0 /* Floating point value 0.000000 */
-#define LVEQNB_Gain_1_dB 250 /* Floating point value 0.122018 */
-#define LVEQNB_Gain_2_dB 530 /* Floating point value 0.258925 */
-#define LVEQNB_Gain_3_dB 845 /* Floating point value 0.412538 */
-#define LVEQNB_Gain_4_dB 1198 /* Floating point value 0.584893 */
-#define LVEQNB_Gain_5_dB 1594 /* Floating point value 0.778279 */
-#define LVEQNB_Gain_6_dB 2038 /* Floating point value 0.995262 */
-#define LVEQNB_Gain_7_dB 2537 /* Floating point value 1.238721 */
-#define LVEQNB_Gain_8_dB 3096 /* Floating point value 1.511886 */
-#define LVEQNB_Gain_9_dB 3724 /* Floating point value 1.818383 */
-#define LVEQNB_Gain_10_dB 4428 /* Floating point value 2.162278 */
-#define LVEQNB_Gain_11_dB 5219 /* Floating point value 2.548134 */
-#define LVEQNB_Gain_12_dB 6105 /* Floating point value 2.981072 */
-#define LVEQNB_Gain_13_dB 7100 /* Floating point value 3.466836 */
-#define LVEQNB_Gain_14_dB 8216 /* Floating point value 4.011872 */
-#define LVEQNB_Gain_15_dB 9469 /* Floating point value 4.623413 */
-#endif
/************************************************************************************/
/* */
/* Frequency table for 2*Pi/Fs */
/* */
/************************************************************************************/
-#ifdef BUILD_FLOAT
#define LVEQNB_2PiOn_8000 0.000785f
#define LVEQNB_2PiOn_11025 0.000570f
#define LVEQNB_2PiOn_12000 0.000524f
@@ -108,32 +70,16 @@
#define LVEQNB_2PiOn_44100 0.000142f
#define LVEQNB_2PiOn_48000 0.000131f
-#ifdef HIGHER_FS
#define LVEQNB_2PiOn_88200 0.000071f
#define LVEQNB_2PiOn_96000 0.000065f
#define LVEQNB_2PiOn_176400 0.000036f
#define LVEQNB_2PiOn_192000 0.000033f
-#endif
-
-#else
-#define LVEQNB_FREQSHIFT 25 /* As a power of 2 */
-#define LVEQNB_2PiOn_8000 26354 /* Floating point value 0.000785 */
-#define LVEQNB_2PiOn_11025 19123 /* Floating point value 0.000570 */
-#define LVEQNB_2PiOn_12000 17569 /* Floating point value 0.000524 */
-#define LVEQNB_2PiOn_16000 13177 /* Floating point value 0.000393 */
-#define LVEQNB_2PiOn_22050 9561 /* Floating point value 0.000285 */
-#define LVEQNB_2PiOn_24000 8785 /* Floating point value 0.000262 */
-#define LVEQNB_2PiOn_32000 6588 /* Floating point value 0.000196 */
-#define LVEQNB_2PiOn_44100 4781 /* Floating point value 0.000142 */
-#define LVEQNB_2PiOn_48000 4392 /* Floating point value 0.000131 */
-#endif
/************************************************************************************/
/* */
/* 50D table for 50 / ( 1 + Gain ) */
/* */
/************************************************************************************/
-#ifdef BUILD_FLOAT
#define LVEQNB_100D_Neg15_dB 5.623413f
#define LVEQNB_100D_Neg14_dB 5.011872f
#define LVEQNB_100D_Neg13_dB 4.466836f
@@ -150,24 +96,5 @@
#define LVEQNB_100D_Neg2_dB 1.258925f
#define LVEQNB_100D_Neg1_dB 1.122018f
#define LVEQNB_100D_0_dB 1.000000f
-#else
-#define LVEQNB_100DSHIFT 5 /* As a power of 2 */
-#define LVEQNB_100D_Neg15_dB 17995 /* Floating point value 5.623413 */
-#define LVEQNB_100D_Neg14_dB 16038 /* Floating point value 5.011872 */
-#define LVEQNB_100D_Neg13_dB 14294 /* Floating point value 4.466836 */
-#define LVEQNB_100D_Neg12_dB 12739 /* Floating point value 3.981072 */
-#define LVEQNB_100D_Neg11_dB 11354 /* Floating point value 3.548134 */
-#define LVEQNB_100D_Neg10_dB 10119 /* Floating point value 3.162278 */
-#define LVEQNB_100D_Neg9_dB 9019 /* Floating point value 2.818383 */
-#define LVEQNB_100D_Neg8_dB 8038 /* Floating point value 2.511886 */
-#define LVEQNB_100D_Neg7_dB 7164 /* Floating point value 2.238721 */
-#define LVEQNB_100D_Neg6_dB 6385 /* Floating point value 1.995262 */
-#define LVEQNB_100D_Neg5_dB 5690 /* Floating point value 1.778279 */
-#define LVEQNB_100D_Neg4_dB 5072 /* Floating point value 1.584893 */
-#define LVEQNB_100D_Neg3_dB 4520 /* Floating point value 1.412538 */
-#define LVEQNB_100D_Neg2_dB 4029 /* Floating point value 1.258925 */
-#define LVEQNB_100D_Neg1_dB 3590 /* Floating point value 1.122018 */
-#define LVEQNB_100D_0_dB 3200 /* Floating point value 1.000000 */
-#endif
#endif
diff --git a/media/libeffects/lvm/lib/Eq/src/LVEQNB_Control.c b/media/libeffects/lvm/lib/Eq/src/LVEQNB_Control.cpp
similarity index 85%
rename from media/libeffects/lvm/lib/Eq/src/LVEQNB_Control.c
rename to media/libeffects/lvm/lib/Eq/src/LVEQNB_Control.cpp
index 7b0f341..6bb4a7e 100644
--- a/media/libeffects/lvm/lib/Eq/src/LVEQNB_Control.c
+++ b/media/libeffects/lvm/lib/Eq/src/LVEQNB_Control.cpp
@@ -26,7 +26,6 @@
#include "VectorArithmetic.h"
#include "BIQUAD.h"
-
/****************************************************************************************/
/* */
/* Defines */
@@ -76,7 +75,6 @@
return(LVEQNB_SUCCESS);
}
-
/************************************************************************************/
/* */
/* FUNCTION: LVEQNB_GetCapabilities */
@@ -114,7 +112,6 @@
return(LVEQNB_SUCCESS);
}
-
/************************************************************************************/
/* */
/* FUNCTION: LVEQNB_SetFilters */
@@ -140,18 +137,13 @@
void LVEQNB_SetFilters(LVEQNB_Instance_t *pInstance,
LVEQNB_Params_t *pParams)
{
-#ifdef HIGHER_FS
extern const LVM_UINT32 LVEQNB_SampleRateTab[]; /* Sample rate table */
-#else
- extern const LVM_UINT16 LVEQNB_SampleRateTab[]; /* Sample rate table */
-#endif
LVM_UINT16 i; /* Filter band index */
LVM_UINT32 fs = (LVM_UINT32)LVEQNB_SampleRateTab[(LVM_UINT16)pParams->SampleRate]; /* Sample rate */
LVM_UINT32 fc; /* Filter centre frequency */
LVM_INT16 QFactor; /* Filter Q factor */
-
pInstance->NBands = pParams->NBands;
for (i=0; i<pParams->NBands; i++)
@@ -162,30 +154,7 @@
fc = (LVM_UINT32)pParams->pBandDefinition[i].Frequency; /* Get the band centre frequency */
QFactor = (LVM_INT16)pParams->pBandDefinition[i].QFactor; /* Get the band Q factor */
-#ifdef BUILD_FLOAT
pInstance->pBiquadType[i] = LVEQNB_SinglePrecision_Float; /* Default to single precision */
-#else
- /*
- * For each filter set the type of biquad required
- */
- pInstance->pBiquadType[i] = LVEQNB_SinglePrecision; /* Default to single precision */
-#endif
-#ifndef BUILD_FLOAT
- if ((fc << 15) <= (LOW_FREQ * fs))
- {
- /*
- * fc <= fs/110
- */
- pInstance->pBiquadType[i] = LVEQNB_DoublePrecision;
- }
- else if (((fc << 15) <= (HIGH_FREQ * fs)) && (QFactor > 300))
- {
- /*
- * (fs/110 < fc < fs/85) & (Q>3)
- */
- pInstance->pBiquadType[i] = LVEQNB_DoublePrecision;
- }
-#endif
/*
* Check for out of range frequencies
@@ -195,7 +164,6 @@
pInstance->pBiquadType[i] = LVEQNB_OutOfRange;
}
-
/*
* Copy the filter definition to persistant memory
*/
@@ -204,7 +172,6 @@
}
}
-
/************************************************************************************/
/* */
/* FUNCTION: LVEQNB_SetCoefficients */
@@ -225,7 +192,6 @@
LVM_UINT16 i; /* Filter band index */
LVEQNB_BiquadType_en BiquadType; /* Filter biquad type */
-
/*
* Set the coefficients for each band by the init function
*/
@@ -238,7 +204,6 @@
BiquadType = pInstance->pBiquadType[i];
switch (BiquadType)
{
-#ifdef BUILD_FLOAT
case LVEQNB_SinglePrecision_Float:
{
PK_FLOAT_Coefs_t Coefficients;
@@ -256,47 +221,6 @@
&Coefficients);
break;
}
-#else
- case LVEQNB_DoublePrecision:
- {
- PK_C32_Coefs_t Coefficients;
-
- /*
- * Calculate the double precision coefficients
- */
- LVEQNB_DoublePrecCoefs((LVM_UINT16)pInstance->Params.SampleRate,
- &pInstance->pBandDefinitions[i],
- &Coefficients);
-
- /*
- * Set the coefficients
- */
- PK_2I_D32F32CllGss_TRC_WRA_01_Init(&pInstance->pEQNB_FilterState[i],
- &pInstance->pEQNB_Taps[i],
- &Coefficients);
- break;
- }
-
- case LVEQNB_SinglePrecision:
- {
- PK_C16_Coefs_t Coefficients;
-
- /*
- * Calculate the single precision coefficients
- */
- LVEQNB_SinglePrecCoefs((LVM_UINT16)pInstance->Params.SampleRate,
- &pInstance->pBandDefinitions[i],
- &Coefficients);
-
- /*
- * Set the coefficients
- */
- PK_2I_D32F32CssGss_TRC_WRA_01_Init(&pInstance->pEQNB_FilterState[i],
- &pInstance->pEQNB_Taps[i],
- &Coefficients);
- break;
- }
-#endif
default:
break;
}
@@ -304,7 +228,6 @@
}
-
/************************************************************************************/
/* */
/* FUNCTION: LVEQNB_ClearFilterHistory */
@@ -316,24 +239,6 @@
/* pInstance Pointer to the instance */
/* */
/************************************************************************************/
-#ifndef BUILD_FLOAT
-void LVEQNB_ClearFilterHistory(LVEQNB_Instance_t *pInstance)
-{
- LVM_INT16 *pTapAddress;
- LVM_INT16 NumTaps;
-
-
- pTapAddress = (LVM_INT16 *)pInstance->pEQNB_Taps;
- NumTaps = (LVM_INT16)((pInstance->Capabilities.MaxBands * sizeof(Biquad_2I_Order2_Taps_t))/sizeof(LVM_INT16));
-
- if (NumTaps != 0)
- {
- LoadConst_16(0, /* Clear the history, value 0 */
- pTapAddress, /* Destination */
- NumTaps); /* Number of words */
- }
-}
-#else
void LVEQNB_ClearFilterHistory(LVEQNB_Instance_t *pInstance)
{
LVM_FLOAT *pTapAddress;
@@ -350,7 +255,6 @@
NumTaps); /* Number of words */
}
}
-#endif
/****************************************************************************************/
/* */
/* FUNCTION: LVEQNB_Control */
@@ -404,7 +308,6 @@
LVC_Mixer_VarSlope_SetTimeConstant(&pInstance->BypassMixer.MixerStream[1],LVEQNB_BYPASS_MIXER_TC,(LVM_Fs_en)pParams->SampleRate,2);
}
-
if( (pInstance->Params.NBands != pParams->NBands ) ||
(pInstance->Params.OperatingMode != pParams->OperatingMode ) ||
(pInstance->Params.pBandDefinition != pParams->pBandDefinition ) ||
@@ -429,7 +332,6 @@
}
}
-
// During operating mode transition, there is a race condition where the mode
// is still LVEQNB_ON, but the effect is considered disabled in the upper layers.
// modeChange handles this special race condition.
@@ -453,7 +355,6 @@
*/
pInstance->Params = *pParams;
-
/*
* Reset the filters except if the algo is switched off
*/
@@ -473,13 +374,8 @@
if (modeChange) {
if(pParams->OperatingMode == LVEQNB_ON)
{
-#ifdef BUILD_FLOAT
LVC_Mixer_SetTarget(&pInstance->BypassMixer.MixerStream[0], 1.0f);
LVC_Mixer_SetTarget(&pInstance->BypassMixer.MixerStream[1], 0.0f);
-#else
- LVC_Mixer_SetTarget(&pInstance->BypassMixer.MixerStream[0],LVM_MAXINT_16);
- LVC_Mixer_SetTarget(&pInstance->BypassMixer.MixerStream[1],0);
-#endif
pInstance->BypassMixer.MixerStream[0].CallbackSet = 1;
pInstance->BypassMixer.MixerStream[1].CallbackSet = 1;
}
@@ -489,13 +385,8 @@
// This may introduce a state race condition if the effect is enabled again
// while in transition. This is fixed in the modeChange logic.
pInstance->Params.OperatingMode = LVEQNB_ON;
-#ifdef BUILD_FLOAT
LVC_Mixer_SetTarget(&pInstance->BypassMixer.MixerStream[0], 0.0f);
LVC_Mixer_SetTarget(&pInstance->BypassMixer.MixerStream[1], 1.0f);
-#else
- LVC_Mixer_SetTarget(&pInstance->BypassMixer.MixerStream[0],0);
- LVC_Mixer_SetTarget(&pInstance->BypassMixer.MixerStream[1],LVM_MAXINT_16);
-#endif
pInstance->BypassMixer.MixerStream[0].CallbackSet = 1;
pInstance->BypassMixer.MixerStream[1].CallbackSet = 1;
}
@@ -508,7 +399,6 @@
return(LVEQNB_SUCCESS);
}
-
/****************************************************************************************/
/* */
/* FUNCTION: LVEQNB_BypassMixerCallBack */
@@ -530,13 +420,8 @@
/*
* Send an ALGOFF event if the ON->OFF switch transition is finished
*/
-#ifdef BUILD_FLOAT
if((LVC_Mixer_GetTarget(&pInstance->BypassMixer.MixerStream[0]) == 0) &&
(CallbackParam == 0)){
-#else
- if((LVC_Mixer_GetTarget(&pInstance->BypassMixer.MixerStream[0]) == 0x00000000) &&
- (CallbackParam == 0)){
-#endif
pInstance->Params.OperatingMode = LVEQNB_BYPASS;
if (CallBack != LVM_NULL){
CallBack(pInstance->Capabilities.pBundleInstance, LVM_NULL, ALGORITHM_EQNB_ID|LVEQNB_EVENT_ALGOFF);
diff --git a/media/libeffects/lvm/lib/Eq/src/LVEQNB_Init.c b/media/libeffects/lvm/lib/Eq/src/LVEQNB_Init.cpp
similarity index 85%
rename from media/libeffects/lvm/lib/Eq/src/LVEQNB_Init.c
rename to media/libeffects/lvm/lib/Eq/src/LVEQNB_Init.cpp
index de1bbb7..271a914 100644
--- a/media/libeffects/lvm/lib/Eq/src/LVEQNB_Init.c
+++ b/media/libeffects/lvm/lib/Eq/src/LVEQNB_Init.cpp
@@ -15,7 +15,6 @@
* limitations under the License.
*/
-
/****************************************************************************************/
/* */
/* Includes */
@@ -67,13 +66,11 @@
INST_ALLOC AllocMem;
LVEQNB_Instance_t *pInstance = (LVEQNB_Instance_t *)hInstance;
-
if((pMemoryTable == LVM_NULL)|| (pCapabilities == LVM_NULL))
{
return LVEQNB_NULLADDRESS;
}
-
/*
* Fill in the memory table
*/
@@ -91,13 +88,11 @@
pMemoryTable->Region[LVEQNB_MEMREGION_INSTANCE].Type = LVEQNB_PERSISTENT;
pMemoryTable->Region[LVEQNB_MEMREGION_INSTANCE].pBaseAddress = LVM_NULL;
-
/*
* Persistant data memory
*/
InstAlloc_Init(&AllocMem,
LVM_NULL);
-#ifdef BUILD_FLOAT
InstAlloc_AddMember(&AllocMem, /* Low pass filter */
sizeof(Biquad_2I_Order2_FLOAT_Taps_t));
InstAlloc_AddMember(&AllocMem, /* High pass filter */
@@ -111,18 +106,6 @@
/* Biquad types */
InstAlloc_AddMember(&AllocMem,
(pCapabilities->MaxBands * sizeof(LVEQNB_BiquadType_en)));
-#else
- InstAlloc_AddMember(&AllocMem, /* Low pass filter */
- sizeof(Biquad_2I_Order2_Taps_t));
- InstAlloc_AddMember(&AllocMem, /* High pass filter */
- sizeof(Biquad_2I_Order2_Taps_t));
- InstAlloc_AddMember(&AllocMem,
- (pCapabilities->MaxBands * sizeof(Biquad_2I_Order2_Taps_t))); /* Equaliser Biquad Taps */
- InstAlloc_AddMember(&AllocMem,
- (pCapabilities->MaxBands * sizeof(LVEQNB_BandDef_t))); /* Filter definitions */
- InstAlloc_AddMember(&AllocMem,
- (pCapabilities->MaxBands * sizeof(LVEQNB_BiquadType_en))); /* Biquad types */
-#endif
pMemoryTable->Region[LVEQNB_MEMREGION_PERSISTENT_DATA].Size = InstAlloc_GetTotal(&AllocMem);
pMemoryTable->Region[LVEQNB_MEMREGION_PERSISTENT_DATA].Alignment = LVEQNB_DATA_ALIGN;
pMemoryTable->Region[LVEQNB_MEMREGION_PERSISTENT_DATA].Type = LVEQNB_PERSISTENT_DATA;
@@ -133,7 +116,6 @@
*/
InstAlloc_Init(&AllocMem,
LVM_NULL);
-#ifdef BUILD_FLOAT
InstAlloc_AddMember(&AllocMem, /* Low pass filter */
sizeof(Biquad_FLOAT_Instance_t));
InstAlloc_AddMember(&AllocMem, /* High pass filter */
@@ -141,14 +123,6 @@
/* Equaliser Biquad Instance */
InstAlloc_AddMember(&AllocMem,
pCapabilities->MaxBands * sizeof(Biquad_FLOAT_Instance_t));
-#else
- InstAlloc_AddMember(&AllocMem, /* Low pass filter */
- sizeof(Biquad_Instance_t));
- InstAlloc_AddMember(&AllocMem, /* High pass filter */
- sizeof(Biquad_Instance_t));
- InstAlloc_AddMember(&AllocMem,
- pCapabilities->MaxBands * sizeof(Biquad_Instance_t)); /* Equaliser Biquad Instance */
-#endif
pMemoryTable->Region[LVEQNB_MEMREGION_PERSISTENT_COEF].Size = InstAlloc_GetTotal(&AllocMem);
pMemoryTable->Region[LVEQNB_MEMREGION_PERSISTENT_COEF].Alignment = LVEQNB_COEF_ALIGN;
pMemoryTable->Region[LVEQNB_MEMREGION_PERSISTENT_COEF].Type = LVEQNB_PERSISTENT_COEF;
@@ -159,14 +133,9 @@
*/
InstAlloc_Init(&AllocMem,
LVM_NULL);
-#ifdef BUILD_FLOAT
InstAlloc_AddMember(&AllocMem, /* Low pass filter */
LVEQNB_SCRATCHBUFFERS * sizeof(LVM_FLOAT) * \
pCapabilities->MaxBlockSize);
-#else
- InstAlloc_AddMember(&AllocMem, /* Low pass filter */
- LVEQNB_SCRATCHBUFFERS*sizeof(LVM_INT16)*pCapabilities->MaxBlockSize);
-#endif
pMemoryTable->Region[LVEQNB_MEMREGION_SCRATCH].Size = InstAlloc_GetTotal(&AllocMem);
pMemoryTable->Region[LVEQNB_MEMREGION_SCRATCH].Alignment = LVEQNB_SCRATCH_ALIGN;
pMemoryTable->Region[LVEQNB_MEMREGION_SCRATCH].Type = LVEQNB_SCRATCH;
@@ -181,7 +150,6 @@
return(LVEQNB_SUCCESS);
}
-
/****************************************************************************************/
/* */
/* FUNCTION: LVEQNB_Init */
@@ -260,14 +228,11 @@
}
pInstance =(LVEQNB_Instance_t *)*phInstance;
-
-
/*
* Save the memory table in the instance structure
*/
pInstance->Capabilities = *pCapabilities;
-
/*
* Save the memory table in the instance structure and
* set the structure pointers
@@ -280,17 +245,10 @@
InstAlloc_Init(&AllocMem,
pMemoryTable->Region[LVEQNB_MEMREGION_PERSISTENT_COEF].pBaseAddress);
-#ifdef BUILD_FLOAT
/* Equaliser Biquad Instance */
- pInstance->pEQNB_FilterState_Float = InstAlloc_AddMember(&AllocMem,
- pCapabilities->MaxBands * \
- sizeof(Biquad_FLOAT_Instance_t));
-#else
- pInstance->pEQNB_FilterState = InstAlloc_AddMember(&AllocMem,
- pCapabilities->MaxBands * sizeof(Biquad_Instance_t)); /* Equaliser Biquad Instance */
-#endif
-
-
+ pInstance->pEQNB_FilterState_Float = (Biquad_FLOAT_Instance_t *)
+ InstAlloc_AddMember(&AllocMem, pCapabilities->MaxBands * \
+ sizeof(Biquad_FLOAT_Instance_t));
/*
* Allocate data memory
@@ -298,15 +256,9 @@
InstAlloc_Init(&AllocMem,
pMemoryTable->Region[LVEQNB_MEMREGION_PERSISTENT_DATA].pBaseAddress);
-#ifdef BUILD_FLOAT
MemSize = (pCapabilities->MaxBands * sizeof(Biquad_2I_Order2_FLOAT_Taps_t));
pInstance->pEQNB_Taps_Float = (Biquad_2I_Order2_FLOAT_Taps_t *)InstAlloc_AddMember(&AllocMem,
MemSize);
-#else
- MemSize = (pCapabilities->MaxBands * sizeof(Biquad_2I_Order2_Taps_t));
- pInstance->pEQNB_Taps = (Biquad_2I_Order2_Taps_t *)InstAlloc_AddMember(&AllocMem,
- MemSize);
-#endif
MemSize = (pCapabilities->MaxBands * sizeof(LVEQNB_BandDef_t));
pInstance->pBandDefinitions = (LVEQNB_BandDef_t *)InstAlloc_AddMember(&AllocMem,
MemSize);
@@ -317,20 +269,14 @@
pInstance->pBiquadType = (LVEQNB_BiquadType_en *)InstAlloc_AddMember(&AllocMem,
MemSize);
-
/*
* Internally map, structure and allign scratch memory
*/
InstAlloc_Init(&AllocMem,
pMemoryTable->Region[LVEQNB_MEMREGION_SCRATCH].pBaseAddress);
-#ifdef BUILD_FLOAT
pInstance->pFastTemporary = (LVM_FLOAT *)InstAlloc_AddMember(&AllocMem,
sizeof(LVM_FLOAT));
-#else
- pInstance->pFastTemporary = (LVM_INT16 *)InstAlloc_AddMember(&AllocMem,
- sizeof(LVM_INT16));
-#endif
/*
* Update the instance parameters
@@ -362,18 +308,12 @@
LVC_Mixer_Init(&pInstance->BypassMixer.MixerStream[0],0,0);
LVC_Mixer_SetTimeConstant(&pInstance->BypassMixer.MixerStream[0],0,LVM_FS_8000,2);
-
pInstance->BypassMixer.MixerStream[1].CallbackSet = 1;
pInstance->BypassMixer.MixerStream[1].CallbackParam = 0;
pInstance->BypassMixer.MixerStream[1].pCallbackHandle = LVM_NULL;
pInstance->BypassMixer.MixerStream[1].pCallBack = LVM_NULL;
-#ifdef BUILD_FLOAT
LVC_Mixer_Init(&pInstance->BypassMixer.MixerStream[1], 0, 1.0f);
LVC_Mixer_SetTimeConstant(&pInstance->BypassMixer.MixerStream[1], 0, LVM_FS_8000, 2);
-#else
- LVC_Mixer_Init(&pInstance->BypassMixer.MixerStream[1],0,LVM_MAXINT_16);
- LVC_Mixer_SetTimeConstant(&pInstance->BypassMixer.MixerStream[1],0,LVM_FS_8000,2);
-#endif
pInstance->bInOperatingModeTransition = LVM_FALSE;
diff --git a/media/libeffects/lvm/lib/Eq/src/LVEQNB_Private.h b/media/libeffects/lvm/lib/Eq/src/LVEQNB_Private.h
index a9cd5fd..40facfb 100644
--- a/media/libeffects/lvm/lib/Eq/src/LVEQNB_Private.h
+++ b/media/libeffects/lvm/lib/Eq/src/LVEQNB_Private.h
@@ -18,11 +18,6 @@
#ifndef __LVEQNB_PRIVATE_H__
#define __LVEQNB_PRIVATE_H__
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
-
/****************************************************************************************/
/* */
/* Includes */
@@ -65,24 +60,19 @@
/* Filter biquad types */
typedef enum
{
-#ifdef BUILD_FLOAT
LVEQNB_SinglePrecision_Float = -1,
-#endif
LVEQNB_SinglePrecision = 0,
LVEQNB_DoublePrecision = 1,
LVEQNB_OutOfRange = 2,
LVEQNB_BIQUADTYPE_MAX = LVM_MAXINT_32
} LVEQNB_BiquadType_en;
-
/****************************************************************************************/
/* */
/* Structures */
/* */
/****************************************************************************************/
-
-
/* Instance structure */
typedef struct
{
@@ -92,20 +82,10 @@
LVEQNB_Capabilities_t Capabilities; /* Instance capabilities */
/* Aligned memory pointers */
-#ifdef BUILD_FLOAT
LVM_FLOAT *pFastTemporary; /* Fast temporary data base address */
-#else
- LVM_INT16 *pFastTemporary; /* Fast temporary data base address */
-#endif
-#ifdef BUILD_FLOAT
Biquad_2I_Order2_FLOAT_Taps_t *pEQNB_Taps_Float; /* Equaliser Taps */
Biquad_FLOAT_Instance_t *pEQNB_FilterState_Float; /* State for each filter band */
-#else
- /* Process variables */
- Biquad_2I_Order2_Taps_t *pEQNB_Taps; /* Equaliser Taps */
- Biquad_Instance_t *pEQNB_FilterState; /* State for each filter band */
-#endif
/* Filter definitions and call back */
LVM_UINT16 NBands; /* Number of bands */
@@ -113,17 +93,12 @@
LVEQNB_BiquadType_en *pBiquadType; /* Filter biquad types */
/* Bypass variable */
-#ifdef BUILD_FLOAT
LVMixer3_2St_FLOAT_st BypassMixer;
-#else
- LVMixer3_2St_st BypassMixer; /* Bypass mixer used in transitions */
-#endif
LVM_INT16 bInOperatingModeTransition; /* Operating mode transition flag */
} LVEQNB_Instance_t;
-
/****************************************************************************************/
/* */
/* Function prototypes */
@@ -136,25 +111,11 @@
void LVEQNB_SetCoefficients(LVEQNB_Instance_t *pInstance);
void LVEQNB_ClearFilterHistory(LVEQNB_Instance_t *pInstance);
-#ifdef BUILD_FLOAT
LVEQNB_ReturnStatus_en LVEQNB_SinglePrecCoefs(LVM_UINT16 Fs,
LVEQNB_BandDef_t *pFilterDefinition,
PK_FLOAT_Coefs_t *pCoefficients);
-#else
-LVEQNB_ReturnStatus_en LVEQNB_SinglePrecCoefs(LVM_UINT16 Fs,
- LVEQNB_BandDef_t *pFilterDefinition,
- PK_C16_Coefs_t *pCoefficients);
-
-LVEQNB_ReturnStatus_en LVEQNB_DoublePrecCoefs(LVM_UINT16 Fs,
- LVEQNB_BandDef_t *pFilterDefinition,
- PK_C32_Coefs_t *pCoefficients);
-#endif
LVM_INT32 LVEQNB_BypassMixerCallBack (void* hInstance, void *pGeneralPurpose, LVM_INT16 CallbackParam);
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
-
#endif /* __LVEQNB_PRIVATE_H__ */
diff --git a/media/libeffects/lvm/lib/Eq/src/LVEQNB_Process.c b/media/libeffects/lvm/lib/Eq/src/LVEQNB_Process.cpp
similarity index 61%
rename from media/libeffects/lvm/lib/Eq/src/LVEQNB_Process.c
rename to media/libeffects/lvm/lib/Eq/src/LVEQNB_Process.cpp
index d188c0e..65eff53 100644
--- a/media/libeffects/lvm/lib/Eq/src/LVEQNB_Process.c
+++ b/media/libeffects/lvm/lib/Eq/src/LVEQNB_Process.cpp
@@ -58,7 +58,6 @@
/* NOTES: */
/* */
/****************************************************************************************/
-#ifdef BUILD_FLOAT
LVEQNB_ReturnStatus_en LVEQNB_Process(LVEQNB_Handle_t hInstance,
const LVM_FLOAT *pInData,
LVM_FLOAT *pOutData,
@@ -123,7 +122,6 @@
*/
Biquad_FLOAT_Instance_t *pBiquad = &pInstance->pEQNB_FilterState_Float[i];
-
/*
* Select single or double precision as required
*/
@@ -152,7 +150,6 @@
}
}
-
if(pInstance->bInOperatingModeTransition == LVM_TRUE){
#ifdef SUPPORT_MC
LVC_MixSoft_2Mc_D16C31_SAT(&pInstance->BypassMixer,
@@ -194,145 +191,3 @@
return LVEQNB_SUCCESS;
}
-#else
-LVEQNB_ReturnStatus_en LVEQNB_Process(LVEQNB_Handle_t hInstance,
- const LVM_INT16 *pInData,
- LVM_INT16 *pOutData,
- LVM_UINT16 NumSamples)
-{
-
- LVM_UINT16 i;
- Biquad_Instance_t *pBiquad;
- LVEQNB_Instance_t *pInstance = (LVEQNB_Instance_t *)hInstance;
- LVM_INT32 *pScratch;
-
-
- /* Check for NULL pointers */
- if((hInstance == LVM_NULL) || (pInData == LVM_NULL) || (pOutData == LVM_NULL))
- {
- return LVEQNB_NULLADDRESS;
- }
-
- /* Check if the input and output data buffers are 32-bit aligned */
- if ((((uintptr_t)pInData % 4) != 0) || (((uintptr_t)pOutData % 4) != 0))
- {
- return LVEQNB_ALIGNMENTERROR;
- }
-
- pScratch = (LVM_INT32 *)pInstance->pFastTemporary;
-
- /*
- * Check the number of samples is not too large
- */
- if (NumSamples > pInstance->Capabilities.MaxBlockSize)
- {
- return(LVEQNB_TOOMANYSAMPLES);
- }
-
- if (pInstance->Params.OperatingMode == LVEQNB_ON)
- {
- /*
- * Convert from 16-bit to 32-bit
- */
- Int16LShiftToInt32_16x32((LVM_INT16 *)pInData, /* Source */
- pScratch, /* Destination */
- (LVM_INT16)(2*NumSamples), /* Left and Right */
- SHIFT); /* Scaling shift */
-
- /*
- * For each section execte the filter unless the gain is 0dB
- */
- if (pInstance->NBands != 0)
- {
- for (i=0; i<pInstance->NBands; i++)
- {
- /*
- * Check if band is non-zero dB gain
- */
- if (pInstance->pBandDefinitions[i].Gain != 0)
- {
- /*
- * Get the address of the biquad instance
- */
- pBiquad = &pInstance->pEQNB_FilterState[i];
-
-
- /*
- * Select single or double precision as required
- */
- switch (pInstance->pBiquadType[i])
- {
- case LVEQNB_SinglePrecision:
- {
- PK_2I_D32F32C14G11_TRC_WRA_01(pBiquad,
- (LVM_INT32 *)pScratch,
- (LVM_INT32 *)pScratch,
- (LVM_INT16)NumSamples);
- break;
- }
-
- case LVEQNB_DoublePrecision:
- {
- PK_2I_D32F32C30G11_TRC_WRA_01(pBiquad,
- (LVM_INT32 *)pScratch,
- (LVM_INT32 *)pScratch,
- (LVM_INT16)NumSamples);
- break;
- }
- default:
- break;
- }
- }
- }
- }
-
-
- if(pInstance->bInOperatingModeTransition == LVM_TRUE){
- /*
- * Convert from 32-bit to 16- bit and saturate
- */
- Int32RShiftToInt16_Sat_32x16(pScratch, /* Source */
- (LVM_INT16 *)pScratch, /* Destination */
- (LVM_INT16)(2*NumSamples), /* Left and Right */
- SHIFT); /* Scaling shift */
-
- LVC_MixSoft_2St_D16C31_SAT(&pInstance->BypassMixer,
- (LVM_INT16 *)pScratch,
- (LVM_INT16 *)pInData,
- (LVM_INT16 *)pScratch,
- (LVM_INT16)(2*NumSamples));
-
- Copy_16((LVM_INT16*)pScratch, /* Source */
- pOutData, /* Destination */
- (LVM_INT16)(2*NumSamples)); /* Left and Right samples */
- }
- else{
-
- /*
- * Convert from 32-bit to 16- bit and saturate
- */
- Int32RShiftToInt16_Sat_32x16(pScratch, /* Source */
- pOutData, /* Destination */
- (LVM_INT16 )(2*NumSamples), /* Left and Right */
- SHIFT); /* Scaling shift */
- }
- }
- else
- {
- /*
- * Mode is OFF so copy the data if necessary
- */
- if (pInData != pOutData)
- {
- Copy_16(pInData, /* Source */
- pOutData, /* Destination */
- (LVM_INT16)(2*NumSamples)); /* Left and Right samples */
- }
- }
-
-
-
- return(LVEQNB_SUCCESS);
-
-}
-#endif
diff --git a/media/libeffects/lvm/lib/Eq/src/LVEQNB_Tables.c b/media/libeffects/lvm/lib/Eq/src/LVEQNB_Tables.cpp
similarity index 67%
rename from media/libeffects/lvm/lib/Eq/src/LVEQNB_Tables.c
rename to media/libeffects/lvm/lib/Eq/src/LVEQNB_Tables.cpp
index 453c42d..0628114 100644
--- a/media/libeffects/lvm/lib/Eq/src/LVEQNB_Tables.c
+++ b/media/libeffects/lvm/lib/Eq/src/LVEQNB_Tables.cpp
@@ -15,7 +15,6 @@
* limitations under the License.
*/
-
/************************************************************************************/
/* */
/* Includes */
@@ -24,7 +23,7 @@
#include "LVEQNB.h"
#include "LVEQNB_Coeffs.h"
-
+#include "LVEQNB_Tables.h"
/************************************************************************************/
/* */
@@ -36,7 +35,6 @@
* Sample rate table for converting between the enumerated type and the actual
* frequency
*/
-#if defined(BUILD_FLOAT) && defined(HIGHER_FS)
const LVM_UINT32 LVEQNB_SampleRateTab[] = {8000, /* 8kS/s */
11025,
12000,
@@ -51,18 +49,6 @@
176400,
192000
};
-#else
-const LVM_UINT16 LVEQNB_SampleRateTab[] = {8000, /* 8kS/s */
- 11025,
- 12000,
- 16000,
- 22050,
- 24000,
- 32000,
- 44100,
- 48000
-};
-#endif
/************************************************************************************/
/* */
@@ -73,7 +59,6 @@
/*
* Table for 2 * Pi / Fs
*/
-#ifdef BUILD_FLOAT
const LVM_FLOAT LVEQNB_TwoPiOnFsTable[] = {LVEQNB_2PiOn_8000, /* 8kS/s */
LVEQNB_2PiOn_11025,
LVEQNB_2PiOn_12000,
@@ -83,29 +68,15 @@
LVEQNB_2PiOn_32000,
LVEQNB_2PiOn_44100,
LVEQNB_2PiOn_48000
-#ifdef HIGHER_FS
,LVEQNB_2PiOn_88200
,LVEQNB_2PiOn_96000
,LVEQNB_2PiOn_176400
,LVEQNB_2PiOn_192000
-#endif
};
-#else
-const LVM_INT16 LVEQNB_TwoPiOnFsTable[] = {LVEQNB_2PiOn_8000, /* 8kS/s */
- LVEQNB_2PiOn_11025,
- LVEQNB_2PiOn_12000,
- LVEQNB_2PiOn_16000,
- LVEQNB_2PiOn_22050,
- LVEQNB_2PiOn_24000,
- LVEQNB_2PiOn_32000,
- LVEQNB_2PiOn_44100,
- LVEQNB_2PiOn_48000}; /* 48kS/s */
-#endif
/*
* Gain table
*/
-#ifdef BUILD_FLOAT
const LVM_FLOAT LVEQNB_GainTable[] = {LVEQNB_Gain_Neg15_dB, /* -15dB gain */
LVEQNB_Gain_Neg14_dB,
LVEQNB_Gain_Neg13_dB,
@@ -137,44 +108,9 @@
LVEQNB_Gain_13_dB,
LVEQNB_Gain_14_dB,
LVEQNB_Gain_15_dB}; /* +15dB gain */
-#else
-const LVM_INT16 LVEQNB_GainTable[] = {LVEQNB_Gain_Neg15_dB, /* -15dB gain */
- LVEQNB_Gain_Neg14_dB,
- LVEQNB_Gain_Neg13_dB,
- LVEQNB_Gain_Neg12_dB,
- LVEQNB_Gain_Neg11_dB,
- LVEQNB_Gain_Neg10_dB,
- LVEQNB_Gain_Neg9_dB,
- LVEQNB_Gain_Neg8_dB,
- LVEQNB_Gain_Neg7_dB,
- LVEQNB_Gain_Neg6_dB,
- LVEQNB_Gain_Neg5_dB,
- LVEQNB_Gain_Neg4_dB,
- LVEQNB_Gain_Neg3_dB,
- LVEQNB_Gain_Neg2_dB,
- LVEQNB_Gain_Neg1_dB,
- LVEQNB_Gain_0_dB, /* 0dB gain */
- LVEQNB_Gain_1_dB,
- LVEQNB_Gain_2_dB,
- LVEQNB_Gain_3_dB,
- LVEQNB_Gain_4_dB,
- LVEQNB_Gain_5_dB,
- LVEQNB_Gain_6_dB,
- LVEQNB_Gain_7_dB,
- LVEQNB_Gain_8_dB,
- LVEQNB_Gain_9_dB,
- LVEQNB_Gain_10_dB,
- LVEQNB_Gain_11_dB,
- LVEQNB_Gain_12_dB,
- LVEQNB_Gain_13_dB,
- LVEQNB_Gain_14_dB,
- LVEQNB_Gain_15_dB}; /* +15dB gain */
-
-#endif
/*
* D table for 100 / (Gain + 1)
*/
-#ifdef BUILD_FLOAT
const LVM_FLOAT LVEQNB_DTable[] = {LVEQNB_100D_Neg15_dB, /* -15dB gain */
LVEQNB_100D_Neg14_dB,
LVEQNB_100D_Neg13_dB,
@@ -191,25 +127,6 @@
LVEQNB_100D_Neg2_dB,
LVEQNB_100D_Neg1_dB,
LVEQNB_100D_0_dB}; /* 0dB gain */
-#else
-const LVM_INT16 LVEQNB_DTable[] = {LVEQNB_100D_Neg15_dB, /* -15dB gain */
- LVEQNB_100D_Neg14_dB,
- LVEQNB_100D_Neg13_dB,
- LVEQNB_100D_Neg12_dB,
- LVEQNB_100D_Neg11_dB,
- LVEQNB_100D_Neg10_dB,
- LVEQNB_100D_Neg9_dB,
- LVEQNB_100D_Neg8_dB,
- LVEQNB_100D_Neg7_dB,
- LVEQNB_100D_Neg6_dB,
- LVEQNB_100D_Neg5_dB,
- LVEQNB_100D_Neg4_dB,
- LVEQNB_100D_Neg3_dB,
- LVEQNB_100D_Neg2_dB,
- LVEQNB_100D_Neg1_dB,
- LVEQNB_100D_0_dB}; /* 0dB gain */
-
-#endif
/************************************************************************************/
/* */
/* Filter polynomial coefficients */
@@ -253,4 +170,3 @@
16586, /* a2 */
-44}; /* a3 */
-
diff --git a/media/libeffects/lvm/lib/Eq/src/LVEQNB_Tables.h b/media/libeffects/lvm/lib/Eq/src/LVEQNB_Tables.h
new file mode 100644
index 0000000..a71eeb9
--- /dev/null
+++ b/media/libeffects/lvm/lib/Eq/src/LVEQNB_Tables.h
@@ -0,0 +1,86 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __LVEQNB_TABLES_H__
+#define __LVEQNB_TABLES_H__
+
+/************************************************************************************/
+/* */
+/* Sample rate table */
+/* */
+/************************************************************************************/
+
+/*
+ * Sample rate table for converting between the enumerated type and the actual
+ * frequency
+ */
+extern const LVM_UINT32 LVEQNB_SampleRateTab[];
+
+/************************************************************************************/
+/* */
+/* Coefficient calculation tables */
+/* */
+/************************************************************************************/
+
+/*
+ * Table for 2 * Pi / Fs
+ */
+extern const LVM_FLOAT LVEQNB_TwoPiOnFsTable[];
+
+/*
+ * Gain table
+ */
+extern const LVM_FLOAT LVEQNB_GainTable[];
+
+/*
+ * D table for 100 / (Gain + 1)
+ */
+extern const LVM_FLOAT LVEQNB_DTable[];
+
+/************************************************************************************/
+/* */
+/* Filter polynomial coefficients */
+/* */
+/************************************************************************************/
+
+/*
+ * Coefficients for calculating the cosine with the equation:
+ *
+ * Cos(x) = (2^Shifts)*(a0 + a1*x + a2*x^2 + a3*x^3 + a4*x^4 + a5*x^5)
+ *
+ * These coefficients expect the input, x, to be in the range 0 to 32768 respresenting
+ * a range of 0 to Pi. The output is in the range 32767 to -32768 representing the range
+ * +1.0 to -1.0
+ */
+extern const LVM_INT16 LVEQNB_CosCoef[];
+
+/*
+ * Coefficients for calculating the cosine error with the equation:
+ *
+ * CosErr(x) = (2^Shifts)*(a0 + a1*x + a2*x^2 + a3*x^3)
+ *
+ * These coefficients expect the input, x, to be in the range 0 to 32768 respresenting
+ * a range of 0 to Pi/25. The output is in the range 0 to 32767 representing the range
+ * 0.0 to 0.0078852986
+ *
+ * This is used to give a double precision cosine over the range 0 to Pi/25 using the
+ * the equation:
+ *
+ * Cos(x) = 1.0 - CosErr(x)
+ */
+extern const LVM_INT16 LVEQNB_DPCosCoef[];
+
+#endif /* __LVEQNB_TABLES_H__ */
diff --git a/media/libeffects/lvm/lib/Reverb/lib/LVREV.h b/media/libeffects/lvm/lib/Reverb/lib/LVREV.h
index 9c2e297..8c91ea9 100644
--- a/media/libeffects/lvm/lib/Reverb/lib/LVREV.h
+++ b/media/libeffects/lvm/lib/Reverb/lib/LVREV.h
@@ -28,11 +28,6 @@
#ifndef __LVREV_H__
#define __LVREV_H__
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
-
/****************************************************************************************/
/* */
/* Includes */
@@ -40,7 +35,6 @@
/****************************************************************************************/
#include "LVM_Types.h"
-
/****************************************************************************************/
/* */
/* Definitions */
@@ -53,7 +47,6 @@
/* Memory table*/
#define LVREV_NR_MEMORY_REGIONS 4 /* Number of memory regions */
-
/****************************************************************************************/
/* */
/* Types */
@@ -62,7 +55,6 @@
/* Instance handle */
typedef void *LVREV_Handle_t;
-
/* Status return values */
typedef enum
{
@@ -73,7 +65,6 @@
LVREV_RETURNSTATUS_DUMMY = LVM_MAXENUM
} LVREV_ReturnStatus_en;
-
/* Reverb delay lines */
typedef enum
{
@@ -83,7 +74,6 @@
LVREV_DELAYLINES_DUMMY = LVM_MAXENUM
} LVREV_NumDelayLines_en;
-
/****************************************************************************************/
/* */
/* Structures */
@@ -96,7 +86,6 @@
LVM_MemoryRegion_st Region[LVREV_NR_MEMORY_REGIONS]; /* One definition for each region */
} LVREV_MemoryTable_st;
-
/* Control Parameter structure */
typedef struct
{
@@ -107,13 +96,8 @@
/* Parameters for REV */
LVM_UINT16 Level; /* Level, 0 to 100 representing percentage of reverb */
-#ifndef HIGHER_FS
- LVM_UINT16 LPF; /* Low pass filter, in Hz */
- LVM_UINT16 HPF; /* High pass filter, in Hz */
-#else
LVM_UINT32 LPF; /* Low pass filter, in Hz */
LVM_UINT32 HPF; /* High pass filter, in Hz */
-#endif
LVM_UINT16 T60; /* Decay time constant, in ms */
LVM_UINT16 Density; /* Echo density, 0 to 100 for minimum to maximum density */
@@ -122,7 +106,6 @@
} LVREV_ControlParams_st;
-
/* Instance Parameter structure */
typedef struct
{
@@ -135,7 +118,6 @@
} LVREV_InstanceParams_st;
-
/****************************************************************************************/
/* */
/* Function Prototypes */
@@ -182,7 +164,6 @@
LVREV_MemoryTable_st *pMemoryTable,
LVREV_InstanceParams_st *pInstanceParams);
-
/****************************************************************************************/
/* */
/* FUNCTION: LVREV_GetInstanceHandle */
@@ -213,7 +194,6 @@
LVREV_MemoryTable_st *pMemoryTable,
LVREV_InstanceParams_st *pInstanceParams);
-
/****************************************************************************************/
/* */
/* FUNCTION: LVXX_GetControlParameters */
@@ -237,7 +217,6 @@
LVREV_ReturnStatus_en LVREV_GetControlParameters(LVREV_Handle_t hInstance,
LVREV_ControlParams_st *pControlParams);
-
/****************************************************************************************/
/* */
/* FUNCTION: LVREV_SetControlParameters */
@@ -260,7 +239,6 @@
LVREV_ReturnStatus_en LVREV_SetControlParameters(LVREV_Handle_t hInstance,
LVREV_ControlParams_st *pNewParams);
-
/****************************************************************************************/
/* */
/* FUNCTION: LVREV_ClearAudioBuffers */
@@ -281,7 +259,6 @@
/****************************************************************************************/
LVREV_ReturnStatus_en LVREV_ClearAudioBuffers(LVREV_Handle_t hInstance);
-
/****************************************************************************************/
/* */
/* FUNCTION: LVREV_Process */
@@ -303,21 +280,10 @@
/* 1. The input and output buffers must be 32-bit aligned */
/* */
/****************************************************************************************/
-#ifdef BUILD_FLOAT
LVREV_ReturnStatus_en LVREV_Process(LVREV_Handle_t hInstance,
const LVM_FLOAT *pInData,
LVM_FLOAT *pOutData,
const LVM_UINT16 NumSamples);
-#else
-LVREV_ReturnStatus_en LVREV_Process(LVREV_Handle_t hInstance,
- const LVM_INT32 *pInData,
- LVM_INT32 *pOutData,
- const LVM_UINT16 NumSamples);
-#endif
-
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
#endif /* __LVREV_H__ */
diff --git a/media/libeffects/lvm/lib/Reverb/src/LVREV_ApplyNewSettings.c b/media/libeffects/lvm/lib/Reverb/src/LVREV_ApplyNewSettings.c
deleted file mode 100644
index e710844..0000000
--- a/media/libeffects/lvm/lib/Reverb/src/LVREV_ApplyNewSettings.c
+++ /dev/null
@@ -1,1224 +0,0 @@
-/*
- * Copyright (C) 2004-2010 NXP Software
- * Copyright (C) 2010 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-/****************************************************************************************/
-/* */
-/* Includes */
-/* */
-/****************************************************************************************/
-#include "LVREV_Private.h"
-#include "Filter.h"
-
-/****************************************************************************************/
-/* */
-/* FUNCTION: LVREV_ApplyNewSettings */
-/* */
-/* DESCRIPTION: */
-/* Applies the new control parameters */
-/* */
-/* PARAMETERS: */
-/* pPrivate Pointer to the instance private parameters */
-/* */
-/* RETURNS: */
-/* LVREV_Success Succeeded */
-/* LVREV_NULLADDRESS When pPrivate is NULL */
-/* */
-/* NOTES: */
-/* */
-/****************************************************************************************/
-
-#ifndef BUILD_FLOAT
-LVREV_ReturnStatus_en LVREV_ApplyNewSettings (LVREV_Instance_st *pPrivate)
-{
-
- LVM_Mode_en OperatingMode;
- LVM_INT32 NumberOfDelayLines;
-
-
- /* Check for NULL pointer */
- if(pPrivate == LVM_NULL)
- {
- return LVREV_NULLADDRESS;
- }
-
- OperatingMode = pPrivate->NewParams.OperatingMode;
-
- if(pPrivate->InstanceParams.NumDelays == LVREV_DELAYLINES_4)
- {
- NumberOfDelayLines = 4;
- }
- else if(pPrivate->InstanceParams.NumDelays == LVREV_DELAYLINES_2)
- {
- NumberOfDelayLines = 2;
- }
- else
- {
- NumberOfDelayLines = 1;
- }
-
- /*
- * Update the high pass filter coefficients
- */
- if((pPrivate->NewParams.HPF != pPrivate->CurrentParams.HPF) ||
- (pPrivate->NewParams.SampleRate != pPrivate->CurrentParams.SampleRate) ||
- (pPrivate->bFirstControl == LVM_TRUE))
- {
- LVM_INT32 Omega;
- FO_C32_Coefs_t Coeffs;
-
- Omega = LVM_GetOmega(pPrivate->NewParams.HPF, pPrivate->NewParams.SampleRate);
- LVM_FO_HPF(Omega, &Coeffs);
- FO_1I_D32F32Cll_TRC_WRA_01_Init( &pPrivate->pFastCoef->HPCoefs, &pPrivate->pFastData->HPTaps, &Coeffs);
- LoadConst_32(0,
- (void *)&pPrivate->pFastData->HPTaps, /* Destination Cast to void: no dereferencing in function*/
- sizeof(Biquad_1I_Order1_Taps_t)/sizeof(LVM_INT32));
- }
-
-
- /*
- * Update the low pass filter coefficients
- */
- if((pPrivate->NewParams.LPF != pPrivate->CurrentParams.LPF) ||
- (pPrivate->NewParams.SampleRate != pPrivate->CurrentParams.SampleRate) ||
- (pPrivate->bFirstControl == LVM_TRUE))
- {
- LVM_INT32 Omega;
- FO_C32_Coefs_t Coeffs;
-
-
- Coeffs.A0 = 0x7FFFFFFF;
- Coeffs.A1 = 0;
- Coeffs.B1 = 0;
- if(pPrivate->NewParams.LPF <= (LVM_FsTable[pPrivate->NewParams.SampleRate] >> 1))
- {
- Omega = LVM_GetOmega(pPrivate->NewParams.LPF, pPrivate->NewParams.SampleRate);
-
- /*
- * Do not apply filter if w =2*pi*fc/fs >= 2.9
- */
- if(Omega<=LVREV_2_9_INQ29)
- {
- LVM_FO_LPF(Omega, &Coeffs);
- }
- }
- FO_1I_D32F32Cll_TRC_WRA_01_Init( &pPrivate->pFastCoef->LPCoefs, &pPrivate->pFastData->LPTaps, &Coeffs);
- LoadConst_32(0,
- (void *)&pPrivate->pFastData->LPTaps, /* Destination Cast to void: no dereferencing in function*/
- sizeof(Biquad_1I_Order1_Taps_t)/sizeof(LVM_INT32));
- }
-
-
- /*
- * Calculate the room size parameter
- */
- if( pPrivate->NewParams.RoomSize != pPrivate->CurrentParams.RoomSize)
- {
- /* Room size range is 10ms to 200ms
- * 0% -- 10ms
- * 50% -- 65ms
- * 100% -- 120ms
- */
- pPrivate->RoomSizeInms = 10 + (((pPrivate->NewParams.RoomSize*11) + 5)/10);
- }
-
-
- /*
- * Update the T delay number of samples and the all pass delay number of samples
- */
- if( (pPrivate->NewParams.RoomSize != pPrivate->CurrentParams.RoomSize) ||
- (pPrivate->NewParams.SampleRate != pPrivate->CurrentParams.SampleRate) ||
- (pPrivate->bFirstControl == LVM_TRUE))
- {
-
- LVM_UINT32 Temp;
- LVM_INT32 APDelaySize;
- LVM_INT32 Fs = LVM_GetFsFromTable(pPrivate->NewParams.SampleRate);
- LVM_UINT32 DelayLengthSamples = (LVM_UINT32)(Fs * pPrivate->RoomSizeInms);
- LVM_INT16 i;
- LVM_INT16 ScaleTable[] = {LVREV_T_3_Power_minus0_on_4, LVREV_T_3_Power_minus1_on_4, LVREV_T_3_Power_minus2_on_4, LVREV_T_3_Power_minus3_on_4};
- LVM_INT16 MaxT_Delay[] = {LVREV_MAX_T0_DELAY, LVREV_MAX_T1_DELAY, LVREV_MAX_T2_DELAY, LVREV_MAX_T3_DELAY};
- LVM_INT16 MaxAP_Delay[] = {LVREV_MAX_AP0_DELAY, LVREV_MAX_AP1_DELAY, LVREV_MAX_AP2_DELAY, LVREV_MAX_AP3_DELAY};
-
-
- /*
- * For each delay line
- */
- for (i=0; i<NumberOfDelayLines; i++)
- {
- if (i != 0)
- {
- LVM_INT32 Temp1; /* to avoid QAC warning on type conversion */
- LVM_INT32 Temp2; /* to avoid QAC warning on type conversion */
-
- Temp2=(LVM_INT32)DelayLengthSamples;
- MUL32x16INTO32(Temp2, ScaleTable[i], Temp1, 15)
- Temp=(LVM_UINT32)Temp1;
- }
- else
- {
- Temp = DelayLengthSamples;
- }
- APDelaySize = Temp / 1500;
-
-
- /*
- * Set the fixed delay
- */
- Temp = (MaxT_Delay[i] - MaxAP_Delay[i]) * Fs / 48000;
- pPrivate->Delay_AP[i] = pPrivate->T[i] - Temp;
-
-
- /*
- * Set the tap selection
- */
- if (pPrivate->AB_Selection)
- {
- /* Smooth from tap A to tap B */
- pPrivate->pOffsetB[i] = &pPrivate->pDelay_T[i][pPrivate->T[i] - Temp - APDelaySize];
- pPrivate->B_DelaySize[i] = APDelaySize;
- pPrivate->Mixer_APTaps[i].Target1 = 0;
- pPrivate->Mixer_APTaps[i].Target2 = 0x7fffffff;
- }
- else
- {
- /* Smooth from tap B to tap A */
- pPrivate->pOffsetA[i] = &pPrivate->pDelay_T[i][pPrivate->T[i] - Temp - APDelaySize];
- pPrivate->A_DelaySize[i] = APDelaySize;
- pPrivate->Mixer_APTaps[i].Target2 = 0;
- pPrivate->Mixer_APTaps[i].Target1 = 0x7fffffff;
- }
-
- /*
- * Set the maximum block size to the smallest delay size
- */
- pPrivate->MaxBlkLen = Temp;
- if (pPrivate->MaxBlkLen > pPrivate->A_DelaySize[i])
- {
- pPrivate->MaxBlkLen = pPrivate->A_DelaySize[i];
- }
- if (pPrivate->MaxBlkLen > pPrivate->B_DelaySize[i])
- {
- pPrivate->MaxBlkLen = pPrivate->B_DelaySize[i];
- }
- }
- if (pPrivate->AB_Selection)
- {
- pPrivate->AB_Selection = 0;
- }
- else
- {
- pPrivate->AB_Selection = 1;
- }
-
-
- /*
- * Limit the maximum block length
- */
- pPrivate->MaxBlkLen=pPrivate->MaxBlkLen-2; /* Just as a precausion, but no problem if we remove this line */
- if(pPrivate->MaxBlkLen > pPrivate->InstanceParams.MaxBlockSize)
- {
- pPrivate->MaxBlkLen = (LVM_INT32)pPrivate->InstanceParams.MaxBlockSize;
- }
- }
-
-
- /*
- * Update the low pass filter coefficient
- */
- if( (pPrivate->NewParams.Damping != pPrivate->CurrentParams.Damping) ||
- (pPrivate->NewParams.SampleRate != pPrivate->CurrentParams.SampleRate) ||
- (pPrivate->bFirstControl == LVM_TRUE))
- {
-
- LVM_INT32 Temp;
- LVM_INT32 Omega;
- FO_C32_Coefs_t Coeffs;
- LVM_INT16 i;
- LVM_INT16 Damping = (LVM_INT16)((pPrivate->NewParams.Damping * 100) + 1000);
- LVM_INT32 ScaleTable[] = {LVREV_T_3_Power_0_on_4, LVREV_T_3_Power_1_on_4, LVREV_T_3_Power_2_on_4, LVREV_T_3_Power_3_on_4};
-
-
- /*
- * For each filter
- */
- for (i=0; i<NumberOfDelayLines; i++)
- {
- if (i != 0)
- {
- MUL32x16INTO32(ScaleTable[i], Damping, Temp, 15)
- }
- else
- {
- Temp = Damping;
- }
- if(Temp <= (LVM_FsTable[pPrivate->NewParams.SampleRate] >> 1))
- {
- Omega = LVM_GetOmega((LVM_UINT16)Temp, pPrivate->NewParams.SampleRate);
- LVM_FO_LPF(Omega, &Coeffs);
- }
- else
- {
- Coeffs.A0 = 0x7FF00000;
- Coeffs.A1 = 0;
- Coeffs.B1 = 0;
- }
- FO_1I_D32F32Cll_TRC_WRA_01_Init(&pPrivate->pFastCoef->RevLPCoefs[i], &pPrivate->pFastData->RevLPTaps[i], &Coeffs);
- }
- }
-
-
- /*
- * Update All-pass filter mixer time constants
- */
- if( (pPrivate->NewParams.RoomSize != pPrivate->CurrentParams.RoomSize) ||
- (pPrivate->NewParams.SampleRate != pPrivate->CurrentParams.SampleRate) ||
- (pPrivate->NewParams.Density != pPrivate->CurrentParams.Density))
- {
- LVM_INT16 i;
- LVM_INT32 Alpha = (LVM_INT32)LVM_Mixer_TimeConstant(LVREV_ALLPASS_TC, LVM_GetFsFromTable(pPrivate->NewParams.SampleRate), 1);
- LVM_INT32 AlphaTap = (LVM_INT32)LVM_Mixer_TimeConstant(LVREV_ALLPASS_TAP_TC, LVM_GetFsFromTable(pPrivate->NewParams.SampleRate), 1);
-
- for (i=0; i<4; i++)
- {
- pPrivate->Mixer_APTaps[i].Alpha1 = AlphaTap;
- pPrivate->Mixer_APTaps[i].Alpha2 = AlphaTap;
- pPrivate->Mixer_SGFeedback[i].Alpha = Alpha;
- pPrivate->Mixer_SGFeedforward[i].Alpha = Alpha;
- }
- }
-
-
- /*
- * Update the feed back gain
- */
- if( (pPrivate->NewParams.RoomSize != pPrivate->CurrentParams.RoomSize) ||
- (pPrivate->NewParams.SampleRate != pPrivate->CurrentParams.SampleRate) ||
- (pPrivate->NewParams.T60 != pPrivate->CurrentParams.T60) ||
- (pPrivate->bFirstControl == LVM_TRUE))
- {
-
- LVM_INT32 G[4]; /* Feedback gain (Q7.24) */
-
- if(pPrivate->NewParams.T60 == 0)
- {
- G[3] = 0;
- G[2] = 0;
- G[1] = 0;
- G[0] = 0;
- }
- else
- {
- LVM_INT32 Temp1;
- LVM_INT32 Temp2;
- LVM_INT16 i;
- LVM_INT16 ScaleTable[] = {LVREV_T_3_Power_minus0_on_4, LVREV_T_3_Power_minus1_on_4, LVREV_T_3_Power_minus2_on_4, LVREV_T_3_Power_minus3_on_4};
-
-
- /*
- * For each delay line
- */
- for (i=0; i<NumberOfDelayLines; i++)
- {
- Temp1 = (3 * pPrivate->RoomSizeInms * ScaleTable[i]) / pPrivate->NewParams.T60;
- if(Temp1 >= (4 << 15))
- {
- G[i] = 0;
- }
- else if((Temp1 >= (2 << 15)))
- {
- Temp2 = LVM_Power10(-(Temp1 << 14));
- Temp1 = LVM_Power10(-(Temp1 << 14));
- MUL32x32INTO32(Temp1,Temp2,Temp1,24)
- }
- else
- {
- Temp1 = LVM_Power10(-(Temp1 << 15));
- }
- if (NumberOfDelayLines == 1)
- {
- G[i] = Temp1;
- }
- else
- {
- LVM_INT32 TempG;
- MUL32x16INTO32(Temp1,ONE_OVER_SQRT_TWO,TempG,15)
- G[i]=TempG;
- }
- }
- }
-
- /* Set up the feedback mixers for four delay lines */
- pPrivate->FeedbackMixer[0].Target=G[0]<<7;
- pPrivate->FeedbackMixer[1].Target=G[1]<<7;
- pPrivate->FeedbackMixer[2].Target=G[2]<<7;
- pPrivate->FeedbackMixer[3].Target=G[3]<<7;
- }
-
-
- /*
- * Calculate the gain correction
- */
- if((pPrivate->NewParams.RoomSize != pPrivate->CurrentParams.RoomSize) ||
- (pPrivate->NewParams.Level != pPrivate->CurrentParams.Level) ||
- (pPrivate->NewParams.T60 != pPrivate->CurrentParams.T60) )
- {
- LVM_INT32 Index=0;
- LVM_INT32 i=0;
- LVM_INT32 Gain=0;
- LVM_INT32 RoomSize=0;
- LVM_INT32 T60;
- LVM_INT32 Coefs[5];
-
- if(pPrivate->NewParams.RoomSize==0)
- {
- RoomSize=1;
- }
- else
- {
- RoomSize=(LVM_INT32)pPrivate->NewParams.RoomSize;
- }
-
- if(pPrivate->NewParams.T60<100)
- {
- T60 = 100 * LVREV_T60_SCALE;
- }
- else
- {
- T60 = pPrivate->NewParams.T60 * LVREV_T60_SCALE;
- }
-
- /* Find the nearest room size in table */
- for(i=0;i<24;i++)
- {
- if(RoomSize<= LVREV_GainPolyTable[i][0])
- {
- Index=i;
- break;
- }
- }
-
-
- if(RoomSize==LVREV_GainPolyTable[Index][0])
- {
- /* Take table values if the room size is in table */
- for(i=1;i<5;i++)
- {
- Coefs[i-1]=LVREV_GainPolyTable[Index][i];
- }
- Coefs[4]=0;
- Gain=LVM_Polynomial(3,Coefs,T60); /* Q.24 result */
- }
- else
- {
- /* Interpolate the gain between nearest room sizes */
-
- LVM_INT32 Gain1,Gain2;
- LVM_INT32 Tot_Dist,Dist;
-
- Tot_Dist=LVREV_GainPolyTable[Index][0]-LVREV_GainPolyTable[Index-1][0];
- Dist=RoomSize-LVREV_GainPolyTable[Index-1][0];
-
-
- /* Get gain for first */
- for(i=1;i<5;i++)
- {
- Coefs[i-1]=LVREV_GainPolyTable[Index-1][i];
- }
- Coefs[4]=0;
-
- Gain1=LVM_Polynomial(3,Coefs,T60); /* Q.24 result */
-
- /* Get gain for second */
- for(i=1;i<5;i++)
- {
- Coefs[i-1]=LVREV_GainPolyTable[Index][i];
- }
- Coefs[4]=0;
-
- Gain2=LVM_Polynomial(3,Coefs,T60); /* Q.24 result */
-
- /* Linear Interpolate the gain */
- Gain = Gain1+ (((Gain2-Gain1)*Dist)/(Tot_Dist));
- }
-
-
- /*
- * Get the inverse of gain: Q.15
- * Gain is mostly above one except few cases, take only gains above 1
- */
- if(Gain < 16777216L)
- {
- pPrivate->Gain= 32767;
- }
- else
- {
- pPrivate->Gain=(LVM_INT16)(LVM_MAXINT_32/(Gain>>8));
- }
-
-
- Index=((32767*100)/(100+pPrivate->NewParams.Level));
- pPrivate->Gain=(LVM_INT16)((pPrivate->Gain*Index)>>15);
- pPrivate->GainMixer.Target = pPrivate->Gain*Index;
- }
-
-
- /*
- * Update the all pass comb filter coefficient
- */
- if( (pPrivate->NewParams.Density != pPrivate->CurrentParams.Density) ||
- (pPrivate->bFirstControl == LVM_TRUE))
- {
- LVM_INT16 i;
- LVM_INT32 b = pPrivate->NewParams.Density * LVREV_B_8_on_1000;
-
- for (i=0;i<4; i++)
- {
- pPrivate->Mixer_SGFeedback[i].Target = b;
- pPrivate->Mixer_SGFeedforward[i].Target = b;
- }
- }
-
-
- /*
- * Update the bypass mixer time constant
- */
- if((pPrivate->NewParams.SampleRate != pPrivate->CurrentParams.SampleRate) ||
- (pPrivate->bFirstControl == LVM_TRUE))
- {
- LVM_UINT16 NumChannels = 1; /* Assume MONO format */
- LVM_INT32 Alpha;
-
- Alpha = (LVM_INT32)LVM_Mixer_TimeConstant(LVREV_FEEDBACKMIXER_TC, LVM_GetFsFromTable(pPrivate->NewParams.SampleRate), NumChannels);
- pPrivate->FeedbackMixer[0].Alpha=Alpha;
- pPrivate->FeedbackMixer[1].Alpha=Alpha;
- pPrivate->FeedbackMixer[2].Alpha=Alpha;
- pPrivate->FeedbackMixer[3].Alpha=Alpha;
-
- NumChannels = 2; /* Always stereo output */
- pPrivate->BypassMixer.Alpha1 = (LVM_INT32)LVM_Mixer_TimeConstant(LVREV_BYPASSMIXER_TC, LVM_GetFsFromTable(pPrivate->NewParams.SampleRate), NumChannels);
- pPrivate->BypassMixer.Alpha2 = pPrivate->BypassMixer.Alpha1;
- pPrivate->GainMixer.Alpha = pPrivate->BypassMixer.Alpha1;
- }
-
-
- /*
- * Update the bypass mixer targets
- */
- if( (pPrivate->NewParams.Level != pPrivate->CurrentParams.Level) &&
- (pPrivate->NewParams.OperatingMode == LVM_MODE_ON))
- {
- pPrivate->BypassMixer.Target2 = ((LVM_INT32)(pPrivate->NewParams.Level * 32767)/100)<<16;
- pPrivate->BypassMixer.Target1 = 0x00000000;
- if ((pPrivate->NewParams.Level == 0) && (pPrivate->bFirstControl == LVM_FALSE))
- {
- pPrivate->BypassMixer.CallbackSet2 = LVM_TRUE;
- }
- if (pPrivate->NewParams.Level != 0)
- {
- pPrivate->bDisableReverb = LVM_FALSE;
- }
- }
-
- if(pPrivate->NewParams.OperatingMode != pPrivate->CurrentParams.OperatingMode)
- {
- if(pPrivate->NewParams.OperatingMode == LVM_MODE_ON)
- {
- pPrivate->BypassMixer.Target2 = ((LVM_INT32)(pPrivate->NewParams.Level * 32767)/100)<<16;
- pPrivate->BypassMixer.Target1 = 0x00000000;
-
- pPrivate->BypassMixer.CallbackSet2 = LVM_FALSE;
- OperatingMode = LVM_MODE_ON;
- if (pPrivate->NewParams.Level == 0)
- {
- pPrivate->bDisableReverb = LVM_TRUE;
- }
- else
- {
- pPrivate->bDisableReverb = LVM_FALSE;
- }
- }
- else if (pPrivate->bFirstControl == LVM_FALSE)
- {
- pPrivate->BypassMixer.Target2 = 0x00000000;
- pPrivate->BypassMixer.Target1 = 0x00000000;
- pPrivate->BypassMixer.CallbackSet2 = LVM_TRUE;
- pPrivate->GainMixer.Target = 0x03FFFFFF;
- OperatingMode = LVM_MODE_ON;
- }
- else
- {
- OperatingMode = LVM_MODE_OFF;
- }
- }
-
-
- /*
- * If it is the first call to ApplyNew settings force the current to the target to begin immediate playback of the effect
- */
- if(pPrivate->bFirstControl == LVM_TRUE)
- {
- pPrivate->BypassMixer.Current1 = pPrivate->BypassMixer.Target1;
- pPrivate->BypassMixer.Current2 = pPrivate->BypassMixer.Target2;
- }
-
-
- /*
- * Copy the new parameters
- */
- pPrivate->CurrentParams = pPrivate->NewParams;
- pPrivate->CurrentParams.OperatingMode = OperatingMode;
-
-
- /*
- * Update flag
- */
- if(pPrivate->bFirstControl == LVM_TRUE)
- {
- pPrivate->bFirstControl = LVM_FALSE;
- }
-
-
- return LVREV_SUCCESS;
-}
-#else /* BUILD_FLOAT*/
-LVREV_ReturnStatus_en LVREV_ApplyNewSettings (LVREV_Instance_st *pPrivate)
-{
-
- LVM_Mode_en OperatingMode;
- LVM_INT32 NumberOfDelayLines;
-
-
- /* Check for NULL pointer */
- if(pPrivate == LVM_NULL)
- {
- return LVREV_NULLADDRESS;
- }
-
- OperatingMode = pPrivate->NewParams.OperatingMode;
-
- if(pPrivate->InstanceParams.NumDelays == LVREV_DELAYLINES_4)
- {
- NumberOfDelayLines = 4;
- }
- else if(pPrivate->InstanceParams.NumDelays == LVREV_DELAYLINES_2)
- {
- NumberOfDelayLines = 2;
- }
- else
- {
- NumberOfDelayLines = 1;
- }
-
- /*
- * Update the high pass filter coefficients
- */
- if((pPrivate->NewParams.HPF != pPrivate->CurrentParams.HPF) ||
- (pPrivate->NewParams.SampleRate != pPrivate->CurrentParams.SampleRate) ||
- (pPrivate->bFirstControl == LVM_TRUE))
- {
- LVM_FLOAT Omega;
- FO_FLOAT_Coefs_t Coeffs;
-
- Omega = LVM_GetOmega(pPrivate->NewParams.HPF, pPrivate->NewParams.SampleRate);
- LVM_FO_HPF(Omega, &Coeffs);
- FO_1I_D32F32Cll_TRC_WRA_01_Init( &pPrivate->pFastCoef->HPCoefs,
- &pPrivate->pFastData->HPTaps, &Coeffs);
- LoadConst_Float(0,
- (void *)&pPrivate->pFastData->HPTaps, /* Destination Cast to void: \
- no dereferencing in function*/
- sizeof(Biquad_1I_Order1_FLOAT_Taps_t) / sizeof(LVM_FLOAT));
- }
-
-
- /*
- * Update the low pass filter coefficients
- */
- if((pPrivate->NewParams.LPF != pPrivate->CurrentParams.LPF) ||
- (pPrivate->NewParams.SampleRate != pPrivate->CurrentParams.SampleRate) ||
- (pPrivate->bFirstControl == LVM_TRUE))
- {
- LVM_FLOAT Omega;
- FO_FLOAT_Coefs_t Coeffs;
-
- Coeffs.A0 = 1;
- Coeffs.A1 = 0;
- Coeffs.B1 = 0;
- if(pPrivate->NewParams.LPF <= (LVM_FsTable[pPrivate->NewParams.SampleRate] >> 1))
- {
- Omega = LVM_GetOmega(pPrivate->NewParams.LPF, pPrivate->NewParams.SampleRate);
-
- /*
- * Do not apply filter if w =2*pi*fc/fs >= 2.9
- */
- if(Omega <= (LVM_FLOAT)LVREV_2_9_INQ29)
- {
- LVM_FO_LPF(Omega, &Coeffs);
- }
- }
- FO_1I_D32F32Cll_TRC_WRA_01_Init( &pPrivate->pFastCoef->LPCoefs,
- &pPrivate->pFastData->LPTaps, &Coeffs);
- LoadConst_Float(0,
- (void *)&pPrivate->pFastData->LPTaps, /* Destination Cast to void: \
- no dereferencing in function*/
- sizeof(Biquad_1I_Order1_FLOAT_Taps_t) / sizeof(LVM_FLOAT));
- }
-
-
- /*
- * Calculate the room size parameter
- */
- if( pPrivate->NewParams.RoomSize != pPrivate->CurrentParams.RoomSize)
- {
- /* Room size range is 10ms to 200ms
- * 0% -- 10ms
- * 50% -- 65ms
- * 100% -- 120ms
- */
- pPrivate->RoomSizeInms = 10 + (((pPrivate->NewParams.RoomSize*11) + 5) / 10);
- }
-
-
- /*
- * Update the T delay number of samples and the all pass delay number of samples
- */
- if( (pPrivate->NewParams.RoomSize != pPrivate->CurrentParams.RoomSize) ||
- (pPrivate->NewParams.SampleRate != pPrivate->CurrentParams.SampleRate) ||
- (pPrivate->bFirstControl == LVM_TRUE))
- {
-
- LVM_UINT32 Temp;
- LVM_INT32 APDelaySize;
- LVM_INT32 Fs = LVM_GetFsFromTable(pPrivate->NewParams.SampleRate);
- LVM_UINT32 DelayLengthSamples = (LVM_UINT32)(Fs * pPrivate->RoomSizeInms);
- LVM_INT16 i;
- LVM_FLOAT ScaleTable[] = {LVREV_T_3_Power_minus0_on_4, LVREV_T_3_Power_minus1_on_4, \
- LVREV_T_3_Power_minus2_on_4, LVREV_T_3_Power_minus3_on_4};
- LVM_INT16 MaxT_Delay[] = {LVREV_MAX_T0_DELAY, LVREV_MAX_T1_DELAY, \
- LVREV_MAX_T2_DELAY, LVREV_MAX_T3_DELAY};
- LVM_INT16 MaxAP_Delay[] = {LVREV_MAX_AP0_DELAY, LVREV_MAX_AP1_DELAY, \
- LVREV_MAX_AP2_DELAY, LVREV_MAX_AP3_DELAY};
-
-
- /*
- * For each delay line
- */
- for (i = 0; i < NumberOfDelayLines; i++)
- {
- if (i != 0)
- {
- LVM_FLOAT Temp1; /* to avoid QAC warning on type conversion */
-
- Temp1=(LVM_FLOAT)DelayLengthSamples;
- Temp = (LVM_UINT32)(Temp1 * ScaleTable[i]);
- }
- else
- {
- Temp = DelayLengthSamples;
- }
- APDelaySize = Temp / 1500;
-
-
- /*
- * Set the fixed delay
- */
-
-#ifdef HIGHER_FS
- Temp = (MaxT_Delay[i] - MaxAP_Delay[i]) * Fs / 192000;
-#else
- Temp = (MaxT_Delay[i] - MaxAP_Delay[i]) * Fs / 48000;
-#endif
- pPrivate->Delay_AP[i] = pPrivate->T[i] - Temp;
-
-
- /*
- * Set the tap selection
- */
- if (pPrivate->AB_Selection)
- {
- /* Smooth from tap A to tap B */
- pPrivate->pOffsetB[i] = &pPrivate->pDelay_T[i][pPrivate->T[i] - \
- Temp - APDelaySize];
- pPrivate->B_DelaySize[i] = APDelaySize;
- pPrivate->Mixer_APTaps[i].Target1 = 0;
- pPrivate->Mixer_APTaps[i].Target2 = 1.0f;
- }
- else
- {
- /* Smooth from tap B to tap A */
- pPrivate->pOffsetA[i] = &pPrivate->pDelay_T[i][pPrivate->T[i] - \
- Temp - APDelaySize];
- pPrivate->A_DelaySize[i] = APDelaySize;
- pPrivate->Mixer_APTaps[i].Target2 = 0;
- pPrivate->Mixer_APTaps[i].Target1 = 1.0f;
- }
-
- /*
- * Set the maximum block size to the smallest delay size
- */
- pPrivate->MaxBlkLen = Temp;
- if (pPrivate->MaxBlkLen > pPrivate->A_DelaySize[i])
- {
- pPrivate->MaxBlkLen = pPrivate->A_DelaySize[i];
- }
- if (pPrivate->MaxBlkLen > pPrivate->B_DelaySize[i])
- {
- pPrivate->MaxBlkLen = pPrivate->B_DelaySize[i];
- }
- }
- if (pPrivate->AB_Selection)
- {
- pPrivate->AB_Selection = 0;
- }
- else
- {
- pPrivate->AB_Selection = 1;
- }
-
-
- /*
- * Limit the maximum block length
- */
- /* Just as a precausion, but no problem if we remove this line */
- pPrivate->MaxBlkLen = pPrivate->MaxBlkLen - 2;
- if(pPrivate->MaxBlkLen > pPrivate->InstanceParams.MaxBlockSize)
- {
- pPrivate->MaxBlkLen = (LVM_INT32)pPrivate->InstanceParams.MaxBlockSize;
- }
- }
-
-
-
- /*
- * Update the low pass filter coefficient
- */
- if( (pPrivate->NewParams.Damping != pPrivate->CurrentParams.Damping) ||
- (pPrivate->NewParams.SampleRate != pPrivate->CurrentParams.SampleRate) ||
- (pPrivate->bFirstControl == LVM_TRUE))
- {
-
- LVM_INT32 Temp;
- LVM_FLOAT Omega;
- FO_FLOAT_Coefs_t Coeffs;
- LVM_INT16 i;
- LVM_INT16 Damping = (LVM_INT16)((pPrivate->NewParams.Damping * 100) + 1000);
- LVM_FLOAT ScaleTable[] = {LVREV_T_3_Power_0_on_4, LVREV_T_3_Power_1_on_4,
- LVREV_T_3_Power_2_on_4, LVREV_T_3_Power_3_on_4};
-
-
- /*
- * For each filter
- */
- for (i = 0; i < NumberOfDelayLines; i++)
- {
- if (i != 0)
- {
- Temp = (LVM_INT32)(ScaleTable[i] * Damping);
- }
- else
- {
- Temp = Damping;
- }
- if(Temp <= (LVM_INT32)(LVM_FsTable[pPrivate->NewParams.SampleRate] >> 1))
- {
- Omega = LVM_GetOmega(Temp, pPrivate->NewParams.SampleRate);
- LVM_FO_LPF(Omega, &Coeffs);
- }
- else
- {
- Coeffs.A0 = 1;
- Coeffs.A1 = 0;
- Coeffs.B1 = 0;
- }
- FO_1I_D32F32Cll_TRC_WRA_01_Init(&pPrivate->pFastCoef->RevLPCoefs[i],
- &pPrivate->pFastData->RevLPTaps[i], &Coeffs);
- }
- }
-
-
- /*
- * Update All-pass filter mixer time constants
- */
- if( (pPrivate->NewParams.RoomSize != pPrivate->CurrentParams.RoomSize) ||
- (pPrivate->NewParams.SampleRate != pPrivate->CurrentParams.SampleRate) ||
- (pPrivate->NewParams.Density != pPrivate->CurrentParams.Density))
- {
- LVM_INT16 i;
- LVM_FLOAT Alpha;
- LVM_FLOAT AlphaTap;
-
- Alpha = LVM_Mixer_TimeConstant(LVREV_ALLPASS_TC,
- LVM_GetFsFromTable(pPrivate->NewParams.SampleRate),
- 1);
-
- AlphaTap = LVM_Mixer_TimeConstant(LVREV_ALLPASS_TAP_TC,
- LVM_GetFsFromTable(pPrivate->NewParams.SampleRate),
- 1);
-
- for (i = 0; i < 4; i++)
- {
- pPrivate->Mixer_APTaps[i].Alpha1 = AlphaTap;
- pPrivate->Mixer_APTaps[i].Alpha2 = AlphaTap;
- pPrivate->Mixer_SGFeedback[i].Alpha = Alpha;
- pPrivate->Mixer_SGFeedforward[i].Alpha = Alpha;
- }
- }
-
-
- /*
- * Update the feed back gain
- */
- if( (pPrivate->NewParams.RoomSize != pPrivate->CurrentParams.RoomSize) ||
- (pPrivate->NewParams.SampleRate != pPrivate->CurrentParams.SampleRate) ||
- (pPrivate->NewParams.T60 != pPrivate->CurrentParams.T60) ||
- (pPrivate->bFirstControl == LVM_TRUE))
- {
-
- LVM_FLOAT G[4]; /* Feedback gain (Q7.24) */
-
- if(pPrivate->NewParams.T60 == 0)
- {
- G[3] = 0;
- G[2] = 0;
- G[1] = 0;
- G[0] = 0;
- }
- else
- {
- LVM_FLOAT Temp1;
- LVM_FLOAT Temp2;
- LVM_INT16 i;
- LVM_FLOAT ScaleTable[] = {LVREV_T_3_Power_minus0_on_4, LVREV_T_3_Power_minus1_on_4,
- LVREV_T_3_Power_minus2_on_4, LVREV_T_3_Power_minus3_on_4};
-
-
- /*
- * For each delay line
- */
- for (i = 0; i < NumberOfDelayLines; i++)
- {
- Temp1 = (3 * pPrivate->RoomSizeInms * ScaleTable[i]) / pPrivate->NewParams.T60;
- if(Temp1 >= (4))
- {
- G[i] = 0;
- }
- else if((Temp1 >= (2)))
- {
- Temp2 = LVM_Power10(-(Temp1 / 2.0f));
- Temp1 = LVM_Power10(-(Temp1 / 2.0f));
- Temp1 = Temp1 * Temp2;
- }
- else
- {
- Temp1 = LVM_Power10(-(Temp1));
- }
- if (NumberOfDelayLines == 1)
- {
- G[i] = Temp1;
- }
- else
- {
- LVM_FLOAT TempG;
- TempG = Temp1 * ONE_OVER_SQRT_TWO;
- G[i]=TempG;
- }
- }
- }
-
- /* Set up the feedback mixers for four delay lines */
- pPrivate->FeedbackMixer[0].Target=G[0];
- pPrivate->FeedbackMixer[1].Target=G[1];
- pPrivate->FeedbackMixer[2].Target=G[2];
- pPrivate->FeedbackMixer[3].Target=G[3];
- }
-
-
- /*
- * Calculate the gain correction
- */
- if((pPrivate->NewParams.RoomSize != pPrivate->CurrentParams.RoomSize) ||
- (pPrivate->NewParams.Level != pPrivate->CurrentParams.Level) ||
- (pPrivate->NewParams.T60 != pPrivate->CurrentParams.T60) )
- {
- LVM_INT32 Index=0;
- LVM_FLOAT Index_FLOAT;
- LVM_INT32 i=0;
- LVM_FLOAT Gain=0;
- LVM_INT32 RoomSize=0;
- LVM_FLOAT T60;
- LVM_FLOAT Coefs[5];
-
-
- if(pPrivate->NewParams.RoomSize == 0)
- {
- RoomSize = 1;
- }
- else
- {
- RoomSize = (LVM_INT32)pPrivate->NewParams.RoomSize;
- }
-
-
- if(pPrivate->NewParams.T60 < 100)
- {
- T60 = 100 * LVREV_T60_SCALE;
- }
- else
- {
- T60 = pPrivate->NewParams.T60 * LVREV_T60_SCALE;
- }
-
- /* Find the nearest room size in table */
- for(i = 0; i < 24; i++)
- {
- if(RoomSize <= LVREV_GainPolyTable[i][0])
- {
- Index = i;
- break;
- }
- }
-
-
- if(RoomSize == LVREV_GainPolyTable[Index][0])
- {
- /* Take table values if the room size is in table */
- for(i = 1; i < 5; i++)
- {
- Coefs[i-1] = LVREV_GainPolyTable[Index][i];
- }
- Coefs[4] = 0;
- Gain = LVM_Polynomial(3, Coefs, T60); /* Q.24 result */
- }
- else
- {
- /* Interpolate the gain between nearest room sizes */
-
- LVM_FLOAT Gain1,Gain2;
- LVM_INT32 Tot_Dist,Dist;
-
- Tot_Dist = (LVM_UINT32)LVREV_GainPolyTable[Index][0] - \
- (LVM_UINT32)LVREV_GainPolyTable[Index-1][0];
- Dist = RoomSize - (LVM_UINT32)LVREV_GainPolyTable[Index - 1][0];
-
-
- /* Get gain for first */
- for(i = 1; i < 5; i++)
- {
- Coefs[i-1] = LVREV_GainPolyTable[Index-1][i];
- }
- Coefs[4] = 0;
-
- Gain1 = LVM_Polynomial(3, Coefs, T60); /* Q.24 result */
-
- /* Get gain for second */
- for(i = 1; i < 5; i++)
- {
- Coefs[i-1] = LVREV_GainPolyTable[Index][i];
- }
- Coefs[4] = 0;
-
- Gain2 = LVM_Polynomial(3, Coefs, T60); /* Q.24 result */
-
- /* Linear Interpolate the gain */
- Gain = Gain1 + (((Gain2 - Gain1) * Dist) / (Tot_Dist));
- }
-
-
- /*
- * Get the inverse of gain: Q.15
- * Gain is mostly above one except few cases, take only gains above 1
- */
- if(Gain < 1)
- {
- pPrivate->Gain = 1;
- }
- else
- {
- pPrivate->Gain = 1 / Gain;
- }
-
- Index_FLOAT = 100.0f / (LVM_FLOAT)(100 + pPrivate->NewParams.Level);
- pPrivate->Gain = pPrivate->Gain * Index_FLOAT;
- pPrivate->GainMixer.Target = (pPrivate->Gain*Index_FLOAT) / 2;
- }
-
-
- /*
- * Update the all pass comb filter coefficient
- */
- if( (pPrivate->NewParams.Density != pPrivate->CurrentParams.Density) ||
- (pPrivate->bFirstControl == LVM_TRUE))
- {
- LVM_INT16 i;
- LVM_FLOAT b = (LVM_FLOAT)pPrivate->NewParams.Density * LVREV_B_8_on_1000;
-
- for (i = 0; i < 4; i++)
- {
- pPrivate->Mixer_SGFeedback[i].Target = b;
- pPrivate->Mixer_SGFeedforward[i].Target = b;
- }
- }
-
-
- /*
- * Update the bypass mixer time constant
- */
- if((pPrivate->NewParams.SampleRate != pPrivate->CurrentParams.SampleRate) ||
- (pPrivate->bFirstControl == LVM_TRUE))
- {
- LVM_UINT16 NumChannels = 1; /* Assume MONO format */
- LVM_FLOAT Alpha;
-
- Alpha = LVM_Mixer_TimeConstant(LVREV_FEEDBACKMIXER_TC,
- LVM_GetFsFromTable(pPrivate->NewParams.SampleRate),
- NumChannels);
- pPrivate->FeedbackMixer[0].Alpha = Alpha;
- pPrivate->FeedbackMixer[1].Alpha = Alpha;
- pPrivate->FeedbackMixer[2].Alpha = Alpha;
- pPrivate->FeedbackMixer[3].Alpha = Alpha;
-
- NumChannels = 2; /* Always stereo output */
- pPrivate->BypassMixer.Alpha1 = LVM_Mixer_TimeConstant(LVREV_BYPASSMIXER_TC,
- LVM_GetFsFromTable(pPrivate->NewParams.SampleRate), NumChannels);
- pPrivate->BypassMixer.Alpha2 = pPrivate->BypassMixer.Alpha1;
- pPrivate->GainMixer.Alpha = pPrivate->BypassMixer.Alpha1;
- }
-
-
- /*
- * Update the bypass mixer targets
- */
- if( (pPrivate->NewParams.Level != pPrivate->CurrentParams.Level) &&
- (pPrivate->NewParams.OperatingMode == LVM_MODE_ON))
- {
- pPrivate->BypassMixer.Target2 = (LVM_FLOAT)(pPrivate->NewParams.Level ) / 100.0f;
- pPrivate->BypassMixer.Target1 = 0x00000000;
- if ((pPrivate->NewParams.Level == 0) && (pPrivate->bFirstControl == LVM_FALSE))
- {
- pPrivate->BypassMixer.CallbackSet2 = LVM_TRUE;
- }
- if (pPrivate->NewParams.Level != 0)
- {
- pPrivate->bDisableReverb = LVM_FALSE;
- }
- }
-
- if(pPrivate->NewParams.OperatingMode != pPrivate->CurrentParams.OperatingMode)
- {
- if(pPrivate->NewParams.OperatingMode == LVM_MODE_ON)
- {
- pPrivate->BypassMixer.Target2 = (LVM_FLOAT)(pPrivate->NewParams.Level ) / 100.0f;
- pPrivate->BypassMixer.Target1 = 0x00000000;
-
- pPrivate->BypassMixer.CallbackSet2 = LVM_FALSE;
- OperatingMode = LVM_MODE_ON;
- if (pPrivate->NewParams.Level == 0)
- {
- pPrivate->bDisableReverb = LVM_TRUE;
- }
- else
- {
- pPrivate->bDisableReverb = LVM_FALSE;
- }
- }
- else if (pPrivate->bFirstControl == LVM_FALSE)
- {
- pPrivate->BypassMixer.Target2 = 0x00000000;
- pPrivate->BypassMixer.Target1 = 0x00000000;
- pPrivate->BypassMixer.CallbackSet2 = LVM_TRUE;
- pPrivate->GainMixer.Target = 0.03125f;
- OperatingMode = LVM_MODE_ON;
- }
- else
- {
- OperatingMode = LVM_MODE_OFF;
- }
- }
-
-
- /* If it is the first call to ApplyNew settings force the current to the target \
- to begin immediate playback of the effect */
- if(pPrivate->bFirstControl == LVM_TRUE)
- {
- pPrivate->BypassMixer.Current1 = pPrivate->BypassMixer.Target1;
- pPrivate->BypassMixer.Current2 = pPrivate->BypassMixer.Target2;
- }
-
-
- /*
- * Copy the new parameters
- */
- pPrivate->CurrentParams = pPrivate->NewParams;
- pPrivate->CurrentParams.OperatingMode = OperatingMode;
-
-
- /*
- * Update flag
- */
- if(pPrivate->bFirstControl == LVM_TRUE)
- {
- pPrivate->bFirstControl = LVM_FALSE;
- }
-
-
- return LVREV_SUCCESS;
-}
-#endif /*BUILD_FLOAT*/
-/****************************************************************************************/
-/* */
-/* FUNCTION: BypassMixer_Callback */
-/* */
-/* DESCRIPTION: */
-/* Controls the On to Off operating mode transition */
-/* */
-/* PARAMETERS: */
-/* pPrivate Pointer to the instance private parameters */
-/* */
-/* RETURNS: */
-/* LVREV_Success Succeeded */
-/* LVREV_NULLADDRESS When pPrivate is NULL */
-/* */
-/* NOTES: */
-/* */
-/****************************************************************************************/
-LVM_INT32 BypassMixer_Callback (void *pCallbackData,
- void *pGeneralPurpose,
- LVM_INT16 GeneralPurpose )
-{
-
- LVREV_Instance_st *pLVREV_Private = (LVREV_Instance_st *)pCallbackData;
-
-
- /*
- * Avoid build warnings
- */
- (void)pGeneralPurpose;
- (void)GeneralPurpose;
-
-
- /*
- * Turn off
- */
- pLVREV_Private->CurrentParams.OperatingMode = LVM_MODE_OFF;
- pLVREV_Private->bDisableReverb = LVM_TRUE;
- LVREV_ClearAudioBuffers((LVREV_Handle_t)pCallbackData);
-
-
- return 0;
-}
-
-/* End of file */
-
diff --git a/media/libeffects/lvm/lib/Reverb/src/LVREV_ApplyNewSettings.cpp b/media/libeffects/lvm/lib/Reverb/src/LVREV_ApplyNewSettings.cpp
new file mode 100644
index 0000000..1f0d39b
--- /dev/null
+++ b/media/libeffects/lvm/lib/Reverb/src/LVREV_ApplyNewSettings.cpp
@@ -0,0 +1,633 @@
+/*
+ * Copyright (C) 2004-2010 NXP Software
+ * Copyright (C) 2010 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+/****************************************************************************************/
+/* */
+/* Includes */
+/* */
+/****************************************************************************************/
+#include "LVREV_Private.h"
+#include "Filter.h"
+
+/****************************************************************************************/
+/* */
+/* FUNCTION: LVREV_ApplyNewSettings */
+/* */
+/* DESCRIPTION: */
+/* Applies the new control parameters */
+/* */
+/* PARAMETERS: */
+/* pPrivate Pointer to the instance private parameters */
+/* */
+/* RETURNS: */
+/* LVREV_Success Succeeded */
+/* LVREV_NULLADDRESS When pPrivate is NULL */
+/* */
+/* NOTES: */
+/* */
+/****************************************************************************************/
+
+LVREV_ReturnStatus_en LVREV_ApplyNewSettings (LVREV_Instance_st *pPrivate)
+{
+
+ LVM_Mode_en OperatingMode;
+ LVM_INT32 NumberOfDelayLines;
+
+ /* Check for NULL pointer */
+ if(pPrivate == LVM_NULL)
+ {
+ return LVREV_NULLADDRESS;
+ }
+
+ OperatingMode = pPrivate->NewParams.OperatingMode;
+
+ if(pPrivate->InstanceParams.NumDelays == LVREV_DELAYLINES_4)
+ {
+ NumberOfDelayLines = 4;
+ }
+ else if(pPrivate->InstanceParams.NumDelays == LVREV_DELAYLINES_2)
+ {
+ NumberOfDelayLines = 2;
+ }
+ else
+ {
+ NumberOfDelayLines = 1;
+ }
+
+ /*
+ * Update the high pass filter coefficients
+ */
+ if((pPrivate->NewParams.HPF != pPrivate->CurrentParams.HPF) ||
+ (pPrivate->NewParams.SampleRate != pPrivate->CurrentParams.SampleRate) ||
+ (pPrivate->bFirstControl == LVM_TRUE))
+ {
+ LVM_FLOAT Omega;
+ FO_FLOAT_Coefs_t Coeffs;
+
+ Omega = LVM_GetOmega(pPrivate->NewParams.HPF, pPrivate->NewParams.SampleRate);
+ LVM_FO_HPF(Omega, &Coeffs);
+ FO_1I_D32F32Cll_TRC_WRA_01_Init( &pPrivate->pFastCoef->HPCoefs,
+ &pPrivate->pFastData->HPTaps, &Coeffs);
+ LoadConst_Float(0,
+ (LVM_FLOAT *)&pPrivate->pFastData->HPTaps,
+ sizeof(Biquad_1I_Order1_FLOAT_Taps_t) / sizeof(LVM_FLOAT));
+ }
+
+ /*
+ * Update the low pass filter coefficients
+ */
+ if((pPrivate->NewParams.LPF != pPrivate->CurrentParams.LPF) ||
+ (pPrivate->NewParams.SampleRate != pPrivate->CurrentParams.SampleRate) ||
+ (pPrivate->bFirstControl == LVM_TRUE))
+ {
+ LVM_FLOAT Omega;
+ FO_FLOAT_Coefs_t Coeffs;
+
+ Coeffs.A0 = 1;
+ Coeffs.A1 = 0;
+ Coeffs.B1 = 0;
+ if(pPrivate->NewParams.LPF <= (LVM_FsTable[pPrivate->NewParams.SampleRate] >> 1))
+ {
+ Omega = LVM_GetOmega(pPrivate->NewParams.LPF, pPrivate->NewParams.SampleRate);
+
+ /*
+ * Do not apply filter if w =2*pi*fc/fs >= 2.9
+ */
+ if(Omega <= (LVM_FLOAT)LVREV_2_9_INQ29)
+ {
+ LVM_FO_LPF(Omega, &Coeffs);
+ }
+ }
+ FO_1I_D32F32Cll_TRC_WRA_01_Init( &pPrivate->pFastCoef->LPCoefs,
+ &pPrivate->pFastData->LPTaps, &Coeffs);
+ LoadConst_Float(0,
+ (LVM_FLOAT *)&pPrivate->pFastData->LPTaps,
+ sizeof(Biquad_1I_Order1_FLOAT_Taps_t) / sizeof(LVM_FLOAT));
+ }
+
+ /*
+ * Calculate the room size parameter
+ */
+ if( pPrivate->NewParams.RoomSize != pPrivate->CurrentParams.RoomSize)
+ {
+ /* Room size range is 10ms to 200ms
+ * 0% -- 10ms
+ * 50% -- 65ms
+ * 100% -- 120ms
+ */
+ pPrivate->RoomSizeInms = 10 + (((pPrivate->NewParams.RoomSize*11) + 5) / 10);
+ }
+
+ /*
+ * Update the T delay number of samples and the all pass delay number of samples
+ */
+ if( (pPrivate->NewParams.RoomSize != pPrivate->CurrentParams.RoomSize) ||
+ (pPrivate->NewParams.SampleRate != pPrivate->CurrentParams.SampleRate) ||
+ (pPrivate->bFirstControl == LVM_TRUE))
+ {
+
+ LVM_UINT32 Temp;
+ LVM_INT32 APDelaySize;
+ LVM_INT32 Fs = LVM_GetFsFromTable(pPrivate->NewParams.SampleRate);
+ LVM_UINT32 DelayLengthSamples = (LVM_UINT32)(Fs * pPrivate->RoomSizeInms);
+ LVM_INT16 i;
+ LVM_FLOAT ScaleTable[] = {LVREV_T_3_Power_minus0_on_4, LVREV_T_3_Power_minus1_on_4, \
+ LVREV_T_3_Power_minus2_on_4, LVREV_T_3_Power_minus3_on_4};
+ LVM_INT16 MaxT_Delay[] = {LVREV_MAX_T0_DELAY, LVREV_MAX_T1_DELAY, \
+ LVREV_MAX_T2_DELAY, LVREV_MAX_T3_DELAY};
+ LVM_INT16 MaxAP_Delay[] = {LVREV_MAX_AP0_DELAY, LVREV_MAX_AP1_DELAY, \
+ LVREV_MAX_AP2_DELAY, LVREV_MAX_AP3_DELAY};
+
+ /*
+ * For each delay line
+ */
+ for (i = 0; i < NumberOfDelayLines; i++)
+ {
+ if (i != 0)
+ {
+ LVM_FLOAT Temp1; /* to avoid QAC warning on type conversion */
+
+ Temp1=(LVM_FLOAT)DelayLengthSamples;
+ Temp = (LVM_UINT32)(Temp1 * ScaleTable[i]);
+ }
+ else
+ {
+ Temp = DelayLengthSamples;
+ }
+ APDelaySize = Temp / 1500;
+
+ /*
+ * Set the fixed delay
+ */
+
+ Temp = (MaxT_Delay[i] - MaxAP_Delay[i]) * Fs / 192000;
+ pPrivate->Delay_AP[i] = pPrivate->T[i] - Temp;
+
+ /*
+ * Set the tap selection
+ */
+ if (pPrivate->AB_Selection)
+ {
+ /* Smooth from tap A to tap B */
+ pPrivate->pOffsetB[i] = &pPrivate->pDelay_T[i][pPrivate->T[i] - \
+ Temp - APDelaySize];
+ pPrivate->B_DelaySize[i] = APDelaySize;
+ pPrivate->Mixer_APTaps[i].Target1 = 0;
+ pPrivate->Mixer_APTaps[i].Target2 = 1.0f;
+ }
+ else
+ {
+ /* Smooth from tap B to tap A */
+ pPrivate->pOffsetA[i] = &pPrivate->pDelay_T[i][pPrivate->T[i] - \
+ Temp - APDelaySize];
+ pPrivate->A_DelaySize[i] = APDelaySize;
+ pPrivate->Mixer_APTaps[i].Target2 = 0;
+ pPrivate->Mixer_APTaps[i].Target1 = 1.0f;
+ }
+
+ /*
+ * Set the maximum block size to the smallest delay size
+ */
+ pPrivate->MaxBlkLen = Temp;
+ if (pPrivate->MaxBlkLen > pPrivate->A_DelaySize[i])
+ {
+ pPrivate->MaxBlkLen = pPrivate->A_DelaySize[i];
+ }
+ if (pPrivate->MaxBlkLen > pPrivate->B_DelaySize[i])
+ {
+ pPrivate->MaxBlkLen = pPrivate->B_DelaySize[i];
+ }
+ }
+ if (pPrivate->AB_Selection)
+ {
+ pPrivate->AB_Selection = 0;
+ }
+ else
+ {
+ pPrivate->AB_Selection = 1;
+ }
+
+ /*
+ * Limit the maximum block length
+ */
+ /* Just as a precausion, but no problem if we remove this line */
+ pPrivate->MaxBlkLen = pPrivate->MaxBlkLen - 2;
+ if(pPrivate->MaxBlkLen > pPrivate->InstanceParams.MaxBlockSize)
+ {
+ pPrivate->MaxBlkLen = (LVM_INT32)pPrivate->InstanceParams.MaxBlockSize;
+ }
+ }
+
+ /*
+ * Update the low pass filter coefficient
+ */
+ if( (pPrivate->NewParams.Damping != pPrivate->CurrentParams.Damping) ||
+ (pPrivate->NewParams.SampleRate != pPrivate->CurrentParams.SampleRate) ||
+ (pPrivate->bFirstControl == LVM_TRUE))
+ {
+
+ LVM_INT32 Temp;
+ LVM_FLOAT Omega;
+ FO_FLOAT_Coefs_t Coeffs;
+ LVM_INT16 i;
+ LVM_INT16 Damping = (LVM_INT16)((pPrivate->NewParams.Damping * 100) + 1000);
+ LVM_FLOAT ScaleTable[] = {LVREV_T_3_Power_0_on_4, LVREV_T_3_Power_1_on_4,
+ LVREV_T_3_Power_2_on_4, LVREV_T_3_Power_3_on_4};
+
+ /*
+ * For each filter
+ */
+ for (i = 0; i < NumberOfDelayLines; i++)
+ {
+ if (i != 0)
+ {
+ Temp = (LVM_INT32)(ScaleTable[i] * Damping);
+ }
+ else
+ {
+ Temp = Damping;
+ }
+ if(Temp <= (LVM_INT32)(LVM_FsTable[pPrivate->NewParams.SampleRate] >> 1))
+ {
+ Omega = LVM_GetOmega(Temp, pPrivate->NewParams.SampleRate);
+ LVM_FO_LPF(Omega, &Coeffs);
+ }
+ else
+ {
+ Coeffs.A0 = 1;
+ Coeffs.A1 = 0;
+ Coeffs.B1 = 0;
+ }
+ FO_1I_D32F32Cll_TRC_WRA_01_Init(&pPrivate->pFastCoef->RevLPCoefs[i],
+ &pPrivate->pFastData->RevLPTaps[i], &Coeffs);
+ }
+ }
+
+ /*
+ * Update All-pass filter mixer time constants
+ */
+ if( (pPrivate->NewParams.RoomSize != pPrivate->CurrentParams.RoomSize) ||
+ (pPrivate->NewParams.SampleRate != pPrivate->CurrentParams.SampleRate) ||
+ (pPrivate->NewParams.Density != pPrivate->CurrentParams.Density))
+ {
+ LVM_INT16 i;
+ LVM_FLOAT Alpha;
+ LVM_FLOAT AlphaTap;
+
+ Alpha = LVM_Mixer_TimeConstant(LVREV_ALLPASS_TC,
+ LVM_GetFsFromTable(pPrivate->NewParams.SampleRate),
+ 1);
+
+ AlphaTap = LVM_Mixer_TimeConstant(LVREV_ALLPASS_TAP_TC,
+ LVM_GetFsFromTable(pPrivate->NewParams.SampleRate),
+ 1);
+
+ for (i = 0; i < 4; i++)
+ {
+ pPrivate->Mixer_APTaps[i].Alpha1 = AlphaTap;
+ pPrivate->Mixer_APTaps[i].Alpha2 = AlphaTap;
+ pPrivate->Mixer_SGFeedback[i].Alpha = Alpha;
+ pPrivate->Mixer_SGFeedforward[i].Alpha = Alpha;
+ }
+ }
+
+ /*
+ * Update the feed back gain
+ */
+ if( (pPrivate->NewParams.RoomSize != pPrivate->CurrentParams.RoomSize) ||
+ (pPrivate->NewParams.SampleRate != pPrivate->CurrentParams.SampleRate) ||
+ (pPrivate->NewParams.T60 != pPrivate->CurrentParams.T60) ||
+ (pPrivate->bFirstControl == LVM_TRUE))
+ {
+
+ LVM_FLOAT G[4]; /* Feedback gain (Q7.24) */
+
+ if(pPrivate->NewParams.T60 == 0)
+ {
+ G[3] = 0;
+ G[2] = 0;
+ G[1] = 0;
+ G[0] = 0;
+ }
+ else
+ {
+ LVM_FLOAT Temp1;
+ LVM_FLOAT Temp2;
+ LVM_INT16 i;
+ LVM_FLOAT ScaleTable[] = {LVREV_T_3_Power_minus0_on_4, LVREV_T_3_Power_minus1_on_4,
+ LVREV_T_3_Power_minus2_on_4, LVREV_T_3_Power_minus3_on_4};
+
+ /*
+ * For each delay line
+ */
+ for (i = 0; i < NumberOfDelayLines; i++)
+ {
+ Temp1 = (3 * pPrivate->RoomSizeInms * ScaleTable[i]) / pPrivate->NewParams.T60;
+ if(Temp1 >= (4))
+ {
+ G[i] = 0;
+ }
+ else if((Temp1 >= (2)))
+ {
+ Temp2 = LVM_Power10(-(Temp1 / 2.0f));
+ Temp1 = LVM_Power10(-(Temp1 / 2.0f));
+ Temp1 = Temp1 * Temp2;
+ }
+ else
+ {
+ Temp1 = LVM_Power10(-(Temp1));
+ }
+ if (NumberOfDelayLines == 1)
+ {
+ G[i] = Temp1;
+ }
+ else
+ {
+ LVM_FLOAT TempG;
+ TempG = Temp1 * ONE_OVER_SQRT_TWO;
+ G[i]=TempG;
+ }
+ }
+ }
+
+ /* Set up the feedback mixers for four delay lines */
+ pPrivate->FeedbackMixer[0].Target=G[0];
+ pPrivate->FeedbackMixer[1].Target=G[1];
+ pPrivate->FeedbackMixer[2].Target=G[2];
+ pPrivate->FeedbackMixer[3].Target=G[3];
+ }
+
+ /*
+ * Calculate the gain correction
+ */
+ if((pPrivate->NewParams.RoomSize != pPrivate->CurrentParams.RoomSize) ||
+ (pPrivate->NewParams.Level != pPrivate->CurrentParams.Level) ||
+ (pPrivate->NewParams.T60 != pPrivate->CurrentParams.T60) )
+ {
+ LVM_INT32 Index=0;
+ LVM_FLOAT Index_FLOAT;
+ LVM_INT32 i=0;
+ LVM_FLOAT Gain=0;
+ LVM_INT32 RoomSize=0;
+ LVM_FLOAT T60;
+ LVM_FLOAT Coefs[5];
+
+ if(pPrivate->NewParams.RoomSize == 0)
+ {
+ RoomSize = 1;
+ }
+ else
+ {
+ RoomSize = (LVM_INT32)pPrivate->NewParams.RoomSize;
+ }
+
+ if(pPrivate->NewParams.T60 < 100)
+ {
+ T60 = 100 * LVREV_T60_SCALE;
+ }
+ else
+ {
+ T60 = pPrivate->NewParams.T60 * LVREV_T60_SCALE;
+ }
+
+ /* Find the nearest room size in table */
+ for(i = 0; i < 24; i++)
+ {
+ if(RoomSize <= LVREV_GainPolyTable[i][0])
+ {
+ Index = i;
+ break;
+ }
+ }
+
+ if(RoomSize == LVREV_GainPolyTable[Index][0])
+ {
+ /* Take table values if the room size is in table */
+ for(i = 1; i < 5; i++)
+ {
+ Coefs[i-1] = LVREV_GainPolyTable[Index][i];
+ }
+ Coefs[4] = 0;
+ Gain = LVM_Polynomial(3, Coefs, T60); /* Q.24 result */
+ }
+ else
+ {
+ /* Interpolate the gain between nearest room sizes */
+
+ LVM_FLOAT Gain1,Gain2;
+ LVM_INT32 Tot_Dist,Dist;
+
+ Tot_Dist = (LVM_UINT32)LVREV_GainPolyTable[Index][0] - \
+ (LVM_UINT32)LVREV_GainPolyTable[Index-1][0];
+ Dist = RoomSize - (LVM_UINT32)LVREV_GainPolyTable[Index - 1][0];
+
+ /* Get gain for first */
+ for(i = 1; i < 5; i++)
+ {
+ Coefs[i-1] = LVREV_GainPolyTable[Index-1][i];
+ }
+ Coefs[4] = 0;
+
+ Gain1 = LVM_Polynomial(3, Coefs, T60); /* Q.24 result */
+
+ /* Get gain for second */
+ for(i = 1; i < 5; i++)
+ {
+ Coefs[i-1] = LVREV_GainPolyTable[Index][i];
+ }
+ Coefs[4] = 0;
+
+ Gain2 = LVM_Polynomial(3, Coefs, T60); /* Q.24 result */
+
+ /* Linear Interpolate the gain */
+ Gain = Gain1 + (((Gain2 - Gain1) * Dist) / (Tot_Dist));
+ }
+
+ /*
+ * Get the inverse of gain: Q.15
+ * Gain is mostly above one except few cases, take only gains above 1
+ */
+ if(Gain < 1)
+ {
+ pPrivate->Gain = 1;
+ }
+ else
+ {
+ pPrivate->Gain = 1 / Gain;
+ }
+
+ Index_FLOAT = 100.0f / (LVM_FLOAT)(100 + pPrivate->NewParams.Level);
+ pPrivate->Gain = pPrivate->Gain * Index_FLOAT;
+ pPrivate->GainMixer.Target = (pPrivate->Gain*Index_FLOAT) / 2;
+ }
+
+ /*
+ * Update the all pass comb filter coefficient
+ */
+ if( (pPrivate->NewParams.Density != pPrivate->CurrentParams.Density) ||
+ (pPrivate->bFirstControl == LVM_TRUE))
+ {
+ LVM_INT16 i;
+ LVM_FLOAT b = (LVM_FLOAT)pPrivate->NewParams.Density * LVREV_B_8_on_1000;
+
+ for (i = 0; i < 4; i++)
+ {
+ pPrivate->Mixer_SGFeedback[i].Target = b;
+ pPrivate->Mixer_SGFeedforward[i].Target = b;
+ }
+ }
+
+ /*
+ * Update the bypass mixer time constant
+ */
+ if((pPrivate->NewParams.SampleRate != pPrivate->CurrentParams.SampleRate) ||
+ (pPrivate->bFirstControl == LVM_TRUE))
+ {
+ LVM_UINT16 NumChannels = 1; /* Assume MONO format */
+ LVM_FLOAT Alpha;
+
+ Alpha = LVM_Mixer_TimeConstant(LVREV_FEEDBACKMIXER_TC,
+ LVM_GetFsFromTable(pPrivate->NewParams.SampleRate),
+ NumChannels);
+ pPrivate->FeedbackMixer[0].Alpha = Alpha;
+ pPrivate->FeedbackMixer[1].Alpha = Alpha;
+ pPrivate->FeedbackMixer[2].Alpha = Alpha;
+ pPrivate->FeedbackMixer[3].Alpha = Alpha;
+
+ NumChannels = 2; /* Always stereo output */
+ pPrivate->BypassMixer.Alpha1 = LVM_Mixer_TimeConstant(LVREV_BYPASSMIXER_TC,
+ LVM_GetFsFromTable(pPrivate->NewParams.SampleRate), NumChannels);
+ pPrivate->BypassMixer.Alpha2 = pPrivate->BypassMixer.Alpha1;
+ pPrivate->GainMixer.Alpha = pPrivate->BypassMixer.Alpha1;
+ }
+
+ /*
+ * Update the bypass mixer targets
+ */
+ if( (pPrivate->NewParams.Level != pPrivate->CurrentParams.Level) &&
+ (pPrivate->NewParams.OperatingMode == LVM_MODE_ON))
+ {
+ pPrivate->BypassMixer.Target2 = (LVM_FLOAT)(pPrivate->NewParams.Level ) / 100.0f;
+ pPrivate->BypassMixer.Target1 = 0x00000000;
+ if ((pPrivate->NewParams.Level == 0) && (pPrivate->bFirstControl == LVM_FALSE))
+ {
+ pPrivate->BypassMixer.CallbackSet2 = LVM_TRUE;
+ }
+ if (pPrivate->NewParams.Level != 0)
+ {
+ pPrivate->bDisableReverb = LVM_FALSE;
+ }
+ }
+
+ if(pPrivate->NewParams.OperatingMode != pPrivate->CurrentParams.OperatingMode)
+ {
+ if(pPrivate->NewParams.OperatingMode == LVM_MODE_ON)
+ {
+ pPrivate->BypassMixer.Target2 = (LVM_FLOAT)(pPrivate->NewParams.Level ) / 100.0f;
+ pPrivate->BypassMixer.Target1 = 0x00000000;
+
+ pPrivate->BypassMixer.CallbackSet2 = LVM_FALSE;
+ OperatingMode = LVM_MODE_ON;
+ if (pPrivate->NewParams.Level == 0)
+ {
+ pPrivate->bDisableReverb = LVM_TRUE;
+ }
+ else
+ {
+ pPrivate->bDisableReverb = LVM_FALSE;
+ }
+ }
+ else if (pPrivate->bFirstControl == LVM_FALSE)
+ {
+ pPrivate->BypassMixer.Target2 = 0x00000000;
+ pPrivate->BypassMixer.Target1 = 0x00000000;
+ pPrivate->BypassMixer.CallbackSet2 = LVM_TRUE;
+ pPrivate->GainMixer.Target = 0.03125f;
+ OperatingMode = LVM_MODE_ON;
+ }
+ else
+ {
+ OperatingMode = LVM_MODE_OFF;
+ }
+ }
+
+ /* If it is the first call to ApplyNew settings force the current to the target \
+ to begin immediate playback of the effect */
+ if(pPrivate->bFirstControl == LVM_TRUE)
+ {
+ pPrivate->BypassMixer.Current1 = pPrivate->BypassMixer.Target1;
+ pPrivate->BypassMixer.Current2 = pPrivate->BypassMixer.Target2;
+ }
+
+ /*
+ * Copy the new parameters
+ */
+ pPrivate->CurrentParams = pPrivate->NewParams;
+ pPrivate->CurrentParams.OperatingMode = OperatingMode;
+
+ /*
+ * Update flag
+ */
+ if(pPrivate->bFirstControl == LVM_TRUE)
+ {
+ pPrivate->bFirstControl = LVM_FALSE;
+ }
+
+ return LVREV_SUCCESS;
+}
+/****************************************************************************************/
+/* */
+/* FUNCTION: BypassMixer_Callback */
+/* */
+/* DESCRIPTION: */
+/* Controls the On to Off operating mode transition */
+/* */
+/* PARAMETERS: */
+/* pPrivate Pointer to the instance private parameters */
+/* */
+/* RETURNS: */
+/* LVREV_Success Succeeded */
+/* LVREV_NULLADDRESS When pPrivate is NULL */
+/* */
+/* NOTES: */
+/* */
+/****************************************************************************************/
+LVM_INT32 BypassMixer_Callback (void *pCallbackData,
+ void *pGeneralPurpose,
+ LVM_INT16 GeneralPurpose )
+{
+
+ LVREV_Instance_st *pLVREV_Private = (LVREV_Instance_st *)pCallbackData;
+
+ /*
+ * Avoid build warnings
+ */
+ (void)pGeneralPurpose;
+ (void)GeneralPurpose;
+
+ /*
+ * Turn off
+ */
+ pLVREV_Private->CurrentParams.OperatingMode = LVM_MODE_OFF;
+ pLVREV_Private->bDisableReverb = LVM_TRUE;
+ LVREV_ClearAudioBuffers((LVREV_Handle_t)pCallbackData);
+
+ return 0;
+}
+
+/* End of file */
+
diff --git a/media/libeffects/lvm/lib/Reverb/src/LVREV_ClearAudioBuffers.c b/media/libeffects/lvm/lib/Reverb/src/LVREV_ClearAudioBuffers.cpp
similarity index 70%
rename from media/libeffects/lvm/lib/Reverb/src/LVREV_ClearAudioBuffers.c
rename to media/libeffects/lvm/lib/Reverb/src/LVREV_ClearAudioBuffers.cpp
index 9491016..586539f 100644
--- a/media/libeffects/lvm/lib/Reverb/src/LVREV_ClearAudioBuffers.c
+++ b/media/libeffects/lvm/lib/Reverb/src/LVREV_ClearAudioBuffers.cpp
@@ -23,7 +23,6 @@
#include "LVREV_Private.h"
#include "VectorArithmetic.h"
-
/****************************************************************************************/
/* */
/* FUNCTION: LVREV_ClearAudioBuffers */
@@ -47,7 +46,6 @@
LVREV_Instance_st *pLVREV_Private = (LVREV_Instance_st *)hInstance;
-
/*
* Check for error conditions
*/
@@ -61,36 +59,14 @@
* Clear all filter tap data, delay-lines and other signal related data
*/
-#ifdef BUILD_FLOAT
LoadConst_Float(0,
- (void *)&pLVREV_Private->pFastData->HPTaps, /* Destination Cast to void: \
- no dereferencing in function*/
- 2);
+ (LVM_FLOAT *)&pLVREV_Private->pFastData->HPTaps,
+ 2);
LoadConst_Float(0,
- (void *)&pLVREV_Private->pFastData->LPTaps, /* Destination Cast to void: \
- no dereferencing in function*/
- 2);
-#else
- LoadConst_32(0,
- (void *)&pLVREV_Private->pFastData->HPTaps, /* Destination Cast to void: no dereferencing in function*/
+ (LVM_FLOAT *)&pLVREV_Private->pFastData->LPTaps,
2);
- LoadConst_32(0,
- (void *)&pLVREV_Private->pFastData->LPTaps, /* Destination Cast to void: no dereferencing in function*/
- 2);
-#endif
if((LVM_UINT16)pLVREV_Private->InstanceParams.NumDelays == LVREV_DELAYLINES_4)
{
-#ifndef BUILD_FLOAT
- LoadConst_32(0, (LVM_INT32 *)&pLVREV_Private->pFastData->RevLPTaps[3], 2);
- LoadConst_32(0, (LVM_INT32 *)&pLVREV_Private->pFastData->RevLPTaps[2], 2);
- LoadConst_32(0, (LVM_INT32 *)&pLVREV_Private->pFastData->RevLPTaps[1], 2);
- LoadConst_32(0, (LVM_INT32 *)&pLVREV_Private->pFastData->RevLPTaps[0], 2);
-
- LoadConst_32(0,pLVREV_Private->pDelay_T[3], (LVM_INT16)LVREV_MAX_T3_DELAY);
- LoadConst_32(0,pLVREV_Private->pDelay_T[2], (LVM_INT16)LVREV_MAX_T2_DELAY);
- LoadConst_32(0,pLVREV_Private->pDelay_T[1], (LVM_INT16)LVREV_MAX_T1_DELAY);
- LoadConst_32(0,pLVREV_Private->pDelay_T[0], (LVM_INT16)LVREV_MAX_T0_DELAY);
-#else
LoadConst_Float(0, (LVM_FLOAT *)&pLVREV_Private->pFastData->RevLPTaps[3], 2);
LoadConst_Float(0, (LVM_FLOAT *)&pLVREV_Private->pFastData->RevLPTaps[2], 2);
LoadConst_Float(0, (LVM_FLOAT *)&pLVREV_Private->pFastData->RevLPTaps[1], 2);
@@ -100,35 +76,21 @@
LoadConst_Float(0, pLVREV_Private->pDelay_T[2], LVREV_MAX_T2_DELAY);
LoadConst_Float(0, pLVREV_Private->pDelay_T[1], LVREV_MAX_T1_DELAY);
LoadConst_Float(0, pLVREV_Private->pDelay_T[0], LVREV_MAX_T0_DELAY);
-#endif
}
if((LVM_UINT16)pLVREV_Private->InstanceParams.NumDelays >= LVREV_DELAYLINES_2)
{
-#ifndef BUILD_FLOAT
- LoadConst_32(0, (LVM_INT32 *)&pLVREV_Private->pFastData->RevLPTaps[1], 2);
- LoadConst_32(0, (LVM_INT32 *)&pLVREV_Private->pFastData->RevLPTaps[0], 2);
-
- LoadConst_32(0,pLVREV_Private->pDelay_T[1], (LVM_INT16)LVREV_MAX_T1_DELAY);
- LoadConst_32(0,pLVREV_Private->pDelay_T[0], (LVM_INT16)LVREV_MAX_T0_DELAY);
-#else
LoadConst_Float(0, (LVM_FLOAT *)&pLVREV_Private->pFastData->RevLPTaps[1], 2);
LoadConst_Float(0, (LVM_FLOAT *)&pLVREV_Private->pFastData->RevLPTaps[0], 2);
LoadConst_Float(0, pLVREV_Private->pDelay_T[1], LVREV_MAX_T1_DELAY);
LoadConst_Float(0, pLVREV_Private->pDelay_T[0], LVREV_MAX_T0_DELAY);
-#endif
}
if((LVM_UINT16)pLVREV_Private->InstanceParams.NumDelays >= LVREV_DELAYLINES_1)
{
-#ifndef BUILD_FLOAT
- LoadConst_32(0, (LVM_INT32 *)&pLVREV_Private->pFastData->RevLPTaps[0], 2);
- LoadConst_32(0,pLVREV_Private->pDelay_T[0], (LVM_INT16)LVREV_MAX_T0_DELAY);
-#else
LoadConst_Float(0, (LVM_FLOAT *)&pLVREV_Private->pFastData->RevLPTaps[0], 2);
LoadConst_Float(0, pLVREV_Private->pDelay_T[0], LVREV_MAX_T0_DELAY);
-#endif
}
return LVREV_SUCCESS;
}
diff --git a/media/libeffects/lvm/lib/Reverb/src/LVREV_GetControlParameters.c b/media/libeffects/lvm/lib/Reverb/src/LVREV_GetControlParameters.cpp
similarity index 99%
rename from media/libeffects/lvm/lib/Reverb/src/LVREV_GetControlParameters.c
rename to media/libeffects/lvm/lib/Reverb/src/LVREV_GetControlParameters.cpp
index 7cee26d..e0b0142 100644
--- a/media/libeffects/lvm/lib/Reverb/src/LVREV_GetControlParameters.c
+++ b/media/libeffects/lvm/lib/Reverb/src/LVREV_GetControlParameters.cpp
@@ -22,7 +22,6 @@
/****************************************************************************************/
#include "LVREV_Private.h"
-
/****************************************************************************************/
/* */
/* FUNCTION: LVREV_GetControlParameters */
@@ -49,7 +48,6 @@
LVREV_Instance_st *pLVREV_Private = (LVREV_Instance_st *)hInstance;
-
/*
* Check for error conditions
*/
diff --git a/media/libeffects/lvm/lib/Reverb/src/LVREV_GetInstanceHandle.c b/media/libeffects/lvm/lib/Reverb/src/LVREV_GetInstanceHandle.cpp
similarity index 74%
rename from media/libeffects/lvm/lib/Reverb/src/LVREV_GetInstanceHandle.c
rename to media/libeffects/lvm/lib/Reverb/src/LVREV_GetInstanceHandle.cpp
index 3366bcb..68f883a 100644
--- a/media/libeffects/lvm/lib/Reverb/src/LVREV_GetInstanceHandle.c
+++ b/media/libeffects/lvm/lib/Reverb/src/LVREV_GetInstanceHandle.cpp
@@ -23,7 +23,6 @@
#include "LVREV_Private.h"
#include "InstAlloc.h"
-
/****************************************************************************************/
/* */
/* FUNCTION: LVREV_GetInstanceHandle */
@@ -59,7 +58,6 @@
LVM_INT16 i;
LVM_UINT16 MaxBlockSize;
-
/*
* Check for error conditions
*/
@@ -108,7 +106,6 @@
/*
* Zero all memory regions
*/
-#ifdef BUILD_FLOAT
LoadConst_Float(0,
(LVM_FLOAT *)pMemoryTable->Region[LVM_PERSISTENT_SLOW_DATA].pBaseAddress,
(LVM_INT16)((pMemoryTable->Region[LVM_PERSISTENT_SLOW_DATA].Size) / \
@@ -125,12 +122,6 @@
(LVM_FLOAT *)pMemoryTable->Region[LVM_TEMPORARY_FAST].pBaseAddress,
(LVM_INT16)((pMemoryTable->Region[LVM_TEMPORARY_FAST].Size) / \
sizeof(LVM_FLOAT)));
-#else
- LoadConst_16(0, (LVM_INT16 *)pMemoryTable->Region[LVM_PERSISTENT_SLOW_DATA].pBaseAddress, (LVM_INT16)((pMemoryTable->Region[LVM_PERSISTENT_SLOW_DATA].Size)/sizeof(LVM_INT16)));
- LoadConst_16(0, (LVM_INT16 *)pMemoryTable->Region[LVM_PERSISTENT_FAST_DATA].pBaseAddress, (LVM_INT16)((pMemoryTable->Region[LVM_PERSISTENT_FAST_DATA].Size)/sizeof(LVM_INT16)));
- LoadConst_16(0, (LVM_INT16 *)pMemoryTable->Region[LVM_PERSISTENT_FAST_COEF].pBaseAddress, (LVM_INT16)((pMemoryTable->Region[LVM_PERSISTENT_FAST_COEF].Size)/sizeof(LVM_INT16)));
- LoadConst_16(0, (LVM_INT16 *)pMemoryTable->Region[LVM_TEMPORARY_FAST].pBaseAddress, (LVM_INT16)((pMemoryTable->Region[LVM_TEMPORARY_FAST].Size)/sizeof(LVM_INT16)));
-#endif
/*
* Set the instance handle if not already initialised
*/
@@ -159,71 +150,31 @@
MaxBlockSize=pInstanceParams->MaxBlockSize;
}
-
/*
* Set the data, coefficient and temporary memory pointers
*/
- pLVREV_Private->pFastData = InstAlloc_AddMember(&FastData, sizeof(LVREV_FastData_st)); /* Fast data memory base address */
-#ifndef BUILD_FLOAT
+ /* Fast data memory base address */
+ pLVREV_Private->pFastData = (LVREV_FastData_st *)
+ InstAlloc_AddMember(&FastData, sizeof(LVREV_FastData_st));
if(pInstanceParams->NumDelays == LVREV_DELAYLINES_4)
{
- pLVREV_Private->pDelay_T[3] = InstAlloc_AddMember(&FastData, LVREV_MAX_T3_DELAY * sizeof(LVM_INT32));
- pLVREV_Private->pDelay_T[2] = InstAlloc_AddMember(&FastData, LVREV_MAX_T2_DELAY * sizeof(LVM_INT32));
- pLVREV_Private->pDelay_T[1] = InstAlloc_AddMember(&FastData, LVREV_MAX_T1_DELAY * sizeof(LVM_INT32));
- pLVREV_Private->pDelay_T[0] = InstAlloc_AddMember(&FastData, LVREV_MAX_T0_DELAY * sizeof(LVM_INT32));
-
- for( i = 0; i < 4; i++)
- {
- pLVREV_Private->pScratchDelayLine[i] = InstAlloc_AddMember(&Temporary, sizeof(LVM_INT32) * MaxBlockSize); /* Scratch for each delay line output */
- }
-
- LoadConst_32(0,pLVREV_Private->pDelay_T[3] ,(LVM_INT16)LVREV_MAX_T3_DELAY);
- LoadConst_32(0,pLVREV_Private->pDelay_T[2] ,(LVM_INT16)LVREV_MAX_T2_DELAY);
- LoadConst_32(0,pLVREV_Private->pDelay_T[1] ,(LVM_INT16)LVREV_MAX_T1_DELAY);
- LoadConst_32(0,pLVREV_Private->pDelay_T[0] ,(LVM_INT16)LVREV_MAX_T0_DELAY);
- }
-
- if(pInstanceParams->NumDelays == LVREV_DELAYLINES_2)
- {
- pLVREV_Private->pDelay_T[1] = InstAlloc_AddMember(&FastData, LVREV_MAX_T1_DELAY * sizeof(LVM_INT32));
- pLVREV_Private->pDelay_T[0] = InstAlloc_AddMember(&FastData, LVREV_MAX_T0_DELAY * sizeof(LVM_INT32));
-
- for( i = 0; i < 2; i++)
- {
- pLVREV_Private->pScratchDelayLine[i] = InstAlloc_AddMember(&Temporary, sizeof(LVM_INT32) * MaxBlockSize); /* Scratch for each delay line output */
- }
-
- LoadConst_32(0,pLVREV_Private->pDelay_T[1] , (LVM_INT16)LVREV_MAX_T1_DELAY);
- LoadConst_32(0,pLVREV_Private->pDelay_T[0] , (LVM_INT16)LVREV_MAX_T0_DELAY);
- }
-
- if(pInstanceParams->NumDelays == LVREV_DELAYLINES_1)
- {
- pLVREV_Private->pDelay_T[0] = InstAlloc_AddMember(&FastData, LVREV_MAX_T0_DELAY * sizeof(LVM_INT32));
-
- for( i = 0; i < 1; i++)
- {
- pLVREV_Private->pScratchDelayLine[i] = InstAlloc_AddMember(&Temporary, sizeof(LVM_INT32) * MaxBlockSize); /* Scratch for each delay line output */
- }
-
- LoadConst_32(0,pLVREV_Private->pDelay_T[0] , (LVM_INT16)LVREV_MAX_T0_DELAY);
- }
-#else
- if(pInstanceParams->NumDelays == LVREV_DELAYLINES_4)
- {
- pLVREV_Private->pDelay_T[3] = InstAlloc_AddMember(&FastData, LVREV_MAX_T3_DELAY * \
+ pLVREV_Private->pDelay_T[3] =
+ (LVM_FLOAT *)InstAlloc_AddMember(&FastData, LVREV_MAX_T3_DELAY * \
sizeof(LVM_FLOAT));
- pLVREV_Private->pDelay_T[2] = InstAlloc_AddMember(&FastData, LVREV_MAX_T2_DELAY * \
+ pLVREV_Private->pDelay_T[2] =
+ (LVM_FLOAT *)InstAlloc_AddMember(&FastData, LVREV_MAX_T2_DELAY * \
sizeof(LVM_FLOAT));
- pLVREV_Private->pDelay_T[1] = InstAlloc_AddMember(&FastData, LVREV_MAX_T1_DELAY * \
+ pLVREV_Private->pDelay_T[1] =
+ (LVM_FLOAT *)InstAlloc_AddMember(&FastData, LVREV_MAX_T1_DELAY * \
sizeof(LVM_FLOAT));
- pLVREV_Private->pDelay_T[0] = InstAlloc_AddMember(&FastData, LVREV_MAX_T0_DELAY * \
+ pLVREV_Private->pDelay_T[0] =
+ (LVM_FLOAT *)InstAlloc_AddMember(&FastData, LVREV_MAX_T0_DELAY * \
sizeof(LVM_FLOAT));
for(i = 0; i < 4; i++)
{
/* Scratch for each delay line output */
- pLVREV_Private->pScratchDelayLine[i] = InstAlloc_AddMember(&Temporary,
+ pLVREV_Private->pScratchDelayLine[i] = (LVM_FLOAT *)InstAlloc_AddMember(&Temporary,
sizeof(LVM_FLOAT) * \
MaxBlockSize);
}
@@ -236,15 +187,17 @@
if(pInstanceParams->NumDelays == LVREV_DELAYLINES_2)
{
- pLVREV_Private->pDelay_T[1] = InstAlloc_AddMember(&FastData, LVREV_MAX_T1_DELAY * \
+ pLVREV_Private->pDelay_T[1] = (LVM_FLOAT *)
+ InstAlloc_AddMember(&FastData, LVREV_MAX_T1_DELAY * \
sizeof(LVM_FLOAT));
- pLVREV_Private->pDelay_T[0] = InstAlloc_AddMember(&FastData, LVREV_MAX_T0_DELAY * \
+ pLVREV_Private->pDelay_T[0] = (LVM_FLOAT *)
+ InstAlloc_AddMember(&FastData, LVREV_MAX_T0_DELAY * \
sizeof(LVM_FLOAT));
for(i = 0; i < 2; i++)
{
/* Scratch for each delay line output */
- pLVREV_Private->pScratchDelayLine[i] = InstAlloc_AddMember(&Temporary,
+ pLVREV_Private->pScratchDelayLine[i] = (LVM_FLOAT *)InstAlloc_AddMember(&Temporary,
sizeof(LVM_FLOAT) * \
MaxBlockSize);
}
@@ -255,20 +208,19 @@
if(pInstanceParams->NumDelays == LVREV_DELAYLINES_1)
{
- pLVREV_Private->pDelay_T[0] = InstAlloc_AddMember(&FastData,
+ pLVREV_Private->pDelay_T[0] = (LVM_FLOAT *)InstAlloc_AddMember(&FastData,
LVREV_MAX_T0_DELAY * sizeof(LVM_FLOAT));
for(i = 0; i < 1; i++)
{
/* Scratch for each delay line output */
- pLVREV_Private->pScratchDelayLine[i] = InstAlloc_AddMember(&Temporary,
+ pLVREV_Private->pScratchDelayLine[i] = (LVM_FLOAT *)InstAlloc_AddMember(&Temporary,
sizeof(LVM_FLOAT) * \
MaxBlockSize);
}
LoadConst_Float(0, pLVREV_Private->pDelay_T[0], (LVM_INT16)LVREV_MAX_T0_DELAY);
}
-#endif
/* All-pass delay buffer addresses and sizes */
pLVREV_Private->T[0] = LVREV_MAX_T0_DELAY;
pLVREV_Private->T[1] = LVREV_MAX_T1_DELAY;
@@ -276,28 +228,24 @@
pLVREV_Private->T[3] = LVREV_MAX_T3_DELAY;
pLVREV_Private->AB_Selection = 1; /* Select smoothing A to B */
-
- pLVREV_Private->pFastCoef = InstAlloc_AddMember(&FastCoef, sizeof(LVREV_FastCoef_st)); /* Fast coefficient memory base address */
-#ifndef BUILD_FLOAT
- pLVREV_Private->pScratch = InstAlloc_AddMember(&Temporary, sizeof(LVM_INT32) * MaxBlockSize); /* General purpose scratch */
- pLVREV_Private->pInputSave = InstAlloc_AddMember(&Temporary, 2 * sizeof(LVM_INT32) * MaxBlockSize); /* Mono->stereo input save for end mix */
- LoadConst_32(0, pLVREV_Private->pInputSave, (LVM_INT16)(MaxBlockSize*2));
-#else
+ /* Fast coefficient memory base address */
+ pLVREV_Private->pFastCoef =
+ (LVREV_FastCoef_st *)InstAlloc_AddMember(&FastCoef, sizeof(LVREV_FastCoef_st));
/* General purpose scratch */
- pLVREV_Private->pScratch = InstAlloc_AddMember(&Temporary, sizeof(LVM_FLOAT) * \
+ pLVREV_Private->pScratch =
+ (LVM_FLOAT *)InstAlloc_AddMember(&Temporary, sizeof(LVM_FLOAT) * \
MaxBlockSize);
/* Mono->stereo input save for end mix */
- pLVREV_Private->pInputSave = InstAlloc_AddMember(&Temporary, 2 * sizeof(LVM_FLOAT) * \
+ pLVREV_Private->pInputSave =
+ (LVM_FLOAT *)InstAlloc_AddMember(&Temporary, 2 * sizeof(LVM_FLOAT) * \
MaxBlockSize);
LoadConst_Float(0, pLVREV_Private->pInputSave, (LVM_INT16)(MaxBlockSize * 2));
-#endif
/*
* Save the instance parameters in the instance structure
*/
pLVREV_Private->InstanceParams = *pInstanceParams;
-
/*
* Set the parameters to invalid
*/
@@ -309,7 +257,6 @@
pLVREV_Private->bFirstControl = LVM_TRUE;
pLVREV_Private->bDisableReverb = LVM_FALSE;
-
/*
* Set mixer parameters
*/
@@ -330,7 +277,6 @@
pLVREV_Private->RoomSizeInms = 100; // 100 msec
-
/*
* Set the output gain mixer parameters
*/
@@ -339,13 +285,8 @@
pLVREV_Private->GainMixer.pGeneralPurpose = LVM_NULL;
pLVREV_Private->GainMixer.pCallBack = LVM_NULL;
pLVREV_Private->GainMixer.CallbackSet = LVM_FALSE;
-#ifndef BUILD_FLOAT
- pLVREV_Private->GainMixer.Current = 0x03ffffff;
- pLVREV_Private->GainMixer.Target = 0x03ffffff;
-#else
pLVREV_Private->GainMixer.Current = 0.03125f;//0x03ffffff;
pLVREV_Private->GainMixer.Target = 0.03125f;//0x03ffffff;
-#endif
/*
* Set the All-Pass Filter mixers
@@ -368,11 +309,7 @@
pLVREV_Private->Mixer_APTaps[i].pCallBack1 = LVM_NULL;
pLVREV_Private->Mixer_APTaps[i].CallbackSet1 = LVM_FALSE;
pLVREV_Private->Mixer_APTaps[i].Current1 = 0;
-#ifndef BUILD_FLOAT
- pLVREV_Private->Mixer_APTaps[i].Target1 = 0x7fffffff;
-#else
pLVREV_Private->Mixer_APTaps[i].Target1 = 1;
-#endif
/* Feedforward mixer */
pLVREV_Private->Mixer_SGFeedforward[i].CallbackParam = 0;
pLVREV_Private->Mixer_SGFeedforward[i].pCallbackHandle = LVM_NULL;
@@ -408,7 +345,6 @@
pLVREV_Private->A_DelaySize[3] = LVREV_MAX_AP3_DELAY;
pLVREV_Private->B_DelaySize[3] = LVREV_MAX_AP3_DELAY;
-
LVREV_ClearAudioBuffers(*phInstance);
return LVREV_SUCCESS;
diff --git a/media/libeffects/lvm/lib/Reverb/src/LVREV_GetMemoryTable.c b/media/libeffects/lvm/lib/Reverb/src/LVREV_GetMemoryTable.cpp
similarity index 87%
rename from media/libeffects/lvm/lib/Reverb/src/LVREV_GetMemoryTable.c
rename to media/libeffects/lvm/lib/Reverb/src/LVREV_GetMemoryTable.cpp
index f6d446b..f59933c 100644
--- a/media/libeffects/lvm/lib/Reverb/src/LVREV_GetMemoryTable.c
+++ b/media/libeffects/lvm/lib/Reverb/src/LVREV_GetMemoryTable.cpp
@@ -68,7 +68,6 @@
LVM_INT16 i;
LVM_UINT16 MaxBlockSize;
-
/*
* Check for error conditions
*/
@@ -109,7 +108,6 @@
InstAlloc_Init(&FastCoef, (void *)LVM_NULL);
InstAlloc_Init(&Temporary, (void *)LVM_NULL);
-
/*
* Fill in the memory table
*/
@@ -123,7 +121,6 @@
return(LVREV_NULLADDRESS);
}
-
/*
* Select the maximum internal block size
*/
@@ -145,7 +142,6 @@
MaxBlockSize=pInstanceParams->MaxBlockSize;
}
-
/*
* Slow data memory
*/
@@ -154,51 +150,33 @@
pMemoryTable->Region[LVM_PERSISTENT_SLOW_DATA].Type = LVM_PERSISTENT_SLOW_DATA;
pMemoryTable->Region[LVM_PERSISTENT_SLOW_DATA].pBaseAddress = LVM_NULL;
-
/*
* Persistent fast data memory
*/
InstAlloc_AddMember(&FastData, sizeof(LVREV_FastData_st));
if(pInstanceParams->NumDelays == LVREV_DELAYLINES_4)
{
-#ifndef BUILD_FLOAT
- InstAlloc_AddMember(&FastData, LVREV_MAX_T3_DELAY * sizeof(LVM_INT32));
- InstAlloc_AddMember(&FastData, LVREV_MAX_T2_DELAY * sizeof(LVM_INT32));
- InstAlloc_AddMember(&FastData, LVREV_MAX_T1_DELAY * sizeof(LVM_INT32));
- InstAlloc_AddMember(&FastData, LVREV_MAX_T0_DELAY * sizeof(LVM_INT32));
-#else
InstAlloc_AddMember(&FastData, LVREV_MAX_T3_DELAY * sizeof(LVM_FLOAT));
InstAlloc_AddMember(&FastData, LVREV_MAX_T2_DELAY * sizeof(LVM_FLOAT));
InstAlloc_AddMember(&FastData, LVREV_MAX_T1_DELAY * sizeof(LVM_FLOAT));
InstAlloc_AddMember(&FastData, LVREV_MAX_T0_DELAY * sizeof(LVM_FLOAT));
-#endif
}
if(pInstanceParams->NumDelays == LVREV_DELAYLINES_2)
{
-#ifndef BUILD_FLOAT
- InstAlloc_AddMember(&FastData, LVREV_MAX_T1_DELAY * sizeof(LVM_INT32));
- InstAlloc_AddMember(&FastData, LVREV_MAX_T0_DELAY * sizeof(LVM_INT32));
-#else
InstAlloc_AddMember(&FastData, LVREV_MAX_T1_DELAY * sizeof(LVM_FLOAT));
InstAlloc_AddMember(&FastData, LVREV_MAX_T0_DELAY * sizeof(LVM_FLOAT));
-#endif
}
if(pInstanceParams->NumDelays == LVREV_DELAYLINES_1)
{
-#ifndef BUILD_FLOAT
- InstAlloc_AddMember(&FastData, LVREV_MAX_T0_DELAY * sizeof(LVM_INT32));
-#else
InstAlloc_AddMember(&FastData, LVREV_MAX_T0_DELAY * sizeof(LVM_FLOAT));
-#endif
}
pMemoryTable->Region[LVM_PERSISTENT_FAST_DATA].Size = InstAlloc_GetTotal(&FastData);
pMemoryTable->Region[LVM_PERSISTENT_FAST_DATA].Type = LVM_PERSISTENT_FAST_DATA;
pMemoryTable->Region[LVM_PERSISTENT_FAST_DATA].pBaseAddress = LVM_NULL;
-
/*
* Persistent fast coefficient memory
*/
@@ -207,29 +185,19 @@
pMemoryTable->Region[LVM_PERSISTENT_FAST_COEF].Type = LVM_PERSISTENT_FAST_COEF;
pMemoryTable->Region[LVM_PERSISTENT_FAST_COEF].pBaseAddress = LVM_NULL;
-
/*
* Temporary fast memory
*/
-#ifndef BUILD_FLOAT
- InstAlloc_AddMember(&Temporary, sizeof(LVM_INT32) * MaxBlockSize); /* General purpose scratch memory */
- InstAlloc_AddMember(&Temporary, 2*sizeof(LVM_INT32) * MaxBlockSize); /* Mono->stereo input saved for end mix */
-#else
/* General purpose scratch memory */
InstAlloc_AddMember(&Temporary, sizeof(LVM_FLOAT) * MaxBlockSize);
/* Mono->stereo input saved for end mix */
InstAlloc_AddMember(&Temporary, 2 * sizeof(LVM_FLOAT) * MaxBlockSize);
-#endif
if(pInstanceParams->NumDelays == LVREV_DELAYLINES_4)
{
for(i=0; i<4; i++)
{
-#ifndef BUILD_FLOAT
- InstAlloc_AddMember(&Temporary, sizeof(LVM_INT32) * MaxBlockSize); /* A Scratch buffer for each delay line */
-#else
/* A Scratch buffer for each delay line */
InstAlloc_AddMember(&Temporary, sizeof(LVM_FLOAT) * MaxBlockSize);
-#endif
}
}
@@ -237,12 +205,8 @@
{
for(i=0; i<2; i++)
{
-#ifndef BUILD_FLOAT
- InstAlloc_AddMember(&Temporary, sizeof(LVM_INT32) * MaxBlockSize); /* A Scratch buffer for each delay line */
-#else
/* A Scratch buffer for each delay line */
InstAlloc_AddMember(&Temporary, sizeof(LVM_FLOAT) * MaxBlockSize);
-#endif
}
}
@@ -250,12 +214,8 @@
{
for(i=0; i<1; i++)
{
-#ifndef BUILD_FLOAT
- InstAlloc_AddMember(&Temporary, sizeof(LVM_INT32) * MaxBlockSize); /* A Scratch buffer for each delay line */
-#else
/* A Scratch buffer for each delay line */
InstAlloc_AddMember(&Temporary, sizeof(LVM_FLOAT) * MaxBlockSize);
-#endif
}
}
@@ -268,14 +228,12 @@
{
LVREV_Instance_st *pLVREV_Private = (LVREV_Instance_st *)hInstance;
-
/*
* Read back memory allocation table
*/
*pMemoryTable = pLVREV_Private->MemoryTable;
}
-
return(LVREV_SUCCESS);
}
diff --git a/media/libeffects/lvm/lib/Reverb/src/LVREV_Private.h b/media/libeffects/lvm/lib/Reverb/src/LVREV_Private.h
index c915ac0..2c27c6e 100644
--- a/media/libeffects/lvm/lib/Reverb/src/LVREV_Private.h
+++ b/media/libeffects/lvm/lib/Reverb/src/LVREV_Private.h
@@ -18,11 +18,6 @@
#ifndef __LVREV_PRIVATE_H__
#define __LVREV_PRIVATE_H__
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-
/****************************************************************************************/
/* */
/* Includes */
@@ -36,43 +31,22 @@
#include "Mixer.h"
#include "LVM_Macros.h"
-
/****************************************************************************************/
/* */
/* Defines */
/* */
/****************************************************************************************/
-#ifndef BUILD_FLOAT
-/* General */
-#define ONE_OVER_SQRT_TWO 23170 /* 1/sqrt(2) * 2^15 */
-#define LVREV_B_8_on_1000 17179869 /* 0.8 * 2^31 */
-#define LVREV_HEADROOM 8192 /* -12dB * 2^15 */
-#define LVREV_2_9_INQ29 1583769190L /* 2.9 in Q29 format */
-#define LVREV_MIN3DB 0x5A82 /* -3dB in Q15 format */
-#else
/* General */
#define ONE_OVER_SQRT_TWO 0.707107f /* 1/sqrt(2) * 2^15 */
#define LVREV_B_8_on_1000 0.008f /* 0.8 * 2^31 */
#define LVREV_HEADROOM 0.25f /* -12dB * 2^15 */
#define LVREV_2_9_INQ29 2.9f /* 2.9 in Q29 format */
#define LVREV_MIN3DB 0.7079457f /* -3dB in Q15 format */
-#endif
/* Intenal constants */
#define LVREV_LP_Poly_Order 4
#define LVREV_LP_Poly_Shift 5
-#ifndef BUILD_FLOAT
-#define LVREV_T_3_Power_0_on_4 32768
-#define LVREV_T_3_Power_1_on_4 43125
-#define LVREV_T_3_Power_2_on_4 56755
-#define LVREV_T_3_Power_3_on_4 74694
-#define LVREV_T60_SCALE 306774 /*(32767/7000)<<16 */
-#define LVREV_T_3_Power_minus0_on_4 32767 /* 3^(-0/4) * 2^15 */
-#define LVREV_T_3_Power_minus1_on_4 24898 /* 3^(-1/4) * 2^15 */
-#define LVREV_T_3_Power_minus2_on_4 18919 /* 3^(-2/4) * 2^15 */
-#define LVREV_T_3_Power_minus3_on_4 14375 /* 3^(-3/4) * 2^15 */
-#else/*BUILD_FLOAT*/
#define LVREV_T60_SCALE 0.000142f /*(1/7000) */
#define LVREV_T_3_Power_0_on_4 1.0f
@@ -83,18 +57,7 @@
#define LVREV_T_3_Power_minus1_on_4 0.759836f /* 3^(-1/4) * 2^15 */
#define LVREV_T_3_Power_minus2_on_4 0.577350f /* 3^(-2/4) * 2^15 */
#define LVREV_T_3_Power_minus3_on_4 0.438691f /* 3^(-3/4) * 2^15 */
-#endif
-#ifndef HIGHER_FS
-#define LVREV_MAX_T3_DELAY 2527 /* ((48000 * 120 * LVREV_T_3_Power_minus3_on_4) >> 15) / 1000 */
-#define LVREV_MAX_T2_DELAY 3326 /* ((48000 * 120 * LVREV_T_3_Power_minus2_on_4) >> 15) / 1000 */
-#define LVREV_MAX_T1_DELAY 4377 /* ((48000 * 120 * LVREV_T_3_Power_minus1_on_4) >> 15) / 1000 */
-#define LVREV_MAX_T0_DELAY 5760 /* ((48000 * 120 * LVREV_T_3_Power_minus0_on_4) >> 15) / 1000 */
-#define LVREV_MAX_AP3_DELAY 1685 /* ((48000 * 120 * LVREV_T_3_Power_minus3_on_4) >> 15) / 1500 */
-#define LVREV_MAX_AP2_DELAY 2218 /* ((48000 * 120 * LVREV_T_3_Power_minus2_on_4) >> 15) / 1500 */
-#define LVREV_MAX_AP1_DELAY 2918 /* ((48000 * 120 * LVREV_T_3_Power_minus1_on_4) >> 15) / 1500 */
-#define LVREV_MAX_AP0_DELAY 3840 /* ((48000 * 120 * LVREV_T_3_Power_minus0_on_4) >> 15) / 1500 */
-#else
/* ((192000 * 120 * LVREV_T_3_Power_minus3_on_4) >> 15) / 1000 */
#define LVREV_MAX_T3_DELAY 10108
/* ((192000 * 120 * LVREV_T_3_Power_minus2_on_4) >> 15) / 1000 */
@@ -111,7 +74,6 @@
#define LVREV_MAX_AP1_DELAY 11672
/* ((192000 * 120 * LVREV_T_3_Power_minus0_on_4) >> 15) / 1500 */
#define LVREV_MAX_AP0_DELAY 15360
-#endif
#define LVREV_BYPASSMIXER_TC 1000 /* Bypass mixer time constant*/
#define LVREV_ALLPASS_TC 1000 /* All-pass filter time constant */
@@ -120,11 +82,7 @@
#define LVREV_OUTPUTGAIN_SHIFT 5 /* Bits shift for output gain correction */
/* Parameter limits */
-#ifndef HIGHER_FS
-#define LVREV_NUM_FS 9 /* Number of supported sample rates */
-#else
#define LVREV_NUM_FS 13 /* Number of supported sample rates */
-#endif
#define LVREV_MAXBLKSIZE_LIMIT 64 /* Maximum block size low limit */
#define LVREV_MAX_LEVEL 100 /* Maximum level, 100% */
@@ -137,81 +95,11 @@
#define LVREV_MAX_DAMPING 100 /* Maximum damping, 100% */
#define LVREV_MAX_ROOMSIZE 100 /* Maximum room size, 100% */
-
-
/****************************************************************************************/
/* */
/* Structures */
/* */
/****************************************************************************************/
-#ifndef BUILD_FLOAT
-/* Fast data structure */
-typedef struct
-{
-
- Biquad_1I_Order1_Taps_t HPTaps; /* High pass filter taps */
- Biquad_1I_Order1_Taps_t LPTaps; /* Low pass filter taps */
- Biquad_1I_Order1_Taps_t RevLPTaps[4]; /* Reverb low pass filters taps */
-
-} LVREV_FastData_st;
-
-
-/* Fast coefficient structure */
-typedef struct
-{
-
- Biquad_Instance_t HPCoefs; /* High pass filter coefficients */
- Biquad_Instance_t LPCoefs; /* Low pass filter coefficients */
- Biquad_Instance_t RevLPCoefs[4]; /* Reverb low pass filters coefficients */
-
-} LVREV_FastCoef_st;
-
-
-/* Instance parameter structure */
-typedef struct
-{
- /* General */
- LVREV_InstanceParams_st InstanceParams; /* Initialisation time instance parameters */
- LVREV_MemoryTable_st MemoryTable; /* Memory table */
- LVREV_ControlParams_st CurrentParams; /* Parameters being used */
- LVREV_ControlParams_st NewParams; /* New parameters from the calling application */
- LVM_CHAR bControlPending; /* Flag to indicate new parameters are available */
- LVM_CHAR bFirstControl; /* Flag to indicate that the control function is called for the first time */
- LVM_CHAR bDisableReverb; /* Flag to indicate that the mix level is 0% and the reverb can be disabled */
- LVM_INT32 RoomSizeInms; /* Room size in msec */
- LVM_INT32 MaxBlkLen; /* Maximum block size for internal processing */
-
- /* Aligned memory pointers */
- LVREV_FastData_st *pFastData; /* Fast data memory base address */
- LVREV_FastCoef_st *pFastCoef; /* Fast coefficient memory base address */
- LVM_INT32 *pScratchDelayLine[4]; /* Delay line scratch memory */
- LVM_INT32 *pScratch; /* Multi ussge scratch */
- LVM_INT32 *pInputSave; /* Reverb block input save for dry/wet mixing*/
-
- /* Feedback matrix */
- Mix_1St_Cll_t FeedbackMixer[4]; /* Mixer for Pop and Click Supression caused by feedback Gain */
-
- /* All-Pass Filter */
- LVM_INT32 T[4]; /* Maximum delay size of buffer */
- LVM_INT32 *pDelay_T[4]; /* Pointer to delay buffers */
- LVM_INT32 Delay_AP[4]; /* Offset to AP delay buffer start */
- LVM_INT16 AB_Selection; /* Smooth from tap A to B when 1 otherwise B to A */
- LVM_INT32 A_DelaySize[4]; /* A delay length in samples */
- LVM_INT32 B_DelaySize[4]; /* B delay length in samples */
- LVM_INT32 *pOffsetA[4]; /* Offset for the A delay tap */
- LVM_INT32 *pOffsetB[4]; /* Offset for the B delay tap */
- Mix_2St_Cll_t Mixer_APTaps[4]; /* Smoothed AP delay mixer */
- Mix_1St_Cll_t Mixer_SGFeedback[4]; /* Smoothed SAfeedback gain */
- Mix_1St_Cll_t Mixer_SGFeedforward[4]; /* Smoothed AP feedforward gain */
-
- /* Output gain */
- Mix_2St_Cll_t BypassMixer; /* Dry/wet mixer */
- LVM_INT16 Gain; /* Gain applied to output to maintain average signal power */
- Mix_1St_Cll_t GainMixer; /* Gain smoothing */
-
-} LVREV_Instance_st;
-
-#else /* BUILD_FLOAT */
/* Fast data structure */
typedef struct
@@ -222,7 +110,6 @@
} LVREV_FastData_st;
-
/* Fast coefficient structure */
typedef struct
{
@@ -262,7 +149,6 @@
Mix_1St_Cll_FLOAT_t FeedbackMixer[4]; /* Mixer for Pop and Click Supression \
caused by feedback Gain */
-
/* All-Pass Filter */
LVM_INT32 T[4]; /* Maximum delay size of buffer */
LVM_FLOAT *pDelay_T[4]; /* Pointer to delay buffers */
@@ -285,7 +171,6 @@
} LVREV_Instance_st;
-#endif
/****************************************************************************************/
/* */
/* Function prototypes */
@@ -293,26 +178,14 @@
/****************************************************************************************/
LVREV_ReturnStatus_en LVREV_ApplyNewSettings(LVREV_Instance_st *pPrivate);
-#ifdef BUILD_FLOAT
void ReverbBlock(LVM_FLOAT *pInput,
LVM_FLOAT *pOutput,
LVREV_Instance_st *pPrivate,
LVM_UINT16 NumSamples);
-#else
-void ReverbBlock(LVM_INT32 *pInput,
- LVM_INT32 *pOutput,
- LVREV_Instance_st *pPrivate,
- LVM_UINT16 NumSamples);
-#endif
LVM_INT32 BypassMixer_Callback(void *pCallbackData,
void *pGeneralPurpose,
LVM_INT16 GeneralPurpose );
-
-#ifdef __cplusplus
-}
-#endif
-
#endif /** __LVREV_PRIVATE_H__ **/
/* End of file */
diff --git a/media/libeffects/lvm/lib/Reverb/src/LVREV_Process.c b/media/libeffects/lvm/lib/Reverb/src/LVREV_Process.cpp
similarity index 61%
rename from media/libeffects/lvm/lib/Reverb/src/LVREV_Process.c
rename to media/libeffects/lvm/lib/Reverb/src/LVREV_Process.cpp
index 1d1283e..35f9ad8 100644
--- a/media/libeffects/lvm/lib/Reverb/src/LVREV_Process.c
+++ b/media/libeffects/lvm/lib/Reverb/src/LVREV_Process.cpp
@@ -23,7 +23,6 @@
#include "LVREV_Private.h"
#include "VectorArithmetic.h"
-
/****************************************************************************************/
/* */
/* FUNCTION: LVREV_Process */
@@ -46,26 +45,14 @@
/* 1. The input and output buffers must be 32-bit aligned */
/* */
/****************************************************************************************/
-#ifdef BUILD_FLOAT
LVREV_ReturnStatus_en LVREV_Process(LVREV_Handle_t hInstance,
const LVM_FLOAT *pInData,
LVM_FLOAT *pOutData,
const LVM_UINT16 NumSamples)
-#else
-LVREV_ReturnStatus_en LVREV_Process(LVREV_Handle_t hInstance,
- const LVM_INT32 *pInData,
- LVM_INT32 *pOutData,
- const LVM_UINT16 NumSamples)
-#endif
{
LVREV_Instance_st *pLVREV_Private = (LVREV_Instance_st *)hInstance;
-#ifdef BUILD_FLOAT
LVM_FLOAT *pInput = (LVM_FLOAT *)pInData;
LVM_FLOAT *pOutput = pOutData;
-#else
- LVM_INT32 *pInput = (LVM_INT32 *)pInData;
- LVM_INT32 *pOutput = pOutData;
-#endif
LVM_INT32 SamplesToProcess, RemainingSamples;
LVM_INT32 format = 1;
@@ -117,15 +104,6 @@
/*
* Copy the data to the output buffer, convert to stereo is required
*/
-#ifndef BUILD_FLOAT
- if(pLVREV_Private->CurrentParams.SourceFormat == LVM_MONO){
- MonoTo2I_32(pInput, pOutput, NumSamples);
- } else {
- Copy_16((LVM_INT16 *)pInput,
- (LVM_INT16 *)pOutput,
- (LVM_INT16)(NumSamples << 2)); // 32 bit data, stereo
- }
-#else
if(pLVREV_Private->CurrentParams.SourceFormat == LVM_MONO){
MonoTo2I_Float(pInput, pOutput, NumSamples);
} else {
@@ -133,7 +111,6 @@
pOutput,
(LVM_INT16)(NumSamples << 1)); // 32 bit data, stereo
}
-#endif
}
return LVREV_SUCCESS;
@@ -164,20 +141,13 @@
}
ReverbBlock(pInput, pOutput, pLVREV_Private, (LVM_UINT16)SamplesToProcess);
-#ifdef BUILD_FLOAT
pInput = (LVM_FLOAT *)(pInput + (SamplesToProcess * format));
pOutput = (LVM_FLOAT *)(pOutput + (SamplesToProcess * 2)); // Always stereo output
-#else
- pInput = (LVM_INT32 *)(pInput +(SamplesToProcess*format));
- pOutput = (LVM_INT32 *)(pOutput+(SamplesToProcess*2));
-#endif
}
return LVREV_SUCCESS;
}
-
-
/****************************************************************************************/
/* */
/* FUNCTION: ReverbBlock */
@@ -200,311 +170,6 @@
/* 1. The input and output buffers must be 32-bit aligned */
/* */
/****************************************************************************************/
-#ifndef BUILD_FLOAT
-void ReverbBlock(LVM_INT32 *pInput, LVM_INT32 *pOutput, LVREV_Instance_st *pPrivate, LVM_UINT16 NumSamples)
-{
- LVM_INT16 j, size;
- LVM_INT32 *pDelayLine;
- LVM_INT32 *pDelayLineInput = pPrivate->pScratch;
- LVM_INT32 *pScratch = pPrivate->pScratch;
- LVM_INT32 *pIn;
- LVM_INT32 *pTemp = pPrivate->pInputSave;
- LVM_INT32 NumberOfDelayLines;
-
- /******************************************************************************
- * All calculations will go into the buffer pointed to by pTemp, this will *
- * then be mixed with the original input to create the final output. *
- * *
- * When INPLACE processing is selected this must be a temporary buffer and *
- * hence this is the worst case, so for simplicity this will ALWAYS be so *
- * *
- * The input buffer will remain untouched until the output of the mixer if *
- * INPLACE processing is selected. *
- * *
- * The temp buffer will always be NumSamples in size regardless of MONO or *
- * STEREO input. In the case of stereo input all processing is done in MONO *
- * and the final output is converted to STEREO after the mixer *
- ******************************************************************************/
-
- if(pPrivate->InstanceParams.NumDelays == LVREV_DELAYLINES_4 )
- {
- NumberOfDelayLines = 4;
- }
- else if(pPrivate->InstanceParams.NumDelays == LVREV_DELAYLINES_2 )
- {
- NumberOfDelayLines = 2;
- }
- else
- {
- NumberOfDelayLines = 1;
- }
-
- if(pPrivate->CurrentParams.SourceFormat == LVM_MONO)
- {
- pIn = pInput;
- }
- else
- {
- /*
- * Stereo to mono conversion
- */
-
- From2iToMono_32( pInput,
- pTemp,
- (LVM_INT16)NumSamples);
-
- pIn = pTemp;
- }
-
- Mult3s_32x16(pIn,
- (LVM_INT16)LVREV_HEADROOM,
- pTemp,
- (LVM_INT16)NumSamples);
-
- /*
- * High pass filter
- */
- FO_1I_D32F32C31_TRC_WRA_01( &pPrivate->pFastCoef->HPCoefs,
- pTemp,
- pTemp,
- (LVM_INT16)NumSamples);
- /*
- * Low pass filter
- */
- FO_1I_D32F32C31_TRC_WRA_01( &pPrivate->pFastCoef->LPCoefs,
- pTemp,
- pTemp,
- (LVM_INT16)NumSamples);
-
- /*
- * Process all delay lines
- */
-
- for(j = 0; j < NumberOfDelayLines; j++)
- {
- pDelayLine = pPrivate->pScratchDelayLine[j];
-
- /*
- * All-pass filter with pop and click suppression
- */
- /* Get the smoothed, delayed output. Put it in the output buffer */
- MixSoft_2St_D32C31_SAT(&pPrivate->Mixer_APTaps[j],
- pPrivate->pOffsetA[j],
- pPrivate->pOffsetB[j],
- pDelayLine,
- (LVM_INT16)NumSamples);
- /* Re-align the all pass filter delay buffer and copying the fixed delay data to the AP delay in the process */
- Copy_16((LVM_INT16 *)&pPrivate->pDelay_T[j][NumSamples],
- (LVM_INT16 *)pPrivate->pDelay_T[j],
- (LVM_INT16)((pPrivate->T[j]-NumSamples) << 1)); /* 32-bit data */
- /* Apply the smoothed feedback and save to fixed delay input (currently empty) */
- MixSoft_1St_D32C31_WRA(&pPrivate->Mixer_SGFeedback[j],
- pDelayLine,
- &pPrivate->pDelay_T[j][pPrivate->T[j]-NumSamples],
- (LVM_INT16)NumSamples);
- /* Sum into the AP delay line */
- Mac3s_Sat_32x16(&pPrivate->pDelay_T[j][pPrivate->T[j]-NumSamples],
- -0x7fff, /* Invert since the feedback coefficient is negative */
- &pPrivate->pDelay_T[j][pPrivate->Delay_AP[j]-NumSamples],
- (LVM_INT16)NumSamples);
- /* Apply smoothed feedforward sand save to fixed delay input (currently empty) */
- MixSoft_1St_D32C31_WRA(&pPrivate->Mixer_SGFeedforward[j],
- &pPrivate->pDelay_T[j][pPrivate->Delay_AP[j]-NumSamples],
- &pPrivate->pDelay_T[j][pPrivate->T[j]-NumSamples],
- (LVM_INT16)NumSamples);
- /* Sum into the AP output */
- Mac3s_Sat_32x16(&pPrivate->pDelay_T[j][pPrivate->T[j]-NumSamples],
- 0x7fff,
- pDelayLine,
- (LVM_INT16)NumSamples);
-
- /*
- * Feedback gain
- */
- MixSoft_1St_D32C31_WRA(&pPrivate->FeedbackMixer[j], pDelayLine, pDelayLine, NumSamples);
-
- /*
- * Low pass filter
- */
- FO_1I_D32F32C31_TRC_WRA_01( &pPrivate->pFastCoef->RevLPCoefs[j],
- pDelayLine,
- pDelayLine,
- (LVM_INT16)NumSamples);
- }
-
- /*
- * Apply rotation matrix and delay samples
- */
- for(j = 0; j < NumberOfDelayLines; j++)
- {
-
- Copy_16( (LVM_INT16*)(pTemp),
- (LVM_INT16*)(pDelayLineInput),
- (LVM_INT16)(NumSamples << 1));
-
- /*
- * Rotation matrix mix
- */
- switch(j)
- {
- case 3:
- /*
- * Add delay line 1 and 2 contribution
- */
- Mac3s_Sat_32x16(pPrivate->pScratchDelayLine[1], -0x8000, pDelayLineInput, (LVM_INT16)NumSamples);
- Mac3s_Sat_32x16(pPrivate->pScratchDelayLine[2], -0x8000, pDelayLineInput, (LVM_INT16)NumSamples);
-
- break;
- case 2:
-
- /*
- * Add delay line 0 and 3 contribution
- */
- Mac3s_Sat_32x16(pPrivate->pScratchDelayLine[0], -0x8000, pDelayLineInput, (LVM_INT16)NumSamples);
- Mac3s_Sat_32x16(pPrivate->pScratchDelayLine[3], -0x8000, pDelayLineInput, (LVM_INT16)NumSamples);
-
- break;
- case 1:
- if(pPrivate->InstanceParams.NumDelays == LVREV_DELAYLINES_4)
- {
- /*
- * Add delay line 0 and 3 contribution
- */
- Mac3s_Sat_32x16(pPrivate->pScratchDelayLine[0], -0x8000, pDelayLineInput, (LVM_INT16)NumSamples);
- Add2_Sat_32x32(pPrivate->pScratchDelayLine[3], pDelayLineInput, (LVM_INT16)NumSamples);
-
- }
- else
- {
- /*
- * Add delay line 0 and 1 contribution
- */
- Mac3s_Sat_32x16(pPrivate->pScratchDelayLine[0], -0x8000, pDelayLineInput, (LVM_INT16)NumSamples);
- Mac3s_Sat_32x16(pPrivate->pScratchDelayLine[1], -0x8000, pDelayLineInput, (LVM_INT16)NumSamples);
-
- }
- break;
- case 0:
- if(pPrivate->InstanceParams.NumDelays == LVREV_DELAYLINES_4)
- {
- /*
- * Add delay line 1 and 2 contribution
- */
- Mac3s_Sat_32x16(pPrivate->pScratchDelayLine[1], -0x8000, pDelayLineInput, (LVM_INT16)NumSamples);
- Add2_Sat_32x32(pPrivate->pScratchDelayLine[2], pDelayLineInput, (LVM_INT16)NumSamples);
-
- }
- else if(pPrivate->InstanceParams.NumDelays == LVREV_DELAYLINES_2)
- {
- /*
- * Add delay line 0 and 1 contribution
- */
- Add2_Sat_32x32(pPrivate->pScratchDelayLine[0], pDelayLineInput, (LVM_INT16)NumSamples);
- Mac3s_Sat_32x16(pPrivate->pScratchDelayLine[1], -0x8000, pDelayLineInput, (LVM_INT16)NumSamples);
-
- }
- else
- {
- /*
- * Add delay line 0 contribution
- */
-
- /* SOURCE DESTINATION*/
- Add2_Sat_32x32(pPrivate->pScratchDelayLine[0], pDelayLineInput, (LVM_INT16)NumSamples);
- }
- break;
- default:
- break;
- }
-
- /*
- * Delay samples
- */
- Copy_16((LVM_INT16 *)pDelayLineInput,
- (LVM_INT16 *)&pPrivate->pDelay_T[j][pPrivate->T[j]-NumSamples],
- (LVM_INT16)(NumSamples << 1)); /* 32-bit data */
-
- }
-
-
- /*
- * Create stereo output
- */
- switch(pPrivate->InstanceParams.NumDelays)
- {
- case LVREV_DELAYLINES_4:
- Add2_Sat_32x32(pPrivate->pScratchDelayLine[3],
- pPrivate->pScratchDelayLine[0],
- (LVM_INT16)NumSamples);
- Add2_Sat_32x32(pPrivate->pScratchDelayLine[2],
- pPrivate->pScratchDelayLine[1],
- (LVM_INT16)NumSamples);
-
-
- JoinTo2i_32x32(pPrivate->pScratchDelayLine[0],
- pPrivate->pScratchDelayLine[1],
- pTemp,
- (LVM_INT16)NumSamples);
-
-
- break;
- case LVREV_DELAYLINES_2:
-
- Copy_16( (LVM_INT16*)pPrivate->pScratchDelayLine[1],
- (LVM_INT16*)pScratch,
- (LVM_INT16)(NumSamples << 1));
-
- Mac3s_Sat_32x16(pPrivate->pScratchDelayLine[0],
- -0x8000,
- pScratch,
- (LVM_INT16)NumSamples);
-
- Add2_Sat_32x32(pPrivate->pScratchDelayLine[1],
- pPrivate->pScratchDelayLine[0],
- (LVM_INT16)NumSamples);
-
-
- JoinTo2i_32x32(pPrivate->pScratchDelayLine[0],
- pScratch,
- pTemp,
- (LVM_INT16)NumSamples);
- break;
- case LVREV_DELAYLINES_1:
- MonoTo2I_32(pPrivate->pScratchDelayLine[0],
- pTemp,
- (LVM_INT16)NumSamples);
- break;
- default:
- break;
- }
-
-
- /*
- * Dry/wet mixer
- */
-
- size = (LVM_INT16)(NumSamples << 1);
- MixSoft_2St_D32C31_SAT(&pPrivate->BypassMixer,
- pTemp,
- pTemp,
- pOutput,
- size);
-
- /* Apply Gain*/
-
- Shift_Sat_v32xv32 (LVREV_OUTPUTGAIN_SHIFT,
- pOutput,
- pOutput,
- size);
-
- MixSoft_1St_D32C31_WRA(&pPrivate->GainMixer,
- pOutput,
- pOutput,
- size);
-
- return;
-}
-#else
void ReverbBlock(LVM_FLOAT *pInput, LVM_FLOAT *pOutput,
LVREV_Instance_st *pPrivate, LVM_UINT16 NumSamples)
{
@@ -742,7 +407,6 @@
(LVM_INT16)(NumSamples)); /* 32-bit data */
}
-
/*
* Create stereo output
*/
@@ -756,13 +420,11 @@
pPrivate->pScratchDelayLine[1],
(LVM_INT16)NumSamples);
-
JoinTo2i_Float(pPrivate->pScratchDelayLine[0],
pPrivate->pScratchDelayLine[1],
pTemp,
(LVM_INT16)NumSamples);
-
break;
case LVREV_DELAYLINES_2:
@@ -779,7 +441,6 @@
pPrivate->pScratchDelayLine[0],
(LVM_INT16)NumSamples);
-
JoinTo2i_Float(pPrivate->pScratchDelayLine[0],
pScratch,
pTemp,
@@ -794,7 +455,6 @@
break;
}
-
/*
* Dry/wet mixer
*/
@@ -820,6 +480,5 @@
return;
}
-#endif
/* End of file */
diff --git a/media/libeffects/lvm/lib/Reverb/src/LVREV_SetControlParameters.c b/media/libeffects/lvm/lib/Reverb/src/LVREV_SetControlParameters.cpp
similarity index 99%
rename from media/libeffects/lvm/lib/Reverb/src/LVREV_SetControlParameters.c
rename to media/libeffects/lvm/lib/Reverb/src/LVREV_SetControlParameters.cpp
index dfed28e..2a75559 100644
--- a/media/libeffects/lvm/lib/Reverb/src/LVREV_SetControlParameters.c
+++ b/media/libeffects/lvm/lib/Reverb/src/LVREV_SetControlParameters.cpp
@@ -48,7 +48,6 @@
LVREV_Instance_st *pLVREV_Private = (LVREV_Instance_st *)hInstance;
-
/*
* Check for error conditions
*/
@@ -67,10 +66,8 @@
(pNewParams->SampleRate != LVM_FS_32000) &&
(pNewParams->SampleRate != LVM_FS_44100) &&
(pNewParams->SampleRate != LVM_FS_48000)
-#ifdef HIGHER_FS
&& (pNewParams->SampleRate != LVM_FS_88200) && (pNewParams->SampleRate != LVM_FS_96000)
&& (pNewParams->SampleRate != LVM_FS_176400) && (pNewParams->SampleRate != LVM_FS_192000)
-#endif
)
#ifdef SUPPORT_MC
|| ((pNewParams->SourceFormat != LVM_STEREO) &&
@@ -84,7 +81,6 @@
return (LVREV_OUTOFRANGE);
}
-
if (pNewParams->Level > LVREV_MAX_LEVEL)
{
return LVREV_OUTOFRANGE;
@@ -120,8 +116,6 @@
return LVREV_OUTOFRANGE;
}
-
-
/*
* Copy the new parameters and set the flag to indicate they are available
*/
diff --git a/media/libeffects/lvm/lib/Reverb/src/LVREV_Tables.c b/media/libeffects/lvm/lib/Reverb/src/LVREV_Tables.cpp
similarity index 66%
rename from media/libeffects/lvm/lib/Reverb/src/LVREV_Tables.c
rename to media/libeffects/lvm/lib/Reverb/src/LVREV_Tables.cpp
index 1058740..5cd623e 100644
--- a/media/libeffects/lvm/lib/Reverb/src/LVREV_Tables.c
+++ b/media/libeffects/lvm/lib/Reverb/src/LVREV_Tables.cpp
@@ -21,6 +21,7 @@
/* */
/****************************************************************************************/
#include "LVREV.h"
+#include "LVREV_Tables.h"
/****************************************************************************************/
/* */
@@ -29,19 +30,6 @@
/****************************************************************************************/
/* Table with supported sampling rates. The table can be indexed using LVM_Fs_en */
-#ifndef HIGHER_FS
-const LVM_UINT16 LVM_FsTable[] = {
- 8000 ,
- 11025,
- 12000,
- 16000,
- 22050,
- 24000,
- 32000,
- 44100,
- 48000
-};
-#else
const LVM_UINT32 LVM_FsTable[] = {
8000 ,
11025,
@@ -57,23 +45,13 @@
176400,
192000
};
-#endif
/* Table with supported sampling rates. The table can be indexed using LVM_Fs_en */
-#ifndef HIGHER_FS
-LVM_UINT16 LVM_GetFsFromTable(LVM_Fs_en FsIndex){
- if (FsIndex > LVM_FS_48000)
- return 0;
-
- return (LVM_FsTable[FsIndex]);
-}
-#else
LVM_UINT32 LVM_GetFsFromTable(LVM_Fs_en FsIndex){
if (FsIndex > LVM_FS_192000)
return 0;
return (LVM_FsTable[FsIndex]);
}
-#endif
/* In order to maintain consistant input and out put signal strengths
output gain/attenuation is applied. This gain depends on T60 and Rooms
@@ -95,33 +73,6 @@
*/
/* Normalizing output including Reverb Level part (only shift up)*/
-#ifndef BUILD_FLOAT
-const LVM_INT32 LVREV_GainPolyTable[24][5]={{1,17547434,128867434,-120988896,50761228,},
- {2,18256869,172666902,-193169292,88345744,},
- {3,16591311,139250151,-149667234,66770059,},
- {4,17379977,170835131,-173579321,76278163,},
- {5,18963512,210364934,-228623519,103435022,},
- {6,17796318,135756417,-144084053,64327698,},
- {7,17454695,174593214,-187513064,85146582,},
- {8,17229257,140715570,-145790588,65361740,},
- {9,17000547,163195946,-176733969,79562130,},
- {10,16711699,142476304,-133339887,58366547,},
- {13,18108419,149223697,-161762020,74397589,},
- {15,16682043,124844884,-134284487,60082180,},
- {17,16627346,120936430,-121766674,53146421,},
- {20,17338325,125432694,-126616983,56534237,},
- {25,16489146,99218217,-94597467,40616506,},
- {30,15582373,84479043,-75365006,30952348,},
- {40,16000669,84896611,-75031127,30696306,},
- {50,15087054,71695031,-59349268,23279669,},
- {60,15830714,68672971,-58211201,23671158,},
- {70,15536061,66657972,-55901437,22560153,},
- {75,15013145,48179917,-24138354,5232074,},
- {80,15688738,50195036,-34206760,11515792,},
- {90,16003322,48323661,-35607378,13153872,},
- {100,15955223,48558201,-33706865,11715792,},
- };
-#else
const LVM_FLOAT LVREV_GainPolyTable[24][5]={{1,1.045909f,7.681098f,-7.211500f,3.025605f,},
{2,1.088194f,10.291749f,-11.513787f,5.265817f,},
{3,0.988919f,8.299956f,-8.920862f,3.979806f,},
@@ -147,6 +98,5 @@
{90,0.953872f,2.880315f,-2.122365f,0.784032f,},
{100,0.951005f,2.894294f,-2.009086f,0.698316f,},
};
-#endif
/* End of file */
diff --git a/media/libeffects/lvm/lib/Reverb/src/LVREV_Tables.h b/media/libeffects/lvm/lib/Reverb/src/LVREV_Tables.h
index 0658186..e100d8a 100644
--- a/media/libeffects/lvm/lib/Reverb/src/LVREV_Tables.h
+++ b/media/libeffects/lvm/lib/Reverb/src/LVREV_Tables.h
@@ -15,15 +15,9 @@
* limitations under the License.
*/
-
#ifndef _LVREV_TABLES_H_
#define _LVREV_TABLES_H_
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-
/****************************************************************************************/
/* */
/* Includes */
@@ -37,22 +31,10 @@
/* */
/****************************************************************************************/
-#ifndef HIGHER_FS
-extern const LVM_UINT16 LVM_FsTable[];
-extern LVM_UINT16 LVM_GetFsFromTable(LVM_Fs_en FsIndex);
-#else
extern const LVM_UINT32 LVM_FsTable[];
extern LVM_UINT32 LVM_GetFsFromTable(LVM_Fs_en FsIndex);
-#endif
-#ifndef BUILD_FLOAT
-extern LVM_INT32 LVREV_GainPolyTable[24][5];
-#else
-extern LVM_FLOAT LVREV_GainPolyTable[24][5];
-#endif
-#ifdef __cplusplus
-}
-#endif
+extern const LVM_FLOAT LVREV_GainPolyTable[24][5];
#endif /** _LVREV_TABLES_H_ **/
diff --git a/media/libeffects/lvm/lib/SpectrumAnalyzer/lib/LVPSA.h b/media/libeffects/lvm/lib/SpectrumAnalyzer/lib/LVPSA.h
index 2038fbb..c9fa7ad 100644
--- a/media/libeffects/lvm/lib/SpectrumAnalyzer/lib/LVPSA.h
+++ b/media/libeffects/lvm/lib/SpectrumAnalyzer/lib/LVPSA.h
@@ -18,14 +18,8 @@
#ifndef _LVPSA_H_
#define _LVPSA_H_
-
#include "LVM_Types.h"
-
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
/****************************************************************************************/
/* */
/* CONSTANTS DEFINITIONS */
@@ -116,8 +110,6 @@
LVPSA_RETURN_DUMMY = LVM_MAXINT_32 /* Force 32 bits enum, don't use it! */
} LVPSA_RETURN;
-
-
/*********************************************************************************************************************************
FUNCTIONS PROTOTYPE
**********************************************************************************************************************************/
@@ -216,17 +208,10 @@
/* otherwise Error due to bad parameters */
/* */
/*********************************************************************************************************************************/
-#ifdef BUILD_FLOAT
LVPSA_RETURN LVPSA_Process ( pLVPSA_Handle_t hInstance,
LVM_FLOAT *pLVPSA_InputSamples,
LVM_UINT16 InputBlockSize,
LVPSA_Time AudioTime );
-#else
-LVPSA_RETURN LVPSA_Process ( pLVPSA_Handle_t hInstance,
- LVM_INT16 *pLVPSA_InputSamples,
- LVM_UINT16 InputBlockSize,
- LVPSA_Time AudioTime );
-#endif
/*********************************************************************************************************************************/
/* */
/* FUNCTION: LVPSA_GetSpectrum */
@@ -288,9 +273,4 @@
LVPSA_RETURN LVPSA_GetInitParams ( pLVPSA_Handle_t hInstance,
LVPSA_InitParams_t *pParams );
-
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
-
#endif /* _LVPSA_H */
diff --git a/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Control.c b/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Control.cpp
similarity index 79%
rename from media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Control.c
rename to media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Control.cpp
index f6c4ea7..deafaa7 100644
--- a/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Control.c
+++ b/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Control.cpp
@@ -28,7 +28,6 @@
LVPSA_RETURN LVPSA_SetQPFCoefficients( LVPSA_InstancePr_t *pInst,
LVPSA_ControlParams_t *pParams );
-#ifdef BUILD_FLOAT
LVPSA_RETURN LVPSA_BPSinglePrecCoefs( LVM_UINT16 Fs,
LVPSA_FilterParam_t *pFilterParams,
BP_FLOAT_Coefs_t *pCoefficients);
@@ -36,27 +35,11 @@
LVPSA_RETURN LVPSA_BPDoublePrecCoefs( LVM_UINT16 Fs,
LVPSA_FilterParam_t *pFilterParams,
BP_FLOAT_Coefs_t *pCoefficients);
-#else
-LVPSA_RETURN LVPSA_BPSinglePrecCoefs( LVM_UINT16 Fs,
- LVPSA_FilterParam_t *pFilterParams,
- BP_C16_Coefs_t *pCoefficients);
-
-LVPSA_RETURN LVPSA_BPDoublePrecCoefs( LVM_UINT16 Fs,
- LVPSA_FilterParam_t *pFilterParams,
- BP_C32_Coefs_t *pCoefficients);
-
-LVPSA_RETURN LVPSA_BPDoublePrecCoefs( LVM_UINT16 Fs,
- LVPSA_FilterParam_t *pFilterParams,
- BP_C32_Coefs_t *pCoefficients);
-#endif
LVPSA_RETURN LVPSA_SetBPFCoefficients( LVPSA_InstancePr_t *pInst,
LVPSA_ControlParams_t *pParams );
LVPSA_RETURN LVPSA_ClearFilterHistory( LVPSA_InstancePr_t *pInst);
-
-
-
/************************************************************************************/
/* */
/* FUNCTION: LVPSA_Control */
@@ -129,7 +112,6 @@
return(LVPSA_OK);
}
-
/************************************************************************************/
/* */
/* FUNCTION: LVPSA_GetInitParams */
@@ -163,7 +145,6 @@
return(LVPSA_OK);
}
-
/************************************************************************************/
/* */
/* FUNCTION: LVPSA_ApplyNewSettings */
@@ -188,14 +169,9 @@
LVM_UINT16 Freq;
LVPSA_ControlParams_t Params;
extern LVM_INT16 LVPSA_nSamplesBufferUpdate[];
-#ifndef HIGHER_FS
- extern LVM_UINT16 LVPSA_SampleRateTab[];
-#else
extern LVM_UINT32 LVPSA_SampleRateTab[];
-#endif
extern LVM_UINT16 LVPSA_DownSamplingFactor[];
-
if(pInst == 0)
{
return(LVPSA_ERROR_NULLADDRESS);
@@ -280,11 +256,7 @@
LVPSA_RETURN LVPSA_SetBPFiltersType ( LVPSA_InstancePr_t *pInst,
LVPSA_ControlParams_t *pParams )
{
-#ifndef HIGHER_FS
- extern LVM_UINT16 LVPSA_SampleRateTab[]; /* Sample rate table */
-#else
extern LVM_UINT32 LVPSA_SampleRateTab[]; /* Sample rate table */
-#endif
LVM_UINT16 ii; /* Filter band index */
LVM_UINT32 fs = (LVM_UINT32)LVPSA_SampleRateTab[(LVM_UINT16)pParams->Fs]; /* Sample rate */
LVM_UINT32 fc; /* Filter centre frequency */
@@ -298,7 +270,6 @@
fc = (LVM_UINT32)pInst->pFiltersParams[ii].CenterFrequency; /* Get the band centre frequency */
QFactor =(LVM_INT16) pInst->pFiltersParams[ii].QFactor; /* Get the band Q factor */
-
/*
* For each filter set the type of biquad required
*/
@@ -358,22 +329,6 @@
{
case LVPSA_DoublePrecisionFilter:
{
-#ifndef BUILD_FLOAT
- BP_C32_Coefs_t Coefficients;
-
- /*
- * Calculate the double precision coefficients
- */
- LVPSA_BPDoublePrecCoefs((LVM_UINT16)pParams->Fs,
- &pInst->pFiltersParams[ii],
- &Coefficients);
- /*
- * Set the coefficients
- */
- BP_1I_D16F32Cll_TRC_WRA_01_Init ( &pInst->pBP_Instances[ii],
- &pInst->pBP_Taps[ii],
- &Coefficients);
-#else
BP_FLOAT_Coefs_t Coefficients;
/*
* Calculate the double precision coefficients
@@ -387,29 +342,11 @@
BP_1I_D16F32Cll_TRC_WRA_01_Init ( &pInst->pBP_Instances[ii],
&pInst->pBP_Taps[ii],
&Coefficients);
-#endif
break;
}
case LVPSA_SimplePrecisionFilter:
{
-#ifndef BUILD_FLOAT
- BP_C16_Coefs_t Coefficients;
-
- /*
- * Calculate the single precision coefficients
- */
- LVPSA_BPSinglePrecCoefs((LVM_UINT16)pParams->Fs,
- &pInst->pFiltersParams[ii],
- &Coefficients);
-
- /*
- * Set the coefficients
- */
- BP_1I_D16F16Css_TRC_WRA_01_Init (&pInst->pBP_Instances[ii],
- &pInst->pBP_Taps[ii],
- &Coefficients);
-#else
BP_FLOAT_Coefs_t Coefficients;
/*
@@ -425,7 +362,6 @@
BP_1I_D16F16Css_TRC_WRA_01_Init (&pInst->pBP_Instances[ii],
&pInst->pBP_Taps[ii],
&Coefficients);
-#endif
break;
}
}
@@ -434,7 +370,6 @@
return(LVPSA_OK);
}
-
/************************************************************************************/
/* */
/* FUNCTION: LVPSA_SetQPFCoefficients */
@@ -458,31 +393,17 @@
{
LVM_UINT16 ii;
LVM_Fs_en Fs = pParams->Fs;
-#ifndef BUILD_FLOAT
- QPD_C32_Coefs *pCoefficients;
- extern QPD_C32_Coefs LVPSA_QPD_Coefs[];
-
- pCoefficients = &LVPSA_QPD_Coefs[(pParams->LevelDetectionSpeed * LVPSA_NR_SUPPORTED_RATE) + Fs];
-#else
QPD_FLOAT_Coefs *pCoefficients;
extern QPD_FLOAT_Coefs LVPSA_QPD_Float_Coefs[];
pCoefficients = &LVPSA_QPD_Float_Coefs[(pParams->LevelDetectionSpeed * \
LVPSA_NR_SUPPORTED_RATE) + Fs];
-#endif
-
for (ii = 0; ii < pInst->nRelevantFilters; ii++)
{
-#ifndef BUILD_FLOAT
- LVPSA_QPD_Init (&pInst->pQPD_States[ii],
- &pInst->pQPD_Taps[ii],
- pCoefficients );
-#else
LVPSA_QPD_Init_Float (&pInst->pQPD_States[ii],
&pInst->pQPD_Taps[ii],
pCoefficients );
-#endif
}
return(LVPSA_OK);
@@ -522,7 +443,6 @@
/* of the n bands equalizer (LVEQNB */
/* */
/****************************************************************************************/
-#ifdef BUILD_FLOAT
LVPSA_RETURN LVPSA_BPSinglePrecCoefs( LVM_UINT16 Fs,
LVPSA_FilterParam_t *pFilterParams,
BP_FLOAT_Coefs_t *pCoefficients)
@@ -531,7 +451,6 @@
extern LVM_FLOAT LVPSA_Float_TwoPiOnFsTable[];
extern LVM_FLOAT LVPSA_Float_CosCoef[];
-
/*
* Intermediate variables and temporary values
*/
@@ -549,7 +468,6 @@
LVM_FLOAT t0;
LVM_INT16 i;
-
/*
* Get the filter definition
*/
@@ -589,7 +507,6 @@
}
COS_T0 = COS_T0 * 8; /*LVPSA_CosCoef_float[0]*/ /* Correct the scaling */
-
B1 = ((LVM_FLOAT)0.5 - B2) * (COS_T0); /* B1 = (0.5 - b2) * cos(t0) */
A0 = ((LVM_FLOAT)0.5 + B2) / 2; /* A0 = (0.5 + b2) / 2 */
@@ -602,89 +519,6 @@
return(LVPSA_OK);
}
-#else
-LVPSA_RETURN LVPSA_BPSinglePrecCoefs( LVM_UINT16 Fs,
- LVPSA_FilterParam_t *pFilterParams,
- BP_C16_Coefs_t *pCoefficients)
-{
-
- extern LVM_INT16 LVPSA_TwoPiOnFsTable[];
- extern LVM_INT16 LVPSA_CosCoef[];
-
-
- /*
- * Intermediate variables and temporary values
- */
- LVM_INT32 T0;
- LVM_INT16 D;
- LVM_INT32 A0;
- LVM_INT32 B1;
- LVM_INT32 B2;
- LVM_INT32 Dt0;
- LVM_INT32 B2_Den;
- LVM_INT32 B2_Num;
- LVM_INT32 COS_T0;
- LVM_INT16 coef;
- LVM_INT32 factor;
- LVM_INT16 t0;
- LVM_INT16 i;
-
-
- /*
- * Get the filter definition
- */
- LVM_UINT16 Frequency = pFilterParams->CenterFrequency;
- LVM_UINT16 QFactor = pFilterParams->QFactor;
-
-
- /*
- * Calculating the intermediate values
- */
- T0 = (LVM_INT32)Frequency * LVPSA_TwoPiOnFsTable[Fs]; /* T0 = 2 * Pi * Fc / Fs */
- D = 3200; /* Floating point value 1.000000 (1*100*2^5) */
- /* Force D = 1 : the function was originally used for a peaking filter.
- The D parameter do not exist for a BandPass filter coefficients */
-
- /*
- * Calculate the B2 coefficient
- */
- Dt0 = D * (T0 >> 10);
- B2_Den = (LVM_INT32)(((LVM_UINT32)QFactor << 19) + (LVM_UINT32)(Dt0 >> 2));
- B2_Num = (LVM_INT32)((LVM_UINT32)(Dt0 >> 3) - ((LVM_UINT32)QFactor << 18));
- B2 = (B2_Num / (B2_Den >> 16)) << 15;
-
- /*
- * Calculate the cosine by a polynomial expansion using the equation:
- *
- * Cos += coef(n) * t0^n For n = 0 to 6
- */
- T0 = (T0 >> 10) * 20859; /* Scale to 1.0 in 16-bit for range 0 to fs/2 */
- t0 = (LVM_INT16)(T0 >> 16);
- factor = 0x7fff; /* Initialise to 1.0 for the a0 coefficient */
- COS_T0 = 0; /* Initialise the error to zero */
- for (i=1; i<7; i++)
- {
- coef = LVPSA_CosCoef[i]; /* Get the nth coefficient */
- COS_T0 += (factor * coef) >> 5; /* The nth partial sum */
- factor = (factor * t0) >> 15; /* Calculate t0^n */
- }
- COS_T0 = COS_T0 << (LVPSA_CosCoef[0]+6); /* Correct the scaling */
-
-
- B1 = ((0x40000000 - B2) >> 16) * (COS_T0 >> 16); /* B1 = (0.5 - b2) * cos(t0) */
- A0 = (0x40000000 + B2) >> 1; /* A0 = (0.5 + b2) / 2 */
-
- /*
- * Write coeff into the data structure
- */
- pCoefficients->A0 = (LVM_INT16)(A0>>16);
- pCoefficients->B1 = (LVM_INT16)(B1>>15);
- pCoefficients->B2 = (LVM_INT16)(B2>>16);
-
-
- return(LVPSA_OK);
-}
-#endif
/****************************************************************************************/
/* */
/* FUNCTION: LVPSA_BPDoublePrecCoefs */
@@ -727,7 +561,6 @@
/* of the n bands equalizer (LVEQNB */
/* */
/****************************************************************************************/
-#ifdef BUILD_FLOAT
LVPSA_RETURN LVPSA_BPDoublePrecCoefs( LVM_UINT16 Fs,
LVPSA_FilterParam_t *pFilterParams,
BP_FLOAT_Coefs_t *pCoefficients)
@@ -759,7 +592,6 @@
LVM_FLOAT Frequency = (LVM_FLOAT)(pFilterParams->CenterFrequency);
LVM_FLOAT QFactor = ((LVM_FLOAT)(pFilterParams->QFactor)) / 100;
-
/*
* Calculating the intermediate values
*/
@@ -810,90 +642,6 @@
return(LVPSA_OK);
}
-#else
-LVPSA_RETURN LVPSA_BPDoublePrecCoefs( LVM_UINT16 Fs,
- LVPSA_FilterParam_t *pFilterParams,
- BP_C32_Coefs_t *pCoefficients)
-{
-
- extern LVM_INT16 LVPSA_TwoPiOnFsTable[];
- extern LVM_INT16 LVPSA_DPCosCoef[];
-
- /*
- * Intermediate variables and temporary values
- */
- LVM_INT32 T0;
- LVM_INT16 D;
- LVM_INT32 A0;
- LVM_INT32 B1;
- LVM_INT32 B2;
- LVM_INT32 Dt0;
- LVM_INT32 B2_Den;
- LVM_INT32 B2_Num;
- LVM_INT32 CosErr;
- LVM_INT16 coef;
- LVM_INT32 factor;
- LVM_INT16 t0;
- LVM_INT16 i;
-
- /*
- * Get the filter definition
- */
- LVM_UINT16 Frequency = pFilterParams->CenterFrequency;
- LVM_UINT16 QFactor = pFilterParams->QFactor;
-
-
- /*
- * Calculating the intermediate values
- */
- T0 = (LVM_INT32)Frequency * LVPSA_TwoPiOnFsTable[Fs]; /* T0 = 2 * Pi * Fc / Fs */
- D = 3200; /* Floating point value 1.000000 (1*100*2^5) */
- /* Force D = 1 : the function was originally used for a peaking filter.
- The D parameter do not exist for a BandPass filter coefficients */
-
- /*
- * Calculate the B2 coefficient
- */
- Dt0 = D * (T0 >> 10);
- B2_Den = (LVM_INT32)(((LVM_UINT32)QFactor << 19) + (LVM_UINT32)(Dt0 >> 2));
- B2_Num = (LVM_INT32)((LVM_UINT32)(Dt0 >> 3) - ((LVM_UINT32)QFactor << 18));
- B2 = (B2_Num / (B2_Den >> 16)) << 15;
-
- /*
- * Calculate the cosine error by a polynomial expansion using the equation:
- *
- * CosErr += coef(n) * t0^n For n = 0 to 4
- */
- T0 = (T0 >> 6) * 0x7f53; /* Scale to 1.0 in 16-bit for range 0 to fs/50 */
- t0 = (LVM_INT16)(T0 >> 16);
- factor = 0x7fff; /* Initialise to 1.0 for the a0 coefficient */
- CosErr = 0; /* Initialise the error to zero */
- for (i=1; i<5; i++)
- {
- coef = LVPSA_DPCosCoef[i]; /* Get the nth coefficient */
- CosErr += (factor * coef) >> 5; /* The nth partial sum */
- factor = (factor * t0) >> 15; /* Calculate t0^n */
- }
- CosErr = CosErr << (LVPSA_DPCosCoef[0]); /* Correct the scaling */
-
- /*
- * Calculate the B1 and A0 coefficients
- */
- B1 = (0x40000000 - B2); /* B1 = (0.5 - b2) */
- A0 = ((B1 >> 16) * (CosErr >> 10)) >> 6; /* Temporary storage for (0.5 - b2) * coserr(t0) */
- B1 -= A0; /* B1 = (0.5 - b2) * (1 - coserr(t0)) */
- A0 = (0x40000000 + B2) >> 1; /* A0 = (0.5 + b2) / 2 */
-
- /*
- * Write coeff into the data structure
- */
- pCoefficients->A0 = A0;
- pCoefficients->B1 = B1;
- pCoefficients->B2 = B2;
-
- return(LVPSA_OK);
-}
-#endif
/************************************************************************************/
/* */
/* FUNCTION: LVPSA_ClearFilterHistory */
@@ -917,17 +665,10 @@
/* Band Pass filters taps */
pTapAddress = (LVM_INT8 *)pInst->pBP_Taps;
-#ifdef BUILD_FLOAT
for(i = 0; i < pInst->nBands * sizeof(Biquad_1I_Order2_FLOAT_Taps_t); i++)
{
pTapAddress[i] = 0;
}
-#else
- for(i = 0; i < pInst->nBands * sizeof(Biquad_1I_Order2_Taps_t); i++)
- {
- pTapAddress[i] = 0;
- }
-#endif
/* Quasi-peak filters taps */
pTapAddress = (LVM_INT8 *)pInst->pQPD_Taps;
for(i = 0; i < pInst->nBands * sizeof(QPD_Taps_t); i++)
diff --git a/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Init.c b/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Init.cpp
similarity index 74%
rename from media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Init.c
rename to media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Init.cpp
index 1c26860..9fcd82f 100644
--- a/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Init.c
+++ b/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Init.cpp
@@ -47,11 +47,7 @@
LVPSA_InstancePr_t *pLVPSA_Inst;
LVPSA_RETURN errorCode = LVPSA_OK;
LVM_UINT32 ii;
-#ifndef BUILD_FLOAT
- extern LVM_INT16 LVPSA_GainTable[];
-#else
extern LVM_FLOAT LVPSA_Float_GainTable[];
-#endif
LVM_UINT32 BufferLength = 0;
/* Ints_Alloc instances, needed for memory alignment management */
@@ -87,14 +83,12 @@
}
}
-
/*Inst_Alloc instances initialization */
InstAlloc_Init( &Instance , pMemoryTable->Region[LVPSA_MEMREGION_INSTANCE].pBaseAddress);
InstAlloc_Init( &Scratch , pMemoryTable->Region[LVPSA_MEMREGION_SCRATCH].pBaseAddress);
InstAlloc_Init( &Data , pMemoryTable->Region[LVPSA_MEMREGION_PERSISTENT_DATA].pBaseAddress);
InstAlloc_Init( &Coef , pMemoryTable->Region[LVPSA_MEMREGION_PERSISTENT_COEF].pBaseAddress);
-
/* Set the instance handle if not already initialised */
if (*phInstance == LVM_NULL)
{
@@ -102,7 +96,6 @@
}
pLVPSA_Inst =(LVPSA_InstancePr_t*)*phInstance;
-
/* Check the memory table for NULL pointers */
for (ii = 0; ii < LVPSA_NR_MEMORY_REGIONS; ii++)
{
@@ -143,39 +136,32 @@
pLVPSA_Inst->SpectralDataBufferLength = BufferLength;
}
-
/* Assign the pointers */
-#ifndef BUILD_FLOAT
- pLVPSA_Inst->pPostGains = InstAlloc_AddMember( &Instance, pInitParams->nBands * sizeof(LVM_UINT16) );
-#else
- pLVPSA_Inst->pPostGains = InstAlloc_AddMember( &Instance, pInitParams->nBands * \
- sizeof(LVM_FLOAT) );
-#endif
- pLVPSA_Inst->pFiltersParams = InstAlloc_AddMember( &Instance, pInitParams->nBands * sizeof(LVPSA_FilterParam_t) );
- pLVPSA_Inst->pSpectralDataBufferStart = InstAlloc_AddMember( &Instance, pInitParams->nBands * pLVPSA_Inst->SpectralDataBufferLength * sizeof(LVM_UINT8) );
- pLVPSA_Inst->pPreviousPeaks = InstAlloc_AddMember( &Instance, pInitParams->nBands * sizeof(LVM_UINT8) );
- pLVPSA_Inst->pBPFiltersPrecision = InstAlloc_AddMember( &Instance, pInitParams->nBands * sizeof(LVPSA_BPFilterPrecision_en) );
-#ifndef BUILD_FLOAT
- pLVPSA_Inst->pBP_Instances = InstAlloc_AddMember( &Coef, pInitParams->nBands * sizeof(Biquad_Instance_t) );
- pLVPSA_Inst->pQPD_States = InstAlloc_AddMember( &Coef, pInitParams->nBands * sizeof(QPD_State_t) );
-#else
- pLVPSA_Inst->pBP_Instances = InstAlloc_AddMember( &Coef, pInitParams->nBands * \
- sizeof(Biquad_FLOAT_Instance_t) );
- pLVPSA_Inst->pQPD_States = InstAlloc_AddMember( &Coef, pInitParams->nBands * \
- sizeof(QPD_FLOAT_State_t) );
-#endif
+ pLVPSA_Inst->pPostGains =
+ (LVM_FLOAT *)InstAlloc_AddMember(&Instance, pInitParams->nBands * sizeof(LVM_FLOAT));
+ pLVPSA_Inst->pFiltersParams = (LVPSA_FilterParam_t *)
+ InstAlloc_AddMember(&Instance, pInitParams->nBands * sizeof(LVPSA_FilterParam_t));
+ pLVPSA_Inst->pSpectralDataBufferStart = (LVM_UINT8 *)
+ InstAlloc_AddMember(&Instance, pInitParams->nBands * \
+ pLVPSA_Inst->SpectralDataBufferLength * sizeof(LVM_UINT8));
+ pLVPSA_Inst->pPreviousPeaks = (LVM_UINT8 *)
+ InstAlloc_AddMember(&Instance, pInitParams->nBands * sizeof(LVM_UINT8));
+ pLVPSA_Inst->pBPFiltersPrecision = (LVPSA_BPFilterPrecision_en *)
+ InstAlloc_AddMember(&Instance, pInitParams->nBands * \
+ sizeof(LVPSA_BPFilterPrecision_en));
+ pLVPSA_Inst->pBP_Instances = (Biquad_FLOAT_Instance_t *)
+ InstAlloc_AddMember(&Coef, pInitParams->nBands * \
+ sizeof(Biquad_FLOAT_Instance_t));
+ pLVPSA_Inst->pQPD_States = (QPD_FLOAT_State_t *)
+ InstAlloc_AddMember(&Coef, pInitParams->nBands * \
+ sizeof(QPD_FLOAT_State_t));
-#ifndef BUILD_FLOAT
- pLVPSA_Inst->pBP_Taps = InstAlloc_AddMember( &Data, pInitParams->nBands * sizeof(Biquad_1I_Order2_Taps_t) );
- pLVPSA_Inst->pQPD_Taps = InstAlloc_AddMember( &Data, pInitParams->nBands * sizeof(QPD_Taps_t) );
-
-#else
- pLVPSA_Inst->pBP_Taps = InstAlloc_AddMember( &Data,
- pInitParams->nBands * \
- sizeof(Biquad_1I_Order2_FLOAT_Taps_t));
- pLVPSA_Inst->pQPD_Taps = InstAlloc_AddMember( &Data, pInitParams->nBands * \
- sizeof(QPD_FLOAT_Taps_t) );
-#endif
+ pLVPSA_Inst->pBP_Taps = (Biquad_1I_Order2_FLOAT_Taps_t *)
+ InstAlloc_AddMember(&Data, pInitParams->nBands * \
+ sizeof(Biquad_1I_Order2_FLOAT_Taps_t));
+ pLVPSA_Inst->pQPD_Taps = (QPD_FLOAT_Taps_t *)
+ InstAlloc_AddMember(&Data, pInitParams->nBands * \
+ sizeof(QPD_FLOAT_Taps_t));
/* Copy filters parameters in the private instance */
for(ii = 0; ii < pLVPSA_Inst->nBands; ii++)
@@ -186,16 +172,11 @@
/* Set Post filters gains*/
for(ii = 0; ii < pLVPSA_Inst->nBands; ii++)
{
-#ifndef BUILD_FLOAT
- pLVPSA_Inst->pPostGains[ii] =(LVM_UINT16) LVPSA_GainTable[pInitParams->pFiltersParams[ii].PostGain + 15];
-#else
pLVPSA_Inst->pPostGains[ii] = LVPSA_Float_GainTable[15 + \
pInitParams->pFiltersParams[ii].PostGain];
-#endif
}
pLVPSA_Inst->pSpectralDataBufferWritePointer = pLVPSA_Inst->pSpectralDataBufferStart;
-
/* Initialize control dependant internal parameters */
errorCode = LVPSA_Control (*phInstance, pControlParams);
diff --git a/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Memory.c b/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Memory.cpp
similarity index 92%
rename from media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Memory.c
rename to media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Memory.cpp
index 06a8f9d..eafcbe6 100644
--- a/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Memory.c
+++ b/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Memory.cpp
@@ -59,19 +59,16 @@
INST_ALLOC Coef;
LVPSA_InstancePr_t *pLVPSA_Inst = (LVPSA_InstancePr_t*)hInstance;
-
InstAlloc_Init( &Instance , LVM_NULL);
InstAlloc_Init( &Scratch , LVM_NULL);
InstAlloc_Init( &Data , LVM_NULL);
InstAlloc_Init( &Coef , LVM_NULL);
-
if((pMemoryTable == LVM_NULL) || (pInitParams == LVM_NULL))
{
return(LVPSA_ERROR_NULLADDRESS);
}
-
/*
* Fill in the memory table
*/
@@ -106,11 +103,7 @@
*/
InstAlloc_AddMember( &Instance, sizeof(LVPSA_InstancePr_t) );
-#ifdef BUILD_FLOAT
InstAlloc_AddMember( &Instance, pInitParams->nBands * sizeof(LVM_FLOAT) );
-#else
- InstAlloc_AddMember( &Instance, pInitParams->nBands * sizeof(LVM_UINT16) );
-#endif
InstAlloc_AddMember( &Instance, pInitParams->nBands * sizeof(LVPSA_FilterParam_t) );
{
@@ -123,7 +116,6 @@
BufferLength=(LVM_UINT32)BL;
}
-
if((BufferLength * LVPSA_InternalRefreshTime) != pInitParams->SpectralDataBufferDuration)
{
BufferLength++;
@@ -138,11 +130,7 @@
/*
* Scratch memory
*/
-#ifndef BUILD_FLOAT
- InstAlloc_AddMember( &Scratch, 2 * pInitParams->MaxInputBlockSize * sizeof(LVM_INT16) );
-#else
InstAlloc_AddMember( &Scratch, 2 * pInitParams->MaxInputBlockSize * sizeof(LVM_FLOAT) );
-#endif
pMemoryTable->Region[LVPSA_MEMREGION_SCRATCH].Size = InstAlloc_GetTotal(&Scratch);
pMemoryTable->Region[LVPSA_MEMREGION_SCRATCH].Type = LVPSA_SCRATCH;
pMemoryTable->Region[LVPSA_MEMREGION_SCRATCH].pBaseAddress = LVM_NULL;
@@ -150,13 +138,8 @@
/*
* Persistent coefficients memory
*/
-#ifndef BUILD_FLOAT
- InstAlloc_AddMember( &Coef, pInitParams->nBands * sizeof(Biquad_Instance_t) );
- InstAlloc_AddMember( &Coef, pInitParams->nBands * sizeof(QPD_State_t) );
-#else
InstAlloc_AddMember( &Coef, pInitParams->nBands * sizeof(Biquad_FLOAT_Instance_t) );
InstAlloc_AddMember( &Coef, pInitParams->nBands * sizeof(QPD_FLOAT_State_t) );
-#endif
pMemoryTable->Region[LVPSA_MEMREGION_PERSISTENT_COEF].Size = InstAlloc_GetTotal(&Coef);
pMemoryTable->Region[LVPSA_MEMREGION_PERSISTENT_COEF].Type = LVPSA_PERSISTENT_COEF;
pMemoryTable->Region[LVPSA_MEMREGION_PERSISTENT_COEF].pBaseAddress = LVM_NULL;
@@ -164,13 +147,8 @@
/*
* Persistent data memory
*/
-#ifndef BUILD_FLOAT
- InstAlloc_AddMember( &Data, pInitParams->nBands * sizeof(Biquad_1I_Order2_Taps_t) );
- InstAlloc_AddMember( &Data, pInitParams->nBands * sizeof(QPD_Taps_t) );
-#else
InstAlloc_AddMember( &Data, pInitParams->nBands * sizeof(Biquad_1I_Order2_FLOAT_Taps_t) );
InstAlloc_AddMember( &Data, pInitParams->nBands * sizeof(QPD_FLOAT_Taps_t) );
-#endif
pMemoryTable->Region[LVPSA_MEMREGION_PERSISTENT_DATA].Size = InstAlloc_GetTotal(&Data);
pMemoryTable->Region[LVPSA_MEMREGION_PERSISTENT_DATA].Type = LVPSA_PERSISTENT_DATA;
pMemoryTable->Region[LVPSA_MEMREGION_PERSISTENT_DATA].pBaseAddress = LVM_NULL;
diff --git a/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Private.h b/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Private.h
index ee07e2e..61987b5 100644
--- a/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Private.h
+++ b/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Private.h
@@ -23,12 +23,6 @@
#include "LVPSA_QPD.h"
#include "LVM_Macros.h"
-
-
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
/**********************************************************************************
CONSTANT DEFINITIONS
***********************************************************************************/
@@ -43,11 +37,7 @@
#define LVPSA_MEMREGION_PERSISTENT_COEF 1 /* Offset to persistent coefficients memory region in memory table */
#define LVPSA_MEMREGION_PERSISTENT_DATA 2 /* Offset to persistent taps memory region in memory table */
#define LVPSA_MEMREGION_SCRATCH 3 /* Offset to scratch memory region in memory table */
-#ifndef HIGHER_FS
-#define LVPSA_NR_SUPPORTED_RATE 9 /* From 8000Hz to 48000Hz*/
-#else
#define LVPSA_NR_SUPPORTED_RATE 13 /* From 8000Hz to 192000Hz*/
-#endif
#define LVPSA_NR_SUPPORTED_SPEED 3 /* LOW, MEDIUM, HIGH */
#define LVPSA_MAXBUFFERDURATION 4000 /* Maximum length in ms of the levels buffer */
@@ -77,7 +67,6 @@
#define LVPSA_InternalRefreshTimeInv 0x0666 /* 1/20ms left shifted by 15 */
#define LVPSA_InternalRefreshTimeShift 15
-
/* Precision of the filter */
typedef enum
{
@@ -96,12 +85,6 @@
LVPSA_MemTab_t MemoryTable;
LVPSA_BPFilterPrecision_en *pBPFiltersPrecision; /* Points a nBands elements array that contains the filter precision for each band */
-#ifndef BUILD_FLOAT
- Biquad_Instance_t *pBP_Instances; /* Points a nBands elements array that contains the band pass filter instance for each band */
- Biquad_1I_Order2_Taps_t *pBP_Taps; /* Points a nBands elements array that contains the band pass filter taps for each band */
- QPD_State_t *pQPD_States; /* Points a nBands elements array that contains the QPD filter instance for each band */
- QPD_Taps_t *pQPD_Taps; /* Points a nBands elements array that contains the QPD filter taps for each band */
-#else
Biquad_FLOAT_Instance_t *pBP_Instances;
/* Points a nBands elements array that contains the band pass filter taps for each band */
Biquad_1I_Order2_FLOAT_Taps_t *pBP_Taps;
@@ -109,17 +92,11 @@
QPD_FLOAT_State_t *pQPD_States;
/* Points a nBands elements array that contains the QPD filter taps for each band */
QPD_FLOAT_Taps_t *pQPD_Taps;
-#endif
-#ifndef BUILD_FLOAT
- LVM_UINT16 *pPostGains; /* Points a nBands elements array that contains the post-filter gains for each band */
-#else
/* Points a nBands elements array that contains the post-filter gains for each band */
LVM_FLOAT *pPostGains;
-#endif
LVPSA_FilterParam_t *pFiltersParams; /* Copy of the filters parameters from the input parameters */
-
LVM_UINT16 nSamplesBufferUpdate; /* Number of samples to make 20ms */
LVM_INT32 BufferUpdateSamplesCount; /* Counter used to know when to put a new value in the buffer */
LVM_UINT16 nRelevantFilters; /* Number of relevent filters depending on sampling frequency and bands center frequency */
@@ -140,8 +117,6 @@
}LVPSA_InstancePr_t, *pLVPSA_InstancePr_t;
-
-
/**********************************************************************************
FUNCTIONS PROTOTYPE
***********************************************************************************/
@@ -162,8 +137,4 @@
/************************************************************************************/
LVPSA_RETURN LVPSA_ApplyNewSettings (LVPSA_InstancePr_t *pInst);
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
-
#endif /* _LVPSA_PRIVATE_H */
diff --git a/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Process.c b/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Process.cpp
similarity index 76%
rename from media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Process.c
rename to media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Process.cpp
index 61899fe..81a88c5 100644
--- a/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Process.c
+++ b/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Process.cpp
@@ -54,7 +54,6 @@
/* otherwise Error due to bad parameters */
/* */
/************************************************************************************/
-#ifdef BUILD_FLOAT
LVPSA_RETURN LVPSA_Process ( pLVPSA_Handle_t hInstance,
LVM_FLOAT *pLVPSA_InputSamples,
LVM_UINT16 InputBlockSize,
@@ -121,7 +120,6 @@
break;
}
-
LVPSA_QPD_Process_Float ( pLVPSA_Inst,
pScratch + InputBlockSize,
(LVM_INT16)InputBlockSize,
@@ -143,95 +141,6 @@
return(LVPSA_OK);
}
-#else
-LVPSA_RETURN LVPSA_Process ( pLVPSA_Handle_t hInstance,
- LVM_INT16 *pLVPSA_InputSamples,
- LVM_UINT16 InputBlockSize,
- LVPSA_Time AudioTime )
-
-{
- LVPSA_InstancePr_t *pLVPSA_Inst = (LVPSA_InstancePr_t*)hInstance;
- LVM_INT16 *pScratch;
- LVM_INT16 ii;
- LVM_INT32 AudioTimeInc;
- extern LVM_UINT32 LVPSA_SampleRateInvTab[];
- LVM_UINT8 *pWrite_Save; /* Position of the write pointer at the beginning of the process */
-
- /******************************************************************************
- CHECK PARAMETERS
- *******************************************************************************/
- if(hInstance == LVM_NULL || pLVPSA_InputSamples == LVM_NULL)
- {
- return(LVPSA_ERROR_NULLADDRESS);
- }
- if(InputBlockSize == 0 || InputBlockSize > pLVPSA_Inst->MaxInputBlockSize)
- {
- return(LVPSA_ERROR_INVALIDPARAM);
- }
-
- pScratch = (LVM_INT16*)pLVPSA_Inst->MemoryTable.Region[LVPSA_MEMREGION_SCRATCH].pBaseAddress;
- pWrite_Save = pLVPSA_Inst->pSpectralDataBufferWritePointer;
-
- /******************************************************************************
- APPLY NEW SETTINGS IF NEEDED
- *******************************************************************************/
- if (pLVPSA_Inst->bControlPending == LVM_TRUE)
- {
- pLVPSA_Inst->bControlPending = 0;
- LVPSA_ApplyNewSettings( pLVPSA_Inst);
- }
-
- /******************************************************************************
- PROCESS SAMPLES
- *******************************************************************************/
- /* Put samples in range [-0.5;0.5[ for BP filters (see Biquads documentation) */
- Copy_16( pLVPSA_InputSamples,pScratch,(LVM_INT16)InputBlockSize);
- Shift_Sat_v16xv16(-1,pScratch,pScratch,(LVM_INT16)InputBlockSize);
-
- for (ii = 0; ii < pLVPSA_Inst->nRelevantFilters; ii++)
- {
- switch(pLVPSA_Inst->pBPFiltersPrecision[ii])
- {
- case LVPSA_SimplePrecisionFilter:
- BP_1I_D16F16C14_TRC_WRA_01 ( &pLVPSA_Inst->pBP_Instances[ii],
- pScratch,
- pScratch + InputBlockSize,
- (LVM_INT16)InputBlockSize);
- break;
-
- case LVPSA_DoublePrecisionFilter:
- BP_1I_D16F32C30_TRC_WRA_01 ( &pLVPSA_Inst->pBP_Instances[ii],
- pScratch,
- pScratch + InputBlockSize,
- (LVM_INT16)InputBlockSize);
- break;
- default:
- break;
- }
-
-
- LVPSA_QPD_Process ( pLVPSA_Inst,
- pScratch + InputBlockSize,
- (LVM_INT16)InputBlockSize,
- ii);
- }
-
- /******************************************************************************
- UPDATE SpectralDataBufferAudioTime
- *******************************************************************************/
-
- if(pLVPSA_Inst->pSpectralDataBufferWritePointer != pWrite_Save)
- {
- MUL32x32INTO32((AudioTime + (LVM_INT32)((LVM_INT32)pLVPSA_Inst->LocalSamplesCount*1000)),
- (LVM_INT32)LVPSA_SampleRateInvTab[pLVPSA_Inst->CurrentParams.Fs],
- AudioTimeInc,
- LVPSA_FsInvertShift)
- pLVPSA_Inst->SpectralDataBufferAudioTime = AudioTime + AudioTimeInc;
- }
-
- return(LVPSA_OK);
-}
-#endif
/************************************************************************************/
/* */
@@ -269,7 +178,6 @@
return(LVPSA_ERROR_NULLADDRESS);
}
-
/* First find the place where to look in the status buffer */
if(GetSpectrumAudioTime <= pLVPSA_Inst->SpectralDataBufferAudioTime)
{
@@ -320,7 +228,6 @@
pRead = pLVPSA_Inst->pSpectralDataBufferWritePointer - StatusDelta * pLVPSA_Inst->nBands;
}
-
/* Read the status buffer and fill the output buffers */
for(ii = 0; ii < pLVPSA_Inst->nBands; ii++)
{
diff --git a/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_QPD.h b/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_QPD.h
index 99d844b..609a485 100644
--- a/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_QPD.h
+++ b/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_QPD.h
@@ -20,25 +20,18 @@
#include "LVM_Types.h"
-
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
typedef struct
{
LVM_INT32 *pDelay; /* pointer to the delayed samples (data of 32 bits) */
LVM_INT32 Coefs[2]; /* pointer to the filter coefficients */
}QPD_State_t, *pQPD_State_t;
-#ifdef BUILD_FLOAT
typedef struct
{
/* pointer to the delayed samples (data of 32 bits) */
LVM_FLOAT *pDelay;
LVM_FLOAT Coefs[2]; /* pointer to the filter coefficients */
}QPD_FLOAT_State_t, *pQPD_FLOAT_State_t;
-#endif
typedef struct
{
@@ -47,15 +40,12 @@
} QPD_C32_Coefs, *PQPD_C32_Coefs;
-#ifdef BUILD_FLOAT
typedef struct
{
LVM_FLOAT KP; /*should store a0*/
LVM_FLOAT KM; /*should store b2*/
} QPD_FLOAT_Coefs, *PQPD_FLOAT_Coefs;
-#endif
-
typedef struct
{
@@ -63,14 +53,12 @@
} QPD_Taps_t, *pQPD_Taps_t;
-#ifdef BUILD_FLOAT
typedef struct
{
LVM_FLOAT Storage[1];
} QPD_FLOAT_Taps_t, *pQPD_FLOAT_Taps_t;
-#endif
/************************************************************************************/
/* */
/* FUNCTION: LVPSA_QPD_Process */
@@ -89,12 +77,10 @@
LVM_INT16 numSamples,
LVM_INT16 BandIndex);
-#ifdef BUILD_FLOAT
void LVPSA_QPD_Process_Float ( void *hInstance,
LVM_FLOAT *pInSamps,
LVM_INT16 numSamples,
LVM_INT16 BandIndex);
-#endif
/************************************************************************************/
/* */
/* FUNCTION: LVPSA_QPD_Init */
@@ -113,15 +99,10 @@
void LVPSA_QPD_Init ( QPD_State_t *pInstance,
QPD_Taps_t *pTaps,
QPD_C32_Coefs *pCoef );
-#ifdef BUILD_FLOAT
void LVPSA_QPD_Init_Float ( QPD_FLOAT_State_t *pInstance,
QPD_FLOAT_Taps_t *pTaps,
QPD_FLOAT_Coefs *pCoef );
-#endif
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
#endif
diff --git a/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_QPD_Init.c b/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_QPD_Init.cpp
similarity index 98%
rename from media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_QPD_Init.c
rename to media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_QPD_Init.cpp
index 2cc32ab..2dbf694 100644
--- a/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_QPD_Init.c
+++ b/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_QPD_Init.cpp
@@ -41,7 +41,6 @@
pQPD_State->Coefs[1] = pCoef->KM;
}
-#ifdef BUILD_FLOAT
void LVPSA_QPD_Init_Float ( pQPD_FLOAT_State_t pQPD_State,
QPD_FLOAT_Taps_t *pTaps,
QPD_FLOAT_Coefs *pCoef )
@@ -50,4 +49,3 @@
pQPD_State->Coefs[0] = ((LVM_FLOAT)pCoef->KP);
pQPD_State->Coefs[1] = ((LVM_FLOAT)pCoef->KM);
}
-#endif
diff --git a/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_QPD_Process.c b/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_QPD_Process.cpp
similarity index 71%
rename from media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_QPD_Process.c
rename to media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_QPD_Process.cpp
index e233172..8805420 100644
--- a/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_QPD_Process.c
+++ b/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_QPD_Process.cpp
@@ -39,12 +39,10 @@
LVM_INT16 BandIndex,
LVM_INT16 Value );
-#ifdef BUILD_FLOAT
void LVPSA_QPD_WritePeak_Float( pLVPSA_InstancePr_t pLVPSA_Inst,
LVM_UINT8 **ppWrite,
LVM_INT16 BandIndex,
LVM_FLOAT Value );
-#endif
/************************************************************************************/
/* */
/* FUNCTION: LVPSA_QPD_Process */
@@ -58,127 +56,6 @@
/* RETURNS: void */
/* */
/************************************************************************************/
-#ifndef BUILD_FLOAT
-void LVPSA_QPD_Process ( void *hInstance,
- LVM_INT16 *pInSamps,
- LVM_INT16 numSamples,
- LVM_INT16 BandIndex)
-{
-
- /******************************************************************************
- PARAMETERS
- *******************************************************************************/
- LVPSA_InstancePr_t *pLVPSA_Inst = (LVPSA_InstancePr_t*)hInstance;
- QPD_State_t *pQPDState = (QPD_State_t*)&pLVPSA_Inst->pQPD_States[BandIndex];
-
- /* Pointer to taps */
- LVM_INT32* pDelay = pQPDState->pDelay;
-
- /* Parameters needed during quasi peak calculations */
- LVM_INT32 X0;
- LVM_INT32 temp,temp2;
- LVM_INT32 accu;
- LVM_INT16 Xg0;
- LVM_INT16 D0;
- LVM_INT16 V0 = (LVM_INT16)(*pDelay);
-
- /* Filter's coef */
- LVM_INT32 Kp = pQPDState->Coefs[0];
- LVM_INT32 Km = pQPDState->Coefs[1];
-
- LVM_INT16 ii = numSamples;
-
- LVM_UINT8 *pWrite = pLVPSA_Inst->pSpectralDataBufferWritePointer;
- LVM_INT32 BufferUpdateSamplesCount = pLVPSA_Inst->BufferUpdateSamplesCount;
- LVM_UINT16 DownSamplingFactor = pLVPSA_Inst->DownSamplingFactor;
-
- /******************************************************************************
- INITIALIZATION
- *******************************************************************************/
- /* Correct the pointer to take the first down sampled signal sample */
- pInSamps += pLVPSA_Inst->DownSamplingCount;
- /* Correct also the number of samples */
- ii = (LVM_INT16)(ii - (LVM_INT16)pLVPSA_Inst->DownSamplingCount);
-
- while (ii > 0)
- {
- /* Apply post gain */
- X0 = ((*pInSamps) * pLVPSA_Inst->pPostGains[BandIndex]) >> (LVPSA_GAINSHIFT-1); /* - 1 to compensate scaling in process function*/
- pInSamps = pInSamps + DownSamplingFactor;
-
- /* Saturate and take absolute value */
- if(X0 < 0)
- X0 = -X0;
- if (X0 > 0x7FFF)
- Xg0 = 0x7FFF;
- else
- Xg0 = (LVM_INT16)(X0);
-
-
- /* Quasi peak filter calculation */
- D0 = (LVM_INT16)(Xg0 - V0);
-
- temp2 = (LVM_INT32)D0;
- MUL32x32INTO32(temp2,Kp,accu,31);
-
- D0 = (LVM_INT16)(D0>>1);
- if (D0 < 0){
- D0 = (LVM_INT16)(-D0);
- }
-
- temp2 = (LVM_INT32)D0;
- MUL32x32INTO32((LVM_INT32)D0,Km,temp,31);
- accu +=temp + Xg0;
-
- if (accu > 0x7FFF)
- accu = 0x7FFF;
- else if(accu < 0)
- accu = 0x0000;
-
- V0 = (LVM_INT16)accu;
-
- if(((pLVPSA_Inst->nSamplesBufferUpdate - BufferUpdateSamplesCount) < DownSamplingFactor))
- {
- LVPSA_QPD_WritePeak( pLVPSA_Inst,
- &pWrite,
- BandIndex,
- V0);
- BufferUpdateSamplesCount -= pLVPSA_Inst->nSamplesBufferUpdate;
- pLVPSA_Inst->LocalSamplesCount = (LVM_UINT16)(numSamples - ii);
- }
- BufferUpdateSamplesCount+=DownSamplingFactor;
-
- ii = (LVM_INT16)(ii-DownSamplingFactor);
-
- }
-
- /* Store last taps in memory */
- *pDelay = (LVM_INT32)(V0);
-
- /* If this is the last call to the function after last band processing,
- update the parameters. */
- if(BandIndex == (pLVPSA_Inst->nRelevantFilters-1))
- {
- pLVPSA_Inst->pSpectralDataBufferWritePointer = pWrite;
- /* Adjustment for 11025Hz input, 220,5 is normally
- the exact number of samples for 20ms.*/
- if((pLVPSA_Inst->pSpectralDataBufferWritePointer != pWrite)&&(pLVPSA_Inst->CurrentParams.Fs == LVM_FS_11025))
- {
- if(pLVPSA_Inst->nSamplesBufferUpdate == 220)
- {
- pLVPSA_Inst->nSamplesBufferUpdate = 221;
- }
- else
- {
- pLVPSA_Inst->nSamplesBufferUpdate = 220;
- }
- }
- pLVPSA_Inst->pSpectralDataBufferWritePointer = pWrite;
- pLVPSA_Inst->BufferUpdateSamplesCount = BufferUpdateSamplesCount;
- pLVPSA_Inst->DownSamplingCount = (LVM_UINT16)(-ii);
- }
-}
-#else
void LVPSA_QPD_Process_Float ( void *hInstance,
LVM_FLOAT *pInSamps,
LVM_INT16 numSamples,
@@ -235,7 +112,6 @@
else
Xg0 =X0;
-
/* Quasi peak filter calculation */
D0 = Xg0 - V0;
@@ -302,7 +178,6 @@
pLVPSA_Inst->DownSamplingCount = (LVM_UINT16)(-ii);
}
}
-#endif
/************************************************************************************/
/* */
/* FUNCTION: LVPSA_QPD_WritePeak */
@@ -326,7 +201,6 @@
{
LVM_UINT8 *pWrite = *ppWrite;
-
/* Write the value and update the write pointer */
*(pWrite + BandIndex) = (LVM_UINT8)(Value>>7);
pWrite += pLVPSA_Inst->nBands;
@@ -338,7 +212,6 @@
*ppWrite = pWrite;
}
-#ifdef BUILD_FLOAT
void LVPSA_QPD_WritePeak_Float( pLVPSA_InstancePr_t pLVPSA_Inst,
LVM_UINT8 **ppWrite,
LVM_INT16 BandIndex,
@@ -357,4 +230,3 @@
*ppWrite = pWrite;
}
-#endif
diff --git a/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Tables.c b/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Tables.cpp
similarity index 87%
rename from media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Tables.c
rename to media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Tables.cpp
index f8af496..9f0aa02 100644
--- a/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Tables.c
+++ b/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Tables.cpp
@@ -15,7 +15,6 @@
* limitations under the License.
*/
-
/************************************************************************************/
/* */
/* Includes */
@@ -24,6 +23,7 @@
#include "LVPSA.h"
#include "LVPSA_QPD.h"
+#include "LVPSA_Tables.h"
/************************************************************************************/
/* */
/* Sample rate table */
@@ -34,17 +34,6 @@
* Sample rate table for converting between the enumerated type and the actual
* frequency
*/
-#ifndef HIGHER_FS
-const LVM_UINT16 LVPSA_SampleRateTab[] = { 8000, /* 8kS/s */
- 11025,
- 12000,
- 16000,
- 22050,
- 24000,
- 32000,
- 44100,
- 48000}; /* 48kS/s */
-#else
const LVM_UINT32 LVPSA_SampleRateTab[] = { 8000, /* 8kS/s */
11025,
12000,
@@ -58,7 +47,6 @@
96000,
176400,
192000}; /* 192kS/s */
-#endif
/************************************************************************************/
/* */
@@ -79,16 +67,12 @@
67109,
48696,
44739
-#ifdef HIGHER_FS
,24348
,22369
,12174
,11185 /* 192kS/s */
-#endif
};
-
-
/************************************************************************************/
/* */
/* Number of samples in 20ms */
@@ -108,12 +92,10 @@
640,
882,
960
-#ifdef HIGHER_FS
,1764
,1920
,3528
,3840 /* 192kS/s */
-#endif
};
/************************************************************************************/
/* */
@@ -133,15 +115,12 @@
21, /* 32000 S/s */
30, /* 44100 S/s */
32 /* 48000 S/s */
-#ifdef HIGHER_FS
,60 /* 88200 S/s */
,64 /* 96000 S/s */
,120 /* 176400 S/s */
,128 /*192000 S/s */
-#endif
};
-
/************************************************************************************/
/* */
/* Coefficient calculation tables */
@@ -160,15 +139,12 @@
6588,
4781,
4392
-#ifdef HIGHER_FS
,2390
,2196
,1195
,1098 /* 192kS/s */
-#endif
};
-#ifdef BUILD_FLOAT
const LVM_FLOAT LVPSA_Float_TwoPiOnFsTable[] = { 0.8042847f, /* 8kS/s */
0.5836054f,
0.5361796f,
@@ -178,15 +154,12 @@
0.2010559f,
0.1459089f,
0.1340372f
-#ifdef HIGHER_FS
,0.0729476f
,0.0670186f
,0.0364738f
,0.0335093f /* 192kS/s */
-#endif
};
-#endif
/*
* Gain table
*/
@@ -222,7 +195,6 @@
10264,
11576}; /* +15dB gain */
-#ifdef BUILD_FLOAT
const LVM_FLOAT LVPSA_Float_GainTable[]={ 0.177734375f, /* -15dB gain */
0.199218750f,
0.223632812f,
@@ -254,7 +226,6 @@
4.466796875f,
5.011718750f,
5.652343750f}; /* +15dB gain */
-#endif
/************************************************************************************/
/* */
/* Cosone polynomial coefficients */
@@ -277,7 +248,6 @@
-2671, /* a3 */
23730, /* a4 */
-9490}; /* a5 */
-#ifdef BUILD_FLOAT
const LVM_FLOAT LVPSA_Float_CosCoef[] = { 3, /* Shifts */
0.1250038f, /* a0 */
-0.0010986f, /* a1 */
@@ -285,7 +255,6 @@
-0.0815149f, /* a3 */
0.7242042f, /* a4 */
-0.2896206f}; /* a5 */
-#endif
/*
* Coefficients for calculating the cosine error with the equation:
*
@@ -305,51 +274,50 @@
-6, /* a1 */
16586, /* a2 */
-44}; /* a3 */
-#ifdef BUILD_FLOAT
const LVM_FLOAT LVPSA_Float_DPCosCoef[] = {1.0f, /* Shifts */
0.0f, /* a0 */
-0.00008311f, /* a1 */
0.50617999f, /* a2 */
-0.00134281f}; /* a3 */
-#endif
/************************************************************************************/
/* */
/* Quasi peak filter coefficients table */
/* */
/************************************************************************************/
const QPD_C32_Coefs LVPSA_QPD_Coefs[] = {
+ /* 8kS/s */ /* LVPSA_SPEED_LOW */
+ {(LVM_INT32)0x80CEFD2B,0x00CB9B17},
+ {(LVM_INT32)0x80D242E7,0x00CED11D},
+ {(LVM_INT32)0x80DCBAF5,0x00D91679},
+ {(LVM_INT32)0x80CEFD2B,0x00CB9B17},
+ {(LVM_INT32)0x80E13739,0x00DD7CD3},
+ {(LVM_INT32)0x80DCBAF5,0x00D91679},
+ {(LVM_INT32)0x80D94BAF,0x00D5B7E7},
+ {(LVM_INT32)0x80E13739,0x00DD7CD3},
+ {(LVM_INT32)0x80DCBAF5,0x00D91679}, /* 48kS/s */
- {0x80CEFD2B,0x00CB9B17}, /* 8kS/s */ /* LVPSA_SPEED_LOW */
- {0x80D242E7,0x00CED11D},
- {0x80DCBAF5,0x00D91679},
- {0x80CEFD2B,0x00CB9B17},
- {0x80E13739,0x00DD7CD3},
- {0x80DCBAF5,0x00D91679},
- {0x80D94BAF,0x00D5B7E7},
- {0x80E13739,0x00DD7CD3},
- {0x80DCBAF5,0x00D91679}, /* 48kS/s */
+ /* 8kS/s */ /* LVPSA_SPEED_MEDIUM */
+ {(LVM_INT32)0x8587513D,0x055C22CF},
+ {(LVM_INT32)0x859D2967,0x0570F007},
+ {(LVM_INT32)0x85E2EFAC,0x05B34D79},
+ {(LVM_INT32)0x8587513D,0x055C22CF},
+ {(LVM_INT32)0x8600C7B9,0x05CFA6CF},
+ {(LVM_INT32)0x85E2EFAC,0x05B34D79},
+ {(LVM_INT32)0x85CC1018,0x059D8F69},
+ {(LVM_INT32)0x8600C7B9,0x05CFA6CF},
+ {(LVM_INT32)0x85E2EFAC,0x05B34D79}, /* 48kS/s */
- {0x8587513D,0x055C22CF}, /* 8kS/s */ /* LVPSA_SPEED_MEDIUM */
- {0x859D2967,0x0570F007},
- {0x85E2EFAC,0x05B34D79},
- {0x8587513D,0x055C22CF},
- {0x8600C7B9,0x05CFA6CF},
- {0x85E2EFAC,0x05B34D79},
- {0x85CC1018,0x059D8F69},
- {0x8600C7B9,0x05CFA6CF},//{0x8600C7B9,0x05CFA6CF},
- {0x85E2EFAC,0x05B34D79}, /* 48kS/s */
+ /* 8kS/s */ /* LVPSA_SPEED_HIGH */
+ {(LVM_INT32)0xA115EA7A,0x1CDB3F5C},
+ {(LVM_INT32)0xA18475F0,0x1D2C83A2},
+ {(LVM_INT32)0xA2E1E950,0x1E2A532E},
+ {(LVM_INT32)0xA115EA7A,0x1CDB3F5C},
+ {(LVM_INT32)0xA375B2C6,0x1E943BBC},
+ {(LVM_INT32)0xA2E1E950,0x1E2A532E},
+ {(LVM_INT32)0xA26FF6BD,0x1DD81530},
+ {(LVM_INT32)0xA375B2C6,0x1E943BBC},
+ {(LVM_INT32)0xA2E1E950,0x1E2A532E}}; /* 48kS/s */
- {0xA115EA7A,0x1CDB3F5C}, /* 8kS/s */ /* LVPSA_SPEED_HIGH */
- {0xA18475F0,0x1D2C83A2},
- {0xA2E1E950,0x1E2A532E},
- {0xA115EA7A,0x1CDB3F5C},
- {0xA375B2C6,0x1E943BBC},
- {0xA2E1E950,0x1E2A532E},
- {0xA26FF6BD,0x1DD81530},
- {0xA375B2C6,0x1E943BBC},
- {0xA2E1E950,0x1E2A532E}}; /* 48kS/s */
-
-#ifdef BUILD_FLOAT
const QPD_FLOAT_Coefs LVPSA_QPD_Float_Coefs[] = {
/* 8kS/s */ /* LVPSA_SPEED_LOW */
@@ -363,12 +331,10 @@
{-0.9931269618682563f,0.0067592649720609f},
/* 48kS/s */
{-0.9932638457976282f,0.0066249934025109f},
-#ifdef HIGHER_FS
{-0.9931269618682563f,0.0067592649720609f},
{-0.9932638457976282f,0.0066249934025109f},
{-0.9931269618682563f,0.0067592649720609f},
{-0.9932638457976282f,0.0066249934025109f},
-#endif
/* 8kS/s */ /* LVPSA_SPEED_MEDIUM */
{-0.9568079425953329f,0.0418742666952312f},
{-0.9561413046903908f,0.0425090822391212f},
@@ -381,12 +347,10 @@
{-0.9531011912040412f,0.0453995238058269f},
/* 48kS/s */
{-0.9540119562298059f,0.0445343819446862f},
-#ifdef HIGHER_FS
{-0.9531011912040412f,0.0453995238058269f},
{-0.9540119562298059f,0.0445343819446862f},
{-0.9531011912040412f,0.0453995238058269f},
{-0.9540119562298059f,0.0445343819446862f},
-#endif
/* 8kS/s */ /* LVPSA_SPEED_HIGH */
{-0.7415186790749431f,0.2254409026354551f},
{-0.7381451204419136f,0.2279209652915597f},
@@ -398,11 +362,8 @@
{-0.7229706319049001f,0.2388987224549055f},
/* 48kS/s */
{-0.7274807319045067f,0.2356666540727019f}
-#ifdef HIGHER_FS
,{-0.7229706319049001f,0.2388987224549055f}
,{-0.7274807319045067f,0.2356666540727019f}
,{-0.7229706319049001f,0.2388987224549055f}
,{-0.7274807319045067f,0.2356666540727019f}
-#endif
};
-#endif
diff --git a/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Tables.h b/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Tables.h
new file mode 100644
index 0000000..65872fe
--- /dev/null
+++ b/media/libeffects/lvm/lib/SpectrumAnalyzer/src/LVPSA_Tables.h
@@ -0,0 +1,128 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __LVPSA_TABLES_H__
+#define __LVPSA_TABLES_H__
+
+/************************************************************************************/
+/* */
+/* Sample rate table */
+/* */
+/************************************************************************************/
+
+/*
+ * Sample rate table for converting between the enumerated type and the actual
+ * frequency
+ */
+extern const LVM_UINT32 LVPSA_SampleRateTab[];
+
+/************************************************************************************/
+/* */
+/* Sample rate inverse table */
+/* */
+/************************************************************************************/
+
+/*
+ * Sample rate table for converting between the enumerated type and the actual
+ * frequency
+ */
+extern const LVM_UINT32 LVPSA_SampleRateInvTab[];
+
+/************************************************************************************/
+/* */
+/* Number of samples in 20ms */
+/* */
+/************************************************************************************/
+
+/*
+ * Table for converting between the enumerated type and the number of samples
+ * during 20ms
+ */
+extern const LVM_UINT16 LVPSA_nSamplesBufferUpdate[];
+
+/************************************************************************************/
+/* */
+/* Down sampling factors */
+/* */
+/************************************************************************************/
+
+/*
+ * Table for converting between the enumerated type and the down sampling factor
+ */
+extern const LVM_UINT16 LVPSA_DownSamplingFactor[];
+
+/************************************************************************************/
+/* */
+/* Coefficient calculation tables */
+/* */
+/************************************************************************************/
+
+/*
+ * Table for 2 * Pi / Fs
+ */
+extern const LVM_INT16 LVPSA_TwoPiOnFsTable[];
+extern const LVM_FLOAT LVPSA_Float_TwoPiOnFsTable[];
+
+/*
+ * Gain table
+ */
+extern const LVM_INT16 LVPSA_GainTable[];
+extern const LVM_FLOAT LVPSA_Float_GainTable[];
+
+/************************************************************************************/
+/* */
+/* Cosone polynomial coefficients */
+/* */
+/************************************************************************************/
+
+/*
+ * Coefficients for calculating the cosine with the equation:
+ *
+ * Cos(x) = (2^Shifts)*(a0 + a1*x + a2*x^2 + a3*x^3 + a4*x^4 + a5*x^5)
+ *
+ * These coefficients expect the input, x, to be in the range 0 to 32768 respresenting
+ * a range of 0 to Pi. The output is in the range 32767 to -32768 representing the range
+ * +1.0 to -1.0
+ */
+extern const LVM_INT16 LVPSA_CosCoef[];
+extern const LVM_FLOAT LVPSA_Float_CosCoef[];
+
+/*
+ * Coefficients for calculating the cosine error with the equation:
+ *
+ * CosErr(x) = (2^Shifts)*(a0 + a1*x + a2*x^2 + a3*x^3)
+ *
+ * These coefficients expect the input, x, to be in the range 0 to 32768 respresenting
+ * a range of 0 to Pi/25. The output is in the range 0 to 32767 representing the range
+ * 0.0 to 0.0078852986
+ *
+ * This is used to give a double precision cosine over the range 0 to Pi/25 using the
+ * the equation:
+ *
+ * Cos(x) = 1.0 - CosErr(x)
+ */
+extern const LVM_INT16 LVPSA_DPCosCoef[];
+extern const LVM_FLOAT LVPSA_Float_DPCosCoef[];
+
+/************************************************************************************/
+/* */
+/* Quasi peak filter coefficients table */
+/* */
+/************************************************************************************/
+extern const QPD_C32_Coefs LVPSA_QPD_Coefs[];
+extern const QPD_FLOAT_Coefs LVPSA_QPD_Float_Coefs[];
+
+#endif /* __LVPSA_TABLES_H__ */
diff --git a/media/libeffects/lvm/lib/StereoWidening/lib/LVCS.h b/media/libeffects/lvm/lib/StereoWidening/lib/LVCS.h
index e507a7c..0adfd1b 100644
--- a/media/libeffects/lvm/lib/StereoWidening/lib/LVCS.h
+++ b/media/libeffects/lvm/lib/StereoWidening/lib/LVCS.h
@@ -56,11 +56,6 @@
#ifndef LVCS_H
#define LVCS_H
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
-
/****************************************************************************************/
/* */
/* Includes */
@@ -70,7 +65,6 @@
#include "LVM_Types.h"
#include "LVM_Common.h"
-
/****************************************************************************************/
/* */
/* Definitions */
@@ -93,7 +87,6 @@
#define LVCS_EVENT_NONE 0x0000 /* Not a valid event */
#define LVCS_EVENT_ALGOFF 0x0001 /* CS has completed switch off */
-
/****************************************************************************************/
/* */
/* Types */
@@ -103,7 +96,6 @@
/* Instance handle */
typedef void *LVCS_Handle_t;
-
/* Operating modes */
typedef enum
{
@@ -112,7 +104,6 @@
LVCS_MAX = LVM_MAXENUM
} LVCS_Modes_en;
-
/* Memory Types */
typedef enum
{
@@ -123,7 +114,6 @@
LVCS_MEMORYTYPE_MAX = LVM_MAXENUM
} LVCS_MemoryTypes_en;
-
/* Function return status */
typedef enum
{
@@ -135,7 +125,6 @@
LVCS_STATUSMAX = LVM_MAXENUM
} LVCS_ReturnStatus_en;
-
/*
* Source data formats
*/
@@ -146,7 +135,6 @@
LVCS_SOURCEMAX = LVM_MAXENUM
} LVCS_SourceFormat_en;
-
/*
* Supported output devices
*/
@@ -172,7 +160,6 @@
void *pTable8;
} LVCS_CSMS_Coef_Tables_t;
-
/****************************************************************************************/
/* */
/* Structures */
@@ -187,14 +174,12 @@
void *pBaseAddress; /* Pointer to the region base address */
} LVCS_MemoryRegion_t;
-
/* Memory table containing the region definitions */
typedef struct
{
LVCS_MemoryRegion_t Region[LVCS_NR_MEMORY_REGIONS]; /* One definition for each region */
} LVCS_MemTab_t;
-
/* Concert Sound parameter structure */
typedef struct
{
@@ -210,7 +195,6 @@
#endif
} LVCS_Params_t;
-
/* Concert Sound Capability structure */
typedef struct
{
@@ -223,7 +207,6 @@
} LVCS_Capabilities_t;
-
/****************************************************************************************/
/* */
/* Function Prototypes */
@@ -270,7 +253,6 @@
LVCS_MemTab_t *pMemoryTable,
LVCS_Capabilities_t *pCapabilities);
-
/****************************************************************************************/
/* */
/* FUNCTION: LVCS_Init */
@@ -308,7 +290,6 @@
LVCS_MemTab_t *pMemoryTable,
LVCS_Capabilities_t *pCapabilities);
-
/****************************************************************************************/
/* */
/* FUNCTION: LVCS_GetParameters */
@@ -332,7 +313,6 @@
LVCS_ReturnStatus_en LVCS_GetParameters(LVCS_Handle_t hInstance,
LVCS_Params_t *pParams);
-
/****************************************************************************************/
/* */
/* FUNCTION: LVCS_Control */
@@ -355,7 +335,6 @@
LVCS_ReturnStatus_en LVCS_Control(LVCS_Handle_t hInstance,
LVCS_Params_t *pParams);
-
/****************************************************************************************/
/* */
/* FUNCTION: LVCS_Process */
@@ -377,20 +356,9 @@
/* NOTES: */
/* */
/****************************************************************************************/
-#ifdef BUILD_FLOAT
LVCS_ReturnStatus_en LVCS_Process(LVCS_Handle_t hInstance,
const LVM_FLOAT *pInData,
LVM_FLOAT *pOutData,
LVM_UINT16 NumSamples);
-#else
-LVCS_ReturnStatus_en LVCS_Process(LVCS_Handle_t hInstance,
- const LVM_INT16 *pInData,
- LVM_INT16 *pOutData,
- LVM_UINT16 NumSamples);
-#endif
-
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
#endif /* LVCS_H */
diff --git a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_BypassMix.c b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_BypassMix.cpp
similarity index 79%
rename from media/libeffects/lvm/lib/StereoWidening/src/LVCS_BypassMix.c
rename to media/libeffects/lvm/lib/StereoWidening/src/LVCS_BypassMix.cpp
index 29e3c9e..ba152c0 100644
--- a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_BypassMix.c
+++ b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_BypassMix.cpp
@@ -70,19 +70,12 @@
{
LVM_UINT16 Offset;
-#ifndef BUILD_FLOAT
- LVM_UINT32 Gain;
- LVM_INT32 Current;
-#else
LVM_FLOAT Gain;
LVM_FLOAT Current;
-#endif
LVCS_Instance_t *pInstance = (LVCS_Instance_t *)hInstance;
LVCS_BypassMix_t *pConfig = (LVCS_BypassMix_t *)&pInstance->BypassMix;
const Gain_t *pOutputGainTable;
-
-
/*
* Set the transition gain
*/
@@ -91,11 +84,7 @@
&& (pInstance->MSTarget1 != 0x7FFF) /* this indicates an off->on transtion */
)
{
-#ifndef BUILD_FLOAT
- pInstance->TransitionGain = pParams->EffectLevel;
-#else
pInstance->TransitionGain = ((LVM_FLOAT)pParams->EffectLevel / 32767);
-#endif
}
else
{
@@ -112,38 +101,21 @@
/*
* Setup the mixer gain for the processed path
*/
-#ifndef BUILD_FLOAT
- Gain = (LVM_UINT32)(pOutputGainTable[Offset].Loss * pInstance->TransitionGain);
-#else
Gain = (LVM_FLOAT)(pOutputGainTable[Offset].Loss * pInstance->TransitionGain);
-#endif
pConfig->Mixer_Instance.MixerStream[0].CallbackParam = 0;
pConfig->Mixer_Instance.MixerStream[0].pCallbackHandle = LVM_NULL;
pConfig->Mixer_Instance.MixerStream[0].pCallBack = LVM_NULL;
pConfig->Mixer_Instance.MixerStream[0].CallbackSet=1;
-#ifndef BUILD_FLOAT
- Current = LVC_Mixer_GetCurrent(&pConfig->Mixer_Instance.MixerStream[0]);
- LVC_Mixer_Init(&pConfig->Mixer_Instance.MixerStream[0],(LVM_INT32)(Gain >> 15),Current);
- LVC_Mixer_VarSlope_SetTimeConstant(&pConfig->Mixer_Instance.MixerStream[0],LVCS_BYPASS_MIXER_TC,pParams->SampleRate,2);
-#else
Current = LVC_Mixer_GetCurrent(&pConfig->Mixer_Instance.MixerStream[0]);
LVC_Mixer_Init(&pConfig->Mixer_Instance.MixerStream[0], (LVM_FLOAT)(Gain), Current);
LVC_Mixer_VarSlope_SetTimeConstant(&pConfig->Mixer_Instance.MixerStream[0],
LVCS_BYPASS_MIXER_TC, pParams->SampleRate, 2);
-#endif
/*
* Setup the mixer gain for the unprocessed path
*/
-#ifndef BUILD_FLOAT
- Gain = (LVM_UINT32)(pOutputGainTable[Offset].Loss * (0x7FFF - pInstance->TransitionGain));
- Gain = (LVM_UINT32)pOutputGainTable[Offset].UnprocLoss * (Gain >> 15);
- Current = LVC_Mixer_GetCurrent(&pConfig->Mixer_Instance.MixerStream[1]);
- LVC_Mixer_Init(&pConfig->Mixer_Instance.MixerStream[1],(LVM_INT32)(Gain >> 15),Current);
- LVC_Mixer_VarSlope_SetTimeConstant(&pConfig->Mixer_Instance.MixerStream[1],LVCS_BYPASS_MIXER_TC,pParams->SampleRate,2);
-#else
Gain = (LVM_FLOAT)(pOutputGainTable[Offset].Loss * (1.0 - \
(LVM_FLOAT)pInstance->TransitionGain));
Gain = (LVM_FLOAT)pOutputGainTable[Offset].UnprocLoss * Gain;
@@ -151,7 +123,6 @@
LVC_Mixer_Init(&pConfig->Mixer_Instance.MixerStream[1], (LVM_FLOAT)(Gain), Current);
LVC_Mixer_VarSlope_SetTimeConstant(&pConfig->Mixer_Instance.MixerStream[1],
LVCS_BYPASS_MIXER_TC, pParams->SampleRate, 2);
-#endif
pConfig->Mixer_Instance.MixerStream[1].CallbackParam = 0;
pConfig->Mixer_Instance.MixerStream[1].pCallbackHandle = hInstance;
pConfig->Mixer_Instance.MixerStream[1].CallbackSet=1;
@@ -162,50 +133,10 @@
*/
pConfig->Output_Shift = pOutputGainTable[Offset].Shift;
-
/*
* Correct gain for the effect level
*/
{
-#ifndef BUILD_FLOAT
- LVM_INT16 GainCorrect;
- LVM_INT32 Gain1;
- LVM_INT32 Gain2;
-
- Gain1 = LVC_Mixer_GetTarget(&pConfig->Mixer_Instance.MixerStream[0]);
- Gain2 = LVC_Mixer_GetTarget(&pConfig->Mixer_Instance.MixerStream[1]);
- /*
- * Calculate the gain correction
- */
- if (pInstance->Params.CompressorMode == LVM_MODE_ON)
- {
- GainCorrect = (LVM_INT16)( pInstance->VolCorrect.GainMin
- - (((LVM_INT32)pInstance->VolCorrect.GainMin * (LVM_INT32)pInstance->TransitionGain) >> 15)
- + (((LVM_INT32)pInstance->VolCorrect.GainFull * (LVM_INT32)pInstance->TransitionGain) >> 15) );
-
- /*
- * Apply the gain correction and shift, note the result is in Q3.13 format
- */
- Gain1 = (Gain1 * GainCorrect) << 4;
- Gain2 = (Gain2 * GainCorrect) << 4;
- }
- else
- {
- Gain1 = Gain1 << 16;
- Gain2 = Gain2 << 16;
- }
-
-
-
- /*
- * Set the gain values
- */
- pConfig->Output_Shift = pConfig->Output_Shift;
- LVC_Mixer_SetTarget(&pConfig->Mixer_Instance.MixerStream[0],Gain1>>16);
- LVC_Mixer_VarSlope_SetTimeConstant(&pConfig->Mixer_Instance.MixerStream[0],LVCS_BYPASS_MIXER_TC,pParams->SampleRate,2);
- LVC_Mixer_SetTarget(&pConfig->Mixer_Instance.MixerStream[1],Gain2>>16);
- LVC_Mixer_VarSlope_SetTimeConstant(&pConfig->Mixer_Instance.MixerStream[1],LVCS_BYPASS_MIXER_TC,pParams->SampleRate,2);
-#else
LVM_FLOAT GainCorrect;
LVM_FLOAT Gain1;
LVM_FLOAT Gain2;
@@ -241,7 +172,6 @@
LVC_Mixer_SetTarget(&pConfig->Mixer_Instance.MixerStream[1],Gain2);
LVC_Mixer_VarSlope_SetTimeConstant(&pConfig->Mixer_Instance.MixerStream[1],
LVCS_BYPASS_MIXER_TC, pParams->SampleRate, 2);
-#endif
}
return(LVCS_SUCCESS);
@@ -276,15 +206,9 @@
/************************************************************************************/
LVCS_ReturnStatus_en LVCS_BypassMixer(LVCS_Handle_t hInstance,
-#ifndef BUILD_FLOAT
- const LVM_INT16 *pProcessed,
- const LVM_INT16 *pUnprocessed,
- LVM_INT16 *pOutData,
-#else
const LVM_FLOAT *pProcessed,
const LVM_FLOAT *pUnprocessed,
LVM_FLOAT *pOutData,
-#endif
LVM_UINT16 NumSamples)
{
@@ -299,21 +223,6 @@
/*
* Apply the bypass mix
*/
-#ifndef BUILD_FLOAT
- LVC_MixSoft_2St_D16C31_SAT(&pConfig->Mixer_Instance,
- pProcessed,
- (LVM_INT16 *) pUnprocessed,
- pOutData,
- (LVM_INT16)(2*NumSamples));
-
- /*
- * Apply output gain correction shift
- */
- Shift_Sat_v16xv16 ((LVM_INT16)pConfig->Output_Shift,
- (LVM_INT16*)pOutData,
- (LVM_INT16*)pOutData,
- (LVM_INT16)(2*NumSamples)); /* Left and right*/
-#else
LVC_MixSoft_2St_D16C31_SAT(&pConfig->Mixer_Instance,
pProcessed,
(LVM_FLOAT *) pUnprocessed,
@@ -326,13 +235,11 @@
(LVM_FLOAT*)pOutData,
(LVM_FLOAT*)pOutData,
(LVM_INT16)(2 * NumSamples)); /* Left and right*/
-#endif
}
return(LVCS_SUCCESS);
}
-
/************************************************************************************/
/* */
/* FUNCTION: LVCS_MixerCallback */
@@ -368,7 +275,6 @@
}
}
-
if ((pInstance->OutputDevice == LVCS_HEADPHONE) &&
(pInstance->MSTarget0 == 1) &&
(pInstance->bTimerDone == LVM_TRUE)){
@@ -380,5 +286,3 @@
return 1;
}
-
-
diff --git a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_BypassMix.h b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_BypassMix.h
index f69ba38..fcd8ee3 100644
--- a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_BypassMix.h
+++ b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_BypassMix.h
@@ -18,11 +18,6 @@
#ifndef __LVCS_BYPASSMIX_H__
#define __LVCS_BYPASSMIX_H__
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
-
/************************************************************************************/
/* */
/* Includes */
@@ -31,7 +26,6 @@
#include "LVC_Mixer.h"
-
/************************************************************************************/
/* */
/* Structures */
@@ -42,25 +36,11 @@
typedef struct
{
/* Mixer settings */
-#ifdef BUILD_FLOAT
LVMixer3_2St_FLOAT_st Mixer_Instance; /* Mixer instance */
-#else
- LVMixer3_2St_st Mixer_Instance; /* Mixer instance */
-#endif
LVM_UINT16 Output_Shift; /* Correcting gain output shift */
} LVCS_BypassMix_t;
-#ifndef BUILD_FLOAT
-/* Output gain type */
-typedef struct
-{
- /* Output gain settings, Gain = (Loss/32768) * 2^Shift */
- LVM_UINT16 Shift; /* Left shifts required */
- LVM_UINT16 Loss; /* Loss required */
- LVM_UINT16 UnprocLoss; /* Unprocessed path loss */
-} Gain_t;
-#else
typedef struct
{
/* Output gain settings, Gain = (Loss/32768) * 2^Shift */
@@ -68,7 +48,6 @@
LVM_FLOAT Loss; /* Loss required */
LVM_FLOAT UnprocLoss; /* Unprocessed path loss */
} Gain_t;
-#endif
/************************************************************************************/
/* */
/* Function prototypes */
@@ -78,21 +57,10 @@
LVCS_ReturnStatus_en LVCS_BypassMixInit(LVCS_Handle_t hInstance,
LVCS_Params_t *pParams);
-#ifndef BUILD_FLOAT
-LVCS_ReturnStatus_en LVCS_BypassMixer(LVCS_Handle_t hInstance,
- const LVM_INT16 *pProcessed,
- const LVM_INT16 *unProcessed,
- LVM_INT16 *pOutData,
- LVM_UINT16 NumSamples);
-#else
LVCS_ReturnStatus_en LVCS_BypassMixer(LVCS_Handle_t hInstance,
const LVM_FLOAT *pProcessed,
const LVM_FLOAT *unProcessed,
LVM_FLOAT *pOutData,
LVM_UINT16 NumSamples);
-#endif
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
#endif /* BYPASSMIX_H */
diff --git a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Control.c b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Control.cpp
similarity index 90%
rename from media/libeffects/lvm/lib/StereoWidening/src/LVCS_Control.c
rename to media/libeffects/lvm/lib/StereoWidening/src/LVCS_Control.cpp
index 3bf6ec6..50db03d 100644
--- a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Control.c
+++ b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Control.cpp
@@ -56,7 +56,6 @@
return(LVCS_SUCCESS);
}
-
/************************************************************************************/
/* */
/* FUNCTION: LVCS_Control */
@@ -120,29 +119,8 @@
pInstance->VolCorrect = pLVCS_VolCorrectTable[Offset];
pInstance->CompressGain = pInstance->VolCorrect.CompMin;
-#ifdef BUILD_FLOAT
LVC_Mixer_Init(&pInstance->BypassMix.Mixer_Instance.MixerStream[0], 0, 0);
-#else
- LVC_Mixer_Init(&pInstance->BypassMix.Mixer_Instance.MixerStream[0],0,0);
-#endif
{
-#ifndef BUILD_FLOAT
- LVM_UINT32 Gain;
- const Gain_t *pOutputGainTable = (Gain_t*)&LVCS_OutputGainTable[0];
- Gain = (LVM_UINT32)(pOutputGainTable[Offset].Loss * LVM_MAXINT_16);
- Gain = (LVM_UINT32)pOutputGainTable[Offset].UnprocLoss * (Gain >> 15);
- Gain=Gain>>15;
- /*
- * Apply the gain correction and shift, note the result is in Q3.13 format
- */
- Gain = (Gain * pInstance->VolCorrect.GainMin) >>12;
-
- LVC_Mixer_Init(&pInstance->BypassMix.Mixer_Instance.MixerStream[1],0,Gain);
- LVC_Mixer_VarSlope_SetTimeConstant(&pInstance->BypassMix.Mixer_Instance.MixerStream[0],
- LVCS_BYPASS_MIXER_TC,pParams->SampleRate,2);
- LVC_Mixer_VarSlope_SetTimeConstant(&pInstance->BypassMix.Mixer_Instance.MixerStream[1],
- LVCS_BYPASS_MIXER_TC,pParams->SampleRate,2);
-#else
LVM_FLOAT Gain;
const Gain_t *pOutputGainTable = (Gain_t*)&LVCS_OutputGainTable[0];
Gain = (LVM_FLOAT)(pOutputGainTable[Offset].Loss);
@@ -158,10 +136,8 @@
LVCS_BYPASS_MIXER_TC, pParams->SampleRate, 2);
LVC_Mixer_VarSlope_SetTimeConstant(&pInstance->BypassMix.Mixer_Instance.MixerStream[1],
LVCS_BYPASS_MIXER_TC, pParams->SampleRate, 2);
-#endif
}
-
err=LVCS_SEnhancerInit(hInstance,
pParams);
@@ -176,7 +152,6 @@
}
-
/*
* Check if the effect level or source format has changed
*/
@@ -243,7 +218,6 @@
pInstance->MSTarget0=0;
}
-
/* Set transition flag */
pInstance->bInOperatingModeTransition = LVM_TRUE;
}
@@ -272,7 +246,6 @@
pInstance->bTimerDone = LVM_TRUE;
-
return;
}
diff --git a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Equaliser.c b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Equaliser.cpp
similarity index 68%
rename from media/libeffects/lvm/lib/StereoWidening/src/LVCS_Equaliser.c
rename to media/libeffects/lvm/lib/StereoWidening/src/LVCS_Equaliser.cpp
index ec5312e..431b7e3 100644
--- a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Equaliser.c
+++ b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Equaliser.cpp
@@ -53,7 +53,6 @@
/* NOTES: */
/* */
/************************************************************************************/
-#ifdef BUILD_FLOAT
LVCS_ReturnStatus_en LVCS_EqualiserInit(LVCS_Handle_t hInstance,
LVCS_Params_t *pParams)
{
@@ -92,8 +91,7 @@
Coeffs.B2 = (LVM_FLOAT)-pEqualiserCoefTable[Offset].B2;
LoadConst_Float((LVM_INT16)0, /* Value */
- (void *)&pData->EqualiserBiquadTaps, /* Destination Cast to void:\
- no dereferencing in function*/
+ (LVM_FLOAT *)&pData->EqualiserBiquadTaps, /* Destination */
/* Number of words */
(LVM_UINT16)(sizeof(pData->EqualiserBiquadTaps) / sizeof(LVM_FLOAT)));
@@ -118,66 +116,6 @@
return(LVCS_SUCCESS);
}
-#else
-LVCS_ReturnStatus_en LVCS_EqualiserInit(LVCS_Handle_t hInstance,
- LVCS_Params_t *pParams)
-{
-
- LVM_UINT16 Offset;
- LVCS_Instance_t *pInstance = (LVCS_Instance_t *)hInstance;
- LVCS_Equaliser_t *pConfig = (LVCS_Equaliser_t *)&pInstance->Equaliser;
- LVCS_Data_t *pData = (LVCS_Data_t *)pInstance->MemoryTable.Region[LVCS_MEMREGION_PERSISTENT_FAST_DATA].pBaseAddress;
- LVCS_Coefficient_t *pCoefficients = (LVCS_Coefficient_t *)pInstance->MemoryTable.Region[LVCS_MEMREGION_PERSISTENT_FAST_COEF].pBaseAddress;
- BQ_C16_Coefs_t Coeffs;
- const BiquadA012B12CoefsSP_t *pEqualiserCoefTable;
-
- /*
- * If the sample rate changes re-initialise the filters
- */
- if ((pInstance->Params.SampleRate != pParams->SampleRate) ||
- (pInstance->Params.SpeakerType != pParams->SpeakerType))
- {
- /*
- * Setup the filter coefficients and clear the history
- */
- Offset = (LVM_UINT16)(pParams->SampleRate + (pParams->SpeakerType * (1+LVM_FS_48000)));
- pEqualiserCoefTable = (BiquadA012B12CoefsSP_t*)&LVCS_EqualiserCoefTable[0];
-
- /* Left and right filters */
- /* Convert incoming coefficients to the required format/ordering */
- Coeffs.A0 = (LVM_INT16) pEqualiserCoefTable[Offset].A0;
- Coeffs.A1 = (LVM_INT16) pEqualiserCoefTable[Offset].A1;
- Coeffs.A2 = (LVM_INT16) pEqualiserCoefTable[Offset].A2;
- Coeffs.B1 = (LVM_INT16)-pEqualiserCoefTable[Offset].B1;
- Coeffs.B2 = (LVM_INT16)-pEqualiserCoefTable[Offset].B2;
-
- LoadConst_16((LVM_INT16)0, /* Value */
- (void *)&pData->EqualiserBiquadTaps, /* Destination Cast to void:\
- no dereferencing in function*/
- (LVM_UINT16)(sizeof(pData->EqualiserBiquadTaps)/sizeof(LVM_INT16))); /* Number of words */
-
- BQ_2I_D16F32Css_TRC_WRA_01_Init(&pCoefficients->EqualiserBiquadInstance,
- &pData->EqualiserBiquadTaps,
- &Coeffs);
-
- /* Callbacks */
- switch(pEqualiserCoefTable[Offset].Scale)
- {
- case 13:
- pConfig->pBiquadCallBack = BQ_2I_D16F32C13_TRC_WRA_01;
- break;
- case 14:
- pConfig->pBiquadCallBack = BQ_2I_D16F32C14_TRC_WRA_01;
- break;
- case 15:
- pConfig->pBiquadCallBack = BQ_2I_D16F32C15_TRC_WRA_01;
- break;
- }
- }
-
- return(LVCS_SUCCESS);
-}
-#endif
/************************************************************************************/
/* */
/* FUNCTION: LVCS_Equaliser */
@@ -197,7 +135,6 @@
/* 1. Always processes in place. */
/* */
/************************************************************************************/
-#ifdef BUILD_FLOAT
LVCS_ReturnStatus_en LVCS_Equaliser(LVCS_Handle_t hInstance,
LVM_FLOAT *pInputOutput,
LVM_UINT16 NumSamples)
@@ -207,11 +144,9 @@
LVCS_Equaliser_t *pConfig = (LVCS_Equaliser_t *)&pInstance->Equaliser;
LVCS_Coefficient_t *pCoefficients;
-
pCoefficients = (LVCS_Coefficient_t *) \
pInstance->MemoryTable.Region[LVCS_MEMREGION_PERSISTENT_FAST_COEF].pBaseAddress;
-
/*
* Check if the equaliser is required
*/
@@ -227,29 +162,3 @@
return(LVCS_SUCCESS);
}
-#else
-LVCS_ReturnStatus_en LVCS_Equaliser(LVCS_Handle_t hInstance,
- LVM_INT16 *pInputOutput,
- LVM_UINT16 NumSamples)
-{
-
- LVCS_Instance_t *pInstance = (LVCS_Instance_t *)hInstance;
- LVCS_Equaliser_t *pConfig = (LVCS_Equaliser_t *)&pInstance->Equaliser;
- LVCS_Coefficient_t *pCoefficients = (LVCS_Coefficient_t *)pInstance->MemoryTable.Region[LVCS_MEMREGION_PERSISTENT_FAST_COEF].pBaseAddress;
-
-
- /*
- * Check if the equaliser is required
- */
- if ((pInstance->Params.OperatingMode & LVCS_EQUALISERSWITCH) != 0)
- {
- /* Apply filter to the left and right channels */
- (pConfig->pBiquadCallBack)((Biquad_Instance_t*)&pCoefficients->EqualiserBiquadInstance,
- (LVM_INT16 *)pInputOutput,
- (LVM_INT16 *)pInputOutput,
- (LVM_INT16)NumSamples);
- }
-
- return(LVCS_SUCCESS);
-}
-#endif
diff --git a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Equaliser.h b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Equaliser.h
index 0e756e7..918d931 100644
--- a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Equaliser.h
+++ b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Equaliser.h
@@ -18,11 +18,6 @@
#ifndef __LVCS_EQUALISER_H__
#define __LVCS_EQUALISER_H__
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
-
/************************************************************************************/
/* */
/* Structures */
@@ -32,14 +27,9 @@
/* Equaliser structure */
typedef struct
{
-#ifndef BUILD_FLOAT
- void (*pBiquadCallBack) (Biquad_Instance_t*, LVM_INT16*, LVM_INT16*, LVM_INT16);
-#else
void (*pBiquadCallBack) (Biquad_FLOAT_Instance_t*, LVM_FLOAT*, LVM_FLOAT*, LVM_INT16);
-#endif
} LVCS_Equaliser_t;
-
/************************************************************************************/
/* */
/* Function prototypes */
@@ -48,17 +38,8 @@
LVCS_ReturnStatus_en LVCS_EqualiserInit(LVCS_Handle_t hInstance,
LVCS_Params_t *pParams);
-#ifndef BUILD_FLOAT
-LVCS_ReturnStatus_en LVCS_Equaliser(LVCS_Handle_t hInstance,
- LVM_INT16 *pInputOutput,
- LVM_UINT16 NumSamples);
-#else
LVCS_ReturnStatus_en LVCS_Equaliser(LVCS_Handle_t hInstance,
LVM_FLOAT *pInputOutput,
LVM_UINT16 NumSamples);
-#endif
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
#endif /* EQUALISER_H */
diff --git a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Headphone_Coeffs.h b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Headphone_Coeffs.h
index ba05577..c7ee232 100644
--- a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Headphone_Coeffs.h
+++ b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Headphone_Coeffs.h
@@ -18,13 +18,11 @@
#ifndef __LVCS_HEADPHONE_COEFFS_H__
#define __LVCS_HEADPHONE_COEFFS_H__
-
/************************************************************************************/
/* */
/* The Stereo Enhancer */
/* */
/************************************************************************************/
-#ifdef BUILD_FLOAT
/* Stereo Enhancer coefficients for 8000 Hz sample rate, scaled with 0.161258 */
#define CS_MIDDLE_8000_A0 0.227720
#define CS_MIDDLE_8000_A1 (-0.215125)
@@ -151,7 +149,6 @@
#define CS_SIDE_48000_B2 0.630405
#define CS_SIDE_48000_SCALE 14
-#ifdef HIGHER_FS
/* Coefficients for 88200Hz sample rate.
* The filter coefficients are obtained by carrying out
* state-space analysis using the coefficients available
@@ -222,7 +219,6 @@
#define CS_SIDE_192000_B1 (-1.891380f)
#define CS_SIDE_192000_B2 0.8923460f
#define CS_SIDE_192000_SCALE 14
-#endif
/************************************************************************************/
/* */
@@ -286,7 +282,6 @@
#define CS_REVERB_22050_B2 (-0.290990)
#define CS_REVERB_22050_SCALE 15
-
/* Reverb coefficients for 24000Hz sample rate, scaled with 1.038030 */
#define CS_REVERB_24000_A0 0.479565
#define CS_REVERB_24000_A1 0.000000
@@ -319,7 +314,6 @@
#define CS_REVERB_48000_B2 0.303347
#define CS_REVERB_48000_SCALE 14
-#ifdef HIGHER_FS
/* Reverb coefficients for 88200Hz sample rate, scaled with 0.8 */
/* Band pass filter with fc1=500 and fc2=8000 */
#define CS_REVERB_88200_A0 0.171901f
@@ -354,9 +348,6 @@
#define CS_REVERB_192000_B2 0.7804076
#define CS_REVERB_192000_SCALE 14
-#endif
-
-
/* Reverb Gain Settings */
#define LVCS_HEADPHONE_DELAYGAIN 0.800000 /* Algorithm delay path gain */
#define LVCS_HEADPHONE_OUTPUTGAIN 1.000000 /* Algorithm output gain */
@@ -505,8 +496,6 @@
#define CSEX_EQUALISER_48000_B2 (-0.347332)
#define CSEX_EQUALISER_48000_SCALE 13
-
-#ifdef HIGHER_FS
/* Equaliser coefficients for 88200Hz sample rate.
* The filter coefficients are obtained by carrying out
* state-space analysis using the coefficients available
@@ -567,8 +556,6 @@
#define CSEX_EQUALISER_192000_B1 (-1.31074)
#define CSEX_EQUALISER_192000_B2 0.31312
#define CSEX_EQUALISER_192000_SCALE 13
-#endif
-
#define LVCS_HEADPHONE_SHIFT 2 /* Output Shift */
#define LVCS_HEADPHONE_SHIFTLOSS 0.8477735 /* Output Shift loss */
@@ -576,376 +563,5 @@
#define LVCS_EX_HEADPHONE_SHIFT 3 /* EX Output Shift */
#define LVCS_EX_HEADPHONE_SHIFTLOSS 0.569225 /* EX Output Shift loss */
#define LVCS_EX_HEADPHONE_GAIN 0.07794425 /* EX Unprocessed path gain */
-#else
-/* Stereo Enhancer coefficients for 8000 Hz sample rate, scaled with 0.161258 */
-#define CS_MIDDLE_8000_A0 7462 /* Floating point value 0.227720 */
-#define CS_MIDDLE_8000_A1 (-7049) /* Floating point value -0.215125 */
-#define CS_MIDDLE_8000_A2 0 /* Floating point value 0.000000 */
-#define CS_MIDDLE_8000_B1 (-30209) /* Floating point value -0.921899 */
-#define CS_MIDDLE_8000_B2 0 /* Floating point value 0.000000 */
-#define CS_MIDDLE_8000_SCALE 15
-#define CS_SIDE_8000_A0 20036 /* Floating point value 0.611441 */
-#define CS_SIDE_8000_A1 (-12463) /* Floating point value -0.380344 */
-#define CS_SIDE_8000_A2 (-7573) /* Floating point value -0.231097 */
-#define CS_SIDE_8000_B1 (-20397) /* Floating point value -0.622470 */
-#define CS_SIDE_8000_B2 (-4285) /* Floating point value -0.130759 */
-#define CS_SIDE_8000_SCALE 15
-
-/* Stereo Enhancer coefficients for 11025Hz sample rate, scaled with 0.162943 */
-#define CS_MIDDLE_11025_A0 7564 /* Floating point value 0.230838 */
-#define CS_MIDDLE_11025_A1 (-7260) /* Floating point value -0.221559 */
-#define CS_MIDDLE_11025_A2 0 /* Floating point value 0.000000 */
-#define CS_MIDDLE_11025_B1 (-30902) /* Floating point value -0.943056 */
-#define CS_MIDDLE_11025_B2 0 /* Floating point value 0.000000 */
-#define CS_MIDDLE_11025_SCALE 15
-#define CS_SIDE_11025_A0 18264 /* Floating point value 0.557372 */
-#define CS_SIDE_11025_A1 (-12828) /* Floating point value -0.391490 */
-#define CS_SIDE_11025_A2 (-5436) /* Floating point value -0.165881 */
-#define CS_SIDE_11025_B1 (-28856) /* Floating point value -0.880608 */
-#define CS_SIDE_11025_B2 1062 /* Floating point value 0.032397 */
-#define CS_SIDE_11025_SCALE 15
-
-/* Stereo Enhancer coefficients for 12000Hz sample rate, scaled with 0.162191 */
-#define CS_MIDDLE_12000_A0 7534 /* Floating point value 0.229932 */
-#define CS_MIDDLE_12000_A1 (-7256) /* Floating point value -0.221436 */
-#define CS_MIDDLE_12000_A2 0 /* Floating point value 0.000000 */
-#define CS_MIDDLE_12000_B1 (-31051) /* Floating point value -0.947616 */
-#define CS_MIDDLE_12000_B2 0 /* Floating point value 0.000000 */
-#define CS_MIDDLE_12000_SCALE 15
-#define CS_SIDE_12000_A0 18298 /* Floating point value 0.558398 */
-#define CS_SIDE_12000_A1 (-12852) /* Floating point value -0.392211 */
-#define CS_SIDE_12000_A2 (-5446) /* Floating point value -0.166187 */
-#define CS_SIDE_12000_B1 (-29247) /* Floating point value -0.892550 */
-#define CS_SIDE_12000_B2 1077 /* Floating point value 0.032856 */
-#define CS_SIDE_12000_SCALE 15
-
-/* Stereo Enhancer coefficients for 16000Hz sample rate, scaled with 0.162371 */
-#define CS_MIDDLE_16000_A0 7558 /* Floating point value 0.230638 */
-#define CS_MIDDLE_16000_A1 (-7348) /* Floating point value -0.224232 */
-#define CS_MIDDLE_16000_A2 0 /* Floating point value 0.000000 */
-#define CS_MIDDLE_16000_B1 (-31475) /* Floating point value -0.960550 */
-#define CS_MIDDLE_16000_B2 0 /* Floating point value 0.000000 */
-#define CS_MIDDLE_16000_SCALE 15
-#define CS_SIDE_16000_A0 8187 /* Floating point value 0.499695 */
-#define CS_SIDE_16000_A1 (-5825) /* Floating point value -0.355543 */
-#define CS_SIDE_16000_A2 (-2362) /* Floating point value -0.144152 */
-#define CS_SIDE_16000_B1 (-17216) /* Floating point value -1.050788 */
-#define CS_SIDE_16000_B2 2361 /* Floating point value 0.144104 */
-#define CS_SIDE_16000_SCALE 14
-
-/* Stereo Enhancer coefficients for 22050Hz sample rate, scaled with 0.160781 */
-#define CS_MIDDLE_22050_A0 7496 /* Floating point value 0.228749 */
-#define CS_MIDDLE_22050_A1 (-7344) /* Floating point value -0.224128 */
-#define CS_MIDDLE_22050_A2 0 /* Floating point value 0.000000 */
-#define CS_MIDDLE_22050_B1 (-31826) /* Floating point value -0.971262 */
-#define CS_MIDDLE_22050_B2 0 /* Floating point value 0.000000 */
-#define CS_MIDDLE_22050_SCALE 15
-#define CS_SIDE_22050_A0 7211 /* Floating point value 0.440112 */
-#define CS_SIDE_22050_A1 (-4278) /* Floating point value -0.261096 */
-#define CS_SIDE_22050_A2 (-2933) /* Floating point value -0.179016 */
-#define CS_SIDE_22050_B1 (-18297) /* Floating point value -1.116786 */
-#define CS_SIDE_22050_B2 2990 /* Floating point value 0.182507 */
-#define CS_SIDE_22050_SCALE 14
-
-/* Stereo Enhancer coefficients for 24000Hz sample rate, scaled with 0.161882 */
-#define CS_MIDDLE_24000_A0 7550 /* Floating point value 0.230395 */
-#define CS_MIDDLE_24000_A1 (-7409) /* Floating point value -0.226117 */
-#define CS_MIDDLE_24000_A2 0 /* Floating point value 0.000000 */
-#define CS_MIDDLE_24000_B1 (-31902) /* Floating point value -0.973573 */
-#define CS_MIDDLE_24000_B2 0 /* Floating point value 0.000000 */
-#define CS_MIDDLE_24000_SCALE 15
-#define CS_SIDE_24000_A0 6796 /* Floating point value 0.414770 */
-#define CS_SIDE_24000_A1 (-4705) /* Floating point value -0.287182 */
-#define CS_SIDE_24000_A2 (-2090) /* Floating point value -0.127588 */
-#define CS_SIDE_24000_B1 (-20147) /* Floating point value -1.229648 */
-#define CS_SIDE_24000_B2 4623 /* Floating point value 0.282177 */
-#define CS_SIDE_24000_SCALE 14
-
-/* Stereo Enhancer coefficients for 32000Hz sample rate, scaled with 0.160322 */
-#define CS_MIDDLE_32000_A0 7484 /* Floating point value 0.228400 */
-#define CS_MIDDLE_32000_A1 (-7380) /* Floating point value -0.225214 */
-#define CS_MIDDLE_32000_A2 0 /* Floating point value 0.000000 */
-#define CS_MIDDLE_32000_B1 (-32117) /* Floating point value -0.980126 */
-#define CS_MIDDLE_32000_B2 0 /* Floating point value 0.000000 */
-#define CS_MIDDLE_32000_SCALE 15
-#define CS_SIDE_32000_A0 5973 /* Floating point value 0.364579 */
-#define CS_SIDE_32000_A1 (-3397) /* Floating point value -0.207355 */
-#define CS_SIDE_32000_A2 (-2576) /* Floating point value -0.157224 */
-#define CS_SIDE_32000_B1 (-20877) /* Floating point value -1.274231 */
-#define CS_SIDE_32000_B2 5120 /* Floating point value 0.312495 */
-#define CS_SIDE_32000_SCALE 14
-
-/* Stereo Enhancer coefficients for 44100Hz sample rate, scaled with 0.163834 */
-#define CS_MIDDLE_44100_A0 7654 /* Floating point value 0.233593 */
-#define CS_MIDDLE_44100_A1 (-7577) /* Floating point value -0.231225 */
-#define CS_MIDDLE_44100_A2 0 /* Floating point value 0.000000 */
-#define CS_MIDDLE_44100_B1 (-32294) /* Floating point value -0.985545 */
-#define CS_MIDDLE_44100_B2 0 /* Floating point value 0.000000 */
-#define CS_MIDDLE_44100_SCALE 15
-#define CS_SIDE_44100_A0 4662 /* Floating point value 0.284573 */
-#define CS_SIDE_44100_A1 (-4242) /* Floating point value -0.258910 */
-#define CS_SIDE_44100_A2 (-420) /* Floating point value -0.025662 */
-#define CS_SIDE_44100_B1 (-25760) /* Floating point value -1.572248 */
-#define CS_SIDE_44100_B2 9640 /* Floating point value 0.588399 */
-#define CS_SIDE_44100_SCALE 14
-
-/* Stereo Enhancer coefficients for 48000Hz sample rate, scaled with 0.164402 */
-#define CS_MIDDLE_48000_A0 7682 /* Floating point value 0.234445 */
-#define CS_MIDDLE_48000_A1 (-7611) /* Floating point value -0.232261 */
-#define CS_MIDDLE_48000_A2 0 /* Floating point value 0.000000 */
-#define CS_MIDDLE_48000_B1 (-32333) /* Floating point value -0.986713 */
-#define CS_MIDDLE_48000_B2 0 /* Floating point value 0.000000 */
-#define CS_MIDDLE_48000_SCALE 15
-#define CS_SIDE_48000_A0 4466 /* Floating point value 0.272606 */
-#define CS_SIDE_48000_A1 (-4374) /* Floating point value -0.266952 */
-#define CS_SIDE_48000_A2 (-93) /* Floating point value -0.005654 */
-#define CS_SIDE_48000_B1 (-26495) /* Floating point value -1.617141 */
-#define CS_SIDE_48000_B2 10329 /* Floating point value 0.630405 */
-#define CS_SIDE_48000_SCALE 14
-
-
-/************************************************************************************/
-/* */
-/* The Reverb Unit */
-/* */
-/************************************************************************************/
-
-/* Reverb delay settings in samples */
-#define LVCS_STEREODELAY_CS_8KHZ 93 /* Sample rate 8kS/s */
-#define LVCS_STEREODELAY_CS_11KHZ 128 /* Sample rate 11kS/s */
-#define LVCS_STEREODELAY_CS_12KHZ 139 /* Sample rate 12kS/s */
-#define LVCS_STEREODELAY_CS_16KHZ 186 /* Sample rate 16kS/s */
-#define LVCS_STEREODELAY_CS_22KHZ 256 /* Sample rate 22kS/s */
-#define LVCS_STEREODELAY_CS_24KHZ 279 /* Sample rate 24kS/s */
-#define LVCS_STEREODELAY_CS_32KHZ 372 /* Sample rate 32kS/s */
-#define LVCS_STEREODELAY_CS_44KHZ 512 /* Sample rate 44kS/s */
-#define LVCS_STEREODELAY_CS_48KHZ 512 /* Sample rate 48kS/s */
-
-/* Reverb coefficients for 8000 Hz sample rate, scaled with 1.038030 */
-#define CS_REVERB_8000_A0 21865 /* Floating point value 0.667271 */
-#define CS_REVERB_8000_A1 (-21865) /* Floating point value -0.667271 */
-#define CS_REVERB_8000_A2 0 /* Floating point value 0.000000 */
-#define CS_REVERB_8000_B1 (-21895) /* Floating point value -0.668179 */
-#define CS_REVERB_8000_B2 0 /* Floating point value 0.000000 */
-#define CS_REVERB_8000_SCALE 15
-
-/* Reverb coefficients for 11025Hz sample rate, scaled with 1.038030 */
-#define CS_REVERB_11025_A0 22926 /* Floating point value 0.699638 */
-#define CS_REVERB_11025_A1 (-22926) /* Floating point value -0.699638 */
-#define CS_REVERB_11025_A2 0 /* Floating point value 0.000000 */
-#define CS_REVERB_11025_B1 (-24546) /* Floating point value -0.749096 */
-#define CS_REVERB_11025_B2 0 /* Floating point value 0.000000 */
-#define CS_REVERB_11025_SCALE 15
-
-/* Reverb coefficients for 12000Hz sample rate, scaled with 1.038030 */
-#define CS_REVERB_12000_A0 23165 /* Floating point value 0.706931 */
-#define CS_REVERB_12000_A1 (-23165) /* Floating point value -0.706931 */
-#define CS_REVERB_12000_A2 0 /* Floating point value 0.000000 */
-#define CS_REVERB_12000_B1 (-25144) /* Floating point value -0.767327 */
-#define CS_REVERB_12000_B2 0 /* Floating point value 0.000000 */
-#define CS_REVERB_12000_SCALE 15
-
-/* Reverb coefficients for 16000Hz sample rate, scaled with 1.038030 */
-#define CS_REVERB_16000_A0 23864 /* Floating point value 0.728272 */
-#define CS_REVERB_16000_A1 (-23864) /* Floating point value -0.728272 */
-#define CS_REVERB_16000_A2 0 /* Floating point value 0.000000 */
-#define CS_REVERB_16000_B1 (-26892) /* Floating point value -0.820679 */
-#define CS_REVERB_16000_B2 0 /* Floating point value 0.000000 */
-#define CS_REVERB_16000_SCALE 15
-
-/* Reverb coefficients for 22050Hz sample rate, scaled with 1.038030 */
-#define CS_REVERB_22050_A0 16921 /* Floating point value 0.516396 */
-#define CS_REVERB_22050_A1 0 /* Floating point value 0.000000 */
-#define CS_REVERB_22050_A2 (-16921) /* Floating point value -0.516396 */
-#define CS_REVERB_22050_B1 (-16991) /* Floating point value -0.518512 */
-#define CS_REVERB_22050_B2 (-9535) /* Floating point value -0.290990 */
-#define CS_REVERB_22050_SCALE 15
-
-/* Reverb coefficients for 24000Hz sample rate, scaled with 1.038030 */
-#define CS_REVERB_24000_A0 15714 /* Floating point value 0.479565 */
-#define CS_REVERB_24000_A1 0 /* Floating point value 0.000000 */
-#define CS_REVERB_24000_A2 (-15714) /* Floating point value -0.479565 */
-#define CS_REVERB_24000_B1 (-20898) /* Floating point value -0.637745 */
-#define CS_REVERB_24000_B2 (-6518) /* Floating point value -0.198912 */
-#define CS_REVERB_24000_SCALE 15
-
-/* Reverb coefficients for 32000Hz sample rate, scaled with 1.038030 */
-#define CS_REVERB_32000_A0 12463 /* Floating point value 0.380349 */
-#define CS_REVERB_32000_A1 0 /* Floating point value 0.000000 */
-#define CS_REVERB_32000_A2 (-12463) /* Floating point value -0.380349 */
-#define CS_REVERB_32000_B1 (-31158) /* Floating point value -0.950873 */
-#define CS_REVERB_32000_B2 1610 /* Floating point value 0.049127 */
-#define CS_REVERB_32000_SCALE 15
-
-/* Reverb coefficients for 44100Hz sample rate, scaled with 1.038030 */
-#define CS_REVERB_44100_A0 4872 /* Floating point value 0.297389 */
-#define CS_REVERB_44100_A1 0 /* Floating point value 0.000000 */
-#define CS_REVERB_44100_A2 (-4872) /* Floating point value -0.297389 */
-#define CS_REVERB_44100_B1 (-19668) /* Floating point value -1.200423 */
-#define CS_REVERB_44100_B2 4203 /* Floating point value 0.256529 */
-#define CS_REVERB_44100_SCALE 14
-
-/* Reverb coefficients for 48000Hz sample rate, scaled with 1.038030 */
-#define CS_REVERB_48000_A0 4566 /* Floating point value 0.278661 */
-#define CS_REVERB_48000_A1 0 /* Floating point value 0.000000 */
-#define CS_REVERB_48000_A2 (-4566) /* Floating point value -0.278661 */
-#define CS_REVERB_48000_B1 (-20562) /* Floating point value -1.254993 */
-#define CS_REVERB_48000_B2 4970 /* Floating point value 0.303347 */
-#define CS_REVERB_48000_SCALE 14
-
-/* Reverb Gain Settings */
-#define LVCS_HEADPHONE_DELAYGAIN 0.800000 /* Algorithm delay path gain */
-#define LVCS_HEADPHONE_OUTPUTGAIN 1.000000 /* Algorithm output gain */
-#define LVCS_HEADPHONE_PROCGAIN 18403 /* Processed path gain */
-#define LVCS_HEADPHONE_UNPROCGAIN 18403 /* Unprocessed path gain */
-#define LVCS_HEADPHONE_GAINCORRECT 1.009343 /* Delay mixer gain correction */
-
-
-/************************************************************************************/
-/* */
-/* The Equaliser */
-/* */
-/************************************************************************************/
-
-/* Equaliser coefficients for 8000 Hz sample rate, CS scaled with 1.038497 and CSEX scaled with 0.775480 */
-#define CS_EQUALISER_8000_A0 20698 /* Floating point value 1.263312 */
-#define CS_EQUALISER_8000_A1 (-9859) /* Floating point value -0.601748 */
-#define CS_EQUALISER_8000_A2 (-4599) /* Floating point value -0.280681 */
-#define CS_EQUALISER_8000_B1 (-7797) /* Floating point value -0.475865 */
-#define CS_EQUALISER_8000_B2 (-6687) /* Floating point value -0.408154 */
-#define CS_EQUALISER_8000_SCALE 14
-#define CSEX_EQUALISER_8000_A0 30912 /* Floating point value 0.943357 */
-#define CSEX_EQUALISER_8000_A1 (-14724) /* Floating point value -0.449345 */
-#define CSEX_EQUALISER_8000_A2 (-6868) /* Floating point value -0.209594 */
-#define CSEX_EQUALISER_8000_B1 (-15593) /* Floating point value -0.475865 */
-#define CSEX_EQUALISER_8000_B2 (-13374) /* Floating point value -0.408154 */
-#define CSEX_EQUALISER_8000_SCALE 15
-
-/* Equaliser coefficients for 11025Hz sample rate, CS scaled with 1.027761 and CSEX scaled with 0.767463 */
-#define CS_EQUALISER_11025_A0 18041 /* Floating point value 1.101145 */
-#define CS_EQUALISER_11025_A1 2278 /* Floating point value 0.139020 */
-#define CS_EQUALISER_11025_A2 (-14163) /* Floating point value -0.864423 */
-#define CS_EQUALISER_11025_B1 402 /* Floating point value 0.024541 */
-#define CS_EQUALISER_11025_B2 (-14892) /* Floating point value -0.908930 */
-#define CS_EQUALISER_11025_SCALE 14
-#define CSEX_EQUALISER_11025_A0 31983 /* Floating point value 0.976058 */
-#define CSEX_EQUALISER_11025_A1 (-22784) /* Floating point value -0.695326 */
-#define CSEX_EQUALISER_11025_A2 (-2976) /* Floating point value -0.090809 */
-#define CSEX_EQUALISER_11025_B1 (-20008) /* Floating point value -0.610594 */
-#define CSEX_EQUALISER_11025_B2 (-10196) /* Floating point value -0.311149 */
-#define CSEX_EQUALISER_11025_SCALE 15
-
-/* Equaliser coefficients for 12000Hz sample rate, CS scaled with 1.032521 and CSEX scaled with 0.771017 */
-#define CS_EQUALISER_12000_A0 20917 /* Floating point value 1.276661 */
-#define CS_EQUALISER_12000_A1 (-16671) /* Floating point value -1.017519 */
-#define CS_EQUALISER_12000_A2 (-723) /* Floating point value -0.044128 */
-#define CS_EQUALISER_12000_B1 (-11954) /* Floating point value -0.729616 */
-#define CS_EQUALISER_12000_B2 (-3351) /* Floating point value -0.204532 */
-#define CS_EQUALISER_12000_SCALE 14
-#define CSEX_EQUALISER_12000_A0 16500 /* Floating point value 1.007095 */
-#define CSEX_EQUALISER_12000_A1 (-14285) /* Floating point value -0.871912 */
-#define CSEX_EQUALISER_12000_A2 381 /* Floating point value 0.023232 */
-#define CSEX_EQUALISER_12000_B1 (-12220) /* Floating point value -0.745857 */
-#define CSEX_EQUALISER_12000_B2 (-3099) /* Floating point value -0.189171 */
-#define CSEX_EQUALISER_12000_SCALE 14
-
-/* Equaliser coefficients for 16000Hz sample rate, CS scaled with 1.031378 and CSEX scaled with 0.770164 */
-#define CS_EQUALISER_16000_A0 20998 /* Floating point value 1.281629 */
-#define CS_EQUALISER_16000_A1 (-17627) /* Floating point value -1.075872 */
-#define CS_EQUALISER_16000_A2 (-678) /* Floating point value -0.041365 */
-#define CS_EQUALISER_16000_B1 (-11882) /* Floating point value -0.725239 */
-#define CS_EQUALISER_16000_B2 (-3676) /* Floating point value -0.224358 */
-#define CS_EQUALISER_16000_SCALE 14
-#define CSEX_EQUALISER_16000_A0 17713 /* Floating point value 1.081091 */
-#define CSEX_EQUALISER_16000_A1 (-14208) /* Floating point value -0.867183 */
-#define CSEX_EQUALISER_16000_A2 (-1151) /* Floating point value -0.070247 */
-#define CSEX_EQUALISER_16000_B1 (-8440) /* Floating point value -0.515121 */
-#define CSEX_EQUALISER_16000_B2 (-6978) /* Floating point value -0.425893 */
-#define CSEX_EQUALISER_16000_SCALE 14
-
-/* Equaliser coefficients for 22050Hz sample rate, CS scaled with 1.041576 and CSEX scaled with 0.777779 */
-#define CS_EQUALISER_22050_A0 22751 /* Floating point value 1.388605 */
-#define CS_EQUALISER_22050_A1 (-21394) /* Floating point value -1.305799 */
-#define CS_EQUALISER_22050_A2 654 /* Floating point value 0.039922 */
-#define CS_EQUALISER_22050_B1 (-11788) /* Floating point value -0.719494 */
-#define CS_EQUALISER_22050_B2 (-3985) /* Floating point value -0.243245 */
-#define CS_EQUALISER_22050_SCALE 14
-#define CSEX_EQUALISER_22050_A0 20855 /* Floating point value 1.272910 */
-#define CSEX_EQUALISER_22050_A1 (-21971) /* Floating point value -1.341014 */
-#define CSEX_EQUALISER_22050_A2 2744 /* Floating point value 0.167462 */
-#define CSEX_EQUALISER_22050_B1 (-10063) /* Floating point value -0.614219 */
-#define CSEX_EQUALISER_22050_B2 (-5659) /* Floating point value -0.345384 */
-#define CSEX_EQUALISER_22050_SCALE 14
-
-/* Equaliser coefficients for 24000Hz sample rate, CS scaled with 1.034495 and CSEX scaled with 0.772491 */
-#define CS_EQUALISER_24000_A0 23099 /* Floating point value 1.409832 */
-#define CS_EQUALISER_24000_A1 (-23863) /* Floating point value -1.456506 */
-#define CS_EQUALISER_24000_A2 2481 /* Floating point value 0.151410 */
-#define CS_EQUALISER_24000_B1 (-13176) /* Floating point value -0.804201 */
-#define CS_EQUALISER_24000_B2 (-2683) /* Floating point value -0.163783 */
-#define CS_EQUALISER_24000_SCALE 14
-#define CSEX_EQUALISER_24000_A0 21286 /* Floating point value 1.299198 */
-#define CSEX_EQUALISER_24000_A1 (-23797) /* Floating point value -1.452447 */
-#define CSEX_EQUALISER_24000_A2 3940 /* Floating point value 0.240489 */
-#define CSEX_EQUALISER_24000_B1 (-10966) /* Floating point value -0.669303 */
-#define CSEX_EQUALISER_24000_B2 (-4833) /* Floating point value -0.294984 */
-#define CSEX_EQUALISER_24000_SCALE 14
-
-/* Equaliser coefficients for 32000Hz sample rate, CS scaled with 1.044559 and CSEX scaled with 0.780006 */
-#define CS_EQUALISER_32000_A0 25575 /* Floating point value 1.560988 */
-#define CS_EQUALISER_32000_A1 (-30765) /* Floating point value -1.877724 */
-#define CS_EQUALISER_32000_A2 6386 /* Floating point value 0.389741 */
-#define CS_EQUALISER_32000_B1 (-14867) /* Floating point value -0.907410 */
-#define CS_EQUALISER_32000_B2 (-1155) /* Floating point value -0.070489 */
-#define CS_EQUALISER_32000_SCALE 14
-#define CSEX_EQUALISER_32000_A0 14623 /* Floating point value 1.785049 */
-#define CSEX_EQUALISER_32000_A1 (-18297) /* Floating point value -2.233497 */
-#define CSEX_EQUALISER_32000_A2 4313 /* Floating point value 0.526431 */
-#define CSEX_EQUALISER_32000_B1 (-3653) /* Floating point value -0.445939 */
-#define CSEX_EQUALISER_32000_B2 (-4280) /* Floating point value -0.522446 */
-#define CSEX_EQUALISER_32000_SCALE 13
-
-/* Equaliser coefficients for 44100Hz sample rate, CS scaled with 1.022170 and CSEX scaled with 0.763288 */
-#define CS_EQUALISER_44100_A0 13304 /* Floating point value 1.623993 */
-#define CS_EQUALISER_44100_A1 (-18602) /* Floating point value -2.270743 */
-#define CS_EQUALISER_44100_A2 5643 /* Floating point value 0.688829 */
-#define CS_EQUALISER_44100_B1 (-9152) /* Floating point value -1.117190 */
-#define CS_EQUALISER_44100_B2 1067 /* Floating point value 0.130208 */
-#define CS_EQUALISER_44100_SCALE 13
-#define CSEX_EQUALISER_44100_A0 16616 /* Floating point value 2.028315 */
-#define CSEX_EQUALISER_44100_A1 (-23613) /* Floating point value -2.882459 */
-#define CSEX_EQUALISER_44100_A2 7410 /* Floating point value 0.904535 */
-#define CSEX_EQUALISER_44100_B1 (-4860) /* Floating point value -0.593308 */
-#define CSEX_EQUALISER_44100_B2 (-3161) /* Floating point value -0.385816 */
-#define CSEX_EQUALISER_44100_SCALE 13
-
-/* Equaliser coefficients for 48000Hz sample rate, CS scaled with 1.018635 and CSEX scaled with 0.760648 */
-#define CS_EQUALISER_48000_A0 13445 /* Floating point value 1.641177 */
-#define CS_EQUALISER_48000_A1 (-19372) /* Floating point value -2.364687 */
-#define CS_EQUALISER_48000_A2 6225 /* Floating point value 0.759910 */
-#define CS_EQUALISER_48000_B1 (-9558) /* Floating point value -1.166774 */
-#define CS_EQUALISER_48000_B2 1459 /* Floating point value 0.178074 */
-#define CS_EQUALISER_48000_SCALE 13
-#define CSEX_EQUALISER_48000_A0 17200 /* Floating point value 2.099655 */
-#define CSEX_EQUALISER_48000_A1 (-25110) /* Floating point value -3.065220 */
-#define CSEX_EQUALISER_48000_A2 8277 /* Floating point value 1.010417 */
-#define CSEX_EQUALISER_48000_B1 (-5194) /* Floating point value -0.634021 */
-#define CSEX_EQUALISER_48000_B2 (-2845) /* Floating point value -0.347332 */
-#define CSEX_EQUALISER_48000_SCALE 13
-
-
-/************************************************************************************/
-/* */
-/* The Output Gain Correction */
-/* */
-/************************************************************************************/
-
-#define LVCS_HEADPHONE_SHIFT 2 /* Output Shift */
-#define LVCS_HEADPHONE_SHIFTLOSS 27779 /* Output Shift loss */
-#define LVCS_HEADPHONE_GAIN 6840 /* Unprocessed path gain */
-#define LVCS_EX_HEADPHONE_SHIFT 3 /* EX Output Shift */
-#define LVCS_EX_HEADPHONE_SHIFTLOSS 18600 /* EX Output Shift loss */
-#define LVCS_EX_HEADPHONE_GAIN 5108 /* EX Unprocessed path gain */
-#endif
#endif
diff --git a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Init.c b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Init.cpp
similarity index 98%
rename from media/libeffects/lvm/lib/StereoWidening/src/LVCS_Init.c
rename to media/libeffects/lvm/lib/StereoWidening/src/LVCS_Init.cpp
index d4c7627..630ecf7 100644
--- a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Init.c
+++ b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Init.cpp
@@ -68,7 +68,6 @@
LVM_UINT32 ScratchSize;
LVCS_Instance_t *pInstance = (LVCS_Instance_t *)hInstance;
-
/*
* Fill in the memory table
*/
@@ -98,13 +97,9 @@
/*
* Scratch memory
*/
-#ifdef BUILD_FLOAT
/* Inplace processing */
ScratchSize = (LVM_UINT32) \
(LVCS_SCRATCHBUFFERS * sizeof(LVM_FLOAT) * pCapabilities->MaxBlockSize);
-#else
- ScratchSize = (LVM_UINT32)(LVCS_SCRATCHBUFFERS*sizeof(LVM_INT16)*pCapabilities->MaxBlockSize); /* Inplace processing */
-#endif
pMemoryTable->Region[LVCS_MEMREGION_TEMPORARY_FAST].Size = ScratchSize;
pMemoryTable->Region[LVCS_MEMREGION_TEMPORARY_FAST].Type = LVCS_SCRATCH;
pMemoryTable->Region[LVCS_MEMREGION_TEMPORARY_FAST].pBaseAddress = LVM_NULL;
@@ -118,7 +113,6 @@
return(LVCS_SUCCESS);
}
-
/************************************************************************************/
/* */
/* FUNCTION: LVCS_Init */
@@ -160,7 +154,6 @@
LVCS_Instance_t *pInstance;
LVCS_VolCorrect_t *pLVCS_VolCorrectTable;
-
/*
* Set the instance handle if not already initialised
*/
@@ -170,7 +163,6 @@
}
pInstance =(LVCS_Instance_t *)*phInstance;
-
/*
* Save the capabilities in the instance structure
*/
@@ -181,7 +173,6 @@
*/
pInstance->MemoryTable = *pMemoryTable;
-
/*
* Set all initial parameters to invalid to force a full initialisation
*/
diff --git a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Private.h b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Private.h
index ab8ccd1..620b341 100644
--- a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Private.h
+++ b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Private.h
@@ -27,11 +27,6 @@
#ifndef __LVCS_PRIVATE_H__
#define __LVCS_PRIVATE_H__
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
-
/************************************************************************************/
/* */
/* Includes */
@@ -45,7 +40,6 @@
#include "LVCS_BypassMix.h" /* Bypass Mixer module definitions */
#include "LVM_Timer.h"
-
/************************************************************************************/
/* */
/* Defines */
@@ -79,7 +73,6 @@
#define LVCS_NR_OF_FS 9
#define LVCS_NR_OF_CHAN_CFG 2
-
/************************************************************************************/
/* */
/* Types */
@@ -94,7 +87,6 @@
LVCS_DEVICE_MAX = LVM_MAXENUM
} LVCS_OutputDevice_en;
-
/************************************************************************************/
/* */
/* Structures */
@@ -104,17 +96,10 @@
/* Volume correction structure */
typedef struct
{
-#ifdef BUILD_FLOAT
LVM_FLOAT CompFull; /* Post CS compression 100% effect */
LVM_FLOAT CompMin; /* Post CS compression 0% effect */
LVM_FLOAT GainFull; /* CS gain correct 100% effect */
LVM_FLOAT GainMin; /* CS gain correct 0% effect */
-#else
- LVM_INT16 CompFull; /* Post CS compression 100% effect */
- LVM_INT16 CompMin; /* Post CS compression 0% effect */
- LVM_INT16 GainFull; /* CS gain correct 100% effect */
- LVM_INT16 GainMin; /* CS gain correct 0% effect */
-#endif
} LVCS_VolCorrect_t;
/* Instance structure */
@@ -128,13 +113,8 @@
/* Private parameters */
LVCS_OutputDevice_en OutputDevice; /* Selected output device type */
LVCS_VolCorrect_t VolCorrect; /* Volume correction settings */
-#ifndef BUILD_FLOAT
- LVM_INT16 TransitionGain; /* Transition gain */
- LVM_INT16 CompressGain; /* Last used compressor gain*/
-#else
LVM_FLOAT TransitionGain; /* Transition gain */
LVM_FLOAT CompressGain; /* Last used compressor gain*/
-#endif
/* Sub-block configurations */
LVCS_StereoEnhancer_t StereoEnhancer; /* Stereo enhancer configuration */
@@ -155,44 +135,24 @@
/* Coefficient Structure */
typedef struct
{
-#ifdef BUILD_FLOAT
Biquad_FLOAT_Instance_t EqualiserBiquadInstance;
Biquad_FLOAT_Instance_t ReverbBiquadInstance;
Biquad_FLOAT_Instance_t SEBiquadInstanceMid;
Biquad_FLOAT_Instance_t SEBiquadInstanceSide;
-#else
- Biquad_Instance_t EqualiserBiquadInstance;
- Biquad_Instance_t ReverbBiquadInstance;
- Biquad_Instance_t SEBiquadInstanceMid;
- Biquad_Instance_t SEBiquadInstanceSide;
-#endif
} LVCS_Coefficient_t;
/* Data Structure */
typedef struct
{
-#ifdef BUILD_FLOAT
Biquad_2I_Order2_FLOAT_Taps_t EqualiserBiquadTaps;
Biquad_2I_Order2_FLOAT_Taps_t ReverbBiquadTaps;
Biquad_1I_Order1_FLOAT_Taps_t SEBiquadTapsMid;
Biquad_1I_Order2_FLOAT_Taps_t SEBiquadTapsSide;
-#else
- Biquad_2I_Order2_Taps_t EqualiserBiquadTaps;
- Biquad_2I_Order2_Taps_t ReverbBiquadTaps;
- Biquad_1I_Order1_Taps_t SEBiquadTapsMid;
- Biquad_1I_Order2_Taps_t SEBiquadTapsSide;
-#endif
} LVCS_Data_t;
void LVCS_TimerCallBack ( void* hInstance,
void* pCallBackParams,
LVM_INT32 CallbackParam);
-
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
-
#endif /* PRIVATE_H */
-
diff --git a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Process.c b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Process.cpp
similarity index 68%
rename from media/libeffects/lvm/lib/StereoWidening/src/LVCS_Process.c
rename to media/libeffects/lvm/lib/StereoWidening/src/LVCS_Process.cpp
index ef1d9eb..ded3bfa 100644
--- a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Process.c
+++ b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Process.cpp
@@ -15,7 +15,6 @@
* limitations under the License.
*/
-
/************************************************************************************/
/* */
/* Includes */
@@ -66,7 +65,6 @@
/* NOTES: */
/* */
/************************************************************************************/
-#ifdef BUILD_FLOAT
LVCS_ReturnStatus_en LVCS_Process_CS(LVCS_Handle_t hInstance,
const LVM_FLOAT *pInData,
LVM_FLOAT *pOutData,
@@ -178,74 +176,6 @@
return(LVCS_SUCCESS);
}
-#else
-LVCS_ReturnStatus_en LVCS_Process_CS(LVCS_Handle_t hInstance,
- const LVM_INT16 *pInData,
- LVM_INT16 *pOutData,
- LVM_UINT16 NumSamples)
-{
- const LVM_INT16 *pInput;
- LVCS_Instance_t *pInstance = (LVCS_Instance_t *)hInstance;
- LVM_INT16 *pScratch = (LVM_INT16 *)pInstance->MemoryTable.Region[LVCS_MEMREGION_TEMPORARY_FAST].pBaseAddress;
- LVCS_ReturnStatus_en err;
-
- /*
- * Check if the processing is inplace
- */
- if (pInData == pOutData)
- {
- /* Processing inplace */
- pInput = pScratch + (2*NumSamples);
- Copy_16((LVM_INT16 *)pInData, /* Source */
- (LVM_INT16 *)pInput, /* Destination */
- (LVM_INT16)(2*NumSamples)); /* Left and right */
- }
- else
- {
- /* Processing outplace */
- pInput = pInData;
- }
-
- /*
- * Call the stereo enhancer
- */
- err=LVCS_StereoEnhancer(hInstance, /* Instance handle */
- pInData, /* Pointer to the input data */
- pOutData, /* Pointer to the output data */
- NumSamples); /* Number of samples to process */
-
- /*
- * Call the reverb generator
- */
- err=LVCS_ReverbGenerator(hInstance, /* Instance handle */
- pOutData, /* Pointer to the input data */
- pOutData, /* Pointer to the output data */
- NumSamples); /* Number of samples to process */
-
- /*
- * Call the equaliser
- */
- err=LVCS_Equaliser(hInstance, /* Instance handle */
- pOutData, /* Pointer to the input data */
- NumSamples); /* Number of samples to process */
-
- /*
- * Call the bypass mixer
- */
- err=LVCS_BypassMixer(hInstance, /* Instance handle */
- pOutData, /* Pointer to the processed data */
- pInput, /* Pointer to the input (unprocessed) data */
- pOutData, /* Pointer to the output data */
- NumSamples); /* Number of samples to process */
-
- if(err !=LVCS_SUCCESS)
- {
- return err;
- }
-
- return(LVCS_SUCCESS);
-}
-#endif
/************************************************************************************/
/* */
/* FUNCTION: LVCS_Process */
@@ -272,7 +202,6 @@
/* NOTES: */
/* */
/************************************************************************************/
-#ifdef BUILD_FLOAT
LVCS_ReturnStatus_en LVCS_Process(LVCS_Handle_t hInstance,
const LVM_FLOAT *pInData,
LVM_FLOAT *pOutData,
@@ -311,7 +240,6 @@
pOutData,
NumSamples);
-
/*
* Compress to reduce expansion effect of Concert Sound and correct volume
* differences for difference settings. Not applied in test modes
@@ -403,7 +331,6 @@
pInstance->CompressGain = Gain;
}
-
if(pInstance->bInOperatingModeTransition == LVM_TRUE){
/*
@@ -455,168 +382,5 @@
}
}
-
return(LVCS_SUCCESS);
}
-#else
-LVCS_ReturnStatus_en LVCS_Process(LVCS_Handle_t hInstance,
- const LVM_INT16 *pInData,
- LVM_INT16 *pOutData,
- LVM_UINT16 NumSamples)
-{
-
- LVCS_Instance_t *pInstance =(LVCS_Instance_t *)hInstance;
- LVCS_ReturnStatus_en err;
-
- /*
- * Check the number of samples is not too large
- */
- if (NumSamples > pInstance->Capabilities.MaxBlockSize)
- {
- return(LVCS_TOOMANYSAMPLES);
- }
-
- /*
- * Check if the algorithm is enabled
- */
- if (pInstance->Params.OperatingMode != LVCS_OFF)
- {
- /*
- * Call CS process function
- */
- err=LVCS_Process_CS(hInstance,
- pInData,
- pOutData,
- NumSamples);
-
- /*
- * Compress to reduce expansion effect of Concert Sound and correct volume
- * differences for difference settings. Not applied in test modes
- */
- if ((pInstance->Params.OperatingMode == LVCS_ON)&&(pInstance->Params.CompressorMode == LVM_MODE_ON))
- {
- LVM_INT16 Gain = pInstance->VolCorrect.CompMin;
- LVM_INT32 Current1;
-
- Current1 = LVC_Mixer_GetCurrent(&pInstance->BypassMix.Mixer_Instance.MixerStream[0]);
- Gain = (LVM_INT16)( pInstance->VolCorrect.CompMin
- - (((LVM_INT32)pInstance->VolCorrect.CompMin * (Current1)) >> 15)
- + (((LVM_INT32)pInstance->VolCorrect.CompFull * (Current1)) >> 15) );
-
- if(NumSamples < LVCS_COMPGAINFRAME)
- {
- NonLinComp_D16(Gain, /* Compressor gain setting */
- pOutData,
- pOutData,
- (LVM_INT32)(2*NumSamples));
- }
- else
- {
- LVM_INT16 GainStep;
- LVM_INT16 FinalGain;
- LVM_INT16 SampleToProcess = NumSamples;
- LVM_INT16 *pOutPtr;
-
- /* Large changes in Gain can cause clicks in output
- Split data into small blocks and use interpolated gain values */
-
- GainStep = (LVM_INT16)(((Gain-pInstance->CompressGain) * LVCS_COMPGAINFRAME)/NumSamples);
-
- if((GainStep ==0)&&(pInstance->CompressGain < Gain))
- {
- GainStep=1;
- }
- else
- {
- if((GainStep ==0)&&(pInstance->CompressGain > Gain))
- {
- GainStep=-1;
- }
- }
-
- FinalGain = Gain;
- Gain = pInstance->CompressGain;
- pOutPtr = pOutData;
-
- while(SampleToProcess > 0)
- {
- Gain = (LVM_INT16)(Gain + GainStep);
- if((GainStep > 0)&& (FinalGain <= Gain))
- {
- Gain = FinalGain;
- GainStep =0;
- }
-
- if((GainStep < 0)&& (FinalGain > Gain))
- {
- Gain = FinalGain;
- GainStep =0;
- }
-
- if(SampleToProcess > LVCS_COMPGAINFRAME)
- {
- NonLinComp_D16(Gain, /* Compressor gain setting */
- pOutPtr,
- pOutPtr,
- (LVM_INT32)(2*LVCS_COMPGAINFRAME));
- pOutPtr +=(2*LVCS_COMPGAINFRAME);
- SampleToProcess = (LVM_INT16)(SampleToProcess-LVCS_COMPGAINFRAME);
- }
- else
- {
- NonLinComp_D16(Gain, /* Compressor gain setting */
- pOutPtr,
- pOutPtr,
- (LVM_INT32)(2*SampleToProcess));
-
- SampleToProcess = 0;
- }
-
- }
- }
-
- /* Store gain value*/
- pInstance->CompressGain = Gain;
- }
-
-
- if(pInstance->bInOperatingModeTransition == LVM_TRUE){
-
- /*
- * Re-init bypass mix when timer has completed
- */
- if ((pInstance->bTimerDone == LVM_TRUE) &&
- (pInstance->BypassMix.Mixer_Instance.MixerStream[1].CallbackSet == 0))
- {
- err=LVCS_BypassMixInit(hInstance,
- &pInstance->Params);
-
- if(err != LVCS_SUCCESS)
- {
- return err;
- }
-
- }
- else{
- LVM_Timer ( &pInstance->TimerInstance,
- (LVM_INT16)NumSamples);
- }
- }
- }
- else
- {
- if (pInData != pOutData)
- {
- /*
- * The algorithm is disabled so just copy the data
- */
- Copy_16((LVM_INT16 *)pInData, /* Source */
- (LVM_INT16 *)pOutData, /* Destination */
- (LVM_INT16)(2*NumSamples)); /* Left and right */
- }
- }
-
-
- return(LVCS_SUCCESS);
-}
-#endif
diff --git a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_ReverbGenerator.c b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_ReverbGenerator.cpp
similarity index 65%
rename from media/libeffects/lvm/lib/StereoWidening/src/LVCS_ReverbGenerator.c
rename to media/libeffects/lvm/lib/StereoWidening/src/LVCS_ReverbGenerator.cpp
index 1085101..d0e6e09 100644
--- a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_ReverbGenerator.c
+++ b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_ReverbGenerator.cpp
@@ -57,7 +57,6 @@
/* 2. The numerator coefficients of the filter are negated to cause an inversion. */
/* */
/************************************************************************************/
-#ifdef BUILD_FLOAT
LVCS_ReturnStatus_en LVCS_ReverbGeneratorInit(LVCS_Handle_t hInstance,
LVCS_Params_t *pParams)
{
@@ -71,7 +70,6 @@
BQ_FLOAT_Coefs_t Coeffs;
const BiquadA012B12CoefsSP_t *pReverbCoefTable;
-
pData = (LVCS_Data_t *) \
pInstance->MemoryTable.Region[LVCS_MEMREGION_PERSISTENT_FAST_DATA].pBaseAddress;
@@ -91,7 +89,6 @@
*/
Delay = (LVM_UINT16)LVCS_StereoDelayCS[(LVM_UINT16)pParams->SampleRate];
-
pConfig->DelaySize = (LVM_INT16)(2 * Delay);
pConfig->DelayOffset = 0;
LoadConst_Float(0, /* Value */
@@ -112,8 +109,7 @@
Coeffs.B2 = (LVM_FLOAT)-pReverbCoefTable[Offset].B2;
LoadConst_Float(0, /* Value */
- (void *)&pData->ReverbBiquadTaps, /* Destination Cast to void:
- no dereferencing in function*/
+ (LVM_FLOAT *)&pData->ReverbBiquadTaps, /* Destination */
/* Number of words */
(LVM_UINT16)(sizeof(pData->ReverbBiquadTaps) / sizeof(LVM_FLOAT)));
@@ -132,7 +128,6 @@
break;
}
-
/*
* Setup the mixer
*/
@@ -148,90 +143,6 @@
}
return(LVCS_SUCCESS);
}
-#else
-LVCS_ReturnStatus_en LVCS_ReverbGeneratorInit(LVCS_Handle_t hInstance,
- LVCS_Params_t *pParams)
-{
-
- LVM_UINT16 Delay;
- LVM_UINT16 Offset;
- LVCS_Instance_t *pInstance = (LVCS_Instance_t *)hInstance;
- LVCS_ReverbGenerator_t *pConfig = (LVCS_ReverbGenerator_t *)&pInstance->Reverberation;
- LVCS_Data_t *pData = (LVCS_Data_t *)pInstance->MemoryTable.Region[LVCS_MEMREGION_PERSISTENT_FAST_DATA].pBaseAddress;
- LVCS_Coefficient_t *pCoefficients = (LVCS_Coefficient_t *)pInstance->MemoryTable.Region[LVCS_MEMREGION_PERSISTENT_FAST_COEF].pBaseAddress;
- BQ_C16_Coefs_t Coeffs;
- const BiquadA012B12CoefsSP_t *pReverbCoefTable;
-
- /*
- * Initialise the delay and filters if:
- * - the sample rate has changed
- * - the speaker type has changed to or from the mobile speaker
- */
- if(pInstance->Params.SampleRate != pParams->SampleRate ) /* Sample rate change test */
-
- {
- /*
- * Setup the delay
- */
- Delay = (LVM_UINT16)LVCS_StereoDelayCS[(LVM_UINT16)pParams->SampleRate];
-
-
- pConfig->DelaySize = (LVM_INT16)(2 * Delay);
- pConfig->DelayOffset = 0;
- LoadConst_16(0, /* Value */
- (LVM_INT16 *)&pConfig->StereoSamples[0], /* Destination */
- (LVM_UINT16)(sizeof(pConfig->StereoSamples)/sizeof(LVM_INT16))); /* Number of words */
-
- /*
- * Setup the filters
- */
- Offset = (LVM_UINT16)pParams->SampleRate;
- pReverbCoefTable = (BiquadA012B12CoefsSP_t*)&LVCS_ReverbCoefTable[0];
-
- /* Convert incoming coefficients to the required format/ordering */
- Coeffs.A0 = (LVM_INT16)pReverbCoefTable[Offset].A0;
- Coeffs.A1 = (LVM_INT16)pReverbCoefTable[Offset].A1;
- Coeffs.A2 = (LVM_INT16)pReverbCoefTable[Offset].A2;
- Coeffs.B1 = (LVM_INT16)-pReverbCoefTable[Offset].B1;
- Coeffs.B2 = (LVM_INT16)-pReverbCoefTable[Offset].B2;
-
- LoadConst_16(0, /* Value */
- (void *)&pData->ReverbBiquadTaps, /* Destination Cast to void: no dereferencing in function*/
- (LVM_UINT16)(sizeof(pData->ReverbBiquadTaps)/sizeof(LVM_INT16))); /* Number of words */
-
- BQ_2I_D16F16Css_TRC_WRA_01_Init(&pCoefficients->ReverbBiquadInstance,
- &pData->ReverbBiquadTaps,
- &Coeffs);
-
- /* Callbacks */
- switch(pReverbCoefTable[Offset].Scale)
- {
- case 14:
- pConfig->pBiquadCallBack = BQ_2I_D16F16C14_TRC_WRA_01;
- break;
- case 15:
- pConfig->pBiquadCallBack = BQ_2I_D16F16C15_TRC_WRA_01;
- break;
- }
-
-
- /*
- * Setup the mixer
- */
- pConfig->ProcGain = (LVM_UINT16)(HEADPHONEGAINPROC);
- pConfig->UnprocGain = (LVM_UINT16)(HEADPHONEGAINUNPROC);
- }
-
- if(pInstance->Params.ReverbLevel != pParams->ReverbLevel)
- {
- LVM_INT32 ReverbPercentage=83886; // 1 Percent Reverb i.e 1/100 in Q 23 format
- ReverbPercentage*=pParams->ReverbLevel; // Actual Reverb Level in Q 23 format
- pConfig->ReverbLevel=(LVM_INT16)(ReverbPercentage>>8); // Reverb Level in Q 15 format
- }
-
- return(LVCS_SUCCESS);
-}
-#endif
/************************************************************************************/
/* */
/* FUNCTION: LVCS_Reverb */
@@ -270,7 +181,6 @@
/* 2. The Gain is combined with the LPF and incorporated in to the coefficients */
/* */
/************************************************************************************/
-#ifdef BUILD_FLOAT
LVCS_ReturnStatus_en LVCS_ReverbGenerator(LVCS_Handle_t hInstance,
const LVM_FLOAT *pInData,
LVM_FLOAT *pOutData,
@@ -301,12 +211,11 @@
(LVM_INT16)(2 * NumSamples)); /* Left and right */
}
-
/*
* Check if the reverb is required
*/
/* Disable when CS4MS in stereo mode */
- if (((pInstance->Params.SpeakerType == LVCS_HEADPHONE) || \
+ if ((((LVCS_OutputDevice_en)pInstance->Params.SpeakerType == LVCS_HEADPHONE) || \
(pInstance->Params.SpeakerType == LVCS_EX_HEADPHONES) ||
(pInstance->Params.SourceFormat != LVCS_STEREO)) &&
/* For validation testing */
@@ -338,7 +247,6 @@
(LVM_FLOAT *)pScratch,
(LVM_INT16)(2 * NumSamples));
-
/*
* Apply the delay mix
*/
@@ -349,87 +257,7 @@
&pConfig->DelayOffset,
(LVM_INT16)NumSamples);
-
}
return(LVCS_SUCCESS);
}
-#else
-LVCS_ReturnStatus_en LVCS_ReverbGenerator(LVCS_Handle_t hInstance,
- const LVM_INT16 *pInData,
- LVM_INT16 *pOutData,
- LVM_UINT16 NumSamples)
-{
-
- LVCS_Instance_t *pInstance = (LVCS_Instance_t *)hInstance;
- LVCS_ReverbGenerator_t *pConfig = (LVCS_ReverbGenerator_t *)&pInstance->Reverberation;
- LVCS_Coefficient_t *pCoefficients = (LVCS_Coefficient_t *)pInstance->MemoryTable.Region[LVCS_MEMREGION_PERSISTENT_FAST_COEF].pBaseAddress;
- LVM_INT16 *pScratch = (LVM_INT16 *)pInstance->MemoryTable.Region[LVCS_MEMREGION_TEMPORARY_FAST].pBaseAddress;
-
-
- /*
- * Copy the data to the output in outplace processing
- */
- if (pInData != pOutData)
- {
- /*
- * Reverb not required so just copy the data
- */
- Copy_16((LVM_INT16 *)pInData, /* Source */
- (LVM_INT16 *)pOutData, /* Destination */
- (LVM_INT16)(2*NumSamples)); /* Left and right */
- }
-
-
- /*
- * Check if the reverb is required
- */
- if (((pInstance->Params.SpeakerType == LVCS_HEADPHONE) || /* Disable when CS4MS in stereo mode */
- (pInstance->Params.SpeakerType == LVCS_EX_HEADPHONES) ||
- (pInstance->Params.SourceFormat != LVCS_STEREO)) &&
- ((pInstance->Params.OperatingMode & LVCS_REVERBSWITCH) !=0)) /* For validation testing */
- {
- /********************************************************************************/
- /* */
- /* Copy the input data to scratch memory and filter it */
- /* */
- /********************************************************************************/
-
- /*
- * Copy the input data to the scratch memory
- */
- Copy_16((LVM_INT16 *)pInData, /* Source */
- (LVM_INT16 *)pScratch, /* Destination */
- (LVM_INT16)(2*NumSamples)); /* Left and right */
-
-
- /*
- * Filter the data
- */
- (pConfig->pBiquadCallBack)((Biquad_Instance_t*)&pCoefficients->ReverbBiquadInstance,
- (LVM_INT16 *)pScratch,
- (LVM_INT16 *)pScratch,
- (LVM_INT16)NumSamples);
-
- Mult3s_16x16( (LVM_INT16 *)pScratch,
- pConfig->ReverbLevel,
- (LVM_INT16 *)pScratch,
- (LVM_INT16)(2*NumSamples));
-
-
- /*
- * Apply the delay mix
- */
- DelayMix_16x16((LVM_INT16 *)pScratch,
- &pConfig->StereoSamples[0],
- pConfig->DelaySize,
- pOutData,
- &pConfig->DelayOffset,
- (LVM_INT16)NumSamples);
-
-
- }
-
- return(LVCS_SUCCESS);
-}
-#endif
diff --git a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_ReverbGenerator.h b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_ReverbGenerator.h
index f94d4e4..1bc4338 100644
--- a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_ReverbGenerator.h
+++ b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_ReverbGenerator.h
@@ -18,11 +18,6 @@
#ifndef __LVCS_REVERBGENERATOR_H__
#define __LVCS_REVERBGENERATOR_H__
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
-
/************************************************************************************/
/* */
/* Includes */
@@ -31,7 +26,6 @@
#include "LVC_Mixer.h"
-
/************************************************************************************/
/* */
/* Defines */
@@ -41,14 +35,12 @@
#define HEADPHONEGAINPROC LVCS_HEADPHONE_PROCGAIN
#define HEADPHONEGAINUNPROC LVCS_HEADPHONE_UNPROCGAIN
-
/************************************************************************************/
/* */
/* Structures */
/* */
/************************************************************************************/
-
/* Reverberation module structure */
typedef struct
{
@@ -58,23 +50,14 @@
LVM_INT16 DelayOffset;
LVM_INT16 ProcGain;
LVM_INT16 UnprocGain;
-#ifndef BUILD_FLOAT
- LVM_INT16 StereoSamples[2*LVCS_STEREODELAY_CS_48KHZ];
- /* Reverb Level */
- LVM_INT16 ReverbLevel;
- /* Filter */
- void (*pBiquadCallBack) (Biquad_Instance_t*, LVM_INT16*, LVM_INT16*, LVM_INT16);
-#else
LVM_FLOAT StereoSamples[2 * LVCS_STEREODELAY_CS_MAX_VAL];
/* Reverb Level */
LVM_FLOAT ReverbLevel;
/* Filter */
void (*pBiquadCallBack) (Biquad_FLOAT_Instance_t*,
LVM_FLOAT*, LVM_FLOAT*, LVM_INT16);
-#endif
} LVCS_ReverbGenerator_t;
-
/************************************************************************************/
/* */
/* Function prototypes */
@@ -83,19 +66,9 @@
LVCS_ReturnStatus_en LVCS_ReverbGeneratorInit(LVCS_Handle_t hInstance,
LVCS_Params_t *pParams);
-#ifdef BUILD_FLOAT
LVCS_ReturnStatus_en LVCS_ReverbGenerator(LVCS_Handle_t hInstance,
const LVM_FLOAT *pInput,
LVM_FLOAT *pOutput,
LVM_UINT16 NumSamples);
-#else
-LVCS_ReturnStatus_en LVCS_ReverbGenerator(LVCS_Handle_t hInstance,
- const LVM_INT16 *pInput,
- LVM_INT16 *pOutput,
- LVM_UINT16 NumSamples);
-#endif
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
#endif /* REVERB_H */
diff --git a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_StereoEnhancer.c b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_StereoEnhancer.cpp
similarity index 64%
rename from media/libeffects/lvm/lib/StereoWidening/src/LVCS_StereoEnhancer.c
rename to media/libeffects/lvm/lib/StereoWidening/src/LVCS_StereoEnhancer.cpp
index 2992c35..7fd8444 100644
--- a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_StereoEnhancer.c
+++ b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_StereoEnhancer.cpp
@@ -49,7 +49,6 @@
/* NOTES: */
/* */
/************************************************************************************/
-#ifdef BUILD_FLOAT
LVCS_ReturnStatus_en LVCS_SEnhancerInit(LVCS_Handle_t hInstance,
LVCS_Params_t *pParams)
{
@@ -63,7 +62,6 @@
BQ_FLOAT_Coefs_t CoeffsSide;
const BiquadA012B12CoefsSP_t *pSESideCoefs;
-
pData = (LVCS_Data_t *) \
pInstance->MemoryTable.Region[LVCS_MEMREGION_PERSISTENT_FAST_DATA].pBaseAddress;
@@ -89,8 +87,7 @@
/* Clear the taps */
LoadConst_Float(0, /* Value */
- (void *)&pData->SEBiquadTapsMid, /* Destination Cast to void:\
- no dereferencing in function*/
+ (LVM_FLOAT *)&pData->SEBiquadTapsMid, /* Destination */
/* Number of words */
(LVM_UINT16)(sizeof(pData->SEBiquadTapsMid) / sizeof(LVM_FLOAT)));
@@ -117,8 +114,7 @@
/* Clear the taps */
LoadConst_Float(0, /* Value */
- (void *)&pData->SEBiquadTapsSide, /* Destination Cast to void:\
- no dereferencing in function*/
+ (LVM_FLOAT *)&pData->SEBiquadTapsSide, /* Destination */
/* Number of words */
(LVM_UINT16)(sizeof(pData->SEBiquadTapsSide) / sizeof(LVM_FLOAT)));
/* Callbacks */
@@ -142,99 +138,8 @@
}
-
return(LVCS_SUCCESS);
}
-#else
-LVCS_ReturnStatus_en LVCS_SEnhancerInit(LVCS_Handle_t hInstance,
- LVCS_Params_t *pParams)
-{
-
- LVM_UINT16 Offset;
- LVCS_Instance_t *pInstance = (LVCS_Instance_t *)hInstance;
- LVCS_StereoEnhancer_t *pConfig = (LVCS_StereoEnhancer_t *)&pInstance->StereoEnhancer;
- LVCS_Data_t *pData = (LVCS_Data_t *)pInstance->MemoryTable.Region[LVCS_MEMREGION_PERSISTENT_FAST_DATA].pBaseAddress;
- LVCS_Coefficient_t *pCoefficient = (LVCS_Coefficient_t *)pInstance->MemoryTable.Region[LVCS_MEMREGION_PERSISTENT_FAST_COEF].pBaseAddress;
- FO_C16_Coefs_t CoeffsMid;
- BQ_C16_Coefs_t CoeffsSide;
- const BiquadA012B12CoefsSP_t *pSESideCoefs;
-
- /*
- * If the sample rate or speaker type has changed update the filters
- */
- if ((pInstance->Params.SampleRate != pParams->SampleRate) ||
- (pInstance->Params.SpeakerType != pParams->SpeakerType))
- {
- /*
- * Set the filter coefficients based on the sample rate
- */
- /* Mid filter */
- Offset = (LVM_UINT16)pParams->SampleRate;
-
- /* Convert incoming coefficients to the required format/ordering */
- CoeffsMid.A0 = (LVM_INT16) LVCS_SEMidCoefTable[Offset].A0;
- CoeffsMid.A1 = (LVM_INT16) LVCS_SEMidCoefTable[Offset].A1;
- CoeffsMid.B1 = (LVM_INT16)-LVCS_SEMidCoefTable[Offset].B1;
-
- /* Clear the taps */
- LoadConst_16(0, /* Value */
- (void *)&pData->SEBiquadTapsMid, /* Destination Cast to void:\
- no dereferencing in function*/
- (LVM_UINT16)(sizeof(pData->SEBiquadTapsMid)/sizeof(LVM_UINT16))); /* Number of words */
-
- FO_1I_D16F16Css_TRC_WRA_01_Init(&pCoefficient->SEBiquadInstanceMid,
- &pData->SEBiquadTapsMid,
- &CoeffsMid);
-
- /* Callbacks */
- if(LVCS_SEMidCoefTable[Offset].Scale==15)
- {
- pConfig->pBiquadCallBack_Mid = FO_1I_D16F16C15_TRC_WRA_01;
- }
-
- Offset = (LVM_UINT16)(pParams->SampleRate);
- pSESideCoefs = (BiquadA012B12CoefsSP_t*)&LVCS_SESideCoefTable[0];
-
- /* Side filter */
- /* Convert incoming coefficients to the required format/ordering */
- CoeffsSide.A0 = (LVM_INT16) pSESideCoefs[Offset].A0;
- CoeffsSide.A1 = (LVM_INT16) pSESideCoefs[Offset].A1;
- CoeffsSide.A2 = (LVM_INT16) pSESideCoefs[Offset].A2;
- CoeffsSide.B1 = (LVM_INT16)-pSESideCoefs[Offset].B1;
- CoeffsSide.B2 = (LVM_INT16)-pSESideCoefs[Offset].B2;
-
- /* Clear the taps */
- LoadConst_16(0, /* Value */
- (void *)&pData->SEBiquadTapsSide, /* Destination Cast to void:\
- no dereferencing in function*/
- (LVM_UINT16)(sizeof(pData->SEBiquadTapsSide)/sizeof(LVM_UINT16))); /* Number of words */
-
-
- /* Callbacks */
- switch(pSESideCoefs[Offset].Scale)
- {
- case 14:
- BQ_1I_D16F32Css_TRC_WRA_01_Init(&pCoefficient->SEBiquadInstanceSide,
- &pData->SEBiquadTapsSide,
- &CoeffsSide);
-
- pConfig->pBiquadCallBack_Side = BQ_1I_D16F32C14_TRC_WRA_01;
- break;
- case 15:
- BQ_1I_D16F16Css_TRC_WRA_01_Init(&pCoefficient->SEBiquadInstanceSide,
- &pData->SEBiquadTapsSide,
- &CoeffsSide);
-
- pConfig->pBiquadCallBack_Side = BQ_1I_D16F16C15_TRC_WRA_01;
- break;
- }
-
- }
-
-
- return(LVCS_SUCCESS);
-}
-#endif
/************************************************************************************/
/* */
/* FUNCTION: LVCS_StereoEnhance */
@@ -273,7 +178,6 @@
/* 1. The side filter is not used in Mobile Speaker mode */
/* */
/************************************************************************************/
-#ifdef BUILD_FLOAT
LVCS_ReturnStatus_en LVCS_StereoEnhancer(LVCS_Handle_t hInstance,
const LVM_FLOAT *pInData,
LVM_FLOAT *pOutData,
@@ -356,81 +260,3 @@
return(LVCS_SUCCESS);
}
-#else
-LVCS_ReturnStatus_en LVCS_StereoEnhancer(LVCS_Handle_t hInstance,
- const LVM_INT16 *pInData,
- LVM_INT16 *pOutData,
- LVM_UINT16 NumSamples)
-{
-
- LVCS_Instance_t *pInstance = (LVCS_Instance_t *)hInstance;
- LVCS_StereoEnhancer_t *pConfig = (LVCS_StereoEnhancer_t *)&pInstance->StereoEnhancer;
- LVCS_Coefficient_t *pCoefficient = (LVCS_Coefficient_t *)pInstance->MemoryTable.Region[LVCS_MEMREGION_PERSISTENT_FAST_COEF].pBaseAddress;
- LVM_INT16 *pScratch = (LVM_INT16 *)pInstance->MemoryTable.Region[LVCS_MEMREGION_TEMPORARY_FAST].pBaseAddress;
-
- /*
- * Check if the Stereo Enhancer is enabled
- */
- if ((pInstance->Params.OperatingMode & LVCS_STEREOENHANCESWITCH) != 0)
- {
- /*
- * Convert from stereo to middle and side
- */
- From2iToMS_16x16(pInData,
- pScratch,
- pScratch+NumSamples,
- (LVM_INT16)NumSamples);
-
- /*
- * Apply filter to the middle signal
- */
- if (pInstance->OutputDevice == LVCS_HEADPHONE)
- {
- (pConfig->pBiquadCallBack_Mid)((Biquad_Instance_t*)&pCoefficient->SEBiquadInstanceMid,
- (LVM_INT16 *)pScratch,
- (LVM_INT16 *)pScratch,
- (LVM_INT16)NumSamples);
- }
- else
- {
- Mult3s_16x16(pScratch, /* Source */
- (LVM_INT16)pConfig->MidGain, /* Gain */
- pScratch, /* Destination */
- (LVM_INT16)NumSamples); /* Number of samples */
- }
-
- /*
- * Apply the filter the side signal only in stereo mode for headphones
- * and in all modes for mobile speakers
- */
- if (pInstance->Params.SourceFormat == LVCS_STEREO)
- {
- (pConfig->pBiquadCallBack_Side)((Biquad_Instance_t*)&pCoefficient->SEBiquadInstanceSide,
- (LVM_INT16 *)(pScratch + NumSamples),
- (LVM_INT16 *)(pScratch + NumSamples),
- (LVM_INT16)NumSamples);
- }
-
- /*
- * Convert from middle and side to stereo
- */
- MSTo2i_Sat_16x16(pScratch,
- pScratch+NumSamples,
- pOutData,
- (LVM_INT16)NumSamples);
-
- }
- else
- {
- /*
- * The stereo enhancer is disabled so just copy the data
- */
- Copy_16((LVM_INT16 *)pInData, /* Source */
- (LVM_INT16 *)pOutData, /* Destination */
- (LVM_INT16)(2*NumSamples)); /* Left and right */
-
- }
-
- return(LVCS_SUCCESS);
-}
-#endif
diff --git a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_StereoEnhancer.h b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_StereoEnhancer.h
index 4125f24..12a5982 100644
--- a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_StereoEnhancer.h
+++ b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_StereoEnhancer.h
@@ -18,11 +18,6 @@
#ifndef __LVCS_STEREOENHANCER_H__
#define __LVCS_STEREOENHANCER_H__
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
-
/************************************************************************************/
/* */
/* Includes */
@@ -33,7 +28,6 @@
#include "LVCS_Headphone_Coeffs.h" /* Headphone coefficients */
#include "BIQUAD.h"
-
/************************************************************************************/
/* */
/* Structures */
@@ -44,17 +38,6 @@
typedef struct
{
-#ifndef BUILD_FLOAT
- /*
- * Middle filter
- */
- void (*pBiquadCallBack_Mid)(Biquad_Instance_t*, LVM_INT16*, LVM_INT16*, LVM_INT16);
- /*
- * Side filter
- */
- void (*pBiquadCallBack_Side)(Biquad_Instance_t*, LVM_INT16*, LVM_INT16*, LVM_INT16);
- LVM_UINT16 MidGain; /* Middle gain in mobile speaker mode */
-#else
/*
* Middle filter
*/
@@ -67,10 +50,8 @@
void (*pBiquadCallBack_Side)(Biquad_FLOAT_Instance_t*,
LVM_FLOAT*, LVM_FLOAT*, LVM_INT16);
LVM_FLOAT MidGain; /* Middle gain in mobile speaker mode */
-#endif
} LVCS_StereoEnhancer_t;
-
/************************************************************************************/
/* */
/* Function prototypes */
@@ -80,19 +61,9 @@
LVCS_ReturnStatus_en LVCS_SEnhancerInit(LVCS_Handle_t hInstance,
LVCS_Params_t *pParams);
-#ifndef BUILD_FLOAT
-LVCS_ReturnStatus_en LVCS_StereoEnhancer(LVCS_Handle_t hInstance,
- const LVM_INT16 *pInData,
- LVM_INT16 *pOutData,
- LVM_UINT16 NumSamples);
-#else
LVCS_ReturnStatus_en LVCS_StereoEnhancer(LVCS_Handle_t hInstance,
const LVM_FLOAT *pInData,
LVM_FLOAT *pOutData,
LVM_UINT16 NumSamples);
-#endif
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
#endif /* STEREOENHANCE_H */
diff --git a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Tables.c b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Tables.cpp
similarity index 93%
rename from media/libeffects/lvm/lib/StereoWidening/src/LVCS_Tables.c
rename to media/libeffects/lvm/lib/StereoWidening/src/LVCS_Tables.cpp
index a1fb48f..d79db61 100644
--- a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Tables.c
+++ b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Tables.cpp
@@ -15,7 +15,6 @@
* limitations under the License.
*/
-
/************************************************************************************/
/* */
/* Includes */
@@ -23,11 +22,11 @@
/************************************************************************************/
#include "LVCS_Private.h"
+#include "LVCS_Tables.h"
#include "Filters.h" /* Filter definitions */
#include "BIQUAD.h" /* Biquad definitions */
#include "LVCS_Headphone_Coeffs.h" /* Headphone coefficients */
-
/************************************************************************************/
/* */
/* Stereo Enhancer coefficient constant tables */
@@ -72,7 +71,6 @@
CS_MIDDLE_48000_A1,
CS_MIDDLE_48000_B1,
(LVM_UINT16 )CS_MIDDLE_48000_SCALE}
-#ifdef HIGHER_FS
,
{CS_MIDDLE_88200_A0, /* 88kS/s coefficients */
CS_MIDDLE_88200_A1,
@@ -90,7 +88,6 @@
CS_MIDDLE_192000_A1,
CS_MIDDLE_192000_B1,
(LVM_UINT16 )CS_MIDDLE_192000_SCALE}
-#endif
};
/* Coefficient table for the side filter */
@@ -150,7 +147,6 @@
CS_SIDE_48000_B1,
CS_SIDE_48000_B2,
(LVM_UINT16 )CS_SIDE_48000_SCALE}
-#ifdef HIGHER_FS
,
{CS_SIDE_88200_A0, /* 88kS/s coefficients */
CS_SIDE_88200_A1,
@@ -176,10 +172,8 @@
CS_SIDE_192000_B1,
CS_SIDE_192000_B2,
(LVM_UINT16 )CS_SIDE_192000_SCALE}
-#endif
};
-
/************************************************************************************/
/* */
/* Equaliser coefficient constant tables */
@@ -242,7 +236,6 @@
CS_EQUALISER_48000_B1,
CS_EQUALISER_48000_B2,
(LVM_UINT16 )CS_EQUALISER_48000_SCALE},
-#ifdef HIGHER_FS
{CS_EQUALISER_88200_A0, /* 88kS/s coeffieients */
CS_EQUALISER_88200_A1,
CS_EQUALISER_88200_A2,
@@ -267,7 +260,6 @@
CS_EQUALISER_192000_B1,
CS_EQUALISER_192000_B2,
(LVM_UINT16 )CS_EQUALISER_192000_SCALE},
-#endif
/* Concert Sound EX Headphone coefficients */
{CSEX_EQUALISER_8000_A0, /* 8kS/s coefficients */
@@ -324,7 +316,6 @@
CSEX_EQUALISER_48000_B1,
CSEX_EQUALISER_48000_B2,
(LVM_UINT16 )CSEX_EQUALISER_48000_SCALE}
-#ifdef HIGHER_FS
,
{CSEX_EQUALISER_88200_A0, /* 88kS/s coefficients */
CSEX_EQUALISER_88200_A1,
@@ -350,10 +341,8 @@
CSEX_EQUALISER_192000_B1,
CSEX_EQUALISER_192000_B2,
(LVM_UINT16 )CSEX_EQUALISER_192000_SCALE}
-#endif
};
-
/************************************************************************************/
/* */
/* Reverb delay constant tables */
@@ -439,7 +428,6 @@
CS_REVERB_48000_B1,
CS_REVERB_48000_B2,
(LVM_UINT16 )CS_REVERB_48000_SCALE}
-#ifdef HIGHER_FS
,
{CS_REVERB_88200_A0, /* 88kS/s coefficients */
CS_REVERB_88200_A1,
@@ -465,10 +453,8 @@
CS_REVERB_192000_B1,
CS_REVERB_192000_B2,
(LVM_UINT16 )CS_REVERB_192000_SCALE}
-#endif
};
-
/************************************************************************************/
/* */
/* Bypass mixer constant tables */
@@ -490,7 +476,6 @@
LVCS_EX_HEADPHONE_GAIN}
};
-
/************************************************************************************/
/* */
/* Volume correction table */
@@ -517,7 +502,6 @@
/* */
/************************************************************************************/
const LVCS_VolCorrect_t LVCS_VolCorrectTable[] = {
-#ifdef BUILD_FLOAT
{0.433362f, /* Headphone, stereo mode */
0.000000f,
1.000024f,
@@ -534,24 +518,6 @@
0.000000f,
1.000024f,
1.412640f}
-#else
- {14200, /* Headphone, stereo mode */
- 0,
- 4096,
- 5786},
- {14200, /* EX Headphone, stereo mode */
- 0,
- 4096,
- 5786},
- {32767, /* Headphone, mono mode */
- 0,
- 4096,
- 5786},
- {32767, /* EX Headphone, mono mode */
- 0,
- 4096,
- 5786}
-#endif
};
/************************************************************************************/
@@ -569,14 +535,11 @@
#define LVCS_VOL_TC_Fs32000 32721 /* Floating point value 0.998565674 */
#define LVCS_VOL_TC_Fs44100 32734 /* Floating point value 0.998962402 */
#define LVCS_VOL_TC_Fs48000 32737 /* Floating point value 0.999053955 */
-#if defined(BUILD_FLOAT) && defined(HIGHER_FS)
#define LVCS_VOL_TC_Fs88200 32751 /* Floating point value 0.999481066 */
#define LVCS_VOL_TC_Fs96000 32751 /* Floating point value 0.999511703 */ /* Todo @ need to re check this value*/
#define LVCS_VOL_TC_Fs176400 32759 /* Floating point value 0.999740499 */
#define LVCS_VOL_TC_Fs192000 32763 /* Floating point value 0.999877925 */ /* Todo @ need to re check this value*/
-#endif
-#if defined(BUILD_FLOAT) && defined(HIGHER_FS)
const LVM_INT16 LVCS_VolumeTCTable[13] = {LVCS_VOL_TC_Fs8000,
LVCS_VOL_TC_Fs11025,
LVCS_VOL_TC_Fs12000,
@@ -591,25 +554,12 @@
LVCS_VOL_TC_Fs176400,
LVCS_VOL_TC_Fs192000
};
-#else
-const LVM_INT16 LVCS_VolumeTCTable[9] = {LVCS_VOL_TC_Fs8000,
- LVCS_VOL_TC_Fs11025,
- LVCS_VOL_TC_Fs12000,
- LVCS_VOL_TC_Fs16000,
- LVCS_VOL_TC_Fs22050,
- LVCS_VOL_TC_Fs24000,
- LVCS_VOL_TC_Fs32000,
- LVCS_VOL_TC_Fs44100,
- LVCS_VOL_TC_Fs48000
-};
-#endif
/************************************************************************************/
/* */
/* Sample rate table */
/* */
/************************************************************************************/
-#if defined(BUILD_FLOAT) && defined(HIGHER_FS)
const LVM_INT32 LVCS_SampleRateTable[13] = {8000,
11025,
12000,
@@ -624,15 +574,3 @@
176400,
192000
};
-#else
-const LVM_INT16 LVCS_SampleRateTable[9] = {8000,
- 11025,
- 12000,
- 16000,
- 22050,
- 24000,
- 32000,
- 44100,
- 48000
-};
-#endif
diff --git a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Tables.h b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Tables.h
index 3f6c4c8..5490699 100644
--- a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Tables.h
+++ b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Tables.h
@@ -18,10 +18,6 @@
#ifndef __LVCS_TABLES_H__
#define __LVCS_TABLES_H__
-#ifdef __cplusplus
-extern "C" {
-#endif /* __cplusplus */
-
/************************************************************************************/
/* */
/* Includes */
@@ -104,15 +100,13 @@
extern const LVCS_VolCorrect_t LVCS_VolCorrectTable[];
extern const LVM_INT16 LVCS_VolumeTCTable[];
-
/************************************************************************************/
/* */
/* Sample rates */
/* */
/************************************************************************************/
-extern LVM_INT32 LVCS_SampleRateTable[];
-
+extern const LVM_INT32 LVCS_SampleRateTable[];
/*Speaker coeffient tables*/
extern LVM_UINT16 LVCS_MS_Small_SEMiddleGainTable[];
@@ -142,11 +136,5 @@
extern LVCS_VolCorrect_t LVCS_MS_Large_VolCorrectTable[];
extern LVM_UINT16 LVCS_MS_Large_ReverbGainTable[];
-
-
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
-
#endif /* __LVCS_TABLES_H__ */
diff --git a/media/libeffects/lvm/tests/Android.bp b/media/libeffects/lvm/tests/Android.bp
index 003ce9e..674c246 100644
--- a/media/libeffects/lvm/tests/Android.bp
+++ b/media/libeffects/lvm/tests/Android.bp
@@ -35,8 +35,6 @@
srcs: ["lvmtest.cpp"],
cflags: [
- "-DBUILD_FLOAT",
- "-DHIGHER_FS",
"-DSUPPORT_MC",
"-Wall",
diff --git a/media/libeffects/lvm/tests/lvmtest.cpp b/media/libeffects/lvm/tests/lvmtest.cpp
index 5b58dd1..a4ace6c 100644
--- a/media/libeffects/lvm/tests/lvmtest.cpp
+++ b/media/libeffects/lvm/tests/lvmtest.cpp
@@ -482,10 +482,6 @@
pContext->pBundledContext->SamplesToExitCountVirt = 0;
pContext->pBundledContext->SamplesToExitCountBb = 0;
pContext->pBundledContext->SamplesToExitCountEq = 0;
-#if defined(BUILD_FLOAT) && !defined(NATIVE_FLOAT_BUFFER)
- pContext->pBundledContext->pInputBuffer = NULL;
- pContext->pBundledContext->pOutputBuffer = NULL;
-#endif
for (int i = 0; i < FIVEBAND_NUMBANDS; i++) {
pContext->pBundledContext->bandGaindB[i] = EQNB_5BandSoftPresets[i];
}
diff --git a/media/libeffects/lvm/wrapper/Android.bp b/media/libeffects/lvm/wrapper/Android.bp
index 16fa126..afc4220 100644
--- a/media/libeffects/lvm/wrapper/Android.bp
+++ b/media/libeffects/lvm/wrapper/Android.bp
@@ -1,6 +1,3 @@
-// The wrapper -DBUILD_FLOAT needs to match
-// the lvm library -DBUILD_FLOAT.
-
// music bundle wrapper
cc_library_shared {
name: "libbundlewrapper",
@@ -14,10 +11,8 @@
vendor: true,
srcs: ["Bundle/EffectBundle.cpp"],
- cflags: [
+ cppflags: [
"-fvisibility=hidden",
- "-DBUILD_FLOAT",
- "-DHIGHER_FS",
"-DSUPPORT_MC",
"-Wall",
@@ -56,10 +51,8 @@
vendor: true,
srcs: ["Reverb/EffectReverb.cpp"],
- cflags: [
+ cppflags: [
"-fvisibility=hidden",
- "-DBUILD_FLOAT",
- "-DHIGHER_FS",
"-Wall",
"-Werror",
diff --git a/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp b/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
index 10dda19..d569c6a 100644
--- a/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
+++ b/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
@@ -81,7 +81,6 @@
} \
}
-
// NXP SW BassBoost UUID
const effect_descriptor_t gBassBoostDescriptor = {
{0x0634f220, 0xddd4, 0x11db, 0xa0fc, { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b }},
@@ -258,26 +257,6 @@
pContext->pBundledContext->firstVolume = LVM_TRUE;
pContext->pBundledContext->volume = 0;
- #ifdef LVM_PCM
- char fileName[256];
- snprintf(fileName, 256, "/data/tmp/bundle_%p_pcm_in.pcm", pContext->pBundledContext);
- pContext->pBundledContext->PcmInPtr = fopen(fileName, "w");
- if (pContext->pBundledContext->PcmInPtr == NULL) {
- ALOGV("cannot open %s", fileName);
- ret = -EINVAL;
- goto exit;
- }
-
- snprintf(fileName, 256, "/data/tmp/bundle_%p_pcm_out.pcm", pContext->pBundledContext);
- pContext->pBundledContext->PcmOutPtr = fopen(fileName, "w");
- if (pContext->pBundledContext->PcmOutPtr == NULL) {
- ALOGV("cannot open %s", fileName);
- fclose(pContext->pBundledContext->PcmInPtr);
- pContext->pBundledContext->PcmInPtr = NULL;
- ret = -EINVAL;
- goto exit;
- }
- #endif
/* Saved strength is used to return the exact strength that was used in the set to the get
* because we map the original strength range of 0:1000 to 1:15, and this will avoid
@@ -295,10 +274,6 @@
pContext->pBundledContext->SamplesToExitCountVirt = 0;
pContext->pBundledContext->SamplesToExitCountBb = 0;
pContext->pBundledContext->SamplesToExitCountEq = 0;
-#if defined(BUILD_FLOAT) && !defined(NATIVE_FLOAT_BUFFER)
- pContext->pBundledContext->pInputBuffer = NULL;
- pContext->pBundledContext->pOutputBuffer = NULL;
-#endif
for (int i = 0; i < FIVEBAND_NUMBANDS; i++) {
pContext->pBundledContext->bandGaindB[i] = EQNB_5BandSoftPresets[i];
}
@@ -443,17 +418,6 @@
(pSessionContext->bEqualizerInstantiated ==LVM_FALSE) &&
(pSessionContext->bVirtualizerInstantiated==LVM_FALSE))
{
-#ifdef LVM_PCM
- if (pContext->pBundledContext->PcmInPtr != NULL) {
- fclose(pContext->pBundledContext->PcmInPtr);
- pContext->pBundledContext->PcmInPtr = NULL;
- }
- if (pContext->pBundledContext->PcmOutPtr != NULL) {
- fclose(pContext->pBundledContext->PcmOutPtr);
- pContext->pBundledContext->PcmOutPtr = NULL;
- }
-#endif
-
// Clear the SessionIndex
for(int i=0; i<LVM_MAX_SESSIONS; i++){
@@ -474,10 +438,6 @@
if (pContext->pBundledContext->workBuffer != NULL) {
free(pContext->pBundledContext->workBuffer);
}
-#if defined(BUILD_FLOAT) && !defined(NATIVE_FLOAT_BUFFER)
- free(pContext->pBundledContext->pInputBuffer);
- free(pContext->pBundledContext->pOutputBuffer);
-#endif
delete pContext->pBundledContext;
pContext->pBundledContext = LVM_NULL;
}
@@ -759,7 +719,6 @@
// pOut: pointer to updated stereo 16 bit output data
//
//----------------------------------------------------------------------------
-#ifdef BUILD_FLOAT
int LvmBundle_process(effect_buffer_t *pIn,
effect_buffer_t *pOut,
int frameCount,
@@ -769,30 +728,6 @@
effect_buffer_t *pOutTmp;
const LVM_INT32 NrChannels =
audio_channel_count_from_out_mask(pContext->config.inputCfg.channels);
-#ifndef NATIVE_FLOAT_BUFFER
- if (pContext->pBundledContext->pInputBuffer == nullptr ||
- pContext->pBundledContext->frameCount < frameCount) {
- free(pContext->pBundledContext->pInputBuffer);
- pContext->pBundledContext->pInputBuffer =
- (LVM_FLOAT *)calloc(frameCount, sizeof(LVM_FLOAT) * NrChannels);
- }
-
- if (pContext->pBundledContext->pOutputBuffer == nullptr ||
- pContext->pBundledContext->frameCount < frameCount) {
- free(pContext->pBundledContext->pOutputBuffer);
- pContext->pBundledContext->pOutputBuffer =
- (LVM_FLOAT *)calloc(frameCount, sizeof(LVM_FLOAT) * NrChannels);
- }
-
- if (pContext->pBundledContext->pInputBuffer == nullptr ||
- pContext->pBundledContext->pOutputBuffer == nullptr) {
- ALOGE("LVM_ERROR : LvmBundle_process memory allocation for float buffer's failed");
- return -EINVAL;
- }
-
- LVM_FLOAT * const pInputBuff = pContext->pBundledContext->pInputBuffer;
- LVM_FLOAT * const pOutputBuff = pContext->pBundledContext->pOutputBuffer;
-#endif
if (pContext->config.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_WRITE){
pOutTmp = pOut;
@@ -814,123 +749,25 @@
return -EINVAL;
}
-#ifdef LVM_PCM
- fwrite(pIn,
- frameCount * sizeof(effect_buffer_t) * NrChannels,
- 1,
- pContext->pBundledContext->PcmInPtr);
- fflush(pContext->pBundledContext->PcmInPtr);
-#endif
-#ifndef NATIVE_FLOAT_BUFFER
- /* Converting input data from fixed point to float point */
- memcpy_to_float_from_i16(pInputBuff, pIn, frameCount * NrChannels);
-
- /* Process the samples */
- LvmStatus = LVM_Process(pContext->pBundledContext->hInstance, /* Instance handle */
- pInputBuff, /* Input buffer */
- pOutputBuff, /* Output buffer */
- (LVM_UINT16)frameCount, /* Number of samples to read */
- 0); /* Audio Time */
-
- /* Converting output data from float point to fixed point */
- memcpy_to_i16_from_float(pOutTmp, pOutputBuff, frameCount * NrChannels);
-
-#else
/* Process the samples */
LvmStatus = LVM_Process(pContext->pBundledContext->hInstance, /* Instance handle */
pIn, /* Input buffer */
pOutTmp, /* Output buffer */
(LVM_UINT16)frameCount, /* Number of samples to read */
0); /* Audio Time */
-#endif
LVM_ERROR_CHECK(LvmStatus, "LVM_Process", "LvmBundle_process")
if(LvmStatus != LVM_SUCCESS) return -EINVAL;
-#ifdef LVM_PCM
- fwrite(pOutTmp,
- frameCount * sizeof(effect_buffer_t) * NrChannels,
- 1,
- pContext->pBundledContext->PcmOutPtr);
- fflush(pContext->pBundledContext->PcmOutPtr);
-#endif
if (pContext->config.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE){
for (int i = 0; i < frameCount * NrChannels; i++) {
-#ifndef NATIVE_FLOAT_BUFFER
- pOut[i] = clamp16((LVM_INT32)pOut[i] + (LVM_INT32)pOutTmp[i]);
-#else
pOut[i] = pOut[i] + pOutTmp[i];
-#endif
}
}
return 0;
} /* end LvmBundle_process */
-#else // BUILD_FLOAT
-
-int LvmBundle_process(LVM_INT16 *pIn,
- LVM_INT16 *pOut,
- int frameCount,
- EffectContext *pContext) {
-
- LVM_ReturnStatus_en LvmStatus = LVM_SUCCESS; /* Function call status */
- LVM_INT16 *pOutTmp;
-
- if (pContext->config.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_WRITE){
- pOutTmp = pOut;
- } else if (pContext->config.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE){
- if (pContext->pBundledContext->frameCount != frameCount) {
- if (pContext->pBundledContext->workBuffer != NULL) {
- free(pContext->pBundledContext->workBuffer);
- }
- pContext->pBundledContext->workBuffer =
- (effect_buffer_t *)calloc(frameCount, sizeof(effect_buffer_t) * FCC_2);
- if (pContext->pBundledContext->workBuffer == NULL) {
- return -ENOMEM;
- }
- pContext->pBundledContext->frameCount = frameCount;
- }
- pOutTmp = pContext->pBundledContext->workBuffer;
- } else {
- ALOGV("LVM_ERROR : LvmBundle_process invalid access mode");
- return -EINVAL;
- }
-
-#ifdef LVM_PCM
- fwrite(pIn, frameCount * sizeof(*pIn) * FCC_2,
- 1 /* nmemb */, pContext->pBundledContext->PcmInPtr);
- fflush(pContext->pBundledContext->PcmInPtr);
-#endif
-
- //ALOGV("Calling LVM_Process");
-
- /* Process the samples */
- LvmStatus = LVM_Process(pContext->pBundledContext->hInstance, /* Instance handle */
- pIn, /* Input buffer */
- pOutTmp, /* Output buffer */
- (LVM_UINT16)frameCount, /* Number of samples to read */
- 0); /* Audio Time */
-
- LVM_ERROR_CHECK(LvmStatus, "LVM_Process", "LvmBundle_process")
- if(LvmStatus != LVM_SUCCESS) return -EINVAL;
-
-#ifdef LVM_PCM
- fwrite(pOutTmp, frameCount * sizeof(*pOutTmp) * FCC_2,
- 1 /* nmemb */, pContext->pBundledContext->PcmOutPtr);
- fflush(pContext->pBundledContext->PcmOutPtr);
-#endif
-
- if (pContext->config.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE){
- for (int i=0; i<frameCount*2; i++){
- pOut[i] = clamp16((LVM_INT32)pOut[i] + (LVM_INT32)pOutTmp[i]);
- }
- }
- return 0;
-} /* end LvmBundle_process */
-
-#endif // BUILD_FLOAT
-
//----------------------------------------------------------------------------
// EqualizerUpdateActiveParams()
//----------------------------------------------------------------------------
@@ -953,7 +790,6 @@
//ALOGV("\tEqualizerUpdateActiveParams just Got -> %d\n",
// ActiveParams.pEQNB_BandDefinition[band].Gain);
-
for (int i = 0; i < FIVEBAND_NUMBANDS; i++) {
ActiveParams.pEQNB_BandDefinition[i].Frequency = EQNB_5BandPresetsFrequencies[i];
ActiveParams.pEQNB_BandDefinition[i].QFactor = EQNB_5BandPresetsQFactors[i];
@@ -1290,7 +1126,6 @@
SampleRate = LVM_FS_48000;
pContext->pBundledContext->SamplesPerSecond = 48000 * NrChannels;
break;
-#if defined(BUILD_FLOAT) && defined(HIGHER_FS)
case 88200:
SampleRate = LVM_FS_88200;
pContext->pBundledContext->SamplesPerSecond = 88200 * NrChannels;
@@ -1307,7 +1142,6 @@
SampleRate = LVM_FS_192000;
pContext->pBundledContext->SamplesPerSecond = 192000 * NrChannels;
break;
-#endif
default:
ALOGV("\tEffect_setConfig invalid sampling rate %d", pConfig->inputCfg.samplingRate);
return -EINVAL;
@@ -2051,8 +1885,6 @@
LVM_ReturnStatus_en LvmStatus=LVM_SUCCESS; /* Function call status */
LVM_INT16 Balance = 0;
-
-
pContext->pBundledContext->positionSaved = position;
Balance = VolumeConvertStereoPosition(pContext->pBundledContext->positionSaved);
@@ -2097,7 +1929,6 @@
return 0;
} /* end VolumeSetStereoPosition */
-
//----------------------------------------------------------------------------
// VolumeGetStereoPosition()
//----------------------------------------------------------------------------
@@ -2970,7 +2801,6 @@
return status;
} /* end Volume_getParameter */
-
//----------------------------------------------------------------------------
// Volume_setParameter()
//----------------------------------------------------------------------------
@@ -3422,17 +3252,10 @@
pContext->pBundledContext->NumberEffectsCalled = 0;
/* Process all the available frames, block processing is
handled internalLY by the LVM bundle */
-#ifdef NATIVE_FLOAT_BUFFER
processStatus = android::LvmBundle_process(inBuffer->f32,
outBuffer->f32,
outBuffer->frameCount,
pContext);
-#else
- processStatus = android::LvmBundle_process(inBuffer->s16,
- outBuffer->s16,
- outBuffer->frameCount,
- pContext);
-#endif
if (processStatus != 0){
ALOGV("\tLVM_ERROR : LvmBundle_process returned error %d", processStatus);
if (status == 0) {
@@ -3447,11 +3270,7 @@
if (pContext->config.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE) {
for (size_t i = 0; i < outBuffer->frameCount * NrChannels; ++i) {
-#ifdef NATIVE_FLOAT_BUFFER
outBuffer->f32[i] += inBuffer->f32[i];
-#else
- outBuffer->s16[i] = clamp16((LVM_INT32)outBuffer->s16[i] + inBuffer->s16[i]);
-#endif
}
} else if (outBuffer->raw != inBuffer->raw) {
memcpy(outBuffer->raw,
diff --git a/media/libeffects/lvm/wrapper/Bundle/EffectBundle.h b/media/libeffects/lvm/wrapper/Bundle/EffectBundle.h
index e4aacd0..524e103 100644
--- a/media/libeffects/lvm/wrapper/Bundle/EffectBundle.h
+++ b/media/libeffects/lvm/wrapper/Bundle/EffectBundle.h
@@ -23,10 +23,6 @@
#include <LVM.h>
#include <limits.h>
-#if __cplusplus
-extern "C" {
-#endif
-
#define FIVEBAND_NUMBANDS 5
#define MAX_NUM_BANDS 5
#define MAX_CALL_SIZE 256
@@ -37,7 +33,6 @@
#define EQUALIZER_CUP_LOAD_ARM9E 220 // Expressed in 0.1 MIPS
#define VOLUME_CUP_LOAD_ARM9E 0 // Expressed in 0.1 MIPS
#define BUNDLE_MEM_USAGE 25 // Expressed in kB
-//#define LVM_PCM
#ifndef OPENSL_ES_H_
static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6,
@@ -99,14 +94,6 @@
int frameCount;
int32_t bandGaindB[FIVEBAND_NUMBANDS];
int volume;
- #ifdef LVM_PCM
- FILE *PcmInPtr;
- FILE *PcmOutPtr;
- #endif
-#if defined(BUILD_FLOAT) && !defined(NATIVE_FLOAT_BUFFER)
- LVM_FLOAT *pInputBuffer;
- LVM_FLOAT *pOutputBuffer;
-#endif
#ifdef SUPPORT_MC
LVM_INT32 ChMask;
#endif
@@ -137,7 +124,6 @@
BundledEffectContext *pBundledContext;
};
-
/* enumerated parameter settings for Volume effect */
typedef enum
{
@@ -228,9 +214,4 @@
static const float LimitLevel_virtualizerContribution = 1.9;
-#if __cplusplus
-} // extern "C"
-#endif
-
-
#endif /*ANDROID_EFFECTBUNDLE_H_*/
diff --git a/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp b/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp
index 602f607..1cb81a6 100644
--- a/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp
+++ b/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp
@@ -79,7 +79,6 @@
{-400, -200, 1300, 900, 0, 2, 0, 10, 1000, 750},
};
-
// NXP SW auxiliary environmental reverb
const effect_descriptor_t gAuxEnvReverbDescriptor = {
{ 0xc2e5d5f0, 0x94bd, 0x4763, 0x9cac, { 0x4e, 0x23, 0x4d, 0x06, 0x83, 0x9e } },
@@ -136,11 +135,7 @@
&gInsertPresetReverbDescriptor
};
-#ifdef BUILD_FLOAT
typedef float process_buffer_t; // process in float
-#else
-typedef int32_t process_buffer_t; // process in Q4_27
-#endif // BUILD_FLOAT
struct ReverbContext{
const struct effect_interface_s *itfe;
@@ -154,10 +149,6 @@
int16_t SavedDiffusion;
int16_t SavedDensity;
bool bEnabled;
- #ifdef LVM_PCM
- FILE *PcmInPtr;
- FILE *PcmOutPtr;
- #endif
LVM_Fs_en SampleRate;
process_buffer_t *InFrames;
process_buffer_t *OutFrames;
@@ -183,11 +174,7 @@
#define REVERB_DEFAULT_PRESET REVERB_PRESET_NONE
-#ifdef BUILD_FLOAT
#define REVERB_SEND_LEVEL 0.75f // 0.75 in 4.12 format
-#else
-#define REVERB_SEND_LEVEL (0x0C00) // 0.75 in 4.12 format
-#endif
#define REVERB_UNIT_VOLUME (0x1000) // 1.0 in 4.12 format
//--- local function prototypes
@@ -269,18 +256,6 @@
*pHandle = (effect_handle_t)pContext;
-#ifdef LVM_PCM
- pContext->PcmInPtr = NULL;
- pContext->PcmOutPtr = NULL;
-
- pContext->PcmInPtr = fopen("/data/tmp/reverb_pcm_in.pcm", "w");
- pContext->PcmOutPtr = fopen("/data/tmp/reverb_pcm_out.pcm", "w");
-
- if((pContext->PcmInPtr == NULL)||
- (pContext->PcmOutPtr == NULL)){
- return -EINVAL;
- }
-#endif
int channels = audio_channel_count_from_out_mask(pContext->config.inputCfg.channels);
@@ -304,10 +279,6 @@
return -EINVAL;
}
- #ifdef LVM_PCM
- fclose(pContext->PcmInPtr);
- fclose(pContext->PcmOutPtr);
- #endif
free(pContext->InFrames);
free(pContext->OutFrames);
pContext->bufferSizeIn = 0;
@@ -378,7 +349,6 @@
return -EINVAL;
}
-#ifdef BUILD_FLOAT
size_t inSize = frameCount * sizeof(process_buffer_t) * channels;
size_t outSize = frameCount * sizeof(process_buffer_t) * FCC_2;
if (pContext->InFrames == NULL ||
@@ -394,10 +364,6 @@
pContext->OutFrames = (process_buffer_t *)calloc(1, pContext->bufferSizeOut);
}
-#ifndef NATIVE_FLOAT_BUFFER
- effect_buffer_t * const OutFrames16 = (effect_buffer_t *)pContext->OutFrames;
-#endif
-#endif
// Check for NULL pointers
if ((pContext->InFrames == NULL) || (pContext->OutFrames == NULL)) {
@@ -405,47 +371,20 @@
return -EINVAL;
}
-#ifdef LVM_PCM
- fwrite(pIn, frameCount * sizeof(*pIn) * channels, 1 /* nmemb */, pContext->PcmInPtr);
- fflush(pContext->PcmInPtr);
-#endif
if (pContext->preset && pContext->nextPreset != pContext->curPreset) {
Reverb_LoadPreset(pContext);
}
if (pContext->auxiliary) {
-#ifdef BUILD_FLOAT
-#ifdef NATIVE_FLOAT_BUFFER
static_assert(std::is_same<decltype(*pIn), decltype(*pContext->InFrames)>::value,
"pIn and InFrames must be same type");
memcpy(pContext->InFrames, pIn, frameCount * channels * sizeof(*pIn));
-#else
- memcpy_to_float_from_i16(
- pContext->InFrames, pIn, frameCount * channels);
-#endif
-#else //no BUILD_FLOAT
- for (int i = 0; i < frameCount * channels; i++) {
- pContext->InFrames[i] = (process_buffer_t)pIn[i]<<8;
- }
-#endif
} else {
// insert reverb input is always stereo
for (int i = 0; i < frameCount; i++) {
-#ifdef BUILD_FLOAT
-#ifdef NATIVE_FLOAT_BUFFER
pContext->InFrames[2 * i] = (process_buffer_t)pIn[2 * i] * REVERB_SEND_LEVEL;
pContext->InFrames[2 * i + 1] = (process_buffer_t)pIn[2 * i + 1] * REVERB_SEND_LEVEL;
-#else
- pContext->InFrames[2 * i] =
- (process_buffer_t)pIn[2 * i] * REVERB_SEND_LEVEL / 32768.0f;
- pContext->InFrames[2 * i + 1] =
- (process_buffer_t)pIn[2 * i + 1] * REVERB_SEND_LEVEL / 32768.0f;
-#endif
-#else
- pContext->InFrames[2*i] = (pIn[2*i] * REVERB_SEND_LEVEL) >> 4; // <<8 + >>12
- pContext->InFrames[2*i+1] = (pIn[2*i+1] * REVERB_SEND_LEVEL) >> 4; // <<8 + >>12
-#endif
}
}
@@ -471,43 +410,16 @@
// Convert to 16 bits
if (pContext->auxiliary) {
-#ifdef BUILD_FLOAT
// nothing to do here
-#ifndef NATIVE_FLOAT_BUFFER
- // pContext->OutFrames and OutFrames16 point to the same buffer
- // make sure the float to int conversion happens in the right order.
- memcpy_to_i16_from_float(OutFrames16, pContext->OutFrames,
- (size_t)frameCount * FCC_2);
-#endif
-#else
- memcpy_to_i16_from_q4_27(OutFrames16, pContext->OutFrames, (size_t)frameCount * FCC_2);
-#endif
} else {
-#ifdef BUILD_FLOAT
-#ifdef NATIVE_FLOAT_BUFFER
for (int i = 0; i < frameCount * FCC_2; i++) { // always stereo here
// Mix with dry input
pContext->OutFrames[i] += pIn[i];
}
-#else
- for (int i = 0; i < frameCount * FCC_2; i++) { // always stereo here
- // pOutputBuff and OutFrames16 point to the same buffer
- // make sure the float to int conversion happens in the right order.
- pContext->OutFrames[i] += (process_buffer_t)pIn[i] / 32768.0f;
- }
- memcpy_to_i16_from_float(OutFrames16, pContext->OutFrames,
- (size_t)frameCount * FCC_2);
-#endif
-#else
- for (int i=0; i < frameCount * FCC_2; i++) { // always stereo here
- OutFrames16[i] = clamp16((pContext->OutFrames[i]>>8) + (process_buffer_t)pIn[i]);
- }
-#endif
// apply volume with ramp if needed
if ((pContext->leftVolume != pContext->prevLeftVolume ||
pContext->rightVolume != pContext->prevRightVolume) &&
pContext->volumeMode == REVERB_VOLUME_RAMP) {
-#if defined (BUILD_FLOAT) && defined (NATIVE_FLOAT_BUFFER)
// FIXME: still using int16 volumes.
// For reference: REVERB_UNIT_VOLUME (0x1000) // 1.0 in 4.12 format
float vl = (float)pContext->prevLeftVolume / 4096;
@@ -522,37 +434,14 @@
vl += incl;
vr += incr;
}
-#else
- LVM_INT32 vl = (LVM_INT32)pContext->prevLeftVolume << 16;
- LVM_INT32 incl = (((LVM_INT32)pContext->leftVolume << 16) - vl) / frameCount;
- LVM_INT32 vr = (LVM_INT32)pContext->prevRightVolume << 16;
- LVM_INT32 incr = (((LVM_INT32)pContext->rightVolume << 16) - vr) / frameCount;
-
- for (int i = 0; i < frameCount; i++) {
- OutFrames16[FCC_2 * i] =
- clamp16((LVM_INT32)((vl >> 16) * OutFrames16[2*i]) >> 12);
- OutFrames16[FCC_2 * i + 1] =
- clamp16((LVM_INT32)((vr >> 16) * OutFrames16[2*i+1]) >> 12);
-
- vl += incl;
- vr += incr;
- }
-#endif
pContext->prevLeftVolume = pContext->leftVolume;
pContext->prevRightVolume = pContext->rightVolume;
} else if (pContext->volumeMode != REVERB_VOLUME_OFF) {
if (pContext->leftVolume != REVERB_UNIT_VOLUME ||
pContext->rightVolume != REVERB_UNIT_VOLUME) {
for (int i = 0; i < frameCount; i++) {
-#if defined(BUILD_FLOAT) && defined(NATIVE_FLOAT_BUFFER)
pContext->OutFrames[FCC_2 * i] *= ((float)pContext->leftVolume / 4096);
pContext->OutFrames[FCC_2 * i + 1] *= ((float)pContext->rightVolume / 4096);
-#else
- OutFrames16[FCC_2 * i] =
- clamp16((LVM_INT32)(pContext->leftVolume * OutFrames16[2*i]) >> 12);
- OutFrames16[FCC_2 * i + 1] =
- clamp16((LVM_INT32)(pContext->rightVolume * OutFrames16[2*i+1]) >> 12);
-#endif
}
}
pContext->prevLeftVolume = pContext->leftVolume;
@@ -561,21 +450,12 @@
}
}
-#ifdef LVM_PCM
- fwrite(pContext->OutFrames, frameCount * sizeof(*pContext->OutFrames) * FCC_2,
- 1 /* nmemb */, pContext->PcmOutPtr);
- fflush(pContext->PcmOutPtr);
-#endif
// Accumulate if required
if (pContext->config.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE){
//ALOGV("\tBuffer access is ACCUMULATE");
for (int i = 0; i < frameCount * FCC_2; i++) { // always stereo here
-#ifndef NATIVE_FLOAT_BUFFER
- pOut[i] = clamp16((int32_t)pOut[i] + (int32_t)OutFrames16[i]);
-#else
pOut[i] += pContext->OutFrames[i];
-#endif
}
}else{
//ALOGV("\tBuffer access is WRITE");
@@ -654,7 +534,6 @@
//ALOGV("\tReverb_setConfig calling memcpy");
pContext->config = *pConfig;
-
switch (pConfig->inputCfg.samplingRate) {
case 8000:
SampleRate = LVM_FS_8000;
@@ -674,7 +553,6 @@
case 48000:
SampleRate = LVM_FS_48000;
break;
-#if defined(BUILD_FLOAT) && defined(HIGHER_FS)
case 88200:
SampleRate = LVM_FS_88200;
break;
@@ -687,7 +565,6 @@
case 192000:
SampleRate = LVM_FS_192000;
break;
-#endif
default:
ALOGV("\rReverb_setConfig invalid sampling rate %d", pConfig->inputCfg.samplingRate);
return -EINVAL;
@@ -1509,7 +1386,6 @@
//ALOGV("\tReverbGetDensity Succesfully returned from LVM_GetControlParameters\n");
//ALOGV("\tReverbGetDensity() just Got -> %d\n", ActiveParams.RoomSize);
-
Temp = (LVM_INT16)(((pContext->SavedDensity * 99) / 1000) + 1);
if(Temp != ActiveParams.RoomSize){
@@ -1557,7 +1433,6 @@
return 0;
}
-
//----------------------------------------------------------------------------
// Reverb_getParameter()
//----------------------------------------------------------------------------
@@ -1903,7 +1778,6 @@
return status;
} /* end Reverb_setParameter */
-
/**
* returns the size in bytes of the value of each environmental reverb parameter
*/
@@ -1951,17 +1825,10 @@
}
//ALOGV("\tReverb_process() Calling process with %d frames", outBuffer->frameCount);
/* Process all the available frames, block processing is handled internalLY by the LVM bundle */
-#if defined (BUILD_FLOAT) && defined (NATIVE_FLOAT_BUFFER)
status = process( inBuffer->f32,
outBuffer->f32,
outBuffer->frameCount,
pContext);
-#else
- status = process( inBuffer->s16,
- outBuffer->s16,
- outBuffer->frameCount,
- pContext);
-#endif
if (pContext->bEnabled == LVM_FALSE) {
if (pContext->SamplesToExitCount > 0) {
@@ -1986,7 +1853,6 @@
LVREV_ControlParams_st ActiveParams; /* Current control Parameters */
LVREV_ReturnStatus_en LvmStatus=LVREV_SUCCESS; /* Function call status */
-
if (pContext == NULL){
ALOGV("\tLVM_ERROR : Reverb_command ERROR pContext == NULL");
return -EINVAL;
@@ -2161,7 +2027,6 @@
return -EINVAL;
}
-
if (pReplyData != NULL) { // we have volume control
pContext->leftVolume = (LVM_INT16)((*(uint32_t *)pCmdData + (1 << 11)) >> 12);
pContext->rightVolume = (LVM_INT16)((*((uint32_t *)pCmdData + 1) + (1 << 11)) >> 12);
diff --git a/media/libeffects/lvm/wrapper/Reverb/EffectReverb.h b/media/libeffects/lvm/wrapper/Reverb/EffectReverb.h
index 8165f5a..96223a8 100644
--- a/media/libeffects/lvm/wrapper/Reverb/EffectReverb.h
+++ b/media/libeffects/lvm/wrapper/Reverb/EffectReverb.h
@@ -20,10 +20,6 @@
#include <audio_effects/effect_environmentalreverb.h>
#include <audio_effects/effect_presetreverb.h>
-#if __cplusplus
-extern "C" {
-#endif
-
#define MAX_NUM_BANDS 5
#define MAX_CALL_SIZE 256
#define LVREV_MAX_T60 7000
@@ -31,16 +27,11 @@
#define LVREV_MAX_FRAME_SIZE 2560
#define LVREV_CUP_LOAD_ARM9E 470 // Expressed in 0.1 MIPS
#define LVREV_MEM_USAGE (71+(LVREV_MAX_FRAME_SIZE>>7)) // Expressed in kB
-//#define LVM_PCM
typedef struct _LPFPair_t
{
int16_t Room_HF;
int16_t LPF;
} LPFPair_t;
-#if __cplusplus
-} // extern "C"
-#endif
-
#endif /*ANDROID_EFFECTREVERB_H_*/
diff --git a/media/libheif/HeifDecoderImpl.cpp b/media/libheif/HeifDecoderImpl.cpp
index a977300..bbc14a9 100644
--- a/media/libheif/HeifDecoderImpl.cpp
+++ b/media/libheif/HeifDecoderImpl.cpp
@@ -66,9 +66,6 @@
void close() {}
uint32_t getFlags() override { return 0; }
String8 toString() override { return String8("HeifDataSource"); }
- sp<DecryptHandle> DrmInitialization(const char*) override {
- return nullptr;
- }
private:
enum {
diff --git a/media/libmedia/Android.bp b/media/libmedia/Android.bp
index 1d33590..778ee44 100644
--- a/media/libmedia/Android.bp
+++ b/media/libmedia/Android.bp
@@ -1,10 +1,3 @@
-cc_defaults {
- name: "libmedia_defaults",
- include_dirs: [
- "bionic/libc/private",
- ],
-}
-
cc_library_headers {
name: "libmedia_headers",
vendor_available: true,
@@ -22,29 +15,13 @@
],
}
-cc_library {
- name: "libmedia_helper",
- vendor_available: true,
- vndk: {
- enabled: true,
- },
- double_loadable: true,
- srcs: ["AudioParameter.cpp", "TypeConverter.cpp"],
- cflags: [
- "-Werror",
- "-Wno-error=deprecated-declarations",
- "-Wall",
+filegroup {
+ name: "libmedia_omx_aidl",
+ srcs: [
+ "aidl/android/IGraphicBufferSource.aidl",
+ "aidl/android/IOMXBufferSource.aidl",
],
- shared_libs: ["libutils", "liblog"],
- header_libs: [
- "libmedia_headers",
- "libaudioclient_headers",
- "libaudio_system_headers",
- ],
- export_header_lib_headers: [
- "libmedia_headers",
- ],
- clang: true,
+ path: "aidl",
}
cc_library_shared {
@@ -56,13 +33,10 @@
double_loadable: true,
srcs: [
- "aidl/android/IGraphicBufferSource.aidl",
- "aidl/android/IOMXBufferSource.aidl",
+ ":libmedia_omx_aidl",
- "IMediaCodecList.cpp",
"IOMX.cpp",
"MediaCodecBuffer.cpp",
- "MediaCodecInfo.cpp",
"OMXBuffer.cpp",
"omx/1.0/WGraphicBufferSource.cpp",
"omx/1.0/WOmxBufferSource.cpp",
@@ -74,7 +48,7 @@
local_include_dirs: ["aidl"],
export_aidl_headers: true,
},
-
+
local_include_dirs: [
"include",
],
@@ -85,7 +59,6 @@
"libbinder",
"libcutils",
"libhidlbase",
- "libhidltransport",
"liblog",
"libstagefright_foundation",
"libui",
@@ -146,7 +119,6 @@
"libcutils",
"libgui",
"libhidlbase",
- "libhidltransport",
"liblog",
"libmedia_omx",
"libstagefright_foundation",
@@ -200,6 +172,7 @@
],
header_libs: [
+ "libmedia_headers",
"media_ndk_headers",
],
@@ -218,11 +191,52 @@
},
}
+cc_library_shared {
+ name: "libmedia_codeclist",
+
+ srcs: [
+ "IMediaCodecList.cpp",
+ "MediaCodecInfo.cpp",
+ ],
+
+ local_include_dirs: [
+ "include",
+ ],
+
+ shared_libs: [
+ "android.hardware.media.omx@1.0",
+ "libbinder",
+ "liblog",
+ "libstagefright_foundation",
+ "libutils",
+ ],
+
+ include_dirs: [
+ "system/libhidl/transport/token/1.0/utils/include",
+ ],
+
+ export_include_dirs: [
+ "include",
+ ],
+
+ cflags: [
+ "-Werror",
+ "-Wno-error=deprecated-declarations",
+ "-Wall",
+ ],
+
+ sanitize: {
+ misc_undefined: [
+ "unsigned-integer-overflow",
+ "signed-integer-overflow",
+ ],
+ cfi: true,
+ },
+}
+
cc_library {
name: "libmedia",
- defaults: [ "libmedia_defaults" ],
-
srcs: [
"IDataSource.cpp",
"BufferingSettings.cpp",
@@ -247,8 +261,6 @@
"mediarecorder.cpp",
"IMediaMetadataRetriever.cpp",
"mediametadataretriever.cpp",
- "MidiDeviceInfo.cpp",
- "JetPlayer.cpp",
"MediaScanner.cpp",
"MediaScannerClient.cpp",
"CharacterEncodingDetector.cpp",
@@ -256,7 +268,6 @@
"MediaProfiles.cpp",
"MediaResource.cpp",
"MediaResourcePolicy.cpp",
- "Visualizer.cpp",
"StringArray.cpp",
"NdkMediaFormatPriv.cpp",
"NdkMediaErrorPriv.cpp",
@@ -268,6 +279,7 @@
},
header_libs: [
+ "bionic_libc_platform_headers",
"libstagefright_headers",
"media_ndk_headers",
],
@@ -291,8 +303,8 @@
"libstagefright_foundation",
"libgui",
"libdl",
- "libaudioutils",
"libaudioclient",
+ "libmedia_codeclist",
"libmedia_omx",
],
@@ -306,7 +318,6 @@
static_libs: [
"libc_malloc_debug_backtrace", // for memory heap analysis
- "libmedia_midiiowrapper",
],
export_include_dirs: [
@@ -329,66 +340,3 @@
cfi: true,
},
}
-
-cc_library_static {
- name: "libmedia_player2_util",
-
- defaults: [ "libmedia_defaults" ],
-
- srcs: [
- "AudioParameter.cpp",
- "BufferingSettings.cpp",
- "DataSourceDesc.cpp",
- "MediaCodecBuffer.cpp",
- "Metadata.cpp",
- "NdkWrapper.cpp",
- ],
-
- shared_libs: [
- "libbinder",
- "libcutils",
- "liblog",
- "libmediandk",
- "libnativewindow",
- "libmediandk_utils",
- "libstagefright_foundation",
- "libui",
- "libutils",
- ],
-
- export_shared_lib_headers: [
- "libbinder",
- "libmediandk",
- ],
-
- header_libs: [
- "media_plugin_headers",
- ],
-
- include_dirs: [
- "frameworks/av/media/ndk",
- ],
-
- static_libs: [
- "libstagefright_rtsp",
- "libstagefright_timedtext",
- ],
-
- export_include_dirs: [
- "include",
- ],
-
- cflags: [
- "-Werror",
- "-Wno-error=deprecated-declarations",
- "-Wall",
- ],
-
- sanitize: {
- misc_undefined: [
- "unsigned-integer-overflow",
- "signed-integer-overflow",
- ],
- cfi: true,
- },
-}
diff --git a/media/libmedia/DataSourceDesc.cpp b/media/libmedia/DataSourceDesc.cpp
deleted file mode 100644
index b7ccbce..0000000
--- a/media/libmedia/DataSourceDesc.cpp
+++ /dev/null
@@ -1,37 +0,0 @@
-/*
- * Copyright 2018 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "DataSourceDesc"
-
-#include <media/DataSource.h>
-#include <media/DataSourceDesc.h>
-#include <media/MediaHTTPService.h>
-
-namespace android {
-
-static const int64_t kLongMax = 0x7ffffffffffffffL;
-
-DataSourceDesc::DataSourceDesc()
- : mType(TYPE_NONE),
- mFDOffset(0),
- mFDLength(kLongMax),
- mId(0),
- mStartPositionMs(0),
- mEndPositionMs(0) {
-}
-
-} // namespace android
diff --git a/media/libmedia/IDataSource.cpp b/media/libmedia/IDataSource.cpp
index 31c85af..61f0a68 100644
--- a/media/libmedia/IDataSource.cpp
+++ b/media/libmedia/IDataSource.cpp
@@ -23,7 +23,6 @@
#include <binder/IMemory.h>
#include <binder/Parcel.h>
-#include <drm/drm_framework_common.h>
#include <media/stagefright/foundation/ADebug.h>
namespace android {
@@ -35,7 +34,6 @@
CLOSE,
GET_FLAGS,
TO_STRING,
- DRM_INITIALIZATION,
};
struct BpDataSource : public BpInterface<IDataSource> {
@@ -95,47 +93,6 @@
remote()->transact(TO_STRING, data, &reply);
return reply.readString8();
}
-
- virtual sp<DecryptHandle> DrmInitialization(const char *mime) {
- Parcel data, reply;
- data.writeInterfaceToken(IDataSource::getInterfaceDescriptor());
- if (mime == NULL) {
- data.writeInt32(0);
- } else {
- data.writeInt32(1);
- data.writeCString(mime);
- }
- remote()->transact(DRM_INITIALIZATION, data, &reply);
- sp<DecryptHandle> handle;
- if (reply.dataAvail() != 0) {
- handle = new DecryptHandle();
- handle->decryptId = reply.readInt32();
- handle->mimeType = reply.readString8();
- handle->decryptApiType = reply.readInt32();
- handle->status = reply.readInt32();
-
- const int bufferLength = data.readInt32();
- if (bufferLength != -1) {
- handle->decryptInfo = new DecryptInfo();
- handle->decryptInfo->decryptBufferLength = bufferLength;
- }
-
- size_t size = data.readInt32();
- for (size_t i = 0; i < size; ++i) {
- DrmCopyControl key = (DrmCopyControl)data.readInt32();
- int value = data.readInt32();
- handle->copyControlVector.add(key, value);
- }
-
- size = data.readInt32();
- for (size_t i = 0; i < size; ++i) {
- String8 key = data.readString8();
- String8 value = data.readString8();
- handle->extendedData.add(key, value);
- }
- }
- return handle;
- }
};
IMPLEMENT_META_INTERFACE(DataSource, "android.media.IDataSource");
@@ -178,42 +135,6 @@
reply->writeString8(toString());
return NO_ERROR;
} break;
- case DRM_INITIALIZATION: {
- CHECK_INTERFACE(IDataSource, data, reply);
- const char *mime = NULL;
- const int32_t flag = data.readInt32();
- if (flag != 0) {
- mime = data.readCString();
- }
- sp<DecryptHandle> handle = DrmInitialization(mime);
- if (handle != NULL) {
- reply->writeInt32(handle->decryptId);
- reply->writeString8(handle->mimeType);
- reply->writeInt32(handle->decryptApiType);
- reply->writeInt32(handle->status);
-
- if (handle->decryptInfo != NULL) {
- reply->writeInt32(handle->decryptInfo->decryptBufferLength);
- } else {
- reply->writeInt32(-1);
- }
-
- size_t size = handle->copyControlVector.size();
- reply->writeInt32(size);
- for (size_t i = 0; i < size; ++i) {
- reply->writeInt32(handle->copyControlVector.keyAt(i));
- reply->writeInt32(handle->copyControlVector.valueAt(i));
- }
-
- size = handle->extendedData.size();
- reply->writeInt32(size);
- for (size_t i = 0; i < size; ++i) {
- reply->writeString8(handle->extendedData.keyAt(i));
- reply->writeString8(handle->extendedData.valueAt(i));
- }
- }
- return NO_ERROR;
- } break;
default:
return BBinder::onTransact(code, data, reply, flags);
diff --git a/media/libmedia/JetPlayer.cpp b/media/libmedia/JetPlayer.cpp
deleted file mode 100644
index 0d3c1ba..0000000
--- a/media/libmedia/JetPlayer.cpp
+++ /dev/null
@@ -1,471 +0,0 @@
-/*
- * Copyright (C) 2008 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "JetPlayer-C"
-
-#include <utils/Log.h>
-#include <media/JetPlayer.h>
-
-
-namespace android
-{
-
-static const int MIX_NUM_BUFFERS = 4;
-static const S_EAS_LIB_CONFIG* pLibConfig = NULL;
-
-//-------------------------------------------------------------------------------------------------
-JetPlayer::JetPlayer(void *javaJetPlayer, int maxTracks, int trackBufferSize) :
- mEventCallback(NULL),
- mJavaJetPlayerRef(javaJetPlayer),
- mTid(-1),
- mRender(false),
- mPaused(false),
- mMaxTracks(maxTracks),
- mEasData(NULL),
- mIoWrapper(NULL),
- mTrackBufferSize(trackBufferSize)
-{
- ALOGV("JetPlayer constructor");
- mPreviousJetStatus.currentUserID = -1;
- mPreviousJetStatus.segmentRepeatCount = -1;
- mPreviousJetStatus.numQueuedSegments = -1;
- mPreviousJetStatus.paused = true;
-}
-
-//-------------------------------------------------------------------------------------------------
-JetPlayer::~JetPlayer()
-{
- ALOGV("~JetPlayer");
- release();
-}
-
-//-------------------------------------------------------------------------------------------------
-int JetPlayer::init()
-{
- //Mutex::Autolock lock(&mMutex);
-
- EAS_RESULT result;
-
- // retrieve the EAS library settings
- if (pLibConfig == NULL)
- pLibConfig = EAS_Config();
- if (pLibConfig == NULL) {
- ALOGE("JetPlayer::init(): EAS library configuration could not be retrieved, aborting.");
- return EAS_FAILURE;
- }
-
- // init the EAS library
- result = EAS_Init(&mEasData);
- if (result != EAS_SUCCESS) {
- ALOGE("JetPlayer::init(): Error initializing Sonivox EAS library, aborting.");
- mState = EAS_STATE_ERROR;
- return result;
- }
- // init the JET library with the default app event controller range
- result = JET_Init(mEasData, NULL, sizeof(S_JET_CONFIG));
- if (result != EAS_SUCCESS) {
- ALOGE("JetPlayer::init(): Error initializing JET library, aborting.");
- mState = EAS_STATE_ERROR;
- return result;
- }
-
- // create the output AudioTrack
- mAudioTrack = new AudioTrack();
- status_t status = mAudioTrack->set(AUDIO_STREAM_MUSIC, //TODO parameterize this
- pLibConfig->sampleRate,
- AUDIO_FORMAT_PCM_16_BIT,
- audio_channel_out_mask_from_count(pLibConfig->numChannels),
- (size_t) mTrackBufferSize,
- AUDIO_OUTPUT_FLAG_NONE);
- if (status != OK) {
- ALOGE("JetPlayer::init(): Error initializing JET library; AudioTrack error %d", status);
- mAudioTrack.clear();
- mState = EAS_STATE_ERROR;
- return EAS_FAILURE;
- }
-
- // create render and playback thread
- {
- Mutex::Autolock l(mMutex);
- ALOGV("JetPlayer::init(): trying to start render thread");
- mThread = new JetPlayerThread(this);
- mThread->run("jetRenderThread", ANDROID_PRIORITY_AUDIO);
- mCondition.wait(mMutex);
- }
- if (mTid > 0) {
- // render thread started, we're ready
- ALOGV("JetPlayer::init(): render thread(%d) successfully started.", mTid);
- mState = EAS_STATE_READY;
- } else {
- ALOGE("JetPlayer::init(): failed to start render thread.");
- mState = EAS_STATE_ERROR;
- return EAS_FAILURE;
- }
-
- return EAS_SUCCESS;
-}
-
-void JetPlayer::setEventCallback(jetevent_callback eventCallback)
-{
- Mutex::Autolock l(mMutex);
- mEventCallback = eventCallback;
-}
-
-//-------------------------------------------------------------------------------------------------
-int JetPlayer::release()
-{
- ALOGV("JetPlayer::release()");
- Mutex::Autolock lock(mMutex);
- mPaused = true;
- mRender = false;
- if (mEasData) {
- JET_Pause(mEasData);
- JET_CloseFile(mEasData);
- JET_Shutdown(mEasData);
- EAS_Shutdown(mEasData);
- }
- delete mIoWrapper;
- mIoWrapper = NULL;
- if (mAudioTrack != 0) {
- mAudioTrack->stop();
- mAudioTrack->flush();
- mAudioTrack.clear();
- }
- if (mAudioBuffer) {
- delete mAudioBuffer;
- mAudioBuffer = NULL;
- }
- mEasData = NULL;
-
- return EAS_SUCCESS;
-}
-
-
-//-------------------------------------------------------------------------------------------------
-int JetPlayer::render() {
- EAS_RESULT result = EAS_FAILURE;
- EAS_I32 count;
- int temp;
- bool audioStarted = false;
-
- ALOGV("JetPlayer::render(): entering");
-
- // allocate render buffer
- mAudioBuffer =
- new EAS_PCM[pLibConfig->mixBufferSize * pLibConfig->numChannels * MIX_NUM_BUFFERS];
-
- // signal main thread that we started
- {
- Mutex::Autolock l(mMutex);
- mTid = gettid();
- ALOGV("JetPlayer::render(): render thread(%d) signal", mTid);
- mCondition.signal();
- }
-
- while (1) {
-
- mMutex.lock(); // [[[[[[[[ LOCK ---------------------------------------
-
- if (mEasData == NULL) {
- mMutex.unlock();
- ALOGV("JetPlayer::render(): NULL EAS data, exiting render.");
- goto threadExit;
- }
-
- // nothing to render, wait for client thread to wake us up
- while (!mRender)
- {
- ALOGV("JetPlayer::render(): signal wait");
- if (audioStarted) {
- mAudioTrack->pause();
- // we have to restart the playback once we start rendering again
- audioStarted = false;
- }
- mCondition.wait(mMutex);
- ALOGV("JetPlayer::render(): signal rx'd");
- }
-
- // render midi data into the input buffer
- int num_output = 0;
- EAS_PCM* p = mAudioBuffer;
- for (int i = 0; i < MIX_NUM_BUFFERS; i++) {
- result = EAS_Render(mEasData, p, pLibConfig->mixBufferSize, &count);
- if (result != EAS_SUCCESS) {
- ALOGE("JetPlayer::render(): EAS_Render returned error %ld", result);
- }
- p += count * pLibConfig->numChannels;
- num_output += count * pLibConfig->numChannels * sizeof(EAS_PCM);
-
- // send events that were generated (if any) to the event callback
- fireEventsFromJetQueue();
- }
-
- // update playback state
- //ALOGV("JetPlayer::render(): updating state");
- JET_Status(mEasData, &mJetStatus);
- fireUpdateOnStatusChange();
- mPaused = mJetStatus.paused;
-
- mMutex.unlock(); // UNLOCK ]]]]]]]] -----------------------------------
-
- // check audio output track
- if (mAudioTrack == NULL) {
- ALOGE("JetPlayer::render(): output AudioTrack was not created");
- goto threadExit;
- }
-
- // Write data to the audio hardware
- //ALOGV("JetPlayer::render(): writing to audio output");
- if ((temp = mAudioTrack->write(mAudioBuffer, num_output)) < 0) {
- ALOGE("JetPlayer::render(): Error in writing:%d",temp);
- return temp;
- }
-
- // start audio output if necessary
- if (!audioStarted) {
- ALOGV("JetPlayer::render(): starting audio playback");
- mAudioTrack->start();
- audioStarted = true;
- }
-
- }//while (1)
-
-threadExit:
- if (mAudioTrack != NULL) {
- mAudioTrack->stop();
- mAudioTrack->flush();
- }
- delete [] mAudioBuffer;
- mAudioBuffer = NULL;
- mMutex.lock();
- mTid = -1;
- mCondition.signal();
- mMutex.unlock();
- return result;
-}
-
-
-//-------------------------------------------------------------------------------------------------
-// fire up an update if any of the status fields has changed
-// precondition: mMutex locked
-void JetPlayer::fireUpdateOnStatusChange()
-{
- if ( (mJetStatus.currentUserID != mPreviousJetStatus.currentUserID)
- ||(mJetStatus.segmentRepeatCount != mPreviousJetStatus.segmentRepeatCount) ) {
- if (mEventCallback) {
- mEventCallback(
- JetPlayer::JET_USERID_UPDATE,
- mJetStatus.currentUserID,
- mJetStatus.segmentRepeatCount,
- mJavaJetPlayerRef);
- }
- mPreviousJetStatus.currentUserID = mJetStatus.currentUserID;
- mPreviousJetStatus.segmentRepeatCount = mJetStatus.segmentRepeatCount;
- }
-
- if (mJetStatus.numQueuedSegments != mPreviousJetStatus.numQueuedSegments) {
- if (mEventCallback) {
- mEventCallback(
- JetPlayer::JET_NUMQUEUEDSEGMENT_UPDATE,
- mJetStatus.numQueuedSegments,
- -1,
- mJavaJetPlayerRef);
- }
- mPreviousJetStatus.numQueuedSegments = mJetStatus.numQueuedSegments;
- }
-
- if (mJetStatus.paused != mPreviousJetStatus.paused) {
- if (mEventCallback) {
- mEventCallback(JetPlayer::JET_PAUSE_UPDATE,
- mJetStatus.paused,
- -1,
- mJavaJetPlayerRef);
- }
- mPreviousJetStatus.paused = mJetStatus.paused;
- }
-
-}
-
-
-//-------------------------------------------------------------------------------------------------
-// fire up all the JET events in the JET engine queue (until the queue is empty)
-// precondition: mMutex locked
-void JetPlayer::fireEventsFromJetQueue()
-{
- if (!mEventCallback) {
- // no callback, just empty the event queue
- while (JET_GetEvent(mEasData, NULL, NULL)) { }
- return;
- }
-
- EAS_U32 rawEvent;
- while (JET_GetEvent(mEasData, &rawEvent, NULL)) {
- mEventCallback(
- JetPlayer::JET_EVENT,
- rawEvent,
- -1,
- mJavaJetPlayerRef);
- }
-}
-
-
-//-------------------------------------------------------------------------------------------------
-int JetPlayer::loadFromFile(const char* path)
-{
- ALOGV("JetPlayer::loadFromFile(): path=%s", path);
-
- Mutex::Autolock lock(mMutex);
-
- delete mIoWrapper;
- mIoWrapper = new MidiIoWrapper(path);
-
- EAS_RESULT result = JET_OpenFile(mEasData, mIoWrapper->getLocator());
- if (result != EAS_SUCCESS)
- mState = EAS_STATE_ERROR;
- else
- mState = EAS_STATE_OPEN;
- return( result );
-}
-
-
-//-------------------------------------------------------------------------------------------------
-int JetPlayer::loadFromFD(const int fd, const long long offset, const long long length)
-{
- ALOGV("JetPlayer::loadFromFD(): fd=%d offset=%lld length=%lld", fd, offset, length);
-
- Mutex::Autolock lock(mMutex);
-
- delete mIoWrapper;
- mIoWrapper = new MidiIoWrapper(fd, offset, length);
-
- EAS_RESULT result = JET_OpenFile(mEasData, mIoWrapper->getLocator());
- if (result != EAS_SUCCESS)
- mState = EAS_STATE_ERROR;
- else
- mState = EAS_STATE_OPEN;
- return( result );
-}
-
-
-//-------------------------------------------------------------------------------------------------
-int JetPlayer::closeFile()
-{
- Mutex::Autolock lock(mMutex);
- return JET_CloseFile(mEasData);
-}
-
-
-//-------------------------------------------------------------------------------------------------
-int JetPlayer::play()
-{
- ALOGV("JetPlayer::play(): entering");
- Mutex::Autolock lock(mMutex);
-
- EAS_RESULT result = JET_Play(mEasData);
-
- mPaused = false;
- mRender = true;
-
- JET_Status(mEasData, &mJetStatus);
- this->dumpJetStatus(&mJetStatus);
-
- fireUpdateOnStatusChange();
-
- // wake up render thread
- ALOGV("JetPlayer::play(): wakeup render thread");
- mCondition.signal();
-
- return result;
-}
-
-//-------------------------------------------------------------------------------------------------
-int JetPlayer::pause()
-{
- Mutex::Autolock lock(mMutex);
- mPaused = true;
- EAS_RESULT result = JET_Pause(mEasData);
-
- mRender = false;
-
- JET_Status(mEasData, &mJetStatus);
- this->dumpJetStatus(&mJetStatus);
- fireUpdateOnStatusChange();
-
-
- return result;
-}
-
-
-//-------------------------------------------------------------------------------------------------
-int JetPlayer::queueSegment(int segmentNum, int libNum, int repeatCount, int transpose,
- EAS_U32 muteFlags, EAS_U8 userID)
-{
- ALOGV("JetPlayer::queueSegment segmentNum=%d, libNum=%d, repeatCount=%d, transpose=%d",
- segmentNum, libNum, repeatCount, transpose);
- Mutex::Autolock lock(mMutex);
- return JET_QueueSegment(mEasData, segmentNum, libNum, repeatCount, transpose, muteFlags,
- userID);
-}
-
-//-------------------------------------------------------------------------------------------------
-int JetPlayer::setMuteFlags(EAS_U32 muteFlags, bool sync)
-{
- Mutex::Autolock lock(mMutex);
- return JET_SetMuteFlags(mEasData, muteFlags, sync);
-}
-
-//-------------------------------------------------------------------------------------------------
-int JetPlayer::setMuteFlag(int trackNum, bool muteFlag, bool sync)
-{
- Mutex::Autolock lock(mMutex);
- return JET_SetMuteFlag(mEasData, trackNum, muteFlag, sync);
-}
-
-//-------------------------------------------------------------------------------------------------
-int JetPlayer::triggerClip(int clipId)
-{
- ALOGV("JetPlayer::triggerClip clipId=%d", clipId);
- Mutex::Autolock lock(mMutex);
- return JET_TriggerClip(mEasData, clipId);
-}
-
-//-------------------------------------------------------------------------------------------------
-int JetPlayer::clearQueue()
-{
- ALOGV("JetPlayer::clearQueue");
- Mutex::Autolock lock(mMutex);
- return JET_Clear_Queue(mEasData);
-}
-
-//-------------------------------------------------------------------------------------------------
-void JetPlayer::dump()
-{
-}
-
-void JetPlayer::dumpJetStatus(S_JET_STATUS* pJetStatus)
-{
- if (pJetStatus!=NULL)
- ALOGV(">> current JET player status: userID=%d segmentRepeatCount=%d numQueuedSegments=%d "
- "paused=%d",
- pJetStatus->currentUserID, pJetStatus->segmentRepeatCount,
- pJetStatus->numQueuedSegments, pJetStatus->paused);
- else
- ALOGE(">> JET player status is NULL");
-}
-
-
-} // end namespace android
diff --git a/media/libmedia/MediaResource.cpp b/media/libmedia/MediaResource.cpp
index e636a50..8626009 100644
--- a/media/libmedia/MediaResource.cpp
+++ b/media/libmedia/MediaResource.cpp
@@ -19,6 +19,8 @@
#include <utils/Log.h>
#include <media/MediaResource.h>
+#include <vector>
+
namespace android {
MediaResource::MediaResource()
@@ -36,26 +38,48 @@
mSubType(subType),
mValue(value) {}
+MediaResource::MediaResource(Type type, const std::vector<uint8_t> &id, uint64_t value)
+ : mType(type),
+ mSubType(kUnspecifiedSubType),
+ mValue(value),
+ mId(id) {}
+
void MediaResource::readFromParcel(const Parcel &parcel) {
mType = static_cast<Type>(parcel.readInt32());
mSubType = static_cast<SubType>(parcel.readInt32());
mValue = parcel.readUint64();
+ parcel.readByteVector(&mId);
}
void MediaResource::writeToParcel(Parcel *parcel) const {
parcel->writeInt32(static_cast<int32_t>(mType));
parcel->writeInt32(static_cast<int32_t>(mSubType));
parcel->writeUint64(mValue);
+ parcel->writeByteVector(mId);
+}
+
+static String8 bytesToHexString(const std::vector<uint8_t> &bytes) {
+ String8 str;
+ for (auto &b : bytes) {
+ str.appendFormat("%02x", b);
+ }
+ return str;
}
String8 MediaResource::toString() const {
String8 str;
- str.appendFormat("%s/%s:%llu", asString(mType), asString(mSubType), (unsigned long long)mValue);
+ str.appendFormat("%s/%s:[%s]:%llu",
+ asString(mType), asString(mSubType),
+ bytesToHexString(mId).c_str(),
+ (unsigned long long)mValue);
return str;
}
bool MediaResource::operator==(const MediaResource &other) const {
- return (other.mType == mType) && (other.mSubType == mSubType) && (other.mValue == mValue);
+ return (other.mType == mType)
+ && (other.mSubType == mSubType)
+ && (other.mValue == mValue)
+ && (other.mId == mId);
}
bool MediaResource::operator!=(const MediaResource &other) const {
diff --git a/media/libmedia/MediaUtils.cpp b/media/libmedia/MediaUtils.cpp
index 31972fa..2efb30e 100644
--- a/media/libmedia/MediaUtils.cpp
+++ b/media/libmedia/MediaUtils.cpp
@@ -22,7 +22,7 @@
#include <sys/resource.h>
#include <unistd.h>
-#include <bionic_malloc.h>
+#include <bionic/malloc.h>
#include "MediaUtils.h"
diff --git a/media/libmedia/MidiDeviceInfo.cpp b/media/libmedia/MidiDeviceInfo.cpp
deleted file mode 100644
index 7588e00..0000000
--- a/media/libmedia/MidiDeviceInfo.cpp
+++ /dev/null
@@ -1,138 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#define LOG_TAG "MidiDeviceInfo"
-
-#include <media/MidiDeviceInfo.h>
-
-#include <binder/Parcel.h>
-#include <log/log.h>
-#include <utils/Errors.h>
-#include <utils/String16.h>
-
-namespace android {
-namespace media {
-namespace midi {
-
-// The constant values need to be kept in sync with MidiDeviceInfo.java.
-// static
-const char* const MidiDeviceInfo::PROPERTY_NAME = "name";
-const char* const MidiDeviceInfo::PROPERTY_MANUFACTURER = "manufacturer";
-const char* const MidiDeviceInfo::PROPERTY_PRODUCT = "product";
-const char* const MidiDeviceInfo::PROPERTY_VERSION = "version";
-const char* const MidiDeviceInfo::PROPERTY_SERIAL_NUMBER = "serial_number";
-const char* const MidiDeviceInfo::PROPERTY_ALSA_CARD = "alsa_card";
-const char* const MidiDeviceInfo::PROPERTY_ALSA_DEVICE = "alsa_device";
-
-String16 MidiDeviceInfo::getProperty(const char* propertyName) {
- String16 value;
- if (mProperties.getString(String16(propertyName), &value)) {
- return value;
- } else {
- return String16();
- }
-}
-
-#define RETURN_IF_FAILED(calledOnce) \
- { \
- status_t returnStatus = calledOnce; \
- if (returnStatus) { \
- ALOGE("Failed at %s:%d (%s)", __FILE__, __LINE__, __func__); \
- return returnStatus; \
- } \
- }
-
-status_t MidiDeviceInfo::writeToParcel(Parcel* parcel) const {
- // Needs to be kept in sync with code in MidiDeviceInfo.java
- RETURN_IF_FAILED(parcel->writeInt32(mType));
- RETURN_IF_FAILED(parcel->writeInt32(mId));
- RETURN_IF_FAILED(parcel->writeInt32((int32_t)mInputPortNames.size()));
- RETURN_IF_FAILED(parcel->writeInt32((int32_t)mOutputPortNames.size()));
- RETURN_IF_FAILED(writeStringVector(parcel, mInputPortNames));
- RETURN_IF_FAILED(writeStringVector(parcel, mOutputPortNames));
- RETURN_IF_FAILED(parcel->writeInt32(mIsPrivate ? 1 : 0));
- RETURN_IF_FAILED(mProperties.writeToParcel(parcel));
- // This corresponds to "extra" properties written by Java code
- RETURN_IF_FAILED(mProperties.writeToParcel(parcel));
- return OK;
-}
-
-status_t MidiDeviceInfo::readFromParcel(const Parcel* parcel) {
- // Needs to be kept in sync with code in MidiDeviceInfo.java
- RETURN_IF_FAILED(parcel->readInt32(&mType));
- RETURN_IF_FAILED(parcel->readInt32(&mId));
- int32_t inputPortCount;
- RETURN_IF_FAILED(parcel->readInt32(&inputPortCount));
- int32_t outputPortCount;
- RETURN_IF_FAILED(parcel->readInt32(&outputPortCount));
- RETURN_IF_FAILED(readStringVector(parcel, &mInputPortNames, inputPortCount));
- RETURN_IF_FAILED(readStringVector(parcel, &mOutputPortNames, outputPortCount));
- int32_t isPrivate;
- RETURN_IF_FAILED(parcel->readInt32(&isPrivate));
- mIsPrivate = isPrivate == 1;
- RETURN_IF_FAILED(mProperties.readFromParcel(parcel));
- // Ignore "extra" properties as they may contain Java Parcelables
- return OK;
-}
-
-status_t MidiDeviceInfo::readStringVector(
- const Parcel* parcel, Vector<String16> *vectorPtr, size_t defaultLength) {
- std::unique_ptr<std::vector<std::unique_ptr<String16>>> v;
- status_t result = parcel->readString16Vector(&v);
- if (result != OK) return result;
- vectorPtr->clear();
- if (v.get() != nullptr) {
- for (const auto& iter : *v) {
- if (iter.get() != nullptr) {
- vectorPtr->push_back(*iter);
- } else {
- vectorPtr->push_back(String16());
- }
- }
- } else {
- vectorPtr->resize(defaultLength);
- }
- return OK;
-}
-
-status_t MidiDeviceInfo::writeStringVector(Parcel* parcel, const Vector<String16>& vector) const {
- std::vector<String16> v;
- for (size_t i = 0; i < vector.size(); ++i) {
- v.push_back(vector[i]);
- }
- return parcel->writeString16Vector(v);
-}
-
-// Vector does not define operator==
-static inline bool areVectorsEqual(const Vector<String16>& lhs, const Vector<String16>& rhs) {
- if (lhs.size() != rhs.size()) return false;
- for (size_t i = 0; i < lhs.size(); ++i) {
- if (lhs[i] != rhs[i]) return false;
- }
- return true;
-}
-
-bool operator==(const MidiDeviceInfo& lhs, const MidiDeviceInfo& rhs) {
- return (lhs.mType == rhs.mType && lhs.mId == rhs.mId &&
- areVectorsEqual(lhs.mInputPortNames, rhs.mInputPortNames) &&
- areVectorsEqual(lhs.mOutputPortNames, rhs.mOutputPortNames) &&
- lhs.mProperties == rhs.mProperties &&
- lhs.mIsPrivate == rhs.mIsPrivate);
-}
-
-} // namespace midi
-} // namespace media
-} // namespace android
diff --git a/media/libmedia/MidiIoWrapper.cpp b/media/libmedia/MidiIoWrapper.cpp
index d8ef9cf..e71ea2c 100644
--- a/media/libmedia/MidiIoWrapper.cpp
+++ b/media/libmedia/MidiIoWrapper.cpp
@@ -17,7 +17,6 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "MidiIoWrapper"
#include <utils/Log.h>
-#include <utils/RefBase.h>
#include <sys/stat.h>
#include <fcntl.h>
@@ -50,7 +49,7 @@
mDataSource = nullptr;
}
-class DataSourceUnwrapper : public DataSourceBase {
+class MidiIoWrapper::DataSourceUnwrapper {
public:
explicit DataSourceUnwrapper(CDataSource *csource) {
diff --git a/media/libmedia/NdkWrapper.cpp b/media/libmedia/NdkWrapper.cpp
deleted file mode 100644
index c150407..0000000
--- a/media/libmedia/NdkWrapper.cpp
+++ /dev/null
@@ -1,1290 +0,0 @@
-/*
- * Copyright 2017, The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "NdkWrapper"
-
-#include <media/NdkWrapper.h>
-
-#include <android/native_window.h>
-#include <log/log.h>
-#include <media/NdkMediaCodec.h>
-#include <media/NdkMediaCrypto.h>
-#include <media/NdkMediaDrm.h>
-#include <media/NdkMediaFormat.h>
-#include <media/NdkMediaExtractor.h>
-#include <media/stagefright/MetaData.h>
-#include <media/stagefright/foundation/ABuffer.h>
-#include <media/stagefright/foundation/AMessage.h>
-#include <utils/Errors.h>
-
-#include "NdkMediaDataSourceCallbacksPriv.h"
-
-namespace android {
-
-static const size_t kAESBlockSize = 16; // AES_BLOCK_SIZE
-
-static const char *AMediaFormatKeyGroupInt32[] = {
- AMEDIAFORMAT_KEY_AAC_DRC_ATTENUATION_FACTOR,
- AMEDIAFORMAT_KEY_AAC_DRC_BOOST_FACTOR,
- AMEDIAFORMAT_KEY_AAC_DRC_HEAVY_COMPRESSION,
- AMEDIAFORMAT_KEY_AAC_DRC_TARGET_REFERENCE_LEVEL,
- AMEDIAFORMAT_KEY_AAC_ENCODED_TARGET_LEVEL,
- AMEDIAFORMAT_KEY_AAC_MAX_OUTPUT_CHANNEL_COUNT,
- AMEDIAFORMAT_KEY_AAC_PROFILE,
- AMEDIAFORMAT_KEY_AAC_SBR_MODE,
- AMEDIAFORMAT_KEY_AUDIO_SESSION_ID,
- AMEDIAFORMAT_KEY_BITRATE_MODE,
- AMEDIAFORMAT_KEY_BIT_RATE,
- AMEDIAFORMAT_KEY_CAPTURE_RATE,
- AMEDIAFORMAT_KEY_CHANNEL_COUNT,
- AMEDIAFORMAT_KEY_CHANNEL_MASK,
- AMEDIAFORMAT_KEY_COLOR_FORMAT,
- AMEDIAFORMAT_KEY_COLOR_RANGE,
- AMEDIAFORMAT_KEY_COLOR_STANDARD,
- AMEDIAFORMAT_KEY_COLOR_TRANSFER,
- AMEDIAFORMAT_KEY_COMPLEXITY,
- AMEDIAFORMAT_KEY_CREATE_INPUT_SURFACE_SUSPENDED,
- AMEDIAFORMAT_KEY_CRYPTO_DEFAULT_IV_SIZE,
- AMEDIAFORMAT_KEY_CRYPTO_ENCRYPTED_BYTE_BLOCK,
- AMEDIAFORMAT_KEY_CRYPTO_MODE,
- AMEDIAFORMAT_KEY_CRYPTO_SKIP_BYTE_BLOCK,
- AMEDIAFORMAT_KEY_FLAC_COMPRESSION_LEVEL,
- AMEDIAFORMAT_KEY_GRID_COLUMNS,
- AMEDIAFORMAT_KEY_GRID_ROWS,
- AMEDIAFORMAT_KEY_HAPTIC_CHANNEL_COUNT,
- AMEDIAFORMAT_KEY_HEIGHT,
- AMEDIAFORMAT_KEY_INTRA_REFRESH_PERIOD,
- AMEDIAFORMAT_KEY_IS_ADTS,
- AMEDIAFORMAT_KEY_IS_AUTOSELECT,
- AMEDIAFORMAT_KEY_IS_DEFAULT,
- AMEDIAFORMAT_KEY_IS_FORCED_SUBTITLE,
- AMEDIAFORMAT_KEY_LATENCY,
- AMEDIAFORMAT_KEY_LEVEL,
- AMEDIAFORMAT_KEY_MAX_HEIGHT,
- AMEDIAFORMAT_KEY_MAX_INPUT_SIZE,
- AMEDIAFORMAT_KEY_MAX_WIDTH,
- AMEDIAFORMAT_KEY_PCM_ENCODING,
- AMEDIAFORMAT_KEY_PRIORITY,
- AMEDIAFORMAT_KEY_PROFILE,
- AMEDIAFORMAT_KEY_PUSH_BLANK_BUFFERS_ON_STOP,
- AMEDIAFORMAT_KEY_ROTATION,
- AMEDIAFORMAT_KEY_SAMPLE_RATE,
- AMEDIAFORMAT_KEY_SLICE_HEIGHT,
- AMEDIAFORMAT_KEY_STRIDE,
- AMEDIAFORMAT_KEY_TRACK_ID,
- AMEDIAFORMAT_KEY_WIDTH,
- AMEDIAFORMAT_KEY_DISPLAY_HEIGHT,
- AMEDIAFORMAT_KEY_DISPLAY_WIDTH,
- AMEDIAFORMAT_KEY_TEMPORAL_LAYER_ID,
- AMEDIAFORMAT_KEY_TILE_HEIGHT,
- AMEDIAFORMAT_KEY_TILE_WIDTH,
- AMEDIAFORMAT_KEY_TRACK_INDEX,
-};
-
-static const char *AMediaFormatKeyGroupInt64[] = {
- AMEDIAFORMAT_KEY_DURATION,
- AMEDIAFORMAT_KEY_MAX_PTS_GAP_TO_ENCODER,
- AMEDIAFORMAT_KEY_REPEAT_PREVIOUS_FRAME_AFTER,
- AMEDIAFORMAT_KEY_TIME_US,
-};
-
-static const char *AMediaFormatKeyGroupString[] = {
- AMEDIAFORMAT_KEY_LANGUAGE,
- AMEDIAFORMAT_KEY_MIME,
- AMEDIAFORMAT_KEY_TEMPORAL_LAYERING,
-};
-
-static const char *AMediaFormatKeyGroupBuffer[] = {
- AMEDIAFORMAT_KEY_CRYPTO_IV,
- AMEDIAFORMAT_KEY_CRYPTO_KEY,
- AMEDIAFORMAT_KEY_HDR_STATIC_INFO,
- AMEDIAFORMAT_KEY_SEI,
- AMEDIAFORMAT_KEY_MPEG_USER_DATA,
-};
-
-static const char *AMediaFormatKeyGroupCsd[] = {
- AMEDIAFORMAT_KEY_CSD_0,
- AMEDIAFORMAT_KEY_CSD_1,
- AMEDIAFORMAT_KEY_CSD_2,
-};
-
-static const char *AMediaFormatKeyGroupRect[] = {
- AMEDIAFORMAT_KEY_DISPLAY_CROP,
-};
-
-static const char *AMediaFormatKeyGroupFloatInt32[] = {
- AMEDIAFORMAT_KEY_FRAME_RATE,
- AMEDIAFORMAT_KEY_I_FRAME_INTERVAL,
- AMEDIAFORMAT_KEY_MAX_FPS_TO_ENCODER,
- AMEDIAFORMAT_KEY_OPERATING_RATE,
-};
-
-static status_t translateErrorCode(media_status_t err) {
- if (err == AMEDIA_OK) {
- return OK;
- } else if (err == AMEDIA_ERROR_END_OF_STREAM) {
- return ERROR_END_OF_STREAM;
- } else if (err == AMEDIA_ERROR_IO) {
- return ERROR_IO;
- } else if (err == AMEDIACODEC_INFO_TRY_AGAIN_LATER) {
- return -EAGAIN;
- }
-
- ALOGE("ndk error code: %d", err);
- return UNKNOWN_ERROR;
-}
-
-static int32_t translateActionCode(int32_t actionCode) {
- if (AMediaCodecActionCode_isTransient(actionCode)) {
- return ACTION_CODE_TRANSIENT;
- } else if (AMediaCodecActionCode_isRecoverable(actionCode)) {
- return ACTION_CODE_RECOVERABLE;
- }
- return ACTION_CODE_FATAL;
-}
-
-static CryptoPlugin::Mode translateToCryptoPluginMode(cryptoinfo_mode_t mode) {
- CryptoPlugin::Mode ret = CryptoPlugin::kMode_Unencrypted;
- switch (mode) {
- case AMEDIACODECRYPTOINFO_MODE_AES_CTR: {
- ret = CryptoPlugin::kMode_AES_CTR;
- break;
- }
-
- case AMEDIACODECRYPTOINFO_MODE_AES_WV: {
- ret = CryptoPlugin::kMode_AES_WV;
- break;
- }
-
- case AMEDIACODECRYPTOINFO_MODE_AES_CBC: {
- ret = CryptoPlugin::kMode_AES_CBC;
- break;
- }
-
- default:
- break;
- }
-
- return ret;
-}
-
-static cryptoinfo_mode_t translateToCryptoInfoMode(CryptoPlugin::Mode mode) {
- cryptoinfo_mode_t ret = AMEDIACODECRYPTOINFO_MODE_CLEAR;
- switch (mode) {
- case CryptoPlugin::kMode_AES_CTR: {
- ret = AMEDIACODECRYPTOINFO_MODE_AES_CTR;
- break;
- }
-
- case CryptoPlugin::kMode_AES_WV: {
- ret = AMEDIACODECRYPTOINFO_MODE_AES_WV;
- break;
- }
-
- case CryptoPlugin::kMode_AES_CBC: {
- ret = AMEDIACODECRYPTOINFO_MODE_AES_CBC;
- break;
- }
-
- default:
- break;
- }
-
- return ret;
-}
-
-//////////// AMediaFormatWrapper
-// static
-sp<AMediaFormatWrapper> AMediaFormatWrapper::Create(const sp<AMessage> &message) {
- sp<AMediaFormatWrapper> aMediaFormat = new AMediaFormatWrapper();
-
- for (size_t i = 0; i < message->countEntries(); ++i) {
- AMessage::Type valueType;
- const char *key = message->getEntryNameAt(i, &valueType);
-
- switch (valueType) {
- case AMessage::kTypeInt32: {
- int32_t val;
- if (!message->findInt32(key, &val)) {
- ALOGE("AMediaFormatWrapper::Create: error at item %zu", i);
- continue;
- }
- aMediaFormat->setInt32(key, val);
- break;
- }
-
- case AMessage::kTypeInt64: {
- int64_t val;
- if (!message->findInt64(key, &val)) {
- ALOGE("AMediaFormatWrapper::Create: error at item %zu", i);
- continue;
- }
- aMediaFormat->setInt64(key, val);
- break;
- }
-
- case AMessage::kTypeFloat: {
- float val;
- if (!message->findFloat(key, &val)) {
- ALOGE("AMediaFormatWrapper::Create: error at item %zu", i);
- continue;
- }
- aMediaFormat->setFloat(key, val);
- break;
- }
-
- case AMessage::kTypeDouble: {
- double val;
- if (!message->findDouble(key, &val)) {
- ALOGE("AMediaFormatWrapper::Create: error at item %zu", i);
- continue;
- }
- aMediaFormat->setDouble(key, val);
- break;
- }
-
- case AMessage::kTypeSize: {
- size_t val;
- if (!message->findSize(key, &val)) {
- ALOGE("AMediaFormatWrapper::Create: error at item %zu", i);
- continue;
- }
- aMediaFormat->setSize(key, val);
- break;
- }
-
- case AMessage::kTypeRect: {
- int32_t left, top, right, bottom;
- if (!message->findRect(key, &left, &top, &right, &bottom)) {
- ALOGE("AMediaFormatWrapper::Create: error at item %zu", i);
- continue;
- }
- aMediaFormat->setRect(key, left, top, right, bottom);
- break;
- }
-
- case AMessage::kTypeString: {
- AString val;
- if (!message->findString(key, &val)) {
- ALOGE("AMediaFormatWrapper::Create: error at item %zu", i);
- continue;
- }
- aMediaFormat->setString(key, val);
- break;
- }
-
- case AMessage::kTypeBuffer: {
- sp<ABuffer> val;
- if (!message->findBuffer(key, &val)) {
- ALOGE("AMediaFormatWrapper::Create: error at item %zu", i);
- continue;
- }
- aMediaFormat->setBuffer(key, val->data(), val->size());
- break;
- }
-
- default: {
- break;
- }
- }
- }
-
- return aMediaFormat;
-}
-
-AMediaFormatWrapper::AMediaFormatWrapper() {
- mAMediaFormat = AMediaFormat_new();
-}
-
-AMediaFormatWrapper::AMediaFormatWrapper(AMediaFormat *aMediaFormat)
- : mAMediaFormat(aMediaFormat) {
-}
-
-AMediaFormatWrapper::~AMediaFormatWrapper() {
- release();
-}
-
-status_t AMediaFormatWrapper::release() {
- if (mAMediaFormat != NULL) {
- media_status_t err = AMediaFormat_delete(mAMediaFormat);
- mAMediaFormat = NULL;
- return translateErrorCode(err);
- }
- return OK;
-}
-
-AMediaFormat *AMediaFormatWrapper::getAMediaFormat() const {
- return mAMediaFormat;
-}
-
-sp<AMessage> AMediaFormatWrapper::toAMessage() const {
- sp<AMessage> msg;
- writeToAMessage(msg);
- return msg;
-}
-
-void AMediaFormatWrapper::writeToAMessage(sp<AMessage> &msg) const {
- if (mAMediaFormat == NULL) {
- msg = NULL;
- }
-
- if (msg == NULL) {
- msg = new AMessage;
- }
- for (auto& key : AMediaFormatKeyGroupInt32) {
- int32_t val;
- if (getInt32(key, &val)) {
- msg->setInt32(key, val);
- }
- }
- for (auto& key : AMediaFormatKeyGroupInt64) {
- int64_t val;
- if (getInt64(key, &val)) {
- msg->setInt64(key, val);
- }
- }
- for (auto& key : AMediaFormatKeyGroupString) {
- AString val;
- if (getString(key, &val)) {
- msg->setString(key, val);
- }
- }
- for (auto& key : AMediaFormatKeyGroupBuffer) {
- void *data;
- size_t size;
- if (getBuffer(key, &data, &size)) {
- sp<ABuffer> buffer = ABuffer::CreateAsCopy(data, size);
- msg->setBuffer(key, buffer);
- }
- }
- for (auto& key : AMediaFormatKeyGroupCsd) {
- void *data;
- size_t size;
- if (getBuffer(key, &data, &size)) {
- sp<ABuffer> buffer = ABuffer::CreateAsCopy(data, size);
- buffer->meta()->setInt32(AMEDIAFORMAT_KEY_CSD, 1);
- buffer->meta()->setInt64(AMEDIAFORMAT_KEY_TIME_US, 0);
- msg->setBuffer(key, buffer);
- }
- }
- for (auto& key : AMediaFormatKeyGroupRect) {
- int32_t left, top, right, bottom;
- if (getRect(key, &left, &top, &right, &bottom)) {
- msg->setRect(key, left, top, right, bottom);
- }
- }
- for (auto& key : AMediaFormatKeyGroupFloatInt32) {
- float valFloat;
- if (getFloat(key, &valFloat)) {
- msg->setFloat(key, valFloat);
- } else {
- int32_t valInt32;
- if (getInt32(key, &valInt32)) {
- msg->setFloat(key, (float)valInt32);
- }
- }
- }
-}
-
-const char* AMediaFormatWrapper::toString() const {
- if (mAMediaFormat == NULL) {
- return NULL;
- }
- return AMediaFormat_toString(mAMediaFormat);
-}
-
-bool AMediaFormatWrapper::getInt32(const char *name, int32_t *out) const {
- if (mAMediaFormat == NULL) {
- return false;
- }
- return AMediaFormat_getInt32(mAMediaFormat, name, out);
-}
-
-bool AMediaFormatWrapper::getInt64(const char *name, int64_t *out) const {
- if (mAMediaFormat == NULL) {
- return false;
- }
- return AMediaFormat_getInt64(mAMediaFormat, name, out);
-}
-
-bool AMediaFormatWrapper::getFloat(const char *name, float *out) const {
- if (mAMediaFormat == NULL) {
- return false;
- }
- return AMediaFormat_getFloat(mAMediaFormat, name, out);
-}
-
-bool AMediaFormatWrapper::getDouble(const char *name, double *out) const {
- if (mAMediaFormat == NULL) {
- return false;
- }
- return AMediaFormat_getDouble(mAMediaFormat, name, out);
-}
-
-bool AMediaFormatWrapper::getSize(const char *name, size_t *out) const {
- if (mAMediaFormat == NULL) {
- return false;
- }
- return AMediaFormat_getSize(mAMediaFormat, name, out);
-}
-
-bool AMediaFormatWrapper::getRect(
- const char *name, int32_t *left, int32_t *top, int32_t *right, int32_t *bottom) const {
- if (mAMediaFormat == NULL) {
- return false;
- }
- return AMediaFormat_getRect(mAMediaFormat, name, left, top, right, bottom);
-}
-
-bool AMediaFormatWrapper::getBuffer(const char *name, void** data, size_t *outSize) const {
- if (mAMediaFormat == NULL) {
- return false;
- }
- return AMediaFormat_getBuffer(mAMediaFormat, name, data, outSize);
-}
-
-bool AMediaFormatWrapper::getString(const char *name, AString *out) const {
- if (mAMediaFormat == NULL) {
- return false;
- }
- const char *outChar = NULL;
- bool ret = AMediaFormat_getString(mAMediaFormat, name, &outChar);
- if (ret) {
- *out = AString(outChar);
- }
- return ret;
-}
-
-void AMediaFormatWrapper::setInt32(const char* name, int32_t value) {
- if (mAMediaFormat != NULL) {
- AMediaFormat_setInt32(mAMediaFormat, name, value);
- }
-}
-
-void AMediaFormatWrapper::setInt64(const char* name, int64_t value) {
- if (mAMediaFormat != NULL) {
- AMediaFormat_setInt64(mAMediaFormat, name, value);
- }
-}
-
-void AMediaFormatWrapper::setFloat(const char* name, float value) {
- if (mAMediaFormat != NULL) {
- AMediaFormat_setFloat(mAMediaFormat, name, value);
- }
-}
-
-void AMediaFormatWrapper::setDouble(const char* name, double value) {
- if (mAMediaFormat != NULL) {
- AMediaFormat_setDouble(mAMediaFormat, name, value);
- }
-}
-
-void AMediaFormatWrapper::setSize(const char* name, size_t value) {
- if (mAMediaFormat != NULL) {
- AMediaFormat_setSize(mAMediaFormat, name, value);
- }
-}
-
-void AMediaFormatWrapper::setRect(
- const char* name, int32_t left, int32_t top, int32_t right, int32_t bottom) {
- if (mAMediaFormat != NULL) {
- AMediaFormat_setRect(mAMediaFormat, name, left, top, right, bottom);
- }
-}
-
-void AMediaFormatWrapper::setString(const char* name, const AString &value) {
- if (mAMediaFormat != NULL) {
- AMediaFormat_setString(mAMediaFormat, name, value.c_str());
- }
-}
-
-void AMediaFormatWrapper::setBuffer(const char* name, void* data, size_t size) {
- if (mAMediaFormat != NULL) {
- AMediaFormat_setBuffer(mAMediaFormat, name, data, size);
- }
-}
-
-
-//////////// ANativeWindowWrapper
-ANativeWindowWrapper::ANativeWindowWrapper(ANativeWindow *aNativeWindow)
- : mANativeWindow(aNativeWindow) {
- if (aNativeWindow != NULL) {
- ANativeWindow_acquire(aNativeWindow);
- }
-}
-
-ANativeWindowWrapper::~ANativeWindowWrapper() {
- release();
-}
-
-status_t ANativeWindowWrapper::release() {
- if (mANativeWindow != NULL) {
- ANativeWindow_release(mANativeWindow);
- mANativeWindow = NULL;
- }
- return OK;
-}
-
-ANativeWindow *ANativeWindowWrapper::getANativeWindow() const {
- return mANativeWindow;
-}
-
-
-//////////// AMediaDrmWrapper
-AMediaDrmWrapper::AMediaDrmWrapper(const uint8_t uuid[16]) {
- mAMediaDrm = AMediaDrm_createByUUID(uuid);
-}
-
-AMediaDrmWrapper::AMediaDrmWrapper(AMediaDrm *aMediaDrm)
- : mAMediaDrm(aMediaDrm) {
-}
-
-AMediaDrmWrapper::~AMediaDrmWrapper() {
- release();
-}
-
-status_t AMediaDrmWrapper::release() {
- if (mAMediaDrm != NULL) {
- AMediaDrm_release(mAMediaDrm);
- mAMediaDrm = NULL;
- }
- return OK;
-}
-
-AMediaDrm *AMediaDrmWrapper::getAMediaDrm() const {
- return mAMediaDrm;
-}
-
-// static
-bool AMediaDrmWrapper::isCryptoSchemeSupported(
- const uint8_t uuid[16],
- const char *mimeType) {
- return AMediaDrm_isCryptoSchemeSupported(uuid, mimeType);
-}
-
-
-//////////// AMediaCryptoWrapper
-AMediaCryptoWrapper::AMediaCryptoWrapper(
- const uint8_t uuid[16], const void *initData, size_t initDataSize) {
- mAMediaCrypto = AMediaCrypto_new(uuid, initData, initDataSize);
-}
-
-AMediaCryptoWrapper::AMediaCryptoWrapper(AMediaCrypto *aMediaCrypto)
- : mAMediaCrypto(aMediaCrypto) {
-}
-
-AMediaCryptoWrapper::~AMediaCryptoWrapper() {
- release();
-}
-
-status_t AMediaCryptoWrapper::release() {
- if (mAMediaCrypto != NULL) {
- AMediaCrypto_delete(mAMediaCrypto);
- mAMediaCrypto = NULL;
- }
- return OK;
-}
-
-AMediaCrypto *AMediaCryptoWrapper::getAMediaCrypto() const {
- return mAMediaCrypto;
-}
-
-bool AMediaCryptoWrapper::isCryptoSchemeSupported(const uint8_t uuid[16]) {
- if (mAMediaCrypto == NULL) {
- return false;
- }
- return AMediaCrypto_isCryptoSchemeSupported(uuid);
-}
-
-bool AMediaCryptoWrapper::requiresSecureDecoderComponent(const char *mime) {
- if (mAMediaCrypto == NULL) {
- return false;
- }
- return AMediaCrypto_requiresSecureDecoderComponent(mime);
-}
-
-
-//////////// AMediaCodecCryptoInfoWrapper
-// static
-sp<AMediaCodecCryptoInfoWrapper> AMediaCodecCryptoInfoWrapper::Create(MetaDataBase &meta) {
-
- uint32_t type;
- const void *crypteddata;
- size_t cryptedsize;
-
- if (!meta.findData(kKeyEncryptedSizes, &type, &crypteddata, &cryptedsize)) {
- return NULL;
- }
-
- int numSubSamples = cryptedsize / sizeof(size_t);
-
- if (numSubSamples <= 0) {
- ALOGE("Create: INVALID numSubSamples: %d", numSubSamples);
- return NULL;
- }
-
- const void *cleardata;
- size_t clearsize;
- if (meta.findData(kKeyPlainSizes, &type, &cleardata, &clearsize)) {
- if (clearsize != cryptedsize) {
- // The two must be of the same length.
- ALOGE("Create: mismatch cryptedsize: %zu != clearsize: %zu", cryptedsize, clearsize);
- return NULL;
- }
- }
-
- const void *key;
- size_t keysize;
- if (meta.findData(kKeyCryptoKey, &type, &key, &keysize)) {
- if (keysize != kAESBlockSize) {
- // Keys must be 16 bytes in length.
- ALOGE("Create: Keys must be %zu bytes in length: %zu", kAESBlockSize, keysize);
- return NULL;
- }
- }
-
- const void *iv;
- size_t ivsize;
- if (meta.findData(kKeyCryptoIV, &type, &iv, &ivsize)) {
- if (ivsize != kAESBlockSize) {
- // IVs must be 16 bytes in length.
- ALOGE("Create: IV must be %zu bytes in length: %zu", kAESBlockSize, ivsize);
- return NULL;
- }
- }
-
- int32_t mode;
- if (!meta.findInt32(kKeyCryptoMode, &mode)) {
- mode = CryptoPlugin::kMode_AES_CTR;
- }
-
- return new AMediaCodecCryptoInfoWrapper(
- numSubSamples,
- (uint8_t*) key,
- (uint8_t*) iv,
- (CryptoPlugin::Mode)mode,
- (size_t*) cleardata,
- (size_t*) crypteddata);
-}
-
-AMediaCodecCryptoInfoWrapper::AMediaCodecCryptoInfoWrapper(
- int numsubsamples,
- uint8_t key[16],
- uint8_t iv[16],
- CryptoPlugin::Mode mode,
- size_t *clearbytes,
- size_t *encryptedbytes) {
- mAMediaCodecCryptoInfo =
- AMediaCodecCryptoInfo_new(numsubsamples,
- key,
- iv,
- translateToCryptoInfoMode(mode),
- clearbytes,
- encryptedbytes);
-}
-
-AMediaCodecCryptoInfoWrapper::AMediaCodecCryptoInfoWrapper(
- AMediaCodecCryptoInfo *aMediaCodecCryptoInfo)
- : mAMediaCodecCryptoInfo(aMediaCodecCryptoInfo) {
-}
-
-AMediaCodecCryptoInfoWrapper::~AMediaCodecCryptoInfoWrapper() {
- release();
-}
-
-status_t AMediaCodecCryptoInfoWrapper::release() {
- if (mAMediaCodecCryptoInfo != NULL) {
- media_status_t err = AMediaCodecCryptoInfo_delete(mAMediaCodecCryptoInfo);
- mAMediaCodecCryptoInfo = NULL;
- return translateErrorCode(err);
- }
- return OK;
-}
-
-AMediaCodecCryptoInfo *AMediaCodecCryptoInfoWrapper::getAMediaCodecCryptoInfo() const {
- return mAMediaCodecCryptoInfo;
-}
-
-void AMediaCodecCryptoInfoWrapper::setPattern(CryptoPlugin::Pattern *pattern) {
- if (mAMediaCodecCryptoInfo == NULL || pattern == NULL) {
- return;
- }
- cryptoinfo_pattern_t ndkPattern = {(int32_t)pattern->mEncryptBlocks,
- (int32_t)pattern->mSkipBlocks };
- return AMediaCodecCryptoInfo_setPattern(mAMediaCodecCryptoInfo, &ndkPattern);
-}
-
-size_t AMediaCodecCryptoInfoWrapper::getNumSubSamples() {
- if (mAMediaCodecCryptoInfo == NULL) {
- return 0;
- }
- return AMediaCodecCryptoInfo_getNumSubSamples(mAMediaCodecCryptoInfo);
-}
-
-status_t AMediaCodecCryptoInfoWrapper::getKey(uint8_t *dst) {
- if (mAMediaCodecCryptoInfo == NULL) {
- return DEAD_OBJECT;
- }
- if (dst == NULL) {
- return BAD_VALUE;
- }
- return translateErrorCode(
- AMediaCodecCryptoInfo_getKey(mAMediaCodecCryptoInfo, dst));
-}
-
-status_t AMediaCodecCryptoInfoWrapper::getIV(uint8_t *dst) {
- if (mAMediaCodecCryptoInfo == NULL) {
- return DEAD_OBJECT;
- }
- if (dst == NULL) {
- return BAD_VALUE;
- }
- return translateErrorCode(
- AMediaCodecCryptoInfo_getIV(mAMediaCodecCryptoInfo, dst));
-}
-
-CryptoPlugin::Mode AMediaCodecCryptoInfoWrapper::getMode() {
- if (mAMediaCodecCryptoInfo == NULL) {
- return CryptoPlugin::kMode_Unencrypted;
- }
- return translateToCryptoPluginMode(
- AMediaCodecCryptoInfo_getMode(mAMediaCodecCryptoInfo));
-}
-
-status_t AMediaCodecCryptoInfoWrapper::getClearBytes(size_t *dst) {
- if (mAMediaCodecCryptoInfo == NULL) {
- return DEAD_OBJECT;
- }
- if (dst == NULL) {
- return BAD_VALUE;
- }
- return translateErrorCode(
- AMediaCodecCryptoInfo_getClearBytes(mAMediaCodecCryptoInfo, dst));
-}
-
-status_t AMediaCodecCryptoInfoWrapper::getEncryptedBytes(size_t *dst) {
- if (mAMediaCodecCryptoInfo == NULL) {
- return DEAD_OBJECT;
- }
- if (dst == NULL) {
- return BAD_VALUE;
- }
- return translateErrorCode(
- AMediaCodecCryptoInfo_getEncryptedBytes(mAMediaCodecCryptoInfo, dst));
-}
-
-
-//////////// AMediaCodecWrapper
-// static
-sp<AMediaCodecWrapper> AMediaCodecWrapper::CreateCodecByName(const AString &name) {
- AMediaCodec *aMediaCodec = AMediaCodec_createCodecByName(name.c_str());
- return new AMediaCodecWrapper(aMediaCodec);
-}
-
-// static
-sp<AMediaCodecWrapper> AMediaCodecWrapper::CreateDecoderByType(const AString &mimeType) {
- AMediaCodec *aMediaCodec = AMediaCodec_createDecoderByType(mimeType.c_str());
- return new AMediaCodecWrapper(aMediaCodec);
-}
-
-// static
-void AMediaCodecWrapper::OnInputAvailableCB(
- AMediaCodec * /* aMediaCodec */,
- void *userdata,
- int32_t index) {
- ALOGV("OnInputAvailableCB: index(%d)", index);
- sp<AMessage> msg = sp<AMessage>((AMessage *)userdata)->dup();
- msg->setInt32("callbackID", CB_INPUT_AVAILABLE);
- msg->setInt32("index", index);
- msg->post();
-}
-
-// static
-void AMediaCodecWrapper::OnOutputAvailableCB(
- AMediaCodec * /* aMediaCodec */,
- void *userdata,
- int32_t index,
- AMediaCodecBufferInfo *bufferInfo) {
- ALOGV("OnOutputAvailableCB: index(%d), (%d, %d, %lld, 0x%x)",
- index, bufferInfo->offset, bufferInfo->size,
- (long long)bufferInfo->presentationTimeUs, bufferInfo->flags);
- sp<AMessage> msg = sp<AMessage>((AMessage *)userdata)->dup();
- msg->setInt32("callbackID", CB_OUTPUT_AVAILABLE);
- msg->setInt32("index", index);
- msg->setSize("offset", (size_t)(bufferInfo->offset));
- msg->setSize("size", (size_t)(bufferInfo->size));
- msg->setInt64("timeUs", bufferInfo->presentationTimeUs);
- msg->setInt32("flags", (int32_t)(bufferInfo->flags));
- msg->post();
-}
-
-// static
-void AMediaCodecWrapper::OnFormatChangedCB(
- AMediaCodec * /* aMediaCodec */,
- void *userdata,
- AMediaFormat *format) {
- sp<AMediaFormatWrapper> formatWrapper = new AMediaFormatWrapper(format);
- sp<AMessage> outputFormat = formatWrapper->toAMessage();
- ALOGV("OnFormatChangedCB: format(%s)", outputFormat->debugString().c_str());
-
- sp<AMessage> msg = sp<AMessage>((AMessage *)userdata)->dup();
- msg->setInt32("callbackID", CB_OUTPUT_FORMAT_CHANGED);
- msg->setMessage("format", outputFormat);
- msg->post();
-}
-
-// static
-void AMediaCodecWrapper::OnErrorCB(
- AMediaCodec * /* aMediaCodec */,
- void *userdata,
- media_status_t err,
- int32_t actionCode,
- const char *detail) {
- ALOGV("OnErrorCB: err(%d), actionCode(%d), detail(%s)", err, actionCode, detail);
- sp<AMessage> msg = sp<AMessage>((AMessage *)userdata)->dup();
- msg->setInt32("callbackID", CB_ERROR);
- msg->setInt32("err", translateErrorCode(err));
- msg->setInt32("actionCode", translateActionCode(actionCode));
- msg->setString("detail", detail);
- msg->post();
-}
-
-AMediaCodecWrapper::AMediaCodecWrapper(AMediaCodec *aMediaCodec)
- : mAMediaCodec(aMediaCodec) {
-}
-
-AMediaCodecWrapper::~AMediaCodecWrapper() {
- release();
-}
-
-status_t AMediaCodecWrapper::release() {
- if (mAMediaCodec != NULL) {
- AMediaCodecOnAsyncNotifyCallback aCB = {};
- AMediaCodec_setAsyncNotifyCallback(mAMediaCodec, aCB, NULL);
- mCallback = NULL;
-
- media_status_t err = AMediaCodec_delete(mAMediaCodec);
- mAMediaCodec = NULL;
- return translateErrorCode(err);
- }
- return OK;
-}
-
-AMediaCodec *AMediaCodecWrapper::getAMediaCodec() const {
- return mAMediaCodec;
-}
-
-status_t AMediaCodecWrapper::getName(AString *outComponentName) const {
- if (mAMediaCodec == NULL) {
- return DEAD_OBJECT;
- }
- char *name = NULL;
- media_status_t err = AMediaCodec_getName(mAMediaCodec, &name);
- if (err != AMEDIA_OK) {
- return translateErrorCode(err);
- }
-
- *outComponentName = AString(name);
- AMediaCodec_releaseName(mAMediaCodec, name);
- return OK;
-}
-
-status_t AMediaCodecWrapper::configure(
- const sp<AMediaFormatWrapper> &format,
- const sp<ANativeWindowWrapper> &nww,
- const sp<AMediaCryptoWrapper> &crypto,
- uint32_t flags) {
- if (mAMediaCodec == NULL) {
- return DEAD_OBJECT;
- }
-
- media_status_t err = AMediaCodec_configure(
- mAMediaCodec,
- format->getAMediaFormat(),
- (nww == NULL ? NULL : nww->getANativeWindow()),
- crypto == NULL ? NULL : crypto->getAMediaCrypto(),
- flags);
-
- return translateErrorCode(err);
-}
-
-status_t AMediaCodecWrapper::setCallback(const sp<AMessage> &callback) {
- if (mAMediaCodec == NULL) {
- return DEAD_OBJECT;
- }
-
- mCallback = callback;
-
- AMediaCodecOnAsyncNotifyCallback aCB = {
- OnInputAvailableCB,
- OnOutputAvailableCB,
- OnFormatChangedCB,
- OnErrorCB
- };
-
- return translateErrorCode(
- AMediaCodec_setAsyncNotifyCallback(mAMediaCodec, aCB, callback.get()));
-}
-
-status_t AMediaCodecWrapper::releaseCrypto() {
- if (mAMediaCodec == NULL) {
- return DEAD_OBJECT;
- }
- return translateErrorCode(AMediaCodec_releaseCrypto(mAMediaCodec));
-}
-
-status_t AMediaCodecWrapper::start() {
- if (mAMediaCodec == NULL) {
- return DEAD_OBJECT;
- }
- return translateErrorCode(AMediaCodec_start(mAMediaCodec));
-}
-
-status_t AMediaCodecWrapper::stop() {
- if (mAMediaCodec == NULL) {
- return DEAD_OBJECT;
- }
- return translateErrorCode(AMediaCodec_stop(mAMediaCodec));
-}
-
-status_t AMediaCodecWrapper::flush() {
- if (mAMediaCodec == NULL) {
- return DEAD_OBJECT;
- }
- return translateErrorCode(AMediaCodec_flush(mAMediaCodec));
-}
-
-uint8_t* AMediaCodecWrapper::getInputBuffer(size_t idx, size_t *out_size) {
- if (mAMediaCodec == NULL) {
- return NULL;
- }
- return AMediaCodec_getInputBuffer(mAMediaCodec, idx, out_size);
-}
-
-uint8_t* AMediaCodecWrapper::getOutputBuffer(size_t idx, size_t *out_size) {
- if (mAMediaCodec == NULL) {
- return NULL;
- }
- return AMediaCodec_getOutputBuffer(mAMediaCodec, idx, out_size);
-}
-
-status_t AMediaCodecWrapper::queueInputBuffer(
- size_t idx,
- size_t offset,
- size_t size,
- uint64_t time,
- uint32_t flags) {
- if (mAMediaCodec == NULL) {
- return DEAD_OBJECT;
- }
- return translateErrorCode(
- AMediaCodec_queueInputBuffer(mAMediaCodec, idx, offset, size, time, flags));
-}
-
-status_t AMediaCodecWrapper::queueSecureInputBuffer(
- size_t idx,
- size_t offset,
- sp<AMediaCodecCryptoInfoWrapper> &codecCryptoInfo,
- uint64_t time,
- uint32_t flags) {
- if (mAMediaCodec == NULL) {
- return DEAD_OBJECT;
- }
- return translateErrorCode(
- AMediaCodec_queueSecureInputBuffer(
- mAMediaCodec,
- idx,
- offset,
- codecCryptoInfo->getAMediaCodecCryptoInfo(),
- time,
- flags));
-}
-
-sp<AMediaFormatWrapper> AMediaCodecWrapper::getOutputFormat() {
- if (mAMediaCodec == NULL) {
- return NULL;
- }
- return new AMediaFormatWrapper(AMediaCodec_getOutputFormat(mAMediaCodec));
-}
-
-sp<AMediaFormatWrapper> AMediaCodecWrapper::getInputFormat() {
- if (mAMediaCodec == NULL) {
- return NULL;
- }
- return new AMediaFormatWrapper(AMediaCodec_getInputFormat(mAMediaCodec));
-}
-
-status_t AMediaCodecWrapper::releaseOutputBuffer(size_t idx, bool render) {
- if (mAMediaCodec == NULL) {
- return DEAD_OBJECT;
- }
- return translateErrorCode(
- AMediaCodec_releaseOutputBuffer(mAMediaCodec, idx, render));
-}
-
-status_t AMediaCodecWrapper::setOutputSurface(const sp<ANativeWindowWrapper> &nww) {
- if (mAMediaCodec == NULL) {
- return DEAD_OBJECT;
- }
- return translateErrorCode(
- AMediaCodec_setOutputSurface(mAMediaCodec,
- (nww == NULL ? NULL : nww->getANativeWindow())));
-}
-
-status_t AMediaCodecWrapper::releaseOutputBufferAtTime(size_t idx, int64_t timestampNs) {
- if (mAMediaCodec == NULL) {
- return DEAD_OBJECT;
- }
- return translateErrorCode(
- AMediaCodec_releaseOutputBufferAtTime(mAMediaCodec, idx, timestampNs));
-}
-
-status_t AMediaCodecWrapper::setParameters(const sp<AMediaFormatWrapper> ¶ms) {
- if (mAMediaCodec == NULL) {
- return DEAD_OBJECT;
- }
- return translateErrorCode(
- AMediaCodec_setParameters(mAMediaCodec, params->getAMediaFormat()));
-}
-
-//////////// AMediaExtractorWrapper
-
-AMediaExtractorWrapper::AMediaExtractorWrapper(AMediaExtractor *aMediaExtractor)
- : mAMediaExtractor(aMediaExtractor) {
-}
-
-AMediaExtractorWrapper::~AMediaExtractorWrapper() {
- release();
-}
-
-status_t AMediaExtractorWrapper::release() {
- if (mAMediaExtractor != NULL) {
- media_status_t err = AMediaExtractor_delete(mAMediaExtractor);
- mAMediaExtractor = NULL;
- return translateErrorCode(err);
- }
- return OK;
-}
-
-AMediaExtractor *AMediaExtractorWrapper::getAMediaExtractor() const {
- return mAMediaExtractor;
-}
-
-status_t AMediaExtractorWrapper::setDataSource(int fd, off64_t offset, off64_t length) {
- if (mAMediaExtractor == NULL) {
- return DEAD_OBJECT;
- }
- return translateErrorCode(AMediaExtractor_setDataSourceFd(
- mAMediaExtractor, fd, offset, length));
-}
-
-status_t AMediaExtractorWrapper::setDataSource(const char *location) {
- if (mAMediaExtractor == NULL) {
- return DEAD_OBJECT;
- }
- return translateErrorCode(AMediaExtractor_setDataSource(mAMediaExtractor, location));
-}
-
-status_t AMediaExtractorWrapper::setDataSource(AMediaDataSource *source) {
- if (mAMediaExtractor == NULL) {
- return DEAD_OBJECT;
- }
- return translateErrorCode(AMediaExtractor_setDataSourceCustom(mAMediaExtractor, source));
-}
-
-size_t AMediaExtractorWrapper::getTrackCount() {
- if (mAMediaExtractor == NULL) {
- return 0;
- }
- return AMediaExtractor_getTrackCount(mAMediaExtractor);
-}
-
-sp<AMediaFormatWrapper> AMediaExtractorWrapper::getFormat() {
- if (mAMediaExtractor == NULL) {
- return NULL;
- }
- return new AMediaFormatWrapper(AMediaExtractor_getFileFormat(mAMediaExtractor));
-}
-
-sp<AMediaFormatWrapper> AMediaExtractorWrapper::getTrackFormat(size_t idx) {
- if (mAMediaExtractor == NULL) {
- return NULL;
- }
- return new AMediaFormatWrapper(AMediaExtractor_getTrackFormat(mAMediaExtractor, idx));
-}
-
-status_t AMediaExtractorWrapper::selectTrack(size_t idx) {
- if (mAMediaExtractor == NULL) {
- return DEAD_OBJECT;
- }
- return translateErrorCode(AMediaExtractor_selectTrack(mAMediaExtractor, idx));
-}
-
-status_t AMediaExtractorWrapper::unselectTrack(size_t idx) {
- if (mAMediaExtractor == NULL) {
- return DEAD_OBJECT;
- }
- return translateErrorCode(AMediaExtractor_unselectTrack(mAMediaExtractor, idx));
-}
-
-status_t AMediaExtractorWrapper::selectSingleTrack(size_t idx) {
- if (mAMediaExtractor == NULL) {
- return DEAD_OBJECT;
- }
- for (size_t i = 0; i < AMediaExtractor_getTrackCount(mAMediaExtractor); ++i) {
- if (i == idx) {
- media_status_t err = AMediaExtractor_selectTrack(mAMediaExtractor, i);
- if (err != AMEDIA_OK) {
- return translateErrorCode(err);
- }
- } else {
- media_status_t err = AMediaExtractor_unselectTrack(mAMediaExtractor, i);
- if (err != AMEDIA_OK) {
- return translateErrorCode(err);
- }
- }
- }
- return OK;
-}
-
-ssize_t AMediaExtractorWrapper::readSampleData(const sp<ABuffer> &buffer) {
- if (mAMediaExtractor == NULL) {
- return -1;
- }
- return AMediaExtractor_readSampleData(mAMediaExtractor, buffer->data(), buffer->capacity());
-}
-
-ssize_t AMediaExtractorWrapper::getSampleSize() {
- if (mAMediaExtractor == NULL) {
- return 0;
- }
- return AMediaExtractor_getSampleSize(mAMediaExtractor);
-}
-
-uint32_t AMediaExtractorWrapper::getSampleFlags() {
- if (mAMediaExtractor == NULL) {
- return 0;
- }
- return AMediaExtractor_getSampleFlags(mAMediaExtractor);
-}
-
-int AMediaExtractorWrapper::getSampleTrackIndex() {
- if (mAMediaExtractor == NULL) {
- return -1;
- }
- return AMediaExtractor_getSampleTrackIndex(mAMediaExtractor);
-}
-
-int64_t AMediaExtractorWrapper::getSampleTime() {
- if (mAMediaExtractor == NULL) {
- return -1;
- }
- return AMediaExtractor_getSampleTime(mAMediaExtractor);
-}
-
-status_t AMediaExtractorWrapper::getSampleFormat(sp<AMediaFormatWrapper> &formatWrapper) {
- if (mAMediaExtractor == NULL) {
- return DEAD_OBJECT;
- }
- AMediaFormat *format = AMediaFormat_new();
- formatWrapper = new AMediaFormatWrapper(format);
- return translateErrorCode(AMediaExtractor_getSampleFormat(mAMediaExtractor, format));
-}
-
-int64_t AMediaExtractorWrapper::getCachedDuration() {
- if (mAMediaExtractor == NULL) {
- return -1;
- }
- return AMediaExtractor_getCachedDuration(mAMediaExtractor);
-}
-
-bool AMediaExtractorWrapper::advance() {
- if (mAMediaExtractor == NULL) {
- return false;
- }
- return AMediaExtractor_advance(mAMediaExtractor);
-}
-
-status_t AMediaExtractorWrapper::seekTo(int64_t seekPosUs, MediaSource::ReadOptions::SeekMode mode) {
- if (mAMediaExtractor == NULL) {
- return DEAD_OBJECT;
- }
-
- SeekMode aMode;
- switch (mode) {
- case MediaSource::ReadOptions::SEEK_PREVIOUS_SYNC: {
- aMode = AMEDIAEXTRACTOR_SEEK_PREVIOUS_SYNC;
- break;
- }
- case MediaSource::ReadOptions::SEEK_NEXT_SYNC: {
- aMode = AMEDIAEXTRACTOR_SEEK_NEXT_SYNC;
- break;
- }
- default: {
- aMode = AMEDIAEXTRACTOR_SEEK_CLOSEST_SYNC;
- break;
- }
- }
- return AMediaExtractor_seekTo(mAMediaExtractor, seekPosUs, aMode);
-}
-
-PsshInfo* AMediaExtractorWrapper::getPsshInfo() {
- if (mAMediaExtractor == NULL) {
- return NULL;
- }
- return AMediaExtractor_getPsshInfo(mAMediaExtractor);
-}
-
-sp<AMediaCodecCryptoInfoWrapper> AMediaExtractorWrapper::getSampleCryptoInfo() {
- if (mAMediaExtractor == NULL) {
- return NULL;
- }
- AMediaCodecCryptoInfo *cryptoInfo = AMediaExtractor_getSampleCryptoInfo(mAMediaExtractor);
- if (cryptoInfo == NULL) {
- return NULL;
- }
- return new AMediaCodecCryptoInfoWrapper(cryptoInfo);
-}
-
-AMediaDataSourceWrapper::AMediaDataSourceWrapper(const sp<DataSource> &dataSource)
- : mDataSource(dataSource),
- mAMediaDataSource(convertDataSourceToAMediaDataSource(dataSource)) {
-}
-
-AMediaDataSourceWrapper::AMediaDataSourceWrapper(AMediaDataSource *aDataSource)
- : mDataSource(NULL),
- mAMediaDataSource(aDataSource) {
-}
-
-AMediaDataSourceWrapper::~AMediaDataSourceWrapper() {
- if (mAMediaDataSource == NULL) {
- return;
- }
- AMediaDataSource_close(mAMediaDataSource);
- AMediaDataSource_delete(mAMediaDataSource);
- mAMediaDataSource = NULL;
-}
-
-AMediaDataSource* AMediaDataSourceWrapper::getAMediaDataSource() {
- return mAMediaDataSource;
-}
-
-void AMediaDataSourceWrapper::close() {
- AMediaDataSource_close(mAMediaDataSource);
-}
-
-} // namespace android
diff --git a/media/libmedia/Visualizer.cpp b/media/libmedia/Visualizer.cpp
deleted file mode 100644
index 2bf0802..0000000
--- a/media/libmedia/Visualizer.cpp
+++ /dev/null
@@ -1,445 +0,0 @@
-/*
-**
-** Copyright 2010, The Android Open Source Project
-**
-** Licensed under the Apache License, Version 2.0 (the "License");
-** you may not use this file except in compliance with the License.
-** You may obtain a copy of the License at
-**
-** http://www.apache.org/licenses/LICENSE-2.0
-**
-** Unless required by applicable law or agreed to in writing, software
-** distributed under the License is distributed on an "AS IS" BASIS,
-** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-** See the License for the specific language governing permissions and
-** limitations under the License.
-*/
-
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "Visualizer"
-#include <utils/Log.h>
-
-#include <stdint.h>
-#include <sys/types.h>
-#include <limits.h>
-
-#include <media/Visualizer.h>
-#include <audio_utils/fixedfft.h>
-#include <utils/Thread.h>
-
-namespace android {
-
-// ---------------------------------------------------------------------------
-
-Visualizer::Visualizer (const String16& opPackageName,
- int32_t priority,
- effect_callback_t cbf,
- void* user,
- audio_session_t sessionId)
- : AudioEffect(SL_IID_VISUALIZATION, opPackageName, NULL, priority, cbf, user, sessionId),
- mCaptureRate(CAPTURE_RATE_DEF),
- mCaptureSize(CAPTURE_SIZE_DEF),
- mSampleRate(44100000),
- mScalingMode(VISUALIZER_SCALING_MODE_NORMALIZED),
- mMeasurementMode(MEASUREMENT_MODE_NONE),
- mCaptureCallBack(NULL),
- mCaptureCbkUser(NULL)
-{
- initCaptureSize();
-}
-
-Visualizer::~Visualizer()
-{
- ALOGV("Visualizer::~Visualizer()");
- setEnabled(false);
- setCaptureCallBack(NULL, NULL, 0, 0);
-}
-
-void Visualizer::release()
-{
- ALOGV("Visualizer::release()");
- setEnabled(false);
- Mutex::Autolock _l(mCaptureLock);
-
- mCaptureThread.clear();
- mCaptureCallBack = NULL;
- mCaptureCbkUser = NULL;
- mCaptureFlags = 0;
- mCaptureRate = 0;
-}
-
-status_t Visualizer::setEnabled(bool enabled)
-{
- Mutex::Autolock _l(mCaptureLock);
-
- sp<CaptureThread> t = mCaptureThread;
- if (t != 0) {
- if (enabled) {
- if (t->exitPending()) {
- mCaptureLock.unlock();
- if (t->requestExitAndWait() == WOULD_BLOCK) {
- mCaptureLock.lock();
- ALOGE("Visualizer::enable() called from thread");
- return INVALID_OPERATION;
- }
- mCaptureLock.lock();
- }
- }
- t->mLock.lock();
- }
-
- status_t status = AudioEffect::setEnabled(enabled);
-
- if (t != 0) {
- if (enabled && status == NO_ERROR) {
- t->run("Visualizer");
- } else {
- t->requestExit();
- }
- }
-
- if (t != 0) {
- t->mLock.unlock();
- }
-
- return status;
-}
-
-status_t Visualizer::setCaptureCallBack(capture_cbk_t cbk, void* user, uint32_t flags,
- uint32_t rate)
-{
- if (rate > CAPTURE_RATE_MAX) {
- return BAD_VALUE;
- }
- Mutex::Autolock _l(mCaptureLock);
-
- if (mEnabled) {
- return INVALID_OPERATION;
- }
-
- if (mCaptureThread != 0) {
- mCaptureLock.unlock();
- mCaptureThread->requestExitAndWait();
- mCaptureLock.lock();
- }
-
- mCaptureThread.clear();
- mCaptureCallBack = cbk;
- mCaptureCbkUser = user;
- mCaptureFlags = flags;
- mCaptureRate = rate;
-
- if (cbk != NULL) {
- mCaptureThread = new CaptureThread(this, rate, ((flags & CAPTURE_CALL_JAVA) != 0));
- }
- ALOGV("setCaptureCallBack() rate: %d thread %p flags 0x%08x",
- rate, mCaptureThread.get(), mCaptureFlags);
- return NO_ERROR;
-}
-
-status_t Visualizer::setCaptureSize(uint32_t size)
-{
- if (size > VISUALIZER_CAPTURE_SIZE_MAX ||
- size < VISUALIZER_CAPTURE_SIZE_MIN ||
- popcount(size) != 1) {
- return BAD_VALUE;
- }
-
- Mutex::Autolock _l(mCaptureLock);
- if (mEnabled) {
- return INVALID_OPERATION;
- }
-
- uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2];
- effect_param_t *p = (effect_param_t *)buf32;
-
- p->psize = sizeof(uint32_t);
- p->vsize = sizeof(uint32_t);
- *(int32_t *)p->data = VISUALIZER_PARAM_CAPTURE_SIZE;
- *((int32_t *)p->data + 1)= size;
- status_t status = setParameter(p);
-
- ALOGV("setCaptureSize size %d status %d p->status %d", size, status, p->status);
-
- if (status == NO_ERROR) {
- status = p->status;
- if (status == NO_ERROR) {
- mCaptureSize = size;
- }
- }
-
- return status;
-}
-
-status_t Visualizer::setScalingMode(uint32_t mode) {
- if ((mode != VISUALIZER_SCALING_MODE_NORMALIZED)
- && (mode != VISUALIZER_SCALING_MODE_AS_PLAYED)) {
- return BAD_VALUE;
- }
-
- Mutex::Autolock _l(mCaptureLock);
-
- uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2];
- effect_param_t *p = (effect_param_t *)buf32;
-
- p->psize = sizeof(uint32_t);
- p->vsize = sizeof(uint32_t);
- *(int32_t *)p->data = VISUALIZER_PARAM_SCALING_MODE;
- *((int32_t *)p->data + 1)= mode;
- status_t status = setParameter(p);
-
- ALOGV("setScalingMode mode %d status %d p->status %d", mode, status, p->status);
-
- if (status == NO_ERROR) {
- status = p->status;
- if (status == NO_ERROR) {
- mScalingMode = mode;
- }
- }
-
- return status;
-}
-
-status_t Visualizer::setMeasurementMode(uint32_t mode) {
- if ((mode != MEASUREMENT_MODE_NONE)
- //Note: needs to be handled as a mask when more measurement modes are added
- && ((mode & MEASUREMENT_MODE_PEAK_RMS) != mode)) {
- return BAD_VALUE;
- }
-
- Mutex::Autolock _l(mCaptureLock);
-
- uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2];
- effect_param_t *p = (effect_param_t *)buf32;
-
- p->psize = sizeof(uint32_t);
- p->vsize = sizeof(uint32_t);
- *(int32_t *)p->data = VISUALIZER_PARAM_MEASUREMENT_MODE;
- *((int32_t *)p->data + 1)= mode;
- status_t status = setParameter(p);
-
- ALOGV("setMeasurementMode mode %d status %d p->status %d", mode, status, p->status);
-
- if (status == NO_ERROR) {
- status = p->status;
- if (status == NO_ERROR) {
- mMeasurementMode = mode;
- }
- }
- return status;
-}
-
-status_t Visualizer::getIntMeasurements(uint32_t type, uint32_t number, int32_t *measurements) {
- if (mMeasurementMode == MEASUREMENT_MODE_NONE) {
- ALOGE("Cannot retrieve int measurements, no measurement mode set");
- return INVALID_OPERATION;
- }
- if (!(mMeasurementMode & type)) {
- // measurement type has not been set on this Visualizer
- ALOGE("Cannot retrieve int measurements, requested measurement mode 0x%x not set(0x%x)",
- type, mMeasurementMode);
- return INVALID_OPERATION;
- }
- // only peak+RMS measurement supported
- if ((type != MEASUREMENT_MODE_PEAK_RMS)
- // for peak+RMS measurement, the results are 2 int32_t values
- || (number != 2)) {
- ALOGE("Cannot retrieve int measurements, MEASUREMENT_MODE_PEAK_RMS returns 2 ints, not %d",
- number);
- return BAD_VALUE;
- }
-
- status_t status = NO_ERROR;
- if (mEnabled) {
- uint32_t replySize = number * sizeof(int32_t);
- status = command(VISUALIZER_CMD_MEASURE,
- sizeof(uint32_t) /*cmdSize*/,
- &type /*cmdData*/,
- &replySize, measurements);
- ALOGV("getMeasurements() command returned %d", status);
- if ((status == NO_ERROR) && (replySize == 0)) {
- status = NOT_ENOUGH_DATA;
- }
- } else {
- ALOGV("getMeasurements() disabled");
- return INVALID_OPERATION;
- }
- return status;
-}
-
-status_t Visualizer::getWaveForm(uint8_t *waveform)
-{
- if (waveform == NULL) {
- return BAD_VALUE;
- }
- if (mCaptureSize == 0) {
- return NO_INIT;
- }
-
- status_t status = NO_ERROR;
- if (mEnabled) {
- uint32_t replySize = mCaptureSize;
- status = command(VISUALIZER_CMD_CAPTURE, 0, NULL, &replySize, waveform);
- ALOGV("getWaveForm() command returned %d", status);
- if ((status == NO_ERROR) && (replySize == 0)) {
- status = NOT_ENOUGH_DATA;
- }
- } else {
- ALOGV("getWaveForm() disabled");
- memset(waveform, 0x80, mCaptureSize);
- }
- return status;
-}
-
-status_t Visualizer::getFft(uint8_t *fft)
-{
- if (fft == NULL) {
- return BAD_VALUE;
- }
- if (mCaptureSize == 0) {
- return NO_INIT;
- }
-
- status_t status = NO_ERROR;
- if (mEnabled) {
- uint8_t buf[mCaptureSize];
- status = getWaveForm(buf);
- if (status == NO_ERROR) {
- status = doFft(fft, buf);
- }
- } else {
- memset(fft, 0, mCaptureSize);
- }
- return status;
-}
-
-status_t Visualizer::doFft(uint8_t *fft, uint8_t *waveform)
-{
- int32_t workspace[mCaptureSize >> 1];
- int32_t nonzero = 0;
-
- for (uint32_t i = 0; i < mCaptureSize; i += 2) {
- workspace[i >> 1] =
- ((waveform[i] ^ 0x80) << 24) | ((waveform[i + 1] ^ 0x80) << 8);
- nonzero |= workspace[i >> 1];
- }
-
- if (nonzero) {
- fixed_fft_real(mCaptureSize >> 1, workspace);
- }
-
- for (uint32_t i = 0; i < mCaptureSize; i += 2) {
- short tmp = workspace[i >> 1] >> 21;
- while (tmp > 127 || tmp < -128) tmp >>= 1;
- fft[i] = tmp;
- tmp = workspace[i >> 1];
- tmp >>= 5;
- while (tmp > 127 || tmp < -128) tmp >>= 1;
- fft[i + 1] = tmp;
- }
-
- return NO_ERROR;
-}
-
-void Visualizer::periodicCapture()
-{
- Mutex::Autolock _l(mCaptureLock);
- ALOGV("periodicCapture() %p mCaptureCallBack %p mCaptureFlags 0x%08x",
- this, mCaptureCallBack, mCaptureFlags);
- if (mCaptureCallBack != NULL &&
- (mCaptureFlags & (CAPTURE_WAVEFORM|CAPTURE_FFT)) &&
- mCaptureSize != 0) {
- uint8_t waveform[mCaptureSize];
- status_t status = getWaveForm(waveform);
- if (status != NO_ERROR) {
- return;
- }
- uint8_t fft[mCaptureSize];
- if (mCaptureFlags & CAPTURE_FFT) {
- status = doFft(fft, waveform);
- }
- if (status != NO_ERROR) {
- return;
- }
- uint8_t *wavePtr = NULL;
- uint8_t *fftPtr = NULL;
- uint32_t waveSize = 0;
- uint32_t fftSize = 0;
- if (mCaptureFlags & CAPTURE_WAVEFORM) {
- wavePtr = waveform;
- waveSize = mCaptureSize;
- }
- if (mCaptureFlags & CAPTURE_FFT) {
- fftPtr = fft;
- fftSize = mCaptureSize;
- }
- mCaptureCallBack(mCaptureCbkUser, waveSize, wavePtr, fftSize, fftPtr, mSampleRate);
- }
-}
-
-uint32_t Visualizer::initCaptureSize()
-{
- uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2];
- effect_param_t *p = (effect_param_t *)buf32;
-
- p->psize = sizeof(uint32_t);
- p->vsize = sizeof(uint32_t);
- *(int32_t *)p->data = VISUALIZER_PARAM_CAPTURE_SIZE;
- status_t status = getParameter(p);
-
- if (status == NO_ERROR) {
- status = p->status;
- }
-
- uint32_t size = 0;
- if (status == NO_ERROR) {
- size = *((int32_t *)p->data + 1);
- }
- mCaptureSize = size;
-
- ALOGV("initCaptureSize size %d status %d", mCaptureSize, status);
-
- return size;
-}
-
-void Visualizer::controlStatusChanged(bool controlGranted) {
- if (controlGranted) {
- // this Visualizer instance regained control of the effect, reset the scaling mode
- // and capture size as has been cached through it.
- ALOGV("controlStatusChanged(true) causes effect parameter reset:");
- ALOGV(" scaling mode reset to %d", mScalingMode);
- setScalingMode(mScalingMode);
- ALOGV(" capture size reset to %d", mCaptureSize);
- setCaptureSize(mCaptureSize);
- }
- AudioEffect::controlStatusChanged(controlGranted);
-}
-
-//-------------------------------------------------------------------------
-
-Visualizer::CaptureThread::CaptureThread(Visualizer* receiver, uint32_t captureRate,
- bool bCanCallJava)
- : Thread(bCanCallJava), mReceiver(receiver)
-{
- mSleepTimeUs = 1000000000 / captureRate;
- ALOGV("CaptureThread cstor %p captureRate %d mSleepTimeUs %d", this, captureRate, mSleepTimeUs);
-}
-
-bool Visualizer::CaptureThread::threadLoop()
-{
- ALOGV("CaptureThread %p enter", this);
- sp<Visualizer> receiver = mReceiver.promote();
- if (receiver == NULL) {
- return false;
- }
- while (!exitPending())
- {
- usleep(mSleepTimeUs);
- receiver->periodicCapture();
- }
- ALOGV("CaptureThread %p exiting", this);
- return false;
-}
-
-} // namespace android
diff --git a/media/libmedia/include/media/DataSourceDesc.h b/media/libmedia/include/media/DataSourceDesc.h
deleted file mode 100644
index 4336767..0000000
--- a/media/libmedia/include/media/DataSourceDesc.h
+++ /dev/null
@@ -1,73 +0,0 @@
-/*
- * Copyright 2018 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_DATASOURCEDESC_H
-#define ANDROID_DATASOURCEDESC_H
-
-#include <media/stagefright/foundation/ABase.h>
-#include <utils/RefBase.h>
-#include <utils/KeyedVector.h>
-#include <utils/String8.h>
-
-namespace android {
-
-class DataSource;
-struct MediaHTTPService;
-
-// A binder interface for implementing a stagefright DataSource remotely.
-struct DataSourceDesc : public RefBase {
-public:
- // intentionally less than INT64_MAX
- // keep consistent with JAVA code
- static const int64_t kMaxTimeMs = 0x7ffffffffffffffll / 1000;
- static const int64_t kMaxTimeUs = kMaxTimeMs * 1000;
-
- enum {
- /* No data source has been set yet */
- TYPE_NONE = 0,
- /* data source is type of MediaDataSource */
- TYPE_CALLBACK = 1,
- /* data source is type of FileDescriptor */
- TYPE_FD = 2,
- /* data source is type of Url */
- TYPE_URL = 3,
- };
-
- DataSourceDesc();
-
- int mType;
-
- sp<MediaHTTPService> mHttpService;
- String8 mUrl;
- KeyedVector<String8, String8> mHeaders;
-
- int mFD;
- int64_t mFDOffset;
- int64_t mFDLength;
-
- sp<DataSource> mCallbackSource;
-
- int64_t mId;
- int64_t mStartPositionMs;
- int64_t mEndPositionMs;
-
-private:
- DISALLOW_EVIL_CONSTRUCTORS(DataSourceDesc);
-};
-
-}; // namespace android
-
-#endif // ANDROID_DATASOURCEDESC_H
diff --git a/media/libmedia/include/media/IDataSource.h b/media/libmedia/include/media/IDataSource.h
index 3858f78..43e2b50 100644
--- a/media/libmedia/include/media/IDataSource.h
+++ b/media/libmedia/include/media/IDataSource.h
@@ -50,8 +50,6 @@
virtual uint32_t getFlags() = 0;
// get a description of the source, e.g. the url or filename it is based on
virtual String8 toString() = 0;
- // Initialize DRM and return a DecryptHandle.
- virtual sp<DecryptHandle> DrmInitialization(const char *mime) = 0;
private:
DISALLOW_EVIL_CONSTRUCTORS(IDataSource);
diff --git a/media/libmedia/include/media/JetPlayer.h b/media/libmedia/include/media/JetPlayer.h
deleted file mode 100644
index bb569bc..0000000
--- a/media/libmedia/include/media/JetPlayer.h
+++ /dev/null
@@ -1,126 +0,0 @@
-/*
- * Copyright (C) 2008 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef JETPLAYER_H_
-#define JETPLAYER_H_
-
-#include <utils/threads.h>
-
-#include <libsonivox/jet.h>
-#include <libsonivox/eas_types.h>
-#include <media/AudioTrack.h>
-#include <media/MidiIoWrapper.h>
-
-
-namespace android {
-
-typedef void (*jetevent_callback)(int eventType, int val1, int val2, void *cookie);
-
-class JetPlayer {
-
-public:
-
- // to keep in sync with the JetPlayer class constants
- // defined in frameworks/base/media/java/android/media/JetPlayer.java
- static const int JET_EVENT = 1;
- static const int JET_USERID_UPDATE = 2;
- static const int JET_NUMQUEUEDSEGMENT_UPDATE = 3;
- static const int JET_PAUSE_UPDATE = 4;
-
- JetPlayer(void *javaJetPlayer,
- int maxTracks = 32,
- int trackBufferSize = 1200);
- ~JetPlayer();
- int init();
- int release();
-
- int loadFromFile(const char* url);
- int loadFromFD(const int fd, const long long offset, const long long length);
- int closeFile();
- int play();
- int pause();
- int queueSegment(int segmentNum, int libNum, int repeatCount, int transpose,
- EAS_U32 muteFlags, EAS_U8 userID);
- int setMuteFlags(EAS_U32 muteFlags, bool sync);
- int setMuteFlag(int trackNum, bool muteFlag, bool sync);
- int triggerClip(int clipId);
- int clearQueue();
-
- void setEventCallback(jetevent_callback callback);
-
- int getMaxTracks() { return mMaxTracks; };
-
-
-private:
- int render();
- void fireUpdateOnStatusChange();
- void fireEventsFromJetQueue();
-
- JetPlayer() {} // no default constructor
- void dump();
- void dumpJetStatus(S_JET_STATUS* pJetStatus);
-
- jetevent_callback mEventCallback;
-
- void* mJavaJetPlayerRef;
- Mutex mMutex; // mutex to sync the render and playback thread with the JET calls
- pid_t mTid;
- Condition mCondition;
- volatile bool mRender;
- bool mPaused;
-
- EAS_STATE mState;
- int* mMemFailedVar;
-
- int mMaxTracks; // max number of MIDI tracks, usually 32
- EAS_DATA_HANDLE mEasData;
- MidiIoWrapper* mIoWrapper;
- EAS_PCM* mAudioBuffer;// EAS renders the MIDI data into this buffer,
- sp<AudioTrack> mAudioTrack; // and we play it in this audio track
- int mTrackBufferSize;
- S_JET_STATUS mJetStatus;
- S_JET_STATUS mPreviousJetStatus;
-
- class JetPlayerThread : public Thread {
- public:
- JetPlayerThread(JetPlayer *player) : mPlayer(player) {
- }
-
- protected:
- virtual ~JetPlayerThread() {}
-
- private:
- JetPlayer *mPlayer;
-
- bool threadLoop() {
- int result;
- result = mPlayer->render();
- return false;
- }
-
- JetPlayerThread(const JetPlayerThread &);
- JetPlayerThread &operator=(const JetPlayerThread &);
- };
-
- sp<JetPlayerThread> mThread;
-
-}; // end class JetPlayer
-
-} // end namespace android
-
-
-
-#endif /*JETPLAYER_H_*/
diff --git a/media/libmedia/include/media/MediaResource.h b/media/libmedia/include/media/MediaResource.h
index 10a07bb..e9684f0 100644
--- a/media/libmedia/include/media/MediaResource.h
+++ b/media/libmedia/include/media/MediaResource.h
@@ -20,6 +20,7 @@
#include <binder/Parcel.h>
#include <utils/String8.h>
+#include <vector>
namespace android {
@@ -32,6 +33,7 @@
kGraphicMemory,
kCpuBoost,
kBattery,
+ kDrmSession,
};
enum SubType {
@@ -43,6 +45,7 @@
MediaResource();
MediaResource(Type type, uint64_t value);
MediaResource(Type type, SubType subType, uint64_t value);
+ MediaResource(Type type, const std::vector<uint8_t> &id, uint64_t value);
void readFromParcel(const Parcel &parcel);
void writeToParcel(Parcel *parcel) const;
@@ -55,6 +58,8 @@
Type mType;
SubType mSubType;
uint64_t mValue;
+ // for kDrmSession-type mId is the unique session id obtained via MediaDrm#openSession
+ std::vector<uint8_t> mId;
};
inline static const char *asString(MediaResource::Type i, const char *def = "??") {
@@ -65,6 +70,7 @@
case MediaResource::kGraphicMemory: return "graphic-memory";
case MediaResource::kCpuBoost: return "cpu-boost";
case MediaResource::kBattery: return "battery";
+ case MediaResource::kDrmSession: return "drm-session";
default: return def;
}
}
diff --git a/media/libmedia/include/media/MidiDeviceInfo.h b/media/libmedia/include/media/MidiDeviceInfo.h
deleted file mode 100644
index 5b4a241..0000000
--- a/media/libmedia/include/media/MidiDeviceInfo.h
+++ /dev/null
@@ -1,81 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_MEDIA_MIDI_DEVICE_INFO_H
-#define ANDROID_MEDIA_MIDI_DEVICE_INFO_H
-
-#include <binder/Parcelable.h>
-#include <binder/PersistableBundle.h>
-#include <utils/String16.h>
-#include <utils/Vector.h>
-
-namespace android {
-namespace media {
-namespace midi {
-
-class MidiDeviceInfo : public Parcelable {
-public:
- MidiDeviceInfo() = default;
- virtual ~MidiDeviceInfo() = default;
- MidiDeviceInfo(const MidiDeviceInfo& midiDeviceInfo) = default;
-
- status_t writeToParcel(Parcel* parcel) const override;
- status_t readFromParcel(const Parcel* parcel) override;
-
- int getType() const { return mType; }
- int getUid() const { return mId; }
- bool isPrivate() const { return mIsPrivate; }
- const Vector<String16>& getInputPortNames() const { return mInputPortNames; }
- const Vector<String16>& getOutputPortNames() const { return mOutputPortNames; }
- String16 getProperty(const char* propertyName);
-
- // The constants need to be kept in sync with MidiDeviceInfo.java
- enum {
- TYPE_USB = 1,
- TYPE_VIRTUAL = 2,
- TYPE_BLUETOOTH = 3,
- };
- static const char* const PROPERTY_NAME;
- static const char* const PROPERTY_MANUFACTURER;
- static const char* const PROPERTY_PRODUCT;
- static const char* const PROPERTY_VERSION;
- static const char* const PROPERTY_SERIAL_NUMBER;
- static const char* const PROPERTY_ALSA_CARD;
- static const char* const PROPERTY_ALSA_DEVICE;
-
- friend bool operator==(const MidiDeviceInfo& lhs, const MidiDeviceInfo& rhs);
- friend bool operator!=(const MidiDeviceInfo& lhs, const MidiDeviceInfo& rhs) {
- return !(lhs == rhs);
- }
-
-private:
- status_t readStringVector(
- const Parcel* parcel, Vector<String16> *vectorPtr, size_t defaultLength);
- status_t writeStringVector(Parcel* parcel, const Vector<String16>& vector) const;
-
- int32_t mType;
- int32_t mId;
- Vector<String16> mInputPortNames;
- Vector<String16> mOutputPortNames;
- os::PersistableBundle mProperties;
- bool mIsPrivate;
-};
-
-} // namespace midi
-} // namespace media
-} // namespace android
-
-#endif // ANDROID_MEDIA_MIDI_DEVICE_INFO_H
diff --git a/media/libmedia/include/media/MidiIoWrapper.h b/media/libmedia/include/media/MidiIoWrapper.h
index b19d49e..d29949e 100644
--- a/media/libmedia/include/media/MidiIoWrapper.h
+++ b/media/libmedia/include/media/MidiIoWrapper.h
@@ -24,7 +24,6 @@
namespace android {
struct CDataSource;
-class DataSourceUnwrapper;
class MidiIoWrapper {
public:
@@ -43,6 +42,7 @@
int mFd;
off64_t mBase;
int64_t mLength;
+ class DataSourceUnwrapper;
DataSourceUnwrapper *mDataSource;
EAS_FILE mEasFile;
};
diff --git a/media/libmedia/include/media/Visualizer.h b/media/libmedia/include/media/Visualizer.h
deleted file mode 100644
index 8078e36..0000000
--- a/media/libmedia/include/media/Visualizer.h
+++ /dev/null
@@ -1,179 +0,0 @@
-/*
- * Copyright (C) 2010 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_MEDIA_VISUALIZER_H
-#define ANDROID_MEDIA_VISUALIZER_H
-
-#include <media/AudioEffect.h>
-#include <system/audio_effects/effect_visualizer.h>
-#include <utils/Thread.h>
-
-/**
- * The Visualizer class enables application to retrieve part of the currently playing audio for
- * visualization purpose. It is not an audio recording interface and only returns partial and low
- * quality audio content. However, to protect privacy of certain audio data (e.g voice mail) the use
- * of the visualizer requires the permission android.permission.RECORD_AUDIO.
- * The audio session ID passed to the constructor indicates which audio content should be
- * visualized:
- * - If the session is 0, the audio output mix is visualized
- * - If the session is not 0, the audio from a particular MediaPlayer or AudioTrack
- * using this audio session is visualized
- * Two types of representation of audio content can be captured:
- * - Waveform data: consecutive 8-bit (unsigned) mono samples by using the getWaveForm() method
- * - Frequency data: 8-bit magnitude FFT by using the getFft() method
- *
- * The length of the capture can be retrieved or specified by calling respectively
- * getCaptureSize() and setCaptureSize() methods. Note that the size of the FFT
- * is half of the specified capture size but both sides of the spectrum are returned yielding in a
- * number of bytes equal to the capture size. The capture size must be a power of 2 in the range
- * returned by getMinCaptureSize() and getMaxCaptureSize().
- * In addition to the polling capture mode, a callback mode is also available by installing a
- * callback function by use of the setCaptureCallBack() method. The rate at which the callback
- * is called as well as the type of data returned is specified.
- * Before capturing data, the Visualizer must be enabled by calling the setEnabled() method.
- * When data capture is not needed any more, the Visualizer should be disabled.
- */
-
-
-namespace android {
-
-// ----------------------------------------------------------------------------
-
-class Visualizer: public AudioEffect {
-public:
-
- enum callback_flags {
- CAPTURE_WAVEFORM = 0x00000001, // capture callback returns a PCM wave form
- CAPTURE_FFT = 0x00000002, // apture callback returns a frequency representation
- CAPTURE_CALL_JAVA = 0x00000004 // the callback thread can call java
- };
-
-
- /* Constructor.
- * See AudioEffect constructor for details on parameters.
- */
- Visualizer(const String16& opPackageName,
- int32_t priority = 0,
- effect_callback_t cbf = NULL,
- void* user = NULL,
- audio_session_t sessionId = AUDIO_SESSION_OUTPUT_MIX);
-
- ~Visualizer();
-
- virtual status_t setEnabled(bool enabled);
-
- // maximum capture size in samples
- static uint32_t getMaxCaptureSize() { return VISUALIZER_CAPTURE_SIZE_MAX; }
- // minimum capture size in samples
- static uint32_t getMinCaptureSize() { return VISUALIZER_CAPTURE_SIZE_MIN; }
- // maximum capture rate in millihertz
- static uint32_t getMaxCaptureRate() { return CAPTURE_RATE_MAX; }
-
- // callback used to return periodic PCM or FFT captures to the application. Either one or both
- // types of data are returned (PCM and FFT) according to flags indicated when installing the
- // callback. When a type of data is not present, the corresponding size (waveformSize or
- // fftSize) is 0.
- typedef void (*capture_cbk_t)(void* user,
- uint32_t waveformSize,
- uint8_t *waveform,
- uint32_t fftSize,
- uint8_t *fft,
- uint32_t samplingrate);
-
- // install a callback to receive periodic captures. The capture rate is specified in milliHertz
- // and the capture format is according to flags (see callback_flags).
- status_t setCaptureCallBack(capture_cbk_t cbk, void* user, uint32_t flags, uint32_t rate);
-
- // set the capture size capture size must be a power of two in the range
- // [VISUALIZER_CAPTURE_SIZE_MAX. VISUALIZER_CAPTURE_SIZE_MIN]
- // must be called when the visualizer is not enabled
- status_t setCaptureSize(uint32_t size);
- uint32_t getCaptureSize() { return mCaptureSize; }
-
- // returns the capture rate indicated when installing the callback
- uint32_t getCaptureRate() { return mCaptureRate; }
-
- // returns the sampling rate of the audio being captured
- uint32_t getSamplingRate() { return mSampleRate; }
-
- // set the way volume affects the captured data
- // mode must one of VISUALIZER_SCALING_MODE_NORMALIZED,
- // VISUALIZER_SCALING_MODE_AS_PLAYED
- status_t setScalingMode(uint32_t mode);
- uint32_t getScalingMode() { return mScalingMode; }
-
- // set which measurements are done on the audio buffers processed by the effect.
- // valid measurements (mask): MEASUREMENT_MODE_PEAK_RMS
- status_t setMeasurementMode(uint32_t mode);
- uint32_t getMeasurementMode() { return mMeasurementMode; }
-
- // return a set of int32_t measurements
- status_t getIntMeasurements(uint32_t type, uint32_t number, int32_t *measurements);
-
- // return a capture in PCM 8 bit unsigned format. The size of the capture is equal to
- // getCaptureSize()
- status_t getWaveForm(uint8_t *waveform);
-
- // return a capture in FFT 8 bit signed format. The size of the capture is equal to
- // getCaptureSize() but the length of the FFT is half of the size (both parts of the spectrum
- // are returned
- status_t getFft(uint8_t *fft);
- void release();
-
-protected:
- // from IEffectClient
- virtual void controlStatusChanged(bool controlGranted);
-
-private:
-
- static const uint32_t CAPTURE_RATE_MAX = 20000;
- static const uint32_t CAPTURE_RATE_DEF = 10000;
- static const uint32_t CAPTURE_SIZE_DEF = VISUALIZER_CAPTURE_SIZE_MAX;
-
- /* internal class to handle the callback */
- class CaptureThread : public Thread
- {
- public:
- CaptureThread(Visualizer* visualizer, uint32_t captureRate, bool bCanCallJava = false);
-
- private:
- friend class Visualizer;
- virtual bool threadLoop();
- wp<Visualizer> mReceiver;
- Mutex mLock;
- uint32_t mSleepTimeUs;
- };
-
- status_t doFft(uint8_t *fft, uint8_t *waveform);
- void periodicCapture();
- uint32_t initCaptureSize();
-
- Mutex mCaptureLock;
- uint32_t mCaptureRate;
- uint32_t mCaptureSize;
- uint32_t mSampleRate;
- uint32_t mScalingMode;
- uint32_t mMeasurementMode;
- capture_cbk_t mCaptureCallBack;
- void *mCaptureCbkUser;
- sp<CaptureThread> mCaptureThread;
- uint32_t mCaptureFlags;
-};
-
-
-}; // namespace android
-
-#endif // ANDROID_MEDIA_VISUALIZER_H
diff --git a/media/libmediahelper/Android.bp b/media/libmediahelper/Android.bp
new file mode 100644
index 0000000..72edeec
--- /dev/null
+++ b/media/libmediahelper/Android.bp
@@ -0,0 +1,29 @@
+cc_library_headers {
+ name: "libmedia_helper_headers",
+ vendor_available: true,
+ export_include_dirs: ["include"],
+}
+
+cc_library {
+ name: "libmedia_helper",
+ vendor_available: true,
+ vndk: {
+ enabled: true,
+ },
+ double_loadable: true,
+ srcs: ["AudioParameter.cpp", "TypeConverter.cpp"],
+ cflags: [
+ "-Werror",
+ "-Wextra",
+ "-Wall",
+ ],
+ shared_libs: ["libutils", "liblog"],
+ header_libs: [
+ "libmedia_helper_headers",
+ "libaudio_system_headers",
+ ],
+ export_header_lib_headers: [
+ "libmedia_helper_headers",
+ ],
+ clang: true,
+}
diff --git a/media/libmedia/AudioParameter.cpp b/media/libmediahelper/AudioParameter.cpp
similarity index 97%
rename from media/libmedia/AudioParameter.cpp
rename to media/libmediahelper/AudioParameter.cpp
index 1c95e27..9f34035 100644
--- a/media/libmedia/AudioParameter.cpp
+++ b/media/libmediahelper/AudioParameter.cpp
@@ -40,6 +40,8 @@
AUDIO_PARAMETER_KEY_AUDIO_LANGUAGE_PREFERRED;
const char * const AudioParameter::keyMonoOutput = AUDIO_PARAMETER_MONO_OUTPUT;
const char * const AudioParameter::keyStreamHwAvSync = AUDIO_PARAMETER_STREAM_HW_AV_SYNC;
+const char * const AudioParameter::keyDeviceConnect = AUDIO_PARAMETER_DEVICE_CONNECT;
+const char * const AudioParameter::keyDeviceDisconnect = AUDIO_PARAMETER_DEVICE_DISCONNECT;
const char * const AudioParameter::keyStreamConnect = AUDIO_PARAMETER_DEVICE_CONNECT;
const char * const AudioParameter::keyStreamDisconnect = AUDIO_PARAMETER_DEVICE_DISCONNECT;
const char * const AudioParameter::keyStreamSupportedFormats = AUDIO_PARAMETER_STREAM_SUP_FORMATS;
diff --git a/media/libmedia/TypeConverter.cpp b/media/libmediahelper/TypeConverter.cpp
similarity index 99%
rename from media/libmedia/TypeConverter.cpp
rename to media/libmediahelper/TypeConverter.cpp
index 5be78d1..c103236 100644
--- a/media/libmedia/TypeConverter.cpp
+++ b/media/libmediahelper/TypeConverter.cpp
@@ -312,6 +312,7 @@
MAKE_STRING_FROM_ENUM(AUDIO_STREAM_DTMF),
MAKE_STRING_FROM_ENUM(AUDIO_STREAM_TTS),
MAKE_STRING_FROM_ENUM(AUDIO_STREAM_ACCESSIBILITY),
+ MAKE_STRING_FROM_ENUM(AUDIO_STREAM_ASSISTANT),
MAKE_STRING_FROM_ENUM(AUDIO_STREAM_REROUTING),
MAKE_STRING_FROM_ENUM(AUDIO_STREAM_PATCH),
TERMINATOR
@@ -393,6 +394,7 @@
MAKE_STRING_FROM_ENUM(AUDIO_FLAG_LOW_LATENCY),
MAKE_STRING_FROM_ENUM(AUDIO_FLAG_DEEP_BUFFER),
MAKE_STRING_FROM_ENUM(AUDIO_FLAG_NO_MEDIA_PROJECTION),
+ MAKE_STRING_FROM_ENUM(AUDIO_FLAG_MUTE_HAPTIC),
MAKE_STRING_FROM_ENUM(AUDIO_FLAG_NO_SYSTEM_CAPTURE),
TERMINATOR
};
diff --git a/media/libaudioclient/include/media/AudioParameter.h b/media/libmediahelper/include/media/AudioParameter.h
similarity index 92%
rename from media/libaudioclient/include/media/AudioParameter.h
rename to media/libmediahelper/include/media/AudioParameter.h
index 24837e3..3c190f2 100644
--- a/media/libaudioclient/include/media/AudioParameter.h
+++ b/media/libmediahelper/include/media/AudioParameter.h
@@ -67,9 +67,12 @@
// keyAudioLanguagePreferred: Preferred audio language
static const char * const keyAudioLanguagePreferred;
- // keyStreamConnect / Disconnect: value is an int in audio_devices_t
- static const char * const keyStreamConnect;
- static const char * const keyStreamDisconnect;
+ // keyDeviceConnect / Disconnect: value is an int in audio_devices_t
+ static const char * const keyDeviceConnect;
+ static const char * const keyDeviceDisconnect;
+ // Need to be here because vendors still use them.
+ static const char * const keyStreamConnect; // Deprecated: DO NOT USE.
+ static const char * const keyStreamDisconnect; // Deprecated: DO NOT USE.
// For querying stream capabilities. All the returned values are lists.
// keyStreamSupportedFormats: audio_format_t
diff --git a/media/libmedia/include/media/TypeConverter.h b/media/libmediahelper/include/media/TypeConverter.h
similarity index 95%
rename from media/libmedia/include/media/TypeConverter.h
rename to media/libmediahelper/include/media/TypeConverter.h
index 2f8c209..011498a 100644
--- a/media/libmedia/include/media/TypeConverter.h
+++ b/media/libmediahelper/include/media/TypeConverter.h
@@ -17,10 +17,11 @@
#ifndef ANDROID_TYPE_CONVERTER_H_
#define ANDROID_TYPE_CONVERTER_H_
+#include <set>
#include <string>
#include <string.h>
-
#include <vector>
+
#include <system/audio.h>
#include <utils/Log.h>
#include <utils/Vector.h>
@@ -42,16 +43,6 @@
}
};
template <typename T>
-struct VectorTraits
-{
- typedef T Type;
- typedef Vector<Type> Collection;
- static void add(Collection &collection, Type value)
- {
- collection.add(value);
- }
-};
-template <typename T>
struct SortedVectorTraits
{
typedef T Type;
@@ -61,18 +52,28 @@
collection.add(value);
}
};
+template <typename T>
+struct SetTraits
+{
+ typedef T Type;
+ typedef std::set<Type> Collection;
+ static void add(Collection &collection, Type value)
+ {
+ collection.insert(value);
+ }
+};
-using SampleRateTraits = SortedVectorTraits<uint32_t>;
+using SampleRateTraits = SetTraits<uint32_t>;
using DeviceTraits = DefaultTraits<audio_devices_t>;
struct OutputDeviceTraits : public DeviceTraits {};
struct InputDeviceTraits : public DeviceTraits {};
-using ChannelTraits = SortedVectorTraits<audio_channel_mask_t>;
+using ChannelTraits = SetTraits<audio_channel_mask_t>;
struct OutputChannelTraits : public ChannelTraits {};
struct InputChannelTraits : public ChannelTraits {};
struct ChannelIndexTraits : public ChannelTraits {};
using InputFlagTraits = DefaultTraits<audio_input_flags_t>;
using OutputFlagTraits = DefaultTraits<audio_output_flags_t>;
-using FormatTraits = VectorTraits<audio_format_t>;
+using FormatTraits = DefaultTraits<audio_format_t>;
using GainModeTraits = DefaultTraits<audio_gain_mode_t>;
using StreamTraits = DefaultTraits<audio_stream_type_t>;
using AudioModeTraits = DefaultTraits<audio_mode_t>;
@@ -259,6 +260,7 @@
|| std::is_same<T, audio_source_t>::value
|| std::is_same<T, audio_stream_type_t>::value
|| std::is_same<T, audio_usage_t>::value
+ || std::is_same<T, audio_format_t>::value
, int> = 0>
static inline std::string toString(const T& value)
{
@@ -291,14 +293,6 @@
return result;
}
-// TODO: Remove when FormatTraits uses DefaultTraits.
-static inline std::string toString(const audio_format_t& format)
-{
- std::string result;
- return TypeConverter<VectorTraits<audio_format_t>>::toString(format, result)
- ? result : std::to_string(static_cast<int>(format));
-}
-
static inline std::string toString(const audio_attributes_t& attributes)
{
std::ostringstream result;
diff --git a/media/libmedia/include/media/convert.h b/media/libmediahelper/include/media/convert.h
similarity index 100%
rename from media/libmedia/include/media/convert.h
rename to media/libmediahelper/include/media/convert.h
diff --git a/media/libmediametrics/MediaAnalyticsItem.cpp b/media/libmediametrics/MediaAnalyticsItem.cpp
index 02c23b1..b7856a6 100644
--- a/media/libmediametrics/MediaAnalyticsItem.cpp
+++ b/media/libmediametrics/MediaAnalyticsItem.cpp
@@ -64,6 +64,16 @@
return item;
}
+MediaAnalyticsItem* MediaAnalyticsItem::convert(mediametrics_handle_t handle) {
+ MediaAnalyticsItem *item = (android::MediaAnalyticsItem *) handle;
+ return item;
+}
+
+mediametrics_handle_t MediaAnalyticsItem::convert(MediaAnalyticsItem *item ) {
+ mediametrics_handle_t handle = (mediametrics_handle_t) item;
+ return handle;
+}
+
// access functions for the class
MediaAnalyticsItem::MediaAnalyticsItem()
: mPid(-1),
diff --git a/media/libmediametrics/MediaMetrics.cpp b/media/libmediametrics/MediaMetrics.cpp
index 6109190..360ae0c 100644
--- a/media/libmediametrics/MediaMetrics.cpp
+++ b/media/libmediametrics/MediaMetrics.cpp
@@ -169,6 +169,11 @@
return item->selfrecord();
}
+mediametrics_handle_t mediametrics_dup(mediametrics_handle_t handle) {
+ android::MediaAnalyticsItem *item = (android::MediaAnalyticsItem *) handle;
+ if (item == NULL) return android::MediaAnalyticsItem::convert(item);
+ return android::MediaAnalyticsItem::convert(item->dup());
+}
const char *mediametrics_readable(mediametrics_handle_t handle) {
android::MediaAnalyticsItem *item = (android::MediaAnalyticsItem *) handle;
diff --git a/media/libmediametrics/include/MediaAnalyticsItem.h b/media/libmediametrics/include/MediaAnalyticsItem.h
index 4a36f6a..42a2f5b 100644
--- a/media/libmediametrics/include/MediaAnalyticsItem.h
+++ b/media/libmediametrics/include/MediaAnalyticsItem.h
@@ -17,6 +17,8 @@
#ifndef ANDROID_MEDIA_MEDIAANALYTICSITEM_H
#define ANDROID_MEDIA_MEDIAANALYTICSITEM_H
+#include "MediaMetrics.h"
+
#include <string>
#include <sys/types.h>
@@ -94,6 +96,9 @@
static MediaAnalyticsItem* create(Key key);
static MediaAnalyticsItem* create();
+ static MediaAnalyticsItem* convert(mediametrics_handle_t);
+ static mediametrics_handle_t convert(MediaAnalyticsItem *);
+
// access functions for the class
~MediaAnalyticsItem();
diff --git a/media/libmediametrics/include/MediaMetrics.h b/media/libmediametrics/include/MediaMetrics.h
index a4e1ed2..29fb241 100644
--- a/media/libmediametrics/include/MediaMetrics.h
+++ b/media/libmediametrics/include/MediaMetrics.h
@@ -79,6 +79,7 @@
// # of attributes set within this record.
int32_t mediametrics_count(mediametrics_handle_t handle);
+mediametrics_handle_t mediametrics_dup(mediametrics_handle_t handle);
bool mediametrics_selfRecord(mediametrics_handle_t handle);
const char *mediametrics_readable(mediametrics_handle_t handle);
diff --git a/media/libmediaplayer2/Android.bp b/media/libmediaplayer2/Android.bp
deleted file mode 100644
index dca6bb6..0000000
--- a/media/libmediaplayer2/Android.bp
+++ /dev/null
@@ -1,129 +0,0 @@
-cc_library_headers {
- name: "libmediaplayer2_headers",
- vendor_available: true,
- export_include_dirs: ["include"],
-}
-
-cc_library_static {
- name: "libmediaplayer2",
-
- srcs: [
- "MediaPlayer2AudioOutput.cpp",
- "mediaplayer2.cpp",
- ],
-
- shared_libs: [
- "libandroid_runtime",
- "libaudioclient",
- "libbinder",
- "libbinder_ndk",
- "libcutils",
- "libgui",
- "liblog",
- "libmedia_omx",
- "libui",
- "libutils",
-
- "libcrypto",
- "libmediametrics",
- "libmediandk",
- "libmediandk_utils",
- "libmediautils",
- "libmemunreachable",
- "libnativewindow",
- "libpowermanager",
- "libstagefright_httplive",
- ],
-
- export_shared_lib_headers: [
- "libaudioclient",
- "libbinder",
- "libgui",
- "libmedia_omx",
- ],
-
- header_libs: [
- "media_plugin_headers",
- ],
-
- include_dirs: [
- "frameworks/base/core/jni",
- ],
-
- static_libs: [
- "libmedia_helper",
- "libmediaplayer2-protos",
- "libmedia_player2_util",
- "libprotobuf-cpp-lite",
- "libstagefright_foundation_without_imemory",
- "libstagefright_nuplayer2",
- "libstagefright_player2",
- "libstagefright_rtsp",
- "libstagefright_timedtext2",
- "libmedia2_jni_core",
- ],
-
- export_include_dirs: [
- "include",
- ],
-
- cflags: [
- "-Werror",
- "-Wno-error=deprecated-declarations",
- "-Wall",
- ],
-
- sanitize: {
- misc_undefined: [
- "unsigned-integer-overflow",
- "signed-integer-overflow",
- ],
- cfi: true,
- },
-}
-
-cc_library {
- name: "libmedia2_jni_core",
-
- srcs: [
- "JavaVMHelper.cpp",
- "JAudioTrack.cpp",
- "JMedia2HTTPService.cpp",
- "JMedia2HTTPConnection.cpp",
- ],
-
- header_libs: [
- "libbinder_headers",
- "libnativehelper_header_only",
- ],
-
- shared_libs: [
- "liblog",
- "libutils",
- "libdl",
- ],
-
- include_dirs: [
- "frameworks/av/media/libmedia/include",
- "frameworks/base/core/jni",
- ],
-
- export_include_dirs: [
- "include",
- ],
-
- cflags: [
- "-Werror",
- "-Wno-error=deprecated-declarations",
- "-Wall",
- ],
-
- sanitize: {
- misc_undefined: [
- "unsigned-integer-overflow",
- "signed-integer-overflow",
- ],
- cfi: true,
- },
-
-}
diff --git a/media/libmediaplayer2/JAudioTrack.cpp b/media/libmediaplayer2/JAudioTrack.cpp
deleted file mode 100644
index fab6c64..0000000
--- a/media/libmediaplayer2/JAudioTrack.cpp
+++ /dev/null
@@ -1,768 +0,0 @@
-/*
- * Copyright 2018 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#define LOG_TAG "JAudioTrack"
-
-#include "media/JAudioAttributes.h"
-#include "media/JAudioFormat.h"
-#include "mediaplayer2/JAudioTrack.h"
-
-#include <android_media_AudioErrors.h>
-#include <mediaplayer2/JavaVMHelper.h>
-
-namespace android {
-
-// TODO: Store Java class/methodID as a member variable in the class.
-// TODO: Add NULL && Exception checks after every JNI call.
-JAudioTrack::JAudioTrack( // < Usages of the arguments are below >
- uint32_t sampleRate, // AudioFormat && bufferSizeInBytes
- audio_format_t format, // AudioFormat && bufferSizeInBytes
- audio_channel_mask_t channelMask, // AudioFormat && bufferSizeInBytes
- callback_t cbf, // Offload
- void* user, // Offload
- size_t frameCount, // bufferSizeInBytes
- int32_t sessionId, // AudioTrack
- const jobject attributes, // AudioAttributes
- float maxRequiredSpeed) { // bufferSizeInBytes
-
- JNIEnv *env = JavaVMHelper::getJNIEnv();
-
- jclass jAudioTrackCls = env->FindClass("android/media/AudioTrack");
- mAudioTrackCls = reinterpret_cast<jclass>(env->NewGlobalRef(jAudioTrackCls));
- env->DeleteLocalRef(jAudioTrackCls);
-
- maxRequiredSpeed = std::min(std::max(maxRequiredSpeed, 1.0f), AUDIO_TIMESTRETCH_SPEED_MAX);
-
- int bufferSizeInBytes = 0;
- if (sampleRate == 0 || frameCount > 0) {
- // Manually calculate buffer size.
- bufferSizeInBytes = audio_channel_count_from_out_mask(channelMask)
- * audio_bytes_per_sample(format) * (frameCount > 0 ? frameCount : 1);
- } else if (sampleRate > 0) {
- // Call Java AudioTrack::getMinBufferSize().
- jmethodID jGetMinBufferSize =
- env->GetStaticMethodID(mAudioTrackCls, "getMinBufferSize", "(III)I");
- bufferSizeInBytes = env->CallStaticIntMethod(mAudioTrackCls, jGetMinBufferSize,
- sampleRate, outChannelMaskFromNative(channelMask), audioFormatFromNative(format));
- }
- bufferSizeInBytes = (int) (bufferSizeInBytes * maxRequiredSpeed);
-
- // Create a Java AudioTrack object through its Builder.
- jclass jBuilderCls = env->FindClass("android/media/AudioTrack$Builder");
- jmethodID jBuilderCtor = env->GetMethodID(jBuilderCls, "<init>", "()V");
- jobject jBuilderObj = env->NewObject(jBuilderCls, jBuilderCtor);
-
- {
- sp<JObjectHolder> audioAttributesObj;
- if (attributes != NULL) {
- audioAttributesObj = new JObjectHolder(attributes);
- } else {
- audioAttributesObj = new JObjectHolder(
- JAudioAttributes::createAudioAttributesObj(env, NULL));
- }
- jmethodID jSetAudioAttributes = env->GetMethodID(jBuilderCls, "setAudioAttributes",
- "(Landroid/media/AudioAttributes;)Landroid/media/AudioTrack$Builder;");
- jBuilderObj = env->CallObjectMethod(jBuilderObj,
- jSetAudioAttributes, audioAttributesObj->getJObject());
- }
-
- jmethodID jSetAudioFormat = env->GetMethodID(jBuilderCls, "setAudioFormat",
- "(Landroid/media/AudioFormat;)Landroid/media/AudioTrack$Builder;");
- jBuilderObj = env->CallObjectMethod(jBuilderObj, jSetAudioFormat,
- JAudioFormat::createAudioFormatObj(env, sampleRate, format, channelMask));
-
- jmethodID jSetBufferSizeInBytes = env->GetMethodID(jBuilderCls, "setBufferSizeInBytes",
- "(I)Landroid/media/AudioTrack$Builder;");
- jBuilderObj = env->CallObjectMethod(jBuilderObj, jSetBufferSizeInBytes, bufferSizeInBytes);
-
- // We only use streaming mode of Java AudioTrack.
- jfieldID jModeStream = env->GetStaticFieldID(mAudioTrackCls, "MODE_STREAM", "I");
- jint transferMode = env->GetStaticIntField(mAudioTrackCls, jModeStream);
- jmethodID jSetTransferMode = env->GetMethodID(jBuilderCls, "setTransferMode",
- "(I)Landroid/media/AudioTrack$Builder;");
- jBuilderObj = env->CallObjectMethod(jBuilderObj, jSetTransferMode,
- transferMode /* Java AudioTrack::MODE_STREAM */);
-
- if (sessionId != 0) {
- jmethodID jSetSessionId = env->GetMethodID(jBuilderCls, "setSessionId",
- "(I)Landroid/media/AudioTrack$Builder;");
- jBuilderObj = env->CallObjectMethod(jBuilderObj, jSetSessionId, sessionId);
- }
-
- mFlags = AUDIO_OUTPUT_FLAG_NONE;
- if (cbf != NULL) {
- jmethodID jSetOffloadedPlayback = env->GetMethodID(jBuilderCls, "setOffloadedPlayback",
- "(Z)Landroid/media/AudioTrack$Builder;");
- jBuilderObj = env->CallObjectMethod(jBuilderObj, jSetOffloadedPlayback, true);
- mFlags = AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD;
- }
-
- jmethodID jBuild = env->GetMethodID(jBuilderCls, "build", "()Landroid/media/AudioTrack;");
- jobject jAudioTrackObj = env->CallObjectMethod(jBuilderObj, jBuild);
- mAudioTrackObj = reinterpret_cast<jobject>(env->NewGlobalRef(jAudioTrackObj));
- env->DeleteLocalRef(jBuilderObj);
-
- if (cbf != NULL) {
- // Set offload mode callback
- jobject jStreamEventCallbackObj = createStreamEventCallback(cbf, user);
- jobject jExecutorObj = createCallbackExecutor();
- jmethodID jSetStreamEventCallback = env->GetMethodID(
- jAudioTrackCls,
- "setStreamEventCallback",
- "(Ljava/util/concurrent/Executor;Landroid/media/AudioTrack$StreamEventCallback;)V");
- env->CallVoidMethod(
- mAudioTrackObj, jSetStreamEventCallback, jExecutorObj, jStreamEventCallbackObj);
- }
-}
-
-JAudioTrack::~JAudioTrack() {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- env->DeleteGlobalRef(mAudioTrackCls);
- env->DeleteGlobalRef(mAudioTrackObj);
-}
-
-size_t JAudioTrack::frameCount() {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jGetBufferSizeInFrames = env->GetMethodID(
- mAudioTrackCls, "getBufferSizeInFrames", "()I");
- return env->CallIntMethod(mAudioTrackObj, jGetBufferSizeInFrames);
-}
-
-size_t JAudioTrack::channelCount() {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jGetChannelCount = env->GetMethodID(mAudioTrackCls, "getChannelCount", "()I");
- return env->CallIntMethod(mAudioTrackObj, jGetChannelCount);
-}
-
-uint32_t JAudioTrack::latency() {
- // TODO: Currently hard-coded as returning zero.
- return 0;
-}
-
-status_t JAudioTrack::getPosition(uint32_t *position) {
- if (position == NULL) {
- return BAD_VALUE;
- }
-
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jGetPlaybackHeadPosition = env->GetMethodID(
- mAudioTrackCls, "getPlaybackHeadPosition", "()I");
- *position = env->CallIntMethod(mAudioTrackObj, jGetPlaybackHeadPosition);
-
- return NO_ERROR;
-}
-
-status_t JAudioTrack::getTimestamp(AudioTimestamp& timestamp) {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
-
- jclass jAudioTimeStampCls = env->FindClass("android/media/AudioTimestamp");
- jobject jAudioTimeStampObj = env->AllocObject(jAudioTimeStampCls);
-
- jfieldID jFramePosition = env->GetFieldID(jAudioTimeStampCls, "framePosition", "J");
- jfieldID jNanoTime = env->GetFieldID(jAudioTimeStampCls, "nanoTime", "J");
-
- jmethodID jGetTimestamp = env->GetMethodID(mAudioTrackCls,
- "getTimestamp", "(Landroid/media/AudioTimestamp;)Z");
- bool success = env->CallBooleanMethod(mAudioTrackObj, jGetTimestamp, jAudioTimeStampObj);
-
- if (!success) {
- return NO_INIT;
- }
-
- long long framePosition = env->GetLongField(jAudioTimeStampObj, jFramePosition);
- long long nanoTime = env->GetLongField(jAudioTimeStampObj, jNanoTime);
-
- struct timespec ts;
- const long long secondToNano = 1000000000LL; // 1E9
- ts.tv_sec = nanoTime / secondToNano;
- ts.tv_nsec = nanoTime % secondToNano;
- timestamp.mTime = ts;
- timestamp.mPosition = (uint32_t) framePosition;
-
- return NO_ERROR;
-}
-
-status_t JAudioTrack::getTimestamp(ExtendedTimestamp *timestamp __unused) {
- // TODO: Implement this after appropriate Java AudioTrack method is available.
- return NO_ERROR;
-}
-
-status_t JAudioTrack::setPlaybackRate(const AudioPlaybackRate &playbackRate) {
- // TODO: existing native AudioTrack returns INVALID_OPERATION on offload/direct/fast tracks.
- // Should we do the same thing?
- JNIEnv *env = JavaVMHelper::getJNIEnv();
-
- jclass jPlaybackParamsCls = env->FindClass("android/media/PlaybackParams");
- jmethodID jPlaybackParamsCtor = env->GetMethodID(jPlaybackParamsCls, "<init>", "()V");
- jobject jPlaybackParamsObj = env->NewObject(jPlaybackParamsCls, jPlaybackParamsCtor);
-
- jmethodID jSetAudioFallbackMode = env->GetMethodID(
- jPlaybackParamsCls, "setAudioFallbackMode", "(I)Landroid/media/PlaybackParams;");
- jPlaybackParamsObj = env->CallObjectMethod(
- jPlaybackParamsObj, jSetAudioFallbackMode, playbackRate.mFallbackMode);
-
- jmethodID jSetAudioStretchMode = env->GetMethodID(
- jPlaybackParamsCls, "setAudioStretchMode", "(I)Landroid/media/PlaybackParams;");
- jPlaybackParamsObj = env->CallObjectMethod(
- jPlaybackParamsObj, jSetAudioStretchMode, playbackRate.mStretchMode);
-
- jmethodID jSetPitch = env->GetMethodID(
- jPlaybackParamsCls, "setPitch", "(F)Landroid/media/PlaybackParams;");
- jPlaybackParamsObj = env->CallObjectMethod(jPlaybackParamsObj, jSetPitch, playbackRate.mPitch);
-
- jmethodID jSetSpeed = env->GetMethodID(
- jPlaybackParamsCls, "setSpeed", "(F)Landroid/media/PlaybackParams;");
- jPlaybackParamsObj = env->CallObjectMethod(jPlaybackParamsObj, jSetSpeed, playbackRate.mSpeed);
-
-
- // Set this Java PlaybackParams object into Java AudioTrack.
- jmethodID jSetPlaybackParams = env->GetMethodID(
- mAudioTrackCls, "setPlaybackParams", "(Landroid/media/PlaybackParams;)V");
- env->CallVoidMethod(mAudioTrackObj, jSetPlaybackParams, jPlaybackParamsObj);
- // TODO: Should we catch the Java IllegalArgumentException?
-
- return NO_ERROR;
-}
-
-const AudioPlaybackRate JAudioTrack::getPlaybackRate() {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
-
- jmethodID jGetPlaybackParams = env->GetMethodID(
- mAudioTrackCls, "getPlaybackParams", "()Landroid/media/PlaybackParams;");
- jobject jPlaybackParamsObj = env->CallObjectMethod(mAudioTrackObj, jGetPlaybackParams);
-
- AudioPlaybackRate playbackRate;
- jclass jPlaybackParamsCls = env->FindClass("android/media/PlaybackParams");
-
- jmethodID jGetAudioFallbackMode = env->GetMethodID(
- jPlaybackParamsCls, "getAudioFallbackMode", "()I");
- // TODO: Should we enable passing AUDIO_TIMESTRETCH_FALLBACK_CUT_REPEAT?
- // The enum is internal only, so it is not defined in PlaybackParmas.java.
- // TODO: Is this right way to convert an int to an enum?
- playbackRate.mFallbackMode = static_cast<AudioTimestretchFallbackMode>(
- env->CallIntMethod(jPlaybackParamsObj, jGetAudioFallbackMode));
-
- jmethodID jGetAudioStretchMode = env->GetMethodID(
- jPlaybackParamsCls, "getAudioStretchMode", "()I");
- playbackRate.mStretchMode = static_cast<AudioTimestretchStretchMode>(
- env->CallIntMethod(jPlaybackParamsObj, jGetAudioStretchMode));
-
- jmethodID jGetPitch = env->GetMethodID(jPlaybackParamsCls, "getPitch", "()F");
- playbackRate.mPitch = env->CallFloatMethod(jPlaybackParamsObj, jGetPitch);
-
- jmethodID jGetSpeed = env->GetMethodID(jPlaybackParamsCls, "getSpeed", "()F");
- playbackRate.mSpeed = env->CallFloatMethod(jPlaybackParamsObj, jGetSpeed);
-
- return playbackRate;
-}
-
-media::VolumeShaper::Status JAudioTrack::applyVolumeShaper(
- const sp<media::VolumeShaper::Configuration>& configuration,
- const sp<media::VolumeShaper::Operation>& operation) {
-
- jobject jConfigurationObj = createVolumeShaperConfigurationObj(configuration);
- jobject jOperationObj = createVolumeShaperOperationObj(operation);
-
- if (jConfigurationObj == NULL || jOperationObj == NULL) {
- return media::VolumeShaper::Status(BAD_VALUE);
- }
-
- JNIEnv *env = JavaVMHelper::getJNIEnv();
-
- jmethodID jCreateVolumeShaper = env->GetMethodID(mAudioTrackCls, "createVolumeShaper",
- "(Landroid/media/VolumeShaper$Configuration;)Landroid/media/VolumeShaper;");
- jobject jVolumeShaperObj = env->CallObjectMethod(
- mAudioTrackObj, jCreateVolumeShaper, jConfigurationObj);
-
- jclass jVolumeShaperCls = env->FindClass("android/media/VolumeShaper");
- jmethodID jApply = env->GetMethodID(jVolumeShaperCls, "apply",
- "(Landroid/media/VolumeShaper$Operation;)V");
- env->CallVoidMethod(jVolumeShaperObj, jApply, jOperationObj);
-
- return media::VolumeShaper::Status(NO_ERROR);
-}
-
-status_t JAudioTrack::setAuxEffectSendLevel(float level) {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jSetAuxEffectSendLevel = env->GetMethodID(
- mAudioTrackCls, "setAuxEffectSendLevel", "(F)I");
- int result = env->CallIntMethod(mAudioTrackObj, jSetAuxEffectSendLevel, level);
- return javaToNativeStatus(result);
-}
-
-status_t JAudioTrack::attachAuxEffect(int effectId) {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jAttachAuxEffect = env->GetMethodID(mAudioTrackCls, "attachAuxEffect", "(I)I");
- int result = env->CallIntMethod(mAudioTrackObj, jAttachAuxEffect, effectId);
- return javaToNativeStatus(result);
-}
-
-status_t JAudioTrack::setVolume(float left, float right) {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- // TODO: Java setStereoVolume is deprecated. Do we really need this method?
- jmethodID jSetStereoVolume = env->GetMethodID(mAudioTrackCls, "setStereoVolume", "(FF)I");
- int result = env->CallIntMethod(mAudioTrackObj, jSetStereoVolume, left, right);
- return javaToNativeStatus(result);
-}
-
-status_t JAudioTrack::setVolume(float volume) {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jSetVolume = env->GetMethodID(mAudioTrackCls, "setVolume", "(F)I");
- int result = env->CallIntMethod(mAudioTrackObj, jSetVolume, volume);
- return javaToNativeStatus(result);
-}
-
-status_t JAudioTrack::start() {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jPlay = env->GetMethodID(mAudioTrackCls, "play", "()V");
- // TODO: Should we catch the Java IllegalStateException from play()?
- env->CallVoidMethod(mAudioTrackObj, jPlay);
- return NO_ERROR;
-}
-
-ssize_t JAudioTrack::write(const void* buffer, size_t size, bool blocking) {
- if (buffer == NULL) {
- return BAD_VALUE;
- }
-
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jbyteArray jAudioData = env->NewByteArray(size);
- env->SetByteArrayRegion(jAudioData, 0, size, (jbyte *) buffer);
-
- jclass jByteBufferCls = env->FindClass("java/nio/ByteBuffer");
- jmethodID jWrap = env->GetStaticMethodID(jByteBufferCls, "wrap", "([B)Ljava/nio/ByteBuffer;");
- jobject jByteBufferObj = env->CallStaticObjectMethod(jByteBufferCls, jWrap, jAudioData);
-
- int writeMode = 0;
- if (blocking) {
- jfieldID jWriteBlocking = env->GetStaticFieldID(mAudioTrackCls, "WRITE_BLOCKING", "I");
- writeMode = env->GetStaticIntField(mAudioTrackCls, jWriteBlocking);
- } else {
- jfieldID jWriteNonBlocking = env->GetStaticFieldID(
- mAudioTrackCls, "WRITE_NON_BLOCKING", "I");
- writeMode = env->GetStaticIntField(mAudioTrackCls, jWriteNonBlocking);
- }
-
- jmethodID jWrite = env->GetMethodID(mAudioTrackCls, "write", "(Ljava/nio/ByteBuffer;II)I");
- int result = env->CallIntMethod(mAudioTrackObj, jWrite, jByteBufferObj, size, writeMode);
-
- if (result >= 0) {
- return result;
- } else {
- return javaToNativeStatus(result);
- }
-}
-
-void JAudioTrack::stop() {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jStop = env->GetMethodID(mAudioTrackCls, "stop", "()V");
- env->CallVoidMethod(mAudioTrackObj, jStop);
- // TODO: Should we catch IllegalStateException?
-}
-
-// TODO: Is the right implementation?
-bool JAudioTrack::stopped() const {
- return !isPlaying();
-}
-
-void JAudioTrack::flush() {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jFlush = env->GetMethodID(mAudioTrackCls, "flush", "()V");
- env->CallVoidMethod(mAudioTrackObj, jFlush);
-}
-
-void JAudioTrack::pause() {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jPause = env->GetMethodID(mAudioTrackCls, "pause", "()V");
- env->CallVoidMethod(mAudioTrackObj, jPause);
- // TODO: Should we catch IllegalStateException?
-}
-
-bool JAudioTrack::isPlaying() const {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jGetPlayState = env->GetMethodID(mAudioTrackCls, "getPlayState", "()I");
- int currentPlayState = env->CallIntMethod(mAudioTrackObj, jGetPlayState);
-
- // TODO: In Java AudioTrack, there is no STOPPING state.
- // This means while stopping, isPlaying() will return different value in two class.
- // - in existing native AudioTrack: true
- // - in JAudioTrack: false
- // If not okay, also modify the implementation of stopped().
- jfieldID jPlayStatePlaying = env->GetStaticFieldID(mAudioTrackCls, "PLAYSTATE_PLAYING", "I");
- int statePlaying = env->GetStaticIntField(mAudioTrackCls, jPlayStatePlaying);
- return currentPlayState == statePlaying;
-}
-
-uint32_t JAudioTrack::getSampleRate() {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jGetSampleRate = env->GetMethodID(mAudioTrackCls, "getSampleRate", "()I");
- return env->CallIntMethod(mAudioTrackObj, jGetSampleRate);
-}
-
-status_t JAudioTrack::getBufferDurationInUs(int64_t *duration) {
- if (duration == nullptr) {
- return BAD_VALUE;
- }
-
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jGetBufferSizeInFrames = env->GetMethodID(
- mAudioTrackCls, "getBufferSizeInFrames", "()I");
- int bufferSizeInFrames = env->CallIntMethod(mAudioTrackObj, jGetBufferSizeInFrames);
-
- const double secondToMicro = 1000000LL; // 1E6
- int sampleRate = JAudioTrack::getSampleRate();
- float speed = JAudioTrack::getPlaybackRate().mSpeed;
-
- *duration = (int64_t) (bufferSizeInFrames * secondToMicro / (sampleRate * speed));
- return NO_ERROR;
-}
-
-audio_format_t JAudioTrack::format() {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jGetAudioFormat = env->GetMethodID(mAudioTrackCls, "getAudioFormat", "()I");
- int javaFormat = env->CallIntMethod(mAudioTrackObj, jGetAudioFormat);
- return audioFormatToNative(javaFormat);
-}
-
-size_t JAudioTrack::frameSize() {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jGetFormat = env->GetMethodID(mAudioTrackCls,
- "getFormat", "()Landroid/media/AudioFormat;");
- jobject jAudioFormatObj = env->CallObjectMethod(mAudioTrackObj, jGetFormat);
-
- jclass jAudioFormatCls = env->FindClass("android/media/AudioFormat");
- jmethodID jGetFrameSizeInBytes = env->GetMethodID(
- jAudioFormatCls, "getFrameSizeInBytes", "()I");
- jint javaFrameSizeInBytes = env->CallIntMethod(jAudioFormatObj, jGetFrameSizeInBytes);
-
- return (size_t)javaFrameSizeInBytes;
-}
-
-status_t JAudioTrack::dump(int fd, const Vector<String16>& args __unused) const
-{
- String8 result;
-
- result.append(" JAudioTrack::dump\n");
-
- // TODO: Remove logs that includes unavailable information from below.
-// result.appendFormat(" status(%d), state(%d), session Id(%d), flags(%#x)\n",
-// mStatus, mState, mSessionId, mFlags);
-// result.appendFormat(" format(%#x), channel mask(%#x), channel count(%u)\n",
-// format(), mChannelMask, channelCount());
-// result.appendFormat(" sample rate(%u), original sample rate(%u), speed(%f)\n",
-// getSampleRate(), mOriginalSampleRate, mPlaybackRate.mSpeed);
-// result.appendFormat(" frame count(%zu), req. frame count(%zu)\n",
-// frameCount(), mReqFrameCount);
-// result.appendFormat(" notif. frame count(%u), req. notif. frame count(%u),"
-// " req. notif. per buff(%u)\n",
-// mNotificationFramesAct, mNotificationFramesReq, mNotificationsPerBufferReq);
-// result.appendFormat(" latency (%d), selected device Id(%d), routed device Id(%d)\n",
-// latency(), mSelectedDeviceId, getRoutedDeviceId());
-// result.appendFormat(" output(%d) AF latency (%u) AF frame count(%zu) AF SampleRate(%u)\n",
-// mOutput, mAfLatency, mAfFrameCount, mAfSampleRate);
- ::write(fd, result.string(), result.size());
- return NO_ERROR;
-}
-
-jobject JAudioTrack::getRoutedDevice() {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jGetRoutedDevice = env->GetMethodID(mAudioTrackCls, "getRoutedDevice",
- "()Landroid/media/AudioDeviceInfo;");
- return env->CallObjectMethod(mAudioTrackObj, jGetRoutedDevice);
-}
-
-int32_t JAudioTrack::getAudioSessionId() {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jGetAudioSessionId = env->GetMethodID(mAudioTrackCls, "getAudioSessionId", "()I");
- jint sessionId = env->CallIntMethod(mAudioTrackObj, jGetAudioSessionId);
- return sessionId;
-}
-
-status_t JAudioTrack::setPreferredDevice(jobject device) {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jSetPreferredDeviceId = env->GetMethodID(mAudioTrackCls, "setPreferredDevice",
- "(Landroid/media/AudioDeviceInfo;)Z");
- jboolean result = env->CallBooleanMethod(mAudioTrackObj, jSetPreferredDeviceId, device);
- return result == true ? NO_ERROR : BAD_VALUE;
-}
-
-audio_stream_type_t JAudioTrack::getAudioStreamType() {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jGetAudioAttributes = env->GetMethodID(mAudioTrackCls, "getAudioAttributes",
- "()Landroid/media/AudioAttributes;");
- jobject jAudioAttributes = env->CallObjectMethod(mAudioTrackObj, jGetAudioAttributes);
- jclass jAudioAttributesCls = env->FindClass("android/media/AudioAttributes");
- jmethodID jGetVolumeControlStream = env->GetMethodID(jAudioAttributesCls,
- "getVolumeControlStream", "()I");
- int javaAudioStreamType = env->CallIntMethod(jAudioAttributes, jGetVolumeControlStream);
- return (audio_stream_type_t)javaAudioStreamType;
-}
-
-status_t JAudioTrack::pendingDuration(int32_t *msec) {
- if (msec == nullptr) {
- return BAD_VALUE;
- }
-
- bool isPurePcmData = audio_is_linear_pcm(format()) && (getFlags() & AUDIO_FLAG_HW_AV_SYNC) == 0;
- if (!isPurePcmData) {
- return INVALID_OPERATION;
- }
-
- // TODO: Need to know the difference btw. client and server time.
- // If getTimestamp(ExtendedTimestamp) is ready, and un-comment below and modify appropriately.
- // (copied from AudioTrack.cpp)
-
-// ExtendedTimestamp ets;
-// ExtendedTimestamp::LOCATION location = ExtendedTimestamp::LOCATION_SERVER;
-// if (getTimestamp_l(&ets) == OK && ets.mTimeNs[location] > 0) {
-// int64_t diff = ets.mPosition[ExtendedTimestamp::LOCATION_CLIENT]
-// - ets.mPosition[location];
-// if (diff < 0) {
-// *msec = 0;
-// } else {
-// // ms is the playback time by frames
-// int64_t ms = (int64_t)((double)diff * 1000 /
-// ((double)mSampleRate * mPlaybackRate.mSpeed));
-// // clockdiff is the timestamp age (negative)
-// int64_t clockdiff = (mState != STATE_ACTIVE) ? 0 :
-// ets.mTimeNs[location]
-// + ets.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_MONOTONIC]
-// - systemTime(SYSTEM_TIME_MONOTONIC);
-//
-// //ALOGV("ms: %lld clockdiff: %lld", (long long)ms, (long long)clockdiff);
-// static const int NANOS_PER_MILLIS = 1000000;
-// *msec = (int32_t)(ms + clockdiff / NANOS_PER_MILLIS);
-// }
-// return NO_ERROR;
-// }
-
- return NO_ERROR;
-}
-
-status_t JAudioTrack::addAudioDeviceCallback(jobject listener, jobject handler) {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jAddOnRoutingChangedListener = env->GetMethodID(mAudioTrackCls,
- "addOnRoutingChangedListener",
- "(Landroid/media/AudioRouting$OnRoutingChangedListener;Landroid/os/Handler;)V");
- env->CallVoidMethod(mAudioTrackObj, jAddOnRoutingChangedListener, listener, handler);
- return NO_ERROR;
-}
-
-status_t JAudioTrack::removeAudioDeviceCallback(jobject listener) {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jRemoveOnRoutingChangedListener = env->GetMethodID(mAudioTrackCls,
- "removeOnRoutingChangedListener",
- "(Landroid/media/AudioRouting$OnRoutingChangedListener;)V");
- env->CallVoidMethod(mAudioTrackObj, jRemoveOnRoutingChangedListener, listener);
- return NO_ERROR;
-}
-
-void JAudioTrack::registerRoutingDelegates(
- Vector<std::pair<sp<JObjectHolder>, sp<JObjectHolder>>>& routingDelegates) {
- for (auto it = routingDelegates.begin(); it != routingDelegates.end(); it++) {
- addAudioDeviceCallback(it->second->getJObject(), getHandler(it->second->getJObject()));
- }
-}
-
-/////////////////////////////////////////////////////////////
-/// Static methods begin ///
-/////////////////////////////////////////////////////////////
-jobject JAudioTrack::getListener(const jobject routingDelegateObj) {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jclass jRoutingDelegateCls = env->FindClass("android/media/RoutingDelegate");
- jmethodID jGetListener = env->GetMethodID(jRoutingDelegateCls,
- "getListener", "()Landroid/media/AudioRouting$OnRoutingChangedListener;");
- return env->CallObjectMethod(routingDelegateObj, jGetListener);
-}
-
-jobject JAudioTrack::getHandler(const jobject routingDelegateObj) {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jclass jRoutingDelegateCls = env->FindClass("android/media/RoutingDelegate");
- jmethodID jGetHandler = env->GetMethodID(jRoutingDelegateCls,
- "getHandler", "()Landroid/os/Handler;");
- return env->CallObjectMethod(routingDelegateObj, jGetHandler);
-}
-
-jobject JAudioTrack::findByKey(
- Vector<std::pair<sp<JObjectHolder>, sp<JObjectHolder>>>& mp, const jobject key) {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- for (auto it = mp.begin(); it != mp.end(); it++) {
- if (env->IsSameObject(it->first->getJObject(), key)) {
- return it->second->getJObject();
- }
- }
- return nullptr;
-}
-
-void JAudioTrack::eraseByKey(
- Vector<std::pair<sp<JObjectHolder>, sp<JObjectHolder>>>& mp, const jobject key) {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- for (auto it = mp.begin(); it != mp.end(); it++) {
- if (env->IsSameObject(it->first->getJObject(), key)) {
- mp.erase(it);
- return;
- }
- }
-}
-
-/////////////////////////////////////////////////////////////
-/// Private method begins ///
-/////////////////////////////////////////////////////////////
-
-jobject JAudioTrack::createVolumeShaperConfigurationObj(
- const sp<media::VolumeShaper::Configuration>& config) {
-
- // TODO: Java VolumeShaper's setId() / setOptionFlags() are hidden.
- if (config == NULL || config->getType() == media::VolumeShaper::Configuration::TYPE_ID) {
- return NULL;
- }
-
- JNIEnv *env = JavaVMHelper::getJNIEnv();
-
- // Referenced "android_media_VolumeShaper.h".
- jfloatArray xarray = nullptr;
- jfloatArray yarray = nullptr;
- if (config->getType() == media::VolumeShaper::Configuration::TYPE_SCALE) {
- // convert curve arrays
- xarray = env->NewFloatArray(config->size());
- yarray = env->NewFloatArray(config->size());
- float * const x = env->GetFloatArrayElements(xarray, nullptr /* isCopy */);
- float * const y = env->GetFloatArrayElements(yarray, nullptr /* isCopy */);
- float *xptr = x, *yptr = y;
- for (const auto &pt : *config.get()) {
- *xptr++ = pt.first;
- *yptr++ = pt.second;
- }
- env->ReleaseFloatArrayElements(xarray, x, 0 /* mode */);
- env->ReleaseFloatArrayElements(yarray, y, 0 /* mode */);
- }
-
- jclass jBuilderCls = env->FindClass("android/media/VolumeShaper$Configuration$Builder");
- jmethodID jBuilderCtor = env->GetMethodID(jBuilderCls, "<init>", "()V");
- jobject jBuilderObj = env->NewObject(jBuilderCls, jBuilderCtor);
-
- jmethodID jSetDuration = env->GetMethodID(jBuilderCls, "setDuration",
- "(L)Landroid/media/VolumeShaper$Configuration$Builder;");
- jBuilderObj = env->CallObjectMethod(jBuilderCls, jSetDuration, (jlong) config->getDurationMs());
-
- jmethodID jSetInterpolatorType = env->GetMethodID(jBuilderCls, "setInterpolatorType",
- "(I)Landroid/media/VolumeShaper$Configuration$Builder;");
- jBuilderObj = env->CallObjectMethod(jBuilderCls, jSetInterpolatorType,
- config->getInterpolatorType());
-
- jmethodID jSetCurve = env->GetMethodID(jBuilderCls, "setCurve",
- "([F[F)Landroid/media/VolumeShaper$Configuration$Builder;");
- jBuilderObj = env->CallObjectMethod(jBuilderCls, jSetCurve, xarray, yarray);
-
- jmethodID jBuild = env->GetMethodID(jBuilderCls, "build",
- "()Landroid/media/VolumeShaper$Configuration;");
- return env->CallObjectMethod(jBuilderObj, jBuild);
-}
-
-jobject JAudioTrack::createVolumeShaperOperationObj(
- const sp<media::VolumeShaper::Operation>& operation) {
-
- JNIEnv *env = JavaVMHelper::getJNIEnv();
-
- jclass jBuilderCls = env->FindClass("android/media/VolumeShaper$Operation$Builder");
- jmethodID jBuilderCtor = env->GetMethodID(jBuilderCls, "<init>", "()V");
- jobject jBuilderObj = env->NewObject(jBuilderCls, jBuilderCtor);
-
- // Set XOffset
- jmethodID jSetXOffset = env->GetMethodID(jBuilderCls, "setXOffset",
- "(F)Landroid/media/VolumeShaper$Operation$Builder;");
- jBuilderObj = env->CallObjectMethod(jBuilderCls, jSetXOffset, operation->getXOffset());
-
- int32_t flags = operation->getFlags();
-
- if (operation->getReplaceId() >= 0) {
- jmethodID jReplace = env->GetMethodID(jBuilderCls, "replace",
- "(IB)Landroid/media/VolumeShaper$Operation$Builder;");
- bool join = (flags | media::VolumeShaper::Operation::FLAG_JOIN) != 0;
- jBuilderObj = env->CallObjectMethod(jBuilderCls, jReplace, operation->getReplaceId(), join);
- }
-
- if (flags | media::VolumeShaper::Operation::FLAG_REVERSE) {
- jmethodID jReverse = env->GetMethodID(jBuilderCls, "reverse",
- "()Landroid/media/VolumeShaper$Operation$Builder;");
- jBuilderObj = env->CallObjectMethod(jBuilderCls, jReverse);
- }
-
- // TODO: VolumeShaper Javadoc says "Do not call terminate() directly". Can we call this?
- if (flags | media::VolumeShaper::Operation::FLAG_TERMINATE) {
- jmethodID jTerminate = env->GetMethodID(jBuilderCls, "terminate",
- "()Landroid/media/VolumeShaper$Operation$Builder;");
- jBuilderObj = env->CallObjectMethod(jBuilderCls, jTerminate);
- }
-
- if (flags | media::VolumeShaper::Operation::FLAG_DELAY) {
- jmethodID jDefer = env->GetMethodID(jBuilderCls, "defer",
- "()Landroid/media/VolumeShaper$Operation$Builder;");
- jBuilderObj = env->CallObjectMethod(jBuilderCls, jDefer);
- }
-
- if (flags | media::VolumeShaper::Operation::FLAG_CREATE_IF_NECESSARY) {
- jmethodID jCreateIfNeeded = env->GetMethodID(jBuilderCls, "createIfNeeded",
- "()Landroid/media/VolumeShaper$Operation$Builder;");
- jBuilderObj = env->CallObjectMethod(jBuilderCls, jCreateIfNeeded);
- }
-
- // TODO: Handle error case (can it be NULL?)
- jmethodID jBuild = env->GetMethodID(jBuilderCls, "build",
- "()Landroid/media/VolumeShaper$Operation;");
- return env->CallObjectMethod(jBuilderObj, jBuild);
-}
-
-jobject JAudioTrack::createStreamEventCallback(callback_t cbf, void* user) {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jclass jCallbackCls = env->FindClass("android/media/MediaPlayer2$StreamEventCallback");
- jmethodID jCallbackCtor = env->GetMethodID(jCallbackCls, "<init>", "(JJJ)V");
- jobject jCallbackObj = env->NewObject(jCallbackCls, jCallbackCtor, this, cbf, user);
- return jCallbackObj;
-}
-
-jobject JAudioTrack::createCallbackExecutor() {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jclass jExecutorsCls = env->FindClass("java/util/concurrent/Executors");
- jmethodID jNewSingleThreadExecutor = env->GetStaticMethodID(jExecutorsCls,
- "newSingleThreadExecutor", "()Ljava/util/concurrent/ExecutorService;");
- jobject jSingleThreadExecutorObj =
- env->CallStaticObjectMethod(jExecutorsCls, jNewSingleThreadExecutor);
- return jSingleThreadExecutorObj;
-}
-
-status_t JAudioTrack::javaToNativeStatus(int javaStatus) {
- switch (javaStatus) {
- case AUDIO_JAVA_SUCCESS:
- return NO_ERROR;
- case AUDIO_JAVA_BAD_VALUE:
- return BAD_VALUE;
- case AUDIO_JAVA_INVALID_OPERATION:
- return INVALID_OPERATION;
- case AUDIO_JAVA_PERMISSION_DENIED:
- return PERMISSION_DENIED;
- case AUDIO_JAVA_NO_INIT:
- return NO_INIT;
- case AUDIO_JAVA_WOULD_BLOCK:
- return WOULD_BLOCK;
- case AUDIO_JAVA_DEAD_OBJECT:
- return DEAD_OBJECT;
- default:
- return UNKNOWN_ERROR;
- }
-}
-
-} // namespace android
diff --git a/media/libmediaplayer2/JMedia2HTTPConnection.cpp b/media/libmediaplayer2/JMedia2HTTPConnection.cpp
deleted file mode 100644
index e1baa10..0000000
--- a/media/libmediaplayer2/JMedia2HTTPConnection.cpp
+++ /dev/null
@@ -1,179 +0,0 @@
-/*
- * Copyright 2017, The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "JMedia2HTTPConnection"
-#include <utils/Log.h>
-
-#include <mediaplayer2/JavaVMHelper.h>
-#include <mediaplayer2/JMedia2HTTPConnection.h>
-#include <media/stagefright/foundation/ADebug.h>
-#include <nativehelper/scoped_local_ref.h>
-
-#include "log/log.h"
-#include "jni.h"
-
-namespace android {
-
-static const size_t kBufferSize = 32768;
-
-JMedia2HTTPConnection::JMedia2HTTPConnection(JNIEnv *env, jobject thiz) {
- mMedia2HTTPConnectionObj = env->NewGlobalRef(thiz);
- CHECK(mMedia2HTTPConnectionObj != NULL);
-
- ScopedLocalRef<jclass> media2HTTPConnectionClass(
- env, env->GetObjectClass(mMedia2HTTPConnectionObj));
- CHECK(media2HTTPConnectionClass.get() != NULL);
-
- mConnectMethod = env->GetMethodID(
- media2HTTPConnectionClass.get(),
- "connect",
- "(Ljava/lang/String;Ljava/lang/String;)Z");
- CHECK(mConnectMethod != NULL);
-
- mDisconnectMethod = env->GetMethodID(
- media2HTTPConnectionClass.get(),
- "disconnect",
- "()V");
- CHECK(mDisconnectMethod != NULL);
-
- mReadAtMethod = env->GetMethodID(
- media2HTTPConnectionClass.get(),
- "readAt",
- "(J[BI)I");
- CHECK(mReadAtMethod != NULL);
-
- mGetSizeMethod = env->GetMethodID(
- media2HTTPConnectionClass.get(),
- "getSize",
- "()J");
- CHECK(mGetSizeMethod != NULL);
-
- mGetMIMETypeMethod = env->GetMethodID(
- media2HTTPConnectionClass.get(),
- "getMIMEType",
- "()Ljava/lang/String;");
- CHECK(mGetMIMETypeMethod != NULL);
-
- mGetUriMethod = env->GetMethodID(
- media2HTTPConnectionClass.get(),
- "getUri",
- "()Ljava/lang/String;");
- CHECK(mGetUriMethod != NULL);
-
- ScopedLocalRef<jbyteArray> tmp(
- env, env->NewByteArray(kBufferSize));
- mByteArrayObj = (jbyteArray)env->NewGlobalRef(tmp.get());
- CHECK(mByteArrayObj != NULL);
-}
-
-JMedia2HTTPConnection::~JMedia2HTTPConnection() {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- env->DeleteGlobalRef(mMedia2HTTPConnectionObj);
- env->DeleteGlobalRef(mByteArrayObj);
-}
-
-bool JMedia2HTTPConnection::connect(
- const char *uri, const KeyedVector<String8, String8> *headers) {
- String8 tmp("");
- if (headers != NULL) {
- for (size_t i = 0; i < headers->size(); ++i) {
- tmp.append(headers->keyAt(i));
- tmp.append(String8(": "));
- tmp.append(headers->valueAt(i));
- tmp.append(String8("\r\n"));
- }
- }
-
- JNIEnv* env = JavaVMHelper::getJNIEnv();
- jstring juri = env->NewStringUTF(uri);
- jstring jheaders = env->NewStringUTF(tmp.string());
-
- jboolean ret =
- env->CallBooleanMethod(mMedia2HTTPConnectionObj, mConnectMethod, juri, jheaders);
-
- env->DeleteLocalRef(juri);
- env->DeleteLocalRef(jheaders);
-
- return (bool)ret;
-}
-
-void JMedia2HTTPConnection::disconnect() {
- JNIEnv* env = JavaVMHelper::getJNIEnv();
- env->CallVoidMethod(mMedia2HTTPConnectionObj, mDisconnectMethod);
-}
-
-ssize_t JMedia2HTTPConnection::readAt(off64_t offset, void *data, size_t size) {
- JNIEnv* env = JavaVMHelper::getJNIEnv();
-
- if (size > kBufferSize) {
- size = kBufferSize;
- }
-
- jint n = env->CallIntMethod(
- mMedia2HTTPConnectionObj, mReadAtMethod, (jlong)offset, mByteArrayObj, (jint)size);
-
- if (n > 0) {
- env->GetByteArrayRegion(
- mByteArrayObj,
- 0,
- n,
- (jbyte *)data);
- }
-
- return n;
-}
-
-off64_t JMedia2HTTPConnection::getSize() {
- JNIEnv* env = JavaVMHelper::getJNIEnv();
- return (off64_t)(env->CallLongMethod(mMedia2HTTPConnectionObj, mGetSizeMethod));
-}
-
-status_t JMedia2HTTPConnection::getMIMEType(String8 *mimeType) {
- JNIEnv* env = JavaVMHelper::getJNIEnv();
- jstring jmime = (jstring)env->CallObjectMethod(mMedia2HTTPConnectionObj, mGetMIMETypeMethod);
- jboolean flag = env->ExceptionCheck();
- if (flag) {
- env->ExceptionClear();
- return UNKNOWN_ERROR;
- }
-
- const char *str = env->GetStringUTFChars(jmime, 0);
- if (str != NULL) {
- *mimeType = String8(str);
- } else {
- *mimeType = "application/octet-stream";
- }
- env->ReleaseStringUTFChars(jmime, str);
- return OK;
-}
-
-status_t JMedia2HTTPConnection::getUri(String8 *uri) {
- JNIEnv* env = JavaVMHelper::getJNIEnv();
- jstring juri = (jstring)env->CallObjectMethod(mMedia2HTTPConnectionObj, mGetUriMethod);
- jboolean flag = env->ExceptionCheck();
- if (flag) {
- env->ExceptionClear();
- return UNKNOWN_ERROR;
- }
-
- const char *str = env->GetStringUTFChars(juri, 0);
- *uri = String8(str);
- env->ReleaseStringUTFChars(juri, str);
- return OK;
-}
-
-} // namespace android
diff --git a/media/libmediaplayer2/JMedia2HTTPService.cpp b/media/libmediaplayer2/JMedia2HTTPService.cpp
deleted file mode 100644
index 20e3573..0000000
--- a/media/libmediaplayer2/JMedia2HTTPService.cpp
+++ /dev/null
@@ -1,59 +0,0 @@
-/*
- * Copyright 2017, The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "JMedia2HTTPService"
-#include <utils/Log.h>
-
-#include <jni.h>
-
-#include <mediaplayer2/JavaVMHelper.h>
-#include <mediaplayer2/JMedia2HTTPService.h>
-#include <mediaplayer2/JMedia2HTTPConnection.h>
-#include <media/stagefright/foundation/ADebug.h>
-
-#include <nativehelper/scoped_local_ref.h>
-
-namespace android {
-
-JMedia2HTTPService::JMedia2HTTPService(JNIEnv *env, jobject thiz) {
- mMedia2HTTPServiceObj = env->NewGlobalRef(thiz);
- CHECK(mMedia2HTTPServiceObj != NULL);
-
- ScopedLocalRef<jclass> media2HTTPServiceClass(env, env->GetObjectClass(mMedia2HTTPServiceObj));
- CHECK(media2HTTPServiceClass.get() != NULL);
-
- mMakeHTTPConnectionMethod = env->GetMethodID(
- media2HTTPServiceClass.get(),
- "makeHTTPConnection",
- "()Landroid/media/Media2HTTPConnection;");
- CHECK(mMakeHTTPConnectionMethod != NULL);
-}
-
-JMedia2HTTPService::~JMedia2HTTPService() {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- env->DeleteGlobalRef(mMedia2HTTPServiceObj);
-}
-
-sp<MediaHTTPConnection> JMedia2HTTPService::makeHTTPConnection() {
- JNIEnv* env = JavaVMHelper::getJNIEnv();
- jobject media2HTTPConnectionObj =
- env->CallObjectMethod(mMedia2HTTPServiceObj, mMakeHTTPConnectionMethod);
-
- return new JMedia2HTTPConnection(env, media2HTTPConnectionObj);
-}
-
-} // namespace android
diff --git a/media/libmediaplayer2/JavaVMHelper.cpp b/media/libmediaplayer2/JavaVMHelper.cpp
deleted file mode 100644
index 8d03ed0..0000000
--- a/media/libmediaplayer2/JavaVMHelper.cpp
+++ /dev/null
@@ -1,162 +0,0 @@
-/*
- * Copyright 2018 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#define LOG_TAG "JavaVMHelper"
-
-#include "mediaplayer2/JavaVMHelper.h"
-
-#include <media/stagefright/foundation/ADebug.h>
-#include <utils/threads.h>
-
-#include <stdlib.h>
-
-namespace android {
-
-// static
-std::atomic<JavaVM *> JavaVMHelper::sJavaVM(NULL);
-
-/*
- * Makes the current thread visible to the VM.
- *
- * The JNIEnv pointer returned is only valid for the current thread, and
- * thus must be tucked into thread-local storage.
- */
-static int javaAttachThread(const char* threadName, JNIEnv** pEnv) {
- JavaVMAttachArgs args;
- JavaVM* vm;
- jint result;
-
- vm = JavaVMHelper::getJavaVM();
- if (vm == NULL) {
- return JNI_ERR;
- }
-
- args.version = JNI_VERSION_1_4;
- args.name = (char*) threadName;
- args.group = NULL;
-
- result = vm->AttachCurrentThread(pEnv, (void*) &args);
- if (result != JNI_OK) {
- ALOGI("NOTE: attach of thread '%s' failed\n", threadName);
- }
-
- return result;
-}
-
-/*
- * Detach the current thread from the set visible to the VM.
- */
-static int javaDetachThread(void) {
- JavaVM* vm;
- jint result;
-
- vm = JavaVMHelper::getJavaVM();
- if (vm == NULL) {
- return JNI_ERR;
- }
-
- result = vm->DetachCurrentThread();
- if (result != JNI_OK) {
- ALOGE("ERROR: thread detach failed\n");
- }
- return result;
-}
-
-/*
- * When starting a native thread that will be visible from the VM, we
- * bounce through this to get the right attach/detach action.
- * Note that this function calls free(args)
- */
-static int javaThreadShell(void* args) {
- void* start = ((void**)args)[0];
- void* userData = ((void **)args)[1];
- char* name = (char*) ((void **)args)[2]; // we own this storage
- free(args);
- JNIEnv* env;
- int result;
-
- /* hook us into the VM */
- if (javaAttachThread(name, &env) != JNI_OK) {
- return -1;
- }
-
- /* start the thread running */
- result = (*(android_thread_func_t)start)(userData);
-
- /* unhook us */
- javaDetachThread();
- free(name);
-
- return result;
-}
-
-/*
- * This is invoked from androidCreateThreadEtc() via the callback
- * set with androidSetCreateThreadFunc().
- *
- * We need to create the new thread in such a way that it gets hooked
- * into the VM before it really starts executing.
- */
-static int javaCreateThreadEtc(
- android_thread_func_t entryFunction,
- void* userData,
- const char* threadName,
- int32_t threadPriority,
- size_t threadStackSize,
- android_thread_id_t* threadId) {
- void** args = (void**) malloc(3 * sizeof(void*)); // javaThreadShell must free
- int result;
-
- LOG_ALWAYS_FATAL_IF(threadName == nullptr, "threadName not provided to javaCreateThreadEtc");
-
- args[0] = (void*) entryFunction;
- args[1] = userData;
- args[2] = (void*) strdup(threadName); // javaThreadShell must free
-
- result = androidCreateRawThreadEtc(javaThreadShell, args,
- threadName, threadPriority, threadStackSize, threadId);
- return result;
-}
-
-// static
-JNIEnv *JavaVMHelper::getJNIEnv() {
- JNIEnv *env;
- JavaVM *vm = sJavaVM.load();
- CHECK(vm != NULL);
-
- if (vm->GetEnv((void **)&env, JNI_VERSION_1_4) != JNI_OK) {
- return NULL;
- }
-
- return env;
-}
-
-//static
-JavaVM *JavaVMHelper::getJavaVM() {
- return sJavaVM.load();
-}
-
-// static
-void JavaVMHelper::setJavaVM(JavaVM *vm) {
- sJavaVM.store(vm);
-
- // Ensure that Thread(/*canCallJava*/ true) in libutils is attached to the VM.
- // This is supposed to be done by runtime, but when libutils is used with linker
- // namespace, CreateThreadFunc should be initialized separately within the namespace.
- androidSetCreateThreadFunc((android_create_thread_fn) javaCreateThreadEtc);
-}
-
-} // namespace android
diff --git a/media/libmediaplayer2/MediaPlayer2AudioOutput.cpp b/media/libmediaplayer2/MediaPlayer2AudioOutput.cpp
deleted file mode 100644
index b4fa0c1..0000000
--- a/media/libmediaplayer2/MediaPlayer2AudioOutput.cpp
+++ /dev/null
@@ -1,656 +0,0 @@
-/*
-**
-** Copyright 2018, The Android Open Source Project
-**
-** Licensed under the Apache License, Version 2.0 (the "License");
-** you may not use this file except in compliance with the License.
-** You may obtain a copy of the License at
-**
-** http://www.apache.org/licenses/LICENSE-2.0
-**
-** Unless required by applicable law or agreed to in writing, software
-** distributed under the License is distributed on an "AS IS" BASIS,
-** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-** See the License for the specific language governing permissions and
-** limitations under the License.
-*/
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "MediaPlayer2AudioOutput"
-#include <mediaplayer2/MediaPlayer2AudioOutput.h>
-
-#include <cutils/properties.h> // for property_get
-#include <utils/Log.h>
-
-#include <media/stagefright/foundation/ADebug.h>
-
-namespace {
-
-const float kMaxRequiredSpeed = 8.0f; // for PCM tracks allow up to 8x speedup.
-
-} // anonymous namespace
-
-namespace android {
-
-// TODO: Find real cause of Audio/Video delay in PV framework and remove this workaround
-/* static */ int MediaPlayer2AudioOutput::mMinBufferCount = 4;
-/* static */ bool MediaPlayer2AudioOutput::mIsOnEmulator = false;
-
-status_t MediaPlayer2AudioOutput::dump(int fd, const Vector<String16>& args) const {
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- result.append(" MediaPlayer2AudioOutput\n");
- snprintf(buffer, 255, " volume(%f)\n", mVolume);
- result.append(buffer);
- snprintf(buffer, 255, " msec per frame(%f), latency (%d)\n",
- mMsecsPerFrame, (mJAudioTrack != nullptr) ? mJAudioTrack->latency() : -1);
- result.append(buffer);
- snprintf(buffer, 255, " aux effect id(%d), send level (%f)\n",
- mAuxEffectId, mSendLevel);
- result.append(buffer);
-
- ::write(fd, result.string(), result.size());
- if (mJAudioTrack != nullptr) {
- mJAudioTrack->dump(fd, args);
- }
- return NO_ERROR;
-}
-
-MediaPlayer2AudioOutput::MediaPlayer2AudioOutput(int32_t sessionId, uid_t uid, int pid,
- const jobject attributes)
- : mCallback(nullptr),
- mCallbackCookie(nullptr),
- mCallbackData(nullptr),
- mVolume(1.0),
- mPlaybackRate(AUDIO_PLAYBACK_RATE_DEFAULT),
- mSampleRateHz(0),
- mMsecsPerFrame(0),
- mFrameSize(0),
- mSessionId(sessionId),
- mUid(uid),
- mPid(pid),
- mSendLevel(0.0),
- mAuxEffectId(0),
- mFlags(AUDIO_OUTPUT_FLAG_NONE) {
- ALOGV("MediaPlayer2AudioOutput(%d)", sessionId);
-
- if (attributes != nullptr) {
- mAttributes = new JObjectHolder(attributes);
- }
-
- setMinBufferCount();
- mRoutingDelegates.clear();
-}
-
-MediaPlayer2AudioOutput::~MediaPlayer2AudioOutput() {
- close();
- delete mCallbackData;
-}
-
-//static
-void MediaPlayer2AudioOutput::setMinBufferCount() {
- char value[PROPERTY_VALUE_MAX];
- if (property_get("ro.kernel.qemu", value, 0)) {
- mIsOnEmulator = true;
- mMinBufferCount = 12; // to prevent systematic buffer underrun for emulator
- }
-}
-
-// static
-bool MediaPlayer2AudioOutput::isOnEmulator() {
- setMinBufferCount(); // benign race wrt other threads
- return mIsOnEmulator;
-}
-
-// static
-int MediaPlayer2AudioOutput::getMinBufferCount() {
- setMinBufferCount(); // benign race wrt other threads
- return mMinBufferCount;
-}
-
-ssize_t MediaPlayer2AudioOutput::bufferSize() const {
- Mutex::Autolock lock(mLock);
- if (mJAudioTrack == nullptr) {
- return NO_INIT;
- }
- return mJAudioTrack->frameCount() * mFrameSize;
-}
-
-ssize_t MediaPlayer2AudioOutput::frameCount() const {
- Mutex::Autolock lock(mLock);
- if (mJAudioTrack == nullptr) {
- return NO_INIT;
- }
- return mJAudioTrack->frameCount();
-}
-
-ssize_t MediaPlayer2AudioOutput::channelCount() const {
- Mutex::Autolock lock(mLock);
- if (mJAudioTrack == nullptr) {
- return NO_INIT;
- }
- return mJAudioTrack->channelCount();
-}
-
-ssize_t MediaPlayer2AudioOutput::frameSize() const {
- Mutex::Autolock lock(mLock);
- if (mJAudioTrack == nullptr) {
- return NO_INIT;
- }
- return mFrameSize;
-}
-
-uint32_t MediaPlayer2AudioOutput::latency () const {
- Mutex::Autolock lock(mLock);
- if (mJAudioTrack == nullptr) {
- return 0;
- }
- return mJAudioTrack->latency();
-}
-
-float MediaPlayer2AudioOutput::msecsPerFrame() const {
- Mutex::Autolock lock(mLock);
- return mMsecsPerFrame;
-}
-
-status_t MediaPlayer2AudioOutput::getPosition(uint32_t *position) const {
- Mutex::Autolock lock(mLock);
- if (mJAudioTrack == nullptr) {
- return NO_INIT;
- }
- return mJAudioTrack->getPosition(position);
-}
-
-status_t MediaPlayer2AudioOutput::getTimestamp(AudioTimestamp &ts) const {
- Mutex::Autolock lock(mLock);
- if (mJAudioTrack == nullptr) {
- return NO_INIT;
- }
- return mJAudioTrack->getTimestamp(ts);
-}
-
-// TODO: Remove unnecessary calls to getPlayedOutDurationUs()
-// as it acquires locks and may query the audio driver.
-//
-// Some calls could conceivably retrieve extrapolated data instead of
-// accessing getTimestamp() or getPosition() every time a data buffer with
-// a media time is received.
-//
-// Calculate duration of played samples if played at normal rate (i.e., 1.0).
-int64_t MediaPlayer2AudioOutput::getPlayedOutDurationUs(int64_t nowUs) const {
- Mutex::Autolock lock(mLock);
- if (mJAudioTrack == nullptr || mSampleRateHz == 0) {
- return 0;
- }
-
- uint32_t numFramesPlayed;
- int64_t numFramesPlayedAtUs;
- AudioTimestamp ts;
-
- status_t res = mJAudioTrack->getTimestamp(ts);
-
- if (res == OK) { // case 1: mixing audio tracks and offloaded tracks.
- numFramesPlayed = ts.mPosition;
- numFramesPlayedAtUs = ts.mTime.tv_sec * 1000000LL + ts.mTime.tv_nsec / 1000;
- //ALOGD("getTimestamp: OK %d %lld", numFramesPlayed, (long long)numFramesPlayedAtUs);
- } else { // case 2: transitory state on start of a new track
- // case 3: transitory at new track or audio fast tracks.
- numFramesPlayed = 0;
- numFramesPlayedAtUs = nowUs;
- //ALOGD("getTimestamp: WOULD_BLOCK %d %lld",
- // numFramesPlayed, (long long)numFramesPlayedAtUs);
- }
-
- // CHECK_EQ(numFramesPlayed & (1 << 31), 0); // can't be negative until 12.4 hrs, test
- // TODO: remove the (int32_t) casting below as it may overflow at 12.4 hours.
- int64_t durationUs = (int64_t)((int32_t)numFramesPlayed * 1000000LL / mSampleRateHz)
- + nowUs - numFramesPlayedAtUs;
- if (durationUs < 0) {
- // Occurs when numFramesPlayed position is very small and the following:
- // (1) In case 1, the time nowUs is computed before getTimestamp() is called and
- // numFramesPlayedAtUs is greater than nowUs by time more than numFramesPlayed.
- // (2) In case 3, using getPosition and adding mAudioSink->latency() to
- // numFramesPlayedAtUs, by a time amount greater than numFramesPlayed.
- //
- // Both of these are transitory conditions.
- ALOGV("getPlayedOutDurationUs: negative duration %lld set to zero", (long long)durationUs);
- durationUs = 0;
- }
- ALOGV("getPlayedOutDurationUs(%lld) nowUs(%lld) frames(%u) framesAt(%lld)",
- (long long)durationUs, (long long)nowUs,
- numFramesPlayed, (long long)numFramesPlayedAtUs);
- return durationUs;
-}
-
-status_t MediaPlayer2AudioOutput::getFramesWritten(uint32_t *frameswritten) const {
- Mutex::Autolock lock(mLock);
- if (mJAudioTrack == nullptr) {
- return NO_INIT;
- }
- ExtendedTimestamp ets;
- status_t status = mJAudioTrack->getTimestamp(&ets);
- if (status == OK || status == WOULD_BLOCK) {
- *frameswritten = (uint32_t)ets.mPosition[ExtendedTimestamp::LOCATION_CLIENT];
- }
- return status;
-}
-
-void MediaPlayer2AudioOutput::setAudioAttributes(const jobject attributes) {
- Mutex::Autolock lock(mLock);
- mAttributes = (attributes == nullptr) ? nullptr : new JObjectHolder(attributes);
-}
-
-audio_stream_type_t MediaPlayer2AudioOutput::getAudioStreamType() const {
- ALOGV("getAudioStreamType");
- Mutex::Autolock lock(mLock);
- if (mJAudioTrack == nullptr) {
- return AUDIO_STREAM_DEFAULT;
- }
- return mJAudioTrack->getAudioStreamType();
-}
-
-void MediaPlayer2AudioOutput::close_l() {
- mJAudioTrack.clear();
-}
-
-status_t MediaPlayer2AudioOutput::open(
- uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
- audio_format_t format,
- AudioCallback cb, void *cookie,
- audio_output_flags_t flags,
- const audio_offload_info_t *offloadInfo,
- uint32_t suggestedFrameCount) {
- ALOGV("open(%u, %d, 0x%x, 0x%x, %d 0x%x)", sampleRate, channelCount, channelMask,
- format, mSessionId, flags);
-
- // offloading is only supported in callback mode for now.
- // offloadInfo must be present if offload flag is set
- if (((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) &&
- ((cb == nullptr) || (offloadInfo == nullptr))) {
- return BAD_VALUE;
- }
-
- // compute frame count for the AudioTrack internal buffer
- const size_t frameCount =
- ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) ? 0 : suggestedFrameCount;
-
- if (channelMask == CHANNEL_MASK_USE_CHANNEL_ORDER) {
- channelMask = audio_channel_out_mask_from_count(channelCount);
- if (0 == channelMask) {
- ALOGE("open() error, can\'t derive mask for %d audio channels", channelCount);
- return NO_INIT;
- }
- }
-
- Mutex::Autolock lock(mLock);
- mCallback = cb;
- mCallbackCookie = cookie;
-
- sp<JAudioTrack> jT;
- CallbackData *newcbd = nullptr;
-
- ALOGV("creating new JAudioTrack");
-
- if (mCallback != nullptr) {
- newcbd = new CallbackData(this);
- jT = new JAudioTrack(
- sampleRate,
- format,
- channelMask,
- CallbackWrapper,
- newcbd,
- frameCount,
- mSessionId,
- mAttributes != nullptr ? mAttributes->getJObject() : nullptr,
- 1.0f); // default value for maxRequiredSpeed
- } else {
- // TODO: Due to buffer memory concerns, we use a max target playback speed
- // based on mPlaybackRate at the time of open (instead of kMaxRequiredSpeed),
- // also clamping the target speed to 1.0 <= targetSpeed <= kMaxRequiredSpeed.
- const float targetSpeed =
- std::min(std::max(mPlaybackRate.mSpeed, 1.0f), kMaxRequiredSpeed);
- ALOGW_IF(targetSpeed != mPlaybackRate.mSpeed,
- "track target speed:%f clamped from playback speed:%f",
- targetSpeed, mPlaybackRate.mSpeed);
- jT = new JAudioTrack(
- sampleRate,
- format,
- channelMask,
- nullptr,
- nullptr,
- frameCount,
- mSessionId,
- mAttributes != nullptr ? mAttributes->getJObject() : nullptr,
- targetSpeed);
- }
-
- if (jT == 0) {
- ALOGE("Unable to create audio track");
- delete newcbd;
- // t goes out of scope, so reference count drops to zero
- return NO_INIT;
- }
-
- CHECK((jT != nullptr) && ((mCallback == nullptr) || (newcbd != nullptr)));
-
- mCallbackData = newcbd;
- ALOGV("setVolume");
- jT->setVolume(mVolume);
-
- mSampleRateHz = sampleRate;
- mFlags = flags;
- mMsecsPerFrame = 1E3f / (mPlaybackRate.mSpeed * sampleRate);
- mFrameSize = jT->frameSize();
- mJAudioTrack = jT;
-
- return updateTrack_l();
-}
-
-status_t MediaPlayer2AudioOutput::updateTrack_l() {
- if (mJAudioTrack == nullptr) {
- return NO_ERROR;
- }
-
- status_t res = NO_ERROR;
- // Note some output devices may give us a direct track even though we don't specify it.
- // Example: Line application b/17459982.
- if ((mJAudioTrack->getFlags()
- & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT)) == 0) {
- res = mJAudioTrack->setPlaybackRate(mPlaybackRate);
- if (res == NO_ERROR) {
- mJAudioTrack->setAuxEffectSendLevel(mSendLevel);
- res = mJAudioTrack->attachAuxEffect(mAuxEffectId);
- }
- }
- if (mPreferredDevice != nullptr) {
- mJAudioTrack->setPreferredDevice(mPreferredDevice->getJObject());
- }
-
- mJAudioTrack->registerRoutingDelegates(mRoutingDelegates);
-
- ALOGV("updateTrack_l() DONE status %d", res);
- return res;
-}
-
-status_t MediaPlayer2AudioOutput::start() {
- ALOGV("start");
- Mutex::Autolock lock(mLock);
- if (mCallbackData != nullptr) {
- mCallbackData->endTrackSwitch();
- }
- if (mJAudioTrack != nullptr) {
- mJAudioTrack->setVolume(mVolume);
- mJAudioTrack->setAuxEffectSendLevel(mSendLevel);
- status_t status = mJAudioTrack->start();
- return status;
- }
- return NO_INIT;
-}
-
-ssize_t MediaPlayer2AudioOutput::write(const void* buffer, size_t size, bool blocking) {
- Mutex::Autolock lock(mLock);
- LOG_ALWAYS_FATAL_IF(mCallback != nullptr, "Don't call write if supplying a callback.");
-
- //ALOGV("write(%p, %u)", buffer, size);
- if (mJAudioTrack != nullptr) {
- return mJAudioTrack->write(buffer, size, blocking);
- }
- return NO_INIT;
-}
-
-void MediaPlayer2AudioOutput::stop() {
- ALOGV("stop");
- Mutex::Autolock lock(mLock);
- if (mJAudioTrack != nullptr) {
- mJAudioTrack->stop();
- }
-}
-
-void MediaPlayer2AudioOutput::flush() {
- ALOGV("flush");
- Mutex::Autolock lock(mLock);
- if (mJAudioTrack != nullptr) {
- mJAudioTrack->flush();
- }
-}
-
-void MediaPlayer2AudioOutput::pause() {
- ALOGV("pause");
- Mutex::Autolock lock(mLock);
- if (mJAudioTrack != nullptr) {
- mJAudioTrack->pause();
- }
-}
-
-void MediaPlayer2AudioOutput::close() {
- ALOGV("close");
- sp<JAudioTrack> track;
- {
- Mutex::Autolock lock(mLock);
- track = mJAudioTrack;
- close_l(); // clears mJAudioTrack
- }
- // destruction of the track occurs outside of mutex.
-}
-
-void MediaPlayer2AudioOutput::setVolume(float volume) {
- ALOGV("setVolume(%f)", volume);
- Mutex::Autolock lock(mLock);
- mVolume = volume;
- if (mJAudioTrack != nullptr) {
- mJAudioTrack->setVolume(volume);
- }
-}
-
-status_t MediaPlayer2AudioOutput::setPlaybackRate(const AudioPlaybackRate &rate) {
- ALOGV("setPlaybackRate(%f %f %d %d)",
- rate.mSpeed, rate.mPitch, rate.mFallbackMode, rate.mStretchMode);
- Mutex::Autolock lock(mLock);
- if (mJAudioTrack == 0) {
- // remember rate so that we can set it when the track is opened
- mPlaybackRate = rate;
- return OK;
- }
- status_t res = mJAudioTrack->setPlaybackRate(rate);
- if (res != NO_ERROR) {
- return res;
- }
- // rate.mSpeed is always greater than 0 if setPlaybackRate succeeded
- CHECK_GT(rate.mSpeed, 0.f);
- mPlaybackRate = rate;
- if (mSampleRateHz != 0) {
- mMsecsPerFrame = 1E3f / (rate.mSpeed * mSampleRateHz);
- }
- return res;
-}
-
-status_t MediaPlayer2AudioOutput::getPlaybackRate(AudioPlaybackRate *rate) {
- ALOGV("getPlaybackRate");
- Mutex::Autolock lock(mLock);
- if (mJAudioTrack == 0) {
- return NO_INIT;
- }
- *rate = mJAudioTrack->getPlaybackRate();
- return NO_ERROR;
-}
-
-status_t MediaPlayer2AudioOutput::setAuxEffectSendLevel(float level) {
- ALOGV("setAuxEffectSendLevel(%f)", level);
- Mutex::Autolock lock(mLock);
- mSendLevel = level;
- if (mJAudioTrack != nullptr) {
- return mJAudioTrack->setAuxEffectSendLevel(level);
- }
- return NO_ERROR;
-}
-
-status_t MediaPlayer2AudioOutput::attachAuxEffect(int effectId) {
- ALOGV("attachAuxEffect(%d)", effectId);
- Mutex::Autolock lock(mLock);
- mAuxEffectId = effectId;
- if (mJAudioTrack != nullptr) {
- return mJAudioTrack->attachAuxEffect(effectId);
- }
- return NO_ERROR;
-}
-
-status_t MediaPlayer2AudioOutput::setPreferredDevice(jobject device) {
- ALOGV("setPreferredDevice");
- Mutex::Autolock lock(mLock);
- status_t ret = NO_ERROR;
- if (mJAudioTrack != nullptr) {
- ret = mJAudioTrack->setPreferredDevice(device);
- }
- if (ret == NO_ERROR) {
- mPreferredDevice = new JObjectHolder(device);
- }
- return ret;
-}
-
-jobject MediaPlayer2AudioOutput::getRoutedDevice() {
- ALOGV("getRoutedDevice");
- Mutex::Autolock lock(mLock);
- if (mJAudioTrack != nullptr) {
- return mJAudioTrack->getRoutedDevice();
- }
- return nullptr;
-}
-
-status_t MediaPlayer2AudioOutput::addAudioDeviceCallback(jobject jRoutingDelegate) {
- ALOGV("addAudioDeviceCallback");
- Mutex::Autolock lock(mLock);
- jobject listener = JAudioTrack::getListener(jRoutingDelegate);
- if (JAudioTrack::findByKey(mRoutingDelegates, listener) == nullptr) {
- sp<JObjectHolder> listenerHolder = new JObjectHolder(listener);
- jobject handler = JAudioTrack::getHandler(jRoutingDelegate);
- sp<JObjectHolder> routingDelegateHolder = new JObjectHolder(jRoutingDelegate);
-
- mRoutingDelegates.push_back(std::pair<sp<JObjectHolder>, sp<JObjectHolder>>(
- listenerHolder, routingDelegateHolder));
-
- if (mJAudioTrack != nullptr) {
- return mJAudioTrack->addAudioDeviceCallback(
- routingDelegateHolder->getJObject(), handler);
- }
- }
- return NO_ERROR;
-}
-
-status_t MediaPlayer2AudioOutput::removeAudioDeviceCallback(jobject listener) {
- ALOGV("removeAudioDeviceCallback");
- Mutex::Autolock lock(mLock);
- jobject routingDelegate = nullptr;
- if ((routingDelegate = JAudioTrack::findByKey(mRoutingDelegates, listener)) != nullptr) {
- if (mJAudioTrack != nullptr) {
- mJAudioTrack->removeAudioDeviceCallback(routingDelegate);
- }
- JAudioTrack::eraseByKey(mRoutingDelegates, listener);
- }
- return NO_ERROR;
-}
-
-// static
-void MediaPlayer2AudioOutput::CallbackWrapper(
- int event, void *cookie, void *info) {
- //ALOGV("callbackwrapper");
- CallbackData *data = (CallbackData*)cookie;
- // lock to ensure we aren't caught in the middle of a track switch.
- data->lock();
- MediaPlayer2AudioOutput *me = data->getOutput();
- JAudioTrack::Buffer *buffer = (JAudioTrack::Buffer *)info;
- if (me == nullptr) {
- // no output set, likely because the track was scheduled to be reused
- // by another player, but the format turned out to be incompatible.
- data->unlock();
- if (buffer != nullptr) {
- buffer->mSize = 0;
- }
- return;
- }
-
- switch(event) {
- case JAudioTrack::EVENT_MORE_DATA: {
- size_t actualSize = (*me->mCallback)(
- me, buffer->mData, buffer->mSize, me->mCallbackCookie,
- CB_EVENT_FILL_BUFFER);
-
- // Log when no data is returned from the callback.
- // (1) We may have no data (especially with network streaming sources).
- // (2) We may have reached the EOS and the audio track is not stopped yet.
- // Note that AwesomePlayer/AudioPlayer will only return zero size when it reaches the EOS.
- // NuPlayer2Renderer will return zero when it doesn't have data (it doesn't block to fill).
- //
- // This is a benign busy-wait, with the next data request generated 10 ms or more later;
- // nevertheless for power reasons, we don't want to see too many of these.
-
- ALOGV_IF(actualSize == 0 && buffer->mSize > 0, "callbackwrapper: empty buffer returned");
-
- buffer->mSize = actualSize;
- } break;
-
- case JAudioTrack::EVENT_STREAM_END:
- // currently only occurs for offloaded callbacks
- ALOGV("callbackwrapper: deliver EVENT_STREAM_END");
- (*me->mCallback)(me, nullptr /* buffer */, 0 /* size */,
- me->mCallbackCookie, CB_EVENT_STREAM_END);
- break;
-
- case JAudioTrack::EVENT_NEW_IAUDIOTRACK :
- ALOGV("callbackwrapper: deliver EVENT_TEAR_DOWN");
- (*me->mCallback)(me, nullptr /* buffer */, 0 /* size */,
- me->mCallbackCookie, CB_EVENT_TEAR_DOWN);
- break;
-
- case JAudioTrack::EVENT_UNDERRUN:
- // This occurs when there is no data available, typically
- // when there is a failure to supply data to the AudioTrack. It can also
- // occur in non-offloaded mode when the audio device comes out of standby.
- //
- // If an AudioTrack underruns it outputs silence. Since this happens suddenly
- // it may sound like an audible pop or glitch.
- //
- // The underrun event is sent once per track underrun; the condition is reset
- // when more data is sent to the AudioTrack.
- ALOGD("callbackwrapper: EVENT_UNDERRUN (discarded)");
- break;
-
- default:
- ALOGE("received unknown event type: %d inside CallbackWrapper !", event);
- }
-
- data->unlock();
-}
-
-int32_t MediaPlayer2AudioOutput::getSessionId() const {
- Mutex::Autolock lock(mLock);
- return mSessionId;
-}
-
-void MediaPlayer2AudioOutput::setSessionId(const int32_t sessionId) {
- Mutex::Autolock lock(mLock);
- mSessionId = sessionId;
-}
-
-uint32_t MediaPlayer2AudioOutput::getSampleRate() const {
- Mutex::Autolock lock(mLock);
- if (mJAudioTrack == 0) {
- return 0;
- }
- return mJAudioTrack->getSampleRate();
-}
-
-int64_t MediaPlayer2AudioOutput::getBufferDurationInUs() const {
- Mutex::Autolock lock(mLock);
- if (mJAudioTrack == 0) {
- return 0;
- }
- int64_t duration;
- if (mJAudioTrack->getBufferDurationInUs(&duration) != OK) {
- return 0;
- }
- return duration;
-}
-
-} // namespace android
diff --git a/media/libmediaplayer2/include/mediaplayer2/JAudioTrack.h b/media/libmediaplayer2/include/mediaplayer2/JAudioTrack.h
deleted file mode 100644
index 2ed4632..0000000
--- a/media/libmediaplayer2/include/mediaplayer2/JAudioTrack.h
+++ /dev/null
@@ -1,461 +0,0 @@
-/*
- * Copyright 2018 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_JAUDIOTRACK_H
-#define ANDROID_JAUDIOTRACK_H
-
-#include <utility>
-#include <jni.h>
-#include <media/AudioResamplerPublic.h>
-#include <media/AudioSystem.h>
-#include <media/VolumeShaper.h>
-#include <system/audio.h>
-#include <utils/Errors.h>
-#include <utils/Vector.h>
-#include <mediaplayer2/JObjectHolder.h>
-#include <media/AudioTimestamp.h> // It has dependency on audio.h/Errors.h, but doesn't
- // include them in it. Therefore it is included here at last.
-
-namespace android {
-
-class JAudioTrack : public RefBase {
-public:
-
- /* Events used by AudioTrack callback function (callback_t).
- * Keep in sync with frameworks/base/media/java/android/media/AudioTrack.java NATIVE_EVENT_*.
- */
- enum event_type {
- EVENT_MORE_DATA = 0, // Request to write more data to buffer.
- EVENT_UNDERRUN = 1, // Buffer underrun occurred. This will not occur for
- // static tracks.
- EVENT_NEW_IAUDIOTRACK = 6, // IAudioTrack was re-created, either due to re-routing and
- // voluntary invalidation by mediaserver, or mediaserver crash.
- EVENT_STREAM_END = 7, // Sent after all the buffers queued in AF and HW are played
- // back (after stop is called) for an offloaded track.
- };
-
- class Buffer
- {
- public:
- size_t mSize; // input/output in bytes.
- void* mData; // pointer to the audio data.
- };
-
- /* As a convenience, if a callback is supplied, a handler thread
- * is automatically created with the appropriate priority. This thread
- * invokes the callback when a new buffer becomes available or various conditions occur.
- *
- * Parameters:
- *
- * event: type of event notified (see enum AudioTrack::event_type).
- * user: Pointer to context for use by the callback receiver.
- * info: Pointer to optional parameter according to event type:
- * - EVENT_MORE_DATA: pointer to JAudioTrack::Buffer struct. The callback must not
- * write more bytes than indicated by 'size' field and update 'size' if fewer bytes
- * are written.
- * - EVENT_NEW_IAUDIOTRACK: unused.
- * - EVENT_STREAM_END: unused.
- */
-
- typedef void (*callback_t)(int event, void* user, void *info);
-
- /* Creates an JAudioTrack object for non-offload mode.
- * Once created, the track needs to be started before it can be used.
- * Unspecified values are set to appropriate default values.
- *
- * Parameters:
- *
- * streamType: Select the type of audio stream this track is attached to
- * (e.g. AUDIO_STREAM_MUSIC).
- * sampleRate: Data source sampling rate in Hz. Zero means to use the sink sample rate.
- * A non-zero value must be specified if AUDIO_OUTPUT_FLAG_DIRECT is set.
- * 0 will not work with current policy implementation for direct output
- * selection where an exact match is needed for sampling rate.
- * (TODO: Check direct output after flags can be used in Java AudioTrack.)
- * format: Audio format. For mixed tracks, any PCM format supported by server is OK.
- * For direct and offloaded tracks, the possible format(s) depends on the
- * output sink.
- * (TODO: How can we check whether a format is supported?)
- * channelMask: Channel mask, such that audio_is_output_channel(channelMask) is true.
- * cbf: Callback function. If not null, this function is called periodically
- * to provide new data and inform of marker, position updates, etc.
- * user: Context for use by the callback receiver.
- * frameCount: Minimum size of track PCM buffer in frames. This defines the
- * application's contribution to the latency of the track.
- * The actual size selected by the JAudioTrack could be larger if the
- * requested size is not compatible with current audio HAL configuration.
- * Zero means to use a default value.
- * sessionId: Specific session ID, or zero to use default.
- * pAttributes: If not NULL, supersedes streamType for use case selection.
- * maxRequiredSpeed: For PCM tracks, this creates an appropriate buffer size that will allow
- * maxRequiredSpeed playback. Values less than 1.0f and greater than
- * AUDIO_TIMESTRETCH_SPEED_MAX will be clamped. For non-PCM tracks
- * and direct or offloaded tracks, this parameter is ignored.
- * (TODO: Handle this after offload / direct track is supported.)
- *
- * TODO: Revive removed arguments after offload mode is supported.
- */
- JAudioTrack(uint32_t sampleRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- callback_t cbf,
- void* user,
- size_t frameCount = 0,
- int32_t sessionId = AUDIO_SESSION_ALLOCATE,
- const jobject pAttributes = NULL,
- float maxRequiredSpeed = 1.0f);
-
- /*
- // Q. May be used in AudioTrack.setPreferredDevice(AudioDeviceInfo)?
- audio_port_handle_t selectedDeviceId,
-
- // TODO: No place to use these values.
- int32_t notificationFrames,
- const audio_offload_info_t *offloadInfo,
- */
-
- virtual ~JAudioTrack();
-
- size_t frameCount();
- size_t channelCount();
-
- /* Returns this track's estimated latency in milliseconds.
- * This includes the latency due to AudioTrack buffer size, AudioMixer (if any)
- * and audio hardware driver.
- */
- uint32_t latency();
-
- /* Return the total number of frames played since playback start.
- * The counter will wrap (overflow) periodically, e.g. every ~27 hours at 44.1 kHz.
- * It is reset to zero by flush(), reload(), and stop().
- *
- * Parameters:
- *
- * position: Address where to return play head position.
- *
- * Returned status (from utils/Errors.h) can be:
- * - NO_ERROR: successful operation
- * - BAD_VALUE: position is NULL
- */
- status_t getPosition(uint32_t *position);
-
- // TODO: Does this comment apply same to Java AudioTrack::getTimestamp?
- // Changed the return type from status_t to bool, since Java AudioTrack::getTimestamp returns
- // boolean. Will Java getTimestampWithStatus() be public?
- /* Poll for a timestamp on demand.
- * Use if EVENT_NEW_TIMESTAMP is not delivered often enough for your needs,
- * or if you need to get the most recent timestamp outside of the event callback handler.
- * Caution: calling this method too often may be inefficient;
- * if you need a high resolution mapping between frame position and presentation time,
- * consider implementing that at application level, based on the low resolution timestamps.
- * Returns NO_ERROR if timestamp is valid.
- * NO_INIT if finds error, and timestamp parameter will be undefined on return.
- */
- status_t getTimestamp(AudioTimestamp& timestamp);
-
- // TODO: This doc is just copied from AudioTrack.h. Revise it after implemenation.
- /* Return the extended timestamp, with additional timebase info and improved drain behavior.
- *
- * This is similar to the AudioTrack.java API:
- * getTimestamp(@NonNull AudioTimestamp timestamp, @AudioTimestamp.Timebase int timebase)
- *
- * Some differences between this method and the getTimestamp(AudioTimestamp& timestamp) method
- *
- * 1. stop() by itself does not reset the frame position.
- * A following start() resets the frame position to 0.
- * 2. flush() by itself does not reset the frame position.
- * The frame position advances by the number of frames flushed,
- * when the first frame after flush reaches the audio sink.
- * 3. BOOTTIME clock offsets are provided to help synchronize with
- * non-audio streams, e.g. sensor data.
- * 4. Position is returned with 64 bits of resolution.
- *
- * Parameters:
- * timestamp: A pointer to the caller allocated ExtendedTimestamp.
- *
- * Returns NO_ERROR on success; timestamp is filled with valid data.
- * BAD_VALUE if timestamp is NULL.
- * WOULD_BLOCK if called immediately after start() when the number
- * of frames consumed is less than the
- * overall hardware latency to physical output. In WOULD_BLOCK cases,
- * one might poll again, or use getPosition(), or use 0 position and
- * current time for the timestamp.
- * If WOULD_BLOCK is returned, the timestamp is still
- * modified with the LOCATION_CLIENT portion filled.
- * DEAD_OBJECT if AudioFlinger dies or the output device changes and
- * the track cannot be automatically restored.
- * The application needs to recreate the AudioTrack
- * because the audio device changed or AudioFlinger died.
- * This typically occurs for direct or offloaded tracks
- * or if mDoNotReconnect is true.
- * INVALID_OPERATION if called on a offloaded or direct track.
- * Use getTimestamp(AudioTimestamp& timestamp) instead.
- */
- status_t getTimestamp(ExtendedTimestamp *timestamp);
-
- /* Set source playback rate for timestretch
- * 1.0 is normal speed: < 1.0 is slower, > 1.0 is faster
- * 1.0 is normal pitch: < 1.0 is lower pitch, > 1.0 is higher pitch
- *
- * AUDIO_TIMESTRETCH_SPEED_MIN <= speed <= AUDIO_TIMESTRETCH_SPEED_MAX
- * AUDIO_TIMESTRETCH_PITCH_MIN <= pitch <= AUDIO_TIMESTRETCH_PITCH_MAX
- *
- * Speed increases the playback rate of media, but does not alter pitch.
- * Pitch increases the "tonal frequency" of media, but does not affect the playback rate.
- */
- status_t setPlaybackRate(const AudioPlaybackRate &playbackRate);
-
- /* Return current playback rate */
- const AudioPlaybackRate getPlaybackRate();
-
- /* Sets the volume shaper object */
- media::VolumeShaper::Status applyVolumeShaper(
- const sp<media::VolumeShaper::Configuration>& configuration,
- const sp<media::VolumeShaper::Operation>& operation);
-
- /* Set the send level for this track. An auxiliary effect should be attached
- * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0.
- */
- status_t setAuxEffectSendLevel(float level);
-
- /* Attach track auxiliary output to specified effect. Use effectId = 0
- * to detach track from effect.
- *
- * Parameters:
- *
- * effectId: effectId obtained from AudioEffect::id().
- *
- * Returned status (from utils/Errors.h) can be:
- * - NO_ERROR: successful operation
- * - INVALID_OPERATION: The effect is not an auxiliary effect.
- * - BAD_VALUE: The specified effect ID is invalid.
- */
- status_t attachAuxEffect(int effectId);
-
- /* Set volume for this track, mostly used for games' sound effects
- * left and right volumes. Levels must be >= 0.0 and <= 1.0.
- * This is the older API. New applications should use setVolume(float) when possible.
- */
- status_t setVolume(float left, float right);
-
- /* Set volume for all channels. This is the preferred API for new applications,
- * especially for multi-channel content.
- */
- status_t setVolume(float volume);
-
- // TODO: Does this comment equally apply to the Java AudioTrack::play()?
- /* After it's created the track is not active. Call start() to
- * make it active. If set, the callback will start being called.
- * If the track was previously paused, volume is ramped up over the first mix buffer.
- */
- status_t start();
-
- // TODO: Does this comment still applies? It seems not. (obtainBuffer, AudioFlinger, ...)
- /* As a convenience we provide a write() interface to the audio buffer.
- * Input parameter 'size' is in byte units.
- * This is implemented on top of obtainBuffer/releaseBuffer. For best
- * performance use callbacks. Returns actual number of bytes written >= 0,
- * or one of the following negative status codes:
- * INVALID_OPERATION AudioTrack is configured for static buffer or streaming mode
- * BAD_VALUE size is invalid
- * WOULD_BLOCK when obtainBuffer() returns same, or
- * AudioTrack was stopped during the write
- * DEAD_OBJECT when AudioFlinger dies or the output device changes and
- * the track cannot be automatically restored.
- * The application needs to recreate the AudioTrack
- * because the audio device changed or AudioFlinger died.
- * This typically occurs for direct or offload tracks
- * or if mDoNotReconnect is true.
- * or any other error code returned by IAudioTrack::start() or restoreTrack_l().
- * Default behavior is to only return when all data has been transferred. Set 'blocking' to
- * false for the method to return immediately without waiting to try multiple times to write
- * the full content of the buffer.
- */
- ssize_t write(const void* buffer, size_t size, bool blocking = true);
-
- // TODO: Does this comment equally apply to the Java AudioTrack::stop()?
- /* Stop a track.
- * In static buffer mode, the track is stopped immediately.
- * In streaming mode, the callback will cease being called. Note that obtainBuffer() still
- * works and will fill up buffers until the pool is exhausted, and then will return WOULD_BLOCK.
- * In streaming mode the stop does not occur immediately: any data remaining in the buffer
- * is first drained, mixed, and output, and only then is the track marked as stopped.
- */
- void stop();
- bool stopped() const;
-
- // TODO: Does this comment equally apply to the Java AudioTrack::flush()?
- /* Flush a stopped or paused track. All previously buffered data is discarded immediately.
- * This has the effect of draining the buffers without mixing or output.
- * Flush is intended for streaming mode, for example before switching to non-contiguous content.
- * This function is a no-op if the track is not stopped or paused, or uses a static buffer.
- */
- void flush();
-
- // TODO: Does this comment equally apply to the Java AudioTrack::pause()?
- // At least we are not using obtainBuffer.
- /* Pause a track. After pause, the callback will cease being called and
- * obtainBuffer returns WOULD_BLOCK. Note that obtainBuffer() still works
- * and will fill up buffers until the pool is exhausted.
- * Volume is ramped down over the next mix buffer following the pause request,
- * and then the track is marked as paused. It can be resumed with ramp up by start().
- */
- void pause();
-
- bool isPlaying() const;
-
- /* Return current source sample rate in Hz.
- * If specified as zero in constructor, this will be the sink sample rate.
- */
- uint32_t getSampleRate();
-
- /* Returns the buffer duration in microseconds at current playback rate. */
- status_t getBufferDurationInUs(int64_t *duration);
-
- audio_format_t format();
-
- size_t frameSize();
-
- /*
- * Dumps the state of an audio track.
- * Not a general-purpose API; intended only for use by media player service to dump its tracks.
- */
- status_t dump(int fd, const Vector<String16>& args) const;
-
- /* Returns the AudioDeviceInfo used by the output to which this AudioTrack is
- * attached.
- */
- jobject getRoutedDevice();
-
- /* Returns the ID of the audio session this AudioTrack belongs to. */
- int32_t getAudioSessionId();
-
- /* Sets the preferred audio device to use for output of this AudioTrack.
- *
- * Parameters:
- * Device: an AudioDeviceInfo object.
- *
- * Returned value:
- * - NO_ERROR: successful operation
- * - BAD_VALUE: failed to set the device
- */
- status_t setPreferredDevice(jobject device);
-
- // TODO: Add AUDIO_OUTPUT_FLAG_DIRECT when it is possible to check.
- // TODO: Add AUDIO_FLAG_HW_AV_SYNC when it is possible to check.
- /* Returns the flags */
- audio_output_flags_t getFlags() const { return mFlags; }
-
- /* We don't keep stream type here,
- * instead, we keep attributes and call getVolumeControlStream() to get stream type
- */
- audio_stream_type_t getAudioStreamType();
-
- /* Obtain the pending duration in milliseconds for playback of pure PCM data remaining in
- * AudioTrack.
- *
- * Returns NO_ERROR if successful.
- * INVALID_OPERATION if the AudioTrack does not contain pure PCM data.
- * BAD_VALUE if msec is nullptr.
- */
- status_t pendingDuration(int32_t *msec);
-
- /* Adds an AudioDeviceCallback. The caller will be notified when the audio device to which this
- * AudioTrack is routed is updated.
- * Replaces any previously installed callback.
- *
- * Parameters:
- * Listener: the listener to receive notification of rerouting events.
- * Handler: the handler to handler the rerouting events.
- *
- * Returns NO_ERROR if successful.
- * (TODO) INVALID_OPERATION if the same callback is already installed.
- * (TODO) NO_INIT or PREMISSION_DENIED if AudioFlinger service is not reachable
- * (TODO) BAD_VALUE if the callback is NULL
- */
- status_t addAudioDeviceCallback(jobject listener, jobject rd);
-
- /* Removes an AudioDeviceCallback.
- *
- * Parameters:
- * Listener: the listener to receive notification of rerouting events.
- *
- * Returns NO_ERROR if successful.
- * (TODO) INVALID_OPERATION if the callback is not installed
- * (TODO) BAD_VALUE if the callback is NULL
- */
- status_t removeAudioDeviceCallback(jobject listener);
-
- /* Register all backed-up routing delegates.
- *
- * Parameters:
- * routingDelegates: backed-up routing delegates
- *
- */
- void registerRoutingDelegates(
- Vector<std::pair<sp<JObjectHolder>, sp<JObjectHolder>>>& routingDelegates);
-
- /* get listener from RoutingDelegate object
- */
- static jobject getListener(const jobject routingDelegateObj);
-
- /* get handler from RoutingDelegate object
- */
- static jobject getHandler(const jobject routingDelegateObj);
-
- /*
- * Parameters:
- * map and key
- *
- * Returns value if key is in the map
- * nullptr if key is not in the map
- */
- static jobject findByKey(
- Vector<std::pair<sp<JObjectHolder>, sp<JObjectHolder>>>& mp, const jobject key);
-
- /*
- * Parameters:
- * map and key
- */
- static void eraseByKey(
- Vector<std::pair<sp<JObjectHolder>, sp<JObjectHolder>>>& mp, const jobject key);
-
-private:
- audio_output_flags_t mFlags;
-
- jclass mAudioTrackCls;
- jobject mAudioTrackObj;
-
- /* Creates a Java VolumeShaper.Configuration object from VolumeShaper::Configuration */
- jobject createVolumeShaperConfigurationObj(
- const sp<media::VolumeShaper::Configuration>& config);
-
- /* Creates a Java VolumeShaper.Operation object from VolumeShaper::Operation */
- jobject createVolumeShaperOperationObj(
- const sp<media::VolumeShaper::Operation>& operation);
-
- /* Creates a Java StreamEventCallback object */
- jobject createStreamEventCallback(callback_t cbf, void* user);
-
- /* Creates a Java Executor object for running a callback */
- jobject createCallbackExecutor();
-
- status_t javaToNativeStatus(int javaStatus);
-};
-
-}; // namespace android
-
-#endif // ANDROID_JAUDIOTRACK_H
diff --git a/media/libmediaplayer2/include/mediaplayer2/JMedia2HTTPConnection.h b/media/libmediaplayer2/include/mediaplayer2/JMedia2HTTPConnection.h
deleted file mode 100644
index 15f7f83..0000000
--- a/media/libmediaplayer2/include/mediaplayer2/JMedia2HTTPConnection.h
+++ /dev/null
@@ -1,58 +0,0 @@
-/*
- * Copyright 2017, The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef _J_MEDIA2_HTTP_CONNECTION_H_
-#define _J_MEDIA2_HTTP_CONNECTION_H_
-
-#include "jni.h"
-
-#include <media/MediaHTTPConnection.h>
-#include <media/stagefright/foundation/ABase.h>
-
-namespace android {
-
-struct JMedia2HTTPConnection : public MediaHTTPConnection {
- JMedia2HTTPConnection(JNIEnv *env, jobject thiz);
-
- virtual bool connect(
- const char *uri, const KeyedVector<String8, String8> *headers) override;
-
- virtual void disconnect() override;
- virtual ssize_t readAt(off64_t offset, void *data, size_t size) override;
- virtual off64_t getSize() override;
- virtual status_t getMIMEType(String8 *mimeType) override;
- virtual status_t getUri(String8 *uri) override;
-
-protected:
- virtual ~JMedia2HTTPConnection();
-
-private:
- jobject mMedia2HTTPConnectionObj;
- jmethodID mConnectMethod;
- jmethodID mDisconnectMethod;
- jmethodID mReadAtMethod;
- jmethodID mGetSizeMethod;
- jmethodID mGetMIMETypeMethod;
- jmethodID mGetUriMethod;
-
- jbyteArray mByteArrayObj;
-
- DISALLOW_EVIL_CONSTRUCTORS(JMedia2HTTPConnection);
-};
-
-} // namespace android
-
-#endif // _J_MEDIA2_HTTP_CONNECTION_H_
diff --git a/media/libmediaplayer2/include/mediaplayer2/JMedia2HTTPService.h b/media/libmediaplayer2/include/mediaplayer2/JMedia2HTTPService.h
deleted file mode 100644
index bf61a7f..0000000
--- a/media/libmediaplayer2/include/mediaplayer2/JMedia2HTTPService.h
+++ /dev/null
@@ -1,47 +0,0 @@
-/*
- * Copyright 2017, The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef _J_MEDIA2_HTTP_SERVICE_H_
-#define _J_MEDIA2_HTTP_SERVICE_H_
-
-#include <jni.h>
-#include <utils/RefBase.h>
-
-#include <media/MediaHTTPService.h>
-#include <media/MediaHTTPConnection.h>
-#include <media/stagefright/foundation/ABase.h>
-
-namespace android {
-
-struct JMedia2HTTPService : public MediaHTTPService {
- JMedia2HTTPService(JNIEnv *env, jobject thiz);
-
- virtual sp<MediaHTTPConnection> makeHTTPConnection() override;
-
-protected:
- virtual ~JMedia2HTTPService();
-
-private:
- jobject mMedia2HTTPServiceObj;
-
- jmethodID mMakeHTTPConnectionMethod;
-
- DISALLOW_EVIL_CONSTRUCTORS(JMedia2HTTPService);
-};
-
-} // namespace android
-
-#endif // _J_MEDIA2_HTTP_SERVICE_H_
diff --git a/media/libmediaplayer2/include/mediaplayer2/JObjectHolder.h b/media/libmediaplayer2/include/mediaplayer2/JObjectHolder.h
deleted file mode 100644
index 93d8b40..0000000
--- a/media/libmediaplayer2/include/mediaplayer2/JObjectHolder.h
+++ /dev/null
@@ -1,47 +0,0 @@
-/*
- * Copyright 2018, The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef JOBJECT_HOLDER_H_
-
-#define JOBJECT_HOLDER_H_
-
-#include "jni.h"
-#include <mediaplayer2/JavaVMHelper.h>
-#include <utils/RefBase.h>
-
-namespace android {
-
-// Helper class for managing global reference of jobject.
-struct JObjectHolder : public RefBase {
- JObjectHolder(jobject obj) {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- mJObject = reinterpret_cast<jobject>(env->NewGlobalRef(obj));
- }
-
- virtual ~JObjectHolder() {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- env->DeleteGlobalRef(mJObject);
- }
-
- jobject getJObject() { return mJObject; }
-
-private:
- jobject mJObject;
-};
-
-} //" android
-
-#endif // JOBJECT_HOLDER_H_
diff --git a/media/libmediaplayer2/include/mediaplayer2/JavaVMHelper.h b/media/libmediaplayer2/include/mediaplayer2/JavaVMHelper.h
deleted file mode 100644
index 4b56aca..0000000
--- a/media/libmediaplayer2/include/mediaplayer2/JavaVMHelper.h
+++ /dev/null
@@ -1,41 +0,0 @@
-/*
- * Copyright 2018 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef JAVA_VM_HELPER_H_
-
-#define JAVA_VM_HELPER_H_
-
-#include "jni.h"
-
-#include <atomic>
-
-namespace android {
-
-struct JavaVMHelper {
- static JNIEnv *getJNIEnv();
- static JavaVM *getJavaVM();
- static void setJavaVM(JavaVM *vm);
-
-private:
- // Once a valid JavaVM has been set, it should never be reset or changed.
- // However, as it may be accessed from multiple threads, access needs to be
- // synchronized.
- static std::atomic<JavaVM *> sJavaVM;
-};
-
-} // namespace android
-
-#endif // JAVA_VM_HELPER_H_
diff --git a/media/libmediaplayer2/include/mediaplayer2/MediaPlayer2AudioOutput.h b/media/libmediaplayer2/include/mediaplayer2/MediaPlayer2AudioOutput.h
deleted file mode 100644
index f38b7cc..0000000
--- a/media/libmediaplayer2/include/mediaplayer2/MediaPlayer2AudioOutput.h
+++ /dev/null
@@ -1,188 +0,0 @@
-/*
-**
-** Copyright 2018, The Android Open Source Project
-**
-** Licensed under the Apache License, Version 2.0 (the "License");
-** you may not use this file except in compliance with the License.
-** You may obtain a copy of the License at
-**
-** http://www.apache.org/licenses/LICENSE-2.0
-**
-** Unless required by applicable law or agreed to in writing, software
-** distributed under the License is distributed on an "AS IS" BASIS,
-** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-** See the License for the specific language governing permissions and
-** limitations under the License.
-*/
-
-#ifndef ANDROID_MEDIAPLAYER2AUDIOOUTPUT_H
-#define ANDROID_MEDIAPLAYER2AUDIOOUTPUT_H
-
-#include <mediaplayer2/MediaPlayer2Interface.h>
-#include <mediaplayer2/JAudioTrack.h>
-#include <mediaplayer2/JObjectHolder.h>
-
-#include <utility>
-#include <utils/String16.h>
-#include <utils/Vector.h>
-
-#include "jni.h"
-
-namespace android {
-
-class AudioTrack;
-
-class MediaPlayer2AudioOutput : public MediaPlayer2Interface::AudioSink
-{
- class CallbackData;
-
-public:
- MediaPlayer2AudioOutput(int32_t sessionId,
- uid_t uid,
- int pid,
- const jobject attributes);
- virtual ~MediaPlayer2AudioOutput();
-
- virtual bool ready() const {
- return mJAudioTrack != nullptr;
- }
- virtual ssize_t bufferSize() const;
- virtual ssize_t frameCount() const;
- virtual ssize_t channelCount() const;
- virtual ssize_t frameSize() const;
- virtual uint32_t latency() const;
- virtual float msecsPerFrame() const;
- virtual status_t getPosition(uint32_t *position) const;
- virtual status_t getTimestamp(AudioTimestamp &ts) const;
- virtual int64_t getPlayedOutDurationUs(int64_t nowUs) const;
- virtual status_t getFramesWritten(uint32_t *frameswritten) const;
- virtual int32_t getSessionId() const;
- virtual void setSessionId(const int32_t id);
- virtual uint32_t getSampleRate() const;
- virtual int64_t getBufferDurationInUs() const;
-
- virtual status_t open(
- uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
- audio_format_t format,
- AudioCallback cb, void *cookie,
- audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
- const audio_offload_info_t *offloadInfo = NULL,
- uint32_t suggestedFrameCount = 0);
-
- virtual status_t start();
- virtual ssize_t write(const void* buffer, size_t size, bool blocking = true);
- virtual void stop();
- virtual void flush();
- virtual void pause();
- virtual void close();
- void setAudioAttributes(const jobject attributes);
- virtual audio_stream_type_t getAudioStreamType() const;
-
- void setVolume(float volume);
- virtual status_t setPlaybackRate(const AudioPlaybackRate& rate);
- virtual status_t getPlaybackRate(AudioPlaybackRate* rate /* nonnull */);
-
- status_t setAuxEffectSendLevel(float level);
- status_t attachAuxEffect(int effectId);
- virtual status_t dump(int fd, const Vector<String16>& args) const;
-
- static bool isOnEmulator();
- static int getMinBufferCount();
- virtual bool needsTrailingPadding() {
- return true;
- // TODO: return correct value.
- //return mNextOutput == NULL;
- }
- // AudioRouting
- virtual status_t setPreferredDevice(jobject device);
- virtual jobject getRoutedDevice();
- virtual status_t addAudioDeviceCallback(jobject routingDelegate);
- virtual status_t removeAudioDeviceCallback(jobject listener);
-
-private:
- static void setMinBufferCount();
- static void CallbackWrapper(int event, void *me, void *info);
- void deleteRecycledTrack_l();
- void close_l();
- status_t updateTrack_l();
-
- sp<JAudioTrack> mJAudioTrack;
- AudioCallback mCallback;
- void * mCallbackCookie;
- CallbackData * mCallbackData;
- sp<JObjectHolder> mAttributes;
- float mVolume;
- AudioPlaybackRate mPlaybackRate;
- uint32_t mSampleRateHz; // sample rate of the content, as set in open()
- float mMsecsPerFrame;
- size_t mFrameSize;
- int32_t mSessionId;
- uid_t mUid;
- int mPid;
- float mSendLevel;
- int mAuxEffectId;
- audio_output_flags_t mFlags;
- sp<JObjectHolder> mPreferredDevice;
- mutable Mutex mLock;
-
- // <listener, routingDelegate>
- Vector<std::pair<sp<JObjectHolder>, sp<JObjectHolder>>> mRoutingDelegates;
-
- // static variables below not protected by mutex
- static bool mIsOnEmulator;
- static int mMinBufferCount; // 12 for emulator; otherwise 4
-
- // CallbackData is what is passed to the AudioTrack as the "user" data.
- // We need to be able to target this to a different Output on the fly,
- // so we can't use the Output itself for this.
- class CallbackData {
- friend MediaPlayer2AudioOutput;
- public:
- explicit CallbackData(MediaPlayer2AudioOutput *cookie) {
- mData = cookie;
- mSwitching = false;
- }
- MediaPlayer2AudioOutput *getOutput() const {
- return mData;
- }
- void setOutput(MediaPlayer2AudioOutput* newcookie) {
- mData = newcookie;
- }
- // lock/unlock are used by the callback before accessing the payload of this object
- void lock() const {
- mLock.lock();
- }
- void unlock() const {
- mLock.unlock();
- }
-
- // tryBeginTrackSwitch/endTrackSwitch are used when the CallbackData is handed over
- // to the next sink.
-
- // tryBeginTrackSwitch() returns true only if it obtains the lock.
- bool tryBeginTrackSwitch() {
- LOG_ALWAYS_FATAL_IF(mSwitching, "tryBeginTrackSwitch() already called");
- if (mLock.tryLock() != OK) {
- return false;
- }
- mSwitching = true;
- return true;
- }
- void endTrackSwitch() {
- if (mSwitching) {
- mLock.unlock();
- }
- mSwitching = false;
- }
-
- private:
- MediaPlayer2AudioOutput *mData;
- mutable Mutex mLock; // a recursive mutex might make this unnecessary.
- bool mSwitching;
- DISALLOW_EVIL_CONSTRUCTORS(CallbackData);
- };
-};
-
-}; // namespace android
-
-#endif // ANDROID_MEDIAPLAYER2AUDIOOUTPUT_H
diff --git a/media/libmediaplayer2/include/mediaplayer2/MediaPlayer2Interface.h b/media/libmediaplayer2/include/mediaplayer2/MediaPlayer2Interface.h
deleted file mode 100644
index 7804a62..0000000
--- a/media/libmediaplayer2/include/mediaplayer2/MediaPlayer2Interface.h
+++ /dev/null
@@ -1,273 +0,0 @@
-/*
- * Copyright 2017 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_MEDIAPLAYER2INTERFACE_H
-#define ANDROID_MEDIAPLAYER2INTERFACE_H
-
-#ifdef __cplusplus
-
-#include <sys/types.h>
-#include <utils/Errors.h>
-#include <utils/String8.h>
-#include <utils/RefBase.h>
-#include <jni.h>
-
-#include <media/AVSyncSettings.h>
-#include <media/AudioResamplerPublic.h>
-#include <media/AudioSystem.h>
-#include <media/AudioTimestamp.h>
-#include <media/BufferingSettings.h>
-#include <media/stagefright/foundation/AHandler.h>
-#include <mediaplayer2/MediaPlayer2Types.h>
-
-#include "jni.h"
-#include "mediaplayer2.pb.h"
-
-using android::media::MediaPlayer2Proto::PlayerMessage;
-
-// Fwd decl to make sure everyone agrees that the scope of struct sockaddr_in is
-// global, and not in android::
-struct sockaddr_in;
-
-namespace android {
-
-struct DataSourceDesc;
-class Parcel;
-struct ANativeWindowWrapper;
-
-#define DEFAULT_AUDIOSINK_BUFFERSIZE 1200
-#define DEFAULT_AUDIOSINK_SAMPLERATE 44100
-
-// when the channel mask isn't known, use the channel count to derive a mask in AudioSink::open()
-#define CHANNEL_MASK_USE_CHANNEL_ORDER 0
-
-// duration below which we do not allow deep audio buffering
-#define AUDIO_SINK_MIN_DEEP_BUFFER_DURATION_US 5000000
-
-class MediaPlayer2InterfaceListener: public RefBase
-{
-public:
- virtual void notify(int64_t srcId, int msg, int ext1, int ext2,
- const PlayerMessage *obj) = 0;
-};
-
-class MediaPlayer2Interface : public AHandler {
-public:
- // AudioSink: abstraction layer for audio output
- class AudioSink : public RefBase {
- public:
- enum cb_event_t {
- CB_EVENT_FILL_BUFFER, // Request to write more data to buffer.
- CB_EVENT_STREAM_END, // Sent after all the buffers queued in AF and HW are played
- // back (after stop is called)
- CB_EVENT_TEAR_DOWN // The AudioTrack was invalidated due to use case change:
- // Need to re-evaluate offloading options
- };
-
- // Callback returns the number of bytes actually written to the buffer.
- typedef size_t (*AudioCallback)(
- AudioSink *audioSink, void *buffer, size_t size, void *cookie, cb_event_t event);
-
- virtual ~AudioSink() {}
- virtual bool ready() const = 0; // audio output is open and ready
- virtual ssize_t bufferSize() const = 0;
- virtual ssize_t frameCount() const = 0;
- virtual ssize_t channelCount() const = 0;
- virtual ssize_t frameSize() const = 0;
- virtual uint32_t latency() const = 0;
- virtual float msecsPerFrame() const = 0;
- virtual status_t getPosition(uint32_t *position) const = 0;
- virtual status_t getTimestamp(AudioTimestamp &ts) const = 0;
- virtual int64_t getPlayedOutDurationUs(int64_t nowUs) const = 0;
- virtual status_t getFramesWritten(uint32_t *frameswritten) const = 0;
- virtual int32_t getSessionId() const = 0;
- virtual audio_stream_type_t getAudioStreamType() const = 0;
- virtual uint32_t getSampleRate() const = 0;
- virtual int64_t getBufferDurationInUs() const = 0;
-
- // If no callback is specified, use the "write" API below to submit
- // audio data.
- virtual status_t open(
- uint32_t sampleRate, int channelCount, audio_channel_mask_t channelMask,
- audio_format_t format=AUDIO_FORMAT_PCM_16_BIT,
- AudioCallback cb = NULL,
- void *cookie = NULL,
- audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
- const audio_offload_info_t *offloadInfo = NULL,
- uint32_t suggestedFrameCount = 0) = 0;
-
- virtual status_t start() = 0;
-
- /* Input parameter |size| is in byte units stored in |buffer|.
- * Data is copied over and actual number of bytes written (>= 0)
- * is returned, or no data is copied and a negative status code
- * is returned (even when |blocking| is true).
- * When |blocking| is false, AudioSink will immediately return after
- * part of or full |buffer| is copied over.
- * When |blocking| is true, AudioSink will wait to copy the entire
- * buffer, unless an error occurs or the copy operation is
- * prematurely stopped.
- */
- virtual ssize_t write(const void* buffer, size_t size, bool blocking = true) = 0;
-
- virtual void stop() = 0;
- virtual void flush() = 0;
- virtual void pause() = 0;
- virtual void close() = 0;
-
- virtual status_t setPlaybackRate(const AudioPlaybackRate& rate) = 0;
- virtual status_t getPlaybackRate(AudioPlaybackRate* rate /* nonnull */) = 0;
- virtual bool needsTrailingPadding() {
- return true;
- }
-
- virtual status_t setParameters(const String8& /* keyValuePairs */) {
- return NO_ERROR;
- }
- virtual String8 getParameters(const String8& /* keys */) {
- return String8::empty();
- }
-
- // AudioRouting
- virtual status_t setPreferredDevice(jobject device);
- virtual jobject getRoutedDevice();
- virtual status_t addAudioDeviceCallback(jobject routingDelegate);
- virtual status_t removeAudioDeviceCallback(jobject listener);
- };
-
- MediaPlayer2Interface() : mListener(NULL) { }
- virtual ~MediaPlayer2Interface() { }
- virtual status_t initCheck() = 0;
-
- virtual void setAudioSink(const sp<AudioSink>& audioSink) {
- mAudioSink = audioSink;
- }
-
- virtual status_t setDataSource(const sp<DataSourceDesc> &dsd) = 0;
-
- virtual status_t prepareNextDataSource(const sp<DataSourceDesc> &dsd) = 0;
-
- virtual status_t playNextDataSource(int64_t srcId) = 0;
-
- // pass the buffered native window to the media player service
- virtual status_t setVideoSurfaceTexture(const sp<ANativeWindowWrapper>& nww) = 0;
-
- virtual status_t getBufferingSettings(BufferingSettings* buffering /* nonnull */) {
- *buffering = BufferingSettings();
- return OK;
- }
- virtual status_t setBufferingSettings(const BufferingSettings& /* buffering */) {
- return OK;
- }
-
- virtual status_t prepareAsync() = 0;
- virtual status_t start() = 0;
- virtual status_t pause() = 0;
- virtual bool isPlaying() = 0;
- virtual status_t setPlaybackSettings(const AudioPlaybackRate& rate) {
- // by default, players only support setting rate to the default
- if (!isAudioPlaybackRateEqual(rate, AUDIO_PLAYBACK_RATE_DEFAULT)) {
- return BAD_VALUE;
- }
- return OK;
- }
- virtual status_t getPlaybackSettings(AudioPlaybackRate* rate /* nonnull */) {
- *rate = AUDIO_PLAYBACK_RATE_DEFAULT;
- return OK;
- }
- virtual status_t setSyncSettings(const AVSyncSettings& sync, float /* videoFps */) {
- // By default, players only support setting sync source to default; all other sync
- // settings are ignored. There is no requirement for getters to return set values.
- if (sync.mSource != AVSYNC_SOURCE_DEFAULT) {
- return BAD_VALUE;
- }
- return OK;
- }
- virtual status_t getSyncSettings(
- AVSyncSettings* sync /* nonnull */, float* videoFps /* nonnull */) {
- *sync = AVSyncSettings();
- *videoFps = -1.f;
- return OK;
- }
- virtual status_t seekTo(
- int64_t msec, MediaPlayer2SeekMode mode = MediaPlayer2SeekMode::SEEK_PREVIOUS_SYNC) = 0;
- virtual status_t getCurrentPosition(int64_t *msec) = 0;
- virtual status_t getDuration(int64_t *msec) = 0;
- virtual status_t reset() = 0;
- virtual status_t notifyAt(int64_t /* mediaTimeUs */) {
- return INVALID_OPERATION;
- }
- virtual status_t setLooping(int loop) = 0;
- virtual status_t setParameter(int key, const Parcel &request) = 0;
- virtual status_t getParameter(int key, Parcel *reply) = 0;
-
- virtual status_t getMetrics(char **buffer, size_t *length) = 0;
-
- // Invoke a generic method on the player by using opaque parcels
- // for the request and reply.
- //
- // @param request Parcel that is positioned at the start of the
- // data sent by the java layer.
- // @param[out] reply Parcel to hold the reply data. Cannot be null.
- // @return OK if the call was successful.
- virtual status_t invoke(const PlayerMessage &request, PlayerMessage *reply) = 0;
-
- void setListener(const sp<MediaPlayer2InterfaceListener> &listener) {
- Mutex::Autolock autoLock(mListenerLock);
- mListener = listener;
- }
-
- void sendEvent(int64_t srcId, int msg, int ext1=0, int ext2=0, const PlayerMessage *obj=NULL) {
- sp<MediaPlayer2InterfaceListener> listener;
- {
- Mutex::Autolock autoLock(mListenerLock);
- listener = mListener;
- }
-
- if (listener) {
- listener->notify(srcId, msg, ext1, ext2, obj);
- }
- }
-
- virtual status_t dump(int /* fd */, const Vector<String16>& /* args */) const {
- return INVALID_OPERATION;
- }
-
- virtual void onMessageReceived(const sp<AMessage> & /* msg */) override { }
-
- // Modular DRM
- virtual status_t prepareDrm(int64_t /*srcId*/, const uint8_t /* uuid */[16],
- const Vector<uint8_t>& /* drmSessionId */) {
- return INVALID_OPERATION;
- }
- virtual status_t releaseDrm(int64_t /*srcId*/) {
- return INVALID_OPERATION;
- }
-
-protected:
- sp<AudioSink> mAudioSink;
-
-private:
- Mutex mListenerLock;
- sp<MediaPlayer2InterfaceListener> mListener;
-};
-
-}; // namespace android
-
-#endif // __cplusplus
-
-
-#endif // ANDROID_MEDIAPLAYER2INTERFACE_H
diff --git a/media/libmediaplayer2/include/mediaplayer2/MediaPlayer2Types.h b/media/libmediaplayer2/include/mediaplayer2/MediaPlayer2Types.h
deleted file mode 100644
index 2430289..0000000
--- a/media/libmediaplayer2/include/mediaplayer2/MediaPlayer2Types.h
+++ /dev/null
@@ -1,204 +0,0 @@
-/*
- * Copyright 2018 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_MEDIAPLAYER2_TYPES_H
-#define ANDROID_MEDIAPLAYER2_TYPES_H
-
-#include <media/mediaplayer_common.h>
-
-#include <media/MediaSource.h>
-
-namespace android {
-
-typedef MediaSource::ReadOptions::SeekMode MediaPlayer2SeekMode;
-
-enum media2_event_type {
- MEDIA2_NOP = 0, // interface test message
- MEDIA2_PREPARED = 1,
- MEDIA2_PLAYBACK_COMPLETE = 2,
- MEDIA2_BUFFERING_UPDATE = 3,
- MEDIA2_SEEK_COMPLETE = 4,
- MEDIA2_SET_VIDEO_SIZE = 5,
- MEDIA2_STARTED = 6,
- MEDIA2_PAUSED = 7,
- MEDIA2_SKIPPED = 8,
- MEDIA2_NOTIFY_TIME = 98,
- MEDIA2_TIMED_TEXT = 99,
- MEDIA2_ERROR = 100,
- MEDIA2_INFO = 200,
- MEDIA2_SUBTITLE_DATA = 201,
- MEDIA2_META_DATA = 202,
- MEDIA2_DRM_INFO = 210,
-};
-
-// Generic error codes for the media player framework. Errors are fatal, the
-// playback must abort.
-//
-// Errors are communicated back to the client using the
-// MediaPlayer2Listener::notify method defined below.
-// In this situation, 'notify' is invoked with the following:
-// 'msg' is set to MEDIA_ERROR.
-// 'ext1' should be a value from the enum media2_error_type.
-// 'ext2' contains an implementation dependant error code to provide
-// more details. Should default to 0 when not used.
-//
-// The codes are distributed as follow:
-// 0xx: Reserved
-// 1xx: Android Player errors. Something went wrong inside the MediaPlayer2.
-// 2xx: Media errors (e.g Codec not supported). There is a problem with the
-// media itself.
-// 3xx: Runtime errors. Some extraordinary condition arose making the playback
-// impossible.
-//
-enum media2_error_type {
- // 0xx
- MEDIA2_ERROR_UNKNOWN = 1,
- // 1xx
- // MEDIA2_ERROR_SERVER_DIED = 100,
- // 2xx
- MEDIA2_ERROR_NOT_VALID_FOR_PROGRESSIVE_PLAYBACK = 200,
- // 3xx
- MEDIA2_ERROR_FAILED_TO_SET_DATA_SOURCE = 300,
-};
-
-
-// Info and warning codes for the media player framework. These are non fatal,
-// the playback is going on but there might be some user visible issues.
-//
-// Info and warning messages are communicated back to the client using the
-// MediaPlayer2Listener::notify method defined below. In this situation,
-// 'notify' is invoked with the following:
-// 'msg' is set to MEDIA_INFO.
-// 'ext1' should be a value from the enum media2_info_type.
-// 'ext2' contains an implementation dependant info code to provide
-// more details. Should default to 0 when not used.
-//
-// The codes are distributed as follow:
-// 0xx: Reserved
-// 7xx: Android Player info/warning (e.g player lagging behind.)
-// 8xx: Media info/warning (e.g media badly interleaved.)
-//
-enum media2_info_type {
- // 0xx
- MEDIA2_INFO_UNKNOWN = 1,
- // The player just started the playback of this data source.
- MEDIA2_INFO_DATA_SOURCE_START = 2,
- // The player just pushed the very first video frame for rendering
- MEDIA2_INFO_VIDEO_RENDERING_START = 3,
- // The player just pushed the very first audio frame for rendering
- MEDIA2_INFO_AUDIO_RENDERING_START = 4,
- // The player just completed the playback of this data source
- MEDIA2_INFO_DATA_SOURCE_END = 5,
- // The player just completed the playback of all data sources.
- // But this is not visible in native code. Just keep this entry for completeness.
- MEDIA2_INFO_DATA_SOURCE_LIST_END = 6,
- // The player just completed an iteration of playback loop. This event is sent only when
- // looping is enabled.
- MEDIA2_INFO_DATA_SOURCE_REPEAT = 7,
-
- //1xx
- // The player just prepared a data source.
- MEDIA2_INFO_PREPARED = 100,
- // The player just completed a call play().
- MEDIA2_INFO_COMPLETE_CALL_PLAY = 101,
- // The player just completed a call pause().
- MEDIA2_INFO_COMPLETE_CALL_PAUSE = 102,
- // The player just completed a call seekTo.
- MEDIA2_INFO_COMPLETE_CALL_SEEK = 103,
-
- // 7xx
- // The video is too complex for the decoder: it can't decode frames fast
- // enough. Possibly only the audio plays fine at this stage.
- MEDIA2_INFO_VIDEO_TRACK_LAGGING = 700,
- // MediaPlayer2 is temporarily pausing playback internally in order to
- // buffer more data.
- MEDIA2_INFO_BUFFERING_START = 701,
- // MediaPlayer2 is resuming playback after filling buffers.
- MEDIA2_INFO_BUFFERING_END = 702,
- // Bandwidth in recent past
- MEDIA2_INFO_NETWORK_BANDWIDTH = 703,
-
- // 8xx
- // Bad interleaving means that a media has been improperly interleaved or not
- // interleaved at all, e.g has all the video samples first then all the audio
- // ones. Video is playing but a lot of disk seek may be happening.
- MEDIA2_INFO_BAD_INTERLEAVING = 800,
- // The media is not seekable (e.g live stream).
- MEDIA2_INFO_NOT_SEEKABLE = 801,
- // New media metadata is available.
- MEDIA2_INFO_METADATA_UPDATE = 802,
- // Audio can not be played.
- MEDIA2_INFO_PLAY_AUDIO_ERROR = 804,
- // Video can not be played.
- MEDIA2_INFO_PLAY_VIDEO_ERROR = 805,
-
- //9xx
- MEDIA2_INFO_TIMED_TEXT_ERROR = 900,
-};
-
-// Do not change these values without updating their counterparts in MediaPlayer2.java
-enum mediaplayer2_states {
- MEDIAPLAYER2_STATE_IDLE = 1001,
- MEDIAPLAYER2_STATE_PREPARED = 1002,
- MEDIAPLAYER2_STATE_PAUSED = 1003,
- MEDIAPLAYER2_STATE_PLAYING = 1004,
- MEDIAPLAYER2_STATE_ERROR = 1005,
-};
-
-enum media_player2_internal_states {
- MEDIA_PLAYER2_STATE_ERROR = 0,
- MEDIA_PLAYER2_IDLE = 1 << 0,
- MEDIA_PLAYER2_INITIALIZED = 1 << 1,
- MEDIA_PLAYER2_PREPARING = 1 << 2,
- MEDIA_PLAYER2_PREPARED = 1 << 3,
- MEDIA_PLAYER2_STARTED = 1 << 4,
- MEDIA_PLAYER2_PAUSED = 1 << 5,
- MEDIA_PLAYER2_PLAYBACK_COMPLETE = 1 << 6
-};
-
-// Keep KEY_PARAMETER_* in sync with MediaPlayer2.java.
-// The same enum space is used for both set and get, in case there are future keys that
-// can be both set and get. But as of now, all parameters are either set only or get only.
-enum media2_parameter_keys {
- // Streaming/buffering parameters
- MEDIA2_KEY_PARAMETER_CACHE_STAT_COLLECT_FREQ_MS = 1100, // set only
-
- // Return a Parcel containing a single int, which is the channel count of the
- // audio track, or zero for error (e.g. no audio track) or unknown.
- MEDIA2_KEY_PARAMETER_AUDIO_CHANNEL_COUNT = 1200, // get only
-
- // Playback rate expressed in permille (1000 is normal speed), saved as int32_t, with negative
- // values used for rewinding or reverse playback.
- MEDIA2_KEY_PARAMETER_PLAYBACK_RATE_PERMILLE = 1300, // set only
-
- // Set a Parcel containing the value of a parcelled Java AudioAttribute instance
- MEDIA2_KEY_PARAMETER_AUDIO_ATTRIBUTES = 1400 // set only
-};
-
-// Keep INVOKE_ID_* in sync with MediaPlayer2.java.
-enum media_player2_invoke_ids {
- MEDIA_PLAYER2_INVOKE_ID_GET_TRACK_INFO = 1,
- MEDIA_PLAYER2_INVOKE_ID_ADD_EXTERNAL_SOURCE = 2,
- MEDIA_PLAYER2_INVOKE_ID_ADD_EXTERNAL_SOURCE_FD = 3,
- MEDIA_PLAYER2_INVOKE_ID_SELECT_TRACK = 4,
- MEDIA_PLAYER2_INVOKE_ID_UNSELECT_TRACK = 5,
- MEDIA_PLAYER2_INVOKE_ID_SET_VIDEO_SCALING_MODE = 6,
- MEDIA_PLAYER2_INVOKE_ID_GET_SELECTED_TRACK = 7
-};
-
-}; // namespace android
-
-#endif // ANDROID_MEDIAPLAYER2_TYPES_H
diff --git a/media/libmediaplayer2/include/mediaplayer2/mediaplayer2.h b/media/libmediaplayer2/include/mediaplayer2/mediaplayer2.h
deleted file mode 100644
index 1e8a1d5..0000000
--- a/media/libmediaplayer2/include/mediaplayer2/mediaplayer2.h
+++ /dev/null
@@ -1,165 +0,0 @@
-/*
- * Copyright 2017 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_MEDIAPLAYER2_H
-#define ANDROID_MEDIAPLAYER2_H
-
-#include <media/AVSyncSettings.h>
-#include <media/AudioResamplerPublic.h>
-#include <media/BufferingSettings.h>
-#include <media/mediaplayer_common.h>
-#include <mediaplayer2/MediaPlayer2Interface.h>
-#include <mediaplayer2/MediaPlayer2Types.h>
-#include <mediaplayer2/JObjectHolder.h>
-
-#include <jni.h>
-#include <utils/Errors.h>
-#include <utils/Mutex.h>
-#include <utils/RefBase.h>
-#include <utils/String16.h>
-#include <utils/Vector.h>
-#include <system/audio-base.h>
-
-#include "jni.h"
-
-namespace android {
-
-struct ANativeWindowWrapper;
-struct DataSourceDesc;
-class MediaPlayer2AudioOutput;
-
-// ref-counted object for callbacks
-class MediaPlayer2Listener: virtual public RefBase
-{
-public:
- virtual void notify(int64_t srcId, int msg, int ext1, int ext2,
- const PlayerMessage *obj = NULL) = 0;
-};
-
-class MediaPlayer2 : public MediaPlayer2InterfaceListener
-{
-public:
- ~MediaPlayer2();
-
- static sp<MediaPlayer2> Create(int32_t sessionId, jobject context);
- static status_t DumpAll(int fd, const Vector<String16>& args);
-
- void disconnect();
-
- status_t getSrcId(int64_t *srcId);
- status_t setDataSource(const sp<DataSourceDesc> &dsd);
- status_t prepareNextDataSource(const sp<DataSourceDesc> &dsd);
- status_t playNextDataSource(int64_t srcId);
- status_t setVideoSurfaceTexture(const sp<ANativeWindowWrapper>& nww);
- status_t setListener(const sp<MediaPlayer2Listener>& listener);
- status_t getBufferingSettings(BufferingSettings* buffering /* nonnull */);
- status_t setBufferingSettings(const BufferingSettings& buffering);
- status_t prepareAsync();
- status_t start();
- status_t pause();
- bool isPlaying();
- mediaplayer2_states getState();
- status_t setPlaybackSettings(const AudioPlaybackRate& rate);
- status_t getPlaybackSettings(AudioPlaybackRate* rate /* nonnull */);
- status_t setSyncSettings(const AVSyncSettings& sync, float videoFpsHint);
- status_t getSyncSettings(
- AVSyncSettings* sync /* nonnull */,
- float* videoFps /* nonnull */);
- status_t getVideoWidth(int *w);
- status_t getVideoHeight(int *h);
- status_t seekTo(
- int64_t msec,
- MediaPlayer2SeekMode mode = MediaPlayer2SeekMode::SEEK_PREVIOUS_SYNC);
- status_t notifyAt(int64_t mediaTimeUs);
- status_t getCurrentPosition(int64_t *msec);
- status_t getDuration(int64_t srcId, int64_t *msec);
- status_t reset();
- status_t setAudioStreamType(audio_stream_type_t type);
- status_t getAudioStreamType(audio_stream_type_t *type);
- status_t setLooping(int loop);
- bool isLooping();
- status_t setVolume(float volume);
- void notify(int64_t srcId, int msg, int ext1, int ext2,
- const PlayerMessage *obj = NULL);
- status_t invoke(const PlayerMessage &request, PlayerMessage *reply);
- status_t setAudioSessionId(int32_t sessionId);
- int32_t getAudioSessionId();
- status_t setAuxEffectSendLevel(float level);
- status_t attachAuxEffect(int effectId);
- status_t setAudioAttributes(const jobject attributes);
- jobject getAudioAttributes();
- status_t getParameter(int key, Parcel* reply);
- status_t getMetrics(char **buffer, size_t *length);
-
- // Modular DRM
- status_t prepareDrm(int64_t srcId,
- const uint8_t uuid[16],
- const Vector<uint8_t>& drmSessionId);
- status_t releaseDrm(int64_t srcId);
- // AudioRouting
- status_t setPreferredDevice(jobject device);
- jobject getRoutedDevice();
- status_t addAudioDeviceCallback(jobject routingDelegate);
- status_t removeAudioDeviceCallback(jobject listener);
-
- status_t dump(int fd, const Vector<String16>& args);
-
-private:
- MediaPlayer2(int32_t sessionId, jobject context);
- bool init();
-
- // Disconnect from the currently connected ANativeWindow.
- void disconnectNativeWindow_l();
-
- status_t setAudioAttributes_l(const jobject attributes);
-
- void clear_l();
- status_t seekTo_l(int64_t msec, MediaPlayer2SeekMode mode);
- status_t prepareAsync_l();
- status_t getDuration_l(int64_t *msec);
- status_t reset_l();
- status_t checkState_l();
-
- pid_t mPid;
- uid_t mUid;
- sp<MediaPlayer2Interface> mPlayer;
- sp<MediaPlayer2AudioOutput> mAudioOutput;
- int64_t mSrcId;
- thread_id_t mLockThreadId;
- mutable Mutex mLock;
- Mutex mNotifyLock;
- sp<MediaPlayer2Listener> mListener;
- media_player2_internal_states mCurrentState;
- bool mTransitionToNext;
- int64_t mCurrentPosition;
- MediaPlayer2SeekMode mCurrentSeekMode;
- int64_t mSeekPosition;
- MediaPlayer2SeekMode mSeekMode;
- audio_stream_type_t mStreamType;
- bool mLoop;
- float mVolume;
- int mVideoWidth;
- int mVideoHeight;
- int32_t mAudioSessionId;
- sp<JObjectHolder> mAudioAttributes;
- sp<JObjectHolder> mContext;
- float mSendLevel;
- sp<ANativeWindowWrapper> mConnectedWindow;
-};
-
-}; // namespace android
-
-#endif // ANDROID_MEDIAPLAYER2_H
diff --git a/media/libmediaplayer2/mediaplayer2.cpp b/media/libmediaplayer2/mediaplayer2.cpp
deleted file mode 100644
index de65f8d..0000000
--- a/media/libmediaplayer2/mediaplayer2.cpp
+++ /dev/null
@@ -1,1261 +0,0 @@
-/*
-**
-** Copyright 2017, The Android Open Source Project
-**
-** Licensed under the Apache License, Version 2.0 (the "License");
-** you may not use this file except in compliance with the License.
-** You may obtain a copy of the License at
-**
-** http://www.apache.org/licenses/LICENSE-2.0
-**
-** Unless required by applicable law or agreed to in writing, software
-** distributed under the License is distributed on an "AS IS" BASIS,
-** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-** See the License for the specific language governing permissions and
-** limitations under the License.
-*/
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "MediaPlayer2Native"
-
-#include <android/binder_ibinder.h>
-#include <media/AudioSystem.h>
-#include <media/DataSourceDesc.h>
-#include <media/MemoryLeakTrackUtil.h>
-#include <media/NdkWrapper.h>
-#include <media/stagefright/foundation/ADebug.h>
-#include <media/stagefright/foundation/ALooperRoster.h>
-#include <mediaplayer2/MediaPlayer2AudioOutput.h>
-#include <mediaplayer2/mediaplayer2.h>
-
-#include <utils/Log.h>
-#include <utils/SortedVector.h>
-#include <utils/String8.h>
-
-#include <system/audio.h>
-#include <system/window.h>
-
-#include <nuplayer2/NuPlayer2Driver.h>
-
-#include <dirent.h>
-#include <sys/stat.h>
-
-namespace android {
-
-extern ALooperRoster gLooperRoster;
-
-namespace {
-
-const int kDumpLockRetries = 50;
-const int kDumpLockSleepUs = 20000;
-
-class proxyListener : public MediaPlayer2InterfaceListener {
-public:
- proxyListener(const wp<MediaPlayer2> &player)
- : mPlayer(player) { }
-
- ~proxyListener() { };
-
- virtual void notify(int64_t srcId, int msg, int ext1, int ext2,
- const PlayerMessage *obj) override {
- sp<MediaPlayer2> player = mPlayer.promote();
- if (player != NULL) {
- player->notify(srcId, msg, ext1, ext2, obj);
- }
- }
-
-private:
- wp<MediaPlayer2> mPlayer;
-};
-
-Mutex sRecordLock;
-SortedVector<wp<MediaPlayer2> > *sPlayers;
-
-void ensureInit_l() {
- if (sPlayers == NULL) {
- sPlayers = new SortedVector<wp<MediaPlayer2> >();
- }
-}
-
-void addPlayer(const wp<MediaPlayer2>& player) {
- Mutex::Autolock lock(sRecordLock);
- ensureInit_l();
- sPlayers->add(player);
-}
-
-void removePlayer(const wp<MediaPlayer2>& player) {
- Mutex::Autolock lock(sRecordLock);
- ensureInit_l();
- sPlayers->remove(player);
-}
-
-/**
- * The only arguments this understands right now are -c, -von and -voff,
- * which are parsed by ALooperRoster::dump()
- */
-status_t dumpPlayers(int fd, const Vector<String16>& args) {
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
- SortedVector< sp<MediaPlayer2> > players; //to serialise the mutex unlock & client destruction.
-
- {
- Mutex::Autolock lock(sRecordLock);
- ensureInit_l();
- for (int i = 0, n = sPlayers->size(); i < n; ++i) {
- sp<MediaPlayer2> p = (*sPlayers)[i].promote();
- if (p != 0) {
- p->dump(fd, args);
- }
- players.add(p);
- }
- }
-
- result.append(" Files opened and/or mapped:\n");
- snprintf(buffer, SIZE, "/proc/%d/maps", getpid());
- FILE *f = fopen(buffer, "r");
- if (f) {
- while (!feof(f)) {
- fgets(buffer, SIZE, f);
- if (strstr(buffer, " /storage/") ||
- strstr(buffer, " /system/sounds/") ||
- strstr(buffer, " /data/") ||
- strstr(buffer, " /system/media/")) {
- result.append(" ");
- result.append(buffer);
- }
- }
- fclose(f);
- } else {
- result.append("couldn't open ");
- result.append(buffer);
- result.append("\n");
- }
-
- snprintf(buffer, SIZE, "/proc/%d/fd", getpid());
- DIR *d = opendir(buffer);
- if (d) {
- struct dirent *ent;
- while((ent = readdir(d)) != NULL) {
- if (strcmp(ent->d_name,".") && strcmp(ent->d_name,"..")) {
- snprintf(buffer, SIZE, "/proc/%d/fd/%s", getpid(), ent->d_name);
- struct stat s;
- if (lstat(buffer, &s) == 0) {
- if ((s.st_mode & S_IFMT) == S_IFLNK) {
- char linkto[256];
- int len = readlink(buffer, linkto, sizeof(linkto));
- if(len > 0) {
- if(len > 255) {
- linkto[252] = '.';
- linkto[253] = '.';
- linkto[254] = '.';
- linkto[255] = 0;
- } else {
- linkto[len] = 0;
- }
- if (strstr(linkto, "/storage/") == linkto ||
- strstr(linkto, "/system/sounds/") == linkto ||
- strstr(linkto, "/data/") == linkto ||
- strstr(linkto, "/system/media/") == linkto) {
- result.append(" ");
- result.append(buffer);
- result.append(" -> ");
- result.append(linkto);
- result.append("\n");
- }
- }
- } else {
- result.append(" unexpected type for ");
- result.append(buffer);
- result.append("\n");
- }
- }
- }
- }
- closedir(d);
- } else {
- result.append("couldn't open ");
- result.append(buffer);
- result.append("\n");
- }
-
- gLooperRoster.dump(fd, args);
-
- bool dumpMem = false;
- bool unreachableMemory = false;
- for (size_t i = 0; i < args.size(); i++) {
- if (args[i] == String16("-m")) {
- dumpMem = true;
- } else if (args[i] == String16("--unreachable")) {
- unreachableMemory = true;
- }
- }
- if (dumpMem) {
- result.append("\nDumping memory:\n");
- std::string s = dumpMemoryAddresses(100 /* limit */);
- result.append(s.c_str(), s.size());
- }
- if (unreachableMemory) {
- result.append("\nDumping unreachable memory:\n");
- // TODO - should limit be an argument parameter?
- // TODO: enable GetUnreachableMemoryString if it's part of stable API
- //std::string s = GetUnreachableMemoryString(true /* contents */, 10000 /* limit */);
- //result.append(s.c_str(), s.size());
- }
-
- write(fd, result.string(), result.size());
- return NO_ERROR;
-}
-
-} // anonymous namespace
-
-//static
-sp<MediaPlayer2> MediaPlayer2::Create(int32_t sessionId, jobject context) {
- sp<MediaPlayer2> player = new MediaPlayer2(sessionId, context);
-
- if (!player->init()) {
- return NULL;
- }
-
- ALOGV("Create new player(%p)", player.get());
-
- addPlayer(player);
- return player;
-}
-
-// static
-status_t MediaPlayer2::DumpAll(int fd, const Vector<String16>& args) {
- return dumpPlayers(fd, args);
-}
-
-MediaPlayer2::MediaPlayer2(int32_t sessionId, jobject context) {
- ALOGV("constructor");
- mSrcId = 0;
- mLockThreadId = 0;
- mListener = NULL;
- mStreamType = AUDIO_STREAM_MUSIC;
- mAudioAttributes = NULL;
- mContext = new JObjectHolder(context);
- mCurrentPosition = -1;
- mCurrentSeekMode = MediaPlayer2SeekMode::SEEK_PREVIOUS_SYNC;
- mSeekPosition = -1;
- mSeekMode = MediaPlayer2SeekMode::SEEK_PREVIOUS_SYNC;
- mCurrentState = MEDIA_PLAYER2_IDLE;
- mTransitionToNext = false;
- mLoop = false;
- mVolume = 1.0;
- mVideoWidth = mVideoHeight = 0;
- mSendLevel = 0;
-
- mPid = AIBinder_getCallingPid();
- mUid = AIBinder_getCallingUid();
-
- mAudioOutput = new MediaPlayer2AudioOutput(sessionId, mUid, mPid, NULL /*attributes*/);
-}
-
-MediaPlayer2::~MediaPlayer2() {
- ALOGV("destructor");
- disconnect();
- removePlayer(this);
-}
-
-bool MediaPlayer2::init() {
- // TODO: after merge with NuPlayer2Driver, MediaPlayer2 will have its own
- // looper for notification.
- return true;
-}
-
-void MediaPlayer2::disconnect() {
- ALOGV("disconnect");
- sp<MediaPlayer2Interface> p;
- {
- Mutex::Autolock _l(mLock);
- p = mPlayer;
- mPlayer.clear();
- }
-
- if (p != 0) {
- p->setListener(NULL);
- p->reset();
- }
-
- {
- Mutex::Autolock _l(mLock);
- disconnectNativeWindow_l();
- }
-}
-
-void MediaPlayer2::clear_l() {
- mCurrentPosition = -1;
- mCurrentSeekMode = MediaPlayer2SeekMode::SEEK_PREVIOUS_SYNC;
- mSeekPosition = -1;
- mSeekMode = MediaPlayer2SeekMode::SEEK_PREVIOUS_SYNC;
- mVideoWidth = mVideoHeight = 0;
-}
-
-status_t MediaPlayer2::setListener(const sp<MediaPlayer2Listener>& listener) {
- ALOGV("setListener");
- Mutex::Autolock _l(mLock);
- mListener = listener;
- return NO_ERROR;
-}
-
-status_t MediaPlayer2::getSrcId(int64_t *srcId) {
- if (srcId == NULL) {
- return BAD_VALUE;
- }
-
- Mutex::Autolock _l(mLock);
- *srcId = mSrcId;
- return OK;
-}
-
-status_t MediaPlayer2::setDataSource(const sp<DataSourceDesc> &dsd) {
- if (dsd == NULL) {
- return BAD_VALUE;
- }
- // Microsecond is used in NuPlayer2.
- if (dsd->mStartPositionMs > DataSourceDesc::kMaxTimeMs) {
- dsd->mStartPositionMs = DataSourceDesc::kMaxTimeMs;
- ALOGW("setDataSource, start poistion clamped to %lld ms", (long long)dsd->mStartPositionMs);
- }
- if (dsd->mEndPositionMs > DataSourceDesc::kMaxTimeMs) {
- dsd->mEndPositionMs = DataSourceDesc::kMaxTimeMs;
- ALOGW("setDataSource, end poistion clamped to %lld ms", (long long)dsd->mStartPositionMs);
- }
- ALOGV("setDataSource type(%d), srcId(%lld)", dsd->mType, (long long)dsd->mId);
-
- sp<MediaPlayer2Interface> oldPlayer;
-
- {
- Mutex::Autolock _l(mLock);
- if (!((mCurrentState & MEDIA_PLAYER2_IDLE)
- || mCurrentState == MEDIA_PLAYER2_STATE_ERROR)) {
- ALOGE("setDataSource called in wrong state %d", mCurrentState);
- return INVALID_OPERATION;
- }
-
- sp<MediaPlayer2Interface> player = new NuPlayer2Driver(mPid, mUid, mContext);
- status_t err = player->initCheck();
- if (err != NO_ERROR) {
- ALOGE("Failed to create player object, initCheck failed(%d)", err);
- return err;
- }
-
- clear_l();
-
- player->setListener(new proxyListener(this));
- player->setAudioSink(mAudioOutput);
-
- err = player->setDataSource(dsd);
- if (err != OK) {
- ALOGE("setDataSource error: %d", err);
- return err;
- }
-
- sp<MediaPlayer2Interface> oldPlayer = mPlayer;
- mPlayer = player;
- mSrcId = dsd->mId;
- mCurrentState = MEDIA_PLAYER2_INITIALIZED;
- }
-
- if (oldPlayer != NULL) {
- oldPlayer->setListener(NULL);
- oldPlayer->reset();
- }
-
- return OK;
-}
-
-status_t MediaPlayer2::prepareNextDataSource(const sp<DataSourceDesc> &dsd) {
- if (dsd == NULL) {
- return BAD_VALUE;
- }
- ALOGV("prepareNextDataSource type(%d), srcId(%lld)", dsd->mType, (long long)dsd->mId);
-
- Mutex::Autolock _l(mLock);
- if (mPlayer == NULL) {
- ALOGE("prepareNextDataSource failed: state %X, mPlayer(%p)", mCurrentState, mPlayer.get());
- return INVALID_OPERATION;
- }
- return mPlayer->prepareNextDataSource(dsd);
-}
-
-status_t MediaPlayer2::playNextDataSource(int64_t srcId) {
- ALOGV("playNextDataSource srcId(%lld)", (long long)srcId);
-
- Mutex::Autolock _l(mLock);
- if (mPlayer == NULL) {
- ALOGE("playNextDataSource failed: state %X, mPlayer(%p)", mCurrentState, mPlayer.get());
- return INVALID_OPERATION;
- }
- mSrcId = srcId;
- mTransitionToNext = true;
- return mPlayer->playNextDataSource(srcId);
-}
-
-status_t MediaPlayer2::invoke(const PlayerMessage &request, PlayerMessage *reply) {
- Mutex::Autolock _l(mLock);
- const bool hasBeenInitialized =
- (mCurrentState != MEDIA_PLAYER2_STATE_ERROR) &&
- ((mCurrentState & MEDIA_PLAYER2_IDLE) != MEDIA_PLAYER2_IDLE);
- if ((mPlayer == NULL) || !hasBeenInitialized) {
- ALOGE("invoke() failed: wrong state %X, mPlayer(%p)", mCurrentState, mPlayer.get());
- return INVALID_OPERATION;
- }
- return mPlayer->invoke(request, reply);
-}
-
-void MediaPlayer2::disconnectNativeWindow_l() {
- if (mConnectedWindow != NULL && mConnectedWindow->getANativeWindow() != NULL) {
- status_t err = native_window_api_disconnect(
- mConnectedWindow->getANativeWindow(), NATIVE_WINDOW_API_MEDIA);
-
- if (err != OK) {
- ALOGW("nativeWindowDisconnect returned an error: %s (%d)",
- strerror(-err), err);
- }
- }
- mConnectedWindow.clear();
-}
-
-status_t MediaPlayer2::setVideoSurfaceTexture(const sp<ANativeWindowWrapper>& nww) {
- ANativeWindow *anw = (nww == NULL ? NULL : nww->getANativeWindow());
- ALOGV("setVideoSurfaceTexture(%p)", anw);
- Mutex::Autolock _l(mLock);
- if (mPlayer == 0) {
- return NO_INIT;
- }
-
- if (anw != NULL) {
- if (mConnectedWindow != NULL
- && mConnectedWindow->getANativeWindow() == anw) {
- return OK;
- }
- status_t err = native_window_api_connect(anw, NATIVE_WINDOW_API_MEDIA);
-
- if (err != OK) {
- ALOGE("setVideoSurfaceTexture failed: %d", err);
- // Note that we must do the reset before disconnecting from the ANW.
- // Otherwise queue/dequeue calls could be made on the disconnected
- // ANW, which may result in errors.
- mPlayer->reset();
- disconnectNativeWindow_l();
- return err;
- }
- }
-
- // Note that we must set the player's new GraphicBufferProducer before
- // disconnecting the old one. Otherwise queue/dequeue calls could be made
- // on the disconnected ANW, which may result in errors.
- status_t err = mPlayer->setVideoSurfaceTexture(nww);
-
- disconnectNativeWindow_l();
-
- if (err == OK) {
- mConnectedWindow = nww;
- mLock.unlock();
- } else if (anw != NULL) {
- mLock.unlock();
- status_t err = native_window_api_disconnect(anw, NATIVE_WINDOW_API_MEDIA);
-
- if (err != OK) {
- ALOGW("nativeWindowDisconnect returned an error: %s (%d)",
- strerror(-err), err);
- }
- }
-
- return err;
-}
-
-status_t MediaPlayer2::getBufferingSettings(BufferingSettings* buffering /* nonnull */) {
- ALOGV("getBufferingSettings");
-
- Mutex::Autolock _l(mLock);
- if (mPlayer == 0) {
- return NO_INIT;
- }
-
- status_t ret = mPlayer->getBufferingSettings(buffering);
- if (ret == NO_ERROR) {
- ALOGV("getBufferingSettings{%s}", buffering->toString().string());
- } else {
- ALOGE("getBufferingSettings returned %d", ret);
- }
- return ret;
-}
-
-status_t MediaPlayer2::setBufferingSettings(const BufferingSettings& buffering) {
- ALOGV("setBufferingSettings{%s}", buffering.toString().string());
-
- Mutex::Autolock _l(mLock);
- if (mPlayer == 0) {
- return NO_INIT;
- }
- return mPlayer->setBufferingSettings(buffering);
-}
-
-status_t MediaPlayer2::setAudioAttributes_l(const jobject attributes) {
- if (mAudioOutput != NULL) {
- mAudioOutput->setAudioAttributes(attributes);
- }
- return NO_ERROR;
-}
-
-status_t MediaPlayer2::prepareAsync() {
- ALOGV("prepareAsync");
- Mutex::Autolock _l(mLock);
- if ((mPlayer != 0) && (mCurrentState & MEDIA_PLAYER2_INITIALIZED)) {
- if (mAudioAttributes != NULL) {
- status_t err = setAudioAttributes_l(mAudioAttributes->getJObject());
- if (err != OK) {
- return err;
- }
- }
- mCurrentState = MEDIA_PLAYER2_PREPARING;
- return mPlayer->prepareAsync();
- }
- ALOGE("prepareAsync called in state %d, mPlayer(%p)", mCurrentState, mPlayer.get());
- return INVALID_OPERATION;
-}
-
-status_t MediaPlayer2::start() {
- ALOGV("start");
-
- status_t ret = NO_ERROR;
- Mutex::Autolock _l(mLock);
-
- mLockThreadId = getThreadId();
-
- if (mCurrentState & MEDIA_PLAYER2_STARTED) {
- ret = NO_ERROR;
- } else if ( (mPlayer != 0) && ( mCurrentState & ( MEDIA_PLAYER2_PREPARED |
- MEDIA_PLAYER2_PLAYBACK_COMPLETE | MEDIA_PLAYER2_PAUSED ) ) ) {
- mPlayer->setLooping(mLoop);
-
- if (mAudioOutput != 0) {
- mAudioOutput->setVolume(mVolume);
- }
-
- if (mAudioOutput != 0) {
- mAudioOutput->setAuxEffectSendLevel(mSendLevel);
- }
- mCurrentState = MEDIA_PLAYER2_STARTED;
- ret = mPlayer->start();
- if (ret != NO_ERROR) {
- mCurrentState = MEDIA_PLAYER2_STATE_ERROR;
- } else {
- if (mCurrentState == MEDIA_PLAYER2_PLAYBACK_COMPLETE) {
- ALOGV("playback completed immediately following start()");
- }
- }
- } else {
- ALOGE("start called in state %d, mPlayer(%p)", mCurrentState, mPlayer.get());
- ret = INVALID_OPERATION;
- }
-
- mLockThreadId = 0;
-
- return ret;
-}
-
-status_t MediaPlayer2::pause() {
- ALOGV("pause");
- Mutex::Autolock _l(mLock);
- if (mCurrentState & (MEDIA_PLAYER2_PAUSED|MEDIA_PLAYER2_PLAYBACK_COMPLETE))
- return NO_ERROR;
- if ((mPlayer != 0) && (mCurrentState & (MEDIA_PLAYER2_STARTED | MEDIA_PLAYER2_PREPARED))) {
- status_t ret = mPlayer->pause();
- if (ret != NO_ERROR) {
- mCurrentState = MEDIA_PLAYER2_STATE_ERROR;
- } else {
- mCurrentState = MEDIA_PLAYER2_PAUSED;
- mTransitionToNext = false;
- }
- return ret;
- }
- ALOGE("pause called in state %d, mPlayer(%p)", mCurrentState, mPlayer.get());
- return INVALID_OPERATION;
-}
-
-bool MediaPlayer2::isPlaying() {
- Mutex::Autolock _l(mLock);
- if (mPlayer != 0) {
- bool temp = mPlayer->isPlaying();
- ALOGV("isPlaying: %d", temp);
- if ((mCurrentState & MEDIA_PLAYER2_STARTED) && ! temp) {
- ALOGE("internal/external state mismatch corrected");
- mCurrentState = MEDIA_PLAYER2_PAUSED;
- } else if ((mCurrentState & MEDIA_PLAYER2_PAUSED) && temp) {
- ALOGE("internal/external state mismatch corrected");
- mCurrentState = MEDIA_PLAYER2_STARTED;
- }
- return temp;
- }
- ALOGV("isPlaying: no active player");
- return false;
-}
-
-mediaplayer2_states MediaPlayer2::getState() {
- Mutex::Autolock _l(mLock);
- if (mCurrentState & MEDIA_PLAYER2_STATE_ERROR) {
- return MEDIAPLAYER2_STATE_ERROR;
- }
- if (mPlayer == 0
- || (mCurrentState &
- (MEDIA_PLAYER2_IDLE | MEDIA_PLAYER2_INITIALIZED | MEDIA_PLAYER2_PREPARING))) {
- return MEDIAPLAYER2_STATE_IDLE;
- }
- if (mCurrentState & MEDIA_PLAYER2_STARTED) {
- return MEDIAPLAYER2_STATE_PLAYING;
- }
- if (mCurrentState & (MEDIA_PLAYER2_PAUSED | MEDIA_PLAYER2_PLAYBACK_COMPLETE)) {
- return MEDIAPLAYER2_STATE_PAUSED;
- }
- // now only mCurrentState & MEDIA_PLAYER2_PREPARED is true
- return MEDIAPLAYER2_STATE_PREPARED;
-}
-
-status_t MediaPlayer2::setPlaybackSettings(const AudioPlaybackRate& rate) {
- ALOGV("setPlaybackSettings: %f %f %d %d",
- rate.mSpeed, rate.mPitch, rate.mFallbackMode, rate.mStretchMode);
- // Negative speed and pitch does not make sense. Further validation will
- // be done by the respective mediaplayers.
- if (rate.mSpeed <= 0.f || rate.mPitch < 0.f) {
- return BAD_VALUE;
- }
- Mutex::Autolock _l(mLock);
- if (mPlayer == 0) {
- return INVALID_OPERATION;
- }
-
- status_t err = mPlayer->setPlaybackSettings(rate);
- return err;
-}
-
-status_t MediaPlayer2::getPlaybackSettings(AudioPlaybackRate* rate /* nonnull */) {
- Mutex::Autolock _l(mLock);
- if (mPlayer == 0) {
- return INVALID_OPERATION;
- }
- status_t ret = mPlayer->getPlaybackSettings(rate);
- if (ret == NO_ERROR) {
- ALOGV("getPlaybackSettings(%f, %f, %d, %d)",
- rate->mSpeed, rate->mPitch, rate->mFallbackMode, rate->mStretchMode);
- } else {
- ALOGV("getPlaybackSettings returned %d", ret);
- }
- return ret;
-}
-
-status_t MediaPlayer2::setSyncSettings(const AVSyncSettings& sync, float videoFpsHint) {
- ALOGV("setSyncSettings: %u %u %f %f",
- sync.mSource, sync.mAudioAdjustMode, sync.mTolerance, videoFpsHint);
- Mutex::Autolock _l(mLock);
- if (mPlayer == 0) return INVALID_OPERATION;
- return mPlayer->setSyncSettings(sync, videoFpsHint);
-}
-
-status_t MediaPlayer2::getSyncSettings(
- AVSyncSettings* sync /* nonnull */, float* videoFps /* nonnull */) {
- Mutex::Autolock _l(mLock);
- if (mPlayer == 0) {
- return INVALID_OPERATION;
- }
- status_t ret = mPlayer->getSyncSettings(sync, videoFps);
- if (ret == NO_ERROR) {
- ALOGV("getSyncSettings(%u, %u, %f, %f)",
- sync->mSource, sync->mAudioAdjustMode, sync->mTolerance, *videoFps);
- } else {
- ALOGV("getSyncSettings returned %d", ret);
- }
- return ret;
-
-}
-
-status_t MediaPlayer2::getVideoWidth(int *w) {
- ALOGV("getVideoWidth");
- Mutex::Autolock _l(mLock);
- if (mPlayer == 0) {
- return INVALID_OPERATION;
- }
- *w = mVideoWidth;
- return NO_ERROR;
-}
-
-status_t MediaPlayer2::getVideoHeight(int *h) {
- ALOGV("getVideoHeight");
- Mutex::Autolock _l(mLock);
- if (mPlayer == 0) {
- return INVALID_OPERATION;
- }
- *h = mVideoHeight;
- return NO_ERROR;
-}
-
-status_t MediaPlayer2::getCurrentPosition(int64_t *msec) {
- ALOGV("getCurrentPosition");
- Mutex::Autolock _l(mLock);
- if (mPlayer == 0) {
- return INVALID_OPERATION;
- }
- if (mCurrentPosition >= 0) {
- ALOGV("Using cached seek position: %lld", (long long)mCurrentPosition);
- *msec = mCurrentPosition;
- return NO_ERROR;
- }
- status_t ret = mPlayer->getCurrentPosition(msec);
- if (ret == NO_ERROR) {
- ALOGV("getCurrentPosition = %lld", (long long)*msec);
- } else {
- ALOGE("getCurrentPosition returned %d", ret);
- }
- return ret;
-}
-
-status_t MediaPlayer2::getDuration(int64_t srcId, int64_t *msec) {
- Mutex::Autolock _l(mLock);
- // TODO: cache duration for currentSrcId and nextSrcId, and return correct
- // value for nextSrcId.
- if (srcId != mSrcId) {
- *msec = -1;
- return OK;
- }
-
- ALOGV("getDuration_l");
- bool isValidState = (mCurrentState & (MEDIA_PLAYER2_PREPARED | MEDIA_PLAYER2_STARTED |
- MEDIA_PLAYER2_PAUSED | MEDIA_PLAYER2_PLAYBACK_COMPLETE));
- if (mPlayer == 0 || !isValidState) {
- ALOGE("Attempt to call getDuration in wrong state: mPlayer=%p, mCurrentState=%u",
- mPlayer.get(), mCurrentState);
- return INVALID_OPERATION;
- }
- int64_t durationMs;
- status_t ret = mPlayer->getDuration(&durationMs);
-
- if (ret == NO_ERROR) {
- ALOGV("getDuration = %lld", (long long)durationMs);
- } else {
- ALOGE("getDuration returned %d", ret);
- // Do not enter error state just because no duration was available.
- durationMs = -1;
- }
-
- if (msec) {
- *msec = durationMs;
- }
- return OK;
-}
-
-status_t MediaPlayer2::seekTo_l(int64_t msec, MediaPlayer2SeekMode mode) {
- ALOGV("seekTo (%lld, %d)", (long long)msec, mode);
- if ((mPlayer == 0) || !(mCurrentState & (MEDIA_PLAYER2_STARTED | MEDIA_PLAYER2_PREPARED |
- MEDIA_PLAYER2_PAUSED | MEDIA_PLAYER2_PLAYBACK_COMPLETE))) {
- ALOGE("Attempt to perform seekTo in wrong state: mPlayer=%p, mCurrentState=%u",
- mPlayer.get(), mCurrentState);
- return INVALID_OPERATION;
- }
- if (msec < 0) {
- ALOGW("Attempt to seek to invalid position: %lld", (long long)msec);
- msec = 0;
- }
-
- int64_t durationMs;
- status_t err = mPlayer->getDuration(&durationMs);
-
- if (err != OK) {
- ALOGW("Stream has no duration and is therefore not seekable.");
- return err;
- }
-
- if (msec > durationMs) {
- ALOGW("Attempt to seek to past end of file: request = %lld, durationMs = %lld",
- (long long)msec, (long long)durationMs);
-
- msec = durationMs;
- }
-
- // cache duration
- mCurrentPosition = msec;
- mCurrentSeekMode = mode;
- if (mSeekPosition < 0) {
- mSeekPosition = msec;
- mSeekMode = mode;
- return mPlayer->seekTo(msec, mode);
- }
- ALOGV("Seek in progress - queue up seekTo[%lld, %d]", (long long)msec, mode);
- return NO_ERROR;
-}
-
-status_t MediaPlayer2::seekTo(int64_t msec, MediaPlayer2SeekMode mode) {
- mLockThreadId = getThreadId();
- Mutex::Autolock _l(mLock);
- status_t result = seekTo_l(msec, mode);
- mLockThreadId = 0;
-
- return result;
-}
-
-status_t MediaPlayer2::notifyAt(int64_t mediaTimeUs) {
- Mutex::Autolock _l(mLock);
- if (mPlayer != 0) {
- return INVALID_OPERATION;
- }
-
- return mPlayer->notifyAt(mediaTimeUs);
-}
-
-status_t MediaPlayer2::reset_l() {
- mLoop = false;
- if (mCurrentState == MEDIA_PLAYER2_IDLE) {
- return NO_ERROR;
- }
- if (mPlayer != 0) {
- status_t ret = mPlayer->reset();
- if (ret != NO_ERROR) {
- ALOGE("reset() failed with return code (%d)", ret);
- mCurrentState = MEDIA_PLAYER2_STATE_ERROR;
- } else {
- mPlayer->setListener(NULL);
- mCurrentState = MEDIA_PLAYER2_IDLE;
- mTransitionToNext = false;
- }
- // setDataSource has to be called again to create a
- // new mediaplayer.
- mPlayer = 0;
- return ret;
- }
- clear_l();
- return NO_ERROR;
-}
-
-status_t MediaPlayer2::reset() {
- ALOGV("reset");
- mLockThreadId = getThreadId();
- Mutex::Autolock _l(mLock);
- status_t result = reset_l();
- mLockThreadId = 0;
-
- return result;
-}
-
-status_t MediaPlayer2::setAudioStreamType(audio_stream_type_t type) {
- ALOGV("MediaPlayer2::setAudioStreamType");
- Mutex::Autolock _l(mLock);
- if (mStreamType == type) return NO_ERROR;
- if (mCurrentState & ( MEDIA_PLAYER2_PREPARED | MEDIA_PLAYER2_STARTED |
- MEDIA_PLAYER2_PAUSED | MEDIA_PLAYER2_PLAYBACK_COMPLETE ) ) {
- // Can't change the stream type after prepare
- ALOGE("setAudioStream called in state %d", mCurrentState);
- return INVALID_OPERATION;
- }
- // cache
- mStreamType = type;
- return OK;
-}
-
-status_t MediaPlayer2::getAudioStreamType(audio_stream_type_t *type) {
- ALOGV("getAudioStreamType");
- Mutex::Autolock _l(mLock);
- *type = mStreamType;
- return OK;
-}
-
-status_t MediaPlayer2::setLooping(int loop) {
- ALOGV("MediaPlayer2::setLooping");
- Mutex::Autolock _l(mLock);
- mLoop = (loop != 0);
- if (mPlayer != 0) {
- return mPlayer->setLooping(loop);
- }
- return OK;
-}
-
-bool MediaPlayer2::isLooping() {
- ALOGV("isLooping");
- Mutex::Autolock _l(mLock);
- if (mPlayer != 0) {
- return mLoop;
- }
- ALOGV("isLooping: no active player");
- return false;
-}
-
-status_t MediaPlayer2::setVolume(float volume) {
- ALOGV("MediaPlayer2::setVolume(%f)", volume);
- Mutex::Autolock _l(mLock);
- mVolume = volume;
- if (mAudioOutput != 0) {
- mAudioOutput->setVolume(volume);
- }
- return OK;
-}
-
-status_t MediaPlayer2::setAudioSessionId(int32_t sessionId) {
- ALOGV("MediaPlayer2::setAudioSessionId(%d)", sessionId);
- Mutex::Autolock _l(mLock);
- if (!(mCurrentState & MEDIA_PLAYER2_IDLE)) {
- ALOGE("setAudioSessionId called in state %d", mCurrentState);
- return INVALID_OPERATION;
- }
- if (sessionId < 0) {
- return BAD_VALUE;
- }
- if (mAudioOutput != NULL && sessionId != mAudioOutput->getSessionId()) {
- mAudioOutput->setSessionId(sessionId);
- }
- return NO_ERROR;
-}
-
-int32_t MediaPlayer2::getAudioSessionId() {
- Mutex::Autolock _l(mLock);
- if (mAudioOutput != NULL) {
- return mAudioOutput->getSessionId();
- }
- return 0;
-}
-
-status_t MediaPlayer2::setAuxEffectSendLevel(float level) {
- ALOGV("MediaPlayer2::setAuxEffectSendLevel(%f)", level);
- Mutex::Autolock _l(mLock);
- mSendLevel = level;
- if (mAudioOutput != 0) {
- return mAudioOutput->setAuxEffectSendLevel(level);
- }
- return OK;
-}
-
-status_t MediaPlayer2::attachAuxEffect(int effectId) {
- ALOGV("MediaPlayer2::attachAuxEffect(%d)", effectId);
- Mutex::Autolock _l(mLock);
- if (mAudioOutput == 0 ||
- (mCurrentState & MEDIA_PLAYER2_IDLE) ||
- (mCurrentState == MEDIA_PLAYER2_STATE_ERROR )) {
- ALOGE("attachAuxEffect called in state %d, mPlayer(%p)", mCurrentState, mPlayer.get());
- return INVALID_OPERATION;
- }
-
- return mAudioOutput->attachAuxEffect(effectId);
-}
-
-// always call with lock held
-status_t MediaPlayer2::checkState_l() {
- if (mCurrentState & ( MEDIA_PLAYER2_PREPARED | MEDIA_PLAYER2_STARTED |
- MEDIA_PLAYER2_PAUSED | MEDIA_PLAYER2_PLAYBACK_COMPLETE) ) {
- // Can't change the audio attributes after prepare
- ALOGE("trying to set audio attributes called in state %d", mCurrentState);
- return INVALID_OPERATION;
- }
- return OK;
-}
-
-status_t MediaPlayer2::setAudioAttributes(const jobject attributes) {
- ALOGV("MediaPlayer2::setAudioAttributes");
- status_t status = INVALID_OPERATION;
- Mutex::Autolock _l(mLock);
- if (checkState_l() != OK) {
- return status;
- }
- mAudioAttributes = new JObjectHolder(attributes);
- status = setAudioAttributes_l(attributes);
- return status;
-}
-
-jobject MediaPlayer2::getAudioAttributes() {
- ALOGV("MediaPlayer2::getAudioAttributes)");
- Mutex::Autolock _l(mLock);
- return mAudioAttributes != NULL ? mAudioAttributes->getJObject() : NULL;
-}
-
-status_t MediaPlayer2::getParameter(int key, Parcel *reply) {
- ALOGV("MediaPlayer2::getParameter(%d)", key);
- Mutex::Autolock _l(mLock);
- if (mPlayer == NULL) {
- ALOGV("getParameter: no active player");
- return INVALID_OPERATION;
- }
-
- status_t status = mPlayer->getParameter(key, reply);
- if (status != OK) {
- ALOGD("getParameter returns %d", status);
- }
- return status;
-}
-
-// for mediametrics
-status_t MediaPlayer2::getMetrics(char **buffer, size_t *length) {
- ALOGD("MediaPlayer2::getMetrics()");
- Mutex::Autolock _l(mLock);
- if (mPlayer == NULL) {
- ALOGV("getMetrics: no active player");
- return INVALID_OPERATION;
- }
-
- status_t status = mPlayer->getMetrics(buffer, length);
- if (status != OK) {
- ALOGD("getMetrics returns %d", status);
- }
- return status;
-}
-
-void MediaPlayer2::notify(int64_t srcId, int msg, int ext1, int ext2, const PlayerMessage *obj) {
- ALOGV("message received srcId=%lld, msg=%d, ext1=%d, ext2=%d",
- (long long)srcId, msg, ext1, ext2);
-
- bool send = true;
- bool locked = false;
-
- // TODO: In the future, we might be on the same thread if the app is
- // running in the same process as the media server. In that case,
- // this will deadlock.
- //
- // The threadId hack below works around this for the care of prepare,
- // seekTo, start, and reset within the same process.
- // FIXME: Remember, this is a hack, it's not even a hack that is applied
- // consistently for all use-cases, this needs to be revisited.
- if (mLockThreadId != getThreadId()) {
- mLock.lock();
- locked = true;
- }
-
- // Allows calls from JNI in idle state to notify errors
- if (!(msg == MEDIA2_ERROR && mCurrentState == MEDIA_PLAYER2_IDLE) && mPlayer == 0) {
- ALOGV("notify(%lld, %d, %d, %d) callback on disconnected mediaplayer",
- (long long)srcId, msg, ext1, ext2);
- if (locked) mLock.unlock(); // release the lock when done.
- return;
- }
-
- switch (msg) {
- case MEDIA2_NOP: // interface test message
- break;
- case MEDIA2_PREPARED:
- ALOGV("MediaPlayer2::notify() prepared, srcId=%lld", (long long)srcId);
- if (srcId == mSrcId) {
- mCurrentState = MEDIA_PLAYER2_PREPARED;
- }
- break;
- case MEDIA2_DRM_INFO:
- ALOGV("MediaPlayer2::notify() MEDIA2_DRM_INFO(%lld, %d, %d, %d, %p)",
- (long long)srcId, msg, ext1, ext2, obj);
- break;
- case MEDIA2_PLAYBACK_COMPLETE:
- ALOGV("playback complete");
- if (mCurrentState == MEDIA_PLAYER2_IDLE) {
- ALOGE("playback complete in idle state");
- }
- if (!mLoop && srcId == mSrcId) {
- mCurrentState = MEDIA_PLAYER2_PLAYBACK_COMPLETE;
- }
- break;
- case MEDIA2_ERROR:
- // Always log errors.
- // ext1: Media framework error code.
- // ext2: Implementation dependant error code.
- ALOGE("error (%d, %d)", ext1, ext2);
- mCurrentState = MEDIA_PLAYER2_STATE_ERROR;
- break;
- case MEDIA2_INFO:
- // ext1: Media framework error code.
- // ext2: Implementation dependant error code.
- if (ext1 != MEDIA2_INFO_VIDEO_TRACK_LAGGING) {
- ALOGW("info/warning (%d, %d)", ext1, ext2);
-
- if (ext1 == MEDIA2_INFO_DATA_SOURCE_START && srcId == mSrcId && mTransitionToNext) {
- mCurrentState = MEDIA_PLAYER2_STARTED;
- mTransitionToNext = false;
- }
- }
- break;
- case MEDIA2_SEEK_COMPLETE:
- ALOGV("Received seek complete");
- if (mSeekPosition != mCurrentPosition || (mSeekMode != mCurrentSeekMode)) {
- ALOGV("Executing queued seekTo(%lld, %d)",
- (long long)mCurrentPosition, mCurrentSeekMode);
- mSeekPosition = -1;
- mSeekMode = MediaPlayer2SeekMode::SEEK_PREVIOUS_SYNC;
- seekTo_l(mCurrentPosition, mCurrentSeekMode);
- }
- else {
- ALOGV("All seeks complete - return to regularly scheduled program");
- mCurrentPosition = mSeekPosition = -1;
- mCurrentSeekMode = mSeekMode = MediaPlayer2SeekMode::SEEK_PREVIOUS_SYNC;
- }
- break;
- case MEDIA2_BUFFERING_UPDATE:
- ALOGV("buffering %d", ext1);
- break;
- case MEDIA2_SET_VIDEO_SIZE:
- ALOGV("New video size %d x %d", ext1, ext2);
- mVideoWidth = ext1;
- mVideoHeight = ext2;
- break;
- case MEDIA2_NOTIFY_TIME:
- ALOGV("Received notify time message");
- break;
- case MEDIA2_TIMED_TEXT:
- ALOGV("Received timed text message");
- break;
- case MEDIA2_SUBTITLE_DATA:
- ALOGV("Received subtitle data message");
- break;
- case MEDIA2_META_DATA:
- ALOGV("Received timed metadata message");
- break;
- default:
- ALOGV("unrecognized message: (%d, %d, %d)", msg, ext1, ext2);
- break;
- }
-
- sp<MediaPlayer2Listener> listener = mListener;
- if (locked) mLock.unlock();
-
- // this prevents re-entrant calls into client code
- if ((listener != 0) && send) {
- Mutex::Autolock _l(mNotifyLock);
- ALOGV("callback application");
- listener->notify(srcId, msg, ext1, ext2, obj);
- ALOGV("back from callback");
- }
-}
-
-// Modular DRM
-status_t MediaPlayer2::prepareDrm(
- int64_t srcId, const uint8_t uuid[16], const Vector<uint8_t>& drmSessionId) {
- // TODO change to ALOGV
- ALOGD("prepareDrm: uuid: %p drmSessionId: %p(%zu)", uuid,
- drmSessionId.array(), drmSessionId.size());
- Mutex::Autolock _l(mLock);
- if (mPlayer == NULL) {
- return NO_INIT;
- }
-
- // Only allowed it in player's preparing/prepared state.
- // We get here only if MEDIA_DRM_INFO has already arrived (e.g., prepare is half-way through or
- // completed) so the state change to "prepared" might not have happened yet (e.g., buffering).
- // Still, we can allow prepareDrm for the use case of being called in OnDrmInfoListener.
- if (!(mCurrentState & (MEDIA_PLAYER2_PREPARING | MEDIA_PLAYER2_PREPARED))) {
- ALOGW("prepareDrm(%lld) called in non-prepare state(%d)", (long long)srcId, mCurrentState);
- if (srcId == mSrcId) {
- return INVALID_OPERATION;
- }
- }
-
- if (drmSessionId.isEmpty()) {
- ALOGE("prepareDrm: Unexpected. Can't proceed with crypto. Empty drmSessionId.");
- return INVALID_OPERATION;
- }
-
- // Passing down to mediaserver mainly for creating the crypto
- status_t status = mPlayer->prepareDrm(srcId, uuid, drmSessionId);
- ALOGE_IF(status != OK, "prepareDrm: Failed at mediaserver with ret: %d", status);
-
- // TODO change to ALOGV
- ALOGD("prepareDrm: mediaserver::prepareDrm ret=%d", status);
-
- return status;
-}
-
-status_t MediaPlayer2::releaseDrm(int64_t srcId) {
- Mutex::Autolock _l(mLock);
- if (mPlayer == NULL) {
- return NO_INIT;
- }
-
- // Not allowing releaseDrm in an active/resumable state
- if (mCurrentState & (MEDIA_PLAYER2_STARTED |
- MEDIA_PLAYER2_PAUSED |
- MEDIA_PLAYER2_PLAYBACK_COMPLETE |
- MEDIA_PLAYER2_STATE_ERROR)) {
- ALOGE("releaseDrm Unexpected state %d. Can only be called in stopped/idle.", mCurrentState);
- return INVALID_OPERATION;
- }
-
- status_t status = mPlayer->releaseDrm(srcId);
- // TODO change to ALOGV
- ALOGD("releaseDrm: mediaserver::releaseDrm ret: %d", status);
- if (status != OK) {
- ALOGE("releaseDrm: Failed at mediaserver with ret: %d", status);
- // Overriding to OK so the client proceed with its own cleanup
- // Client can't do more cleanup. mediaserver release its crypto at end of session anyway.
- status = OK;
- }
-
- return status;
-}
-
-status_t MediaPlayer2::setPreferredDevice(jobject device) {
- Mutex::Autolock _l(mLock);
- if (mAudioOutput == NULL) {
- ALOGV("setPreferredDevice: audio sink not init");
- return NO_INIT;
- }
- return mAudioOutput->setPreferredDevice(device);
-}
-
-jobject MediaPlayer2::getRoutedDevice() {
- Mutex::Autolock _l(mLock);
- if (mAudioOutput == NULL) {
- ALOGV("getRoutedDevice: audio sink not init");
- return nullptr;
- }
- return mAudioOutput->getRoutedDevice();
-}
-
-status_t MediaPlayer2::addAudioDeviceCallback(jobject routingDelegate) {
- Mutex::Autolock _l(mLock);
- if (mAudioOutput == NULL) {
- ALOGV("addAudioDeviceCallback: player not init");
- return NO_INIT;
- }
- return mAudioOutput->addAudioDeviceCallback(routingDelegate);
-}
-
-status_t MediaPlayer2::removeAudioDeviceCallback(jobject listener) {
- Mutex::Autolock _l(mLock);
- if (mAudioOutput == NULL) {
- ALOGV("addAudioDeviceCallback: player not init");
- return NO_INIT;
- }
- return mAudioOutput->removeAudioDeviceCallback(listener);
-}
-
-status_t MediaPlayer2::dump(int fd, const Vector<String16>& args) {
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
- result.append(" MediaPlayer2\n");
- snprintf(buffer, 255, " pid(%d), looping(%s)\n", mPid, mLoop?"true": "false");
- result.append(buffer);
-
- sp<MediaPlayer2Interface> player;
- sp<MediaPlayer2AudioOutput> audioOutput;
- bool locked = false;
- for (int i = 0; i < kDumpLockRetries; ++i) {
- if (mLock.tryLock() == NO_ERROR) {
- locked = true;
- break;
- }
- usleep(kDumpLockSleepUs);
- }
-
- if (locked) {
- player = mPlayer;
- audioOutput = mAudioOutput;
- mLock.unlock();
- } else {
- result.append(" lock is taken, no dump from player and audio output\n");
- }
- write(fd, result.string(), result.size());
-
- if (player != NULL) {
- player->dump(fd, args);
- }
- if (audioOutput != 0) {
- audioOutput->dump(fd, args);
- }
- write(fd, "\n", 1);
- return NO_ERROR;
-}
-
-} // namespace android
diff --git a/media/libmediaplayer2/nuplayer2/Android.bp b/media/libmediaplayer2/nuplayer2/Android.bp
deleted file mode 100644
index 0f69b2e..0000000
--- a/media/libmediaplayer2/nuplayer2/Android.bp
+++ /dev/null
@@ -1,72 +0,0 @@
-cc_library_static {
-
- srcs: [
- "JMediaPlayer2Utils.cpp",
- "JWakeLock.cpp",
- "GenericSource2.cpp",
- "HTTPLiveSource2.cpp",
- "NuPlayer2.cpp",
- "NuPlayer2CCDecoder.cpp",
- "NuPlayer2Decoder.cpp",
- "NuPlayer2DecoderBase.cpp",
- "NuPlayer2DecoderPassThrough.cpp",
- "NuPlayer2Driver.cpp",
- "NuPlayer2Drm.cpp",
- "NuPlayer2Renderer.cpp",
- "RTSPSource2.cpp",
- ],
-
- header_libs: [
- "libbase_headers",
- "libmediaplayer2_headers",
- "media_plugin_headers",
- ],
-
- include_dirs: [
- "frameworks/av/media/libstagefright",
- "frameworks/av/media/libstagefright/httplive",
- "frameworks/av/media/libstagefright/include",
- "frameworks/av/media/libstagefright/mpeg2ts",
- "frameworks/av/media/libstagefright/rtsp",
- "frameworks/av/media/libstagefright/timedtext",
- "frameworks/av/media/ndk",
- "frameworks/base/core/jni",
- ],
-
- cflags: [
- "-Werror",
- "-Wall",
- ],
-
- product_variables: {
- debuggable: {
- cflags: [
- "-DENABLE_STAGEFRIGHT_EXPERIMENTS",
- ],
- }
- },
-
- shared_libs: [
- "libbinder",
- "libui",
- "libgui",
- "libmedia",
- "libmediametrics",
- "libmediandk",
- "libmediandk_utils",
- "libpowermanager",
- ],
-
- static_libs: [
- "libmedia_helper",
- "libmediaplayer2-protos",
- "libmedia2_jni_core",
- ],
-
- name: "libstagefright_nuplayer2",
-
- sanitize: {
- cfi: true,
- },
-
-}
diff --git a/media/libmediaplayer2/nuplayer2/GenericSource2.cpp b/media/libmediaplayer2/nuplayer2/GenericSource2.cpp
deleted file mode 100644
index 9552580..0000000
--- a/media/libmediaplayer2/nuplayer2/GenericSource2.cpp
+++ /dev/null
@@ -1,1547 +0,0 @@
-/*
- * Copyright 2017 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "GenericSource2"
-
-#include "GenericSource2.h"
-#include "NuPlayer2Drm.h"
-
-#include "AnotherPacketSource.h"
-#include <cutils/properties.h>
-#include <media/DataSource.h>
-#include <media/MediaBufferHolder.h>
-#include <media/NdkWrapper.h>
-#include <media/stagefright/foundation/ABuffer.h>
-#include <media/stagefright/foundation/ADebug.h>
-#include <media/stagefright/foundation/AMessage.h>
-#include <media/stagefright/MediaBuffer.h>
-#include <media/stagefright/MediaClock.h>
-#include <media/stagefright/MediaDefs.h>
-#include <media/stagefright/MetaData.h>
-#include <media/stagefright/NdkUtils.h>
-#include <media/stagefright/Utils.h>
-
-namespace android {
-
-static const int kInitialMarkMs = 5000; // 5secs
-
-//static const int kPausePlaybackMarkMs = 2000; // 2secs
-static const int kResumePlaybackMarkMs = 15000; // 15secs
-
-NuPlayer2::GenericSource2::GenericSource2(
- const sp<AMessage> ¬ify,
- uid_t uid,
- const sp<MediaClock> &mediaClock)
- : Source(notify),
- mAudioTimeUs(0),
- mAudioLastDequeueTimeUs(0),
- mVideoTimeUs(0),
- mVideoLastDequeueTimeUs(0),
- mPrevBufferPercentage(-1),
- mPollBufferingGeneration(0),
- mSentPauseOnBuffering(false),
- mAudioDataGeneration(0),
- mVideoDataGeneration(0),
- mFetchSubtitleDataGeneration(0),
- mFetchTimedTextDataGeneration(0),
- mDurationUs(-1ll),
- mAudioIsVorbis(false),
- mIsSecure(false),
- mIsStreaming(false),
- mUID(uid),
- mMediaClock(mediaClock),
- mFd(-1),
- mBitrate(-1ll),
- mPendingReadBufferTypes(0) {
- ALOGV("GenericSource2");
- CHECK(mediaClock != NULL);
-
- mBufferingSettings.mInitialMarkMs = kInitialMarkMs;
- mBufferingSettings.mResumePlaybackMarkMs = kResumePlaybackMarkMs;
- resetDataSource();
-}
-
-void NuPlayer2::GenericSource2::resetDataSource() {
- ALOGV("resetDataSource");
-
- mDisconnected = false;
- mUri.clear();
- mUriHeaders.clear();
- if (mFd >= 0) {
- close(mFd);
- mFd = -1;
- }
- mOffset = 0;
- mLength = 0;
- mStarted = false;
- mPreparing = false;
-
- mIsDrmProtected = false;
- mIsDrmReleased = false;
- mIsSecure = false;
- mMimes.clear();
-}
-
-status_t NuPlayer2::GenericSource2::setDataSource(
- const char *url,
- const KeyedVector<String8, String8> *headers) {
- Mutex::Autolock _l(mLock);
- ALOGV("setDataSource url: %s", url);
-
- resetDataSource();
-
- mUri = url;
-
- if (headers) {
- mUriHeaders = *headers;
- }
-
- // delay data source creation to prepareAsync() to avoid blocking
- // the calling thread in setDataSource for any significant time.
- return OK;
-}
-
-status_t NuPlayer2::GenericSource2::setDataSource(
- int fd, int64_t offset, int64_t length) {
- Mutex::Autolock _l(mLock);
- ALOGV("setDataSource %d/%lld/%lld", fd, (long long)offset, (long long)length);
-
- resetDataSource();
-
- mFd = dup(fd);
- mOffset = offset;
- mLength = length;
-
- // delay data source creation to prepareAsync() to avoid blocking
- // the calling thread in setDataSource for any significant time.
- return OK;
-}
-
-status_t NuPlayer2::GenericSource2::setDataSource(const sp<DataSource>& source) {
- Mutex::Autolock _l(mLock);
- ALOGV("setDataSource (source: %p)", source.get());
-
- resetDataSource();
- mDataSourceWrapper = new AMediaDataSourceWrapper(source);
- return OK;
-}
-
-sp<MetaData> NuPlayer2::GenericSource2::getFileFormatMeta() const {
- Mutex::Autolock _l(mLock);
- return mFileMeta;
-}
-
-status_t NuPlayer2::GenericSource2::initFromDataSource() {
- mExtractor = new AMediaExtractorWrapper(AMediaExtractor_new());
- CHECK(mFd >=0 || mDataSourceWrapper != NULL);
- sp<AMediaDataSourceWrapper> aSourceWrapper = mDataSourceWrapper;
- const int fd = mFd;
-
- mLock.unlock();
- // This might take long time if data source is not reliable.
- status_t err;
- if (aSourceWrapper != NULL) {
- err = mExtractor->setDataSource(aSourceWrapper->getAMediaDataSource());
- } else {
- err = mExtractor->setDataSource(fd, mOffset, mLength);
- }
-
- if (err != OK) {
- ALOGE("initFromDataSource, failed to set extractor data source!");
- mLock.lock();
- return UNKNOWN_ERROR;
- }
-
- size_t numtracks = mExtractor->getTrackCount();
- if (numtracks == 0) {
- ALOGE("initFromDataSource, source has no track!");
- mLock.lock();
- return UNKNOWN_ERROR;
- }
-
- mFileMeta = convertMediaFormatWrapperToMetaData(mExtractor->getFormat());
- mLock.lock();
- if (mFileMeta != NULL) {
- int64_t duration;
- if (mFileMeta->findInt64(kKeyDuration, &duration)) {
- mDurationUs = duration;
- }
- }
-
- int32_t totalBitrate = 0;
-
- mMimes.clear();
-
- for (size_t i = 0; i < numtracks; ++i) {
-
- sp<AMediaFormatWrapper> trackFormat = mExtractor->getTrackFormat(i);
- if (trackFormat == NULL) {
- ALOGE("no metadata for track %zu", i);
- return UNKNOWN_ERROR;
- }
-
- sp<AMediaExtractorWrapper> trackExtractor = new AMediaExtractorWrapper(AMediaExtractor_new());
- if (aSourceWrapper != NULL) {
- trackExtractor->setDataSource(aSourceWrapper->getAMediaDataSource());
- } else {
- trackExtractor->setDataSource(fd, mOffset, mLength);
- }
-
- const char *mime;
- sp<MetaData> meta = convertMediaFormatWrapperToMetaData(trackFormat);
- CHECK(meta->findCString(kKeyMIMEType, &mime));
-
- ALOGV("initFromDataSource track[%zu]: %s", i, mime);
-
- // Do the string compare immediately with "mime",
- // we can't assume "mime" would stay valid after another
- // extractor operation, some extractors might modify meta
- // during getTrack() and make it invalid.
- if (!strncasecmp(mime, "audio/", 6)) {
- if (mAudioTrack.mExtractor == NULL) {
- mAudioTrack.mIndex = i;
- mAudioTrack.mExtractor = trackExtractor;
- mAudioTrack.mExtractor->selectTrack(i);
- mAudioTrack.mPackets = new AnotherPacketSource(meta);
-
- if (!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_VORBIS)) {
- mAudioIsVorbis = true;
- } else {
- mAudioIsVorbis = false;
- }
-
- mMimes.add(String8(mime));
- }
- } else if (!strncasecmp(mime, "video/", 6)) {
- if (mVideoTrack.mExtractor == NULL) {
- mVideoTrack.mIndex = i;
- mVideoTrack.mExtractor = trackExtractor;
- mVideoTrack.mExtractor->selectTrack(i);
- mVideoTrack.mPackets = new AnotherPacketSource(meta);
-
- // video always at the beginning
- mMimes.insertAt(String8(mime), 0);
- }
- }
-
- mExtractors.push(trackExtractor);
- int64_t durationUs;
- if (meta->findInt64(kKeyDuration, &durationUs)) {
- if (durationUs > mDurationUs) {
- mDurationUs = durationUs;
- }
- }
-
- int32_t bitrate;
- if (totalBitrate >= 0 && meta->findInt32(kKeyBitRate, &bitrate)) {
- totalBitrate += bitrate;
- } else {
- totalBitrate = -1;
- }
- }
-
- ALOGV("initFromDataSource mExtractors.size(): %zu mIsSecure: %d mime[0]: %s", mExtractors.size(),
- mIsSecure, (mMimes.isEmpty() ? "NONE" : mMimes[0].string()));
-
- if (mExtractors.size() == 0) {
- ALOGE("b/23705695");
- return UNKNOWN_ERROR;
- }
-
- // Modular DRM: The return value doesn't affect source initialization.
- (void)checkDrmInfo();
-
- mBitrate = totalBitrate;
-
- return OK;
-}
-
-status_t NuPlayer2::GenericSource2::getBufferingSettings(
- BufferingSettings* buffering /* nonnull */) {
- {
- Mutex::Autolock _l(mLock);
- *buffering = mBufferingSettings;
- }
-
- ALOGV("getBufferingSettings{%s}", buffering->toString().string());
- return OK;
-}
-
-status_t NuPlayer2::GenericSource2::setBufferingSettings(const BufferingSettings& buffering) {
- ALOGV("setBufferingSettings{%s}", buffering.toString().string());
-
- Mutex::Autolock _l(mLock);
- mBufferingSettings = buffering;
- return OK;
-}
-
-int64_t NuPlayer2::GenericSource2::getLastReadPosition() {
- if (mAudioTrack.mExtractor != NULL) {
- return mAudioTimeUs;
- } else if (mVideoTrack.mExtractor != NULL) {
- return mVideoTimeUs;
- } else {
- return 0;
- }
-}
-
-bool NuPlayer2::GenericSource2::isStreaming() const {
- Mutex::Autolock _l(mLock);
- return mIsStreaming;
-}
-
-NuPlayer2::GenericSource2::~GenericSource2() {
- ALOGV("~GenericSource2");
- if (mLooper != NULL) {
- mLooper->unregisterHandler(id());
- mLooper->stop();
- }
- if (mDataSourceWrapper != NULL) {
- mDataSourceWrapper->close();
- }
- resetDataSource();
-}
-
-void NuPlayer2::GenericSource2::prepareAsync(int64_t startTimeUs) {
- Mutex::Autolock _l(mLock);
- ALOGV("prepareAsync: (looper: %d)", (mLooper != NULL));
-
- if (mLooper == NULL) {
- mLooper = new ALooper;
- mLooper->setName("generic2");
- mLooper->start(false, /* runOnCallingThread */
- true, /* canCallJava */
- PRIORITY_DEFAULT);
-
- mLooper->registerHandler(this);
- }
-
- sp<AMessage> msg = new AMessage(kWhatPrepareAsync, this);
- msg->setInt64("startTimeUs", startTimeUs);
-
- msg->post();
-}
-
-void NuPlayer2::GenericSource2::onPrepareAsync(int64_t startTimeUs) {
- ALOGV("onPrepareAsync: mFd %d mUri %s mDataSourceWrapper: %p",
- mFd, mUri.c_str(), mDataSourceWrapper.get());
-
- if (!mUri.empty()) {
- const char* uri = mUri.c_str();
- size_t numheaders = mUriHeaders.size();
- const char **key_values = numheaders ? new const char *[numheaders * 2] : NULL;
- for (size_t i = 0; i < numheaders; ++i) {
- key_values[i * 2] = mUriHeaders.keyAt(i).c_str();
- key_values[i * 2 + 1] = mUriHeaders.valueAt(i).c_str();
- }
- mLock.unlock();
- AMediaDataSource *aSource = AMediaDataSource_newUri(uri, numheaders, key_values);
- mLock.lock();
- mDataSourceWrapper = aSource ? new AMediaDataSourceWrapper(aSource) : NULL;
- delete[] key_values;
- // For cached streaming cases, we need to wait for enough
- // buffering before reporting prepared.
- mIsStreaming = !strncasecmp("http://", uri, 7) || !strncasecmp("https://", uri, 8);
- }
-
- if (mDisconnected || (mFd < 0 && mDataSourceWrapper == NULL)) {
- ALOGE("mDisconnected(%d) or Failed to create data source!", mDisconnected);
- notifyPreparedAndCleanup(UNKNOWN_ERROR);
- return;
- }
-
- // init extractor from data source
- status_t err = initFromDataSource();
- if (mFd >= 0) {
- close(mFd);
- mFd = -1;
- }
-
- if (err != OK) {
- ALOGE("Failed to init from data source!");
- notifyPreparedAndCleanup(err);
- return;
- }
-
- if (mVideoTrack.mExtractor != NULL) {
- sp<MetaData> meta = getFormatMeta_l(false /* audio */);
- sp<AMessage> msg = new AMessage;
- err = convertMetaDataToMessage(meta, &msg);
- if(err != OK) {
- notifyPreparedAndCleanup(err);
- return;
- }
- notifyVideoSizeChanged(msg);
- }
-
- notifyFlagsChanged(
- // FLAG_SECURE will be known if/when prepareDrm is called by the app
- // FLAG_PROTECTED will be known if/when prepareDrm is called by the app
- FLAG_CAN_PAUSE |
- FLAG_CAN_SEEK_BACKWARD |
- FLAG_CAN_SEEK_FORWARD |
- FLAG_CAN_SEEK);
-
- doSeek(startTimeUs, MediaPlayer2SeekMode::SEEK_CLOSEST);
- finishPrepareAsync();
-
- ALOGV("onPrepareAsync: Done");
-}
-
-void NuPlayer2::GenericSource2::finishPrepareAsync() {
- ALOGV("finishPrepareAsync");
-
- if (mIsStreaming) {
- mPreparing = true;
- ++mPollBufferingGeneration;
- schedulePollBuffering();
- } else {
- notifyPrepared();
- }
-
- if (mAudioTrack.mExtractor != NULL) {
- postReadBuffer(MEDIA_TRACK_TYPE_AUDIO);
- }
-
- if (mVideoTrack.mExtractor != NULL) {
- postReadBuffer(MEDIA_TRACK_TYPE_VIDEO);
- }
-}
-
-void NuPlayer2::GenericSource2::notifyPreparedAndCleanup(status_t err) {
- if (err != OK) {
- mDataSourceWrapper.clear();
-
- mBitrate = -1;
- mPrevBufferPercentage = -1;
- ++mPollBufferingGeneration;
- }
- notifyPrepared(err);
-}
-
-void NuPlayer2::GenericSource2::start() {
- Mutex::Autolock _l(mLock);
- ALOGI("start");
-
- if (mAudioTrack.mExtractor != NULL) {
- postReadBuffer(MEDIA_TRACK_TYPE_AUDIO);
- }
-
- if (mVideoTrack.mExtractor != NULL) {
- postReadBuffer(MEDIA_TRACK_TYPE_VIDEO);
- }
-
- mStarted = true;
-}
-
-void NuPlayer2::GenericSource2::stop() {
- Mutex::Autolock _l(mLock);
- mStarted = false;
-}
-
-void NuPlayer2::GenericSource2::pause() {
- Mutex::Autolock _l(mLock);
- mStarted = false;
-}
-
-void NuPlayer2::GenericSource2::resume() {
- Mutex::Autolock _l(mLock);
- mStarted = true;
-}
-
-void NuPlayer2::GenericSource2::disconnect() {
- {
- Mutex::Autolock _l(mLock);
- mDisconnected = true;
- }
- if (mDataSourceWrapper != NULL) {
- mDataSourceWrapper->close();
- }
-}
-
-status_t NuPlayer2::GenericSource2::feedMoreTSData() {
- return OK;
-}
-
-void NuPlayer2::GenericSource2::onMessageReceived(const sp<AMessage> &msg) {
- Mutex::Autolock _l(mLock);
- switch (msg->what()) {
- case kWhatPrepareAsync:
- {
- int64_t startTimeUs;
- CHECK(msg->findInt64("startTimeUs", &startTimeUs));
- onPrepareAsync(startTimeUs);
- break;
- }
- case kWhatFetchSubtitleData:
- {
- fetchTextData(kWhatSendSubtitleData, MEDIA_TRACK_TYPE_SUBTITLE,
- mFetchSubtitleDataGeneration, mSubtitleTrack.mPackets, msg);
- break;
- }
-
- case kWhatFetchTimedTextData:
- {
- fetchTextData(kWhatSendTimedTextData, MEDIA_TRACK_TYPE_TIMEDTEXT,
- mFetchTimedTextDataGeneration, mTimedTextTrack.mPackets, msg);
- break;
- }
-
- case kWhatSendSubtitleData:
- {
- sendTextData(kWhatSubtitleData, MEDIA_TRACK_TYPE_SUBTITLE,
- mFetchSubtitleDataGeneration, mSubtitleTrack.mPackets, msg);
- break;
- }
-
- case kWhatSendGlobalTimedTextData:
- {
- sendGlobalTextData(kWhatTimedTextData, mFetchTimedTextDataGeneration, msg);
- break;
- }
- case kWhatSendTimedTextData:
- {
- sendTextData(kWhatTimedTextData, MEDIA_TRACK_TYPE_TIMEDTEXT,
- mFetchTimedTextDataGeneration, mTimedTextTrack.mPackets, msg);
- break;
- }
-
- case kWhatChangeAVSource:
- {
- int32_t trackIndex;
- CHECK(msg->findInt32("trackIndex", &trackIndex));
- const sp<AMediaExtractorWrapper> extractor = mExtractors.itemAt(trackIndex);
-
- Track* track;
- AString mime;
- media_track_type trackType, counterpartType;
- sp<AMediaFormatWrapper> format = extractor->getTrackFormat(trackIndex);
- format->getString(AMEDIAFORMAT_KEY_MIME, &mime);
- if (!strncasecmp(mime.c_str(), "audio/", 6)) {
- track = &mAudioTrack;
- trackType = MEDIA_TRACK_TYPE_AUDIO;
- counterpartType = MEDIA_TRACK_TYPE_VIDEO;;
- } else {
- CHECK(!strncasecmp(mime.c_str(), "video/", 6));
- track = &mVideoTrack;
- trackType = MEDIA_TRACK_TYPE_VIDEO;
- counterpartType = MEDIA_TRACK_TYPE_AUDIO;;
- }
-
-
- track->mExtractor = extractor;
- track->mExtractor->selectSingleTrack(trackIndex);
- track->mIndex = trackIndex;
- ++mAudioDataGeneration;
- ++mVideoDataGeneration;
-
- int64_t timeUs, actualTimeUs;
- const bool formatChange = true;
- if (trackType == MEDIA_TRACK_TYPE_AUDIO) {
- timeUs = mAudioLastDequeueTimeUs;
- } else {
- timeUs = mVideoLastDequeueTimeUs;
- }
- readBuffer(trackType, timeUs, MediaPlayer2SeekMode::SEEK_PREVIOUS_SYNC /* mode */,
- &actualTimeUs, formatChange);
- readBuffer(counterpartType, -1, MediaPlayer2SeekMode::SEEK_PREVIOUS_SYNC /* mode */,
- NULL, !formatChange);
- ALOGV("timeUs %lld actualTimeUs %lld", (long long)timeUs, (long long)actualTimeUs);
-
- break;
- }
-
- case kWhatSeek:
- {
- onSeek(msg);
- break;
- }
-
- case kWhatReadBuffer:
- {
- onReadBuffer(msg);
- break;
- }
-
- case kWhatPollBuffering:
- {
- int32_t generation;
- CHECK(msg->findInt32("generation", &generation));
- if (generation == mPollBufferingGeneration) {
- onPollBuffering();
- }
- break;
- }
-
- default:
- Source::onMessageReceived(msg);
- break;
- }
-}
-
-void NuPlayer2::GenericSource2::fetchTextData(
- uint32_t sendWhat,
- media_track_type type,
- int32_t curGen,
- const sp<AnotherPacketSource>& packets,
- const sp<AMessage>& msg) {
- int32_t msgGeneration;
- CHECK(msg->findInt32("generation", &msgGeneration));
- if (msgGeneration != curGen) {
- // stale
- return;
- }
-
- int32_t avail;
- if (packets->hasBufferAvailable(&avail)) {
- return;
- }
-
- int64_t timeUs;
- CHECK(msg->findInt64("timeUs", &timeUs));
-
- int64_t subTimeUs = 0;
- readBuffer(type, timeUs, MediaPlayer2SeekMode::SEEK_PREVIOUS_SYNC /* mode */, &subTimeUs);
-
- status_t eosResult;
- if (!packets->hasBufferAvailable(&eosResult)) {
- return;
- }
-
- if (msg->what() == kWhatFetchSubtitleData) {
- subTimeUs -= 1000000ll; // send subtile data one second earlier
- }
- sp<AMessage> msg2 = new AMessage(sendWhat, this);
- msg2->setInt32("generation", msgGeneration);
- mMediaClock->addTimer(msg2, subTimeUs);
-}
-
-void NuPlayer2::GenericSource2::sendTextData(
- uint32_t what,
- media_track_type type,
- int32_t curGen,
- const sp<AnotherPacketSource>& packets,
- const sp<AMessage>& msg) {
- int32_t msgGeneration;
- CHECK(msg->findInt32("generation", &msgGeneration));
- if (msgGeneration != curGen) {
- // stale
- return;
- }
-
- int64_t subTimeUs;
- if (packets->nextBufferTime(&subTimeUs) != OK) {
- return;
- }
-
- int64_t nextSubTimeUs;
- readBuffer(type, -1, MediaPlayer2SeekMode::SEEK_PREVIOUS_SYNC /* mode */, &nextSubTimeUs);
-
- sp<ABuffer> buffer;
- status_t dequeueStatus = packets->dequeueAccessUnit(&buffer);
- if (dequeueStatus == OK) {
- sp<AMessage> notify = dupNotify();
- notify->setInt32("what", what);
- notify->setBuffer("buffer", buffer);
- notify->post();
-
- if (msg->what() == kWhatSendSubtitleData) {
- nextSubTimeUs -= 1000000ll; // send subtile data one second earlier
- }
- mMediaClock->addTimer(msg, nextSubTimeUs);
- }
-}
-
-void NuPlayer2::GenericSource2::sendGlobalTextData(
- uint32_t what,
- int32_t curGen,
- sp<AMessage> msg) {
- int32_t msgGeneration;
- CHECK(msg->findInt32("generation", &msgGeneration));
- if (msgGeneration != curGen) {
- // stale
- return;
- }
-
- void *data = NULL;
- size_t size = 0;
- if (mTimedTextTrack.mExtractor->getTrackFormat(mTimedTextTrack.mIndex)->getBuffer(
- "text", &data, &size)) {
- mGlobalTimedText = new ABuffer(size);
- if (mGlobalTimedText->data()) {
- memcpy(mGlobalTimedText->data(), data, size);
- sp<AMessage> globalMeta = mGlobalTimedText->meta();
- globalMeta->setInt64("timeUs", 0);
- globalMeta->setString("mime", MEDIA_MIMETYPE_TEXT_3GPP);
- globalMeta->setInt32("global", 1);
- sp<AMessage> notify = dupNotify();
- notify->setInt32("what", what);
- notify->setBuffer("buffer", mGlobalTimedText);
- notify->post();
- }
- }
-}
-
-sp<AMessage> NuPlayer2::GenericSource2::getFormat(bool audio) {
- Mutex::Autolock _l(mLock);
- return getFormat_l(audio);
-}
-
-sp<MetaData> NuPlayer2::GenericSource2::getFormatMeta(bool audio) {
- Mutex::Autolock _l(mLock);
- return getFormatMeta_l(audio);
-}
-
-sp<AMessage> NuPlayer2::GenericSource2::getFormat_l(bool audio) {
- sp<AMediaExtractorWrapper> extractor = audio ? mAudioTrack.mExtractor : mVideoTrack.mExtractor;
- size_t trackIndex = audio ? mAudioTrack.mIndex : mVideoTrack.mIndex;
-
- if (extractor == NULL) {
- return NULL;
- }
-
- return extractor->getTrackFormat(trackIndex)->toAMessage();
-}
-
-sp<MetaData> NuPlayer2::GenericSource2::getFormatMeta_l(bool audio) {
- sp<AMediaExtractorWrapper> extractor = audio ? mAudioTrack.mExtractor : mVideoTrack.mExtractor;
- size_t trackIndex = audio ? mAudioTrack.mIndex : mVideoTrack.mIndex;
-
- if (extractor == NULL) {
- return NULL;
- }
-
- return convertMediaFormatWrapperToMetaData(extractor->getTrackFormat(trackIndex));
-}
-
-status_t NuPlayer2::GenericSource2::dequeueAccessUnit(
- bool audio, sp<ABuffer> *accessUnit) {
- Mutex::Autolock _l(mLock);
- // If has gone through stop/releaseDrm sequence, we no longer send down any buffer b/c
- // the codec's crypto object has gone away (b/37960096).
- // Note: This will be unnecessary when stop() changes behavior and releases codec (b/35248283).
- if (!mStarted && mIsDrmReleased) {
- return -EWOULDBLOCK;
- }
-
- Track *track = audio ? &mAudioTrack : &mVideoTrack;
-
- if (track->mExtractor == NULL) {
- return -EWOULDBLOCK;
- }
-
- status_t finalResult;
- if (!track->mPackets->hasBufferAvailable(&finalResult)) {
- if (finalResult == OK) {
- postReadBuffer(
- audio ? MEDIA_TRACK_TYPE_AUDIO : MEDIA_TRACK_TYPE_VIDEO);
- return -EWOULDBLOCK;
- }
- return finalResult;
- }
-
- status_t result = track->mPackets->dequeueAccessUnit(accessUnit);
-
- // start pulling in more buffers if cache is running low
- // so that decoder has less chance of being starved
- if (!mIsStreaming) {
- if (track->mPackets->getAvailableBufferCount(&finalResult) < 2) {
- postReadBuffer(audio? MEDIA_TRACK_TYPE_AUDIO : MEDIA_TRACK_TYPE_VIDEO);
- }
- } else {
- int64_t durationUs = track->mPackets->getBufferedDurationUs(&finalResult);
- // TODO: maxRebufferingMarkMs could be larger than
- // mBufferingSettings.mResumePlaybackMarkMs
- int64_t restartBufferingMarkUs =
- mBufferingSettings.mResumePlaybackMarkMs * 1000ll / 2;
- if (finalResult == OK) {
- if (durationUs < restartBufferingMarkUs) {
- postReadBuffer(audio? MEDIA_TRACK_TYPE_AUDIO : MEDIA_TRACK_TYPE_VIDEO);
- }
- if (track->mPackets->getAvailableBufferCount(&finalResult) < 2
- && !mSentPauseOnBuffering && !mPreparing) {
- mSentPauseOnBuffering = true;
- sp<AMessage> notify = dupNotify();
- notify->setInt32("what", kWhatPauseOnBufferingStart);
- notify->post();
- }
- }
- }
-
- if (result != OK) {
- if (mSubtitleTrack.mExtractor != NULL) {
- mSubtitleTrack.mPackets->clear();
- mFetchSubtitleDataGeneration++;
- }
- if (mTimedTextTrack.mExtractor != NULL) {
- mTimedTextTrack.mPackets->clear();
- mFetchTimedTextDataGeneration++;
- }
- return result;
- }
-
- int64_t timeUs;
- status_t eosResult; // ignored
- CHECK((*accessUnit)->meta()->findInt64("timeUs", &timeUs));
- if (audio) {
- mAudioLastDequeueTimeUs = timeUs;
- } else {
- mVideoLastDequeueTimeUs = timeUs;
- }
-
- if (mSubtitleTrack.mExtractor != NULL
- && !mSubtitleTrack.mPackets->hasBufferAvailable(&eosResult)) {
- sp<AMessage> msg = new AMessage(kWhatFetchSubtitleData, this);
- msg->setInt64("timeUs", timeUs);
- msg->setInt32("generation", mFetchSubtitleDataGeneration);
- msg->post();
- }
-
- if (mTimedTextTrack.mExtractor != NULL
- && !mTimedTextTrack.mPackets->hasBufferAvailable(&eosResult)) {
- sp<AMessage> msg = new AMessage(kWhatFetchTimedTextData, this);
- msg->setInt64("timeUs", timeUs);
- msg->setInt32("generation", mFetchTimedTextDataGeneration);
- msg->post();
- }
-
- return result;
-}
-
-status_t NuPlayer2::GenericSource2::getDuration(int64_t *durationUs) {
- Mutex::Autolock _l(mLock);
- *durationUs = mDurationUs;
- return OK;
-}
-
-size_t NuPlayer2::GenericSource2::getTrackCount() const {
- Mutex::Autolock _l(mLock);
- return mExtractors.size();
-}
-
-sp<AMessage> NuPlayer2::GenericSource2::getTrackInfo(size_t trackIndex) const {
- Mutex::Autolock _l(mLock);
- size_t trackCount = mExtractors.size();
- if (trackIndex >= trackCount) {
- return NULL;
- }
-
- sp<AMessage> format = mExtractors.itemAt(trackIndex)->getTrackFormat(trackIndex)->toAMessage();
- if (format == NULL) {
- ALOGE("no metadata for track %zu", trackIndex);
- return NULL;
- }
-
- AString mime;
- CHECK(format->findString(AMEDIAFORMAT_KEY_MIME, &mime));
-
- int32_t trackType;
- if (!strncasecmp(mime.c_str(), "video/", 6)) {
- trackType = MEDIA_TRACK_TYPE_VIDEO;
- } else if (!strncasecmp(mime.c_str(), "audio/", 6)) {
- trackType = MEDIA_TRACK_TYPE_AUDIO;
- } else if (!strcasecmp(mime.c_str(), MEDIA_MIMETYPE_TEXT_3GPP)) {
- trackType = MEDIA_TRACK_TYPE_TIMEDTEXT;
- } else {
- trackType = MEDIA_TRACK_TYPE_UNKNOWN;
- }
- format->setInt32("type", trackType);
-
- AString lang;
- if (!format->findString("language", &lang)) {
- format->setString("language", "und");
- }
-
- if (trackType == MEDIA_TRACK_TYPE_SUBTITLE) {
- int32_t isAutoselect = 1, isDefault = 0, isForced = 0;
- format->findInt32(AMEDIAFORMAT_KEY_IS_AUTOSELECT, &isAutoselect);
- format->findInt32(AMEDIAFORMAT_KEY_IS_DEFAULT, &isDefault);
- format->findInt32(AMEDIAFORMAT_KEY_IS_FORCED_SUBTITLE, &isForced);
-
- format->setInt32("auto", !!isAutoselect);
- format->setInt32("default", !!isDefault);
- format->setInt32("forced", !!isForced);
- }
-
- return format;
-}
-
-ssize_t NuPlayer2::GenericSource2::getSelectedTrack(media_track_type type) const {
- Mutex::Autolock _l(mLock);
- const Track *track = NULL;
- switch (type) {
- case MEDIA_TRACK_TYPE_VIDEO:
- track = &mVideoTrack;
- break;
- case MEDIA_TRACK_TYPE_AUDIO:
- track = &mAudioTrack;
- break;
- case MEDIA_TRACK_TYPE_TIMEDTEXT:
- track = &mTimedTextTrack;
- break;
- case MEDIA_TRACK_TYPE_SUBTITLE:
- track = &mSubtitleTrack;
- break;
- default:
- break;
- }
-
- if (track != NULL && track->mExtractor != NULL) {
- return track->mIndex;
- }
-
- return -1;
-}
-
-status_t NuPlayer2::GenericSource2::selectTrack(size_t trackIndex, bool select, int64_t timeUs) {
- Mutex::Autolock _l(mLock);
- ALOGV("%s track: %zu", select ? "select" : "deselect", trackIndex);
-
- if (trackIndex >= mExtractors.size()) {
- return BAD_INDEX;
- }
-
- if (!select) {
- Track* track = NULL;
- if (mSubtitleTrack.mExtractor != NULL && trackIndex == mSubtitleTrack.mIndex) {
- track = &mSubtitleTrack;
- mFetchSubtitleDataGeneration++;
- } else if (mTimedTextTrack.mExtractor != NULL && trackIndex == mTimedTextTrack.mIndex) {
- track = &mTimedTextTrack;
- mFetchTimedTextDataGeneration++;
- }
- if (track == NULL) {
- return INVALID_OPERATION;
- }
- track->mExtractor = NULL;
- track->mPackets->clear();
- return OK;
- }
-
- const sp<AMediaExtractorWrapper> extractor = mExtractors.itemAt(trackIndex);
- sp<MetaData> meta = convertMediaFormatWrapperToMetaData(extractor->getTrackFormat(trackIndex));
- const char *mime;
- CHECK(meta->findCString(kKeyMIMEType, &mime));
- if (!strncasecmp(mime, "text/", 5)) {
- bool isSubtitle = strcasecmp(mime, MEDIA_MIMETYPE_TEXT_3GPP);
- Track *track = isSubtitle ? &mSubtitleTrack : &mTimedTextTrack;
- if (track->mExtractor != NULL && track->mIndex == trackIndex) {
- return OK;
- }
- track->mIndex = trackIndex;
- track->mExtractor = mExtractors.itemAt(trackIndex);
- track->mExtractor->selectSingleTrack(trackIndex);
- if (track->mPackets == NULL) {
- track->mPackets = new AnotherPacketSource(meta);
- } else {
- track->mPackets->clear();
- track->mPackets->setFormat(meta);
-
- }
-
- if (isSubtitle) {
- mFetchSubtitleDataGeneration++;
- } else {
- mFetchTimedTextDataGeneration++;
- }
-
- status_t eosResult; // ignored
- if (mSubtitleTrack.mExtractor != NULL
- && !mSubtitleTrack.mPackets->hasBufferAvailable(&eosResult)) {
- sp<AMessage> msg = new AMessage(kWhatFetchSubtitleData, this);
- msg->setInt64("timeUs", timeUs);
- msg->setInt32("generation", mFetchSubtitleDataGeneration);
- msg->post();
- }
-
- sp<AMessage> msg2 = new AMessage(kWhatSendGlobalTimedTextData, this);
- msg2->setInt32("generation", mFetchTimedTextDataGeneration);
- msg2->post();
-
- if (mTimedTextTrack.mExtractor != NULL
- && !mTimedTextTrack.mPackets->hasBufferAvailable(&eosResult)) {
- sp<AMessage> msg = new AMessage(kWhatFetchTimedTextData, this);
- msg->setInt64("timeUs", timeUs);
- msg->setInt32("generation", mFetchTimedTextDataGeneration);
- msg->post();
- }
-
- return OK;
- } else if (!strncasecmp(mime, "audio/", 6) || !strncasecmp(mime, "video/", 6)) {
- bool audio = !strncasecmp(mime, "audio/", 6);
- Track *track = audio ? &mAudioTrack : &mVideoTrack;
- if (track->mExtractor != NULL && track->mIndex == trackIndex) {
- return OK;
- }
-
- sp<AMessage> msg = new AMessage(kWhatChangeAVSource, this);
- msg->setInt32("trackIndex", trackIndex);
- msg->post();
- return OK;
- }
-
- return INVALID_OPERATION;
-}
-
-status_t NuPlayer2::GenericSource2::seekTo(int64_t seekTimeUs, MediaPlayer2SeekMode mode) {
- ALOGV("seekTo: %lld, %d", (long long)seekTimeUs, mode);
- sp<AMessage> msg = new AMessage(kWhatSeek, this);
- msg->setInt64("seekTimeUs", seekTimeUs);
- msg->setInt32("mode", mode);
-
- // Need to call readBuffer on |mLooper| to ensure the calls to
- // IMediaSource::read* are serialized. Note that IMediaSource::read*
- // is called without |mLock| acquired and MediaSource is not thread safe.
- sp<AMessage> response;
- status_t err = msg->postAndAwaitResponse(&response);
- if (err == OK && response != NULL) {
- CHECK(response->findInt32("err", &err));
- }
-
- return err;
-}
-
-void NuPlayer2::GenericSource2::onSeek(const sp<AMessage>& msg) {
- int64_t seekTimeUs;
- int32_t mode;
- CHECK(msg->findInt64("seekTimeUs", &seekTimeUs));
- CHECK(msg->findInt32("mode", &mode));
-
- sp<AMessage> response = new AMessage;
- status_t err = doSeek(seekTimeUs, (MediaPlayer2SeekMode)mode);
- response->setInt32("err", err);
-
- sp<AReplyToken> replyID;
- CHECK(msg->senderAwaitsResponse(&replyID));
- response->postReply(replyID);
-}
-
-status_t NuPlayer2::GenericSource2::doSeek(int64_t seekTimeUs, MediaPlayer2SeekMode mode) {
- if (mVideoTrack.mExtractor != NULL) {
- ++mVideoDataGeneration;
-
- int64_t actualTimeUs;
- readBuffer(MEDIA_TRACK_TYPE_VIDEO, seekTimeUs, mode, &actualTimeUs);
-
- if (mode != MediaPlayer2SeekMode::SEEK_CLOSEST) {
- seekTimeUs = actualTimeUs;
- }
- mVideoLastDequeueTimeUs = actualTimeUs;
- }
-
- if (mAudioTrack.mExtractor != NULL) {
- ++mAudioDataGeneration;
- readBuffer(MEDIA_TRACK_TYPE_AUDIO, seekTimeUs, MediaPlayer2SeekMode::SEEK_CLOSEST);
- mAudioLastDequeueTimeUs = seekTimeUs;
- }
-
- if (mSubtitleTrack.mExtractor != NULL) {
- mSubtitleTrack.mPackets->clear();
- mFetchSubtitleDataGeneration++;
- }
-
- if (mTimedTextTrack.mExtractor != NULL) {
- mTimedTextTrack.mPackets->clear();
- mFetchTimedTextDataGeneration++;
- }
-
- ++mPollBufferingGeneration;
- schedulePollBuffering();
- return OK;
-}
-
-sp<ABuffer> NuPlayer2::GenericSource2::mediaBufferToABuffer(
- MediaBufferBase* mb,
- media_track_type trackType) {
- bool audio = trackType == MEDIA_TRACK_TYPE_AUDIO;
- size_t outLength = mb->range_length();
-
- if (audio && mAudioIsVorbis) {
- outLength += sizeof(int32_t);
- }
-
- sp<ABuffer> ab;
-
- if (mIsDrmProtected) {
- // Modular DRM
- // Enabled for both video/audio so 1) media buffer is reused without extra copying
- // 2) meta data can be retrieved in onInputBufferFetched for calling queueSecureInputBuffer.
-
- // data is already provided in the buffer
- ab = new ABuffer(NULL, mb->range_length());
- ab->meta()->setObject("mediaBufferHolder", new MediaBufferHolder(mb));
-
- // Modular DRM: Required b/c of the above add_ref.
- // If ref>0, there must be an observer, or it'll crash at release().
- // TODO: MediaBuffer might need to be revised to ease such need.
- mb->setObserver(this);
- // setMediaBufferBase() interestingly doesn't increment the ref count on its own.
- // Extra increment (since we want to keep mb alive and attached to ab beyond this function
- // call. This is to counter the effect of mb->release() towards the end.
- mb->add_ref();
-
- } else {
- ab = new ABuffer(outLength);
- memcpy(ab->data(),
- (const uint8_t *)mb->data() + mb->range_offset(),
- mb->range_length());
- }
-
- if (audio && mAudioIsVorbis) {
- int32_t numPageSamples;
- if (!mb->meta_data().findInt32(kKeyValidSamples, &numPageSamples)) {
- numPageSamples = -1;
- }
-
- uint8_t* abEnd = ab->data() + mb->range_length();
- memcpy(abEnd, &numPageSamples, sizeof(numPageSamples));
- }
-
- sp<AMessage> meta = ab->meta();
-
- int64_t timeUs;
- CHECK(mb->meta_data().findInt64(kKeyTime, &timeUs));
- meta->setInt64("timeUs", timeUs);
-
- if (trackType == MEDIA_TRACK_TYPE_VIDEO) {
- int32_t layerId;
- if (mb->meta_data().findInt32(kKeyTemporalLayerId, &layerId)) {
- meta->setInt32("temporal-layer-id", layerId);
- }
- }
-
- if (trackType == MEDIA_TRACK_TYPE_TIMEDTEXT) {
- AString mime;
- sp<AMediaExtractorWrapper> extractor = mTimedTextTrack.mExtractor;
- size_t trackIndex = mTimedTextTrack.mIndex;
- CHECK(extractor != NULL
- && extractor->getTrackFormat(trackIndex)->getString(AMEDIAFORMAT_KEY_MIME, &mime));
- meta->setString("mime", mime.c_str());
- }
-
- int64_t durationUs;
- if (mb->meta_data().findInt64(kKeyDuration, &durationUs)) {
- meta->setInt64("durationUs", durationUs);
- }
-
- if (trackType == MEDIA_TRACK_TYPE_SUBTITLE) {
- meta->setInt32(AMEDIAFORMAT_KEY_TRACK_INDEX, mSubtitleTrack.mIndex);
- }
-
- uint32_t dataType; // unused
- const void *seiData;
- size_t seiLength;
- if (mb->meta_data().findData(kKeySEI, &dataType, &seiData, &seiLength)) {
- sp<ABuffer> sei = ABuffer::CreateAsCopy(seiData, seiLength);;
- meta->setBuffer("sei", sei);
- }
-
- const void *mpegUserDataPointer;
- size_t mpegUserDataLength;
- if (mb->meta_data().findData(
- kKeyMpegUserData, &dataType, &mpegUserDataPointer, &mpegUserDataLength)) {
- sp<ABuffer> mpegUserData = ABuffer::CreateAsCopy(mpegUserDataPointer, mpegUserDataLength);
- meta->setBuffer(AMEDIAFORMAT_KEY_MPEG_USER_DATA, mpegUserData);
- }
-
- mb->release();
- mb = NULL;
-
- return ab;
-}
-
-int32_t NuPlayer2::GenericSource2::getDataGeneration(media_track_type type) const {
- int32_t generation = -1;
- switch (type) {
- case MEDIA_TRACK_TYPE_VIDEO:
- generation = mVideoDataGeneration;
- break;
- case MEDIA_TRACK_TYPE_AUDIO:
- generation = mAudioDataGeneration;
- break;
- case MEDIA_TRACK_TYPE_TIMEDTEXT:
- generation = mFetchTimedTextDataGeneration;
- break;
- case MEDIA_TRACK_TYPE_SUBTITLE:
- generation = mFetchSubtitleDataGeneration;
- break;
- default:
- break;
- }
-
- return generation;
-}
-
-void NuPlayer2::GenericSource2::postReadBuffer(media_track_type trackType) {
- if ((mPendingReadBufferTypes & (1 << trackType)) == 0) {
- mPendingReadBufferTypes |= (1 << trackType);
- sp<AMessage> msg = new AMessage(kWhatReadBuffer, this);
- msg->setInt32("trackType", trackType);
- msg->post();
- }
-}
-
-void NuPlayer2::GenericSource2::onReadBuffer(const sp<AMessage>& msg) {
- int32_t tmpType;
- CHECK(msg->findInt32("trackType", &tmpType));
- media_track_type trackType = (media_track_type)tmpType;
- mPendingReadBufferTypes &= ~(1 << trackType);
- readBuffer(trackType);
-}
-
-void NuPlayer2::GenericSource2::readBuffer(
- media_track_type trackType, int64_t seekTimeUs, MediaPlayer2SeekMode mode,
- int64_t *actualTimeUs, bool formatChange) {
- Track *track;
- size_t maxBuffers = 1;
- switch (trackType) {
- case MEDIA_TRACK_TYPE_VIDEO:
- track = &mVideoTrack;
- maxBuffers = 8; // too large of a number may influence seeks
- break;
- case MEDIA_TRACK_TYPE_AUDIO:
- track = &mAudioTrack;
- maxBuffers = 64;
- break;
- case MEDIA_TRACK_TYPE_SUBTITLE:
- track = &mSubtitleTrack;
- break;
- case MEDIA_TRACK_TYPE_TIMEDTEXT:
- track = &mTimedTextTrack;
- break;
- default:
- TRESPASS();
- }
-
- if (track->mExtractor == NULL) {
- return;
- }
-
- if (actualTimeUs) {
- *actualTimeUs = seekTimeUs;
- }
-
-
- bool seeking = false;
- sp<AMediaExtractorWrapper> extractor = track->mExtractor;
- if (seekTimeUs >= 0) {
- extractor->seekTo(seekTimeUs, mode);
- seeking = true;
- }
-
- int32_t generation = getDataGeneration(trackType);
- for (size_t numBuffers = 0; numBuffers < maxBuffers; ) {
- Vector<sp<ABuffer> > aBuffers;
-
- mLock.unlock();
-
- sp<AMediaFormatWrapper> format;
- ssize_t sampleSize = -1;
- status_t err = extractor->getSampleFormat(format);
- if (err == OK) {
- sampleSize = extractor->getSampleSize();
- }
-
- if (err != OK || sampleSize < 0) {
- mLock.lock();
- track->mPackets->signalEOS(err != OK ? err : ERROR_END_OF_STREAM);
- break;
- }
-
- sp<ABuffer> abuf = new ABuffer(sampleSize);
- sampleSize = extractor->readSampleData(abuf);
- mLock.lock();
-
- // in case track has been changed since we don't have lock for some time.
- if (generation != getDataGeneration(trackType)) {
- break;
- }
-
- int64_t timeUs = extractor->getSampleTime();
- if (timeUs < 0) {
- track->mPackets->signalEOS(ERROR_MALFORMED);
- break;
- }
-
- sp<AMessage> meta = abuf->meta();
- format->writeToAMessage(meta);
- meta->setInt64("timeUs", timeUs);
- if (trackType == MEDIA_TRACK_TYPE_AUDIO) {
- mAudioTimeUs = timeUs;
- } else if (trackType == MEDIA_TRACK_TYPE_VIDEO) {
- mVideoTimeUs = timeUs;
- }
-
- sp<AMediaCodecCryptoInfoWrapper> cryptInfo = extractor->getSampleCryptoInfo();
- if (cryptInfo != NULL) {
- meta->setObject("cryptInfo", cryptInfo);
- }
-
- queueDiscontinuityIfNeeded(seeking, formatChange, trackType, track);
-
- if (numBuffers == 0 && actualTimeUs != nullptr) {
- *actualTimeUs = timeUs;
- }
- if (seeking) {
- if (meta != nullptr && mode == MediaPlayer2SeekMode::SEEK_CLOSEST
- && seekTimeUs > timeUs) {
- sp<AMessage> extra = new AMessage;
- extra->setInt64("resume-at-mediaTimeUs", seekTimeUs);
- meta->setMessage("extra", extra);
- }
- }
-
- track->mPackets->queueAccessUnit(abuf);
- formatChange = false;
- seeking = false;
- ++numBuffers;
- extractor->advance();
-
- }
-
- if (mIsStreaming
- && (trackType == MEDIA_TRACK_TYPE_VIDEO || trackType == MEDIA_TRACK_TYPE_AUDIO)) {
- status_t finalResult;
- int64_t durationUs = track->mPackets->getBufferedDurationUs(&finalResult);
-
- // TODO: maxRebufferingMarkMs could be larger than
- // mBufferingSettings.mResumePlaybackMarkMs
- int64_t markUs = (mPreparing ? mBufferingSettings.mInitialMarkMs
- : mBufferingSettings.mResumePlaybackMarkMs) * 1000ll;
- if (finalResult == ERROR_END_OF_STREAM || durationUs >= markUs) {
- if (mPreparing || mSentPauseOnBuffering) {
- Track *counterTrack =
- (trackType == MEDIA_TRACK_TYPE_VIDEO ? &mAudioTrack : &mVideoTrack);
- if (counterTrack->mExtractor != NULL) {
- durationUs = counterTrack->mPackets->getBufferedDurationUs(&finalResult);
- }
- if (finalResult == ERROR_END_OF_STREAM || durationUs >= markUs) {
- if (mPreparing) {
- notifyPrepared();
- mPreparing = false;
- } else {
- mSentPauseOnBuffering = false;
- sp<AMessage> notify = dupNotify();
- notify->setInt32("what", kWhatResumeOnBufferingEnd);
- notify->post();
- }
- }
- }
- return;
- }
-
- postReadBuffer(trackType);
- }
-}
-
-void NuPlayer2::GenericSource2::queueDiscontinuityIfNeeded(
- bool seeking, bool formatChange, media_track_type trackType, Track *track) {
- // formatChange && seeking: track whose source is changed during selection
- // formatChange && !seeking: track whose source is not changed during selection
- // !formatChange: normal seek
- if ((seeking || formatChange)
- && (trackType == MEDIA_TRACK_TYPE_AUDIO
- || trackType == MEDIA_TRACK_TYPE_VIDEO)) {
- ATSParser::DiscontinuityType type = (formatChange && seeking)
- ? ATSParser::DISCONTINUITY_FORMATCHANGE
- : ATSParser::DISCONTINUITY_NONE;
- track->mPackets->queueDiscontinuity(type, NULL /* extra */, true /* discard */);
- }
-}
-
-void NuPlayer2::GenericSource2::notifyBufferingUpdate(int32_t percentage) {
- // Buffering percent could go backward as it's estimated from remaining
- // data and last access time. This could cause the buffering position
- // drawn on media control to jitter slightly. Remember previously reported
- // percentage and don't allow it to go backward.
- if (percentage < mPrevBufferPercentage) {
- percentage = mPrevBufferPercentage;
- } else if (percentage > 100) {
- percentage = 100;
- }
-
- mPrevBufferPercentage = percentage;
-
- ALOGV("notifyBufferingUpdate: buffering %d%%", percentage);
-
- sp<AMessage> notify = dupNotify();
- notify->setInt32("what", kWhatBufferingUpdate);
- notify->setInt32("percentage", percentage);
- notify->post();
-}
-
-void NuPlayer2::GenericSource2::schedulePollBuffering() {
- if (mIsStreaming) {
- sp<AMessage> msg = new AMessage(kWhatPollBuffering, this);
- msg->setInt32("generation", mPollBufferingGeneration);
- // Enquires buffering status every second.
- msg->post(1000000ll);
- }
-}
-
-void NuPlayer2::GenericSource2::onPollBuffering() {
- int64_t cachedDurationUs = -1ll;
-
- sp<AMediaExtractorWrapper> extractor;
- if (mVideoTrack.mExtractor != NULL) {
- extractor = mVideoTrack.mExtractor;
- } else if (mAudioTrack.mExtractor != NULL) {
- extractor = mAudioTrack.mExtractor;
- }
-
- if (extractor != NULL) {
- cachedDurationUs = extractor->getCachedDuration();
- }
-
- if (cachedDurationUs >= 0ll) {
- ssize_t sampleSize = extractor->getSampleSize();
- if (sampleSize >= 0ll) {
- int64_t cachedPosUs = getLastReadPosition() + cachedDurationUs;
- int percentage = 100.0 * cachedPosUs / mDurationUs;
- if (percentage > 100) {
- percentage = 100;
- }
-
- notifyBufferingUpdate(percentage);
- ALOGV("onPollBuffering: cachedDurationUs %.1f sec", cachedDurationUs / 1000000.0f);
- } else {
- notifyBufferingUpdate(100);
- ALOGV("onPollBuffering: EOS");
- }
- }
-
- schedulePollBuffering();
-}
-
-// Modular DRM
-status_t NuPlayer2::GenericSource2::prepareDrm(
- const uint8_t uuid[16],
- const Vector<uint8_t> &drmSessionId,
- sp<AMediaCryptoWrapper> *outCrypto) {
- Mutex::Autolock _l(mLock);
- ALOGV("prepareDrm");
-
- mIsDrmProtected = false;
- mIsDrmReleased = false;
- mIsSecure = false;
-
- status_t status = OK;
- sp<AMediaCryptoWrapper> crypto =
- new AMediaCryptoWrapper(uuid, drmSessionId.array(), drmSessionId.size());
- if (crypto == NULL) {
- ALOGE("prepareDrm: failed to create crypto.");
- return UNKNOWN_ERROR;
- }
- ALOGV("prepareDrm: crypto created for uuid: %s",
- DrmUUID::toHexString(uuid).string());
-
- *outCrypto = crypto;
- // as long a there is an active crypto
- mIsDrmProtected = true;
-
- if (mMimes.size() == 0) {
- status = UNKNOWN_ERROR;
- ALOGE("prepareDrm: Unexpected. Must have at least one track. status: %d", status);
- return status;
- }
-
- // first mime in this list is either the video track, or the first audio track
- const char *mime = mMimes[0].string();
- mIsSecure = crypto->requiresSecureDecoderComponent(mime);
- ALOGV("prepareDrm: requiresSecureDecoderComponent mime: %s isSecure: %d",
- mime, mIsSecure);
-
- // Checking the member flags while in the looper to send out the notification.
- // The legacy mDecryptHandle!=NULL check (for FLAG_PROTECTED) is equivalent to mIsDrmProtected.
- notifyFlagsChanged(
- (mIsSecure ? FLAG_SECURE : 0) |
- // Setting "protected screen" only for L1: b/38390836
- (mIsSecure ? FLAG_PROTECTED : 0) |
- FLAG_CAN_PAUSE |
- FLAG_CAN_SEEK_BACKWARD |
- FLAG_CAN_SEEK_FORWARD |
- FLAG_CAN_SEEK);
-
- if (status == OK) {
- ALOGV("prepareDrm: mCrypto: %p", outCrypto->get());
- ALOGD("prepareDrm ret: %d ", status);
- } else {
- ALOGE("prepareDrm err: %d", status);
- }
- return status;
-}
-
-status_t NuPlayer2::GenericSource2::releaseDrm() {
- Mutex::Autolock _l(mLock);
- ALOGV("releaseDrm");
-
- if (mIsDrmProtected) {
- mIsDrmProtected = false;
- // to prevent returning any more buffer after stop/releaseDrm (b/37960096)
- mIsDrmReleased = true;
- ALOGV("releaseDrm: mIsDrmProtected is reset.");
- } else {
- ALOGE("releaseDrm: mIsDrmProtected is already false.");
- }
-
- return OK;
-}
-
-status_t NuPlayer2::GenericSource2::checkDrmInfo()
-{
- // clearing the flag at prepare in case the player is reused after stop/releaseDrm with the
- // same source without being reset (called by prepareAsync/initFromDataSource)
- mIsDrmReleased = false;
-
- if (mExtractor == NULL) {
- ALOGV("checkDrmInfo: No extractor");
- return OK; // letting the caller responds accordingly
- }
-
- PsshInfo *psshInfo = mExtractor->getPsshInfo();
- if (psshInfo == NULL) {
- ALOGV("checkDrmInfo: No PSSH");
- return OK; // source without DRM info
- }
-
- PlayerMessage playerMsg;
- status_t ret = NuPlayer2Drm::retrieveDrmInfo(psshInfo, &playerMsg);
- ALOGV("checkDrmInfo: MEDIA_DRM_INFO PSSH drm info size: %d", (int)playerMsg.ByteSize());
-
- if (ret != OK) {
- ALOGE("checkDrmInfo: failed to retrive DrmInfo %d", ret);
- return UNKNOWN_ERROR;
- }
-
- int size = playerMsg.ByteSize();
- sp<ABuffer> drmInfoBuf = new ABuffer(size);
- playerMsg.SerializeToArray(drmInfoBuf->data(), size);
- drmInfoBuf->setRange(0, size);
- notifyDrmInfo(drmInfoBuf);
-
- return OK;
-}
-
-void NuPlayer2::GenericSource2::signalBufferReturned(MediaBufferBase *buffer)
-{
- //ALOGV("signalBufferReturned %p refCount: %d", buffer, buffer->localRefcount());
-
- buffer->setObserver(NULL);
- buffer->release(); // this leads to delete since that there is no observor
-}
-
-} // namespace android
diff --git a/media/libmediaplayer2/nuplayer2/GenericSource2.h b/media/libmediaplayer2/nuplayer2/GenericSource2.h
deleted file mode 100644
index ade1aa3..0000000
--- a/media/libmediaplayer2/nuplayer2/GenericSource2.h
+++ /dev/null
@@ -1,246 +0,0 @@
-/*
- * Copyright 2017 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef GENERIC_SOURCE2_H_
-
-#define GENERIC_SOURCE2_H_
-
-#include "NuPlayer2.h"
-#include "NuPlayer2Source.h"
-
-#include "ATSParser.h"
-
-#include <media/stagefright/MediaBuffer.h>
-#include <mediaplayer2/mediaplayer2.h>
-#include <media/NdkMediaDataSource.h>
-#include <media/NdkMediaExtractor.h>
-#include <media/NdkWrapper.h>
-
-namespace android {
-
-class DecryptHandle;
-struct AnotherPacketSource;
-struct ARTSPController;
-class DataSource;
-class IDataSource;
-class IMediaSource;
-struct MediaSource;
-class MediaBuffer;
-struct MediaClock;
-
-struct NuPlayer2::GenericSource2 : public NuPlayer2::Source,
- public MediaBufferObserver // Modular DRM
-{
- GenericSource2(const sp<AMessage> ¬ify, uid_t uid,
- const sp<MediaClock> &mediaClock);
-
- status_t setDataSource(
- const char *url,
- const KeyedVector<String8, String8> *headers);
-
- status_t setDataSource(int fd, int64_t offset, int64_t length);
-
- status_t setDataSource(const sp<DataSource>& dataSource);
-
- virtual status_t getBufferingSettings(
- BufferingSettings* buffering /* nonnull */) override;
- virtual status_t setBufferingSettings(const BufferingSettings& buffering) override;
-
- virtual void prepareAsync(int64_t startTimeUs);
-
- virtual void start();
- virtual void stop();
- virtual void pause();
- virtual void resume();
-
- virtual void disconnect();
-
- virtual status_t feedMoreTSData();
-
- virtual sp<MetaData> getFileFormatMeta() const;
-
- virtual status_t dequeueAccessUnit(bool audio, sp<ABuffer> *accessUnit);
-
- virtual status_t getDuration(int64_t *durationUs);
- virtual size_t getTrackCount() const;
- virtual sp<AMessage> getTrackInfo(size_t trackIndex) const;
- virtual ssize_t getSelectedTrack(media_track_type type) const;
- virtual status_t selectTrack(size_t trackIndex, bool select, int64_t timeUs);
- virtual status_t seekTo(
- int64_t seekTimeUs,
- MediaPlayer2SeekMode mode = MediaPlayer2SeekMode::SEEK_PREVIOUS_SYNC) override;
-
- virtual bool isStreaming() const;
-
- // Modular DRM
- virtual void signalBufferReturned(MediaBufferBase *buffer);
-
- virtual status_t prepareDrm(
- const uint8_t uuid[16],
- const Vector<uint8_t> &drmSessionId,
- sp<AMediaCryptoWrapper> *outCrypto);
-
- virtual status_t releaseDrm();
-
-
-protected:
- virtual ~GenericSource2();
-
- virtual void onMessageReceived(const sp<AMessage> &msg);
-
- virtual sp<AMessage> getFormat(bool audio);
- virtual sp<MetaData> getFormatMeta(bool audio);
-
-private:
- enum {
- kWhatPrepareAsync,
- kWhatFetchSubtitleData,
- kWhatFetchTimedTextData,
- kWhatSendSubtitleData,
- kWhatSendGlobalTimedTextData,
- kWhatSendTimedTextData,
- kWhatChangeAVSource,
- kWhatPollBuffering,
- kWhatSeek,
- kWhatReadBuffer,
- kWhatStart,
- kWhatResume,
- kWhatSecureDecodersInstantiated,
- };
-
- struct Track {
- size_t mIndex;
- sp<AMediaExtractorWrapper> mExtractor;
- sp<AnotherPacketSource> mPackets;
- };
-
- int64_t mAudioTimeUs;
- int64_t mAudioLastDequeueTimeUs;
- int64_t mVideoTimeUs;
- int64_t mVideoLastDequeueTimeUs;
-
- BufferingSettings mBufferingSettings;
- int32_t mPrevBufferPercentage;
- int32_t mPollBufferingGeneration;
- bool mSentPauseOnBuffering;
-
- int32_t mAudioDataGeneration;
- int32_t mVideoDataGeneration;
- int32_t mFetchSubtitleDataGeneration;
- int32_t mFetchTimedTextDataGeneration;
- int64_t mDurationUs;
- bool mAudioIsVorbis;
- // Secure codec is required.
- bool mIsSecure;
- bool mIsStreaming;
- uid_t mUID;
- const sp<MediaClock> mMediaClock;
- AString mUri;
- KeyedVector<String8, String8> mUriHeaders;
- int mFd;
- int64_t mOffset;
- int64_t mLength;
-
- bool mDisconnected;
- sp<MetaData> mFileMeta;
- sp<AMediaDataSourceWrapper> mDataSourceWrapper;
- sp<AMediaExtractorWrapper> mExtractor;
- Vector<sp<AMediaExtractorWrapper> > mExtractors;
- bool mStarted;
- bool mPreparing;
- int64_t mBitrate;
- uint32_t mPendingReadBufferTypes;
- sp<ABuffer> mGlobalTimedText;
-
- Track mVideoTrack;
- Track mAudioTrack;
- Track mSubtitleTrack;
- Track mTimedTextTrack;
-
- mutable Mutex mLock;
-
- sp<ALooper> mLooper;
-
- void resetDataSource();
-
- status_t initFromDataSource();
- int64_t getLastReadPosition();
-
- void notifyPreparedAndCleanup(status_t err);
- void onSecureDecodersInstantiated(status_t err);
- void finishPrepareAsync();
- status_t startSources();
-
- void onSeek(const sp<AMessage>& msg);
- status_t doSeek(int64_t seekTimeUs, MediaPlayer2SeekMode mode);
-
- void onPrepareAsync(int64_t startTimeUs);
-
- void fetchTextData(
- uint32_t what, media_track_type type,
- int32_t curGen, const sp<AnotherPacketSource>& packets, const sp<AMessage>& msg);
-
- void sendGlobalTextData(
- uint32_t what,
- int32_t curGen, sp<AMessage> msg);
-
- void sendTextData(
- uint32_t what, media_track_type type,
- int32_t curGen, const sp<AnotherPacketSource>& packets, const sp<AMessage>& msg);
-
- sp<ABuffer> mediaBufferToABuffer(
- MediaBufferBase *mbuf,
- media_track_type trackType);
-
- void postReadBuffer(media_track_type trackType);
- void onReadBuffer(const sp<AMessage>& msg);
- // When |mode| is MediaPlayer2SeekMode::SEEK_CLOSEST, the buffer read shall
- // include an item indicating skipping rendering all buffers with timestamp
- // earlier than |seekTimeUs|.
- // For other modes, the buffer read will not include the item as above in order
- // to facilitate fast seek operation.
- void readBuffer(
- media_track_type trackType,
- int64_t seekTimeUs = -1ll,
- MediaPlayer2SeekMode mode = MediaPlayer2SeekMode::SEEK_PREVIOUS_SYNC,
- int64_t *actualTimeUs = NULL, bool formatChange = false);
-
- void queueDiscontinuityIfNeeded(
- bool seeking, bool formatChange, media_track_type trackType, Track *track);
-
- void schedulePollBuffering();
- void onPollBuffering();
- void notifyBufferingUpdate(int32_t percentage);
-
- sp<AMessage> getFormat_l(bool audio);
- sp<MetaData> getFormatMeta_l(bool audio);
- int32_t getDataGeneration(media_track_type type) const;
-
- // Modular DRM
- // The source is DRM protected and is prepared for DRM.
- bool mIsDrmProtected;
- // releaseDrm has been processed.
- bool mIsDrmReleased;
- Vector<String8> mMimes;
-
- status_t checkDrmInfo();
-
- DISALLOW_EVIL_CONSTRUCTORS(GenericSource2);
-};
-
-} // namespace android
-
-#endif // GENERIC_SOURCE2_H_
diff --git a/media/libmediaplayer2/nuplayer2/HTTPLiveSource2.cpp b/media/libmediaplayer2/nuplayer2/HTTPLiveSource2.cpp
deleted file mode 100644
index e53900b..0000000
--- a/media/libmediaplayer2/nuplayer2/HTTPLiveSource2.cpp
+++ /dev/null
@@ -1,450 +0,0 @@
-/*
- * Copyright 2017 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "HTTPLiveSource2"
-#include <utils/Log.h>
-
-#include "HTTPLiveSource2.h"
-
-#include "AnotherPacketSource.h"
-#include "LiveDataSource.h"
-
-#include <media/MediaHTTPService.h>
-#include <media/stagefright/foundation/ABuffer.h>
-#include <media/stagefright/foundation/ADebug.h>
-#include <media/stagefright/foundation/AMessage.h>
-#include <media/stagefright/MediaErrors.h>
-#include <media/stagefright/MetaData.h>
-#include <media/stagefright/MediaDefs.h>
-#include <media/stagefright/Utils.h>
-
-// default buffer prepare/ready/underflow marks
-static const int kReadyMarkMs = 5000; // 5 seconds
-static const int kPrepareMarkMs = 1500; // 1.5 seconds
-
-namespace android {
-
-NuPlayer2::HTTPLiveSource2::HTTPLiveSource2(
- const sp<AMessage> ¬ify,
- const sp<MediaHTTPService> &httpService,
- const char *url,
- const KeyedVector<String8, String8> *headers)
- : Source(notify),
- mHTTPService(httpService),
- mURL(url),
- mFlags(0),
- mFinalResult(OK),
- mOffset(0),
- mFetchSubtitleDataGeneration(0),
- mFetchMetaDataGeneration(0),
- mHasMetadata(false),
- mMetadataSelected(false) {
- mBufferingSettings.mInitialMarkMs = kPrepareMarkMs;
- mBufferingSettings.mResumePlaybackMarkMs = kReadyMarkMs;
- if (headers) {
- mExtraHeaders = *headers;
-
- ssize_t index =
- mExtraHeaders.indexOfKey(String8("x-hide-urls-from-log"));
-
- if (index >= 0) {
- mFlags |= kFlagIncognito;
-
- mExtraHeaders.removeItemsAt(index);
- }
- }
-}
-
-NuPlayer2::HTTPLiveSource2::~HTTPLiveSource2() {
- if (mLiveSession != NULL) {
- mLiveSession->disconnect();
-
- mLiveLooper->unregisterHandler(mLiveSession->id());
- mLiveLooper->unregisterHandler(id());
- mLiveLooper->stop();
-
- mLiveSession.clear();
- mLiveLooper.clear();
- }
-}
-
-status_t NuPlayer2::HTTPLiveSource2::getBufferingSettings(
- BufferingSettings* buffering /* nonnull */) {
- *buffering = mBufferingSettings;
-
- return OK;
-}
-
-status_t NuPlayer2::HTTPLiveSource2::setBufferingSettings(const BufferingSettings& buffering) {
- mBufferingSettings = buffering;
-
- if (mLiveSession != NULL) {
- mLiveSession->setBufferingSettings(mBufferingSettings);
- }
-
- return OK;
-}
-
-// TODO: fetch data starting from |startTimeUs|
-void NuPlayer2::HTTPLiveSource2::prepareAsync(int64_t /* startTimeUs */) {
- if (mLiveLooper == NULL) {
- mLiveLooper = new ALooper;
- mLiveLooper->setName("http live2");
- mLiveLooper->start(false, /* runOnCallingThread */
- true /* canCallJava */);
-
- mLiveLooper->registerHandler(this);
- }
-
- sp<AMessage> notify = new AMessage(kWhatSessionNotify, this);
-
- mLiveSession = new LiveSession(
- notify,
- (mFlags & kFlagIncognito) ? LiveSession::kFlagIncognito : 0,
- mHTTPService);
-
- mLiveLooper->registerHandler(mLiveSession);
-
- mLiveSession->setBufferingSettings(mBufferingSettings);
- mLiveSession->connectAsync(
- mURL.c_str(), mExtraHeaders.isEmpty() ? NULL : &mExtraHeaders);
-}
-
-void NuPlayer2::HTTPLiveSource2::start() {
-}
-
-sp<MetaData> NuPlayer2::HTTPLiveSource2::getFormatMeta(bool audio) {
- sp<MetaData> meta;
- if (mLiveSession != NULL) {
- mLiveSession->getStreamFormatMeta(
- audio ? LiveSession::STREAMTYPE_AUDIO
- : LiveSession::STREAMTYPE_VIDEO,
- &meta);
- }
-
- return meta;
-}
-
-sp<AMessage> NuPlayer2::HTTPLiveSource2::getFormat(bool audio) {
- sp<MetaData> meta;
- status_t err = -EWOULDBLOCK;
- if (mLiveSession != NULL) {
- err = mLiveSession->getStreamFormatMeta(
- audio ? LiveSession::STREAMTYPE_AUDIO
- : LiveSession::STREAMTYPE_VIDEO,
- &meta);
- }
-
- sp<AMessage> format;
- if (err == -EWOULDBLOCK) {
- format = new AMessage();
- format->setInt32("err", err);
- return format;
- }
-
- if (err != OK || convertMetaDataToMessage(meta, &format) != OK) {
- return NULL;
- }
- return format;
-}
-
-status_t NuPlayer2::HTTPLiveSource2::feedMoreTSData() {
- return OK;
-}
-
-status_t NuPlayer2::HTTPLiveSource2::dequeueAccessUnit(
- bool audio, sp<ABuffer> *accessUnit) {
- return mLiveSession->dequeueAccessUnit(
- audio ? LiveSession::STREAMTYPE_AUDIO
- : LiveSession::STREAMTYPE_VIDEO,
- accessUnit);
-}
-
-status_t NuPlayer2::HTTPLiveSource2::getDuration(int64_t *durationUs) {
- return mLiveSession->getDuration(durationUs);
-}
-
-size_t NuPlayer2::HTTPLiveSource2::getTrackCount() const {
- return mLiveSession->getTrackCount();
-}
-
-sp<AMessage> NuPlayer2::HTTPLiveSource2::getTrackInfo(size_t trackIndex) const {
- return mLiveSession->getTrackInfo(trackIndex);
-}
-
-ssize_t NuPlayer2::HTTPLiveSource2::getSelectedTrack(media_track_type type) const {
- if (mLiveSession == NULL) {
- return -1;
- } else if (type == MEDIA_TRACK_TYPE_METADATA) {
- // MEDIA_TRACK_TYPE_METADATA is always last track
- // mMetadataSelected can only be true when mHasMetadata is true
- return mMetadataSelected ? (mLiveSession->getTrackCount() - 1) : -1;
- } else {
- return mLiveSession->getSelectedTrack(type);
- }
-}
-
-status_t NuPlayer2::HTTPLiveSource2::selectTrack(size_t trackIndex, bool select, int64_t /*timeUs*/) {
- if (mLiveSession == NULL) {
- return INVALID_OPERATION;
- }
-
- status_t err = INVALID_OPERATION;
- bool postFetchMsg = false, isSub = false;
- if (!mHasMetadata || trackIndex != mLiveSession->getTrackCount() - 1) {
- err = mLiveSession->selectTrack(trackIndex, select);
- postFetchMsg = select;
- isSub = true;
- } else {
- // metadata track; i.e. (mHasMetadata && trackIndex == mLiveSession->getTrackCount() - 1)
- if (mMetadataSelected && !select) {
- err = OK;
- } else if (!mMetadataSelected && select) {
- postFetchMsg = true;
- err = OK;
- } else {
- err = BAD_VALUE; // behave as LiveSession::selectTrack
- }
-
- mMetadataSelected = select;
- }
-
- if (err == OK) {
- int32_t &generation = isSub ? mFetchSubtitleDataGeneration : mFetchMetaDataGeneration;
- generation++;
- if (postFetchMsg) {
- int32_t what = isSub ? kWhatFetchSubtitleData : kWhatFetchMetaData;
- sp<AMessage> msg = new AMessage(what, this);
- msg->setInt32("generation", generation);
- msg->post();
- }
- }
-
- // LiveSession::selectTrack returns BAD_VALUE when selecting the currently
- // selected track, or unselecting a non-selected track. In this case it's an
- // no-op so we return OK.
- return (err == OK || err == BAD_VALUE) ? (status_t)OK : err;
-}
-
-status_t NuPlayer2::HTTPLiveSource2::seekTo(int64_t seekTimeUs, MediaPlayer2SeekMode mode) {
- if (mLiveSession->isSeekable()) {
- return mLiveSession->seekTo(seekTimeUs, mode);
- } else {
- return INVALID_OPERATION;
- }
-}
-
-void NuPlayer2::HTTPLiveSource2::pollForRawData(
- const sp<AMessage> &msg, int32_t currentGeneration,
- LiveSession::StreamType fetchType, int32_t pushWhat) {
-
- int32_t generation;
- CHECK(msg->findInt32("generation", &generation));
-
- if (generation != currentGeneration) {
- return;
- }
-
- sp<ABuffer> buffer;
- while (mLiveSession->dequeueAccessUnit(fetchType, &buffer) == OK) {
-
- sp<AMessage> notify = dupNotify();
- notify->setInt32("what", pushWhat);
- notify->setBuffer("buffer", buffer);
-
- int64_t timeUs, baseUs, delayUs;
- CHECK(buffer->meta()->findInt64("baseUs", &baseUs));
- CHECK(buffer->meta()->findInt64("timeUs", &timeUs));
- delayUs = baseUs + timeUs - ALooper::GetNowUs();
-
- if (fetchType == LiveSession::STREAMTYPE_SUBTITLES) {
- notify->post();
- msg->post(delayUs > 0LL ? delayUs : 0LL);
- return;
- } else if (fetchType == LiveSession::STREAMTYPE_METADATA) {
- if (delayUs < -1000000LL) { // 1 second
- continue;
- }
- notify->post();
- // push all currently available metadata buffers in each invocation of pollForRawData
- // continue;
- } else {
- TRESPASS();
- }
- }
-
- // try again in 1 second
- msg->post(1000000LL);
-}
-
-void NuPlayer2::HTTPLiveSource2::onMessageReceived(const sp<AMessage> &msg) {
- switch (msg->what()) {
- case kWhatSessionNotify:
- {
- onSessionNotify(msg);
- break;
- }
-
- case kWhatFetchSubtitleData:
- {
- pollForRawData(
- msg, mFetchSubtitleDataGeneration,
- /* fetch */ LiveSession::STREAMTYPE_SUBTITLES,
- /* push */ kWhatSubtitleData);
-
- break;
- }
-
- case kWhatFetchMetaData:
- {
- if (!mMetadataSelected) {
- break;
- }
-
- pollForRawData(
- msg, mFetchMetaDataGeneration,
- /* fetch */ LiveSession::STREAMTYPE_METADATA,
- /* push */ kWhatTimedMetaData);
-
- break;
- }
-
- default:
- Source::onMessageReceived(msg);
- break;
- }
-}
-
-void NuPlayer2::HTTPLiveSource2::onSessionNotify(const sp<AMessage> &msg) {
- int32_t what;
- CHECK(msg->findInt32("what", &what));
-
- switch (what) {
- case LiveSession::kWhatPrepared:
- {
- // notify the current size here if we have it, otherwise report an initial size of (0,0)
- sp<AMessage> format = getFormat(false /* audio */);
- int32_t width;
- int32_t height;
- if (format != NULL &&
- format->findInt32("width", &width) && format->findInt32("height", &height)) {
- notifyVideoSizeChanged(format);
- } else {
- notifyVideoSizeChanged();
- }
-
- uint32_t flags = 0;
- if (mLiveSession->isSeekable()) {
- flags |= FLAG_CAN_PAUSE;
- flags |= FLAG_CAN_SEEK;
- flags |= FLAG_CAN_SEEK_BACKWARD;
- flags |= FLAG_CAN_SEEK_FORWARD;
- }
-
- if (mLiveSession->hasDynamicDuration()) {
- flags |= FLAG_DYNAMIC_DURATION;
- }
-
- notifyFlagsChanged(flags);
-
- notifyPrepared();
- break;
- }
-
- case LiveSession::kWhatPreparationFailed:
- {
- status_t err;
- CHECK(msg->findInt32("err", &err));
-
- notifyPrepared(err);
- break;
- }
-
- case LiveSession::kWhatStreamsChanged:
- {
- uint32_t changedMask;
- CHECK(msg->findInt32(
- "changedMask", (int32_t *)&changedMask));
-
- bool audio = changedMask & LiveSession::STREAMTYPE_AUDIO;
- bool video = changedMask & LiveSession::STREAMTYPE_VIDEO;
-
- sp<AMessage> reply;
- CHECK(msg->findMessage("reply", &reply));
-
- sp<AMessage> notify = dupNotify();
- notify->setInt32("what", kWhatQueueDecoderShutdown);
- notify->setInt32("audio", audio);
- notify->setInt32("video", video);
- notify->setMessage("reply", reply);
- notify->post();
- break;
- }
-
- case LiveSession::kWhatBufferingStart:
- {
- sp<AMessage> notify = dupNotify();
- notify->setInt32("what", kWhatPauseOnBufferingStart);
- notify->post();
- break;
- }
-
- case LiveSession::kWhatBufferingEnd:
- {
- sp<AMessage> notify = dupNotify();
- notify->setInt32("what", kWhatResumeOnBufferingEnd);
- notify->post();
- break;
- }
-
-
- case LiveSession::kWhatBufferingUpdate:
- {
- sp<AMessage> notify = dupNotify();
- int32_t percentage;
- CHECK(msg->findInt32("percentage", &percentage));
- notify->setInt32("what", kWhatBufferingUpdate);
- notify->setInt32("percentage", percentage);
- notify->post();
- break;
- }
-
- case LiveSession::kWhatMetadataDetected:
- {
- if (!mHasMetadata) {
- mHasMetadata = true;
-
- sp<AMessage> notify = dupNotify();
- // notification without buffer triggers MEDIA2_INFO_METADATA_UPDATE
- notify->setInt32("what", kWhatTimedMetaData);
- notify->post();
- }
- break;
- }
-
- case LiveSession::kWhatError:
- {
- break;
- }
-
- default:
- TRESPASS();
- }
-}
-
-} // namespace android
-
diff --git a/media/libmediaplayer2/nuplayer2/HTTPLiveSource2.h b/media/libmediaplayer2/nuplayer2/HTTPLiveSource2.h
deleted file mode 100644
index 8fc71e2..0000000
--- a/media/libmediaplayer2/nuplayer2/HTTPLiveSource2.h
+++ /dev/null
@@ -1,99 +0,0 @@
-/*
- * Copyright 2017 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef HTTP_LIVE_SOURCE2_H_
-
-#define HTTP_LIVE_SOURCE2_H_
-
-#include "NuPlayer2.h"
-#include "NuPlayer2Source.h"
-
-#include "LiveSession.h"
-
-namespace android {
-
-struct LiveSession;
-
-struct NuPlayer2::HTTPLiveSource2 : public NuPlayer2::Source {
- HTTPLiveSource2(
- const sp<AMessage> ¬ify,
- const sp<MediaHTTPService> &httpService,
- const char *url,
- const KeyedVector<String8, String8> *headers);
-
- virtual status_t getBufferingSettings(
- BufferingSettings* buffering /* nonnull */) override;
- virtual status_t setBufferingSettings(const BufferingSettings& buffering) override;
-
- virtual void prepareAsync(int64_t startTimeUs);
- virtual void start();
-
- virtual status_t dequeueAccessUnit(bool audio, sp<ABuffer> *accessUnit);
- virtual sp<MetaData> getFormatMeta(bool audio);
- virtual sp<AMessage> getFormat(bool audio);
-
- virtual status_t feedMoreTSData();
- virtual status_t getDuration(int64_t *durationUs);
- virtual size_t getTrackCount() const;
- virtual sp<AMessage> getTrackInfo(size_t trackIndex) const;
- virtual ssize_t getSelectedTrack(media_track_type /* type */) const;
- virtual status_t selectTrack(size_t trackIndex, bool select, int64_t timeUs);
- virtual status_t seekTo(
- int64_t seekTimeUs,
- MediaPlayer2SeekMode mode = MediaPlayer2SeekMode::SEEK_PREVIOUS_SYNC) override;
-
-protected:
- virtual ~HTTPLiveSource2();
-
- virtual void onMessageReceived(const sp<AMessage> &msg);
-
-private:
- enum Flags {
- // Don't log any URLs.
- kFlagIncognito = 1,
- };
-
- enum {
- kWhatSessionNotify,
- kWhatFetchSubtitleData,
- kWhatFetchMetaData,
- };
-
- sp<MediaHTTPService> mHTTPService;
- AString mURL;
- KeyedVector<String8, String8> mExtraHeaders;
- uint32_t mFlags;
- status_t mFinalResult;
- off64_t mOffset;
- sp<ALooper> mLiveLooper;
- sp<LiveSession> mLiveSession;
- int32_t mFetchSubtitleDataGeneration;
- int32_t mFetchMetaDataGeneration;
- bool mHasMetadata;
- bool mMetadataSelected;
- BufferingSettings mBufferingSettings;
-
- void onSessionNotify(const sp<AMessage> &msg);
- void pollForRawData(
- const sp<AMessage> &msg, int32_t currentGeneration,
- LiveSession::StreamType fetchType, int32_t pushWhat);
-
- DISALLOW_EVIL_CONSTRUCTORS(HTTPLiveSource2);
-};
-
-} // namespace android
-
-#endif // HTTP_LIVE_SOURCE2_H_
diff --git a/media/libmediaplayer2/nuplayer2/JMediaPlayer2Utils.cpp b/media/libmediaplayer2/nuplayer2/JMediaPlayer2Utils.cpp
deleted file mode 100644
index 89703de..0000000
--- a/media/libmediaplayer2/nuplayer2/JMediaPlayer2Utils.cpp
+++ /dev/null
@@ -1,65 +0,0 @@
-/*
- * Copyright 2018, The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "JMediaPlayer2Utils"
-
-#include "JMediaPlayer2Utils.h"
-#include <mediaplayer2/JavaVMHelper.h>
-
-#include <media/stagefright/MediaDefs.h>
-#include <media/stagefright/Utils.h>
-#include <utils/Log.h>
-
-#include "log/log.h"
-
-namespace android {
-
-static const int64_t kOffloadMinDurationSec = 60;
-
-// static
-bool JMediaPlayer2Utils::isOffloadedAudioPlaybackSupported(
- const sp<MetaData>& meta, bool hasVideo, bool isStreaming, audio_stream_type_t streamType)
-{
- if (hasVideo || streamType != AUDIO_STREAM_MUSIC) {
- return false;
- }
-
- audio_offload_info_t info = AUDIO_INFO_INITIALIZER;
- if (OK != getAudioOffloadInfo(meta, hasVideo, isStreaming, streamType, &info)) {
- return false;
- }
-
- if (info.duration_us < kOffloadMinDurationSec * 1000000) {
- return false;
- }
-
- int32_t audioFormat = audioFormatFromNative(info.format);
- int32_t channelMask = outChannelMaskFromNative(info.channel_mask);
- if (audioFormat == ENCODING_INVALID || channelMask == CHANNEL_INVALID) {
- return false;
- }
-
- JNIEnv* env = JavaVMHelper::getJNIEnv();
- jclass jMP2UtilsCls = env->FindClass("android/media/MediaPlayer2Utils");
- jmethodID jSetAudioOutputDeviceById = env->GetStaticMethodID(
- jMP2UtilsCls, "isOffloadedAudioPlaybackSupported", "(III)Z");
- jboolean result = env->CallStaticBooleanMethod(
- jMP2UtilsCls, jSetAudioOutputDeviceById, audioFormat, info.sample_rate, channelMask);
- return result;
-}
-
-} // namespace android
diff --git a/media/libmediaplayer2/nuplayer2/JMediaPlayer2Utils.h b/media/libmediaplayer2/nuplayer2/JMediaPlayer2Utils.h
deleted file mode 100644
index fcbd43c..0000000
--- a/media/libmediaplayer2/nuplayer2/JMediaPlayer2Utils.h
+++ /dev/null
@@ -1,35 +0,0 @@
-/*
- * Copyright 2018, The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef _J_MEDIAPLAYER2_UTILS2_H_
-#define _J_MEDIAPLAYER2_UTILS2_H_
-
-#include <media/stagefright/MetaData.h>
-
-#include "jni.h"
-#include "android_media_AudioFormat.h"
-
-namespace android {
-
-struct JMediaPlayer2Utils {
- static bool isOffloadedAudioPlaybackSupported(
- const sp<MetaData>& meta, bool hasVideo, bool isStreaming,
- audio_stream_type_t streamType);
-};
-
-} // namespace android
-
-#endif // _J_MEDIAPLAYER2_UTILS2_H_
diff --git a/media/libmediaplayer2/nuplayer2/JWakeLock.cpp b/media/libmediaplayer2/nuplayer2/JWakeLock.cpp
deleted file mode 100644
index 983d77e..0000000
--- a/media/libmediaplayer2/nuplayer2/JWakeLock.cpp
+++ /dev/null
@@ -1,97 +0,0 @@
-/*
- * Copyright 2017 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "JWakeLock"
-#include <utils/Log.h>
-
-#include "JWakeLock.h"
-
-#include <media/stagefright/foundation/ADebug.h>
-
-namespace android {
-
-JWakeLock::JWakeLock(const sp<JObjectHolder> &context) :
- mWakeLockCount(0),
- mWakeLock(NULL),
- mContext(context) {}
-
-JWakeLock::~JWakeLock() {
- clearJavaWakeLock();
-}
-
-bool JWakeLock::acquire() {
- if (mWakeLockCount == 0) {
- if (mWakeLock == NULL) {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jclass jContextCls = env->FindClass("android/content/Context");
- jclass jPowerManagerCls = env->FindClass("android/os/PowerManager");
-
- jmethodID jGetSystemService = env->GetMethodID(jContextCls,
- "getSystemService", "(Ljava/lang/String;)Ljava/lang/Object;");
- jobject javaPowerManagerObj = env->CallObjectMethod(mContext->getJObject(),
- jGetSystemService, env->NewStringUTF("power"));
-
- jfieldID jPARTIAL_WAKE_LOCK = env->GetStaticFieldID(jPowerManagerCls,
- "PARTIAL_WAKE_LOCK", "I");
- jint PARTIAL_WAKE_LOCK = env->GetStaticIntField(jPowerManagerCls, jPARTIAL_WAKE_LOCK);
-
- jmethodID jNewWakeLock = env->GetMethodID(jPowerManagerCls,
- "newWakeLock", "(ILjava/lang/String;)Landroid/os/PowerManager$WakeLock;");
- jobject javaWakeLock = env->CallObjectMethod(javaPowerManagerObj,
- jNewWakeLock, PARTIAL_WAKE_LOCK, env->NewStringUTF("JWakeLock"));
- mWakeLock = new JObjectHolder(javaWakeLock);
- env->DeleteLocalRef(javaPowerManagerObj);
- env->DeleteLocalRef(javaWakeLock);
- }
- if (mWakeLock != NULL) {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jclass wakeLockCls = env->FindClass("android/os/PowerManager$WakeLock");
- jmethodID jAcquire = env->GetMethodID(wakeLockCls, "acquire", "()V");
- env->CallVoidMethod(mWakeLock->getJObject(), jAcquire);
- mWakeLockCount++;
- return true;
- }
- } else {
- mWakeLockCount++;
- return true;
- }
- return false;
-}
-
-void JWakeLock::release(bool force) {
- if (mWakeLockCount == 0) {
- return;
- }
- if (force) {
- // Force wakelock release below by setting reference count to 1.
- mWakeLockCount = 1;
- }
- if (--mWakeLockCount == 0) {
- if (mWakeLock != NULL) {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jclass wakeLockCls = env->FindClass("android/os/PowerManager$WakeLock");
- jmethodID jRelease = env->GetMethodID(wakeLockCls, "release", "()V");
- env->CallVoidMethod(mWakeLock->getJObject(), jRelease);
- }
- }
-}
-
-void JWakeLock::clearJavaWakeLock() {
- release(true);
-}
-
-} // namespace android
diff --git a/media/libmediaplayer2/nuplayer2/JWakeLock.h b/media/libmediaplayer2/nuplayer2/JWakeLock.h
deleted file mode 100644
index 36c542e..0000000
--- a/media/libmediaplayer2/nuplayer2/JWakeLock.h
+++ /dev/null
@@ -1,51 +0,0 @@
-/*
- * Copyright 2017 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef J_WAKELOCK_H_
-#define J_WAKELOCK_H_
-
-#include <media/stagefright/foundation/ABase.h>
-#include <mediaplayer2/JObjectHolder.h>
-#include <utils/RefBase.h>
-
-namespace android {
-
-class JWakeLock : public RefBase {
-
-public:
- JWakeLock(const sp<JObjectHolder> &context);
-
- // NOTE: acquire and release are not thread safe
-
- // returns true if wakelock was acquired
- bool acquire();
- void release(bool force = false);
-
- virtual ~JWakeLock();
-
-private:
- uint32_t mWakeLockCount;
- sp<JObjectHolder> mWakeLock;
- const sp<JObjectHolder> mContext;
-
- void clearJavaWakeLock();
-
- DISALLOW_EVIL_CONSTRUCTORS(JWakeLock);
-};
-
-} // namespace android
-
-#endif // J_WAKELOCK_H_
diff --git a/media/libmediaplayer2/nuplayer2/MODULE_LICENSE_APACHE2 b/media/libmediaplayer2/nuplayer2/MODULE_LICENSE_APACHE2
deleted file mode 100644
index e69de29..0000000
--- a/media/libmediaplayer2/nuplayer2/MODULE_LICENSE_APACHE2
+++ /dev/null
diff --git a/media/libmediaplayer2/nuplayer2/NOTICE b/media/libmediaplayer2/nuplayer2/NOTICE
deleted file mode 100644
index c5b1efa..0000000
--- a/media/libmediaplayer2/nuplayer2/NOTICE
+++ /dev/null
@@ -1,190 +0,0 @@
-
- Copyright (c) 2005-2008, The Android Open Source Project
-
- Licensed under the Apache License, Version 2.0 (the "License");
- you may not use this file except in compliance with the License.
-
- Unless required by applicable law or agreed to in writing, software
- distributed under the License is distributed on an "AS IS" BASIS,
- WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- See the License for the specific language governing permissions and
- limitations under the License.
-
-
- Apache License
- Version 2.0, January 2004
- http://www.apache.org/licenses/
-
- TERMS AND CONDITIONS FOR USE, REPRODUCTION, AND DISTRIBUTION
-
- 1. Definitions.
-
- "License" shall mean the terms and conditions for use, reproduction,
- and distribution as defined by Sections 1 through 9 of this document.
-
- "Licensor" shall mean the copyright owner or entity authorized by
- the copyright owner that is granting the License.
-
- "Legal Entity" shall mean the union of the acting entity and all
- other entities that control, are controlled by, or are under common
- control with that entity. For the purposes of this definition,
- "control" means (i) the power, direct or indirect, to cause the
- direction or management of such entity, whether by contract or
- otherwise, or (ii) ownership of fifty percent (50%) or more of the
- outstanding shares, or (iii) beneficial ownership of such entity.
-
- "You" (or "Your") shall mean an individual or Legal Entity
- exercising permissions granted by this License.
-
- "Source" form shall mean the preferred form for making modifications,
- including but not limited to software source code, documentation
- source, and configuration files.
-
- "Object" form shall mean any form resulting from mechanical
- transformation or translation of a Source form, including but
- not limited to compiled object code, generated documentation,
- and conversions to other media types.
-
- "Work" shall mean the work of authorship, whether in Source or
- Object form, made available under the License, as indicated by a
- copyright notice that is included in or attached to the work
- (an example is provided in the Appendix below).
-
- "Derivative Works" shall mean any work, whether in Source or Object
- form, that is based on (or derived from) the Work and for which the
- editorial revisions, annotations, elaborations, or other modifications
- represent, as a whole, an original work of authorship. For the purposes
- of this License, Derivative Works shall not include works that remain
- separable from, or merely link (or bind by name) to the interfaces of,
- the Work and Derivative Works thereof.
-
- "Contribution" shall mean any work of authorship, including
- the original version of the Work and any modifications or additions
- to that Work or Derivative Works thereof, that is intentionally
- submitted to Licensor for inclusion in the Work by the copyright owner
- or by an individual or Legal Entity authorized to submit on behalf of
- the copyright owner. For the purposes of this definition, "submitted"
- means any form of electronic, verbal, or written communication sent
- to the Licensor or its representatives, including but not limited to
- communication on electronic mailing lists, source code control systems,
- and issue tracking systems that are managed by, or on behalf of, the
- Licensor for the purpose of discussing and improving the Work, but
- excluding communication that is conspicuously marked or otherwise
- designated in writing by the copyright owner as "Not a Contribution."
-
- "Contributor" shall mean Licensor and any individual or Legal Entity
- on behalf of whom a Contribution has been received by Licensor and
- subsequently incorporated within the Work.
-
- 2. Grant of Copyright License. Subject to the terms and conditions of
- this License, each Contributor hereby grants to You a perpetual,
- worldwide, non-exclusive, no-charge, royalty-free, irrevocable
- copyright license to reproduce, prepare Derivative Works of,
- publicly display, publicly perform, sublicense, and distribute the
- Work and such Derivative Works in Source or Object form.
-
- 3. Grant of Patent License. Subject to the terms and conditions of
- this License, each Contributor hereby grants to You a perpetual,
- worldwide, non-exclusive, no-charge, royalty-free, irrevocable
- (except as stated in this section) patent license to make, have made,
- use, offer to sell, sell, import, and otherwise transfer the Work,
- where such license applies only to those patent claims licensable
- by such Contributor that are necessarily infringed by their
- Contribution(s) alone or by combination of their Contribution(s)
- with the Work to which such Contribution(s) was submitted. If You
- institute patent litigation against any entity (including a
- cross-claim or counterclaim in a lawsuit) alleging that the Work
- or a Contribution incorporated within the Work constitutes direct
- or contributory patent infringement, then any patent licenses
- granted to You under this License for that Work shall terminate
- as of the date such litigation is filed.
-
- 4. Redistribution. You may reproduce and distribute copies of the
- Work or Derivative Works thereof in any medium, with or without
- modifications, and in Source or Object form, provided that You
- meet the following conditions:
-
- (a) You must give any other recipients of the Work or
- Derivative Works a copy of this License; and
-
- (b) You must cause any modified files to carry prominent notices
- stating that You changed the files; and
-
- (c) You must retain, in the Source form of any Derivative Works
- that You distribute, all copyright, patent, trademark, and
- attribution notices from the Source form of the Work,
- excluding those notices that do not pertain to any part of
- the Derivative Works; and
-
- (d) If the Work includes a "NOTICE" text file as part of its
- distribution, then any Derivative Works that You distribute must
- include a readable copy of the attribution notices contained
- within such NOTICE file, excluding those notices that do not
- pertain to any part of the Derivative Works, in at least one
- of the following places: within a NOTICE text file distributed
- as part of the Derivative Works; within the Source form or
- documentation, if provided along with the Derivative Works; or,
- within a display generated by the Derivative Works, if and
- wherever such third-party notices normally appear. The contents
- of the NOTICE file are for informational purposes only and
- do not modify the License. You may add Your own attribution
- notices within Derivative Works that You distribute, alongside
- or as an addendum to the NOTICE text from the Work, provided
- that such additional attribution notices cannot be construed
- as modifying the License.
-
- You may add Your own copyright statement to Your modifications and
- may provide additional or different license terms and conditions
- for use, reproduction, or distribution of Your modifications, or
- for any such Derivative Works as a whole, provided Your use,
- reproduction, and distribution of the Work otherwise complies with
- the conditions stated in this License.
-
- 5. Submission of Contributions. Unless You explicitly state otherwise,
- any Contribution intentionally submitted for inclusion in the Work
- by You to the Licensor shall be under the terms and conditions of
- this License, without any additional terms or conditions.
- Notwithstanding the above, nothing herein shall supersede or modify
- the terms of any separate license agreement you may have executed
- with Licensor regarding such Contributions.
-
- 6. Trademarks. This License does not grant permission to use the trade
- names, trademarks, service marks, or product names of the Licensor,
- except as required for reasonable and customary use in describing the
- origin of the Work and reproducing the content of the NOTICE file.
-
- 7. Disclaimer of Warranty. Unless required by applicable law or
- agreed to in writing, Licensor provides the Work (and each
- Contributor provides its Contributions) on an "AS IS" BASIS,
- WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or
- implied, including, without limitation, any warranties or conditions
- of TITLE, NON-INFRINGEMENT, MERCHANTABILITY, or FITNESS FOR A
- PARTICULAR PURPOSE. You are solely responsible for determining the
- appropriateness of using or redistributing the Work and assume any
- risks associated with Your exercise of permissions under this License.
-
- 8. Limitation of Liability. In no event and under no legal theory,
- whether in tort (including negligence), contract, or otherwise,
- unless required by applicable law (such as deliberate and grossly
- negligent acts) or agreed to in writing, shall any Contributor be
- liable to You for damages, including any direct, indirect, special,
- incidental, or consequential damages of any character arising as a
- result of this License or out of the use or inability to use the
- Work (including but not limited to damages for loss of goodwill,
- work stoppage, computer failure or malfunction, or any and all
- other commercial damages or losses), even if such Contributor
- has been advised of the possibility of such damages.
-
- 9. Accepting Warranty or Additional Liability. While redistributing
- the Work or Derivative Works thereof, You may choose to offer,
- and charge a fee for, acceptance of support, warranty, indemnity,
- or other liability obligations and/or rights consistent with this
- License. However, in accepting such obligations, You may act only
- on Your own behalf and on Your sole responsibility, not on behalf
- of any other Contributor, and only if You agree to indemnify,
- defend, and hold each Contributor harmless for any liability
- incurred by, or claims asserted against, such Contributor by reason
- of your accepting any such warranty or additional liability.
-
- END OF TERMS AND CONDITIONS
-
diff --git a/media/libmediaplayer2/nuplayer2/NuPlayer2.cpp b/media/libmediaplayer2/nuplayer2/NuPlayer2.cpp
deleted file mode 100644
index d608d4a..0000000
--- a/media/libmediaplayer2/nuplayer2/NuPlayer2.cpp
+++ /dev/null
@@ -1,3308 +0,0 @@
-/*
- * Copyright 2017 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "NuPlayer2"
-
-#include <inttypes.h>
-
-#include <utils/Log.h>
-
-#include "NuPlayer2.h"
-
-#include "HTTPLiveSource2.h"
-#include "JMediaPlayer2Utils.h"
-#include "NuPlayer2CCDecoder.h"
-#include "NuPlayer2Decoder.h"
-#include "NuPlayer2DecoderBase.h"
-#include "NuPlayer2DecoderPassThrough.h"
-#include "NuPlayer2Driver.h"
-#include "NuPlayer2Renderer.h"
-#include "NuPlayer2Source.h"
-#include "RTSPSource2.h"
-#include "GenericSource2.h"
-#include "TextDescriptions2.h"
-
-#include "ATSParser.h"
-
-#include <cutils/properties.h>
-
-#include <media/AudioParameter.h>
-#include <media/AudioResamplerPublic.h>
-#include <media/AVSyncSettings.h>
-#include <media/DataSourceDesc.h>
-#include <media/MediaCodecBuffer.h>
-#include <media/NdkWrapper.h>
-
-#include <media/stagefright/foundation/hexdump.h>
-#include <media/stagefright/foundation/ABuffer.h>
-#include <media/stagefright/foundation/ADebug.h>
-#include <media/stagefright/foundation/AMessage.h>
-#include <media/stagefright/foundation/avc_utils.h>
-#include <media/stagefright/MediaBuffer.h>
-#include <media/stagefright/MediaClock.h>
-#include <media/stagefright/MediaDefs.h>
-#include <media/stagefright/MediaErrors.h>
-#include <media/stagefright/MetaData.h>
-
-#include "ESDS.h"
-#include <media/stagefright/Utils.h>
-
-#include <system/window.h>
-
-namespace android {
-
-static status_t sendMetaDataToHal(sp<MediaPlayer2Interface::AudioSink>& sink,
- const sp<MetaData>& meta) {
- int32_t sampleRate = 0;
- int32_t bitRate = 0;
- int32_t channelMask = 0;
- int32_t delaySamples = 0;
- int32_t paddingSamples = 0;
-
- AudioParameter param = AudioParameter();
-
- if (meta->findInt32(kKeySampleRate, &sampleRate)) {
- param.addInt(String8(AUDIO_OFFLOAD_CODEC_SAMPLE_RATE), sampleRate);
- }
- if (meta->findInt32(kKeyChannelMask, &channelMask)) {
- param.addInt(String8(AUDIO_OFFLOAD_CODEC_NUM_CHANNEL), channelMask);
- }
- if (meta->findInt32(kKeyBitRate, &bitRate)) {
- param.addInt(String8(AUDIO_OFFLOAD_CODEC_AVG_BIT_RATE), bitRate);
- }
- if (meta->findInt32(kKeyEncoderDelay, &delaySamples)) {
- param.addInt(String8(AUDIO_OFFLOAD_CODEC_DELAY_SAMPLES), delaySamples);
- }
- if (meta->findInt32(kKeyEncoderPadding, &paddingSamples)) {
- param.addInt(String8(AUDIO_OFFLOAD_CODEC_PADDING_SAMPLES), paddingSamples);
- }
-
- ALOGV("sendMetaDataToHal: bitRate %d, sampleRate %d, chanMask %d,"
- "delaySample %d, paddingSample %d", bitRate, sampleRate,
- channelMask, delaySamples, paddingSamples);
-
- sink->setParameters(param.toString());
- return OK;
-}
-
-
-struct NuPlayer2::Action : public RefBase {
- Action() {}
-
- virtual void execute(NuPlayer2 *player) = 0;
-
-private:
- DISALLOW_EVIL_CONSTRUCTORS(Action);
-};
-
-struct NuPlayer2::SeekAction : public Action {
- explicit SeekAction(int64_t seekTimeUs, MediaPlayer2SeekMode mode)
- : mSeekTimeUs(seekTimeUs),
- mMode(mode) {
- }
-
- virtual void execute(NuPlayer2 *player) {
- player->performSeek(mSeekTimeUs, mMode);
- }
-
-private:
- int64_t mSeekTimeUs;
- MediaPlayer2SeekMode mMode;
-
- DISALLOW_EVIL_CONSTRUCTORS(SeekAction);
-};
-
-struct NuPlayer2::ResumeDecoderAction : public Action {
- explicit ResumeDecoderAction(bool needNotify)
- : mNeedNotify(needNotify) {
- }
-
- virtual void execute(NuPlayer2 *player) {
- player->performResumeDecoders(mNeedNotify);
- }
-
-private:
- bool mNeedNotify;
-
- DISALLOW_EVIL_CONSTRUCTORS(ResumeDecoderAction);
-};
-
-struct NuPlayer2::SetSurfaceAction : public Action {
- explicit SetSurfaceAction(const sp<ANativeWindowWrapper> &nww)
- : mNativeWindow(nww) {
- }
-
- virtual void execute(NuPlayer2 *player) {
- player->performSetSurface(mNativeWindow);
- }
-
-private:
- sp<ANativeWindowWrapper> mNativeWindow;
-
- DISALLOW_EVIL_CONSTRUCTORS(SetSurfaceAction);
-};
-
-struct NuPlayer2::FlushDecoderAction : public Action {
- FlushDecoderAction(FlushCommand audio, FlushCommand video)
- : mAudio(audio),
- mVideo(video) {
- }
-
- virtual void execute(NuPlayer2 *player) {
- player->performDecoderFlush(mAudio, mVideo);
- }
-
-private:
- FlushCommand mAudio;
- FlushCommand mVideo;
-
- DISALLOW_EVIL_CONSTRUCTORS(FlushDecoderAction);
-};
-
-struct NuPlayer2::PostMessageAction : public Action {
- explicit PostMessageAction(const sp<AMessage> &msg)
- : mMessage(msg) {
- }
-
- virtual void execute(NuPlayer2 *) {
- mMessage->post();
- }
-
-private:
- sp<AMessage> mMessage;
-
- DISALLOW_EVIL_CONSTRUCTORS(PostMessageAction);
-};
-
-// Use this if there's no state necessary to save in order to execute
-// the action.
-struct NuPlayer2::SimpleAction : public Action {
- typedef void (NuPlayer2::*ActionFunc)();
-
- explicit SimpleAction(ActionFunc func)
- : mFunc(func) {
- }
-
- virtual void execute(NuPlayer2 *player) {
- (player->*mFunc)();
- }
-
-private:
- ActionFunc mFunc;
-
- DISALLOW_EVIL_CONSTRUCTORS(SimpleAction);
-};
-
-////////////////////////////////////////////////////////////////////////////////
-
-NuPlayer2::NuPlayer2(
- pid_t pid, uid_t uid, const sp<MediaClock> &mediaClock, const sp<JObjectHolder> &context)
- : mPID(pid),
- mUID(uid),
- mMediaClock(mediaClock),
- mOffloadAudio(false),
- mAudioDecoderGeneration(0),
- mVideoDecoderGeneration(0),
- mRendererGeneration(0),
- mEOSMonitorGeneration(0),
- mLastStartedPlayingTimeNs(0),
- mPreviousSeekTimeUs(0),
- mAudioEOS(false),
- mVideoEOS(false),
- mScanSourcesPending(false),
- mScanSourcesGeneration(0),
- mPollDurationGeneration(0),
- mTimedTextGeneration(0),
- mFlushingAudio(NONE),
- mFlushingVideo(NONE),
- mResumePending(false),
- mVideoScalingMode(NATIVE_WINDOW_SCALING_MODE_SCALE_TO_WINDOW),
- mPlaybackSettings(AUDIO_PLAYBACK_RATE_DEFAULT),
- mVideoFpsHint(-1.f),
- mStarted(false),
- mPrepared(false),
- mResetting(false),
- mSourceStarted(false),
- mAudioDecoderError(false),
- mVideoDecoderError(false),
- mPaused(false),
- mPausedByClient(true),
- mPausedForBuffering(false),
- mContext(context) {
- CHECK(mediaClock != NULL);
- clearFlushComplete();
-}
-
-NuPlayer2::~NuPlayer2() {
-}
-
-void NuPlayer2::setDriver(const wp<NuPlayer2Driver> &driver) {
- mDriver = driver;
-}
-
-static bool IsHTTPLiveURL(const char *url) {
- if (!strncasecmp("http://", url, 7)
- || !strncasecmp("https://", url, 8)
- || !strncasecmp("file://", url, 7)) {
- size_t len = strlen(url);
- if (len >= 5 && !strcasecmp(".m3u8", &url[len - 5])) {
- return true;
- }
-
- if (strstr(url,"m3u8")) {
- return true;
- }
- }
-
- return false;
-}
-
-status_t NuPlayer2::createNuPlayer2Source(const sp<DataSourceDesc> &dsd,
- sp<Source> *source,
- DATA_SOURCE_TYPE *dataSourceType) {
- status_t err = NO_ERROR;
- sp<AMessage> notify = new AMessage(kWhatSourceNotify, this);
- notify->setInt64("srcId", dsd->mId);
-
- switch (dsd->mType) {
- case DataSourceDesc::TYPE_URL:
- {
- const char *url = dsd->mUrl.c_str();
- size_t len = strlen(url);
-
- const sp<MediaHTTPService> &httpService = dsd->mHttpService;
- KeyedVector<String8, String8> *headers = &(dsd->mHeaders);
-
- if (IsHTTPLiveURL(url)) {
- *source = new HTTPLiveSource2(notify, httpService, url, headers);
- ALOGV("createNuPlayer2Source HTTPLiveSource2 %s", url);
- *dataSourceType = DATA_SOURCE_TYPE_HTTP_LIVE;
- } else if (!strncasecmp(url, "rtsp://", 7)) {
- *source = new RTSPSource2(
- notify, httpService, url, headers, mUID);
- ALOGV("createNuPlayer2Source RTSPSource2 %s", url);
- *dataSourceType = DATA_SOURCE_TYPE_RTSP;
- } else if ((!strncasecmp(url, "http://", 7)
- || !strncasecmp(url, "https://", 8))
- && ((len >= 4 && !strcasecmp(".sdp", &url[len - 4]))
- || strstr(url, ".sdp?"))) {
- *source = new RTSPSource2(
- notify, httpService, url, headers, mUID, true);
- ALOGV("createNuPlayer2Source RTSPSource2 http/https/.sdp %s", url);
- *dataSourceType = DATA_SOURCE_TYPE_RTSP;
- } else {
- ALOGV("createNuPlayer2Source GenericSource2 %s", url);
-
- sp<GenericSource2> genericSource =
- new GenericSource2(notify, mUID, mMediaClock);
-
- err = genericSource->setDataSource(url, headers);
-
- if (err == OK) {
- *source = genericSource;
- } else {
- *source = NULL;
- ALOGE("Failed to create NuPlayer2Source!");
- }
-
- // regardless of success/failure
- *dataSourceType = DATA_SOURCE_TYPE_GENERIC_URL;
- }
- break;
- }
-
- case DataSourceDesc::TYPE_FD:
- {
- sp<GenericSource2> genericSource =
- new GenericSource2(notify, mUID, mMediaClock);
-
- ALOGV("createNuPlayer2Source fd %d/%lld/%lld source: %p",
- dsd->mFD, (long long)dsd->mFDOffset, (long long)dsd->mFDLength,
- genericSource.get());
-
- err = genericSource->setDataSource(dsd->mFD, dsd->mFDOffset, dsd->mFDLength);
-
- if (err != OK) {
- ALOGE("Failed to create NuPlayer2Source!");
- *source = NULL;
- } else {
- *source = genericSource;
- }
-
- *dataSourceType = DATA_SOURCE_TYPE_GENERIC_FD;
- break;
- }
-
- case DataSourceDesc::TYPE_CALLBACK:
- {
- sp<GenericSource2> genericSource =
- new GenericSource2(notify, mUID, mMediaClock);
- err = genericSource->setDataSource(dsd->mCallbackSource);
-
- if (err != OK) {
- ALOGE("Failed to create NuPlayer2Source!");
- *source = NULL;
- } else {
- *source = genericSource;
- }
-
- *dataSourceType = DATA_SOURCE_TYPE_MEDIA;
- break;
- }
-
- default:
- err = BAD_TYPE;
- *source = NULL;
- *dataSourceType = DATA_SOURCE_TYPE_NONE;
- ALOGE("invalid data source type!");
- break;
- }
-
- return err;
-}
-
-void NuPlayer2::setDataSourceAsync(const sp<DataSourceDesc> &dsd) {
- DATA_SOURCE_TYPE dataSourceType;
- sp<Source> source;
- createNuPlayer2Source(dsd, &source, &dataSourceType);
-
- // TODO: currently NuPlayer2Driver makes blocking call to setDataSourceAsync
- // and expects notifySetDataSourceCompleted regardless of success or failure.
- // This will be changed since setDataSource should be asynchronous at JAVA level.
- // When it succeeds, app will get onInfo notification. Otherwise, onError
- // will be called.
- /*
- if (err != OK) {
- notifyListener(dsd->mId, MEDIA2_ERROR, MEDIA2_ERROR_FAILED_TO_SET_DATA_SOURCE, err);
- return;
- }
-
- // Now, source != NULL.
- */
-
- mCurrentSourceInfo.mDataSourceType = dataSourceType;
-
- sp<AMessage> msg = new AMessage(kWhatSetDataSource, this);
- msg->setObject("source", source);
- msg->setInt64("srcId", dsd->mId);
- msg->setInt64("startTimeUs", dsd->mStartPositionMs * 1000);
- msg->setInt64("endTimeUs", dsd->mEndPositionMs * 1000);
- msg->post();
-}
-
-void NuPlayer2::prepareNextDataSourceAsync(const sp<DataSourceDesc> &dsd) {
- DATA_SOURCE_TYPE dataSourceType;
- sp<Source> source;
- createNuPlayer2Source(dsd, &source, &dataSourceType);
-
- /*
- if (err != OK) {
- notifyListener(dsd->mId, MEDIA2_ERROR, MEDIA2_ERROR_FAILED_TO_SET_DATA_SOURCE, err);
- return;
- }
-
- // Now, source != NULL.
- */
-
- mNextSourceInfo.mDataSourceType = dataSourceType;
-
- sp<AMessage> msg = new AMessage(kWhatPrepareNextDataSource, this);
- msg->setObject("source", source);
- msg->setInt64("srcId", dsd->mId);
- msg->setInt64("startTimeUs", dsd->mStartPositionMs * 1000);
- msg->setInt64("endTimeUs", dsd->mEndPositionMs * 1000);
- msg->post();
-}
-
-void NuPlayer2::playNextDataSource(int64_t srcId) {
- disconnectSource();
-
- sp<AMessage> msg = new AMessage(kWhatPlayNextDataSource, this);
- msg->setInt64("srcId", srcId);
- msg->post();
-}
-
-status_t NuPlayer2::getBufferingSettings(
- BufferingSettings *buffering /* nonnull */) {
- sp<AMessage> msg = new AMessage(kWhatGetBufferingSettings, this);
- sp<AMessage> response;
- status_t err = msg->postAndAwaitResponse(&response);
- if (err == OK && response != NULL) {
- CHECK(response->findInt32("err", &err));
- if (err == OK) {
- readFromAMessage(response, buffering);
- }
- }
- return err;
-}
-
-status_t NuPlayer2::setBufferingSettings(const BufferingSettings& buffering) {
- sp<AMessage> msg = new AMessage(kWhatSetBufferingSettings, this);
- writeToAMessage(msg, buffering);
- sp<AMessage> response;
- status_t err = msg->postAndAwaitResponse(&response);
- if (err == OK && response != NULL) {
- CHECK(response->findInt32("err", &err));
- }
- return err;
-}
-
-void NuPlayer2::prepareAsync() {
- ALOGV("prepareAsync");
-
- (new AMessage(kWhatPrepare, this))->post();
-}
-
-void NuPlayer2::setVideoSurfaceTextureAsync(const sp<ANativeWindowWrapper> &nww) {
- sp<AMessage> msg = new AMessage(kWhatSetVideoSurface, this);
-
- if (nww == NULL || nww->getANativeWindow() == NULL) {
- msg->setObject("surface", NULL);
- } else {
- msg->setObject("surface", nww);
- }
-
- msg->post();
-}
-
-void NuPlayer2::setAudioSink(const sp<MediaPlayer2Interface::AudioSink> &sink) {
- sp<AMessage> msg = new AMessage(kWhatSetAudioSink, this);
- msg->setObject("sink", sink);
- msg->post();
-}
-
-void NuPlayer2::start() {
- (new AMessage(kWhatStart, this))->post();
-}
-
-status_t NuPlayer2::setPlaybackSettings(const AudioPlaybackRate &rate) {
- // do some cursory validation of the settings here. audio modes are
- // only validated when set on the audiosink.
- if (rate.mSpeed < AUDIO_TIMESTRETCH_SPEED_MIN
- || rate.mSpeed > AUDIO_TIMESTRETCH_SPEED_MAX
- || rate.mPitch < AUDIO_TIMESTRETCH_SPEED_MIN
- || rate.mPitch > AUDIO_TIMESTRETCH_SPEED_MAX) {
- return BAD_VALUE;
- }
- sp<AMessage> msg = new AMessage(kWhatConfigPlayback, this);
- writeToAMessage(msg, rate);
- sp<AMessage> response;
- status_t err = msg->postAndAwaitResponse(&response);
- if (err == OK && response != NULL) {
- CHECK(response->findInt32("err", &err));
- }
- return err;
-}
-
-status_t NuPlayer2::getPlaybackSettings(AudioPlaybackRate *rate /* nonnull */) {
- sp<AMessage> msg = new AMessage(kWhatGetPlaybackSettings, this);
- sp<AMessage> response;
- status_t err = msg->postAndAwaitResponse(&response);
- if (err == OK && response != NULL) {
- CHECK(response->findInt32("err", &err));
- if (err == OK) {
- readFromAMessage(response, rate);
- }
- }
- return err;
-}
-
-status_t NuPlayer2::setSyncSettings(const AVSyncSettings &sync, float videoFpsHint) {
- sp<AMessage> msg = new AMessage(kWhatConfigSync, this);
- writeToAMessage(msg, sync, videoFpsHint);
- sp<AMessage> response;
- status_t err = msg->postAndAwaitResponse(&response);
- if (err == OK && response != NULL) {
- CHECK(response->findInt32("err", &err));
- }
- return err;
-}
-
-status_t NuPlayer2::getSyncSettings(
- AVSyncSettings *sync /* nonnull */, float *videoFps /* nonnull */) {
- sp<AMessage> msg = new AMessage(kWhatGetSyncSettings, this);
- sp<AMessage> response;
- status_t err = msg->postAndAwaitResponse(&response);
- if (err == OK && response != NULL) {
- CHECK(response->findInt32("err", &err));
- if (err == OK) {
- readFromAMessage(response, sync, videoFps);
- }
- }
- return err;
-}
-
-void NuPlayer2::pause() {
- (new AMessage(kWhatPause, this))->post();
-}
-
-void NuPlayer2::resetAsync() {
- disconnectSource();
- (new AMessage(kWhatReset, this))->post();
-}
-
-void NuPlayer2::disconnectSource() {
- sp<Source> source;
- {
- Mutex::Autolock autoLock(mSourceLock);
- source = mCurrentSourceInfo.mSource;
- }
-
- if (source != NULL) {
- // During a reset, the data source might be unresponsive already, we need to
- // disconnect explicitly so that reads exit promptly.
- // We can't queue the disconnect request to the looper, as it might be
- // queued behind a stuck read and never gets processed.
- // Doing a disconnect outside the looper to allows the pending reads to exit
- // (either successfully or with error).
- source->disconnect();
- }
-
-}
-
-status_t NuPlayer2::notifyAt(int64_t mediaTimeUs) {
- sp<AMessage> notify = new AMessage(kWhatNotifyTime, this);
- notify->setInt64("timerUs", mediaTimeUs);
- mMediaClock->addTimer(notify, mediaTimeUs);
- return OK;
-}
-
-void NuPlayer2::seekToAsync(int64_t seekTimeUs, MediaPlayer2SeekMode mode, bool needNotify) {
- sp<AMessage> msg = new AMessage(kWhatSeek, this);
- msg->setInt64("seekTimeUs", seekTimeUs);
- msg->setInt32("mode", mode);
- msg->setInt32("needNotify", needNotify);
- msg->post();
-}
-
-void NuPlayer2::rewind() {
- sp<AMessage> msg = new AMessage(kWhatRewind, this);
- msg->post();
-}
-
-void NuPlayer2::writeTrackInfo(
- PlayerMessage* reply, const sp<AMessage>& format) const {
- if (format == NULL) {
- ALOGE("NULL format");
- return;
- }
- int32_t trackType;
- if (!format->findInt32("type", &trackType)) {
- ALOGE("no track type");
- return;
- }
-
- AString mime;
- if (!format->findString("mime", &mime)) {
- // Java MediaPlayer only uses mimetype for subtitle and timedtext tracks.
- // If we can't find the mimetype here it means that we wouldn't be needing
- // the mimetype on the Java end. We still write a placeholder mime to keep the
- // (de)serialization logic simple.
- if (trackType == MEDIA_TRACK_TYPE_AUDIO) {
- mime = "audio/";
- } else if (trackType == MEDIA_TRACK_TYPE_VIDEO) {
- mime = "video/";
- } else {
- ALOGE("unknown track type: %d", trackType);
- return;
- }
- }
-
- AString lang;
- if (!format->findString("language", &lang)) {
- ALOGE("no language");
- return;
- }
-
- reply->add_values()->set_int32_value(trackType);
- reply->add_values()->set_string_value(mime.c_str());
- reply->add_values()->set_string_value(lang.c_str());
-
- if (trackType == MEDIA_TRACK_TYPE_SUBTITLE) {
- int32_t isAuto, isDefault, isForced;
- CHECK(format->findInt32("auto", &isAuto));
- CHECK(format->findInt32("default", &isDefault));
- CHECK(format->findInt32("forced", &isForced));
-
- reply->add_values()->set_int32_value(isAuto);
- reply->add_values()->set_int32_value(isDefault);
- reply->add_values()->set_int32_value(isForced);
- }
-}
-
-void NuPlayer2::onMessageReceived(const sp<AMessage> &msg) {
-
- switch (msg->what()) {
- case kWhatSetDataSource:
- {
- ALOGV("kWhatSetDataSource");
-
- CHECK(mCurrentSourceInfo.mSource == NULL);
-
- status_t err = OK;
- sp<RefBase> obj;
- CHECK(msg->findObject("source", &obj));
- if (obj != NULL) {
- Mutex::Autolock autoLock(mSourceLock);
- CHECK(msg->findInt64("srcId", &mCurrentSourceInfo.mSrcId));
- CHECK(msg->findInt64("startTimeUs", &mCurrentSourceInfo.mStartTimeUs));
- CHECK(msg->findInt64("endTimeUs", &mCurrentSourceInfo.mEndTimeUs));
- mCurrentSourceInfo.mSource = static_cast<Source *>(obj.get());
- } else {
- err = UNKNOWN_ERROR;
- ALOGE("kWhatSetDataSource, source should not be NULL");
- }
-
- CHECK(mDriver != NULL);
- sp<NuPlayer2Driver> driver = mDriver.promote();
- if (driver != NULL) {
- driver->notifySetDataSourceCompleted(mCurrentSourceInfo.mSrcId, err);
- }
- break;
- }
-
- case kWhatPrepareNextDataSource:
- {
- ALOGV("kWhatPrepareNextDataSource");
-
- status_t err = OK;
- sp<RefBase> obj;
- CHECK(msg->findObject("source", &obj));
- if (obj != NULL) {
- Mutex::Autolock autoLock(mSourceLock);
- CHECK(msg->findInt64("srcId", &mNextSourceInfo.mSrcId));
- CHECK(msg->findInt64("startTimeUs", &mNextSourceInfo.mStartTimeUs));
- CHECK(msg->findInt64("endTimeUs", &mNextSourceInfo.mEndTimeUs));
- mNextSourceInfo.mSource = static_cast<Source *>(obj.get());
- mNextSourceInfo.mSource->prepareAsync(mNextSourceInfo.mStartTimeUs);
- } else {
- err = UNKNOWN_ERROR;
- }
-
- break;
- }
-
- case kWhatPlayNextDataSource:
- {
- ALOGV("kWhatPlayNextDataSource");
- int64_t srcId;
- CHECK(msg->findInt64("srcId", &srcId));
- if (srcId != mNextSourceInfo.mSrcId) {
- notifyListener(srcId, MEDIA2_ERROR, MEDIA2_ERROR_UNKNOWN, 0);
- return;
- }
-
- mResetting = true;
- stopPlaybackTimer("kWhatPlayNextDataSource");
- stopRebufferingTimer(true);
-
- mDeferredActions.push_back(
- new FlushDecoderAction(
- FLUSH_CMD_SHUTDOWN /* audio */,
- FLUSH_CMD_SHUTDOWN /* video */));
-
- mDeferredActions.push_back(
- new SimpleAction(&NuPlayer2::performPlayNextDataSource));
-
- processDeferredActions();
- break;
- }
-
- case kWhatEOSMonitor:
- {
- int32_t generation;
- CHECK(msg->findInt32("generation", &generation));
- int32_t reason;
- CHECK(msg->findInt32("reason", &reason));
-
- if (generation != mEOSMonitorGeneration || reason != MediaClock::TIMER_REASON_REACHED) {
- break; // stale or reset
- }
-
- ALOGV("kWhatEOSMonitor");
- notifyListener(mCurrentSourceInfo.mSrcId, MEDIA2_PLAYBACK_COMPLETE, 0, 0);
- break;
- }
-
- case kWhatGetBufferingSettings:
- {
- sp<AReplyToken> replyID;
- CHECK(msg->senderAwaitsResponse(&replyID));
-
- ALOGV("kWhatGetBufferingSettings");
- BufferingSettings buffering;
- status_t err = OK;
- if (mCurrentSourceInfo.mSource != NULL) {
- err = mCurrentSourceInfo.mSource->getBufferingSettings(&buffering);
- } else {
- err = INVALID_OPERATION;
- }
- sp<AMessage> response = new AMessage;
- if (err == OK) {
- writeToAMessage(response, buffering);
- }
- response->setInt32("err", err);
- response->postReply(replyID);
- break;
- }
-
- case kWhatSetBufferingSettings:
- {
- sp<AReplyToken> replyID;
- CHECK(msg->senderAwaitsResponse(&replyID));
-
- ALOGV("kWhatSetBufferingSettings");
- BufferingSettings buffering;
- readFromAMessage(msg, &buffering);
- status_t err = OK;
- if (mCurrentSourceInfo.mSource != NULL) {
- err = mCurrentSourceInfo.mSource->setBufferingSettings(buffering);
- } else {
- err = INVALID_OPERATION;
- }
- sp<AMessage> response = new AMessage;
- response->setInt32("err", err);
- response->postReply(replyID);
- break;
- }
-
- case kWhatPrepare:
- {
- ALOGV("onMessageReceived kWhatPrepare");
-
- mCurrentSourceInfo.mSource->prepareAsync(mCurrentSourceInfo.mStartTimeUs);
- break;
- }
-
- case kWhatGetTrackInfo:
- {
- sp<AReplyToken> replyID;
- CHECK(msg->senderAwaitsResponse(&replyID));
-
- int64_t srcId;
- CHECK(msg->findInt64("srcId", (int64_t*)&srcId));
-
- PlayerMessage* reply;
- CHECK(msg->findPointer("reply", (void**)&reply));
-
- // TODO: use correct source info based on srcId.
- size_t inbandTracks = 0;
- if (mCurrentSourceInfo.mSource != NULL) {
- inbandTracks = mCurrentSourceInfo.mSource->getTrackCount();
- }
-
- size_t ccTracks = 0;
- if (mCCDecoder != NULL) {
- ccTracks = mCCDecoder->getTrackCount();
- }
-
- // total track count
- reply->add_values()->set_int32_value(inbandTracks + ccTracks);
-
- // write inband tracks
- for (size_t i = 0; i < inbandTracks; ++i) {
- writeTrackInfo(reply, mCurrentSourceInfo.mSource->getTrackInfo(i));
- }
-
- // write CC track
- for (size_t i = 0; i < ccTracks; ++i) {
- writeTrackInfo(reply, mCCDecoder->getTrackInfo(i));
- }
-
- sp<AMessage> response = new AMessage;
- response->postReply(replyID);
- break;
- }
-
- case kWhatGetSelectedTrack:
- {
- int64_t srcId;
- CHECK(msg->findInt64("srcId", (int64_t*)&srcId));
-
- int32_t type32;
- CHECK(msg->findInt32("type", (int32_t*)&type32));
- media_track_type type = (media_track_type)type32;
-
- // TODO: use correct source info based on srcId.
- size_t inbandTracks = 0;
- status_t err = INVALID_OPERATION;
- ssize_t selectedTrack = -1;
- if (mCurrentSourceInfo.mSource != NULL) {
- err = OK;
- inbandTracks = mCurrentSourceInfo.mSource->getTrackCount();
- selectedTrack = mCurrentSourceInfo.mSource->getSelectedTrack(type);
- }
-
- if (selectedTrack == -1 && mCCDecoder != NULL) {
- err = OK;
- selectedTrack = mCCDecoder->getSelectedTrack(type);
- if (selectedTrack != -1) {
- selectedTrack += inbandTracks;
- }
- }
-
- PlayerMessage* reply;
- CHECK(msg->findPointer("reply", (void**)&reply));
- reply->add_values()->set_int32_value(selectedTrack);
-
- sp<AMessage> response = new AMessage;
- response->setInt32("err", err);
-
- sp<AReplyToken> replyID;
- CHECK(msg->senderAwaitsResponse(&replyID));
- response->postReply(replyID);
- break;
- }
-
- case kWhatSelectTrack:
- {
- sp<AReplyToken> replyID;
- CHECK(msg->senderAwaitsResponse(&replyID));
-
- int64_t srcId;
- size_t trackIndex;
- int32_t select;
- int64_t timeUs;
- CHECK(msg->findInt64("srcId", (int64_t*)&srcId));
- CHECK(msg->findSize("trackIndex", &trackIndex));
- CHECK(msg->findInt32("select", &select));
- CHECK(msg->findInt64("timeUs", &timeUs));
-
- status_t err = INVALID_OPERATION;
-
- // TODO: use correct source info based on srcId.
- size_t inbandTracks = 0;
- if (mCurrentSourceInfo.mSource != NULL) {
- inbandTracks = mCurrentSourceInfo.mSource->getTrackCount();
- }
- size_t ccTracks = 0;
- if (mCCDecoder != NULL) {
- ccTracks = mCCDecoder->getTrackCount();
- }
-
- if (trackIndex < inbandTracks) {
- err = mCurrentSourceInfo.mSource->selectTrack(trackIndex, select, timeUs);
-
- if (!select && err == OK) {
- int32_t type;
- sp<AMessage> info = mCurrentSourceInfo.mSource->getTrackInfo(trackIndex);
- if (info != NULL
- && info->findInt32("type", &type)
- && type == MEDIA_TRACK_TYPE_TIMEDTEXT) {
- ++mTimedTextGeneration;
- }
- }
- } else {
- trackIndex -= inbandTracks;
-
- if (trackIndex < ccTracks) {
- err = mCCDecoder->selectTrack(trackIndex, select);
- }
- }
-
- sp<AMessage> response = new AMessage;
- response->setInt32("err", err);
-
- response->postReply(replyID);
- break;
- }
-
- case kWhatPollDuration:
- {
- int32_t generation;
- CHECK(msg->findInt32("generation", &generation));
-
- if (generation != mPollDurationGeneration) {
- // stale
- break;
- }
-
- int64_t durationUs;
- if (mDriver != NULL && mCurrentSourceInfo.mSource->getDuration(&durationUs) == OK) {
- sp<NuPlayer2Driver> driver = mDriver.promote();
- if (driver != NULL) {
- driver->notifyDuration(mCurrentSourceInfo.mSrcId, durationUs);
- }
- }
-
- msg->post(1000000LL); // poll again in a second.
- break;
- }
-
- case kWhatSetVideoSurface:
- {
-
- sp<RefBase> obj;
- CHECK(msg->findObject("surface", &obj));
- sp<ANativeWindowWrapper> nww = static_cast<ANativeWindowWrapper *>(obj.get());
-
- ALOGD("onSetVideoSurface(%p, %s video decoder)",
- (nww == NULL ? NULL : nww->getANativeWindow()),
- (mCurrentSourceInfo.mSource != NULL && mStarted
- && mCurrentSourceInfo.mSource->getFormat(false /* audio */) != NULL
- && mVideoDecoder != NULL) ? "have" : "no");
-
- // Need to check mStarted before calling mCurrentSourceInfo.mSource->getFormat
- // because NuPlayer2 might be in preparing state and it could take long time.
- // When mStarted is true, mCurrentSourceInfo.mSource must have been set.
- if (mCurrentSourceInfo.mSource == NULL || !mStarted
- || mCurrentSourceInfo.mSource->getFormat(false /* audio */) == NULL
- // NOTE: mVideoDecoder's mNativeWindow is always non-null
- || (mVideoDecoder != NULL && mVideoDecoder->setVideoSurface(nww) == OK)) {
- performSetSurface(nww);
- break;
- }
-
- mDeferredActions.push_back(
- new FlushDecoderAction(
- (obj != NULL ? FLUSH_CMD_FLUSH : FLUSH_CMD_NONE) /* audio */,
- FLUSH_CMD_SHUTDOWN /* video */));
-
- mDeferredActions.push_back(new SetSurfaceAction(nww));
-
- if (obj != NULL) {
- if (mStarted) {
- // Issue a seek to refresh the video screen only if started otherwise
- // the extractor may not yet be started and will assert.
- // If the video decoder is not set (perhaps audio only in this case)
- // do not perform a seek as it is not needed.
- int64_t currentPositionUs = 0;
- if (getCurrentPosition(¤tPositionUs) == OK) {
- mDeferredActions.push_back(
- new SeekAction(currentPositionUs,
- MediaPlayer2SeekMode::SEEK_PREVIOUS_SYNC /* mode */));
- }
- }
-
- // If there is a new surface texture, instantiate decoders
- // again if possible.
- mDeferredActions.push_back(
- new SimpleAction(&NuPlayer2::performScanSources));
-
- // After a flush without shutdown, decoder is paused.
- // Don't resume it until source seek is done, otherwise it could
- // start pulling stale data too soon.
- mDeferredActions.push_back(
- new ResumeDecoderAction(false /* needNotify */));
- }
-
- processDeferredActions();
- break;
- }
-
- case kWhatSetAudioSink:
- {
- ALOGV("kWhatSetAudioSink");
-
- sp<RefBase> obj;
- CHECK(msg->findObject("sink", &obj));
-
- mAudioSink = static_cast<MediaPlayer2Interface::AudioSink *>(obj.get());
- break;
- }
-
- case kWhatStart:
- {
- ALOGV("kWhatStart");
- if (mStarted) {
- // do not resume yet if the source is still buffering
- if (!mPausedForBuffering) {
- onResume();
- }
- } else {
- onStart(true /* play */);
- }
- mPausedByClient = false;
- notifyListener(mCurrentSourceInfo.mSrcId, MEDIA2_STARTED, 0, 0);
- break;
- }
-
- case kWhatConfigPlayback:
- {
- sp<AReplyToken> replyID;
- CHECK(msg->senderAwaitsResponse(&replyID));
- AudioPlaybackRate rate /* sanitized */;
- readFromAMessage(msg, &rate);
- status_t err = OK;
- if (mRenderer != NULL) {
- // AudioSink allows only 1.f and 0.f for offload mode.
- // For other speed, switch to non-offload mode.
- if (mOffloadAudio && (rate.mSpeed != 1.f || rate.mPitch != 1.f)) {
- int64_t currentPositionUs;
- if (getCurrentPosition(¤tPositionUs) != OK) {
- currentPositionUs = mPreviousSeekTimeUs;
- }
-
- // Set mPlaybackSettings so that the new audio decoder can
- // be created correctly.
- mPlaybackSettings = rate;
- if (!mPaused) {
- mRenderer->pause();
- }
- restartAudio(
- currentPositionUs, true /* forceNonOffload */,
- true /* needsToCreateAudioDecoder */);
- if (!mPaused) {
- mRenderer->resume();
- }
- }
-
- err = mRenderer->setPlaybackSettings(rate);
- }
- if (err == OK) {
- mPlaybackSettings = rate;
-
- if (mVideoDecoder != NULL) {
- sp<AMessage> params = new AMessage();
- params->setFloat("playback-speed", mPlaybackSettings.mSpeed);
- mVideoDecoder->setParameters(params);
- }
- }
-
- sp<AMessage> response = new AMessage;
- response->setInt32("err", err);
- response->postReply(replyID);
- break;
- }
-
- case kWhatGetPlaybackSettings:
- {
- sp<AReplyToken> replyID;
- CHECK(msg->senderAwaitsResponse(&replyID));
- AudioPlaybackRate rate = mPlaybackSettings;
- status_t err = OK;
- if (mRenderer != NULL) {
- err = mRenderer->getPlaybackSettings(&rate);
- }
- if (err == OK) {
- // get playback settings used by renderer, as it may be
- // slightly off due to audiosink not taking small changes.
- mPlaybackSettings = rate;
- }
- sp<AMessage> response = new AMessage;
- if (err == OK) {
- writeToAMessage(response, rate);
- }
- response->setInt32("err", err);
- response->postReply(replyID);
- break;
- }
-
- case kWhatConfigSync:
- {
- sp<AReplyToken> replyID;
- CHECK(msg->senderAwaitsResponse(&replyID));
-
- ALOGV("kWhatConfigSync");
- AVSyncSettings sync;
- float videoFpsHint;
- readFromAMessage(msg, &sync, &videoFpsHint);
- status_t err = OK;
- if (mRenderer != NULL) {
- err = mRenderer->setSyncSettings(sync, videoFpsHint);
- }
- if (err == OK) {
- mSyncSettings = sync;
- mVideoFpsHint = videoFpsHint;
- }
- sp<AMessage> response = new AMessage;
- response->setInt32("err", err);
- response->postReply(replyID);
- break;
- }
-
- case kWhatGetSyncSettings:
- {
- sp<AReplyToken> replyID;
- CHECK(msg->senderAwaitsResponse(&replyID));
- AVSyncSettings sync = mSyncSettings;
- float videoFps = mVideoFpsHint;
- status_t err = OK;
- if (mRenderer != NULL) {
- err = mRenderer->getSyncSettings(&sync, &videoFps);
- if (err == OK) {
- mSyncSettings = sync;
- mVideoFpsHint = videoFps;
- }
- }
- sp<AMessage> response = new AMessage;
- if (err == OK) {
- writeToAMessage(response, sync, videoFps);
- }
- response->setInt32("err", err);
- response->postReply(replyID);
- break;
- }
-
- case kWhatScanSources:
- {
- int32_t generation;
- CHECK(msg->findInt32("generation", &generation));
- if (generation != mScanSourcesGeneration) {
- // Drop obsolete msg.
- break;
- }
-
- mScanSourcesPending = false;
-
- ALOGV("scanning sources haveAudio=%d, haveVideo=%d",
- mAudioDecoder != NULL, mVideoDecoder != NULL);
-
- bool mHadAnySourcesBefore =
- (mAudioDecoder != NULL) || (mVideoDecoder != NULL);
- bool rescan = false;
-
- // initialize video before audio because successful initialization of
- // video may change deep buffer mode of audio.
- if (mNativeWindow != NULL && mNativeWindow->getANativeWindow() != NULL) {
- if (instantiateDecoder(false, &mVideoDecoder) == -EWOULDBLOCK) {
- rescan = true;
- }
- }
-
- // Don't try to re-open audio sink if there's an existing decoder.
- if (mAudioSink != NULL && mAudioDecoder == NULL) {
- if (instantiateDecoder(true, &mAudioDecoder) == -EWOULDBLOCK) {
- rescan = true;
- }
- }
-
- if (!mHadAnySourcesBefore
- && (mAudioDecoder != NULL || mVideoDecoder != NULL)) {
- // This is the first time we've found anything playable.
-
- if (mCurrentSourceInfo.mSourceFlags & Source::FLAG_DYNAMIC_DURATION) {
- schedulePollDuration();
- }
- }
-
- status_t err;
- if ((err = mCurrentSourceInfo.mSource->feedMoreTSData()) != OK) {
- if (mAudioDecoder == NULL && mVideoDecoder == NULL) {
- // We're not currently decoding anything (no audio or
- // video tracks found) and we just ran out of input data.
-
- if (err == ERROR_END_OF_STREAM) {
- notifyListener(mCurrentSourceInfo.mSrcId, MEDIA2_PLAYBACK_COMPLETE, 0, 0);
- } else {
- notifyListener(
- mCurrentSourceInfo.mSrcId, MEDIA2_ERROR, MEDIA2_ERROR_UNKNOWN, err);
- }
- }
- break;
- }
-
- if (rescan) {
- msg->post(100000LL);
- mScanSourcesPending = true;
- }
- break;
- }
-
- case kWhatVideoNotify:
- case kWhatAudioNotify:
- {
- bool audio = msg->what() == kWhatAudioNotify;
-
- int32_t currentDecoderGeneration =
- (audio? mAudioDecoderGeneration : mVideoDecoderGeneration);
- int32_t requesterGeneration = currentDecoderGeneration - 1;
- CHECK(msg->findInt32("generation", &requesterGeneration));
-
- if (requesterGeneration != currentDecoderGeneration) {
- ALOGV("got message from old %s decoder, generation(%d:%d)",
- audio ? "audio" : "video", requesterGeneration,
- currentDecoderGeneration);
- sp<AMessage> reply;
- if (!(msg->findMessage("reply", &reply))) {
- return;
- }
-
- reply->setInt32("err", INFO_DISCONTINUITY);
- reply->post();
- return;
- }
-
- int32_t what;
- CHECK(msg->findInt32("what", &what));
-
- if (what == DecoderBase::kWhatInputDiscontinuity) {
- int32_t formatChange;
- CHECK(msg->findInt32("formatChange", &formatChange));
-
- ALOGV("%s discontinuity: formatChange %d",
- audio ? "audio" : "video", formatChange);
-
- if (formatChange) {
- mDeferredActions.push_back(
- new FlushDecoderAction(
- audio ? FLUSH_CMD_SHUTDOWN : FLUSH_CMD_NONE,
- audio ? FLUSH_CMD_NONE : FLUSH_CMD_SHUTDOWN));
- }
-
- mDeferredActions.push_back(
- new SimpleAction(
- &NuPlayer2::performScanSources));
-
- processDeferredActions();
- } else if (what == DecoderBase::kWhatEOS) {
- int32_t err;
- CHECK(msg->findInt32("err", &err));
-
- if (err == ERROR_END_OF_STREAM) {
- ALOGV("got %s decoder EOS", audio ? "audio" : "video");
- } else {
- ALOGV("got %s decoder EOS w/ error %d",
- audio ? "audio" : "video",
- err);
- }
-
- mRenderer->queueEOS(audio, err);
- } else if (what == DecoderBase::kWhatFlushCompleted) {
- ALOGV("decoder %s flush completed", audio ? "audio" : "video");
-
- handleFlushComplete(audio, true /* isDecoder */);
- finishFlushIfPossible();
- } else if (what == DecoderBase::kWhatVideoSizeChanged) {
- sp<AMessage> format;
- CHECK(msg->findMessage("format", &format));
-
- sp<AMessage> inputFormat =
- mCurrentSourceInfo.mSource->getFormat(false /* audio */);
-
- setVideoScalingMode(mVideoScalingMode);
- updateVideoSize(mCurrentSourceInfo.mSrcId, inputFormat, format);
- } else if (what == DecoderBase::kWhatShutdownCompleted) {
- ALOGV("%s shutdown completed", audio ? "audio" : "video");
- if (audio) {
- Mutex::Autolock autoLock(mDecoderLock);
- mAudioDecoder.clear();
- mAudioDecoderError = false;
- ++mAudioDecoderGeneration;
-
- CHECK_EQ((int)mFlushingAudio, (int)SHUTTING_DOWN_DECODER);
- mFlushingAudio = SHUT_DOWN;
- } else {
- Mutex::Autolock autoLock(mDecoderLock);
- mVideoDecoder.clear();
- mVideoDecoderError = false;
- ++mVideoDecoderGeneration;
-
- CHECK_EQ((int)mFlushingVideo, (int)SHUTTING_DOWN_DECODER);
- mFlushingVideo = SHUT_DOWN;
- }
-
- finishFlushIfPossible();
- } else if (what == DecoderBase::kWhatResumeCompleted) {
- finishResume();
- } else if (what == DecoderBase::kWhatError) {
- status_t err;
- if (!msg->findInt32("err", &err) || err == OK) {
- err = UNKNOWN_ERROR;
- }
-
- // Decoder errors can be due to Source (e.g. from streaming),
- // or from decoding corrupted bitstreams, or from other decoder
- // MediaCodec operations (e.g. from an ongoing reset or seek).
- // They may also be due to openAudioSink failure at
- // decoder start or after a format change.
- //
- // We try to gracefully shut down the affected decoder if possible,
- // rather than trying to force the shutdown with something
- // similar to performReset(). This method can lead to a hang
- // if MediaCodec functions block after an error, but they should
- // typically return INVALID_OPERATION instead of blocking.
-
- FlushStatus *flushing = audio ? &mFlushingAudio : &mFlushingVideo;
- ALOGE("received error(%#x) from %s decoder, flushing(%d), now shutting down",
- err, audio ? "audio" : "video", *flushing);
-
- switch (*flushing) {
- case NONE:
- mDeferredActions.push_back(
- new FlushDecoderAction(
- audio ? FLUSH_CMD_SHUTDOWN : FLUSH_CMD_NONE,
- audio ? FLUSH_CMD_NONE : FLUSH_CMD_SHUTDOWN));
- processDeferredActions();
- break;
- case FLUSHING_DECODER:
- *flushing = FLUSHING_DECODER_SHUTDOWN; // initiate shutdown after flush.
- break; // Wait for flush to complete.
- case FLUSHING_DECODER_SHUTDOWN:
- break; // Wait for flush to complete.
- case SHUTTING_DOWN_DECODER:
- break; // Wait for shutdown to complete.
- case FLUSHED:
- getDecoder(audio)->initiateShutdown(); // In the middle of a seek.
- *flushing = SHUTTING_DOWN_DECODER; // Shut down.
- break;
- case SHUT_DOWN:
- finishFlushIfPossible(); // Should not occur.
- break; // Finish anyways.
- }
- if (mCurrentSourceInfo.mSource != nullptr) {
- if (audio) {
- if (mVideoDecoderError
- || mCurrentSourceInfo.mSource->getFormat(false /* audio */) == NULL
- || mNativeWindow == NULL
- || mNativeWindow->getANativeWindow() == NULL
- || mVideoDecoder == NULL) {
- // When both audio and video have error, or this stream has only audio
- // which has error, notify client of error.
- notifyListener(
- mCurrentSourceInfo.mSrcId, MEDIA2_ERROR,
- MEDIA2_ERROR_UNKNOWN, err);
- } else {
- // Only audio track has error. Video track could be still good to play.
- notifyListener(
- mCurrentSourceInfo.mSrcId, MEDIA2_INFO,
- MEDIA2_INFO_PLAY_AUDIO_ERROR, err);
- }
- mAudioDecoderError = true;
- } else {
- if (mAudioDecoderError
- || mCurrentSourceInfo.mSource->getFormat(true /* audio */) == NULL
- || mAudioSink == NULL || mAudioDecoder == NULL) {
- // When both audio and video have error, or this stream has only video
- // which has error, notify client of error.
- notifyListener(
- mCurrentSourceInfo.mSrcId, MEDIA2_ERROR,
- MEDIA2_ERROR_UNKNOWN, err);
- } else {
- // Only video track has error. Audio track could be still good to play.
- notifyListener(
- mCurrentSourceInfo.mSrcId, MEDIA2_INFO,
- MEDIA2_INFO_PLAY_VIDEO_ERROR, err);
- }
- mVideoDecoderError = true;
- }
- }
- } else {
- ALOGV("Unhandled decoder notification %d '%c%c%c%c'.",
- what,
- what >> 24,
- (what >> 16) & 0xff,
- (what >> 8) & 0xff,
- what & 0xff);
- }
-
- break;
- }
-
- case kWhatRendererNotify:
- {
- int32_t requesterGeneration = mRendererGeneration - 1;
- CHECK(msg->findInt32("generation", &requesterGeneration));
- if (requesterGeneration != mRendererGeneration) {
- ALOGV("got message from old renderer, generation(%d:%d)",
- requesterGeneration, mRendererGeneration);
- return;
- }
-
- int32_t what;
- CHECK(msg->findInt32("what", &what));
-
- if (what == Renderer::kWhatEOS) {
- int32_t audio;
- CHECK(msg->findInt32("audio", &audio));
-
- int32_t finalResult;
- CHECK(msg->findInt32("finalResult", &finalResult));
-
- if (audio) {
- mAudioEOS = true;
- } else {
- mVideoEOS = true;
- }
-
- if (finalResult == ERROR_END_OF_STREAM) {
- ALOGV("reached %s EOS", audio ? "audio" : "video");
- } else {
- ALOGE("%s track encountered an error (%d)",
- audio ? "audio" : "video", finalResult);
-
- notifyListener(
- mCurrentSourceInfo.mSrcId, MEDIA2_ERROR,
- MEDIA2_ERROR_UNKNOWN, finalResult);
- }
-
- if ((mAudioEOS || mAudioDecoder == NULL)
- && (mVideoEOS || mVideoDecoder == NULL)) {
- notifyListener(mCurrentSourceInfo.mSrcId, MEDIA2_PLAYBACK_COMPLETE, 0, 0);
- }
- } else if (what == Renderer::kWhatFlushComplete) {
- int32_t audio;
- CHECK(msg->findInt32("audio", &audio));
-
- if (audio) {
- mAudioEOS = false;
- } else {
- mVideoEOS = false;
- }
-
- ALOGV("renderer %s flush completed.", audio ? "audio" : "video");
- if (audio && (mFlushingAudio == NONE || mFlushingAudio == FLUSHED
- || mFlushingAudio == SHUT_DOWN)) {
- // Flush has been handled by tear down.
- break;
- }
- handleFlushComplete(audio, false /* isDecoder */);
- finishFlushIfPossible();
- } else if (what == Renderer::kWhatVideoRenderingStart) {
- notifyListener(mCurrentSourceInfo.mSrcId, MEDIA2_INFO,
- MEDIA2_INFO_VIDEO_RENDERING_START, 0);
- } else if (what == Renderer::kWhatMediaRenderingStart) {
- ALOGV("media rendering started");
- notifyListener(mCurrentSourceInfo.mSrcId, MEDIA2_STARTED, 0, 0);
- } else if (what == Renderer::kWhatAudioTearDown) {
- int32_t reason;
- CHECK(msg->findInt32("reason", &reason));
- ALOGV("Tear down audio with reason %d.", reason);
- if (reason == Renderer::kDueToTimeout && !(mPaused && mOffloadAudio)) {
- // TimeoutWhenPaused is only for offload mode.
- ALOGW("Receive a stale message for teardown.");
- break;
- }
- int64_t positionUs;
- if (!msg->findInt64("positionUs", &positionUs)) {
- positionUs = mPreviousSeekTimeUs;
- }
-
- restartAudio(
- positionUs, reason == Renderer::kForceNonOffload /* forceNonOffload */,
- reason != Renderer::kDueToTimeout /* needsToCreateAudioDecoder */);
- }
- break;
- }
-
- case kWhatMoreDataQueued:
- {
- break;
- }
-
- case kWhatReset:
- {
- ALOGV("kWhatReset");
-
- mResetting = true;
- stopPlaybackTimer("kWhatReset");
- stopRebufferingTimer(true);
-
- mDeferredActions.push_back(
- new FlushDecoderAction(
- FLUSH_CMD_SHUTDOWN /* audio */,
- FLUSH_CMD_SHUTDOWN /* video */));
-
- mDeferredActions.push_back(
- new SimpleAction(&NuPlayer2::performReset));
-
- processDeferredActions();
- break;
- }
-
- case kWhatNotifyTime:
- {
- ALOGV("kWhatNotifyTime");
- int64_t timerUs;
- CHECK(msg->findInt64("timerUs", &timerUs));
-
- notifyListener(mCurrentSourceInfo.mSrcId, MEDIA2_NOTIFY_TIME, timerUs, 0);
- break;
- }
-
- case kWhatSeek:
- {
- int64_t seekTimeUs;
- int32_t mode;
- int32_t needNotify;
- CHECK(msg->findInt64("seekTimeUs", &seekTimeUs));
- CHECK(msg->findInt32("mode", &mode));
- CHECK(msg->findInt32("needNotify", &needNotify));
-
- ALOGV("kWhatSeek seekTimeUs=%lld us, mode=%d, needNotify=%d",
- (long long)seekTimeUs, mode, needNotify);
-
- if (!mStarted) {
- if (!mSourceStarted) {
- mSourceStarted = true;
- mCurrentSourceInfo.mSource->start();
- }
- if (seekTimeUs > 0) {
- performSeek(seekTimeUs, (MediaPlayer2SeekMode)mode);
- }
-
- if (needNotify) {
- notifyDriverSeekComplete(mCurrentSourceInfo.mSrcId);
- }
- break;
- }
-
- // seeks can take a while, so we essentially paused
- notifyListener(mCurrentSourceInfo.mSrcId, MEDIA2_PAUSED, 0, 0);
-
- mDeferredActions.push_back(
- new FlushDecoderAction(FLUSH_CMD_FLUSH /* audio */,
- FLUSH_CMD_FLUSH /* video */));
-
- mDeferredActions.push_back(
- new SeekAction(seekTimeUs, (MediaPlayer2SeekMode)mode));
-
- // After a flush without shutdown, decoder is paused.
- // Don't resume it until source seek is done, otherwise it could
- // start pulling stale data too soon.
- mDeferredActions.push_back(
- new ResumeDecoderAction(needNotify));
-
- processDeferredActions();
- break;
- }
-
- case kWhatRewind:
- {
- ALOGV("kWhatRewind");
-
- int64_t seekTimeUs = mCurrentSourceInfo.mStartTimeUs;
- int32_t mode = MediaPlayer2SeekMode::SEEK_CLOSEST;
-
- if (!mStarted) {
- if (!mSourceStarted) {
- mSourceStarted = true;
- mCurrentSourceInfo.mSource->start();
- }
- performSeek(seekTimeUs, (MediaPlayer2SeekMode)mode);
- break;
- }
-
- // seeks can take a while, so we essentially paused
- notifyListener(mCurrentSourceInfo.mSrcId, MEDIA2_PAUSED, 0, 0);
-
- mDeferredActions.push_back(
- new FlushDecoderAction(FLUSH_CMD_FLUSH /* audio */,
- FLUSH_CMD_FLUSH /* video */));
-
- mDeferredActions.push_back(
- new SeekAction(seekTimeUs, (MediaPlayer2SeekMode)mode));
-
- // After a flush without shutdown, decoder is paused.
- // Don't resume it until source seek is done, otherwise it could
- // start pulling stale data too soon.
- mDeferredActions.push_back(
- new ResumeDecoderAction(false /* needNotify */));
-
- processDeferredActions();
- break;
- }
-
- case kWhatPause:
- {
- if (!mStarted) {
- onStart(false /* play */);
- }
- onPause();
- notifyListener(mCurrentSourceInfo.mSrcId, MEDIA2_PAUSED, 0, 0);
- mPausedByClient = true;
- break;
- }
-
- case kWhatSourceNotify:
- {
- onSourceNotify(msg);
- break;
- }
-
- case kWhatClosedCaptionNotify:
- {
- onClosedCaptionNotify(msg);
- break;
- }
-
- case kWhatPrepareDrm:
- {
- status_t status = onPrepareDrm(msg);
-
- sp<AMessage> response = new AMessage;
- response->setInt32("status", status);
- sp<AReplyToken> replyID;
- CHECK(msg->senderAwaitsResponse(&replyID));
- response->postReply(replyID);
- break;
- }
-
- case kWhatReleaseDrm:
- {
- status_t status = onReleaseDrm(msg);
-
- sp<AMessage> response = new AMessage;
- response->setInt32("status", status);
- sp<AReplyToken> replyID;
- CHECK(msg->senderAwaitsResponse(&replyID));
- response->postReply(replyID);
- break;
- }
-
- default:
- TRESPASS();
- break;
- }
-}
-
-void NuPlayer2::onResume() {
- if (!mPaused || mResetting) {
- ALOGD_IF(mResetting, "resetting, onResume discarded");
- return;
- }
- mPaused = false;
- if (mCurrentSourceInfo.mSource != NULL) {
- mCurrentSourceInfo.mSource->resume();
- } else {
- ALOGW("resume called when source is gone or not set");
- }
- // |mAudioDecoder| may have been released due to the pause timeout, so re-create it if
- // needed.
- if (audioDecoderStillNeeded() && mAudioDecoder == NULL) {
- instantiateDecoder(true /* audio */, &mAudioDecoder);
- }
- if (mRenderer != NULL) {
- mRenderer->resume();
- } else {
- ALOGW("resume called when renderer is gone or not set");
- }
-
- startPlaybackTimer("onresume");
-}
-
-void NuPlayer2::onStart(bool play) {
- ALOGV("onStart: mCrypto: %p", mCurrentSourceInfo.mCrypto.get());
-
- if (!mSourceStarted) {
- mSourceStarted = true;
- mCurrentSourceInfo.mSource->start();
- }
-
- mOffloadAudio = false;
- mAudioEOS = false;
- mVideoEOS = false;
- mStarted = true;
- mPaused = false;
-
- uint32_t flags = 0;
-
- if (mCurrentSourceInfo.mSource->isRealTime()) {
- flags |= Renderer::FLAG_REAL_TIME;
- }
-
- bool hasAudio = (mCurrentSourceInfo.mSource->getFormat(true /* audio */) != NULL);
- bool hasVideo = (mCurrentSourceInfo.mSource->getFormat(false /* audio */) != NULL);
- if (!hasAudio && !hasVideo) {
- ALOGE("no metadata for either audio or video source");
- mCurrentSourceInfo.mSource->stop();
- mSourceStarted = false;
- notifyListener(mCurrentSourceInfo.mSrcId, MEDIA2_ERROR,
- MEDIA2_ERROR_UNKNOWN, ERROR_MALFORMED);
- return;
- }
- ALOGV_IF(!hasAudio, "no metadata for audio source"); // video only stream
-
- sp<MetaData> audioMeta = mCurrentSourceInfo.mSource->getFormatMeta(true /* audio */);
-
- audio_stream_type_t streamType = AUDIO_STREAM_MUSIC;
- if (mAudioSink != NULL) {
- streamType = mAudioSink->getAudioStreamType();
- }
-
- mOffloadAudio =
- JMediaPlayer2Utils::isOffloadedAudioPlaybackSupported(
- audioMeta, hasVideo, mCurrentSourceInfo.mSource->isStreaming(), streamType)
- && (mPlaybackSettings.mSpeed == 1.f && mPlaybackSettings.mPitch == 1.f);
-
- // Modular DRM: Disabling audio offload if the source is protected
- if (mOffloadAudio && mCurrentSourceInfo.mIsDrmProtected) {
- mOffloadAudio = false;
- ALOGV("onStart: Disabling mOffloadAudio now that the source is protected.");
- }
-
- if (mOffloadAudio) {
- flags |= Renderer::FLAG_OFFLOAD_AUDIO;
- }
-
- sp<AMessage> notify = new AMessage(kWhatRendererNotify, this);
- ++mRendererGeneration;
- notify->setInt32("generation", mRendererGeneration);
- mRenderer = new Renderer(mAudioSink, mMediaClock, notify, mContext, flags);
- mRendererLooper = new ALooper;
- mRendererLooper->setName("NuPlayer2Renderer");
- mRendererLooper->start(false, true, ANDROID_PRIORITY_AUDIO);
- mRendererLooper->registerHandler(mRenderer);
-
- status_t err = mRenderer->setPlaybackSettings(mPlaybackSettings);
- if (err != OK) {
- mCurrentSourceInfo.mSource->stop();
- mSourceStarted = false;
- notifyListener(mCurrentSourceInfo.mSrcId, MEDIA2_ERROR, MEDIA2_ERROR_UNKNOWN, err);
- return;
- }
-
- float rate = getFrameRate();
- if (rate > 0) {
- mRenderer->setVideoFrameRate(rate);
- }
-
- addEndTimeMonitor();
- // Renderer is created in paused state.
- if (play) {
- mRenderer->resume();
- }
-
- if (mVideoDecoder != NULL) {
- mVideoDecoder->setRenderer(mRenderer);
- }
- if (mAudioDecoder != NULL) {
- mAudioDecoder->setRenderer(mRenderer);
- }
-
- startPlaybackTimer("onstart");
- notifyListener(mCurrentSourceInfo.mSrcId, MEDIA2_INFO, MEDIA2_INFO_DATA_SOURCE_START, 0);
-
- postScanSources();
-}
-
-void NuPlayer2::addEndTimeMonitor() {
- ++mEOSMonitorGeneration;
-
- if (mCurrentSourceInfo.mEndTimeUs == DataSourceDesc::kMaxTimeUs) {
- return;
- }
-
- sp<AMessage> msg = new AMessage(kWhatEOSMonitor, this);
- msg->setInt32("generation", mEOSMonitorGeneration);
- mMediaClock->addTimer(msg, mCurrentSourceInfo.mEndTimeUs);
-}
-
-void NuPlayer2::startPlaybackTimer(const char *where) {
- Mutex::Autolock autoLock(mPlayingTimeLock);
- if (mLastStartedPlayingTimeNs == 0) {
- mLastStartedPlayingTimeNs = systemTime();
- ALOGV("startPlaybackTimer() time %20" PRId64 " (%s)", mLastStartedPlayingTimeNs, where);
- }
-}
-
-void NuPlayer2::stopPlaybackTimer(const char *where) {
- Mutex::Autolock autoLock(mPlayingTimeLock);
-
- ALOGV("stopPlaybackTimer() time %20" PRId64 " (%s)", mLastStartedPlayingTimeNs, where);
-
- if (mLastStartedPlayingTimeNs != 0) {
- sp<NuPlayer2Driver> driver = mDriver.promote();
- if (driver != NULL) {
- int64_t now = systemTime();
- int64_t played = now - mLastStartedPlayingTimeNs;
- ALOGV("stopPlaybackTimer() log %20" PRId64 "", played);
-
- if (played > 0) {
- driver->notifyMorePlayingTimeUs(mCurrentSourceInfo.mSrcId, (played+500)/1000);
- }
- }
- mLastStartedPlayingTimeNs = 0;
- }
-}
-
-void NuPlayer2::startRebufferingTimer() {
- Mutex::Autolock autoLock(mPlayingTimeLock);
- if (mLastStartedRebufferingTimeNs == 0) {
- mLastStartedRebufferingTimeNs = systemTime();
- ALOGV("startRebufferingTimer() time %20" PRId64 "", mLastStartedRebufferingTimeNs);
- }
-}
-
-void NuPlayer2::stopRebufferingTimer(bool exitingPlayback) {
- Mutex::Autolock autoLock(mPlayingTimeLock);
-
- ALOGV("stopRebufferTimer() time %20" PRId64 " (exiting %d)",
- mLastStartedRebufferingTimeNs, exitingPlayback);
-
- if (mLastStartedRebufferingTimeNs != 0) {
- sp<NuPlayer2Driver> driver = mDriver.promote();
- if (driver != NULL) {
- int64_t now = systemTime();
- int64_t rebuffered = now - mLastStartedRebufferingTimeNs;
- ALOGV("stopRebufferingTimer() log %20" PRId64 "", rebuffered);
-
- if (rebuffered > 0) {
- driver->notifyMoreRebufferingTimeUs(
- mCurrentSourceInfo.mSrcId, (rebuffered+500)/1000);
- if (exitingPlayback) {
- driver->notifyRebufferingWhenExit(mCurrentSourceInfo.mSrcId, true);
- }
- }
- }
- mLastStartedRebufferingTimeNs = 0;
- }
-}
-
-void NuPlayer2::onPause() {
-
- stopPlaybackTimer("onPause");
-
- if (mPaused) {
- return;
- }
- mPaused = true;
- if (mCurrentSourceInfo.mSource != NULL) {
- mCurrentSourceInfo.mSource->pause();
- } else {
- ALOGW("pause called when source is gone or not set");
- }
- if (mRenderer != NULL) {
- mRenderer->pause();
- } else {
- ALOGW("pause called when renderer is gone or not set");
- }
-
-}
-
-bool NuPlayer2::audioDecoderStillNeeded() {
- // Audio decoder is no longer needed if it's in shut/shutting down status.
- return ((mFlushingAudio != SHUT_DOWN) && (mFlushingAudio != SHUTTING_DOWN_DECODER));
-}
-
-void NuPlayer2::handleFlushComplete(bool audio, bool isDecoder) {
- // We wait for both the decoder flush and the renderer flush to complete
- // before entering either the FLUSHED or the SHUTTING_DOWN_DECODER state.
-
- mFlushComplete[audio][isDecoder] = true;
- if (!mFlushComplete[audio][!isDecoder]) {
- return;
- }
-
- FlushStatus *state = audio ? &mFlushingAudio : &mFlushingVideo;
- switch (*state) {
- case FLUSHING_DECODER:
- {
- *state = FLUSHED;
- break;
- }
-
- case FLUSHING_DECODER_SHUTDOWN:
- {
- *state = SHUTTING_DOWN_DECODER;
-
- ALOGV("initiating %s decoder shutdown", audio ? "audio" : "video");
- getDecoder(audio)->initiateShutdown();
- break;
- }
-
- default:
- // decoder flush completes only occur in a flushing state.
- LOG_ALWAYS_FATAL_IF(isDecoder, "decoder flush in invalid state %d", *state);
- break;
- }
-}
-
-void NuPlayer2::finishFlushIfPossible() {
- if (mFlushingAudio != NONE && mFlushingAudio != FLUSHED
- && mFlushingAudio != SHUT_DOWN) {
- return;
- }
-
- if (mFlushingVideo != NONE && mFlushingVideo != FLUSHED
- && mFlushingVideo != SHUT_DOWN) {
- return;
- }
-
- ALOGV("both audio and video are flushed now.");
-
- mFlushingAudio = NONE;
- mFlushingVideo = NONE;
-
- clearFlushComplete();
-
- processDeferredActions();
-}
-
-void NuPlayer2::postScanSources() {
- if (mScanSourcesPending) {
- return;
- }
-
- sp<AMessage> msg = new AMessage(kWhatScanSources, this);
- msg->setInt32("generation", mScanSourcesGeneration);
- msg->post();
-
- mScanSourcesPending = true;
-}
-
-void NuPlayer2::tryOpenAudioSinkForOffload(
- const sp<AMessage> &format, const sp<MetaData> &audioMeta, bool hasVideo) {
- // Note: This is called early in NuPlayer2 to determine whether offloading
- // is possible; otherwise the decoders call the renderer openAudioSink directly.
-
- status_t err = mRenderer->openAudioSink(
- format, true /* offloadOnly */, hasVideo,
- AUDIO_OUTPUT_FLAG_NONE, &mOffloadAudio, mCurrentSourceInfo.mSource->isStreaming());
- if (err != OK) {
- // Any failure we turn off mOffloadAudio.
- mOffloadAudio = false;
- } else if (mOffloadAudio) {
- sendMetaDataToHal(mAudioSink, audioMeta);
- }
-}
-
-void NuPlayer2::closeAudioSink() {
- mRenderer->closeAudioSink();
-}
-
-void NuPlayer2::restartAudio(
- int64_t currentPositionUs, bool forceNonOffload, bool needsToCreateAudioDecoder) {
- if (mAudioDecoder != NULL) {
- mAudioDecoder->pause();
- Mutex::Autolock autoLock(mDecoderLock);
- mAudioDecoder.clear();
- mAudioDecoderError = false;
- ++mAudioDecoderGeneration;
- }
- if (mFlushingAudio == FLUSHING_DECODER) {
- mFlushComplete[1 /* audio */][1 /* isDecoder */] = true;
- mFlushingAudio = FLUSHED;
- finishFlushIfPossible();
- } else if (mFlushingAudio == FLUSHING_DECODER_SHUTDOWN
- || mFlushingAudio == SHUTTING_DOWN_DECODER) {
- mFlushComplete[1 /* audio */][1 /* isDecoder */] = true;
- mFlushingAudio = SHUT_DOWN;
- finishFlushIfPossible();
- needsToCreateAudioDecoder = false;
- }
- if (mRenderer == NULL) {
- return;
- }
- closeAudioSink();
- mRenderer->flush(true /* audio */, false /* notifyComplete */);
- if (mVideoDecoder != NULL) {
- mDeferredActions.push_back(
- new FlushDecoderAction(FLUSH_CMD_NONE /* audio */,
- FLUSH_CMD_FLUSH /* video */));
- mDeferredActions.push_back(
- new SeekAction(currentPositionUs,
- MediaPlayerSeekMode::SEEK_PREVIOUS_SYNC /* mode */));
- // After a flush without shutdown, decoder is paused.
- // Don't resume it until source seek is done, otherwise it could
- // start pulling stale data too soon.
- mDeferredActions.push_back(new ResumeDecoderAction(false));
- processDeferredActions();
- } else {
- performSeek(currentPositionUs, MediaPlayerSeekMode::SEEK_PREVIOUS_SYNC /* mode */);
- }
-
- if (forceNonOffload) {
- mRenderer->signalDisableOffloadAudio();
- mOffloadAudio = false;
- }
- if (needsToCreateAudioDecoder) {
- instantiateDecoder(true /* audio */, &mAudioDecoder, !forceNonOffload);
- }
-}
-
-void NuPlayer2::determineAudioModeChange(const sp<AMessage> &audioFormat) {
- if (mCurrentSourceInfo.mSource == NULL || mAudioSink == NULL) {
- return;
- }
-
- if (mRenderer == NULL) {
- ALOGW("No renderer can be used to determine audio mode. Use non-offload for safety.");
- mOffloadAudio = false;
- return;
- }
-
- sp<MetaData> audioMeta = mCurrentSourceInfo.mSource->getFormatMeta(true /* audio */);
- sp<AMessage> videoFormat = mCurrentSourceInfo.mSource->getFormat(false /* audio */);
- audio_stream_type_t streamType = mAudioSink->getAudioStreamType();
- const bool hasVideo = (videoFormat != NULL);
- bool canOffload = JMediaPlayer2Utils::isOffloadedAudioPlaybackSupported(
- audioMeta, hasVideo, mCurrentSourceInfo.mSource->isStreaming(), streamType)
- && (mPlaybackSettings.mSpeed == 1.f && mPlaybackSettings.mPitch == 1.f);
-
- // Modular DRM: Disabling audio offload if the source is protected
- if (canOffload && mCurrentSourceInfo.mIsDrmProtected) {
- canOffload = false;
- ALOGV("determineAudioModeChange: Disabling mOffloadAudio b/c the source is protected.");
- }
-
- if (canOffload) {
- if (!mOffloadAudio) {
- mRenderer->signalEnableOffloadAudio();
- }
- // open audio sink early under offload mode.
- tryOpenAudioSinkForOffload(audioFormat, audioMeta, hasVideo);
- } else {
- if (mOffloadAudio) {
- mRenderer->signalDisableOffloadAudio();
- mOffloadAudio = false;
- }
- }
-}
-
-status_t NuPlayer2::instantiateDecoder(
- bool audio, sp<DecoderBase> *decoder, bool checkAudioModeChange) {
- // The audio decoder could be cleared by tear down. If still in shut down
- // process, no need to create a new audio decoder.
- if (*decoder != NULL || (audio && mFlushingAudio == SHUT_DOWN)) {
- return OK;
- }
-
- sp<AMessage> format = mCurrentSourceInfo.mSource->getFormat(audio);
-
- if (format == NULL) {
- return UNKNOWN_ERROR;
- } else {
- status_t err;
- if (format->findInt32("err", &err) && err) {
- return err;
- }
- }
-
- format->setInt32("priority", 0 /* realtime */);
-
- if (!audio) {
- AString mime;
- CHECK(format->findString("mime", &mime));
-
- sp<AMessage> ccNotify = new AMessage(kWhatClosedCaptionNotify, this);
- if (mCCDecoder == NULL) {
- mCCDecoder = new CCDecoder(ccNotify);
- }
-
- if (mCurrentSourceInfo.mSourceFlags & Source::FLAG_SECURE) {
- format->setInt32("secure", true);
- }
-
- if (mCurrentSourceInfo.mSourceFlags & Source::FLAG_PROTECTED) {
- format->setInt32("protected", true);
- }
-
- float rate = getFrameRate();
- if (rate > 0) {
- format->setFloat("operating-rate", rate * mPlaybackSettings.mSpeed);
- }
- }
-
- Mutex::Autolock autoLock(mDecoderLock);
-
- if (audio) {
- sp<AMessage> notify = new AMessage(kWhatAudioNotify, this);
- ++mAudioDecoderGeneration;
- notify->setInt32("generation", mAudioDecoderGeneration);
-
- if (checkAudioModeChange) {
- determineAudioModeChange(format);
- }
- if (mOffloadAudio) {
- mCurrentSourceInfo.mSource->setOffloadAudio(true /* offload */);
-
- const bool hasVideo = (mCurrentSourceInfo.mSource->getFormat(false /*audio */) != NULL);
- format->setInt32("has-video", hasVideo);
- *decoder = new DecoderPassThrough(notify, mCurrentSourceInfo.mSource, mRenderer);
- ALOGV("instantiateDecoder audio DecoderPassThrough hasVideo: %d", hasVideo);
- } else {
- mCurrentSourceInfo.mSource->setOffloadAudio(false /* offload */);
-
- *decoder = new Decoder(notify, mCurrentSourceInfo.mSource, mPID, mUID, mRenderer);
- ALOGV("instantiateDecoder audio Decoder");
- }
- mAudioDecoderError = false;
- } else {
- sp<AMessage> notify = new AMessage(kWhatVideoNotify, this);
- ++mVideoDecoderGeneration;
- notify->setInt32("generation", mVideoDecoderGeneration);
-
- *decoder = new Decoder(
- notify, mCurrentSourceInfo.mSource, mPID, mUID, mRenderer, mNativeWindow,
- mCCDecoder);
- mVideoDecoderError = false;
-
- // enable FRC if high-quality AV sync is requested, even if not
- // directly queuing to display, as this will even improve textureview
- // playback.
- {
- if (property_get_bool("persist.sys.media.avsync", false)) {
- format->setInt32("auto-frc", 1);
- }
- }
- }
- (*decoder)->init();
-
- // Modular DRM
- if (mCurrentSourceInfo.mIsDrmProtected) {
- format->setObject("crypto", mCurrentSourceInfo.mCrypto);
- ALOGV("instantiateDecoder: mCrypto: %p isSecure: %d",
- mCurrentSourceInfo.mCrypto.get(),
- (mCurrentSourceInfo.mSourceFlags & Source::FLAG_SECURE) != 0);
- }
-
- (*decoder)->configure(format);
-
- if (!audio) {
- sp<AMessage> params = new AMessage();
- float rate = getFrameRate();
- if (rate > 0) {
- params->setFloat("frame-rate-total", rate);
- }
-
- sp<MetaData> fileMeta = getFileMeta();
- if (fileMeta != NULL) {
- int32_t videoTemporalLayerCount;
- if (fileMeta->findInt32(kKeyTemporalLayerCount, &videoTemporalLayerCount)
- && videoTemporalLayerCount > 0) {
- params->setInt32("temporal-layer-count", videoTemporalLayerCount);
- }
- }
-
- if (params->countEntries() > 0) {
- (*decoder)->setParameters(params);
- }
- }
- return OK;
-}
-
-void NuPlayer2::updateVideoSize(
- int64_t srcId,
- const sp<AMessage> &inputFormat,
- const sp<AMessage> &outputFormat) {
- if (inputFormat == NULL) {
- ALOGW("Unknown video size, reporting 0x0!");
- notifyListener(srcId, MEDIA2_SET_VIDEO_SIZE, 0, 0);
- return;
- }
- int32_t err = OK;
- inputFormat->findInt32("err", &err);
- if (err == -EWOULDBLOCK) {
- ALOGW("Video meta is not available yet!");
- return;
- }
- if (err != OK) {
- ALOGW("Something is wrong with video meta!");
- return;
- }
-
- int32_t displayWidth, displayHeight;
- if (outputFormat != NULL) {
- int32_t width, height;
- CHECK(outputFormat->findInt32("width", &width));
- CHECK(outputFormat->findInt32("height", &height));
-
- int32_t cropLeft, cropTop, cropRight, cropBottom;
- CHECK(outputFormat->findRect(
- "crop",
- &cropLeft, &cropTop, &cropRight, &cropBottom));
-
- displayWidth = cropRight - cropLeft + 1;
- displayHeight = cropBottom - cropTop + 1;
-
- ALOGV("Video output format changed to %d x %d "
- "(crop: %d x %d @ (%d, %d))",
- width, height,
- displayWidth,
- displayHeight,
- cropLeft, cropTop);
- } else {
- CHECK(inputFormat->findInt32("width", &displayWidth));
- CHECK(inputFormat->findInt32("height", &displayHeight));
-
- ALOGV("Video input format %d x %d", displayWidth, displayHeight);
- }
-
- // Take into account sample aspect ratio if necessary:
- int32_t sarWidth, sarHeight;
- if (inputFormat->findInt32("sar-width", &sarWidth)
- && inputFormat->findInt32("sar-height", &sarHeight)
- && sarWidth > 0 && sarHeight > 0) {
- ALOGV("Sample aspect ratio %d : %d", sarWidth, sarHeight);
-
- displayWidth = (displayWidth * sarWidth) / sarHeight;
-
- ALOGV("display dimensions %d x %d", displayWidth, displayHeight);
- } else {
- int32_t width, height;
- if (inputFormat->findInt32("display-width", &width)
- && inputFormat->findInt32("display-height", &height)
- && width > 0 && height > 0
- && displayWidth > 0 && displayHeight > 0) {
- if (displayHeight * (int64_t)width / height > (int64_t)displayWidth) {
- displayHeight = (int32_t)(displayWidth * (int64_t)height / width);
- } else {
- displayWidth = (int32_t)(displayHeight * (int64_t)width / height);
- }
- ALOGV("Video display width and height are overridden to %d x %d",
- displayWidth, displayHeight);
- }
- }
-
- int32_t rotationDegrees;
- if (!inputFormat->findInt32("rotation-degrees", &rotationDegrees)) {
- rotationDegrees = 0;
- }
-
- if (rotationDegrees == 90 || rotationDegrees == 270) {
- int32_t tmp = displayWidth;
- displayWidth = displayHeight;
- displayHeight = tmp;
- }
-
- notifyListener(
- srcId,
- MEDIA2_SET_VIDEO_SIZE,
- displayWidth,
- displayHeight);
-}
-
-void NuPlayer2::notifyListener(
- int64_t srcId, int msg, int ext1, int ext2, const PlayerMessage *in) {
- if (mDriver == NULL) {
- return;
- }
-
- sp<NuPlayer2Driver> driver = mDriver.promote();
-
- if (driver == NULL) {
- return;
- }
-
- driver->notifyListener(srcId, msg, ext1, ext2, in);
-}
-
-void NuPlayer2::flushDecoder(bool audio, bool needShutdown) {
- ALOGV("[%s] flushDecoder needShutdown=%d",
- audio ? "audio" : "video", needShutdown);
-
- const sp<DecoderBase> &decoder = getDecoder(audio);
- if (decoder == NULL) {
- ALOGI("flushDecoder %s without decoder present",
- audio ? "audio" : "video");
- return;
- }
-
- // Make sure we don't continue to scan sources until we finish flushing.
- ++mScanSourcesGeneration;
- if (mScanSourcesPending) {
- if (!needShutdown) {
- mDeferredActions.push_back(
- new SimpleAction(&NuPlayer2::performScanSources));
- }
- mScanSourcesPending = false;
- }
-
- decoder->signalFlush();
-
- FlushStatus newStatus =
- needShutdown ? FLUSHING_DECODER_SHUTDOWN : FLUSHING_DECODER;
-
- mFlushComplete[audio][false /* isDecoder */] = (mRenderer == NULL);
- mFlushComplete[audio][true /* isDecoder */] = false;
- if (audio) {
- ALOGE_IF(mFlushingAudio != NONE,
- "audio flushDecoder() is called in state %d", mFlushingAudio);
- mFlushingAudio = newStatus;
- } else {
- ALOGE_IF(mFlushingVideo != NONE,
- "video flushDecoder() is called in state %d", mFlushingVideo);
- mFlushingVideo = newStatus;
- }
-}
-
-void NuPlayer2::queueDecoderShutdown(
- bool audio, bool video, const sp<AMessage> &reply) {
- ALOGI("queueDecoderShutdown audio=%d, video=%d", audio, video);
-
- mDeferredActions.push_back(
- new FlushDecoderAction(
- audio ? FLUSH_CMD_SHUTDOWN : FLUSH_CMD_NONE,
- video ? FLUSH_CMD_SHUTDOWN : FLUSH_CMD_NONE));
-
- mDeferredActions.push_back(
- new SimpleAction(&NuPlayer2::performScanSources));
-
- mDeferredActions.push_back(new PostMessageAction(reply));
-
- processDeferredActions();
-}
-
-status_t NuPlayer2::setVideoScalingMode(int32_t mode) {
- mVideoScalingMode = mode;
- if (mNativeWindow != NULL && mNativeWindow->getANativeWindow() != NULL) {
- status_t ret = native_window_set_scaling_mode(
- mNativeWindow->getANativeWindow(), mVideoScalingMode);
- if (ret != OK) {
- ALOGE("Failed to set scaling mode (%d): %s",
- -ret, strerror(-ret));
- return ret;
- }
- }
- return OK;
-}
-
-status_t NuPlayer2::getTrackInfo(int64_t srcId, PlayerMessage* reply) const {
- sp<AMessage> msg = new AMessage(kWhatGetTrackInfo, this);
- msg->setInt64("srcId", srcId);
- msg->setPointer("reply", reply);
-
- sp<AMessage> response;
- status_t err = msg->postAndAwaitResponse(&response);
- return err;
-}
-
-status_t NuPlayer2::getSelectedTrack(int64_t srcId, int32_t type, PlayerMessage* reply) const {
- sp<AMessage> msg = new AMessage(kWhatGetSelectedTrack, this);
- msg->setPointer("reply", reply);
- msg->setInt64("srcId", srcId);
- msg->setInt32("type", type);
-
- sp<AMessage> response;
- status_t err = msg->postAndAwaitResponse(&response);
- if (err == OK && response != NULL) {
- CHECK(response->findInt32("err", &err));
- }
- return err;
-}
-
-status_t NuPlayer2::selectTrack(int64_t srcId, size_t trackIndex, bool select, int64_t timeUs) {
- sp<AMessage> msg = new AMessage(kWhatSelectTrack, this);
- msg->setInt64("srcId", srcId);
- msg->setSize("trackIndex", trackIndex);
- msg->setInt32("select", select);
- msg->setInt64("timeUs", timeUs);
-
- sp<AMessage> response;
- status_t err = msg->postAndAwaitResponse(&response);
-
- if (err != OK) {
- return err;
- }
-
- if (!response->findInt32("err", &err)) {
- err = OK;
- }
-
- return err;
-}
-
-status_t NuPlayer2::getCurrentPosition(int64_t *mediaUs) {
- sp<Renderer> renderer = mRenderer;
- if (renderer == NULL) {
- return NO_INIT;
- }
-
- return renderer->getCurrentPosition(mediaUs);
-}
-
-void NuPlayer2::getStats(Vector<sp<AMessage> > *mTrackStats) {
- CHECK(mTrackStats != NULL);
-
- mTrackStats->clear();
-
- Mutex::Autolock autoLock(mDecoderLock);
- if (mVideoDecoder != NULL) {
- mTrackStats->push_back(mVideoDecoder->getStats());
- }
- if (mAudioDecoder != NULL) {
- mTrackStats->push_back(mAudioDecoder->getStats());
- }
-}
-
-sp<MetaData> NuPlayer2::getFileMeta() {
- return mCurrentSourceInfo.mSource->getFileFormatMeta();
-}
-
-float NuPlayer2::getFrameRate() {
- sp<MetaData> meta = mCurrentSourceInfo.mSource->getFormatMeta(false /* audio */);
- if (meta == NULL) {
- return 0;
- }
- int32_t rate;
- if (!meta->findInt32(kKeyFrameRate, &rate)) {
- // fall back to try file meta
- sp<MetaData> fileMeta = getFileMeta();
- if (fileMeta == NULL) {
- ALOGW("source has video meta but not file meta");
- return -1;
- }
- int32_t fileMetaRate;
- if (!fileMeta->findInt32(kKeyFrameRate, &fileMetaRate)) {
- return -1;
- }
- return fileMetaRate;
- }
- return rate;
-}
-
-void NuPlayer2::schedulePollDuration() {
- sp<AMessage> msg = new AMessage(kWhatPollDuration, this);
- msg->setInt32("generation", mPollDurationGeneration);
- msg->post();
-}
-
-void NuPlayer2::cancelPollDuration() {
- ++mPollDurationGeneration;
-}
-
-void NuPlayer2::processDeferredActions() {
- while (!mDeferredActions.empty()) {
- // We won't execute any deferred actions until we're no longer in
- // an intermediate state, i.e. one more more decoders are currently
- // flushing or shutting down.
-
- if (mFlushingAudio != NONE || mFlushingVideo != NONE) {
- // We're currently flushing, postpone the reset until that's
- // completed.
-
- ALOGV("postponing action mFlushingAudio=%d, mFlushingVideo=%d",
- mFlushingAudio, mFlushingVideo);
-
- break;
- }
-
- sp<Action> action = *mDeferredActions.begin();
- mDeferredActions.erase(mDeferredActions.begin());
-
- action->execute(this);
- }
-}
-
-void NuPlayer2::performSeek(int64_t seekTimeUs, MediaPlayer2SeekMode mode) {
- ALOGV("performSeek seekTimeUs=%lld us (%.2f secs), mode=%d",
- (long long)seekTimeUs, seekTimeUs / 1E6, mode);
-
- if (mCurrentSourceInfo.mSource == NULL) {
- // This happens when reset occurs right before the loop mode
- // asynchronously seeks to the start of the stream.
- LOG_ALWAYS_FATAL_IF(mAudioDecoder != NULL || mVideoDecoder != NULL,
- "mCurrentSourceInfo.mSource is NULL and decoders not NULL audio(%p) video(%p)",
- mAudioDecoder.get(), mVideoDecoder.get());
- return;
- }
- mPreviousSeekTimeUs = seekTimeUs;
- mCurrentSourceInfo.mSource->seekTo(seekTimeUs, mode);
- ++mTimedTextGeneration;
-
- // everything's flushed, continue playback.
-}
-
-void NuPlayer2::performDecoderFlush(FlushCommand audio, FlushCommand video) {
- ALOGV("performDecoderFlush audio=%d, video=%d", audio, video);
-
- if ((audio == FLUSH_CMD_NONE || mAudioDecoder == NULL)
- && (video == FLUSH_CMD_NONE || mVideoDecoder == NULL)) {
- return;
- }
-
- if (audio != FLUSH_CMD_NONE && mAudioDecoder != NULL) {
- flushDecoder(true /* audio */, (audio == FLUSH_CMD_SHUTDOWN));
- }
-
- if (video != FLUSH_CMD_NONE && mVideoDecoder != NULL) {
- flushDecoder(false /* audio */, (video == FLUSH_CMD_SHUTDOWN));
- }
-}
-
-void NuPlayer2::performReset() {
- ALOGV("performReset");
-
- CHECK(mAudioDecoder == NULL);
- CHECK(mVideoDecoder == NULL);
-
- stopPlaybackTimer("performReset");
- stopRebufferingTimer(true);
-
- cancelPollDuration();
-
- ++mScanSourcesGeneration;
- mScanSourcesPending = false;
-
- if (mRendererLooper != NULL) {
- if (mRenderer != NULL) {
- mRendererLooper->unregisterHandler(mRenderer->id());
- }
- mRendererLooper->stop();
- mRendererLooper.clear();
- }
- mRenderer.clear();
- ++mRendererGeneration;
-
- resetSourceInfo(mCurrentSourceInfo);
- resetSourceInfo(mNextSourceInfo);
-
- if (mDriver != NULL) {
- sp<NuPlayer2Driver> driver = mDriver.promote();
- if (driver != NULL) {
- driver->notifyResetComplete(mCurrentSourceInfo.mSrcId);
- }
- }
-
- mStarted = false;
- mPrepared = false;
- mResetting = false;
- mSourceStarted = false;
-
-}
-
-void NuPlayer2::performPlayNextDataSource() {
- ALOGV("performPlayNextDataSource");
-
- CHECK(mAudioDecoder == NULL);
- CHECK(mVideoDecoder == NULL);
-
- stopPlaybackTimer("performPlayNextDataSource");
- stopRebufferingTimer(true);
-
- cancelPollDuration();
-
- ++mScanSourcesGeneration;
- mScanSourcesPending = false;
-
- ++mRendererGeneration;
-
- if (mCurrentSourceInfo.mSource != NULL) {
- mCurrentSourceInfo.mSource->stop();
- }
-
- long previousSrcId;
- {
- Mutex::Autolock autoLock(mSourceLock);
- previousSrcId = mCurrentSourceInfo.mSrcId;
-
- mCurrentSourceInfo = mNextSourceInfo;
- mNextSourceInfo = SourceInfo();
- mNextSourceInfo.mSrcId = ~mCurrentSourceInfo.mSrcId; // to distinguish the two sources.
- }
-
- if (mDriver != NULL) {
- sp<NuPlayer2Driver> driver = mDriver.promote();
- if (driver != NULL) {
- notifyListener(previousSrcId, MEDIA2_INFO, MEDIA2_INFO_DATA_SOURCE_END, 0);
-
- int64_t durationUs;
- if (mCurrentSourceInfo.mSource->getDuration(&durationUs) == OK) {
- driver->notifyDuration(mCurrentSourceInfo.mSrcId, durationUs);
- }
- notifyListener(
- mCurrentSourceInfo.mSrcId, MEDIA2_INFO, MEDIA2_INFO_DATA_SOURCE_START, 0);
- }
- }
-
- mStarted = false;
- mPrepared = true; // TODO: what if it's not prepared
- mResetting = false;
- mSourceStarted = false;
-
- addEndTimeMonitor();
-
- if (mRenderer != NULL) {
- mRenderer->resume();
- }
-
- onStart(true /* play */);
- mPausedByClient = false;
- notifyListener(mCurrentSourceInfo.mSrcId, MEDIA2_STARTED, 0, 0);
-}
-
-void NuPlayer2::performScanSources() {
- ALOGV("performScanSources");
-
- if (!mStarted) {
- return;
- }
-
- if (mAudioDecoder == NULL || mVideoDecoder == NULL) {
- postScanSources();
- }
-}
-
-void NuPlayer2::performSetSurface(const sp<ANativeWindowWrapper> &nww) {
- ALOGV("performSetSurface");
-
- mNativeWindow = nww;
-
- // XXX - ignore error from setVideoScalingMode for now
- setVideoScalingMode(mVideoScalingMode);
-
- if (mDriver != NULL) {
- sp<NuPlayer2Driver> driver = mDriver.promote();
- if (driver != NULL) {
- driver->notifySetSurfaceComplete(mCurrentSourceInfo.mSrcId);
- }
- }
-}
-
-void NuPlayer2::performResumeDecoders(bool needNotify) {
- if (needNotify) {
- mResumePending = true;
- if (mVideoDecoder == NULL) {
- // if audio-only, we can notify seek complete now,
- // as the resume operation will be relatively fast.
- finishResume();
- }
- }
-
- if (mVideoDecoder != NULL) {
- // When there is continuous seek, MediaPlayer will cache the seek
- // position, and send down new seek request when previous seek is
- // complete. Let's wait for at least one video output frame before
- // notifying seek complete, so that the video thumbnail gets updated
- // when seekbar is dragged.
- mVideoDecoder->signalResume(needNotify);
- }
-
- if (mAudioDecoder != NULL) {
- mAudioDecoder->signalResume(false /* needNotify */);
- }
-}
-
-void NuPlayer2::finishResume() {
- if (mResumePending) {
- mResumePending = false;
- notifyDriverSeekComplete(mCurrentSourceInfo.mSrcId);
- }
-}
-
-void NuPlayer2::notifyDriverSeekComplete(int64_t srcId) {
- if (mDriver != NULL) {
- sp<NuPlayer2Driver> driver = mDriver.promote();
- if (driver != NULL) {
- driver->notifySeekComplete(srcId);
- }
- }
-}
-
-void NuPlayer2::onSourceNotify(const sp<AMessage> &msg) {
- int32_t what;
- CHECK(msg->findInt32("what", &what));
-
- int64_t srcId;
- CHECK(msg->findInt64("srcId", &srcId));
- switch (what) {
- case Source::kWhatPrepared:
- {
- ALOGV("NuPlayer2::onSourceNotify Source::kWhatPrepared source:%p, Id(%lld)",
- mCurrentSourceInfo.mSource.get(), (long long)srcId);
- if (srcId == mCurrentSourceInfo.mSrcId) {
- if (mCurrentSourceInfo.mSource == NULL) {
- // This is a stale notification from a source that was
- // asynchronously preparing when the client called reset().
- // We handled the reset, the source is gone.
- break;
- }
-
- int32_t err;
- CHECK(msg->findInt32("err", &err));
-
- if (err != OK) {
- // shut down potential secure codecs in case client never calls reset
- mDeferredActions.push_back(
- new FlushDecoderAction(FLUSH_CMD_SHUTDOWN /* audio */,
- FLUSH_CMD_SHUTDOWN /* video */));
- processDeferredActions();
- } else {
- mPrepared = true;
- }
-
- sp<NuPlayer2Driver> driver = mDriver.promote();
- if (driver != NULL) {
- // notify duration first, so that it's definitely set when
- // the app received the "prepare complete" callback.
- int64_t durationUs;
- if (mCurrentSourceInfo.mSource->getDuration(&durationUs) == OK) {
- driver->notifyDuration(srcId, durationUs);
- }
- driver->notifyPrepareCompleted(srcId, err);
- }
- } else if (srcId == mNextSourceInfo.mSrcId) {
- if (mNextSourceInfo.mSource == NULL) {
- break; // stale
- }
-
- sp<NuPlayer2Driver> driver = mDriver.promote();
- if (driver != NULL) {
- int32_t err;
- CHECK(msg->findInt32("err", &err));
- driver->notifyPrepareCompleted(srcId, err);
- }
- }
-
- break;
- }
-
- // Modular DRM
- case Source::kWhatDrmInfo:
- {
- PlayerMessage playerMsg;
- sp<ABuffer> drmInfo;
- CHECK(msg->findBuffer("drmInfo", &drmInfo));
- playerMsg.ParseFromArray(drmInfo->data(), drmInfo->size());
-
- ALOGV("onSourceNotify() kWhatDrmInfo MEDIA2_DRM_INFO drmInfo: %p playerMsg size: %d",
- drmInfo.get(), playerMsg.ByteSize());
-
- notifyListener(srcId, MEDIA2_DRM_INFO, 0 /* ext1 */, 0 /* ext2 */, &playerMsg);
-
- break;
- }
-
- case Source::kWhatFlagsChanged:
- {
- uint32_t flags;
- CHECK(msg->findInt32("flags", (int32_t *)&flags));
-
- sp<NuPlayer2Driver> driver = mDriver.promote();
- if (driver != NULL) {
-
- ALOGV("onSourceNotify() kWhatFlagsChanged FLAG_CAN_PAUSE: %d "
- "FLAG_CAN_SEEK_BACKWARD: %d \n\t\t\t\t FLAG_CAN_SEEK_FORWARD: %d "
- "FLAG_CAN_SEEK: %d FLAG_DYNAMIC_DURATION: %d \n"
- "\t\t\t\t FLAG_SECURE: %d FLAG_PROTECTED: %d",
- (flags & Source::FLAG_CAN_PAUSE) != 0,
- (flags & Source::FLAG_CAN_SEEK_BACKWARD) != 0,
- (flags & Source::FLAG_CAN_SEEK_FORWARD) != 0,
- (flags & Source::FLAG_CAN_SEEK) != 0,
- (flags & Source::FLAG_DYNAMIC_DURATION) != 0,
- (flags & Source::FLAG_SECURE) != 0,
- (flags & Source::FLAG_PROTECTED) != 0);
-
- if ((flags & NuPlayer2::Source::FLAG_CAN_SEEK) == 0) {
- driver->notifyListener(
- srcId, MEDIA2_INFO, MEDIA2_INFO_NOT_SEEKABLE, 0);
- }
- if (srcId == mCurrentSourceInfo.mSrcId) {
- driver->notifyFlagsChanged(srcId, flags);
- }
- }
-
- if (srcId == mCurrentSourceInfo.mSrcId) {
- if ((mCurrentSourceInfo.mSourceFlags & Source::FLAG_DYNAMIC_DURATION)
- && (!(flags & Source::FLAG_DYNAMIC_DURATION))) {
- cancelPollDuration();
- } else if (!(mCurrentSourceInfo.mSourceFlags & Source::FLAG_DYNAMIC_DURATION)
- && (flags & Source::FLAG_DYNAMIC_DURATION)
- && (mAudioDecoder != NULL || mVideoDecoder != NULL)) {
- schedulePollDuration();
- }
-
- mCurrentSourceInfo.mSourceFlags = flags;
- } else if (srcId == mNextSourceInfo.mSrcId) {
- // TODO: handle duration polling for next source.
- mNextSourceInfo.mSourceFlags = flags;
- }
- break;
- }
-
- case Source::kWhatVideoSizeChanged:
- {
- sp<AMessage> format;
- CHECK(msg->findMessage("format", &format));
-
- updateVideoSize(srcId, format);
- break;
- }
-
- case Source::kWhatBufferingUpdate:
- {
- int32_t percentage;
- CHECK(msg->findInt32("percentage", &percentage));
-
- notifyListener(srcId, MEDIA2_BUFFERING_UPDATE, percentage, 0);
- break;
- }
-
- case Source::kWhatPauseOnBufferingStart:
- {
- // ignore if not playing
- if (mStarted) {
- ALOGI("buffer low, pausing...");
-
- startRebufferingTimer();
- mPausedForBuffering = true;
- onPause();
- }
- notifyListener(srcId, MEDIA2_INFO, MEDIA2_INFO_BUFFERING_START, 0);
- break;
- }
-
- case Source::kWhatResumeOnBufferingEnd:
- {
- // ignore if not playing
- if (mStarted) {
- ALOGI("buffer ready, resuming...");
-
- stopRebufferingTimer(false);
- mPausedForBuffering = false;
-
- // do not resume yet if client didn't unpause
- if (!mPausedByClient) {
- onResume();
- }
- }
- notifyListener(srcId, MEDIA2_INFO, MEDIA2_INFO_BUFFERING_END, 0);
- break;
- }
-
- case Source::kWhatCacheStats:
- {
- int32_t kbps;
- CHECK(msg->findInt32("bandwidth", &kbps));
-
- notifyListener(srcId, MEDIA2_INFO, MEDIA2_INFO_NETWORK_BANDWIDTH, kbps);
- break;
- }
-
- case Source::kWhatSubtitleData:
- {
- sp<ABuffer> buffer;
- CHECK(msg->findBuffer("buffer", &buffer));
-
- sendSubtitleData(buffer, 0 /* baseIndex */);
- break;
- }
-
- case Source::kWhatTimedMetaData:
- {
- sp<ABuffer> buffer;
- if (!msg->findBuffer("buffer", &buffer)) {
- notifyListener(srcId, MEDIA2_INFO, MEDIA2_INFO_METADATA_UPDATE, 0);
- } else {
- sendTimedMetaData(buffer);
- }
- break;
- }
-
- case Source::kWhatTimedTextData:
- {
- int32_t generation;
- if (msg->findInt32("generation", &generation)
- && generation != mTimedTextGeneration) {
- break;
- }
-
- sp<ABuffer> buffer;
- CHECK(msg->findBuffer("buffer", &buffer));
-
- sp<NuPlayer2Driver> driver = mDriver.promote();
- if (driver == NULL) {
- break;
- }
-
- int64_t posMs;
- int64_t timeUs, posUs;
- driver->getCurrentPosition(&posMs);
- posUs = posMs * 1000LL;
- CHECK(buffer->meta()->findInt64("timeUs", &timeUs));
-
- if (posUs < timeUs) {
- if (!msg->findInt32("generation", &generation)) {
- msg->setInt32("generation", mTimedTextGeneration);
- }
- msg->post(timeUs - posUs);
- } else {
- sendTimedTextData(buffer);
- }
- break;
- }
-
- case Source::kWhatQueueDecoderShutdown:
- {
- int32_t audio, video;
- CHECK(msg->findInt32("audio", &audio));
- CHECK(msg->findInt32("video", &video));
-
- sp<AMessage> reply;
- CHECK(msg->findMessage("reply", &reply));
-
- queueDecoderShutdown(audio, video, reply);
- break;
- }
-
- case Source::kWhatDrmNoLicense:
- {
- notifyListener(srcId, MEDIA2_ERROR, MEDIA2_ERROR_UNKNOWN, ERROR_DRM_NO_LICENSE);
- break;
- }
-
- default:
- TRESPASS();
- }
-}
-
-void NuPlayer2::onClosedCaptionNotify(const sp<AMessage> &msg) {
- int32_t what;
- CHECK(msg->findInt32("what", &what));
-
- switch (what) {
- case NuPlayer2::CCDecoder::kWhatClosedCaptionData:
- {
- sp<ABuffer> buffer;
- CHECK(msg->findBuffer("buffer", &buffer));
-
- size_t inbandTracks = 0;
- if (mCurrentSourceInfo.mSource != NULL) {
- inbandTracks = mCurrentSourceInfo.mSource->getTrackCount();
- }
-
- sendSubtitleData(buffer, inbandTracks);
- break;
- }
-
- case NuPlayer2::CCDecoder::kWhatTrackAdded:
- {
- notifyListener(mCurrentSourceInfo.mSrcId, MEDIA2_INFO, MEDIA2_INFO_METADATA_UPDATE, 0);
-
- break;
- }
-
- default:
- TRESPASS();
- }
-
-
-}
-
-void NuPlayer2::sendSubtitleData(const sp<ABuffer> &buffer, int32_t baseIndex) {
- int32_t trackIndex;
- int64_t timeUs, durationUs;
- CHECK(buffer->meta()->findInt32(AMEDIAFORMAT_KEY_TRACK_INDEX, &trackIndex));
- CHECK(buffer->meta()->findInt64("timeUs", &timeUs));
- CHECK(buffer->meta()->findInt64("durationUs", &durationUs));
-
- PlayerMessage playerMsg;
- playerMsg.add_values()->set_int32_value(trackIndex + baseIndex);
- playerMsg.add_values()->set_int64_value(timeUs);
- playerMsg.add_values()->set_int64_value(durationUs);
- playerMsg.add_values()->set_bytes_value(buffer->data(), buffer->size());
-
- notifyListener(mCurrentSourceInfo.mSrcId, MEDIA2_SUBTITLE_DATA, 0, 0, &playerMsg);
-}
-
-void NuPlayer2::sendTimedMetaData(const sp<ABuffer> &buffer) {
- int64_t timeUs;
- CHECK(buffer->meta()->findInt64("timeUs", &timeUs));
-
- PlayerMessage playerMsg;
- playerMsg.add_values()->set_int64_value(timeUs);
- playerMsg.add_values()->set_bytes_value(buffer->data(), buffer->size());
-
- notifyListener(mCurrentSourceInfo.mSrcId, MEDIA2_META_DATA, 0, 0, &playerMsg);
-}
-
-void NuPlayer2::sendTimedTextData(const sp<ABuffer> &buffer) {
- const void *data;
- size_t size = 0;
- int64_t timeUs;
- int32_t flag = TextDescriptions2::IN_BAND_TEXT_3GPP;
-
- AString mime;
- CHECK(buffer->meta()->findString("mime", &mime));
- CHECK(strcasecmp(mime.c_str(), MEDIA_MIMETYPE_TEXT_3GPP) == 0);
-
- data = buffer->data();
- size = buffer->size();
-
- PlayerMessage playerMsg;
- if (size > 0) {
- CHECK(buffer->meta()->findInt64("timeUs", &timeUs));
- int32_t global = 0;
- if (buffer->meta()->findInt32("global", &global) && global) {
- flag |= TextDescriptions2::GLOBAL_DESCRIPTIONS;
- } else {
- flag |= TextDescriptions2::LOCAL_DESCRIPTIONS;
- }
- TextDescriptions2::getPlayerMessageOfDescriptions(
- (const uint8_t *)data, size, flag, timeUs / 1000, &playerMsg);
- }
-
- if (playerMsg.values_size() > 0) {
- notifyListener(mCurrentSourceInfo.mSrcId, MEDIA2_TIMED_TEXT, 0, 0, &playerMsg);
- } else { // send an empty timed text
- notifyListener(mCurrentSourceInfo.mSrcId, MEDIA2_TIMED_TEXT, 0, 0);
- }
-}
-
-const char *NuPlayer2::getDataSourceType() {
- switch (mCurrentSourceInfo.mDataSourceType) {
- case DATA_SOURCE_TYPE_HTTP_LIVE:
- return "HTTPLive";
-
- case DATA_SOURCE_TYPE_RTSP:
- return "RTSP";
-
- case DATA_SOURCE_TYPE_GENERIC_URL:
- return "GenURL";
-
- case DATA_SOURCE_TYPE_GENERIC_FD:
- return "GenFD";
-
- case DATA_SOURCE_TYPE_MEDIA:
- return "Media";
-
- case DATA_SOURCE_TYPE_NONE:
- default:
- return "None";
- }
- }
-
-NuPlayer2::SourceInfo* NuPlayer2::getSourceInfoByIdInMsg(const sp<AMessage> &msg) {
- int64_t srcId;
- CHECK(msg->findInt64("srcId", &srcId));
- if (mCurrentSourceInfo.mSrcId == srcId) {
- return &mCurrentSourceInfo;
- } else if (mNextSourceInfo.mSrcId == srcId) {
- return &mNextSourceInfo;
- } else {
- return NULL;
- }
-}
-
-void NuPlayer2::resetSourceInfo(NuPlayer2::SourceInfo &srcInfo) {
- if (srcInfo.mSource != NULL) {
- srcInfo.mSource->stop();
-
- Mutex::Autolock autoLock(mSourceLock);
- srcInfo.mSource.clear();
- }
- // Modular DRM
- ALOGD("performReset mCrypto: %p", srcInfo.mCrypto.get());
- srcInfo.mCrypto.clear();
- srcInfo.mIsDrmProtected = false;
-}
-
-// Modular DRM begin
-status_t NuPlayer2::prepareDrm(
- int64_t srcId, const uint8_t uuid[16], const Vector<uint8_t> &drmSessionId)
-{
- ALOGV("prepareDrm ");
-
- // Passing to the looper anyway; called in a pre-config prepared state so no race on mCrypto
- sp<AMessage> msg = new AMessage(kWhatPrepareDrm, this);
- // synchronous call so just passing the address but with local copies of "const" args
- uint8_t UUID[16];
- memcpy(UUID, uuid, sizeof(UUID));
- Vector<uint8_t> sessionId = drmSessionId;
- msg->setInt64("srcId", srcId);
- msg->setPointer("uuid", (void*)UUID);
- msg->setPointer("drmSessionId", (void*)&sessionId);
-
- sp<AMessage> response;
- status_t status = msg->postAndAwaitResponse(&response);
-
- if (status == OK && response != NULL) {
- CHECK(response->findInt32("status", &status));
- ALOGV("prepareDrm ret: %d ", status);
- } else {
- ALOGE("prepareDrm err: %d", status);
- }
-
- return status;
-}
-
-status_t NuPlayer2::releaseDrm(int64_t srcId)
-{
- ALOGV("releaseDrm ");
-
- sp<AMessage> msg = new AMessage(kWhatReleaseDrm, this);
- msg->setInt64("srcId", srcId);
-
- sp<AMessage> response;
- status_t status = msg->postAndAwaitResponse(&response);
-
- if (status == OK && response != NULL) {
- CHECK(response->findInt32("status", &status));
- ALOGV("releaseDrm ret: %d ", status);
- } else {
- ALOGE("releaseDrm err: %d", status);
- }
-
- return status;
-}
-
-status_t NuPlayer2::onPrepareDrm(const sp<AMessage> &msg)
-{
- // TODO change to ALOGV
- ALOGD("onPrepareDrm ");
-
- status_t status = INVALID_OPERATION;
- SourceInfo *srcInfo = getSourceInfoByIdInMsg(msg);
- if (srcInfo == NULL) {
- return status;
- }
-
- int64_t srcId = srcInfo->mSrcId;
- if (srcInfo->mSource == NULL) {
- ALOGE("onPrepareDrm: srcInfo(%lld) No source. onPrepareDrm failed with %d.",
- (long long)srcId, status);
- return status;
- }
-
- uint8_t *uuid;
- Vector<uint8_t> *drmSessionId;
- CHECK(msg->findPointer("uuid", (void**)&uuid));
- CHECK(msg->findPointer("drmSessionId", (void**)&drmSessionId));
-
- status = OK;
- sp<AMediaCryptoWrapper> crypto = NULL;
-
- status = srcInfo->mSource->prepareDrm(uuid, *drmSessionId, &crypto);
- if (crypto == NULL) {
- ALOGE("onPrepareDrm: srcInfo(%lld).mSource->prepareDrm failed. status: %d",
- (long long)srcId, status);
- return status;
- }
- ALOGV("onPrepareDrm: srcInfo(%lld).mSource->prepareDrm succeeded", (long long)srcId);
-
- if (srcInfo->mCrypto != NULL) {
- ALOGE("onPrepareDrm: srcInfo(%lld) Unexpected. Already having mCrypto: %p",
- (long long)srcId, srcInfo->mCrypto.get());
- srcInfo->mCrypto.clear();
- }
-
- srcInfo->mCrypto = crypto;
- srcInfo->mIsDrmProtected = true;
- // TODO change to ALOGV
- ALOGD("onPrepareDrm: mCrypto: %p", srcInfo->mCrypto.get());
-
- return status;
-}
-
-status_t NuPlayer2::onReleaseDrm(const sp<AMessage> &msg)
-{
- // TODO change to ALOGV
- ALOGD("onReleaseDrm ");
- SourceInfo *srcInfo = getSourceInfoByIdInMsg(msg);;
- if (srcInfo == NULL) {
- return INVALID_OPERATION;
- }
-
- int64_t srcId = srcInfo->mSrcId;
- if (!srcInfo->mIsDrmProtected) {
- ALOGW("onReleaseDrm: srcInfo(%lld) Unexpected. mIsDrmProtected is already false.",
- (long long)srcId);
- }
-
- srcInfo->mIsDrmProtected = false;
-
- status_t status;
- if (srcInfo->mCrypto != NULL) {
- // notifying the source first before removing crypto from codec
- if (srcInfo->mSource != NULL) {
- srcInfo->mSource->releaseDrm();
- }
-
- status=OK;
- // first making sure the codecs have released their crypto reference
- const sp<DecoderBase> &videoDecoder = getDecoder(false/*audio*/);
- if (videoDecoder != NULL) {
- status = videoDecoder->releaseCrypto();
- ALOGV("onReleaseDrm: video decoder ret: %d", status);
- }
-
- const sp<DecoderBase> &audioDecoder = getDecoder(true/*audio*/);
- if (audioDecoder != NULL) {
- status_t status_audio = audioDecoder->releaseCrypto();
- if (status == OK) { // otherwise, returning the first error
- status = status_audio;
- }
- ALOGV("onReleaseDrm: audio decoder ret: %d", status_audio);
- }
-
- // TODO change to ALOGV
- ALOGD("onReleaseDrm: mCrypto: %p", srcInfo->mCrypto.get());
- srcInfo->mCrypto.clear();
- } else { // srcInfo->mCrypto == NULL
- ALOGE("onReleaseDrm: Unexpected. There is no crypto.");
- status = INVALID_OPERATION;
- }
-
- return status;
-}
-// Modular DRM end
-////////////////////////////////////////////////////////////////////////////////
-
-sp<AMessage> NuPlayer2::Source::getFormat(bool audio) {
- sp<MetaData> meta = getFormatMeta(audio);
-
- if (meta == NULL) {
- return NULL;
- }
-
- sp<AMessage> msg = new AMessage;
-
- if(convertMetaDataToMessage(meta, &msg) == OK) {
- return msg;
- }
- return NULL;
-}
-
-void NuPlayer2::Source::notifyFlagsChanged(uint32_t flags) {
- sp<AMessage> notify = dupNotify();
- notify->setInt32("what", kWhatFlagsChanged);
- notify->setInt32("flags", flags);
- notify->post();
-}
-
-void NuPlayer2::Source::notifyVideoSizeChanged(const sp<AMessage> &format) {
- sp<AMessage> notify = dupNotify();
- notify->setInt32("what", kWhatVideoSizeChanged);
- notify->setMessage("format", format);
- notify->post();
-}
-
-void NuPlayer2::Source::notifyPrepared(status_t err) {
- ALOGV("Source::notifyPrepared %d", err);
- sp<AMessage> notify = dupNotify();
- notify->setInt32("what", kWhatPrepared);
- notify->setInt32("err", err);
- notify->post();
-}
-
-void NuPlayer2::Source::notifyDrmInfo(const sp<ABuffer> &drmInfoBuffer)
-{
- ALOGV("Source::notifyDrmInfo");
-
- sp<AMessage> notify = dupNotify();
- notify->setInt32("what", kWhatDrmInfo);
- notify->setBuffer("drmInfo", drmInfoBuffer);
-
- notify->post();
-}
-
-void NuPlayer2::Source::onMessageReceived(const sp<AMessage> & /* msg */) {
- TRESPASS();
-}
-
-NuPlayer2::SourceInfo::SourceInfo()
- : mDataSourceType(DATA_SOURCE_TYPE_NONE),
- mSrcId(0),
- mSourceFlags(0),
- mStartTimeUs(0),
- mEndTimeUs(DataSourceDesc::kMaxTimeUs) {
-}
-
-NuPlayer2::SourceInfo & NuPlayer2::SourceInfo::operator=(const NuPlayer2::SourceInfo &other) {
- mSource = other.mSource;
- mCrypto = other.mCrypto;
- mDataSourceType = (DATA_SOURCE_TYPE)other.mDataSourceType;
- mSrcId = other.mSrcId;
- mSourceFlags = other.mSourceFlags;
- mStartTimeUs = other.mStartTimeUs;
- mEndTimeUs = other.mEndTimeUs;
- mIsDrmProtected = other.mIsDrmProtected;
- return *this;
-}
-
-} // namespace android
diff --git a/media/libmediaplayer2/nuplayer2/NuPlayer2.h b/media/libmediaplayer2/nuplayer2/NuPlayer2.h
deleted file mode 100644
index b8fb988..0000000
--- a/media/libmediaplayer2/nuplayer2/NuPlayer2.h
+++ /dev/null
@@ -1,369 +0,0 @@
-/*
- * Copyright 2017 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef NU_PLAYER2_H_
-
-#define NU_PLAYER2_H_
-
-#include <media/AudioResamplerPublic.h>
-#include <media/stagefright/foundation/AHandler.h>
-
-#include <mediaplayer2/MediaPlayer2Interface.h>
-#include <mediaplayer2/JObjectHolder.h>
-
-#include "mediaplayer2.pb.h"
-
-using android::media::MediaPlayer2Proto::PlayerMessage;
-
-namespace android {
-
-struct ABuffer;
-struct AMediaCryptoWrapper;
-struct AMessage;
-struct ANativeWindowWrapper;
-struct AudioPlaybackRate;
-struct AVSyncSettings;
-struct DataSourceDesc;
-struct MediaClock;
-struct MediaHTTPService;
-class MetaData;
-struct NuPlayer2Driver;
-
-struct NuPlayer2 : public AHandler {
- explicit NuPlayer2(pid_t pid, uid_t uid,
- const sp<MediaClock> &mediaClock, const sp<JObjectHolder> &context);
-
- void setDriver(const wp<NuPlayer2Driver> &driver);
-
- void setDataSourceAsync(const sp<DataSourceDesc> &dsd);
- void prepareNextDataSourceAsync(const sp<DataSourceDesc> &dsd);
- void playNextDataSource(int64_t srcId);
-
- status_t getBufferingSettings(BufferingSettings* buffering /* nonnull */);
- status_t setBufferingSettings(const BufferingSettings& buffering);
-
- void prepareAsync();
-
- void setVideoSurfaceTextureAsync(const sp<ANativeWindowWrapper> &nww);
-
- void setAudioSink(const sp<MediaPlayer2Interface::AudioSink> &sink);
- status_t setPlaybackSettings(const AudioPlaybackRate &rate);
- status_t getPlaybackSettings(AudioPlaybackRate *rate /* nonnull */);
- status_t setSyncSettings(const AVSyncSettings &sync, float videoFpsHint);
- status_t getSyncSettings(AVSyncSettings *sync /* nonnull */, float *videoFps /* nonnull */);
-
- void start();
-
- void pause();
-
- // Will notify the driver through "notifyResetComplete" once finished.
- void resetAsync();
-
- // Request a notification when specified media time is reached.
- status_t notifyAt(int64_t mediaTimeUs);
-
- // Will notify the driver through "notifySeekComplete" once finished
- // and needNotify is true.
- void seekToAsync(
- int64_t seekTimeUs,
- MediaPlayer2SeekMode mode = MediaPlayer2SeekMode::SEEK_PREVIOUS_SYNC,
- bool needNotify = false);
- void rewind();
-
- status_t setVideoScalingMode(int32_t mode);
- status_t getTrackInfo(int64_t srcId, PlayerMessage* reply) const;
- status_t getSelectedTrack(int64_t srcId, int32_t type, PlayerMessage* reply) const;
- status_t selectTrack(int64_t srcId, size_t trackIndex, bool select, int64_t timeUs);
- status_t getCurrentPosition(int64_t *mediaUs);
- void getStats(Vector<sp<AMessage> > *mTrackStats);
-
- sp<MetaData> getFileMeta();
- float getFrameRate();
-
- // Modular DRM
- status_t prepareDrm(int64_t srcId, const uint8_t uuid[16], const Vector<uint8_t> &drmSessionId);
- status_t releaseDrm(int64_t srcId);
-
- const char *getDataSourceType();
-
-protected:
- virtual ~NuPlayer2();
-
- virtual void onMessageReceived(const sp<AMessage> &msg);
-
-public:
- struct StreamListener;
- struct Source;
-
-private:
- struct Decoder;
- struct DecoderBase;
- struct DecoderPassThrough;
- struct CCDecoder;
- struct GenericSource2;
- struct HTTPLiveSource2;
- struct Renderer;
- struct RTSPSource2;
- struct Action;
- struct SeekAction;
- struct SetSurfaceAction;
- struct ResumeDecoderAction;
- struct FlushDecoderAction;
- struct PostMessageAction;
- struct SimpleAction;
-
- enum {
- kWhatSetDataSource = '=DaS',
- kWhatPrepare = 'prep',
- kWhatPrepareNextDataSource = 'pNDS',
- kWhatPlayNextDataSource = 'plNS',
- kWhatSetVideoSurface = '=VSu',
- kWhatSetAudioSink = '=AuS',
- kWhatMoreDataQueued = 'more',
- kWhatConfigPlayback = 'cfPB',
- kWhatConfigSync = 'cfSy',
- kWhatGetPlaybackSettings = 'gPbS',
- kWhatGetSyncSettings = 'gSyS',
- kWhatStart = 'strt',
- kWhatScanSources = 'scan',
- kWhatVideoNotify = 'vidN',
- kWhatAudioNotify = 'audN',
- kWhatClosedCaptionNotify = 'capN',
- kWhatRendererNotify = 'renN',
- kWhatReset = 'rset',
- kWhatNotifyTime = 'nfyT',
- kWhatSeek = 'seek',
- kWhatPause = 'paus',
- kWhatResume = 'rsme',
- kWhatPollDuration = 'polD',
- kWhatSourceNotify = 'srcN',
- kWhatGetTrackInfo = 'gTrI',
- kWhatGetSelectedTrack = 'gSel',
- kWhatSelectTrack = 'selT',
- kWhatGetBufferingSettings = 'gBus',
- kWhatSetBufferingSettings = 'sBuS',
- kWhatPrepareDrm = 'pDrm',
- kWhatReleaseDrm = 'rDrm',
- kWhatRewind = 'reWd',
- kWhatEOSMonitor = 'eosM',
- };
-
- typedef enum {
- DATA_SOURCE_TYPE_NONE,
- DATA_SOURCE_TYPE_HTTP_LIVE,
- DATA_SOURCE_TYPE_RTSP,
- DATA_SOURCE_TYPE_GENERIC_URL,
- DATA_SOURCE_TYPE_GENERIC_FD,
- DATA_SOURCE_TYPE_MEDIA,
- } DATA_SOURCE_TYPE;
-
- struct SourceInfo {
- SourceInfo();
- SourceInfo &operator=(const SourceInfo &);
-
- sp<Source> mSource;
- std::atomic<DATA_SOURCE_TYPE> mDataSourceType;
- int64_t mSrcId;
- uint32_t mSourceFlags;
- int64_t mStartTimeUs;
- int64_t mEndTimeUs;
- // Modular DRM
- sp<AMediaCryptoWrapper> mCrypto;
- bool mIsDrmProtected = false;
- };
-
- wp<NuPlayer2Driver> mDriver;
- pid_t mPID;
- uid_t mUID;
- const sp<MediaClock> mMediaClock;
- Mutex mSourceLock; // guard |mSource|.
- SourceInfo mCurrentSourceInfo;
- SourceInfo mNextSourceInfo;
- sp<ANativeWindowWrapper> mNativeWindow;
- sp<MediaPlayer2Interface::AudioSink> mAudioSink;
- sp<DecoderBase> mVideoDecoder;
- bool mOffloadAudio;
- sp<DecoderBase> mAudioDecoder;
- Mutex mDecoderLock; // guard |mAudioDecoder| and |mVideoDecoder|.
- sp<CCDecoder> mCCDecoder;
- sp<Renderer> mRenderer;
- sp<ALooper> mRendererLooper;
- int32_t mAudioDecoderGeneration;
- int32_t mVideoDecoderGeneration;
- int32_t mRendererGeneration;
- int32_t mEOSMonitorGeneration;
-
- Mutex mPlayingTimeLock;
- int64_t mLastStartedPlayingTimeNs;
- void stopPlaybackTimer(const char *where);
- void startPlaybackTimer(const char *where);
-
- int64_t mLastStartedRebufferingTimeNs;
- void startRebufferingTimer();
- void stopRebufferingTimer(bool exitingPlayback);
-
- int64_t mPreviousSeekTimeUs;
-
- List<sp<Action> > mDeferredActions;
-
- bool mAudioEOS;
- bool mVideoEOS;
-
- bool mScanSourcesPending;
- int32_t mScanSourcesGeneration;
-
- int32_t mPollDurationGeneration;
- int32_t mTimedTextGeneration;
-
- enum FlushStatus {
- NONE,
- FLUSHING_DECODER,
- FLUSHING_DECODER_SHUTDOWN,
- SHUTTING_DOWN_DECODER,
- FLUSHED,
- SHUT_DOWN,
- };
-
- enum FlushCommand {
- FLUSH_CMD_NONE,
- FLUSH_CMD_FLUSH,
- FLUSH_CMD_SHUTDOWN,
- };
-
- // Status of flush responses from the decoder and renderer.
- bool mFlushComplete[2][2];
-
- FlushStatus mFlushingAudio;
- FlushStatus mFlushingVideo;
-
- // Status of flush responses from the decoder and renderer.
- bool mResumePending;
-
- int32_t mVideoScalingMode;
-
- AudioPlaybackRate mPlaybackSettings;
- AVSyncSettings mSyncSettings;
- float mVideoFpsHint;
- bool mStarted;
- bool mPrepared;
- bool mResetting;
- bool mSourceStarted;
- bool mAudioDecoderError;
- bool mVideoDecoderError;
-
- // Actual pause state, either as requested by client or due to buffering.
- bool mPaused;
-
- // Pause state as requested by client. Note that if mPausedByClient is
- // true, mPaused is always true; if mPausedByClient is false, mPaused could
- // still become true, when we pause internally due to buffering.
- bool mPausedByClient;
-
- // Pause state as requested by source (internally) due to buffering
- bool mPausedForBuffering;
-
- // Passed from JAVA
- const sp<JObjectHolder> mContext;
-
- inline const sp<DecoderBase> &getDecoder(bool audio) {
- return audio ? mAudioDecoder : mVideoDecoder;
- }
-
- inline void clearFlushComplete() {
- mFlushComplete[0][0] = false;
- mFlushComplete[0][1] = false;
- mFlushComplete[1][0] = false;
- mFlushComplete[1][1] = false;
- }
-
- void disconnectSource();
-
- status_t createNuPlayer2Source(const sp<DataSourceDesc> &dsd,
- sp<Source> *source,
- DATA_SOURCE_TYPE *dataSourceType);
-
- void tryOpenAudioSinkForOffload(
- const sp<AMessage> &format, const sp<MetaData> &audioMeta, bool hasVideo);
- void closeAudioSink();
- void restartAudio(
- int64_t currentPositionUs, bool forceNonOffload, bool needsToCreateAudioDecoder);
- void determineAudioModeChange(const sp<AMessage> &audioFormat);
-
- status_t instantiateDecoder(
- bool audio, sp<DecoderBase> *decoder, bool checkAudioModeChange = true);
-
- void updateVideoSize(
- int64_t srcId,
- const sp<AMessage> &inputFormat,
- const sp<AMessage> &outputFormat = NULL);
-
- void notifyListener(int64_t srcId, int msg, int ext1, int ext2, const PlayerMessage *in = NULL);
-
- void addEndTimeMonitor();
-
- void handleFlushComplete(bool audio, bool isDecoder);
- void finishFlushIfPossible();
-
- void onStart(bool play);
- void onResume();
- void onPause();
-
- bool audioDecoderStillNeeded();
-
- void flushDecoder(bool audio, bool needShutdown);
-
- void finishResume();
- void notifyDriverSeekComplete(int64_t srcId);
-
- void postScanSources();
-
- void schedulePollDuration();
- void cancelPollDuration();
-
- void processDeferredActions();
-
- void performSeek(int64_t seekTimeUs, MediaPlayer2SeekMode mode);
- void performDecoderFlush(FlushCommand audio, FlushCommand video);
- void performReset();
- void performPlayNextDataSource();
- void performScanSources();
- void performSetSurface(const sp<ANativeWindowWrapper> &nw);
- void performResumeDecoders(bool needNotify);
-
- void onSourceNotify(const sp<AMessage> &msg);
- void onClosedCaptionNotify(const sp<AMessage> &msg);
-
- void queueDecoderShutdown(
- bool audio, bool video, const sp<AMessage> &reply);
-
- void sendSubtitleData(const sp<ABuffer> &buffer, int32_t baseIndex);
- void sendTimedMetaData(const sp<ABuffer> &buffer);
- void sendTimedTextData(const sp<ABuffer> &buffer);
-
- void writeTrackInfo(PlayerMessage* reply, const sp<AMessage>& format) const;
-
- status_t onPrepareDrm(const sp<AMessage> &msg);
- status_t onReleaseDrm(const sp<AMessage> &msg);
-
- SourceInfo* getSourceInfoByIdInMsg(const sp<AMessage> &msg);
- void resetSourceInfo(SourceInfo &srcInfo);
-
- DISALLOW_EVIL_CONSTRUCTORS(NuPlayer2);
-};
-
-} // namespace android
-
-#endif // NU_PLAYER2_H_
diff --git a/media/libmediaplayer2/nuplayer2/NuPlayer2CCDecoder.cpp b/media/libmediaplayer2/nuplayer2/NuPlayer2CCDecoder.cpp
deleted file mode 100644
index 98c3403..0000000
--- a/media/libmediaplayer2/nuplayer2/NuPlayer2CCDecoder.cpp
+++ /dev/null
@@ -1,606 +0,0 @@
-/*
- * Copyright 2017 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "NuPlayer2CCDecoder"
-#include <utils/Log.h>
-#include <inttypes.h>
-
-#include "NuPlayer2CCDecoder.h"
-
-#include <media/NdkMediaFormat.h>
-#include <media/stagefright/foundation/ABitReader.h>
-#include <media/stagefright/foundation/ABuffer.h>
-#include <media/stagefright/foundation/ADebug.h>
-#include <media/stagefright/foundation/AMessage.h>
-#include <media/stagefright/foundation/avc_utils.h>
-#include <media/stagefright/MediaDefs.h>
-
-namespace android {
-
-// In CEA-708B, the maximum bandwidth of CC is set to 9600bps.
-static const size_t kMaxBandwithSizeBytes = 9600 / 8;
-
-struct CCData {
- CCData(uint8_t type, uint8_t data1, uint8_t data2)
- : mType(type), mData1(data1), mData2(data2) {
- }
- bool getChannel(size_t *channel) const {
- if (mData1 >= 0x10 && mData1 <= 0x1f) {
- *channel = (mData1 >= 0x18 ? 1 : 0) + (mType ? 2 : 0);
- return true;
- }
- return false;
- }
-
- uint8_t mType;
- uint8_t mData1;
- uint8_t mData2;
-};
-
-static bool isNullPad(CCData *cc) {
- return cc->mData1 < 0x10 && cc->mData2 < 0x10;
-}
-
-static void dumpBytePair(const sp<ABuffer> &ccBuf) __attribute__ ((unused));
-static void dumpBytePair(const sp<ABuffer> &ccBuf) {
- size_t offset = 0;
- AString out;
-
- while (offset < ccBuf->size()) {
- char tmp[128];
-
- CCData *cc = (CCData *) (ccBuf->data() + offset);
-
- if (isNullPad(cc)) {
- // 1 null pad or XDS metadata, ignore
- offset += sizeof(CCData);
- continue;
- }
-
- if (cc->mData1 >= 0x20 && cc->mData1 <= 0x7f) {
- // 2 basic chars
- snprintf(tmp, sizeof(tmp), "[%d]Basic: %c %c", cc->mType, cc->mData1, cc->mData2);
- } else if ((cc->mData1 == 0x11 || cc->mData1 == 0x19)
- && cc->mData2 >= 0x30 && cc->mData2 <= 0x3f) {
- // 1 special char
- snprintf(tmp, sizeof(tmp), "[%d]Special: %02x %02x", cc->mType, cc->mData1, cc->mData2);
- } else if ((cc->mData1 == 0x12 || cc->mData1 == 0x1A)
- && cc->mData2 >= 0x20 && cc->mData2 <= 0x3f){
- // 1 Spanish/French char
- snprintf(tmp, sizeof(tmp), "[%d]Spanish: %02x %02x", cc->mType, cc->mData1, cc->mData2);
- } else if ((cc->mData1 == 0x13 || cc->mData1 == 0x1B)
- && cc->mData2 >= 0x20 && cc->mData2 <= 0x3f){
- // 1 Portuguese/German/Danish char
- snprintf(tmp, sizeof(tmp), "[%d]German: %02x %02x", cc->mType, cc->mData1, cc->mData2);
- } else if ((cc->mData1 == 0x11 || cc->mData1 == 0x19)
- && cc->mData2 >= 0x20 && cc->mData2 <= 0x2f){
- // Mid-Row Codes (Table 69)
- snprintf(tmp, sizeof(tmp), "[%d]Mid-row: %02x %02x", cc->mType, cc->mData1, cc->mData2);
- } else if (((cc->mData1 == 0x14 || cc->mData1 == 0x1c)
- && cc->mData2 >= 0x20 && cc->mData2 <= 0x2f)
- ||
- ((cc->mData1 == 0x17 || cc->mData1 == 0x1f)
- && cc->mData2 >= 0x21 && cc->mData2 <= 0x23)){
- // Misc Control Codes (Table 70)
- snprintf(tmp, sizeof(tmp), "[%d]Ctrl: %02x %02x", cc->mType, cc->mData1, cc->mData2);
- } else if ((cc->mData1 & 0x70) == 0x10
- && (cc->mData2 & 0x40) == 0x40
- && ((cc->mData1 & 0x07) || !(cc->mData2 & 0x20)) ) {
- // Preamble Address Codes (Table 71)
- snprintf(tmp, sizeof(tmp), "[%d]PAC: %02x %02x", cc->mType, cc->mData1, cc->mData2);
- } else {
- snprintf(tmp, sizeof(tmp), "[%d]Invalid: %02x %02x", cc->mType, cc->mData1, cc->mData2);
- }
-
- if (out.size() > 0) {
- out.append(", ");
- }
-
- out.append(tmp);
-
- offset += sizeof(CCData);
- }
-
- ALOGI("%s", out.c_str());
-}
-
-NuPlayer2::CCDecoder::CCDecoder(const sp<AMessage> ¬ify)
- : mNotify(notify),
- mSelectedTrack(-1),
- mDTVCCPacket(new ABuffer(kMaxBandwithSizeBytes)) {
- mDTVCCPacket->setRange(0, 0);
-
- // In CEA-608, streams from packets which have the value 0 of cc_type contain CC1 and CC2, and
- // streams from packets which have the value 1 of cc_type contain CC3 and CC4.
- // The following array indicates the current transmitting channels for each value of cc_type.
- mLine21Channels[0] = 0; // CC1
- mLine21Channels[1] = 2; // CC3
-}
-
-size_t NuPlayer2::CCDecoder::getTrackCount() const {
- return mTracks.size();
-}
-
-sp<AMessage> NuPlayer2::CCDecoder::getTrackInfo(size_t index) const {
- if (!isTrackValid(index)) {
- return NULL;
- }
-
- sp<AMessage> format = new AMessage();
-
- CCTrack track = mTracks[index];
-
- format->setInt32("type", MEDIA_TRACK_TYPE_SUBTITLE);
- format->setString("language", "und");
-
- switch (track.mTrackType) {
- case kTrackTypeCEA608:
- format->setString("mime", MEDIA_MIMETYPE_TEXT_CEA_608);
- break;
- case kTrackTypeCEA708:
- format->setString("mime", MEDIA_MIMETYPE_TEXT_CEA_708);
- break;
- default:
- ALOGE("Unknown track type: %d", track.mTrackType);
- return NULL;
- }
-
- // For CEA-608 CC1, field 0 channel 0
- bool isDefaultAuto = track.mTrackType == kTrackTypeCEA608
- && track.mTrackChannel == 0;
- // For CEA-708, Primary Caption Service.
- bool isDefaultOnly = track.mTrackType == kTrackTypeCEA708
- && track.mTrackChannel == 1;
- format->setInt32("auto", isDefaultAuto);
- format->setInt32("default", isDefaultAuto || isDefaultOnly);
- format->setInt32("forced", 0);
-
- return format;
-}
-
-status_t NuPlayer2::CCDecoder::selectTrack(size_t index, bool select) {
- if (!isTrackValid(index)) {
- return BAD_VALUE;
- }
-
- if (select) {
- if (mSelectedTrack == (ssize_t)index) {
- ALOGE("track %zu already selected", index);
- return BAD_VALUE;
- }
- ALOGV("selected track %zu", index);
- mSelectedTrack = index;
- } else {
- if (mSelectedTrack != (ssize_t)index) {
- ALOGE("track %zu is not selected", index);
- return BAD_VALUE;
- }
- ALOGV("unselected track %zu", index);
- mSelectedTrack = -1;
- }
-
- // Clear the previous track payloads
- mCCMap.clear();
-
- return OK;
-}
-
-ssize_t NuPlayer2::CCDecoder::getSelectedTrack(media_track_type type) const {
- if (mSelectedTrack != -1) {
- CCTrack track = mTracks[mSelectedTrack];
- if (track.mTrackType == kTrackTypeCEA608 || track.mTrackType == kTrackTypeCEA708) {
- return (type == MEDIA_TRACK_TYPE_SUBTITLE ? mSelectedTrack : -1);
- }
- return (type == MEDIA_TRACK_TYPE_UNKNOWN ? mSelectedTrack : -1);
- }
-
- return -1;
-}
-
-bool NuPlayer2::CCDecoder::isSelected() const {
- return mSelectedTrack >= 0 && mSelectedTrack < (int32_t)getTrackCount();
-}
-
-bool NuPlayer2::CCDecoder::isTrackValid(size_t index) const {
- return index < getTrackCount();
-}
-
-// returns true if a new CC track is found
-bool NuPlayer2::CCDecoder::extractFromSEI(const sp<ABuffer> &accessUnit) {
- sp<ABuffer> sei;
- if (!accessUnit->meta()->findBuffer("sei", &sei) || sei == NULL) {
- return false;
- }
-
- int64_t timeUs;
- CHECK(accessUnit->meta()->findInt64("timeUs", &timeUs));
-
- bool trackAdded = false;
-
- const NALPosition *nal = (NALPosition *)sei->data();
-
- for (size_t i = 0; i < sei->size() / sizeof(NALPosition); ++i, ++nal) {
- trackAdded |= parseSEINalUnit(
- timeUs, accessUnit->data() + nal->nalOffset, nal->nalSize);
- }
-
- return trackAdded;
-}
-
-// returns true if a new CC track is found
-bool NuPlayer2::CCDecoder::parseSEINalUnit(int64_t timeUs, const uint8_t *data, size_t size) {
- unsigned nalType = data[0] & 0x1f;
-
- // the buffer should only have SEI in it
- if (nalType != 6) {
- return false;
- }
-
- bool trackAdded = false;
- NALBitReader br(data + 1, size - 1);
-
- // sei_message()
- while (br.atLeastNumBitsLeft(16)) { // at least 16-bit for sei_message()
- uint32_t payload_type = 0;
- size_t payload_size = 0;
- uint8_t last_byte;
-
- do {
- last_byte = br.getBits(8);
- payload_type += last_byte;
- } while (last_byte == 0xFF);
-
- do {
- last_byte = br.getBits(8);
- payload_size += last_byte;
- } while (last_byte == 0xFF);
-
- if (payload_size > SIZE_MAX / 8
- || !br.atLeastNumBitsLeft(payload_size * 8)) {
- ALOGV("Malformed SEI payload");
- break;
- }
-
- // sei_payload()
- if (payload_type == 4) {
- bool isCC = false;
- if (payload_size > 1 + 2 + 4 + 1) {
- // user_data_registered_itu_t_t35()
-
- // ATSC A/72: 6.4.2
- uint8_t itu_t_t35_country_code = br.getBits(8);
- uint16_t itu_t_t35_provider_code = br.getBits(16);
- uint32_t user_identifier = br.getBits(32);
- uint8_t user_data_type_code = br.getBits(8);
-
- payload_size -= 1 + 2 + 4 + 1;
-
- isCC = itu_t_t35_country_code == 0xB5
- && itu_t_t35_provider_code == 0x0031
- && user_identifier == 'GA94'
- && user_data_type_code == 0x3;
- }
-
- if (isCC && payload_size > 2) {
- trackAdded |= parseMPEGCCData(timeUs, br.data(), br.numBitsLeft() / 8);
- } else {
- ALOGV("Malformed SEI payload type 4");
- }
- } else {
- ALOGV("Unsupported SEI payload type %d", payload_type);
- }
-
- // skipping remaining bits of this payload
- br.skipBits(payload_size * 8);
- }
-
- return trackAdded;
-}
-
-// returns true if a new CC track is found
-bool NuPlayer2::CCDecoder::extractFromMPEGUserData(const sp<ABuffer> &accessUnit) {
- sp<ABuffer> mpegUserData;
- if (!accessUnit->meta()->findBuffer(AMEDIAFORMAT_KEY_MPEG_USER_DATA, &mpegUserData)
- || mpegUserData == NULL) {
- return false;
- }
-
- int64_t timeUs;
- CHECK(accessUnit->meta()->findInt64("timeUs", &timeUs));
-
- bool trackAdded = false;
-
- const size_t *userData = (size_t *)mpegUserData->data();
-
- for (size_t i = 0; i < mpegUserData->size() / sizeof(size_t); ++i) {
- trackAdded |= parseMPEGUserDataUnit(
- timeUs, accessUnit->data() + userData[i], accessUnit->size() - userData[i]);
- }
-
- return trackAdded;
-}
-
-// returns true if a new CC track is found
-bool NuPlayer2::CCDecoder::parseMPEGUserDataUnit(int64_t timeUs, const uint8_t *data, size_t size) {
- ABitReader br(data + 4, 5);
-
- uint32_t user_identifier = br.getBits(32);
- uint8_t user_data_type = br.getBits(8);
-
- if (user_identifier == 'GA94' && user_data_type == 0x3) {
- return parseMPEGCCData(timeUs, data + 9, size - 9);
- }
-
- return false;
-}
-
-// returns true if a new CC track is found
-bool NuPlayer2::CCDecoder::parseMPEGCCData(int64_t timeUs, const uint8_t *data, size_t size) {
- bool trackAdded = false;
-
- // MPEG_cc_data()
- // ATSC A/53 Part 4: 6.2.3.1
- ABitReader br(data, size);
-
- if (br.numBitsLeft() <= 16) {
- return false;
- }
-
- br.skipBits(1);
- bool process_cc_data_flag = br.getBits(1);
- br.skipBits(1);
- size_t cc_count = br.getBits(5);
- br.skipBits(8);
-
- if (!process_cc_data_flag || 3 * 8 * cc_count >= br.numBitsLeft()) {
- return false;
- }
-
- sp<ABuffer> line21CCBuf = NULL;
-
- for (size_t i = 0; i < cc_count; ++i) {
- br.skipBits(5);
- bool cc_valid = br.getBits(1);
- uint8_t cc_type = br.getBits(2);
-
- if (cc_valid) {
- if (cc_type == 3) {
- if (mDTVCCPacket->size() > 0) {
- trackAdded |= parseDTVCCPacket(
- timeUs, mDTVCCPacket->data(), mDTVCCPacket->size());
- mDTVCCPacket->setRange(0, 0);
- }
- if (mDTVCCPacket->size() + 2 > mDTVCCPacket->capacity()) {
- return false;
- }
- memcpy(mDTVCCPacket->data() + mDTVCCPacket->size(), br.data(), 2);
- mDTVCCPacket->setRange(0, mDTVCCPacket->size() + 2);
- br.skipBits(16);
- } else if (mDTVCCPacket->size() > 0 && cc_type == 2) {
- if (mDTVCCPacket->size() + 2 > mDTVCCPacket->capacity()) {
- return false;
- }
- memcpy(mDTVCCPacket->data() + mDTVCCPacket->size(), br.data(), 2);
- mDTVCCPacket->setRange(0, mDTVCCPacket->size() + 2);
- br.skipBits(16);
- } else if (cc_type == 0 || cc_type == 1) {
- uint8_t cc_data_1 = br.getBits(8) & 0x7f;
- uint8_t cc_data_2 = br.getBits(8) & 0x7f;
-
- CCData cc(cc_type, cc_data_1, cc_data_2);
-
- if (isNullPad(&cc)) {
- continue;
- }
-
- size_t channel;
- if (cc.getChannel(&channel)) {
- mLine21Channels[cc_type] = channel;
-
- // create a new track if it does not exist.
- getTrackIndex(kTrackTypeCEA608, channel, &trackAdded);
- }
-
- if (isSelected() && mTracks[mSelectedTrack].mTrackType == kTrackTypeCEA608
- && mTracks[mSelectedTrack].mTrackChannel == mLine21Channels[cc_type]) {
- if (line21CCBuf == NULL) {
- line21CCBuf = new ABuffer((cc_count - i) * sizeof(CCData));
- line21CCBuf->setRange(0, 0);
- }
- if (line21CCBuf->size() + sizeof(cc) > line21CCBuf->capacity()) {
- return false;
- }
- memcpy(line21CCBuf->data() + line21CCBuf->size(), &cc, sizeof(cc));
- line21CCBuf->setRange(0, line21CCBuf->size() + sizeof(CCData));
- }
- } else {
- br.skipBits(16);
- }
- } else {
- if ((cc_type == 3 || cc_type == 2) && mDTVCCPacket->size() > 0) {
- trackAdded |= parseDTVCCPacket(timeUs, mDTVCCPacket->data(), mDTVCCPacket->size());
- mDTVCCPacket->setRange(0, 0);
- }
- br.skipBits(16);
- }
- }
-
- if (isSelected() && mTracks[mSelectedTrack].mTrackType == kTrackTypeCEA608
- && line21CCBuf != NULL && line21CCBuf->size() > 0) {
- mCCMap.add(timeUs, line21CCBuf);
- }
-
- return trackAdded;
-}
-
-// returns true if a new CC track is found
-bool NuPlayer2::CCDecoder::parseDTVCCPacket(int64_t timeUs, const uint8_t *data, size_t size) {
- // CEA-708B 5 DTVCC Packet Layer.
- ABitReader br(data, size);
- br.skipBits(2);
-
- size_t packet_size = br.getBits(6);
- if (packet_size == 0) packet_size = 64;
- packet_size *= 2;
-
- if (size != packet_size) {
- return false;
- }
-
- bool trackAdded = false;
-
- while (br.numBitsLeft() >= 16) {
- // CEA-708B Figure 5 and 6.
- uint8_t service_number = br.getBits(3);
- size_t block_size = br.getBits(5);
-
- if (service_number == 64) {
- br.skipBits(2);
- service_number = br.getBits(6);
-
- if (service_number < 64) {
- return trackAdded;
- }
- }
-
- if (br.numBitsLeft() < block_size * 8) {
- return trackAdded;
- }
-
- if (block_size > 0) {
- size_t trackIndex = getTrackIndex(kTrackTypeCEA708, service_number, &trackAdded);
- if (mSelectedTrack == (ssize_t)trackIndex) {
- sp<ABuffer> ccPacket = new ABuffer(block_size);
- if (ccPacket->capacity() == 0) {
- return false;
- }
- memcpy(ccPacket->data(), br.data(), block_size);
- mCCMap.add(timeUs, ccPacket);
- }
- }
- br.skipBits(block_size * 8);
- }
-
- return trackAdded;
-}
-
-// return the track index for a given type and channel.
-// if the track does not exist, creates a new one.
-size_t NuPlayer2::CCDecoder::getTrackIndex(
- int32_t trackType, size_t channel, bool *trackAdded) {
- CCTrack track(trackType, channel);
- ssize_t index = mTrackIndices.indexOfKey(track);
-
- if (index < 0) {
- // A new track is added.
- index = mTracks.size();
- mTrackIndices.add(track, index);
- mTracks.add(track);
- *trackAdded = true;
- return index;
- }
-
- return mTrackIndices.valueAt(index);
-}
-
-void NuPlayer2::CCDecoder::decode(const sp<ABuffer> &accessUnit) {
- if (extractFromMPEGUserData(accessUnit) || extractFromSEI(accessUnit)) {
- sp<AMessage> msg = mNotify->dup();
- msg->setInt32("what", kWhatTrackAdded);
- msg->post();
- }
- // TODO: extract CC from other sources
-}
-
-void NuPlayer2::CCDecoder::display(int64_t timeUs) {
- if (!isSelected()) {
- return;
- }
-
- ssize_t index = mCCMap.indexOfKey(timeUs);
- if (index < 0) {
- ALOGV("cc for timestamp %" PRId64 " not found", timeUs);
- return;
- }
-
- sp<ABuffer> ccBuf;
-
- if (index == 0) {
- ccBuf = mCCMap.valueAt(index);
- } else {
- size_t size = 0;
-
- for (ssize_t i = 0; i <= index; ++i) {
- size += mCCMap.valueAt(i)->size();
- }
-
- ccBuf = new ABuffer(size);
- ccBuf->setRange(0, 0);
-
- if (ccBuf->capacity() > 0) {
- for (ssize_t i = 0; i <= index; ++i) {
- sp<ABuffer> buf = mCCMap.valueAt(i);
- memcpy(ccBuf->data() + ccBuf->size(), buf->data(), buf->size());
- ccBuf->setRange(0, ccBuf->size() + buf->size());
- }
- }
- }
-
- if (ccBuf->size() > 0) {
-#if 0
- dumpBytePair(ccBuf);
-#endif
-
- ccBuf->meta()->setInt32(AMEDIAFORMAT_KEY_TRACK_INDEX, mSelectedTrack);
- ccBuf->meta()->setInt64("timeUs", timeUs);
- ccBuf->meta()->setInt64("durationUs", 0LL);
-
- sp<AMessage> msg = mNotify->dup();
- msg->setInt32("what", kWhatClosedCaptionData);
- msg->setBuffer("buffer", ccBuf);
- msg->post();
- }
-
- // remove all entries before timeUs
- mCCMap.removeItemsAt(0, index + 1);
-}
-
-void NuPlayer2::CCDecoder::flush() {
- mCCMap.clear();
- mDTVCCPacket->setRange(0, 0);
-}
-
-int32_t NuPlayer2::CCDecoder::CCTrack::compare(const NuPlayer2::CCDecoder::CCTrack& rhs) const {
- int32_t cmp = mTrackType - rhs.mTrackType;
- if (cmp != 0) return cmp;
- return mTrackChannel - rhs.mTrackChannel;
-}
-
-bool NuPlayer2::CCDecoder::CCTrack::operator<(const NuPlayer2::CCDecoder::CCTrack& rhs) const {
- return compare(rhs) < 0;
-}
-
-bool NuPlayer2::CCDecoder::CCTrack::operator==(const NuPlayer2::CCDecoder::CCTrack& rhs) const {
- return compare(rhs) == 0;
-}
-
-bool NuPlayer2::CCDecoder::CCTrack::operator!=(const NuPlayer2::CCDecoder::CCTrack& rhs) const {
- return compare(rhs) != 0;
-}
-
-} // namespace android
-
diff --git a/media/libmediaplayer2/nuplayer2/NuPlayer2CCDecoder.h b/media/libmediaplayer2/nuplayer2/NuPlayer2CCDecoder.h
deleted file mode 100644
index 97834d1..0000000
--- a/media/libmediaplayer2/nuplayer2/NuPlayer2CCDecoder.h
+++ /dev/null
@@ -1,98 +0,0 @@
-/*
- * Copyright 2017 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef NUPLAYER2_CCDECODER_H_
-
-#define NUPLAYER2_CCDECODER_H_
-
-#include "NuPlayer2.h"
-
-namespace android {
-
-struct NuPlayer2::CCDecoder : public RefBase {
- enum {
- kWhatClosedCaptionData,
- kWhatTrackAdded,
- };
-
- enum {
- kTrackTypeCEA608,
- kTrackTypeCEA708,
- };
-
- explicit CCDecoder(const sp<AMessage> ¬ify);
-
- size_t getTrackCount() const;
- sp<AMessage> getTrackInfo(size_t index) const;
- status_t selectTrack(size_t index, bool select);
- ssize_t getSelectedTrack(media_track_type type) const;
- bool isSelected() const;
- void decode(const sp<ABuffer> &accessUnit);
- void display(int64_t timeUs);
- void flush();
-
-private:
- // CC track identifier.
- struct CCTrack {
- CCTrack() : mTrackType(0), mTrackChannel(0) { }
-
- CCTrack(const int32_t trackType, const size_t trackChannel)
- : mTrackType(trackType), mTrackChannel(trackChannel) { }
-
- int32_t mTrackType;
- size_t mTrackChannel;
-
- // The ordering of CCTracks is to build a map of track to index.
- // It is necessary to find the index of the matched CCTrack when CC data comes.
- int compare(const NuPlayer2::CCDecoder::CCTrack& rhs) const;
- inline bool operator<(const NuPlayer2::CCDecoder::CCTrack& rhs) const;
- inline bool operator==(const NuPlayer2::CCDecoder::CCTrack& rhs) const;
- inline bool operator!=(const NuPlayer2::CCDecoder::CCTrack& rhs) const;
- };
-
- sp<AMessage> mNotify;
- KeyedVector<int64_t, sp<ABuffer> > mCCMap;
- ssize_t mSelectedTrack;
- KeyedVector<CCTrack, size_t> mTrackIndices;
- Vector<CCTrack> mTracks;
-
- // CEA-608 closed caption
- size_t mLine21Channels[2]; // The current channels of NTSC_CC_FIELD_{1, 2}
-
- // CEA-708 closed caption
- sp<ABuffer> mDTVCCPacket;
-
- bool isTrackValid(size_t index) const;
- size_t getTrackIndex(int32_t trackType, size_t channel, bool *trackAdded);
-
- // Extract from H.264 SEIs
- bool extractFromSEI(const sp<ABuffer> &accessUnit);
- bool parseSEINalUnit(int64_t timeUs, const uint8_t *data, size_t size);
-
- // Extract from MPEG user data
- bool extractFromMPEGUserData(const sp<ABuffer> &accessUnit);
- bool parseMPEGUserDataUnit(int64_t timeUs, const uint8_t *data, size_t size);
-
- // Extract CC tracks from MPEG_cc_data
- bool parseMPEGCCData(int64_t timeUs, const uint8_t *data, size_t size);
- bool parseDTVCCPacket(int64_t timeUs, const uint8_t *data, size_t size);
-
- DISALLOW_EVIL_CONSTRUCTORS(CCDecoder);
-};
-
-} // namespace android
-
-#endif // NUPLAYER2_CCDECODER_H_
diff --git a/media/libmediaplayer2/nuplayer2/NuPlayer2Decoder.cpp b/media/libmediaplayer2/nuplayer2/NuPlayer2Decoder.cpp
deleted file mode 100644
index 66bfae5..0000000
--- a/media/libmediaplayer2/nuplayer2/NuPlayer2Decoder.cpp
+++ /dev/null
@@ -1,1315 +0,0 @@
-/*
- * Copyright 2017 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "NuPlayer2Decoder"
-#include <utils/Log.h>
-#include <inttypes.h>
-
-#include <algorithm>
-
-#include "NuPlayer2CCDecoder.h"
-#include "NuPlayer2Decoder.h"
-#include "NuPlayer2Drm.h"
-#include "NuPlayer2Renderer.h"
-#include "NuPlayer2Source.h"
-
-#include <cutils/properties.h>
-#include <media/MediaBufferHolder.h>
-#include <media/MediaCodecBuffer.h>
-#include <media/NdkMediaCodec.h>
-#include <media/NdkWrapper.h>
-#include <media/stagefright/foundation/ABuffer.h>
-#include <media/stagefright/foundation/ADebug.h>
-#include <media/stagefright/foundation/AMessage.h>
-#include <media/stagefright/foundation/avc_utils.h>
-#include <media/stagefright/MediaBuffer.h>
-#include <media/stagefright/MediaDefs.h>
-#include <media/stagefright/MediaErrors.h>
-#include <media/stagefright/SurfaceUtils.h>
-
-#include <system/window.h>
-#include "ATSParser.h"
-
-namespace android {
-
-static float kDisplayRefreshingRate = 60.f; // TODO: get this from the display
-
-// The default total video frame rate of a stream when that info is not available from
-// the source.
-static float kDefaultVideoFrameRateTotal = 30.f;
-
-static inline bool getAudioDeepBufferSetting() {
- return property_get_bool("media.stagefright.audio.deep", false /* default_value */);
-}
-
-NuPlayer2::Decoder::Decoder(
- const sp<AMessage> ¬ify,
- const sp<Source> &source,
- pid_t pid,
- uid_t uid,
- const sp<Renderer> &renderer,
- const sp<ANativeWindowWrapper> &nww,
- const sp<CCDecoder> &ccDecoder)
- : DecoderBase(notify),
- mNativeWindow(nww),
- mSource(source),
- mRenderer(renderer),
- mCCDecoder(ccDecoder),
- mPid(pid),
- mUid(uid),
- mSkipRenderingUntilMediaTimeUs(-1LL),
- mNumFramesTotal(0LL),
- mNumInputFramesDropped(0LL),
- mNumOutputFramesDropped(0LL),
- mVideoWidth(0),
- mVideoHeight(0),
- mIsAudio(true),
- mIsVideoAVC(false),
- mIsSecure(false),
- mIsEncrypted(false),
- mIsEncryptedObservedEarlier(false),
- mFormatChangePending(false),
- mTimeChangePending(false),
- mFrameRateTotal(kDefaultVideoFrameRateTotal),
- mPlaybackSpeed(1.0f),
- mNumVideoTemporalLayerTotal(1), // decode all layers
- mNumVideoTemporalLayerAllowed(1),
- mCurrentMaxVideoTemporalLayerId(0),
- mResumePending(false),
- mComponentName("decoder") {
- mVideoTemporalLayerAggregateFps[0] = mFrameRateTotal;
-}
-
-NuPlayer2::Decoder::~Decoder() {
- // Need to stop looper first since mCodec could be accessed on the mDecoderLooper.
- stopLooper();
- if (mCodec != NULL) {
- mCodec->release();
- }
- releaseAndResetMediaBuffers();
-}
-
-sp<AMessage> NuPlayer2::Decoder::getStats() const {
- mStats->setInt64("frames-total", mNumFramesTotal);
- mStats->setInt64("frames-dropped-input", mNumInputFramesDropped);
- mStats->setInt64("frames-dropped-output", mNumOutputFramesDropped);
- mStats->setFloat("frame-rate-total", mFrameRateTotal);
-
- // i'm mutexed right now.
- // make our own copy, so we aren't victim to any later changes.
- sp<AMessage> copiedStats = mStats->dup();
- return copiedStats;
-}
-
-status_t NuPlayer2::Decoder::setVideoSurface(const sp<ANativeWindowWrapper> &nww) {
- if (nww == NULL || nww->getANativeWindow() == NULL
- || ADebug::isExperimentEnabled("legacy-setsurface")) {
- return BAD_VALUE;
- }
-
- sp<AMessage> msg = new AMessage(kWhatSetVideoSurface, this);
-
- msg->setObject("surface", nww);
- sp<AMessage> response;
- status_t err = msg->postAndAwaitResponse(&response);
- if (err == OK && response != NULL) {
- CHECK(response->findInt32("err", &err));
- }
- return err;
-}
-
-void NuPlayer2::Decoder::onMessageReceived(const sp<AMessage> &msg) {
- ALOGV("[%s] onMessage: %s", mComponentName.c_str(), msg->debugString().c_str());
-
- switch (msg->what()) {
- case kWhatCodecNotify:
- {
- int32_t cbID;
- CHECK(msg->findInt32("callbackID", &cbID));
-
- ALOGV("[%s] kWhatCodecNotify: cbID = %d, paused = %d",
- mIsAudio ? "audio" : "video", cbID, mPaused);
-
- if (mPaused) {
- break;
- }
-
- switch (cbID) {
- case AMediaCodecWrapper::CB_INPUT_AVAILABLE:
- {
- int32_t index;
- CHECK(msg->findInt32("index", &index));
-
- handleAnInputBuffer(index);
- break;
- }
-
- case AMediaCodecWrapper::CB_OUTPUT_AVAILABLE:
- {
- int32_t index;
- size_t offset;
- size_t size;
- int64_t timeUs;
- int32_t flags;
-
- CHECK(msg->findInt32("index", &index));
- CHECK(msg->findSize("offset", &offset));
- CHECK(msg->findSize("size", &size));
- CHECK(msg->findInt64("timeUs", &timeUs));
- CHECK(msg->findInt32("flags", &flags));
-
- handleAnOutputBuffer(index, offset, size, timeUs, flags);
- break;
- }
-
- case AMediaCodecWrapper::CB_OUTPUT_FORMAT_CHANGED:
- {
- sp<AMessage> format;
- CHECK(msg->findMessage("format", &format));
-
- handleOutputFormatChange(format);
- break;
- }
-
- case AMediaCodecWrapper::CB_ERROR:
- {
- status_t err;
- CHECK(msg->findInt32("err", &err));
- ALOGE("Decoder (%s) reported error : 0x%x",
- mIsAudio ? "audio" : "video", err);
-
- handleError(err);
- break;
- }
-
- default:
- {
- TRESPASS();
- break;
- }
- }
-
- break;
- }
-
- case kWhatRenderBuffer:
- {
- if (!isStaleReply(msg)) {
- onRenderBuffer(msg);
- }
- break;
- }
-
- case kWhatAudioOutputFormatChanged:
- {
- if (!isStaleReply(msg)) {
- status_t err;
- if (msg->findInt32("err", &err) && err != OK) {
- ALOGE("Renderer reported 0x%x when changing audio output format", err);
- handleError(err);
- }
- }
- break;
- }
-
- case kWhatSetVideoSurface:
- {
- sp<AReplyToken> replyID;
- CHECK(msg->senderAwaitsResponse(&replyID));
-
- sp<RefBase> obj;
- CHECK(msg->findObject("surface", &obj));
- sp<ANativeWindowWrapper> nww =
- static_cast<ANativeWindowWrapper *>(obj.get()); // non-null
- if (nww == NULL || nww->getANativeWindow() == NULL) {
- break;
- }
- int32_t err = INVALID_OPERATION;
- // NOTE: in practice mNativeWindow is always non-null,
- // but checking here for completeness
- if (mCodec != NULL
- && mNativeWindow != NULL && mNativeWindow->getANativeWindow() != NULL) {
- // TODO: once AwesomePlayer is removed, remove this automatic connecting
- // to the surface by MediaPlayerService.
- //
- // at this point MediaPlayer2Manager::client has already connected to the
- // surface, which MediaCodec does not expect
- err = native_window_api_disconnect(nww->getANativeWindow(),
- NATIVE_WINDOW_API_MEDIA);
- if (err == OK) {
- err = mCodec->setOutputSurface(nww);
- ALOGI_IF(err, "codec setOutputSurface returned: %d", err);
- if (err == OK) {
- // reconnect to the old surface as MPS::Client will expect to
- // be able to disconnect from it.
- (void)native_window_api_connect(mNativeWindow->getANativeWindow(),
- NATIVE_WINDOW_API_MEDIA);
-
- mNativeWindow = nww;
- }
- }
- if (err != OK) {
- // reconnect to the new surface on error as MPS::Client will expect to
- // be able to disconnect from it.
- (void)native_window_api_connect(nww->getANativeWindow(),
- NATIVE_WINDOW_API_MEDIA);
- }
- }
-
- sp<AMessage> response = new AMessage;
- response->setInt32("err", err);
- response->postReply(replyID);
- break;
- }
-
- case kWhatDrmReleaseCrypto:
- {
- ALOGV("kWhatDrmReleaseCrypto");
- onReleaseCrypto(msg);
- break;
- }
-
- default:
- DecoderBase::onMessageReceived(msg);
- break;
- }
-}
-
-void NuPlayer2::Decoder::onConfigure(const sp<AMessage> &format) {
- ALOGV("[%s] onConfigure (format=%s)", mComponentName.c_str(), format->debugString().c_str());
- CHECK(mCodec == NULL);
-
- mFormatChangePending = false;
- mTimeChangePending = false;
-
- ++mBufferGeneration;
-
- AString mime;
- CHECK(format->findString("mime", &mime));
-
- mIsAudio = !strncasecmp("audio/", mime.c_str(), 6);
- mIsVideoAVC = !strcasecmp(MEDIA_MIMETYPE_VIDEO_AVC, mime.c_str());
-
- mComponentName = mime;
- mComponentName.append(" decoder");
- ALOGV("[%s] onConfigure (nww=%p)", mComponentName.c_str(),
- (mNativeWindow == NULL ? NULL : mNativeWindow->getANativeWindow()));
-
- mCodec = AMediaCodecWrapper::CreateDecoderByType(mime);
- int32_t secure = 0;
- if (format->findInt32("secure", &secure) && secure != 0) {
- if (mCodec != NULL) {
- if (mCodec->getName(&mComponentName) == OK) {
- mComponentName.append(".secure");
- mCodec->release();
- ALOGI("[%s] creating", mComponentName.c_str());
- mCodec = AMediaCodecWrapper::CreateCodecByName(mComponentName);
- } else {
- mCodec = NULL;
- }
- }
- }
- if (mCodec == NULL) {
- ALOGE("Failed to create %s%s decoder",
- (secure ? "secure " : ""), mime.c_str());
- handleError(NO_INIT);
- return;
- }
- mIsSecure = secure;
-
- mCodec->getName(&mComponentName);
-
- status_t err;
- if (mNativeWindow != NULL && mNativeWindow->getANativeWindow() != NULL) {
- // disconnect from surface as MediaCodec will reconnect
- err = native_window_api_disconnect(mNativeWindow->getANativeWindow(),
- NATIVE_WINDOW_API_MEDIA);
- // We treat this as a warning, as this is a preparatory step.
- // Codec will try to connect to the surface, which is where
- // any error signaling will occur.
- ALOGW_IF(err != OK, "failed to disconnect from surface: %d", err);
- }
-
- // Modular DRM
- sp<RefBase> objCrypto;
- format->findObject("crypto", &objCrypto);
- sp<AMediaCryptoWrapper> crypto = static_cast<AMediaCryptoWrapper *>(objCrypto.get());
- // non-encrypted source won't have a crypto
- mIsEncrypted = (crypto != NULL);
- // configure is called once; still using OR in case the behavior changes.
- mIsEncryptedObservedEarlier = mIsEncryptedObservedEarlier || mIsEncrypted;
- ALOGV("onConfigure mCrypto: %p, mIsSecure: %d", crypto.get(), mIsSecure);
-
- err = mCodec->configure(
- AMediaFormatWrapper::Create(format),
- mNativeWindow,
- crypto,
- 0 /* flags */);
-
- if (err != OK) {
- ALOGE("Failed to configure [%s] decoder (err=%d)", mComponentName.c_str(), err);
- mCodec->release();
- mCodec.clear();
- handleError(err);
- return;
- }
- rememberCodecSpecificData(format);
-
- // the following should work in configured state
- sp<AMediaFormatWrapper> outputFormat = mCodec->getOutputFormat();
- if (outputFormat == NULL) {
- handleError(INVALID_OPERATION);
- return;
- }
- mInputFormat = mCodec->getInputFormat();
- if (mInputFormat == NULL) {
- handleError(INVALID_OPERATION);
- return;
- }
-
- mStats->setString("mime", mime.c_str());
- mStats->setString("component-name", mComponentName.c_str());
-
- if (!mIsAudio) {
- int32_t width, height;
- if (outputFormat->getInt32("width", &width)
- && outputFormat->getInt32("height", &height)) {
- mStats->setInt32("width", width);
- mStats->setInt32("height", height);
- }
- }
-
- sp<AMessage> reply = new AMessage(kWhatCodecNotify, this);
- mCodec->setCallback(reply);
-
- err = mCodec->start();
- if (err != OK) {
- ALOGE("Failed to start [%s] decoder (err=%d)", mComponentName.c_str(), err);
- mCodec->release();
- mCodec.clear();
- handleError(err);
- return;
- }
-
- releaseAndResetMediaBuffers();
-
- mPaused = false;
- mResumePending = false;
-}
-
-void NuPlayer2::Decoder::onSetParameters(const sp<AMessage> ¶ms) {
- bool needAdjustLayers = false;
- float frameRateTotal;
- if (params->findFloat("frame-rate-total", &frameRateTotal)
- && mFrameRateTotal != frameRateTotal) {
- needAdjustLayers = true;
- mFrameRateTotal = frameRateTotal;
- }
-
- int32_t numVideoTemporalLayerTotal;
- if (params->findInt32("temporal-layer-count", &numVideoTemporalLayerTotal)
- && numVideoTemporalLayerTotal >= 0
- && numVideoTemporalLayerTotal <= kMaxNumVideoTemporalLayers
- && mNumVideoTemporalLayerTotal != numVideoTemporalLayerTotal) {
- needAdjustLayers = true;
- mNumVideoTemporalLayerTotal = std::max(numVideoTemporalLayerTotal, 1);
- }
-
- if (needAdjustLayers && mNumVideoTemporalLayerTotal > 1) {
- // TODO: For now, layer fps is calculated for some specific architectures.
- // But it really should be extracted from the stream.
- mVideoTemporalLayerAggregateFps[0] =
- mFrameRateTotal / (float)(1LL << (mNumVideoTemporalLayerTotal - 1));
- for (int32_t i = 1; i < mNumVideoTemporalLayerTotal; ++i) {
- mVideoTemporalLayerAggregateFps[i] =
- mFrameRateTotal / (float)(1LL << (mNumVideoTemporalLayerTotal - i))
- + mVideoTemporalLayerAggregateFps[i - 1];
- }
- }
-
- float playbackSpeed;
- if (params->findFloat("playback-speed", &playbackSpeed)
- && mPlaybackSpeed != playbackSpeed) {
- needAdjustLayers = true;
- mPlaybackSpeed = playbackSpeed;
- }
-
- if (needAdjustLayers) {
- float decodeFrameRate = mFrameRateTotal;
- // enable temporal layering optimization only if we know the layering depth
- if (mNumVideoTemporalLayerTotal > 1) {
- int32_t layerId;
- for (layerId = 0; layerId < mNumVideoTemporalLayerTotal - 1; ++layerId) {
- if (mVideoTemporalLayerAggregateFps[layerId] * mPlaybackSpeed
- >= kDisplayRefreshingRate * 0.9) {
- break;
- }
- }
- mNumVideoTemporalLayerAllowed = layerId + 1;
- decodeFrameRate = mVideoTemporalLayerAggregateFps[layerId];
- }
- ALOGV("onSetParameters: allowed layers=%d, decodeFps=%g",
- mNumVideoTemporalLayerAllowed, decodeFrameRate);
-
- if (mCodec == NULL) {
- ALOGW("onSetParameters called before codec is created.");
- return;
- }
-
- sp<AMediaFormatWrapper> codecParams = new AMediaFormatWrapper();
- codecParams->setFloat("operating-rate", decodeFrameRate * mPlaybackSpeed);
- mCodec->setParameters(codecParams);
- }
-}
-
-void NuPlayer2::Decoder::onSetRenderer(const sp<Renderer> &renderer) {
- mRenderer = renderer;
-}
-
-void NuPlayer2::Decoder::onResume(bool notifyComplete) {
- mPaused = false;
-
- if (notifyComplete) {
- mResumePending = true;
- }
-
- if (mCodec == NULL) {
- ALOGE("[%s] onResume without a valid codec", mComponentName.c_str());
- handleError(NO_INIT);
- return;
- }
- mCodec->start();
-}
-
-void NuPlayer2::Decoder::doFlush(bool notifyComplete) {
- if (mCCDecoder != NULL) {
- mCCDecoder->flush();
- }
-
- if (mRenderer != NULL) {
- mRenderer->flush(mIsAudio, notifyComplete);
- mRenderer->signalTimeDiscontinuity();
- }
-
- status_t err = OK;
- if (mCodec != NULL) {
- err = mCodec->flush();
- mCSDsToSubmit = mCSDsForCurrentFormat; // copy operator
- ++mBufferGeneration;
- }
-
- if (err != OK) {
- ALOGE("failed to flush [%s] (err=%d)", mComponentName.c_str(), err);
- handleError(err);
- // finish with posting kWhatFlushCompleted.
- // we attempt to release the buffers even if flush fails.
- }
- releaseAndResetMediaBuffers();
- mPaused = true;
-}
-
-
-void NuPlayer2::Decoder::onFlush() {
- doFlush(true);
-
- if (isDiscontinuityPending()) {
- // This could happen if the client starts seeking/shutdown
- // after we queued an EOS for discontinuities.
- // We can consider discontinuity handled.
- finishHandleDiscontinuity(false /* flushOnTimeChange */);
- }
-
- sp<AMessage> notify = mNotify->dup();
- notify->setInt32("what", kWhatFlushCompleted);
- notify->post();
-}
-
-void NuPlayer2::Decoder::onShutdown(bool notifyComplete) {
- status_t err = OK;
-
- // if there is a pending resume request, notify complete now
- notifyResumeCompleteIfNecessary();
-
- if (mCodec != NULL) {
- err = mCodec->release();
- mCodec = NULL;
- ++mBufferGeneration;
-
- if (mNativeWindow != NULL && mNativeWindow->getANativeWindow() != NULL) {
- // reconnect to surface as MediaCodec disconnected from it
- status_t error = native_window_api_connect(mNativeWindow->getANativeWindow(),
- NATIVE_WINDOW_API_MEDIA);
- ALOGW_IF(error != NO_ERROR,
- "[%s] failed to connect to native window, error=%d",
- mComponentName.c_str(), error);
- }
- mComponentName = "decoder";
- }
-
- releaseAndResetMediaBuffers();
-
- if (err != OK) {
- ALOGE("failed to release [%s] (err=%d)", mComponentName.c_str(), err);
- handleError(err);
- // finish with posting kWhatShutdownCompleted.
- }
-
- if (notifyComplete) {
- sp<AMessage> notify = mNotify->dup();
- notify->setInt32("what", kWhatShutdownCompleted);
- notify->post();
- mPaused = true;
- }
-}
-
-/*
- * returns true if we should request more data
- */
-bool NuPlayer2::Decoder::doRequestBuffers() {
- if (isDiscontinuityPending()) {
- return false;
- }
- status_t err = OK;
- while (err == OK && !mDequeuedInputBuffers.empty()) {
- size_t bufferIx = *mDequeuedInputBuffers.begin();
- sp<AMessage> msg = new AMessage();
- msg->setSize("buffer-ix", bufferIx);
- err = fetchInputData(msg);
- if (err != OK && err != ERROR_END_OF_STREAM) {
- // if EOS, need to queue EOS buffer
- break;
- }
- mDequeuedInputBuffers.erase(mDequeuedInputBuffers.begin());
-
- if (!mPendingInputMessages.empty()
- || !onInputBufferFetched(msg)) {
- mPendingInputMessages.push_back(msg);
- }
- }
-
- return err == -EWOULDBLOCK
- && mSource->feedMoreTSData() == OK;
-}
-
-void NuPlayer2::Decoder::handleError(int32_t err)
-{
- // We cannot immediately release the codec due to buffers still outstanding
- // in the renderer. We signal to the player the error so it can shutdown/release the
- // decoder after flushing and increment the generation to discard unnecessary messages.
-
- ++mBufferGeneration;
-
- sp<AMessage> notify = mNotify->dup();
- notify->setInt32("what", kWhatError);
- notify->setInt32("err", err);
- notify->post();
-}
-
-status_t NuPlayer2::Decoder::releaseCrypto()
-{
- ALOGV("releaseCrypto");
-
- sp<AMessage> msg = new AMessage(kWhatDrmReleaseCrypto, this);
-
- sp<AMessage> response;
- status_t status = msg->postAndAwaitResponse(&response);
- if (status == OK && response != NULL) {
- CHECK(response->findInt32("status", &status));
- ALOGV("releaseCrypto ret: %d ", status);
- } else {
- ALOGE("releaseCrypto err: %d", status);
- }
-
- return status;
-}
-
-void NuPlayer2::Decoder::onReleaseCrypto(const sp<AMessage>& msg)
-{
- status_t status = INVALID_OPERATION;
- if (mCodec != NULL) {
- status = mCodec->releaseCrypto();
- } else {
- // returning OK if the codec has been already released
- status = OK;
- ALOGE("onReleaseCrypto No mCodec. err: %d", status);
- }
-
- sp<AMessage> response = new AMessage;
- response->setInt32("status", status);
- // Clearing the state as it's tied to crypto. mIsEncryptedObservedEarlier is sticky though
- // and lasts for the lifetime of this codec. See its use in fetchInputData.
- mIsEncrypted = false;
-
- sp<AReplyToken> replyID;
- CHECK(msg->senderAwaitsResponse(&replyID));
- response->postReply(replyID);
-}
-
-bool NuPlayer2::Decoder::handleAnInputBuffer(size_t index) {
- if (isDiscontinuityPending()) {
- return false;
- }
-
- if (mCodec == NULL) {
- ALOGE("[%s] handleAnInputBuffer without a valid codec", mComponentName.c_str());
- handleError(NO_INIT);
- return false;
- }
-
- size_t bufferSize = 0;
- uint8_t *bufferBase = mCodec->getInputBuffer(index, &bufferSize);
-
- if (bufferBase == NULL) {
- ALOGE("[%s] handleAnInputBuffer, failed to get input buffer", mComponentName.c_str());
- handleError(UNKNOWN_ERROR);
- return false;
- }
-
- sp<MediaCodecBuffer> buffer =
- new MediaCodecBuffer(NULL /* format */, new ABuffer(bufferBase, bufferSize));
-
- if (index >= mInputBuffers.size()) {
- for (size_t i = mInputBuffers.size(); i <= index; ++i) {
- mInputBuffers.add();
- mMediaBuffers.add();
- mInputBufferIsDequeued.add();
- mMediaBuffers.editItemAt(i) = NULL;
- mInputBufferIsDequeued.editItemAt(i) = false;
- }
- }
- mInputBuffers.editItemAt(index) = buffer;
-
- //CHECK_LT(bufferIx, mInputBuffers.size());
-
- if (mMediaBuffers[index] != NULL) {
- mMediaBuffers[index]->release();
- mMediaBuffers.editItemAt(index) = NULL;
- }
- mInputBufferIsDequeued.editItemAt(index) = true;
-
- if (!mCSDsToSubmit.isEmpty()) {
- sp<AMessage> msg = new AMessage();
- msg->setSize("buffer-ix", index);
-
- sp<ABuffer> buffer = mCSDsToSubmit.itemAt(0);
- ALOGI("[%s] resubmitting CSD", mComponentName.c_str());
- msg->setBuffer("buffer", buffer);
- mCSDsToSubmit.removeAt(0);
- if (!onInputBufferFetched(msg)) {
- handleError(UNKNOWN_ERROR);
- return false;
- }
- return true;
- }
-
- while (!mPendingInputMessages.empty()) {
- sp<AMessage> msg = *mPendingInputMessages.begin();
- if (!onInputBufferFetched(msg)) {
- break;
- }
- mPendingInputMessages.erase(mPendingInputMessages.begin());
- }
-
- if (!mInputBufferIsDequeued.editItemAt(index)) {
- return true;
- }
-
- mDequeuedInputBuffers.push_back(index);
-
- onRequestInputBuffers();
- return true;
-}
-
-bool NuPlayer2::Decoder::handleAnOutputBuffer(
- size_t index,
- size_t offset,
- size_t size,
- int64_t timeUs,
- int32_t flags) {
- if (mCodec == NULL) {
- ALOGE("[%s] handleAnOutputBuffer without a valid codec", mComponentName.c_str());
- handleError(NO_INIT);
- return false;
- }
-
-// CHECK_LT(bufferIx, mOutputBuffers.size());
-
- size_t bufferSize = 0;
- uint8_t *bufferBase = mCodec->getOutputBuffer(index, &bufferSize);
-
- if (bufferBase == NULL) {
- ALOGE("[%s] handleAnOutputBuffer, failed to get output buffer", mComponentName.c_str());
- handleError(UNKNOWN_ERROR);
- return false;
- }
-
- sp<MediaCodecBuffer> buffer =
- new MediaCodecBuffer(NULL /* format */, new ABuffer(bufferBase, bufferSize));
-
- if (index >= mOutputBuffers.size()) {
- for (size_t i = mOutputBuffers.size(); i <= index; ++i) {
- mOutputBuffers.add();
- }
- }
-
- mOutputBuffers.editItemAt(index) = buffer;
-
- buffer->setRange(offset, size);
- buffer->meta()->clear();
- buffer->meta()->setInt64("timeUs", timeUs);
-
- bool eos = flags & AMEDIACODEC_BUFFER_FLAG_END_OF_STREAM;
- // we do not expect CODECCONFIG or SYNCFRAME for decoder
-
- sp<AMessage> reply = new AMessage(kWhatRenderBuffer, this);
- reply->setSize("buffer-ix", index);
- reply->setInt32("generation", mBufferGeneration);
-
- if (eos) {
- ALOGI("[%s] saw output EOS", mIsAudio ? "audio" : "video");
-
- buffer->meta()->setInt32("eos", true);
- reply->setInt32("eos", true);
- }
-
- mNumFramesTotal += !mIsAudio;
-
- if (mSkipRenderingUntilMediaTimeUs >= 0) {
- if (timeUs < mSkipRenderingUntilMediaTimeUs) {
- ALOGV("[%s] dropping buffer at time %lld as requested.",
- mComponentName.c_str(), (long long)timeUs);
-
- reply->post();
- if (eos) {
- notifyResumeCompleteIfNecessary();
- if (mRenderer != NULL && !isDiscontinuityPending()) {
- mRenderer->queueEOS(mIsAudio, ERROR_END_OF_STREAM);
- }
- }
- return true;
- }
-
- mSkipRenderingUntilMediaTimeUs = -1;
- }
-
- // wait until 1st frame comes out to signal resume complete
- notifyResumeCompleteIfNecessary();
-
- if (mRenderer != NULL) {
- // send the buffer to renderer.
- mRenderer->queueBuffer(mIsAudio, buffer, reply);
- if (eos && !isDiscontinuityPending()) {
- mRenderer->queueEOS(mIsAudio, ERROR_END_OF_STREAM);
- }
- }
-
- return true;
-}
-
-void NuPlayer2::Decoder::handleOutputFormatChange(const sp<AMessage> &format) {
- if (!mIsAudio) {
- int32_t width, height;
- if (format->findInt32("width", &width)
- && format->findInt32("height", &height)) {
- mStats->setInt32("width", width);
- mStats->setInt32("height", height);
- }
- sp<AMessage> notify = mNotify->dup();
- notify->setInt32("what", kWhatVideoSizeChanged);
- notify->setMessage("format", format);
- notify->post();
- } else if (mRenderer != NULL) {
- uint32_t flags;
- int64_t durationUs;
- bool hasVideo = (mSource->getFormat(false /* audio */) != NULL);
- if (getAudioDeepBufferSetting() // override regardless of source duration
- || (mSource->getDuration(&durationUs) == OK
- && durationUs > AUDIO_SINK_MIN_DEEP_BUFFER_DURATION_US)) {
- flags = AUDIO_OUTPUT_FLAG_DEEP_BUFFER;
- } else {
- flags = AUDIO_OUTPUT_FLAG_NONE;
- }
-
- sp<AMessage> reply = new AMessage(kWhatAudioOutputFormatChanged, this);
- reply->setInt32("generation", mBufferGeneration);
- mRenderer->changeAudioFormat(
- format, false /* offloadOnly */, hasVideo,
- flags, mSource->isStreaming(), reply);
- }
-}
-
-void NuPlayer2::Decoder::releaseAndResetMediaBuffers() {
- for (size_t i = 0; i < mMediaBuffers.size(); i++) {
- if (mMediaBuffers[i] != NULL) {
- mMediaBuffers[i]->release();
- mMediaBuffers.editItemAt(i) = NULL;
- }
- }
- mMediaBuffers.resize(mInputBuffers.size());
- for (size_t i = 0; i < mMediaBuffers.size(); i++) {
- mMediaBuffers.editItemAt(i) = NULL;
- }
- mInputBufferIsDequeued.clear();
- mInputBufferIsDequeued.resize(mInputBuffers.size());
- for (size_t i = 0; i < mInputBufferIsDequeued.size(); i++) {
- mInputBufferIsDequeued.editItemAt(i) = false;
- }
-
- mPendingInputMessages.clear();
- mDequeuedInputBuffers.clear();
- mSkipRenderingUntilMediaTimeUs = -1;
-}
-
-bool NuPlayer2::Decoder::isStaleReply(const sp<AMessage> &msg) {
- int32_t generation;
- CHECK(msg->findInt32("generation", &generation));
- return generation != mBufferGeneration;
-}
-
-status_t NuPlayer2::Decoder::fetchInputData(sp<AMessage> &reply) {
- sp<ABuffer> accessUnit;
- bool dropAccessUnit = true;
- do {
- status_t err = mSource->dequeueAccessUnit(mIsAudio, &accessUnit);
-
- if (err == -EWOULDBLOCK) {
- return err;
- } else if (err != OK) {
- if (err == INFO_DISCONTINUITY) {
- int32_t type;
- CHECK(accessUnit->meta()->findInt32("discontinuity", &type));
-
- bool formatChange =
- (mIsAudio &&
- (type & ATSParser::DISCONTINUITY_AUDIO_FORMAT))
- || (!mIsAudio &&
- (type & ATSParser::DISCONTINUITY_VIDEO_FORMAT));
-
- bool timeChange = (type & ATSParser::DISCONTINUITY_TIME) != 0;
-
- ALOGI("%s discontinuity (format=%d, time=%d)",
- mIsAudio ? "audio" : "video", formatChange, timeChange);
-
- bool seamlessFormatChange = false;
- sp<AMessage> newFormat = mSource->getFormat(mIsAudio);
- if (formatChange) {
- seamlessFormatChange =
- supportsSeamlessFormatChange(newFormat);
- // treat seamless format change separately
- formatChange = !seamlessFormatChange;
- }
-
- // For format or time change, return EOS to queue EOS input,
- // then wait for EOS on output.
- if (formatChange /* not seamless */) {
- mFormatChangePending = true;
- err = ERROR_END_OF_STREAM;
- } else if (timeChange) {
- rememberCodecSpecificData(newFormat);
- mTimeChangePending = true;
- err = ERROR_END_OF_STREAM;
- } else if (seamlessFormatChange) {
- // reuse existing decoder and don't flush
- rememberCodecSpecificData(newFormat);
- continue;
- } else {
- // This stream is unaffected by the discontinuity
- return -EWOULDBLOCK;
- }
- }
-
- // reply should only be returned without a buffer set
- // when there is an error (including EOS)
- CHECK(err != OK);
-
- reply->setInt32("err", err);
- return ERROR_END_OF_STREAM;
- }
-
- dropAccessUnit = false;
- if (!mIsAudio && !mIsEncrypted) {
- // Extra safeguard if higher-level behavior changes. Otherwise, not required now.
- // Preventing the buffer from being processed (and sent to codec) if this is a later
- // round of playback but this time without prepareDrm. Or if there is a race between
- // stop (which is not blocking) and releaseDrm allowing buffers being processed after
- // Crypto has been released (GenericSource currently prevents this race though).
- // Particularly doing this check before IsAVCReferenceFrame call to prevent parsing
- // of encrypted data.
- if (mIsEncryptedObservedEarlier) {
- ALOGE("fetchInputData: mismatched mIsEncrypted/mIsEncryptedObservedEarlier (0/1)");
-
- return INVALID_OPERATION;
- }
-
- int32_t layerId = 0;
- bool haveLayerId = accessUnit->meta()->findInt32("temporal-layer-id", &layerId);
- if (mRenderer->getVideoLateByUs() > 100000LL
- && mIsVideoAVC
- && !IsAVCReferenceFrame(accessUnit)) {
- dropAccessUnit = true;
- } else if (haveLayerId && mNumVideoTemporalLayerTotal > 1) {
- // Add only one layer each time.
- if (layerId > mCurrentMaxVideoTemporalLayerId + 1
- || layerId >= mNumVideoTemporalLayerAllowed) {
- dropAccessUnit = true;
- ALOGV("dropping layer(%d), speed=%g, allowed layer count=%d, max layerId=%d",
- layerId, mPlaybackSpeed, mNumVideoTemporalLayerAllowed,
- mCurrentMaxVideoTemporalLayerId);
- } else if (layerId > mCurrentMaxVideoTemporalLayerId) {
- mCurrentMaxVideoTemporalLayerId = layerId;
- } else if (layerId == 0 && mNumVideoTemporalLayerTotal > 1
- && IsIDR(accessUnit->data(), accessUnit->size())) {
- mCurrentMaxVideoTemporalLayerId = mNumVideoTemporalLayerTotal - 1;
- }
- }
- if (dropAccessUnit) {
- if (layerId <= mCurrentMaxVideoTemporalLayerId && layerId > 0) {
- mCurrentMaxVideoTemporalLayerId = layerId - 1;
- }
- ++mNumInputFramesDropped;
- }
- }
- } while (dropAccessUnit);
-
- // ALOGV("returned a valid buffer of %s data", mIsAudio ? "mIsAudio" : "video");
-#if 0
- int64_t mediaTimeUs;
- CHECK(accessUnit->meta()->findInt64("timeUs", &mediaTimeUs));
- ALOGV("[%s] feeding input buffer at media time %.3f",
- mIsAudio ? "audio" : "video",
- mediaTimeUs / 1E6);
-#endif
-
- if (mCCDecoder != NULL) {
- mCCDecoder->decode(accessUnit);
- }
-
- reply->setBuffer("buffer", accessUnit);
-
- return OK;
-}
-
-bool NuPlayer2::Decoder::onInputBufferFetched(const sp<AMessage> &msg) {
- if (mCodec == NULL) {
- ALOGE("[%s] onInputBufferFetched without a valid codec", mComponentName.c_str());
- handleError(NO_INIT);
- return false;
- }
-
- size_t bufferIx;
- CHECK(msg->findSize("buffer-ix", &bufferIx));
- CHECK_LT(bufferIx, mInputBuffers.size());
- sp<MediaCodecBuffer> codecBuffer = mInputBuffers[bufferIx];
-
- sp<ABuffer> buffer;
- bool hasBuffer = msg->findBuffer("buffer", &buffer);
- bool needsCopy = true;
-
- if (buffer == NULL /* includes !hasBuffer */) {
- int32_t streamErr = ERROR_END_OF_STREAM;
- CHECK(msg->findInt32("err", &streamErr) || !hasBuffer);
-
- CHECK(streamErr != OK);
-
- // attempt to queue EOS
- status_t err = mCodec->queueInputBuffer(
- bufferIx,
- 0,
- 0,
- 0,
- AMEDIACODEC_BUFFER_FLAG_END_OF_STREAM);
- if (err == OK) {
- mInputBufferIsDequeued.editItemAt(bufferIx) = false;
- } else if (streamErr == ERROR_END_OF_STREAM) {
- streamErr = err;
- // err will not be ERROR_END_OF_STREAM
- }
-
- if (streamErr != ERROR_END_OF_STREAM) {
- ALOGE("Stream error for [%s] (err=%d), EOS %s queued",
- mComponentName.c_str(),
- streamErr,
- err == OK ? "successfully" : "unsuccessfully");
- handleError(streamErr);
- }
- } else {
- sp<AMessage> extra;
- if (buffer->meta()->findMessage("extra", &extra) && extra != NULL) {
- int64_t resumeAtMediaTimeUs;
- if (extra->findInt64(
- "resume-at-mediaTimeUs", &resumeAtMediaTimeUs)) {
- ALOGI("[%s] suppressing rendering until %lld us",
- mComponentName.c_str(), (long long)resumeAtMediaTimeUs);
- mSkipRenderingUntilMediaTimeUs = resumeAtMediaTimeUs;
- }
- }
-
- int64_t timeUs = 0;
- uint32_t flags = 0;
- CHECK(buffer->meta()->findInt64("timeUs", &timeUs));
-
- int32_t eos, csd;
- // we do not expect SYNCFRAME for decoder
- if (buffer->meta()->findInt32("eos", &eos) && eos) {
- flags |= AMEDIACODEC_BUFFER_FLAG_END_OF_STREAM;
- } else if (buffer->meta()->findInt32("csd", &csd) && csd) {
- flags |= AMEDIACODEC_BUFFER_FLAG_CODEC_CONFIG;
- }
-
- // Modular DRM
- MediaBufferBase *mediaBuf = NULL;
- sp<AMediaCodecCryptoInfoWrapper> cryptInfo;
-
- // copy into codec buffer
- if (needsCopy) {
- if (buffer->size() > codecBuffer->capacity()) {
- handleError(ERROR_BUFFER_TOO_SMALL);
- mDequeuedInputBuffers.push_back(bufferIx);
- return false;
- }
-
- if (buffer->data() != NULL) {
- codecBuffer->setRange(0, buffer->size());
- memcpy(codecBuffer->data(), buffer->data(), buffer->size());
- } else { // No buffer->data()
- //Modular DRM
- sp<RefBase> holder;
- if (buffer->meta()->findObject("mediaBufferHolder", &holder)) {
- mediaBuf = (holder != nullptr) ?
- static_cast<MediaBufferHolder*>(holder.get())->mediaBuffer() : nullptr;
- }
- if (mediaBuf != NULL) {
- if (mediaBuf->size() > codecBuffer->capacity()) {
- handleError(ERROR_BUFFER_TOO_SMALL);
- mDequeuedInputBuffers.push_back(bufferIx);
- return false;
- }
-
- codecBuffer->setRange(0, mediaBuf->size());
- memcpy(codecBuffer->data(), mediaBuf->data(), mediaBuf->size());
-
- MetaDataBase &meta_data = mediaBuf->meta_data();
- cryptInfo = AMediaCodecCryptoInfoWrapper::Create(meta_data);
- } else { // No mediaBuf
- ALOGE("onInputBufferFetched: buffer->data()/mediaBuf are NULL for %p",
- buffer.get());
- handleError(UNKNOWN_ERROR);
- return false;
- }
- } // buffer->data()
- } // needsCopy
-
- sp<RefBase> cryptInfoObj;
- if (buffer->meta()->findObject("cryptInfo", &cryptInfoObj)) {
- cryptInfo = static_cast<AMediaCodecCryptoInfoWrapper *>(cryptInfoObj.get());
- }
-
- status_t err;
- if (cryptInfo != NULL) {
- err = mCodec->queueSecureInputBuffer(
- bufferIx,
- codecBuffer->offset(),
- cryptInfo,
- timeUs,
- flags);
- // synchronous call so done with cryptInfo here
- } else {
- err = mCodec->queueInputBuffer(
- bufferIx,
- codecBuffer->offset(),
- codecBuffer->size(),
- timeUs,
- flags);
- } // no cryptInfo
-
- if (err != OK) {
- ALOGE("onInputBufferFetched: queue%sInputBuffer failed for [%s] (err=%d)",
- (cryptInfo != NULL ? "Secure" : ""),
- mComponentName.c_str(), err);
- handleError(err);
- } else {
- mInputBufferIsDequeued.editItemAt(bufferIx) = false;
- }
-
- } // buffer != NULL
- return true;
-}
-
-void NuPlayer2::Decoder::onRenderBuffer(const sp<AMessage> &msg) {
- status_t err;
- int32_t render;
- size_t bufferIx;
- int32_t eos;
- CHECK(msg->findSize("buffer-ix", &bufferIx));
-
- if (!mIsAudio) {
- int64_t timeUs;
- sp<MediaCodecBuffer> buffer = mOutputBuffers[bufferIx];
- buffer->meta()->findInt64("timeUs", &timeUs);
-
- if (mCCDecoder != NULL && mCCDecoder->isSelected()) {
- mCCDecoder->display(timeUs);
- }
- }
-
- if (mCodec == NULL) {
- err = NO_INIT;
- } else if (msg->findInt32("render", &render) && render) {
- int64_t timestampNs;
- CHECK(msg->findInt64("timestampNs", ×tampNs));
- err = mCodec->releaseOutputBufferAtTime(bufferIx, timestampNs);
- } else {
- mNumOutputFramesDropped += !mIsAudio;
- err = mCodec->releaseOutputBuffer(bufferIx, false /* render */);
- }
- if (err != OK) {
- ALOGE("failed to release output buffer for [%s] (err=%d)",
- mComponentName.c_str(), err);
- handleError(err);
- }
- if (msg->findInt32("eos", &eos) && eos
- && isDiscontinuityPending()) {
- finishHandleDiscontinuity(true /* flushOnTimeChange */);
- }
-}
-
-bool NuPlayer2::Decoder::isDiscontinuityPending() const {
- return mFormatChangePending || mTimeChangePending;
-}
-
-void NuPlayer2::Decoder::finishHandleDiscontinuity(bool flushOnTimeChange) {
- ALOGV("finishHandleDiscontinuity: format %d, time %d, flush %d",
- mFormatChangePending, mTimeChangePending, flushOnTimeChange);
-
- // If we have format change, pause and wait to be killed;
- // If we have time change only, flush and restart fetching.
-
- if (mFormatChangePending) {
- mPaused = true;
- } else if (mTimeChangePending) {
- if (flushOnTimeChange) {
- doFlush(false /* notifyComplete */);
- signalResume(false /* notifyComplete */);
- }
- }
-
- // Notify NuPlayer2 to either shutdown decoder, or rescan sources
- sp<AMessage> msg = mNotify->dup();
- msg->setInt32("what", kWhatInputDiscontinuity);
- msg->setInt32("formatChange", mFormatChangePending);
- msg->post();
-
- mFormatChangePending = false;
- mTimeChangePending = false;
-}
-
-bool NuPlayer2::Decoder::supportsSeamlessAudioFormatChange(
- const sp<AMessage> &targetFormat) const {
- if (targetFormat == NULL) {
- return true;
- }
-
- AString mime;
- if (!targetFormat->findString("mime", &mime)) {
- return false;
- }
-
- if (!strcasecmp(mime.c_str(), MEDIA_MIMETYPE_AUDIO_AAC)) {
- // field-by-field comparison
- const char * keys[] = { "channel-count", "sample-rate", "is-adts" };
- for (unsigned int i = 0; i < sizeof(keys) / sizeof(keys[0]); i++) {
- int32_t oldVal, newVal;
- if (!mInputFormat->getInt32(keys[i], &oldVal) ||
- !targetFormat->findInt32(keys[i], &newVal) ||
- oldVal != newVal) {
- return false;
- }
- }
-
- sp<ABuffer> newBuf;
- uint8_t *oldBufData = NULL;
- size_t oldBufSize = 0;
- if (mInputFormat->getBuffer("csd-0", (void**)&oldBufData, &oldBufSize) &&
- targetFormat->findBuffer("csd-0", &newBuf)) {
- if (oldBufSize != newBuf->size()) {
- return false;
- }
- return !memcmp(oldBufData, newBuf->data(), oldBufSize);
- }
- }
- return false;
-}
-
-bool NuPlayer2::Decoder::supportsSeamlessFormatChange(const sp<AMessage> &targetFormat) const {
- if (mInputFormat == NULL) {
- return false;
- }
-
- if (targetFormat == NULL) {
- return true;
- }
-
- AString oldMime, newMime;
- if (!mInputFormat->getString("mime", &oldMime)
- || !targetFormat->findString("mime", &newMime)
- || !(oldMime == newMime)) {
- return false;
- }
-
- bool audio = !strncasecmp(oldMime.c_str(), "audio/", strlen("audio/"));
- bool seamless;
- if (audio) {
- seamless = supportsSeamlessAudioFormatChange(targetFormat);
- } else {
- int32_t isAdaptive;
- seamless = (mCodec != NULL &&
- mInputFormat->getInt32("adaptive-playback", &isAdaptive) &&
- isAdaptive);
- }
-
- ALOGV("%s seamless support for %s", seamless ? "yes" : "no", oldMime.c_str());
- return seamless;
-}
-
-void NuPlayer2::Decoder::rememberCodecSpecificData(const sp<AMessage> &format) {
- if (format == NULL) {
- return;
- }
- mCSDsForCurrentFormat.clear();
- for (int32_t i = 0; ; ++i) {
- AString tag = "csd-";
- tag.append(i);
- sp<ABuffer> buffer;
- if (!format->findBuffer(tag.c_str(), &buffer)) {
- break;
- }
- mCSDsForCurrentFormat.push(buffer);
- }
-}
-
-void NuPlayer2::Decoder::notifyResumeCompleteIfNecessary() {
- if (mResumePending) {
- mResumePending = false;
-
- sp<AMessage> notify = mNotify->dup();
- notify->setInt32("what", kWhatResumeCompleted);
- notify->post();
- }
-}
-
-} // namespace android
-
diff --git a/media/libmediaplayer2/nuplayer2/NuPlayer2Decoder.h b/media/libmediaplayer2/nuplayer2/NuPlayer2Decoder.h
deleted file mode 100644
index fdfb10e..0000000
--- a/media/libmediaplayer2/nuplayer2/NuPlayer2Decoder.h
+++ /dev/null
@@ -1,150 +0,0 @@
-/*
- * Copyright 2017 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef NUPLAYER2_DECODER_H_
-#define NUPLAYER2_DECODER_H_
-
-#include "NuPlayer2.h"
-
-#include "NuPlayer2DecoderBase.h"
-
-namespace android {
-
-class MediaCodecBuffer;
-
-struct AMediaCodecWrapper;
-struct AMediaFormatWrapper;
-
-struct NuPlayer2::Decoder : public DecoderBase {
- Decoder(const sp<AMessage> ¬ify,
- const sp<Source> &source,
- pid_t pid,
- uid_t uid,
- const sp<Renderer> &renderer = NULL,
- const sp<ANativeWindowWrapper> &nww = NULL,
- const sp<CCDecoder> &ccDecoder = NULL);
-
- virtual sp<AMessage> getStats() const;
-
- // sets the output surface of video decoders.
- virtual status_t setVideoSurface(const sp<ANativeWindowWrapper> &nww);
-
- virtual status_t releaseCrypto();
-
-protected:
- virtual ~Decoder();
-
- virtual void onMessageReceived(const sp<AMessage> &msg);
-
- virtual void onConfigure(const sp<AMessage> &format);
- virtual void onSetParameters(const sp<AMessage> ¶ms);
- virtual void onSetRenderer(const sp<Renderer> &renderer);
- virtual void onResume(bool notifyComplete);
- virtual void onFlush();
- virtual void onShutdown(bool notifyComplete);
- virtual bool doRequestBuffers();
-
-private:
- enum {
- kWhatCodecNotify = 'cdcN',
- kWhatRenderBuffer = 'rndr',
- kWhatSetVideoSurface = 'sSur',
- kWhatAudioOutputFormatChanged = 'aofc',
- kWhatDrmReleaseCrypto = 'rDrm',
- };
-
- enum {
- kMaxNumVideoTemporalLayers = 32,
- };
-
- sp<ANativeWindowWrapper> mNativeWindow;
-
- sp<Source> mSource;
- sp<Renderer> mRenderer;
- sp<CCDecoder> mCCDecoder;
-
- sp<AMediaFormatWrapper> mInputFormat;
- sp<AMediaCodecWrapper> mCodec;
-
- List<sp<AMessage> > mPendingInputMessages;
-
- Vector<sp<MediaCodecBuffer> > mInputBuffers;
- Vector<sp<MediaCodecBuffer> > mOutputBuffers;
- Vector<sp<ABuffer> > mCSDsForCurrentFormat;
- Vector<sp<ABuffer> > mCSDsToSubmit;
- Vector<bool> mInputBufferIsDequeued;
- Vector<MediaBuffer *> mMediaBuffers;
- Vector<size_t> mDequeuedInputBuffers;
-
- const pid_t mPid;
- const uid_t mUid;
- int64_t mSkipRenderingUntilMediaTimeUs;
- int64_t mNumFramesTotal;
- int64_t mNumInputFramesDropped;
- int64_t mNumOutputFramesDropped;
- int32_t mVideoWidth;
- int32_t mVideoHeight;
- bool mIsAudio;
- bool mIsVideoAVC;
- bool mIsSecure;
- bool mIsEncrypted;
- bool mIsEncryptedObservedEarlier;
- bool mFormatChangePending;
- bool mTimeChangePending;
- float mFrameRateTotal;
- float mPlaybackSpeed;
- int32_t mNumVideoTemporalLayerTotal;
- int32_t mNumVideoTemporalLayerAllowed;
- int32_t mCurrentMaxVideoTemporalLayerId;
- float mVideoTemporalLayerAggregateFps[kMaxNumVideoTemporalLayers];
-
- bool mResumePending;
- AString mComponentName;
-
- void handleError(int32_t err);
- bool handleAnInputBuffer(size_t index);
- bool handleAnOutputBuffer(
- size_t index,
- size_t offset,
- size_t size,
- int64_t timeUs,
- int32_t flags);
- void handleOutputFormatChange(const sp<AMessage> &format);
-
- void releaseAndResetMediaBuffers();
- bool isStaleReply(const sp<AMessage> &msg);
-
- void doFlush(bool notifyComplete);
- status_t fetchInputData(sp<AMessage> &reply);
- bool onInputBufferFetched(const sp<AMessage> &msg);
- void onRenderBuffer(const sp<AMessage> &msg);
-
- bool supportsSeamlessFormatChange(const sp<AMessage> &to) const;
- bool supportsSeamlessAudioFormatChange(const sp<AMessage> &targetFormat) const;
- void rememberCodecSpecificData(const sp<AMessage> &format);
- bool isDiscontinuityPending() const;
- void finishHandleDiscontinuity(bool flushOnTimeChange);
-
- void notifyResumeCompleteIfNecessary();
-
- void onReleaseCrypto(const sp<AMessage>& msg);
-
- DISALLOW_EVIL_CONSTRUCTORS(Decoder);
-};
-
-} // namespace android
-
-#endif // NUPLAYER2_DECODER_H_
diff --git a/media/libmediaplayer2/nuplayer2/NuPlayer2DecoderBase.cpp b/media/libmediaplayer2/nuplayer2/NuPlayer2DecoderBase.cpp
deleted file mode 100644
index 914f29f..0000000
--- a/media/libmediaplayer2/nuplayer2/NuPlayer2DecoderBase.cpp
+++ /dev/null
@@ -1,216 +0,0 @@
-/*
- * Copyright 2017 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "NuPlayer2DecoderBase"
-#include <utils/Log.h>
-#include <inttypes.h>
-
-#include "NuPlayer2DecoderBase.h"
-
-#include "NuPlayer2Renderer.h"
-
-#include <media/MediaCodecBuffer.h>
-#include <media/stagefright/foundation/ADebug.h>
-#include <media/stagefright/foundation/AMessage.h>
-
-namespace android {
-
-NuPlayer2::DecoderBase::DecoderBase(const sp<AMessage> ¬ify)
- : mNotify(notify),
- mBufferGeneration(0),
- mPaused(false),
- mStats(new AMessage),
- mRequestInputBuffersPending(false) {
- // Every decoder has its own looper because MediaCodec operations
- // are blocking, but NuPlayer2 needs asynchronous operations.
- mDecoderLooper = new ALooper;
- mDecoderLooper->setName("NP2Decoder");
- mDecoderLooper->start(false, /* runOnCallingThread */
- true, /* canCallJava */
- ANDROID_PRIORITY_AUDIO);
-}
-
-NuPlayer2::DecoderBase::~DecoderBase() {
- stopLooper();
-}
-
-static
-status_t PostAndAwaitResponse(
- const sp<AMessage> &msg, sp<AMessage> *response) {
- status_t err = msg->postAndAwaitResponse(response);
-
- if (err != OK) {
- return err;
- }
-
- if (!(*response)->findInt32("err", &err)) {
- err = OK;
- }
-
- return err;
-}
-
-void NuPlayer2::DecoderBase::configure(const sp<AMessage> &format) {
- sp<AMessage> msg = new AMessage(kWhatConfigure, this);
- msg->setMessage("format", format);
- msg->post();
-}
-
-void NuPlayer2::DecoderBase::init() {
- mDecoderLooper->registerHandler(this);
-}
-
-void NuPlayer2::DecoderBase::stopLooper() {
- mDecoderLooper->unregisterHandler(id());
- mDecoderLooper->stop();
-}
-
-void NuPlayer2::DecoderBase::setParameters(const sp<AMessage> ¶ms) {
- sp<AMessage> msg = new AMessage(kWhatSetParameters, this);
- msg->setMessage("params", params);
- msg->post();
-}
-
-void NuPlayer2::DecoderBase::setRenderer(const sp<Renderer> &renderer) {
- sp<AMessage> msg = new AMessage(kWhatSetRenderer, this);
- msg->setObject("renderer", renderer);
- msg->post();
-}
-
-void NuPlayer2::DecoderBase::pause() {
- sp<AMessage> msg = new AMessage(kWhatPause, this);
-
- sp<AMessage> response;
- PostAndAwaitResponse(msg, &response);
-}
-
-void NuPlayer2::DecoderBase::signalFlush() {
- (new AMessage(kWhatFlush, this))->post();
-}
-
-void NuPlayer2::DecoderBase::signalResume(bool notifyComplete) {
- sp<AMessage> msg = new AMessage(kWhatResume, this);
- msg->setInt32("notifyComplete", notifyComplete);
- msg->post();
-}
-
-void NuPlayer2::DecoderBase::initiateShutdown() {
- (new AMessage(kWhatShutdown, this))->post();
-}
-
-void NuPlayer2::DecoderBase::onRequestInputBuffers() {
- if (mRequestInputBuffersPending) {
- return;
- }
-
- // doRequestBuffers() return true if we should request more data
- if (doRequestBuffers()) {
- mRequestInputBuffersPending = true;
-
- sp<AMessage> msg = new AMessage(kWhatRequestInputBuffers, this);
- msg->post(10 * 1000LL);
- }
-}
-
-void NuPlayer2::DecoderBase::onMessageReceived(const sp<AMessage> &msg) {
-
- switch (msg->what()) {
- case kWhatConfigure:
- {
- sp<AMessage> format;
- CHECK(msg->findMessage("format", &format));
- onConfigure(format);
- break;
- }
-
- case kWhatSetParameters:
- {
- sp<AMessage> params;
- CHECK(msg->findMessage("params", ¶ms));
- onSetParameters(params);
- break;
- }
-
- case kWhatSetRenderer:
- {
- sp<RefBase> obj;
- CHECK(msg->findObject("renderer", &obj));
- onSetRenderer(static_cast<Renderer *>(obj.get()));
- break;
- }
-
- case kWhatPause:
- {
- sp<AReplyToken> replyID;
- CHECK(msg->senderAwaitsResponse(&replyID));
-
- mPaused = true;
-
- (new AMessage)->postReply(replyID);
- break;
- }
-
- case kWhatRequestInputBuffers:
- {
- mRequestInputBuffersPending = false;
- onRequestInputBuffers();
- break;
- }
-
- case kWhatFlush:
- {
- onFlush();
- break;
- }
-
- case kWhatResume:
- {
- int32_t notifyComplete;
- CHECK(msg->findInt32("notifyComplete", ¬ifyComplete));
-
- onResume(notifyComplete);
- break;
- }
-
- case kWhatShutdown:
- {
- onShutdown(true);
- break;
- }
-
- default:
- TRESPASS();
- break;
- }
-}
-
-void NuPlayer2::DecoderBase::handleError(int32_t err)
-{
- // We cannot immediately release the codec due to buffers still outstanding
- // in the renderer. We signal to the player the error so it can shutdown/release the
- // decoder after flushing and increment the generation to discard unnecessary messages.
-
- ++mBufferGeneration;
-
- sp<AMessage> notify = mNotify->dup();
- notify->setInt32("what", kWhatError);
- notify->setInt32("err", err);
- notify->post();
-}
-
-} // namespace android
-
diff --git a/media/libmediaplayer2/nuplayer2/NuPlayer2DecoderBase.h b/media/libmediaplayer2/nuplayer2/NuPlayer2DecoderBase.h
deleted file mode 100644
index 1e57f0d..0000000
--- a/media/libmediaplayer2/nuplayer2/NuPlayer2DecoderBase.h
+++ /dev/null
@@ -1,111 +0,0 @@
-/*
- * Copyright 2017 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef NUPLAYER2_DECODER_BASE_H_
-
-#define NUPLAYER2_DECODER_BASE_H_
-
-#include "NuPlayer2.h"
-
-#include <media/stagefright/foundation/AHandler.h>
-
-namespace android {
-
-struct ABuffer;
-struct ANativeWindowWrapper;
-struct MediaCodec;
-class MediaBuffer;
-class MediaCodecBuffer;
-
-struct NuPlayer2::DecoderBase : public AHandler {
- explicit DecoderBase(const sp<AMessage> ¬ify);
-
- void configure(const sp<AMessage> &format);
- void init();
- void setParameters(const sp<AMessage> ¶ms);
-
- // Synchronous call to ensure decoder will not request or send out data.
- void pause();
-
- void setRenderer(const sp<Renderer> &renderer);
- virtual status_t setVideoSurface(const sp<ANativeWindowWrapper> &) { return INVALID_OPERATION; }
-
- void signalFlush();
- void signalResume(bool notifyComplete);
- void initiateShutdown();
-
- virtual sp<AMessage> getStats() const {
- return mStats;
- }
-
- virtual status_t releaseCrypto() {
- return INVALID_OPERATION;
- }
-
- enum {
- kWhatInputDiscontinuity = 'inDi',
- kWhatVideoSizeChanged = 'viSC',
- kWhatFlushCompleted = 'flsC',
- kWhatShutdownCompleted = 'shDC',
- kWhatResumeCompleted = 'resC',
- kWhatEOS = 'eos ',
- kWhatError = 'err ',
- };
-
-protected:
-
- virtual ~DecoderBase();
-
- void stopLooper();
-
- virtual void onMessageReceived(const sp<AMessage> &msg);
-
- virtual void onConfigure(const sp<AMessage> &format) = 0;
- virtual void onSetParameters(const sp<AMessage> ¶ms) = 0;
- virtual void onSetRenderer(const sp<Renderer> &renderer) = 0;
- virtual void onResume(bool notifyComplete) = 0;
- virtual void onFlush() = 0;
- virtual void onShutdown(bool notifyComplete) = 0;
-
- void onRequestInputBuffers();
- virtual bool doRequestBuffers() = 0;
- virtual void handleError(int32_t err);
-
- sp<AMessage> mNotify;
- int32_t mBufferGeneration;
- bool mPaused;
- sp<AMessage> mStats;
-
-private:
- enum {
- kWhatConfigure = 'conf',
- kWhatSetParameters = 'setP',
- kWhatSetRenderer = 'setR',
- kWhatPause = 'paus',
- kWhatRequestInputBuffers = 'reqB',
- kWhatFlush = 'flus',
- kWhatShutdown = 'shuD',
- };
-
- sp<ALooper> mDecoderLooper;
- bool mRequestInputBuffersPending;
-
- DISALLOW_EVIL_CONSTRUCTORS(DecoderBase);
-};
-
-} // namespace android
-
-#endif // NUPLAYER2_DECODER_BASE_H_
diff --git a/media/libmediaplayer2/nuplayer2/NuPlayer2DecoderPassThrough.cpp b/media/libmediaplayer2/nuplayer2/NuPlayer2DecoderPassThrough.cpp
deleted file mode 100644
index 0514e88..0000000
--- a/media/libmediaplayer2/nuplayer2/NuPlayer2DecoderPassThrough.cpp
+++ /dev/null
@@ -1,434 +0,0 @@
-/*
- * Copyright 2017 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "NuPlayer2DecoderPassThrough"
-#include <utils/Log.h>
-#include <inttypes.h>
-
-#include "NuPlayer2DecoderPassThrough.h"
-
-#include "NuPlayer2Renderer.h"
-#include "NuPlayer2Source.h"
-
-#include <media/MediaCodecBuffer.h>
-#include <media/stagefright/foundation/ABuffer.h>
-#include <media/stagefright/foundation/ADebug.h>
-#include <media/stagefright/foundation/AMessage.h>
-#include <media/stagefright/MediaErrors.h>
-
-#include "ATSParser.h"
-
-namespace android {
-
-// TODO optimize buffer size for power consumption
-// The offload read buffer size is 32 KB but 24 KB uses less power.
-static const size_t kAggregateBufferSizeBytes = 24 * 1024;
-static const size_t kMaxCachedBytes = 200000;
-
-NuPlayer2::DecoderPassThrough::DecoderPassThrough(
- const sp<AMessage> ¬ify,
- const sp<Source> &source,
- const sp<Renderer> &renderer)
- : DecoderBase(notify),
- mSource(source),
- mRenderer(renderer),
- mSkipRenderingUntilMediaTimeUs(-1LL),
- mReachedEOS(true),
- mPendingAudioErr(OK),
- mPendingBuffersToDrain(0),
- mCachedBytes(0),
- mComponentName("pass through decoder") {
- ALOGW_IF(renderer == NULL, "expect a non-NULL renderer");
-}
-
-NuPlayer2::DecoderPassThrough::~DecoderPassThrough() {
-}
-
-void NuPlayer2::DecoderPassThrough::onConfigure(const sp<AMessage> &format) {
- ALOGV("[%s] onConfigure", mComponentName.c_str());
- mCachedBytes = 0;
- mPendingBuffersToDrain = 0;
- mReachedEOS = false;
- ++mBufferGeneration;
-
- onRequestInputBuffers();
-
- int32_t hasVideo = 0;
- format->findInt32("has-video", &hasVideo);
-
- // The audio sink is already opened before the PassThrough decoder is created.
- // Opening again might be relevant if decoder is instantiated after shutdown and
- // format is different.
- status_t err = mRenderer->openAudioSink(
- format, true /* offloadOnly */, hasVideo,
- AUDIO_OUTPUT_FLAG_NONE /* flags */, NULL /* isOffloaded */, mSource->isStreaming());
- if (err != OK) {
- handleError(err);
- }
-}
-
-void NuPlayer2::DecoderPassThrough::onSetParameters(const sp<AMessage> &/*params*/) {
- ALOGW("onSetParameters() called unexpectedly");
-}
-
-void NuPlayer2::DecoderPassThrough::onSetRenderer(
- const sp<Renderer> &renderer) {
- // renderer can't be changed during offloading
- ALOGW_IF(renderer != mRenderer,
- "ignoring request to change renderer");
-}
-
-bool NuPlayer2::DecoderPassThrough::isStaleReply(const sp<AMessage> &msg) {
- int32_t generation;
- CHECK(msg->findInt32("generation", &generation));
- return generation != mBufferGeneration;
-}
-
-bool NuPlayer2::DecoderPassThrough::isDoneFetching() const {
- ALOGV("[%s] mCachedBytes = %zu, mReachedEOS = %d mPaused = %d",
- mComponentName.c_str(), mCachedBytes, mReachedEOS, mPaused);
-
- return mCachedBytes >= kMaxCachedBytes || mReachedEOS || mPaused;
-}
-
-/*
- * returns true if we should request more data
- */
-bool NuPlayer2::DecoderPassThrough::doRequestBuffers() {
- status_t err = OK;
- while (!isDoneFetching()) {
- sp<AMessage> msg = new AMessage();
-
- err = fetchInputData(msg);
- if (err != OK) {
- break;
- }
-
- onInputBufferFetched(msg);
- }
-
- return err == -EWOULDBLOCK
- && mSource->feedMoreTSData() == OK;
-}
-
-status_t NuPlayer2::DecoderPassThrough::dequeueAccessUnit(sp<ABuffer> *accessUnit) {
- status_t err;
-
- // Did we save an accessUnit earlier because of a discontinuity?
- if (mPendingAudioAccessUnit != NULL) {
- *accessUnit = mPendingAudioAccessUnit;
- mPendingAudioAccessUnit.clear();
- err = mPendingAudioErr;
- ALOGV("feedDecoderInputData() use mPendingAudioAccessUnit");
- } else {
- err = mSource->dequeueAccessUnit(true /* audio */, accessUnit);
- }
-
- if (err == INFO_DISCONTINUITY || err == ERROR_END_OF_STREAM) {
- if (mAggregateBuffer != NULL) {
- // We already have some data so save this for later.
- mPendingAudioErr = err;
- mPendingAudioAccessUnit = *accessUnit;
- (*accessUnit).clear();
- ALOGD("return aggregated buffer and save err(=%d) for later", err);
- err = OK;
- }
- }
-
- return err;
-}
-
-sp<ABuffer> NuPlayer2::DecoderPassThrough::aggregateBuffer(
- const sp<ABuffer> &accessUnit) {
- sp<ABuffer> aggregate;
-
- if (accessUnit == NULL) {
- // accessUnit is saved to mPendingAudioAccessUnit
- // return current mAggregateBuffer
- aggregate = mAggregateBuffer;
- mAggregateBuffer.clear();
- return aggregate;
- }
-
- size_t smallSize = accessUnit->size();
- if ((mAggregateBuffer == NULL)
- // Don't bother if only room for a few small buffers.
- && (smallSize < (kAggregateBufferSizeBytes / 3))) {
- // Create a larger buffer for combining smaller buffers from the extractor.
- mAggregateBuffer = new ABuffer(kAggregateBufferSizeBytes);
- mAggregateBuffer->setRange(0, 0); // start empty
- }
-
- if (mAggregateBuffer != NULL) {
- int64_t timeUs;
- int64_t dummy;
- bool smallTimestampValid = accessUnit->meta()->findInt64("timeUs", &timeUs);
- bool bigTimestampValid = mAggregateBuffer->meta()->findInt64("timeUs", &dummy);
- // Will the smaller buffer fit?
- size_t bigSize = mAggregateBuffer->size();
- size_t roomLeft = mAggregateBuffer->capacity() - bigSize;
- // Should we save this small buffer for the next big buffer?
- // If the first small buffer did not have a timestamp then save
- // any buffer that does have a timestamp until the next big buffer.
- if ((smallSize > roomLeft)
- || (!bigTimestampValid && (bigSize > 0) && smallTimestampValid)) {
- mPendingAudioErr = OK;
- mPendingAudioAccessUnit = accessUnit;
- aggregate = mAggregateBuffer;
- mAggregateBuffer.clear();
- } else {
- // Grab time from first small buffer if available.
- if ((bigSize == 0) && smallTimestampValid) {
- mAggregateBuffer->meta()->setInt64("timeUs", timeUs);
- }
- // Append small buffer to the bigger buffer.
- memcpy(mAggregateBuffer->base() + bigSize, accessUnit->data(), smallSize);
- bigSize += smallSize;
- mAggregateBuffer->setRange(0, bigSize);
-
- ALOGV("feedDecoderInputData() smallSize = %zu, bigSize = %zu, capacity = %zu",
- smallSize, bigSize, mAggregateBuffer->capacity());
- }
- } else {
- // decided not to aggregate
- aggregate = accessUnit;
- }
-
- return aggregate;
-}
-
-status_t NuPlayer2::DecoderPassThrough::fetchInputData(sp<AMessage> &reply) {
- sp<ABuffer> accessUnit;
-
- do {
- status_t err = dequeueAccessUnit(&accessUnit);
-
- if (err == -EWOULDBLOCK) {
- // Flush out the aggregate buffer to try to avoid underrun.
- accessUnit = aggregateBuffer(NULL /* accessUnit */);
- if (accessUnit != NULL) {
- break;
- }
- return err;
- } else if (err != OK) {
- if (err == INFO_DISCONTINUITY) {
- int32_t type;
- CHECK(accessUnit->meta()->findInt32("discontinuity", &type));
-
- bool formatChange =
- (type & ATSParser::DISCONTINUITY_AUDIO_FORMAT) != 0;
-
- bool timeChange =
- (type & ATSParser::DISCONTINUITY_TIME) != 0;
-
- ALOGI("audio discontinuity (formatChange=%d, time=%d)",
- formatChange, timeChange);
-
- if (formatChange || timeChange) {
- sp<AMessage> msg = mNotify->dup();
- msg->setInt32("what", kWhatInputDiscontinuity);
- // will perform seamless format change,
- // only notify NuPlayer2 to scan sources
- msg->setInt32("formatChange", false);
- msg->post();
- }
-
- if (timeChange) {
- doFlush(false /* notifyComplete */);
- err = OK;
- } else if (formatChange) {
- // do seamless format change
- err = OK;
- } else {
- // This stream is unaffected by the discontinuity
- return -EWOULDBLOCK;
- }
- }
-
- reply->setInt32("err", err);
- return OK;
- }
-
- accessUnit = aggregateBuffer(accessUnit);
- } while (accessUnit == NULL);
-
-#if 0
- int64_t mediaTimeUs;
- CHECK(accessUnit->meta()->findInt64("timeUs", &mediaTimeUs));
- ALOGV("feeding audio input buffer at media time %.2f secs",
- mediaTimeUs / 1E6);
-#endif
-
- reply->setBuffer("buffer", accessUnit);
-
- return OK;
-}
-
-void NuPlayer2::DecoderPassThrough::onInputBufferFetched(
- const sp<AMessage> &msg) {
- if (mReachedEOS) {
- return;
- }
-
- sp<ABuffer> buffer;
- bool hasBuffer = msg->findBuffer("buffer", &buffer);
- if (buffer == NULL) {
- int32_t streamErr = ERROR_END_OF_STREAM;
- CHECK(msg->findInt32("err", &streamErr) || !hasBuffer);
- if (streamErr == OK) {
- return;
- }
-
- if (streamErr != ERROR_END_OF_STREAM) {
- handleError(streamErr);
- }
- mReachedEOS = true;
- if (mRenderer != NULL) {
- mRenderer->queueEOS(true /* audio */, ERROR_END_OF_STREAM);
- }
- return;
- }
-
- sp<AMessage> extra;
- if (buffer->meta()->findMessage("extra", &extra) && extra != NULL) {
- int64_t resumeAtMediaTimeUs;
- if (extra->findInt64(
- "resume-at-mediatimeUs", &resumeAtMediaTimeUs)) {
- ALOGI("[%s] suppressing rendering until %lld us",
- mComponentName.c_str(), (long long)resumeAtMediaTimeUs);
- mSkipRenderingUntilMediaTimeUs = resumeAtMediaTimeUs;
- }
- }
-
- int32_t bufferSize = buffer->size();
- mCachedBytes += bufferSize;
-
- int64_t timeUs = 0;
- CHECK(buffer->meta()->findInt64("timeUs", &timeUs));
- if (mSkipRenderingUntilMediaTimeUs >= 0) {
- if (timeUs < mSkipRenderingUntilMediaTimeUs) {
- ALOGV("[%s] dropping buffer at time %lld as requested.",
- mComponentName.c_str(), (long long)timeUs);
-
- onBufferConsumed(bufferSize);
- return;
- }
-
- mSkipRenderingUntilMediaTimeUs = -1;
- }
-
- if (mRenderer == NULL) {
- onBufferConsumed(bufferSize);
- return;
- }
-
- sp<AMessage> reply = new AMessage(kWhatBufferConsumed, this);
- reply->setInt32("generation", mBufferGeneration);
- reply->setInt32("size", bufferSize);
-
- sp<MediaCodecBuffer> mcBuffer = new MediaCodecBuffer(nullptr, buffer);
- mcBuffer->meta()->setInt64("timeUs", timeUs);
-
- mRenderer->queueBuffer(true /* audio */, mcBuffer, reply);
-
- ++mPendingBuffersToDrain;
- ALOGV("onInputBufferFilled: #ToDrain = %zu, cachedBytes = %zu",
- mPendingBuffersToDrain, mCachedBytes);
-}
-
-void NuPlayer2::DecoderPassThrough::onBufferConsumed(int32_t size) {
- --mPendingBuffersToDrain;
- mCachedBytes -= size;
- ALOGV("onBufferConsumed: #ToDrain = %zu, cachedBytes = %zu",
- mPendingBuffersToDrain, mCachedBytes);
- onRequestInputBuffers();
-}
-
-void NuPlayer2::DecoderPassThrough::onResume(bool notifyComplete) {
- mPaused = false;
-
- onRequestInputBuffers();
-
- if (notifyComplete) {
- sp<AMessage> notify = mNotify->dup();
- notify->setInt32("what", kWhatResumeCompleted);
- notify->post();
- }
-}
-
-void NuPlayer2::DecoderPassThrough::doFlush(bool notifyComplete) {
- ++mBufferGeneration;
- mSkipRenderingUntilMediaTimeUs = -1;
- mPendingAudioAccessUnit.clear();
- mPendingAudioErr = OK;
- mAggregateBuffer.clear();
-
- if (mRenderer != NULL) {
- mRenderer->flush(true /* audio */, notifyComplete);
- mRenderer->signalTimeDiscontinuity();
- }
-
- mPendingBuffersToDrain = 0;
- mCachedBytes = 0;
- mReachedEOS = false;
-}
-
-void NuPlayer2::DecoderPassThrough::onFlush() {
- doFlush(true /* notifyComplete */);
-
- mPaused = true;
- sp<AMessage> notify = mNotify->dup();
- notify->setInt32("what", kWhatFlushCompleted);
- notify->post();
-
-}
-
-void NuPlayer2::DecoderPassThrough::onShutdown(bool notifyComplete) {
- ++mBufferGeneration;
- mSkipRenderingUntilMediaTimeUs = -1;
-
- if (notifyComplete) {
- sp<AMessage> notify = mNotify->dup();
- notify->setInt32("what", kWhatShutdownCompleted);
- notify->post();
- }
-
- mReachedEOS = true;
-}
-
-void NuPlayer2::DecoderPassThrough::onMessageReceived(const sp<AMessage> &msg) {
- ALOGV("[%s] onMessage: %s", mComponentName.c_str(),
- msg->debugString().c_str());
-
- switch (msg->what()) {
- case kWhatBufferConsumed:
- {
- if (!isStaleReply(msg)) {
- int32_t size;
- CHECK(msg->findInt32("size", &size));
- onBufferConsumed(size);
- }
- break;
- }
-
- default:
- DecoderBase::onMessageReceived(msg);
- break;
- }
-}
-
-} // namespace android
diff --git a/media/libmediaplayer2/nuplayer2/NuPlayer2DecoderPassThrough.h b/media/libmediaplayer2/nuplayer2/NuPlayer2DecoderPassThrough.h
deleted file mode 100644
index 838c60a..0000000
--- a/media/libmediaplayer2/nuplayer2/NuPlayer2DecoderPassThrough.h
+++ /dev/null
@@ -1,85 +0,0 @@
-/*
- * Copyright 2017 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef NUPLAYER2_DECODER_PASS_THROUGH_H_
-
-#define NUPLAYER2_DECODER_PASS_THROUGH_H_
-
-#include "NuPlayer2.h"
-
-#include "NuPlayer2DecoderBase.h"
-
-namespace android {
-
-struct NuPlayer2::DecoderPassThrough : public DecoderBase {
- DecoderPassThrough(const sp<AMessage> ¬ify,
- const sp<Source> &source,
- const sp<Renderer> &renderer);
-
-protected:
-
- virtual ~DecoderPassThrough();
-
- virtual void onMessageReceived(const sp<AMessage> &msg);
-
- virtual void onConfigure(const sp<AMessage> &format);
- virtual void onSetParameters(const sp<AMessage> ¶ms);
- virtual void onSetRenderer(const sp<Renderer> &renderer);
- virtual void onResume(bool notifyComplete);
- virtual void onFlush();
- virtual void onShutdown(bool notifyComplete);
- virtual bool doRequestBuffers();
-
-private:
- enum {
- kWhatBufferConsumed = 'bufC',
- };
-
- sp<Source> mSource;
- sp<Renderer> mRenderer;
- int64_t mSkipRenderingUntilMediaTimeUs;
-
- bool mReachedEOS;
-
- // Used by feedDecoderInputData to aggregate small buffers into
- // one large buffer.
- sp<ABuffer> mPendingAudioAccessUnit;
- status_t mPendingAudioErr;
- sp<ABuffer> mAggregateBuffer;
-
- // mPendingBuffersToDrain are only for debugging. It can be removed
- // when the power investigation is done.
- size_t mPendingBuffersToDrain;
- size_t mCachedBytes;
- AString mComponentName;
-
- bool isStaleReply(const sp<AMessage> &msg);
- bool isDoneFetching() const;
-
- status_t dequeueAccessUnit(sp<ABuffer> *accessUnit);
- sp<ABuffer> aggregateBuffer(const sp<ABuffer> &accessUnit);
- status_t fetchInputData(sp<AMessage> &reply);
- void doFlush(bool notifyComplete);
-
- void onInputBufferFetched(const sp<AMessage> &msg);
- void onBufferConsumed(int32_t size);
-
- DISALLOW_EVIL_CONSTRUCTORS(DecoderPassThrough);
-};
-
-} // namespace android
-
-#endif // NUPLAYER2_DECODER_PASS_THROUGH_H_
diff --git a/media/libmediaplayer2/nuplayer2/NuPlayer2Driver.cpp b/media/libmediaplayer2/nuplayer2/NuPlayer2Driver.cpp
deleted file mode 100644
index 1876496..0000000
--- a/media/libmediaplayer2/nuplayer2/NuPlayer2Driver.cpp
+++ /dev/null
@@ -1,1010 +0,0 @@
-/*
- * Copyright 2017 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "NuPlayer2Driver"
-#include <inttypes.h>
-#include <android-base/macros.h>
-#include <utils/Log.h>
-#include <cutils/properties.h>
-
-#include "NuPlayer2Driver.h"
-
-#include "NuPlayer2.h"
-#include "NuPlayer2Source.h"
-
-#include <media/DataSourceDesc.h>
-#include <media/stagefright/foundation/ADebug.h>
-#include <media/stagefright/foundation/ALooper.h>
-#include <media/stagefright/foundation/AUtils.h>
-#include <media/stagefright/foundation/ByteUtils.h>
-#include <media/stagefright/MediaClock.h>
-#include <media/stagefright/MetaData.h>
-#include <media/stagefright/Utils.h>
-
-#include <media/IMediaAnalyticsService.h>
-
-using google::protobuf::RepeatedPtrField;
-using android::media::MediaPlayer2Proto::Value;
-
-static const int kDumpLockRetries = 50;
-static const int kDumpLockSleepUs = 20000;
-
-namespace android {
-
-struct PlayerMessageWrapper : public RefBase {
- static sp<PlayerMessageWrapper> Create(const PlayerMessage *p) {
- if (p != NULL) {
- sp<PlayerMessageWrapper> pw = new PlayerMessageWrapper();
- pw->copyFrom(p);
- return pw;
- }
- return NULL;
- }
-
- const PlayerMessage *getPlayerMessage() {
- return mPlayerMessage;
- }
-
-protected:
- virtual ~PlayerMessageWrapper() {
- if (mPlayerMessage != NULL) {
- delete mPlayerMessage;
- }
- }
-
-private:
- PlayerMessageWrapper()
- : mPlayerMessage(NULL) { }
-
- void copyFrom(const PlayerMessage *p) {
- if (mPlayerMessage == NULL) {
- mPlayerMessage = new PlayerMessage;
- }
- mPlayerMessage->CopyFrom(*p);
- }
-
- PlayerMessage *mPlayerMessage;
-};
-
-// key for media statistics
-static const char *kKeyPlayer = "nuplayer2";
-// attrs for media statistics
- // NB: these are matched with public Java API constants defined
- // in frameworks/base/media/java/android/media/MediaPlayer2.java
- // These must be kept synchronized with the constants there.
-static const char *kPlayerVMime = "android.media.mediaplayer.video.mime";
-static const char *kPlayerVCodec = "android.media.mediaplayer.video.codec";
-static const char *kPlayerWidth = "android.media.mediaplayer.width";
-static const char *kPlayerHeight = "android.media.mediaplayer.height";
-static const char *kPlayerFrames = "android.media.mediaplayer.frames";
-static const char *kPlayerFramesDropped = "android.media.mediaplayer.dropped";
-static const char *kPlayerFrameRate = "android.media.mediaplayer.fps";
-static const char *kPlayerAMime = "android.media.mediaplayer.audio.mime";
-static const char *kPlayerACodec = "android.media.mediaplayer.audio.codec";
-static const char *kPlayerDuration = "android.media.mediaplayer.durationMs";
-static const char *kPlayerPlaying = "android.media.mediaplayer.playingMs";
-static const char *kPlayerError = "android.media.mediaplayer.err";
-static const char *kPlayerErrorCode = "android.media.mediaplayer.errcode";
-
-// NB: These are not yet exposed as public Java API constants.
-static const char *kPlayerErrorState = "android.media.mediaplayer.errstate";
-static const char *kPlayerDataSourceType = "android.media.mediaplayer.dataSource";
-//
-static const char *kPlayerRebuffering = "android.media.mediaplayer.rebufferingMs";
-static const char *kPlayerRebufferingCount = "android.media.mediaplayer.rebuffers";
-static const char *kPlayerRebufferingAtExit = "android.media.mediaplayer.rebufferExit";
-
-static const char *kPlayerVersion = "android.media.mediaplayer.version";
-
-
-NuPlayer2Driver::NuPlayer2Driver(pid_t pid, uid_t uid, const sp<JObjectHolder> &context)
- : mState(STATE_IDLE),
- mAsyncResult(UNKNOWN_ERROR),
- mSrcId(0),
- mSetSurfaceInProgress(false),
- mDurationUs(-1),
- mPositionUs(-1),
- mSeekInProgress(false),
- mPlayingTimeUs(0),
- mRebufferingTimeUs(0),
- mRebufferingEvents(0),
- mRebufferingAtExit(false),
- mLooper(new ALooper),
- mNuPlayer2Looper(new ALooper),
- mMediaClock(new MediaClock),
- mPlayer(new NuPlayer2(pid, uid, mMediaClock, context)),
- mPlayerFlags(0),
- mMetricsHandle(0),
- mPlayerVersion(0),
- mClientUid(uid),
- mAtEOS(false),
- mLooping(false),
- mAutoLoop(false) {
- ALOGD("NuPlayer2Driver(%p) created, clientPid(%d)", this, pid);
- mLooper->setName("NuPlayer2Driver Looper");
- mNuPlayer2Looper->setName("NuPlayer2 Looper");
-
- mMediaClock->init();
-
- // XXX: what version are we?
- // Ideally, this ticks with the apk version info for the APEX packaging
-
- // set up media metrics record
- mMetricsHandle = mediametrics_create(kKeyPlayer);
- mediametrics_setUid(mMetricsHandle, mClientUid);
- mediametrics_setInt64(mMetricsHandle, kPlayerVersion, mPlayerVersion);
-
- mNuPlayer2Looper->start(
- false, /* runOnCallingThread */
- true, /* canCallJava */
- PRIORITY_AUDIO);
-
- mNuPlayer2Looper->registerHandler(mPlayer);
-
- mPlayer->setDriver(this);
-}
-
-NuPlayer2Driver::~NuPlayer2Driver() {
- ALOGV("~NuPlayer2Driver(%p)", this);
- mNuPlayer2Looper->stop();
- mLooper->stop();
-
- // finalize any pending metrics, usually a no-op.
- updateMetrics("destructor");
- logMetrics("destructor");
-
- mediametrics_delete(mMetricsHandle);
-}
-
-status_t NuPlayer2Driver::initCheck() {
- mLooper->start(
- false, /* runOnCallingThread */
- true, /* canCallJava */
- PRIORITY_AUDIO);
-
- mLooper->registerHandler(this);
- return OK;
-}
-
-status_t NuPlayer2Driver::setDataSource(const sp<DataSourceDesc> &dsd) {
- ALOGV("setDataSource(%p)", this);
- Mutex::Autolock autoLock(mLock);
-
- if (mState != STATE_IDLE) {
- return INVALID_OPERATION;
- }
-
- mSrcId = dsd->mId;
- mState = STATE_SET_DATASOURCE_PENDING;
-
- mPlayer->setDataSourceAsync(dsd);
-
- while (mState == STATE_SET_DATASOURCE_PENDING) {
- mCondition.wait(mLock);
- }
-
- return mAsyncResult;
-}
-
-status_t NuPlayer2Driver::prepareNextDataSource(const sp<DataSourceDesc> &dsd) {
- ALOGV("prepareNextDataSource(%p)", this);
- Mutex::Autolock autoLock(mLock);
-
- mPlayer->prepareNextDataSourceAsync(dsd);
-
- return OK;
-}
-
-status_t NuPlayer2Driver::playNextDataSource(int64_t srcId) {
- ALOGV("playNextDataSource(%p)", this);
- Mutex::Autolock autoLock(mLock);
-
- mSrcId = srcId;
- mPlayer->playNextDataSource(srcId);
-
- return OK;
-}
-
-status_t NuPlayer2Driver::setVideoSurfaceTexture(const sp<ANativeWindowWrapper> &nww) {
- ALOGV("setVideoSurfaceTexture(%p)", this);
- Mutex::Autolock autoLock(mLock);
-
- if (mSetSurfaceInProgress) {
- return INVALID_OPERATION;
- }
-
- switch (mState) {
- case STATE_SET_DATASOURCE_PENDING:
- case STATE_RESET_IN_PROGRESS:
- return INVALID_OPERATION;
-
- default:
- break;
- }
-
- mSetSurfaceInProgress = true;
-
- mPlayer->setVideoSurfaceTextureAsync(nww);
-
- while (mSetSurfaceInProgress) {
- mCondition.wait(mLock);
- }
-
- return OK;
-}
-
-status_t NuPlayer2Driver::getBufferingSettings(BufferingSettings* buffering) {
- ALOGV("getBufferingSettings(%p)", this);
- {
- Mutex::Autolock autoLock(mLock);
- if (mState == STATE_IDLE) {
- return INVALID_OPERATION;
- }
- }
-
- return mPlayer->getBufferingSettings(buffering);
-}
-
-status_t NuPlayer2Driver::setBufferingSettings(const BufferingSettings& buffering) {
- ALOGV("setBufferingSettings(%p)", this);
- {
- Mutex::Autolock autoLock(mLock);
- if (mState == STATE_IDLE) {
- return INVALID_OPERATION;
- }
- }
-
- return mPlayer->setBufferingSettings(buffering);
-}
-
-status_t NuPlayer2Driver::prepareAsync() {
- ALOGV("prepareAsync(%p)", this);
- Mutex::Autolock autoLock(mLock);
-
- switch (mState) {
- case STATE_UNPREPARED:
- mState = STATE_PREPARING;
- mPlayer->prepareAsync();
- return OK;
- default:
- return INVALID_OPERATION;
- };
-}
-
-status_t NuPlayer2Driver::start() {
- ALOGD("start(%p), state is %d, eos is %d", this, mState, mAtEOS);
- Mutex::Autolock autoLock(mLock);
- return start_l();
-}
-
-status_t NuPlayer2Driver::start_l() {
- switch (mState) {
- case STATE_PAUSED:
- case STATE_PREPARED:
- {
- mPlayer->start();
- FALLTHROUGH_INTENDED;
- }
-
- case STATE_RUNNING:
- {
- if (mAtEOS) {
- mPlayer->rewind();
- mAtEOS = false;
- mPositionUs = -1;
- }
- break;
- }
-
- default:
- return INVALID_OPERATION;
- }
-
- mState = STATE_RUNNING;
-
- return OK;
-}
-
-status_t NuPlayer2Driver::pause() {
- ALOGD("pause(%p)", this);
- // The NuPlayerRenderer may get flushed if pause for long enough, e.g. the pause timeout tear
- // down for audio offload mode. If that happens, the NuPlayerRenderer will no longer know the
- // current position. So similar to seekTo, update |mPositionUs| to the pause position by calling
- // getCurrentPosition here.
- int64_t unused;
- getCurrentPosition(&unused);
-
- Mutex::Autolock autoLock(mLock);
-
- switch (mState) {
- case STATE_PAUSED:
- return OK;
-
- case STATE_PREPARED:
- case STATE_RUNNING:
- mState = STATE_PAUSED;
- mPlayer->pause();
- break;
-
- default:
- return INVALID_OPERATION;
- }
-
- return OK;
-}
-
-bool NuPlayer2Driver::isPlaying() {
- return mState == STATE_RUNNING && !mAtEOS;
-}
-
-status_t NuPlayer2Driver::setPlaybackSettings(const AudioPlaybackRate &rate) {
- status_t err = mPlayer->setPlaybackSettings(rate);
- if (err == OK) {
- // try to update position
- int64_t unused;
- getCurrentPosition(&unused);
- }
- return err;
-}
-
-status_t NuPlayer2Driver::getPlaybackSettings(AudioPlaybackRate *rate) {
- return mPlayer->getPlaybackSettings(rate);
-}
-
-status_t NuPlayer2Driver::setSyncSettings(const AVSyncSettings &sync, float videoFpsHint) {
- return mPlayer->setSyncSettings(sync, videoFpsHint);
-}
-
-status_t NuPlayer2Driver::getSyncSettings(AVSyncSettings *sync, float *videoFps) {
- return mPlayer->getSyncSettings(sync, videoFps);
-}
-
-status_t NuPlayer2Driver::seekTo(int64_t msec, MediaPlayer2SeekMode mode) {
- ALOGD("seekTo(%p) (%lld ms, %d) at state %d", this, (long long)msec, mode, mState);
- Mutex::Autolock autoLock(mLock);
-
- int64_t seekTimeUs = msec * 1000LL;
-
- switch (mState) {
- case STATE_PREPARED:
- case STATE_PAUSED:
- case STATE_RUNNING:
- {
- mAtEOS = false;
- mSeekInProgress = true;
- mPlayer->seekToAsync(seekTimeUs, mode, true /* needNotify */);
- break;
- }
-
- default:
- return INVALID_OPERATION;
- }
-
- mPositionUs = seekTimeUs;
- return OK;
-}
-
-status_t NuPlayer2Driver::getCurrentPosition(int64_t *msec) {
- int64_t tempUs = 0;
- {
- Mutex::Autolock autoLock(mLock);
- if (mSeekInProgress || (mState == STATE_PAUSED && !mAtEOS)) {
- tempUs = (mPositionUs <= 0) ? 0 : mPositionUs;
- *msec = divRound(tempUs, (int64_t)(1000));
- return OK;
- }
- }
-
- status_t ret = mPlayer->getCurrentPosition(&tempUs);
-
- Mutex::Autolock autoLock(mLock);
- // We need to check mSeekInProgress here because mPlayer->seekToAsync is an async call, which
- // means getCurrentPosition can be called before seek is completed. Iow, renderer may return a
- // position value that's different the seek to position.
- if (ret != OK) {
- tempUs = (mPositionUs <= 0) ? 0 : mPositionUs;
- } else {
- mPositionUs = tempUs;
- }
- *msec = divRound(tempUs, (int64_t)(1000));
- return OK;
-}
-
-status_t NuPlayer2Driver::getDuration(int64_t *msec) {
- Mutex::Autolock autoLock(mLock);
-
- if (mDurationUs < 0) {
- return UNKNOWN_ERROR;
- }
-
- *msec = (mDurationUs + 500LL) / 1000;
-
- return OK;
-}
-
-void NuPlayer2Driver::updateMetrics(const char *where) {
- if (where == NULL) {
- where = "unknown";
- }
- ALOGV("updateMetrics(%p) from %s at state %d", this, where, mState);
-
- // gather the final stats for this record
- Vector<sp<AMessage>> trackStats;
- mPlayer->getStats(&trackStats);
-
- if (trackStats.size() > 0) {
- for (size_t i = 0; i < trackStats.size(); ++i) {
- const sp<AMessage> &stats = trackStats.itemAt(i);
-
- AString mime;
- stats->findString("mime", &mime);
-
- AString name;
- stats->findString("component-name", &name);
-
- if (mime.startsWith("video/")) {
- int32_t width, height;
- mediametrics_setCString(mMetricsHandle, kPlayerVMime, mime.c_str());
- if (!name.empty()) {
- mediametrics_setCString(mMetricsHandle, kPlayerVCodec, name.c_str());
- }
-
- if (stats->findInt32("width", &width)
- && stats->findInt32("height", &height)) {
- mediametrics_setInt32(mMetricsHandle, kPlayerWidth, width);
- mediametrics_setInt32(mMetricsHandle, kPlayerHeight, height);
- }
-
- int64_t numFramesTotal = 0;
- int64_t numFramesDropped = 0;
- stats->findInt64("frames-total", &numFramesTotal);
- stats->findInt64("frames-dropped-output", &numFramesDropped);
-
- mediametrics_setInt64(mMetricsHandle, kPlayerFrames, numFramesTotal);
- mediametrics_setInt64(mMetricsHandle, kPlayerFramesDropped, numFramesDropped);
-
- float frameRate = 0;
- if (stats->findFloat("frame-rate-output", &frameRate)) {
- mediametrics_setInt64(mMetricsHandle, kPlayerFrameRate, frameRate);
- }
-
- } else if (mime.startsWith("audio/")) {
- mediametrics_setCString(mMetricsHandle, kPlayerAMime, mime.c_str());
- if (!name.empty()) {
- mediametrics_setCString(mMetricsHandle, kPlayerACodec, name.c_str());
- }
- }
- }
- }
-
- // always provide duration and playing time, even if they have 0/unknown values.
-
- // getDuration() uses mLock for mutex -- careful where we use it.
- int64_t duration_ms = -1;
- getDuration(&duration_ms);
- mediametrics_setInt64(mMetricsHandle, kPlayerDuration, duration_ms);
-
- mediametrics_setInt64(mMetricsHandle, kPlayerPlaying, (mPlayingTimeUs+500)/1000 );
-
- if (mRebufferingEvents != 0) {
- mediametrics_setInt64(mMetricsHandle, kPlayerRebuffering, (mRebufferingTimeUs+500)/1000 );
- mediametrics_setInt32(mMetricsHandle, kPlayerRebufferingCount, mRebufferingEvents);
- mediametrics_setInt32(mMetricsHandle, kPlayerRebufferingAtExit, mRebufferingAtExit);
- }
-
- mediametrics_setCString(mMetricsHandle, kPlayerDataSourceType, mPlayer->getDataSourceType());
-}
-
-
-void NuPlayer2Driver::logMetrics(const char *where) {
- if (where == NULL) {
- where = "unknown";
- }
- ALOGV("logMetrics(%p) from %s at state %d", this, where, mState);
-
- if (mMetricsHandle == 0 || mediametrics_isEnabled() == false) {
- return;
- }
-
- // log only non-empty records
- // we always updateMetrics() before we get here
- // and that always injects 3 fields (duration, playing time, and
- // datasource) into the record.
- // So the canonical "empty" record has 3 elements in it.
- if (mediametrics_count(mMetricsHandle) > 3) {
- mediametrics_selfRecord(mMetricsHandle);
- // re-init in case we prepare() and start() again.
- mediametrics_delete(mMetricsHandle);
- mMetricsHandle = mediametrics_create(kKeyPlayer);
- mediametrics_setUid(mMetricsHandle, mClientUid);
- mediametrics_setInt64(mMetricsHandle, kPlayerVersion, mPlayerVersion);
- } else {
- ALOGV("did not have anything to record");
- }
-}
-
-status_t NuPlayer2Driver::reset() {
- ALOGD("reset(%p) at state %d", this, mState);
-
- updateMetrics("reset");
- logMetrics("reset");
-
- Mutex::Autolock autoLock(mLock);
-
- switch (mState) {
- case STATE_IDLE:
- return OK;
-
- case STATE_SET_DATASOURCE_PENDING:
- case STATE_RESET_IN_PROGRESS:
- return INVALID_OPERATION;
-
- case STATE_PREPARING:
- {
- notifyListener_l(mSrcId, MEDIA2_PREPARED);
- break;
- }
-
- default:
- break;
- }
-
- mState = STATE_RESET_IN_PROGRESS;
- mPlayer->resetAsync();
-
- while (mState == STATE_RESET_IN_PROGRESS) {
- mCondition.wait(mLock);
- }
-
- mDurationUs = -1;
- mPositionUs = -1;
- mLooping = false;
- mPlayingTimeUs = 0;
- mRebufferingTimeUs = 0;
- mRebufferingEvents = 0;
- mRebufferingAtExit = false;
-
- return OK;
-}
-
-status_t NuPlayer2Driver::notifyAt(int64_t mediaTimeUs) {
- ALOGV("notifyAt(%p), time:%lld", this, (long long)mediaTimeUs);
- return mPlayer->notifyAt(mediaTimeUs);
-}
-
-status_t NuPlayer2Driver::setLooping(int loop) {
- mLooping = loop != 0;
- return OK;
-}
-
-status_t NuPlayer2Driver::invoke(const PlayerMessage &request, PlayerMessage *response) {
- if (response == NULL) {
- ALOGE("reply is a NULL pointer");
- return BAD_VALUE;
- }
-
- RepeatedPtrField<const Value>::const_iterator it = request.values().cbegin();
- int32_t methodId = (it++)->int32_value();
-
- switch (methodId) {
- case MEDIA_PLAYER2_INVOKE_ID_SET_VIDEO_SCALING_MODE:
- {
- int mode = (it++)->int32_value();
- return mPlayer->setVideoScalingMode(mode);
- }
-
- case MEDIA_PLAYER2_INVOKE_ID_GET_TRACK_INFO:
- {
- int64_t srcId = (it++)->int64_value();
- return mPlayer->getTrackInfo(srcId, response);
- }
-
- case MEDIA_PLAYER2_INVOKE_ID_SELECT_TRACK:
- {
- int64_t srcId = (it++)->int64_value();
- int trackIndex = (it++)->int32_value();
- int64_t msec = 0;
- // getCurrentPosition should always return OK
- getCurrentPosition(&msec);
- return mPlayer->selectTrack(srcId, trackIndex, true /* select */, msec * 1000LL);
- }
-
- case MEDIA_PLAYER2_INVOKE_ID_UNSELECT_TRACK:
- {
- int64_t srcId = (it++)->int64_value();
- int trackIndex = (it++)->int32_value();
- return mPlayer->selectTrack(
- srcId, trackIndex, false /* select */, 0xdeadbeef /* not used */);
- }
-
- case MEDIA_PLAYER2_INVOKE_ID_GET_SELECTED_TRACK:
- {
- int64_t srcId = (it++)->int64_value();
- int32_t type = (it++)->int32_value();
- return mPlayer->getSelectedTrack(srcId, type, response);
- }
-
- default:
- {
- return INVALID_OPERATION;
- }
- }
-}
-
-void NuPlayer2Driver::setAudioSink(const sp<AudioSink> &audioSink) {
- mPlayer->setAudioSink(audioSink);
- mAudioSink = audioSink;
-}
-
-status_t NuPlayer2Driver::setParameter(
- int /* key */, const Parcel & /* request */) {
- return INVALID_OPERATION;
-}
-
-status_t NuPlayer2Driver::getParameter(int key __unused, Parcel *reply __unused) {
- return INVALID_OPERATION;
-}
-
-status_t NuPlayer2Driver::getMetrics(char **buffer, size_t *length) {
- updateMetrics("api");
- if (mediametrics_getAttributes(mMetricsHandle, buffer, length))
- return OK;
- else
- return FAILED_TRANSACTION;
-}
-
-void NuPlayer2Driver::notifyResetComplete(int64_t /* srcId */) {
- ALOGD("notifyResetComplete(%p)", this);
- Mutex::Autolock autoLock(mLock);
-
- CHECK_EQ(mState, STATE_RESET_IN_PROGRESS);
- mState = STATE_IDLE;
- mCondition.broadcast();
-}
-
-void NuPlayer2Driver::notifySetSurfaceComplete(int64_t /* srcId */) {
- ALOGV("notifySetSurfaceComplete(%p)", this);
- Mutex::Autolock autoLock(mLock);
-
- CHECK(mSetSurfaceInProgress);
- mSetSurfaceInProgress = false;
-
- mCondition.broadcast();
-}
-
-void NuPlayer2Driver::notifyDuration(int64_t /* srcId */, int64_t durationUs) {
- Mutex::Autolock autoLock(mLock);
- mDurationUs = durationUs;
-}
-
-void NuPlayer2Driver::notifyMorePlayingTimeUs(int64_t /* srcId */, int64_t playingUs) {
- Mutex::Autolock autoLock(mLock);
- mPlayingTimeUs += playingUs;
-}
-
-void NuPlayer2Driver::notifyMoreRebufferingTimeUs(int64_t /* srcId */, int64_t rebufferingUs) {
- Mutex::Autolock autoLock(mLock);
- mRebufferingTimeUs += rebufferingUs;
- mRebufferingEvents++;
-}
-
-void NuPlayer2Driver::notifyRebufferingWhenExit(int64_t /* srcId */, bool status) {
- Mutex::Autolock autoLock(mLock);
- mRebufferingAtExit = status;
-}
-
-void NuPlayer2Driver::notifySeekComplete(int64_t srcId) {
- ALOGV("notifySeekComplete(%p)", this);
- Mutex::Autolock autoLock(mLock);
- mSeekInProgress = false;
- notifyListener_l(srcId, MEDIA2_SEEK_COMPLETE);
-}
-
-status_t NuPlayer2Driver::dump(
- int fd, const Vector<String16> & /* args */) const {
-
- Vector<sp<AMessage> > trackStats;
- mPlayer->getStats(&trackStats);
-
- AString logString(" NuPlayer2\n");
- char buf[256] = {0};
-
- bool locked = false;
- for (int i = 0; i < kDumpLockRetries; ++i) {
- if (mLock.tryLock() == NO_ERROR) {
- locked = true;
- break;
- }
- usleep(kDumpLockSleepUs);
- }
-
- if (locked) {
- snprintf(buf, sizeof(buf), " state(%d), atEOS(%d), looping(%d), autoLoop(%d)\n",
- mState, mAtEOS, mLooping, mAutoLoop);
- mLock.unlock();
- } else {
- snprintf(buf, sizeof(buf), " NPD(%p) lock is taken\n", this);
- }
- logString.append(buf);
-
- for (size_t i = 0; i < trackStats.size(); ++i) {
- const sp<AMessage> &stats = trackStats.itemAt(i);
-
- AString mime;
- if (stats->findString("mime", &mime)) {
- snprintf(buf, sizeof(buf), " mime(%s)\n", mime.c_str());
- logString.append(buf);
- }
-
- AString name;
- if (stats->findString("component-name", &name)) {
- snprintf(buf, sizeof(buf), " decoder(%s)\n", name.c_str());
- logString.append(buf);
- }
-
- if (mime.startsWith("video/")) {
- int32_t width, height;
- if (stats->findInt32("width", &width)
- && stats->findInt32("height", &height)) {
- snprintf(buf, sizeof(buf), " resolution(%d x %d)\n", width, height);
- logString.append(buf);
- }
-
- int64_t numFramesTotal = 0;
- int64_t numFramesDropped = 0;
-
- stats->findInt64("frames-total", &numFramesTotal);
- stats->findInt64("frames-dropped-output", &numFramesDropped);
- snprintf(buf, sizeof(buf), " numFramesTotal(%lld), numFramesDropped(%lld), "
- "percentageDropped(%.2f%%)\n",
- (long long)numFramesTotal,
- (long long)numFramesDropped,
- numFramesTotal == 0
- ? 0.0 : (double)(numFramesDropped * 100) / numFramesTotal);
- logString.append(buf);
- }
- }
-
- ALOGI("%s", logString.c_str());
-
- if (fd >= 0) {
- FILE *out = fdopen(dup(fd), "w");
- fprintf(out, "%s", logString.c_str());
- fclose(out);
- out = NULL;
- }
-
- return OK;
-}
-
-void NuPlayer2Driver::onMessageReceived(const sp<AMessage> &msg) {
- switch (msg->what()) {
- case kWhatNotifyListener: {
- int64_t srcId;
- int32_t msgId;
- int32_t ext1 = 0;
- int32_t ext2 = 0;
- CHECK(msg->findInt64("srcId", &srcId));
- CHECK(msg->findInt32("messageId", &msgId));
- msg->findInt32("ext1", &ext1);
- msg->findInt32("ext2", &ext2);
- sp<PlayerMessageWrapper> in;
- sp<RefBase> obj;
- if (msg->findObject("obj", &obj) && obj != NULL) {
- in = static_cast<PlayerMessageWrapper *>(obj.get());
- }
- sendEvent(srcId, msgId, ext1, ext2, (in == NULL ? NULL : in->getPlayerMessage()));
- break;
- }
- default:
- break;
- }
-}
-
-void NuPlayer2Driver::notifyListener(
- int64_t srcId, int msg, int ext1, int ext2, const PlayerMessage *in) {
- Mutex::Autolock autoLock(mLock);
- notifyListener_l(srcId, msg, ext1, ext2, in);
-}
-
-void NuPlayer2Driver::notifyListener_l(
- int64_t srcId, int msg, int ext1, int ext2, const PlayerMessage *in) {
- ALOGD("notifyListener_l(%p), (%lld, %d, %d, %d, %d), loop setting(%d, %d)",
- this, (long long)srcId, msg, ext1, ext2,
- (in == NULL ? -1 : (int)in->ByteSize()), mAutoLoop, mLooping);
- if (srcId == mSrcId) {
- switch (msg) {
- case MEDIA2_PLAYBACK_COMPLETE:
- {
- if (mState != STATE_RESET_IN_PROGRESS) {
- if (mAutoLoop) {
- audio_stream_type_t streamType = AUDIO_STREAM_MUSIC;
- if (mAudioSink != NULL) {
- streamType = mAudioSink->getAudioStreamType();
- }
- if (streamType == AUDIO_STREAM_NOTIFICATION) {
- ALOGW("disabling auto-loop for notification");
- mAutoLoop = false;
- }
- }
- if (mLooping || mAutoLoop) {
- mPlayer->rewind();
- if (mAudioSink != NULL) {
- // The renderer has stopped the sink at the end in order to play out
- // the last little bit of audio. In looping mode, we need to restart it.
- mAudioSink->start();
- }
-
- sp<AMessage> notify = new AMessage(kWhatNotifyListener, this);
- notify->setInt64("srcId", srcId);
- notify->setInt32("messageId", MEDIA2_INFO);
- notify->setInt32("ext1", MEDIA2_INFO_DATA_SOURCE_REPEAT);
- notify->post();
- return;
- }
- if (property_get_bool("persist.debug.sf.stats", false)) {
- Vector<String16> args;
- dump(-1, args);
- }
- mPlayer->pause();
- mState = STATE_PAUSED;
- }
- FALLTHROUGH_INTENDED;
- }
-
- case MEDIA2_ERROR:
- {
- // when we have an error, add it to the analytics for this playback.
- // ext1 is our primary 'error type' value. Only add ext2 when non-zero.
- // [test against msg is due to fall through from previous switch value]
- if (msg == MEDIA2_ERROR) {
- mediametrics_setInt32(mMetricsHandle, kPlayerError, ext1);
- if (ext2 != 0) {
- mediametrics_setInt32(mMetricsHandle, kPlayerErrorCode, ext2);
- }
- mediametrics_setCString(mMetricsHandle, kPlayerErrorState, stateString(mState).c_str());
- }
- mAtEOS = true;
- break;
- }
-
- default:
- break;
- }
- }
-
- sp<AMessage> notify = new AMessage(kWhatNotifyListener, this);
- notify->setInt64("srcId", srcId);
- notify->setInt32("messageId", msg);
- notify->setInt32("ext1", ext1);
- notify->setInt32("ext2", ext2);
- notify->setObject("obj", PlayerMessageWrapper::Create((PlayerMessage*)in));
- notify->post();
-}
-
-void NuPlayer2Driver::notifySetDataSourceCompleted(int64_t /* srcId */, status_t err) {
- Mutex::Autolock autoLock(mLock);
-
- CHECK_EQ(mState, STATE_SET_DATASOURCE_PENDING);
-
- mAsyncResult = err;
- mState = (err == OK) ? STATE_UNPREPARED : STATE_IDLE;
- mCondition.broadcast();
-}
-
-void NuPlayer2Driver::notifyPrepareCompleted(int64_t srcId, status_t err) {
- ALOGV("notifyPrepareCompleted %d", err);
-
- Mutex::Autolock autoLock(mLock);
-
- if (srcId != mSrcId) {
- if (err == OK) {
- notifyListener_l(srcId, MEDIA2_PREPARED);
- } else {
- notifyListener_l(srcId, MEDIA2_ERROR, MEDIA2_ERROR_UNKNOWN, err);
- }
- return;
- }
-
- if (mState != STATE_PREPARING) {
- // We were preparing asynchronously when the client called
- // reset(), we sent a premature "prepared" notification and
- // then initiated the reset. This notification is stale.
- CHECK(mState == STATE_RESET_IN_PROGRESS || mState == STATE_IDLE);
- return;
- }
-
- CHECK_EQ(mState, STATE_PREPARING);
-
- mAsyncResult = err;
-
- if (err == OK) {
- // update state before notifying client, so that if client calls back into NuPlayer2Driver
- // in response, NuPlayer2Driver has the right state
- mState = STATE_PREPARED;
- notifyListener_l(srcId, MEDIA2_PREPARED);
- } else {
- mState = STATE_UNPREPARED;
- notifyListener_l(srcId, MEDIA2_ERROR, MEDIA2_ERROR_UNKNOWN, err);
- }
-
- sp<MetaData> meta = mPlayer->getFileMeta();
- int32_t loop;
- if (meta != NULL
- && meta->findInt32(kKeyAutoLoop, &loop) && loop != 0) {
- mAutoLoop = true;
- }
-
- mCondition.broadcast();
-}
-
-void NuPlayer2Driver::notifyFlagsChanged(int64_t /* srcId */, uint32_t flags) {
- Mutex::Autolock autoLock(mLock);
-
- mPlayerFlags = flags;
-}
-
-// Modular DRM
-status_t NuPlayer2Driver::prepareDrm(
- int64_t srcId, const uint8_t uuid[16], const Vector<uint8_t> &drmSessionId)
-{
- ALOGV("prepareDrm(%p) state: %d", this, mState);
-
- // leaving the state verification for mediaplayer.cpp
- status_t ret = mPlayer->prepareDrm(srcId, uuid, drmSessionId);
-
- ALOGV("prepareDrm ret: %d", ret);
-
- return ret;
-}
-
-status_t NuPlayer2Driver::releaseDrm(int64_t srcId)
-{
- ALOGV("releaseDrm(%p) state: %d", this, mState);
-
- // leaving the state verification for mediaplayer.cpp
- status_t ret = mPlayer->releaseDrm(srcId);
-
- ALOGV("releaseDrm ret: %d", ret);
-
- return ret;
-}
-
-std::string NuPlayer2Driver::stateString(State state) {
- const char *rval = NULL;
- char rawbuffer[16]; // allows "%d"
-
- switch (state) {
- case STATE_IDLE: rval = "IDLE"; break;
- case STATE_SET_DATASOURCE_PENDING: rval = "SET_DATASOURCE_PENDING"; break;
- case STATE_UNPREPARED: rval = "UNPREPARED"; break;
- case STATE_PREPARING: rval = "PREPARING"; break;
- case STATE_PREPARED: rval = "PREPARED"; break;
- case STATE_RUNNING: rval = "RUNNING"; break;
- case STATE_PAUSED: rval = "PAUSED"; break;
- case STATE_RESET_IN_PROGRESS: rval = "RESET_IN_PROGRESS"; break;
- default:
- // yes, this buffer is shared and vulnerable to races
- snprintf(rawbuffer, sizeof(rawbuffer), "%d", state);
- rval = rawbuffer;
- break;
- }
-
- return rval;
-}
-
-} // namespace android
diff --git a/media/libmediaplayer2/nuplayer2/NuPlayer2Driver.h b/media/libmediaplayer2/nuplayer2/NuPlayer2Driver.h
deleted file mode 100644
index c97e247..0000000
--- a/media/libmediaplayer2/nuplayer2/NuPlayer2Driver.h
+++ /dev/null
@@ -1,156 +0,0 @@
-/*
- * Copyright 2017 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include <mediaplayer2/MediaPlayer2Interface.h>
-
-#include <media/MediaMetrics.h>
-#include <media/stagefright/foundation/ABase.h>
-#include <mediaplayer2/JObjectHolder.h>
-
-namespace android {
-
-struct ALooper;
-struct MediaClock;
-struct NuPlayer2;
-
-struct NuPlayer2Driver : public MediaPlayer2Interface {
- explicit NuPlayer2Driver(pid_t pid, uid_t uid, const sp<JObjectHolder> &context);
-
- virtual status_t initCheck() override;
-
- virtual status_t setDataSource(const sp<DataSourceDesc> &dsd) override;
- virtual status_t prepareNextDataSource(const sp<DataSourceDesc> &dsd) override;
- virtual status_t playNextDataSource(int64_t srcId) override;
-
- virtual status_t setVideoSurfaceTexture(const sp<ANativeWindowWrapper> &nww) override;
-
- virtual status_t getBufferingSettings(
- BufferingSettings* buffering /* nonnull */) override;
- virtual status_t setBufferingSettings(const BufferingSettings& buffering) override;
-
- virtual status_t prepareAsync() override;
- virtual status_t start() override;
- virtual status_t pause() override;
- virtual bool isPlaying() override;
- virtual status_t setPlaybackSettings(const AudioPlaybackRate &rate) override;
- virtual status_t getPlaybackSettings(AudioPlaybackRate *rate) override;
- virtual status_t setSyncSettings(const AVSyncSettings &sync, float videoFpsHint) override;
- virtual status_t getSyncSettings(AVSyncSettings *sync, float *videoFps) override;
- virtual status_t seekTo(
- int64_t msec,
- MediaPlayer2SeekMode mode = MediaPlayer2SeekMode::SEEK_PREVIOUS_SYNC) override;
- virtual status_t getCurrentPosition(int64_t *msec) override;
- virtual status_t getDuration(int64_t *msec) override;
- virtual status_t reset() override;
- virtual status_t notifyAt(int64_t mediaTimeUs) override;
- virtual status_t setLooping(int loop) override;
- virtual status_t invoke(const PlayerMessage &request, PlayerMessage *response) override;
- virtual void setAudioSink(const sp<AudioSink> &audioSink) override;
- virtual status_t setParameter(int key, const Parcel &request) override;
- virtual status_t getParameter(int key, Parcel *reply) override;
- virtual status_t getMetrics(char **buf, size_t *length) override;
-
- virtual status_t dump(int fd, const Vector<String16> &args) const override;
-
- virtual void onMessageReceived(const sp<AMessage> &msg) override;
-
- void notifySetDataSourceCompleted(int64_t srcId, status_t err);
- void notifyPrepareCompleted(int64_t srcId, status_t err);
- void notifyResetComplete(int64_t srcId);
- void notifySetSurfaceComplete(int64_t srcId);
- void notifyDuration(int64_t srcId, int64_t durationUs);
- void notifyMorePlayingTimeUs(int64_t srcId, int64_t timeUs);
- void notifyMoreRebufferingTimeUs(int64_t srcId, int64_t timeUs);
- void notifyRebufferingWhenExit(int64_t srcId, bool status);
- void notifySeekComplete(int64_t srcId);
- void notifyListener(int64_t srcId, int msg, int ext1 = 0, int ext2 = 0,
- const PlayerMessage *in = NULL);
- void notifyFlagsChanged(int64_t srcId, uint32_t flags);
-
- // Modular DRM
- virtual status_t prepareDrm(
- int64_t srcId, const uint8_t uuid[16], const Vector<uint8_t> &drmSessionId);
- virtual status_t releaseDrm(int64_t srcId);
-
-protected:
- virtual ~NuPlayer2Driver();
-
-private:
- enum State {
- STATE_IDLE,
- STATE_SET_DATASOURCE_PENDING,
- STATE_UNPREPARED,
- STATE_PREPARING,
- STATE_PREPARED,
- STATE_RUNNING,
- STATE_PAUSED,
- STATE_RESET_IN_PROGRESS,
- };
-
- std::string stateString(State state);
-
- enum {
- kWhatNotifyListener,
- };
-
- mutable Mutex mLock;
- Condition mCondition;
-
- State mState;
-
- status_t mAsyncResult;
-
- // The following are protected through "mLock"
- // >>>
- int64_t mSrcId;
- bool mSetSurfaceInProgress;
- int64_t mDurationUs;
- int64_t mPositionUs;
- bool mSeekInProgress;
- int64_t mPlayingTimeUs;
- int64_t mRebufferingTimeUs;
- int32_t mRebufferingEvents;
- bool mRebufferingAtExit;
- // <<<
-
- sp<ALooper> mLooper;
- sp<ALooper> mNuPlayer2Looper;
- const sp<MediaClock> mMediaClock;
- const sp<NuPlayer2> mPlayer;
- sp<AudioSink> mAudioSink;
- uint32_t mPlayerFlags;
-
- mediametrics_handle_t mMetricsHandle;
- int64_t mPlayerVersion;
- uid_t mClientUid;
-
- bool mAtEOS;
- bool mLooping;
- bool mAutoLoop;
-
- void updateMetrics(const char *where);
- void logMetrics(const char *where);
-
- status_t start_l();
- void notifyListener_l(int64_t srcId, int msg, int ext1 = 0, int ext2 = 0,
- const PlayerMessage *in = NULL);
-
- DISALLOW_EVIL_CONSTRUCTORS(NuPlayer2Driver);
-};
-
-} // namespace android
-
-
diff --git a/media/libmediaplayer2/nuplayer2/NuPlayer2Drm.cpp b/media/libmediaplayer2/nuplayer2/NuPlayer2Drm.cpp
deleted file mode 100644
index f41a431..0000000
--- a/media/libmediaplayer2/nuplayer2/NuPlayer2Drm.cpp
+++ /dev/null
@@ -1,172 +0,0 @@
-/*
- * Copyright 2017 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "NuPlayer2Drm"
-
-#include "NuPlayer2Drm.h"
-
-#include <media/NdkWrapper.h>
-#include <utils/Log.h>
-#include <sstream>
-
-namespace android {
-
-Vector<DrmUUID> NuPlayer2Drm::parsePSSH(const void *pssh, size_t psshsize)
-{
- Vector<DrmUUID> drmSchemes, empty;
- const int DATALEN_SIZE = 4;
-
- // the format of the buffer is 1 or more of:
- // {
- // 16 byte uuid
- // 4 byte data length N
- // N bytes of data
- // }
- // Determine the number of entries in the source data.
- // Since we got the data from stagefright, we trust it is valid and properly formatted.
-
- const uint8_t *data = (const uint8_t*)pssh;
- size_t len = psshsize;
- size_t numentries = 0;
- while (len > 0) {
- if (len < DrmUUID::UUID_SIZE) {
- ALOGE("ParsePSSH: invalid PSSH data");
- return empty;
- }
-
- const uint8_t *uuidPtr = data;
-
- // skip uuid
- data += DrmUUID::UUID_SIZE;
- len -= DrmUUID::UUID_SIZE;
-
- // get data length
- if (len < DATALEN_SIZE) {
- ALOGE("ParsePSSH: invalid PSSH data");
- return empty;
- }
-
- uint32_t datalen = *((uint32_t*)data);
- data += DATALEN_SIZE;
- len -= DATALEN_SIZE;
-
- if (len < datalen) {
- ALOGE("ParsePSSH: invalid PSSH data");
- return empty;
- }
-
- // skip the data
- data += datalen;
- len -= datalen;
-
- DrmUUID _uuid(uuidPtr);
- drmSchemes.add(_uuid);
-
- ALOGV("ParsePSSH[%zu]: %s: %s", numentries,
- _uuid.toHexString().string(),
- DrmUUID::arrayToHex(data, datalen).string()
- );
-
- numentries++;
- }
-
- return drmSchemes;
-}
-
-Vector<DrmUUID> NuPlayer2Drm::getSupportedDrmSchemes(const void *pssh, size_t psshsize)
-{
- Vector<DrmUUID> psshDRMs = parsePSSH(pssh, psshsize);
-
- Vector<DrmUUID> supportedDRMs;
- for (size_t i = 0; i < psshDRMs.size(); i++) {
- DrmUUID uuid = psshDRMs[i];
- if (AMediaDrmWrapper::isCryptoSchemeSupported(uuid.ptr(), NULL)) {
- supportedDRMs.add(uuid);
- }
- }
-
- ALOGV("getSupportedDrmSchemes: psshDRMs: %zu supportedDRMs: %zu",
- psshDRMs.size(), supportedDRMs.size());
-
- return supportedDRMs;
-}
-
-sp<ABuffer> NuPlayer2Drm::retrieveDrmInfo(const void *pssh, uint32_t psshsize)
-{
- std::ostringstream buf;
-
- // 1) PSSH bytes
- buf.write(reinterpret_cast<const char *>(&psshsize), sizeof(psshsize));
- buf.write(reinterpret_cast<const char *>(pssh), psshsize);
-
- ALOGV("retrieveDrmInfo: MEDIA2_DRM_INFO PSSH: size: %u %s", psshsize,
- DrmUUID::arrayToHex((uint8_t*)pssh, psshsize).string());
-
- // 2) supportedDRMs
- Vector<DrmUUID> supportedDRMs = getSupportedDrmSchemes(pssh, psshsize);
- uint32_t n = supportedDRMs.size();
- buf.write(reinterpret_cast<char *>(&n), sizeof(n));
- for (size_t i = 0; i < n; i++) {
- DrmUUID uuid = supportedDRMs[i];
- buf.write(reinterpret_cast<const char *>(&n), sizeof(n));
- buf.write(reinterpret_cast<const char *>(uuid.ptr()), DrmUUID::UUID_SIZE);
-
- ALOGV("retrieveDrmInfo: MEDIA2_DRM_INFO supportedScheme[%zu] %s", i,
- uuid.toHexString().string());
- }
-
- sp<ABuffer> drmInfoBuffer = ABuffer::CreateAsCopy(buf.str().c_str(), buf.tellp());
- return drmInfoBuffer;
-}
-
-status_t NuPlayer2Drm::retrieveDrmInfo(PsshInfo *psshInfo, PlayerMessage *playerMsg)
-{
- std::ostringstream pssh, drmInfo;
-
- // 0) Generate PSSH bytes
- for (size_t i = 0; i < psshInfo->numentries; i++) {
- PsshEntry *entry = &psshInfo->entries[i];
- uint32_t datalen = entry->datalen;
- pssh.write(reinterpret_cast<const char *>(&entry->uuid), sizeof(entry->uuid));
- pssh.write(reinterpret_cast<const char *>(&datalen), sizeof(datalen));
- pssh.write(reinterpret_cast<const char *>(entry->data), datalen);
- }
-
- uint32_t psshSize = pssh.tellp();
- std::string psshBase = pssh.str();
- const auto* psshPtr = reinterpret_cast<const uint8_t*>(psshBase.c_str());
- ALOGV("retrieveDrmInfo: MEDIA_DRM_INFO PSSH: size: %u %s", psshSize,
- DrmUUID::arrayToHex(psshPtr, psshSize).string());
-
- // 1) Write PSSH bytes
- playerMsg->add_values()->set_bytes_value(
- reinterpret_cast<const char *>(pssh.str().c_str()), psshSize);
-
- // 2) Write supportedDRMs
- uint32_t numentries = psshInfo->numentries;
- playerMsg->add_values()->set_int32_value(numentries);
- for (size_t i = 0; i < numentries; i++) {
- PsshEntry *entry = &psshInfo->entries[i];
- playerMsg->add_values()->set_bytes_value(
- reinterpret_cast<const char *>(&entry->uuid), sizeof(entry->uuid));
- ALOGV("retrieveDrmInfo: MEDIA_DRM_INFO supportedScheme[%zu] %s", i,
- DrmUUID::arrayToHex((const uint8_t*)&entry->uuid, sizeof(AMediaUUID)).string());
- }
- return OK;
-}
-
-} // namespace android
diff --git a/media/libmediaplayer2/nuplayer2/NuPlayer2Drm.h b/media/libmediaplayer2/nuplayer2/NuPlayer2Drm.h
deleted file mode 100644
index 968d1be..0000000
--- a/media/libmediaplayer2/nuplayer2/NuPlayer2Drm.h
+++ /dev/null
@@ -1,93 +0,0 @@
-/*
- * Copyright 2017 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef NUPLAYER2_DRM_H_
-#define NUPLAYER2_DRM_H_
-
-#include <media/NdkMediaExtractor.h>
-#include <media/stagefright/foundation/ABuffer.h>
-
-#include <utils/String8.h>
-#include <utils/Vector.h>
-
-#include "mediaplayer2.pb.h"
-
-using android::media::MediaPlayer2Proto::PlayerMessage;
-
-namespace android {
-
- struct DrmUUID {
- static const int UUID_SIZE = 16;
-
- DrmUUID() {
- memset(this->uuid, 0, sizeof(uuid));
- }
-
- // to allow defining Vector/KeyedVector of UUID type
- DrmUUID(const DrmUUID &a) {
- memcpy(this->uuid, a.uuid, sizeof(uuid));
- }
-
- // to allow defining Vector/KeyedVector of UUID type
- DrmUUID(const uint8_t uuid_in[UUID_SIZE]) {
- memcpy(this->uuid, uuid_in, sizeof(uuid));
- }
-
- const uint8_t *ptr() const {
- return uuid;
- }
-
- String8 toHexString() const {
- return arrayToHex(uuid, UUID_SIZE);
- }
-
- static String8 toHexString(const uint8_t uuid_in[UUID_SIZE]) {
- return arrayToHex(uuid_in, UUID_SIZE);
- }
-
- static String8 arrayToHex(const uint8_t *array, int bytes) {
- String8 result;
- for (int i = 0; i < bytes; i++) {
- result.appendFormat("%02x", array[i]);
- }
-
- return result;
- }
-
- protected:
- uint8_t uuid[UUID_SIZE];
- };
-
-
- struct NuPlayer2Drm {
-
- // static helpers - internal
-
- protected:
- static Vector<DrmUUID> parsePSSH(const void *pssh, size_t psshsize);
- static Vector<DrmUUID> getSupportedDrmSchemes(const void *pssh, size_t psshsize);
-
- // static helpers - public
-
- public:
- static sp<ABuffer> retrieveDrmInfo(const void *pssh, uint32_t psshsize);
- static status_t retrieveDrmInfo(PsshInfo *, PlayerMessage *);
-
- }; // NuPlayer2Drm
-
-} // android
-
-#endif //NUPLAYER2_DRM_H_
diff --git a/media/libmediaplayer2/nuplayer2/NuPlayer2Renderer.cpp b/media/libmediaplayer2/nuplayer2/NuPlayer2Renderer.cpp
deleted file mode 100644
index fd459df..0000000
--- a/media/libmediaplayer2/nuplayer2/NuPlayer2Renderer.cpp
+++ /dev/null
@@ -1,2096 +0,0 @@
-/*
- * Copyright (C) 2010 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "NuPlayer2Renderer"
-#include <utils/Log.h>
-
-#include "JWakeLock.h"
-#include "NuPlayer2Renderer.h"
-#include <algorithm>
-#include <cutils/properties.h>
-#include <media/stagefright/foundation/ADebug.h>
-#include <media/stagefright/foundation/AMessage.h>
-#include <media/stagefright/foundation/AUtils.h>
-#include <media/stagefright/MediaClock.h>
-#include <media/stagefright/MediaCodecConstants.h>
-#include <media/stagefright/MediaDefs.h>
-#include <media/stagefright/MediaErrors.h>
-#include <media/stagefright/Utils.h>
-#include <media/stagefright/VideoFrameScheduler2.h>
-#include <media/MediaCodecBuffer.h>
-
-#include <inttypes.h>
-
-namespace android {
-
-/*
- * Example of common configuration settings in shell script form
-
- #Turn offload audio off (use PCM for Play Music) -- AudioPolicyManager
- adb shell setprop audio.offload.disable 1
-
- #Allow offload audio with video (requires offloading to be enabled) -- AudioPolicyManager
- adb shell setprop audio.offload.video 1
-
- #Use audio callbacks for PCM data
- adb shell setprop media.stagefright.audio.cbk 1
-
- #Use deep buffer for PCM data with video (it is generally enabled for audio-only)
- adb shell setprop media.stagefright.audio.deep 1
-
- #Set size of buffers for pcm audio sink in msec (example: 1000 msec)
- adb shell setprop media.stagefright.audio.sink 1000
-
- * These configurations take effect for the next track played (not the current track).
- */
-
-static inline bool getUseAudioCallbackSetting() {
- return property_get_bool("media.stagefright.audio.cbk", false /* default_value */);
-}
-
-static inline int32_t getAudioSinkPcmMsSetting() {
- return property_get_int32(
- "media.stagefright.audio.sink", 500 /* default_value */);
-}
-
-// Maximum time in paused state when offloading audio decompression. When elapsed, the AudioSink
-// is closed to allow the audio DSP to power down.
-static const int64_t kOffloadPauseMaxUs = 10000000LL;
-
-// Maximum allowed delay from AudioSink, 1.5 seconds.
-static const int64_t kMaxAllowedAudioSinkDelayUs = 1500000LL;
-
-static const int64_t kMinimumAudioClockUpdatePeriodUs = 20 /* msec */ * 1000;
-
-// Default video frame display duration when only video exists.
-// Used to set max media time in MediaClock.
-static const int64_t kDefaultVideoFrameIntervalUs = 100000LL;
-
-// static
-const NuPlayer2::Renderer::PcmInfo NuPlayer2::Renderer::AUDIO_PCMINFO_INITIALIZER = {
- AUDIO_CHANNEL_NONE,
- AUDIO_OUTPUT_FLAG_NONE,
- AUDIO_FORMAT_INVALID,
- 0, // mNumChannels
- 0 // mSampleRate
-};
-
-// static
-const int64_t NuPlayer2::Renderer::kMinPositionUpdateDelayUs = 100000LL;
-
-static audio_format_t constexpr audioFormatFromEncoding(int32_t pcmEncoding) {
- switch (pcmEncoding) {
- case kAudioEncodingPcmFloat:
- return AUDIO_FORMAT_PCM_FLOAT;
- case kAudioEncodingPcm16bit:
- return AUDIO_FORMAT_PCM_16_BIT;
- case kAudioEncodingPcm8bit:
- return AUDIO_FORMAT_PCM_8_BIT; // TODO: do we want to support this?
- default:
- ALOGE("%s: Invalid encoding: %d", __func__, pcmEncoding);
- return AUDIO_FORMAT_INVALID;
- }
-}
-
-NuPlayer2::Renderer::Renderer(
- const sp<MediaPlayer2Interface::AudioSink> &sink,
- const sp<MediaClock> &mediaClock,
- const sp<AMessage> ¬ify,
- const sp<JObjectHolder> &context,
- uint32_t flags)
- : mAudioSink(sink),
- mUseVirtualAudioSink(false),
- mNotify(notify),
- mFlags(flags),
- mNumFramesWritten(0),
- mDrainAudioQueuePending(false),
- mDrainVideoQueuePending(false),
- mAudioQueueGeneration(0),
- mVideoQueueGeneration(0),
- mAudioDrainGeneration(0),
- mVideoDrainGeneration(0),
- mAudioEOSGeneration(0),
- mMediaClock(mediaClock),
- mPlaybackSettings(AUDIO_PLAYBACK_RATE_DEFAULT),
- mAudioFirstAnchorTimeMediaUs(-1),
- mAnchorTimeMediaUs(-1),
- mAnchorNumFramesWritten(-1),
- mVideoLateByUs(0LL),
- mNextVideoTimeMediaUs(-1),
- mHasAudio(false),
- mHasVideo(false),
- mNotifyCompleteAudio(false),
- mNotifyCompleteVideo(false),
- mSyncQueues(false),
- mPaused(true),
- mPauseDrainAudioAllowedUs(0),
- mVideoSampleReceived(false),
- mVideoRenderingStarted(false),
- mVideoRenderingStartGeneration(0),
- mAudioRenderingStartGeneration(0),
- mRenderingDataDelivered(false),
- mNextAudioClockUpdateTimeUs(-1),
- mLastAudioMediaTimeUs(-1),
- mAudioOffloadPauseTimeoutGeneration(0),
- mAudioTornDown(false),
- mCurrentOffloadInfo(AUDIO_INFO_INITIALIZER),
- mCurrentPcmInfo(AUDIO_PCMINFO_INITIALIZER),
- mTotalBuffersQueued(0),
- mLastAudioBufferDrained(0),
- mUseAudioCallback(false),
- mWakeLock(new JWakeLock(context)) {
- CHECK(mediaClock != NULL);
- mMediaClock->setPlaybackRate(mPlaybackSettings.mSpeed);
-}
-
-NuPlayer2::Renderer::~Renderer() {
- if (offloadingAudio()) {
- mAudioSink->stop();
- mAudioSink->flush();
- mAudioSink->close();
- }
-
- // Try to avoid racing condition in case callback is still on.
- Mutex::Autolock autoLock(mLock);
- if (mUseAudioCallback) {
- flushQueue(&mAudioQueue);
- flushQueue(&mVideoQueue);
- }
- mWakeLock.clear();
- mVideoScheduler.clear();
- mNotify.clear();
- mAudioSink.clear();
-}
-
-void NuPlayer2::Renderer::queueBuffer(
- bool audio,
- const sp<MediaCodecBuffer> &buffer,
- const sp<AMessage> ¬ifyConsumed) {
- sp<AMessage> msg = new AMessage(kWhatQueueBuffer, this);
- msg->setInt32("queueGeneration", getQueueGeneration(audio));
- msg->setInt32("audio", static_cast<int32_t>(audio));
- msg->setObject("buffer", buffer);
- msg->setMessage("notifyConsumed", notifyConsumed);
- msg->post();
-}
-
-void NuPlayer2::Renderer::queueEOS(bool audio, status_t finalResult) {
- CHECK_NE(finalResult, (status_t)OK);
-
- sp<AMessage> msg = new AMessage(kWhatQueueEOS, this);
- msg->setInt32("queueGeneration", getQueueGeneration(audio));
- msg->setInt32("audio", static_cast<int32_t>(audio));
- msg->setInt32("finalResult", finalResult);
- msg->post();
-}
-
-status_t NuPlayer2::Renderer::setPlaybackSettings(const AudioPlaybackRate &rate) {
- sp<AMessage> msg = new AMessage(kWhatConfigPlayback, this);
- writeToAMessage(msg, rate);
- sp<AMessage> response;
- status_t err = msg->postAndAwaitResponse(&response);
- if (err == OK && response != NULL) {
- CHECK(response->findInt32("err", &err));
- }
- return err;
-}
-
-status_t NuPlayer2::Renderer::onConfigPlayback(const AudioPlaybackRate &rate /* sanitized */) {
- if (rate.mSpeed <= 0.f) {
- ALOGW("playback rate cannot be %f", rate.mSpeed);
- return BAD_VALUE;
- }
-
- if (mAudioSink != NULL && mAudioSink->ready()) {
- status_t err = mAudioSink->setPlaybackRate(rate);
- if (err != OK) {
- ALOGW("failed to get playback rate from audio sink, err(%d)", err);
- return err;
- }
- }
- mPlaybackSettings = rate;
- mMediaClock->setPlaybackRate(mPlaybackSettings.mSpeed);
- return OK;
-}
-
-status_t NuPlayer2::Renderer::getPlaybackSettings(AudioPlaybackRate *rate /* nonnull */) {
- sp<AMessage> msg = new AMessage(kWhatGetPlaybackSettings, this);
- sp<AMessage> response;
- status_t err = msg->postAndAwaitResponse(&response);
- if (err == OK && response != NULL) {
- CHECK(response->findInt32("err", &err));
- if (err == OK) {
- readFromAMessage(response, rate);
- }
- }
- return err;
-}
-
-status_t NuPlayer2::Renderer::onGetPlaybackSettings(AudioPlaybackRate *rate /* nonnull */) {
- if (mAudioSink != NULL && mAudioSink->ready()) {
- status_t err = mAudioSink->getPlaybackRate(rate);
- if (err == OK) {
- if (!isAudioPlaybackRateEqual(*rate, mPlaybackSettings)) {
- ALOGW("correcting mismatch in internal/external playback rate, %f vs %f",
- rate->mSpeed, mPlaybackSettings.mSpeed);
- }
- // get playback settings used by audiosink, as it may be
- // slightly off due to audiosink not taking small changes.
- mPlaybackSettings = *rate;
- }
- return err;
- }
- *rate = mPlaybackSettings;
- return OK;
-}
-
-status_t NuPlayer2::Renderer::setSyncSettings(const AVSyncSettings &sync, float videoFpsHint) {
- sp<AMessage> msg = new AMessage(kWhatConfigSync, this);
- writeToAMessage(msg, sync, videoFpsHint);
- sp<AMessage> response;
- status_t err = msg->postAndAwaitResponse(&response);
- if (err == OK && response != NULL) {
- CHECK(response->findInt32("err", &err));
- }
- return err;
-}
-
-status_t NuPlayer2::Renderer::onConfigSync(const AVSyncSettings &sync, float videoFpsHint __unused) {
- if (sync.mSource != AVSYNC_SOURCE_DEFAULT) {
- return BAD_VALUE;
- }
- // TODO: support sync sources
- return INVALID_OPERATION;
-}
-
-status_t NuPlayer2::Renderer::getSyncSettings(AVSyncSettings *sync, float *videoFps) {
- sp<AMessage> msg = new AMessage(kWhatGetSyncSettings, this);
- sp<AMessage> response;
- status_t err = msg->postAndAwaitResponse(&response);
- if (err == OK && response != NULL) {
- CHECK(response->findInt32("err", &err));
- if (err == OK) {
- readFromAMessage(response, sync, videoFps);
- }
- }
- return err;
-}
-
-status_t NuPlayer2::Renderer::onGetSyncSettings(
- AVSyncSettings *sync /* nonnull */, float *videoFps /* nonnull */) {
- *sync = mSyncSettings;
- *videoFps = -1.f;
- return OK;
-}
-
-void NuPlayer2::Renderer::flush(bool audio, bool notifyComplete) {
- {
- Mutex::Autolock autoLock(mLock);
- if (audio) {
- mNotifyCompleteAudio |= notifyComplete;
- clearAudioFirstAnchorTime_l();
- ++mAudioQueueGeneration;
- ++mAudioDrainGeneration;
- } else {
- mNotifyCompleteVideo |= notifyComplete;
- ++mVideoQueueGeneration;
- ++mVideoDrainGeneration;
- mNextVideoTimeMediaUs = -1;
- }
-
- mMediaClock->clearAnchor();
- mVideoLateByUs = 0;
- mSyncQueues = false;
- }
-
- sp<AMessage> msg = new AMessage(kWhatFlush, this);
- msg->setInt32("audio", static_cast<int32_t>(audio));
- msg->post();
-}
-
-void NuPlayer2::Renderer::signalTimeDiscontinuity() {
-}
-
-void NuPlayer2::Renderer::signalDisableOffloadAudio() {
- (new AMessage(kWhatDisableOffloadAudio, this))->post();
-}
-
-void NuPlayer2::Renderer::signalEnableOffloadAudio() {
- (new AMessage(kWhatEnableOffloadAudio, this))->post();
-}
-
-void NuPlayer2::Renderer::pause() {
- (new AMessage(kWhatPause, this))->post();
-}
-
-void NuPlayer2::Renderer::resume() {
- (new AMessage(kWhatResume, this))->post();
-}
-
-void NuPlayer2::Renderer::setVideoFrameRate(float fps) {
- sp<AMessage> msg = new AMessage(kWhatSetVideoFrameRate, this);
- msg->setFloat("frame-rate", fps);
- msg->post();
-}
-
-// Called on any threads without mLock acquired.
-status_t NuPlayer2::Renderer::getCurrentPosition(int64_t *mediaUs) {
- status_t result = mMediaClock->getMediaTime(ALooper::GetNowUs(), mediaUs);
- if (result == OK) {
- return result;
- }
-
- // MediaClock has not started yet. Try to start it if possible.
- {
- Mutex::Autolock autoLock(mLock);
- if (mAudioFirstAnchorTimeMediaUs == -1) {
- return result;
- }
-
- AudioTimestamp ts;
- status_t res = mAudioSink->getTimestamp(ts);
- if (res != OK) {
- return result;
- }
-
- // AudioSink has rendered some frames.
- int64_t nowUs = ALooper::GetNowUs();
- int64_t nowMediaUs = mAudioSink->getPlayedOutDurationUs(nowUs)
- + mAudioFirstAnchorTimeMediaUs;
- mMediaClock->updateAnchor(nowMediaUs, nowUs, -1);
- }
-
- return mMediaClock->getMediaTime(ALooper::GetNowUs(), mediaUs);
-}
-
-void NuPlayer2::Renderer::clearAudioFirstAnchorTime_l() {
- mAudioFirstAnchorTimeMediaUs = -1;
- mMediaClock->setStartingTimeMedia(-1);
-}
-
-void NuPlayer2::Renderer::setAudioFirstAnchorTimeIfNeeded_l(int64_t mediaUs) {
- if (mAudioFirstAnchorTimeMediaUs == -1) {
- mAudioFirstAnchorTimeMediaUs = mediaUs;
- mMediaClock->setStartingTimeMedia(mediaUs);
- }
-}
-
-// Called on renderer looper.
-void NuPlayer2::Renderer::clearAnchorTime() {
- mMediaClock->clearAnchor();
- mAnchorTimeMediaUs = -1;
- mAnchorNumFramesWritten = -1;
-}
-
-void NuPlayer2::Renderer::setVideoLateByUs(int64_t lateUs) {
- Mutex::Autolock autoLock(mLock);
- mVideoLateByUs = lateUs;
-}
-
-int64_t NuPlayer2::Renderer::getVideoLateByUs() {
- Mutex::Autolock autoLock(mLock);
- return mVideoLateByUs;
-}
-
-status_t NuPlayer2::Renderer::openAudioSink(
- const sp<AMessage> &format,
- bool offloadOnly,
- bool hasVideo,
- uint32_t flags,
- bool *isOffloaded,
- bool isStreaming) {
- sp<AMessage> msg = new AMessage(kWhatOpenAudioSink, this);
- msg->setMessage("format", format);
- msg->setInt32("offload-only", offloadOnly);
- msg->setInt32("has-video", hasVideo);
- msg->setInt32("flags", flags);
- msg->setInt32("isStreaming", isStreaming);
-
- sp<AMessage> response;
- status_t postStatus = msg->postAndAwaitResponse(&response);
-
- int32_t err;
- if (postStatus != OK || response.get() == nullptr || !response->findInt32("err", &err)) {
- err = INVALID_OPERATION;
- } else if (err == OK && isOffloaded != NULL) {
- int32_t offload;
- CHECK(response->findInt32("offload", &offload));
- *isOffloaded = (offload != 0);
- }
- return err;
-}
-
-void NuPlayer2::Renderer::closeAudioSink() {
- sp<AMessage> msg = new AMessage(kWhatCloseAudioSink, this);
-
- sp<AMessage> response;
- msg->postAndAwaitResponse(&response);
-}
-
-void NuPlayer2::Renderer::changeAudioFormat(
- const sp<AMessage> &format,
- bool offloadOnly,
- bool hasVideo,
- uint32_t flags,
- bool isStreaming,
- const sp<AMessage> ¬ify) {
- sp<AMessage> meta = new AMessage;
- meta->setMessage("format", format);
- meta->setInt32("offload-only", offloadOnly);
- meta->setInt32("has-video", hasVideo);
- meta->setInt32("flags", flags);
- meta->setInt32("isStreaming", isStreaming);
-
- sp<AMessage> msg = new AMessage(kWhatChangeAudioFormat, this);
- msg->setInt32("queueGeneration", getQueueGeneration(true /* audio */));
- msg->setMessage("notify", notify);
- msg->setMessage("meta", meta);
- msg->post();
-}
-
-void NuPlayer2::Renderer::onMessageReceived(const sp<AMessage> &msg) {
- switch (msg->what()) {
- case kWhatOpenAudioSink:
- {
- sp<AMessage> format;
- CHECK(msg->findMessage("format", &format));
-
- int32_t offloadOnly;
- CHECK(msg->findInt32("offload-only", &offloadOnly));
-
- int32_t hasVideo;
- CHECK(msg->findInt32("has-video", &hasVideo));
-
- uint32_t flags;
- CHECK(msg->findInt32("flags", (int32_t *)&flags));
-
- uint32_t isStreaming;
- CHECK(msg->findInt32("isStreaming", (int32_t *)&isStreaming));
-
- status_t err = onOpenAudioSink(format, offloadOnly, hasVideo, flags, isStreaming);
-
- sp<AMessage> response = new AMessage;
- response->setInt32("err", err);
- response->setInt32("offload", offloadingAudio());
-
- sp<AReplyToken> replyID;
- CHECK(msg->senderAwaitsResponse(&replyID));
- response->postReply(replyID);
-
- break;
- }
-
- case kWhatCloseAudioSink:
- {
- sp<AReplyToken> replyID;
- CHECK(msg->senderAwaitsResponse(&replyID));
-
- onCloseAudioSink();
-
- sp<AMessage> response = new AMessage;
- response->postReply(replyID);
- break;
- }
-
- case kWhatStopAudioSink:
- {
- mAudioSink->stop();
- break;
- }
-
- case kWhatChangeAudioFormat:
- {
- int32_t queueGeneration;
- CHECK(msg->findInt32("queueGeneration", &queueGeneration));
-
- sp<AMessage> notify;
- CHECK(msg->findMessage("notify", ¬ify));
-
- if (offloadingAudio()) {
- ALOGW("changeAudioFormat should NOT be called in offload mode");
- notify->setInt32("err", INVALID_OPERATION);
- notify->post();
- break;
- }
-
- sp<AMessage> meta;
- CHECK(msg->findMessage("meta", &meta));
-
- if (queueGeneration != getQueueGeneration(true /* audio */)
- || mAudioQueue.empty()) {
- onChangeAudioFormat(meta, notify);
- break;
- }
-
- QueueEntry entry;
- entry.mNotifyConsumed = notify;
- entry.mMeta = meta;
-
- Mutex::Autolock autoLock(mLock);
- mAudioQueue.push_back(entry);
- postDrainAudioQueue_l();
-
- break;
- }
-
- case kWhatDrainAudioQueue:
- {
- mDrainAudioQueuePending = false;
-
- int32_t generation;
- CHECK(msg->findInt32("drainGeneration", &generation));
- if (generation != getDrainGeneration(true /* audio */)) {
- break;
- }
-
- if (onDrainAudioQueue()) {
- uint32_t numFramesPlayed;
- CHECK_EQ(mAudioSink->getPosition(&numFramesPlayed),
- (status_t)OK);
-
- // Handle AudioTrack race when start is immediately called after flush.
- uint32_t numFramesPendingPlayout =
- (mNumFramesWritten > numFramesPlayed ?
- mNumFramesWritten - numFramesPlayed : 0);
-
- // This is how long the audio sink will have data to
- // play back.
- int64_t delayUs =
- mAudioSink->msecsPerFrame()
- * numFramesPendingPlayout * 1000ll;
- if (mPlaybackSettings.mSpeed > 1.0f) {
- delayUs /= mPlaybackSettings.mSpeed;
- }
-
- // Let's give it more data after about half that time
- // has elapsed.
- delayUs /= 2;
- // check the buffer size to estimate maximum delay permitted.
- const int64_t maxDrainDelayUs = std::max(
- mAudioSink->getBufferDurationInUs(), (int64_t)500000 /* half second */);
- ALOGD_IF(delayUs > maxDrainDelayUs, "postDrainAudioQueue long delay: %lld > %lld",
- (long long)delayUs, (long long)maxDrainDelayUs);
- Mutex::Autolock autoLock(mLock);
- postDrainAudioQueue_l(delayUs);
- }
- break;
- }
-
- case kWhatDrainVideoQueue:
- {
- int32_t generation;
- CHECK(msg->findInt32("drainGeneration", &generation));
- if (generation != getDrainGeneration(false /* audio */)) {
- break;
- }
-
- mDrainVideoQueuePending = false;
-
- onDrainVideoQueue();
-
- postDrainVideoQueue();
- break;
- }
-
- case kWhatPostDrainVideoQueue:
- {
- int32_t generation;
- CHECK(msg->findInt32("drainGeneration", &generation));
- if (generation != getDrainGeneration(false /* audio */)) {
- break;
- }
-
- mDrainVideoQueuePending = false;
- postDrainVideoQueue();
- break;
- }
-
- case kWhatQueueBuffer:
- {
- onQueueBuffer(msg);
- break;
- }
-
- case kWhatQueueEOS:
- {
- onQueueEOS(msg);
- break;
- }
-
- case kWhatEOS:
- {
- int32_t generation;
- CHECK(msg->findInt32("audioEOSGeneration", &generation));
- if (generation != mAudioEOSGeneration) {
- break;
- }
- status_t finalResult;
- CHECK(msg->findInt32("finalResult", &finalResult));
- notifyEOS(true /* audio */, finalResult);
- break;
- }
-
- case kWhatConfigPlayback:
- {
- sp<AReplyToken> replyID;
- CHECK(msg->senderAwaitsResponse(&replyID));
- AudioPlaybackRate rate;
- readFromAMessage(msg, &rate);
- status_t err = onConfigPlayback(rate);
- sp<AMessage> response = new AMessage;
- response->setInt32("err", err);
- response->postReply(replyID);
- break;
- }
-
- case kWhatGetPlaybackSettings:
- {
- sp<AReplyToken> replyID;
- CHECK(msg->senderAwaitsResponse(&replyID));
- AudioPlaybackRate rate = AUDIO_PLAYBACK_RATE_DEFAULT;
- status_t err = onGetPlaybackSettings(&rate);
- sp<AMessage> response = new AMessage;
- if (err == OK) {
- writeToAMessage(response, rate);
- }
- response->setInt32("err", err);
- response->postReply(replyID);
- break;
- }
-
- case kWhatConfigSync:
- {
- sp<AReplyToken> replyID;
- CHECK(msg->senderAwaitsResponse(&replyID));
- AVSyncSettings sync;
- float videoFpsHint;
- readFromAMessage(msg, &sync, &videoFpsHint);
- status_t err = onConfigSync(sync, videoFpsHint);
- sp<AMessage> response = new AMessage;
- response->setInt32("err", err);
- response->postReply(replyID);
- break;
- }
-
- case kWhatGetSyncSettings:
- {
- sp<AReplyToken> replyID;
- CHECK(msg->senderAwaitsResponse(&replyID));
-
- ALOGV("kWhatGetSyncSettings");
- AVSyncSettings sync;
- float videoFps = -1.f;
- status_t err = onGetSyncSettings(&sync, &videoFps);
- sp<AMessage> response = new AMessage;
- if (err == OK) {
- writeToAMessage(response, sync, videoFps);
- }
- response->setInt32("err", err);
- response->postReply(replyID);
- break;
- }
-
- case kWhatFlush:
- {
- onFlush(msg);
- break;
- }
-
- case kWhatDisableOffloadAudio:
- {
- onDisableOffloadAudio();
- break;
- }
-
- case kWhatEnableOffloadAudio:
- {
- onEnableOffloadAudio();
- break;
- }
-
- case kWhatPause:
- {
- onPause();
- break;
- }
-
- case kWhatResume:
- {
- onResume();
- break;
- }
-
- case kWhatSetVideoFrameRate:
- {
- float fps;
- CHECK(msg->findFloat("frame-rate", &fps));
- onSetVideoFrameRate(fps);
- break;
- }
-
- case kWhatAudioTearDown:
- {
- int32_t reason;
- CHECK(msg->findInt32("reason", &reason));
-
- onAudioTearDown((AudioTearDownReason)reason);
- break;
- }
-
- case kWhatAudioOffloadPauseTimeout:
- {
- int32_t generation;
- CHECK(msg->findInt32("drainGeneration", &generation));
- if (generation != mAudioOffloadPauseTimeoutGeneration) {
- break;
- }
- ALOGV("Audio Offload tear down due to pause timeout.");
- onAudioTearDown(kDueToTimeout);
- mWakeLock->release();
- break;
- }
-
- default:
- TRESPASS();
- break;
- }
-}
-
-void NuPlayer2::Renderer::postDrainAudioQueue_l(int64_t delayUs) {
- if (mDrainAudioQueuePending || mSyncQueues || mUseAudioCallback) {
- return;
- }
-
- if (mAudioQueue.empty()) {
- return;
- }
-
- // FIXME: if paused, wait until AudioTrack stop() is complete before delivering data.
- if (mPaused) {
- const int64_t diffUs = mPauseDrainAudioAllowedUs - ALooper::GetNowUs();
- if (diffUs > delayUs) {
- delayUs = diffUs;
- }
- }
-
- mDrainAudioQueuePending = true;
- sp<AMessage> msg = new AMessage(kWhatDrainAudioQueue, this);
- msg->setInt32("drainGeneration", mAudioDrainGeneration);
- msg->post(delayUs);
-}
-
-void NuPlayer2::Renderer::prepareForMediaRenderingStart_l() {
- mAudioRenderingStartGeneration = mAudioDrainGeneration;
- mVideoRenderingStartGeneration = mVideoDrainGeneration;
- mRenderingDataDelivered = false;
-}
-
-void NuPlayer2::Renderer::notifyIfMediaRenderingStarted_l() {
- if (mVideoRenderingStartGeneration == mVideoDrainGeneration &&
- mAudioRenderingStartGeneration == mAudioDrainGeneration) {
- mRenderingDataDelivered = true;
- if (mPaused) {
- return;
- }
- mVideoRenderingStartGeneration = -1;
- mAudioRenderingStartGeneration = -1;
-
- sp<AMessage> notify = mNotify->dup();
- notify->setInt32("what", kWhatMediaRenderingStart);
- notify->post();
- }
-}
-
-// static
-size_t NuPlayer2::Renderer::AudioSinkCallback(
- MediaPlayer2Interface::AudioSink * /* audioSink */,
- void *buffer,
- size_t size,
- void *cookie,
- MediaPlayer2Interface::AudioSink::cb_event_t event) {
- NuPlayer2::Renderer *me = (NuPlayer2::Renderer *)cookie;
-
- switch (event) {
- case MediaPlayer2Interface::AudioSink::CB_EVENT_FILL_BUFFER:
- {
- return me->fillAudioBuffer(buffer, size);
- break;
- }
-
- case MediaPlayer2Interface::AudioSink::CB_EVENT_STREAM_END:
- {
- ALOGV("AudioSink::CB_EVENT_STREAM_END");
- me->notifyEOSCallback();
- break;
- }
-
- case MediaPlayer2Interface::AudioSink::CB_EVENT_TEAR_DOWN:
- {
- ALOGV("AudioSink::CB_EVENT_TEAR_DOWN");
- me->notifyAudioTearDown(kDueToError);
- break;
- }
- }
-
- return 0;
-}
-
-void NuPlayer2::Renderer::notifyEOSCallback() {
- Mutex::Autolock autoLock(mLock);
-
- if (!mUseAudioCallback) {
- return;
- }
-
- notifyEOS_l(true /* audio */, ERROR_END_OF_STREAM);
-}
-
-size_t NuPlayer2::Renderer::fillAudioBuffer(void *buffer, size_t size) {
- Mutex::Autolock autoLock(mLock);
-
- if (!mUseAudioCallback) {
- return 0;
- }
-
- bool hasEOS = false;
-
- size_t sizeCopied = 0;
- bool firstEntry = true;
- QueueEntry *entry; // will be valid after while loop if hasEOS is set.
- while (sizeCopied < size && !mAudioQueue.empty()) {
- entry = &*mAudioQueue.begin();
-
- if (entry->mBuffer == NULL) { // EOS
- hasEOS = true;
- mAudioQueue.erase(mAudioQueue.begin());
- break;
- }
-
- if (firstEntry && entry->mOffset == 0) {
- firstEntry = false;
- int64_t mediaTimeUs;
- CHECK(entry->mBuffer->meta()->findInt64("timeUs", &mediaTimeUs));
- ALOGV("fillAudioBuffer: rendering audio at media time %.2f secs", mediaTimeUs / 1E6);
- setAudioFirstAnchorTimeIfNeeded_l(mediaTimeUs);
- }
-
- size_t copy = entry->mBuffer->size() - entry->mOffset;
- size_t sizeRemaining = size - sizeCopied;
- if (copy > sizeRemaining) {
- copy = sizeRemaining;
- }
-
- memcpy((char *)buffer + sizeCopied,
- entry->mBuffer->data() + entry->mOffset,
- copy);
-
- entry->mOffset += copy;
- if (entry->mOffset == entry->mBuffer->size()) {
- entry->mNotifyConsumed->post();
- mAudioQueue.erase(mAudioQueue.begin());
- entry = NULL;
- }
- sizeCopied += copy;
-
- notifyIfMediaRenderingStarted_l();
- }
-
- if (mAudioFirstAnchorTimeMediaUs >= 0) {
- int64_t nowUs = ALooper::GetNowUs();
- int64_t nowMediaUs =
- mAudioFirstAnchorTimeMediaUs + mAudioSink->getPlayedOutDurationUs(nowUs);
- // we don't know how much data we are queueing for offloaded tracks.
- mMediaClock->updateAnchor(nowMediaUs, nowUs, INT64_MAX);
- }
-
- // for non-offloaded audio, we need to compute the frames written because
- // there is no EVENT_STREAM_END notification. The frames written gives
- // an estimate on the pending played out duration.
- if (!offloadingAudio()) {
- mNumFramesWritten += sizeCopied / mAudioSink->frameSize();
- }
-
- if (hasEOS) {
- (new AMessage(kWhatStopAudioSink, this))->post();
- // As there is currently no EVENT_STREAM_END callback notification for
- // non-offloaded audio tracks, we need to post the EOS ourselves.
- if (!offloadingAudio()) {
- int64_t postEOSDelayUs = 0;
- if (mAudioSink->needsTrailingPadding()) {
- postEOSDelayUs = getPendingAudioPlayoutDurationUs(ALooper::GetNowUs());
- }
- ALOGV("fillAudioBuffer: notifyEOS_l "
- "mNumFramesWritten:%u finalResult:%d postEOSDelay:%lld",
- mNumFramesWritten, entry->mFinalResult, (long long)postEOSDelayUs);
- notifyEOS_l(true /* audio */, entry->mFinalResult, postEOSDelayUs);
- }
- }
- return sizeCopied;
-}
-
-void NuPlayer2::Renderer::drainAudioQueueUntilLastEOS() {
- List<QueueEntry>::iterator it = mAudioQueue.begin(), itEOS = it;
- bool foundEOS = false;
- while (it != mAudioQueue.end()) {
- int32_t eos;
- QueueEntry *entry = &*it++;
- if ((entry->mBuffer == nullptr && entry->mNotifyConsumed == nullptr)
- || (entry->mNotifyConsumed->findInt32("eos", &eos) && eos != 0)) {
- itEOS = it;
- foundEOS = true;
- }
- }
-
- if (foundEOS) {
- // post all replies before EOS and drop the samples
- for (it = mAudioQueue.begin(); it != itEOS; it++) {
- if (it->mBuffer == nullptr) {
- if (it->mNotifyConsumed == nullptr) {
- // delay doesn't matter as we don't even have an AudioTrack
- notifyEOS(true /* audio */, it->mFinalResult);
- } else {
- // TAG for re-opening audio sink.
- onChangeAudioFormat(it->mMeta, it->mNotifyConsumed);
- }
- } else {
- it->mNotifyConsumed->post();
- }
- }
- mAudioQueue.erase(mAudioQueue.begin(), itEOS);
- }
-}
-
-bool NuPlayer2::Renderer::onDrainAudioQueue() {
- // do not drain audio during teardown as queued buffers may be invalid.
- if (mAudioTornDown) {
- return false;
- }
- // TODO: This call to getPosition checks if AudioTrack has been created
- // in AudioSink before draining audio. If AudioTrack doesn't exist, then
- // CHECKs on getPosition will fail.
- // We still need to figure out why AudioTrack is not created when
- // this function is called. One possible reason could be leftover
- // audio. Another possible place is to check whether decoder
- // has received INFO_FORMAT_CHANGED as the first buffer since
- // AudioSink is opened there, and possible interactions with flush
- // immediately after start. Investigate error message
- // "vorbis_dsp_synthesis returned -135", along with RTSP.
- uint32_t numFramesPlayed;
- if (mAudioSink->getPosition(&numFramesPlayed) != OK) {
- // When getPosition fails, renderer will not reschedule the draining
- // unless new samples are queued.
- // If we have pending EOS (or "eos" marker for discontinuities), we need
- // to post these now as NuPlayer2Decoder might be waiting for it.
- drainAudioQueueUntilLastEOS();
-
- ALOGW("onDrainAudioQueue(): audio sink is not ready");
- return false;
- }
-
-#if 0
- ssize_t numFramesAvailableToWrite =
- mAudioSink->frameCount() - (mNumFramesWritten - numFramesPlayed);
-
- if (numFramesAvailableToWrite == mAudioSink->frameCount()) {
- ALOGI("audio sink underrun");
- } else {
- ALOGV("audio queue has %d frames left to play",
- mAudioSink->frameCount() - numFramesAvailableToWrite);
- }
-#endif
-
- uint32_t prevFramesWritten = mNumFramesWritten;
- while (!mAudioQueue.empty()) {
- QueueEntry *entry = &*mAudioQueue.begin();
-
- if (entry->mBuffer == NULL) {
- if (entry->mNotifyConsumed != nullptr) {
- // TAG for re-open audio sink.
- onChangeAudioFormat(entry->mMeta, entry->mNotifyConsumed);
- mAudioQueue.erase(mAudioQueue.begin());
- continue;
- }
-
- // EOS
- if (mPaused) {
- // Do not notify EOS when paused.
- // This is needed to avoid switch to next clip while in pause.
- ALOGV("onDrainAudioQueue(): Do not notify EOS when paused");
- return false;
- }
-
- int64_t postEOSDelayUs = 0;
- if (mAudioSink->needsTrailingPadding()) {
- postEOSDelayUs = getPendingAudioPlayoutDurationUs(ALooper::GetNowUs());
- }
- notifyEOS(true /* audio */, entry->mFinalResult, postEOSDelayUs);
- mLastAudioMediaTimeUs = getDurationUsIfPlayedAtSampleRate(mNumFramesWritten);
-
- mAudioQueue.erase(mAudioQueue.begin());
- entry = NULL;
- if (mAudioSink->needsTrailingPadding()) {
- // If we're not in gapless playback (i.e. through setNextPlayer), we
- // need to stop the track here, because that will play out the last
- // little bit at the end of the file. Otherwise short files won't play.
- mAudioSink->stop();
- mNumFramesWritten = 0;
- }
- return false;
- }
-
- mLastAudioBufferDrained = entry->mBufferOrdinal;
-
- // ignore 0-sized buffer which could be EOS marker with no data
- if (entry->mOffset == 0 && entry->mBuffer->size() > 0) {
- int64_t mediaTimeUs;
- CHECK(entry->mBuffer->meta()->findInt64("timeUs", &mediaTimeUs));
- ALOGV("onDrainAudioQueue: rendering audio at media time %.2f secs",
- mediaTimeUs / 1E6);
- onNewAudioMediaTime(mediaTimeUs);
- }
-
- size_t copy = entry->mBuffer->size() - entry->mOffset;
-
- ssize_t written = mAudioSink->write(entry->mBuffer->data() + entry->mOffset,
- copy, false /* blocking */);
- if (written < 0) {
- // An error in AudioSink write. Perhaps the AudioSink was not properly opened.
- if (written == WOULD_BLOCK) {
- ALOGV("AudioSink write would block when writing %zu bytes", copy);
- } else {
- ALOGE("AudioSink write error(%zd) when writing %zu bytes", written, copy);
- // This can only happen when AudioSink was opened with doNotReconnect flag set to
- // true, in which case the NuPlayer2 will handle the reconnect.
- notifyAudioTearDown(kDueToError);
- }
- break;
- }
-
- entry->mOffset += written;
- size_t remainder = entry->mBuffer->size() - entry->mOffset;
- if ((ssize_t)remainder < mAudioSink->frameSize()) {
- if (remainder > 0) {
- ALOGW("Corrupted audio buffer has fractional frames, discarding %zu bytes.",
- remainder);
- entry->mOffset += remainder;
- copy -= remainder;
- }
-
- entry->mNotifyConsumed->post();
- mAudioQueue.erase(mAudioQueue.begin());
-
- entry = NULL;
- }
-
- size_t copiedFrames = written / mAudioSink->frameSize();
- mNumFramesWritten += copiedFrames;
-
- {
- Mutex::Autolock autoLock(mLock);
- int64_t maxTimeMedia;
- maxTimeMedia =
- mAnchorTimeMediaUs +
- (int64_t)(max((long long)mNumFramesWritten - mAnchorNumFramesWritten, 0LL)
- * 1000LL * mAudioSink->msecsPerFrame());
- mMediaClock->updateMaxTimeMedia(maxTimeMedia);
-
- notifyIfMediaRenderingStarted_l();
- }
-
- if (written != (ssize_t)copy) {
- // A short count was received from AudioSink::write()
- //
- // AudioSink write is called in non-blocking mode.
- // It may return with a short count when:
- //
- // 1) Size to be copied is not a multiple of the frame size. Fractional frames are
- // discarded.
- // 2) The data to be copied exceeds the available buffer in AudioSink.
- // 3) An error occurs and data has been partially copied to the buffer in AudioSink.
- // 4) AudioSink is an AudioCache for data retrieval, and the AudioCache is exceeded.
-
- // (Case 1)
- // Must be a multiple of the frame size. If it is not a multiple of a frame size, it
- // needs to fail, as we should not carry over fractional frames between calls.
- CHECK_EQ(copy % mAudioSink->frameSize(), 0u);
-
- // (Case 2, 3, 4)
- // Return early to the caller.
- // Beware of calling immediately again as this may busy-loop if you are not careful.
- ALOGV("AudioSink write short frame count %zd < %zu", written, copy);
- break;
- }
- }
-
- // calculate whether we need to reschedule another write.
- bool reschedule = !mAudioQueue.empty()
- && (!mPaused
- || prevFramesWritten != mNumFramesWritten); // permit pause to fill buffers
- //ALOGD("reschedule:%d empty:%d mPaused:%d prevFramesWritten:%u mNumFramesWritten:%u",
- // reschedule, mAudioQueue.empty(), mPaused, prevFramesWritten, mNumFramesWritten);
- return reschedule;
-}
-
-int64_t NuPlayer2::Renderer::getDurationUsIfPlayedAtSampleRate(uint32_t numFrames) {
- int32_t sampleRate = offloadingAudio() ?
- mCurrentOffloadInfo.sample_rate : mCurrentPcmInfo.mSampleRate;
- if (sampleRate == 0) {
- ALOGE("sampleRate is 0 in %s mode", offloadingAudio() ? "offload" : "non-offload");
- return 0;
- }
- return (int64_t)(numFrames * 1000000LL / sampleRate);
-}
-
-// Calculate duration of pending samples if played at normal rate (i.e., 1.0).
-int64_t NuPlayer2::Renderer::getPendingAudioPlayoutDurationUs(int64_t nowUs) {
- int64_t writtenAudioDurationUs = getDurationUsIfPlayedAtSampleRate(mNumFramesWritten);
- if (mUseVirtualAudioSink) {
- int64_t nowUs = ALooper::GetNowUs();
- int64_t mediaUs;
- if (mMediaClock->getMediaTime(nowUs, &mediaUs) != OK) {
- return 0LL;
- } else {
- return writtenAudioDurationUs - (mediaUs - mAudioFirstAnchorTimeMediaUs);
- }
- }
-
- const int64_t audioSinkPlayedUs = mAudioSink->getPlayedOutDurationUs(nowUs);
- int64_t pendingUs = writtenAudioDurationUs - audioSinkPlayedUs;
- if (pendingUs < 0) {
- // This shouldn't happen unless the timestamp is stale.
- ALOGW("%s: pendingUs %lld < 0, clamping to zero, potential resume after pause "
- "writtenAudioDurationUs: %lld, audioSinkPlayedUs: %lld",
- __func__, (long long)pendingUs,
- (long long)writtenAudioDurationUs, (long long)audioSinkPlayedUs);
- pendingUs = 0;
- }
- return pendingUs;
-}
-
-int64_t NuPlayer2::Renderer::getRealTimeUs(int64_t mediaTimeUs, int64_t nowUs) {
- int64_t realUs;
- if (mMediaClock->getRealTimeFor(mediaTimeUs, &realUs) != OK) {
- // If failed to get current position, e.g. due to audio clock is
- // not ready, then just play out video immediately without delay.
- return nowUs;
- }
- return realUs;
-}
-
-void NuPlayer2::Renderer::onNewAudioMediaTime(int64_t mediaTimeUs) {
- Mutex::Autolock autoLock(mLock);
- // TRICKY: vorbis decoder generates multiple frames with the same
- // timestamp, so only update on the first frame with a given timestamp
- if (mediaTimeUs == mAnchorTimeMediaUs) {
- return;
- }
- setAudioFirstAnchorTimeIfNeeded_l(mediaTimeUs);
-
- // mNextAudioClockUpdateTimeUs is -1 if we're waiting for audio sink to start
- if (mNextAudioClockUpdateTimeUs == -1) {
- AudioTimestamp ts;
- if (mAudioSink->getTimestamp(ts) == OK && ts.mPosition > 0) {
- mNextAudioClockUpdateTimeUs = 0; // start our clock updates
- }
- }
- int64_t nowUs = ALooper::GetNowUs();
- if (mNextAudioClockUpdateTimeUs >= 0) {
- if (nowUs >= mNextAudioClockUpdateTimeUs) {
- int64_t nowMediaUs = mediaTimeUs - getPendingAudioPlayoutDurationUs(nowUs);
- mMediaClock->updateAnchor(nowMediaUs, nowUs, mediaTimeUs);
- mUseVirtualAudioSink = false;
- mNextAudioClockUpdateTimeUs = nowUs + kMinimumAudioClockUpdatePeriodUs;
- }
- } else {
- int64_t unused;
- if ((mMediaClock->getMediaTime(nowUs, &unused) != OK)
- && (getDurationUsIfPlayedAtSampleRate(mNumFramesWritten)
- > kMaxAllowedAudioSinkDelayUs)) {
- // Enough data has been sent to AudioSink, but AudioSink has not rendered
- // any data yet. Something is wrong with AudioSink, e.g., the device is not
- // connected to audio out.
- // Switch to system clock. This essentially creates a virtual AudioSink with
- // initial latenty of getDurationUsIfPlayedAtSampleRate(mNumFramesWritten).
- // This virtual AudioSink renders audio data starting from the very first sample
- // and it's paced by system clock.
- ALOGW("AudioSink stuck. ARE YOU CONNECTED TO AUDIO OUT? Switching to system clock.");
- mMediaClock->updateAnchor(mAudioFirstAnchorTimeMediaUs, nowUs, mediaTimeUs);
- mUseVirtualAudioSink = true;
- }
- }
- mAnchorNumFramesWritten = mNumFramesWritten;
- mAnchorTimeMediaUs = mediaTimeUs;
-}
-
-// Called without mLock acquired.
-void NuPlayer2::Renderer::postDrainVideoQueue() {
- if (mDrainVideoQueuePending
- || getSyncQueues()
- || (mPaused && mVideoSampleReceived)) {
- return;
- }
-
- if (mVideoQueue.empty()) {
- return;
- }
-
- QueueEntry &entry = *mVideoQueue.begin();
-
- sp<AMessage> msg = new AMessage(kWhatDrainVideoQueue, this);
- msg->setInt32("drainGeneration", getDrainGeneration(false /* audio */));
-
- if (entry.mBuffer == NULL) {
- // EOS doesn't carry a timestamp.
- msg->post();
- mDrainVideoQueuePending = true;
- return;
- }
-
- int64_t nowUs = ALooper::GetNowUs();
- if (mFlags & FLAG_REAL_TIME) {
- int64_t realTimeUs;
- CHECK(entry.mBuffer->meta()->findInt64("timeUs", &realTimeUs));
-
- realTimeUs = mVideoScheduler->schedule(realTimeUs * 1000) / 1000;
-
- int64_t twoVsyncsUs = 2 * (mVideoScheduler->getVsyncPeriod() / 1000);
-
- int64_t delayUs = realTimeUs - nowUs;
-
- ALOGW_IF(delayUs > 500000, "unusually high delayUs: %lld", (long long)delayUs);
- // post 2 display refreshes before rendering is due
- msg->post(delayUs > twoVsyncsUs ? delayUs - twoVsyncsUs : 0);
-
- mDrainVideoQueuePending = true;
- return;
- }
-
- int64_t mediaTimeUs;
- CHECK(entry.mBuffer->meta()->findInt64("timeUs", &mediaTimeUs));
-
- {
- Mutex::Autolock autoLock(mLock);
- if (mAnchorTimeMediaUs < 0) {
- mMediaClock->updateAnchor(mediaTimeUs, nowUs, mediaTimeUs);
- mAnchorTimeMediaUs = mediaTimeUs;
- }
- }
- mNextVideoTimeMediaUs = mediaTimeUs;
- if (!mHasAudio) {
- // smooth out videos >= 10fps
- mMediaClock->updateMaxTimeMedia(mediaTimeUs + kDefaultVideoFrameIntervalUs);
- }
-
- if (!mVideoSampleReceived || mediaTimeUs < mAudioFirstAnchorTimeMediaUs) {
- msg->post();
- } else {
- int64_t twoVsyncsUs = 2 * (mVideoScheduler->getVsyncPeriod() / 1000);
-
- // post 2 display refreshes before rendering is due
- mMediaClock->addTimer(msg, mediaTimeUs, -twoVsyncsUs);
- }
-
- mDrainVideoQueuePending = true;
-}
-
-void NuPlayer2::Renderer::onDrainVideoQueue() {
- if (mVideoQueue.empty()) {
- return;
- }
-
- QueueEntry *entry = &*mVideoQueue.begin();
-
- if (entry->mBuffer == NULL) {
- // EOS
-
- notifyEOS(false /* audio */, entry->mFinalResult);
-
- mVideoQueue.erase(mVideoQueue.begin());
- entry = NULL;
-
- setVideoLateByUs(0);
- return;
- }
-
- int64_t nowUs = ALooper::GetNowUs();
- int64_t realTimeUs;
- int64_t mediaTimeUs = -1;
- if (mFlags & FLAG_REAL_TIME) {
- CHECK(entry->mBuffer->meta()->findInt64("timeUs", &realTimeUs));
- } else {
- CHECK(entry->mBuffer->meta()->findInt64("timeUs", &mediaTimeUs));
-
- realTimeUs = getRealTimeUs(mediaTimeUs, nowUs);
- }
- realTimeUs = mVideoScheduler->schedule(realTimeUs * 1000) / 1000;
-
- bool tooLate = false;
-
- if (!mPaused) {
- setVideoLateByUs(nowUs - realTimeUs);
- tooLate = (mVideoLateByUs > 40000);
-
- if (tooLate) {
- ALOGV("video late by %lld us (%.2f secs)",
- (long long)mVideoLateByUs, mVideoLateByUs / 1E6);
- } else {
- int64_t mediaUs = 0;
- mMediaClock->getMediaTime(realTimeUs, &mediaUs);
- ALOGV("rendering video at media time %.2f secs",
- (mFlags & FLAG_REAL_TIME ? realTimeUs :
- mediaUs) / 1E6);
-
- if (!(mFlags & FLAG_REAL_TIME)
- && mLastAudioMediaTimeUs != -1
- && mediaTimeUs > mLastAudioMediaTimeUs) {
- // If audio ends before video, video continues to drive media clock.
- // Also smooth out videos >= 10fps.
- mMediaClock->updateMaxTimeMedia(mediaTimeUs + kDefaultVideoFrameIntervalUs);
- }
- }
- } else {
- setVideoLateByUs(0);
- if (!mVideoSampleReceived && !mHasAudio) {
- // This will ensure that the first frame after a flush won't be used as anchor
- // when renderer is in paused state, because resume can happen any time after seek.
- clearAnchorTime();
- }
- }
-
- // Always render the first video frame while keeping stats on A/V sync.
- if (!mVideoSampleReceived) {
- realTimeUs = nowUs;
- tooLate = false;
- }
-
- entry->mNotifyConsumed->setInt64("timestampNs", realTimeUs * 1000LL);
- entry->mNotifyConsumed->setInt32("render", !tooLate);
- entry->mNotifyConsumed->post();
- mVideoQueue.erase(mVideoQueue.begin());
- entry = NULL;
-
- mVideoSampleReceived = true;
-
- if (!mPaused) {
- if (!mVideoRenderingStarted) {
- mVideoRenderingStarted = true;
- notifyVideoRenderingStart();
- }
- Mutex::Autolock autoLock(mLock);
- notifyIfMediaRenderingStarted_l();
- }
-}
-
-void NuPlayer2::Renderer::notifyVideoRenderingStart() {
- sp<AMessage> notify = mNotify->dup();
- notify->setInt32("what", kWhatVideoRenderingStart);
- notify->post();
-}
-
-void NuPlayer2::Renderer::notifyEOS(bool audio, status_t finalResult, int64_t delayUs) {
- Mutex::Autolock autoLock(mLock);
- notifyEOS_l(audio, finalResult, delayUs);
-}
-
-void NuPlayer2::Renderer::notifyEOS_l(bool audio, status_t finalResult, int64_t delayUs) {
- if (audio && delayUs > 0) {
- sp<AMessage> msg = new AMessage(kWhatEOS, this);
- msg->setInt32("audioEOSGeneration", mAudioEOSGeneration);
- msg->setInt32("finalResult", finalResult);
- msg->post(delayUs);
- return;
- }
- sp<AMessage> notify = mNotify->dup();
- notify->setInt32("what", kWhatEOS);
- notify->setInt32("audio", static_cast<int32_t>(audio));
- notify->setInt32("finalResult", finalResult);
- notify->post(delayUs);
-
- if (audio) {
- // Video might outlive audio. Clear anchor to enable video only case.
- mAnchorTimeMediaUs = -1;
- mHasAudio = false;
- if (mNextVideoTimeMediaUs >= 0) {
- int64_t mediaUs = 0;
- int64_t nowUs = ALooper::GetNowUs();
- status_t result = mMediaClock->getMediaTime(nowUs, &mediaUs);
- if (result == OK) {
- if (mNextVideoTimeMediaUs > mediaUs) {
- mMediaClock->updateMaxTimeMedia(mNextVideoTimeMediaUs);
- }
- } else {
- mMediaClock->updateAnchor(
- mNextVideoTimeMediaUs, nowUs,
- mNextVideoTimeMediaUs + kDefaultVideoFrameIntervalUs);
- }
- }
- }
-}
-
-void NuPlayer2::Renderer::notifyAudioTearDown(AudioTearDownReason reason) {
- sp<AMessage> msg = new AMessage(kWhatAudioTearDown, this);
- msg->setInt32("reason", reason);
- msg->post();
-}
-
-void NuPlayer2::Renderer::onQueueBuffer(const sp<AMessage> &msg) {
- int32_t audio;
- CHECK(msg->findInt32("audio", &audio));
-
- if (dropBufferIfStale(audio, msg)) {
- return;
- }
-
- if (audio) {
- mHasAudio = true;
- } else {
- mHasVideo = true;
- }
-
- if (mHasVideo) {
- if (mVideoScheduler == NULL) {
- mVideoScheduler = new VideoFrameScheduler2();
- mVideoScheduler->init();
- }
- }
-
- sp<RefBase> obj;
- CHECK(msg->findObject("buffer", &obj));
- sp<MediaCodecBuffer> buffer = static_cast<MediaCodecBuffer *>(obj.get());
-
- sp<AMessage> notifyConsumed;
- CHECK(msg->findMessage("notifyConsumed", ¬ifyConsumed));
-
- QueueEntry entry;
- entry.mBuffer = buffer;
- entry.mNotifyConsumed = notifyConsumed;
- entry.mOffset = 0;
- entry.mFinalResult = OK;
- entry.mBufferOrdinal = ++mTotalBuffersQueued;
-
- if (audio) {
- Mutex::Autolock autoLock(mLock);
- mAudioQueue.push_back(entry);
- postDrainAudioQueue_l();
- } else {
- mVideoQueue.push_back(entry);
- postDrainVideoQueue();
- }
-
- Mutex::Autolock autoLock(mLock);
- if (!mSyncQueues || mAudioQueue.empty() || mVideoQueue.empty()) {
- return;
- }
-
- sp<MediaCodecBuffer> firstAudioBuffer = (*mAudioQueue.begin()).mBuffer;
- sp<MediaCodecBuffer> firstVideoBuffer = (*mVideoQueue.begin()).mBuffer;
-
- if (firstAudioBuffer == NULL || firstVideoBuffer == NULL) {
- // EOS signalled on either queue.
- syncQueuesDone_l();
- return;
- }
-
- int64_t firstAudioTimeUs;
- int64_t firstVideoTimeUs;
- CHECK(firstAudioBuffer->meta()
- ->findInt64("timeUs", &firstAudioTimeUs));
- CHECK(firstVideoBuffer->meta()
- ->findInt64("timeUs", &firstVideoTimeUs));
-
- int64_t diff = firstVideoTimeUs - firstAudioTimeUs;
-
- ALOGV("queueDiff = %.2f secs", diff / 1E6);
-
- if (diff > 100000LL) {
- // Audio data starts More than 0.1 secs before video.
- // Drop some audio.
-
- (*mAudioQueue.begin()).mNotifyConsumed->post();
- mAudioQueue.erase(mAudioQueue.begin());
- return;
- }
-
- syncQueuesDone_l();
-}
-
-void NuPlayer2::Renderer::syncQueuesDone_l() {
- if (!mSyncQueues) {
- return;
- }
-
- mSyncQueues = false;
-
- if (!mAudioQueue.empty()) {
- postDrainAudioQueue_l();
- }
-
- if (!mVideoQueue.empty()) {
- mLock.unlock();
- postDrainVideoQueue();
- mLock.lock();
- }
-}
-
-void NuPlayer2::Renderer::onQueueEOS(const sp<AMessage> &msg) {
- int32_t audio;
- CHECK(msg->findInt32("audio", &audio));
-
- if (dropBufferIfStale(audio, msg)) {
- return;
- }
-
- int32_t finalResult;
- CHECK(msg->findInt32("finalResult", &finalResult));
-
- QueueEntry entry;
- entry.mOffset = 0;
- entry.mFinalResult = finalResult;
-
- if (audio) {
- Mutex::Autolock autoLock(mLock);
- if (mAudioQueue.empty() && mSyncQueues) {
- syncQueuesDone_l();
- }
- mAudioQueue.push_back(entry);
- postDrainAudioQueue_l();
- } else {
- if (mVideoQueue.empty() && getSyncQueues()) {
- Mutex::Autolock autoLock(mLock);
- syncQueuesDone_l();
- }
- mVideoQueue.push_back(entry);
- postDrainVideoQueue();
- }
-}
-
-void NuPlayer2::Renderer::onFlush(const sp<AMessage> &msg) {
- int32_t audio, notifyComplete;
- CHECK(msg->findInt32("audio", &audio));
-
- {
- Mutex::Autolock autoLock(mLock);
- if (audio) {
- notifyComplete = mNotifyCompleteAudio;
- mNotifyCompleteAudio = false;
- mLastAudioMediaTimeUs = -1;
-
- mHasAudio = false;
- if (mNextVideoTimeMediaUs >= 0) {
- int64_t nowUs = ALooper::GetNowUs();
- mMediaClock->updateAnchor(
- mNextVideoTimeMediaUs, nowUs,
- mNextVideoTimeMediaUs + kDefaultVideoFrameIntervalUs);
- }
- } else {
- notifyComplete = mNotifyCompleteVideo;
- mNotifyCompleteVideo = false;
- mVideoRenderingStarted = false;
- }
-
- // If we're currently syncing the queues, i.e. dropping audio while
- // aligning the first audio/video buffer times and only one of the
- // two queues has data, we may starve that queue by not requesting
- // more buffers from the decoder. If the other source then encounters
- // a discontinuity that leads to flushing, we'll never find the
- // corresponding discontinuity on the other queue.
- // Therefore we'll stop syncing the queues if at least one of them
- // is flushed.
- syncQueuesDone_l();
- }
- clearAnchorTime();
-
- ALOGV("flushing %s", audio ? "audio" : "video");
- if (audio) {
- {
- Mutex::Autolock autoLock(mLock);
- flushQueue(&mAudioQueue);
-
- ++mAudioDrainGeneration;
- ++mAudioEOSGeneration;
- prepareForMediaRenderingStart_l();
-
- // the frame count will be reset after flush.
- clearAudioFirstAnchorTime_l();
- }
-
- mDrainAudioQueuePending = false;
-
- if (offloadingAudio()) {
- mAudioSink->pause();
- mAudioSink->flush();
- if (!mPaused) {
- mAudioSink->start();
- }
- } else {
- mAudioSink->pause();
- mAudioSink->flush();
- // Call stop() to signal to the AudioSink to completely fill the
- // internal buffer before resuming playback.
- // FIXME: this is ignored after flush().
- mAudioSink->stop();
- if (mPaused) {
- // Race condition: if renderer is paused and audio sink is stopped,
- // we need to make sure that the audio track buffer fully drains
- // before delivering data.
- // FIXME: remove this if we can detect if stop() is complete.
- const int delayUs = 2 * 50 * 1000; // (2 full mixer thread cycles at 50ms)
- mPauseDrainAudioAllowedUs = ALooper::GetNowUs() + delayUs;
- } else {
- mAudioSink->start();
- }
- mNumFramesWritten = 0;
- }
- mNextAudioClockUpdateTimeUs = -1;
- } else {
- flushQueue(&mVideoQueue);
-
- mDrainVideoQueuePending = false;
-
- if (mVideoScheduler != NULL) {
- mVideoScheduler->restart();
- }
-
- Mutex::Autolock autoLock(mLock);
- ++mVideoDrainGeneration;
- prepareForMediaRenderingStart_l();
- }
-
- mVideoSampleReceived = false;
-
- if (notifyComplete) {
- notifyFlushComplete(audio);
- }
-}
-
-void NuPlayer2::Renderer::flushQueue(List<QueueEntry> *queue) {
- while (!queue->empty()) {
- QueueEntry *entry = &*queue->begin();
-
- if (entry->mBuffer != NULL) {
- entry->mNotifyConsumed->post();
- } else if (entry->mNotifyConsumed != nullptr) {
- // Is it needed to open audio sink now?
- onChangeAudioFormat(entry->mMeta, entry->mNotifyConsumed);
- }
-
- queue->erase(queue->begin());
- entry = NULL;
- }
-}
-
-void NuPlayer2::Renderer::notifyFlushComplete(bool audio) {
- sp<AMessage> notify = mNotify->dup();
- notify->setInt32("what", kWhatFlushComplete);
- notify->setInt32("audio", static_cast<int32_t>(audio));
- notify->post();
-}
-
-bool NuPlayer2::Renderer::dropBufferIfStale(
- bool audio, const sp<AMessage> &msg) {
- int32_t queueGeneration;
- CHECK(msg->findInt32("queueGeneration", &queueGeneration));
-
- if (queueGeneration == getQueueGeneration(audio)) {
- return false;
- }
-
- sp<AMessage> notifyConsumed;
- if (msg->findMessage("notifyConsumed", ¬ifyConsumed)) {
- notifyConsumed->post();
- }
-
- return true;
-}
-
-void NuPlayer2::Renderer::onAudioSinkChanged() {
- if (offloadingAudio()) {
- return;
- }
- CHECK(!mDrainAudioQueuePending);
- mNumFramesWritten = 0;
- mAnchorNumFramesWritten = -1;
- uint32_t written;
- if (mAudioSink->getFramesWritten(&written) == OK) {
- mNumFramesWritten = written;
- }
-}
-
-void NuPlayer2::Renderer::onDisableOffloadAudio() {
- Mutex::Autolock autoLock(mLock);
- mFlags &= ~FLAG_OFFLOAD_AUDIO;
- ++mAudioDrainGeneration;
- if (mAudioRenderingStartGeneration != -1) {
- prepareForMediaRenderingStart_l();
- }
-}
-
-void NuPlayer2::Renderer::onEnableOffloadAudio() {
- Mutex::Autolock autoLock(mLock);
- mFlags |= FLAG_OFFLOAD_AUDIO;
- ++mAudioDrainGeneration;
- if (mAudioRenderingStartGeneration != -1) {
- prepareForMediaRenderingStart_l();
- }
-}
-
-void NuPlayer2::Renderer::onPause() {
- if (mPaused) {
- return;
- }
-
- {
- Mutex::Autolock autoLock(mLock);
- // we do not increment audio drain generation so that we fill audio buffer during pause.
- ++mVideoDrainGeneration;
- prepareForMediaRenderingStart_l();
- mPaused = true;
- mMediaClock->setPlaybackRate(0.0);
- }
-
- mDrainAudioQueuePending = false;
- mDrainVideoQueuePending = false;
-
- // Note: audio data may not have been decoded, and the AudioSink may not be opened.
- mAudioSink->pause();
- startAudioOffloadPauseTimeout();
-
- ALOGV("now paused audio queue has %zu entries, video has %zu entries",
- mAudioQueue.size(), mVideoQueue.size());
-}
-
-void NuPlayer2::Renderer::onResume() {
- if (!mPaused) {
- return;
- }
-
- // Note: audio data may not have been decoded, and the AudioSink may not be opened.
- cancelAudioOffloadPauseTimeout();
- if (mAudioSink->ready()) {
- status_t err = mAudioSink->start();
- if (err != OK) {
- ALOGE("cannot start AudioSink err %d", err);
- notifyAudioTearDown(kDueToError);
- }
- }
-
- {
- Mutex::Autolock autoLock(mLock);
- mPaused = false;
- // rendering started message may have been delayed if we were paused.
- if (mRenderingDataDelivered) {
- notifyIfMediaRenderingStarted_l();
- }
- // configure audiosink as we did not do it when pausing
- if (mAudioSink != NULL && mAudioSink->ready()) {
- mAudioSink->setPlaybackRate(mPlaybackSettings);
- }
-
- mMediaClock->setPlaybackRate(mPlaybackSettings.mSpeed);
-
- if (!mAudioQueue.empty()) {
- postDrainAudioQueue_l();
- }
- }
-
- if (!mVideoQueue.empty()) {
- postDrainVideoQueue();
- }
-}
-
-void NuPlayer2::Renderer::onSetVideoFrameRate(float fps) {
- if (mVideoScheduler == NULL) {
- mVideoScheduler = new VideoFrameScheduler2();
- }
- mVideoScheduler->init(fps);
-}
-
-int32_t NuPlayer2::Renderer::getQueueGeneration(bool audio) {
- Mutex::Autolock autoLock(mLock);
- return (audio ? mAudioQueueGeneration : mVideoQueueGeneration);
-}
-
-int32_t NuPlayer2::Renderer::getDrainGeneration(bool audio) {
- Mutex::Autolock autoLock(mLock);
- return (audio ? mAudioDrainGeneration : mVideoDrainGeneration);
-}
-
-bool NuPlayer2::Renderer::getSyncQueues() {
- Mutex::Autolock autoLock(mLock);
- return mSyncQueues;
-}
-
-void NuPlayer2::Renderer::onAudioTearDown(AudioTearDownReason reason) {
- if (mAudioTornDown) {
- return;
- }
- mAudioTornDown = true;
-
- int64_t currentPositionUs;
- sp<AMessage> notify = mNotify->dup();
- if (getCurrentPosition(¤tPositionUs) == OK) {
- notify->setInt64("positionUs", currentPositionUs);
- }
-
- mAudioSink->stop();
- mAudioSink->flush();
-
- notify->setInt32("what", kWhatAudioTearDown);
- notify->setInt32("reason", reason);
- notify->post();
-}
-
-void NuPlayer2::Renderer::startAudioOffloadPauseTimeout() {
- if (offloadingAudio()) {
- mWakeLock->acquire();
- sp<AMessage> msg = new AMessage(kWhatAudioOffloadPauseTimeout, this);
- msg->setInt32("drainGeneration", mAudioOffloadPauseTimeoutGeneration);
- msg->post(kOffloadPauseMaxUs);
- }
-}
-
-void NuPlayer2::Renderer::cancelAudioOffloadPauseTimeout() {
- // We may have called startAudioOffloadPauseTimeout() without
- // the AudioSink open and with offloadingAudio enabled.
- //
- // When we cancel, it may be that offloadingAudio is subsequently disabled, so regardless
- // we always release the wakelock and increment the pause timeout generation.
- //
- // Note: The acquired wakelock prevents the device from suspending
- // immediately after offload pause (in case a resume happens shortly thereafter).
- mWakeLock->release(true);
- ++mAudioOffloadPauseTimeoutGeneration;
-}
-
-status_t NuPlayer2::Renderer::onOpenAudioSink(
- const sp<AMessage> &format,
- bool offloadOnly,
- bool hasVideo,
- uint32_t flags,
- bool isStreaming) {
- ALOGV("openAudioSink: offloadOnly(%d) offloadingAudio(%d)",
- offloadOnly, offloadingAudio());
-
- bool audioSinkChanged = false;
-
- int32_t numChannels;
- CHECK(format->findInt32("channel-count", &numChannels));
-
- int32_t channelMask;
- if (!format->findInt32("channel-mask", &channelMask)) {
- // signal to the AudioSink to derive the mask from count.
- channelMask = CHANNEL_MASK_USE_CHANNEL_ORDER;
- }
-
- int32_t sampleRate;
- CHECK(format->findInt32("sample-rate", &sampleRate));
-
- // read pcm encoding from MediaCodec output format, if available
- int32_t pcmEncoding;
- audio_format_t audioFormat =
- format->findInt32(KEY_PCM_ENCODING, &pcmEncoding) ?
- audioFormatFromEncoding(pcmEncoding) : AUDIO_FORMAT_PCM_16_BIT;
-
- if (offloadingAudio()) {
- AString mime;
- CHECK(format->findString("mime", &mime));
- status_t err = mapMimeToAudioFormat(audioFormat, mime.c_str());
-
- if (err != OK) {
- ALOGE("Couldn't map mime \"%s\" to a valid "
- "audio_format", mime.c_str());
- onDisableOffloadAudio();
- } else {
- ALOGV("Mime \"%s\" mapped to audio_format 0x%x",
- mime.c_str(), audioFormat);
-
- int avgBitRate = -1;
- format->findInt32("bitrate", &avgBitRate);
-
- int32_t aacProfile = -1;
- if (audioFormat == AUDIO_FORMAT_AAC
- && format->findInt32("aac-profile", &aacProfile)) {
- // Redefine AAC format as per aac profile
- mapAACProfileToAudioFormat(
- audioFormat,
- aacProfile);
- }
-
- audio_offload_info_t offloadInfo = AUDIO_INFO_INITIALIZER;
- offloadInfo.duration_us = -1;
- format->findInt64(
- "durationUs", &offloadInfo.duration_us);
- offloadInfo.sample_rate = sampleRate;
- offloadInfo.channel_mask = channelMask;
- offloadInfo.format = audioFormat;
- offloadInfo.stream_type = AUDIO_STREAM_MUSIC;
- offloadInfo.bit_rate = avgBitRate;
- offloadInfo.has_video = hasVideo;
- offloadInfo.is_streaming = isStreaming;
-
- if (memcmp(&mCurrentOffloadInfo, &offloadInfo, sizeof(offloadInfo)) == 0) {
- ALOGV("openAudioSink: no change in offload mode");
- // no change from previous configuration, everything ok.
- return OK;
- }
- mCurrentPcmInfo = AUDIO_PCMINFO_INITIALIZER;
-
- ALOGV("openAudioSink: try to open AudioSink in offload mode");
- uint32_t offloadFlags = flags;
- offloadFlags |= AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD;
- offloadFlags &= ~AUDIO_OUTPUT_FLAG_DEEP_BUFFER;
- audioSinkChanged = true;
- mAudioSink->close();
-
- err = mAudioSink->open(
- sampleRate,
- numChannels,
- (audio_channel_mask_t)channelMask,
- audioFormat,
- &NuPlayer2::Renderer::AudioSinkCallback,
- this,
- (audio_output_flags_t)offloadFlags,
- &offloadInfo);
-
- if (err == OK) {
- err = mAudioSink->setPlaybackRate(mPlaybackSettings);
- }
-
- if (err == OK) {
- // If the playback is offloaded to h/w, we pass
- // the HAL some metadata information.
- // We don't want to do this for PCM because it
- // will be going through the AudioFlinger mixer
- // before reaching the hardware.
- // TODO
- mCurrentOffloadInfo = offloadInfo;
- if (!mPaused) { // for preview mode, don't start if paused
- err = mAudioSink->start();
- }
- ALOGV_IF(err == OK, "openAudioSink: offload succeeded");
- }
- if (err != OK) {
- // Clean up, fall back to non offload mode.
- mAudioSink->close();
- onDisableOffloadAudio();
- mCurrentOffloadInfo = AUDIO_INFO_INITIALIZER;
- ALOGV("openAudioSink: offload failed");
- if (offloadOnly) {
- notifyAudioTearDown(kForceNonOffload);
- }
- } else {
- mUseAudioCallback = true; // offload mode transfers data through callback
- ++mAudioDrainGeneration; // discard pending kWhatDrainAudioQueue message.
- }
- }
- }
- if (!offloadOnly && !offloadingAudio()) {
- ALOGV("openAudioSink: open AudioSink in NON-offload mode");
- uint32_t pcmFlags = flags;
- pcmFlags &= ~AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD;
-
- const PcmInfo info = {
- (audio_channel_mask_t)channelMask,
- (audio_output_flags_t)pcmFlags,
- audioFormat,
- numChannels,
- sampleRate
- };
- if (memcmp(&mCurrentPcmInfo, &info, sizeof(info)) == 0) {
- ALOGV("openAudioSink: no change in pcm mode");
- // no change from previous configuration, everything ok.
- return OK;
- }
-
- audioSinkChanged = true;
- mAudioSink->close();
- mCurrentOffloadInfo = AUDIO_INFO_INITIALIZER;
- // Note: It is possible to set up the callback, but not use it to send audio data.
- // This requires a fix in AudioSink to explicitly specify the transfer mode.
- mUseAudioCallback = getUseAudioCallbackSetting();
- if (mUseAudioCallback) {
- ++mAudioDrainGeneration; // discard pending kWhatDrainAudioQueue message.
- }
-
- // Compute the desired buffer size.
- // For callback mode, the amount of time before wakeup is about half the buffer size.
- const uint32_t frameCount =
- (unsigned long long)sampleRate * getAudioSinkPcmMsSetting() / 1000;
-
- // We should always be able to set our playback settings if the sink is closed.
- LOG_ALWAYS_FATAL_IF(mAudioSink->setPlaybackRate(mPlaybackSettings) != OK,
- "onOpenAudioSink: can't set playback rate on closed sink");
- status_t err = mAudioSink->open(
- sampleRate,
- numChannels,
- (audio_channel_mask_t)channelMask,
- audioFormat,
- mUseAudioCallback ? &NuPlayer2::Renderer::AudioSinkCallback : NULL,
- mUseAudioCallback ? this : NULL,
- (audio_output_flags_t)pcmFlags,
- NULL,
- frameCount);
- if (err != OK) {
- ALOGW("openAudioSink: non offloaded open failed status: %d", err);
- mAudioSink->close();
- mCurrentPcmInfo = AUDIO_PCMINFO_INITIALIZER;
- return err;
- }
- mCurrentPcmInfo = info;
- if (!mPaused) { // for preview mode, don't start if paused
- mAudioSink->start();
- }
- }
- if (audioSinkChanged) {
- onAudioSinkChanged();
- }
- mAudioTornDown = false;
- return OK;
-}
-
-void NuPlayer2::Renderer::onCloseAudioSink() {
- mAudioSink->close();
- mCurrentOffloadInfo = AUDIO_INFO_INITIALIZER;
- mCurrentPcmInfo = AUDIO_PCMINFO_INITIALIZER;
-}
-
-void NuPlayer2::Renderer::onChangeAudioFormat(
- const sp<AMessage> &meta, const sp<AMessage> ¬ify) {
- sp<AMessage> format;
- CHECK(meta->findMessage("format", &format));
-
- int32_t offloadOnly;
- CHECK(meta->findInt32("offload-only", &offloadOnly));
-
- int32_t hasVideo;
- CHECK(meta->findInt32("has-video", &hasVideo));
-
- uint32_t flags;
- CHECK(meta->findInt32("flags", (int32_t *)&flags));
-
- uint32_t isStreaming;
- CHECK(meta->findInt32("isStreaming", (int32_t *)&isStreaming));
-
- status_t err = onOpenAudioSink(format, offloadOnly, hasVideo, flags, isStreaming);
-
- if (err != OK) {
- notify->setInt32("err", err);
- }
- notify->post();
-}
-
-} // namespace android
diff --git a/media/libmediaplayer2/nuplayer2/NuPlayer2Renderer.h b/media/libmediaplayer2/nuplayer2/NuPlayer2Renderer.h
deleted file mode 100644
index d065dee..0000000
--- a/media/libmediaplayer2/nuplayer2/NuPlayer2Renderer.h
+++ /dev/null
@@ -1,304 +0,0 @@
-/*
- * Copyright 2017 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef NUPLAYER2_RENDERER_H_
-
-#define NUPLAYER2_RENDERER_H_
-
-#include <media/AudioResamplerPublic.h>
-#include <media/AVSyncSettings.h>
-#include <mediaplayer2/JObjectHolder.h>
-
-#include "NuPlayer2.h"
-
-namespace android {
-
-class JWakeLock;
-struct MediaClock;
-class MediaCodecBuffer;
-struct VideoFrameSchedulerBase;
-
-struct NuPlayer2::Renderer : public AHandler {
- enum Flags {
- FLAG_REAL_TIME = 1,
- FLAG_OFFLOAD_AUDIO = 2,
- };
- Renderer(const sp<MediaPlayer2Interface::AudioSink> &sink,
- const sp<MediaClock> &mediaClock,
- const sp<AMessage> ¬ify,
- const sp<JObjectHolder> &context,
- uint32_t flags = 0);
-
- static size_t AudioSinkCallback(
- MediaPlayer2Interface::AudioSink *audioSink,
- void *data, size_t size, void *me,
- MediaPlayer2Interface::AudioSink::cb_event_t event);
-
- void queueBuffer(
- bool audio,
- const sp<MediaCodecBuffer> &buffer,
- const sp<AMessage> ¬ifyConsumed);
-
- void queueEOS(bool audio, status_t finalResult);
-
- status_t setPlaybackSettings(const AudioPlaybackRate &rate /* sanitized */);
- status_t getPlaybackSettings(AudioPlaybackRate *rate /* nonnull */);
- status_t setSyncSettings(const AVSyncSettings &sync, float videoFpsHint);
- status_t getSyncSettings(AVSyncSettings *sync /* nonnull */, float *videoFps /* nonnull */);
-
- void flush(bool audio, bool notifyComplete);
-
- void signalTimeDiscontinuity();
-
- void signalDisableOffloadAudio();
- void signalEnableOffloadAudio();
-
- void pause();
- void resume();
-
- void setVideoFrameRate(float fps);
-
- status_t getCurrentPosition(int64_t *mediaUs);
- int64_t getVideoLateByUs();
-
- status_t openAudioSink(
- const sp<AMessage> &format,
- bool offloadOnly,
- bool hasVideo,
- uint32_t flags,
- bool *isOffloaded,
- bool isStreaming);
- void closeAudioSink();
-
- // re-open audio sink after all pending audio buffers played.
- void changeAudioFormat(
- const sp<AMessage> &format,
- bool offloadOnly,
- bool hasVideo,
- uint32_t flags,
- bool isStreaming,
- const sp<AMessage> ¬ify);
-
- enum {
- kWhatEOS = 'eos ',
- kWhatFlushComplete = 'fluC',
- kWhatPosition = 'posi',
- kWhatVideoRenderingStart = 'vdrd',
- kWhatMediaRenderingStart = 'mdrd',
- kWhatAudioTearDown = 'adTD',
- kWhatAudioOffloadPauseTimeout = 'aOPT',
- };
-
- enum AudioTearDownReason {
- kDueToError = 0, // Could restart with either offload or non-offload.
- kDueToTimeout,
- kForceNonOffload, // Restart only with non-offload.
- };
-
-protected:
- virtual ~Renderer();
-
- virtual void onMessageReceived(const sp<AMessage> &msg);
-
-private:
- enum {
- kWhatDrainAudioQueue = 'draA',
- kWhatDrainVideoQueue = 'draV',
- kWhatPostDrainVideoQueue = 'pDVQ',
- kWhatQueueBuffer = 'queB',
- kWhatQueueEOS = 'qEOS',
- kWhatConfigPlayback = 'cfPB',
- kWhatConfigSync = 'cfSy',
- kWhatGetPlaybackSettings = 'gPbS',
- kWhatGetSyncSettings = 'gSyS',
- kWhatFlush = 'flus',
- kWhatPause = 'paus',
- kWhatResume = 'resm',
- kWhatOpenAudioSink = 'opnA',
- kWhatCloseAudioSink = 'clsA',
- kWhatChangeAudioFormat = 'chgA',
- kWhatStopAudioSink = 'stpA',
- kWhatDisableOffloadAudio = 'noOA',
- kWhatEnableOffloadAudio = 'enOA',
- kWhatSetVideoFrameRate = 'sVFR',
- };
-
- // if mBuffer != nullptr, it's a buffer containing real data.
- // else if mNotifyConsumed == nullptr, it's EOS.
- // else it's a tag for re-opening audio sink in different format.
- struct QueueEntry {
- sp<MediaCodecBuffer> mBuffer;
- sp<AMessage> mMeta;
- sp<AMessage> mNotifyConsumed;
- size_t mOffset;
- status_t mFinalResult;
- int32_t mBufferOrdinal;
- };
-
- static const int64_t kMinPositionUpdateDelayUs;
-
- sp<MediaPlayer2Interface::AudioSink> mAudioSink;
- bool mUseVirtualAudioSink;
- sp<AMessage> mNotify;
- Mutex mLock;
- uint32_t mFlags;
- List<QueueEntry> mAudioQueue;
- List<QueueEntry> mVideoQueue;
- uint32_t mNumFramesWritten;
- sp<VideoFrameSchedulerBase> mVideoScheduler;
-
- bool mDrainAudioQueuePending;
- bool mDrainVideoQueuePending;
- int32_t mAudioQueueGeneration;
- int32_t mVideoQueueGeneration;
- int32_t mAudioDrainGeneration;
- int32_t mVideoDrainGeneration;
- int32_t mAudioEOSGeneration;
-
- const sp<MediaClock> mMediaClock;
-
- AudioPlaybackRate mPlaybackSettings;
- AVSyncSettings mSyncSettings;
- float mVideoFpsHint;
-
- int64_t mAudioFirstAnchorTimeMediaUs;
- int64_t mAnchorTimeMediaUs;
- int64_t mAnchorNumFramesWritten;
- int64_t mVideoLateByUs;
- int64_t mNextVideoTimeMediaUs;
- bool mHasAudio;
- bool mHasVideo;
-
- bool mNotifyCompleteAudio;
- bool mNotifyCompleteVideo;
-
- bool mSyncQueues;
-
- // modified on only renderer's thread.
- bool mPaused;
- int64_t mPauseDrainAudioAllowedUs; // time when we can drain/deliver audio in pause mode.
-
- bool mVideoSampleReceived;
- bool mVideoRenderingStarted;
- int32_t mVideoRenderingStartGeneration;
- int32_t mAudioRenderingStartGeneration;
- bool mRenderingDataDelivered;
-
- int64_t mNextAudioClockUpdateTimeUs;
- // the media timestamp of last audio sample right before EOS.
- int64_t mLastAudioMediaTimeUs;
-
- int32_t mAudioOffloadPauseTimeoutGeneration;
- bool mAudioTornDown;
- audio_offload_info_t mCurrentOffloadInfo;
-
- struct PcmInfo {
- audio_channel_mask_t mChannelMask;
- audio_output_flags_t mFlags;
- audio_format_t mFormat;
- int32_t mNumChannels;
- int32_t mSampleRate;
- };
- PcmInfo mCurrentPcmInfo;
- static const PcmInfo AUDIO_PCMINFO_INITIALIZER;
-
- int32_t mTotalBuffersQueued;
- int32_t mLastAudioBufferDrained;
- bool mUseAudioCallback;
-
- sp<JWakeLock> mWakeLock;
-
- status_t getCurrentPositionOnLooper(int64_t *mediaUs);
- status_t getCurrentPositionOnLooper(
- int64_t *mediaUs, int64_t nowUs, bool allowPastQueuedVideo = false);
- bool getCurrentPositionIfPaused_l(int64_t *mediaUs);
- status_t getCurrentPositionFromAnchor(
- int64_t *mediaUs, int64_t nowUs, bool allowPastQueuedVideo = false);
-
- void notifyEOSCallback();
- size_t fillAudioBuffer(void *buffer, size_t size);
-
- bool onDrainAudioQueue();
- void drainAudioQueueUntilLastEOS();
- int64_t getPendingAudioPlayoutDurationUs(int64_t nowUs);
- void postDrainAudioQueue_l(int64_t delayUs = 0);
-
- void clearAnchorTime();
- void clearAudioFirstAnchorTime_l();
- void setAudioFirstAnchorTimeIfNeeded_l(int64_t mediaUs);
- void setVideoLateByUs(int64_t lateUs);
-
- void onNewAudioMediaTime(int64_t mediaTimeUs);
- int64_t getRealTimeUs(int64_t mediaTimeUs, int64_t nowUs);
-
- void onDrainVideoQueue();
- void postDrainVideoQueue();
-
- void prepareForMediaRenderingStart_l();
- void notifyIfMediaRenderingStarted_l();
-
- void onQueueBuffer(const sp<AMessage> &msg);
- void onQueueEOS(const sp<AMessage> &msg);
- void onFlush(const sp<AMessage> &msg);
- void onAudioSinkChanged();
- void onDisableOffloadAudio();
- void onEnableOffloadAudio();
- status_t onConfigPlayback(const AudioPlaybackRate &rate /* sanitized */);
- status_t onGetPlaybackSettings(AudioPlaybackRate *rate /* nonnull */);
- status_t onConfigSync(const AVSyncSettings &sync, float videoFpsHint);
- status_t onGetSyncSettings(AVSyncSettings *sync /* nonnull */, float *videoFps /* nonnull */);
-
- void onPause();
- void onResume();
- void onSetVideoFrameRate(float fps);
- int32_t getQueueGeneration(bool audio);
- int32_t getDrainGeneration(bool audio);
- bool getSyncQueues();
- void onAudioTearDown(AudioTearDownReason reason);
- status_t onOpenAudioSink(
- const sp<AMessage> &format,
- bool offloadOnly,
- bool hasVideo,
- uint32_t flags,
- bool isStreaming);
- void onCloseAudioSink();
- void onChangeAudioFormat(const sp<AMessage> &meta, const sp<AMessage> ¬ify);
-
- void notifyEOS(bool audio, status_t finalResult, int64_t delayUs = 0);
- void notifyEOS_l(bool audio, status_t finalResult, int64_t delayUs = 0);
- void notifyFlushComplete(bool audio);
- void notifyPosition();
- void notifyVideoLateBy(int64_t lateByUs);
- void notifyVideoRenderingStart();
- void notifyAudioTearDown(AudioTearDownReason reason);
-
- void flushQueue(List<QueueEntry> *queue);
- bool dropBufferIfStale(bool audio, const sp<AMessage> &msg);
- void syncQueuesDone_l();
-
- bool offloadingAudio() const { return (mFlags & FLAG_OFFLOAD_AUDIO) != 0; }
-
- void startAudioOffloadPauseTimeout();
- void cancelAudioOffloadPauseTimeout();
-
- int64_t getDurationUsIfPlayedAtSampleRate(uint32_t numFrames);
-
- DISALLOW_EVIL_CONSTRUCTORS(Renderer);
-};
-
-} // namespace android
-
-#endif // NUPLAYER2_RENDERER_H_
diff --git a/media/libmediaplayer2/nuplayer2/NuPlayer2Source.h b/media/libmediaplayer2/nuplayer2/NuPlayer2Source.h
deleted file mode 100644
index 9298a99..0000000
--- a/media/libmediaplayer2/nuplayer2/NuPlayer2Source.h
+++ /dev/null
@@ -1,166 +0,0 @@
-/*
- * Copyright 2017 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef NUPLAYER2_SOURCE_H_
-
-#define NUPLAYER2_SOURCE_H_
-
-#include "NuPlayer2.h"
-
-#include <media/stagefright/foundation/AMessage.h>
-#include <media/stagefright/MetaData.h>
-#include <mediaplayer2/mediaplayer2.h>
-#include <utils/Vector.h>
-
-namespace android {
-
-struct ABuffer;
-struct AMediaCryptoWrapper;
-class MediaBuffer;
-
-struct NuPlayer2::Source : public AHandler {
- enum Flags {
- FLAG_CAN_PAUSE = 1,
- FLAG_CAN_SEEK_BACKWARD = 2, // the "10 sec back button"
- FLAG_CAN_SEEK_FORWARD = 4, // the "10 sec forward button"
- FLAG_CAN_SEEK = 8, // the "seek bar"
- FLAG_DYNAMIC_DURATION = 16,
- FLAG_SECURE = 32, // Secure codec is required.
- FLAG_PROTECTED = 64, // The screen needs to be protected (screenshot is disabled).
- };
-
- enum {
- kWhatPrepared,
- kWhatFlagsChanged,
- kWhatVideoSizeChanged,
- kWhatBufferingUpdate,
- kWhatPauseOnBufferingStart,
- kWhatResumeOnBufferingEnd,
- kWhatCacheStats,
- kWhatSubtitleData,
- kWhatTimedTextData,
- kWhatTimedMetaData,
- kWhatQueueDecoderShutdown,
- kWhatDrmNoLicense,
- // Modular DRM
- kWhatDrmInfo,
- };
-
- // The provides message is used to notify the player about various
- // events.
- explicit Source(const sp<AMessage> ¬ify)
- : mNotify(notify) {
- }
-
- virtual status_t getBufferingSettings(
- BufferingSettings* buffering /* nonnull */) = 0;
- virtual status_t setBufferingSettings(const BufferingSettings& buffering) = 0;
-
- virtual void prepareAsync(int64_t startTimeUs) = 0;
-
- virtual void start() = 0;
- virtual void stop() {}
- virtual void pause() {}
- virtual void resume() {}
-
- // Explicitly disconnect the underling data source
- virtual void disconnect() {}
-
- // Returns OK iff more data was available,
- // an error or ERROR_END_OF_STREAM if not.
- virtual status_t feedMoreTSData() = 0;
-
- // Returns non-NULL format when the specified track exists.
- // When the format has "err" set to -EWOULDBLOCK, source needs more time to get valid meta data.
- // Returns NULL if the specified track doesn't exist or is invalid;
- virtual sp<AMessage> getFormat(bool audio);
-
- virtual sp<MetaData> getFormatMeta(bool /* audio */) { return NULL; }
- virtual sp<MetaData> getFileFormatMeta() const { return NULL; }
-
- virtual status_t dequeueAccessUnit(
- bool audio, sp<ABuffer> *accessUnit) = 0;
-
- virtual status_t getDuration(int64_t * /* durationUs */) {
- return INVALID_OPERATION;
- }
-
- virtual size_t getTrackCount() const {
- return 0;
- }
-
- virtual sp<AMessage> getTrackInfo(size_t /* trackIndex */) const {
- return NULL;
- }
-
- virtual ssize_t getSelectedTrack(media_track_type /* type */) const {
- return INVALID_OPERATION;
- }
-
- virtual status_t selectTrack(size_t /* trackIndex */, bool /* select */, int64_t /* timeUs*/) {
- return INVALID_OPERATION;
- }
-
- virtual status_t seekTo(
- int64_t /* seekTimeUs */,
- MediaPlayer2SeekMode /* mode */ = MediaPlayer2SeekMode::SEEK_PREVIOUS_SYNC) {
- return INVALID_OPERATION;
- }
-
- virtual bool isRealTime() const {
- return false;
- }
-
- virtual bool isStreaming() const {
- return true;
- }
-
- virtual void setOffloadAudio(bool /* offload */) {}
-
- // Modular DRM
- virtual status_t prepareDrm(
- const uint8_t /* uuid */[16], const Vector<uint8_t> & /* drmSessionId */,
- sp<AMediaCryptoWrapper> * /* crypto */) {
- return INVALID_OPERATION;
- }
-
- virtual status_t releaseDrm() {
- return INVALID_OPERATION;
- }
-
-protected:
- virtual ~Source() {}
-
- virtual void onMessageReceived(const sp<AMessage> &msg);
-
- sp<AMessage> dupNotify() const { return mNotify->dup(); }
-
- void notifyFlagsChanged(uint32_t flags);
- void notifyVideoSizeChanged(const sp<AMessage> &format = NULL);
- void notifyPrepared(status_t err = OK);
- // Modular DRM
- void notifyDrmInfo(const sp<ABuffer> &buffer);
-
-private:
- sp<AMessage> mNotify;
-
- DISALLOW_EVIL_CONSTRUCTORS(Source);
-};
-
-} // namespace android
-
-#endif // NUPLAYER2_SOURCE_H_
-
diff --git a/media/libmediaplayer2/nuplayer2/RTSPSource2.cpp b/media/libmediaplayer2/nuplayer2/RTSPSource2.cpp
deleted file mode 100644
index a70269e..0000000
--- a/media/libmediaplayer2/nuplayer2/RTSPSource2.cpp
+++ /dev/null
@@ -1,903 +0,0 @@
-/*
- * Copyright 2017 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "RTSPSource2"
-#include <utils/Log.h>
-
-#include "RTSPSource2.h"
-
-#include "AnotherPacketSource.h"
-#include "MyHandler.h"
-#include "SDPLoader.h"
-
-#include <media/MediaHTTPService.h>
-#include <media/stagefright/MediaDefs.h>
-#include <media/stagefright/MetaData.h>
-
-namespace android {
-
-const int64_t kNearEOSTimeoutUs = 2000000LL; // 2 secs
-
-// Default Buffer Underflow/Prepare/StartServer/Overflow Marks
-static const int kUnderflowMarkMs = 1000; // 1 second
-static const int kPrepareMarkMs = 3000; // 3 seconds
-//static const int kStartServerMarkMs = 5000;
-static const int kOverflowMarkMs = 10000; // 10 seconds
-
-NuPlayer2::RTSPSource2::RTSPSource2(
- const sp<AMessage> ¬ify,
- const sp<MediaHTTPService> &httpService,
- const char *url,
- const KeyedVector<String8, String8> *headers,
- uid_t uid,
- bool isSDP)
- : Source(notify),
- mHTTPService(httpService),
- mURL(url),
- mUID(uid),
- mFlags(0),
- mIsSDP(isSDP),
- mState(DISCONNECTED),
- mFinalResult(OK),
- mDisconnectReplyID(0),
- mBuffering(false),
- mInPreparationPhase(true),
- mEOSPending(false),
- mSeekGeneration(0),
- mEOSTimeoutAudio(0),
- mEOSTimeoutVideo(0) {
- mBufferingSettings.mInitialMarkMs = kPrepareMarkMs;
- mBufferingSettings.mResumePlaybackMarkMs = kOverflowMarkMs;
- if (headers) {
- mExtraHeaders = *headers;
-
- ssize_t index =
- mExtraHeaders.indexOfKey(String8("x-hide-urls-from-log"));
-
- if (index >= 0) {
- mFlags |= kFlagIncognito;
-
- mExtraHeaders.removeItemsAt(index);
- }
- }
-}
-
-NuPlayer2::RTSPSource2::~RTSPSource2() {
- if (mLooper != NULL) {
- mLooper->unregisterHandler(id());
- mLooper->stop();
- }
-}
-
-status_t NuPlayer2::RTSPSource2::getBufferingSettings(
- BufferingSettings* buffering /* nonnull */) {
- Mutex::Autolock _l(mBufferingSettingsLock);
- *buffering = mBufferingSettings;
- return OK;
-}
-
-status_t NuPlayer2::RTSPSource2::setBufferingSettings(const BufferingSettings& buffering) {
- Mutex::Autolock _l(mBufferingSettingsLock);
- mBufferingSettings = buffering;
- return OK;
-}
-
-// TODO: fetch data starting from |startTimeUs|
-void NuPlayer2::RTSPSource2::prepareAsync(int64_t /* startTimeUs */) {
- if (mIsSDP && mHTTPService == NULL) {
- notifyPrepared(BAD_VALUE);
- return;
- }
-
- if (mLooper == NULL) {
- mLooper = new ALooper;
- mLooper->setName("rtsp2");
- mLooper->start();
-
- mLooper->registerHandler(this);
- }
-
- CHECK(mHandler == NULL);
- CHECK(mSDPLoader == NULL);
-
- sp<AMessage> notify = new AMessage(kWhatNotify, this);
-
- CHECK_EQ(mState, (int)DISCONNECTED);
- mState = CONNECTING;
-
- if (mIsSDP) {
- mSDPLoader = new SDPLoader(notify,
- (mFlags & kFlagIncognito) ? SDPLoader::kFlagIncognito : 0,
- mHTTPService);
-
- mSDPLoader->load(
- mURL.c_str(), mExtraHeaders.isEmpty() ? NULL : &mExtraHeaders);
- } else {
- mHandler = new MyHandler(mURL.c_str(), notify, true /* uidValid */, mUID);
- mLooper->registerHandler(mHandler);
-
- mHandler->connect();
- }
-
- startBufferingIfNecessary();
-}
-
-void NuPlayer2::RTSPSource2::start() {
-}
-
-void NuPlayer2::RTSPSource2::stop() {
- if (mLooper == NULL) {
- return;
- }
- sp<AMessage> msg = new AMessage(kWhatDisconnect, this);
-
- sp<AMessage> dummy;
- msg->postAndAwaitResponse(&dummy);
-}
-
-status_t NuPlayer2::RTSPSource2::feedMoreTSData() {
- Mutex::Autolock _l(mBufferingLock);
- return mFinalResult;
-}
-
-sp<MetaData> NuPlayer2::RTSPSource2::getFormatMeta(bool audio) {
- sp<AnotherPacketSource> source = getSource(audio);
-
- if (source == NULL) {
- return NULL;
- }
-
- return source->getFormat();
-}
-
-bool NuPlayer2::RTSPSource2::haveSufficientDataOnAllTracks() {
- // We're going to buffer at least 2 secs worth data on all tracks before
- // starting playback (both at startup and after a seek).
-
- static const int64_t kMinDurationUs = 2000000LL;
-
- int64_t mediaDurationUs = 0;
- getDuration(&mediaDurationUs);
- if ((mAudioTrack != NULL && mAudioTrack->isFinished(mediaDurationUs))
- || (mVideoTrack != NULL && mVideoTrack->isFinished(mediaDurationUs))) {
- return true;
- }
-
- status_t err;
- int64_t durationUs;
- if (mAudioTrack != NULL
- && (durationUs = mAudioTrack->getBufferedDurationUs(&err))
- < kMinDurationUs
- && err == OK) {
- ALOGV("audio track doesn't have enough data yet. (%.2f secs buffered)",
- durationUs / 1E6);
- return false;
- }
-
- if (mVideoTrack != NULL
- && (durationUs = mVideoTrack->getBufferedDurationUs(&err))
- < kMinDurationUs
- && err == OK) {
- ALOGV("video track doesn't have enough data yet. (%.2f secs buffered)",
- durationUs / 1E6);
- return false;
- }
-
- return true;
-}
-
-status_t NuPlayer2::RTSPSource2::dequeueAccessUnit(
- bool audio, sp<ABuffer> *accessUnit) {
- if (!stopBufferingIfNecessary()) {
- return -EWOULDBLOCK;
- }
-
- sp<AnotherPacketSource> source = getSource(audio);
-
- if (source == NULL) {
- return -EWOULDBLOCK;
- }
-
- status_t finalResult;
- if (!source->hasBufferAvailable(&finalResult)) {
- if (finalResult == OK) {
-
- // If other source already signaled EOS, this source should also return EOS
- if (sourceReachedEOS(!audio)) {
- return ERROR_END_OF_STREAM;
- }
-
- // If this source has detected near end, give it some time to retrieve more
- // data before returning EOS
- int64_t mediaDurationUs = 0;
- getDuration(&mediaDurationUs);
- if (source->isFinished(mediaDurationUs)) {
- int64_t eosTimeout = audio ? mEOSTimeoutAudio : mEOSTimeoutVideo;
- if (eosTimeout == 0) {
- setEOSTimeout(audio, ALooper::GetNowUs());
- } else if ((ALooper::GetNowUs() - eosTimeout) > kNearEOSTimeoutUs) {
- setEOSTimeout(audio, 0);
- return ERROR_END_OF_STREAM;
- }
- return -EWOULDBLOCK;
- }
-
- if (!sourceNearEOS(!audio)) {
- // We should not enter buffering mode
- // if any of the sources already have detected EOS.
- startBufferingIfNecessary();
- }
-
- return -EWOULDBLOCK;
- }
- return finalResult;
- }
-
- setEOSTimeout(audio, 0);
-
- return source->dequeueAccessUnit(accessUnit);
-}
-
-sp<AnotherPacketSource> NuPlayer2::RTSPSource2::getSource(bool audio) {
- if (mTSParser != NULL) {
- sp<MediaSource> source = mTSParser->getSource(
- audio ? ATSParser::AUDIO : ATSParser::VIDEO);
-
- return static_cast<AnotherPacketSource *>(source.get());
- }
-
- return audio ? mAudioTrack : mVideoTrack;
-}
-
-void NuPlayer2::RTSPSource2::setEOSTimeout(bool audio, int64_t timeout) {
- if (audio) {
- mEOSTimeoutAudio = timeout;
- } else {
- mEOSTimeoutVideo = timeout;
- }
-}
-
-status_t NuPlayer2::RTSPSource2::getDuration(int64_t *durationUs) {
- *durationUs = -1LL;
-
- int64_t audioDurationUs;
- if (mAudioTrack != NULL
- && mAudioTrack->getFormat()->findInt64(
- kKeyDuration, &audioDurationUs)
- && audioDurationUs > *durationUs) {
- *durationUs = audioDurationUs;
- }
-
- int64_t videoDurationUs;
- if (mVideoTrack != NULL
- && mVideoTrack->getFormat()->findInt64(
- kKeyDuration, &videoDurationUs)
- && videoDurationUs > *durationUs) {
- *durationUs = videoDurationUs;
- }
-
- return OK;
-}
-
-status_t NuPlayer2::RTSPSource2::seekTo(int64_t seekTimeUs, MediaPlayer2SeekMode mode) {
- sp<AMessage> msg = new AMessage(kWhatPerformSeek, this);
- msg->setInt32("generation", ++mSeekGeneration);
- msg->setInt64("timeUs", seekTimeUs);
- msg->setInt32("mode", mode);
-
- sp<AMessage> response;
- status_t err = msg->postAndAwaitResponse(&response);
- if (err == OK && response != NULL) {
- CHECK(response->findInt32("err", &err));
- }
-
- return err;
-}
-
-void NuPlayer2::RTSPSource2::performSeek(int64_t seekTimeUs) {
- if (mState != CONNECTED) {
- finishSeek(INVALID_OPERATION);
- return;
- }
-
- mState = SEEKING;
- mHandler->seek(seekTimeUs);
- mEOSPending = false;
-}
-
-void NuPlayer2::RTSPSource2::schedulePollBuffering() {
- sp<AMessage> msg = new AMessage(kWhatPollBuffering, this);
- msg->post(1000000LL); // 1 second intervals
-}
-
-void NuPlayer2::RTSPSource2::checkBuffering(
- bool *prepared, bool *underflow, bool *overflow, bool *startServer, bool *finished) {
- size_t numTracks = mTracks.size();
- size_t preparedCount, underflowCount, overflowCount, startCount, finishedCount;
- preparedCount = underflowCount = overflowCount = startCount = finishedCount = 0;
-
- size_t count = numTracks;
- for (size_t i = 0; i < count; ++i) {
- status_t finalResult;
- TrackInfo *info = &mTracks.editItemAt(i);
- sp<AnotherPacketSource> src = info->mSource;
- if (src == NULL) {
- --numTracks;
- continue;
- }
- int64_t bufferedDurationUs = src->getBufferedDurationUs(&finalResult);
-
- int64_t initialMarkUs;
- int64_t maxRebufferingMarkUs;
- {
- Mutex::Autolock _l(mBufferingSettingsLock);
- initialMarkUs = mBufferingSettings.mInitialMarkMs * 1000LL;
- // TODO: maxRebufferingMarkUs could be larger than
- // mBufferingSettings.mResumePlaybackMarkMs * 1000ll.
- maxRebufferingMarkUs = mBufferingSettings.mResumePlaybackMarkMs * 1000LL;
- }
- // isFinished when duration is 0 checks for EOS result only
- if (bufferedDurationUs > initialMarkUs
- || src->isFinished(/* duration */ 0)) {
- ++preparedCount;
- }
-
- if (src->isFinished(/* duration */ 0)) {
- ++overflowCount;
- ++finishedCount;
- } else {
- // TODO: redefine kUnderflowMarkMs to a fair value,
- if (bufferedDurationUs < kUnderflowMarkMs * 1000) {
- ++underflowCount;
- }
- if (bufferedDurationUs > maxRebufferingMarkUs) {
- ++overflowCount;
- }
- int64_t startServerMarkUs =
- (kUnderflowMarkMs * 1000LL + maxRebufferingMarkUs) / 2;
- if (bufferedDurationUs < startServerMarkUs) {
- ++startCount;
- }
- }
- }
-
- *prepared = (preparedCount == numTracks);
- *underflow = (underflowCount > 0);
- *overflow = (overflowCount == numTracks);
- *startServer = (startCount > 0);
- *finished = (finishedCount > 0);
-}
-
-void NuPlayer2::RTSPSource2::onPollBuffering() {
- bool prepared, underflow, overflow, startServer, finished;
- checkBuffering(&prepared, &underflow, &overflow, &startServer, &finished);
-
- if (prepared && mInPreparationPhase) {
- mInPreparationPhase = false;
- notifyPrepared();
- }
-
- if (!mInPreparationPhase && underflow) {
- startBufferingIfNecessary();
- }
-
- if (haveSufficientDataOnAllTracks()) {
- stopBufferingIfNecessary();
- }
-
- if (overflow && mHandler != NULL) {
- mHandler->pause();
- }
-
- if (startServer && mHandler != NULL) {
- mHandler->resume();
- }
-
- if (finished && mHandler != NULL) {
- mHandler->cancelAccessUnitTimeoutCheck();
- }
-
- schedulePollBuffering();
-}
-
-void NuPlayer2::RTSPSource2::signalSourceEOS(status_t result) {
- const bool audio = true;
- const bool video = false;
-
- sp<AnotherPacketSource> source = getSource(audio);
- if (source != NULL) {
- source->signalEOS(result);
- }
-
- source = getSource(video);
- if (source != NULL) {
- source->signalEOS(result);
- }
-}
-
-bool NuPlayer2::RTSPSource2::sourceReachedEOS(bool audio) {
- sp<AnotherPacketSource> source = getSource(audio);
- status_t finalResult;
- return (source != NULL &&
- !source->hasBufferAvailable(&finalResult) &&
- finalResult == ERROR_END_OF_STREAM);
-}
-
-bool NuPlayer2::RTSPSource2::sourceNearEOS(bool audio) {
- sp<AnotherPacketSource> source = getSource(audio);
- int64_t mediaDurationUs = 0;
- getDuration(&mediaDurationUs);
- return (source != NULL && source->isFinished(mediaDurationUs));
-}
-
-void NuPlayer2::RTSPSource2::onSignalEOS(const sp<AMessage> &msg) {
- int32_t generation;
- CHECK(msg->findInt32("generation", &generation));
-
- if (generation != mSeekGeneration) {
- return;
- }
-
- if (mEOSPending) {
- signalSourceEOS(ERROR_END_OF_STREAM);
- mEOSPending = false;
- }
-}
-
-void NuPlayer2::RTSPSource2::postSourceEOSIfNecessary() {
- const bool audio = true;
- const bool video = false;
- // If a source has detected near end, give it some time to retrieve more
- // data before signaling EOS
- if (sourceNearEOS(audio) || sourceNearEOS(video)) {
- if (!mEOSPending) {
- sp<AMessage> msg = new AMessage(kWhatSignalEOS, this);
- msg->setInt32("generation", mSeekGeneration);
- msg->post(kNearEOSTimeoutUs);
- mEOSPending = true;
- }
- }
-}
-
-void NuPlayer2::RTSPSource2::onMessageReceived(const sp<AMessage> &msg) {
- if (msg->what() == kWhatDisconnect) {
- sp<AReplyToken> replyID;
- CHECK(msg->senderAwaitsResponse(&replyID));
-
- mDisconnectReplyID = replyID;
- finishDisconnectIfPossible();
- return;
- } else if (msg->what() == kWhatPerformSeek) {
- int32_t generation;
- CHECK(msg->findInt32("generation", &generation));
- CHECK(msg->senderAwaitsResponse(&mSeekReplyID));
-
- if (generation != mSeekGeneration) {
- // obsolete.
- finishSeek(OK);
- return;
- }
-
- int64_t seekTimeUs;
- int32_t mode;
- CHECK(msg->findInt64("timeUs", &seekTimeUs));
- CHECK(msg->findInt32("mode", &mode));
-
- // TODO: add "mode" to performSeek.
- performSeek(seekTimeUs/*, (MediaPlayer2SeekMode)mode */);
- return;
- } else if (msg->what() == kWhatPollBuffering) {
- onPollBuffering();
- return;
- } else if (msg->what() == kWhatSignalEOS) {
- onSignalEOS(msg);
- return;
- }
-
- CHECK_EQ(msg->what(), kWhatNotify);
-
- int32_t what;
- CHECK(msg->findInt32("what", &what));
-
- switch (what) {
- case MyHandler::kWhatConnected:
- {
- onConnected();
-
- notifyVideoSizeChanged();
-
- uint32_t flags = 0;
-
- if (mHandler->isSeekable()) {
- flags = FLAG_CAN_PAUSE
- | FLAG_CAN_SEEK
- | FLAG_CAN_SEEK_BACKWARD
- | FLAG_CAN_SEEK_FORWARD;
- }
-
- notifyFlagsChanged(flags);
- schedulePollBuffering();
- break;
- }
-
- case MyHandler::kWhatDisconnected:
- {
- onDisconnected(msg);
- break;
- }
-
- case MyHandler::kWhatSeekDone:
- {
- mState = CONNECTED;
- // Unblock seekTo here in case we attempted to seek in a live stream
- finishSeek(OK);
- break;
- }
-
- case MyHandler::kWhatSeekPaused:
- {
- sp<AnotherPacketSource> source = getSource(true /* audio */);
- if (source != NULL) {
- source->queueDiscontinuity(ATSParser::DISCONTINUITY_NONE,
- /* extra */ NULL,
- /* discard */ true);
- }
- source = getSource(false /* video */);
- if (source != NULL) {
- source->queueDiscontinuity(ATSParser::DISCONTINUITY_NONE,
- /* extra */ NULL,
- /* discard */ true);
- };
-
- status_t err = OK;
- msg->findInt32("err", &err);
-
- if (err == OK) {
- int64_t timeUs;
- CHECK(msg->findInt64("time", &timeUs));
- mHandler->continueSeekAfterPause(timeUs);
- } else {
- finishSeek(err);
- }
- break;
- }
-
- case MyHandler::kWhatAccessUnit:
- {
- size_t trackIndex;
- CHECK(msg->findSize("trackIndex", &trackIndex));
-
- if (mTSParser == NULL) {
- CHECK_LT(trackIndex, mTracks.size());
- } else {
- CHECK_EQ(trackIndex, 0u);
- }
-
- sp<ABuffer> accessUnit;
- CHECK(msg->findBuffer("accessUnit", &accessUnit));
-
- int32_t damaged;
- if (accessUnit->meta()->findInt32("damaged", &damaged)
- && damaged) {
- ALOGI("dropping damaged access unit.");
- break;
- }
-
- if (mTSParser != NULL) {
- size_t offset = 0;
- status_t err = OK;
- while (offset + 188 <= accessUnit->size()) {
- err = mTSParser->feedTSPacket(
- accessUnit->data() + offset, 188);
- if (err != OK) {
- break;
- }
-
- offset += 188;
- }
-
- if (offset < accessUnit->size()) {
- err = ERROR_MALFORMED;
- }
-
- if (err != OK) {
- signalSourceEOS(err);
- }
-
- postSourceEOSIfNecessary();
- break;
- }
-
- TrackInfo *info = &mTracks.editItemAt(trackIndex);
-
- sp<AnotherPacketSource> source = info->mSource;
- if (source != NULL) {
- uint32_t rtpTime;
- CHECK(accessUnit->meta()->findInt32("rtp-time", (int32_t *)&rtpTime));
-
- if (!info->mNPTMappingValid) {
- // This is a live stream, we didn't receive any normal
- // playtime mapping. We won't map to npt time.
- source->queueAccessUnit(accessUnit);
- break;
- }
-
- int64_t nptUs =
- ((double)rtpTime - (double)info->mRTPTime)
- / info->mTimeScale
- * 1000000LL
- + info->mNormalPlaytimeUs;
-
- accessUnit->meta()->setInt64("timeUs", nptUs);
-
- source->queueAccessUnit(accessUnit);
- }
- postSourceEOSIfNecessary();
- break;
- }
-
- case MyHandler::kWhatEOS:
- {
- int32_t finalResult;
- CHECK(msg->findInt32("finalResult", &finalResult));
- CHECK_NE(finalResult, (status_t)OK);
-
- if (mTSParser != NULL) {
- signalSourceEOS(finalResult);
- }
-
- size_t trackIndex;
- CHECK(msg->findSize("trackIndex", &trackIndex));
- CHECK_LT(trackIndex, mTracks.size());
-
- TrackInfo *info = &mTracks.editItemAt(trackIndex);
- sp<AnotherPacketSource> source = info->mSource;
- if (source != NULL) {
- source->signalEOS(finalResult);
- }
-
- break;
- }
-
- case MyHandler::kWhatSeekDiscontinuity:
- {
- size_t trackIndex;
- CHECK(msg->findSize("trackIndex", &trackIndex));
- CHECK_LT(trackIndex, mTracks.size());
-
- TrackInfo *info = &mTracks.editItemAt(trackIndex);
- sp<AnotherPacketSource> source = info->mSource;
- if (source != NULL) {
- source->queueDiscontinuity(
- ATSParser::DISCONTINUITY_TIME,
- NULL,
- true /* discard */);
- }
-
- break;
- }
-
- case MyHandler::kWhatNormalPlayTimeMapping:
- {
- size_t trackIndex;
- CHECK(msg->findSize("trackIndex", &trackIndex));
- CHECK_LT(trackIndex, mTracks.size());
-
- uint32_t rtpTime;
- CHECK(msg->findInt32("rtpTime", (int32_t *)&rtpTime));
-
- int64_t nptUs;
- CHECK(msg->findInt64("nptUs", &nptUs));
-
- TrackInfo *info = &mTracks.editItemAt(trackIndex);
- info->mRTPTime = rtpTime;
- info->mNormalPlaytimeUs = nptUs;
- info->mNPTMappingValid = true;
- break;
- }
-
- case SDPLoader::kWhatSDPLoaded:
- {
- onSDPLoaded(msg);
- break;
- }
-
- default:
- TRESPASS();
- }
-}
-
-void NuPlayer2::RTSPSource2::onConnected() {
- CHECK(mAudioTrack == NULL);
- CHECK(mVideoTrack == NULL);
-
- size_t numTracks = mHandler->countTracks();
- for (size_t i = 0; i < numTracks; ++i) {
- int32_t timeScale;
- sp<MetaData> format = mHandler->getTrackFormat(i, &timeScale);
-
- const char *mime;
- CHECK(format->findCString(kKeyMIMEType, &mime));
-
- if (!strcasecmp(mime, MEDIA_MIMETYPE_CONTAINER_MPEG2TS)) {
- // Very special case for MPEG2 Transport Streams.
- CHECK_EQ(numTracks, 1u);
-
- mTSParser = new ATSParser;
- return;
- }
-
- bool isAudio = !strncasecmp(mime, "audio/", 6);
- bool isVideo = !strncasecmp(mime, "video/", 6);
-
- TrackInfo info;
- info.mTimeScale = timeScale;
- info.mRTPTime = 0;
- info.mNormalPlaytimeUs = 0LL;
- info.mNPTMappingValid = false;
-
- if ((isAudio && mAudioTrack == NULL)
- || (isVideo && mVideoTrack == NULL)) {
- sp<AnotherPacketSource> source = new AnotherPacketSource(format);
-
- if (isAudio) {
- mAudioTrack = source;
- } else {
- mVideoTrack = source;
- }
-
- info.mSource = source;
- }
-
- mTracks.push(info);
- }
-
- mState = CONNECTED;
-}
-
-void NuPlayer2::RTSPSource2::onSDPLoaded(const sp<AMessage> &msg) {
- status_t err;
- CHECK(msg->findInt32("result", &err));
-
- mSDPLoader.clear();
-
- if (mDisconnectReplyID != 0) {
- err = UNKNOWN_ERROR;
- }
-
- if (err == OK) {
- sp<ASessionDescription> desc;
- sp<RefBase> obj;
- CHECK(msg->findObject("description", &obj));
- desc = static_cast<ASessionDescription *>(obj.get());
-
- AString rtspUri;
- if (!desc->findAttribute(0, "a=control", &rtspUri)) {
- ALOGE("Unable to find url in SDP");
- err = UNKNOWN_ERROR;
- } else {
- sp<AMessage> notify = new AMessage(kWhatNotify, this);
-
- mHandler = new MyHandler(rtspUri.c_str(), notify, true /* uidValid */, mUID);
- mLooper->registerHandler(mHandler);
-
- mHandler->loadSDP(desc);
- }
- }
-
- if (err != OK) {
- if (mState == CONNECTING) {
- // We're still in the preparation phase, signal that it
- // failed.
- notifyPrepared(err);
- }
-
- mState = DISCONNECTED;
- setError(err);
-
- if (mDisconnectReplyID != 0) {
- finishDisconnectIfPossible();
- }
- }
-}
-
-void NuPlayer2::RTSPSource2::onDisconnected(const sp<AMessage> &msg) {
- if (mState == DISCONNECTED) {
- return;
- }
-
- status_t err;
- CHECK(msg->findInt32("result", &err));
- CHECK_NE(err, (status_t)OK);
-
- mLooper->unregisterHandler(mHandler->id());
- mHandler.clear();
-
- if (mState == CONNECTING) {
- // We're still in the preparation phase, signal that it
- // failed.
- notifyPrepared(err);
- }
-
- mState = DISCONNECTED;
- setError(err);
-
- if (mDisconnectReplyID != 0) {
- finishDisconnectIfPossible();
- }
-}
-
-void NuPlayer2::RTSPSource2::finishDisconnectIfPossible() {
- if (mState != DISCONNECTED) {
- if (mHandler != NULL) {
- mHandler->disconnect();
- } else if (mSDPLoader != NULL) {
- mSDPLoader->cancel();
- }
- return;
- }
-
- (new AMessage)->postReply(mDisconnectReplyID);
- mDisconnectReplyID = 0;
-}
-
-void NuPlayer2::RTSPSource2::setError(status_t err) {
- Mutex::Autolock _l(mBufferingLock);
- mFinalResult = err;
-}
-
-void NuPlayer2::RTSPSource2::startBufferingIfNecessary() {
- Mutex::Autolock _l(mBufferingLock);
-
- if (!mBuffering) {
- mBuffering = true;
-
- sp<AMessage> notify = dupNotify();
- notify->setInt32("what", kWhatPauseOnBufferingStart);
- notify->post();
- }
-}
-
-bool NuPlayer2::RTSPSource2::stopBufferingIfNecessary() {
- Mutex::Autolock _l(mBufferingLock);
-
- if (mBuffering) {
- if (!haveSufficientDataOnAllTracks()) {
- return false;
- }
-
- mBuffering = false;
-
- sp<AMessage> notify = dupNotify();
- notify->setInt32("what", kWhatResumeOnBufferingEnd);
- notify->post();
- }
-
- return true;
-}
-
-void NuPlayer2::RTSPSource2::finishSeek(status_t err) {
- if (mSeekReplyID == NULL) {
- return;
- }
- sp<AMessage> seekReply = new AMessage;
- seekReply->setInt32("err", err);
- seekReply->postReply(mSeekReplyID);
- mSeekReplyID = NULL;
-}
-
-} // namespace android
diff --git a/media/libmediaplayer2/nuplayer2/RTSPSource2.h b/media/libmediaplayer2/nuplayer2/RTSPSource2.h
deleted file mode 100644
index e5f1716..0000000
--- a/media/libmediaplayer2/nuplayer2/RTSPSource2.h
+++ /dev/null
@@ -1,167 +0,0 @@
-/*
- * Copyright 2017 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef RTSP_SOURCE2_H_
-
-#define RTSP_SOURCE2_H_
-
-#include "NuPlayer2Source.h"
-
-#include "ATSParser.h"
-
-namespace android {
-
-struct ALooper;
-struct AReplyToken;
-struct AnotherPacketSource;
-struct MyHandler;
-struct SDPLoader;
-
-struct NuPlayer2::RTSPSource2 : public NuPlayer2::Source {
- RTSPSource2(
- const sp<AMessage> ¬ify,
- const sp<MediaHTTPService> &httpService,
- const char *url,
- const KeyedVector<String8, String8> *headers,
- uid_t uid = 0,
- bool isSDP = false);
-
- virtual status_t getBufferingSettings(
- BufferingSettings* buffering /* nonnull */) override;
- virtual status_t setBufferingSettings(const BufferingSettings& buffering) override;
-
- virtual void prepareAsync(int64_t startTimeUs);
- virtual void start();
- virtual void stop();
-
- virtual status_t feedMoreTSData();
-
- virtual status_t dequeueAccessUnit(bool audio, sp<ABuffer> *accessUnit);
-
- virtual status_t getDuration(int64_t *durationUs);
- virtual status_t seekTo(
- int64_t seekTimeUs,
- MediaPlayer2SeekMode mode = MediaPlayer2SeekMode::SEEK_PREVIOUS_SYNC) override;
-
- void onMessageReceived(const sp<AMessage> &msg);
-
-protected:
- virtual ~RTSPSource2();
-
- virtual sp<MetaData> getFormatMeta(bool audio);
-
-private:
- enum {
- kWhatNotify = 'noti',
- kWhatDisconnect = 'disc',
- kWhatPerformSeek = 'seek',
- kWhatPollBuffering = 'poll',
- kWhatSignalEOS = 'eos ',
- };
-
- enum State {
- DISCONNECTED,
- CONNECTING,
- CONNECTED,
- SEEKING,
- };
-
- enum Flags {
- // Don't log any URLs.
- kFlagIncognito = 1,
- };
-
- struct TrackInfo {
- sp<AnotherPacketSource> mSource;
-
- int32_t mTimeScale;
- uint32_t mRTPTime;
- int64_t mNormalPlaytimeUs;
- bool mNPTMappingValid;
- };
-
- sp<MediaHTTPService> mHTTPService;
- AString mURL;
- KeyedVector<String8, String8> mExtraHeaders;
- uid_t mUID;
- uint32_t mFlags;
- bool mIsSDP;
- State mState;
- status_t mFinalResult;
- sp<AReplyToken> mDisconnectReplyID;
- Mutex mBufferingLock;
- bool mBuffering;
- bool mInPreparationPhase;
- bool mEOSPending;
-
- Mutex mBufferingSettingsLock;
- BufferingSettings mBufferingSettings;
-
- sp<ALooper> mLooper;
- sp<MyHandler> mHandler;
- sp<SDPLoader> mSDPLoader;
-
- Vector<TrackInfo> mTracks;
- sp<AnotherPacketSource> mAudioTrack;
- sp<AnotherPacketSource> mVideoTrack;
-
- sp<ATSParser> mTSParser;
-
- int32_t mSeekGeneration;
-
- int64_t mEOSTimeoutAudio;
- int64_t mEOSTimeoutVideo;
-
- sp<AReplyToken> mSeekReplyID;
-
- sp<AnotherPacketSource> getSource(bool audio);
-
- void onConnected();
- void onSDPLoaded(const sp<AMessage> &msg);
- void onDisconnected(const sp<AMessage> &msg);
- void finishDisconnectIfPossible();
-
- void performSeek(int64_t seekTimeUs);
- void schedulePollBuffering();
- void checkBuffering(
- bool *prepared,
- bool *underflow,
- bool *overflow,
- bool *startServer,
- bool *finished);
- void onPollBuffering();
-
- bool haveSufficientDataOnAllTracks();
-
- void setEOSTimeout(bool audio, int64_t timeout);
- void setError(status_t err);
- void startBufferingIfNecessary();
- bool stopBufferingIfNecessary();
- void finishSeek(status_t err);
-
- void postSourceEOSIfNecessary();
- void signalSourceEOS(status_t result);
- void onSignalEOS(const sp<AMessage> &msg);
-
- bool sourceNearEOS(bool audio);
- bool sourceReachedEOS(bool audio);
-
- DISALLOW_EVIL_CONSTRUCTORS(RTSPSource2);
-};
-
-} // namespace android
-
-#endif // RTSP_SOURCE2_H_
diff --git a/media/libmediaplayerservice/Android.bp b/media/libmediaplayerservice/Android.bp
index 6709585..5301f5c 100644
--- a/media/libmediaplayerservice/Android.bp
+++ b/media/libmediaplayerservice/Android.bp
@@ -7,6 +7,7 @@
"MediaPlayerService.cpp",
"MediaRecorderClient.cpp",
"MetadataRetrieverClient.cpp",
+ "StagefrightMetadataRetriever.cpp",
"StagefrightRecorder.cpp",
"TestPlayerStub.cpp",
],
@@ -21,11 +22,14 @@
"libcodec2_client",
"libcrypto",
"libcutils",
+ "libdatasource",
"libdl",
+ "libdrmframework",
"libgui",
"libhidlbase",
"liblog",
"libmedia",
+ "libmedia_codeclist",
"libmedia_omx",
"libmediadrm",
"libmediametrics",
@@ -44,6 +48,7 @@
],
static_libs: [
+ "libplayerservice_datasource",
"libstagefright_nuplayer",
"libstagefright_rtsp",
"libstagefright_timedtext",
diff --git a/media/libmediaplayerservice/MediaPlayerFactory.cpp b/media/libmediaplayerservice/MediaPlayerFactory.cpp
index 1376ccc..05f7365 100644
--- a/media/libmediaplayerservice/MediaPlayerFactory.cpp
+++ b/media/libmediaplayerservice/MediaPlayerFactory.cpp
@@ -20,9 +20,9 @@
#include <utils/Log.h>
#include <cutils/properties.h>
+#include <datasource/FileSource.h>
#include <media/DataSource.h>
#include <media/IMediaPlayer.h>
-#include <media/stagefright/FileSource.h>
#include <media/stagefright/foundation/ADebug.h>
#include <utils/Errors.h>
#include <utils/misc.h>
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index dfd3933..46c130f 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -48,6 +48,7 @@
#include <utils/Vector.h>
#include <codec2/hidl/client.h>
+#include <datasource/HTTPBase.h>
#include <media/IMediaHTTPService.h>
#include <media/IRemoteDisplay.h>
#include <media/IRemoteDisplayClient.h>
@@ -61,6 +62,7 @@
#include <media/stagefright/MediaCodecList.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/Utils.h>
+#include <media/stagefright/FoundationUtils.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/ALooperRoster.h>
#include <media/stagefright/SurfaceUtils.h>
@@ -80,7 +82,6 @@
#include "TestPlayerStub.h"
#include "nuplayer/NuPlayerDriver.h"
-#include "HTTPBase.h"
static const int kDumpLockRetries = 50;
static const int kDumpLockSleepUs = 20000;
diff --git a/media/libmediaplayerservice/MediaPlayerService.h b/media/libmediaplayerservice/MediaPlayerService.h
index 49688ce..2562b8f 100644
--- a/media/libmediaplayerservice/MediaPlayerService.h
+++ b/media/libmediaplayerservice/MediaPlayerService.h
@@ -26,10 +26,12 @@
#include <utils/String8.h>
#include <utils/Vector.h>
+#include <media/AudioSystem.h>
#include <media/MediaPlayerInterface.h>
#include <media/Metadata.h>
#include <media/stagefright/foundation/ABase.h>
+
#include <system/audio.h>
namespace android {
diff --git a/media/libmediaplayerservice/MetadataRetrieverClient.cpp b/media/libmediaplayerservice/MetadataRetrieverClient.cpp
index 40b17bf..4dbab0a 100644
--- a/media/libmediaplayerservice/MetadataRetrieverClient.cpp
+++ b/media/libmediaplayerservice/MetadataRetrieverClient.cpp
@@ -37,6 +37,7 @@
#include <media/MediaPlayerInterface.h>
#include <media/stagefright/InterfaceUtils.h>
#include <media/stagefright/Utils.h>
+#include <media/stagefright/FoundationUtils.h>
#include <private/media/VideoFrame.h>
#include "MetadataRetrieverClient.h"
#include "StagefrightMetadataRetriever.h"
diff --git a/media/libstagefright/StagefrightMetadataRetriever.cpp b/media/libmediaplayerservice/StagefrightMetadataRetriever.cpp
similarity index 98%
rename from media/libstagefright/StagefrightMetadataRetriever.cpp
rename to media/libmediaplayerservice/StagefrightMetadataRetriever.cpp
index fa3d372..1aae241 100644
--- a/media/libstagefright/StagefrightMetadataRetriever.cpp
+++ b/media/libmediaplayerservice/StagefrightMetadataRetriever.cpp
@@ -22,14 +22,14 @@
#include <utils/Log.h>
#include <cutils/properties.h>
-#include "include/FrameDecoder.h"
-#include "include/StagefrightMetadataRetriever.h"
+#include "StagefrightMetadataRetriever.h"
+#include "FrameDecoder.h"
+#include <datasource/PlayerServiceDataSourceFactory.h>
+#include <datasource/PlayerServiceFileSource.h>
#include <media/IMediaHTTPService.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/AMessage.h>
-#include <media/stagefright/DataSourceFactory.h>
-#include <media/stagefright/FileSource.h>
#include <media/stagefright/MediaCodecList.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
@@ -63,7 +63,8 @@
ALOGV("setDataSource(%s)", uri);
clearMetadata();
- mSource = DataSourceFactory::CreateFromURI(httpService, uri, headers);
+ mSource = PlayerServiceDataSourceFactory::getInstance()->CreateFromURI(
+ httpService, uri, headers);
if (mSource == NULL) {
ALOGE("Unable to create data source for '%s'.", uri);
@@ -91,7 +92,7 @@
ALOGV("setDataSource(%d, %" PRId64 ", %" PRId64 ")", fd, offset, length);
clearMetadata();
- mSource = new FileSource(fd, offset, length);
+ mSource = new PlayerServiceFileSource(fd, offset, length);
status_t err;
if ((err = mSource->initCheck()) != OK) {
diff --git a/media/libstagefright/include/StagefrightMetadataRetriever.h b/media/libmediaplayerservice/StagefrightMetadataRetriever.h
similarity index 100%
rename from media/libstagefright/include/StagefrightMetadataRetriever.h
rename to media/libmediaplayerservice/StagefrightMetadataRetriever.h
diff --git a/media/libmediaplayerservice/datasource/Android.bp b/media/libmediaplayerservice/datasource/Android.bp
new file mode 100644
index 0000000..71fa50b
--- /dev/null
+++ b/media/libmediaplayerservice/datasource/Android.bp
@@ -0,0 +1,43 @@
+cc_library_static {
+ name: "libplayerservice_datasource",
+
+ srcs: [
+ "PlayerServiceDataSourceFactory.cpp",
+ "PlayerServiceFileSource.cpp",
+ "PlayerServiceMediaHTTP.cpp",
+ ],
+
+ header_libs: [
+ "media_ndk_headers",
+ "libmedia_headers",
+ ],
+
+ shared_libs: [
+ "libdatasource",
+ "libdrmframework",
+ "liblog",
+ "libutils",
+ ],
+
+ local_include_dirs: [
+ "include",
+ ],
+
+ export_include_dirs: [
+ "include",
+ ],
+
+ cflags: [
+ "-Werror",
+ "-Wno-error=deprecated-declarations",
+ "-Wall",
+ ],
+
+ sanitize: {
+ misc_undefined: [
+ "unsigned-integer-overflow",
+ "signed-integer-overflow",
+ ],
+ cfi: true,
+ },
+}
diff --git a/media/libmediaplayerservice/datasource/PlayerServiceDataSourceFactory.cpp b/media/libmediaplayerservice/datasource/PlayerServiceDataSourceFactory.cpp
new file mode 100644
index 0000000..ef946e9
--- /dev/null
+++ b/media/libmediaplayerservice/datasource/PlayerServiceDataSourceFactory.cpp
@@ -0,0 +1,61 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+//#define LOG_NDEBUG 0
+#define LOG_TAG "PlayerServuceDataSourceFactory"
+
+
+#include <datasource/PlayerServiceDataSourceFactory.h>
+#include <datasource/PlayerServiceFileSource.h>
+#include <datasource/PlayerServiceMediaHTTP.h>
+#include <media/MediaHTTPConnection.h>
+#include <media/MediaHTTPService.h>
+
+namespace android {
+
+// static
+sp<PlayerServiceDataSourceFactory> PlayerServiceDataSourceFactory::sInstance;
+// static
+Mutex PlayerServiceDataSourceFactory::sInstanceLock;
+
+// static
+sp<PlayerServiceDataSourceFactory> PlayerServiceDataSourceFactory::getInstance() {
+ Mutex::Autolock l(sInstanceLock);
+ if (!sInstance) {
+ sInstance = new PlayerServiceDataSourceFactory();
+ }
+ return sInstance;
+}
+
+sp<DataSource> PlayerServiceDataSourceFactory::CreateMediaHTTP(
+ const sp<MediaHTTPService> &httpService) {
+ if (httpService == NULL) {
+ return NULL;
+ }
+
+ sp<MediaHTTPConnection> conn = httpService->makeHTTPConnection();
+ if (conn == NULL) {
+ ALOGE("Failed to make http connection from http service!");
+ return NULL;
+ } else {
+ return new PlayerServiceMediaHTTP(conn);
+ }
+}
+
+sp<DataSource> PlayerServiceDataSourceFactory::CreateFileSource(const char *uri) {
+ return new PlayerServiceFileSource(uri);
+}
+
+} // namespace android
diff --git a/media/libstagefright/FileSource.cpp b/media/libmediaplayerservice/datasource/PlayerServiceFileSource.cpp
similarity index 81%
rename from media/libstagefright/FileSource.cpp
rename to media/libmediaplayerservice/datasource/PlayerServiceFileSource.cpp
index aee7fd8..bb4ba75 100644
--- a/media/libstagefright/FileSource.cpp
+++ b/media/libmediaplayerservice/datasource/PlayerServiceFileSource.cpp
@@ -15,35 +15,36 @@
*/
//#define LOG_NDEBUG 0
-#define LOG_TAG "FileSource"
+#define LOG_TAG "PlayerServiceFileSource"
#include <utils/Log.h>
+#include <datasource/PlayerServiceFileSource.h>
#include <media/stagefright/foundation/ADebug.h>
-#include <media/stagefright/FileSource.h>
-#include <media/stagefright/Utils.h>
#include <private/android_filesystem_config.h>
namespace android {
-FileSource::FileSource(const char *filename)
- : ClearFileSource(filename),
+PlayerServiceFileSource::PlayerServiceFileSource(const char *filename)
+ : FileSource(filename),
mDecryptHandle(NULL),
mDrmManagerClient(NULL),
mDrmBufOffset(0),
mDrmBufSize(0),
mDrmBuf(NULL){
+ (void) DrmInitialization(nullptr);
}
-FileSource::FileSource(int fd, int64_t offset, int64_t length)
- : ClearFileSource(fd, offset, length),
+PlayerServiceFileSource::PlayerServiceFileSource(int fd, int64_t offset, int64_t length)
+ : FileSource(fd, offset, length),
mDecryptHandle(NULL),
mDrmManagerClient(NULL),
mDrmBufOffset(0),
mDrmBufSize(0),
mDrmBuf(NULL) {
+ (void) DrmInitialization(nullptr);
}
-FileSource::~FileSource() {
+PlayerServiceFileSource::~PlayerServiceFileSource() {
if (mDrmBuf != NULL) {
delete[] mDrmBuf;
mDrmBuf = NULL;
@@ -62,7 +63,7 @@
}
}
-ssize_t FileSource::readAt(off64_t offset, void *data, size_t size) {
+ssize_t PlayerServiceFileSource::readAt(off64_t offset, void *data, size_t size) {
if (mFd < 0) {
return NO_INIT;
}
@@ -87,8 +88,10 @@
}
}
-sp<DecryptHandle> FileSource::DrmInitialization(const char *mime) {
- if (getuid() == AID_MEDIA_EX) return nullptr; // no DRM in media extractor
+sp<DecryptHandle> PlayerServiceFileSource::DrmInitialization(const char *mime) {
+ if (getuid() == AID_MEDIA_EX) {
+ return NULL; // no DRM in media extractor
+ }
if (mDrmManagerClient == NULL) {
mDrmManagerClient = new DrmManagerClient();
}
@@ -110,7 +113,7 @@
return mDecryptHandle;
}
-ssize_t FileSource::readAtDRM_l(off64_t offset, void *data, size_t size) {
+ssize_t PlayerServiceFileSource::readAtDRM_l(off64_t offset, void *data, size_t size) {
size_t DRM_CACHE_SIZE = 1024;
if (mDrmBuf == NULL) {
mDrmBuf = new unsigned char[DRM_CACHE_SIZE];
@@ -141,7 +144,7 @@
}
/* static */
-bool FileSource::requiresDrm(int fd, int64_t offset, int64_t length, const char *mime) {
+bool PlayerServiceFileSource::requiresDrm(int fd, int64_t offset, int64_t length, const char *mime) {
std::unique_ptr<DrmManagerClient> drmClient(new DrmManagerClient());
sp<DecryptHandle> decryptHandle =
drmClient->openDecryptSession(fd, offset, length, mime);
diff --git a/media/libstagefright/http/MediaHTTP.cpp b/media/libmediaplayerservice/datasource/PlayerServiceMediaHTTP.cpp
similarity index 77%
rename from media/libstagefright/http/MediaHTTP.cpp
rename to media/libmediaplayerservice/datasource/PlayerServiceMediaHTTP.cpp
index 0fba3dc..f99a861 100644
--- a/media/libstagefright/http/MediaHTTP.cpp
+++ b/media/libmediaplayerservice/datasource/PlayerServiceMediaHTTP.cpp
@@ -15,32 +15,33 @@
*/
//#define LOG_NDEBUG 0
-#define LOG_TAG "MediaHTTP"
+#define LOG_TAG "PlayerServiceMediaHTTP"
#include <utils/Log.h>
-#include <media/stagefright/MediaHTTP.h>
+#include <datasource/PlayerServiceMediaHTTP.h>
#include <binder/IServiceManager.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/ALooper.h>
-#include <media/stagefright/Utils.h>
+#include <media/stagefright/FoundationUtils.h>
#include <media/MediaHTTPConnection.h>
namespace android {
-MediaHTTP::MediaHTTP(const sp<MediaHTTPConnection> &conn)
- : ClearMediaHTTP(conn),
+PlayerServiceMediaHTTP::PlayerServiceMediaHTTP(const sp<MediaHTTPConnection> &conn)
+ : MediaHTTP(conn),
mDrmManagerClient(NULL) {
+ (void) DrmInitialization(nullptr);
}
-MediaHTTP::~MediaHTTP() {
+PlayerServiceMediaHTTP::~PlayerServiceMediaHTTP() {
clearDRMState_l();
}
// DRM...
-sp<DecryptHandle> MediaHTTP::DrmInitialization(const char* mime) {
+sp<DecryptHandle> PlayerServiceMediaHTTP::DrmInitialization(const char *mime) {
if (mDrmManagerClient == NULL) {
mDrmManagerClient = new DrmManagerClient();
}
@@ -62,7 +63,7 @@
return mDecryptHandle;
}
-void MediaHTTP::clearDRMState_l() {
+void PlayerServiceMediaHTTP::clearDRMState_l() {
if (mDecryptHandle != NULL) {
// To release mDecryptHandle
CHECK(mDrmManagerClient);
diff --git a/media/libmediaplayerservice/datasource/include/datasource/PlayerServiceDataSourceFactory.h b/media/libmediaplayerservice/datasource/include/datasource/PlayerServiceDataSourceFactory.h
new file mode 100644
index 0000000..7d58c5c
--- /dev/null
+++ b/media/libmediaplayerservice/datasource/include/datasource/PlayerServiceDataSourceFactory.h
@@ -0,0 +1,48 @@
+/*
+ * Copyright (C) 2017 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef PLAYER_SERVICE_DATA_SOURCE_FACTORY_H_
+
+#define PLAYER_SERVICE_DATA_SOURCE_FACTORY_H_
+
+#include <datasource/DataSourceFactory.h>
+#include <media/DataSource.h>
+#include <sys/types.h>
+#include <utils/RefBase.h>
+
+namespace android {
+
+struct MediaHTTPService;
+class String8;
+struct HTTPBase;
+
+class PlayerServiceDataSourceFactory : public DataSourceFactory {
+public:
+ static sp<PlayerServiceDataSourceFactory> getInstance();
+ virtual sp<DataSource> CreateMediaHTTP(const sp<MediaHTTPService> &httpService);
+
+protected:
+ virtual sp<DataSource> CreateFileSource(const char *uri);
+
+private:
+ static sp<PlayerServiceDataSourceFactory> sInstance;
+ static Mutex sInstanceLock;
+ PlayerServiceDataSourceFactory() {};
+};
+
+} // namespace android
+
+#endif // PLAYER_SERVICE_DATA_SOURCE_FACTORY_H_
diff --git a/media/libstagefright/include/media/stagefright/FileSource.h b/media/libmediaplayerservice/datasource/include/datasource/PlayerServiceFileSource.h
similarity index 62%
rename from media/libstagefright/include/media/stagefright/FileSource.h
rename to media/libmediaplayerservice/datasource/include/datasource/PlayerServiceFileSource.h
index b610eef..7ae8dda 100644
--- a/media/libstagefright/include/media/stagefright/FileSource.h
+++ b/media/libmediaplayerservice/datasource/include/datasource/PlayerServiceFileSource.h
@@ -14,33 +14,33 @@
* limitations under the License.
*/
-#ifndef FILE_SOURCE_H_
+#ifndef PLAYER_SERVICE_FILE_SOURCE_H_
-#define FILE_SOURCE_H_
+#define PLAYER_SERVICE_FILE_SOURCE_H_
#include <stdio.h>
-#include <media/stagefright/ClearFileSource.h>
+#include <datasource/FileSource.h>
#include <media/stagefright/MediaErrors.h>
#include <utils/threads.h>
#include <drm/DrmManagerClient.h>
namespace android {
-class FileSource : public ClearFileSource {
+// FileSource implementation which works on MediaPlayerService.
+// Supports OMA(forword-lock) files.
+class PlayerServiceFileSource : public FileSource {
public:
- FileSource(const char *filename);
- // FileSource takes ownership and will close the fd
- FileSource(int fd, int64_t offset, int64_t length);
+ PlayerServiceFileSource(const char *filename);
+ // PlayerServiceFileSource takes ownership and will close the fd
+ PlayerServiceFileSource(int fd, int64_t offset, int64_t length);
virtual ssize_t readAt(off64_t offset, void *data, size_t size);
- virtual sp<DecryptHandle> DrmInitialization(const char *mime);
-
static bool requiresDrm(int fd, int64_t offset, int64_t length, const char *mime);
protected:
- virtual ~FileSource();
+ virtual ~PlayerServiceFileSource();
private:
/*for DRM*/
@@ -50,13 +50,14 @@
ssize_t mDrmBufSize;
unsigned char *mDrmBuf;
+ sp<DecryptHandle> DrmInitialization(const char *mime);
ssize_t readAtDRM_l(off64_t offset, void *data, size_t size);
- FileSource(const FileSource &);
- FileSource &operator=(const FileSource &);
+ PlayerServiceFileSource(const PlayerServiceFileSource &);
+ PlayerServiceFileSource &operator=(const PlayerServiceFileSource &);
};
} // namespace android
-#endif // FILE_SOURCE_H_
+#endif // PLAYER_SERVICE_FILE_SOURCE_H_
diff --git a/media/libstagefright/include/media/stagefright/MediaHTTP.h b/media/libmediaplayerservice/datasource/include/datasource/PlayerServiceMediaHTTP.h
similarity index 62%
rename from media/libstagefright/include/media/stagefright/MediaHTTP.h
rename to media/libmediaplayerservice/datasource/include/datasource/PlayerServiceMediaHTTP.h
index acaa6c4..b5124dc 100644
--- a/media/libstagefright/include/media/stagefright/MediaHTTP.h
+++ b/media/libmediaplayerservice/datasource/include/datasource/PlayerServiceMediaHTTP.h
@@ -14,34 +14,35 @@
* limitations under the License.
*/
-#ifndef MEDIA_HTTP_H_
+#ifndef PLAYER_SERVICE_MEDIA_HTTP_H_
-#define MEDIA_HTTP_H_
+#define PLAYER_SERVICE_MEDIA_HTTP_H_
+#include <datasource/MediaHTTP.h>
#include <media/stagefright/foundation/AString.h>
-#include <media/stagefright/ClearMediaHTTP.h>
namespace android {
struct MediaHTTPConnection;
-struct MediaHTTP : public ClearMediaHTTP {
- MediaHTTP(const sp<MediaHTTPConnection> &conn);
+// MediaHTTP implementation which works on MediaPlayerService.
+// Supports OMA(forword-lock) stream.
+struct PlayerServiceMediaHTTP : public MediaHTTP {
+ PlayerServiceMediaHTTP(const sp<MediaHTTPConnection> &conn);
protected:
- virtual ~MediaHTTP();
-
- virtual sp<DecryptHandle> DrmInitialization(const char* mime);
+ virtual ~PlayerServiceMediaHTTP();
private:
sp<DecryptHandle> mDecryptHandle;
DrmManagerClient *mDrmManagerClient;
+ sp<DecryptHandle> DrmInitialization(const char *mime);
void clearDRMState_l();
- DISALLOW_EVIL_CONSTRUCTORS(MediaHTTP);
+ DISALLOW_EVIL_CONSTRUCTORS(PlayerServiceMediaHTTP);
};
} // namespace android
-#endif // MEDIA_HTTP_H_
+#endif // PLAYER_SERVICE_MEDIA_HTTP_H_
diff --git a/media/libmediaplayerservice/include/MediaPlayerInterface.h b/media/libmediaplayerservice/include/MediaPlayerInterface.h
index 0ad4d04..436cb31 100644
--- a/media/libmediaplayerservice/include/MediaPlayerInterface.h
+++ b/media/libmediaplayerservice/include/MediaPlayerInterface.h
@@ -27,7 +27,6 @@
#include <media/mediaplayer.h>
#include <media/AudioResamplerPublic.h>
-#include <media/AudioSystem.h>
#include <media/AudioTimestamp.h>
#include <media/AVSyncSettings.h>
#include <media/BufferingSettings.h>
diff --git a/media/libmediaplayerservice/nuplayer/Android.bp b/media/libmediaplayerservice/nuplayer/Android.bp
index 23a19e7..c8f48a2 100644
--- a/media/libmediaplayerservice/nuplayer/Android.bp
+++ b/media/libmediaplayerservice/nuplayer/Android.bp
@@ -18,6 +18,7 @@
],
header_libs: [
+ "libmediadrm_headers",
"media_plugin_headers",
],
@@ -45,6 +46,7 @@
shared_libs: [
"libbinder",
+ "libdatasource",
"libui",
"libgui",
"libmedia",
@@ -52,6 +54,10 @@
"libpowermanager",
],
+ static_libs: [
+ "libplayerservice_datasource",
+ ],
+
name: "libstagefright_nuplayer",
sanitize: {
diff --git a/media/libmediaplayerservice/nuplayer/GenericSource.cpp b/media/libmediaplayerservice/nuplayer/GenericSource.cpp
index 4653711..00e3443 100644
--- a/media/libmediaplayerservice/nuplayer/GenericSource.cpp
+++ b/media/libmediaplayerservice/nuplayer/GenericSource.cpp
@@ -23,6 +23,10 @@
#include "AnotherPacketSource.h"
#include <binder/IServiceManager.h>
#include <cutils/properties.h>
+#include <datasource/PlayerServiceDataSourceFactory.h>
+#include <datasource/PlayerServiceFileSource.h>
+#include <datasource/HTTPBase.h>
+#include <datasource/NuCachedSource2.h>
#include <media/DataSource.h>
#include <media/MediaBufferHolder.h>
#include <media/MediaSource.h>
@@ -31,8 +35,6 @@
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/AMessage.h>
-#include <media/stagefright/DataSourceFactory.h>
-#include <media/stagefright/FileSource.h>
#include <media/stagefright/InterfaceUtils.h>
#include <media/stagefright/MediaBuffer.h>
#include <media/stagefright/MediaClock.h>
@@ -41,8 +43,6 @@
#include <media/stagefright/MediaExtractorFactory.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/Utils.h>
-#include "../../libstagefright/include/NuCachedSource2.h"
-#include "../../libstagefright/include/HTTPBase.h"
namespace android {
@@ -385,7 +385,8 @@
if (!strncasecmp("http://", uri, 7) || !strncasecmp("https://", uri, 8)) {
sp<DataSource> httpSource;
mDisconnectLock.unlock();
- httpSource = DataSourceFactory::CreateMediaHTTP(mHTTPService);
+ httpSource = PlayerServiceDataSourceFactory::getInstance()
+ ->CreateMediaHTTP(mHTTPService);
if (httpSource == NULL) {
ALOGE("Failed to create http source!");
notifyPreparedAndCleanup(UNKNOWN_ERROR);
@@ -401,9 +402,9 @@
mLock.unlock();
mDisconnectLock.unlock();
// This might take long time if connection has some issue.
- sp<DataSource> dataSource = DataSourceFactory::CreateFromURI(
- mHTTPService, uri, &mUriHeaders, &contentType,
- static_cast<HTTPBase *>(mHttpSource.get()));
+ sp<DataSource> dataSource = PlayerServiceDataSourceFactory::getInstance()
+ ->CreateFromURI(mHTTPService, uri, &mUriHeaders, &contentType,
+ static_cast<HTTPBase *>(mHttpSource.get()));
mDisconnectLock.lock();
mLock.lock();
if (!mDisconnected) {
@@ -411,7 +412,8 @@
}
} else {
if (property_get_bool("media.stagefright.extractremote", true) &&
- !FileSource::requiresDrm(mFd, mOffset, mLength, nullptr /* mime */)) {
+ !PlayerServiceFileSource::requiresDrm(
+ mFd, mOffset, mLength, nullptr /* mime */)) {
sp<IBinder> binder =
defaultServiceManager()->getService(String16("media.extractor"));
if (binder != nullptr) {
@@ -438,7 +440,7 @@
}
if (mDataSource == nullptr) {
ALOGD("FileSource local");
- mDataSource = new FileSource(mFd, mOffset, mLength);
+ mDataSource = new PlayerServiceFileSource(mFd, mOffset, mLength);
}
// TODO: close should always be done on mFd, see the lines following
// CreateDataSourceFromIDataSource above,
@@ -782,7 +784,7 @@
return;
}
- int64_t nextSubTimeUs;
+ int64_t nextSubTimeUs = 0;
readBuffer(type, -1, MediaPlayerSeekMode::SEEK_PREVIOUS_SYNC /* mode */, &nextSubTimeUs);
sp<ABuffer> buffer;
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
index 3388097..c1c4b55 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
@@ -1798,7 +1798,9 @@
}
void NuPlayer::closeAudioSink() {
- mRenderer->closeAudioSink();
+ if (mRenderer != NULL) {
+ mRenderer->closeAudioSink();
+ }
}
void NuPlayer::restartAudio(
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.h b/media/libmediaplayerservice/nuplayer/NuPlayer.h
index 9f5be06..0e58ec2 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.h
@@ -19,7 +19,7 @@
#define NU_PLAYER_H_
#include <media/AudioResamplerPublic.h>
-#include <media/ICrypto.h>
+#include <mediadrm/ICrypto.h>
#include <media/MediaPlayerInterface.h>
#include <media/stagefright/foundation/AHandler.h>
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
index 2f0da2d..bd2b884 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
@@ -28,7 +28,7 @@
#include "NuPlayerSource.h"
#include <cutils/properties.h>
-#include <media/ICrypto.h>
+#include <mediadrm/ICrypto.h>
#include <media/MediaBufferHolder.h>
#include <media/MediaCodecBuffer.h>
#include <media/stagefright/foundation/ABuffer.h>
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDecoderPassThrough.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDecoderPassThrough.cpp
index 0997e7d..793014e 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDecoderPassThrough.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDecoderPassThrough.cpp
@@ -24,7 +24,7 @@
#include "NuPlayerRenderer.h"
#include "NuPlayerSource.h"
-#include <media/ICrypto.h>
+#include <mediadrm/ICrypto.h>
#include <media/MediaCodecBuffer.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/ADebug.h>
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
index 865cb2a..95c973a 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
@@ -33,6 +33,7 @@
#include <media/stagefright/MediaClock.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/Utils.h>
+#include <media/stagefright/FoundationUtils.h>
#include <media/IMediaAnalyticsService.h>
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDrm.h b/media/libmediaplayerservice/nuplayer/NuPlayerDrm.h
index 50f69ff..4360656 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDrm.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDrm.h
@@ -18,8 +18,8 @@
#define NUPLAYER_DRM_H_
#include <binder/Parcel.h>
-#include <media/ICrypto.h>
-#include <media/IDrm.h>
+#include <mediadrm/ICrypto.h>
+#include <mediadrm/IDrm.h>
#include <media/stagefright/MetaData.h> // for CryptInfo
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerSource.h b/media/libmediaplayerservice/nuplayer/NuPlayerSource.h
index 9f5ef78..f137c52 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerSource.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerSource.h
@@ -20,7 +20,7 @@
#include "NuPlayer.h"
-#include <media/ICrypto.h>
+#include <mediadrm/ICrypto.h>
#include <media/mediaplayer.h>
#include <media/stagefright/foundation/AMessage.h>
#include <media/stagefright/MetaData.h>
diff --git a/media/libmediaplayerservice/tests/Android.bp b/media/libmediaplayerservice/tests/Android.bp
index f8c89e5..8357925 100644
--- a/media/libmediaplayerservice/tests/Android.bp
+++ b/media/libmediaplayerservice/tests/Android.bp
@@ -6,14 +6,22 @@
shared_libs: [
"liblog",
+ "libbinder",
+ "libmedia",
"libmediaplayerservice",
"libmediadrm",
+ "libresourcemanagerservice",
"libutils",
"android.hardware.drm@1.0",
"android.hardware.drm@1.1",
"android.hardware.drm@1.2",
],
+ include_dirs: [
+ "frameworks/av/include",
+ "frameworks/av/services/mediaresourcemanager",
+ ],
+
cflags: [
"-Werror",
"-Wall",
diff --git a/media/libmediaplayerservice/tests/DrmSessionManager_test.cpp b/media/libmediaplayerservice/tests/DrmSessionManager_test.cpp
index d81ee05..58e4bee 100644
--- a/media/libmediaplayerservice/tests/DrmSessionManager_test.cpp
+++ b/media/libmediaplayerservice/tests/DrmSessionManager_test.cpp
@@ -20,14 +20,29 @@
#include <gtest/gtest.h>
+#include <media/IResourceManagerService.h>
+#include <media/IResourceManagerClient.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/ProcessInfoInterface.h>
#include <mediadrm/DrmHal.h>
#include <mediadrm/DrmSessionClientInterface.h>
#include <mediadrm/DrmSessionManager.h>
+#include <algorithm>
+#include <vector>
+
+#include "ResourceManagerService.h"
+
namespace android {
+static Vector<uint8_t> toAndroidVector(const std::vector<uint8_t> &vec) {
+ Vector<uint8_t> aVec;
+ for (auto b : vec) {
+ aVec.push_back(b);
+ }
+ return aVec;
+}
+
struct FakeProcessInfo : public ProcessInfoInterface {
FakeProcessInfo() {}
virtual ~FakeProcessInfo() {}
@@ -47,173 +62,128 @@
DISALLOW_EVIL_CONSTRUCTORS(FakeProcessInfo);
};
-struct FakeDrm : public DrmSessionClientInterface {
- FakeDrm() {}
+struct FakeDrm : public BnResourceManagerClient {
+ FakeDrm(const std::vector<uint8_t>& sessionId, const sp<DrmSessionManager>& manager)
+ : mSessionId(toAndroidVector(sessionId)),
+ mReclaimed(false),
+ mDrmSessionManager(manager) {}
+
virtual ~FakeDrm() {}
- virtual bool reclaimSession(const Vector<uint8_t>& sessionId) {
- mReclaimedSessions.push_back(sessionId);
+ virtual bool reclaimResource() {
+ mReclaimed = true;
+ mDrmSessionManager->removeSession(mSessionId);
return true;
}
- const Vector<Vector<uint8_t> >& reclaimedSessions() const {
- return mReclaimedSessions;
+ virtual String8 getName() {
+ String8 name("FakeDrm[");
+ for (size_t i = 0; i < mSessionId.size(); ++i) {
+ name.appendFormat("%02x", mSessionId[i]);
+ }
+ name.append("]");
+ return name;
}
+ bool isReclaimed() const {
+ return mReclaimed;
+ }
+
+ const Vector<uint8_t> mSessionId;
+
private:
- Vector<Vector<uint8_t> > mReclaimedSessions;
+ bool mReclaimed;
+ const sp<DrmSessionManager> mDrmSessionManager;
DISALLOW_EVIL_CONSTRUCTORS(FakeDrm);
};
+struct FakeSystemCallback :
+ public ResourceManagerService::SystemCallbackInterface {
+ FakeSystemCallback() {}
+
+ virtual void noteStartVideo(int /*uid*/) override {}
+
+ virtual void noteStopVideo(int /*uid*/) override {}
+
+ virtual void noteResetVideo() override {}
+
+ virtual bool requestCpusetBoost(
+ bool /*enable*/, const sp<IInterface> &/*client*/) override {
+ return true;
+ }
+
+protected:
+ virtual ~FakeSystemCallback() {}
+
+private:
+
+ DISALLOW_EVIL_CONSTRUCTORS(FakeSystemCallback);
+};
+
static const int kTestPid1 = 30;
static const int kTestPid2 = 20;
-static const uint8_t kTestSessionId1[] = {1, 2, 3};
-static const uint8_t kTestSessionId2[] = {4, 5, 6, 7, 8};
-static const uint8_t kTestSessionId3[] = {9, 0};
+static const std::vector<uint8_t> kTestSessionId1{1, 2, 3};
+static const std::vector<uint8_t> kTestSessionId2{4, 5, 6, 7, 8};
+static const std::vector<uint8_t> kTestSessionId3{9, 0};
class DrmSessionManagerTest : public ::testing::Test {
public:
DrmSessionManagerTest()
- : mDrmSessionManager(new DrmSessionManager(new FakeProcessInfo())),
- mTestDrm1(new FakeDrm()),
- mTestDrm2(new FakeDrm()) {
- GetSessionId(kTestSessionId1, ARRAY_SIZE(kTestSessionId1), &mSessionId1);
- GetSessionId(kTestSessionId2, ARRAY_SIZE(kTestSessionId2), &mSessionId2);
- GetSessionId(kTestSessionId3, ARRAY_SIZE(kTestSessionId3), &mSessionId3);
+ : mService(new ResourceManagerService(new FakeProcessInfo(), new FakeSystemCallback())),
+ mDrmSessionManager(new DrmSessionManager(mService)),
+ mTestDrm1(new FakeDrm(kTestSessionId1, mDrmSessionManager)),
+ mTestDrm2(new FakeDrm(kTestSessionId2, mDrmSessionManager)),
+ mTestDrm3(new FakeDrm(kTestSessionId3, mDrmSessionManager)) {
+ DrmSessionManager *ptr = new DrmSessionManager(mService);
+ EXPECT_NE(ptr, nullptr);
+ /* mDrmSessionManager = ptr; */
}
protected:
- static void GetSessionId(const uint8_t* ids, size_t num, Vector<uint8_t>* sessionId) {
- for (size_t i = 0; i < num; ++i) {
- sessionId->push_back(ids[i]);
- }
- }
-
- static void ExpectEqSessionInfo(const SessionInfo& info, sp<DrmSessionClientInterface> drm,
- const Vector<uint8_t>& sessionId, int64_t timeStamp) {
- EXPECT_EQ(drm, info.drm);
- EXPECT_TRUE(isEqualSessionId(sessionId, info.sessionId));
- EXPECT_EQ(timeStamp, info.timeStamp);
- }
-
void addSession() {
- mDrmSessionManager->addSession(kTestPid1, mTestDrm1, mSessionId1);
- mDrmSessionManager->addSession(kTestPid2, mTestDrm2, mSessionId2);
- mDrmSessionManager->addSession(kTestPid2, mTestDrm2, mSessionId3);
- const PidSessionInfosMap& map = sessionMap();
- EXPECT_EQ(2u, map.size());
- ssize_t index1 = map.indexOfKey(kTestPid1);
- ASSERT_GE(index1, 0);
- const SessionInfos& infos1 = map[index1];
- EXPECT_EQ(1u, infos1.size());
- ExpectEqSessionInfo(infos1[0], mTestDrm1, mSessionId1, 0);
-
- ssize_t index2 = map.indexOfKey(kTestPid2);
- ASSERT_GE(index2, 0);
- const SessionInfos& infos2 = map[index2];
- EXPECT_EQ(2u, infos2.size());
- ExpectEqSessionInfo(infos2[0], mTestDrm2, mSessionId2, 1);
- ExpectEqSessionInfo(infos2[1], mTestDrm2, mSessionId3, 2);
+ mDrmSessionManager->addSession(kTestPid1, mTestDrm1, mTestDrm1->mSessionId);
+ mDrmSessionManager->addSession(kTestPid2, mTestDrm2, mTestDrm2->mSessionId);
+ mDrmSessionManager->addSession(kTestPid2, mTestDrm3, mTestDrm3->mSessionId);
}
- const PidSessionInfosMap& sessionMap() {
- return mDrmSessionManager->mSessionMap;
- }
-
- void testGetLowestPriority() {
- int pid;
- int priority;
- EXPECT_FALSE(mDrmSessionManager->getLowestPriority_l(&pid, &priority));
-
- addSession();
- EXPECT_TRUE(mDrmSessionManager->getLowestPriority_l(&pid, &priority));
-
- EXPECT_EQ(kTestPid1, pid);
- FakeProcessInfo processInfo;
- int priority1;
- processInfo.getPriority(kTestPid1, &priority1);
- EXPECT_EQ(priority1, priority);
- }
-
- void testGetLeastUsedSession() {
- sp<DrmSessionClientInterface> drm;
- Vector<uint8_t> sessionId;
- EXPECT_FALSE(mDrmSessionManager->getLeastUsedSession_l(kTestPid1, &drm, &sessionId));
-
- addSession();
-
- EXPECT_TRUE(mDrmSessionManager->getLeastUsedSession_l(kTestPid1, &drm, &sessionId));
- EXPECT_EQ(mTestDrm1, drm);
- EXPECT_TRUE(isEqualSessionId(mSessionId1, sessionId));
-
- EXPECT_TRUE(mDrmSessionManager->getLeastUsedSession_l(kTestPid2, &drm, &sessionId));
- EXPECT_EQ(mTestDrm2, drm);
- EXPECT_TRUE(isEqualSessionId(mSessionId2, sessionId));
-
- // mSessionId2 is no longer the least used session.
- mDrmSessionManager->useSession(mSessionId2);
- EXPECT_TRUE(mDrmSessionManager->getLeastUsedSession_l(kTestPid2, &drm, &sessionId));
- EXPECT_EQ(mTestDrm2, drm);
- EXPECT_TRUE(isEqualSessionId(mSessionId3, sessionId));
- }
-
+ sp<IResourceManagerService> mService;
sp<DrmSessionManager> mDrmSessionManager;
sp<FakeDrm> mTestDrm1;
sp<FakeDrm> mTestDrm2;
- Vector<uint8_t> mSessionId1;
- Vector<uint8_t> mSessionId2;
- Vector<uint8_t> mSessionId3;
+ sp<FakeDrm> mTestDrm3;
};
TEST_F(DrmSessionManagerTest, addSession) {
addSession();
+
+ EXPECT_EQ(3u, mDrmSessionManager->getSessionCount());
+ EXPECT_TRUE(mDrmSessionManager->containsSession(mTestDrm1->mSessionId));
+ EXPECT_TRUE(mDrmSessionManager->containsSession(mTestDrm2->mSessionId));
+ EXPECT_TRUE(mDrmSessionManager->containsSession(mTestDrm3->mSessionId));
}
TEST_F(DrmSessionManagerTest, useSession) {
addSession();
- mDrmSessionManager->useSession(mSessionId1);
- mDrmSessionManager->useSession(mSessionId3);
+ mDrmSessionManager->useSession(mTestDrm1->mSessionId);
+ mDrmSessionManager->useSession(mTestDrm3->mSessionId);
- const PidSessionInfosMap& map = sessionMap();
- const SessionInfos& infos1 = map.valueFor(kTestPid1);
- const SessionInfos& infos2 = map.valueFor(kTestPid2);
- ExpectEqSessionInfo(infos1[0], mTestDrm1, mSessionId1, 3);
- ExpectEqSessionInfo(infos2[1], mTestDrm2, mSessionId3, 4);
+ EXPECT_EQ(3u, mDrmSessionManager->getSessionCount());
+ EXPECT_TRUE(mDrmSessionManager->containsSession(mTestDrm1->mSessionId));
+ EXPECT_TRUE(mDrmSessionManager->containsSession(mTestDrm2->mSessionId));
+ EXPECT_TRUE(mDrmSessionManager->containsSession(mTestDrm3->mSessionId));
}
TEST_F(DrmSessionManagerTest, removeSession) {
addSession();
- mDrmSessionManager->removeSession(mSessionId2);
+ mDrmSessionManager->removeSession(mTestDrm2->mSessionId);
- const PidSessionInfosMap& map = sessionMap();
- EXPECT_EQ(2u, map.size());
- const SessionInfos& infos1 = map.valueFor(kTestPid1);
- const SessionInfos& infos2 = map.valueFor(kTestPid2);
- EXPECT_EQ(1u, infos1.size());
- EXPECT_EQ(1u, infos2.size());
- // mSessionId2 has been removed.
- ExpectEqSessionInfo(infos2[0], mTestDrm2, mSessionId3, 2);
-}
-
-TEST_F(DrmSessionManagerTest, removeDrm) {
- addSession();
-
- sp<FakeDrm> drm = new FakeDrm;
- const uint8_t ids[] = {123};
- Vector<uint8_t> sessionId;
- GetSessionId(ids, ARRAY_SIZE(ids), &sessionId);
- mDrmSessionManager->addSession(kTestPid2, drm, sessionId);
-
- mDrmSessionManager->removeDrm(mTestDrm2);
-
- const PidSessionInfosMap& map = sessionMap();
- const SessionInfos& infos2 = map.valueFor(kTestPid2);
- EXPECT_EQ(1u, infos2.size());
- // mTestDrm2 has been removed.
- ExpectEqSessionInfo(infos2[0], drm, sessionId, 3);
+ EXPECT_EQ(2u, mDrmSessionManager->getSessionCount());
+ EXPECT_TRUE(mDrmSessionManager->containsSession(mTestDrm1->mSessionId));
+ EXPECT_FALSE(mDrmSessionManager->containsSession(mTestDrm2->mSessionId));
+ EXPECT_TRUE(mDrmSessionManager->containsSession(mTestDrm3->mSessionId));
}
TEST_F(DrmSessionManagerTest, reclaimSession) {
@@ -224,30 +194,63 @@
EXPECT_FALSE(mDrmSessionManager->reclaimSession(50));
EXPECT_TRUE(mDrmSessionManager->reclaimSession(10));
- EXPECT_EQ(1u, mTestDrm1->reclaimedSessions().size());
- EXPECT_TRUE(isEqualSessionId(mSessionId1, mTestDrm1->reclaimedSessions()[0]));
-
- mDrmSessionManager->removeSession(mSessionId1);
+ EXPECT_TRUE(mTestDrm1->isReclaimed());
// add a session from a higher priority process.
- sp<FakeDrm> drm = new FakeDrm;
- const uint8_t ids[] = {1, 3, 5};
- Vector<uint8_t> sessionId;
- GetSessionId(ids, ARRAY_SIZE(ids), &sessionId);
- mDrmSessionManager->addSession(15, drm, sessionId);
+ const std::vector<uint8_t> sid{1, 3, 5};
+ sp<FakeDrm> drm = new FakeDrm(sid, mDrmSessionManager);
+ mDrmSessionManager->addSession(15, drm, drm->mSessionId);
+ // make sure mTestDrm2 is reclaimed next instead of mTestDrm3
+ mDrmSessionManager->useSession(mTestDrm3->mSessionId);
EXPECT_TRUE(mDrmSessionManager->reclaimSession(18));
- EXPECT_EQ(1u, mTestDrm2->reclaimedSessions().size());
- // mSessionId2 is reclaimed.
- EXPECT_TRUE(isEqualSessionId(mSessionId2, mTestDrm2->reclaimedSessions()[0]));
+ EXPECT_TRUE(mTestDrm2->isReclaimed());
+
+ EXPECT_EQ(2u, mDrmSessionManager->getSessionCount());
+ EXPECT_FALSE(mDrmSessionManager->containsSession(mTestDrm1->mSessionId));
+ EXPECT_FALSE(mDrmSessionManager->containsSession(mTestDrm2->mSessionId));
+ EXPECT_TRUE(mDrmSessionManager->containsSession(mTestDrm3->mSessionId));
+ EXPECT_TRUE(mDrmSessionManager->containsSession(drm->mSessionId));
}
-TEST_F(DrmSessionManagerTest, getLowestPriority) {
- testGetLowestPriority();
-}
+TEST_F(DrmSessionManagerTest, reclaimAfterUse) {
+ // nothing to reclaim yet
+ EXPECT_FALSE(mDrmSessionManager->reclaimSession(kTestPid1));
+ EXPECT_FALSE(mDrmSessionManager->reclaimSession(kTestPid2));
-TEST_F(DrmSessionManagerTest, getLeastUsedSession_l) {
- testGetLeastUsedSession();
+ // add sessions from same pid
+ mDrmSessionManager->addSession(kTestPid2, mTestDrm1, mTestDrm1->mSessionId);
+ mDrmSessionManager->addSession(kTestPid2, mTestDrm2, mTestDrm2->mSessionId);
+ mDrmSessionManager->addSession(kTestPid2, mTestDrm3, mTestDrm3->mSessionId);
+
+ // use some but not all sessions
+ mDrmSessionManager->useSession(mTestDrm1->mSessionId);
+ mDrmSessionManager->useSession(mTestDrm1->mSessionId);
+ mDrmSessionManager->useSession(mTestDrm2->mSessionId);
+
+ // calling pid priority is too low
+ int lowPriorityPid = kTestPid2 + 1;
+ EXPECT_FALSE(mDrmSessionManager->reclaimSession(lowPriorityPid));
+
+ // unused session is reclaimed first
+ int highPriorityPid = kTestPid2 - 1;
+ EXPECT_TRUE(mDrmSessionManager->reclaimSession(highPriorityPid));
+ EXPECT_FALSE(mTestDrm1->isReclaimed());
+ EXPECT_FALSE(mTestDrm2->isReclaimed());
+ EXPECT_TRUE(mTestDrm3->isReclaimed());
+ mDrmSessionManager->removeSession(mTestDrm3->mSessionId);
+
+ // less-used session is reclaimed next
+ EXPECT_TRUE(mDrmSessionManager->reclaimSession(highPriorityPid));
+ EXPECT_FALSE(mTestDrm1->isReclaimed());
+ EXPECT_TRUE(mTestDrm2->isReclaimed());
+ EXPECT_TRUE(mTestDrm3->isReclaimed());
+
+ // most-used session still open
+ EXPECT_EQ(1u, mDrmSessionManager->getSessionCount());
+ EXPECT_TRUE(mDrmSessionManager->containsSession(mTestDrm1->mSessionId));
+ EXPECT_FALSE(mDrmSessionManager->containsSession(mTestDrm2->mSessionId));
+ EXPECT_FALSE(mDrmSessionManager->containsSession(mTestDrm3->mSessionId));
}
} // namespace android
diff --git a/media/libnbaio/Android.bp b/media/libnbaio/Android.bp
index a4df38d..04ddcff 100644
--- a/media/libnbaio/Android.bp
+++ b/media/libnbaio/Android.bp
@@ -1,4 +1,3 @@
-
cc_defaults {
name: "libnbaio_mono_defaults",
srcs: [
@@ -9,20 +8,27 @@
header_libs: [
"libaudioclient_headers",
"libaudio_system_headers",
- "libmedia_headers",
],
export_header_lib_headers: [
"libaudioclient_headers",
- "libmedia_headers",
],
shared_libs: [
"libaudioutils",
+ "libcutils",
"liblog",
"libutils",
],
+ export_shared_lib_headers: [
+ "libaudioutils",
+ ],
export_include_dirs: ["include_mono"],
+
+ cflags: [
+ "-Werror",
+ "-Wall",
+ ],
}
// libnbaio_mono is the part of libnbaio that is available for vendors to use. Vendor modules can't
@@ -53,20 +59,7 @@
// ],
// static_libs: ["libsndfile"],
- shared_libs: [
- "libaudioutils",
- "libbinder",
- "libcutils",
- "liblog",
- "libutils",
- ],
-
- cflags: [
- "-Werror",
- "-Wall",
- ],
-
- include_dirs: ["system/media/audio_utils/include"],
+ header_libs: ["libaudiohal_headers"],
export_include_dirs: ["include"],
}
diff --git a/media/libnbaio/include_mono/media/nbaio/MonoPipe.h b/media/libnbaio/include_mono/media/nbaio/MonoPipe.h
index c51d0fe..926d84a 100644
--- a/media/libnbaio/include_mono/media/nbaio/MonoPipe.h
+++ b/media/libnbaio/include_mono/media/nbaio/MonoPipe.h
@@ -19,7 +19,7 @@
#include <time.h>
#include <audio_utils/fifo.h>
-#include <media/SingleStateQueue.h>
+#include <media/nbaio/SingleStateQueue.h>
#include <media/nbaio/NBAIO.h>
namespace android {
diff --git a/media/libmedia/include/media/SingleStateQueue.h b/media/libnbaio/include_mono/media/nbaio/SingleStateQueue.h
similarity index 100%
rename from media/libmedia/include/media/SingleStateQueue.h
rename to media/libnbaio/include_mono/media/nbaio/SingleStateQueue.h
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index 44f246d..eacaea8 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -2410,7 +2410,7 @@
}
rate = (OMX_U32)(rateFloat * 65536.0f + 0.5f);
} else {
- if (rateFloat > UINT_MAX) {
+ if (rateFloat > static_cast<float>(UINT_MAX)) {
return BAD_VALUE;
}
rate = (OMX_U32)(rateFloat);
@@ -3316,6 +3316,7 @@
{ MEDIA_MIMETYPE_VIDEO_VP9, OMX_VIDEO_CodingVP9 },
{ MEDIA_MIMETYPE_VIDEO_DOLBY_VISION, OMX_VIDEO_CodingDolbyVision },
{ MEDIA_MIMETYPE_IMAGE_ANDROID_HEIC, OMX_VIDEO_CodingImageHEIC },
+ { MEDIA_MIMETYPE_VIDEO_AV1, OMX_VIDEO_CodingAV1 },
};
static status_t GetVideoCodingTypeFromMime(
diff --git a/media/libstagefright/Android.bp b/media/libstagefright/Android.bp
index 7eab230..18dacb8 100644
--- a/media/libstagefright/Android.bp
+++ b/media/libstagefright/Android.bp
@@ -19,8 +19,10 @@
],
cfi: true,
},
-
- shared_libs: ["libmedia"],
+ shared_libs: [
+ "libstagefright_foundation",
+ "libutils"
+ ],
}
cc_library_static {
@@ -58,10 +60,14 @@
"-Wall",
],
+ header_libs: [
+ "libmediadrm_headers",
+ ],
+
shared_libs: [
"libgui",
"liblog",
- "libmedia_omx",
+ "libmedia_codeclist",
"libstagefright_foundation",
"libui",
"libutils",
@@ -96,6 +102,10 @@
"include",
],
+ header_libs: [
+ "libmedia_helper_headers",
+ ],
+
cflags: [
"-Wno-multichar",
"-Werror",
@@ -121,7 +131,6 @@
"ACodecBufferChannel.cpp",
"AHierarchicalStateMachine.cpp",
"AMRWriter.cpp",
- "AudioPlayer.cpp",
"AudioSource.cpp",
"BufferImpl.cpp",
"CallbackDataSource.cpp",
@@ -129,12 +138,7 @@
"CameraSource.cpp",
"CameraSourceTimeLapse.cpp",
"DataConverter.cpp",
- "DataSourceFactory.cpp",
- "DataURISource.cpp",
- "ClearFileSource.cpp",
- "FileSource.cpp",
"FrameDecoder.cpp",
- "HTTPBase.cpp",
"HevcUtils.cpp",
"InterfaceUtils.cpp",
"JPEGSource.cpp",
@@ -151,10 +155,7 @@
"MediaSource.cpp",
"MediaSync.cpp",
"MediaTrack.cpp",
- "http/ClearMediaHTTP.cpp",
- "http/MediaHTTP.cpp",
"MediaMuxer.cpp",
- "NuCachedSource2.cpp",
"NuMediaExtractor.cpp",
"OggWriter.cpp",
"OMXClient.cpp",
@@ -164,11 +165,10 @@
"SimpleDecodingSource.cpp",
"SkipCutBuffer.cpp",
"StagefrightMediaScanner.cpp",
- "StagefrightMetadataRetriever.cpp",
"StagefrightPluginLoader.cpp",
"SurfaceUtils.cpp",
- "Utils.cpp",
"ThrottledSource.cpp",
+ "Utils.cpp",
"VideoFrameSchedulerBase.cpp",
"VideoFrameScheduler.cpp",
],
@@ -179,12 +179,13 @@
"libbinder",
"libcamera_client",
"libcutils",
+ "libdatasource",
"libdl",
"libdl_android",
- "libdrmframework",
"libgui",
"liblog",
"libmedia",
+ "libmedia_codeclist",
"libmedia_omx",
"libmedia_omx_client",
"libaudioclient",
@@ -206,6 +207,7 @@
],
static_libs: [
+ "libstagefright_esds",
"libstagefright_color_conversion",
"libyuv_static",
"libstagefright_mediafilter",
@@ -213,13 +215,12 @@
"libstagefright_timedtext",
"libogg",
"libwebm",
- "libstagefright_esds",
"libstagefright_id3",
- "libFLAC",
],
header_libs:[
- "libnativeloader-dummy-headers",
+ "libmediadrm_headers",
+ "libnativeloader-headers",
"libstagefright_xmlparser_headers",
"media_ndk_headers",
],
@@ -259,62 +260,3 @@
],
},
}
-
-cc_library_static {
- name: "libstagefright_player2",
-
- srcs: [
- "ClearFileSource.cpp",
- "DataURISource.cpp",
- "HTTPBase.cpp",
- "HevcUtils.cpp",
- "MediaClock.cpp",
- "MediaSource.cpp",
- "NdkUtils.cpp",
- "Utils.cpp",
- "VideoFrameSchedulerBase.cpp",
- "VideoFrameScheduler2.cpp",
- "http/ClearMediaHTTP.cpp",
- ],
-
- shared_libs: [
- "libgui",
- "liblog",
- "libnetd_client",
- "libutils",
- "libstagefright_foundation",
- "libandroid",
- ],
-
- static_libs: [
- "libmedia_player2_util",
- "libmedia2_jni_core",
- ],
-
- export_include_dirs: [
- "include",
- ],
-
- cflags: [
- "-Wno-multichar",
- "-Werror",
- "-Wno-error=deprecated-declarations",
- "-Wall",
- ],
-
- product_variables: {
- debuggable: {
- // enable experiments only in userdebug and eng builds
- cflags: ["-DENABLE_STAGEFRIGHT_EXPERIMENTS"],
- },
- },
-
- sanitize: {
- cfi: true,
- misc_undefined: [
- "unsigned-integer-overflow",
- "signed-integer-overflow",
- ],
- },
-}
-
diff --git a/media/libstagefright/BufferImpl.cpp b/media/libstagefright/BufferImpl.cpp
index b760273..f73b625 100644
--- a/media/libstagefright/BufferImpl.cpp
+++ b/media/libstagefright/BufferImpl.cpp
@@ -21,7 +21,7 @@
#include <binder/IMemory.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/AMessage.h>
-#include <media/ICrypto.h>
+#include <mediadrm/ICrypto.h>
#include <utils/NativeHandle.h>
#include "include/SecureBuffer.h"
diff --git a/media/libstagefright/CallbackDataSource.cpp b/media/libstagefright/CallbackDataSource.cpp
index 92e6eb9..265f21b 100644
--- a/media/libstagefright/CallbackDataSource.cpp
+++ b/media/libstagefright/CallbackDataSource.cpp
@@ -113,10 +113,6 @@
}
}
-sp<DecryptHandle> CallbackDataSource::DrmInitialization(const char *mime) {
- return mIDataSource->DrmInitialization(mime);
-}
-
sp<IDataSource> CallbackDataSource::getIDataSource() const {
return mIDataSource;
}
@@ -190,14 +186,6 @@
return mSource->flags();
}
-sp<DecryptHandle> TinyCacheSource::DrmInitialization(const char *mime) {
- // flush cache when DrmInitialization occurs since decrypted
- // data may differ from what is in cache.
- mCachedOffset = 0;
- mCachedSize = 0;
- return mSource->DrmInitialization(mime);
-}
-
sp<IDataSource> TinyCacheSource::getIDataSource() const {
return mSource->getIDataSource();
}
diff --git a/media/libstagefright/CodecBase.cpp b/media/libstagefright/CodecBase.cpp
index d0610b2..97f38f8 100644
--- a/media/libstagefright/CodecBase.cpp
+++ b/media/libstagefright/CodecBase.cpp
@@ -18,7 +18,7 @@
#define LOG_TAG "CodecBase"
#include <android/hardware/cas/native/1.0/IDescrambler.h>
-#include <media/ICrypto.h>
+#include <mediadrm/ICrypto.h>
#include <media/stagefright/CodecBase.h>
#include <utils/Log.h>
diff --git a/media/libstagefright/FrameDecoder.cpp b/media/libstagefright/FrameDecoder.cpp
index 18a6bd8..08f690b 100644
--- a/media/libstagefright/FrameDecoder.cpp
+++ b/media/libstagefright/FrameDecoder.cpp
@@ -22,7 +22,7 @@
#include <binder/MemoryHeapBase.h>
#include <gui/Surface.h>
#include <inttypes.h>
-#include <media/ICrypto.h>
+#include <mediadrm/ICrypto.h>
#include <media/IMediaSource.h>
#include <media/MediaCodecBuffer.h>
#include <media/stagefright/foundation/avc_utils.h>
@@ -725,12 +725,6 @@
}
converter.setSrcColorSpace(standard, range, transfer);
- int32_t dstLeft, dstTop, dstRight, dstBottom;
- dstLeft = mTilesDecoded % mGridCols * width;
- dstTop = mTilesDecoded / mGridCols * height;
- dstRight = dstLeft + width - 1;
- dstBottom = dstTop + height - 1;
-
int32_t crop_left, crop_top, crop_right, crop_bottom;
if (!outputFormat->findRect("crop", &crop_left, &crop_top, &crop_right, &crop_bottom)) {
crop_left = crop_top = 0;
@@ -738,15 +732,25 @@
crop_bottom = height - 1;
}
+ int32_t crop_width, crop_height;
+ crop_width = crop_right - crop_left + 1;
+ crop_height = crop_bottom - crop_top + 1;
+
+ int32_t dstLeft, dstTop, dstRight, dstBottom;
+ dstLeft = mTilesDecoded % mGridCols * crop_width;
+ dstTop = mTilesDecoded / mGridCols * crop_height;
+ dstRight = dstLeft + crop_width - 1;
+ dstBottom = dstTop + crop_height - 1;
+
// apply crop on bottom-right
// TODO: need to move this into the color converter itself.
if (dstRight >= mWidth) {
- crop_right = mWidth - dstLeft - 1;
- dstRight = dstLeft + crop_right;
+ crop_right = crop_left + mWidth - dstLeft - 1;
+ dstRight = mWidth - 1;
}
if (dstBottom >= mHeight) {
- crop_bottom = mHeight - dstTop - 1;
- dstBottom = dstTop + crop_bottom;
+ crop_bottom = crop_top + mHeight - dstTop - 1;
+ dstBottom = mHeight - 1;
}
*done = (++mTilesDecoded >= mTargetTiles);
diff --git a/media/libstagefright/MPEG4Writer.cpp b/media/libstagefright/MPEG4Writer.cpp
index f130c9b..258bed8 100644
--- a/media/libstagefright/MPEG4Writer.cpp
+++ b/media/libstagefright/MPEG4Writer.cpp
@@ -613,8 +613,9 @@
CHECK(source.get() != NULL);
- const char *mime;
- source->getFormat()->findCString(kKeyMIMEType, &mime);
+ const char *mime = NULL;
+ sp<MetaData> meta = source->getFormat();
+ meta->findCString(kKeyMIMEType, &mime);
if (Track::getFourCCForMime(mime) == NULL) {
ALOGE("Unsupported mime '%s'", mime);
diff --git a/media/libstagefright/MediaClock.cpp b/media/libstagefright/MediaClock.cpp
index 4f9bc6d..24608a7 100644
--- a/media/libstagefright/MediaClock.cpp
+++ b/media/libstagefright/MediaClock.cpp
@@ -281,7 +281,7 @@
it = mTimers.erase(it);
} else {
if (mPlaybackRate != 0.0
- && (double)diffMediaUs < INT64_MAX * (double)mPlaybackRate) {
+ && (double)diffMediaUs < (double)INT64_MAX * (double)mPlaybackRate) {
int64_t targetRealUs = diffMediaUs / (double)mPlaybackRate;
if (targetRealUs < nextLapseRealUs) {
nextLapseRealUs = targetRealUs;
diff --git a/media/libstagefright/MediaCodec.cpp b/media/libstagefright/MediaCodec.cpp
index eceb84e..77eace9 100644
--- a/media/libstagefright/MediaCodec.cpp
+++ b/media/libstagefright/MediaCodec.cpp
@@ -35,7 +35,7 @@
#include <cutils/properties.h>
#include <gui/BufferQueue.h>
#include <gui/Surface.h>
-#include <media/ICrypto.h>
+#include <mediadrm/ICrypto.h>
#include <media/IOMX.h>
#include <media/IResourceManagerService.h>
#include <media/MediaCodecBuffer.h>
@@ -527,7 +527,7 @@
mFlags(0),
mStickyError(OK),
mSoftRenderer(NULL),
- mAnalyticsItem(NULL),
+ mMetricsHandle(0),
mIsVideo(false),
mVideoWidth(0),
mVideoHeight(0),
@@ -548,19 +548,19 @@
mResourceManagerClient = new ResourceManagerClient(this);
mResourceManagerService = new ResourceManagerServiceProxy(pid, mUid);
- initAnalyticsItem();
+ initMediametrics();
}
MediaCodec::~MediaCodec() {
CHECK_EQ(mState, UNINITIALIZED);
mResourceManagerService->removeClient(getId(mResourceManagerClient));
- flushAnalyticsItem();
+ flushMediametrics();
}
-void MediaCodec::initAnalyticsItem() {
- if (mAnalyticsItem == NULL) {
- mAnalyticsItem = MediaAnalyticsItem::create(kCodecKeyName);
+void MediaCodec::initMediametrics() {
+ if (mMetricsHandle == 0) {
+ mMetricsHandle = mediametrics_create(kCodecKeyName);
}
mLatencyHist.setup(kLatencyHistBuckets, kLatencyHistWidth, kLatencyHistFloor);
@@ -574,38 +574,39 @@
}
}
-void MediaCodec::updateAnalyticsItem() {
- ALOGV("MediaCodec::updateAnalyticsItem");
- if (mAnalyticsItem == NULL) {
+void MediaCodec::updateMediametrics() {
+ ALOGV("MediaCodec::updateMediametrics");
+ if (mMetricsHandle == 0) {
return;
}
+
if (mLatencyHist.getCount() != 0 ) {
- mAnalyticsItem->setInt64(kCodecLatencyMax, mLatencyHist.getMax());
- mAnalyticsItem->setInt64(kCodecLatencyMin, mLatencyHist.getMin());
- mAnalyticsItem->setInt64(kCodecLatencyAvg, mLatencyHist.getAvg());
- mAnalyticsItem->setInt64(kCodecLatencyCount, mLatencyHist.getCount());
+ mediametrics_setInt64(mMetricsHandle, kCodecLatencyMax, mLatencyHist.getMax());
+ mediametrics_setInt64(mMetricsHandle, kCodecLatencyMin, mLatencyHist.getMin());
+ mediametrics_setInt64(mMetricsHandle, kCodecLatencyAvg, mLatencyHist.getAvg());
+ mediametrics_setInt64(mMetricsHandle, kCodecLatencyCount, mLatencyHist.getCount());
if (kEmitHistogram) {
// and the histogram itself
std::string hist = mLatencyHist.emit();
- mAnalyticsItem->setCString(kCodecLatencyHist, hist.c_str());
+ mediametrics_setCString(mMetricsHandle, kCodecLatencyHist, hist.c_str());
}
}
if (mLatencyUnknown > 0) {
- mAnalyticsItem->setInt64(kCodecLatencyUnknown, mLatencyUnknown);
+ mediametrics_setInt64(mMetricsHandle, kCodecLatencyUnknown, mLatencyUnknown);
}
#if 0
// enable for short term, only while debugging
- updateEphemeralAnalytics(mAnalyticsItem);
+ updateEphemeralMediametrics(mMetricsHandle);
#endif
}
-void MediaCodec::updateEphemeralAnalytics(MediaAnalyticsItem *item) {
- ALOGD("MediaCodec::updateEphemeralAnalytics()");
+void MediaCodec::updateEphemeralMediametrics(mediametrics_handle_t item) {
+ ALOGD("MediaCodec::updateEphemeralMediametrics()");
- if (item == NULL) {
+ if (item == 0) {
return;
}
@@ -628,28 +629,27 @@
// spit the data (if any) into the supplied analytics record
if (recentHist.getCount()!= 0 ) {
- item->setInt64(kCodecRecentLatencyMax, recentHist.getMax());
- item->setInt64(kCodecRecentLatencyMin, recentHist.getMin());
- item->setInt64(kCodecRecentLatencyAvg, recentHist.getAvg());
- item->setInt64(kCodecRecentLatencyCount, recentHist.getCount());
+ mediametrics_setInt64(item, kCodecRecentLatencyMax, recentHist.getMax());
+ mediametrics_setInt64(item, kCodecRecentLatencyMin, recentHist.getMin());
+ mediametrics_setInt64(item, kCodecRecentLatencyAvg, recentHist.getAvg());
+ mediametrics_setInt64(item, kCodecRecentLatencyCount, recentHist.getCount());
if (kEmitHistogram) {
// and the histogram itself
std::string hist = recentHist.emit();
- item->setCString(kCodecRecentLatencyHist, hist.c_str());
+ mediametrics_setCString(item, kCodecRecentLatencyHist, hist.c_str());
}
}
}
-void MediaCodec::flushAnalyticsItem() {
- updateAnalyticsItem();
- if (mAnalyticsItem != NULL) {
- // don't log empty records
- if (mAnalyticsItem->count() > 0) {
- mAnalyticsItem->selfrecord();
+void MediaCodec::flushMediametrics() {
+ updateMediametrics();
+ if (mMetricsHandle != 0) {
+ if (mediametrics_count(mMetricsHandle) > 0) {
+ mediametrics_selfRecord(mMetricsHandle);
}
- delete mAnalyticsItem;
- mAnalyticsItem = NULL;
+ mediametrics_delete(mMetricsHandle);
+ mMetricsHandle = 0;
}
}
@@ -981,9 +981,10 @@
// ".secure"
msg->setString("name", name);
- if (mAnalyticsItem != NULL) {
- mAnalyticsItem->setCString(kCodecCodec, name.c_str());
- mAnalyticsItem->setCString(kCodecMode, mIsVideo ? kCodecModeVideo : kCodecModeAudio);
+ if (mMetricsHandle != 0) {
+ mediametrics_setCString(mMetricsHandle, kCodecCodec, name.c_str());
+ mediametrics_setCString(mMetricsHandle, kCodecMode,
+ mIsVideo ? kCodecModeVideo : kCodecModeAudio);
}
if (mIsVideo) {
@@ -1044,16 +1045,17 @@
uint32_t flags) {
sp<AMessage> msg = new AMessage(kWhatConfigure, this);
- if (mAnalyticsItem != NULL) {
+ if (mMetricsHandle != 0) {
int32_t profile = 0;
if (format->findInt32("profile", &profile)) {
- mAnalyticsItem->setInt32(kCodecProfile, profile);
+ mediametrics_setInt32(mMetricsHandle, kCodecProfile, profile);
}
int32_t level = 0;
if (format->findInt32("level", &level)) {
- mAnalyticsItem->setInt32(kCodecLevel, level);
+ mediametrics_setInt32(mMetricsHandle, kCodecLevel, level);
}
- mAnalyticsItem->setInt32(kCodecEncoder, (flags & CONFIGURE_FLAG_ENCODE) ? 1 : 0);
+ mediametrics_setInt32(mMetricsHandle, kCodecEncoder,
+ (flags & CONFIGURE_FLAG_ENCODE) ? 1 : 0);
}
if (mIsVideo) {
@@ -1063,17 +1065,17 @@
mRotationDegrees = 0;
}
- if (mAnalyticsItem != NULL) {
- mAnalyticsItem->setInt32(kCodecWidth, mVideoWidth);
- mAnalyticsItem->setInt32(kCodecHeight, mVideoHeight);
- mAnalyticsItem->setInt32(kCodecRotation, mRotationDegrees);
+ if (mMetricsHandle != 0) {
+ mediametrics_setInt32(mMetricsHandle, kCodecWidth, mVideoWidth);
+ mediametrics_setInt32(mMetricsHandle, kCodecHeight, mVideoHeight);
+ mediametrics_setInt32(mMetricsHandle, kCodecRotation, mRotationDegrees);
int32_t maxWidth = 0;
if (format->findInt32("max-width", &maxWidth)) {
- mAnalyticsItem->setInt32(kCodecMaxWidth, maxWidth);
+ mediametrics_setInt32(mMetricsHandle, kCodecMaxWidth, maxWidth);
}
int32_t maxHeight = 0;
if (format->findInt32("max-height", &maxHeight)) {
- mAnalyticsItem->setInt32(kCodecMaxHeight, maxHeight);
+ mediametrics_setInt32(mMetricsHandle, kCodecMaxHeight, maxHeight);
}
}
@@ -1095,8 +1097,8 @@
} else {
msg->setPointer("descrambler", descrambler.get());
}
- if (mAnalyticsItem != NULL) {
- mAnalyticsItem->setInt32(kCodecCrypto, 1);
+ if (mMetricsHandle != 0) {
+ mediametrics_setInt32(mMetricsHandle, kCodecCrypto, 1);
}
} else if (mFlags & kFlagIsSecure) {
ALOGW("Crypto or descrambler should be given for secure codec");
@@ -1561,22 +1563,22 @@
return OK;
}
-status_t MediaCodec::getMetrics(MediaAnalyticsItem * &reply) {
+status_t MediaCodec::getMetrics(mediametrics_handle_t &reply) {
- reply = NULL;
+ reply = 0;
// shouldn't happen, but be safe
- if (mAnalyticsItem == NULL) {
+ if (mMetricsHandle == 0) {
return UNKNOWN_ERROR;
}
// update any in-flight data that's not carried within the record
- updateAnalyticsItem();
+ updateMediametrics();
// send it back to the caller.
- reply = mAnalyticsItem->dup();
+ reply = mediametrics_dup(mMetricsHandle);
- updateEphemeralAnalytics(reply);
+ updateEphemeralMediametrics(reply);
return OK;
}
@@ -1890,10 +1892,11 @@
case CONFIGURING:
{
if (actionCode == ACTION_CODE_FATAL) {
- mAnalyticsItem->setInt32(kCodecError, err);
- mAnalyticsItem->setCString(kCodecErrorState, stateString(mState).c_str());
- flushAnalyticsItem();
- initAnalyticsItem();
+ mediametrics_setInt32(mMetricsHandle, kCodecError, err);
+ mediametrics_setCString(mMetricsHandle, kCodecErrorState,
+ stateString(mState).c_str());
+ flushMediametrics();
+ initMediametrics();
}
setState(actionCode == ACTION_CODE_FATAL ?
UNINITIALIZED : INITIALIZED);
@@ -1903,10 +1906,11 @@
case STARTING:
{
if (actionCode == ACTION_CODE_FATAL) {
- mAnalyticsItem->setInt32(kCodecError, err);
- mAnalyticsItem->setCString(kCodecErrorState, stateString(mState).c_str());
- flushAnalyticsItem();
- initAnalyticsItem();
+ mediametrics_setInt32(mMetricsHandle, kCodecError, err);
+ mediametrics_setCString(mMetricsHandle, kCodecErrorState,
+ stateString(mState).c_str());
+ flushMediametrics();
+ initMediametrics();
}
setState(actionCode == ACTION_CODE_FATAL ?
UNINITIALIZED : CONFIGURED);
@@ -1944,10 +1948,11 @@
case FLUSHING:
{
if (actionCode == ACTION_CODE_FATAL) {
- mAnalyticsItem->setInt32(kCodecError, err);
- mAnalyticsItem->setCString(kCodecErrorState, stateString(mState).c_str());
- flushAnalyticsItem();
- initAnalyticsItem();
+ mediametrics_setInt32(mMetricsHandle, kCodecError, err);
+ mediametrics_setCString(mMetricsHandle, kCodecErrorState,
+ stateString(mState).c_str());
+ flushMediametrics();
+ initMediametrics();
setState(UNINITIALIZED);
} else {
@@ -1977,10 +1982,11 @@
setState(INITIALIZED);
break;
default:
- mAnalyticsItem->setInt32(kCodecError, err);
- mAnalyticsItem->setCString(kCodecErrorState, stateString(mState).c_str());
- flushAnalyticsItem();
- initAnalyticsItem();
+ mediametrics_setInt32(mMetricsHandle, kCodecError, err);
+ mediametrics_setCString(mMetricsHandle, kCodecErrorState,
+ stateString(mState).c_str());
+ flushMediametrics();
+ initMediametrics();
setState(UNINITIALIZED);
break;
}
@@ -2037,7 +2043,8 @@
CHECK(msg->findString("componentName", &mComponentName));
if (mComponentName.c_str()) {
- mAnalyticsItem->setCString(kCodecCodec, mComponentName.c_str());
+ mediametrics_setCString(mMetricsHandle, kCodecCodec,
+ mComponentName.c_str());
}
const char *owner = mCodecInfo->getOwnerName();
@@ -2053,11 +2060,11 @@
if (mComponentName.endsWith(".secure")) {
mFlags |= kFlagIsSecure;
resourceType = MediaResource::kSecureCodec;
- mAnalyticsItem->setInt32(kCodecSecure, 1);
+ mediametrics_setInt32(mMetricsHandle, kCodecSecure, 1);
} else {
mFlags &= ~kFlagIsSecure;
resourceType = MediaResource::kNonSecureCodec;
- mAnalyticsItem->setInt32(kCodecSecure, 0);
+ mediametrics_setInt32(mMetricsHandle, kCodecSecure, 0);
}
if (mIsVideo) {
@@ -2105,14 +2112,15 @@
(new AMessage)->postReply(mReplyID);
// augment our media metrics info, now that we know more things
- if (mAnalyticsItem != NULL) {
+ if (mMetricsHandle != 0) {
sp<AMessage> format;
if (mConfigureMsg != NULL &&
mConfigureMsg->findMessage("format", &format)) {
// format includes: mime
AString mime;
if (format->findString("mime", &mime)) {
- mAnalyticsItem->setCString(kCodecMime, mime.c_str());
+ mediametrics_setCString(mMetricsHandle, kCodecMime,
+ mime.c_str());
}
}
}
diff --git a/media/libstagefright/MediaCodecList.cpp b/media/libstagefright/MediaCodecList.cpp
index 3d58d4b..a267f7e 100644
--- a/media/libstagefright/MediaCodecList.cpp
+++ b/media/libstagefright/MediaCodecList.cpp
@@ -170,6 +170,7 @@
sp<IMediaCodecList> MediaCodecList::sRemoteList;
sp<MediaCodecList::BinderDeathObserver> MediaCodecList::sBinderDeathObserver;
+sp<IBinder> MediaCodecList::sMediaPlayer; // kept since linked to death
void MediaCodecList::BinderDeathObserver::binderDied(const wp<IBinder> &who __unused) {
Mutex::Autolock _l(sRemoteInitMutex);
@@ -181,15 +182,14 @@
sp<IMediaCodecList> MediaCodecList::getInstance() {
Mutex::Autolock _l(sRemoteInitMutex);
if (sRemoteList == nullptr) {
- sp<IBinder> binder =
- defaultServiceManager()->getService(String16("media.player"));
+ sMediaPlayer = defaultServiceManager()->getService(String16("media.player"));
sp<IMediaPlayerService> service =
- interface_cast<IMediaPlayerService>(binder);
+ interface_cast<IMediaPlayerService>(sMediaPlayer);
if (service.get() != nullptr) {
sRemoteList = service->getCodecList();
if (sRemoteList != nullptr) {
sBinderDeathObserver = new BinderDeathObserver();
- binder->linkToDeath(sBinderDeathObserver.get());
+ sMediaPlayer->linkToDeath(sBinderDeathObserver.get());
}
}
if (sRemoteList == nullptr) {
diff --git a/media/libstagefright/MediaCodecListOverrides.cpp b/media/libstagefright/MediaCodecListOverrides.cpp
index dd7c3e6..b027a97 100644
--- a/media/libstagefright/MediaCodecListOverrides.cpp
+++ b/media/libstagefright/MediaCodecListOverrides.cpp
@@ -22,7 +22,7 @@
#include <cutils/properties.h>
#include <gui/Surface.h>
-#include <media/ICrypto.h>
+#include <mediadrm/ICrypto.h>
#include <media/IMediaCodecList.h>
#include <media/MediaCodecInfo.h>
#include <media/MediaResourcePolicy.h>
diff --git a/media/libstagefright/MediaCodecSource.cpp b/media/libstagefright/MediaCodecSource.cpp
index 50e454c..7243b82 100644
--- a/media/libstagefright/MediaCodecSource.cpp
+++ b/media/libstagefright/MediaCodecSource.cpp
@@ -22,7 +22,7 @@
#include <gui/IGraphicBufferProducer.h>
#include <gui/Surface.h>
-#include <media/ICrypto.h>
+#include <mediadrm/ICrypto.h>
#include <media/MediaBufferHolder.h>
#include <media/MediaCodecBuffer.h>
#include <media/MediaSource.h>
diff --git a/media/libstagefright/MediaExtractorFactory.cpp b/media/libstagefright/MediaExtractorFactory.cpp
index 4c8be1f..9e85475 100644
--- a/media/libstagefright/MediaExtractorFactory.cpp
+++ b/media/libstagefright/MediaExtractorFactory.cpp
@@ -71,9 +71,6 @@
ALOGV("MediaExtractorFactory::CreateFromService %s", mime);
- // initialize source decryption if needed
- source->DrmInitialization(nullptr /* mime */);
-
void *meta = nullptr;
void *creator = NULL;
FreeMetaFunc freeMeta = nullptr;
@@ -279,7 +276,7 @@
std::shared_ptr<std::list<sp<ExtractorPlugin>>> newList(new std::list<sp<ExtractorPlugin>>());
- android_namespace_t *mediaNs = android_get_exported_namespace("media");
+ android_namespace_t *mediaNs = android_get_exported_namespace("com_android_media");
if (mediaNs != NULL) {
const android_dlextinfo dlextinfo = {
.flags = ANDROID_DLEXT_USE_NAMESPACE,
diff --git a/media/libstagefright/NdkUtils.cpp b/media/libstagefright/NdkUtils.cpp
deleted file mode 100644
index 904fe72..0000000
--- a/media/libstagefright/NdkUtils.cpp
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- * Copyright (C) 2018 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-
-#include <media/stagefright/NdkUtils.h>
-#include <media/stagefright/Utils.h>
-#include <media/stagefright/foundation/AMessage.h>
-
-namespace android {
-
-sp<MetaData> convertMediaFormatWrapperToMetaData(const sp<AMediaFormatWrapper> &fmt) {
- sp<AMessage> msg = fmt->toAMessage();
- sp<MetaData> meta = new MetaData;
- convertMessageToMetaData(msg, meta);
- return meta;
-}
-
-} // namespace android
-
diff --git a/media/libstagefright/NuMediaExtractor.cpp b/media/libstagefright/NuMediaExtractor.cpp
index 680d426..66fb4b0 100644
--- a/media/libstagefright/NuMediaExtractor.cpp
+++ b/media/libstagefright/NuMediaExtractor.cpp
@@ -22,13 +22,13 @@
#include "include/ESDS.h"
+#include <datasource/DataSourceFactory.h>
+#include <datasource/FileSource.h>
#include <media/DataSource.h>
#include <media/MediaSource.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/AMessage.h>
-#include <media/stagefright/DataSourceFactory.h>
-#include <media/stagefright/FileSource.h>
#include <media/stagefright/MediaBuffer.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
@@ -36,6 +36,7 @@
#include <media/stagefright/MediaExtractorFactory.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/Utils.h>
+#include <media/stagefright/FoundationUtils.h>
namespace android {
@@ -81,7 +82,7 @@
}
sp<DataSource> dataSource =
- DataSourceFactory::CreateFromURI(httpService, path, headers);
+ DataSourceFactory::getInstance()->CreateFromURI(httpService, path, headers);
if (dataSource == NULL) {
return -ENOENT;
diff --git a/media/libstagefright/SimpleDecodingSource.cpp b/media/libstagefright/SimpleDecodingSource.cpp
index babdc7a..b809848 100644
--- a/media/libstagefright/SimpleDecodingSource.cpp
+++ b/media/libstagefright/SimpleDecodingSource.cpp
@@ -20,7 +20,7 @@
#include <gui/Surface.h>
-#include <media/ICrypto.h>
+#include <mediadrm/ICrypto.h>
#include <media/MediaCodecBuffer.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/foundation/ALooper.h>
diff --git a/media/libstagefright/Utils.cpp b/media/libstagefright/Utils.cpp
index 135151f..02e8ab5 100644
--- a/media/libstagefright/Utils.cpp
+++ b/media/libstagefright/Utils.cpp
@@ -689,6 +689,7 @@
{ "temporal-layer-id", kKeyTemporalLayerId },
{ "thumbnail-width", kKeyThumbnailWidth },
{ "thumbnail-height", kKeyThumbnailHeight },
+ { "track-id", kKeyTrackID },
{ "valid-samples", kKeyValidSamples },
}
};
@@ -896,12 +897,6 @@
msg->setInt32("is-sync-frame", 1);
}
- // this only needs to be translated from meta to message as it is an extractor key
- int32_t trackID;
- if (meta->findInt32(kKeyTrackID, &trackID)) {
- msg->setInt32("track-id", trackID);
- }
-
const char *lang;
if (meta->findCString(kKeyMediaLanguage, &lang)) {
msg->setString("language", lang);
@@ -1806,7 +1801,7 @@
if (msg->findInt32("frame-rate", &fps) && fps > 0) {
meta->setInt32(kKeyFrameRate, fps);
} else if (msg->findFloat("frame-rate", &fpsFloat)
- && fpsFloat >= 1 && fpsFloat <= INT32_MAX) {
+ && fpsFloat >= 1 && static_cast<int32_t>(fpsFloat) <= INT32_MAX) {
// truncate values to distinguish between e.g. 24 vs 23.976 fps
meta->setInt32(kKeyFrameRate, (int32_t)fpsFloat);
}
@@ -1895,22 +1890,6 @@
#endif
}
-AString MakeUserAgent() {
- AString ua;
- ua.append("stagefright/1.2 (Linux;Android ");
-
-#if (PROPERTY_VALUE_MAX < 8)
-#error "PROPERTY_VALUE_MAX must be at least 8"
-#endif
-
- char value[PROPERTY_VALUE_MAX];
- property_get("ro.build.version.release", value, "Unknown");
- ua.append(value);
- ua.append(")");
-
- return ua;
-}
-
status_t sendMetaDataToHal(sp<MediaPlayerBase::AudioSink>& sink,
const sp<MetaData>& meta)
{
@@ -2099,39 +2078,6 @@
return AudioSystem::isOffloadSupported(info);
}
-AString uriDebugString(const AString &uri, bool incognito) {
- if (incognito) {
- return AString("<URI suppressed>");
- }
-
- if (property_get_bool("media.stagefright.log-uri", false)) {
- return uri;
- }
-
- // find scheme
- AString scheme;
- const char *chars = uri.c_str();
- for (size_t i = 0; i < uri.size(); i++) {
- const char c = chars[i];
- if (!isascii(c)) {
- break;
- } else if (isalpha(c)) {
- continue;
- } else if (i == 0) {
- // first character must be a letter
- break;
- } else if (isdigit(c) || c == '+' || c == '.' || c =='-') {
- continue;
- } else if (c != ':') {
- break;
- }
- scheme = AString(uri, 0, i);
- scheme.append("://<suppressed>");
- return scheme;
- }
- return AString("<no-scheme URI suppressed>");
-}
-
HLSTime::HLSTime(const sp<AMessage>& meta) :
mSeq(-1),
mTimeUs(-1LL),
@@ -2230,36 +2176,4 @@
}
}
-AString nameForFd(int fd) {
- const size_t SIZE = 256;
- char buffer[SIZE];
- AString result;
- snprintf(buffer, SIZE, "/proc/%d/fd/%d", getpid(), fd);
- struct stat s;
- if (lstat(buffer, &s) == 0) {
- if ((s.st_mode & S_IFMT) == S_IFLNK) {
- char linkto[256];
- int len = readlink(buffer, linkto, sizeof(linkto));
- if(len > 0) {
- if(len > 255) {
- linkto[252] = '.';
- linkto[253] = '.';
- linkto[254] = '.';
- linkto[255] = 0;
- } else {
- linkto[len] = 0;
- }
- result.append(linkto);
- }
- } else {
- result.append("unexpected type for ");
- result.append(buffer);
- }
- } else {
- result.append("couldn't open ");
- result.append(buffer);
- }
- return result;
-}
-
} // namespace android
diff --git a/media/libstagefright/VideoFrameScheduler2.cpp b/media/libstagefright/VideoFrameScheduler2.cpp
deleted file mode 100644
index 23671f2..0000000
--- a/media/libstagefright/VideoFrameScheduler2.cpp
+++ /dev/null
@@ -1,305 +0,0 @@
-/*
- * Copyright (C) 2018 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "VideoFrameScheduler2"
-#include <utils/Log.h>
-#define ATRACE_TAG ATRACE_TAG_VIDEO
-#include <utils/Mutex.h>
-#include <utils/Thread.h>
-#include <utils/Trace.h>
-
-#include <algorithm>
-#include <jni.h>
-#include <math.h>
-
-#include <android/choreographer.h>
-#include <android/looper.h>
-#include <media/stagefright/VideoFrameScheduler2.h>
-#include <mediaplayer2/JavaVMHelper.h>
-
-#include <media/stagefright/foundation/ADebug.h>
-#include <media/stagefright/foundation/AUtils.h>
-
-namespace android {
-
-static void getVsyncOffset(nsecs_t* appVsyncOffsetPtr, nsecs_t* sfVsyncOffsetPtr);
-
-/* ======================================================================= */
-/* VsyncTracker */
-/* ======================================================================= */
-
-class VsyncTracker : public RefBase{
-public:
- VsyncTracker();
- ~VsyncTracker() {}
- nsecs_t getVsyncPeriod();
- nsecs_t getVsyncTime(nsecs_t periodOffset);
- void addSample(nsecs_t timestamp);
-
-private:
- static const int kMaxSamples = 32;
- static const int kMinSamplesForUpdate = 6;
- int mNumSamples;
- int mFirstSample;
- nsecs_t mReferenceTime;
- nsecs_t mPhase;
- nsecs_t mPeriod;
- nsecs_t mTimestampSamples[kMaxSamples];
- Mutex mLock;
-
- void updateModelLocked();
-};
-
-VsyncTracker::VsyncTracker()
- : mNumSamples(0),
- mFirstSample(0),
- mReferenceTime(0),
- mPhase(0),
- mPeriod(0) {
- for (int i = 0; i < kMaxSamples; i++) {
- mTimestampSamples[i] = 0;
- }
-}
-
-nsecs_t VsyncTracker::getVsyncPeriod() {
- Mutex::Autolock dataLock(mLock);
- return mPeriod;
-}
-
-nsecs_t VsyncTracker::getVsyncTime(nsecs_t periodOffset) {
- Mutex::Autolock dataLock(mLock);
- const nsecs_t now = systemTime();
- nsecs_t phase = mReferenceTime + mPhase;
-
- // result = (((now - phase) / mPeriod) + periodOffset + 1) * mPeriod + phase
- // prevent overflow
- nsecs_t result = (now - phase) / mPeriod;
- if (result > LONG_LONG_MAX - periodOffset - 1) {
- return LONG_LONG_MAX;
- } else {
- result += periodOffset + 1;
- }
- if (result > LONG_LONG_MAX / mPeriod) {
- return LONG_LONG_MAX;
- } else {
- result *= mPeriod;
- }
- if (result > LONG_LONG_MAX - phase) {
- return LONG_LONG_MAX;
- } else {
- result += phase;
- }
-
- return result;
-}
-
-void VsyncTracker::addSample(nsecs_t timestamp) {
- Mutex::Autolock dataLock(mLock);
- if (mNumSamples == 0) {
- mPhase = 0;
- mReferenceTime = timestamp;
- }
- int idx = (mFirstSample + mNumSamples) % kMaxSamples;
- mTimestampSamples[idx] = timestamp;
- if (mNumSamples < kMaxSamples) {
- mNumSamples++;
- } else {
- mFirstSample = (mFirstSample + 1) % kMaxSamples;
- }
- updateModelLocked();
-}
-
-void VsyncTracker::updateModelLocked() {
- if (mNumSamples < kMinSamplesForUpdate) {
- return;
- }
- nsecs_t durationSum = 0;
- nsecs_t minDuration = LONG_MAX;
- nsecs_t maxDuration = 0;
-
- for (int i = 1; i < mNumSamples; i++) {
- int idx = (mFirstSample + i) % kMaxSamples;
- int prev = (idx + kMaxSamples - 1) % kMaxSamples;
- long duration = mTimestampSamples[idx] - mTimestampSamples[prev];
- durationSum += duration;
- if (minDuration > duration) { minDuration = duration; }
- if (maxDuration < duration) { maxDuration = duration; }
- }
-
- durationSum -= (minDuration + maxDuration);
- mPeriod = durationSum / (mNumSamples - 3);
-
- double sampleAvgX = 0.0;
- double sampleAvgY = 0.0;
- double scale = 2.0 * M_PI / (double) mPeriod;
-
- for (int i = 1; i < mNumSamples; i++) {
- int idx = (mFirstSample + i) % kMaxSamples;
- long sample = mTimestampSamples[idx] - mReferenceTime;
- double samplePhase = (double) (sample % mPeriod) * scale;
- sampleAvgX += cos(samplePhase);
- sampleAvgY += sin(samplePhase);
- }
-
- sampleAvgX /= (double) mNumSamples - 1.0;
- sampleAvgY /= (double) mNumSamples - 1.0;
- mPhase = (long) (atan2(sampleAvgY, sampleAvgX) / scale);
-}
-
-static void frameCallback(int64_t frameTimeNanos, void* data) {
- if (data == NULL) {
- return;
- }
- sp<VsyncTracker> vsyncTracker(static_cast<VsyncTracker*>(data));
- vsyncTracker->addSample(frameTimeNanos);
- AChoreographer_postFrameCallback64(AChoreographer_getInstance(),
- frameCallback, static_cast<void*>(vsyncTracker.get()));
-}
-
-/* ======================================================================= */
-/* JNI */
-/* ======================================================================= */
-
-static void getVsyncOffset(nsecs_t* appVsyncOffsetPtr, nsecs_t* sfVsyncOffsetPtr) {
- static const nsecs_t kOneMillisecInNanosec = 1000000;
- static const nsecs_t kOneSecInNanosec = kOneMillisecInNanosec * 1000;
-
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jclass jDisplayManagerGlobalCls = env->FindClass(
- "android/hardware/display/DisplayManagerGlobal");
- jclass jDisplayCls = env->FindClass("android/view/Display");
-
- jmethodID jGetInstance = env->GetStaticMethodID(jDisplayManagerGlobalCls,
- "getInstance", "()Landroid/hardware/display/DisplayManagerGlobal;");
- jobject javaDisplayManagerGlobalObj = env->CallStaticObjectMethod(
- jDisplayManagerGlobalCls, jGetInstance);
-
- jfieldID jDEFAULT_DISPLAY = env->GetStaticFieldID(jDisplayCls, "DEFAULT_DISPLAY", "I");
- jint DEFAULT_DISPLAY = env->GetStaticIntField(jDisplayCls, jDEFAULT_DISPLAY);
-
- jmethodID jgetRealDisplay = env->GetMethodID(jDisplayManagerGlobalCls,
- "getRealDisplay", "(I)Landroid/view/Display;");
- jobject javaDisplayObj = env->CallObjectMethod(
- javaDisplayManagerGlobalObj, jgetRealDisplay, DEFAULT_DISPLAY);
-
- jmethodID jGetRefreshRate = env->GetMethodID(jDisplayCls, "getRefreshRate", "()F");
- jfloat javaRefreshRate = env->CallFloatMethod(javaDisplayObj, jGetRefreshRate);
- nsecs_t vsyncPeriod = (nsecs_t) (kOneSecInNanosec / (float) javaRefreshRate);
-
- jmethodID jGetAppVsyncOffsetNanos = env->GetMethodID(
- jDisplayCls, "getAppVsyncOffsetNanos", "()J");
- jlong javaAppVsyncOffset = env->CallLongMethod(javaDisplayObj, jGetAppVsyncOffsetNanos);
- *appVsyncOffsetPtr = (nsecs_t) javaAppVsyncOffset;
-
- jmethodID jGetPresentationDeadlineNanos = env->GetMethodID(
- jDisplayCls, "getPresentationDeadlineNanos", "()J");
- jlong javaPresentationDeadline = env->CallLongMethod(
- javaDisplayObj, jGetPresentationDeadlineNanos);
-
- *sfVsyncOffsetPtr = vsyncPeriod - ((nsecs_t) javaPresentationDeadline - kOneMillisecInNanosec);
-}
-
-/* ======================================================================= */
-/* Choreographer Thread */
-/* ======================================================================= */
-
-struct ChoreographerThread : public Thread {
- ChoreographerThread(bool canCallJava);
- status_t init(void* data);
- virtual status_t readyToRun() override;
- virtual bool threadLoop() override;
-
-protected:
- virtual ~ChoreographerThread() {}
-
-private:
- DISALLOW_EVIL_CONSTRUCTORS(ChoreographerThread);
- void* mData;
-};
-
-ChoreographerThread::ChoreographerThread(bool canCallJava) : Thread(canCallJava) {
-}
-
-status_t ChoreographerThread::init(void* data) {
- if (data == NULL) {
- return NO_INIT;
- }
- mData = data;
- return OK;
-}
-
-status_t ChoreographerThread::readyToRun() {
- ALooper_prepare(ALOOPER_PREPARE_ALLOW_NON_CALLBACKS);
- if (AChoreographer_getInstance() == NULL) {
- return NO_INIT;
- }
- AChoreographer_postFrameCallback64(AChoreographer_getInstance(), frameCallback, mData);
- return OK;
-}
-
-bool ChoreographerThread::threadLoop() {
- ALooper_pollOnce(-1, nullptr, nullptr, nullptr);
- return true;
-}
-
-/* ======================================================================= */
-/* Frame Scheduler */
-/* ======================================================================= */
-
-VideoFrameScheduler2::VideoFrameScheduler2() : VideoFrameSchedulerBase() {
-
- getVsyncOffset(&mAppVsyncOffset, &mSfVsyncOffset);
-
- Mutex::Autolock threadLock(mLock);
- mChoreographerThread = new ChoreographerThread(true);
-
- mVsyncTracker = new VsyncTracker();
- if (mChoreographerThread->init(static_cast<void*>(mVsyncTracker.get())) != OK) {
- mChoreographerThread.clear();
- }
- if (mChoreographerThread != NULL && mChoreographerThread->run("Choreographer") != OK) {
- mChoreographerThread.clear();
- }
-}
-
-void VideoFrameScheduler2::updateVsync() {
- mVsyncTime = 0;
- mVsyncPeriod = 0;
-
- if (mVsyncTracker != NULL) {
- mVsyncPeriod = mVsyncTracker->getVsyncPeriod();
- mVsyncTime = mVsyncTracker->getVsyncTime(mSfVsyncOffset - mAppVsyncOffset);
- }
- mVsyncRefreshAt = systemTime(SYSTEM_TIME_MONOTONIC) + kVsyncRefreshPeriod;
-}
-
-void VideoFrameScheduler2::release() {
- // Do not change order
- {
- Mutex::Autolock threadLock(mLock);
- mChoreographerThread->requestExitAndWait();
- mChoreographerThread.clear();
- }
-
- mVsyncTracker.clear();
-}
-
-VideoFrameScheduler2::~VideoFrameScheduler2() {
- release();
-}
-
-} // namespace android
diff --git a/media/libstagefright/bqhelper/Android.bp b/media/libstagefright/bqhelper/Android.bp
index db67034..6719bab 100644
--- a/media/libstagefright/bqhelper/Android.bp
+++ b/media/libstagefright/bqhelper/Android.bp
@@ -27,7 +27,6 @@
"libcutils",
"libhidlbase",
"libhidlmemory",
- "libhidltransport",
"liblog",
"libstagefright_foundation",
"libui",
@@ -39,7 +38,6 @@
"android.hidl.token@1.0-utils",
"libbase",
"libEGL",
- "libhwbinder",
"libnativewindow",
"libvndksupport",
],
diff --git a/media/libstagefright/codecs/amrnb/common/Android.bp b/media/libstagefright/codecs/amrnb/common/Android.bp
index 772ebf9..ea8b073 100644
--- a/media/libstagefright/codecs/amrnb/common/Android.bp
+++ b/media/libstagefright/codecs/amrnb/common/Android.bp
@@ -1,4 +1,4 @@
-cc_library_shared {
+cc_library {
name: "libstagefright_amrnb_common",
vendor_available: true,
diff --git a/media/libstagefright/codecs/amrnb/dec/test/AmrnbDecTestEnvironment.h b/media/libstagefright/codecs/amrnb/dec/test/AmrnbDecTestEnvironment.h
new file mode 100644
index 0000000..0344ac5
--- /dev/null
+++ b/media/libstagefright/codecs/amrnb/dec/test/AmrnbDecTestEnvironment.h
@@ -0,0 +1,72 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __AMRNBDEC_TEST_ENVIRONMENT_H__
+#define __AMRNBDEC_TEST_ENVIRONMENT_H__
+
+#include <gtest/gtest.h>
+
+#include <getopt.h>
+
+using namespace std;
+
+class AmrnbDecTestEnvironment : public ::testing::Environment {
+ public:
+ AmrnbDecTestEnvironment() : res("/data/local/tmp/") {}
+
+ // Parses the command line arguments
+ int initFromOptions(int argc, char **argv);
+
+ void setRes(const char *_res) { res = _res; }
+
+ const string getRes() const { return res; }
+
+ private:
+ string res;
+};
+
+int AmrnbDecTestEnvironment::initFromOptions(int argc, char **argv) {
+ static struct option options[] = {{"res", required_argument, 0, 'P'}, {0, 0, 0, 0}};
+
+ while (true) {
+ int index = 0;
+ int c = getopt_long(argc, argv, "P:", options, &index);
+ if (c == -1) {
+ break;
+ }
+
+ switch (c) {
+ case 'P':
+ setRes(optarg);
+ break;
+ default:
+ break;
+ }
+ }
+
+ if (optind < argc) {
+ fprintf(stderr,
+ "unrecognized option: %s\n\n"
+ "usage: %s <gtest options> <test options>\n\n"
+ "test options are:\n\n"
+ "-P, --path: Resource files directory location\n",
+ argv[optind ?: 1], argv[0]);
+ return 2;
+ }
+ return 0;
+}
+
+#endif // __AMRNBDEC_TEST_ENVIRONMENT_H__
diff --git a/media/libstagefright/codecs/amrnb/dec/test/AmrnbDecoderTest.cpp b/media/libstagefright/codecs/amrnb/dec/test/AmrnbDecoderTest.cpp
new file mode 100644
index 0000000..af62074
--- /dev/null
+++ b/media/libstagefright/codecs/amrnb/dec/test/AmrnbDecoderTest.cpp
@@ -0,0 +1,175 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "AmrnbDecoderTest"
+#define OUTPUT_FILE "/data/local/tmp/amrnbDecode.out"
+
+#include <utils/Log.h>
+
+#include <audio_utils/sndfile.h>
+#include <stdio.h>
+
+#include "gsmamr_dec.h"
+
+#include "AmrnbDecTestEnvironment.h"
+
+// Constants for AMR-NB
+constexpr int32_t kInputBufferSize = 64;
+constexpr int32_t kSamplesPerFrame = L_FRAME;
+constexpr int32_t kBitsPerSample = 16;
+constexpr int32_t kSampleRate = 8000;
+constexpr int32_t kChannels = 1;
+constexpr int32_t kOutputBufferSize = kSamplesPerFrame * kBitsPerSample / 8;
+const int32_t kFrameSizes[] = {12, 13, 15, 17, 19, 20, 26, 31, -1, -1, -1, -1, -1, -1, -1, -1};
+
+constexpr int32_t kNumFrameReset = 150;
+
+static AmrnbDecTestEnvironment *gEnv = nullptr;
+
+class AmrnbDecoderTest : public ::testing::TestWithParam<string> {
+ public:
+ AmrnbDecoderTest() : mFpInput(nullptr) {}
+
+ ~AmrnbDecoderTest() {
+ if (mFpInput) {
+ fclose(mFpInput);
+ mFpInput = nullptr;
+ }
+ }
+
+ FILE *mFpInput;
+ SNDFILE *openOutputFile(SF_INFO *sfInfo);
+ int32_t DecodeFrames(void *amrHandle, SNDFILE *outFileHandle, int32_t frameCount = INT32_MAX);
+};
+
+SNDFILE *AmrnbDecoderTest::openOutputFile(SF_INFO *sfInfo) {
+ memset(sfInfo, 0, sizeof(SF_INFO));
+ sfInfo->channels = kChannels;
+ sfInfo->format = SF_FORMAT_WAV | SF_FORMAT_PCM_16;
+ sfInfo->samplerate = kSampleRate;
+ SNDFILE *outFileHandle = sf_open(OUTPUT_FILE, SFM_WRITE, sfInfo);
+ return outFileHandle;
+}
+
+int32_t AmrnbDecoderTest::DecodeFrames(void *amrHandle, SNDFILE *outFileHandle,
+ int32_t frameCount) {
+ uint8_t inputBuf[kInputBufferSize];
+ int16_t outputBuf[kOutputBufferSize];
+
+ while (frameCount > 0) {
+ uint8_t mode;
+ int32_t bytesRead = fread(&mode, 1, 1, mFpInput);
+ if (bytesRead != 1) break;
+
+ // Find frame type
+ Frame_Type_3GPP frameType = (Frame_Type_3GPP)((mode >> 3) & 0x0f);
+ int32_t frameSize = kFrameSizes[frameType];
+ if (frameSize < 0) {
+ ALOGE("Illegal frame type");
+ return -1;
+ }
+ bytesRead = fread(inputBuf, 1, frameSize, mFpInput);
+ if (bytesRead != frameSize) break;
+
+ int32_t bytesDecoded = AMRDecode(amrHandle, frameType, inputBuf, outputBuf, MIME_IETF);
+ if (bytesDecoded == -1) {
+ ALOGE("Failed to decode the input file");
+ return -1;
+ }
+
+ sf_writef_short(outFileHandle, outputBuf, kSamplesPerFrame);
+ frameCount--;
+ }
+ return 0;
+}
+
+TEST_F(AmrnbDecoderTest, CreateAmrnbDecoderTest) {
+ void *amrHandle;
+ int32_t status = GSMInitDecode(&amrHandle, (Word8 *)"AMRNBDecoder");
+ ASSERT_EQ(status, 0) << "Error creating AMR-NB decoder";
+ GSMDecodeFrameExit(&amrHandle);
+ ASSERT_EQ(amrHandle, nullptr) << "Error deleting AMR-NB decoder";
+}
+
+TEST_P(AmrnbDecoderTest, DecodeTest) {
+ string inputFile = gEnv->getRes() + GetParam();
+ mFpInput = fopen(inputFile.c_str(), "rb");
+ ASSERT_NE(mFpInput, nullptr) << "Error opening input file " << inputFile;
+
+ // Open the output file.
+ SF_INFO sfInfo;
+ SNDFILE *outFileHandle = openOutputFile(&sfInfo);
+ ASSERT_NE(outFileHandle, nullptr) << "Error opening output file for writing decoded output";
+
+ void *amrHandle;
+ int32_t status = GSMInitDecode(&amrHandle, (Word8 *)"AMRNBDecoder");
+ ASSERT_EQ(status, 0) << "Error creating AMR-NB decoder";
+
+ // Decode
+ int32_t decoderErr = DecodeFrames(amrHandle, outFileHandle);
+ ASSERT_EQ(decoderErr, 0) << "DecodeFrames returned error";
+
+ sf_close(outFileHandle);
+ GSMDecodeFrameExit(&amrHandle);
+ ASSERT_EQ(amrHandle, nullptr) << "Error deleting AMR-NB decoder";
+}
+
+TEST_P(AmrnbDecoderTest, ResetDecodeTest) {
+ string inputFile = gEnv->getRes() + GetParam();
+ mFpInput = fopen(inputFile.c_str(), "rb");
+ ASSERT_NE(mFpInput, nullptr) << "Error opening input file " << inputFile;
+
+ // Open the output file.
+ SF_INFO sfInfo;
+ SNDFILE *outFileHandle = openOutputFile(&sfInfo);
+ ASSERT_NE(outFileHandle, nullptr) << "Error opening output file for writing decoded output";
+
+ void *amrHandle;
+ int32_t status = GSMInitDecode(&amrHandle, (Word8 *)"AMRNBDecoder");
+ ASSERT_EQ(status, 0) << "Error creating AMR-NB decoder";
+
+ // Decode kNumFrameReset first
+ int32_t decoderErr = DecodeFrames(amrHandle, outFileHandle, kNumFrameReset);
+ ASSERT_EQ(decoderErr, 0) << "DecodeFrames returned error";
+
+ status = Speech_Decode_Frame_reset(amrHandle);
+ ASSERT_EQ(status, 0) << "Error resting AMR-NB decoder";
+
+ // Start decoding again
+ decoderErr = DecodeFrames(amrHandle, outFileHandle);
+ ASSERT_EQ(decoderErr, 0) << "DecodeFrames returned error";
+
+ sf_close(outFileHandle);
+ GSMDecodeFrameExit(&amrHandle);
+ ASSERT_EQ(amrHandle, nullptr) << "Error deleting AMR-NB decoder";
+}
+
+INSTANTIATE_TEST_SUITE_P(AmrnbDecoderTestAll, AmrnbDecoderTest,
+ ::testing::Values(("bbb_8000hz_1ch_8kbps_amrnb_30sec.amrnb"),
+ ("sine_amrnb_1ch_12kbps_8000hz.amrnb")));
+
+int main(int argc, char **argv) {
+ gEnv = new AmrnbDecTestEnvironment();
+ ::testing::AddGlobalTestEnvironment(gEnv);
+ ::testing::InitGoogleTest(&argc, argv);
+ int status = gEnv->initFromOptions(argc, argv);
+ if (status == 0) {
+ status = RUN_ALL_TESTS();
+ ALOGV("Test result = %d\n", status);
+ }
+ return status;
+}
diff --git a/media/libstagefright/codecs/amrnb/dec/test/Android.bp b/media/libstagefright/codecs/amrnb/dec/test/Android.bp
new file mode 100644
index 0000000..7a95cfa
--- /dev/null
+++ b/media/libstagefright/codecs/amrnb/dec/test/Android.bp
@@ -0,0 +1,48 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+cc_test {
+ name: "AmrnbDecoderTest",
+ gtest: true,
+
+ srcs: [
+ "AmrnbDecoderTest.cpp",
+ ],
+
+ static_libs: [
+ "libstagefright_amrnb_common",
+ "libstagefright_amrnbdec",
+ "libaudioutils",
+ "libsndfile",
+ ],
+
+ shared_libs: [
+ "liblog",
+ ],
+
+ cflags: [
+ "-Werror",
+ "-Wall",
+ ],
+
+ sanitize: {
+ cfi: true,
+ misc_undefined: [
+ "unsigned-integer-overflow",
+ "signed-integer-overflow",
+ ],
+ },
+}
diff --git a/media/libstagefright/codecs/amrnb/dec/test/AndroidTest.xml b/media/libstagefright/codecs/amrnb/dec/test/AndroidTest.xml
new file mode 100644
index 0000000..1a9e678
--- /dev/null
+++ b/media/libstagefright/codecs/amrnb/dec/test/AndroidTest.xml
@@ -0,0 +1,31 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!-- Copyright (C) 2020 The Android Open Source Project
+
+ Licensed under the Apache License, Version 2.0 (the "License");
+ you may not use this file except in compliance with the License.
+ You may obtain a copy of the License at
+
+ http://www.apache.org/licenses/LICENSE-2.0
+
+ Unless required by applicable law or agreed to in writing, software
+ distributed under the License is distributed on an "AS IS" BASIS,
+ WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ See the License for the specific language governing permissions and
+ limitations under the License.
+-->
+<configuration description="Test module config for Amr-nb Decoder unit test">
+ <option name="test-suite-tag" value="AmrnbDecoderTest" />
+ <target_preparer class="com.android.tradefed.targetprep.PushFilePreparer">
+ <option name="cleanup" value="true" />
+ <option name="push" value="AmrnbDecoderTest->/data/local/tmp/AmrnbDecoderTest" />
+ <option name="push-file"
+ key="https://storage.googleapis.com/android_media/frameworks/av/media/libstagefright/codecs/amrnb/dec/test/AmrnbDecoderTest.zip?unzip=true"
+ value="/data/local/tmp/AmrnbDecoderTestRes/" />
+ </target_preparer>
+
+ <test class="com.android.tradefed.testtype.GTest" >
+ <option name="native-test-device-path" value="/data/local/tmp" />
+ <option name="module-name" value="AmrnbDecoderTest" />
+ <option name="native-test-flag" value="-P /data/local/tmp/AmrnbDecoderTestRes/" />
+ </test>
+</configuration>
diff --git a/media/libstagefright/codecs/amrnb/dec/test/README.md b/media/libstagefright/codecs/amrnb/dec/test/README.md
new file mode 100644
index 0000000..e9073e4
--- /dev/null
+++ b/media/libstagefright/codecs/amrnb/dec/test/README.md
@@ -0,0 +1,39 @@
+## Media Testing ##
+---
+#### AMR-NB Decoder :
+The Amr-Nb Decoder Test Suite validates the amrnb decoder available in libstagefright.
+
+Run the following steps to build the test suite:
+```
+m AmrnbDecoderTest
+```
+
+The 32-bit binaries will be created in the following path : ${OUT}/data/nativetest/
+
+The 64-bit binaries will be created in the following path : ${OUT}/data/nativetest64/
+
+To test 64-bit binary push binaries from nativetest64.
+```
+adb push ${OUT}/data/nativetest64/AmrnbDecoderTest/AmrnbDecoderTest /data/local/tmp/
+```
+
+To test 32-bit binary push binaries from nativetest.
+```
+adb push ${OUT}/data/nativetest/AmrnbDecoderTest/AmrnbDecoderTest /data/local/tmp/
+```
+
+The resource file for the tests is taken from [here](https://storage.googleapis.com/android_media/frameworks/av/media/libstagefright/codecs/amrnb/dec/test/AmrnbDecoderTest.zip). Download, unzip and push these files into device for testing.
+
+```
+adb push AmrnbDecoderTestRes/. /data/local/tmp/
+```
+
+usage: AmrnbDecoderTest -P \<path_to_folder\>
+```
+adb shell /data/local/tmp/AmrnbDecoderTest -P /data/local/tmp/AmrnbDecoderTestRes/
+```
+Alternatively, the test can also be run using atest command.
+
+```
+atest AmrnbDecoderTest -- --enable-module-dynamic-download=true
+```
diff --git a/media/libstagefright/codecs/amrnb/enc/test/AmrnbEncTestEnvironment.h b/media/libstagefright/codecs/amrnb/enc/test/AmrnbEncTestEnvironment.h
new file mode 100644
index 0000000..5a2fcd1
--- /dev/null
+++ b/media/libstagefright/codecs/amrnb/enc/test/AmrnbEncTestEnvironment.h
@@ -0,0 +1,72 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __AMRNBENC_TEST_ENVIRONMENT_H__
+#define __AMRNBENC_TEST_ENVIRONMENT_H__
+
+#include <gtest/gtest.h>
+
+#include <getopt.h>
+
+using namespace std;
+
+class AmrnbEncTestEnvironment : public ::testing::Environment {
+ public:
+ AmrnbEncTestEnvironment() : res("/data/local/tmp/") {}
+
+ // Parses the command line arguments
+ int initFromOptions(int argc, char **argv);
+
+ void setRes(const char *_res) { res = _res; }
+
+ const string getRes() const { return res; }
+
+ private:
+ string res;
+};
+
+int AmrnbEncTestEnvironment::initFromOptions(int argc, char **argv) {
+ static struct option options[] = {{"res", required_argument, 0, 'P'}, {0, 0, 0, 0}};
+
+ while (true) {
+ int index = 0;
+ int c = getopt_long(argc, argv, "P:", options, &index);
+ if (c == -1) {
+ break;
+ }
+
+ switch (c) {
+ case 'P':
+ setRes(optarg);
+ break;
+ default:
+ break;
+ }
+ }
+
+ if (optind < argc) {
+ fprintf(stderr,
+ "unrecognized option: %s\n\n"
+ "usage: %s <gtest options> <test options>\n\n"
+ "test options are:\n\n"
+ "-P, --path: Resource files directory location\n",
+ argv[optind ?: 1], argv[0]);
+ return 2;
+ }
+ return 0;
+}
+
+#endif // __AMRNBENC_TEST_ENVIRONMENT_H__
diff --git a/media/libstagefright/codecs/amrnb/enc/test/AmrnbEncoderTest.cpp b/media/libstagefright/codecs/amrnb/enc/test/AmrnbEncoderTest.cpp
new file mode 100644
index 0000000..fb72998
--- /dev/null
+++ b/media/libstagefright/codecs/amrnb/enc/test/AmrnbEncoderTest.cpp
@@ -0,0 +1,207 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "AmrnbEncoderTest"
+
+#include <utils/Log.h>
+
+#include <audio_utils/sndfile.h>
+#include <stdio.h>
+
+#include "gsmamr_enc.h"
+
+#include "AmrnbEncTestEnvironment.h"
+
+#define OUTPUT_FILE "/data/local/tmp/amrnbEncode.out"
+
+constexpr int32_t kInputBufferSize = L_FRAME * 2; // 160 samples * 16-bit per sample.
+constexpr int32_t kOutputBufferSize = 1024;
+constexpr int32_t kNumFrameReset = 200;
+constexpr int32_t kMaxCount = 10;
+struct AmrNbEncState {
+ void *encCtx;
+ void *pidSyncCtx;
+};
+
+static AmrnbEncTestEnvironment *gEnv = nullptr;
+
+class AmrnbEncoderTest : public ::testing::TestWithParam<pair<string, int32_t>> {
+ public:
+ AmrnbEncoderTest() : mAmrEncHandle(nullptr) {}
+
+ ~AmrnbEncoderTest() {
+ if (mAmrEncHandle) {
+ free(mAmrEncHandle);
+ mAmrEncHandle = nullptr;
+ }
+ }
+
+ AmrNbEncState *mAmrEncHandle;
+ int32_t EncodeFrames(int32_t mode, FILE *fpInput, FILE *mFpOutput,
+ int32_t frameCount = INT32_MAX);
+};
+
+int32_t AmrnbEncoderTest::EncodeFrames(int32_t mode, FILE *fpInput, FILE *mFpOutput,
+ int32_t frameCount) {
+ int32_t frameNum = 0;
+ uint16_t inputBuf[kInputBufferSize];
+ uint8_t outputBuf[kOutputBufferSize];
+ while (frameNum < frameCount) {
+ int32_t bytesRead = fread(inputBuf, 1, kInputBufferSize, fpInput);
+ if (bytesRead != kInputBufferSize && !feof(fpInput)) {
+ ALOGE("Unable to read data from input file");
+ return -1;
+ } else if (feof(fpInput) && bytesRead == 0) {
+ break;
+ }
+ Frame_Type_3GPP frame_type = (Frame_Type_3GPP)mode;
+ int32_t bytesGenerated =
+ AMREncode(mAmrEncHandle->encCtx, mAmrEncHandle->pidSyncCtx, (Mode)mode,
+ (Word16 *)inputBuf, outputBuf, &frame_type, AMR_TX_WMF);
+ frameNum++;
+ if (bytesGenerated < 0) {
+ ALOGE("Error in encoging the file: Invalid output format");
+ return -1;
+ }
+
+ // Convert from WMF to RFC 3267 format.
+ if (bytesGenerated > 0) {
+ outputBuf[0] = ((outputBuf[0] << 3) | 4) & 0x7c;
+ }
+ fwrite(outputBuf, 1, bytesGenerated, mFpOutput);
+ }
+ return 0;
+}
+
+TEST_F(AmrnbEncoderTest, CreateAmrnbEncoderTest) {
+ mAmrEncHandle = (AmrNbEncState *)malloc(sizeof(AmrNbEncState));
+ ASSERT_NE(mAmrEncHandle, nullptr) << "Error in allocating memory to Codec handle";
+ for (int count = 0; count < kMaxCount; count++) {
+ int32_t status = AMREncodeInit(&mAmrEncHandle->encCtx, &mAmrEncHandle->pidSyncCtx, 0);
+ ASSERT_EQ(status, 0) << "Error creating AMR-NB encoder";
+ ALOGV("Successfully created encoder");
+ }
+ if (mAmrEncHandle) {
+ AMREncodeExit(&mAmrEncHandle->encCtx, &mAmrEncHandle->pidSyncCtx);
+ ASSERT_EQ(mAmrEncHandle->encCtx, nullptr) << "Error deleting AMR-NB encoder";
+ ASSERT_EQ(mAmrEncHandle->pidSyncCtx, nullptr) << "Error deleting AMR-NB encoder";
+ free(mAmrEncHandle);
+ mAmrEncHandle = nullptr;
+ ALOGV("Successfully deleted encoder");
+ }
+}
+
+TEST_P(AmrnbEncoderTest, EncodeTest) {
+ mAmrEncHandle = (AmrNbEncState *)malloc(sizeof(AmrNbEncState));
+ ASSERT_NE(mAmrEncHandle, nullptr) << "Error in allocating memory to Codec handle";
+ int32_t status = AMREncodeInit(&mAmrEncHandle->encCtx, &mAmrEncHandle->pidSyncCtx, 0);
+ ASSERT_EQ(status, 0) << "Error creating AMR-NB encoder";
+
+ string inputFile = gEnv->getRes() + GetParam().first;
+ FILE *fpInput = fopen(inputFile.c_str(), "rb");
+ ASSERT_NE(fpInput, nullptr) << "Error opening input file " << inputFile;
+
+ FILE *fpOutput = fopen(OUTPUT_FILE, "wb");
+ ASSERT_NE(fpOutput, nullptr) << "Error opening output file " << OUTPUT_FILE;
+
+ // Write file header.
+ fwrite("#!AMR\n", 1, 6, fpOutput);
+
+ int32_t mode = GetParam().second;
+ int32_t encodeErr = EncodeFrames(mode, fpInput, fpOutput);
+ ASSERT_EQ(encodeErr, 0) << "EncodeFrames returned error for Codec mode: " << mode;
+
+ fclose(fpOutput);
+ fclose(fpInput);
+
+ AMREncodeExit(&mAmrEncHandle->encCtx, &mAmrEncHandle->pidSyncCtx);
+ ASSERT_EQ(mAmrEncHandle->encCtx, nullptr) << "Error deleting AMR-NB encoder";
+ ASSERT_EQ(mAmrEncHandle->pidSyncCtx, nullptr) << "Error deleting AMR-NB encoder";
+ free(mAmrEncHandle);
+ mAmrEncHandle = nullptr;
+ ALOGV("Successfully deleted encoder");
+}
+
+TEST_P(AmrnbEncoderTest, ResetEncoderTest) {
+ mAmrEncHandle = (AmrNbEncState *)malloc(sizeof(AmrNbEncState));
+ ASSERT_NE(mAmrEncHandle, nullptr) << "Error in allocating memory to Codec handle";
+ int32_t status = AMREncodeInit(&mAmrEncHandle->encCtx, &mAmrEncHandle->pidSyncCtx, 0);
+ ASSERT_EQ(status, 0) << "Error creating AMR-NB encoder";
+
+ string inputFile = gEnv->getRes() + GetParam().first;
+ FILE *fpInput = fopen(inputFile.c_str(), "rb");
+ ASSERT_NE(fpInput, nullptr) << "Error opening input file " << inputFile;
+
+ FILE *fpOutput = fopen(OUTPUT_FILE, "wb");
+ ASSERT_NE(fpOutput, nullptr) << "Error opening output file " << OUTPUT_FILE;
+
+ // Write file header.
+ fwrite("#!AMR\n", 1, 6, fpOutput);
+
+ int32_t mode = GetParam().second;
+ // Encode kNumFrameReset first
+ int32_t encodeErr = EncodeFrames(mode, fpInput, fpOutput, kNumFrameReset);
+ ASSERT_EQ(encodeErr, 0) << "EncodeFrames returned error for Codec mode: " << mode;
+
+ status = AMREncodeReset(mAmrEncHandle->encCtx, mAmrEncHandle->pidSyncCtx);
+ ASSERT_EQ(status, 0) << "Error resting AMR-NB encoder";
+
+ // Start encoding again
+ encodeErr = EncodeFrames(mode, fpInput, fpOutput);
+ ASSERT_EQ(encodeErr, 0) << "EncodeFrames returned error for Codec mode: " << mode;
+
+ fclose(fpOutput);
+ fclose(fpInput);
+
+ AMREncodeExit(&mAmrEncHandle->encCtx, &mAmrEncHandle->pidSyncCtx);
+ ASSERT_EQ(mAmrEncHandle->encCtx, nullptr) << "Error deleting AMR-NB encoder";
+ ASSERT_EQ(mAmrEncHandle->pidSyncCtx, nullptr) << "Error deleting AMR-NB encoder";
+ free(mAmrEncHandle);
+ mAmrEncHandle = nullptr;
+ ALOGV("Successfully deleted encoder");
+}
+
+// TODO: Add more test vectors
+INSTANTIATE_TEST_SUITE_P(AmrnbEncoderTestAll, AmrnbEncoderTest,
+ ::testing::Values(make_pair("bbb_raw_1ch_8khz_s16le.raw", MR475),
+ make_pair("bbb_raw_1ch_8khz_s16le.raw", MR515),
+ make_pair("bbb_raw_1ch_8khz_s16le.raw", MR59),
+ make_pair("bbb_raw_1ch_8khz_s16le.raw", MR67),
+ make_pair("bbb_raw_1ch_8khz_s16le.raw", MR74),
+ make_pair("bbb_raw_1ch_8khz_s16le.raw", MR795),
+ make_pair("bbb_raw_1ch_8khz_s16le.raw", MR102),
+ make_pair("bbb_raw_1ch_8khz_s16le.raw", MR122),
+ make_pair("sinesweepraw.raw", MR475),
+ make_pair("sinesweepraw.raw", MR515),
+ make_pair("sinesweepraw.raw", MR59),
+ make_pair("sinesweepraw.raw", MR67),
+ make_pair("sinesweepraw.raw", MR74),
+ make_pair("sinesweepraw.raw", MR795),
+ make_pair("sinesweepraw.raw", MR102),
+ make_pair("sinesweepraw.raw", MR122)));
+
+int main(int argc, char **argv) {
+ gEnv = new AmrnbEncTestEnvironment();
+ ::testing::AddGlobalTestEnvironment(gEnv);
+ ::testing::InitGoogleTest(&argc, argv);
+ int status = gEnv->initFromOptions(argc, argv);
+ if (status == 0) {
+ status = RUN_ALL_TESTS();
+ ALOGV("Test result = %d\n", status);
+ }
+ return status;
+}
diff --git a/media/libstagefright/codecs/amrnb/enc/test/Android.bp b/media/libstagefright/codecs/amrnb/enc/test/Android.bp
new file mode 100644
index 0000000..e8982fe
--- /dev/null
+++ b/media/libstagefright/codecs/amrnb/enc/test/Android.bp
@@ -0,0 +1,48 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+cc_test {
+ name: "AmrnbEncoderTest",
+ gtest: true,
+
+ srcs: [
+ "AmrnbEncoderTest.cpp",
+ ],
+
+ static_libs: [
+ "libstagefright_amrnb_common",
+ "libstagefright_amrnbenc",
+ "libaudioutils",
+ "libsndfile",
+ ],
+
+ shared_libs: [
+ "liblog",
+ ],
+
+ cflags: [
+ "-Werror",
+ "-Wall",
+ ],
+
+ sanitize: {
+ cfi: true,
+ misc_undefined: [
+ "unsigned-integer-overflow",
+ "signed-integer-overflow",
+ ],
+ },
+}
diff --git a/media/libstagefright/codecs/amrnb/enc/test/AndroidTest.xml b/media/libstagefright/codecs/amrnb/enc/test/AndroidTest.xml
new file mode 100644
index 0000000..9fe61b1
--- /dev/null
+++ b/media/libstagefright/codecs/amrnb/enc/test/AndroidTest.xml
@@ -0,0 +1,31 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!-- Copyright (C) 2020 The Android Open Source Project
+
+ Licensed under the Apache License, Version 2.0 (the "License");
+ you may not use this file except in compliance with the License.
+ You may obtain a copy of the License at
+
+ http://www.apache.org/licenses/LICENSE-2.0
+
+ Unless required by applicable law or agreed to in writing, software
+ distributed under the License is distributed on an "AS IS" BASIS,
+ WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ See the License for the specific language governing permissions and
+ limitations under the License.
+-->
+<configuration description="Test module config for Amr-nb Encoder unit test">
+ <option name="test-suite-tag" value="AmrnbEncoderTest" />
+ <target_preparer class="com.android.tradefed.targetprep.PushFilePreparer">
+ <option name="cleanup" value="true" />
+ <option name="push" value="AmrnbEncoderTest->/data/local/tmp/AmrnbEncoderTest" />
+ <option name="push-file"
+ key="https://storage.googleapis.com/android_media/frameworks/av/media/libstagefright/codecs/amrnb/enc/test/AmrnbEncoderTest.zip?unzip=true"
+ value="/data/local/tmp/AmrnbEncoderTestRes/" />
+ </target_preparer>
+
+ <test class="com.android.tradefed.testtype.GTest" >
+ <option name="native-test-device-path" value="/data/local/tmp" />
+ <option name="module-name" value="AmrnbEncoderTest" />
+ <option name="native-test-flag" value="-P /data/local/tmp/AmrnbEncoderTestRes/" />
+ </test>
+</configuration>
diff --git a/media/libstagefright/codecs/amrnb/enc/test/README.md b/media/libstagefright/codecs/amrnb/enc/test/README.md
new file mode 100644
index 0000000..e9d2c95
--- /dev/null
+++ b/media/libstagefright/codecs/amrnb/enc/test/README.md
@@ -0,0 +1,39 @@
+## Media Testing ##
+---
+#### AMR-NB Encoder :
+The Amr-Nb Encoder Test Suite validates the amrnb encoder available in libstagefright.
+
+Run the following steps to build the test suite:
+```
+m AmrnbEncoderTest
+```
+
+The 32-bit binaries will be created in the following path : ${OUT}/data/nativetest/
+
+The 64-bit binaries will be created in the following path : ${OUT}/data/nativetest64/
+
+To test 64-bit binary push binaries from nativetest64.
+```
+adb push ${OUT}/data/nativetest64/AmrnbEncoderTest/AmrnbEncoderTest /data/local/tmp/
+```
+
+To test 32-bit binary push binaries from nativetest.
+```
+adb push ${OUT}/data/nativetest/AmrnbEncoderTest/AmrnbEncoderTest /data/local/tmp/
+```
+
+The resource file for the tests is taken from [here](https://storage.googleapis.com/android_media/frameworks/av/media/libstagefright/codecs/amrnb/enc/test/AmrnbEncoderTest.zip). Download, unzip and push these files into device for testing.
+
+```
+adb push AmrnbEncoderTestRes/. /data/local/tmp/
+```
+
+usage: AmrnbEncoderTest -P \<path_to_folder\>
+```
+adb shell /data/local/tmp/AmrnbEncoderTest -P /data/local/tmp/AmrnbEncoderTestRes/
+```
+Alternatively, the test can also be run using atest command.
+
+```
+atest AmrnbEncoderTest -- --enable-module-dynamic-download=true
+```
diff --git a/media/libstagefright/codecs/amrwb/test/AmrwbDecTestEnvironment.h b/media/libstagefright/codecs/amrwb/test/AmrwbDecTestEnvironment.h
new file mode 100644
index 0000000..84d337d
--- /dev/null
+++ b/media/libstagefright/codecs/amrwb/test/AmrwbDecTestEnvironment.h
@@ -0,0 +1,72 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __AMRWBDEC_TEST_ENVIRONMENT_H__
+#define __AMRWBDEC_TEST_ENVIRONMENT_H__
+
+#include <gtest/gtest.h>
+
+#include <getopt.h>
+
+using namespace std;
+
+class AmrwbDecTestEnvironment : public ::testing::Environment {
+ public:
+ AmrwbDecTestEnvironment() : res("/data/local/tmp/") {}
+
+ // Parses the command line arguments
+ int initFromOptions(int argc, char **argv);
+
+ void setRes(const char *_res) { res = _res; }
+
+ const string getRes() const { return res; }
+
+ private:
+ string res;
+};
+
+int AmrwbDecTestEnvironment::initFromOptions(int argc, char **argv) {
+ static struct option options[] = {{"res", required_argument, 0, 'P'}, {0, 0, 0, 0}};
+
+ while (true) {
+ int index = 0;
+ int c = getopt_long(argc, argv, "P:", options, &index);
+ if (c == -1) {
+ break;
+ }
+
+ switch (c) {
+ case 'P':
+ setRes(optarg);
+ break;
+ default:
+ break;
+ }
+ }
+
+ if (optind < argc) {
+ fprintf(stderr,
+ "unrecognized option: %s\n\n"
+ "usage: %s <gtest options> <test options>\n\n"
+ "test options are:\n\n"
+ "-P, --path: Resource files directory location\n",
+ argv[optind ?: 1], argv[0]);
+ return 2;
+ }
+ return 0;
+}
+
+#endif // __AMRWBDEC_TEST_ENVIRONMENT_H__
diff --git a/media/libstagefright/codecs/amrwb/test/AmrwbDecoderTest.cpp b/media/libstagefright/codecs/amrwb/test/AmrwbDecoderTest.cpp
new file mode 100644
index 0000000..2aad81b
--- /dev/null
+++ b/media/libstagefright/codecs/amrwb/test/AmrwbDecoderTest.cpp
@@ -0,0 +1,223 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "AmrwbDecoderTest"
+#define OUTPUT_FILE "/data/local/tmp/amrwbDecode.out"
+
+#include <utils/Log.h>
+
+#include <audio_utils/sndfile.h>
+#include <stdio.h>
+
+#include "pvamrwbdecoder.h"
+#include "pvamrwbdecoder_api.h"
+
+#include "AmrwbDecTestEnvironment.h"
+
+// Constants for AMR-WB.
+constexpr int32_t kInputBufferSize = 64;
+constexpr int32_t kSamplesPerFrame = 320;
+constexpr int32_t kBitsPerSample = 16;
+constexpr int32_t kSampleRate = 16000;
+constexpr int32_t kChannels = 1;
+constexpr int32_t kMaxSourceDataUnitSize = KAMRWB_NB_BITS_MAX * sizeof(int16_t);
+constexpr int32_t kOutputBufferSize = kSamplesPerFrame * kBitsPerSample / 8;
+const int32_t kFrameSizes[16] = {17, 23, 32, 36, 40, 46, 50, 58, 60, -1, -1, -1, -1, -1, -1, -1};
+constexpr int32_t kNumFrameReset = 150;
+
+constexpr int32_t kMaxCount = 10;
+
+static AmrwbDecTestEnvironment *gEnv = nullptr;
+
+class AmrwbDecoderTest : public ::testing::TestWithParam<string> {
+ public:
+ AmrwbDecoderTest() : mFpInput(nullptr) {}
+
+ ~AmrwbDecoderTest() {
+ if (mFpInput) {
+ fclose(mFpInput);
+ mFpInput = nullptr;
+ }
+ }
+
+ FILE *mFpInput;
+ int32_t DecodeFrames(int16_t *decoderCookie, void *decoderBuf, SNDFILE *outFileHandle,
+ int32_t frameCount = INT32_MAX);
+ SNDFILE *openOutputFile(SF_INFO *sfInfo);
+};
+
+SNDFILE *AmrwbDecoderTest::openOutputFile(SF_INFO *sfInfo) {
+ memset(sfInfo, 0, sizeof(SF_INFO));
+ sfInfo->channels = kChannels;
+ sfInfo->format = SF_FORMAT_WAV | SF_FORMAT_PCM_16;
+ sfInfo->samplerate = kSampleRate;
+ SNDFILE *outFileHandle = sf_open(OUTPUT_FILE, SFM_WRITE, sfInfo);
+ return outFileHandle;
+}
+
+int32_t AmrwbDecoderTest::DecodeFrames(int16_t *decoderCookie, void *decoderBuf,
+ SNDFILE *outFileHandle, int32_t frameCount) {
+ uint8_t inputBuf[kInputBufferSize];
+ int16_t inputSampleBuf[kMaxSourceDataUnitSize];
+ int16_t outputBuf[kOutputBufferSize];
+
+ while (frameCount > 0) {
+ uint8_t modeByte;
+ int32_t bytesRead = fread(&modeByte, 1, 1, mFpInput);
+ if (bytesRead != 1) break;
+
+ int16 mode = ((modeByte >> 3) & 0x0f);
+ if (mode >= 9) {
+ // Produce silence for comfort noise, speech lost and no data.
+ int32_t outputBufferSize = kSamplesPerFrame * kBitsPerSample / 8;
+ memset(outputBuf, 0, outputBufferSize);
+ } else {
+ // Read rest of the frame.
+ int32_t frameSize = kFrameSizes[mode];
+ // AMR-WB file format cannot have mode 10, 11, 12 and 13.
+ if (frameSize < 0) {
+ ALOGE("Illegal frame mode");
+ return -1;
+ }
+ bytesRead = fread(inputBuf, 1, frameSize, mFpInput);
+ if (bytesRead != frameSize) break;
+
+ int16 frameMode = mode;
+ int16 frameType;
+ RX_State_wb rx_state;
+ mime_unsorting(inputBuf, inputSampleBuf, &frameType, &frameMode, 1, &rx_state);
+
+ int16_t numSamplesOutput;
+ pvDecoder_AmrWb(frameMode, inputSampleBuf, outputBuf, &numSamplesOutput, decoderBuf,
+ frameType, decoderCookie);
+ if (numSamplesOutput != kSamplesPerFrame) {
+ ALOGE("Failed to decode the input file");
+ return -1;
+ }
+ for (int count = 0; count < kSamplesPerFrame; ++count) {
+ /* Delete the 2 LSBs (14-bit output) */
+ outputBuf[count] &= 0xfffc;
+ }
+ }
+ sf_writef_short(outFileHandle, outputBuf, kSamplesPerFrame / kChannels);
+ frameCount--;
+ }
+ return 0;
+}
+
+TEST_F(AmrwbDecoderTest, MultiCreateAmrwbDecoderTest) {
+ uint32_t memRequirements = pvDecoder_AmrWbMemRequirements();
+ void *decoderBuf = malloc(memRequirements);
+ ASSERT_NE(decoderBuf, nullptr)
+ << "Failed to allocate decoder memory of size " << memRequirements;
+
+ // Create AMR-WB decoder instance.
+ void *amrHandle = nullptr;
+ int16_t *decoderCookie;
+ for (int count = 0; count < kMaxCount; count++) {
+ pvDecoder_AmrWb_Init(&amrHandle, decoderBuf, &decoderCookie);
+ ASSERT_NE(amrHandle, nullptr) << "Failed to initialize decoder";
+ ALOGV("Decoder created successfully");
+ }
+ if (decoderBuf) {
+ free(decoderBuf);
+ decoderBuf = nullptr;
+ }
+}
+
+TEST_P(AmrwbDecoderTest, DecodeTest) {
+ uint32_t memRequirements = pvDecoder_AmrWbMemRequirements();
+ void *decoderBuf = malloc(memRequirements);
+ ASSERT_NE(decoderBuf, nullptr)
+ << "Failed to allocate decoder memory of size " << memRequirements;
+
+ void *amrHandle = nullptr;
+ int16_t *decoderCookie;
+ pvDecoder_AmrWb_Init(&amrHandle, decoderBuf, &decoderCookie);
+ ASSERT_NE(amrHandle, nullptr) << "Failed to initialize decoder";
+
+ string inputFile = gEnv->getRes() + GetParam();
+ mFpInput = fopen(inputFile.c_str(), "rb");
+ ASSERT_NE(mFpInput, nullptr) << "Error opening input file " << inputFile;
+
+ // Open the output file.
+ SF_INFO sfInfo;
+ SNDFILE *outFileHandle = openOutputFile(&sfInfo);
+ ASSERT_NE(outFileHandle, nullptr) << "Error opening output file for writing decoded output";
+
+ int32_t decoderErr = DecodeFrames(decoderCookie, decoderBuf, outFileHandle);
+ ASSERT_EQ(decoderErr, 0) << "DecodeFrames returned error";
+
+ sf_close(outFileHandle);
+ if (decoderBuf) {
+ free(decoderBuf);
+ decoderBuf = nullptr;
+ }
+}
+
+TEST_P(AmrwbDecoderTest, ResetDecoderTest) {
+ uint32_t memRequirements = pvDecoder_AmrWbMemRequirements();
+ void *decoderBuf = malloc(memRequirements);
+ ASSERT_NE(decoderBuf, nullptr)
+ << "Failed to allocate decoder memory of size " << memRequirements;
+
+ void *amrHandle = nullptr;
+ int16_t *decoderCookie;
+ pvDecoder_AmrWb_Init(&amrHandle, decoderBuf, &decoderCookie);
+ ASSERT_NE(amrHandle, nullptr) << "Failed to initialize decoder";
+
+ string inputFile = gEnv->getRes() + GetParam();
+ mFpInput = fopen(inputFile.c_str(), "rb");
+ ASSERT_NE(mFpInput, nullptr) << "Error opening input file " << inputFile;
+
+ // Open the output file.
+ SF_INFO sfInfo;
+ SNDFILE *outFileHandle = openOutputFile(&sfInfo);
+ ASSERT_NE(outFileHandle, nullptr) << "Error opening output file for writing decoded output";
+
+ // Decode 150 frames first
+ int32_t decoderErr = DecodeFrames(decoderCookie, decoderBuf, outFileHandle, kNumFrameReset);
+ ASSERT_EQ(decoderErr, 0) << "DecodeFrames returned error";
+
+ // Reset Decoder
+ pvDecoder_AmrWb_Reset(decoderBuf, 1);
+
+ // Start decoding again
+ decoderErr = DecodeFrames(decoderCookie, decoderBuf, outFileHandle);
+ ASSERT_EQ(decoderErr, 0) << "DecodeFrames returned error";
+
+ sf_close(outFileHandle);
+ if (decoderBuf) {
+ free(decoderBuf);
+ }
+}
+
+INSTANTIATE_TEST_SUITE_P(AmrwbDecoderTestAll, AmrwbDecoderTest,
+ ::testing::Values(("bbb_amrwb_1ch_14kbps_16000hz.amrwb"),
+ ("bbb_16000hz_1ch_9kbps_amrwb_30sec.amrwb")));
+
+int main(int argc, char **argv) {
+ gEnv = new AmrwbDecTestEnvironment();
+ ::testing::AddGlobalTestEnvironment(gEnv);
+ ::testing::InitGoogleTest(&argc, argv);
+ int status = gEnv->initFromOptions(argc, argv);
+ if (status == 0) {
+ status = RUN_ALL_TESTS();
+ ALOGV("Test result = %d\n", status);
+ }
+ return status;
+}
diff --git a/media/libstagefright/codecs/amrwb/test/Android.bp b/media/libstagefright/codecs/amrwb/test/Android.bp
new file mode 100644
index 0000000..968215a
--- /dev/null
+++ b/media/libstagefright/codecs/amrwb/test/Android.bp
@@ -0,0 +1,47 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+cc_test {
+ name: "AmrwbDecoderTest",
+ gtest: true,
+
+ srcs: [
+ "AmrwbDecoderTest.cpp",
+ ],
+
+ static_libs: [
+ "libstagefright_amrwbdec",
+ "libsndfile",
+ "libaudioutils",
+ ],
+
+ shared_libs: [
+ "liblog",
+ ],
+
+ cflags: [
+ "-Werror",
+ "-Wall",
+ ],
+
+ sanitize: {
+ cfi: true,
+ misc_undefined: [
+ "unsigned-integer-overflow",
+ "signed-integer-overflow",
+ ],
+ },
+}
diff --git a/media/libstagefright/codecs/amrwb/test/AndroidTest.xml b/media/libstagefright/codecs/amrwb/test/AndroidTest.xml
new file mode 100644
index 0000000..e211a1f
--- /dev/null
+++ b/media/libstagefright/codecs/amrwb/test/AndroidTest.xml
@@ -0,0 +1,31 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!-- Copyright (C) 2020 The Android Open Source Project
+
+ Licensed under the Apache License, Version 2.0 (the "License");
+ you may not use this file except in compliance with the License.
+ You may obtain a copy of the License at
+
+ http://www.apache.org/licenses/LICENSE-2.0
+
+ Unless required by applicable law or agreed to in writing, software
+ distributed under the License is distributed on an "AS IS" BASIS,
+ WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ See the License for the specific language governing permissions and
+ limitations under the License.
+-->
+<configuration description="Test module config for Amr-wb Decoder unit test">
+ <option name="test-suite-tag" value="AmrwbDecoderTest" />
+ <target_preparer class="com.android.tradefed.targetprep.PushFilePreparer">
+ <option name="cleanup" value="true" />
+ <option name="push" value="AmrwbDecoderTest->/data/local/tmp/AmrwbDecoderTest" />
+ <option name="push-file"
+ key="https://storage.googleapis.com/android_media/frameworks/av/media/libstagefright/codecs/amrwb/test/AmrwbDecoderTest.zip?unzip=true"
+ value="/data/local/tmp/AmrwbDecoderTestRes/" />
+ </target_preparer>
+
+ <test class="com.android.tradefed.testtype.GTest" >
+ <option name="native-test-device-path" value="/data/local/tmp" />
+ <option name="module-name" value="AmrwbDecoderTest" />
+ <option name="native-test-flag" value="-P /data/local/tmp/AmrwbDecoderTestRes/" />
+ </test>
+</configuration>
diff --git a/media/libstagefright/codecs/amrwb/test/README.md b/media/libstagefright/codecs/amrwb/test/README.md
new file mode 100644
index 0000000..a9d5c06
--- /dev/null
+++ b/media/libstagefright/codecs/amrwb/test/README.md
@@ -0,0 +1,39 @@
+## Media Testing ##
+---
+#### AMR-WB Decoder :
+The Amr-Wb Decoder Test Suite validates the amrwb decoder available in libstagefright.
+
+Run the following steps to build the test suite:
+```
+m AmrwbDecoderTest
+```
+
+The 32-bit binaries will be created in the following path : ${OUT}/data/nativetest/
+
+The 64-bit binaries will be created in the following path : ${OUT}/data/nativetest64/
+
+To test 64-bit binary push binaries from nativetest64.
+```
+adb push ${OUT}/data/nativetest64/AmrwbDecoderTest/AmrwbDecoderTest /data/local/tmp/
+```
+
+To test 32-bit binary push binaries from nativetest.
+```
+adb push ${OUT}/data/nativetest/AmrwbDecoderTest/AmrwbDecoderTest /data/local/tmp/
+```
+
+The resource file for the tests is taken from [here](https://storage.googleapis.com/android_media/frameworks/av/media/libstagefright/codecs/amrwb/test/AmrwbDecoderTest.zip). Download, unzip and push these files into device for testing.
+
+```
+adb push AmrwbDecoderTestRes/. /data/local/tmp/
+```
+
+usage: AmrwbDecoderTest -P \<path_to_folder\>
+```
+adb shell /data/local/tmp/AmrwbDecoderTest -P /data/local/tmp/AmrwbDecoderTestRes/
+```
+Alternatively, the test can also be run using atest command.
+
+```
+atest AmrwbDecoderTest -- --enable-module-dynamic-download=true
+```
diff --git a/media/libstagefright/codecs/amrwbenc/test/AmrwbEncTestEnvironment.h b/media/libstagefright/codecs/amrwbenc/test/AmrwbEncTestEnvironment.h
new file mode 100644
index 0000000..08ada66
--- /dev/null
+++ b/media/libstagefright/codecs/amrwbenc/test/AmrwbEncTestEnvironment.h
@@ -0,0 +1,72 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __AMRWBENC_TEST_ENVIRONMENT_H__
+#define __AMRWBENC_TEST_ENVIRONMENT_H__
+
+#include <gtest/gtest.h>
+
+#include <getopt.h>
+
+using namespace std;
+
+class AmrwbEncTestEnvironment : public ::testing::Environment {
+ public:
+ AmrwbEncTestEnvironment() : res("/data/local/tmp/") {}
+
+ // Parses the command line arguments
+ int initFromOptions(int argc, char **argv);
+
+ void setRes(const char *_res) { res = _res; }
+
+ const string getRes() const { return res; }
+
+ private:
+ string res;
+};
+
+int AmrwbEncTestEnvironment::initFromOptions(int argc, char **argv) {
+ static struct option options[] = {{"res", required_argument, 0, 'P'}, {0, 0, 0, 0}};
+
+ while (true) {
+ int index = 0;
+ int c = getopt_long(argc, argv, "P:", options, &index);
+ if (c == -1) {
+ break;
+ }
+
+ switch (c) {
+ case 'P':
+ setRes(optarg);
+ break;
+ default:
+ break;
+ }
+ }
+
+ if (optind < argc) {
+ fprintf(stderr,
+ "unrecognized option: %s\n\n"
+ "usage: %s <gtest options> <test options>\n\n"
+ "test options are:\n\n"
+ "-P, --path: Resource files directory location\n",
+ argv[optind ?: 1], argv[0]);
+ return 2;
+ }
+ return 0;
+}
+
+#endif // __AMRWBENC_TEST_ENVIRONMENT_H__
diff --git a/media/libstagefright/codecs/amrwbenc/test/AmrwbEncoderTest.cpp b/media/libstagefright/codecs/amrwbenc/test/AmrwbEncoderTest.cpp
new file mode 100644
index 0000000..1a6ee27
--- /dev/null
+++ b/media/libstagefright/codecs/amrwbenc/test/AmrwbEncoderTest.cpp
@@ -0,0 +1,198 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "AmrwbEncoderTest"
+
+#include <utils/Log.h>
+
+#include <stdio.h>
+
+#include "cmnMemory.h"
+#include "voAMRWB.h"
+
+#include "AmrwbEncTestEnvironment.h"
+
+#define OUTPUT_FILE "/data/local/tmp/amrwbEncode.out"
+#define VOAMRWB_RFC3267_HEADER_INFO "#!AMR-WB\n"
+
+constexpr int32_t kInputBufferSize = 640;
+constexpr int32_t kOutputBufferSize = 1024;
+
+static AmrwbEncTestEnvironment *gEnv = nullptr;
+
+class AmrwbEncoderTest : public ::testing::TestWithParam<tuple<string, int32_t, VOAMRWBFRAMETYPE>> {
+ public:
+ AmrwbEncoderTest() : mEncoderHandle(nullptr) {
+ tuple<string, int32_t, VOAMRWBFRAMETYPE> params = GetParam();
+ mInputFile = gEnv->getRes() + get<0>(params);
+ mMode = get<1>(params);
+ mFrameType = get<2>(params);
+ mMemOperator.Alloc = cmnMemAlloc;
+ mMemOperator.Copy = cmnMemCopy;
+ mMemOperator.Free = cmnMemFree;
+ mMemOperator.Set = cmnMemSet;
+ mMemOperator.Check = cmnMemCheck;
+
+ mUserData.memflag = VO_IMF_USERMEMOPERATOR;
+ mUserData.memData = (VO_PTR)(&mMemOperator);
+ }
+
+ ~AmrwbEncoderTest() {
+ if (mEncoderHandle) {
+ mEncoderHandle = nullptr;
+ }
+ }
+
+ string mInputFile;
+ unsigned char mOutputBuf[kOutputBufferSize];
+ unsigned char mInputBuf[kInputBufferSize];
+ VOAMRWBFRAMETYPE mFrameType;
+ VO_AUDIO_CODECAPI mApiHandle;
+ VO_MEM_OPERATOR mMemOperator;
+ VO_CODEC_INIT_USERDATA mUserData;
+ VO_HANDLE mEncoderHandle;
+ int32_t mMode;
+};
+
+TEST_P(AmrwbEncoderTest, CreateAmrwbEncoderTest) {
+ int32_t status = voGetAMRWBEncAPI(&mApiHandle);
+ ASSERT_EQ(status, VO_ERR_NONE) << "Failed to get api handle";
+
+ status = mApiHandle.Init(&mEncoderHandle, VO_AUDIO_CodingAMRWB, &mUserData);
+ ASSERT_EQ(status, VO_ERR_NONE) << "Failed to init AMRWB encoder";
+
+ status = mApiHandle.SetParam(mEncoderHandle, VO_PID_AMRWB_FRAMETYPE, &mFrameType);
+ ASSERT_EQ(status, VO_ERR_NONE) << "Failed to set AMRWB encoder frame type to " << mFrameType;
+
+ status = mApiHandle.SetParam(mEncoderHandle, VO_PID_AMRWB_MODE, &mMode);
+ ASSERT_EQ(status, VO_ERR_NONE) << "Failed to set AMRWB encoder mode to %d" << mMode;
+ ALOGV("AMR-WB encoder created successfully");
+
+ status = mApiHandle.Uninit(mEncoderHandle);
+ ASSERT_EQ(status, VO_ERR_NONE) << "Failed to delete AMRWB encoder";
+ ALOGV("AMR-WB encoder deleted successfully");
+}
+
+TEST_P(AmrwbEncoderTest, AmrwbEncodeTest) {
+ VO_CODECBUFFER inData;
+ VO_CODECBUFFER outData;
+ VO_AUDIO_OUTPUTINFO outFormat;
+
+ FILE *fpInput = fopen(mInputFile.c_str(), "rb");
+ ASSERT_NE(fpInput, nullptr) << "Error opening input file " << mInputFile;
+
+ FILE *fpOutput = fopen(OUTPUT_FILE, "wb");
+ ASSERT_NE(fpOutput, nullptr) << "Error opening output file " << OUTPUT_FILE;
+
+ uint32_t status = voGetAMRWBEncAPI(&mApiHandle);
+ ASSERT_EQ(status, VO_ERR_NONE) << "Failed to get api handle";
+
+ status = mApiHandle.Init(&mEncoderHandle, VO_AUDIO_CodingAMRWB, &mUserData);
+ ASSERT_EQ(status, VO_ERR_NONE) << "Failed to init AMRWB encoder";
+
+ status = mApiHandle.SetParam(mEncoderHandle, VO_PID_AMRWB_FRAMETYPE, &mFrameType);
+ ASSERT_EQ(status, VO_ERR_NONE) << "Failed to set AMRWB encoder frame type to " << mFrameType;
+
+ status = mApiHandle.SetParam(mEncoderHandle, VO_PID_AMRWB_MODE, &mMode);
+ ASSERT_EQ(status, VO_ERR_NONE) << "Failed to set AMRWB encoder mode to " << mMode;
+
+ if (mFrameType == VOAMRWB_RFC3267) {
+ /* write RFC3267 Header info to indicate single channel AMR file storage format */
+ int32_t size = strlen(VOAMRWB_RFC3267_HEADER_INFO);
+ memcpy(mOutputBuf, VOAMRWB_RFC3267_HEADER_INFO, size);
+ fwrite(mOutputBuf, 1, size, fpOutput);
+ }
+
+ int32_t frameNum = 0;
+ while (1) {
+ int32_t buffLength =
+ (int32_t)fread(mInputBuf, sizeof(signed char), kInputBufferSize, fpInput);
+
+ if (buffLength == 0 || feof(fpInput)) break;
+ ASSERT_EQ(buffLength, kInputBufferSize) << "Error in reading input file";
+
+ inData.Buffer = (unsigned char *)mInputBuf;
+ inData.Length = buffLength;
+ outData.Buffer = mOutputBuf;
+ status = mApiHandle.SetInputData(mEncoderHandle, &inData);
+ ASSERT_EQ(status, VO_ERR_NONE) << "Failed to setup Input data";
+
+ do {
+ status = mApiHandle.GetOutputData(mEncoderHandle, &outData, &outFormat);
+ ASSERT_NE(status, VO_ERR_LICENSE_ERROR) << "Failed to encode the file";
+ if (status == 0) {
+ frameNum++;
+ fwrite(outData.Buffer, 1, outData.Length, fpOutput);
+ fflush(fpOutput);
+ }
+ } while (status != VO_ERR_INPUT_BUFFER_SMALL);
+ }
+
+ ALOGV("Number of frames processed: %d", frameNum);
+ status = mApiHandle.Uninit(mEncoderHandle);
+ ASSERT_EQ(status, VO_ERR_NONE) << "Failed to delete AMRWB encoder";
+
+ if (fpInput) {
+ fclose(fpInput);
+ }
+ if (fpOutput) {
+ fclose(fpOutput);
+ }
+}
+
+INSTANTIATE_TEST_SUITE_P(
+ AmrwbEncoderTestAll, AmrwbEncoderTest,
+ ::testing::Values(
+ make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD66, VOAMRWB_DEFAULT),
+ make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD885, VOAMRWB_DEFAULT),
+ make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1265, VOAMRWB_DEFAULT),
+ make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1425, VOAMRWB_DEFAULT),
+ make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1585, VOAMRWB_DEFAULT),
+ make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1825, VOAMRWB_DEFAULT),
+ make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1985, VOAMRWB_DEFAULT),
+ make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD2305, VOAMRWB_DEFAULT),
+ make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD2385, VOAMRWB_DEFAULT),
+ make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD66, VOAMRWB_ITU),
+ make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD885, VOAMRWB_ITU),
+ make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1265, VOAMRWB_ITU),
+ make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1425, VOAMRWB_ITU),
+ make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1585, VOAMRWB_ITU),
+ make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1825, VOAMRWB_ITU),
+ make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1985, VOAMRWB_ITU),
+ make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD2305, VOAMRWB_ITU),
+ make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD2385, VOAMRWB_ITU),
+ make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD66, VOAMRWB_RFC3267),
+ make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD885, VOAMRWB_RFC3267),
+ make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1265, VOAMRWB_RFC3267),
+ make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1425, VOAMRWB_RFC3267),
+ make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1585, VOAMRWB_RFC3267),
+ make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1825, VOAMRWB_RFC3267),
+ make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD1985, VOAMRWB_RFC3267),
+ make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD2305, VOAMRWB_RFC3267),
+ make_tuple("bbb_raw_1ch_16khz_s16le.raw", VOAMRWB_MD2385, VOAMRWB_RFC3267)));
+
+int main(int argc, char **argv) {
+ gEnv = new AmrwbEncTestEnvironment();
+ ::testing::AddGlobalTestEnvironment(gEnv);
+ ::testing::InitGoogleTest(&argc, argv);
+ int status = gEnv->initFromOptions(argc, argv);
+ if (status == 0) {
+ status = RUN_ALL_TESTS();
+ ALOGV("Test result = %d\n", status);
+ }
+ return status;
+}
diff --git a/media/libstagefright/codecs/amrwbenc/test/Android.bp b/media/libstagefright/codecs/amrwbenc/test/Android.bp
new file mode 100644
index 0000000..7042bc5
--- /dev/null
+++ b/media/libstagefright/codecs/amrwbenc/test/Android.bp
@@ -0,0 +1,48 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+cc_test {
+ name: "AmrwbEncoderTest",
+ gtest: true,
+
+ srcs: [
+ "AmrwbEncoderTest.cpp",
+ ],
+
+ static_libs: [
+ "libstagefright_enc_common",
+ "libstagefright_amrwbenc",
+ "libaudioutils",
+ "libsndfile",
+ ],
+
+ shared_libs: [
+ "liblog",
+ ],
+
+ cflags: [
+ "-Werror",
+ "-Wall",
+ ],
+
+ sanitize: {
+ cfi: true,
+ misc_undefined: [
+ "unsigned-integer-overflow",
+ "signed-integer-overflow",
+ ],
+ },
+}
diff --git a/media/libstagefright/codecs/amrwbenc/test/AndroidTest.xml b/media/libstagefright/codecs/amrwbenc/test/AndroidTest.xml
new file mode 100644
index 0000000..46f147c
--- /dev/null
+++ b/media/libstagefright/codecs/amrwbenc/test/AndroidTest.xml
@@ -0,0 +1,31 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!-- Copyright (C) 2020 The Android Open Source Project
+
+ Licensed under the Apache License, Version 2.0 (the "License");
+ you may not use this file except in compliance with the License.
+ You may obtain a copy of the License at
+
+ http://www.apache.org/licenses/LICENSE-2.0
+
+ Unless required by applicable law or agreed to in writing, software
+ distributed under the License is distributed on an "AS IS" BASIS,
+ WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ See the License for the specific language governing permissions and
+ limitations under the License.
+-->
+<configuration description="Test module config for Amr-wb Encoder unit test">
+ <option name="test-suite-tag" value="AmrwbEncoderTest" />
+ <target_preparer class="com.android.tradefed.targetprep.PushFilePreparer">
+ <option name="cleanup" value="true" />
+ <option name="push" value="AmrwbEncoderTest->/data/local/tmp/AmrwbEncoderTest" />
+ <option name="push-file"
+ key="https://storage.googleapis.com/android_media/frameworks/av/media/libstagefright/codecs/amrwbenc/test/AmrwbEncoderTest.zip?unzip=true"
+ value="/data/local/tmp/AmrwbEncoderTestRes/" />
+ </target_preparer>
+
+ <test class="com.android.tradefed.testtype.GTest" >
+ <option name="native-test-device-path" value="/data/local/tmp" />
+ <option name="module-name" value="AmrwbEncoderTest" />
+ <option name="native-test-flag" value="-P /data/local/tmp/AmrwbEncoderTestRes/" />
+ </test>
+</configuration>
diff --git a/media/libstagefright/codecs/amrwbenc/test/README.md b/media/libstagefright/codecs/amrwbenc/test/README.md
new file mode 100644
index 0000000..78762cb
--- /dev/null
+++ b/media/libstagefright/codecs/amrwbenc/test/README.md
@@ -0,0 +1,39 @@
+## Media Testing ##
+---
+#### AMR-WB Encoder :
+The Amr-Wb Encoder Test Suite validates the amrwb encoder available in libstagefright.
+
+Run the following steps to build the test suite:
+```
+m AmrwbEncoderTest
+```
+
+The 32-bit binaries will be created in the following path : ${OUT}/data/nativetest/
+
+The 64-bit binaries will be created in the following path : ${OUT}/data/nativetest64/
+
+To test 64-bit binary push binaries from nativetest64.
+```
+adb push ${OUT}/data/nativetest64/AmrwbEncoderTest/AmrwbEncoderTest /data/local/tmp/
+```
+
+To test 32-bit binary push binaries from nativetest.
+```
+adb push ${OUT}/data/nativetest/AmrwbEncoderTest/AmrwbEncoderTest /data/local/tmp/
+```
+
+The resource file for the tests is taken from [here](https://storage.googleapis.com/android_media/frameworks/av/media/libstagefright/codecs/amrwbenc/test/AmrwbEncoderTest.zip). Download, unzip and push these files into device for testing.
+
+```
+adb push AmrwbEncoderTestRes/. /data/local/tmp/
+```
+
+usage: AmrwbEncoderTest -P \<path_to_folder\>
+```
+adb shell /data/local/tmp/AmrwbEncoderTest -P /data/local/tmp/AmrwbEncoderTestRes/
+```
+Alternatively, the test can also be run using atest command.
+
+```
+atest AmrwbEncoderTest -- --enable-module-dynamic-download=true
+```
diff --git a/media/libstagefright/codecs/m4v_h263/dec/test/Android.bp b/media/libstagefright/codecs/m4v_h263/dec/test/Android.bp
new file mode 100644
index 0000000..e335c9b
--- /dev/null
+++ b/media/libstagefright/codecs/m4v_h263/dec/test/Android.bp
@@ -0,0 +1,47 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+cc_test {
+ name: "Mpeg4H263DecoderTest",
+ gtest: true,
+
+ srcs: [
+ "Mpeg4H263DecoderTest.cpp",
+ ],
+
+ shared_libs: [
+ "liblog",
+ ],
+
+ static_libs: [
+ "libstagefright_m4vh263dec",
+ "libstagefright_foundation",
+ ],
+
+ cflags: [
+ "-DOSCL_IMPORT_REF=",
+ "-Werror",
+ "-Wall",
+ ],
+
+ sanitize: {
+ misc_undefined: [
+ "unsigned-integer-overflow",
+ "signed-integer-overflow",
+ ],
+ cfi: true,
+ },
+}
diff --git a/media/libstagefright/codecs/m4v_h263/dec/test/AndroidTest.xml b/media/libstagefright/codecs/m4v_h263/dec/test/AndroidTest.xml
new file mode 100755
index 0000000..47e10ca
--- /dev/null
+++ b/media/libstagefright/codecs/m4v_h263/dec/test/AndroidTest.xml
@@ -0,0 +1,31 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!-- Copyright (C) 2020 The Android Open Source Project
+
+ Licensed under the Apache License, Version 2.0 (the "License");
+ you may not use this file except in compliance with the License.
+ You may obtain a copy of the License at
+
+ http://www.apache.org/licenses/LICENSE-2.0
+
+ Unless required by applicable law or agreed to in writing, software
+ distributed under the License is distributed on an "AS IS" BASIS,
+ WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ See the License for the specific language governing permissions and
+ limitations under the License.
+-->
+<configuration description="Test module config for Mpeg4H263 Decoder unit tests">
+ <option name="test-suite-tag" value="Mpeg4H263DecoderTest" />
+ <target_preparer class="com.android.tradefed.targetprep.PushFilePreparer">
+ <option name="cleanup" value="true" />
+ <option name="push" value="Mpeg4H263DecoderTest->/data/local/tmp/Mpeg4H263DecoderTest" />
+ <option name="push-file"
+ key="https://storage.googleapis.com/android_media/frameworks/av/media/libstagefright/codecs/m4v_h263/dec/test/Mpeg4H263Decoder.zip?unzip=true"
+ value="/data/local/tmp/Mpeg4H263DecoderTestRes/" />
+ </target_preparer>
+
+ <test class="com.android.tradefed.testtype.GTest" >
+ <option name="native-test-device-path" value="/data/local/tmp" />
+ <option name="module-name" value="Mpeg4H263DecoderTest" />
+ <option name="native-test-flag" value="-P /data/local/tmp/Mpeg4H263DecoderTestRes/" />
+ </test>
+</configuration>
diff --git a/media/libstagefright/codecs/m4v_h263/dec/test/Mpeg4H263DecoderTest.cpp b/media/libstagefright/codecs/m4v_h263/dec/test/Mpeg4H263DecoderTest.cpp
new file mode 100644
index 0000000..967c1ea
--- /dev/null
+++ b/media/libstagefright/codecs/m4v_h263/dec/test/Mpeg4H263DecoderTest.cpp
@@ -0,0 +1,423 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "Mpeg4H263DecoderTest"
+#include <utils/Log.h>
+
+#include <stdio.h>
+#include <string.h>
+#include <utils/String8.h>
+#include <fstream>
+
+#include <media/stagefright/foundation/AUtils.h>
+#include "mp4dec_api.h"
+
+#include "Mpeg4H263DecoderTestEnvironment.h"
+
+using namespace android;
+
+#define OUTPUT_FILE_NAME "/data/local/tmp/Output.yuv"
+#define CODEC_CONFIG_FLAG 32
+#define SYNC_FRAME 1
+#define MPEG4_MAX_WIDTH 1920
+#define MPEG4_MAX_HEIGHT 1080
+#define H263_MAX_WIDTH 352
+#define H263_MAX_HEIGHT 288
+
+constexpr uint32_t kNumOutputBuffers = 2;
+
+struct FrameInfo {
+ int32_t bytesCount;
+ uint32_t flags;
+ int64_t timestamp;
+};
+
+struct tagvideoDecControls;
+
+static Mpeg4H263DecoderTestEnvironment *gEnv = nullptr;
+
+class Mpeg4H263DecoderTest : public ::testing::TestWithParam<tuple<string, string, bool>> {
+ public:
+ Mpeg4H263DecoderTest()
+ : mDecHandle(nullptr),
+ mInputBuffer(nullptr),
+ mInitialized(false),
+ mFramesConfigured(false),
+ mNumSamplesOutput(0),
+ mWidth(352),
+ mHeight(288) {
+ memset(mOutputBuffer, 0x0, sizeof(mOutputBuffer));
+ }
+
+ ~Mpeg4H263DecoderTest() {
+ if (mEleStream.is_open()) mEleStream.close();
+ if (mDecHandle) {
+ delete mDecHandle;
+ mDecHandle = nullptr;
+ }
+ if (mInputBuffer) {
+ free(mInputBuffer);
+ mInputBuffer = nullptr;
+ }
+ freeOutputBuffer();
+ }
+
+ status_t initDecoder();
+ void allocOutputBuffer(size_t outputBufferSize);
+ void dumpOutput(ofstream &ostrm);
+ void freeOutputBuffer();
+ void processMpeg4H263Decoder(vector<FrameInfo> Info, int32_t offset, int32_t range,
+ ifstream &mEleStream, ofstream &ostrm, MP4DecodingMode inputMode);
+ void deInitDecoder();
+
+ ifstream mEleStream;
+ tagvideoDecControls *mDecHandle;
+ char *mInputBuffer;
+ uint8_t *mOutputBuffer[kNumOutputBuffers];
+ bool mInitialized;
+ bool mFramesConfigured;
+ uint32_t mNumSamplesOutput;
+ uint32_t mWidth;
+ uint32_t mHeight;
+};
+
+status_t Mpeg4H263DecoderTest::initDecoder() {
+ if (!mDecHandle) {
+ mDecHandle = new tagvideoDecControls;
+ }
+ if (!mDecHandle) {
+ return NO_MEMORY;
+ }
+ memset(mDecHandle, 0, sizeof(tagvideoDecControls));
+
+ return OK;
+}
+
+void Mpeg4H263DecoderTest::allocOutputBuffer(size_t outputBufferSize) {
+ for (int32_t i = 0; i < kNumOutputBuffers; ++i) {
+ if (!mOutputBuffer[i]) {
+ mOutputBuffer[i] = (uint8_t *)malloc(outputBufferSize);
+ ASSERT_NE(mOutputBuffer[i], nullptr) << "Output buffer allocation failed";
+ }
+ }
+}
+
+void Mpeg4H263DecoderTest::dumpOutput(ofstream &ostrm) {
+ uint8_t *src = mOutputBuffer[mNumSamplesOutput & 1];
+ size_t vStride = align(mHeight, 16);
+ size_t srcYStride = align(mWidth, 16);
+ size_t srcUVStride = srcYStride / 2;
+ uint8_t *srcStart = src;
+
+ /* Y buffer */
+ for (size_t i = 0; i < mHeight; ++i) {
+ ostrm.write(reinterpret_cast<char *>(src), mWidth);
+ src += srcYStride;
+ }
+ /* U buffer */
+ src = srcStart + vStride * srcYStride;
+ for (size_t i = 0; i < mHeight / 2; ++i) {
+ ostrm.write(reinterpret_cast<char *>(src), mWidth / 2);
+ src += srcUVStride;
+ }
+ /* V buffer */
+ src = srcStart + vStride * srcYStride * 5 / 4;
+ for (size_t i = 0; i < mHeight / 2; ++i) {
+ ostrm.write(reinterpret_cast<char *>(src), mWidth / 2);
+ src += srcUVStride;
+ }
+}
+
+void Mpeg4H263DecoderTest::freeOutputBuffer() {
+ for (int32_t i = 0; i < kNumOutputBuffers; ++i) {
+ if (mOutputBuffer[i]) {
+ free(mOutputBuffer[i]);
+ mOutputBuffer[i] = nullptr;
+ }
+ }
+}
+
+void Mpeg4H263DecoderTest::processMpeg4H263Decoder(vector<FrameInfo> Info, int32_t offset,
+ int32_t range, ifstream &mEleStream,
+ ofstream &ostrm, MP4DecodingMode inputMode) {
+ size_t maxWidth = (inputMode == MPEG4_MODE) ? MPEG4_MAX_WIDTH : H263_MAX_WIDTH;
+ size_t maxHeight = (inputMode == MPEG4_MODE) ? MPEG4_MAX_HEIGHT : H263_MAX_HEIGHT;
+ size_t outputBufferSize = align(maxWidth, 16) * align(maxHeight, 16) * 3 / 2;
+ uint32_t frameIndex = offset;
+ bool status = true;
+ ASSERT_GE(range, 0) << "Invalid range";
+ ASSERT_TRUE(offset >= 0 && offset <= Info.size() - 1) << "Invalid offset";
+ ASSERT_LE(range + offset, Info.size()) << "range+offset can't be greater than the no of frames";
+
+ while (1) {
+ if (frameIndex == Info.size() || frameIndex == (offset + range)) break;
+
+ int32_t bytesCount = Info[frameIndex].bytesCount;
+ ASSERT_GT(bytesCount, 0) << "Size for the memory allocation is negative";
+ mInputBuffer = (char *)malloc(bytesCount);
+ ASSERT_NE(mInputBuffer, nullptr) << "Insufficient memory to read frame";
+ mEleStream.read(mInputBuffer, bytesCount);
+ ASSERT_EQ(mEleStream.gcount(), bytesCount) << "mEleStream.gcount() != bytesCount";
+ static const uint8_t volInfo[] = {0x00, 0x00, 0x01, 0xB0};
+ bool volHeader = memcmp(mInputBuffer, volInfo, 4) == 0;
+ if (volHeader) {
+ PVCleanUpVideoDecoder(mDecHandle);
+ mInitialized = false;
+ }
+
+ if (!mInitialized) {
+ uint8_t *volData[1]{};
+ int32_t volSize = 0;
+
+ uint32_t flags = Info[frameIndex].flags;
+ bool codecConfig = flags == CODEC_CONFIG_FLAG;
+ if (codecConfig || volHeader) {
+ volData[0] = reinterpret_cast<uint8_t *>(mInputBuffer);
+ volSize = bytesCount;
+ }
+
+ status = PVInitVideoDecoder(mDecHandle, volData, &volSize, 1, maxWidth, maxHeight,
+ inputMode);
+ ASSERT_TRUE(status) << "PVInitVideoDecoder failed. Unsupported content";
+
+ mInitialized = true;
+ MP4DecodingMode actualMode = PVGetDecBitstreamMode(mDecHandle);
+ ASSERT_EQ(inputMode, actualMode)
+ << "Decoded mode not same as actual mode of the decoder";
+
+ PVSetPostProcType(mDecHandle, 0);
+
+ int32_t dispWidth, dispHeight;
+ PVGetVideoDimensions(mDecHandle, &dispWidth, &dispHeight);
+
+ int32_t bufWidth, bufHeight;
+ PVGetBufferDimensions(mDecHandle, &bufWidth, &bufHeight);
+
+ ASSERT_LE(dispWidth, bufWidth) << "Display width is greater than buffer width";
+ ASSERT_LE(dispHeight, bufHeight) << "Display height is greater than buffer height";
+
+ if (dispWidth != mWidth || dispHeight != mHeight) {
+ mWidth = dispWidth;
+ mHeight = dispHeight;
+ freeOutputBuffer();
+ if (inputMode == H263_MODE) {
+ PVCleanUpVideoDecoder(mDecHandle);
+
+ uint8_t *volData[1]{};
+ int32_t volSize = 0;
+
+ status = PVInitVideoDecoder(mDecHandle, volData, &volSize, 1, maxWidth,
+ maxHeight, H263_MODE);
+ ASSERT_TRUE(status) << "PVInitVideoDecoder failed for H263";
+ }
+ mFramesConfigured = false;
+ }
+
+ if (codecConfig) {
+ frameIndex++;
+ continue;
+ }
+ }
+
+ uint32_t yFrameSize = sizeof(uint8) * mDecHandle->size;
+ ASSERT_GE(outputBufferSize, yFrameSize * 3 / 2)
+ << "Too small output buffer: " << outputBufferSize << " bytes";
+ ASSERT_NO_FATAL_FAILURE(allocOutputBuffer(outputBufferSize));
+
+ if (!mFramesConfigured) {
+ PVSetReferenceYUV(mDecHandle, mOutputBuffer[1]);
+ mFramesConfigured = true;
+ }
+
+ // Need to check if header contains new info, e.g., width/height, etc.
+ VopHeaderInfo headerInfo;
+ uint32_t useExtTimestamp = 1;
+ int32_t inputSize = (Info)[frameIndex].bytesCount;
+ uint32_t timestamp = frameIndex;
+
+ uint8_t *bitstreamTmp = reinterpret_cast<uint8_t *>(mInputBuffer);
+
+ status = PVDecodeVopHeader(mDecHandle, &bitstreamTmp, ×tamp, &inputSize, &headerInfo,
+ &useExtTimestamp, mOutputBuffer[mNumSamplesOutput & 1]);
+ ASSERT_EQ(status, PV_TRUE) << "failed to decode vop header";
+
+ // H263 doesn't have VOL header, the frame size information is in short header, i.e. the
+ // decoder may detect size change after PVDecodeVopHeader.
+ int32_t dispWidth, dispHeight;
+ PVGetVideoDimensions(mDecHandle, &dispWidth, &dispHeight);
+
+ int32_t bufWidth, bufHeight;
+ PVGetBufferDimensions(mDecHandle, &bufWidth, &bufHeight);
+
+ ASSERT_LE(dispWidth, bufWidth) << "Display width is greater than buffer width";
+ ASSERT_LE(dispHeight, bufHeight) << "Display height is greater than buffer height";
+ if (dispWidth != mWidth || dispHeight != mHeight) {
+ mWidth = dispWidth;
+ mHeight = dispHeight;
+ }
+
+ status = PVDecodeVopBody(mDecHandle, &inputSize);
+ ASSERT_EQ(status, PV_TRUE) << "failed to decode video frame No = %d" << frameIndex;
+
+ dumpOutput(ostrm);
+
+ ++mNumSamplesOutput;
+ ++frameIndex;
+ }
+ freeOutputBuffer();
+}
+
+void Mpeg4H263DecoderTest::deInitDecoder() {
+ if (mInitialized) {
+ if (mDecHandle) {
+ PVCleanUpVideoDecoder(mDecHandle);
+ delete mDecHandle;
+ mDecHandle = nullptr;
+ }
+ mInitialized = false;
+ }
+ freeOutputBuffer();
+}
+
+void getInfo(string infoFileName, vector<FrameInfo> &Info) {
+ ifstream eleInfo;
+ eleInfo.open(infoFileName);
+ ASSERT_EQ(eleInfo.is_open(), true) << "Failed to open " << infoFileName;
+ int32_t bytesCount = 0;
+ uint32_t flags = 0;
+ uint32_t timestamp = 0;
+ while (1) {
+ if (!(eleInfo >> bytesCount)) {
+ break;
+ }
+ eleInfo >> flags;
+ eleInfo >> timestamp;
+ Info.push_back({bytesCount, flags, timestamp});
+ }
+ if (eleInfo.is_open()) eleInfo.close();
+}
+
+TEST_P(Mpeg4H263DecoderTest, DecodeTest) {
+ tuple<string /* InputFileName */, string /* InfoFileName */, bool /* mode */> params =
+ GetParam();
+
+ string inputFileName = gEnv->getRes() + get<0>(params);
+ mEleStream.open(inputFileName, ifstream::binary);
+ ASSERT_EQ(mEleStream.is_open(), true) << "Failed to open " << get<0>(params);
+
+ string infoFileName = gEnv->getRes() + get<1>(params);
+ vector<FrameInfo> Info;
+ ASSERT_NO_FATAL_FAILURE(getInfo(infoFileName, Info));
+ ASSERT_NE(Info.empty(), true) << "Invalid Info file";
+
+ ofstream ostrm;
+ ostrm.open(OUTPUT_FILE_NAME, std::ofstream::binary);
+ ASSERT_EQ(ostrm.is_open(), true) << "Failed to open output stream for " << get<0>(params);
+
+ status_t err = initDecoder();
+ ASSERT_EQ(err, OK) << "initDecoder: failed to create decoder " << err;
+
+ bool isMpeg4 = get<2>(params);
+ MP4DecodingMode inputMode = isMpeg4 ? MPEG4_MODE : H263_MODE;
+ ASSERT_NO_FATAL_FAILURE(
+ processMpeg4H263Decoder(Info, 0, Info.size(), mEleStream, ostrm, inputMode));
+ deInitDecoder();
+ ostrm.close();
+ Info.clear();
+}
+
+TEST_P(Mpeg4H263DecoderTest, FlushTest) {
+ tuple<string /* InputFileName */, string /* InfoFileName */, bool /* mode */> params =
+ GetParam();
+
+ string inputFileName = gEnv->getRes() + get<0>(params);
+ mEleStream.open(inputFileName, ifstream::binary);
+ ASSERT_EQ(mEleStream.is_open(), true) << "Failed to open " << get<0>(params);
+
+ string infoFileName = gEnv->getRes() + get<1>(params);
+ vector<FrameInfo> Info;
+ ASSERT_NO_FATAL_FAILURE(getInfo(infoFileName, Info));
+ ASSERT_NE(Info.empty(), true) << "Invalid Info file";
+
+ ofstream ostrm;
+ ostrm.open(OUTPUT_FILE_NAME, std::ofstream::binary);
+ ASSERT_EQ(ostrm.is_open(), true) << "Failed to open output stream for " << get<0>(params);
+
+ status_t err = initDecoder();
+ ASSERT_EQ(err, OK) << "initDecoder: failed to create decoder " << err;
+
+ bool isMpeg4 = get<2>(params);
+ MP4DecodingMode inputMode = isMpeg4 ? MPEG4_MODE : H263_MODE;
+ // Number of frames to be decoded before flush
+ int32_t numFrames = Info.size() / 3;
+ ASSERT_NO_FATAL_FAILURE(
+ processMpeg4H263Decoder(Info, 0, numFrames, mEleStream, ostrm, inputMode));
+
+ if (mInitialized) {
+ int32_t status = PVResetVideoDecoder(mDecHandle);
+ ASSERT_EQ(status, PV_TRUE);
+ }
+
+ // Seek to next key frame and start decoding till the end
+ int32_t index = numFrames;
+ bool keyFrame = false;
+ uint32_t flags = 0;
+ while (index < (int32_t)Info.size()) {
+ if (Info[index].flags) flags = 1u << (Info[index].flags - 1);
+ if ((flags & SYNC_FRAME) == SYNC_FRAME) {
+ keyFrame = true;
+ break;
+ }
+ flags = 0;
+ mEleStream.ignore(Info[index].bytesCount);
+ index++;
+ }
+ ALOGV("Index= %d", index);
+ if (keyFrame) {
+ mNumSamplesOutput = 0;
+ ASSERT_NO_FATAL_FAILURE(processMpeg4H263Decoder(Info, index, (int32_t)Info.size() - index,
+ mEleStream, ostrm, inputMode));
+ }
+ deInitDecoder();
+ ostrm.close();
+ Info.clear();
+}
+
+INSTANTIATE_TEST_SUITE_P(
+ Mpeg4H263DecoderTestAll, Mpeg4H263DecoderTest,
+ ::testing::Values(make_tuple("swirl_128x96_h263.h263", "swirl_128x96_h263.info", false),
+ make_tuple("swirl_176x144_h263.h263", "swirl_176x144_h263.info", false),
+ make_tuple("swirl_352x288_h263.h263", "swirl_352x288_h263.info", false),
+ make_tuple("bbb_352x288_h263.h263", "bbb_352x288_h263.info", false),
+ make_tuple("bbb_352x288_mpeg4.m4v", "bbb_352x288_mpeg4.info", true),
+ make_tuple("swirl_128x128_mpeg4.m4v", "swirl_128x128_mpeg4.info", true),
+ make_tuple("swirl_130x132_mpeg4.m4v", "swirl_130x132_mpeg4.info", true),
+ make_tuple("swirl_132x130_mpeg4.m4v", "swirl_132x130_mpeg4.info", true),
+ make_tuple("swirl_136x144_mpeg4.m4v", "swirl_136x144_mpeg4.info", true),
+ make_tuple("swirl_144x136_mpeg4.m4v", "swirl_144x136_mpeg4.info", true)));
+
+int main(int argc, char **argv) {
+ gEnv = new Mpeg4H263DecoderTestEnvironment();
+ ::testing::AddGlobalTestEnvironment(gEnv);
+ ::testing::InitGoogleTest(&argc, argv);
+ int status = gEnv->initFromOptions(argc, argv);
+ if (status == 0) {
+ status = RUN_ALL_TESTS();
+ ALOGD("Decoder Test Result = %d\n", status);
+ }
+ return status;
+}
diff --git a/media/libstagefright/codecs/m4v_h263/dec/test/Mpeg4H263DecoderTestEnvironment.h b/media/libstagefright/codecs/m4v_h263/dec/test/Mpeg4H263DecoderTestEnvironment.h
new file mode 100644
index 0000000..f085845
--- /dev/null
+++ b/media/libstagefright/codecs/m4v_h263/dec/test/Mpeg4H263DecoderTestEnvironment.h
@@ -0,0 +1,73 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __MPEG4_H263_DECODER_TEST_ENVIRONMENT_H__
+#define __MPEG4_H263_DECODER_TEST_ENVIRONMENT_H__
+
+#include <gtest/gtest.h>
+
+#include <getopt.h>
+
+using namespace std;
+
+class Mpeg4H263DecoderTestEnvironment : public ::testing::Environment {
+ public:
+ Mpeg4H263DecoderTestEnvironment() : res("/data/local/tmp/") {}
+
+ // Parses the command line arguments
+ int initFromOptions(int argc, char **argv);
+
+ void setRes(const char *_res) { res = _res; }
+
+ const string getRes() const { return res; }
+
+ private:
+ string res;
+};
+
+int Mpeg4H263DecoderTestEnvironment::initFromOptions(int argc, char **argv) {
+ static struct option options[] = {{"path", required_argument, 0, 'P'}, {0, 0, 0, 0}};
+
+ while (true) {
+ int index = 0;
+ int c = getopt_long(argc, argv, "P:", options, &index);
+ if (c == -1) {
+ break;
+ }
+
+ switch (c) {
+ case 'P': {
+ setRes(optarg);
+ break;
+ }
+ default:
+ break;
+ }
+ }
+
+ if (optind < argc) {
+ fprintf(stderr,
+ "unrecognized option: %s\n\n"
+ "usage: %s <gtest options> <test options>\n\n"
+ "test options are:\n\n"
+ "-P, --path: Resource files directory location\n",
+ argv[optind ?: 1], argv[0]);
+ return 2;
+ }
+ return 0;
+}
+
+#endif // __MPEG4_H263_DECODER_TEST_ENVIRONMENT_H__
diff --git a/media/libstagefright/codecs/m4v_h263/dec/test/README.md b/media/libstagefright/codecs/m4v_h263/dec/test/README.md
new file mode 100644
index 0000000..7e4aea1
--- /dev/null
+++ b/media/libstagefright/codecs/m4v_h263/dec/test/README.md
@@ -0,0 +1,39 @@
+## Media Testing ##
+---
+#### Mpeg4H263Decoder :
+The Mpeg4H263Decoder Test Suite validates the Mpeg4 and H263 decoder available in libstagefright.
+
+Run the following steps to build the test suite:
+```
+m Mpeg4H263DecoderTest
+```
+
+The 32-bit binaries will be created in the following path : ${OUT}/data/nativetest/
+
+The 64-bit binaries will be created in the following path : ${OUT}/data/nativetest64/
+
+To test 64-bit binary push binaries from nativetest64.
+```
+adb push ${OUT}/data/nativetest64/Mpeg4H263DecoderTest/Mpeg4H263DecoderTest /data/local/tmp/
+```
+
+To test 32-bit binary push binaries from nativetest.
+```
+adb push ${OUT}/data/nativetest/Mpeg4H263DecoderTest/Mpeg4H263DecoderTest /data/local/tmp/
+```
+
+The resource file for the tests is taken from [here](https://storage.googleapis.com/android_media/frameworks/av/media/libstagefright/codecs/m4v_h263/dec/test/Mpeg4H263Decoder.zip). Download, unzip and push these files into device for testing.
+
+```
+adb push Mpeg4H263Decoder /data/local/tmp/
+```
+
+usage: Mpeg4H263DecoderTest -P \<path_to_folder\>
+```
+adb shell /data/local/tmp/Mpeg4H263DecoderTest -P /data/local/tmp/Mpeg4H263Decoder/
+```
+Alternatively, the test can also be run using atest command.
+
+```
+atest Mpeg4H263DecoderTest -- --enable-module-dynamic-download=true
+```
diff --git a/media/libstagefright/codecs/m4v_h263/enc/test/Android.bp b/media/libstagefright/codecs/m4v_h263/enc/test/Android.bp
new file mode 100644
index 0000000..b9a8117
--- /dev/null
+++ b/media/libstagefright/codecs/m4v_h263/enc/test/Android.bp
@@ -0,0 +1,45 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+cc_test {
+ name: "Mpeg4H263EncoderTest",
+ gtest: true,
+
+ srcs : [ "Mpeg4H263EncoderTest.cpp" ],
+
+ shared_libs: [
+ "libutils",
+ "liblog",
+ ],
+
+ static_libs: [
+ "libstagefright_m4vh263enc",
+ ],
+
+ cflags: [
+ "-DOSCL_IMPORT_REF=",
+ "-Wall",
+ "-Werror",
+ ],
+
+ sanitize: {
+ misc_undefined: [
+ "signed-integer-overflow",
+ "unsigned-integer-overflow",
+ ],
+ cfi: true,
+ },
+}
diff --git a/media/libstagefright/codecs/m4v_h263/enc/test/AndroidTest.xml b/media/libstagefright/codecs/m4v_h263/enc/test/AndroidTest.xml
new file mode 100644
index 0000000..5218932
--- /dev/null
+++ b/media/libstagefright/codecs/m4v_h263/enc/test/AndroidTest.xml
@@ -0,0 +1,31 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!-- Copyright (C) 2020 The Android Open Source Project
+
+ Licensed under the Apache License, Version 2.0 (the "License");
+ you may not use this file except in compliance with the License.
+ You may obtain a copy of the License at
+
+ http://www.apache.org/licenses/LICENSE-2.0
+
+ Unless required by applicable law or agreed to in writing, software
+ distributed under the License is distributed on an "AS IS" BASIS,
+ WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ See the License for the specific language governing permissions and
+ limitations under the License.
+-->
+<configuration description="Test module config for MPEG4H263 encoder unit tests">
+ <option name="test-suite-tag" value="Mpeg4H263EncoderTest" />
+ <target_preparer class="com.android.tradefed.targetprep.PushFilePreparer">
+ <option name="cleanup" value="true" />
+ <option name="push" value="Mpeg4H263EncoderTest->/data/local/tmp/Mpeg4H263EncoderTest/" />
+ <option name="push-file"
+ key="https://storage.googleapis.com/android_media/frameworks/av/media/libstagefright/codecs/m4v_h263/enc/test/Mpeg4H263Encoder.zip?unzip=true"
+ value="/data/local/tmp/Mpeg4H263EncoderTestRes/" />
+ </target_preparer>
+
+ <test class="com.android.tradefed.testtype.GTest" >
+ <option name="native-test-device-path" value="/data/local/tmp" />
+ <option name="module-name" value="Mpeg4H263EncoderTest" />
+ <option name="native-test-flag" value="-P /data/local/tmp/Mpeg4H263EncoderTestRes/" />
+ </test>
+</configuration>
\ No newline at end of file
diff --git a/media/libstagefright/codecs/m4v_h263/enc/test/Mpeg4H263EncoderTest.cpp b/media/libstagefright/codecs/m4v_h263/enc/test/Mpeg4H263EncoderTest.cpp
new file mode 100644
index 0000000..78c705a
--- /dev/null
+++ b/media/libstagefright/codecs/m4v_h263/enc/test/Mpeg4H263EncoderTest.cpp
@@ -0,0 +1,250 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "Mpeg4H263EncoderTest"
+#include <utils/Log.h>
+
+#include <stdint.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <sys/stat.h>
+
+#include "mp4enc_api.h"
+
+#include "Mpeg4H263EncoderTestEnvironment.h"
+
+#define ENCODED_FILE "/data/local/tmp/Mpeg4H263Output"
+
+// assuming a worst case compression of 2X
+constexpr int16_t kCompressionRatio = 2;
+constexpr int8_t kIDRFrameRefreshIntervalInSec = 1;
+
+static Mpeg4H263EncoderTestEnvironment *gEnv = nullptr;
+
+class Mpeg4H263EncoderTest
+ : public ::testing::TestWithParam<tuple<string, bool, int32_t, int32_t, float, int32_t>> {
+ private:
+ void initEncoderParams();
+
+ public:
+ Mpeg4H263EncoderTest()
+ : mInputBuffer(nullptr),
+ mOutputBuffer(nullptr),
+ mFpInput(nullptr),
+ mFpOutput(nullptr),
+ mEncodeHandle(nullptr),
+ mEncodeControl(nullptr) {}
+
+ ~Mpeg4H263EncoderTest() {
+ if(mFpInput) {
+ fclose(mFpInput);
+ }
+ if(mFpOutput) {
+ fclose(mFpOutput);
+ }
+ if(mInputBuffer) free(mInputBuffer);
+ if(mOutputBuffer) free(mOutputBuffer);
+ if(mEncodeHandle) free(mEncodeHandle);
+ if(mEncodeControl) free(mEncodeControl);
+ }
+
+ void SetUp() override {
+ tuple<string /* fileName */, bool /* isMpeg4 */, int32_t /* frameWidth */,
+ int32_t /* frameHeight */, float /* frameRate */, int32_t /* bitRate */>
+ params = GetParam();
+ mFileName = gEnv->getRes() + get<0>(params);
+ mIsMpeg4 = get<1>(params);
+ mFrameWidth = get<2>(params);
+ mFrameHeight = get<3>(params);
+ mFrameRate = get<4>(params);
+ mBitRate = get<5>(params);
+
+ ASSERT_TRUE(mFrameWidth % 16 == 0) << "Frame Width should be multiple of 16";
+ ASSERT_TRUE(mFrameHeight % 16 == 0) << "Frame Height should be multiple of 16";
+ ASSERT_LE(mFrameWidth, (mIsMpeg4 ? 720 : 352))
+ << "Frame Width <= 720 for Mpeg4 and <= 352 for H263";
+ ASSERT_LE(mFrameHeight, (mIsMpeg4 ? 480 : 288))
+ << "Frame Height <= 480 for Mpeg4 and <= 288 for H263";
+ ASSERT_LE(mFrameRate, 30) << "Frame rate less than or equal to 30";
+ ASSERT_LE(mBitRate, 2048) << "Bit rate less than or equal to 2048 kbps";
+
+ mOutputBufferSize = ( mFrameWidth * mFrameHeight * 3/2 ) / kCompressionRatio;
+ mEncodeHandle = new VideoEncOptions;
+ ASSERT_NE(mEncodeHandle, nullptr) << "Failed to get Video Encoding options object";
+ memset(mEncodeHandle, 0, sizeof(VideoEncOptions));
+ mEncodeControl = new VideoEncControls;
+ ASSERT_NE(mEncodeControl, nullptr) << "Failed to get Video Encoding control object";
+ memset(mEncodeControl, 0, sizeof(VideoEncControls));
+ ASSERT_NO_FATAL_FAILURE(initEncoderParams())
+ << "Failed to get the default Encoding parameters!";
+ }
+
+ int64_t getTotalFrames();
+ void processEncoder(int32_t);
+ bool mIsMpeg4;
+ int32_t mFrameWidth, mFrameHeight, mBitRate;
+ int64_t mOutputBufferSize;
+ float mFrameRate;
+ string mFileName;
+ uint8_t *mInputBuffer, *mOutputBuffer;
+ FILE *mFpInput, *mFpOutput;
+ VideoEncOptions *mEncodeHandle;
+ VideoEncControls *mEncodeControl;
+};
+
+void Mpeg4H263EncoderTest::initEncoderParams() {
+ bool status = PVGetDefaultEncOption(mEncodeHandle, 0);
+ ASSERT_TRUE(status);
+
+ mEncodeHandle->rcType = VBR_1;
+ mEncodeHandle->vbvDelay = 5.0f;
+ mEncodeHandle->profile_level = CORE_PROFILE_LEVEL2;
+ mEncodeHandle->packetSize = 32;
+ mEncodeHandle->rvlcEnable = PV_OFF;
+ mEncodeHandle->numLayers = 1;
+ mEncodeHandle->timeIncRes = 1000;
+ mEncodeHandle->iQuant[0] = 15;
+ mEncodeHandle->pQuant[0] = 12;
+ mEncodeHandle->quantType[0] = 0;
+ mEncodeHandle->noFrameSkipped = PV_OFF;
+ mEncodeHandle->numIntraMB = 0;
+ mEncodeHandle->sceneDetect = PV_ON;
+ mEncodeHandle->searchRange = 16;
+ mEncodeHandle->mv8x8Enable = PV_OFF;
+ mEncodeHandle->gobHeaderInterval = 0;
+ mEncodeHandle->useACPred = PV_ON;
+ mEncodeHandle->intraDCVlcTh = 0;
+ if(!mIsMpeg4) {
+ mEncodeHandle->encMode = H263_MODE;
+ } else {
+ mEncodeHandle->encMode = COMBINE_MODE_WITH_ERR_RES;
+ }
+ mEncodeHandle->encWidth[0] = mFrameWidth;
+ mEncodeHandle->encHeight[0] = mFrameHeight;
+ mEncodeHandle->encFrameRate[0] = mFrameRate;
+ mEncodeHandle->bitRate[0] = mBitRate * 1024;
+ mEncodeHandle->tickPerSrc = mEncodeHandle->timeIncRes / mFrameRate;
+ if (kIDRFrameRefreshIntervalInSec == 0) {
+ // All I frames.
+ mEncodeHandle->intraPeriod = 1;
+ } else {
+ mEncodeHandle->intraPeriod = (kIDRFrameRefreshIntervalInSec * mFrameRate);
+ }
+}
+
+int64_t Mpeg4H263EncoderTest::getTotalFrames() {
+ int32_t frameSize = (mFrameWidth * mFrameHeight * 3) / 2;
+ struct stat buf;
+ stat(mFileName.c_str(), &buf);
+ size_t fileSize = buf.st_size;
+ int64_t totalFrames = (int64_t)(fileSize/frameSize);
+ return totalFrames;
+}
+
+void Mpeg4H263EncoderTest::processEncoder(int32_t numFramesToEncode) {
+ bool status;
+ int64_t numEncodedFrames = 0;
+ int32_t bytesRead;
+ int32_t frameSize = (mFrameWidth * mFrameHeight * 3) / 2;
+ while(numFramesToEncode != 0) {
+ bytesRead = fread(mInputBuffer, 1, frameSize, mFpInput);
+ // End of file.
+ if (bytesRead != frameSize) {
+ break;
+ }
+
+ VideoEncFrameIO videoIn, videoOut;
+ videoIn.height = mFrameHeight;
+ videoIn.pitch = mFrameWidth;
+ videoIn.timestamp = (numEncodedFrames * 1000) / mFrameRate; // in ms.
+ videoIn.yChan = mInputBuffer;
+ videoIn.uChan = videoIn.yChan + videoIn.height * videoIn.pitch;
+ videoIn.vChan = videoIn.uChan + ((videoIn.height * videoIn.pitch) >> 2);
+ uint32_t modTimeMs = 0;
+ int32_t dataLength = mOutputBufferSize;
+ int32_t nLayer = 0;
+ status = PVEncodeVideoFrame(mEncodeControl, &videoIn, &videoOut, &modTimeMs, mOutputBuffer,
+ &dataLength, &nLayer);
+ ASSERT_TRUE(status) << "Failed to Encode: " << mFileName;
+
+ MP4HintTrack hintTrack;
+ status = PVGetHintTrack(mEncodeControl, &hintTrack);
+ ASSERT_TRUE(status) << "Failed to get hint track!";
+ UChar *overrunBuffer = PVGetOverrunBuffer(mEncodeControl);
+ ASSERT_EQ(overrunBuffer, nullptr) << "Overrun of buffer!";
+
+ int64_t numBytes = fwrite(mOutputBuffer, 1, dataLength, mFpOutput);
+ ASSERT_EQ(numBytes, dataLength) << "Failed to write to the output file!";
+ numEncodedFrames++;
+ numFramesToEncode--;
+ }
+}
+
+TEST_P(Mpeg4H263EncoderTest, EncodeTest) {
+ mInputBuffer = (uint8_t *)malloc((mFrameWidth * mFrameWidth * 3) / 2);
+ ASSERT_NE(mInputBuffer, nullptr) << "Failed to allocate the input buffer!";
+
+ mOutputBuffer = (uint8_t *)malloc(mOutputBufferSize);
+ ASSERT_NE(mOutputBuffer, nullptr) << "Failed to allocate the output buffer!";
+
+ mFpInput = fopen(mFileName.c_str(), "rb");
+ ASSERT_NE(mFpInput, nullptr) << "Failed to open the input file: " << mFileName;
+
+ mFpOutput = fopen(ENCODED_FILE, "wb");
+ ASSERT_NE(mFpOutput, nullptr) << "Failed to open the output file:" << ENCODED_FILE;
+
+ bool status = PVInitVideoEncoder(mEncodeControl, mEncodeHandle);
+ ASSERT_TRUE(status) << "Failed to initialize the encoder!";
+
+ // Get VOL header.
+ int32_t size = mOutputBufferSize;
+ status = PVGetVolHeader(mEncodeControl, mOutputBuffer, &size, 0);
+ ASSERT_TRUE(status) << "Failed to get the VOL header!";
+
+ // Write the VOL header on the first frame.
+ int32_t numBytes = fwrite(mOutputBuffer, 1, size, mFpOutput);
+ ASSERT_EQ(numBytes, size) << "Failed to write the VOL header!";
+
+ int64_t totalFrames = getTotalFrames();
+ ASSERT_NO_FATAL_FAILURE(processEncoder(totalFrames)) << "Failed to Encode: " << mFileName;
+ status = PVCleanUpVideoEncoder(mEncodeControl);
+ ASSERT_TRUE(status) << "Failed to clean up the encoder resources!";
+}
+
+INSTANTIATE_TEST_SUITE_P(
+ EncodeTest, Mpeg4H263EncoderTest,
+ ::testing::Values(
+ make_tuple("bbb_352x288_420p_30fps_32frames.yuv", false, 352, 288, 25, 1024),
+ make_tuple("bbb_352x288_420p_30fps_32frames.yuv", true, 352, 288, 25, 1024),
+ make_tuple("bbb_352x288_420p_30fps_32frames.yuv", false, 176, 144, 25, 1024),
+ make_tuple("bbb_352x288_420p_30fps_32frames.yuv", true, 176, 144, 25, 1024),
+ make_tuple("football_qvga.yuv", false, 352, 288, 25, 1024),
+ make_tuple("football_qvga.yuv", true, 352, 288, 25, 1024),
+ make_tuple("football_qvga.yuv", false, 176, 144, 30, 1024),
+ make_tuple("football_qvga.yuv", true, 176, 144, 30, 1024)));
+
+int32_t main(int argc, char **argv) {
+ gEnv = new Mpeg4H263EncoderTestEnvironment();
+ ::testing::AddGlobalTestEnvironment(gEnv);
+ ::testing::InitGoogleTest(&argc, argv);
+ uint8_t status = gEnv->initFromOptions(argc, argv);
+ if (status == 0) {
+ status = RUN_ALL_TESTS();
+ ALOGI("Encoder Test Result = %d\n", status);
+ }
+ return status;
+}
diff --git a/media/libstagefright/codecs/m4v_h263/enc/test/Mpeg4H263EncoderTestEnvironment.h b/media/libstagefright/codecs/m4v_h263/enc/test/Mpeg4H263EncoderTestEnvironment.h
new file mode 100644
index 0000000..7ee36e0
--- /dev/null
+++ b/media/libstagefright/codecs/m4v_h263/enc/test/Mpeg4H263EncoderTestEnvironment.h
@@ -0,0 +1,73 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __MPEG4_H263_ENCODER_TEST_ENVIRONMENT_H__
+#define __MPEG4_H263_ENCODER_TEST_ENVIRONMENT_H__
+
+#include <gtest/gtest.h>
+
+#include <getopt.h>
+
+using namespace std;
+
+class Mpeg4H263EncoderTestEnvironment : public::testing::Environment {
+ public:
+ Mpeg4H263EncoderTestEnvironment() : res("/data/local/tmp/Mpeg4H263EncoderTest/") {}
+
+ // Parses the command line arguments
+ int initFromOptions(int argc, char **argv);
+
+ void setRes(const char *_res) { res = _res; }
+
+ const string getRes() const { return res; }
+
+ private:
+ string res;
+};
+
+int Mpeg4H263EncoderTestEnvironment::initFromOptions(int argc, char **argv) {
+ static struct option options[] = {{"path", required_argument, 0, 'P'}, {0, 0, 0, 0}};
+
+ while (true) {
+ int index = 0;
+ int c = getopt_long(argc, argv, "P:", options, &index);
+ if (c == -1) {
+ break;
+ }
+
+ switch (c) {
+ case 'P': {
+ setRes(optarg);
+ break;
+ }
+ default:
+ break;
+ }
+ }
+
+ if (optind < argc) {
+ fprintf(stderr,
+ "unrecognized option: %s\n\n"
+ "usage: %s <gtest options> <test options>\n\n"
+ "test options are:\n\n"
+ "-P, --path: Resource files directory location\n",
+ argv[optind ?: 1], argv[0]);
+ return 2;
+ }
+ return 0;
+}
+
+#endif // __MPEG4_H263_ENCODER_TEST_ENVIRONMENT_H__
diff --git a/media/libstagefright/codecs/m4v_h263/enc/test/README.md b/media/libstagefright/codecs/m4v_h263/enc/test/README.md
new file mode 100644
index 0000000..25de878
--- /dev/null
+++ b/media/libstagefright/codecs/m4v_h263/enc/test/README.md
@@ -0,0 +1,38 @@
+## Media Testing ##
+---
+
+#### Mpeg4H263Encoder :
+The Mpeg4H263Encoder Test Suite validates the Mpeg4 and H263 encoder available in libstagefright.
+Run the following steps to build the test suite:
+```
+m Mpeg4H263EncoderTest
+```
+
+The 32-bit binaries will be created in the following path : ${OUT}/data/nativetest/
+The 64-bit binaries will be created in the following path : ${OUT}/data/nativetest64/
+
+To test 64-bit binary push binaries from nativetest64.
+```
+adb push ${OUT}/data/nativetest64/Mpeg4H263EncoderTest/Mpeg4H263EncoderTest /data/local/tmp/
+```
+
+To test 32-bit binary push binaries from nativetest.
+```
+adb push ${OUT}/data/nativetest/Mpeg4H263EncoderTest/Mpeg4H263EncoderTest /data/local/tmp/
+```
+
+The resource file for the tests is taken from [here](https://storage.googleapis.com/android_media/frameworks/av/media/libstagefright/codecs/m4v_h263/enc/test/Mpeg4H263Encoder.zip ) Download, unzip and push these files into device for testing.
+
+```
+adb push Mpeg4H263Encoder/. /data/local/tmp/
+```
+
+usage: Mpeg4H263EncoderTest -P \<path_to_folder\>
+```
+adb shell /data/local/tmp/Mpeg4H263EncoderTest -P /data/local/tmp/
+```
+Alternatively, the test can also be run using atest command.
+
+```
+atest Mpeg4H263EncoderTest -- --enable-module-dynamic-download=true
+```
diff --git a/media/libstagefright/codecs/mp3dec/src/pv_mp3dec_fxd_op_c_equivalent.h b/media/libstagefright/codecs/mp3dec/src/pv_mp3dec_fxd_op_c_equivalent.h
index adb0dd4..f9d91b1 100644
--- a/media/libstagefright/codecs/mp3dec/src/pv_mp3dec_fxd_op_c_equivalent.h
+++ b/media/libstagefright/codecs/mp3dec/src/pv_mp3dec_fxd_op_c_equivalent.h
@@ -44,7 +44,7 @@
#endif
#include "pvmp3_audio_type_defs.h"
-#define Qfmt_31(a) (Int32)((float)(a)*0x7FFFFFFF)
+#define Qfmt_31(a) (Int32)((float)(a)*(float)0x7FFFFFFF)
#define Qfmt15(x) (Int16)((x)*((Int32)1<<15) + ((x)>=0?0.5F:-0.5F))
diff --git a/media/libstagefright/codecs/mp3dec/src/pvmp3_alias_reduction.cpp b/media/libstagefright/codecs/mp3dec/src/pvmp3_alias_reduction.cpp
index af738ba..a4f798e 100644
--- a/media/libstagefright/codecs/mp3dec/src/pvmp3_alias_reduction.cpp
+++ b/media/libstagefright/codecs/mp3dec/src/pvmp3_alias_reduction.cpp
@@ -169,7 +169,7 @@
int32 i, j;
- *used_freq_lines = fxp_mul32_Q32(*used_freq_lines << 16, (int32)(0x7FFFFFFF / (float)18 - 1.0f)) >> 15;
+ *used_freq_lines = fxp_mul32_Q32(*used_freq_lines << 16, (int32)((float)0x7FFFFFFF / 18.0f - 1.0f)) >> 15;
if (gr_info->window_switching_flag && gr_info->block_type == 2)
diff --git a/media/libstagefright/codecs/mp3dec/src/pvmp3_dct_9.cpp b/media/libstagefright/codecs/mp3dec/src/pvmp3_dct_9.cpp
index bbb247d..9cd0e91 100644
--- a/media/libstagefright/codecs/mp3dec/src/pvmp3_dct_9.cpp
+++ b/media/libstagefright/codecs/mp3dec/src/pvmp3_dct_9.cpp
@@ -77,7 +77,7 @@
; Include all pre-processor statements here. Include conditional
; compile variables also.
----------------------------------------------------------------------------*/
-#define Qfmt31(a) (int32)((a)*(0x7FFFFFFF))
+#define Qfmt31(a) (int32)((a)*((float)0x7FFFFFFF))
#define cos_pi_9 Qfmt31( 0.93969262078591f)
#define cos_2pi_9 Qfmt31( 0.76604444311898f)
diff --git a/media/libstagefright/codecs/mp3dec/test/Android.bp b/media/libstagefright/codecs/mp3dec/test/Android.bp
new file mode 100644
index 0000000..0ff8b12
--- /dev/null
+++ b/media/libstagefright/codecs/mp3dec/test/Android.bp
@@ -0,0 +1,48 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+cc_test {
+ name: "Mp3DecoderTest",
+ gtest: true,
+
+ srcs: [
+ "mp3reader.cpp",
+ "Mp3DecoderTest.cpp",
+ ],
+
+ static_libs: [
+ "libstagefright_mp3dec",
+ "libsndfile",
+ "libaudioutils",
+ ],
+
+ shared_libs: [
+ "liblog",
+ ],
+
+ cflags: [
+ "-Werror",
+ "-Wall",
+ ],
+
+ sanitize: {
+ cfi: true,
+ misc_undefined: [
+ "unsigned-integer-overflow",
+ "signed-integer-overflow",
+ ],
+ },
+}
diff --git a/media/libstagefright/codecs/mp3dec/test/AndroidTest.xml b/media/libstagefright/codecs/mp3dec/test/AndroidTest.xml
new file mode 100644
index 0000000..7ff9732
--- /dev/null
+++ b/media/libstagefright/codecs/mp3dec/test/AndroidTest.xml
@@ -0,0 +1,31 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!-- Copyright (C) 2020 The Android Open Source Project
+
+ Licensed under the Apache License, Version 2.0 (the "License");
+ you may not use this file except in compliance with the License.
+ You may obtain a copy of the License at
+
+ http://www.apache.org/licenses/LICENSE-2.0
+
+ Unless required by applicable law or agreed to in writing, software
+ distributed under the License is distributed on an "AS IS" BASIS,
+ WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ See the License for the specific language governing permissions and
+ limitations under the License.
+-->
+<configuration description="Test module config for Mp3 Decoder unit test">
+ <option name="test-suite-tag" value="Mp3DecoderTest" />
+ <target_preparer class="com.android.tradefed.targetprep.PushFilePreparer">
+ <option name="cleanup" value="true" />
+ <option name="push" value="Mp3DecoderTest->/data/local/tmp/Mp3DecoderTest" />
+ <option name="push-file"
+ key="https://storage.googleapis.com/android_media/frameworks/av/media/libstagefright/mp3dec/test/Mp3DecoderTest.zip?unzip=true"
+ value="/data/local/tmp/Mp3DecoderTestRes/" />
+ </target_preparer>
+
+ <test class="com.android.tradefed.testtype.GTest" >
+ <option name="native-test-device-path" value="/data/local/tmp" />
+ <option name="module-name" value="Mp3DecoderTest" />
+ <option name="native-test-flag" value="-P /data/local/tmp/Mp3DecoderTestRes/" />
+ </test>
+</configuration>
diff --git a/media/libstagefright/codecs/mp3dec/test/Mp3DecoderTest.cpp b/media/libstagefright/codecs/mp3dec/test/Mp3DecoderTest.cpp
new file mode 100644
index 0000000..99553ec
--- /dev/null
+++ b/media/libstagefright/codecs/mp3dec/test/Mp3DecoderTest.cpp
@@ -0,0 +1,200 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "Mp3DecoderTest"
+
+#include <utils/Log.h>
+
+#include <audio_utils/sndfile.h>
+#include <stdio.h>
+
+#include "mp3reader.h"
+#include "pvmp3decoder_api.h"
+
+#include "Mp3DecoderTestEnvironment.h"
+
+#define OUTPUT_FILE "/data/local/tmp/mp3Decode.out"
+
+constexpr int32_t kInputBufferSize = 1024 * 10;
+constexpr int32_t kOutputBufferSize = 4608 * 2;
+constexpr int32_t kMaxCount = 10;
+constexpr int32_t kNumFrameReset = 150;
+
+static Mp3DecoderTestEnvironment *gEnv = nullptr;
+
+class Mp3DecoderTest : public ::testing::TestWithParam<string> {
+ public:
+ Mp3DecoderTest() : mConfig(nullptr) {}
+
+ ~Mp3DecoderTest() {
+ if (mConfig) {
+ delete mConfig;
+ mConfig = nullptr;
+ }
+ }
+
+ virtual void SetUp() override {
+ mConfig = new tPVMP3DecoderExternal{};
+ ASSERT_NE(mConfig, nullptr) << "Failed to initialize config. No Memory available";
+ mConfig->equalizerType = flat;
+ mConfig->crcEnabled = false;
+ }
+
+ tPVMP3DecoderExternal *mConfig;
+ Mp3Reader mMp3Reader;
+
+ ERROR_CODE DecodeFrames(void *decoderbuf, SNDFILE *outFileHandle, SF_INFO sfInfo,
+ int32_t frameCount = INT32_MAX);
+ SNDFILE *openOutputFile(SF_INFO *sfInfo);
+};
+
+ERROR_CODE Mp3DecoderTest::DecodeFrames(void *decoderBuf, SNDFILE *outFileHandle, SF_INFO sfInfo,
+ int32_t frameCount) {
+ uint8_t inputBuf[kInputBufferSize];
+ int16_t outputBuf[kOutputBufferSize];
+ uint32_t bytesRead;
+ ERROR_CODE decoderErr;
+ while (frameCount > 0) {
+ bool success = mMp3Reader.getFrame(inputBuf, &bytesRead);
+ if (!success) {
+ break;
+ }
+ mConfig->inputBufferCurrentLength = bytesRead;
+ mConfig->inputBufferMaxLength = 0;
+ mConfig->inputBufferUsedLength = 0;
+ mConfig->pInputBuffer = inputBuf;
+ mConfig->pOutputBuffer = outputBuf;
+ mConfig->outputFrameSize = kOutputBufferSize / sizeof(int16_t);
+ decoderErr = pvmp3_framedecoder(mConfig, decoderBuf);
+ if (decoderErr != NO_DECODING_ERROR) break;
+ sf_writef_short(outFileHandle, outputBuf, mConfig->outputFrameSize / sfInfo.channels);
+ frameCount--;
+ }
+ return decoderErr;
+}
+
+SNDFILE *Mp3DecoderTest::openOutputFile(SF_INFO *sfInfo) {
+ memset(sfInfo, 0, sizeof(SF_INFO));
+ sfInfo->channels = mMp3Reader.getNumChannels();
+ sfInfo->format = SF_FORMAT_WAV | SF_FORMAT_PCM_16;
+ sfInfo->samplerate = mMp3Reader.getSampleRate();
+ SNDFILE *outFileHandle = sf_open(OUTPUT_FILE, SFM_WRITE, sfInfo);
+ return outFileHandle;
+}
+
+TEST_F(Mp3DecoderTest, MultiCreateMp3DecoderTest) {
+ size_t memRequirements = pvmp3_decoderMemRequirements();
+ ASSERT_NE(memRequirements, 0) << "Failed to get the memory requirement size";
+ void *decoderBuf = malloc(memRequirements);
+ ASSERT_NE(decoderBuf, nullptr)
+ << "Failed to allocate decoder memory of size " << memRequirements;
+ for (int count = 0; count < kMaxCount; count++) {
+ pvmp3_InitDecoder(mConfig, decoderBuf);
+ ALOGV("Decoder created successfully");
+ }
+ if (decoderBuf) {
+ free(decoderBuf);
+ decoderBuf = nullptr;
+ }
+}
+
+TEST_P(Mp3DecoderTest, DecodeTest) {
+ size_t memRequirements = pvmp3_decoderMemRequirements();
+ ASSERT_NE(memRequirements, 0) << "Failed to get the memory requirement size";
+ void *decoderBuf = malloc(memRequirements);
+ ASSERT_NE(decoderBuf, nullptr)
+ << "Failed to allocate decoder memory of size " << memRequirements;
+
+ pvmp3_InitDecoder(mConfig, decoderBuf);
+ ALOGV("Decoder created successfully");
+ string inputFile = gEnv->getRes() + GetParam();
+ bool status = mMp3Reader.init(inputFile.c_str());
+ ASSERT_TRUE(status) << "Unable to initialize the mp3Reader";
+
+ // Open the output file.
+ SF_INFO sfInfo;
+ SNDFILE *outFileHandle = openOutputFile(&sfInfo);
+ ASSERT_NE(outFileHandle, nullptr) << "Error opening output file for writing decoded output";
+
+ ERROR_CODE decoderErr = DecodeFrames(decoderBuf, outFileHandle, sfInfo);
+ ASSERT_EQ(decoderErr, NO_DECODING_ERROR) << "Failed to decode the frames";
+ ASSERT_EQ(sfInfo.channels, mConfig->num_channels) << "Number of channels does not match";
+ ASSERT_EQ(sfInfo.samplerate, mConfig->samplingRate) << "Sample rate does not match";
+
+ mMp3Reader.close();
+ sf_close(outFileHandle);
+ if (decoderBuf) {
+ free(decoderBuf);
+ decoderBuf = nullptr;
+ }
+}
+
+TEST_P(Mp3DecoderTest, ResetDecoderTest) {
+ size_t memRequirements = pvmp3_decoderMemRequirements();
+ ASSERT_NE(memRequirements, 0) << "Failed to get the memory requirement size";
+ void *decoderBuf = malloc(memRequirements);
+ ASSERT_NE(decoderBuf, nullptr)
+ << "Failed to allocate decoder memory of size " << memRequirements;
+
+ pvmp3_InitDecoder(mConfig, decoderBuf);
+ ALOGV("Decoder created successfully.");
+ string inputFile = gEnv->getRes() + GetParam();
+ bool status = mMp3Reader.init(inputFile.c_str());
+ ASSERT_TRUE(status) << "Unable to initialize the mp3Reader";
+
+ // Open the output file.
+ SF_INFO sfInfo;
+ SNDFILE *outFileHandle = openOutputFile(&sfInfo);
+ ASSERT_NE(outFileHandle, nullptr) << "Error opening output file for writing decoded output";
+
+ ERROR_CODE decoderErr;
+ decoderErr = DecodeFrames(decoderBuf, outFileHandle, sfInfo, kNumFrameReset);
+ ASSERT_EQ(decoderErr, NO_DECODING_ERROR) << "Failed to decode the frames";
+ ASSERT_EQ(sfInfo.channels, mConfig->num_channels) << "Number of channels does not match";
+ ASSERT_EQ(sfInfo.samplerate, mConfig->samplingRate) << "Sample rate does not match";
+
+ pvmp3_resetDecoder(decoderBuf);
+ // Decode the same file.
+ decoderErr = DecodeFrames(decoderBuf, outFileHandle, sfInfo);
+ ASSERT_EQ(decoderErr, NO_DECODING_ERROR) << "Failed to decode the frames";
+ ASSERT_EQ(sfInfo.channels, mConfig->num_channels) << "Number of channels does not match";
+ ASSERT_EQ(sfInfo.samplerate, mConfig->samplingRate) << "Sample rate does not match";
+
+ mMp3Reader.close();
+ sf_close(outFileHandle);
+ if (decoderBuf) {
+ free(decoderBuf);
+ decoderBuf = nullptr;
+ }
+}
+
+INSTANTIATE_TEST_SUITE_P(Mp3DecoderTestAll, Mp3DecoderTest,
+ ::testing::Values(("bbb_44100hz_2ch_128kbps_mp3_30sec.mp3"),
+ ("bbb_44100hz_2ch_128kbps_mp3_5mins.mp3"),
+ ("bbb_mp3_stereo_192kbps_48000hz.mp3")));
+
+int main(int argc, char **argv) {
+ gEnv = new Mp3DecoderTestEnvironment();
+ ::testing::AddGlobalTestEnvironment(gEnv);
+ ::testing::InitGoogleTest(&argc, argv);
+ int status = gEnv->initFromOptions(argc, argv);
+ if (status == 0) {
+ status = RUN_ALL_TESTS();
+ ALOGV("Test result = %d\n", status);
+ }
+ return status;
+}
diff --git a/media/libstagefright/codecs/mp3dec/test/Mp3DecoderTestEnvironment.h b/media/libstagefright/codecs/mp3dec/test/Mp3DecoderTestEnvironment.h
new file mode 100644
index 0000000..a54b34c
--- /dev/null
+++ b/media/libstagefright/codecs/mp3dec/test/Mp3DecoderTestEnvironment.h
@@ -0,0 +1,72 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __MP3DECODER_TEST_ENVIRONMENT_H__
+#define __MP3DECODER_TEST_ENVIRONMENT_H__
+
+#include <gtest/gtest.h>
+
+#include <getopt.h>
+
+using namespace std;
+
+class Mp3DecoderTestEnvironment : public ::testing::Environment {
+ public:
+ Mp3DecoderTestEnvironment() : res("/data/local/tmp/") {}
+
+ // Parses the command line arguments
+ int initFromOptions(int argc, char **argv);
+
+ void setRes(const char *_res) { res = _res; }
+
+ const string getRes() const { return res; }
+
+ private:
+ string res;
+};
+
+int Mp3DecoderTestEnvironment::initFromOptions(int argc, char **argv) {
+ static struct option options[] = {{"res", required_argument, 0, 'P'}, {0, 0, 0, 0}};
+
+ while (true) {
+ int index = 0;
+ int c = getopt_long(argc, argv, "P:", options, &index);
+ if (c == -1) {
+ break;
+ }
+
+ switch (c) {
+ case 'P':
+ setRes(optarg);
+ break;
+ default:
+ break;
+ }
+ }
+
+ if (optind < argc) {
+ fprintf(stderr,
+ "unrecognized option: %s\n\n"
+ "usage: %s <gtest options> <test options>\n\n"
+ "test options are:\n\n"
+ "-P, --path: Resource files directory location\n",
+ argv[optind ?: 1], argv[0]);
+ return 2;
+ }
+ return 0;
+}
+
+#endif // __MP3DECODER_TEST_ENVIRONMENT_H__
diff --git a/media/libstagefright/codecs/mp3dec/test/README.md b/media/libstagefright/codecs/mp3dec/test/README.md
new file mode 100644
index 0000000..f59fec7
--- /dev/null
+++ b/media/libstagefright/codecs/mp3dec/test/README.md
@@ -0,0 +1,39 @@
+## Media Testing ##
+---
+#### Mp3Decoder :
+The Mp3Decoder Test Suite validates the mp3decoder available in libstagefright.
+
+Run the following steps to build the test suite:
+```
+m Mp3DecoderTest
+```
+
+The 32-bit binaries will be created in the following path : ${OUT}/data/nativetest/
+
+The 64-bit binaries will be created in the following path : ${OUT}/data/nativetest64/
+
+To test 64-bit binary push binaries from nativetest64.
+```
+adb push ${OUT}/data/nativetest64/Mp3DecoderTest/Mp3DecoderTest /data/local/tmp/
+```
+
+To test 32-bit binary push binaries from nativetest.
+```
+adb push ${OUT}/data/nativetest/Mp3DecoderTest/Mp3DecoderTest /data/local/tmp/
+```
+
+The resource file for the tests is taken from [here](https://storage.googleapis.com/android_media/frameworks/av/media/libstagefright/mp3dec/test/Mp3DecoderTest.zip). Download, unzip and push these files into device for testing.
+
+```
+adb push Mp3DecoderTestRes/. /data/local/tmp/
+```
+
+usage: Mp3DecoderTest -P \<path_to_folder\>
+```
+adb shell /data/local/tmp/Mp3DecoderTest -P /data/local/tmp/Mp3DecoderTestRes/
+```
+Alternatively, the test can also be run using atest command.
+
+```
+atest Mp3DecoderTest -- --enable-module-dynamic-download=true
+```
diff --git a/media/libstagefright/codecs/on2/enc/Android.bp b/media/libstagefright/codecs/on2/enc/Android.bp
index cd69e0d..705e554 100644
--- a/media/libstagefright/codecs/on2/enc/Android.bp
+++ b/media/libstagefright/codecs/on2/enc/Android.bp
@@ -21,4 +21,5 @@
},
shared_libs: ["libvpx"],
+ header_libs: ["libbase_headers"],
}
diff --git a/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp b/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp
index 1293a74..08e20cc 100644
--- a/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp
+++ b/media/libstagefright/codecs/vorbis/dec/SoftVorbis.cpp
@@ -572,16 +572,17 @@
}
void SoftVorbis::onPortFlushCompleted(OMX_U32 portIndex) {
- if (portIndex == 0 && mState != NULL) {
- // Make sure that the next buffer output does not still
- // depend on fragments from the last one decoded.
-
+ if (portIndex == 0) {
mInputBufferCount = 0;
mNumFramesOutput = 0;
mSawInputEos = false;
mSignalledOutputEos = false;
mNumFramesLeftOnPage = -1;
- vorbis_dsp_restart(mState);
+ if (mState != NULL) {
+ // Make sure that the next buffer output does not still
+ // depend on fragments from the last one decoded.
+ vorbis_dsp_restart(mState);
+ }
}
}
@@ -603,6 +604,7 @@
mSawInputEos = false;
mSignalledOutputEos = false;
mSignalledError = false;
+ mNumFramesLeftOnPage = -1;
mOutputPortSettingsChange = NONE;
}
diff --git a/media/libstagefright/colorconversion/SoftwareRenderer.cpp b/media/libstagefright/colorconversion/SoftwareRenderer.cpp
index 359df3d..4711315 100644
--- a/media/libstagefright/colorconversion/SoftwareRenderer.cpp
+++ b/media/libstagefright/colorconversion/SoftwareRenderer.cpp
@@ -31,9 +31,14 @@
namespace android {
-static int ALIGN(int x, int y) {
- // y must be a power of 2.
- return (x + y - 1) & ~(y - 1);
+inline void initDstYUV(
+ const android_ycbcr &ycbcr, int32_t cropTop, int32_t cropLeft,
+ uint8_t **dst_y, uint8_t **dst_u, uint8_t **dst_v) {
+ *dst_y = (uint8_t *)ycbcr.y + cropTop * ycbcr.ystride + cropLeft;
+
+ int32_t c_offset = (cropTop / 2) * ycbcr.cstride + cropLeft / 2;
+ *dst_v = (uint8_t *)ycbcr.cr + c_offset;
+ *dst_u = (uint8_t *)ycbcr.cb + c_offset;
}
SoftwareRenderer::SoftwareRenderer(
@@ -269,10 +274,21 @@
Rect bounds(mCropWidth, mCropHeight);
- void *dst;
- CHECK_EQ(0, mapper.lock(buf->handle,
- GRALLOC_USAGE_SW_READ_NEVER | GRALLOC_USAGE_SW_WRITE_RARELY,
- bounds, &dst));
+ void *dst = NULL;
+ struct android_ycbcr ycbcr;
+ if ( !mConverter &&
+ (mColorFormat == OMX_COLOR_FormatYUV420Planar ||
+ mColorFormat == OMX_COLOR_FormatYUV420SemiPlanar ||
+ mColorFormat == OMX_TI_COLOR_FormatYUV420PackedSemiPlanar ||
+ mColorFormat == OMX_COLOR_FormatYUV420Planar16)) {
+ CHECK_EQ(0, mapper.lockYCbCr(buf->handle,
+ GRALLOC_USAGE_SW_READ_NEVER | GRALLOC_USAGE_SW_WRITE_RARELY,
+ bounds, &ycbcr));
+ } else {
+ CHECK_EQ(0, mapper.lock(buf->handle,
+ GRALLOC_USAGE_SW_READ_NEVER | GRALLOC_USAGE_SW_WRITE_RARELY,
+ bounds, &dst));
+ }
// TODO move the other conversions also into ColorConverter, and
// fix cropping issues (when mCropLeft/Top != 0 or mWidth != mCropWidth)
@@ -289,22 +305,14 @@
const uint8_t *src_u = (const uint8_t *)data + mStride * mHeight + mCropTop * mStride / 4;
const uint8_t *src_v = (const uint8_t *)src_u + mStride * mHeight / 4;
- uint8_t *dst_y = (uint8_t *)dst;
- size_t dst_y_size = buf->stride * buf->height;
- size_t dst_c_stride = ALIGN(buf->stride / 2, 16);
- size_t dst_c_size = dst_c_stride * buf->height / 2;
- uint8_t *dst_v = dst_y + dst_y_size;
- uint8_t *dst_u = dst_v + dst_c_size;
-
- dst_y += mCropTop * buf->stride + mCropLeft;
- dst_v += (mCropTop/2) * dst_c_stride + mCropLeft/2;
- dst_u += (mCropTop/2) * dst_c_stride + mCropLeft/2;
+ uint8_t *dst_y, *dst_u, *dst_v;
+ initDstYUV(ycbcr, mCropTop, mCropLeft, &dst_y, &dst_u, &dst_v);
for (int y = 0; y < mCropHeight; ++y) {
memcpy(dst_y, src_y, mCropWidth);
src_y += mStride;
- dst_y += buf->stride;
+ dst_y += ycbcr.ystride;
}
for (int y = 0; y < (mCropHeight + 1) / 2; ++y) {
@@ -313,24 +321,16 @@
src_u += mStride / 2;
src_v += mStride / 2;
- dst_u += dst_c_stride;
- dst_v += dst_c_stride;
+ dst_u += ycbcr.cstride;
+ dst_v += ycbcr.cstride;
}
} else if (mColorFormat == OMX_COLOR_FormatYUV420Planar16) {
const uint8_t *src_y = (const uint8_t *)data + mCropTop * mStride + mCropLeft * 2;
const uint8_t *src_u = (const uint8_t *)data + mStride * mHeight + mCropTop * mStride / 4;
const uint8_t *src_v = (const uint8_t *)src_u + mStride * mHeight / 4;
- uint8_t *dst_y = (uint8_t *)dst;
- size_t dst_y_size = buf->stride * buf->height;
- size_t dst_c_stride = ALIGN(buf->stride / 2, 16);
- size_t dst_c_size = dst_c_stride * buf->height / 2;
- uint8_t *dst_v = dst_y + dst_y_size;
- uint8_t *dst_u = dst_v + dst_c_size;
-
- dst_y += mCropTop * buf->stride + mCropLeft;
- dst_v += (mCropTop / 2) * dst_c_stride + mCropLeft / 2;
- dst_u += (mCropTop / 2) * dst_c_stride + mCropLeft / 2;
+ uint8_t *dst_y, *dst_u, *dst_v;
+ initDstYUV(ycbcr, mCropTop, mCropLeft, &dst_y, &dst_u, &dst_v);
for (int y = 0; y < mCropHeight; ++y) {
for (int x = 0; x < mCropWidth; ++x) {
@@ -338,7 +338,7 @@
}
src_y += mStride;
- dst_y += buf->stride;
+ dst_y += ycbcr.ystride;
}
for (int y = 0; y < (mCropHeight + 1) / 2; ++y) {
@@ -349,8 +349,8 @@
src_u += mStride / 2;
src_v += mStride / 2;
- dst_u += dst_c_stride;
- dst_v += dst_c_stride;
+ dst_u += ycbcr.cstride;
+ dst_v += ycbcr.cstride;
}
} else if (mColorFormat == OMX_TI_COLOR_FormatYUV420PackedSemiPlanar
|| mColorFormat == OMX_COLOR_FormatYUV420SemiPlanar) {
@@ -361,23 +361,14 @@
src_y += mCropLeft + mCropTop * mWidth;
src_uv += (mCropLeft + mCropTop * mWidth) / 2;
- uint8_t *dst_y = (uint8_t *)dst;
-
- size_t dst_y_size = buf->stride * buf->height;
- size_t dst_c_stride = ALIGN(buf->stride / 2, 16);
- size_t dst_c_size = dst_c_stride * buf->height / 2;
- uint8_t *dst_v = dst_y + dst_y_size;
- uint8_t *dst_u = dst_v + dst_c_size;
-
- dst_y += mCropTop * buf->stride + mCropLeft;
- dst_v += (mCropTop/2) * dst_c_stride + mCropLeft/2;
- dst_u += (mCropTop/2) * dst_c_stride + mCropLeft/2;
+ uint8_t *dst_y, *dst_u, *dst_v;
+ initDstYUV(ycbcr, mCropTop, mCropLeft, &dst_y, &dst_u, &dst_v);
for (int y = 0; y < mCropHeight; ++y) {
memcpy(dst_y, src_y, mCropWidth);
src_y += mWidth;
- dst_y += buf->stride;
+ dst_y += ycbcr.ystride;
}
for (int y = 0; y < (mCropHeight + 1) / 2; ++y) {
@@ -388,8 +379,8 @@
}
src_uv += mWidth;
- dst_u += dst_c_stride;
- dst_v += dst_c_stride;
+ dst_u += ycbcr.cstride;
+ dst_v += ycbcr.cstride;
}
} else if (mColorFormat == OMX_COLOR_Format24bitRGB888) {
uint8_t* srcPtr = (uint8_t*)data + mWidth * mCropTop * 3 + mCropLeft * 3;
diff --git a/media/libstagefright/filters/Android.bp b/media/libstagefright/filters/Android.bp
index 7a67e55..88f30c4 100644
--- a/media/libstagefright/filters/Android.bp
+++ b/media/libstagefright/filters/Android.bp
@@ -8,7 +8,7 @@
"MediaFilter.cpp",
"RSFilter.cpp",
"SaturationFilter.cpp",
- "saturationARGB.rs",
+ "saturationARGB.rscript",
"SimpleFilter.cpp",
"ZeroFilter.cpp",
],
@@ -23,6 +23,10 @@
"-Wall",
],
+ header_libs: [
+ "libmediadrm_headers",
+ ],
+
shared_libs: [
"libgui",
"libmedia",
diff --git a/media/libstagefright/filters/saturation.rs b/media/libstagefright/filters/saturation.rscript
similarity index 100%
rename from media/libstagefright/filters/saturation.rs
rename to media/libstagefright/filters/saturation.rscript
diff --git a/media/libstagefright/filters/saturationARGB.rs b/media/libstagefright/filters/saturationARGB.rscript
similarity index 100%
rename from media/libstagefright/filters/saturationARGB.rs
rename to media/libstagefright/filters/saturationARGB.rscript
diff --git a/media/libstagefright/flac/dec/test/Android.bp b/media/libstagefright/flac/dec/test/Android.bp
new file mode 100644
index 0000000..70ca80a
--- /dev/null
+++ b/media/libstagefright/flac/dec/test/Android.bp
@@ -0,0 +1,50 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+cc_test {
+ name: "FlacDecoderTest",
+ gtest: true,
+
+ srcs: [
+ "FlacDecoderTest.cpp",
+ ],
+
+ shared_libs: [
+ "liblog",
+ ],
+
+ static_libs: [
+ "libstagefright_flacdec",
+ "libFLAC",
+ ],
+
+ header_libs: [
+ "libstagefright_foundation_headers",
+ ],
+
+ cflags: [
+ "-Werror",
+ "-Wall",
+ ],
+
+ sanitize: {
+ misc_undefined: [
+ "unsigned-integer-overflow",
+ "signed-integer-overflow",
+ ],
+ cfi: true,
+ },
+}
diff --git a/media/libstagefright/flac/dec/test/AndroidTest.xml b/media/libstagefright/flac/dec/test/AndroidTest.xml
new file mode 100644
index 0000000..bebba8e
--- /dev/null
+++ b/media/libstagefright/flac/dec/test/AndroidTest.xml
@@ -0,0 +1,31 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!-- Copyright (C) 2020 The Android Open Source Project
+
+ Licensed under the Apache License, Version 2.0 (the "License");
+ you may not use this file except in compliance with the License.
+ You may obtain a copy of the License at
+
+ http://www.apache.org/licenses/LICENSE-2.0
+
+ Unless required by applicable law or agreed to in writing, software
+ distributed under the License is distributed on an "AS IS" BASIS,
+ WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ See the License for the specific language governing permissions and
+ limitations under the License.
+-->
+<configuration description="Test module config for flac decoder unit tests">
+ <option name="test-suite-tag" value="FlacDecoderTest" />
+ <target_preparer class="com.android.tradefed.targetprep.PushFilePreparer">
+ <option name="cleanup" value="true" />
+ <option name="push" value="FlacDecoderTest->/data/local/tmp/FlacDecoderTest/" />
+ <option name="push-file"
+ key="https://storage.googleapis.com/android_media/frameworks/av/media/libstagefright/flac/dec/test/FlacDecoder.zip?unzip=true"
+ value="/data/local/tmp/FlacDecoderTestRes/" />
+ </target_preparer>
+
+ <test class="com.android.tradefed.testtype.GTest" >
+ <option name="native-test-device-path" value="/data/local/tmp" />
+ <option name="module-name" value="FlacDecoderTest" />
+ <option name="native-test-flag" value="-P /data/local/tmp/FlacDecoderTestRes/" />
+ </test>
+</configuration>
\ No newline at end of file
diff --git a/media/libstagefright/flac/dec/test/FlacDecoderTest.cpp b/media/libstagefright/flac/dec/test/FlacDecoderTest.cpp
new file mode 100644
index 0000000..34f12db
--- /dev/null
+++ b/media/libstagefright/flac/dec/test/FlacDecoderTest.cpp
@@ -0,0 +1,270 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "FlacDecoderTest"
+
+#include <utils/Log.h>
+#include <fstream>
+
+#include "FLACDecoder.h"
+
+#include "FlacDecoderTestEnvironment.h"
+
+#define OUTPUT_FILE_NAME "/data/local/tmp/FlacDecoderOutput.raw"
+#define CODEC_CONFIG_FLAG 32
+
+constexpr uint32_t kMaxCount = 10;
+constexpr int32_t kMaxBlockSize = 4096;
+
+using namespace android;
+
+struct FrameInfo {
+ int32_t bytesCount;
+ uint32_t flags;
+ int64_t timestamp;
+};
+
+static FlacDecoderTestEnvironment *gEnv = nullptr;
+
+class FLACDecoderTest : public ::testing::TestWithParam<tuple<string, string, bool>> {
+ public:
+ FLACDecoderTest() : mFLACDecoder(nullptr), mHasStreamInfo(false), mInputBufferCount(0) {}
+
+ ~FLACDecoderTest() {
+ if (mEleStream.is_open()) mEleStream.close();
+ if (mFLACDecoder) delete mFLACDecoder;
+ mFLACDecoder = nullptr;
+ }
+
+ virtual void SetUp() override {
+ mFLACDecoder = FLACDecoder::Create();
+ ASSERT_NE(mFLACDecoder, nullptr) << "initDecoder: failed to create FLACDecoder";
+ }
+
+ int32_t processFlacDecoder(vector<FrameInfo> Info, int32_t offset, int32_t range,
+ bool outputFloat, ofstream &ostrm);
+
+ FLACDecoder *mFLACDecoder;
+ FLAC__StreamMetadata_StreamInfo mStreamInfo;
+
+ bool mHasStreamInfo;
+ int32_t mInputBufferCount;
+ ifstream mEleStream;
+};
+
+void getInfo(string infoFileName, vector<FrameInfo> &Info) {
+ ifstream eleInfo;
+ eleInfo.open(infoFileName);
+ ASSERT_EQ(eleInfo.is_open(), true);
+ int32_t bytesCount = 0;
+ uint32_t flags = 0;
+ uint32_t timestamp = 0;
+ while (1) {
+ if (!(eleInfo >> bytesCount)) break;
+ eleInfo >> flags;
+ eleInfo >> timestamp;
+ Info.push_back({bytesCount, flags, timestamp});
+ }
+ if (eleInfo.is_open()) eleInfo.close();
+}
+
+int32_t FLACDecoderTest::processFlacDecoder(vector<FrameInfo> Info, int32_t offset, int32_t range,
+ bool outputFloat, ofstream &ostrm) {
+ memset(&mStreamInfo, 0, sizeof(mStreamInfo));
+
+ int32_t frameID = offset;
+ if (range + offset > Info.size() || range < 0 || offset > Info.size() - 1 || offset < 0) {
+ ALOGE("Invalid offset or range or both passed for decoding");
+ ALOGE("offset = %d \t range = %d \t Info.size() = %zu", offset, range, Info.size());
+ return -1;
+ }
+
+ while (1) {
+ if (frameID == Info.size() || frameID == (offset + range)) break;
+ int64_t flags = (Info)[frameID].flags;
+ int32_t size = (Info)[frameID].bytesCount;
+ if (size < 0) {
+ ALOGE("Size for the memory allocation is negative");
+ return -1;
+ }
+ char *data = (char *)malloc(size);
+ if (!data) {
+ ALOGE("Insufficient memory to read frame");
+ return -1;
+ }
+
+ mEleStream.read(data, size);
+ if (mEleStream.gcount() != size) {
+ if (data) {
+ free(data);
+ data = nullptr;
+ }
+ ALOGE("Invalid size read, requested: %d and read: %zu", size, mEleStream.gcount());
+ return -1;
+ }
+
+ if (flags == CODEC_CONFIG_FLAG && mInputBufferCount == 0) {
+ status_t decoderErr = mFLACDecoder->parseMetadata((uint8_t *)data, size);
+ if (decoderErr == WOULD_BLOCK) {
+ ALOGV("process: parseMetadata is Blocking, Continue %d", decoderErr);
+ } else if (decoderErr == OK) {
+ mStreamInfo = mFLACDecoder->getStreamInfo();
+ if (mStreamInfo.sample_rate && mStreamInfo.max_blocksize && mStreamInfo.channels) {
+ mHasStreamInfo = true;
+ }
+ ALOGV("decoder configuration : %d Hz, %d channels, %d samples,"
+ " %d block size",
+ mStreamInfo.sample_rate, mStreamInfo.channels,
+ (int32_t)mStreamInfo.total_samples, mStreamInfo.max_blocksize);
+ } else {
+ ALOGE("FLACDecoder parseMetaData returns error %d", decoderErr);
+ if (data) {
+ free(data);
+ data = nullptr;
+ }
+ return decoderErr;
+ }
+ } else {
+ const size_t sampleSize = outputFloat ? sizeof(float) : sizeof(int16_t);
+ size_t outSize = mHasStreamInfo
+ ? mStreamInfo.max_blocksize * mStreamInfo.channels * sampleSize
+ : kMaxBlockSize * FLACDecoder::kMaxChannels * sampleSize;
+
+ void *out_buf = malloc(outSize);
+ if (!out_buf) {
+ if (data) {
+ free(data);
+ data = nullptr;
+ }
+ ALOGE("Output buffer allocation failed");
+ return -1;
+ }
+ status_t decoderErr = mFLACDecoder->decodeOneFrame((uint8_t *)data, size, out_buf,
+ &outSize, outputFloat);
+ if (decoderErr != OK) {
+ ALOGE("decodeOneFrame returns error %d", decoderErr);
+ if (data) {
+ free(data);
+ data = nullptr;
+ }
+ if (out_buf) {
+ free(out_buf);
+ out_buf = nullptr;
+ }
+ return decoderErr;
+ }
+ ostrm.write(reinterpret_cast<char *>(out_buf), outSize);
+ free(out_buf);
+ out_buf = nullptr;
+ }
+ mInputBufferCount++;
+ frameID++;
+ free(data);
+ data = nullptr;
+ }
+ ALOGV("frameID=%d", frameID);
+ return 0;
+}
+
+TEST_F(FLACDecoderTest, CreateDeleteTest) {
+ if (mFLACDecoder) delete mFLACDecoder;
+ mFLACDecoder = nullptr;
+
+ for (int32_t i = 0; i < kMaxCount; i++) {
+ mFLACDecoder = FLACDecoder::Create();
+ ASSERT_NE(mFLACDecoder, nullptr) << "FLACDecoder Creation Failed";
+ if (mFLACDecoder) delete mFLACDecoder;
+ mFLACDecoder = nullptr;
+ }
+}
+
+TEST_P(FLACDecoderTest, FlushTest) {
+ tuple<string /* InputFileName */, string /* InfoFileName */, bool /* outputfloat */> params =
+ GetParam();
+
+ string inputFileName = gEnv->getRes() + get<0>(params);
+ string infoFileName = gEnv->getRes() + get<1>(params);
+ bool outputFloat = get<2>(params);
+
+ vector<FrameInfo> Info;
+ getInfo(infoFileName, Info);
+
+ mEleStream.open(inputFileName, ifstream::binary);
+ ASSERT_EQ(mEleStream.is_open(), true);
+
+ ofstream ostrm;
+ ostrm.open(OUTPUT_FILE_NAME, std::ofstream::binary);
+ ASSERT_EQ(ostrm.is_open(), true);
+
+ int32_t status = processFlacDecoder(Info, 0, Info.size() / 3, outputFloat, ostrm);
+ ASSERT_EQ(status, 0) << "Test Failed. Decode returned error = " << status << endl;
+ mFLACDecoder->flush();
+ mHasStreamInfo = false;
+ status = processFlacDecoder(Info, (Info.size() / 3), Info.size() - (Info.size() / 3),
+ outputFloat, ostrm);
+ ostrm.close();
+ Info.clear();
+ ASSERT_EQ(status, 0) << "Test Failed. Decode returned error = " << status << endl;
+}
+
+TEST_P(FLACDecoderTest, DecodeTest) {
+ tuple<string /* InputFileName */, string /* InfoFileName */, bool /* outputfloat */> params =
+ GetParam();
+
+ string inputFileName = gEnv->getRes() + get<0>(params);
+ string infoFileName = gEnv->getRes() + get<1>(params);
+ bool outputFloat = get<2>(params);
+
+ vector<FrameInfo> Info;
+ getInfo(infoFileName, Info);
+
+ mEleStream.open(inputFileName, ifstream::binary);
+ ASSERT_EQ(mEleStream.is_open(), true);
+
+ ofstream ostrm;
+ ostrm.open(OUTPUT_FILE_NAME, std::ofstream::binary);
+ ASSERT_EQ(ostrm.is_open(), true);
+
+ int32_t status = processFlacDecoder(Info, 0, Info.size(), outputFloat, ostrm);
+ ostrm.close();
+ Info.clear();
+ ASSERT_EQ(status, 0) << "Test Failed. Decode returned error = " << status << endl;
+}
+
+// TODO: Add remaining tests
+INSTANTIATE_TEST_SUITE_P(
+ FLACDecoderTestAll, FLACDecoderTest,
+ ::testing::Values(make_tuple("bbb_flac_stereo_680kbps_48000hz.flac",
+ "bbb_flac_stereo_680kbps_48000hz.info", true),
+ make_tuple("bbb_flac_stereo_680kbps_48000hz.flac",
+ "bbb_flac_stereo_680kbps_48000hz.info", false),
+ make_tuple("bbb_flac_stereo_600kbps_44100hz.flac",
+ "bbb_flac_stereo_600kbps_44100hz.info", true),
+ make_tuple("bbb_flac_stereo_600kbps_44100hz.flac",
+ "bbb_flac_stereo_600kbps_44100hz.info", false)));
+
+int main(int argc, char **argv) {
+ gEnv = new FlacDecoderTestEnvironment();
+ ::testing::AddGlobalTestEnvironment(gEnv);
+ ::testing::InitGoogleTest(&argc, argv);
+ int status = gEnv->initFromOptions(argc, argv);
+ if (status == 0) {
+ status = RUN_ALL_TESTS();
+ ALOGV("Flac Decoder Test Result = %d\n", status);
+ }
+ return status;
+}
diff --git a/media/libstagefright/flac/dec/test/FlacDecoderTestEnvironment.h b/media/libstagefright/flac/dec/test/FlacDecoderTestEnvironment.h
new file mode 100644
index 0000000..1334bba
--- /dev/null
+++ b/media/libstagefright/flac/dec/test/FlacDecoderTestEnvironment.h
@@ -0,0 +1,73 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __FLAC_DECODER_TEST_ENVIRONMENT_H__
+#define __FLAC_DECODER_TEST_ENVIRONMENT_H__
+
+#include <gtest/gtest.h>
+
+#include <getopt.h>
+
+using namespace std;
+
+class FlacDecoderTestEnvironment : public ::testing::Environment {
+ public:
+ FlacDecoderTestEnvironment() : res("/data/local/tmp/") {}
+
+ // Parses the command line arguments
+ int initFromOptions(int argc, char **argv);
+
+ void setRes(const char *_res) { res = _res; }
+
+ const string getRes() const { return res; }
+
+ private:
+ string res;
+};
+
+int FlacDecoderTestEnvironment::initFromOptions(int argc, char **argv) {
+ static struct option options[] = {{"path", required_argument, 0, 'P'}, {0, 0, 0, 0}};
+
+ while (true) {
+ int index = 0;
+ int c = getopt_long(argc, argv, "P:", options, &index);
+ if (c == -1) {
+ break;
+ }
+
+ switch (c) {
+ case 'P': {
+ setRes(optarg);
+ break;
+ }
+ default:
+ break;
+ }
+ }
+
+ if (optind < argc) {
+ fprintf(stderr,
+ "unrecognized option: %s\n\n"
+ "usage: %s <gtest options> <test options>\n\n"
+ "test options are:\n\n"
+ "-P, --path: Resource files directory location\n",
+ argv[optind ?: 1], argv[0]);
+ return 2;
+ }
+ return 0;
+}
+
+#endif // __FLAC_DECODER_TEST_ENVIRONMENT_H__
diff --git a/media/libstagefright/flac/dec/test/README.md b/media/libstagefright/flac/dec/test/README.md
new file mode 100644
index 0000000..4d194cd
--- /dev/null
+++ b/media/libstagefright/flac/dec/test/README.md
@@ -0,0 +1,40 @@
+## Media Testing ##
+---
+#### FlacDecoder :
+The FlacDecoder Test Suite validates the FlacDecoder available in libstagefright.
+
+Run the following steps to build the test suite:
+```
+m FlacDecoderTest
+```
+
+The 32-bit binaries will be created in the following path : ${OUT}/data/nativetest/
+
+The 64-bit binaries will be created in the following path : ${OUT}/data/nativetest64/
+
+To test 64-bit binary push binaries from nativetest64.
+```
+adb push ${OUT}/data/nativetest64/FlacDecoderTest/FlacDecoderTest /data/local/tmp/
+```
+
+To test 32-bit binary push binaries from nativetest.
+```
+adb push ${OUT}/data/nativetest/FlacDecoderTest/FlacDecoderTest /data/local/tmp/
+```
+
+The resource file for the tests is taken from [here](https://storage.googleapis.com/android_media/frameworks/av/media/libstagefright/flac/dec/test/FlacDecoder.zip).
+Download, unzip and push these files into device for testing.
+
+```
+adb push FlacDecoder /data/local/tmp/
+```
+
+usage: FlacDecoderTest -P \<path_to_folder\>
+```
+adb shell /data/local/tmp/FlacDecoderTest -P /data/local/tmp/FlacDecoder/
+```
+Alternatively, the test can also be run using atest command.
+
+```
+atest FlacDecoderTest -- --enable-module-dynamic-download=true
+```
diff --git a/media/libstagefright/foundation/AString.cpp b/media/libstagefright/foundation/AString.cpp
index fb51cc5..a8adff5 100644
--- a/media/libstagefright/foundation/AString.cpp
+++ b/media/libstagefright/foundation/AString.cpp
@@ -365,8 +365,6 @@
// static
AString AString::FromParcel(const Parcel &parcel) {
size_t size = static_cast<size_t>(parcel.readInt32());
- // The static analyzer incorrectly reports a false-positive here in c++17.
- // https://bugs.llvm.org/show_bug.cgi?id=38176 . NOLINTNEXTLINE
return AString(static_cast<const char *>(parcel.readInplace(size)), size);
}
diff --git a/media/libstagefright/foundation/Android.bp b/media/libstagefright/foundation/Android.bp
index 533cd72..effbb4e 100644
--- a/media/libstagefright/foundation/Android.bp
+++ b/media/libstagefright/foundation/Android.bp
@@ -34,10 +34,6 @@
"media_plugin_headers",
],
- export_shared_lib_headers: [
- "libbinder",
- ],
-
cflags: [
"-Wno-multichar",
"-Werror",
@@ -65,6 +61,7 @@
"AudioPresentationInfo.cpp",
"ByteUtils.cpp",
"ColorUtils.cpp",
+ "FoundationUtils.cpp",
"MediaBuffer.cpp",
"MediaBufferBase.cpp",
"MediaBufferGroup.cpp",
diff --git a/media/libstagefright/foundation/FoundationUtils.cpp b/media/libstagefright/foundation/FoundationUtils.cpp
new file mode 100644
index 0000000..8285e4c
--- /dev/null
+++ b/media/libstagefright/foundation/FoundationUtils.cpp
@@ -0,0 +1,112 @@
+/*
+ * Copyright (C) 2009 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "FoundationUtils"
+#include <utils/Log.h>
+#include <ctype.h>
+#include <stdio.h>
+#include <unistd.h>
+#include <sys/stat.h>
+#include <sys/types.h>
+
+#include <cutils/properties.h>
+#include <media/stagefright/foundation/AString.h>
+
+namespace android {
+
+AString uriDebugString(const AString &uri, bool incognito) {
+ if (incognito) {
+ return AString("<URI suppressed>");
+ }
+
+ if (property_get_bool("media.stagefright.log-uri", false)) {
+ return uri;
+ }
+
+ // find scheme
+ AString scheme;
+ const char *chars = uri.c_str();
+ for (size_t i = 0; i < uri.size(); i++) {
+ const char c = chars[i];
+ if (!isascii(c)) {
+ break;
+ } else if (isalpha(c)) {
+ continue;
+ } else if (i == 0) {
+ // first character must be a letter
+ break;
+ } else if (isdigit(c) || c == '+' || c == '.' || c =='-') {
+ continue;
+ } else if (c != ':') {
+ break;
+ }
+ scheme = AString(uri, 0, i);
+ scheme.append("://<suppressed>");
+ return scheme;
+ }
+ return AString("<no-scheme URI suppressed>");
+}
+
+AString MakeUserAgent() {
+ AString ua;
+ ua.append("stagefright/1.2 (Linux;Android ");
+
+#if (PROPERTY_VALUE_MAX < 8)
+#error "PROPERTY_VALUE_MAX must be at least 8"
+#endif
+
+ char value[PROPERTY_VALUE_MAX];
+ property_get("ro.build.version.release", value, "Unknown");
+ ua.append(value);
+ ua.append(")");
+
+ return ua;
+}
+
+AString nameForFd(int fd) {
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ AString result;
+ snprintf(buffer, SIZE, "/proc/%d/fd/%d", getpid(), fd);
+ struct stat s;
+ if (lstat(buffer, &s) == 0) {
+ if ((s.st_mode & S_IFMT) == S_IFLNK) {
+ char linkto[256];
+ int len = readlink(buffer, linkto, sizeof(linkto));
+ if(len > 0) {
+ if(len > 255) {
+ linkto[252] = '.';
+ linkto[253] = '.';
+ linkto[254] = '.';
+ linkto[255] = 0;
+ } else {
+ linkto[len] = 0;
+ }
+ result.append(linkto);
+ }
+ } else {
+ result.append("unexpected type for ");
+ result.append(buffer);
+ }
+ } else {
+ result.append("couldn't open ");
+ result.append(buffer);
+ }
+ return result;
+}
+
+} // namespace android
diff --git a/media/libstagefright/httplive/Android.bp b/media/libstagefright/httplive/Android.bp
index c0ee14e..12e7ca6 100644
--- a/media/libstagefright/httplive/Android.bp
+++ b/media/libstagefright/httplive/Android.bp
@@ -31,6 +31,7 @@
"liblog",
"libcrypto",
"libcutils",
+ "libdatasource",
"libmedia",
"libmediandk",
"libstagefright",
diff --git a/media/libstagefright/httplive/HTTPDownloader.cpp b/media/libstagefright/httplive/HTTPDownloader.cpp
index c7e92cd..68f1de9 100644
--- a/media/libstagefright/httplive/HTTPDownloader.cpp
+++ b/media/libstagefright/httplive/HTTPDownloader.cpp
@@ -21,13 +21,13 @@
#include "HTTPDownloader.h"
#include "M3UParser.h"
+#include <datasource/MediaHTTP.h>
+#include <datasource/FileSource.h>
#include <media/DataSource.h>
#include <media/MediaHTTPConnection.h>
#include <media/MediaHTTPService.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/ADebug.h>
-#include <media/stagefright/ClearMediaHTTP.h>
-#include <media/stagefright/ClearFileSource.h>
#include <openssl/aes.h>
#include <openssl/md5.h>
#include <utils/Mutex.h>
@@ -38,7 +38,7 @@
HTTPDownloader::HTTPDownloader(
const sp<MediaHTTPService> &httpService,
const KeyedVector<String8, String8> &headers) :
- mHTTPDataSource(new ClearMediaHTTP(httpService->makeHTTPConnection())),
+ mHTTPDataSource(new MediaHTTP(httpService->makeHTTPConnection())),
mExtraHeaders(headers),
mDisconnecting(false) {
}
@@ -91,7 +91,7 @@
if (reconnect) {
if (!strncasecmp(url, "file://", 7)) {
- mDataSource = new ClearFileSource(url + 7);
+ mDataSource = new FileSource(url + 7);
} else if (strncasecmp(url, "http://", 7)
&& strncasecmp(url, "https://", 8)) {
return ERROR_UNSUPPORTED;
diff --git a/media/libstagefright/httplive/LiveSession.cpp b/media/libstagefright/httplive/LiveSession.cpp
index 9cf97c7..3bad015 100644
--- a/media/libstagefright/httplive/LiveSession.cpp
+++ b/media/libstagefright/httplive/LiveSession.cpp
@@ -34,6 +34,7 @@
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/Utils.h>
+#include <media/stagefright/FoundationUtils.h>
#include <utils/Mutex.h>
diff --git a/media/libstagefright/httplive/M3UParser.cpp b/media/libstagefright/httplive/M3UParser.cpp
index cb97a3c..e0324e3 100644
--- a/media/libstagefright/httplive/M3UParser.cpp
+++ b/media/libstagefright/httplive/M3UParser.cpp
@@ -27,6 +27,7 @@
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/Utils.h>
+#include <media/stagefright/FoundationUtils.h>
#include <media/mediaplayer.h>
namespace android {
diff --git a/media/libstagefright/httplive/PlaylistFetcher.cpp b/media/libstagefright/httplive/PlaylistFetcher.cpp
index 635ecfe..4d0848a 100644
--- a/media/libstagefright/httplive/PlaylistFetcher.cpp
+++ b/media/libstagefright/httplive/PlaylistFetcher.cpp
@@ -28,17 +28,18 @@
#include "mpeg2ts/AnotherPacketSource.h"
#include "mpeg2ts/HlsSampleDecryptor.h"
+#include <datasource/DataURISource.h>
#include <media/stagefright/foundation/ABitReader.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/ByteUtils.h>
#include <media/stagefright/foundation/MediaKeys.h>
#include <media/stagefright/foundation/avc_utils.h>
-#include <media/stagefright/DataURISource.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/MetaDataUtils.h>
#include <media/stagefright/Utils.h>
+#include <media/stagefright/FoundationUtils.h>
#include <ctype.h>
#include <inttypes.h>
diff --git a/media/libstagefright/id3/Android.bp b/media/libstagefright/id3/Android.bp
index 7151d07..c8173cf 100644
--- a/media/libstagefright/id3/Android.bp
+++ b/media/libstagefright/id3/Android.bp
@@ -4,6 +4,7 @@
srcs: ["ID3.cpp"],
header_libs: [
+ "libmedia_headers",
"media_ndk_headers",
],
@@ -33,6 +34,7 @@
],
shared_libs: [
+ "libdatasource",
"libstagefright",
"libutils",
"liblog",
diff --git a/media/libstagefright/id3/ID3.cpp b/media/libstagefright/id3/ID3.cpp
index 792a68a..425468f 100644
--- a/media/libstagefright/id3/ID3.cpp
+++ b/media/libstagefright/id3/ID3.cpp
@@ -32,7 +32,7 @@
static const size_t kMaxMetadataSize = 3 * 1024 * 1024;
-struct MemorySource : public DataSourceBase {
+struct ID3::MemorySource : public DataSourceBase {
MemorySource(const uint8_t *data, size_t size)
: mData(data),
mSize(size) {
@@ -58,7 +58,7 @@
DISALLOW_EVIL_CONSTRUCTORS(MemorySource);
};
-class DataSourceUnwrapper : public DataSourceBase {
+class ID3::DataSourceUnwrapper : public DataSourceBase {
public:
explicit DataSourceUnwrapper(DataSourceHelper *sourcehelper) {
diff --git a/media/libstagefright/id3/test/Android.bp b/media/libstagefright/id3/test/Android.bp
new file mode 100644
index 0000000..9d26eec
--- /dev/null
+++ b/media/libstagefright/id3/test/Android.bp
@@ -0,0 +1,48 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+cc_test {
+ name: "ID3Test",
+ gtest: true,
+
+ srcs: ["ID3Test.cpp"],
+
+ static_libs: [
+ "libdatasource",
+ "libstagefright_id3",
+ "libstagefright",
+ "libstagefright_foundation",
+ ],
+
+ shared_libs: [
+ "libutils",
+ "liblog",
+ "libbinder",
+ ],
+
+ cflags: [
+ "-Werror",
+ "-Wall",
+ ],
+
+ sanitize: {
+ cfi: true,
+ misc_undefined: [
+ "unsigned-integer-overflow",
+ "signed-integer-overflow",
+ ],
+ },
+}
diff --git a/media/libstagefright/id3/test/AndroidTest.xml b/media/libstagefright/id3/test/AndroidTest.xml
new file mode 100644
index 0000000..6c6697d
--- /dev/null
+++ b/media/libstagefright/id3/test/AndroidTest.xml
@@ -0,0 +1,31 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!-- Copyright (C) 2020 The Android Open Source Project
+
+ Licensed under the Apache License, Version 2.0 (the "License");
+ you may not use this file except in compliance with the License.
+ You may obtain a copy of the License at
+
+ http://www.apache.org/licenses/LICENSE-2.0
+
+ Unless required by applicable law or agreed to in writing, software
+ distributed under the License is distributed on an "AS IS" BASIS,
+ WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ See the License for the specific language governing permissions and
+ limitations under the License.
+-->
+<configuration description="Test module config for ID3 unit tests">
+ <option name="test-suite-tag" value="ID3Test" />
+ <target_preparer class="com.android.tradefed.targetprep.PushFilePreparer">
+ <option name="cleanup" value="true" />
+ <option name="push" value="ID3Test->/data/local/tmp/ID3Test" />
+ <option name="push-file"
+ key="https://storage.googleapis.com/android_media/frameworks/av/media/libstagefright/id3/test/ID3Test.zip?unzip=true"
+ value="/data/local/tmp/ID3TestRes/" />
+ </target_preparer>
+
+ <test class="com.android.tradefed.testtype.GTest" >
+ <option name="native-test-device-path" value="/data/local/tmp" />
+ <option name="module-name" value="ID3Test" />
+ <option name="native-test-flag" value="-P /data/local/tmp/ID3TestRes/" />
+ </test>
+</configuration>
diff --git a/media/libstagefright/id3/test/ID3Test.cpp b/media/libstagefright/id3/test/ID3Test.cpp
new file mode 100644
index 0000000..a8f1470
--- /dev/null
+++ b/media/libstagefright/id3/test/ID3Test.cpp
@@ -0,0 +1,217 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "ID3Test"
+#include <utils/Log.h>
+
+#include <ctype.h>
+#include <string>
+#include <sys/stat.h>
+#include <datasource/FileSource.h>
+
+#include <media/stagefright/foundation/hexdump.h>
+#include <ID3.h>
+
+#include "ID3TestEnvironment.h"
+
+using namespace android;
+
+static ID3TestEnvironment *gEnv = nullptr;
+
+class ID3tagTest : public ::testing::TestWithParam<string> {};
+class ID3versionTest : public ::testing::TestWithParam<pair<string, int>> {};
+class ID3textTagTest : public ::testing::TestWithParam<pair<string, int>> {};
+class ID3albumArtTest : public ::testing::TestWithParam<pair<string, bool>> {};
+class ID3multiAlbumArtTest : public ::testing::TestWithParam<pair<string, int>> {};
+
+TEST_P(ID3tagTest, TagTest) {
+ string path = gEnv->getRes() + GetParam();
+ sp<FileSource> file = new FileSource(path.c_str());
+ ASSERT_EQ(file->initCheck(), (status_t)OK) << "File initialization failed! \n";
+ ID3 tag(file.get());
+ ASSERT_TRUE(tag.isValid()) << "No valid ID3 tag found for " << path.c_str() << "\n";
+
+ ID3::Iterator it(tag, nullptr);
+ while (!it.done()) {
+ String8 id;
+ it.getID(&id);
+ ASSERT_GT(id.length(), 0) << "No ID tag found! \n";
+ ALOGV("Found ID tag: %s\n", String8(id).c_str());
+ it.next();
+ }
+}
+
+TEST_P(ID3versionTest, VersionTest) {
+ int versionNumber = GetParam().second;
+ string path = gEnv->getRes() + GetParam().first;
+ sp<android::FileSource> file = new FileSource(path.c_str());
+ ASSERT_EQ(file->initCheck(), (status_t)OK) << "File initialization failed! \n";
+
+ ID3 tag(file.get());
+ ASSERT_TRUE(tag.isValid()) << "No valid ID3 tag found for " << path.c_str() << "\n";
+ ASSERT_TRUE(tag.version() >= versionNumber)
+ << "Expected version: " << tag.version() << " Found version: " << versionNumber;
+}
+
+TEST_P(ID3textTagTest, TextTagTest) {
+ int numTextFrames = GetParam().second;
+ string path = gEnv->getRes() + GetParam().first;
+ sp<android::FileSource> file = new FileSource(path.c_str());
+ ASSERT_EQ(file->initCheck(), (status_t)OK) << "File initialization failed! \n";
+
+ ID3 tag(file.get());
+ ASSERT_TRUE(tag.isValid()) << "No valid ID3 tag found for " << path.c_str() << "\n";
+ int countTextFrames = 0;
+ ID3::Iterator it(tag, nullptr);
+ while (!it.done()) {
+ String8 id;
+ it.getID(&id);
+ ASSERT_GT(id.length(), 0);
+ if (id[0] == 'T') {
+ String8 text;
+ countTextFrames++;
+ it.getString(&text);
+ ALOGV("Found text frame %s : %s \n", id.string(), text.string());
+ }
+ it.next();
+ }
+ ASSERT_EQ(countTextFrames, numTextFrames)
+ << "Expected " << numTextFrames << " text frames, found " << countTextFrames;
+}
+
+TEST_P(ID3albumArtTest, AlbumArtTest) {
+ bool albumArtPresent = GetParam().second;
+ string path = gEnv->getRes() + GetParam().first;
+ sp<android::FileSource> file = new FileSource(path.c_str());
+ ASSERT_EQ(file->initCheck(), (status_t)OK) << "File initialization failed! \n";
+
+ ID3 tag(file.get());
+ ASSERT_TRUE(tag.isValid()) << "No valid ID3 tag found for " << path.c_str() << "\n";
+ size_t dataSize;
+ String8 mime;
+ const void *data = tag.getAlbumArt(&dataSize, &mime);
+
+ if (albumArtPresent) {
+ if (data) {
+ ALOGV("Found album art: size = %zu mime = %s \n", dataSize, mime.string());
+ }
+ ASSERT_NE(data, nullptr) << "Expected album art, found none!" << path;
+ } else {
+ ASSERT_EQ(data, nullptr) << "Found album art when expected none!";
+ }
+#if (LOG_NDEBUG == 0)
+ hexdump(data, dataSize > 128 ? 128 : dataSize);
+#endif
+}
+
+TEST_P(ID3multiAlbumArtTest, MultiAlbumArtTest) {
+ int numAlbumArt = GetParam().second;
+ string path = gEnv->getRes() + GetParam().first;
+ sp<android::FileSource> file = new FileSource(path.c_str());
+ ASSERT_EQ(file->initCheck(), (status_t)OK) << "File initialization failed! \n";
+
+ ID3 tag(file.get());
+ ASSERT_TRUE(tag.isValid()) << "No valid ID3 tag found for " << path.c_str() << "\n";
+ int count = 0;
+ ID3::Iterator it(tag, nullptr);
+ while (!it.done()) {
+ String8 id;
+ it.getID(&id);
+ ASSERT_GT(id.length(), 0);
+ // Check if the tag is an "APIC/PIC" tag.
+ if (String8(id) == "APIC" || String8(id) == "PIC") {
+ count++;
+ size_t dataSize;
+ String8 mime;
+ const void *data = tag.getAlbumArt(&dataSize, &mime);
+ if (data) {
+ ALOGV("Found album art: size = %zu mime = %s \n", dataSize, mime.string());
+#if (LOG_NDEBUG == 0)
+ hexdump(data, dataSize > 128 ? 128 : dataSize);
+#endif
+ }
+ ASSERT_NE(data, nullptr) << "Expected album art, found none!" << path;
+ }
+ it.next();
+ }
+ ASSERT_EQ(count, numAlbumArt) << "Found " << count << " album arts, expected " << numAlbumArt
+ << " album arts! \n";
+}
+
+INSTANTIATE_TEST_SUITE_P(id3TestAll, ID3tagTest,
+ ::testing::Values("bbb_44100hz_2ch_128kbps_mp3_30sec.mp3",
+ "bbb_44100hz_2ch_128kbps_mp3_30sec_1_image.mp3",
+ "bbb_44100hz_2ch_128kbps_mp3_30sec_2_image.mp3",
+ "bbb_44100hz_2ch_128kbps_mp3_5mins.mp3",
+ "bbb_44100hz_2ch_128kbps_mp3_5mins_1_image.mp3",
+ "bbb_44100hz_2ch_128kbps_mp3_5mins_2_image.mp3",
+ "bbb_44100hz_2ch_128kbps_mp3_5mins_largeSize.mp3",
+ "bbb_44100hz_2ch_128kbps_mp3_30sec_moreTextFrames.mp3"));
+
+INSTANTIATE_TEST_SUITE_P(
+ id3TestAll, ID3versionTest,
+ ::testing::Values(make_pair("bbb_44100hz_2ch_128kbps_mp3_30sec.mp3", 4),
+ make_pair("bbb_44100hz_2ch_128kbps_mp3_30sec_1_image.mp3", 4),
+ make_pair("bbb_44100hz_2ch_128kbps_mp3_30sec_2_image.mp3", 4),
+ make_pair("bbb_44100hz_2ch_128kbps_mp3_5mins.mp3", 4),
+ make_pair("bbb_44100hz_2ch_128kbps_mp3_5mins_1_image.mp3", 4),
+ make_pair("bbb_44100hz_2ch_128kbps_mp3_5mins_2_image.mp3", 4),
+ make_pair("bbb_44100hz_2ch_128kbps_mp3_5mins_largeSize.mp3", 4),
+ make_pair("bbb_44100hz_2ch_128kbps_mp3_30sec_moreTextFrames.mp3", 4)));
+
+INSTANTIATE_TEST_SUITE_P(
+ id3TestAll, ID3textTagTest,
+ ::testing::Values(make_pair("bbb_44100hz_2ch_128kbps_mp3_30sec.mp3", 1),
+ make_pair("bbb_44100hz_2ch_128kbps_mp3_30sec_1_image.mp3", 1),
+ make_pair("bbb_44100hz_2ch_128kbps_mp3_30sec_2_image.mp3", 1),
+ make_pair("bbb_44100hz_2ch_128kbps_mp3_5mins.mp3", 1),
+ make_pair("bbb_44100hz_2ch_128kbps_mp3_5mins_1_image.mp3", 1),
+ make_pair("bbb_44100hz_2ch_128kbps_mp3_5mins_2_image.mp3", 1),
+ make_pair("bbb_44100hz_2ch_128kbps_mp3_5mins_largeSize.mp3", 1),
+ make_pair("bbb_44100hz_2ch_128kbps_mp3_30sec_moreTextFrames.mp3", 5)));
+
+INSTANTIATE_TEST_SUITE_P(
+ id3TestAll, ID3albumArtTest,
+ ::testing::Values(make_pair("bbb_44100hz_2ch_128kbps_mp3_30sec.mp3", false),
+ make_pair("bbb_44100hz_2ch_128kbps_mp3_30sec_1_image.mp3", true),
+ make_pair("bbb_44100hz_2ch_128kbps_mp3_30sec_2_image.mp3", true),
+ make_pair("bbb_44100hz_2ch_128kbps_mp3_5mins.mp3", false),
+ make_pair("bbb_44100hz_2ch_128kbps_mp3_5mins_1_image.mp3", true),
+ make_pair("bbb_44100hz_2ch_128kbps_mp3_5mins_2_image.mp3", true),
+ make_pair("bbb_44100hz_2ch_128kbps_mp3_5mins_largeSize.mp3", true)));
+
+INSTANTIATE_TEST_SUITE_P(
+ id3TestAll, ID3multiAlbumArtTest,
+ ::testing::Values(make_pair("bbb_44100hz_2ch_128kbps_mp3_30sec.mp3", 0),
+ make_pair("bbb_44100hz_2ch_128kbps_mp3_5mins.mp3", 0),
+ make_pair("bbb_44100hz_2ch_128kbps_mp3_30sec_1_image.mp3", 1),
+ make_pair("bbb_44100hz_2ch_128kbps_mp3_5mins_1_image.mp3", 1),
+ make_pair("bbb_44100hz_2ch_128kbps_mp3_30sec_2_image.mp3", 2),
+ make_pair("bbb_44100hz_2ch_128kbps_mp3_5mins_2_image.mp3", 2),
+ make_pair("bbb_44100hz_2ch_128kbps_mp3_5mins_largeSize.mp3", 3)));
+
+int main(int argc, char **argv) {
+ gEnv = new ID3TestEnvironment();
+ ::testing::AddGlobalTestEnvironment(gEnv);
+ ::testing::InitGoogleTest(&argc, argv);
+ int status = gEnv->initFromOptions(argc, argv);
+ if (status == 0) {
+ status = RUN_ALL_TESTS();
+ ALOGI("ID3 Test result = %d\n", status);
+ }
+ return status;
+}
diff --git a/media/libstagefright/id3/test/ID3TestEnvironment.h b/media/libstagefright/id3/test/ID3TestEnvironment.h
new file mode 100644
index 0000000..2229718
--- /dev/null
+++ b/media/libstagefright/id3/test/ID3TestEnvironment.h
@@ -0,0 +1,73 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __ID3_TEST_ENVIRONMENT_H__
+#define __ID3_TEST_ENVIRONMENT_H__
+
+#include <gtest/gtest.h>
+
+#include <getopt.h>
+
+using namespace std;
+
+class ID3TestEnvironment : public::testing::Environment {
+ public:
+ ID3TestEnvironment() : res("/data/local/tmp/") {}
+
+ // Parses the command line arguments
+ int initFromOptions(int argc, char **argv);
+
+ void setRes(const char *_res) { res = _res; }
+
+ const string getRes() const { return res; }
+
+ private:
+ string res;
+};
+
+int ID3TestEnvironment::initFromOptions(int argc, char **argv) {
+ static struct option options[] = {{"path", required_argument, 0, 'P'}, {0, 0, 0, 0}};
+
+ while (true) {
+ int index = 0;
+ int c = getopt_long(argc, argv, "P:", options, &index);
+ if (c == -1) {
+ break;
+ }
+
+ switch (c) {
+ case 'P': {
+ setRes(optarg);
+ break;
+ }
+ default:
+ break;
+ }
+ }
+
+ if (optind < argc) {
+ fprintf(stderr,
+ "unrecognized option: %s\n\n"
+ "usage: %s <gtest options> <test options>\n\n"
+ "test options are:\n\n"
+ "-P, --path: Resource files directory location\n",
+ argv[optind ?: 1], argv[0]);
+ return 2;
+ }
+ return 0;
+}
+
+#endif // __ID3_TEST_ENVIRONMENT_H__
diff --git a/media/libstagefright/id3/test/README.md b/media/libstagefright/id3/test/README.md
new file mode 100644
index 0000000..7fd8901
--- /dev/null
+++ b/media/libstagefright/id3/test/README.md
@@ -0,0 +1,40 @@
+## Media Testing ##
+---
+#### ID3 Test :
+The ID3 Test Suite validates the ID3 parser available in libstagefright.
+
+Run the following command in the id3 folder to build the test suite:
+```
+m ID3Test
+```
+
+The 32-bit binaries will be created in the following path : ${OUT}/data/nativetest/
+
+The 64-bit binaries will be created in the following path : ${OUT}/data/nativetest64/
+
+To test 64-bit binary push binaries from nativetest64.
+```
+adb push ${OUT}/data/nativetest64/ID3Test/ID3Test /data/local/tmp/
+```
+
+To test 32-bit binary push binaries from nativetest.
+```
+adb push ${OUT}/data/nativetest/ID3Test/ID3Test /data/local/tmp/
+```
+
+The resource file for the tests is taken from [here](https://storage.googleapis.com/android_media/frameworks/av/media/libstagefright/id3/test/ID3Test.zip ).
+Download, unzip and push these files into device for testing.
+
+```
+adb push ID3Test /data/local/tmp/
+```
+
+usage: ID3Test -P \<path_to_folder\>
+```
+adb shell /data/local/tmp/ID3Test -P /data/local/tmp/ID3/
+```
+Alternatively, the test can also be run using atest command.
+
+```
+atest ID3Test -- --enable-module-dynamic-download=true
+```
diff --git a/media/libstagefright/id3/testid3.cpp b/media/libstagefright/id3/testid3.cpp
index 86e6adf..9984d85 100644
--- a/media/libstagefright/id3/testid3.cpp
+++ b/media/libstagefright/id3/testid3.cpp
@@ -22,7 +22,7 @@
#include <dirent.h>
#include <binder/ProcessState.h>
-#include <media/stagefright/FileSource.h>
+#include <datasource/FileSource.h>
#include <media/stagefright/foundation/ADebug.h>
#define MAXPATHLEN 256
diff --git a/media/libstagefright/include/ACodecBufferChannel.h b/media/libstagefright/include/ACodecBufferChannel.h
index 7c01e45..3a087d1 100644
--- a/media/libstagefright/include/ACodecBufferChannel.h
+++ b/media/libstagefright/include/ACodecBufferChannel.h
@@ -25,7 +25,7 @@
#include <media/openmax/OMX_Types.h>
#include <media/stagefright/CodecBase.h>
-#include <media/ICrypto.h>
+#include <mediadrm/ICrypto.h>
#include <media/IOMX.h>
namespace android {
diff --git a/media/libstagefright/include/CallbackDataSource.h b/media/libstagefright/include/CallbackDataSource.h
index 9f413cd..e428494 100644
--- a/media/libstagefright/include/CallbackDataSource.h
+++ b/media/libstagefright/include/CallbackDataSource.h
@@ -41,7 +41,6 @@
virtual String8 toString() {
return mName;
}
- virtual sp<DecryptHandle> DrmInitialization(const char *mime = NULL);
virtual sp<IDataSource> getIDataSource() const;
private:
@@ -70,7 +69,6 @@
virtual String8 toString() {
return mName;
}
- virtual sp<DecryptHandle> DrmInitialization(const char *mime = NULL);
virtual sp<IDataSource> getIDataSource() const;
private:
diff --git a/media/libstagefright/include/ID3.h b/media/libstagefright/include/ID3.h
index 5e433ea..2843a7a 100644
--- a/media/libstagefright/include/ID3.h
+++ b/media/libstagefright/include/ID3.h
@@ -77,6 +77,8 @@
size_t rawSize() const { return mRawSize; }
private:
+ class DataSourceUnwrapper;
+ struct MemorySource;
bool mIsValid;
uint8_t *mData;
size_t mSize;
diff --git a/media/libstagefright/include/SecureBuffer.h b/media/libstagefright/include/SecureBuffer.h
index cf7933a..c45e0e5 100644
--- a/media/libstagefright/include/SecureBuffer.h
+++ b/media/libstagefright/include/SecureBuffer.h
@@ -18,7 +18,7 @@
#define SECURE_BUFFER_H_
-#include <media/ICrypto.h>
+#include <mediadrm/ICrypto.h>
#include <media/MediaCodecBuffer.h>
namespace android {
diff --git a/media/libstagefright/include/ThrottledSource.h b/media/libstagefright/include/ThrottledSource.h
index 71e62f7..5ae0653 100644
--- a/media/libstagefright/include/ThrottledSource.h
+++ b/media/libstagefright/include/ThrottledSource.h
@@ -54,10 +54,6 @@
return mSource->reconnectAtOffset(offset);
}
- virtual sp<DecryptHandle> DrmInitialization(const char *mime = NULL) {
- return mSource->DrmInitialization(mime);
- }
-
virtual String8 getMIMEType() const {
return mSource->getMIMEType();
}
diff --git a/media/libstagefright/include/media/stagefright/DataSource.h b/media/libstagefright/include/media/stagefright/DataSource.h
index 1f7a473..83d3e5d 100644
--- a/media/libstagefright/include/media/stagefright/DataSource.h
+++ b/media/libstagefright/include/media/stagefright/DataSource.h
@@ -52,11 +52,6 @@
////////////////////////////////////////////////////////////////////////////
- // for DRM
- virtual sp<DecryptHandle> DrmInitialization(const char * /*mime*/ = NULL) {
- return NULL;
- }
-
virtual String8 getUri() {
return String8();
}
diff --git a/media/libstagefright/include/media/stagefright/NdkUtils.h b/media/libstagefright/include/media/stagefright/FoundationUtils.h
similarity index 66%
rename from media/libstagefright/include/media/stagefright/NdkUtils.h
rename to media/libstagefright/include/media/stagefright/FoundationUtils.h
index a68884a..1548981 100644
--- a/media/libstagefright/include/media/stagefright/NdkUtils.h
+++ b/media/libstagefright/include/media/stagefright/FoundationUtils.h
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2018 The Android Open Source Project
+ * Copyright (C) 2019 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
@@ -14,18 +14,19 @@
* limitations under the License.
*/
-#ifndef NDK_UTILS_H_
+#ifndef FOUNDATION_UTILS_H_
-#define NDK_UTILS_H_
+#define FOUNDATION_UTILS_H_
-#include <media/stagefright/MetaData.h>
-#include <media/NdkWrapper.h>
+#include <media/stagefright/foundation/AString.h>
namespace android {
-sp<MetaData> convertMediaFormatWrapperToMetaData(
- const sp<AMediaFormatWrapper> &fmt);
+AString MakeUserAgent();
+AString uriDebugString(const AString &uri, bool incognito = false);
+
+AString nameForFd(int fd);
} // namespace android
-#endif // NDK_UTILS_H_
+#endif // FOUNDATION_UTILS_H_
diff --git a/media/libstagefright/include/media/stagefright/MediaCodec.h b/media/libstagefright/include/media/stagefright/MediaCodec.h
index cd30347..01d0325 100644
--- a/media/libstagefright/include/media/stagefright/MediaCodec.h
+++ b/media/libstagefright/include/media/stagefright/MediaCodec.h
@@ -25,7 +25,7 @@
#include <media/hardware/CryptoAPI.h>
#include <media/MediaCodecInfo.h>
#include <media/MediaResource.h>
-#include <media/MediaAnalyticsItem.h>
+#include <media/MediaMetrics.h>
#include <media/stagefright/foundation/AHandler.h>
#include <media/stagefright/FrameRenderTracker.h>
#include <utils/Vector.h>
@@ -189,7 +189,7 @@
status_t getCodecInfo(sp<MediaCodecInfo> *codecInfo) const;
- status_t getMetrics(MediaAnalyticsItem * &reply);
+ status_t getMetrics(mediametrics_handle_t &reply);
status_t setParameters(const sp<AMessage> ¶ms);
@@ -328,11 +328,11 @@
sp<Surface> mSurface;
SoftwareRenderer *mSoftRenderer;
- MediaAnalyticsItem *mAnalyticsItem;
- void initAnalyticsItem();
- void updateAnalyticsItem();
- void flushAnalyticsItem();
- void updateEphemeralAnalytics(MediaAnalyticsItem *item);
+ mediametrics_handle_t mMetricsHandle;
+ void initMediametrics();
+ void updateMediametrics();
+ void flushMediametrics();
+ void updateEphemeralMediametrics(mediametrics_handle_t item);
sp<AMessage> mOutputFormat;
sp<AMessage> mInputFormat;
diff --git a/media/libstagefright/include/media/stagefright/MediaCodecList.h b/media/libstagefright/include/media/stagefright/MediaCodecList.h
index e44b0a4..e681d25 100644
--- a/media/libstagefright/include/media/stagefright/MediaCodecList.h
+++ b/media/libstagefright/include/media/stagefright/MediaCodecList.h
@@ -83,6 +83,7 @@
};
static sp<BinderDeathObserver> sBinderDeathObserver;
+ static sp<IBinder> sMediaPlayer;
static sp<IMediaCodecList> sCodecList;
static sp<IMediaCodecList> sRemoteList;
diff --git a/media/libstagefright/include/media/stagefright/MediaErrors.h b/media/libstagefright/include/media/stagefright/MediaErrors.h
index 09639e2..6f48c5d 100644
--- a/media/libstagefright/include/media/stagefright/MediaErrors.h
+++ b/media/libstagefright/include/media/stagefright/MediaErrors.h
@@ -99,7 +99,13 @@
ERROR_CAS_DEVICE_REVOKED = CAS_ERROR_BASE - 9,
ERROR_CAS_RESOURCE_BUSY = CAS_ERROR_BASE - 10,
ERROR_CAS_INSUFFICIENT_OUTPUT_PROTECTION = CAS_ERROR_BASE - 11,
- ERROR_CAS_LAST_USED_ERRORCODE = CAS_ERROR_BASE - 11,
+ ERROR_CAS_NEED_ACTIVATION = CAS_ERROR_BASE - 12,
+ ERROR_CAS_NEED_PAIRING = CAS_ERROR_BASE - 13,
+ ERROR_CAS_NO_CARD = CAS_ERROR_BASE - 14,
+ ERROR_CAS_CARD_MUTE = CAS_ERROR_BASE - 15,
+ ERROR_CAS_CARD_INVALID = CAS_ERROR_BASE - 16,
+ ERROR_CAS_BLACKOUT = CAS_ERROR_BASE - 17,
+ ERROR_CAS_LAST_USED_ERRORCODE = CAS_ERROR_BASE - 17,
ERROR_CAS_VENDOR_MAX = CAS_ERROR_BASE - 500,
ERROR_CAS_VENDOR_MIN = CAS_ERROR_BASE - 999,
diff --git a/media/libstagefright/include/media/stagefright/RemoteDataSource.h b/media/libstagefright/include/media/stagefright/RemoteDataSource.h
index e191e6a..5a69bd7 100644
--- a/media/libstagefright/include/media/stagefright/RemoteDataSource.h
+++ b/media/libstagefright/include/media/stagefright/RemoteDataSource.h
@@ -66,9 +66,6 @@
virtual String8 toString() {
return mName;
}
- virtual sp<DecryptHandle> DrmInitialization(const char *mime) {
- return mSource->DrmInitialization(mime);
- }
private:
enum {
diff --git a/media/libstagefright/include/media/stagefright/Utils.h b/media/libstagefright/include/media/stagefright/Utils.h
index e8e0a11..2b9b759 100644
--- a/media/libstagefright/include/media/stagefright/Utils.h
+++ b/media/libstagefright/include/media/stagefright/Utils.h
@@ -41,8 +41,6 @@
// TODO: combine this with avc_utils::getNextNALUnit
const uint8_t *findNextNalStartCode(const uint8_t *data, size_t length);
-AString MakeUserAgent();
-
// Convert a MIME type to a AudioSystem::audio_format
status_t mapMimeToAudioFormat(audio_format_t& format, const char* mime);
@@ -60,8 +58,6 @@
bool canOffloadStream(const sp<MetaData>& meta, bool hasVideo,
bool isStreaming, audio_stream_type_t streamType);
-AString uriDebugString(const AString &uri, bool incognito = false);
-
struct HLSTime {
int32_t mSeq;
int64_t mTimeUs;
@@ -85,7 +81,6 @@
void writeToAMessage(const sp<AMessage> &msg, const BufferingSettings &buffering);
void readFromAMessage(const sp<AMessage> &msg, BufferingSettings *buffering /* nonnull */);
-AString nameForFd(int fd);
} // namespace android
#endif // UTILS_H_
diff --git a/media/libstagefright/mpeg2ts/Android.bp b/media/libstagefright/mpeg2ts/Android.bp
index a507b91..cab841c 100644
--- a/media/libstagefright/mpeg2ts/Android.bp
+++ b/media/libstagefright/mpeg2ts/Android.bp
@@ -29,7 +29,6 @@
shared_libs: [
"libcrypto",
- "libmedia",
"libhidlmemory",
"android.hardware.cas.native@1.0",
"android.hidl.memory@1.0",
@@ -37,6 +36,8 @@
],
header_libs: [
+ "libmedia_headers",
+ "libaudioclient_headers",
"media_ndk_headers",
],
diff --git a/media/libstagefright/omx/Android.bp b/media/libstagefright/omx/Android.bp
index 7d03d98..7d612b4 100644
--- a/media/libstagefright/omx/Android.bp
+++ b/media/libstagefright/omx/Android.bp
@@ -45,7 +45,6 @@
"libdl",
"libhidlbase",
"libhidlmemory",
- "libhidltransport",
"libvndksupport",
"android.hardware.media.omx@1.0",
"android.hardware.graphics.bufferqueue@1.0",
diff --git a/media/libstagefright/omx/tests/Android.bp b/media/libstagefright/omx/tests/Android.bp
index 569fa88..eb01543 100644
--- a/media/libstagefright/omx/tests/Android.bp
+++ b/media/libstagefright/omx/tests/Android.bp
@@ -7,6 +7,7 @@
shared_libs: [
"libstagefright",
"libbinder",
+ "libdatasource",
"libmedia",
"libmedia_omx",
"libutils",
diff --git a/media/libstagefright/omx/tests/OMXHarness.cpp b/media/libstagefright/omx/tests/OMXHarness.cpp
index cc8c234..6848a83 100644
--- a/media/libstagefright/omx/tests/OMXHarness.cpp
+++ b/media/libstagefright/omx/tests/OMXHarness.cpp
@@ -27,13 +27,13 @@
#include <binder/ProcessState.h>
#include <binder/IServiceManager.h>
#include <cutils/properties.h>
+#include <datasource/DataSourceFactory.h>
#include <media/DataSource.h>
#include <media/IMediaHTTPService.h>
#include <media/MediaSource.h>
#include <media/OMXBuffer.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/ALooper.h>
-#include <media/stagefright/DataSourceFactory.h>
#include <media/stagefright/InterfaceUtils.h>
#include <media/stagefright/MediaBuffer.h>
#include <media/stagefright/MediaDefs.h>
@@ -278,7 +278,7 @@
static sp<IMediaExtractor> CreateExtractorFromURI(const char *uri) {
sp<DataSource> source =
- DataSourceFactory::CreateFromURI(NULL /* httpService */, uri);
+ DataSourceFactory::getInstance()->CreateFromURI(NULL /* httpService */, uri);
if (source == NULL) {
return NULL;
diff --git a/media/libstagefright/rtsp/ARTSPConnection.cpp b/media/libstagefright/rtsp/ARTSPConnection.cpp
index 789e62a..bb66f4c 100644
--- a/media/libstagefright/rtsp/ARTSPConnection.cpp
+++ b/media/libstagefright/rtsp/ARTSPConnection.cpp
@@ -21,12 +21,14 @@
#include "ARTSPConnection.h"
#include "NetworkUtils.h"
+#include <datasource/HTTPBase.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/AMessage.h>
#include <media/stagefright/foundation/base64.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/Utils.h>
+#include <media/stagefright/FoundationUtils.h>
#include <arpa/inet.h>
#include <fcntl.h>
@@ -34,7 +36,6 @@
#include <openssl/md5.h>
#include <sys/socket.h>
-#include "include/HTTPBase.h"
namespace android {
@@ -953,7 +954,7 @@
CHECK_GE(space2, 0);
method->setTo(request, 0, space1);
- url->setTo(request, space1 + 1, space2 - space1);
+ url->setTo(request, space1 + 1, space2 - space1 - 1);
}
void ARTSPConnection::addAuthentication(AString *request) {
diff --git a/media/libstagefright/rtsp/Android.bp b/media/libstagefright/rtsp/Android.bp
index 9bc9c89..a5a895e 100644
--- a/media/libstagefright/rtsp/Android.bp
+++ b/media/libstagefright/rtsp/Android.bp
@@ -21,6 +21,7 @@
shared_libs: [
"libcrypto",
+ "libdatasource",
"libmedia",
],
diff --git a/media/libstagefright/rtsp/MyHandler.h b/media/libstagefright/rtsp/MyHandler.h
index 48bc8ce..9c30623 100644
--- a/media/libstagefright/rtsp/MyHandler.h
+++ b/media/libstagefright/rtsp/MyHandler.h
@@ -36,18 +36,19 @@
#include <ctype.h>
#include <cutils/properties.h>
+#include <datasource/HTTPBase.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/ALooper.h>
#include <media/stagefright/foundation/AMessage.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/Utils.h>
+#include <media/stagefright/FoundationUtils.h>
#include <arpa/inet.h>
#include <sys/socket.h>
#include <netdb.h>
-#include "HTTPBase.h"
#if LOG_NDEBUG
#define UNUSED_UNLESS_VERBOSE(x) (void)(x)
diff --git a/media/libstagefright/rtsp/SDPLoader.cpp b/media/libstagefright/rtsp/SDPLoader.cpp
index 665d51a..e236267 100644
--- a/media/libstagefright/rtsp/SDPLoader.cpp
+++ b/media/libstagefright/rtsp/SDPLoader.cpp
@@ -22,12 +22,13 @@
#include "ASessionDescription.h"
+#include <datasource/MediaHTTP.h>
#include <media/MediaHTTPConnection.h>
#include <media/MediaHTTPService.h>
-#include <media/stagefright/ClearMediaHTTP.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/Utils.h>
+#include <media/stagefright/FoundationUtils.h>
#define DEFAULT_SDP_SIZE 100000
@@ -41,7 +42,7 @@
mFlags(flags),
mNetLooper(new ALooper),
mCancelled(false),
- mHTTPDataSource(new ClearMediaHTTP(httpService->makeHTTPConnection())) {
+ mHTTPDataSource(new MediaHTTP(httpService->makeHTTPConnection())) {
mNetLooper->setName("sdp net");
mNetLooper->start(false /* runOnCallingThread */,
false /* canCallJava */,
diff --git a/media/libstagefright/tests/writer/Android.bp b/media/libstagefright/tests/writer/Android.bp
new file mode 100644
index 0000000..7e169cb
--- /dev/null
+++ b/media/libstagefright/tests/writer/Android.bp
@@ -0,0 +1,58 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+cc_test {
+ name: "writerTest",
+ gtest: true,
+
+ srcs: [
+ "WriterUtility.cpp",
+ "WriterTest.cpp",
+ ],
+
+ shared_libs: [
+ "libbinder",
+ "libcutils",
+ "liblog",
+ "libutils",
+ ],
+
+ static_libs: [
+ "libstagefright_webm",
+ "libdatasource",
+ "libstagefright",
+ "libstagefright_foundation",
+ "libstagefright_esds",
+ "libogg",
+ ],
+
+ include_dirs: [
+ "frameworks/av/media/libstagefright",
+ ],
+
+ cflags: [
+ "-Werror",
+ "-Wall",
+ ],
+
+ sanitize: {
+ cfi: true,
+ misc_undefined: [
+ "unsigned-integer-overflow",
+ "signed-integer-overflow",
+ ],
+ },
+}
diff --git a/media/libstagefright/tests/writer/AndroidTest.xml b/media/libstagefright/tests/writer/AndroidTest.xml
new file mode 100644
index 0000000..d831555
--- /dev/null
+++ b/media/libstagefright/tests/writer/AndroidTest.xml
@@ -0,0 +1,30 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!-- Copyright (C) 2019 The Android Open Source Project
+
+ Licensed under the Apache License, Version 2.0 (the "License");
+ you may not use this file except in compliance with the License.
+ You may obtain a copy of the License at
+
+ http://www.apache.org/licenses/LICENSE-2.0
+
+ Unless required by applicable law or agreed to in writing, software
+ distributed under the License is distributed on an "AS IS" BASIS,
+ WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ See the License for the specific language governing permissions and
+ limitations under the License.
+-->
+<configuration description="Test module config for writer tests">
+ <option name="test-suite-tag" value="writerTest" />
+ <target_preparer class="com.android.tradefed.targetprep.PushFilePreparer">
+ <option name="cleanup" value="true" />
+ <option name="push" value="writerTest->/data/local/tmp/writerTest" />
+ <option name="push-file"
+ key="https://storage.googleapis.com/android_media/frameworks/av/media/libstagefright/tests/writer/Writer.zip?unzip=true"
+ value="/data/local/tmp/writerTestRes/" />
+ </target_preparer>
+ <test class="com.android.tradefed.testtype.GTest" >
+ <option name="native-test-device-path" value="/data/local/tmp" />
+ <option name="module-name" value="writerTest" />
+ <option name="native-test-flag" value="-P /data/local/tmp/writerTestRes/" />
+ </test>
+</configuration>
diff --git a/media/libstagefright/tests/writer/README.md b/media/libstagefright/tests/writer/README.md
new file mode 100644
index 0000000..ae07917
--- /dev/null
+++ b/media/libstagefright/tests/writer/README.md
@@ -0,0 +1,31 @@
+## Media Testing ##
+---
+#### Writer :
+The Writer Test Suite validates the writers available in libstagefright.
+
+Run the following steps to build the test suite:
+```
+mmm frameworks/av/media/libstagefright/tests/writer/
+```
+
+The 32-bit binaries will be created in the following path : ${OUT}/data/nativetest/
+The 64-bit binaries will be created in the following path : ${OUT}/data/nativetest64/
+
+To test 64-bit binary push binaries from nativetest64.
+
+adb push ${OUT}/data/nativetest64/writerTest/writerTest /data/local/tmp/
+
+To test 32-bit binary push binaries from nativetest.
+
+adb push ${OUT}/data/nativetest/writerTest/writerTest /data/local/tmp/
+
+The resource file for the tests is taken from [here](https://storage.googleapis.com/android_media/frameworks/av/media/libstagefright/tests/writer/writerTestRes.zip).
+Download and extract the folder. Push all the files in this folder to /data/local/tmp/ on the device.
+```
+adb push writerTestRes /data/local/tmp/
+```
+
+usage: writerTest -P \<path_to_res_folder\>
+```
+adb shell /data/local/tmp/writerTest -P /data/local/tmp/
+```
diff --git a/media/libstagefright/tests/writer/WriterTest.cpp b/media/libstagefright/tests/writer/WriterTest.cpp
new file mode 100644
index 0000000..ff063e3
--- /dev/null
+++ b/media/libstagefright/tests/writer/WriterTest.cpp
@@ -0,0 +1,476 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "WriterTest"
+#include <utils/Log.h>
+
+#include <fstream>
+#include <iostream>
+
+#include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/MetaData.h>
+#include <media/stagefright/Utils.h>
+
+#include <media/mediarecorder.h>
+
+#include <media/stagefright/AACWriter.h>
+#include <media/stagefright/AMRWriter.h>
+#include <media/stagefright/MPEG2TSWriter.h>
+#include <media/stagefright/MPEG4Writer.h>
+#include <media/stagefright/OggWriter.h>
+#include <webm/WebmWriter.h>
+
+#include "WriterTestEnvironment.h"
+#include "WriterUtility.h"
+
+#define OUTPUT_FILE_NAME "/data/local/tmp/writer.out"
+
+static WriterTestEnvironment *gEnv = nullptr;
+
+struct configFormat {
+ char mime[128];
+ int32_t width;
+ int32_t height;
+ int32_t sampleRate;
+ int32_t channelCount;
+};
+
+// LookUpTable of clips and metadata for component testing
+static const struct InputData {
+ const char *mime;
+ string inputFile;
+ string info;
+ int32_t firstParam;
+ int32_t secondParam;
+ bool isAudio;
+} kInputData[] = {
+ {MEDIA_MIMETYPE_AUDIO_OPUS, "bbb_opus_stereo_128kbps_48000hz.opus",
+ "bbb_opus_stereo_128kbps_48000hz.info", 48000, 2, true},
+ {MEDIA_MIMETYPE_AUDIO_AAC, "bbb_aac_stereo_128kbps_48000hz.aac",
+ "bbb_aac_stereo_128kbps_48000hz.info", 48000, 2, true},
+ {MEDIA_MIMETYPE_AUDIO_AAC_ADTS, "Mps_2_c2_fr1_Sc1_Dc2_0x03_raw.adts",
+ "Mps_2_c2_fr1_Sc1_Dc2_0x03_raw.info", 48000, 2, true},
+ {MEDIA_MIMETYPE_AUDIO_AMR_NB, "sine_amrnb_1ch_12kbps_8000hz.amrnb",
+ "sine_amrnb_1ch_12kbps_8000hz.info", 8000, 1, true},
+ {MEDIA_MIMETYPE_AUDIO_AMR_WB, "bbb_amrwb_1ch_14kbps_16000hz.amrwb",
+ "bbb_amrwb_1ch_14kbps_16000hz.info", 16000, 1, true},
+ {MEDIA_MIMETYPE_AUDIO_VORBIS, "bbb_vorbis_stereo_128kbps_48000hz.vorbis",
+ "bbb_vorbis_stereo_128kbps_48000hz.info", 48000, 2, true},
+ {MEDIA_MIMETYPE_AUDIO_FLAC, "bbb_flac_stereo_680kbps_48000hz.flac",
+ "bbb_flac_stereo_680kbps_48000hz.info", 48000, 2, true},
+ {MEDIA_MIMETYPE_VIDEO_VP9, "bbb_vp9_176x144_285kbps_60fps.vp9",
+ "bbb_vp9_176x144_285kbps_60fps.info", 176, 144, false},
+ {MEDIA_MIMETYPE_VIDEO_VP8, "bbb_vp8_176x144_240kbps_60fps.vp8",
+ "bbb_vp8_176x144_240kbps_60fps.info", 176, 144, false},
+ {MEDIA_MIMETYPE_VIDEO_AVC, "bbb_avc_176x144_300kbps_60fps.h264",
+ "bbb_avc_176x144_300kbps_60fps.info", 176, 144, false},
+ {MEDIA_MIMETYPE_VIDEO_HEVC, "bbb_hevc_176x144_176kbps_60fps.hevc",
+ "bbb_hevc_176x144_176kbps_60fps.info", 176, 144, false},
+ {MEDIA_MIMETYPE_VIDEO_AV1, "bbb_av1_176_144.av1", "bbb_av1_176_144.info", 176, 144, false},
+ {MEDIA_MIMETYPE_VIDEO_H263, "bbb_h263_352x288_300kbps_12fps.h263",
+ "bbb_h263_352x288_300kbps_12fps.info", 352, 288, false},
+ {MEDIA_MIMETYPE_VIDEO_MPEG4, "bbb_mpeg4_352x288_512kbps_30fps.m4v",
+ "bbb_mpeg4_352x288_512kbps_30fps.info", 352, 288, false},
+};
+
+class WriterTest : public ::testing::TestWithParam<pair<string, int32_t>> {
+ public:
+ WriterTest() : mWriter(nullptr), mFileMeta(nullptr), mCurrentTrack(nullptr) {}
+
+ ~WriterTest() {
+ if (mWriter) {
+ mWriter.clear();
+ mWriter = nullptr;
+ }
+ if (mFileMeta) {
+ mFileMeta.clear();
+ mFileMeta = nullptr;
+ }
+ if (mCurrentTrack) {
+ mCurrentTrack.clear();
+ mCurrentTrack = nullptr;
+ }
+ }
+
+ virtual void SetUp() override {
+ mNumCsds = 0;
+ mInputFrameId = 0;
+ mWriterName = unknown_comp;
+ mDisableTest = false;
+
+ static const std::map<std::string, standardWriters> mapWriter = {
+ {"ogg", OGG}, {"aac", AAC}, {"aac_adts", AAC_ADTS}, {"webm", WEBM},
+ {"mpeg4", MPEG4}, {"amrnb", AMR_NB}, {"amrwb", AMR_WB}, {"mpeg2Ts", MPEG2TS}};
+ // Find the component type
+ string writerFormat = GetParam().first;
+ if (mapWriter.find(writerFormat) != mapWriter.end()) {
+ mWriterName = mapWriter.at(writerFormat);
+ }
+ if (mWriterName == standardWriters::unknown_comp) {
+ cout << "[ WARN ] Test Skipped. No specific writer mentioned\n";
+ mDisableTest = true;
+ }
+ }
+
+ virtual void TearDown() override {
+ mBufferInfo.clear();
+ if (mInputStream.is_open()) mInputStream.close();
+ }
+
+ void getInputBufferInfo(string inputFileName, string inputInfo);
+
+ int32_t createWriter(int32_t fd);
+
+ int32_t addWriterSource(bool isAudio, configFormat params);
+
+ enum standardWriters {
+ OGG,
+ AAC,
+ AAC_ADTS,
+ WEBM,
+ MPEG4,
+ AMR_NB,
+ AMR_WB,
+ MPEG2TS,
+ unknown_comp,
+ };
+
+ standardWriters mWriterName;
+ sp<MediaWriter> mWriter;
+ sp<MetaData> mFileMeta;
+ sp<MediaAdapter> mCurrentTrack;
+
+ bool mDisableTest;
+ int32_t mNumCsds;
+ int32_t mInputFrameId;
+ ifstream mInputStream;
+ vector<BufferInfo> mBufferInfo;
+};
+
+void WriterTest::getInputBufferInfo(string inputFileName, string inputInfo) {
+ std::ifstream eleInfo;
+ eleInfo.open(inputInfo.c_str());
+ ASSERT_EQ(eleInfo.is_open(), true);
+ int32_t bytesCount = 0;
+ uint32_t flags = 0;
+ int64_t timestamp = 0;
+ while (1) {
+ if (!(eleInfo >> bytesCount)) break;
+ eleInfo >> flags;
+ eleInfo >> timestamp;
+ mBufferInfo.push_back({bytesCount, flags, timestamp});
+ if (flags == CODEC_CONFIG_FLAG) mNumCsds++;
+ }
+ eleInfo.close();
+ mInputStream.open(inputFileName.c_str(), std::ifstream::binary);
+ ASSERT_EQ(mInputStream.is_open(), true);
+}
+
+int32_t WriterTest::createWriter(int32_t fd) {
+ mFileMeta = new MetaData;
+ switch (mWriterName) {
+ case OGG:
+ mWriter = new OggWriter(fd);
+ mFileMeta->setInt32(kKeyFileType, output_format::OUTPUT_FORMAT_OGG);
+ break;
+ case AAC:
+ mWriter = new AACWriter(fd);
+ mFileMeta->setInt32(kKeyFileType, output_format::OUTPUT_FORMAT_AAC_ADIF);
+ break;
+ case AAC_ADTS:
+ mWriter = new AACWriter(fd);
+ mFileMeta->setInt32(kKeyFileType, output_format::OUTPUT_FORMAT_AAC_ADTS);
+ break;
+ case WEBM:
+ mWriter = new WebmWriter(fd);
+ mFileMeta->setInt32(kKeyFileType, output_format::OUTPUT_FORMAT_WEBM);
+ break;
+ case MPEG4:
+ mWriter = new MPEG4Writer(fd);
+ mFileMeta->setInt32(kKeyFileType, output_format::OUTPUT_FORMAT_MPEG_4);
+ break;
+ case AMR_NB:
+ mWriter = new AMRWriter(fd);
+ mFileMeta->setInt32(kKeyFileType, output_format::OUTPUT_FORMAT_AMR_NB);
+ break;
+ case AMR_WB:
+ mWriter = new AMRWriter(fd);
+ mFileMeta->setInt32(kKeyFileType, output_format::OUTPUT_FORMAT_AMR_WB);
+ break;
+ case MPEG2TS:
+ mWriter = new MPEG2TSWriter(fd);
+ mFileMeta->setInt32(kKeyFileType, output_format::OUTPUT_FORMAT_MPEG2TS);
+ break;
+ default:
+ return -1;
+ }
+ if (mWriter == nullptr) return -1;
+ mFileMeta->setInt32(kKeyRealTimeRecording, false);
+ return 0;
+}
+
+int32_t WriterTest::addWriterSource(bool isAudio, configFormat params) {
+ if (mInputFrameId) return -1;
+ sp<AMessage> format = new AMessage;
+ if (mInputStream.is_open()) {
+ format->setString("mime", params.mime);
+ if (isAudio) {
+ format->setInt32("channel-count", params.channelCount);
+ format->setInt32("sample-rate", params.sampleRate);
+ } else {
+ format->setInt32("width", params.width);
+ format->setInt32("height", params.height);
+ }
+
+ int32_t status =
+ writeHeaderBuffers(mInputStream, mBufferInfo, mInputFrameId, format, mNumCsds);
+ if (status != 0) return -1;
+ }
+ sp<MetaData> trackMeta = new MetaData;
+ convertMessageToMetaData(format, trackMeta);
+ mCurrentTrack = new MediaAdapter(trackMeta);
+ if (mCurrentTrack == nullptr) {
+ ALOGE("MediaAdapter returned nullptr");
+ return -1;
+ }
+ status_t result = mWriter->addSource(mCurrentTrack);
+ return result;
+}
+
+void getFileDetails(string &inputFilePath, string &info, configFormat ¶ms, bool &isAudio,
+ int32_t streamIndex = 0) {
+ if (streamIndex >= sizeof(kInputData) / sizeof(kInputData[0])) {
+ return;
+ }
+ inputFilePath += kInputData[streamIndex].inputFile;
+ info += kInputData[streamIndex].info;
+ strcpy(params.mime, kInputData[streamIndex].mime);
+ isAudio = kInputData[streamIndex].isAudio;
+ if (isAudio) {
+ params.sampleRate = kInputData[streamIndex].firstParam;
+ params.channelCount = kInputData[streamIndex].secondParam;
+ } else {
+ params.width = kInputData[streamIndex].firstParam;
+ params.height = kInputData[streamIndex].secondParam;
+ }
+ return;
+}
+
+TEST_P(WriterTest, CreateWriterTest) {
+ if (mDisableTest) return;
+ ALOGV("Tests the creation of writers");
+
+ string outputFile = OUTPUT_FILE_NAME;
+ int32_t fd =
+ open(outputFile.c_str(), O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR);
+ ASSERT_GE(fd, 0) << "Failed to open output file to dump writer's data";
+
+ // Creating writer within a test scope. Destructor should be called when the test ends
+ ASSERT_EQ((status_t)OK, createWriter(fd))
+ << "Failed to create writer for output format:" << GetParam().first;
+}
+
+TEST_P(WriterTest, WriterTest) {
+ if (mDisableTest) return;
+ ALOGV("Checks if for a given input, a valid muxed file has been created or not");
+
+ string writerFormat = GetParam().first;
+ string outputFile = OUTPUT_FILE_NAME;
+ int32_t fd =
+ open(outputFile.c_str(), O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR);
+ ASSERT_GE(fd, 0) << "Failed to open output file to dump writer's data";
+
+ int32_t status = createWriter(fd);
+ ASSERT_EQ((status_t)OK, status) << "Failed to create writer for output format:" << writerFormat;
+
+ string inputFile = gEnv->getRes();
+ string inputInfo = gEnv->getRes();
+ configFormat param;
+ bool isAudio;
+ int32_t inputFileIdx = GetParam().second;
+ getFileDetails(inputFile, inputInfo, param, isAudio, inputFileIdx);
+ ASSERT_NE(inputFile.compare(gEnv->getRes()), 0) << "No input file specified";
+
+ ASSERT_NO_FATAL_FAILURE(getInputBufferInfo(inputFile, inputInfo));
+ status = addWriterSource(isAudio, param);
+ ASSERT_EQ((status_t)OK, status) << "Failed to add source for " << writerFormat << "Writer";
+
+ status = mWriter->start(mFileMeta.get());
+ ASSERT_EQ((status_t)OK, status);
+ status = sendBuffersToWriter(mInputStream, mBufferInfo, mInputFrameId, mCurrentTrack, 0,
+ mBufferInfo.size());
+ ASSERT_EQ((status_t)OK, status) << writerFormat << " writer failed";
+ mCurrentTrack->stop();
+
+ status = mWriter->stop();
+ ASSERT_EQ((status_t)OK, status) << "Failed to stop the writer";
+ close(fd);
+}
+
+TEST_P(WriterTest, PauseWriterTest) {
+ if (mDisableTest) return;
+ ALOGV("Validates the pause() api of writers");
+
+ string writerFormat = GetParam().first;
+ string outputFile = OUTPUT_FILE_NAME;
+ int32_t fd =
+ open(outputFile.c_str(), O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR);
+ ASSERT_GE(fd, 0) << "Failed to open output file to dump writer's data";
+
+ int32_t status = createWriter(fd);
+ ASSERT_EQ((status_t)OK, status) << "Failed to create writer for output format:" << writerFormat;
+
+ string inputFile = gEnv->getRes();
+ string inputInfo = gEnv->getRes();
+ configFormat param;
+ bool isAudio;
+ int32_t inputFileIdx = GetParam().second;
+ getFileDetails(inputFile, inputInfo, param, isAudio, inputFileIdx);
+ ASSERT_NE(inputFile.compare(gEnv->getRes()), 0) << "No input file specified";
+
+ ASSERT_NO_FATAL_FAILURE(getInputBufferInfo(inputFile, inputInfo));
+ status = addWriterSource(isAudio, param);
+ ASSERT_EQ((status_t)OK, status) << "Failed to add source for " << writerFormat << "Writer";
+
+ status = mWriter->start(mFileMeta.get());
+ ASSERT_EQ((status_t)OK, status);
+ status = sendBuffersToWriter(mInputStream, mBufferInfo, mInputFrameId, mCurrentTrack, 0,
+ mBufferInfo.size() / 4);
+ ASSERT_EQ((status_t)OK, status) << writerFormat << " writer failed";
+
+ bool isPaused = false;
+ if ((mWriterName != standardWriters::MPEG2TS) && (mWriterName != standardWriters::MPEG4)) {
+ status = mWriter->pause();
+ ASSERT_EQ((status_t)OK, status);
+ isPaused = true;
+ }
+ // In the pause state, writers shouldn't write anything. Testing the writers for the same
+ int32_t numFramesPaused = mBufferInfo.size() / 4;
+ status = sendBuffersToWriter(mInputStream, mBufferInfo, mInputFrameId, mCurrentTrack,
+ mInputFrameId, numFramesPaused, isPaused);
+ ASSERT_EQ((status_t)OK, status) << writerFormat << " writer failed";
+
+ if (isPaused) {
+ status = mWriter->start(mFileMeta.get());
+ ASSERT_EQ((status_t)OK, status);
+ }
+ status = sendBuffersToWriter(mInputStream, mBufferInfo, mInputFrameId, mCurrentTrack,
+ mInputFrameId, mBufferInfo.size());
+ ASSERT_EQ((status_t)OK, status) << writerFormat << " writer failed";
+ mCurrentTrack->stop();
+
+ status = mWriter->stop();
+ ASSERT_EQ((status_t)OK, status) << "Failed to stop the writer";
+ close(fd);
+}
+
+TEST_P(WriterTest, MultiStartStopPauseTest) {
+ // TODO: (b/144821804)
+ // Enable the test for MPE2TS writer
+ if (mDisableTest || mWriterName == standardWriters::MPEG2TS) return;
+ ALOGV("Test writers for multiple start, stop and pause calls");
+
+ string outputFile = OUTPUT_FILE_NAME;
+ int32_t fd =
+ open(outputFile.c_str(), O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR);
+ ASSERT_GE(fd, 0) << "Failed to open output file to dump writer's data";
+
+ string writerFormat = GetParam().first;
+ int32_t status = createWriter(fd);
+ ASSERT_EQ(status, (status_t)OK) << "Failed to create writer for output format:" << writerFormat;
+
+ string inputFile = gEnv->getRes();
+ string inputInfo = gEnv->getRes();
+ configFormat param;
+ bool isAudio;
+ int32_t inputFileIdx = GetParam().second;
+ getFileDetails(inputFile, inputInfo, param, isAudio, inputFileIdx);
+ ASSERT_NE(inputFile.compare(gEnv->getRes()), 0) << "No input file specified";
+
+ ASSERT_NO_FATAL_FAILURE(getInputBufferInfo(inputFile, inputInfo));
+ status = addWriterSource(isAudio, param);
+ ASSERT_EQ((status_t)OK, status) << "Failed to add source for " << writerFormat << "Writer";
+
+ // first start should succeed.
+ status = mWriter->start(mFileMeta.get());
+ ASSERT_EQ((status_t)OK, status) << "Could not start the writer";
+
+ // Multiple start() may/may not succeed.
+ // Writers are expected to not crash on multiple start() calls.
+ for (int32_t count = 0; count < kMaxCount; count++) {
+ mWriter->start(mFileMeta.get());
+ }
+
+ status = sendBuffersToWriter(mInputStream, mBufferInfo, mInputFrameId, mCurrentTrack, 0,
+ mBufferInfo.size() / 4);
+ ASSERT_EQ((status_t)OK, status) << writerFormat << " writer failed";
+
+ for (int32_t count = 0; count < kMaxCount; count++) {
+ mWriter->pause();
+ mWriter->start(mFileMeta.get());
+ }
+
+ mWriter->pause();
+ int32_t numFramesPaused = mBufferInfo.size() / 4;
+ status = sendBuffersToWriter(mInputStream, mBufferInfo, mInputFrameId, mCurrentTrack,
+ mInputFrameId, numFramesPaused, true);
+ ASSERT_EQ((status_t)OK, status) << writerFormat << " writer failed";
+
+ for (int32_t count = 0; count < kMaxCount; count++) {
+ mWriter->start(mFileMeta.get());
+ }
+
+ status = sendBuffersToWriter(mInputStream, mBufferInfo, mInputFrameId, mCurrentTrack,
+ mInputFrameId, mBufferInfo.size());
+ ASSERT_EQ((status_t)OK, status) << writerFormat << " writer failed";
+
+ mCurrentTrack->stop();
+
+ // first stop should succeed.
+ status = mWriter->stop();
+ ASSERT_EQ((status_t)OK, status) << "Failed to stop the writer";
+ // Multiple stop() may/may not succeed.
+ // Writers are expected to not crash on multiple stop() calls.
+ for (int32_t count = 0; count < kMaxCount; count++) {
+ mWriter->stop();
+ }
+ close(fd);
+}
+
+// TODO: (b/144476164)
+// Add AAC_ADTS, FLAC, AV1 input
+INSTANTIATE_TEST_SUITE_P(WriterTestAll, WriterTest,
+ ::testing::Values(make_pair("ogg", 0), make_pair("webm", 0),
+ make_pair("aac", 1), make_pair("mpeg4", 1),
+ make_pair("amrnb", 3), make_pair("amrwb", 4),
+ make_pair("webm", 5), make_pair("webm", 7),
+ make_pair("webm", 8), make_pair("mpeg4", 9),
+ make_pair("mpeg4", 10), make_pair("mpeg4", 12),
+ make_pair("mpeg4", 13), make_pair("mpeg2Ts", 1),
+ make_pair("mpeg2Ts", 9)));
+
+int main(int argc, char **argv) {
+ gEnv = new WriterTestEnvironment();
+ ::testing::AddGlobalTestEnvironment(gEnv);
+ ::testing::InitGoogleTest(&argc, argv);
+ int status = gEnv->initFromOptions(argc, argv);
+ if (status == 0) {
+ status = RUN_ALL_TESTS();
+ ALOGV("Test result = %d\n", status);
+ }
+ return status;
+}
diff --git a/media/libstagefright/tests/writer/WriterTestEnvironment.h b/media/libstagefright/tests/writer/WriterTestEnvironment.h
new file mode 100644
index 0000000..99e686f
--- /dev/null
+++ b/media/libstagefright/tests/writer/WriterTestEnvironment.h
@@ -0,0 +1,72 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __WRITER_TEST_ENVIRONMENT_H__
+#define __WRITER_TEST_ENVIRONMENT_H__
+
+#include <gtest/gtest.h>
+
+#include <getopt.h>
+
+using namespace std;
+
+class WriterTestEnvironment : public ::testing::Environment {
+ public:
+ WriterTestEnvironment() : res("/data/local/tmp/") {}
+
+ // Parses the command line arguments
+ int initFromOptions(int argc, char **argv);
+
+ void setRes(const char *_res) { res = _res; }
+
+ const string getRes() const { return res; }
+
+ private:
+ string res;
+};
+
+int WriterTestEnvironment::initFromOptions(int argc, char **argv) {
+ static struct option options[] = {{"res", required_argument, 0, 'P'}, {0, 0, 0, 0}};
+
+ while (true) {
+ int index = 0;
+ int c = getopt_long(argc, argv, "P:", options, &index);
+ if (c == -1) {
+ break;
+ }
+
+ switch (c) {
+ case 'P':
+ setRes(optarg);
+ break;
+ default:
+ break;
+ }
+ }
+
+ if (optind < argc) {
+ fprintf(stderr,
+ "unrecognized option: %s\n\n"
+ "usage: %s <gtest options> <test options>\n\n"
+ "test options are:\n\n"
+ "-P, --path: Resource files directory location\n",
+ argv[optind ?: 1], argv[0]);
+ return 2;
+ }
+ return 0;
+}
+
+#endif // __WRITER_TEST_ENVIRONMENT_H__
diff --git a/media/libstagefright/tests/writer/WriterUtility.cpp b/media/libstagefright/tests/writer/WriterUtility.cpp
new file mode 100644
index 0000000..f24ccb6
--- /dev/null
+++ b/media/libstagefright/tests/writer/WriterUtility.cpp
@@ -0,0 +1,102 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "WriterUtility"
+#include <utils/Log.h>
+
+#include <media/stagefright/MediaBuffer.h>
+
+#include "WriterUtility.h"
+
+int32_t sendBuffersToWriter(ifstream &inputStream, vector<BufferInfo> &bufferInfo,
+ int32_t &inputFrameId, sp<MediaAdapter> ¤tTrack, int32_t offset,
+ int32_t range, bool isPaused) {
+ while (1) {
+ if (inputFrameId >= (int)bufferInfo.size() || inputFrameId >= (offset + range)) break;
+ int32_t size = bufferInfo[inputFrameId].size;
+ char *data = (char *)malloc(size);
+ if (!data) {
+ ALOGE("Insufficient memeory to read input");
+ return -1;
+ }
+
+ inputStream.read(data, size);
+ CHECK_EQ(inputStream.gcount(), size);
+
+ sp<ABuffer> buffer = new ABuffer((void *)data, size);
+ if (buffer.get() == nullptr) {
+ ALOGE("sendBuffersToWriter() got a nullptr buffer.");
+ return -1;
+ }
+ MediaBuffer *mediaBuffer = new MediaBuffer(buffer);
+
+ // Released in MediaAdapter::signalBufferReturned().
+ mediaBuffer->add_ref();
+ mediaBuffer->set_range(buffer->offset(), buffer->size());
+
+ MetaDataBase &sampleMetaData = mediaBuffer->meta_data();
+ sampleMetaData.setInt64(kKeyTime, bufferInfo[inputFrameId].timeUs);
+ // Just set the kKeyDecodingTime as the presentation time for now.
+ sampleMetaData.setInt64(kKeyDecodingTime, bufferInfo[inputFrameId].timeUs);
+
+ if (bufferInfo[inputFrameId].flags == 1) {
+ sampleMetaData.setInt32(kKeyIsSyncFrame, true);
+ }
+
+ // This pushBuffer will wait until the mediaBuffer is consumed.
+ int status = currentTrack->pushBuffer(mediaBuffer);
+ free(data);
+ inputFrameId++;
+
+ if (OK != status) {
+ if (!isPaused) return status;
+ else {
+ ALOGD("Writer is in paused state. Input buffers won't get consumed");
+ return 0;
+ }
+ }
+ }
+ return 0;
+}
+
+int32_t writeHeaderBuffers(ifstream &inputStream, vector<BufferInfo> &bufferInfo,
+ int32_t &inputFrameId, sp<AMessage> &format, int32_t numCsds) {
+ char csdName[kMaxCSDStrlen];
+ for (int csdId = 0; csdId < numCsds; csdId++) {
+ int32_t flags = bufferInfo[inputFrameId].flags;
+ if (flags == CODEC_CONFIG_FLAG) {
+ int32_t size = bufferInfo[inputFrameId].size;
+ char *data = (char *)malloc(size);
+ if (!data) {
+ ALOGE("Insufficient memeory to read input");
+ return -1;
+ }
+ inputStream.read(data, size);
+ CHECK_EQ(inputStream.gcount(), size);
+
+ sp<ABuffer> csdBuffer = ABuffer::CreateAsCopy((void *)data, size);
+ if (csdBuffer.get() == nullptr || csdBuffer->base() == nullptr) {
+ return -1;
+ }
+ snprintf(csdName, sizeof(csdName), "csd-%d", csdId);
+ format->setBuffer(csdName, csdBuffer);
+ inputFrameId++;
+ free(data);
+ }
+ }
+ return 0;
+}
diff --git a/media/libstagefright/tests/writer/WriterUtility.h b/media/libstagefright/tests/writer/WriterUtility.h
new file mode 100644
index 0000000..cdd6246
--- /dev/null
+++ b/media/libstagefright/tests/writer/WriterUtility.h
@@ -0,0 +1,51 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef WRITER_UTILITY_H_
+#define WRITER_UTILITY_H_
+
+#include <fstream>
+#include <iostream>
+#include <vector>
+
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+
+#include <media/stagefright/MediaAdapter.h>
+
+using namespace android;
+using namespace std;
+
+#define CODEC_CONFIG_FLAG 32
+
+constexpr uint32_t kMaxCSDStrlen = 16;
+constexpr uint32_t kMaxCount = 20;
+
+struct BufferInfo {
+ int32_t size;
+ uint32_t flags;
+ int64_t timeUs;
+};
+
+int32_t sendBuffersToWriter(ifstream &inputStream, vector<BufferInfo> &bufferInfo,
+ int32_t &inputFrameId, sp<MediaAdapter> ¤tTrack, int32_t offset,
+ int32_t range, bool isPaused = false);
+
+int32_t writeHeaderBuffers(ifstream &inputStream, vector<BufferInfo> &bufferInfo,
+ int32_t &inputFrameId, sp<AMessage> &format, int32_t numCsds);
+
+#endif // WRITER_UTILITY_H_
diff --git a/media/libstagefright/timedtext/Android.bp b/media/libstagefright/timedtext/Android.bp
index 6935655..4f4ceb1 100644
--- a/media/libstagefright/timedtext/Android.bp
+++ b/media/libstagefright/timedtext/Android.bp
@@ -23,32 +23,4 @@
shared_libs: ["libmedia"],
}
-cc_library_static {
- name: "libstagefright_timedtext2",
- srcs: ["TextDescriptions2.cpp"],
-
- static_libs: [
- "libmediaplayer2-protos",
- "libprotobuf-cpp-lite",
- ],
-
- cflags: [
- "-Wno-multichar",
- "-Werror",
- "-Wall",
- ],
-
- sanitize: {
- misc_undefined: [
- "signed-integer-overflow",
- ],
- cfi: true,
- },
-
- include_dirs: [
- "frameworks/av/media/libstagefright",
- ],
-
- shared_libs: ["libmedia"],
-}
diff --git a/media/libstagefright/timedtext/TextDescriptions2.cpp b/media/libstagefright/timedtext/TextDescriptions2.cpp
deleted file mode 100644
index f48eacc..0000000
--- a/media/libstagefright/timedtext/TextDescriptions2.cpp
+++ /dev/null
@@ -1,188 +0,0 @@
-/*
- * Copyright (C) 2018 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include "TextDescriptions2.h"
-#include <media/stagefright/foundation/ByteUtils.h>
-#include <media/stagefright/MediaErrors.h>
-
-namespace android {
-
-TextDescriptions2::TextDescriptions2() {
-}
-
-status_t TextDescriptions2::getPlayerMessageOfDescriptions(
- const uint8_t *data, ssize_t size,
- uint32_t flags, int timeMs, PlayerMessage *playerMsg) {
- if (flags & IN_BAND_TEXT_3GPP) {
- if (flags & GLOBAL_DESCRIPTIONS) {
- return extract3GPPGlobalDescriptions(data, size, playerMsg);
- } else if (flags & LOCAL_DESCRIPTIONS) {
- return extract3GPPLocalDescriptions(data, size, timeMs, playerMsg);
- }
- } else if (flags & OUT_OF_BAND_TEXT_SRT) {
- if (flags & LOCAL_DESCRIPTIONS) {
- return extractSRTLocalDescriptions(data, size, timeMs, playerMsg);
- }
- }
-
- return ERROR_UNSUPPORTED;
-}
-
-// Parse the SRT text sample, and store the timing and text sample in a PlayerMessage.
-// The PlayerMessage will be sent to MediaPlayer2.java through event, and will be
-// parsed in TimedText.java.
-status_t TextDescriptions2::extractSRTLocalDescriptions(
- const uint8_t *data, ssize_t size, int timeMs, PlayerMessage *playerMsg) {
- playerMsg->add_values()->set_int32_value(KEY_LOCAL_SETTING);
- playerMsg->add_values()->set_int32_value(KEY_START_TIME);
- playerMsg->add_values()->set_int32_value(timeMs);
-
- playerMsg->add_values()->set_int32_value(KEY_STRUCT_TEXT);
- playerMsg->add_values()->set_bytes_value(data, size);
-
- return OK;
-}
-
-// Extract the local 3GPP display descriptions. 3GPP local descriptions
-// are appended to the text sample if any.
-status_t TextDescriptions2::extract3GPPLocalDescriptions(
- const uint8_t *data, ssize_t size,
- int timeMs, PlayerMessage *playerMsg) {
-
- playerMsg->add_values()->set_int32_value(KEY_LOCAL_SETTING);
-
- // write start time to display this text sample
- playerMsg->add_values()->set_int32_value(KEY_START_TIME);
- playerMsg->add_values()->set_int32_value(timeMs);
-
- if (size < 2) {
- return OK;
- }
- ssize_t textLen = (*data) << 8 | (*(data + 1));
-
- if (size < textLen + 2) {
- return OK;
- }
-
- // write text sample length and text sample itself
- playerMsg->add_values()->set_int32_value(KEY_STRUCT_TEXT);
- playerMsg->add_values()->set_bytes_value(data + 2, textLen);
-
- if (size > textLen + 2) {
- data += (textLen + 2);
- size -= (textLen + 2);
- } else {
- return OK;
- }
-
- while (size >= 8) {
- const uint8_t *tmpData = data;
- ssize_t chunkSize = U32_AT(tmpData); // size includes size and type
- uint32_t chunkType = U32_AT(tmpData + 4);
-
- if (chunkSize <= 8 || chunkSize > size) {
- return OK;
- }
-
- size_t remaining = chunkSize - 8;
-
- tmpData += 8;
-
- switch(chunkType) {
- // 'tbox' box to indicate the position of the text with values
- // of top, left, bottom and right
- case FOURCC('t', 'b', 'o', 'x'):
- {
- if (remaining < 8) {
- return OK;
- }
- playerMsg->add_values()->set_int32_value(KEY_STRUCT_TEXT_POS);
- playerMsg->add_values()->set_int32_value(U16_AT(tmpData));
- playerMsg->add_values()->set_int32_value(U16_AT(tmpData + 2));
- playerMsg->add_values()->set_int32_value(U16_AT(tmpData + 4));
- playerMsg->add_values()->set_int32_value(U16_AT(tmpData + 6));
-
- tmpData += 8;
- remaining -= 8;
- break;
- }
- default:
- {
- break;
- }
- }
-
- data += chunkSize;
- size -= chunkSize;
- }
-
- return OK;
-}
-
-// To extract box 'tx3g' defined in 3GPP TS 26.245, and store it in a PlayerMessage
-status_t TextDescriptions2::extract3GPPGlobalDescriptions(
- const uint8_t *data, ssize_t size, PlayerMessage *playerMsg) {
-
- playerMsg->add_values()->set_int32_value(KEY_GLOBAL_SETTING);
-
- while (size >= 8) {
- ssize_t chunkSize = U32_AT(data);
- uint32_t chunkType = U32_AT(data + 4);
- const uint8_t *tmpData = data;
- tmpData += 8;
- size_t remaining = size - 8;
-
- if (size < chunkSize) {
- return OK;
- }
- switch(chunkType) {
- case FOURCC('t', 'x', '3', 'g'):
- {
- if (remaining < 18) {
- return OK;
- }
- // Skip DISPLAY_FLAGS, STRUCT_JUSTIFICATION, and BACKGROUND_COLOR_RGBA
- tmpData += 18;
- remaining -= 18;
-
- if (remaining < 8) {
- return OK;
- }
- playerMsg->add_values()->set_int32_value(KEY_STRUCT_TEXT_POS);
- playerMsg->add_values()->set_int32_value(U16_AT(tmpData));
- playerMsg->add_values()->set_int32_value(U16_AT(tmpData + 2));
- playerMsg->add_values()->set_int32_value(U16_AT(tmpData + 4));
- playerMsg->add_values()->set_int32_value(U16_AT(tmpData + 6));
-
- tmpData += 8;
- remaining -= 18;
- // Ignore remaining data.
- break;
- }
- default:
- {
- break;
- }
- }
-
- data += chunkSize;
- size -= chunkSize;
- }
-
- return OK;
-}
-
-} // namespace android
diff --git a/media/libstagefright/timedtext/TextDescriptions2.h b/media/libstagefright/timedtext/TextDescriptions2.h
deleted file mode 100644
index 7c7d2d0..0000000
--- a/media/libstagefright/timedtext/TextDescriptions2.h
+++ /dev/null
@@ -1,88 +0,0 @@
- /*
- * Copyright (C) 2018 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef TEXT_DESCRIPTIONS2_H_
-
-#define TEXT_DESCRIPTIONS2_H_
-
-#include <binder/Parcel.h>
-#include <media/stagefright/foundation/ABase.h>
-
-#include "mediaplayer2.pb.h"
-
-using android::media::MediaPlayer2Proto::PlayerMessage;
-
-namespace android {
-
-class TextDescriptions2 {
-public:
- enum {
- IN_BAND_TEXT_3GPP = 0x01,
- OUT_OF_BAND_TEXT_SRT = 0x02,
-
- GLOBAL_DESCRIPTIONS = 0x100,
- LOCAL_DESCRIPTIONS = 0x200,
- };
-
- static status_t getPlayerMessageOfDescriptions(
- const uint8_t *data, ssize_t size,
- uint32_t flags, int timeMs, PlayerMessage *playerMsg);
-private:
- TextDescriptions2();
-
- enum {
- // These keys must be in sync with the keys in TimedText.java
- KEY_DISPLAY_FLAGS = 1, // int
- KEY_STYLE_FLAGS = 2, // int
- KEY_BACKGROUND_COLOR_RGBA = 3, // int
- KEY_HIGHLIGHT_COLOR_RGBA = 4, // int
- KEY_SCROLL_DELAY = 5, // int
- KEY_WRAP_TEXT = 6, // int
- KEY_START_TIME = 7, // int
- KEY_STRUCT_BLINKING_TEXT_LIST = 8, // List<CharPos>
- KEY_STRUCT_FONT_LIST = 9, // List<Font>
- KEY_STRUCT_HIGHLIGHT_LIST = 10, // List<CharPos>
- KEY_STRUCT_HYPER_TEXT_LIST = 11, // List<HyperText>
- KEY_STRUCT_KARAOKE_LIST = 12, // List<Karaoke>
- KEY_STRUCT_STYLE_LIST = 13, // List<Style>
- KEY_STRUCT_TEXT_POS = 14, // TextPos
- KEY_STRUCT_JUSTIFICATION = 15, // Justification
- KEY_STRUCT_TEXT = 16, // Text
-
- KEY_GLOBAL_SETTING = 101,
- KEY_LOCAL_SETTING = 102,
- KEY_START_CHAR = 103,
- KEY_END_CHAR = 104,
- KEY_FONT_ID = 105,
- KEY_FONT_SIZE = 106,
- KEY_TEXT_COLOR_RGBA = 107,
- };
-
- static status_t extractSRTLocalDescriptions(
- const uint8_t *data, ssize_t size,
- int timeMs, PlayerMessage *playerMsg);
- static status_t extract3GPPGlobalDescriptions(
- const uint8_t *data, ssize_t size,
- PlayerMessage *playerMsg);
- static status_t extract3GPPLocalDescriptions(
- const uint8_t *data, ssize_t size,
- int timeMs, PlayerMessage *playerMsg);
-
- DISALLOW_EVIL_CONSTRUCTORS(TextDescriptions2);
-};
-
-} // namespace android
-#endif // TEXT_DESCRIPTIONS2_H_
diff --git a/media/libstagefright/webm/Android.bp b/media/libstagefright/webm/Android.bp
index 64ecc2d..2cebe8f 100644
--- a/media/libstagefright/webm/Android.bp
+++ b/media/libstagefright/webm/Android.bp
@@ -27,12 +27,14 @@
include_dirs: ["frameworks/av/include"],
shared_libs: [
+ "libdatasource",
"libstagefright_foundation",
"libutils",
"liblog",
],
header_libs: [
+ "libmedia_headers",
"media_ndk_headers",
],
}
diff --git a/media/libstagefright/webm/WebmFrameThread.h b/media/libstagefright/webm/WebmFrameThread.h
index 1ddaf9a..2dde20a 100644
--- a/media/libstagefright/webm/WebmFrameThread.h
+++ b/media/libstagefright/webm/WebmFrameThread.h
@@ -20,8 +20,8 @@
#include "WebmFrame.h"
#include "LinkedBlockingQueue.h"
+#include <datasource/FileSource.h>
#include <media/MediaSource.h>
-#include <media/stagefright/FileSource.h>
#include <utils/List.h>
#include <utils/Errors.h>
diff --git a/media/libstagefright/webm/tests/Android.bp b/media/libstagefright/webm/tests/Android.bp
new file mode 100644
index 0000000..5183a49
--- /dev/null
+++ b/media/libstagefright/webm/tests/Android.bp
@@ -0,0 +1,54 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+cc_test {
+ name: "WebmFrameThreadUnitTest",
+ gtest: true,
+
+ srcs: [
+ "WebmFrameThreadUnitTest.cpp",
+ ],
+
+ include_dirs: [
+ "frameworks/av/media/libstagefright",
+ ],
+
+ static_libs: [
+ "libdatasource",
+ "libstagefright",
+ "libstagefright_webm",
+ "libstagefright_foundation",
+ ],
+
+ shared_libs: [
+ "libbinder",
+ "liblog",
+ "libutils",
+ ],
+
+ cflags: [
+ "-Werror",
+ "-Wall",
+ ],
+
+ sanitize: {
+ cfi: true,
+ misc_undefined: [
+ "unsigned-integer-overflow",
+ "signed-integer-overflow",
+ ],
+ },
+}
diff --git a/media/libstagefright/webm/tests/README.md b/media/libstagefright/webm/tests/README.md
new file mode 100644
index 0000000..2e74f34
--- /dev/null
+++ b/media/libstagefright/webm/tests/README.md
@@ -0,0 +1,26 @@
+## Media Testing ##
+---
+#### Webm Writer Utility Tests :
+The Webm Writer Utility Test Suite validates the APIs being used by the WebmWriter.
+
+Run the following steps to build the test suite:
+```
+mmm frameworks/av/media/libstagefright/webm/tests/
+```
+
+The 32-bit binaries will be created in the following path : ${OUT}/data/nativetest/
+
+The 64-bit binaries will be created in the following path : ${OUT}/data/nativetest64/
+
+#### WebmFrameThread
+To test 64-bit binary push binaries from nativetest64.
+
+adb push ${OUT}/data/nativetest64/WebmFrameThreadUnitTest/WebmFrameThreadUnitTest /data/local/tmp/
+
+To test 32-bit binary push binaries from nativetest.
+
+adb push ${OUT}/data/nativetest/WebmFrameThreadUnitTest/WebmFrameThreadUnitTest /data/local/tmp/
+
+```
+adb shell /data/local/tmp/WebmFrameThreadUnitTest
+```
diff --git a/media/libstagefright/webm/tests/WebmFrameThreadUnitTest.cpp b/media/libstagefright/webm/tests/WebmFrameThreadUnitTest.cpp
new file mode 100644
index 0000000..89cd2ca
--- /dev/null
+++ b/media/libstagefright/webm/tests/WebmFrameThreadUnitTest.cpp
@@ -0,0 +1,314 @@
+/*
+ * Copyright (C) 2020 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "WebmFrameThreadUnitTest"
+#include <utils/Log.h>
+
+#include <gtest/gtest.h>
+
+#include <media/stagefright/MediaAdapter.h>
+#include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/MetaData.h>
+#include <media/stagefright/Utils.h>
+
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/foundation/OpusHeader.h>
+
+#include "webm/EbmlUtil.h"
+#include "webm/WebmConstants.h"
+#include "webm/WebmFrameThread.h"
+
+using namespace android;
+using namespace webm;
+
+static constexpr int32_t kVideoIdx = 0;
+static constexpr int32_t kAudioIdx = 1;
+static constexpr int32_t kMaxStreamCount = 2;
+
+static constexpr int32_t kCsdSize = 32;
+static constexpr int32_t kFrameSize = 128;
+
+static constexpr int32_t kMaxLoopCount = 20;
+static constexpr int32_t kNumFramesToWrite = 32;
+static constexpr int32_t kSyncFrameInterval = 10;
+static constexpr uint64_t kDefaultTimeCodeScaleUs = 1000000; /* 1sec */
+
+#define OUTPUT_FILE_NAME "/data/local/tmp/webmFrameThreadOutput.webm"
+
+// LookUpTable of clips and metadata for component testing
+static const struct InputData {
+ const char *mime;
+ int32_t firstParam;
+ int32_t secondParam;
+ bool isAudio;
+} kInputData[] = {
+ {MEDIA_MIMETYPE_AUDIO_OPUS, 48000, 6, true},
+ {MEDIA_MIMETYPE_AUDIO_VORBIS, 44100, 1, true},
+ {MEDIA_MIMETYPE_VIDEO_VP9, 176, 144, false},
+ {MEDIA_MIMETYPE_VIDEO_VP8, 1920, 1080, false},
+};
+
+class WebmFrameThreadUnitTest : public ::testing::TestWithParam<std::pair<int32_t, int32_t>> {
+ public:
+ WebmFrameThreadUnitTest()
+ : mSinkThread(nullptr), mAudioThread(nullptr), mVideoThread(nullptr), mSource{} {}
+
+ ~WebmFrameThreadUnitTest() {
+ if (mSinkThread) mSinkThread.clear();
+ if (mAudioThread) mAudioThread.clear();
+ if (mVideoThread) mVideoThread.clear();
+ }
+
+ virtual void SetUp() override {
+ mSegmentDataStart = 0;
+ mFd = open(OUTPUT_FILE_NAME, O_CREAT | O_LARGEFILE | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR);
+ ASSERT_GE(mFd, 0) << "Failed to open output file " << OUTPUT_FILE_NAME;
+ }
+
+ virtual void TearDown() override {
+ if (mFd >= 0) close(mFd);
+ for (int32_t idx = 0; idx < kMaxStreamCount; idx++) {
+ if (mSource[idx] != nullptr) {
+ mSource[idx].clear();
+ }
+ }
+ mVSink.clear();
+ mASink.clear();
+ mCuePoints.clear();
+ }
+
+ void addTrack(bool isAudio, int32_t index);
+ void writeFileData(int32_t inputFrameId, int32_t range);
+
+ void createWebmThreads(std::initializer_list<int32_t> indexList);
+ void startWebmFrameThreads();
+ void stopWebmFrameThreads();
+
+ int32_t mFd;
+ uint64_t mSegmentDataStart;
+
+ sp<WebmFrameSinkThread> mSinkThread;
+ sp<WebmFrameSourceThread> mAudioThread;
+ sp<WebmFrameSourceThread> mVideoThread;
+
+ List<sp<WebmElement>> mCuePoints;
+ sp<MediaAdapter> mSource[kMaxStreamCount];
+ LinkedBlockingQueue<const sp<WebmFrame>> mVSink;
+ LinkedBlockingQueue<const sp<WebmFrame>> mASink;
+};
+
+void writeAudioHeaderData(const sp<AMessage> &format, const char *mimeType) {
+ if (strncasecmp(mimeType, MEDIA_MIMETYPE_AUDIO_OPUS, strlen(MEDIA_MIMETYPE_AUDIO_OPUS) + 1) &&
+ strncasecmp(mimeType, MEDIA_MIMETYPE_AUDIO_VORBIS,
+ strlen(MEDIA_MIMETYPE_AUDIO_VORBIS) + 1)) {
+ ASSERT_TRUE(false) << "Unsupported mime type";
+ }
+
+ // Dummy CSD buffers for Opus and Vorbis
+ char csdBuffer[kCsdSize];
+ memset(csdBuffer, 0xFF, sizeof(csdBuffer));
+
+ sp<ABuffer> csdBuffer0 = ABuffer::CreateAsCopy((void *)csdBuffer, kCsdSize);
+ ASSERT_NE(csdBuffer0.get(), nullptr) << "Unable to allocate buffer for CSD0 data";
+ ASSERT_NE(csdBuffer0->base(), nullptr) << "ABuffer base is null for CSD0";
+
+ sp<ABuffer> csdBuffer1 = ABuffer::CreateAsCopy((void *)csdBuffer, kCsdSize);
+ ASSERT_NE(csdBuffer1.get(), nullptr) << "Unable to allocate buffer for CSD1 data";
+ ASSERT_NE(csdBuffer1->base(), nullptr) << "ABuffer base is null for CSD1";
+
+ sp<ABuffer> csdBuffer2 = ABuffer::CreateAsCopy((void *)csdBuffer, kCsdSize);
+ ASSERT_NE(csdBuffer2.get(), nullptr) << "Unable to allocate buffer for CSD2 data";
+ ASSERT_NE(csdBuffer2->base(), nullptr) << "ABuffer base is null for CSD2";
+
+ format->setBuffer("csd-0", csdBuffer0);
+ format->setBuffer("csd-1", csdBuffer1);
+ format->setBuffer("csd-2", csdBuffer2);
+}
+
+void WebmFrameThreadUnitTest::addTrack(bool isAudio, int32_t index) {
+ ASSERT_LT(index, sizeof(kInputData) / sizeof(kInputData[0]))
+ << "Invalid index for loopup table";
+
+ sp<AMessage> format = new AMessage;
+ format->setString("mime", kInputData[index].mime);
+ if (!isAudio) {
+ format->setInt32("width", kInputData[index].firstParam);
+ format->setInt32("height", kInputData[index].secondParam);
+ } else {
+ format->setInt32("sample-rate", kInputData[index].firstParam);
+ format->setInt32("channel-count", kInputData[index].secondParam);
+ ASSERT_NO_FATAL_FAILURE(writeAudioHeaderData(format, kInputData[index].mime));
+ }
+
+ sp<MetaData> trackMeta = new MetaData;
+ convertMessageToMetaData(format, trackMeta);
+
+ if (!isAudio) {
+ mSource[kVideoIdx] = new MediaAdapter(trackMeta);
+ ASSERT_NE(mSource[kVideoIdx], nullptr) << "Unable to create source";
+ } else {
+ mSource[kAudioIdx] = new MediaAdapter(trackMeta);
+ ASSERT_NE(mSource[kAudioIdx], nullptr) << "Unable to create source";
+ }
+}
+
+void WebmFrameThreadUnitTest::createWebmThreads(std::initializer_list<int32_t> indexList) {
+ mSinkThread = new WebmFrameSinkThread(mFd, mSegmentDataStart, mVSink, mASink, mCuePoints);
+ ASSERT_NE(mSinkThread, nullptr) << "Failed to create Sink Thread";
+
+ bool isAudio;
+ // MultiTrack input
+ for (int32_t index : indexList) {
+ isAudio = kInputData[index].isAudio;
+ ASSERT_NO_FATAL_FAILURE(addTrack(isAudio, index));
+ if (!isAudio) {
+ mVideoThread = new WebmFrameMediaSourceThread(mSource[kVideoIdx], kVideoType, mVSink,
+ kDefaultTimeCodeScaleUs, 0, 0, 1, 0);
+ } else {
+ mAudioThread = new WebmFrameMediaSourceThread(mSource[kAudioIdx], kAudioType, mASink,
+ kDefaultTimeCodeScaleUs, 0, 0, 1, 0);
+ }
+ }
+ // To handle single track file
+ if (!mVideoThread) {
+ mVideoThread = new WebmFrameEmptySourceThread(kVideoType, mVSink);
+ } else if (!mAudioThread) {
+ mAudioThread = new WebmFrameEmptySourceThread(kAudioType, mASink);
+ }
+ ASSERT_NE(mVideoThread, nullptr) << "Failed to create Video Thread";
+ ASSERT_NE(mAudioThread, nullptr) << "Failed to create Audio Thread";
+}
+
+void WebmFrameThreadUnitTest::startWebmFrameThreads() {
+ status_t status = mAudioThread->start();
+ ASSERT_EQ(status, AMEDIA_OK) << "Failed to start Audio Thread";
+ status = mVideoThread->start();
+ ASSERT_EQ(status, AMEDIA_OK) << "Failed to start Video Thread";
+ status = mSinkThread->start();
+ ASSERT_EQ(status, AMEDIA_OK) << "Failed to start Sink Thread";
+}
+
+void WebmFrameThreadUnitTest::stopWebmFrameThreads() {
+ status_t status = mAudioThread->stop();
+ ASSERT_EQ(status, AMEDIA_OK) << "Failed to stop Audio Thread";
+ status = mVideoThread->stop();
+ ASSERT_EQ(status, AMEDIA_OK) << "Failed to stop Video Thread";
+ status = mSinkThread->stop();
+ ASSERT_EQ(status, AMEDIA_OK) << "Failed to stop Sink Thread";
+}
+
+// Write dummy data to a file
+void WebmFrameThreadUnitTest::writeFileData(int32_t inputFrameId, int32_t range) {
+ char data[kFrameSize];
+ memset(data, 0xFF, sizeof(data));
+ int32_t status = OK;
+ do {
+ // Queue frames for both A/V tracks
+ for (int32_t idx = kVideoIdx; idx < kMaxStreamCount; idx++) {
+ sp<ABuffer> buffer = new ABuffer((void *)data, kFrameSize);
+ ASSERT_NE(buffer.get(), nullptr) << "ABuffer returned nullptr";
+
+ // Released in MediaAdapter::signalBufferReturned().
+ MediaBuffer *mediaBuffer = new MediaBuffer(buffer);
+ ASSERT_NE(mediaBuffer, nullptr) << "MediaBuffer returned nullptr";
+
+ mediaBuffer->add_ref();
+ mediaBuffer->set_range(buffer->offset(), buffer->size());
+
+ MetaDataBase &sampleMetaData = mediaBuffer->meta_data();
+ sampleMetaData.setInt64(kKeyTime, inputFrameId * kDefaultTimeCodeScaleUs);
+
+ // For audio codecs, treat all frame as sync frame
+ if ((idx == kAudioIdx) || (inputFrameId % kSyncFrameInterval == 0)) {
+ sampleMetaData.setInt32(kKeyIsSyncFrame, true);
+ }
+
+ // This pushBuffer will wait until the mediaBuffer is consumed.
+ if (mSource[idx] != nullptr) {
+ status = mSource[idx]->pushBuffer(mediaBuffer);
+ }
+ ASSERT_EQ(status, OK);
+ }
+ inputFrameId++;
+ } while (inputFrameId < range);
+}
+
+TEST_P(WebmFrameThreadUnitTest, WriteTest) {
+ int32_t index1 = GetParam().first;
+ int32_t index2 = GetParam().second;
+ ASSERT_NO_FATAL_FAILURE(createWebmThreads({index1, index2}));
+
+ ASSERT_NO_FATAL_FAILURE(startWebmFrameThreads());
+
+ ASSERT_NO_FATAL_FAILURE(writeFileData(0, kNumFramesToWrite));
+
+ if (mSource[kAudioIdx]) mSource[kAudioIdx]->stop();
+ if (mSource[kVideoIdx]) mSource[kVideoIdx]->stop();
+
+ ASSERT_NO_FATAL_FAILURE(stopWebmFrameThreads());
+}
+
+TEST_P(WebmFrameThreadUnitTest, PauseTest) {
+ int32_t index1 = GetParam().first;
+ int32_t index2 = GetParam().second;
+ ASSERT_NO_FATAL_FAILURE(createWebmThreads({index1, index2}));
+
+ ASSERT_NO_FATAL_FAILURE(startWebmFrameThreads());
+
+ int32_t offset = 0;
+ ASSERT_NO_FATAL_FAILURE(writeFileData(offset, kNumFramesToWrite));
+ offset += kNumFramesToWrite;
+
+ for (int idx = 0; idx < kMaxLoopCount; idx++) {
+ // pause the threads
+ status_t status = mAudioThread->pause();
+ ASSERT_EQ(status, AMEDIA_OK) << "Failed to pause Audio Thread";
+ status = mVideoThread->pause();
+ ASSERT_EQ(status, AMEDIA_OK) << "Failed to pause Video Thread";
+
+ // Under pause state, no write should happen
+ ASSERT_NO_FATAL_FAILURE(writeFileData(offset, kNumFramesToWrite));
+ offset += kNumFramesToWrite;
+
+ status = mAudioThread->resume();
+ ASSERT_EQ(status, AMEDIA_OK) << "Failed to resume Audio Thread";
+ status = mVideoThread->resume();
+ ASSERT_EQ(status, AMEDIA_OK) << "Failed to resume Video Thread";
+
+ ASSERT_NO_FATAL_FAILURE(writeFileData(offset, kNumFramesToWrite));
+ offset += kNumFramesToWrite;
+ }
+
+ if (mSource[kAudioIdx]) mSource[kAudioIdx]->stop();
+ if (mSource[kVideoIdx]) mSource[kVideoIdx]->stop();
+ ASSERT_NO_FATAL_FAILURE(stopWebmFrameThreads());
+}
+
+INSTANTIATE_TEST_SUITE_P(WebmFrameThreadUnitTestAll, WebmFrameThreadUnitTest,
+ ::testing::Values(std::make_pair(0, 1), std::make_pair(0, 2),
+ std::make_pair(0, 3), std::make_pair(1, 0),
+ std::make_pair(1, 2), std::make_pair(1, 3),
+ std::make_pair(2, 3)));
+
+int main(int argc, char **argv) {
+ ::testing::InitGoogleTest(&argc, argv);
+ int status = RUN_ALL_TESTS();
+ ALOGV("Test result = %d\n", status);
+ return status;
+}
diff --git a/media/libstagefright/xmlparser/MediaCodecsXmlParser.cpp b/media/libstagefright/xmlparser/MediaCodecsXmlParser.cpp
index 9783e9b..d905b8d 100644
--- a/media/libstagefright/xmlparser/MediaCodecsXmlParser.cpp
+++ b/media/libstagefright/xmlparser/MediaCodecsXmlParser.cpp
@@ -1269,7 +1269,7 @@
void MediaCodecsXmlParser::Impl::State::addDetail(
const std::string &key, const std::string &value) {
CHECK(inType());
- ALOGI("limit: %s = %s", key.c_str(), value.c_str());
+ ALOGV("limit: %s = %s", key.c_str(), value.c_str());
const StringSet &variants = mVariantsStack.back();
if (variants.empty()) {
type()[key] = value;
diff --git a/media/mtp/IMtpHandle.h b/media/mtp/IMtpHandle.h
index fd14b18..0435e82 100644
--- a/media/mtp/IMtpHandle.h
+++ b/media/mtp/IMtpHandle.h
@@ -16,7 +16,7 @@
#ifndef _IMTP_HANDLE_H
#define _IMTP_HANDLE_H
-#include <linux/usb/f_mtp.h>
+#include "f_mtp.h"
namespace android {
diff --git a/media/mtp/MtpServer.cpp b/media/mtp/MtpServer.cpp
index ca8cb78..a291939 100644
--- a/media/mtp/MtpServer.cpp
+++ b/media/mtp/MtpServer.cpp
@@ -42,6 +42,7 @@
#include "MtpServer.h"
#include "MtpStorage.h"
#include "MtpStringBuffer.h"
+#include "android-base/strings.h"
namespace android {
@@ -955,6 +956,11 @@
if (!mData.getString(modified)) return MTP_RESPONSE_INVALID_PARAMETER; // date modified
// keywords follow
+ int type = storage->getType();
+ if (type == MTP_STORAGE_REMOVABLE_RAM) {
+ std::string str = android::base::Trim((const char*)name);
+ name.set(str.c_str());
+ }
ALOGV("name: %s format: 0x%04X (%s)\n", (const char*)name, format,
MtpDebug::getFormatCodeName(format));
time_t modifiedTime;
diff --git a/media/mtp/PosixAsyncIO.cpp b/media/mtp/PosixAsyncIO.cpp
index 72c07cc..8205e3b 100644
--- a/media/mtp/PosixAsyncIO.cpp
+++ b/media/mtp/PosixAsyncIO.cpp
@@ -47,10 +47,10 @@
CHECK(aiocbp->queued);
int ret;
if (aiocbp->read) {
- ret = TEMP_FAILURE_RETRY(pread(aiocbp->aio_fildes,
+ ret = TEMP_FAILURE_RETRY(pread64(aiocbp->aio_fildes,
aiocbp->aio_buf, aiocbp->aio_nbytes, aiocbp->aio_offset));
} else {
- ret = TEMP_FAILURE_RETRY(pwrite(aiocbp->aio_fildes,
+ ret = TEMP_FAILURE_RETRY(pwrite64(aiocbp->aio_fildes,
aiocbp->aio_buf, aiocbp->aio_nbytes, aiocbp->aio_offset));
}
{
@@ -139,7 +139,7 @@
return 0;
}
-void aio_prepare(struct aiocb *aiocbp, void* buf, size_t count, off_t offset) {
+void aio_prepare(struct aiocb *aiocbp, void* buf, size_t count, off64_t offset) {
aiocbp->aio_buf = buf;
aiocbp->aio_offset = offset;
aiocbp->aio_nbytes = count;
diff --git a/media/mtp/PosixAsyncIO.h b/media/mtp/PosixAsyncIO.h
index 2bb5735..2bcae4c 100644
--- a/media/mtp/PosixAsyncIO.h
+++ b/media/mtp/PosixAsyncIO.h
@@ -32,7 +32,7 @@
int aio_fildes;
void *aio_buf;
- off_t aio_offset;
+ off64_t aio_offset;
size_t aio_nbytes;
// Used internally
@@ -61,7 +61,7 @@
ssize_t aio_return(struct aiocb *);
// Helper method for setting aiocb members
-void aio_prepare(struct aiocb *, void*, size_t, off_t);
+void aio_prepare(struct aiocb *, void*, size_t, off64_t);
#endif // POSIXASYNCIO_H
diff --git a/media/mtp/f_mtp.h b/media/mtp/f_mtp.h
new file mode 100644
index 0000000..22ec771
--- /dev/null
+++ b/media/mtp/f_mtp.h
@@ -0,0 +1,43 @@
+/****************************************************************************
+ ****************************************************************************
+ ***
+ *** This header was automatically generated from a Linux kernel header
+ *** of the same name, to make information necessary for userspace to
+ *** call into the kernel available to libc. It contains only constants,
+ *** structures, and macros generated from the original header, and thus,
+ *** contains no copyrightable information.
+ ***
+ *** To edit the content of this header, modify the corresponding
+ *** source file (e.g. under external/kernel-headers/original/) then
+ *** run bionic/libc/kernel/tools/update_all.py
+ ***
+ *** Any manual change here will be lost the next time this script will
+ *** be run. You've been warned!
+ ***
+ ****************************************************************************
+ ****************************************************************************/
+#ifndef _UAPI_LINUX_USB_F_MTP_H
+#define _UAPI_LINUX_USB_F_MTP_H
+#include <linux/ioctl.h>
+#include <linux/types.h>
+/* WARNING: DO NOT EDIT, AUTO-GENERATED CODE - SEE TOP FOR INSTRUCTIONS */
+struct mtp_file_range {
+ int fd;
+ loff_t offset;
+ int64_t length;
+/* WARNING: DO NOT EDIT, AUTO-GENERATED CODE - SEE TOP FOR INSTRUCTIONS */
+ uint16_t command;
+ uint32_t transaction_id;
+};
+struct mtp_event {
+/* WARNING: DO NOT EDIT, AUTO-GENERATED CODE - SEE TOP FOR INSTRUCTIONS */
+ size_t length;
+ void * data;
+};
+#define MTP_SEND_FILE _IOW('M', 0, struct mtp_file_range)
+/* WARNING: DO NOT EDIT, AUTO-GENERATED CODE - SEE TOP FOR INSTRUCTIONS */
+#define MTP_RECEIVE_FILE _IOW('M', 1, struct mtp_file_range)
+#define MTP_SEND_EVENT _IOW('M', 3, struct mtp_event)
+#define MTP_SEND_FILE_WITH_HEADER _IOW('M', 4, struct mtp_file_range)
+#endif
+/* WARNING: DO NOT EDIT, AUTO-GENERATED CODE - SEE TOP FOR INSTRUCTIONS */
diff --git a/media/ndk/Android.bp b/media/ndk/Android.bp
index 9c7a630..24cad4d 100644
--- a/media/ndk/Android.bp
+++ b/media/ndk/Android.bp
@@ -54,7 +54,6 @@
],
include_dirs: [
- "bionic/libc/private",
"frameworks/base/core/jni",
"frameworks/native/include/media/openmax",
"system/media/camera/include",
@@ -70,16 +69,21 @@
"libgrallocusage",
],
+ header_libs: [
+ "libmediadrm_headers",
+ ],
+
shared_libs: [
"android.hardware.graphics.bufferqueue@1.0",
"android.hidl.token@1.0-utils",
"libandroid_runtime_lazy",
"libbase",
"libbinder",
+ "libdatasource",
"libmedia",
+ "libmediadrm",
"libmedia_omx",
"libmedia_jni_utils",
- "libmediadrm",
"libstagefright",
"libstagefright_foundation",
"liblog",
@@ -90,7 +94,6 @@
"libhidlbase",
"libgui",
"libui",
- "libmedia2_jni_core",
"libmediandk_utils",
],
@@ -142,6 +145,10 @@
"-Wall",
],
+ header_libs: [
+ "libmedia_headers",
+ ],
+
shared_libs: [
],
diff --git a/media/ndk/NdkMediaCodec.cpp b/media/ndk/NdkMediaCodec.cpp
index e041533..af21a99 100644
--- a/media/ndk/NdkMediaCodec.cpp
+++ b/media/ndk/NdkMediaCodec.cpp
@@ -250,8 +250,8 @@
ALOGE("CB_ERROR: err is expected.");
break;
}
- if (!msg->findInt32("action", &actionCode)) {
- ALOGE("CB_ERROR: action is expected.");
+ if (!msg->findInt32("actionCode", &actionCode)) {
+ ALOGE("CB_ERROR: actionCode is expected.");
break;
}
msg->findString("detail", &detail);
diff --git a/media/ndk/NdkMediaCrypto.cpp b/media/ndk/NdkMediaCrypto.cpp
index ce2c660..792fc00 100644
--- a/media/ndk/NdkMediaCrypto.cpp
+++ b/media/ndk/NdkMediaCrypto.cpp
@@ -27,8 +27,8 @@
#include <utils/Log.h>
#include <utils/StrongPointer.h>
#include <binder/IServiceManager.h>
-#include <media/ICrypto.h>
-#include <media/IMediaDrmService.h>
+#include <mediadrm/ICrypto.h>
+#include <mediadrm/IMediaDrmService.h>
#include <android_util_Binder.h>
#include <jni.h>
diff --git a/media/ndk/NdkMediaCryptoPriv.h b/media/ndk/NdkMediaCryptoPriv.h
index 14ea928..8664d95 100644
--- a/media/ndk/NdkMediaCryptoPriv.h
+++ b/media/ndk/NdkMediaCryptoPriv.h
@@ -30,7 +30,7 @@
#include <sys/types.h>
#include <utils/StrongPointer.h>
-#include <media/ICrypto.h>
+#include <mediadrm/ICrypto.h>
using namespace android;
diff --git a/media/ndk/NdkMediaDataSource.cpp b/media/ndk/NdkMediaDataSource.cpp
index 7979c2f..c1d4686 100644
--- a/media/ndk/NdkMediaDataSource.cpp
+++ b/media/ndk/NdkMediaDataSource.cpp
@@ -26,18 +26,16 @@
#include <android_runtime/AndroidRuntime.h>
#include <android_util_Binder.h>
#include <cutils/properties.h>
-#include <utils/Log.h>
-#include <utils/StrongPointer.h>
+#include <datasource/DataSourceFactory.h>
+#include <datasource/HTTPBase.h>
+#include <datasource/NuCachedSource2.h>
#include <media/IMediaHTTPService.h>
#include <media/NdkMediaError.h>
#include <media/NdkMediaDataSource.h>
-#include <media/stagefright/DataSourceFactory.h>
#include <media/stagefright/InterfaceUtils.h>
-#include <mediaplayer2/JavaVMHelper.h>
-#include <mediaplayer2/JMedia2HTTPService.h>
+#include <utils/Log.h>
+#include <utils/StrongPointer.h>
-#include "../../libstagefright/include/HTTPBase.h"
-#include "../../libstagefright/include/NuCachedSource2.h"
#include "NdkMediaDataSourceCallbacksPriv.h"
@@ -120,18 +118,11 @@
return size >= 0 ? OK : UNKNOWN_ERROR;
}
-static sp<MediaHTTPService> createMediaHttpServiceFromJavaObj(JNIEnv *env, jobject obj, int version) {
+static sp<MediaHTTPService> createMediaHttpServiceFromJavaObj(JNIEnv *env, jobject obj) {
if (obj == NULL) {
return NULL;
}
- switch (version) {
- case 1:
- return interface_cast<IMediaHTTPService>(ibinderForJavaObject(env, obj));
- case 2:
- return new JMedia2HTTPService(env, obj);
- default:
- return NULL;
- }
+ return interface_cast<IMediaHTTPService>(ibinderForJavaObject(env, obj));
}
static sp<MediaHTTPService> createMediaHttpServiceTemplate(
@@ -139,8 +130,7 @@
const char *uri,
const char *clazz,
const char *method,
- const char *signature,
- int version) {
+ const char *signature) {
jobject service = NULL;
if (env == NULL) {
ALOGE("http service must be created from Java thread");
@@ -167,34 +157,22 @@
env->DeleteLocalRef(juri);
env->ExceptionClear();
- sp<MediaHTTPService> httpService = createMediaHttpServiceFromJavaObj(env, service, version);
+ sp<MediaHTTPService> httpService = createMediaHttpServiceFromJavaObj(env, service);
return httpService;
}
-sp<MediaHTTPService> createMediaHttpService(const char *uri, int version) {
+sp<MediaHTTPService> createMediaHttpService(const char *uri) {
JNIEnv *env;
const char *clazz, *method, *signature;
- switch (version) {
- case 1:
- env = AndroidRuntime::getJNIEnv();
- clazz = "android/media/MediaHTTPService";
- method = "createHttpServiceBinderIfNecessary";
- signature = "(Ljava/lang/String;)Landroid/os/IBinder;";
- break;
- case 2:
- env = JavaVMHelper::getJNIEnv();
- clazz = "android/media/Media2HTTPService";
- method = "createHTTPService";
- signature = "(Ljava/lang/String;)Landroid/media/Media2HTTPService;";
- break;
- default:
- return NULL;
- }
+ env = AndroidRuntime::getJNIEnv();
+ clazz = "android/media/MediaHTTPService";
+ method = "createHttpServiceBinderIfNecessary";
+ signature = "(Ljava/lang/String;)Landroid/os/IBinder;";
- return createMediaHttpServiceTemplate(env, uri, clazz, method, signature, version);
+ return createMediaHttpServiceTemplate(env, uri, clazz, method, signature);
}
@@ -216,7 +194,7 @@
int numheaders,
const char * const *key_values) {
- sp<MediaHTTPService> service = createMediaHttpService(uri, /* version = */ 1);
+ sp<MediaHTTPService> service = createMediaHttpService(uri);
KeyedVector<String8, String8> headers;
for (int i = 0; i < numheaders; ++i) {
String8 key8(key_values[i * 2]);
@@ -224,7 +202,7 @@
headers.add(key8, value8);
}
- sp<DataSource> source = DataSourceFactory::CreateFromURI(service, uri, &headers);
+ sp<DataSource> source = DataSourceFactory::getInstance()->CreateFromURI(service, uri, &headers);
if (source == NULL) {
ALOGE("AMediaDataSource_newUri source is null");
return NULL;
diff --git a/media/ndk/NdkMediaDataSourcePriv.h b/media/ndk/NdkMediaDataSourcePriv.h
index 16ff974..ddcd7da 100644
--- a/media/ndk/NdkMediaDataSourcePriv.h
+++ b/media/ndk/NdkMediaDataSourcePriv.h
@@ -62,7 +62,7 @@
};
-sp<MediaHTTPService> createMediaHttpService(const char *uri, int version);
+sp<MediaHTTPService> createMediaHttpService(const char *uri);
#endif // _NDK_MEDIA_DATASOURCE_PRIV_H
diff --git a/media/ndk/NdkMediaDrm.cpp b/media/ndk/NdkMediaDrm.cpp
index cd5a23a..842216c 100644
--- a/media/ndk/NdkMediaDrm.cpp
+++ b/media/ndk/NdkMediaDrm.cpp
@@ -29,12 +29,12 @@
#include <android-base/properties.h>
#include <binder/PermissionController.h>
-#include <media/IDrm.h>
-#include <media/IDrmClient.h>
+#include <mediadrm/IDrm.h>
+#include <mediadrm/IDrmClient.h>
#include <media/stagefright/MediaErrors.h>
#include <binder/IServiceManager.h>
-#include <media/IMediaDrmService.h>
#include <media/NdkMediaCrypto.h>
+#include <mediadrm/IMediaDrmService.h>
using namespace android;
diff --git a/media/ndk/NdkMediaExtractor.cpp b/media/ndk/NdkMediaExtractor.cpp
index c83b255..0da0740 100644
--- a/media/ndk/NdkMediaExtractor.cpp
+++ b/media/ndk/NdkMediaExtractor.cpp
@@ -89,7 +89,7 @@
ALOGV("setDataSource(%s)", uri);
- sp<MediaHTTPService> httpService = createMediaHttpService(uri, /* version = */ 1);
+ sp<MediaHTTPService> httpService = createMediaHttpService(uri);
if (httpService == NULL) {
ALOGE("can't create http service");
return AMEDIA_ERROR_UNSUPPORTED;
diff --git a/media/ndk/include/media/NdkImage.h b/media/ndk/include/media/NdkImage.h
index 3e60de0..62b8624 100644
--- a/media/ndk/include/media/NdkImage.h
+++ b/media/ndk/include/media/NdkImage.h
@@ -570,6 +570,8 @@
* return {@link AMEDIA_ERROR_INVALID_OBJECT}. Application still needs to call this method on those
* {@link AImage} objects to fully delete the {@link AImage} object from memory.</p>
*
+ * Available since API level 24.
+ *
* @param image The {@link AImage} to be deleted.
*/
void AImage_delete(AImage* image) __INTRODUCED_IN(24);
@@ -577,6 +579,8 @@
/**
* Query the width of the input {@link AImage}.
*
+ * Available since API level 24.
+ *
* @param image the {@link AImage} of interest.
* @param width the width of the image will be filled here if the method call succeeeds.
*
@@ -591,6 +595,8 @@
/**
* Query the height of the input {@link AImage}.
*
+ * Available since API level 24.
+ *
* @param image the {@link AImage} of interest.
* @param height the height of the image will be filled here if the method call succeeeds.
*
@@ -607,6 +613,8 @@
*
* <p>The format value will be one of AIMAGE_FORMAT_* enum value.</p>
*
+ * Available since API level 24.
+ *
* @param image the {@link AImage} of interest.
* @param format the format of the image will be filled here if the method call succeeeds.
*
@@ -624,6 +632,8 @@
* <p>The crop rectangle specifies the region of valid pixels in the image, using coordinates in the
* largest-resolution plane.</p>
*
+ * Available since API level 24.
+ *
* @param image the {@link AImage} of interest.
* @param rect the cropped rectangle of the image will be filled here if the method call succeeeds.
*
@@ -648,6 +658,8 @@
* {@link ACameraCaptureSession_captureCallbacks#onCaptureCompleted} callback.
* </p>
*
+ * Available since API level 24.
+ *
* @param image the {@link AImage} of interest.
* @param timestampNs the timestamp of the image will be filled here if the method call succeeeds.
*
@@ -665,6 +677,8 @@
* <p>The number of plane of an {@link AImage} is determined by its format, which can be queried by
* {@link AImage_getFormat} method.</p>
*
+ * Available since API level 24.
+ *
* @param image the {@link AImage} of interest.
* @param numPlanes the number of planes of the image will be filled here if the method call
* succeeeds.
@@ -687,6 +701,8 @@
* being returned.
* For formats where pixel stride is well defined, the pixel stride is always greater than 0.</p>
*
+ * Available since API level 24.
+ *
* @param image the {@link AImage} of interest.
* @param planeIdx the index of the plane. Must be less than the number of planes of input image.
* @param pixelStride the pixel stride of the image will be filled here if the method call succeeeds.
@@ -714,6 +730,8 @@
* being returned.
* For formats where row stride is well defined, the row stride is always greater than 0.</p>
*
+ * Available since API level 24.
+ *
* @param image the {@link AImage} of interest.
* @param planeIdx the index of the plane. Must be less than the number of planes of input image.
* @param rowStride the row stride of the image will be filled here if the method call succeeeds.
@@ -739,6 +757,8 @@
* pointer from previous AImage_getPlaneData call becomes invalid. Do NOT use it after the
* {@link AImage} or the parent {@link AImageReader} is deleted.</p>
*
+ * Available since API level 24.
+ *
* @param image the {@link AImage} of interest.
* @param planeIdx the index of the plane. Must be less than the number of planes of input image.
* @param data the data pointer of the image will be filled here if the method call succeeeds.
@@ -769,6 +789,8 @@
* signal the release of the hardware buffer back to the {@link AImageReader}'s queue using
* releaseFenceFd.</p>
*
+ * Available since API level 26.
+ *
* @param image The {@link AImage} to be deleted.
* @param releaseFenceFd A sync fence fd defined in {@link sync.h}, which signals the release of
* underlying {@link AHardwareBuffer}.
@@ -794,6 +816,8 @@
* {@link AImageReader_setBufferRemovedListener} to be notified when the buffer is no longer used
* by {@link AImageReader}.</p>
*
+ * Available since API level 26.
+ *
* @param image the {@link AImage} of interest.
* @param outBuffer The memory area pointed to by buffer will contain the acquired AHardwareBuffer
* handle.
diff --git a/media/ndk/include/media/NdkImageReader.h b/media/ndk/include/media/NdkImageReader.h
index e5d863c..600ffc9 100644
--- a/media/ndk/include/media/NdkImageReader.h
+++ b/media/ndk/include/media/NdkImageReader.h
@@ -67,6 +67,8 @@
* The valid sizes and formats depend on the source of the image data.
* </p>
*
+ * Available since API level 24.
+ *
* @param width The default width in pixels of the Images that this reader will produce.
* @param height The default height in pixels of the Images that this reader will produce.
* @param format The format of the Image that this reader will produce. This must be one of the
@@ -101,6 +103,8 @@
* making any of data pointers obtained from {@link AImage_getPlaneData} invalid. Do NOT access
* the reader object or any of those data pointers after this method returns.</p>
*
+ * Available since API level 24.
+ *
* @param reader The image reader to be deleted.
*/
void AImageReader_delete(AImageReader* reader) __INTRODUCED_IN(24);
@@ -108,6 +112,8 @@
/**
* Get a {@link ANativeWindow} that can be used to produce {@link AImage} for this image reader.
*
+ * Available since API level 24.
+ *
* @param reader The image reader of interest.
* @param window The output {@link ANativeWindow} will be filled here if the method call succeeds.
* The {@link ANativeWindow} is managed by this image reader. Do NOT call
@@ -126,6 +132,8 @@
* {@link ANativeWindow}. If so, the actual width of the images can be found using
* {@link AImage_getWidth}.</p>
*
+ * Available since API level 24.
+ *
* @param reader The image reader of interest.
* @param width the default width of the reader will be filled here if the method call succeeeds.
*
@@ -142,6 +150,8 @@
* {@link ANativeWindow}. If so, the actual height of the images can be found using
* {@link AImage_getHeight}.</p>
*
+ * Available since API level 24.
+ *
* @param reader The image reader of interest.
* @param height the default height of the reader will be filled here if the method call succeeeds.
*
@@ -154,6 +164,8 @@
/**
* Query the format of the {@link AImage} generated by this reader.
*
+ * Available since API level 24.
+ *
* @param reader The image reader of interest.
* @param format the fromat of the reader will be filled here if the method call succeeeds. The
* value will be one of the AIMAGE_FORMAT_* enum value defiend in {@link NdkImage.h}.
@@ -167,6 +179,8 @@
/**
* Query the maximum number of concurrently acquired {@link AImage}s of this reader.
*
+ * Available since API level 24.
+ *
* @param reader The image reader of interest.
* @param maxImages the maximum number of concurrently acquired images of the reader will be filled
* here if the method call succeeeds.
@@ -197,6 +211,8 @@
* {@link AImage_delete}.
* </p>
*
+ * Available since API level 24.
+ *
* @param reader The image reader of interest.
* @param image the acquired {@link AImage} will be filled here if the method call succeeeds.
*
@@ -214,7 +230,6 @@
media_status_t AImageReader_acquireNextImage(AImageReader* reader, /*out*/AImage** image) __INTRODUCED_IN(24);
/**
-
* Acquire the latest {@link AImage} from the image reader's queue, dropping older images.
*
* <p>
@@ -241,6 +256,8 @@
* {@link AImage_delete}.
* </p>
*
+ * Available since API level 24.
+ *
* @param reader The image reader of interest.
* @param image the acquired {@link AImage} will be filled here if the method call succeeeds.
*
@@ -290,6 +307,8 @@
*
* Calling this method will replace previously registered listeners.
*
+ * Available since API level 24.
+ *
* @param reader The image reader of interest.
* @param listener The {@link AImageReader_ImageListener} to be registered. Set this to NULL if
* the application no longer needs to listen to new images.
@@ -356,6 +375,9 @@
* {@link AHARDWAREBUFFER_USAGE_GPU_SAMPLED_IMAGE}, or combined</td>
* </tr>
* </table>
+ *
+ * Available since API level 26.
+ *
* @return <ul>
* <li>{@link AMEDIA_OK} if the method call succeeds.</li>
* <li>{@link AMEDIA_ERROR_INVALID_PARAMETER} if reader is NULL, or one or more of width,
@@ -377,6 +399,8 @@
* additional parameter for the sync fence. All other parameters and the return values are
* identical to those passed to {@link AImageReader_acquireNextImage}.</p>
*
+ * Available since API level 26.
+ *
* @param acquireFenceFd A sync fence fd defined in {@link sync.h}, which is used to signal when the
* buffer is ready to consume. When synchronization fence is not needed, fence will be set
* to -1 and the {@link AImage} returned is ready for use immediately. Otherwise, user shall
@@ -397,6 +421,8 @@
* additional parameter for the sync fence. All other parameters and the return values are
* identical to those passed to {@link AImageReader_acquireLatestImage}.</p>
*
+ * Available since API level 26.
+ *
* @param acquireFenceFd A sync fence fd defined in {@link sync.h}, which is used to signal when the
* buffer is ready to consume. When synchronization fence is not needed, fence will be set
* to -1 and the {@link AImage} returned is ready for use immediately. Otherwise, user shall
@@ -408,6 +434,7 @@
*/
media_status_t AImageReader_acquireLatestImageAsync(
AImageReader* reader, /*out*/AImage** image, /*out*/int* acquireFenceFd) __INTRODUCED_IN(26);
+
/**
* Signature of the callback which is called when {@link AImageReader} is about to remove a buffer.
*
@@ -451,6 +478,8 @@
*
* <p>Note that calling this method will replace previously registered listeners.</p>
*
+ * Available since API level 26.
+ *
* @param reader The image reader of interest.
* @param listener the {@link AImageReader_BufferRemovedListener} to be registered. Set this to
* NULL if application no longer needs to listen to buffer removed events.
diff --git a/media/ndk/include/media/NdkMediaCodec.h b/media/ndk/include/media/NdkMediaCodec.h
index b3ee853..8fb6a87 100644
--- a/media/ndk/include/media/NdkMediaCodec.h
+++ b/media/ndk/include/media/NdkMediaCodec.h
@@ -127,27 +127,37 @@
* Create codec by name. Use this if you know the exact codec you want to use.
* When configuring, you will need to specify whether to use the codec as an
* encoder or decoder.
+ *
+ * Available since API level 21.
*/
AMediaCodec* AMediaCodec_createCodecByName(const char *name) __INTRODUCED_IN(21);
/**
* Create codec by mime type. Most applications will use this, specifying a
* mime type obtained from media extractor.
+ *
+ * Available since API level 21.
*/
AMediaCodec* AMediaCodec_createDecoderByType(const char *mime_type) __INTRODUCED_IN(21);
/**
* Create encoder by name.
+ *
+ * Available since API level 21.
*/
AMediaCodec* AMediaCodec_createEncoderByType(const char *mime_type) __INTRODUCED_IN(21);
/**
- * delete the codec and free its resources
+ * Delete the codec and free its resources.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaCodec_delete(AMediaCodec*) __INTRODUCED_IN(21);
/**
* Configure the codec. For decoding you would typically get the format from an extractor.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaCodec_configure(
AMediaCodec*,
@@ -159,29 +169,39 @@
/**
* Start the codec. A codec must be configured before it can be started, and must be started
* before buffers can be sent to it.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaCodec_start(AMediaCodec*) __INTRODUCED_IN(21);
/**
* Stop the codec.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaCodec_stop(AMediaCodec*) __INTRODUCED_IN(21);
/*
* Flush the codec's input and output. All indices previously returned from calls to
* AMediaCodec_dequeueInputBuffer and AMediaCodec_dequeueOutputBuffer become invalid.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaCodec_flush(AMediaCodec*) __INTRODUCED_IN(21);
/**
* Get an input buffer. The specified buffer index must have been previously obtained from
* dequeueInputBuffer, and not yet queued.
+ *
+ * Available since API level 21.
*/
uint8_t* AMediaCodec_getInputBuffer(AMediaCodec*, size_t idx, size_t *out_size) __INTRODUCED_IN(21);
/**
* Get an output buffer. The specified buffer index must have been previously obtained from
* dequeueOutputBuffer, and not yet queued.
+ *
+ * Available since API level 21.
*/
uint8_t* AMediaCodec_getOutputBuffer(AMediaCodec*, size_t idx, size_t *out_size) __INTRODUCED_IN(21);
@@ -189,6 +209,8 @@
* Get the index of the next available input buffer. An app will typically use this with
* getInputBuffer() to get a pointer to the buffer, then copy the data to be encoded or decoded
* into the buffer before passing it to the codec.
+ *
+ * Available since API level 21.
*/
ssize_t AMediaCodec_dequeueInputBuffer(AMediaCodec*, int64_t timeoutUs) __INTRODUCED_IN(21);
@@ -218,6 +240,8 @@
/**
* Send the specified buffer to the codec for processing.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaCodec_queueInputBuffer(AMediaCodec*, size_t idx,
_off_t_compat offset, size_t size,
@@ -225,6 +249,8 @@
/**
* Send the specified buffer to the codec for processing.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaCodec_queueSecureInputBuffer(AMediaCodec*, size_t idx,
_off_t_compat offset,
@@ -235,15 +261,26 @@
/**
* Get the index of the next available buffer of processed data.
+ *
+ * Available since API level 21.
*/
ssize_t AMediaCodec_dequeueOutputBuffer(AMediaCodec*, AMediaCodecBufferInfo *info,
int64_t timeoutUs) __INTRODUCED_IN(21);
+
+/**
+ * Returns the format of the codec's output.
+ * The caller must free the returned format.
+ *
+ * Available since API level 21.
+ */
AMediaFormat* AMediaCodec_getOutputFormat(AMediaCodec*) __INTRODUCED_IN(21);
/**
* If you are done with a buffer, use this call to return the buffer to
* the codec. If you previously specified a surface when configuring this
* video decoder you can optionally render the buffer.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaCodec_releaseOutputBuffer(AMediaCodec*, size_t idx, bool render) __INTRODUCED_IN(21);
@@ -256,6 +293,8 @@
* to ImageReader (software readable) output.
*
* For more details, see the Java documentation for MediaCodec.setOutputSurface.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaCodec_setOutputSurface(AMediaCodec*, ANativeWindow* surface) __INTRODUCED_IN(21);
@@ -266,6 +305,8 @@
* this call will simply return the buffer to the codec.
*
* For more details, see the Java documentation for MediaCodec.releaseOutputBuffer.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaCodec_releaseOutputBufferAtTime(
AMediaCodec *mData, size_t idx, int64_t timestampNs) __INTRODUCED_IN(21);
@@ -282,6 +323,8 @@
* ANativeWindow_release() when done.
*
* For more details, see the Java documentation for MediaCodec.createInputSurface.
+ *
+ * Available since API level 26.
*/
media_status_t AMediaCodec_createInputSurface(
AMediaCodec *mData, ANativeWindow **surface) __INTRODUCED_IN(26);
@@ -298,6 +341,8 @@
* ANativeWindow_release() when done.
*
* For more details, see the Java documentation for MediaCodec.createPersistentInputSurface.
+ *
+ * Available since API level 26.
*/
media_status_t AMediaCodec_createPersistentInputSurface(
ANativeWindow **surface) __INTRODUCED_IN(26);
@@ -311,6 +356,8 @@
* AMediaCodec_configure(..); and before AMediaCodec_start() has been called.
*
* For more details, see the Java documentation for MediaCodec.setInputSurface.
+ *
+ * Available since API level 26.
*/
media_status_t AMediaCodec_setInputSurface(
AMediaCodec *mData, ANativeWindow *surface) __INTRODUCED_IN(26);
@@ -322,6 +369,8 @@
* after AMediaCodec_start() has been called.
*
* NOTE: Some of these parameter changes may silently fail to apply.
+ *
+ * Available since API level 26.
*/
media_status_t AMediaCodec_setParameters(
AMediaCodec *mData, const AMediaFormat* params) __INTRODUCED_IN(26);
@@ -339,6 +388,8 @@
* Returns AMEDIA_OK when completed succesfully.
*
* For more details, see the Java documentation for MediaCodec.signalEndOfInputStream.
+ *
+ * Available since API level 26.
*/
media_status_t AMediaCodec_signalEndOfInputStream(AMediaCodec *mData) __INTRODUCED_IN(26);
@@ -349,6 +400,9 @@
/**
* Get format of the buffer. The specified buffer index must have been previously obtained from
* dequeueOutputBuffer.
+ * The caller must free the returned format.
+ *
+ * Available since API level 28.
*/
AMediaFormat* AMediaCodec_getBufferFormat(AMediaCodec*, size_t index) __INTRODUCED_IN(28);
@@ -356,11 +410,15 @@
* Get the component name. If the codec was created by createDecoderByType
* or createEncoderByType, what component is chosen is not known beforehand.
* Caller shall call AMediaCodec_releaseName to free the returned pointer.
+ *
+ * Available since API level 28.
*/
media_status_t AMediaCodec_getName(AMediaCodec*, char** out_name) __INTRODUCED_IN(28);
/**
* Free the memory pointed by name which is returned by AMediaCodec_getName.
+ *
+ * Available since API level 28.
*/
void AMediaCodec_releaseName(AMediaCodec*, char* name) __INTRODUCED_IN(28);
@@ -382,6 +440,8 @@
* All callbacks are fired on one NDK internal thread.
* AMediaCodec_setAsyncNotifyCallback should not be called on the callback thread.
* No heavy duty task should be performed on callback thread.
+ *
+ * Available since API level 28.
*/
media_status_t AMediaCodec_setAsyncNotifyCallback(
AMediaCodec*,
@@ -390,6 +450,8 @@
/**
* Release the crypto if applicable.
+ *
+ * Available since API level 28.
*/
media_status_t AMediaCodec_releaseCrypto(AMediaCodec*) __INTRODUCED_IN(28);
@@ -397,12 +459,17 @@
* Call this after AMediaCodec_configure() returns successfully to get the input
* format accepted by the codec. Do this to determine what optional configuration
* parameters were supported by the codec.
+ * The caller must free the returned format.
+ *
+ * Available since API level 28.
*/
AMediaFormat* AMediaCodec_getInputFormat(AMediaCodec*) __INTRODUCED_IN(28);
/**
* Returns true if the codec cannot proceed further, but can be recovered by stopping,
* configuring, and starting again.
+ *
+ * Available since API level 28.
*/
bool AMediaCodecActionCode_isRecoverable(int32_t actionCode) __INTRODUCED_IN(28);
@@ -410,6 +477,8 @@
* Returns true if the codec error is a transient issue, perhaps due to
* resource constraints, and that the method (or encoding/decoding) may be
* retried at a later time.
+ *
+ * Available since API level 28.
*/
bool AMediaCodecActionCode_isTransient(int32_t actionCode) __INTRODUCED_IN(28);
@@ -440,6 +509,8 @@
* numBytesOfClearData can be null to indicate that all data is encrypted.
* This information encapsulates per-sample metadata as outlined in
* ISO/IEC FDIS 23001-7:2011 "Common encryption in ISO base media file format files".
+ *
+ * Available since API level 21.
*/
AMediaCodecCryptoInfo *AMediaCodecCryptoInfo_new(
int numsubsamples,
@@ -450,13 +521,17 @@
size_t *encryptedbytes) __INTRODUCED_IN(21);
/**
- * delete an AMediaCodecCryptoInfo created previously with AMediaCodecCryptoInfo_new, or
- * obtained from AMediaExtractor
+ * Delete an AMediaCodecCryptoInfo created previously with AMediaCodecCryptoInfo_new, or
+ * obtained from AMediaExtractor.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaCodecCryptoInfo_delete(AMediaCodecCryptoInfo*) __INTRODUCED_IN(21);
/**
- * Set the crypto pattern on an AMediaCryptoInfo object
+ * Set the crypto pattern on an AMediaCryptoInfo object.
+ *
+ * Available since API level 21.
*/
void AMediaCodecCryptoInfo_setPattern(
AMediaCodecCryptoInfo *info,
@@ -464,32 +539,44 @@
/**
* The number of subsamples that make up the buffer's contents.
+ *
+ * Available since API level 21.
*/
size_t AMediaCodecCryptoInfo_getNumSubSamples(AMediaCodecCryptoInfo*) __INTRODUCED_IN(21);
/**
- * A 16-byte opaque key
+ * A 16-byte opaque key.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaCodecCryptoInfo_getKey(AMediaCodecCryptoInfo*, uint8_t *dst) __INTRODUCED_IN(21);
/**
- * A 16-byte initialization vector
+ * A 16-byte initialization vector.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaCodecCryptoInfo_getIV(AMediaCodecCryptoInfo*, uint8_t *dst) __INTRODUCED_IN(21);
/**
* The type of encryption that has been applied,
* one of AMEDIACODECRYPTOINFO_MODE_CLEAR or AMEDIACODECRYPTOINFO_MODE_AES_CTR.
+ *
+ * Available since API level 21.
*/
cryptoinfo_mode_t AMediaCodecCryptoInfo_getMode(AMediaCodecCryptoInfo*) __INTRODUCED_IN(21);
/**
* The number of leading unencrypted bytes in each subsample.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaCodecCryptoInfo_getClearBytes(AMediaCodecCryptoInfo*, size_t *dst) __INTRODUCED_IN(21);
/**
* The number of trailing encrypted bytes in each subsample.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaCodecCryptoInfo_getEncryptedBytes(AMediaCodecCryptoInfo*, size_t *dst) __INTRODUCED_IN(21);
diff --git a/media/ndk/include/media/NdkMediaCrypto.h b/media/ndk/include/media/NdkMediaCrypto.h
index bcdf9a0..3fa07c7 100644
--- a/media/ndk/include/media/NdkMediaCrypto.h
+++ b/media/ndk/include/media/NdkMediaCrypto.h
@@ -49,12 +49,24 @@
#if __ANDROID_API__ >= 21
+/**
+ * Available since API level 21.
+ */
bool AMediaCrypto_isCryptoSchemeSupported(const AMediaUUID uuid) __INTRODUCED_IN(21);
+/**
+ * Available since API level 21.
+ */
bool AMediaCrypto_requiresSecureDecoderComponent(const char *mime) __INTRODUCED_IN(21);
+/**
+ * Available since API level 21.
+ */
AMediaCrypto* AMediaCrypto_new(const AMediaUUID uuid, const void *initData, size_t initDataSize) __INTRODUCED_IN(21);
+/**
+ * Available since API level 21.
+ */
void AMediaCrypto_delete(AMediaCrypto* crypto) __INTRODUCED_IN(21);
#endif /* __ANDROID_API__ >= 21 */
diff --git a/media/ndk/include/media/NdkMediaDataSource.h b/media/ndk/include/media/NdkMediaDataSource.h
index 16b1eb3..0577df2 100644
--- a/media/ndk/include/media/NdkMediaDataSource.h
+++ b/media/ndk/include/media/NdkMediaDataSource.h
@@ -88,6 +88,8 @@
/**
* Create new media data source. Returns NULL if memory allocation
* for the new data source object fails.
+ *
+ * Available since API level 28.
*/
AMediaDataSource* AMediaDataSource_new() __INTRODUCED_IN(28);
@@ -116,6 +118,7 @@
* ...
* key_values[(numheaders - 1) * 2]:key_values[(numheaders - 1) * 2 + 1]
*
+ * Available since API level 29.
*/
AMediaDataSource* AMediaDataSource_newUri(const char *uri,
int numheaders,
@@ -125,12 +128,16 @@
/**
* Delete a previously created media data source.
+ *
+ * Available since API level 28.
*/
void AMediaDataSource_delete(AMediaDataSource*) __INTRODUCED_IN(28);
/**
* Set an user provided opaque handle. This opaque handle is passed as
* the first argument to the data source callbacks.
+ *
+ * Available since API level 28.
*/
void AMediaDataSource_setUserdata(
AMediaDataSource*, void *userdata) __INTRODUCED_IN(28);
@@ -145,6 +152,8 @@
*
* Please refer to the definition of AMediaDataSourceReadAt for
* additional details.
+ *
+ * Available since API level 28.
*/
void AMediaDataSource_setReadAt(
AMediaDataSource*,
@@ -156,6 +165,8 @@
*
* Please refer to the definition of AMediaDataSourceGetSize for
* additional details.
+ *
+ * Available since API level 28.
*/
void AMediaDataSource_setGetSize(
AMediaDataSource*,
@@ -167,6 +178,8 @@
*
* Please refer to the definition of AMediaDataSourceClose for
* additional details.
+ *
+ * Available since API level 28.
*/
void AMediaDataSource_setClose(
AMediaDataSource*,
@@ -181,6 +194,8 @@
*
* Please refer to the definition of AMediaDataSourceClose for
* additional details.
+ *
+ * Available since API level 29.
*/
void AMediaDataSource_close(AMediaDataSource*) __INTRODUCED_IN(29);
@@ -191,6 +206,8 @@
*
* Please refer to the definition of AMediaDataSourceGetAvailableSize
* for additional details.
+ *
+ * Available since API level 29.
*/
void AMediaDataSource_setGetAvailableSize(
AMediaDataSource*,
diff --git a/media/ndk/include/media/NdkMediaDrm.h b/media/ndk/include/media/NdkMediaDrm.h
index 2e438d9..31f5c7d 100644
--- a/media/ndk/include/media/NdkMediaDrm.h
+++ b/media/ndk/include/media/NdkMediaDrm.h
@@ -174,41 +174,53 @@
* uuid identifies the universal unique ID of the crypto scheme. uuid must be 16 bytes.
* mimeType is the MIME type of the media container, e.g. "video/mp4". If mimeType
* is not known or required, it can be provided as NULL.
+ *
+ * Available since API level 21.
*/
bool AMediaDrm_isCryptoSchemeSupported(const uint8_t *uuid,
const char *mimeType) __INTRODUCED_IN(21);
/**
- * Create a MediaDrm instance from a UUID
+ * Create a MediaDrm instance from a UUID.
* uuid identifies the universal unique ID of the crypto scheme. uuid must be 16 bytes.
+ *
+ * Available since API level 21.
*/
AMediaDrm* AMediaDrm_createByUUID(const uint8_t *uuid) __INTRODUCED_IN(21);
/**
- * Release a MediaDrm object
+ * Release a MediaDrm object.
+ *
+ * Available since API level 21.
*/
void AMediaDrm_release(AMediaDrm *) __INTRODUCED_IN(21);
/**
- * Register a callback to be invoked when an event occurs
+ * Register a callback to be invoked when an event occurs.
*
- * listener is the callback that will be invoked on event
+ * listener is the callback that will be invoked on event.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaDrm_setOnEventListener(AMediaDrm *,
AMediaDrmEventListener listener) __INTRODUCED_IN(21);
/**
- * Register a callback to be invoked when an expiration update event occurs
+ * Register a callback to be invoked when an expiration update event occurs.
*
- * listener is the callback that will be invoked on event
+ * listener is the callback that will be invoked on event.
+ *
+ * Available since API level 29.
*/
media_status_t AMediaDrm_setOnExpirationUpdateListener(AMediaDrm *,
AMediaDrmExpirationUpdateListener listener) __INTRODUCED_IN(29);
/**
- * Register a callback to be invoked when a key status change event occurs
+ * Register a callback to be invoked when a key status change event occurs.
*
- * listener is the callback that will be invoked on event
+ * listener is the callback that will be invoked on event.
+ *
+ * Available since API level 29.
*/
media_status_t AMediaDrm_setOnKeysChangeListener(AMediaDrm *,
AMediaDrmKeysChangeListener listener) __INTRODUCED_IN(29);
@@ -216,8 +228,10 @@
/**
* Open a new session with the MediaDrm object. A session ID is returned.
*
- * returns MEDIADRM_NOT_PROVISIONED_ERROR if provisioning is needed
- * returns MEDIADRM_RESOURCE_BUSY_ERROR if required resources are in use
+ * Returns MEDIADRM_NOT_PROVISIONED_ERROR if provisioning is needed.
+ * Returns MEDIADRM_RESOURCE_BUSY_ERROR if required resources are in use.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaDrm_openSession(AMediaDrm *,
AMediaDrmSessionId *sessionId) __INTRODUCED_IN(21);
@@ -225,6 +239,8 @@
/**
* Close a session on the MediaDrm object that was previously opened
* with AMediaDrm_openSession.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaDrm_closeSession(AMediaDrm *,
const AMediaDrmSessionId *sessionId) __INTRODUCED_IN(21);
@@ -272,9 +288,11 @@
* MediaDrm object is released.
* 2. keyRequestSize will be set to the size of the request
*
- * returns MEDIADRM_NOT_PROVISIONED_ERROR if reprovisioning is needed, due to a
+ * Returns MEDIADRM_NOT_PROVISIONED_ERROR if reprovisioning is needed, due to a
* problem with the device certificate.
-*/
+ *
+ * Available since API level 21.
+ */
media_status_t AMediaDrm_getKeyRequest(AMediaDrm *, const AMediaDrmScope *scope,
const uint8_t *init, size_t initSize, const char *mimeType, AMediaDrmKeyType keyType,
const AMediaDrmKeyValue *optionalParameters, size_t numOptionalParameters,
@@ -295,8 +313,9 @@
*
* response points to the opaque response from the server
* responseSize should be set to the size of the response in bytes
+ *
+ * Available since API level 21.
*/
-
media_status_t AMediaDrm_provideKeyResponse(AMediaDrm *, const AMediaDrmScope *scope,
const uint8_t *response, size_t responseSize,
AMediaDrmKeySetId *keySetId) __INTRODUCED_IN(21);
@@ -305,8 +324,10 @@
* Restore persisted offline keys into a new session. keySetId identifies the
* keys to load, obtained from a prior call to AMediaDrm_provideKeyResponse.
*
- * sessionId is the session ID for the DRM session
- * keySetId identifies the saved key set to restore
+ * sessionId is the session ID for the DRM session.
+ * keySetId identifies the saved key set to restore.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaDrm_restoreKeys(AMediaDrm *, const AMediaDrmSessionId *sessionId,
const AMediaDrmKeySetId *keySetId) __INTRODUCED_IN(21);
@@ -314,7 +335,9 @@
/**
* Remove the current keys from a session.
*
- * keySetId identifies keys to remove
+ * keySetId identifies keys to remove.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaDrm_removeKeys(AMediaDrm *,
const AMediaDrmSessionId *keySetId) __INTRODUCED_IN(21);
@@ -331,6 +354,8 @@
* to the number of entries written to the array. If the number of {key, value} pairs
* to be returned is greater than *numPairs, MEDIADRM_SHORT_BUFFER will be returned
* and numPairs will be set to the number of pairs available.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaDrm_queryKeyStatus(AMediaDrm *, const AMediaDrmSessionId *sessionId,
AMediaDrmKeyValue *keyValuePairs, size_t *numPairs) __INTRODUCED_IN(21);
@@ -350,6 +375,8 @@
* 3. serverUrl will reference a NULL terminated string containing the URL
* the provisioning request should be sent to. It will remain accessible until
* the next call to getProvisionRequest.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaDrm_getProvisionRequest(AMediaDrm *, const uint8_t **provisionRequest,
size_t *provisionRequestSize, const char **serverUrl) __INTRODUCED_IN(21);
@@ -363,8 +390,10 @@
* DRM engine plugin.
* responseSize is the length of the provisioning response in bytes.
*
- * returns MEDIADRM_DEVICE_REVOKED_ERROR if the response indicates that the
+ * Returns MEDIADRM_DEVICE_REVOKED_ERROR if the response indicates that the
* server rejected the request
+ *
+ * Available since API level 21.
*/
media_status_t AMediaDrm_provideProvisionResponse(AMediaDrm *,
const uint8_t *response, size_t responseSize) __INTRODUCED_IN(21);
@@ -390,6 +419,8 @@
* If *numSecureStops is too small for the number of secure stops available,
* MEDIADRM_SHORT_BUFFER will be returned and *numSecureStops will be set to the
* number required.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaDrm_getSecureStops(AMediaDrm *,
AMediaDrmSecureStop *secureStops, size_t *numSecureStops) __INTRODUCED_IN(21);
@@ -399,6 +430,8 @@
* the message, remove the SecureStops identified in the response.
*
* ssRelease is the server response indicating which secure stops to release
+ *
+ * Available since API level 21.
*/
media_status_t AMediaDrm_releaseSecureStops(AMediaDrm *,
const AMediaDrmSecureStop *ssRelease) __INTRODUCED_IN(21);
@@ -432,6 +465,8 @@
* On return, propertyValue will be set to point to the property value. The
* memory that the value resides in is owned by the NDK MediaDrm API and
* will remain valid until the next call to AMediaDrm_getPropertyString.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaDrm_getPropertyString(AMediaDrm *, const char *propertyName,
const char **propertyValue) __INTRODUCED_IN(21);
@@ -447,18 +482,24 @@
* On return, *propertyValue will be set to point to the property value. The
* memory that the value resides in is owned by the NDK MediaDrm API and
* will remain valid until the next call to AMediaDrm_getPropertyByteArray.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaDrm_getPropertyByteArray(AMediaDrm *, const char *propertyName,
AMediaDrmByteArray *propertyValue) __INTRODUCED_IN(21);
/**
* Set a DRM engine plugin String property value.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaDrm_setPropertyString(AMediaDrm *, const char *propertyName,
const char *value) __INTRODUCED_IN(21);
/**
* Set a DRM engine plugin byte array property value.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaDrm_setPropertyByteArray(AMediaDrm *, const char *propertyName,
const uint8_t *value, size_t valueSize) __INTRODUCED_IN(21);
@@ -487,6 +528,8 @@
* ensure that the output buffer is large enough to accept dataSize bytes. The key
* to use is identified by the 16 byte keyId. The key must have been loaded into
* the session using provideKeyResponse.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaDrm_encrypt(AMediaDrm *, const AMediaDrmSessionId *sessionId,
const char *cipherAlgorithm, uint8_t *keyId, uint8_t *iv,
@@ -498,6 +541,8 @@
* ensure that the output buffer is large enough to accept dataSize bytes. The key
* to use is identified by the 16 byte keyId. The key must have been loaded into
* the session using provideKeyResponse.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaDrm_decrypt(AMediaDrm *, const AMediaDrmSessionId *sessionId,
const char *cipherAlgorithm, uint8_t *keyId, uint8_t *iv,
@@ -511,6 +556,8 @@
* *signatureSize is set to the buffer size required. The key to use is identified
* by the 16 byte keyId. The key must have been loaded into the session using
* provideKeyResponse.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaDrm_sign(AMediaDrm *, const AMediaDrmSessionId *sessionId,
const char *macAlgorithm, uint8_t *keyId, uint8_t *message, size_t messageSize,
@@ -522,6 +569,8 @@
* if the signature matches, otherwise MEDAIDRM_VERIFY_FAILED is returned. The key to
* use is identified by the 16 byte keyId. The key must have been loaded into the
* session using provideKeyResponse.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaDrm_verify(AMediaDrm *, const AMediaDrmSessionId *sessionId,
const char *macAlgorithm, uint8_t *keyId, const uint8_t *message, size_t messageSize,
diff --git a/media/ndk/include/media/NdkMediaExtractor.h b/media/ndk/include/media/NdkMediaExtractor.h
index e3d9fe6..14319c4 100644
--- a/media/ndk/include/media/NdkMediaExtractor.h
+++ b/media/ndk/include/media/NdkMediaExtractor.h
@@ -52,23 +52,31 @@
#if __ANDROID_API__ >= 21
/**
- * Create new media extractor
+ * Create new media extractor.
+ *
+ * Available since API level 21.
*/
AMediaExtractor* AMediaExtractor_new() __INTRODUCED_IN(21);
/**
- * Delete a previously created media extractor
+ * Delete a previously created media extractor.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaExtractor_delete(AMediaExtractor*) __INTRODUCED_IN(21);
/**
- * Set the file descriptor from which the extractor will read.
+ * Set the file descriptor from which the extractor will read.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaExtractor_setDataSourceFd(AMediaExtractor*, int fd, off64_t offset,
off64_t length) __INTRODUCED_IN(21);
/**
* Set the URI from which the extractor will read.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaExtractor_setDataSource(AMediaExtractor*,
const char *location) __INTRODUCED_IN(21);
@@ -77,6 +85,8 @@
/**
* Set the custom data source implementation from which the extractor will read.
+ *
+ * Available since API level 28.
*/
media_status_t AMediaExtractor_setDataSourceCustom(AMediaExtractor*,
AMediaDataSource *src) __INTRODUCED_IN(28);
@@ -85,11 +95,15 @@
/**
* Return the number of tracks in the previously specified media file
+ *
+ * Available since API level 21.
*/
size_t AMediaExtractor_getTrackCount(AMediaExtractor*) __INTRODUCED_IN(21);
/**
* Return the format of the specified track. The caller must free the returned format
+ *
+ * Available since API level 21.
*/
AMediaFormat* AMediaExtractor_getTrackFormat(AMediaExtractor*, size_t idx) __INTRODUCED_IN(21);
@@ -98,41 +112,55 @@
* getSampleTime only retrieve information for the subset of tracks selected.
* Selecting the same track multiple times has no effect, the track is
* only selected once.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaExtractor_selectTrack(AMediaExtractor*, size_t idx) __INTRODUCED_IN(21);
/**
* Unselect the specified track. Subsequent calls to readSampleData, getSampleTrackIndex and
- * getSampleTime only retrieve information for the subset of tracks selected..
+ * getSampleTime only retrieve information for the subset of tracks selected.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaExtractor_unselectTrack(AMediaExtractor*, size_t idx) __INTRODUCED_IN(21);
/**
* Read the current sample.
+ *
+ * Available since API level 21.
*/
ssize_t AMediaExtractor_readSampleData(AMediaExtractor*,
uint8_t *buffer, size_t capacity) __INTRODUCED_IN(21);
/**
* Read the current sample's flags.
+ *
+ * Available since API level 21.
*/
uint32_t AMediaExtractor_getSampleFlags(AMediaExtractor*) __INTRODUCED_IN(21);
/**
* Returns the track index the current sample originates from (or -1
* if no more samples are available)
+ *
+ * Available since API level 21.
*/
int AMediaExtractor_getSampleTrackIndex(AMediaExtractor*) __INTRODUCED_IN(21);
/**
* Returns the current sample's presentation time in microseconds.
* or -1 if no more samples are available.
+ *
+ * Available since API level 21.
*/
int64_t AMediaExtractor_getSampleTime(AMediaExtractor*) __INTRODUCED_IN(21);
/**
* Advance to the next sample. Returns false if no more sample data
* is available (end of stream).
+ *
+ * Available since API level 21.
*/
bool AMediaExtractor_advance(AMediaExtractor*) __INTRODUCED_IN(21);
@@ -143,7 +171,7 @@
} SeekMode;
/**
- *
+ * Available since API level 21.
*/
media_status_t AMediaExtractor_seekTo(AMediaExtractor*,
int64_t seekPosUs, SeekMode mode) __INTRODUCED_IN(21);
@@ -167,10 +195,14 @@
/**
* Get the PSSH info if present.
+ *
+ * Available since API level 21.
*/
PsshInfo* AMediaExtractor_getPsshInfo(AMediaExtractor*) __INTRODUCED_IN(21);
-
+/**
+ * Available since API level 21.
+ */
AMediaCodecCryptoInfo *AMediaExtractor_getSampleCryptoInfo(AMediaExtractor *) __INTRODUCED_IN(21);
enum {
@@ -186,6 +218,8 @@
*
* This function will always return a format; however, the format could be empty
* (no key-value pairs) if the media container does not provide format information.
+ *
+ * Available since API level 28.
*/
AMediaFormat* AMediaExtractor_getFileFormat(AMediaExtractor*) __INTRODUCED_IN(28);
@@ -198,6 +232,7 @@
* uint8_t *buf = new uint8_t[sampleSize];
* AMediaExtractor_readSampleData(ex, buf, sampleSize);
*
+ * Available since API level 28.
*/
ssize_t AMediaExtractor_getSampleSize(AMediaExtractor*) __INTRODUCED_IN(28);
@@ -211,6 +246,8 @@
* Returns -1 when the extractor is not reading from a network data source, or when the
* cached duration cannot be calculated (bitrate, duration, and file size information
* not available).
+ *
+ * Available since API level 28.
*/
int64_t AMediaExtractor_getCachedDuration(AMediaExtractor *) __INTRODUCED_IN(28);
@@ -222,6 +259,8 @@
* Returns AMEDIA_OK on success or AMEDIA_ERROR_* to indicate failure reason.
* Existing key-value pairs in |fmt| would be removed if this API returns AMEDIA_OK.
* The contents of |fmt| is undefined if this API returns AMEDIA_ERROR_*.
+ *
+ * Available since API level 28.
*/
media_status_t AMediaExtractor_getSampleFormat(AMediaExtractor *ex,
AMediaFormat *fmt) __INTRODUCED_IN(28);
diff --git a/media/ndk/include/media/NdkMediaFormat.h b/media/ndk/include/media/NdkMediaFormat.h
index fd43f36..41c2378 100644
--- a/media/ndk/include/media/NdkMediaFormat.h
+++ b/media/ndk/include/media/NdkMediaFormat.h
@@ -48,40 +48,78 @@
#if __ANDROID_API__ >= 21
+/**
+ * Available since API level 21.
+ */
AMediaFormat *AMediaFormat_new() __INTRODUCED_IN(21);
+
+/**
+ * Available since API level 21.
+ */
media_status_t AMediaFormat_delete(AMediaFormat*) __INTRODUCED_IN(21);
/**
* Human readable representation of the format. The returned string is owned by the format,
* and remains valid until the next call to toString, or until the format is deleted.
+ *
+ * Available since API level 21.
*/
const char* AMediaFormat_toString(AMediaFormat*) __INTRODUCED_IN(21);
+/**
+ * Available since API level 21.
+ */
bool AMediaFormat_getInt32(AMediaFormat*, const char *name, int32_t *out) __INTRODUCED_IN(21);
+/**
+ * Available since API level 21.
+ */
bool AMediaFormat_getInt64(AMediaFormat*, const char *name, int64_t *out) __INTRODUCED_IN(21);
+/**
+ * Available since API level 21.
+ */
bool AMediaFormat_getFloat(AMediaFormat*, const char *name, float *out) __INTRODUCED_IN(21);
+/**
+ * Available since API level 21.
+ */
bool AMediaFormat_getSize(AMediaFormat*, const char *name, size_t *out) __INTRODUCED_IN(21);
/**
* The returned data is owned by the format and remains valid as long as the named entry
* is part of the format.
+ *
+ * Available since API level 21.
*/
bool AMediaFormat_getBuffer(AMediaFormat*, const char *name, void** data, size_t *size) __INTRODUCED_IN(21);
/**
* The returned string is owned by the format, and remains valid until the next call to getString,
* or until the format is deleted.
+ *
+ * Available since API level 21.
*/
bool AMediaFormat_getString(AMediaFormat*, const char *name, const char **out) __INTRODUCED_IN(21);
+/**
+ * Available since API level 21.
+ */
void AMediaFormat_setInt32(AMediaFormat*, const char* name, int32_t value) __INTRODUCED_IN(21);
+/**
+ * Available since API level 21.
+ */
void AMediaFormat_setInt64(AMediaFormat*, const char* name, int64_t value) __INTRODUCED_IN(21);
+/**
+ * Available since API level 21.
+ */
void AMediaFormat_setFloat(AMediaFormat*, const char* name, float value) __INTRODUCED_IN(21);
/**
* The provided string is copied into the format.
+ *
+ * Available since API level 21.
*/
void AMediaFormat_setString(AMediaFormat*, const char* name, const char* value) __INTRODUCED_IN(21);
/**
* The provided data is copied into the format.
+ *
+ * Available since API level 21.
*/
void AMediaFormat_setBuffer(AMediaFormat*, const char* name, const void* data, size_t size) __INTRODUCED_IN(21);
@@ -155,24 +193,43 @@
#endif /* __ANDROID_API__ >= 21 */
#if __ANDROID_API__ >= 28
+/**
+ * Available since API level 28.
+ */
bool AMediaFormat_getDouble(AMediaFormat*, const char *name, double *out) __INTRODUCED_IN(28);
+/**
+ * Available since API level 28.
+ */
bool AMediaFormat_getRect(AMediaFormat*, const char *name,
int32_t *left, int32_t *top, int32_t *right, int32_t *bottom) __INTRODUCED_IN(28);
+/**
+ * Available since API level 28.
+ */
void AMediaFormat_setDouble(AMediaFormat*, const char* name, double value) __INTRODUCED_IN(28);
+/**
+ * Available since API level 28.
+ */
void AMediaFormat_setSize(AMediaFormat*, const char* name, size_t value) __INTRODUCED_IN(28);
+/**
+ * Available since API level 28.
+ */
void AMediaFormat_setRect(AMediaFormat*, const char* name,
int32_t left, int32_t top, int32_t right, int32_t bottom) __INTRODUCED_IN(28);
#endif /* __ANDROID_API__ >= 28 */
#if __ANDROID_API__ >= 29
/**
- * remove all key/value pairs from the given AMediaFormat
+ * Remove all key/value pairs from the given AMediaFormat.
+ *
+ * Available since API level 29.
*/
void AMediaFormat_clear(AMediaFormat*) __INTRODUCED_IN(29);
/**
- * copy one AMediaFormat to another
+ * Copy one AMediaFormat to another.
+ *
+ * Available since API level 29.
*/
media_status_t AMediaFormat_copy(AMediaFormat *to, AMediaFormat *from) __INTRODUCED_IN(29);
diff --git a/media/ndk/include/media/NdkMediaMuxer.h b/media/ndk/include/media/NdkMediaMuxer.h
index 7393867..3fdeea4 100644
--- a/media/ndk/include/media/NdkMediaMuxer.h
+++ b/media/ndk/include/media/NdkMediaMuxer.h
@@ -56,12 +56,16 @@
#if __ANDROID_API__ >= 21
/**
- * Create new media muxer
+ * Create new media muxer.
+ *
+ * Available since API level 21.
*/
AMediaMuxer* AMediaMuxer_new(int fd, OutputFormat format) __INTRODUCED_IN(21);
/**
- * Delete a previously created media muxer
+ * Delete a previously created media muxer.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaMuxer_delete(AMediaMuxer*) __INTRODUCED_IN(21);
@@ -75,6 +79,8 @@
* Both values are specified in degrees.
* Latitude must be in the range [-90, 90].
* Longitude must be in the range [-180, 180].
+ *
+ * Available since API level 21.
*/
media_status_t AMediaMuxer_setLocation(AMediaMuxer*,
float latitude, float longitude) __INTRODUCED_IN(21);
@@ -90,6 +96,8 @@
* during playback.
* The angle is specified in degrees, clockwise.
* The supported angles are 0, 90, 180, and 270 degrees.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaMuxer_setOrientationHint(AMediaMuxer*, int degrees) __INTRODUCED_IN(21);
@@ -97,18 +105,24 @@
* Adds a track with the specified format.
* Returns the index of the new track or a negative value in case of failure,
* which can be interpreted as a media_status_t.
+ *
+ * Available since API level 21.
*/
ssize_t AMediaMuxer_addTrack(AMediaMuxer*, const AMediaFormat* format) __INTRODUCED_IN(21);
/**
* Start the muxer. Should be called after AMediaMuxer_addTrack and
* before AMediaMuxer_writeSampleData.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaMuxer_start(AMediaMuxer*) __INTRODUCED_IN(21);
/**
* Stops the muxer.
* Once the muxer stops, it can not be restarted.
+ *
+ * Available since API level 21.
*/
media_status_t AMediaMuxer_stop(AMediaMuxer*) __INTRODUCED_IN(21);
@@ -118,6 +132,8 @@
* the right tracks. Also, it needs to make sure the samples for each track
* are written in chronological order (e.g. in the order they are provided
* by the encoder.)
+ *
+ * Available since API level 21.
*/
media_status_t AMediaMuxer_writeSampleData(AMediaMuxer *muxer,
size_t trackIdx, const uint8_t *data,
diff --git a/media/ndk/libmediandk.map.txt b/media/ndk/libmediandk.map.txt
index f666ad0..7531578 100644
--- a/media/ndk/libmediandk.map.txt
+++ b/media/ndk/libmediandk.map.txt
@@ -4,7 +4,7 @@
AImageReader_acquireLatestImageAsync; # introduced=26
AImageReader_acquireNextImage; # introduced=24
AImageReader_acquireNextImageAsync; # introduced=26
- AImageReader_getWindowNativeHandle; #vndk
+ AImageReader_getWindowNativeHandle; # llndk
AImageReader_delete; # introduced=24
AImageReader_getFormat; # introduced=24
AImageReader_getHeight; # introduced=24
diff --git a/media/tests/benchmark/.clang-format b/media/tests/benchmark/.clang-format
new file mode 100644
index 0000000..bf1e355
--- /dev/null
+++ b/media/tests/benchmark/.clang-format
@@ -0,0 +1,13 @@
+BasedOnStyle: Google
+Standard: Cpp11
+AccessModifierOffset: -2
+AllowShortFunctionsOnASingleLine: Inline
+ColumnLimit: 100
+CommentPragmas: NOLINT:.*
+DerivePointerAlignment: false
+IncludeBlocks: Preserve
+IndentWidth: 4
+ContinuationIndentWidth: 8
+PointerAlignment: Right
+TabWidth: 4
+UseTab: Never
diff --git a/media/libstagefright/include/media/stagefright/NdkUtils.h b/media/tests/benchmark/Android.bp
similarity index 62%
copy from media/libstagefright/include/media/stagefright/NdkUtils.h
copy to media/tests/benchmark/Android.bp
index a68884a..de408dd 100644
--- a/media/libstagefright/include/media/stagefright/NdkUtils.h
+++ b/media/tests/benchmark/Android.bp
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2018 The Android Open Source Project
+ * Copyright (C) 2019 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
@@ -14,18 +14,8 @@
* limitations under the License.
*/
-#ifndef NDK_UTILS_H_
-
-#define NDK_UTILS_H_
-
-#include <media/stagefright/MetaData.h>
-#include <media/NdkWrapper.h>
-
-namespace android {
-
-sp<MetaData> convertMediaFormatWrapperToMetaData(
- const sp<AMediaFormatWrapper> &fmt);
-
-} // namespace android
-
-#endif // NDK_UTILS_H_
+subdirs = [
+ "src",
+ "tests",
+ "MediaBenchmarkTest",
+]
diff --git a/media/tests/benchmark/MediaBenchmarkTest/Android.bp b/media/tests/benchmark/MediaBenchmarkTest/Android.bp
new file mode 100644
index 0000000..d80d9a5
--- /dev/null
+++ b/media/tests/benchmark/MediaBenchmarkTest/Android.bp
@@ -0,0 +1,66 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+android_test {
+ name: "MediaBenchmarkTest",
+
+ defaults: [
+ "MediaBenchmark-defaults",
+ ],
+
+ // Include all the test code
+ srcs: ["src/androidTest/**/*.java"],
+
+ resource_dirs: ["res"],
+
+ libs: [
+ "android.test.runner",
+ "android.test.base",
+ ],
+
+ jni_libs: [
+ "libmediabenchmark_jni",
+ ],
+
+ static_libs: [
+ "libMediaBenchmark",
+ "junit",
+ "androidx.test.runner",
+ ],
+}
+
+android_library {
+ name: "libMediaBenchmark",
+
+ defaults: [
+ "MediaBenchmark-defaults",
+ ],
+
+ // Include all the libraries
+ srcs: ["src/main/**/*.java"],
+
+ static_libs: [
+ "androidx.test.core",
+ ],
+}
+
+java_defaults {
+ name: "MediaBenchmark-defaults",
+
+ sdk_version: "system_current",
+ min_sdk_version: "28",
+ target_sdk_version: "29",
+}
diff --git a/media/tests/benchmark/MediaBenchmarkTest/AndroidManifest.xml b/media/tests/benchmark/MediaBenchmarkTest/AndroidManifest.xml
new file mode 100644
index 0000000..eea9914
--- /dev/null
+++ b/media/tests/benchmark/MediaBenchmarkTest/AndroidManifest.xml
@@ -0,0 +1,34 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!--
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+-->
+
+<manifest xmlns:android="http://schemas.android.com/apk/res/android"
+ xmlns:tools="http://schemas.android.com/tools"
+ package="com.android.media.benchmark">
+ <uses-permission android:name="android.permission.READ_EXTERNAL_STORAGE" />
+ <uses-permission android:name="android.permission.WRITE_EXTERNAL_STORAGE" />
+ <uses-permission android:name="android.permission.READ_INTERNAL_STORAGE" />
+ <uses-permission android:name="android.permission.WRITE_INTERNAL_STORAGE" />
+
+ <application
+ tools:ignore="AllowBackup,GoogleAppIndexingWarning,MissingApplicationIcon"
+ tools:remove="android:appComponentFactory">
+ </application>
+
+ <instrumentation android:name="androidx.test.runner.AndroidJUnitRunner"
+ android:targetPackage="com.android.media.benchmark"
+ android:label="Benchmark Media Test"/>
+</manifest>
\ No newline at end of file
diff --git a/media/tests/benchmark/MediaBenchmarkTest/AndroidTest.xml b/media/tests/benchmark/MediaBenchmarkTest/AndroidTest.xml
new file mode 100644
index 0000000..1890661
--- /dev/null
+++ b/media/tests/benchmark/MediaBenchmarkTest/AndroidTest.xml
@@ -0,0 +1,34 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!-- Copyright (C) 2018 The Android Open Source Project
+
+ Licensed under the Apache License, Version 2.0 (the "License");
+ you may not use this file except in compliance with the License.
+ You may obtain a copy of the License at
+
+ http://www.apache.org/licenses/LICENSE-2.0
+
+ Unless required by applicable law or agreed to in writing, software
+ distributed under the License is distributed on an "AS IS" BASIS,
+ WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ See the License for the specific language governing permissions and
+ limitations under the License.
+-->
+<configuration description="Runs Media Benchmark Tests">
+ <target_preparer class="com.android.tradefed.targetprep.PushFilePreparer">
+ <option name="cleanup" value="true" />
+ <option name="push-file"
+ key="https://storage.googleapis.com/android_media/frameworks/av/media/tests/benchmark/MediaBenchmark.zip?unzip=true"
+ value="/data/local/tmp/MediaBenchmark/res/" />
+ </target_preparer>
+ <target_preparer class="com.android.tradefed.targetprep.TestAppInstallSetup">
+ <option name="cleanup-apks" value="false" />
+ <option name="test-file-name" value="MediaBenchmarkTest.apk" />
+ </target_preparer>
+
+ <option name="test-tag" value="MediaBenchmarkTest" />
+ <test class="com.android.tradefed.testtype.AndroidJUnitTest" >
+ <option name="package" value="com.android.media.benchmark" />
+ <option name="runner" value="androidx.test.runner.AndroidJUnitRunner" />
+ <option name="hidden-api-checks" value="false"/>
+ </test>
+</configuration>
diff --git a/media/tests/benchmark/MediaBenchmarkTest/build.gradle b/media/tests/benchmark/MediaBenchmarkTest/build.gradle
new file mode 100644
index 0000000..b2aee1a
--- /dev/null
+++ b/media/tests/benchmark/MediaBenchmarkTest/build.gradle
@@ -0,0 +1,76 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+buildscript {
+ repositories {
+ google()
+ jcenter()
+ }
+ dependencies {
+ classpath 'com.android.tools.build:gradle:3.5.0'
+ }
+}
+
+apply plugin: 'com.android.application'
+
+android {
+ compileSdkVersion 29
+ defaultConfig {
+ applicationId "com.android.media.benchmark"
+ minSdkVersion 28
+ targetSdkVersion 29
+ versionCode 1
+ versionName "1.0"
+ testInstrumentationRunner "androidx.test.runner.AndroidJUnitRunner"
+ }
+ sourceSets {
+ main {
+ java.srcDirs 'src/main/java'
+ res.srcDirs 'res'
+ manifest.srcFile 'AndroidManifest.xml'
+ }
+ androidTest {
+ java.srcDirs 'src/androidTest/java'
+ res.srcDirs 'res'
+ manifest.srcFile 'AndroidManifest.xml'
+ }
+ }
+ buildTypes {
+ release {
+ minifyEnabled false
+ proguardFiles getDefaultProguardFile('proguard-android.txt'), 'proguard-rules.pro'
+ }
+ }
+ externalNativeBuild {
+ cmake {
+ path "src/main/cpp/CMakeLists.txt"
+ version "3.10.2"
+ }
+ }
+}
+
+repositories {
+ google()
+ jcenter()
+}
+
+dependencies {
+ implementation fileTree(dir: 'libs', include: ['*.jar'])
+ implementation 'androidx.appcompat:appcompat:1.1.0'
+ testImplementation 'junit:junit:4.12'
+ androidTestImplementation 'androidx.test:runner:1.2.0'
+ androidTestImplementation 'androidx.test.ext:junit:1.1.1'
+}
\ No newline at end of file
diff --git a/media/tests/benchmark/MediaBenchmarkTest/res/values/strings.xml b/media/tests/benchmark/MediaBenchmarkTest/res/values/strings.xml
new file mode 100644
index 0000000..24dbccc
--- /dev/null
+++ b/media/tests/benchmark/MediaBenchmarkTest/res/values/strings.xml
@@ -0,0 +1,4 @@
+<resources>
+ <string name="input_file_path">/data/local/tmp/MediaBenchmark/res/</string>
+ <string name="output_file_path">/data/local/tmp/MediaBenchmark/output/</string>
+</resources>
diff --git a/media/tests/benchmark/MediaBenchmarkTest/src/androidTest/java/com/android/media/benchmark/tests/DecoderTest.java b/media/tests/benchmark/MediaBenchmarkTest/src/androidTest/java/com/android/media/benchmark/tests/DecoderTest.java
new file mode 100644
index 0000000..afd70a3
--- /dev/null
+++ b/media/tests/benchmark/MediaBenchmarkTest/src/androidTest/java/com/android/media/benchmark/tests/DecoderTest.java
@@ -0,0 +1,220 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package com.android.media.benchmark.tests;
+
+import android.content.Context;
+import android.media.MediaCodec;
+import android.media.MediaFormat;
+import android.util.Log;
+
+import androidx.test.platform.app.InstrumentationRegistry;
+
+import com.android.media.benchmark.R;
+import com.android.media.benchmark.library.CodecUtils;
+import com.android.media.benchmark.library.Decoder;
+import com.android.media.benchmark.library.Extractor;
+import com.android.media.benchmark.library.Native;
+import com.android.media.benchmark.library.Stats;
+
+import org.junit.BeforeClass;
+import org.junit.Test;
+import org.junit.runner.RunWith;
+import org.junit.runners.Parameterized;
+
+import java.io.File;
+import java.io.FileDescriptor;
+import java.io.FileInputStream;
+import java.io.FileOutputStream;
+import java.io.IOException;
+import java.nio.ByteBuffer;
+import java.nio.file.Files;
+import java.nio.file.Paths;
+import java.util.ArrayList;
+import java.util.Arrays;
+import java.util.Collection;
+
+import static org.junit.Assert.assertTrue;
+import static org.junit.Assert.assertEquals;
+
+@RunWith(Parameterized.class)
+public class DecoderTest {
+ private static final Context mContext =
+ InstrumentationRegistry.getInstrumentation().getTargetContext();
+ private static final String mInputFilePath = mContext.getString(R.string.input_file_path);
+ private static final String mOutputFilePath = mContext.getString(R.string.output_file_path);
+ private static final String mStatsFile =
+ mContext.getExternalFilesDir(null) + "/Decoder." + System.currentTimeMillis() + ".csv";
+ private static final String TAG = "DecoderTest";
+ private static final long PER_TEST_TIMEOUT_MS = 60000;
+ private static final boolean DEBUG = false;
+ private static final boolean WRITE_OUTPUT = false;
+ private String mInputFile;
+ private boolean mAsyncMode;
+
+ public DecoderTest(String inputFile, boolean asyncMode) {
+ this.mInputFile = inputFile;
+ this.mAsyncMode = asyncMode;
+ }
+
+ @Parameterized.Parameters
+ public static Collection<Object[]> input() {
+ return Arrays.asList(new Object[][]{
+ //Audio Sync Test
+ {"bbb_44100hz_2ch_128kbps_aac_30sec.mp4", false},
+ {"bbb_44100hz_2ch_128kbps_mp3_30sec.mp3", false},
+ {"bbb_8000hz_1ch_8kbps_amrnb_30sec.3gp", false},
+ {"bbb_16000hz_1ch_9kbps_amrwb_30sec.3gp", false},
+ {"bbb_44100hz_2ch_80kbps_vorbis_30sec.webm", false},
+ {"bbb_44100hz_2ch_600kbps_flac_30sec.mp4", false},
+ {"bbb_48000hz_2ch_100kbps_opus_30sec.webm", false},
+ // Audio Async Test
+ {"bbb_44100hz_2ch_128kbps_aac_30sec.mp4", true},
+ {"bbb_44100hz_2ch_128kbps_mp3_30sec.mp3", true},
+ {"bbb_8000hz_1ch_8kbps_amrnb_30sec.3gp", true},
+ {"bbb_16000hz_1ch_9kbps_amrwb_30sec.3gp", true},
+ {"bbb_44100hz_2ch_80kbps_vorbis_30sec.webm", true},
+ {"bbb_44100hz_2ch_600kbps_flac_30sec.mp4", true},
+ {"bbb_48000hz_2ch_100kbps_opus_30sec.webm", true},
+ // Video Sync Test
+ {"crowd_1920x1080_25fps_4000kbps_vp9.webm", false},
+ {"crowd_1920x1080_25fps_4000kbps_vp8.webm", false},
+ {"crowd_1920x1080_25fps_4000kbps_av1.webm", false},
+ {"crowd_1920x1080_25fps_7300kbps_mpeg2.mp4", false},
+ {"crowd_1920x1080_25fps_6000kbps_mpeg4.mp4", false},
+ {"crowd_352x288_25fps_6000kbps_h263.3gp", false},
+ {"crowd_1920x1080_25fps_6700kbps_h264.ts", false},
+ {"crowd_1920x1080_25fps_4000kbps_h265.mkv", false},
+ // Video Async Test
+ {"crowd_1920x1080_25fps_4000kbps_vp9.webm", true},
+ {"crowd_1920x1080_25fps_4000kbps_vp8.webm", true},
+ {"crowd_1920x1080_25fps_4000kbps_av1.webm", true},
+ {"crowd_1920x1080_25fps_7300kbps_mpeg2.mp4", true},
+ {"crowd_1920x1080_25fps_6000kbps_mpeg4.mp4", true},
+ {"crowd_352x288_25fps_6000kbps_h263.3gp", true},
+ {"crowd_1920x1080_25fps_6700kbps_h264.ts", true},
+ {"crowd_1920x1080_25fps_4000kbps_h265.mkv", true}});
+ }
+
+ @BeforeClass
+ public static void writeStatsHeaderToFile() throws IOException {
+ Stats mStats = new Stats();
+ boolean status = mStats.writeStatsHeader(mStatsFile);
+ assertTrue("Unable to open stats file for writing!", status);
+ Log.d(TAG, "Saving Benchmark results in: " + mStatsFile);
+ }
+
+ @Test(timeout = PER_TEST_TIMEOUT_MS)
+ public void testDecoder() throws IOException {
+ File inputFile = new File(mInputFilePath + mInputFile);
+ assertTrue("Cannot find " + mInputFile + " in directory " + mInputFilePath,
+ inputFile.exists());
+ FileInputStream fileInput = new FileInputStream(inputFile);
+ FileDescriptor fileDescriptor = fileInput.getFD();
+ Extractor extractor = new Extractor();
+ int trackCount = extractor.setUpExtractor(fileDescriptor);
+ assertTrue("Extraction failed. No tracks for file: " + mInputFile, (trackCount > 0));
+ ArrayList<ByteBuffer> inputBuffer = new ArrayList<>();
+ ArrayList<MediaCodec.BufferInfo> frameInfo = new ArrayList<>();
+ for (int currentTrack = 0; currentTrack < trackCount; currentTrack++) {
+ extractor.selectExtractorTrack(currentTrack);
+ MediaFormat format = extractor.getFormat(currentTrack);
+ String mime = format.getString(MediaFormat.KEY_MIME);
+ ArrayList<String> mediaCodecs = CodecUtils.selectCodecs(mime, false);
+ assertTrue("No suitable codecs found for file: " + mInputFile + " track : " +
+ currentTrack + " mime: " + mime, (mediaCodecs.size() > 0));
+
+ // Get samples from extractor
+ int sampleSize;
+ do {
+ sampleSize = extractor.getFrameSample();
+ MediaCodec.BufferInfo bufInfo = new MediaCodec.BufferInfo();
+ MediaCodec.BufferInfo info = extractor.getBufferInfo();
+ ByteBuffer dataBuffer = ByteBuffer.allocate(info.size);
+ dataBuffer.put(extractor.getFrameBuffer().array(), 0, info.size);
+ bufInfo.set(info.offset, info.size, info.presentationTimeUs, info.flags);
+ inputBuffer.add(dataBuffer);
+ frameInfo.add(bufInfo);
+ if (DEBUG) {
+ Log.d(TAG, "Extracted bufInfo: flag = " + bufInfo.flags + " timestamp = " +
+ bufInfo.presentationTimeUs + " size = " + bufInfo.size);
+ }
+ } while (sampleSize > 0);
+ for (String codecName : mediaCodecs) {
+ FileOutputStream decodeOutputStream = null;
+ if (WRITE_OUTPUT) {
+ if (!Paths.get(mOutputFilePath).toFile().exists()) {
+ Files.createDirectories(Paths.get(mOutputFilePath));
+ }
+ File outFile = new File(mOutputFilePath + "decoder.out");
+ if (outFile.exists()) {
+ assertTrue(" Unable to delete existing file" + outFile.toString(),
+ outFile.delete());
+ }
+ assertTrue("Unable to create file: " + outFile.toString(),
+ outFile.createNewFile());
+ decodeOutputStream = new FileOutputStream(outFile);
+ }
+ Decoder decoder = new Decoder();
+ decoder.setupDecoder(decodeOutputStream);
+ int status = decoder.decode(inputBuffer, frameInfo, mAsyncMode, format, codecName);
+ decoder.deInitCodec();
+ assertEquals("Decoder returned error " + status + " for file: " + mInputFile +
+ " with codec: " + codecName, 0, status);
+ decoder.dumpStatistics(mInputFile, codecName, (mAsyncMode ? "async" : "sync"),
+ extractor.getClipDuration(), mStatsFile);
+ Log.i(TAG, "Decoding Successful for file: " + mInputFile + " with codec: " +
+ codecName);
+ decoder.resetDecoder();
+ if (decodeOutputStream != null) {
+ decodeOutputStream.close();
+ }
+ }
+ extractor.unselectExtractorTrack(currentTrack);
+ inputBuffer.clear();
+ frameInfo.clear();
+ }
+ extractor.deinitExtractor();
+ fileInput.close();
+ }
+
+ @Test
+ public void testNativeDecoder() throws IOException {
+ File inputFile = new File(mInputFilePath + mInputFile);
+ assertTrue("Cannot find " + mInputFile + " in directory " + mInputFilePath,
+ inputFile.exists());
+ FileInputStream fileInput = new FileInputStream(inputFile);
+ FileDescriptor fileDescriptor = fileInput.getFD();
+ Extractor extractor = new Extractor();
+ int trackCount = extractor.setUpExtractor(fileDescriptor);
+ assertTrue("Extraction failed. No tracks for file: ", trackCount > 0);
+ for (int currentTrack = 0; currentTrack < trackCount; currentTrack++) {
+ extractor.selectExtractorTrack(currentTrack);
+ MediaFormat format = extractor.getFormat(currentTrack);
+ String mime = format.getString(MediaFormat.KEY_MIME);
+ ArrayList<String> mediaCodecs = CodecUtils.selectCodecs(mime, false);
+ for (String codecName : mediaCodecs) {
+ Log.i("Test: %s\n", mInputFile);
+ Native nativeDecoder = new Native();
+ int status = nativeDecoder.Decode(
+ mInputFilePath, mInputFile, mStatsFile, codecName, mAsyncMode);
+ assertEquals("Decoder returned error " + status + " for file: " + mInputFile, 0,
+ status);
+ }
+ }
+ fileInput.close();
+ }
+}
diff --git a/media/tests/benchmark/MediaBenchmarkTest/src/androidTest/java/com/android/media/benchmark/tests/EncoderTest.java b/media/tests/benchmark/MediaBenchmarkTest/src/androidTest/java/com/android/media/benchmark/tests/EncoderTest.java
new file mode 100644
index 0000000..48e1422
--- /dev/null
+++ b/media/tests/benchmark/MediaBenchmarkTest/src/androidTest/java/com/android/media/benchmark/tests/EncoderTest.java
@@ -0,0 +1,308 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package com.android.media.benchmark.tests;
+
+import android.content.Context;
+import android.media.MediaCodec;
+import android.media.MediaFormat;
+
+import static android.media.MediaCodecInfo.CodecCapabilities.COLOR_FormatYUV420Flexible;
+
+import android.util.Log;
+
+import androidx.test.platform.app.InstrumentationRegistry;
+
+import com.android.media.benchmark.R;
+import com.android.media.benchmark.library.CodecUtils;
+import com.android.media.benchmark.library.Decoder;
+import com.android.media.benchmark.library.Encoder;
+import com.android.media.benchmark.library.Extractor;
+import com.android.media.benchmark.library.Native;
+import com.android.media.benchmark.library.Stats;
+
+import org.junit.BeforeClass;
+import org.junit.Test;
+import org.junit.runner.RunWith;
+import org.junit.runners.Parameterized;
+
+import java.io.File;
+import java.io.FileDescriptor;
+import java.io.FileInputStream;
+import java.io.FileOutputStream;
+import java.io.IOException;
+import java.nio.ByteBuffer;
+
+import java.util.ArrayList;
+import java.util.Arrays;
+import java.util.Collection;
+
+import static org.junit.Assert.assertTrue;
+import static org.junit.Assert.assertEquals;
+
+@RunWith(Parameterized.class)
+public class EncoderTest {
+ private static final Context mContext =
+ InstrumentationRegistry.getInstrumentation().getTargetContext();
+ private static final String mInputFilePath = mContext.getString(R.string.input_file_path);
+ private static final String mOutputFilePath = mContext.getString(R.string.output_file_path);
+ private static final String mStatsFile =
+ mContext.getExternalFilesDir(null) + "/Encoder." + System.currentTimeMillis() + ".csv";
+ private static final String TAG = "EncoderTest";
+ private static final long PER_TEST_TIMEOUT_MS = 120000;
+ private static final boolean DEBUG = false;
+ private static final boolean WRITE_OUTPUT = false;
+ private static final int ENCODE_DEFAULT_FRAME_RATE = 25;
+ private static final int ENCODE_DEFAULT_BIT_RATE = 8000000 /* 8 Mbps */;
+ private static final int ENCODE_MIN_BIT_RATE = 600000 /* 600 Kbps */;
+ private static final int ENCODE_DEFAULT_AUDIO_BIT_RATE = 128000 /* 128 Kbps */;
+ private String mInputFile;
+
+ @Parameterized.Parameters
+ public static Collection<Object[]> inputFiles() {
+ return Arrays.asList(new Object[][]{
+ // Audio Test
+ {"bbb_44100hz_2ch_128kbps_aac_30sec.mp4"},
+ {"bbb_8000hz_1ch_8kbps_amrnb_30sec.3gp"},
+ {"bbb_16000hz_1ch_9kbps_amrwb_30sec.3gp"},
+ {"bbb_44100hz_2ch_600kbps_flac_30sec.mp4"},
+ {"bbb_48000hz_2ch_100kbps_opus_30sec.webm"},
+ // Video Test
+ {"crowd_1920x1080_25fps_4000kbps_vp8.webm"},
+ {"crowd_1920x1080_25fps_6700kbps_h264.ts"},
+ {"crowd_1920x1080_25fps_4000kbps_h265.mkv"},
+ {"crowd_1920x1080_25fps_4000kbps_vp9.webm"},
+ {"crowd_176x144_25fps_6000kbps_mpeg4.mp4"},
+ {"crowd_176x144_25fps_6000kbps_h263.3gp"}});
+ }
+
+ public EncoderTest(String inputFileName) {
+ this.mInputFile = inputFileName;
+ }
+
+ @BeforeClass
+ public static void writeStatsHeaderToFile() throws IOException {
+ Stats mStats = new Stats();
+ boolean status = mStats.writeStatsHeader(mStatsFile);
+ assertTrue("Unable to open stats file for writing!", status);
+ Log.d(TAG, "Saving Benchmark results in: " + mStatsFile);
+ }
+
+ @Test(timeout = PER_TEST_TIMEOUT_MS)
+ public void testEncoder() throws Exception {
+ int status;
+ int frameSize;
+ //Parameters for video
+ int width = 0;
+ int height = 0;
+ int profile = 0;
+ int level = 0;
+ int frameRate = 0;
+
+ //Parameters for audio
+ int bitRate = 0;
+ int sampleRate = 0;
+ int numChannels = 0;
+ File inputFile = new File(mInputFilePath + mInputFile);
+ assertTrue("Cannot find " + mInputFile + " in directory " + mInputFilePath,
+ inputFile.exists());
+ FileInputStream fileInput = new FileInputStream(inputFile);
+ FileDescriptor fileDescriptor = fileInput.getFD();
+ Extractor extractor = new Extractor();
+ int trackCount = extractor.setUpExtractor(fileDescriptor);
+ assertTrue("Extraction failed. No tracks for file: " + mInputFile, (trackCount > 0));
+ ArrayList<ByteBuffer> inputBuffer = new ArrayList<>();
+ ArrayList<MediaCodec.BufferInfo> frameInfo = new ArrayList<>();
+ for (int currentTrack = 0; currentTrack < trackCount; currentTrack++) {
+ int colorFormat = COLOR_FormatYUV420Flexible;
+ extractor.selectExtractorTrack(currentTrack);
+ MediaFormat format = extractor.getFormat(currentTrack);
+ // Get samples from extractor
+ int sampleSize;
+ do {
+ sampleSize = extractor.getFrameSample();
+ MediaCodec.BufferInfo bufInfo = new MediaCodec.BufferInfo();
+ MediaCodec.BufferInfo info = extractor.getBufferInfo();
+ ByteBuffer dataBuffer = ByteBuffer.allocate(info.size);
+ dataBuffer.put(extractor.getFrameBuffer().array(), 0, info.size);
+ bufInfo.set(info.offset, info.size, info.presentationTimeUs, info.flags);
+ inputBuffer.add(dataBuffer);
+ frameInfo.add(bufInfo);
+ if (DEBUG) {
+ Log.d(TAG, "Extracted bufInfo: flag = " + bufInfo.flags + " timestamp = " +
+ bufInfo.presentationTimeUs + " size = " + bufInfo.size);
+ }
+ } while (sampleSize > 0);
+ int tid = android.os.Process.myTid();
+ File decodedFile = new File(mContext.getFilesDir() + "/decoder_" + tid + ".out");
+ FileOutputStream decodeOutputStream = new FileOutputStream(decodedFile);
+ Decoder decoder = new Decoder();
+ decoder.setupDecoder(decodeOutputStream);
+ status = decoder.decode(inputBuffer, frameInfo, false, format, "");
+ assertEquals("Decoder returned error " + status + " for file: " + mInputFile, 0,
+ status);
+ MediaFormat decoderFormat = decoder.getFormat();
+ decoder.deInitCodec();
+ extractor.unselectExtractorTrack(currentTrack);
+ inputBuffer.clear();
+ frameInfo.clear();
+ if (decodeOutputStream != null) {
+ decodeOutputStream.close();
+ }
+ String mime = format.getString(MediaFormat.KEY_MIME);
+ ArrayList<String> mediaCodecs = CodecUtils.selectCodecs(mime, true);
+ assertTrue("No suitable codecs found for file: " + mInputFile + " track : " +
+ currentTrack + " mime: " + mime, (mediaCodecs.size() > 0));
+ Boolean[] encodeMode = {true, false};
+ /* Encoding the decoder's output */
+ for (Boolean asyncMode : encodeMode) {
+ for (String codecName : mediaCodecs) {
+ FileOutputStream encodeOutputStream = null;
+ if (WRITE_OUTPUT) {
+ File outEncodeFile = new File(mOutputFilePath + "encoder.out");
+ if (outEncodeFile.exists()) {
+ assertTrue(" Unable to delete existing file" + outEncodeFile.toString(),
+ outEncodeFile.delete());
+ }
+ assertTrue("Unable to create file to write encoder output: " +
+ outEncodeFile.toString(), outEncodeFile.createNewFile());
+ encodeOutputStream = new FileOutputStream(outEncodeFile);
+ }
+ File rawFile = new File(mContext.getFilesDir() + "/decoder_" + tid + ".out");
+ assertTrue("Cannot open file to write decoded output", rawFile.exists());
+ if (DEBUG) {
+ Log.i(TAG, "Path of decoded input file: " + rawFile.toString());
+ }
+ FileInputStream eleStream = new FileInputStream(rawFile);
+ if (mime.startsWith("video/")) {
+ width = format.getInteger(MediaFormat.KEY_WIDTH);
+ height = format.getInteger(MediaFormat.KEY_HEIGHT);
+ if (format.containsKey(MediaFormat.KEY_FRAME_RATE)) {
+ frameRate = format.getInteger(MediaFormat.KEY_FRAME_RATE);
+ } else if (frameRate <= 0) {
+ frameRate = ENCODE_DEFAULT_FRAME_RATE;
+ }
+ if (format.containsKey(MediaFormat.KEY_BIT_RATE)) {
+ bitRate = format.getInteger(MediaFormat.KEY_BIT_RATE);
+ } else if (bitRate <= 0) {
+ if (mime.contains("video/3gpp") || mime.contains("video/mp4v-es")) {
+ bitRate = ENCODE_MIN_BIT_RATE;
+ } else {
+ bitRate = ENCODE_DEFAULT_BIT_RATE;
+ }
+ }
+ if (format.containsKey(MediaFormat.KEY_PROFILE)) {
+ profile = format.getInteger(MediaFormat.KEY_PROFILE);
+ }
+ if (format.containsKey(MediaFormat.KEY_PROFILE)) {
+ level = format.getInteger(MediaFormat.KEY_LEVEL);
+ }
+ if (decoderFormat.containsKey(MediaFormat.KEY_COLOR_FORMAT)) {
+ colorFormat = decoderFormat.getInteger(MediaFormat.KEY_COLOR_FORMAT);
+ }
+ } else {
+ sampleRate = format.getInteger(MediaFormat.KEY_SAMPLE_RATE);
+ numChannels = format.getInteger(MediaFormat.KEY_CHANNEL_COUNT);
+ if (decoderFormat.containsKey(MediaFormat.KEY_BIT_RATE)) {
+ bitRate = decoderFormat.getInteger(MediaFormat.KEY_BIT_RATE);
+ } else {
+ bitRate = ENCODE_DEFAULT_AUDIO_BIT_RATE;
+ }
+ }
+ /*Setup Encode Format*/
+ MediaFormat encodeFormat;
+ if (mime.startsWith("video/")) {
+ frameSize = width * height * 3 / 2;
+ encodeFormat = MediaFormat.createVideoFormat(mime, width, height);
+ encodeFormat.setInteger(MediaFormat.KEY_FRAME_RATE, frameRate);
+ encodeFormat.setInteger(MediaFormat.KEY_BIT_RATE, bitRate);
+ encodeFormat.setInteger(MediaFormat.KEY_PROFILE, profile);
+ encodeFormat.setInteger(MediaFormat.KEY_LEVEL, level);
+ encodeFormat.setInteger(MediaFormat.KEY_I_FRAME_INTERVAL, 1);
+ encodeFormat.setInteger(MediaFormat.KEY_MAX_INPUT_SIZE, frameSize);
+ encodeFormat.setInteger(MediaFormat.KEY_COLOR_FORMAT, colorFormat);
+ } else {
+ encodeFormat = MediaFormat.createAudioFormat(mime, sampleRate, numChannels);
+ encodeFormat.setInteger(MediaFormat.KEY_BIT_RATE, bitRate);
+ frameSize = 4096;
+ }
+ Encoder encoder = new Encoder();
+ encoder.setupEncoder(encodeOutputStream, eleStream);
+ status = encoder.encode(codecName, encodeFormat, mime, frameRate, sampleRate,
+ frameSize, asyncMode);
+ encoder.deInitEncoder();
+ assertEquals(
+ codecName + " encoder returned error " + status + " for " + "file:" +
+ " " + mInputFile, 0, status);
+ encoder.dumpStatistics(mInputFile, codecName, (asyncMode ? "async" : "sync"),
+ extractor.getClipDuration(), mStatsFile);
+ Log.i(TAG, "Encoding complete for file: " + mInputFile + " with codec: " +
+ codecName + " for aSyncMode = " + asyncMode);
+ encoder.resetEncoder();
+ eleStream.close();
+ if (encodeOutputStream != null) {
+ encodeOutputStream.close();
+ }
+
+ }
+ }
+ //Cleanup temporary input file
+ if (decodedFile.exists()) {
+ assertTrue(" Unable to delete decoded file" + decodedFile.toString(),
+ decodedFile.delete());
+ Log.i(TAG, "Successfully deleted decoded file");
+ }
+ }
+ extractor.deinitExtractor();
+ fileInput.close();
+ }
+
+ @Test(timeout = PER_TEST_TIMEOUT_MS)
+ public void testNativeEncoder() throws Exception {
+ File inputFile = new File(mInputFilePath + mInputFile);
+ assertTrue("Cannot find " + mInputFile + " in directory " + mInputFilePath,
+ inputFile.exists());
+ int tid = android.os.Process.myTid();
+ final String mDecodedFile = mContext.getFilesDir() + "/decoder_" + tid + ".out";
+ FileInputStream fileInput = new FileInputStream(inputFile);
+ FileDescriptor fileDescriptor = fileInput.getFD();
+ Extractor extractor = new Extractor();
+ int trackCount = extractor.setUpExtractor(fileDescriptor);
+ assertTrue("Extraction failed. No tracks for file: ", trackCount > 0);
+ for (int currentTrack = 0; currentTrack < trackCount; currentTrack++) {
+ extractor.selectExtractorTrack(currentTrack);
+ MediaFormat format = extractor.getFormat(currentTrack);
+ String mime = format.getString(MediaFormat.KEY_MIME);
+ ArrayList<String> mediaCodecs = CodecUtils.selectCodecs(mime, true);
+ // Encoding the decoder's output
+ for (String codecName : mediaCodecs) {
+ Native nativeEncoder = new Native();
+ int status = nativeEncoder
+ .Encode(mInputFilePath, mInputFile, mDecodedFile, mStatsFile, codecName);
+ assertEquals(
+ codecName + " encoder returned error " + status + " for " + "file:" + " " +
+ mInputFile, 0, status);
+ }
+ }
+ File decodedFile = new File(mDecodedFile);
+ // Cleanup temporary input file
+ if (decodedFile.exists()) {
+ assertTrue("Unable to delete - " + mDecodedFile, decodedFile.delete());
+ Log.i(TAG, "Successfully deleted - " + mDecodedFile);
+ }
+ fileInput.close();
+ }
+}
diff --git a/media/tests/benchmark/MediaBenchmarkTest/src/androidTest/java/com/android/media/benchmark/tests/ExtractorTest.java b/media/tests/benchmark/MediaBenchmarkTest/src/androidTest/java/com/android/media/benchmark/tests/ExtractorTest.java
new file mode 100644
index 0000000..4d026c1
--- /dev/null
+++ b/media/tests/benchmark/MediaBenchmarkTest/src/androidTest/java/com/android/media/benchmark/tests/ExtractorTest.java
@@ -0,0 +1,121 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package com.android.media.benchmark.tests;
+
+import com.android.media.benchmark.R;
+import com.android.media.benchmark.library.Extractor;
+import com.android.media.benchmark.library.Native;
+import com.android.media.benchmark.library.Stats;
+
+import android.content.Context;
+import android.media.MediaFormat;
+import android.util.Log;
+
+import androidx.test.platform.app.InstrumentationRegistry;
+
+import org.junit.BeforeClass;
+import org.junit.Test;
+import org.junit.runner.RunWith;
+import org.junit.runners.Parameterized;
+
+import java.io.File;
+import java.io.FileDescriptor;
+import java.io.FileInputStream;
+import java.io.FileOutputStream;
+import java.io.IOException;
+import java.util.Arrays;
+import java.util.Collection;
+
+import static org.junit.Assert.assertTrue;
+import static org.junit.Assert.assertEquals;
+
+import static org.junit.Assert.assertTrue;
+
+@RunWith(Parameterized.class)
+public class ExtractorTest {
+ private static Context mContext =
+ InstrumentationRegistry.getInstrumentation().getTargetContext();
+ private static final String mInputFilePath = mContext.getString(R.string.input_file_path);
+ private static final String mStatsFile = mContext.getExternalFilesDir(null) + "/Extractor."
+ + System.currentTimeMillis() + ".csv";
+ private static final String TAG = "ExtractorTest";
+ private String mInputFileName;
+ private int mTrackId;
+
+ @Parameterized.Parameters
+ public static Collection<Object[]> inputFiles() {
+ return Arrays.asList(new Object[][]{/* Parameters: filename, trackId*/
+ {"crowd_1920x1080_25fps_6000kbps_mpeg4.mp4", 0},
+ {"crowd_1920x1080_25fps_6700kbps_h264.ts", 0},
+ {"crowd_1920x1080_25fps_7300kbps_mpeg2.mp4", 0},
+ {"crowd_1920x1080_25fps_4000kbps_av1.webm", 0},
+ {"crowd_1920x1080_25fps_4000kbps_h265.mkv", 0},
+ {"crowd_1920x1080_25fps_4000kbps_vp8.webm", 0},
+ {"bbb_44100hz_2ch_128kbps_aac_5mins.mp4", 0},
+ {"bbb_44100hz_2ch_128kbps_mp3_5mins.mp3", 0},
+ {"bbb_44100hz_2ch_600kbps_flac_5mins.flac", 0},
+ {"bbb_8000hz_1ch_8kbps_amrnb_5mins.3gp", 0},
+ {"bbb_16000hz_1ch_9kbps_amrwb_5mins.3gp", 0},
+ {"bbb_44100hz_2ch_80kbps_vorbis_5mins.webm", 0},
+ {"bbb_48000hz_2ch_100kbps_opus_5mins.webm", 0}});
+ }
+
+ public ExtractorTest(String filename, int track) {
+ this.mInputFileName = filename;
+ this.mTrackId = track;
+ }
+
+ @BeforeClass
+ public static void writeStatsHeaderToFile() throws IOException {
+ Stats mStats = new Stats();
+ boolean status = mStats.writeStatsHeader(mStatsFile);
+ assertTrue("Unable to open stats file for writing!", status);
+ Log.d(TAG, "Saving Benchmark results in: " + mStatsFile);
+ }
+
+ @Test
+ public void testExtractor() throws IOException {
+ File inputFile = new File(mInputFilePath + mInputFileName);
+ assertTrue("Cannot find " + mInputFileName + " in directory " + mInputFilePath,
+ inputFile.exists());
+ FileInputStream fileInput = new FileInputStream(inputFile);
+ FileDescriptor fileDescriptor = fileInput.getFD();
+ Extractor extractor = new Extractor();
+ extractor.setUpExtractor(fileDescriptor);
+ MediaFormat format = extractor.getFormat(mTrackId);
+ String mime = format.getString(MediaFormat.KEY_MIME);
+ int status = extractor.extractSample(mTrackId);
+ assertEquals("Extraction failed for " + mInputFileName, 0, status);
+ Log.i(TAG, "Extracted " + mInputFileName + " successfully.");
+ extractor.deinitExtractor();
+ extractor.dumpStatistics(mInputFileName, mime, mStatsFile);
+ fileInput.close();
+ }
+
+ @Test
+ public void testNativeExtractor() throws IOException {
+ Native nativeExtractor = new Native();
+ File inputFile = new File(mInputFilePath + mInputFileName);
+ assertTrue("Cannot find " + mInputFileName + " in directory " + mInputFilePath,
+ inputFile.exists());
+ FileInputStream fileInput = new FileInputStream(inputFile);
+ int status = nativeExtractor.Extract(mInputFilePath, mInputFileName, mStatsFile);
+ fileInput.close();
+ assertEquals("Extraction failed for " + mInputFileName, 0, status);
+ Log.i(TAG, "Extracted " + mInputFileName + " successfully.");
+ }
+}
diff --git a/media/tests/benchmark/MediaBenchmarkTest/src/androidTest/java/com/android/media/benchmark/tests/MuxerTest.java b/media/tests/benchmark/MediaBenchmarkTest/src/androidTest/java/com/android/media/benchmark/tests/MuxerTest.java
new file mode 100644
index 0000000..21ba957
--- /dev/null
+++ b/media/tests/benchmark/MediaBenchmarkTest/src/androidTest/java/com/android/media/benchmark/tests/MuxerTest.java
@@ -0,0 +1,180 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+package com.android.media.benchmark.tests;
+
+import com.android.media.benchmark.R;
+import com.android.media.benchmark.library.Extractor;
+import com.android.media.benchmark.library.Muxer;
+import com.android.media.benchmark.library.Native;
+import com.android.media.benchmark.library.Stats;
+
+import androidx.test.platform.app.InstrumentationRegistry;
+
+import android.content.Context;
+import android.media.MediaCodec;
+import android.media.MediaFormat;
+import android.media.MediaMuxer;
+import android.util.Log;
+
+import org.junit.BeforeClass;
+import org.junit.Test;
+import org.junit.runner.RunWith;
+import org.junit.runners.Parameterized;
+
+import java.io.File;
+import java.io.FileDescriptor;
+import java.io.FileInputStream;
+import java.io.FileOutputStream;
+import java.io.IOException;
+import java.nio.ByteBuffer;
+import java.util.ArrayList;
+import java.util.Arrays;
+import java.util.Collection;
+import java.util.Hashtable;
+import java.util.Map;
+
+import static org.junit.Assert.assertTrue;
+import static org.junit.Assert.assertEquals;
+import static org.junit.Assert.assertNotEquals;
+
+import static org.junit.Assert.assertTrue;
+
+@RunWith(Parameterized.class)
+public class MuxerTest {
+ private static Context mContext =
+ InstrumentationRegistry.getInstrumentation().getTargetContext();
+ private static final String mInputFilePath = mContext.getString(R.string.input_file_path);
+ private static final String mStatsFile =
+ mContext.getExternalFilesDir(null) + "/Muxer." + System.currentTimeMillis() + ".csv";
+ private static final String TAG = "MuxerTest";
+ private static final Map<String, Integer> mMapFormat = new Hashtable<String, Integer>() {
+ {
+ put("mp4", MediaMuxer.OutputFormat.MUXER_OUTPUT_MPEG_4);
+ put("webm", MediaMuxer.OutputFormat.MUXER_OUTPUT_WEBM);
+ put("3gpp", MediaMuxer.OutputFormat.MUXER_OUTPUT_3GPP);
+ put("ogg", MediaMuxer.OutputFormat.MUXER_OUTPUT_OGG);
+ }
+ };
+ private String mInputFileName;
+ private String mFormat;
+
+ @Parameterized.Parameters
+ public static Collection<Object[]> inputFiles() {
+ return Arrays.asList(new Object[][]{
+ /* Parameters: filename, format */
+ {"crowd_1920x1080_25fps_4000kbps_vp8.webm", "webm"},
+ {"crowd_1920x1080_25fps_4000kbps_vp9.webm", "webm"},
+ {"crowd_1920x1080_25fps_6000kbps_mpeg4.mp4", "mp4"},
+ {"crowd_352x288_25fps_6000kbps_h263.3gp", "mp4"},
+ {"crowd_1920x1080_25fps_6700kbps_h264.ts", "mp4"},
+ {"crowd_1920x1080_25fps_4000kbps_h265.mkv", "mp4"},
+ {"crowd_1920x1080_25fps_6000kbps_mpeg4.mp4", "3gpp"},
+ {"crowd_352x288_25fps_6000kbps_h263.3gp", "3gpp"},
+ {"crowd_1920x1080_25fps_6700kbps_h264.ts", "3gpp"},
+ {"crowd_1920x1080_25fps_4000kbps_h265.mkv", "3gpp"},
+ {"bbb_48000hz_2ch_100kbps_opus_5mins.webm", "ogg"},
+ {"bbb_44100hz_2ch_80kbps_vorbis_5mins.webm", "webm"},
+ {"bbb_48000hz_2ch_100kbps_opus_5mins.webm", "webm"},
+ {"bbb_44100hz_2ch_128kbps_aac_5mins.mp4", "mp4"},
+ {"bbb_8000hz_1ch_8kbps_amrnb_5mins.3gp", "mp4"},
+ {"bbb_16000hz_1ch_9kbps_amrwb_5mins.3gp", "mp4"},
+ {"bbb_44100hz_2ch_128kbps_aac_5mins.mp4", "3gpp"},
+ {"bbb_8000hz_1ch_8kbps_amrnb_5mins.3gp", "3gpp"},
+ {"bbb_16000hz_1ch_9kbps_amrwb_5mins.3gp", "3gpp"}});
+ }
+
+ public MuxerTest(String filename, String outputFormat) {
+ this.mInputFileName = filename;
+ this.mFormat = outputFormat;
+ }
+
+ @BeforeClass
+ public static void writeStatsHeaderToFile() throws IOException {
+ Stats mStats = new Stats();
+ boolean status = mStats.writeStatsHeader(mStatsFile);
+ assertTrue("Unable to open stats file for writing!", status);
+ Log.d(TAG, "Saving Benchmark results in: " + mStatsFile);
+ }
+
+ @Test
+ public void testMuxer() throws IOException {
+ File inputFile = new File(mInputFilePath + mInputFileName);
+ assertTrue("Cannot find " + mInputFileName + " in directory " + mInputFilePath,
+ inputFile.exists());
+ FileInputStream fileInput = new FileInputStream(inputFile);
+ FileDescriptor fileDescriptor = fileInput.getFD();
+ ArrayList<ByteBuffer> inputBuffer = new ArrayList<>();
+ ArrayList<MediaCodec.BufferInfo> inputBufferInfo = new ArrayList<>();
+ Extractor extractor = new Extractor();
+ int trackCount = extractor.setUpExtractor(fileDescriptor);
+ for (int currentTrack = 0; currentTrack < trackCount; currentTrack++) {
+ extractor.selectExtractorTrack(currentTrack);
+ while (true) {
+ int sampleSize = extractor.getFrameSample();
+ MediaCodec.BufferInfo bufferInfo = extractor.getBufferInfo();
+ MediaCodec.BufferInfo tempBufferInfo = new MediaCodec.BufferInfo();
+ tempBufferInfo
+ .set(bufferInfo.offset, bufferInfo.size, bufferInfo.presentationTimeUs,
+ bufferInfo.flags);
+ inputBufferInfo.add(tempBufferInfo);
+ ByteBuffer tempSampleBuffer = ByteBuffer.allocate(tempBufferInfo.size);
+ tempSampleBuffer.put(extractor.getFrameBuffer().array(), 0, bufferInfo.size);
+ inputBuffer.add(tempSampleBuffer);
+ if (sampleSize < 0) {
+ break;
+ }
+ }
+ MediaFormat format = extractor.getFormat(currentTrack);
+ int outputFormat = mMapFormat.getOrDefault(mFormat, -1);
+ assertNotEquals("Test failed for " + mInputFileName + ". Returned invalid " +
+ "output format for given " + mFormat + " format.", -1, outputFormat);
+ Muxer muxer = new Muxer();
+ int trackIndex = muxer.setUpMuxer(mContext, outputFormat, format);
+ int status = muxer.mux(trackIndex, inputBuffer, inputBufferInfo);
+ assertEquals("Cannot perform write operation for " + mInputFileName, 0, status);
+ Log.i(TAG, "Muxed " + mInputFileName + " successfully.");
+ muxer.deInitMuxer();
+ muxer.dumpStatistics(mInputFileName, mFormat, extractor.getClipDuration(), mStatsFile);
+ muxer.resetMuxer();
+ extractor.unselectExtractorTrack(currentTrack);
+ inputBufferInfo.clear();
+ inputBuffer.clear();
+
+ }
+ extractor.deinitExtractor();
+ fileInput.close();
+ }
+
+ @Test
+ public void testNativeMuxer() {
+ Native nativeMuxer = new Native();
+ File inputFile = new File(mInputFilePath + mInputFileName);
+ assertTrue("Cannot find " + mInputFileName + " in directory " + mInputFilePath,
+ inputFile.exists());
+ int tid = android.os.Process.myTid();
+ String mMuxOutputFile = (mContext.getFilesDir() + "/mux_" + tid + ".out");
+ int status = nativeMuxer.Mux(
+ mInputFilePath, mInputFileName, mMuxOutputFile, mStatsFile, mFormat);
+ assertEquals("Cannot perform write operation for " + mInputFileName, 0, status);
+ Log.i(TAG, "Muxed " + mInputFileName + " successfully.");
+ File muxedFile = new File(mMuxOutputFile);
+ // Cleanup temporary output file
+ if (muxedFile.exists()) {
+ assertTrue("Unable to delete" + mMuxOutputFile + " file.",
+ muxedFile.delete());
+ }
+ }
+}
diff --git a/media/tests/benchmark/MediaBenchmarkTest/src/main/cpp/Android.bp b/media/tests/benchmark/MediaBenchmarkTest/src/main/cpp/Android.bp
new file mode 100644
index 0000000..3e5e4c8
--- /dev/null
+++ b/media/tests/benchmark/MediaBenchmarkTest/src/main/cpp/Android.bp
@@ -0,0 +1,33 @@
+cc_test_library {
+ name: "libmediabenchmark_jni",
+ sdk_version: "current",
+
+ defaults: [
+ "libmediabenchmark_common-defaults",
+ "libmediabenchmark_soft_sanitize_all-defaults",
+ ],
+
+ srcs: [
+ "NativeExtractor.cpp",
+ "NativeMuxer.cpp",
+ "NativeEncoder.cpp",
+ "NativeDecoder.cpp",
+ ],
+
+ shared_libs: [
+ "liblog",
+ ],
+
+ static_libs: [
+ "libmediabenchmark_common",
+ "libmediabenchmark_extractor",
+ "libmediabenchmark_muxer",
+ "libmediabenchmark_decoder",
+ "libmediabenchmark_encoder",
+ ],
+
+ cflags: [
+ "-Wall",
+ "-Werror",
+ ],
+}
diff --git a/media/tests/benchmark/MediaBenchmarkTest/src/main/cpp/CMakeLists.txt b/media/tests/benchmark/MediaBenchmarkTest/src/main/cpp/CMakeLists.txt
new file mode 100644
index 0000000..5823883
--- /dev/null
+++ b/media/tests/benchmark/MediaBenchmarkTest/src/main/cpp/CMakeLists.txt
@@ -0,0 +1,44 @@
+#
+# Copyright (C) 2019 The Android Open Source Project
+#
+# Licensed under the Apache License, Version 2.0 (the "License"); you may not
+# use this file except in compliance with the License. You may obtain a copy of
+# the License at
+#
+# http://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS, WITHOUT
+# WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. See the
+# License for the specific language governing permissions and limitations under
+# the License.
+#
+
+cmake_minimum_required(VERSION 3.4.1)
+
+set(native_source_path "../../../../src/native")
+set(CMAKE_CXX_FLAGS "${CMAKE_CXX_FLAGS} -Wall -Werror")
+
+add_library(
+ mediabenchmark_jni SHARED
+ NativeExtractor.cpp
+ NativeMuxer.cpp
+ NativeDecoder.cpp
+ NativeEncoder.cpp
+ ${native_source_path}/common/BenchmarkCommon.cpp
+ ${native_source_path}/common/Stats.cpp
+ ${native_source_path}/common/utils/Timers.cpp
+ ${native_source_path}/extractor/Extractor.cpp
+ ${native_source_path}/muxer/Muxer.cpp
+ ${native_source_path}/decoder/Decoder.cpp
+ ${native_source_path}/encoder/Encoder.cpp)
+
+include_directories(${native_source_path}/common)
+include_directories(${native_source_path}/extractor)
+include_directories(${native_source_path}/muxer)
+include_directories(${native_source_path}/decoder)
+include_directories(${native_source_path}/encoder)
+
+find_library(log-lib log)
+
+target_link_libraries(mediabenchmark_jni mediandk ${log-lib})
diff --git a/media/tests/benchmark/MediaBenchmarkTest/src/main/cpp/NativeDecoder.cpp b/media/tests/benchmark/MediaBenchmarkTest/src/main/cpp/NativeDecoder.cpp
new file mode 100644
index 0000000..043bc9e
--- /dev/null
+++ b/media/tests/benchmark/MediaBenchmarkTest/src/main/cpp/NativeDecoder.cpp
@@ -0,0 +1,130 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "NativeDecoder"
+
+#include <jni.h>
+#include <fstream>
+#include <stdio.h>
+#include <string.h>
+#include <sys/stat.h>
+
+#include <android/log.h>
+
+#include "Decoder.h"
+
+extern "C" JNIEXPORT int JNICALL Java_com_android_media_benchmark_library_Native_Decode(
+ JNIEnv *env, jobject thiz, jstring jFilePath, jstring jFileName, jstring jStatsFile,
+ jstring jCodecName, jboolean asyncMode) {
+ const char *filePath = env->GetStringUTFChars(jFilePath, nullptr);
+ const char *fileName = env->GetStringUTFChars(jFileName, nullptr);
+ string sFilePath = string(filePath) + string(fileName);
+ UNUSED(thiz);
+ FILE *inputFp = fopen(sFilePath.c_str(), "rb");
+ env->ReleaseStringUTFChars(jFileName, fileName);
+ env->ReleaseStringUTFChars(jFilePath, filePath);
+ if (!inputFp) {
+ ALOGE("Unable to open input file for reading");
+ return -1;
+ }
+
+ Decoder *decoder = new Decoder();
+ Extractor *extractor = decoder->getExtractor();
+ if (!extractor) {
+ ALOGE("Extractor creation failed");
+ return -1;
+ }
+
+ // Read file properties
+ struct stat buf;
+ stat(sFilePath.c_str(), &buf);
+ size_t fileSize = buf.st_size;
+ if (fileSize > kMaxBufferSize) {
+ ALOGE("File size greater than maximum buffer size");
+ return -1;
+ }
+ int32_t fd = fileno(inputFp);
+ int32_t trackCount = extractor->initExtractor(fd, fileSize);
+ if (trackCount <= 0) {
+ ALOGE("initExtractor failed");
+ return -1;
+ }
+ for (int curTrack = 0; curTrack < trackCount; curTrack++) {
+ int32_t status = extractor->setupTrackFormat(curTrack);
+ if (status != 0) {
+ ALOGE("Track Format invalid");
+ return -1;
+ }
+
+ uint8_t *inputBuffer = (uint8_t *) malloc(fileSize);
+ if (!inputBuffer) {
+ ALOGE("Insufficient memory");
+ return -1;
+ }
+
+ vector<AMediaCodecBufferInfo> frameInfo;
+ AMediaCodecBufferInfo info;
+ uint32_t inputBufferOffset = 0;
+
+ // Get frame data
+ while (1) {
+ status = extractor->getFrameSample(info);
+ if (status || !info.size) break;
+ // copy the meta data and buffer to be passed to decoder
+ if (inputBufferOffset + info.size > kMaxBufferSize) {
+ ALOGE("Memory allocated not sufficient");
+ free(inputBuffer);
+ return -1;
+ }
+ memcpy(inputBuffer + inputBufferOffset, extractor->getFrameBuf(), info.size);
+ frameInfo.push_back(info);
+ inputBufferOffset += info.size;
+ }
+
+ const char *codecName = env->GetStringUTFChars(jCodecName, nullptr);
+ string sCodecName = string(codecName);
+ decoder->setupDecoder();
+ status = decoder->decode(inputBuffer, frameInfo, sCodecName, asyncMode);
+ if (status != AMEDIA_OK) {
+ ALOGE("Decode returned error");
+ free(inputBuffer);
+ env->ReleaseStringUTFChars(jCodecName, codecName);
+ return -1;
+ }
+ decoder->deInitCodec();
+ const char *inputReference = env->GetStringUTFChars(jFileName, nullptr);
+ const char *statsFile = env->GetStringUTFChars(jStatsFile, nullptr);
+ string sInputReference = string(inputReference);
+ decoder->dumpStatistics(sInputReference, sCodecName, (asyncMode ? "async" : "sync"),
+ statsFile);
+ env->ReleaseStringUTFChars(jCodecName, codecName);
+ env->ReleaseStringUTFChars(jStatsFile, statsFile);
+ env->ReleaseStringUTFChars(jFileName, inputReference);
+ if (inputBuffer) {
+ free(inputBuffer);
+ inputBuffer = nullptr;
+ }
+ decoder->resetDecoder();
+ }
+ if (inputFp) {
+ fclose(inputFp);
+ inputFp = nullptr;
+ }
+ extractor->deInitExtractor();
+ delete decoder;
+ return 0;
+}
diff --git a/media/tests/benchmark/MediaBenchmarkTest/src/main/cpp/NativeEncoder.cpp b/media/tests/benchmark/MediaBenchmarkTest/src/main/cpp/NativeEncoder.cpp
new file mode 100644
index 0000000..1277c8b
--- /dev/null
+++ b/media/tests/benchmark/MediaBenchmarkTest/src/main/cpp/NativeEncoder.cpp
@@ -0,0 +1,218 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "NativeEncoder"
+
+#include <jni.h>
+#include <sys/stat.h>
+#include <fstream>
+#include <iostream>
+
+#include <android/log.h>
+
+#include "Decoder.h"
+#include "Encoder.h"
+
+#include <stdio.h>
+
+constexpr int32_t ENCODE_DEFAULT_FRAME_RATE = 25;
+constexpr int32_t ENCODE_DEFAULT_AUDIO_BIT_RATE = 128000 /* 128 Kbps */;
+constexpr int32_t ENCODE_DEFAULT_BIT_RATE = 8000000 /* 8 Mbps */;
+constexpr int32_t ENCODE_MIN_BIT_RATE = 600000 /* 600 Kbps */;
+
+extern "C" JNIEXPORT int JNICALL Java_com_android_media_benchmark_library_Native_Encode(
+ JNIEnv *env, jobject thiz, jstring jFilePath, jstring jFileName, jstring jOutFilePath,
+ jstring jStatsFile, jstring jCodecName) {
+ const char *filePath = env->GetStringUTFChars(jFilePath, nullptr);
+ const char *fileName = env->GetStringUTFChars(jFileName, nullptr);
+ string sFilePath = string(filePath) + string(fileName);
+ UNUSED(thiz);
+ FILE *inputFp = fopen(sFilePath.c_str(), "rb");
+ env->ReleaseStringUTFChars(jFileName, fileName);
+ env->ReleaseStringUTFChars(jFilePath, filePath);
+ if (!inputFp) {
+ ALOGE("Unable to open input file for reading");
+ return -1;
+ }
+
+ Decoder *decoder = new Decoder();
+ Extractor *extractor = decoder->getExtractor();
+ if (!extractor) {
+ ALOGE("Extractor creation failed");
+ return -1;
+ }
+
+ // Read file properties
+ struct stat buf;
+ stat(sFilePath.c_str(), &buf);
+ size_t fileSize = buf.st_size;
+ if (fileSize > kMaxBufferSize) {
+ ALOGE("File size greater than maximum buffer size");
+ return -1;
+ }
+ int32_t fd = fileno(inputFp);
+ int32_t trackCount = extractor->initExtractor(fd, fileSize);
+ if (trackCount <= 0) {
+ ALOGE("initExtractor failed");
+ return -1;
+ }
+
+ for (int curTrack = 0; curTrack < trackCount; curTrack++) {
+ int32_t status = extractor->setupTrackFormat(curTrack);
+ if (status != 0) {
+ ALOGE("Track Format invalid");
+ return -1;
+ }
+ uint8_t *inputBuffer = (uint8_t *)malloc(fileSize);
+ if (!inputBuffer) {
+ ALOGE("Insufficient memory");
+ return -1;
+ }
+ vector<AMediaCodecBufferInfo> frameInfo;
+ AMediaCodecBufferInfo info;
+ uint32_t inputBufferOffset = 0;
+
+ // Get frame data
+ while (1) {
+ status = extractor->getFrameSample(info);
+ if (status || !info.size) break;
+ // copy the meta data and buffer to be passed to decoder
+ if (inputBufferOffset + info.size > kMaxBufferSize) {
+ ALOGE("Memory allocated not sufficient");
+ free(inputBuffer);
+ return -1;
+ }
+ memcpy(inputBuffer + inputBufferOffset, extractor->getFrameBuf(), info.size);
+ frameInfo.push_back(info);
+ inputBufferOffset += info.size;
+ }
+ string decName = "";
+ const char *outputFilePath = env->GetStringUTFChars(jOutFilePath, nullptr);
+ FILE *outFp = fopen(outputFilePath, "wb");
+ if (outFp == nullptr) {
+ ALOGE("%s - File failed to open for writing!", outputFilePath);
+ free(inputBuffer);
+ return -1;
+ }
+ decoder->setupDecoder();
+ status = decoder->decode(inputBuffer, frameInfo, decName, false /*asyncMode */, outFp);
+ if (status != AMEDIA_OK) {
+ ALOGE("Decode returned error");
+ free(inputBuffer);
+ return -1;
+ }
+
+ AMediaFormat *decoderFormat = decoder->getFormat();
+ AMediaFormat *format = extractor->getFormat();
+ if (inputBuffer) {
+ free(inputBuffer);
+ inputBuffer = nullptr;
+ }
+ const char *mime = nullptr;
+ AMediaFormat_getString(format, AMEDIAFORMAT_KEY_MIME, &mime);
+ if (!mime) {
+ ALOGE("Error in AMediaFormat_getString");
+ return -1;
+ }
+ ifstream eleStream;
+ eleStream.open(outputFilePath, ifstream::binary | ifstream::ate);
+ if (!eleStream.is_open()) {
+ ALOGE("%s - File failed to open for reading!", outputFilePath);
+ env->ReleaseStringUTFChars(jOutFilePath, outputFilePath);
+ return -1;
+ }
+ const char *codecName = env->GetStringUTFChars(jCodecName, NULL);
+ const char *inputReference = env->GetStringUTFChars(jFileName, nullptr);
+ string sCodecName = string(codecName);
+ string sInputReference = string(inputReference);
+
+ bool asyncMode[2] = {true, false};
+ for (int i = 0; i < 2; i++) {
+ size_t eleSize = eleStream.tellg();
+ eleStream.seekg(0, ifstream::beg);
+
+ // Get encoder params
+ encParameter encParams;
+ if (!strncmp(mime, "video/", 6)) {
+ AMediaFormat_getInt32(format, AMEDIAFORMAT_KEY_WIDTH, &encParams.width);
+ AMediaFormat_getInt32(format, AMEDIAFORMAT_KEY_HEIGHT, &encParams.height);
+ AMediaFormat_getInt32(format, AMEDIAFORMAT_KEY_FRAME_RATE, &encParams.frameRate);
+ AMediaFormat_getInt32(format, AMEDIAFORMAT_KEY_BIT_RATE, &encParams.bitrate);
+ if (encParams.bitrate <= 0 || encParams.frameRate <= 0) {
+ encParams.frameRate = ENCODE_DEFAULT_FRAME_RATE;
+ if (!strcmp(mime, "video/3gpp") || !strcmp(mime, "video/mp4v-es")) {
+ encParams.bitrate = ENCODE_MIN_BIT_RATE /* 600 Kbps */;
+ } else {
+ encParams.bitrate = ENCODE_DEFAULT_BIT_RATE /* 8 Mbps */;
+ }
+ }
+ AMediaFormat_getInt32(format, AMEDIAFORMAT_KEY_PROFILE, &encParams.profile);
+ AMediaFormat_getInt32(format, AMEDIAFORMAT_KEY_LEVEL, &encParams.level);
+ AMediaFormat_getInt32(decoderFormat, AMEDIAFORMAT_KEY_COLOR_FORMAT,
+ &encParams.colorFormat);
+ } else {
+ AMediaFormat_getInt32(format, AMEDIAFORMAT_KEY_SAMPLE_RATE, &encParams.sampleRate);
+ AMediaFormat_getInt32(format, AMEDIAFORMAT_KEY_CHANNEL_COUNT,
+ &encParams.numChannels);
+ encParams.bitrate = ENCODE_DEFAULT_AUDIO_BIT_RATE;
+ }
+ Encoder *encoder = new Encoder();
+ encoder->setupEncoder();
+ status = encoder->encode(sCodecName, eleStream, eleSize, asyncMode[i], encParams,
+ (char *)mime);
+ if (status != AMEDIA_OK) {
+ ALOGE("Encoder returned error");
+ return -1;
+ }
+ ALOGV("Encoding complete with codec %s for asyncMode = %d", sCodecName.c_str(),
+ asyncMode[i]);
+ encoder->deInitCodec();
+ const char *statsFile = env->GetStringUTFChars(jStatsFile, nullptr);
+ encoder->dumpStatistics(sInputReference, extractor->getClipDuration(), sCodecName,
+ (asyncMode[i] ? "async" : "sync"), statsFile);
+ env->ReleaseStringUTFChars(jStatsFile, statsFile);
+ encoder->resetEncoder();
+ delete encoder;
+ encoder = nullptr;
+ }
+ eleStream.close();
+ if (outFp) {
+ fclose(outFp);
+ outFp = nullptr;
+ }
+ env->ReleaseStringUTFChars(jFileName, inputReference);
+ env->ReleaseStringUTFChars(jCodecName, codecName);
+ env->ReleaseStringUTFChars(jOutFilePath, outputFilePath);
+ if (format) {
+ AMediaFormat_delete(format);
+ format = nullptr;
+ }
+ if (decoderFormat) {
+ AMediaFormat_delete(decoderFormat);
+ decoderFormat = nullptr;
+ }
+ decoder->deInitCodec();
+ decoder->resetDecoder();
+ }
+ if (inputFp) {
+ fclose(inputFp);
+ inputFp = nullptr;
+ }
+ extractor->deInitExtractor();
+ delete decoder;
+ return 0;
+}
diff --git a/media/tests/benchmark/MediaBenchmarkTest/src/main/cpp/NativeExtractor.cpp b/media/tests/benchmark/MediaBenchmarkTest/src/main/cpp/NativeExtractor.cpp
new file mode 100644
index 0000000..a762760
--- /dev/null
+++ b/media/tests/benchmark/MediaBenchmarkTest/src/main/cpp/NativeExtractor.cpp
@@ -0,0 +1,81 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "NativeExtractor"
+
+#include <jni.h>
+#include <fstream>
+#include <string>
+#include <sys/stat.h>
+
+#include "Extractor.h"
+
+extern "C" JNIEXPORT int32_t JNICALL Java_com_android_media_benchmark_library_Native_Extract(
+ JNIEnv *env, jobject thiz, jstring jInputFilePath, jstring jInputFileName,
+ jstring jStatsFile) {
+ UNUSED(thiz);
+ const char *inputFilePath = env->GetStringUTFChars(jInputFilePath, nullptr);
+ const char *inputFileName = env->GetStringUTFChars(jInputFileName, nullptr);
+ string sFilePath = string(inputFilePath) + string(inputFileName);
+ FILE *inputFp = fopen(sFilePath.c_str(), "rb");
+
+ // Read file properties
+ struct stat buf;
+ stat(sFilePath.c_str(), &buf);
+ size_t fileSize = buf.st_size;
+ int32_t fd = fileno(inputFp);
+
+ Extractor *extractObj = new Extractor();
+ int32_t trackCount = extractObj->initExtractor((long) fd, fileSize);
+ if (trackCount <= 0) {
+ ALOGE("initExtractor failed");
+ return -1;
+ }
+
+ int32_t trackID = 0;
+ const char *mime = nullptr;
+ int32_t status = extractObj->extract(trackID);
+ if (status != AMEDIA_OK) {
+ ALOGE("Extraction failed");
+ return -1;
+ }
+
+ if (inputFp) {
+ fclose(inputFp);
+ inputFp = nullptr;
+ }
+ status = extractObj->setupTrackFormat(trackID);
+ AMediaFormat *format = extractObj->getFormat();
+ if (!format) {
+ ALOGE("format is null!");
+ return -1;
+ }
+ AMediaFormat_getString(format, AMEDIAFORMAT_KEY_MIME, &mime);
+ if (!mime) {
+ ALOGE("mime is null!");
+ return -1;
+ }
+ extractObj->deInitExtractor();
+ const char *statsFile = env->GetStringUTFChars(jStatsFile, nullptr);
+ extractObj->dumpStatistics(string(inputFileName), string(mime), statsFile);
+ env->ReleaseStringUTFChars(jStatsFile, statsFile);
+ env->ReleaseStringUTFChars(jInputFilePath, inputFilePath);
+ env->ReleaseStringUTFChars(jInputFileName, inputFileName);
+
+ delete extractObj;
+ return status;
+}
diff --git a/media/tests/benchmark/MediaBenchmarkTest/src/main/cpp/NativeMuxer.cpp b/media/tests/benchmark/MediaBenchmarkTest/src/main/cpp/NativeMuxer.cpp
new file mode 100644
index 0000000..a5ef5b8
--- /dev/null
+++ b/media/tests/benchmark/MediaBenchmarkTest/src/main/cpp/NativeMuxer.cpp
@@ -0,0 +1,184 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "NativeMuxer"
+
+#include <jni.h>
+#include <fstream>
+#include <string>
+#include <sys/stat.h>
+
+#include "Muxer.h"
+
+MUXER_OUTPUT_T getMuxerOutFormat(const char *fmt);
+
+extern "C" JNIEXPORT int32_t JNICALL Java_com_android_media_benchmark_library_Native_Mux(
+ JNIEnv *env, jobject thiz, jstring jInputFilePath, jstring jInputFileName,
+ jstring jOutputFilePath, jstring jStatsFile, jstring jFormat) {
+ UNUSED(thiz);
+ ALOGV("Mux the samples given by extractor");
+ const char *inputFilePath = env->GetStringUTFChars(jInputFilePath, nullptr);
+ const char *inputFileName = env->GetStringUTFChars(jInputFileName, nullptr);
+ string sInputFile = string(inputFilePath) + string(inputFileName);
+ FILE *inputFp = fopen(sInputFile.c_str(), "rb");
+ if (!inputFp) {
+ ALOGE("Unable to open input file for reading");
+ return -1;
+ }
+
+ const char *fmt = env->GetStringUTFChars(jFormat, nullptr);
+ MUXER_OUTPUT_T outputFormat = getMuxerOutFormat(fmt);
+ if (outputFormat == MUXER_OUTPUT_FORMAT_INVALID) {
+ ALOGE("output format is MUXER_OUTPUT_FORMAT_INVALID");
+ return MUXER_OUTPUT_FORMAT_INVALID;
+ }
+
+ Muxer *muxerObj = new Muxer();
+ Extractor *extractor = muxerObj->getExtractor();
+ if (!extractor) {
+ ALOGE("Extractor creation failed");
+ return -1;
+ }
+
+ // Read file properties
+ struct stat buf;
+ stat(sInputFile.c_str(), &buf);
+ size_t fileSize = buf.st_size;
+ int32_t fd = fileno(inputFp);
+
+ int32_t trackCount = extractor->initExtractor(fd, fileSize);
+ if (trackCount <= 0) {
+ ALOGE("initExtractor failed");
+ return -1;
+ }
+
+ for (int curTrack = 0; curTrack < trackCount; curTrack++) {
+ int32_t status = extractor->setupTrackFormat(curTrack);
+ if (status != 0) {
+ ALOGE("Track Format invalid");
+ return -1;
+ }
+
+ uint8_t *inputBuffer = (uint8_t *) malloc(fileSize);
+ if (!inputBuffer) {
+ ALOGE("Allocation Failed");
+ return -1;
+ }
+ vector<AMediaCodecBufferInfo> frameInfos;
+ AMediaCodecBufferInfo info;
+ uint32_t inputBufferOffset = 0;
+
+ // Get Frame Data
+ while (1) {
+ status = extractor->getFrameSample(info);
+ if (status || !info.size) break;
+ // copy the meta data and buffer to be passed to muxer
+ if (inputBufferOffset + info.size > fileSize) {
+ ALOGE("Memory allocated not sufficient");
+ if (inputBuffer) {
+ free(inputBuffer);
+ inputBuffer = nullptr;
+ }
+ return -1;
+ }
+ memcpy(inputBuffer + inputBufferOffset, extractor->getFrameBuf(),
+ static_cast<size_t>(info.size));
+ info.offset = inputBufferOffset;
+ frameInfos.push_back(info);
+ inputBufferOffset += info.size;
+ }
+
+ const char *outputFilePath = env->GetStringUTFChars(jOutputFilePath, nullptr);
+ FILE *outputFp = fopen(((string) outputFilePath).c_str(), "w+b");
+ env->ReleaseStringUTFChars(jOutputFilePath, outputFilePath);
+
+ if (!outputFp) {
+ ALOGE("Unable to open output file for writing");
+ if (inputBuffer) {
+ free(inputBuffer);
+ inputBuffer = nullptr;
+ }
+ return -1;
+ }
+ int32_t outFd = fileno(outputFp);
+
+ status = muxerObj->initMuxer(outFd, (MUXER_OUTPUT_T) outputFormat);
+ if (status != 0) {
+ ALOGE("initMuxer failed");
+ if (inputBuffer) {
+ free(inputBuffer);
+ inputBuffer = nullptr;
+ }
+ return -1;
+ }
+
+ status = muxerObj->mux(inputBuffer, frameInfos);
+ if (status != 0) {
+ ALOGE("Mux failed");
+ if (inputBuffer) {
+ free(inputBuffer);
+ inputBuffer = nullptr;
+ }
+ return -1;
+ }
+ muxerObj->deInitMuxer();
+ const char *statsFile = env->GetStringUTFChars(jStatsFile, nullptr);
+ string muxFormat(fmt);
+ muxerObj->dumpStatistics(string(inputFileName), muxFormat, statsFile);
+ env->ReleaseStringUTFChars(jStatsFile, statsFile);
+ env->ReleaseStringUTFChars(jInputFilePath, inputFilePath);
+ env->ReleaseStringUTFChars(jInputFileName, inputFileName);
+
+ if (inputBuffer) {
+ free(inputBuffer);
+ inputBuffer = nullptr;
+ }
+ if (outputFp) {
+ fclose(outputFp);
+ outputFp = nullptr;
+ }
+ muxerObj->resetMuxer();
+ }
+ if (inputFp) {
+ fclose(inputFp);
+ inputFp = nullptr;
+ }
+ env->ReleaseStringUTFChars(jFormat, fmt);
+ extractor->deInitExtractor();
+ delete muxerObj;
+
+ return 0;
+}
+
+MUXER_OUTPUT_T getMuxerOutFormat(const char *fmt) {
+ static const struct {
+ const char *name;
+ int value;
+ } kFormatMaps[] = {{"mp4", MUXER_OUTPUT_FORMAT_MPEG_4},
+ {"webm", MUXER_OUTPUT_FORMAT_WEBM},
+ {"3gpp", MUXER_OUTPUT_FORMAT_3GPP},
+ {"ogg", MUXER_OUTPUT_FORMAT_OGG}};
+
+ int32_t muxOutputFormat = MUXER_OUTPUT_FORMAT_INVALID;
+ for (auto kFormatMap : kFormatMaps) {
+ if (!strcmp(fmt, kFormatMap.name)) {
+ muxOutputFormat = kFormatMap.value;
+ break;
+ }
+ }
+ return (MUXER_OUTPUT_T) muxOutputFormat;
+}
diff --git a/media/tests/benchmark/MediaBenchmarkTest/src/main/java/com/android/media/benchmark/library/CodecUtils.java b/media/tests/benchmark/MediaBenchmarkTest/src/main/java/com/android/media/benchmark/library/CodecUtils.java
new file mode 100644
index 0000000..08035c9
--- /dev/null
+++ b/media/tests/benchmark/MediaBenchmarkTest/src/main/java/com/android/media/benchmark/library/CodecUtils.java
@@ -0,0 +1,39 @@
+package com.android.media.benchmark.library;
+
+import android.media.MediaCodecInfo;
+import android.media.MediaCodecList;
+import android.os.Build;
+
+import java.util.ArrayList;
+
+public class CodecUtils {
+ private CodecUtils() {}
+
+ /**
+ * Queries the MediaCodecList and returns codec names of supported codecs.
+ *
+ * @param mimeType Mime type of input
+ * @param isEncoder Specifies encoder or decoder
+ * @return ArrayList of codec names
+ */
+ public static ArrayList<String> selectCodecs(String mimeType, boolean isEncoder) {
+ MediaCodecList codecList = new MediaCodecList(MediaCodecList.REGULAR_CODECS);
+ MediaCodecInfo[] codecInfos = codecList.getCodecInfos();
+ ArrayList<String> supportedCodecs = new ArrayList<>();
+ for (MediaCodecInfo codecInfo : codecInfos) {
+ if (isEncoder != codecInfo.isEncoder()) {
+ continue;
+ }
+ if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.Q && codecInfo.isAlias()) {
+ continue;
+ }
+ String[] types = codecInfo.getSupportedTypes();
+ for (String type : types) {
+ if (type.equalsIgnoreCase(mimeType)) {
+ supportedCodecs.add(codecInfo.getName());
+ }
+ }
+ }
+ return supportedCodecs;
+ }
+}
diff --git a/media/tests/benchmark/MediaBenchmarkTest/src/main/java/com/android/media/benchmark/library/Decoder.java b/media/tests/benchmark/MediaBenchmarkTest/src/main/java/com/android/media/benchmark/library/Decoder.java
new file mode 100644
index 0000000..66fee33
--- /dev/null
+++ b/media/tests/benchmark/MediaBenchmarkTest/src/main/java/com/android/media/benchmark/library/Decoder.java
@@ -0,0 +1,312 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package com.android.media.benchmark.library;
+
+import android.media.MediaCodec;
+import android.media.MediaCodec.BufferInfo;
+import android.media.MediaFormat;
+import android.util.Log;
+
+import androidx.annotation.NonNull;
+
+import java.io.FileOutputStream;
+import java.io.IOException;
+import java.nio.ByteBuffer;
+import java.util.ArrayList;
+
+public class Decoder {
+ private static final String TAG = "Decoder";
+ private static final boolean DEBUG = false;
+ private static final int kQueueDequeueTimeoutUs = 1000;
+
+ private final Object mLock = new Object();
+ private MediaCodec mCodec;
+ private ArrayList<BufferInfo> mInputBufferInfo;
+ private Stats mStats;
+
+ private boolean mSawInputEOS;
+ private boolean mSawOutputEOS;
+ private boolean mSignalledError;
+
+ private int mNumOutputFrame;
+ private int mIndex;
+
+ private ArrayList<ByteBuffer> mInputBuffer;
+ private FileOutputStream mOutputStream;
+
+ public Decoder() { mStats = new Stats(); }
+
+ /**
+ * Setup of decoder
+ *
+ * @param outputStream Will dump the output in this stream if not null.
+ */
+ public void setupDecoder(FileOutputStream outputStream) {
+ mSignalledError = false;
+ mOutputStream = outputStream;
+ }
+
+ private MediaCodec createCodec(String codecName, MediaFormat format) throws IOException {
+ String mime = format.getString(MediaFormat.KEY_MIME);
+ try {
+ MediaCodec codec;
+ if (codecName.isEmpty()) {
+ Log.i(TAG, "File mime type: " + mime);
+ if (mime != null) {
+ codec = MediaCodec.createDecoderByType(mime);
+ Log.i(TAG, "Decoder created for mime type " + mime);
+ return codec;
+ } else {
+ Log.e(TAG, "Mime type is null, please specify a mime type to create decoder");
+ return null;
+ }
+ } else {
+ codec = MediaCodec.createByCodecName(codecName);
+ Log.i(TAG, "Decoder created with codec name: " + codecName + " mime: " + mime);
+ return codec;
+ }
+ } catch (IllegalArgumentException ex) {
+ ex.printStackTrace();
+ Log.e(TAG, "Failed to create decoder for " + codecName + " mime:" + mime);
+ return null;
+ }
+ }
+
+ /**
+ * Decodes the given input buffer,
+ * provided valid list of buffer info and format are passed as inputs.
+ *
+ * @param inputBuffer Decode the provided list of ByteBuffers
+ * @param inputBufferInfo List of buffer info corresponding to provided input buffers
+ * @param asyncMode Will run on async implementation if true
+ * @param format For creating the decoder if codec name is empty and configuring it
+ * @param codecName Will create the decoder with codecName
+ * @return 0 if decode was successful , -1 for fail, -2 for decoder not created
+ * @throws IOException if the codec cannot be created.
+ */
+ public int decode(@NonNull ArrayList<ByteBuffer> inputBuffer,
+ @NonNull ArrayList<BufferInfo> inputBufferInfo, final boolean asyncMode,
+ @NonNull MediaFormat format, String codecName) throws IOException {
+ mInputBuffer = new ArrayList<>(inputBuffer.size());
+ mInputBuffer.addAll(inputBuffer);
+ mInputBufferInfo = new ArrayList<>(inputBufferInfo.size());
+ mInputBufferInfo.addAll(inputBufferInfo);
+ mSawInputEOS = false;
+ mSawOutputEOS = false;
+ mNumOutputFrame = 0;
+ mIndex = 0;
+ long sTime = mStats.getCurTime();
+ mCodec = createCodec(codecName, format);
+ if (mCodec == null) {
+ return -2;
+ }
+ if (asyncMode) {
+ mCodec.setCallback(new MediaCodec.Callback() {
+ @Override
+ public void onInputBufferAvailable(
+ @NonNull MediaCodec mediaCodec, int inputBufferId) {
+ try {
+ mStats.addInputTime();
+ onInputAvailable(inputBufferId, mediaCodec);
+ } catch (Exception e) {
+ e.printStackTrace();
+ Log.e(TAG, e.toString());
+ }
+ }
+
+ @Override
+ public void onOutputBufferAvailable(@NonNull MediaCodec mediaCodec,
+ int outputBufferId, @NonNull MediaCodec.BufferInfo bufferInfo) {
+ mStats.addOutputTime();
+ onOutputAvailable(mediaCodec, outputBufferId, bufferInfo);
+ if (mSawOutputEOS) {
+ synchronized (mLock) { mLock.notify(); }
+ }
+ }
+
+ @Override
+ public void onOutputFormatChanged(
+ @NonNull MediaCodec mediaCodec, @NonNull MediaFormat format) {
+ Log.i(TAG, "Output format changed. Format: " + format.toString());
+ }
+
+ @Override
+ public void onError(
+ @NonNull MediaCodec mediaCodec, @NonNull MediaCodec.CodecException e) {
+ mSignalledError = true;
+ Log.e(TAG, "Codec Error: " + e.toString());
+ e.printStackTrace();
+ synchronized (mLock) { mLock.notify(); }
+ }
+ });
+ }
+ int isEncoder = 0;
+ if (DEBUG) {
+ Log.d(TAG, "Media Format : " + format.toString());
+ }
+ mCodec.configure(format, null, null, isEncoder);
+ mCodec.start();
+ Log.i(TAG, "Codec started ");
+ long eTime = mStats.getCurTime();
+ mStats.setInitTime(mStats.getTimeDiff(sTime, eTime));
+ mStats.setStartTime();
+ if (asyncMode) {
+ try {
+ synchronized (mLock) { mLock.wait(); }
+ if (mSignalledError) {
+ return -1;
+ }
+ } catch (InterruptedException e) {
+ e.printStackTrace();
+ }
+ } else {
+ while (!mSawOutputEOS && !mSignalledError) {
+ /* Queue input data */
+ if (!mSawInputEOS) {
+ int inputBufferId = mCodec.dequeueInputBuffer(kQueueDequeueTimeoutUs);
+ if (inputBufferId < 0 && inputBufferId != MediaCodec.INFO_TRY_AGAIN_LATER) {
+ Log.e(TAG,
+ "MediaCodec.dequeueInputBuffer "
+ + " returned invalid index : " + inputBufferId);
+ return -1;
+ }
+ mStats.addInputTime();
+ onInputAvailable(inputBufferId, mCodec);
+ }
+ /* Dequeue output data */
+ BufferInfo outputBufferInfo = new BufferInfo();
+ int outputBufferId =
+ mCodec.dequeueOutputBuffer(outputBufferInfo, kQueueDequeueTimeoutUs);
+ if (outputBufferId < 0) {
+ if (outputBufferId == MediaCodec.INFO_OUTPUT_FORMAT_CHANGED) {
+ MediaFormat outFormat = mCodec.getOutputFormat();
+ Log.i(TAG, "Output format changed. Format: " + outFormat.toString());
+ } else if (outputBufferId == MediaCodec.INFO_OUTPUT_BUFFERS_CHANGED) {
+ Log.i(TAG, "Ignoring deprecated flag: INFO_OUTPUT_BUFFERS_CHANGED");
+ } else if (outputBufferId != MediaCodec.INFO_TRY_AGAIN_LATER) {
+ Log.e(TAG,
+ "MediaCodec.dequeueOutputBuffer"
+ + " returned invalid index " + outputBufferId);
+ return -1;
+ }
+ } else {
+ mStats.addOutputTime();
+ if (DEBUG) {
+ Log.d(TAG, "Dequeue O/P buffer with BufferID " + outputBufferId);
+ }
+ onOutputAvailable(mCodec, outputBufferId, outputBufferInfo);
+ }
+ }
+ }
+ mInputBuffer.clear();
+ mInputBufferInfo.clear();
+ return 0;
+ }
+
+ /**
+ * Stops the codec and releases codec resources.
+ */
+ public void deInitCodec() {
+ long sTime = mStats.getCurTime();
+ if (mCodec != null) {
+ mCodec.stop();
+ mCodec.release();
+ mCodec = null;
+ }
+ long eTime = mStats.getCurTime();
+ mStats.setDeInitTime(mStats.getTimeDiff(sTime, eTime));
+ }
+
+ /**
+ * Prints out the statistics in the information log
+ *
+ * @param inputReference The operation being performed, in this case decode
+ * @param componentName Name of the component/codec
+ * @param mode The operating mode: Sync/Async
+ * @param durationUs Duration of the clip in microseconds
+ * @param statsFile The output file where the stats data is written
+ */
+ public void dumpStatistics(String inputReference, String componentName, String mode,
+ long durationUs, String statsFile) throws IOException {
+ String operation = "decode";
+ mStats.dumpStatistics(
+ inputReference, operation, componentName, mode, durationUs, statsFile);
+ }
+
+ /**
+ * Resets the stats
+ */
+ public void resetDecoder() { mStats.reset(); }
+
+ /**
+ * Returns the format of the output buffers
+ */
+ public MediaFormat getFormat() {
+ return mCodec.getOutputFormat();
+ }
+
+ private void onInputAvailable(int inputBufferId, MediaCodec mediaCodec) {
+ if ((inputBufferId >= 0) && !mSawInputEOS) {
+ ByteBuffer inputCodecBuffer = mediaCodec.getInputBuffer(inputBufferId);
+ BufferInfo bufInfo = mInputBufferInfo.get(mIndex);
+ inputCodecBuffer.put(mInputBuffer.get(mIndex).array());
+ mIndex++;
+ mSawInputEOS = (bufInfo.flags & MediaCodec.BUFFER_FLAG_END_OF_STREAM) != 0;
+ if (mSawInputEOS) {
+ Log.i(TAG, "Saw input EOS");
+ }
+ mStats.addFrameSize(bufInfo.size);
+ mediaCodec.queueInputBuffer(inputBufferId, bufInfo.offset, bufInfo.size,
+ bufInfo.presentationTimeUs, bufInfo.flags);
+ if (DEBUG) {
+ Log.d(TAG,
+ "Codec Input: "
+ + "flag = " + bufInfo.flags + " timestamp = "
+ + bufInfo.presentationTimeUs + " size = " + bufInfo.size);
+ }
+ }
+ }
+
+ private void onOutputAvailable(
+ MediaCodec mediaCodec, int outputBufferId, BufferInfo outputBufferInfo) {
+ if (mSawOutputEOS || outputBufferId < 0) {
+ return;
+ }
+ mNumOutputFrame++;
+ if (DEBUG) {
+ Log.d(TAG,
+ "In OutputBufferAvailable ,"
+ + " output frame number = " + mNumOutputFrame);
+ }
+ if (mOutputStream != null) {
+ try {
+ ByteBuffer outputBuffer = mediaCodec.getOutputBuffer(outputBufferId);
+ byte[] bytesOutput = new byte[outputBuffer.remaining()];
+ outputBuffer.get(bytesOutput);
+ mOutputStream.write(bytesOutput);
+ } catch (IOException e) {
+ e.printStackTrace();
+ Log.d(TAG, "Error Dumping File: Exception " + e.toString());
+ }
+ }
+ mediaCodec.releaseOutputBuffer(outputBufferId, false);
+ mSawOutputEOS = (outputBufferInfo.flags & MediaCodec.BUFFER_FLAG_END_OF_STREAM) != 0;
+ if (mSawOutputEOS) {
+ Log.i(TAG, "Saw output EOS");
+ }
+ }
+}
diff --git a/media/tests/benchmark/MediaBenchmarkTest/src/main/java/com/android/media/benchmark/library/Encoder.java b/media/tests/benchmark/MediaBenchmarkTest/src/main/java/com/android/media/benchmark/library/Encoder.java
new file mode 100644
index 0000000..45e5574
--- /dev/null
+++ b/media/tests/benchmark/MediaBenchmarkTest/src/main/java/com/android/media/benchmark/library/Encoder.java
@@ -0,0 +1,364 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package com.android.media.benchmark.library;
+
+import android.media.MediaCodec;
+import android.media.MediaCodec.CodecException;
+import android.media.MediaFormat;
+import android.util.Log;
+
+import androidx.annotation.NonNull;
+
+import java.io.FileInputStream;
+import java.io.FileOutputStream;
+import java.io.IOException;
+import java.nio.ByteBuffer;
+
+public class Encoder {
+ // Change in AUDIO_ENCODE_DEFAULT_MAX_INPUT_SIZE should also be taken to
+ // kDefaultAudioEncodeFrameSize present in BenchmarkCommon.h
+ private static final int AUDIO_ENCODE_DEFAULT_MAX_INPUT_SIZE = 4096;
+ private static final String TAG = "Encoder";
+ private static final boolean DEBUG = false;
+ private static final int kQueueDequeueTimeoutUs = 1000;
+
+ private final Object mLock = new Object();
+ private MediaCodec mCodec;
+ private String mMime;
+ private Stats mStats;
+
+ private int mOffset;
+ private int mFrameSize;
+ private int mNumInputFrame;
+ private int mNumFrames;
+ private int mFrameRate;
+ private int mSampleRate;
+ private long mInputBufferSize;
+
+ private boolean mSawInputEOS;
+ private boolean mSawOutputEOS;
+ private boolean mSignalledError;
+
+ private FileInputStream mInputStream;
+ private FileOutputStream mOutputStream;
+
+ public Encoder() {
+ mStats = new Stats();
+ mNumInputFrame = 0;
+ mSawInputEOS = false;
+ mSawOutputEOS = false;
+ mSignalledError = false;
+ }
+
+ /**
+ * Setup of encoder
+ *
+ * @param encoderOutputStream Will dump the encoder output in this stream if not null.
+ * @param fileInputStream Will read the decoded output from this stream
+ */
+ public void setupEncoder(FileOutputStream encoderOutputStream,
+ FileInputStream fileInputStream) {
+ this.mInputStream = fileInputStream;
+ this.mOutputStream = encoderOutputStream;
+ }
+
+ private MediaCodec createCodec(String codecName, String mime) throws IOException {
+ try {
+ MediaCodec codec;
+ if (codecName.isEmpty()) {
+ Log.i(TAG, "Mime type: " + mime);
+ if (mime != null) {
+ codec = MediaCodec.createEncoderByType(mime);
+ Log.i(TAG, "Encoder created for mime type " + mime);
+ return codec;
+ } else {
+ Log.e(TAG, "Mime type is null, please specify a mime type to create encoder");
+ return null;
+ }
+ } else {
+ codec = MediaCodec.createByCodecName(codecName);
+ Log.i(TAG, "Encoder created with codec name: " + codecName + " and mime: " + mime);
+ return codec;
+ }
+ } catch (IllegalArgumentException ex) {
+ ex.printStackTrace();
+ Log.e(TAG, "Failed to create encoder for " + codecName + " mime: " + mime);
+ return null;
+ }
+ }
+
+ /**
+ * Encodes the given raw input file and measures the performance of encode operation,
+ * provided a valid list of parameters are passed as inputs.
+ *
+ * @param codecName Will create the encoder with codecName
+ * @param mime For creating encode format
+ * @param encodeFormat Format of the output data
+ * @param frameSize Size of the frame
+ * @param asyncMode Will run on async implementation if true
+ * @return 0 if encode was successful , -1 for fail, -2 for encoder not created
+ * @throws IOException If the codec cannot be created.
+ */
+ public int encode(String codecName, MediaFormat encodeFormat, String mime, int frameRate,
+ int sampleRate, int frameSize, boolean asyncMode) throws IOException {
+ mInputBufferSize = mInputStream.getChannel().size();
+ mMime = mime;
+ mOffset = 0;
+ mFrameRate = frameRate;
+ mSampleRate = sampleRate;
+ long sTime = mStats.getCurTime();
+ mCodec = createCodec(codecName, mime);
+ if (mCodec == null) {
+ return -2;
+ }
+ /*Configure Codec*/
+ try {
+ mCodec.configure(encodeFormat, null, null, MediaCodec.CONFIGURE_FLAG_ENCODE);
+ } catch (IllegalArgumentException | IllegalStateException | MediaCodec.CryptoException e) {
+ Log.e(TAG, "Failed to configure " + mCodec.getName() + " encoder.");
+ e.printStackTrace();
+ return -2;
+ }
+ if (mMime.startsWith("video/")) {
+ mFrameSize = frameSize;
+ } else {
+ int maxInputSize = AUDIO_ENCODE_DEFAULT_MAX_INPUT_SIZE;
+ MediaFormat format = mCodec.getInputFormat();
+ if (format.containsKey(MediaFormat.KEY_MAX_INPUT_SIZE)) {
+ maxInputSize = format.getInteger(MediaFormat.KEY_MAX_INPUT_SIZE);
+ }
+ mFrameSize = frameSize;
+ if (mFrameSize > maxInputSize && maxInputSize > 0) {
+ mFrameSize = maxInputSize;
+ }
+ }
+ mNumFrames = (int) ((mInputBufferSize + mFrameSize - 1) / mFrameSize);
+ if (asyncMode) {
+ mCodec.setCallback(new MediaCodec.Callback() {
+ @Override
+ public void onInputBufferAvailable(@NonNull MediaCodec mediaCodec,
+ int inputBufferId) {
+ try {
+ mStats.addInputTime();
+ onInputAvailable(mediaCodec, inputBufferId);
+ } catch (Exception e) {
+ e.printStackTrace();
+ Log.e(TAG, e.toString());
+ }
+ }
+
+ @Override
+ public void onOutputBufferAvailable(@NonNull MediaCodec mediaCodec,
+ int outputBufferId,
+ @NonNull MediaCodec.BufferInfo bufferInfo) {
+ mStats.addOutputTime();
+ onOutputAvailable(mediaCodec, outputBufferId, bufferInfo);
+ if (mSawOutputEOS) {
+ Log.i(TAG, "Saw output EOS");
+ synchronized (mLock) { mLock.notify(); }
+ }
+ }
+
+ @Override
+ public void onError(@NonNull MediaCodec mediaCodec, @NonNull CodecException e) {
+ mediaCodec.stop();
+ mediaCodec.release();
+ Log.e(TAG, "CodecError: " + e.toString());
+ e.printStackTrace();
+ }
+
+ @Override
+ public void onOutputFormatChanged(@NonNull MediaCodec mediaCodec,
+ @NonNull MediaFormat format) {
+ Log.i(TAG, "Output format changed. Format: " + format.toString());
+ }
+ });
+ }
+ mCodec.start();
+ long eTime = mStats.getCurTime();
+ mStats.setInitTime(mStats.getTimeDiff(sTime, eTime));
+ mStats.setStartTime();
+ if (asyncMode) {
+ try {
+ synchronized (mLock) { mLock.wait(); }
+ if (mSignalledError) {
+ return -1;
+ }
+ } catch (InterruptedException e) {
+ e.printStackTrace();
+ }
+ } else {
+ while (!mSawOutputEOS && !mSignalledError) {
+ /* Queue input data */
+ if (!mSawInputEOS) {
+ int inputBufferId = mCodec.dequeueInputBuffer(kQueueDequeueTimeoutUs);
+ if (inputBufferId < 0 && inputBufferId != MediaCodec.INFO_TRY_AGAIN_LATER) {
+ Log.e(TAG, "MediaCodec.dequeueInputBuffer " + "returned invalid index : " +
+ inputBufferId);
+ return -1;
+ }
+ mStats.addInputTime();
+ onInputAvailable(mCodec, inputBufferId);
+ }
+ /* Dequeue output data */
+ MediaCodec.BufferInfo outputBufferInfo = new MediaCodec.BufferInfo();
+ int outputBufferId =
+ mCodec.dequeueOutputBuffer(outputBufferInfo, kQueueDequeueTimeoutUs);
+ if (outputBufferId < 0) {
+ if (outputBufferId == MediaCodec.INFO_OUTPUT_FORMAT_CHANGED) {
+ MediaFormat outFormat = mCodec.getOutputFormat();
+ Log.i(TAG, "Output format changed. Format: " + outFormat.toString());
+ } else if (outputBufferId != MediaCodec.INFO_TRY_AGAIN_LATER) {
+ Log.e(TAG, "MediaCodec.dequeueOutputBuffer" + " returned invalid index " +
+ outputBufferId);
+ return -1;
+ }
+ } else {
+ mStats.addOutputTime();
+ if (DEBUG) {
+ Log.d(TAG, "Dequeue O/P buffer with BufferID " + outputBufferId);
+ }
+ onOutputAvailable(mCodec, outputBufferId, outputBufferInfo);
+ }
+ }
+ }
+ return 0;
+ }
+
+ private void onOutputAvailable(MediaCodec mediaCodec, int outputBufferId,
+ MediaCodec.BufferInfo outputBufferInfo) {
+ if (mSawOutputEOS || outputBufferId < 0) {
+ if (mSawOutputEOS) {
+ Log.i(TAG, "Saw output EOS");
+ }
+ return;
+ }
+ ByteBuffer outputBuffer = mediaCodec.getOutputBuffer(outputBufferId);
+ if (mOutputStream != null) {
+ try {
+
+ byte[] bytesOutput = new byte[outputBuffer.remaining()];
+ outputBuffer.get(bytesOutput);
+ mOutputStream.write(bytesOutput);
+ } catch (IOException e) {
+ e.printStackTrace();
+ Log.d(TAG, "Error Dumping File: Exception " + e.toString());
+ return;
+ }
+ }
+ mStats.addFrameSize(outputBuffer.remaining());
+ mediaCodec.releaseOutputBuffer(outputBufferId, false);
+ mSawOutputEOS = (outputBufferInfo.flags & MediaCodec.BUFFER_FLAG_END_OF_STREAM) != 0;
+ }
+
+ private void onInputAvailable(MediaCodec mediaCodec, int inputBufferId) throws IOException {
+ if (mSawInputEOS || inputBufferId < 0) {
+ if (mSawInputEOS) {
+ Log.i(TAG, "Saw input EOS");
+ }
+ return;
+ }
+ if (mInputBufferSize < mOffset) {
+ Log.e(TAG, "Out of bound access of input buffer");
+ mSignalledError = true;
+ return;
+ }
+ ByteBuffer inputBuffer = mCodec.getInputBuffer(inputBufferId);
+ if (inputBuffer == null) {
+ mSignalledError = true;
+ return;
+ }
+ int bufSize = inputBuffer.capacity();
+ int bytesToRead = mFrameSize;
+ if (mInputBufferSize - mOffset < mFrameSize) {
+ bytesToRead = (int) (mInputBufferSize - mOffset);
+ }
+ //b/148655275 - Update Frame size, as Format value may not be valid
+ if (bufSize < bytesToRead) {
+ if(mNumInputFrame == 0) {
+ mFrameSize = bufSize;
+ bytesToRead = bufSize;
+ mNumFrames = (int) ((mInputBufferSize + mFrameSize - 1) / mFrameSize);
+ } else {
+ mSignalledError = true;
+ return;
+ }
+ }
+
+ byte[] inputArray = new byte[bytesToRead];
+ mInputStream.read(inputArray, 0, bytesToRead);
+ inputBuffer.put(inputArray);
+ int flag = 0;
+ if (mNumInputFrame >= mNumFrames - 1 || bytesToRead == 0) {
+ Log.i(TAG, "Sending EOS on input last frame");
+ mSawInputEOS = true;
+ flag = MediaCodec.BUFFER_FLAG_END_OF_STREAM;
+ }
+ int presentationTimeUs;
+ if (mMime.startsWith("video/")) {
+ presentationTimeUs = mNumInputFrame * (1000000 / mFrameRate);
+ } else {
+ presentationTimeUs = mNumInputFrame * mFrameSize * 1000000 / mSampleRate;
+ }
+ mediaCodec.queueInputBuffer(inputBufferId, 0, bytesToRead, presentationTimeUs, flag);
+ mNumInputFrame++;
+ mOffset += bytesToRead;
+ }
+
+ /**
+ * Stops the codec and releases codec resources.
+ */
+ public void deInitEncoder() {
+ long sTime = mStats.getCurTime();
+ if (mCodec != null) {
+ mCodec.stop();
+ mCodec.release();
+ mCodec = null;
+ }
+ long eTime = mStats.getCurTime();
+ mStats.setDeInitTime(mStats.getTimeDiff(sTime, eTime));
+ }
+
+ /**
+ * Prints out the statistics in the information log
+ *
+ * @param inputReference The operation being performed, in this case decode
+ * @param componentName Name of the component/codec
+ * @param mode The operating mode: Sync/Async
+ * @param durationUs Duration of the clip in microseconds
+ * @param statsFile The output file where the stats data is written
+ */
+ public void dumpStatistics(String inputReference, String componentName, String mode,
+ long durationUs, String statsFile) throws IOException {
+ String operation = "encode";
+ mStats.dumpStatistics(
+ inputReference, operation, componentName, mode, durationUs, statsFile);
+ }
+
+ /**
+ * Resets the stats
+ */
+ public void resetEncoder() {
+ mOffset = 0;
+ mInputBufferSize = 0;
+ mNumInputFrame = 0;
+ mSawInputEOS = false;
+ mSawOutputEOS = false;
+ mSignalledError = false;
+ mStats.reset();
+ }
+}
diff --git a/media/tests/benchmark/MediaBenchmarkTest/src/main/java/com/android/media/benchmark/library/Extractor.java b/media/tests/benchmark/MediaBenchmarkTest/src/main/java/com/android/media/benchmark/library/Extractor.java
new file mode 100644
index 0000000..f3024e7
--- /dev/null
+++ b/media/tests/benchmark/MediaBenchmarkTest/src/main/java/com/android/media/benchmark/library/Extractor.java
@@ -0,0 +1,178 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package com.android.media.benchmark.library;
+
+import android.media.MediaCodec;
+import android.media.MediaExtractor;
+import android.media.MediaFormat;
+import android.util.Log;
+
+import java.io.FileDescriptor;
+import java.io.IOException;
+import java.nio.ByteBuffer;
+
+public class Extractor {
+ private static final String TAG = "Extractor";
+ private static final int kMaxBufSize = 1024 * 1024 * 16;
+ private MediaExtractor mExtractor;
+ private ByteBuffer mFrameBuffer;
+ private MediaCodec.BufferInfo mBufferInfo;
+ private Stats mStats;
+ private long mDurationUs;
+
+ public Extractor() {
+ mFrameBuffer = ByteBuffer.allocate(kMaxBufSize);
+ mBufferInfo = new MediaCodec.BufferInfo();
+ mStats = new Stats();
+ }
+
+ /**
+ * Creates a Media Extractor and sets data source(FileDescriptor)to use
+ *
+ * @param fileDescriptor FileDescriptor for the file which is to be extracted
+ * @return TrackCount of the sample
+ * @throws IOException If FileDescriptor is null
+ */
+ public int setUpExtractor(FileDescriptor fileDescriptor) throws IOException {
+ long sTime = mStats.getCurTime();
+ mExtractor = new MediaExtractor();
+ mExtractor.setDataSource(fileDescriptor);
+ long eTime = mStats.getCurTime();
+ long timeTaken = mStats.getTimeDiff(sTime, eTime);
+ mStats.setInitTime(timeTaken);
+ return mExtractor.getTrackCount();
+ }
+
+ /**
+ * Returns the track format of the specified index
+ *
+ * @param trackID Index of the track
+ * @return Format of the track
+ */
+ public MediaFormat getFormat(int trackID) { return mExtractor.getTrackFormat(trackID); }
+
+ /**
+ * Returns the extracted buffer for the input clip
+ */
+ public ByteBuffer getFrameBuffer() { return this.mFrameBuffer; }
+
+ /**
+ * Returns the information of buffer related to sample
+ */
+ public MediaCodec.BufferInfo getBufferInfo() { return this.mBufferInfo; }
+
+ /**
+ * Returns the duration of the sample
+ */
+ public long getClipDuration() { return this.mDurationUs; }
+
+ /**
+ * Retrieve the current sample and store it in the byte buffer
+ * Also, sets the information related to extracted sample and store it in buffer info
+ *
+ * @return Sample size of the extracted sample
+ */
+ public int getFrameSample() {
+ int sampleSize = mExtractor.readSampleData(mFrameBuffer, 0);
+ if (sampleSize < 0) {
+ mBufferInfo.flags = MediaCodec.BUFFER_FLAG_END_OF_STREAM;
+ mBufferInfo.size = 0;
+ } else {
+ mBufferInfo.size = sampleSize;
+ mBufferInfo.offset = 0;
+ mBufferInfo.flags = mExtractor.getSampleFlags();
+ mBufferInfo.presentationTimeUs = mExtractor.getSampleTime();
+ mExtractor.advance();
+ }
+ return sampleSize;
+ }
+
+ /**
+ * Setup the track format and get the duration of the sample
+ * Track is selected here for extraction
+ *
+ * @param trackId Track index to be selected
+ * @return 0 for valid track, otherwise -1
+ */
+ public int selectExtractorTrack(int trackId) {
+ MediaFormat trackFormat = mExtractor.getTrackFormat(trackId);
+ mDurationUs = trackFormat.getLong(MediaFormat.KEY_DURATION);
+ if (mDurationUs < 0) {
+ Log.e(TAG, "Invalid Clip");
+ return -1;
+ }
+ mExtractor.selectTrack(trackId);
+ return 0;
+ }
+
+ /**
+ * Unselect the track
+ *
+ * @param trackId Track Index to be unselected
+ */
+ public void unselectExtractorTrack(int trackId) { mExtractor.unselectTrack(trackId); }
+
+ /**
+ * Free up the resources
+ */
+ public void deinitExtractor() {
+ long sTime = mStats.getCurTime();
+ mExtractor.release();
+ long eTime = mStats.getCurTime();
+ long timeTaken = mStats.getTimeDiff(sTime, eTime);
+ mStats.setDeInitTime(timeTaken);
+ }
+
+ /**
+ * Performs extract operation
+ *
+ * @param currentTrack Track index to be extracted
+ * @return Status as 0 if extraction is successful, -1 otherwise
+ */
+ public int extractSample(int currentTrack) {
+ int status;
+ status = selectExtractorTrack(currentTrack);
+ if (status == -1) {
+ Log.e(TAG, "Failed to select track");
+ return -1;
+ }
+ mStats.setStartTime();
+ while (true) {
+ int readSampleSize = getFrameSample();
+ if (readSampleSize <= 0) {
+ break;
+ }
+ mStats.addOutputTime();
+ mStats.addFrameSize(readSampleSize);
+ }
+ unselectExtractorTrack(currentTrack);
+ return 0;
+ }
+
+ /**
+ * Write the benchmark logs for the given input file
+ *
+ * @param inputReference Name of the input file
+ * @param mimeType Mime type of the muxed file
+ * @param statsFile The output file where the stats data is written
+ */
+ public void dumpStatistics(String inputReference, String mimeType, String statsFile)
+ throws IOException {
+ String operation = "extract";
+ mStats.dumpStatistics(inputReference, operation, mimeType, "", mDurationUs, statsFile);
+ }
+}
diff --git a/media/tests/benchmark/MediaBenchmarkTest/src/main/java/com/android/media/benchmark/library/Muxer.java b/media/tests/benchmark/MediaBenchmarkTest/src/main/java/com/android/media/benchmark/library/Muxer.java
new file mode 100644
index 0000000..340b539
--- /dev/null
+++ b/media/tests/benchmark/MediaBenchmarkTest/src/main/java/com/android/media/benchmark/library/Muxer.java
@@ -0,0 +1,113 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+package com.android.media.benchmark.library;
+
+import android.content.Context;
+import android.media.MediaCodec;
+import android.media.MediaFormat;
+import android.media.MediaMuxer;
+
+import java.io.IOException;
+import java.nio.ByteBuffer;
+import java.util.ArrayList;
+
+public class Muxer {
+ private Stats mStats;
+ private MediaMuxer mMuxer;
+
+ /**
+ * Creates a Media Muxer for the specified path
+ *
+ * @param context App context to specify the output file path
+ * @param outputFormat Format of the output media file
+ * @param trackFormat Format of the current track
+ * @return Returns the track index of the newly added track, -1 otherwise
+ */
+ public int setUpMuxer(Context context, int outputFormat, MediaFormat trackFormat) {
+ try {
+ mStats = new Stats();
+ long sTime = mStats.getCurTime();
+ mMuxer = new MediaMuxer(context.getFilesDir().getPath() + "/mux.out.", outputFormat);
+ int trackIndex = mMuxer.addTrack(trackFormat);
+ mMuxer.start();
+ long eTime = mStats.getCurTime();
+ long timeTaken = mStats.getTimeDiff(sTime, eTime);
+ mStats.setInitTime(timeTaken);
+ return trackIndex;
+ } catch (IllegalArgumentException | IOException e) {
+ e.printStackTrace();
+ return -1;
+ }
+ }
+
+ /**
+ * Performs the Mux operation
+ *
+ * @param trackIndex Track index of the sample
+ * @param inputExtractedBuffer Buffer containing encoded samples
+ * @param inputBufferInfo Buffer information related to these samples
+ * @return Returns Status as 0 if write operation is successful, -1 otherwise
+ */
+ public int mux(int trackIndex, ArrayList<ByteBuffer> inputExtractedBuffer,
+ ArrayList<MediaCodec.BufferInfo> inputBufferInfo) {
+ mStats.setStartTime();
+ for (int sampleCount = 0; sampleCount < inputExtractedBuffer.size(); sampleCount++) {
+ try {
+ mMuxer.writeSampleData(trackIndex, inputExtractedBuffer.get(sampleCount),
+ inputBufferInfo.get(sampleCount));
+ mStats.addOutputTime();
+ mStats.addFrameSize(inputBufferInfo.get(sampleCount).size);
+ } catch (IllegalArgumentException | IllegalStateException e) {
+ e.printStackTrace();
+ return -1;
+ }
+ }
+ return 0;
+ }
+
+ /**
+ * Stops the muxer and free up the resources
+ */
+ public void deInitMuxer() {
+ long sTime = mStats.getCurTime();
+ mMuxer.stop();
+ mMuxer.release();
+ long eTime = mStats.getCurTime();
+ long timeTaken = mStats.getTimeDiff(sTime, eTime);
+ mStats.setDeInitTime(timeTaken);
+ }
+
+ /**
+ * Resets the stats
+ */
+ public void resetMuxer() {
+ mStats.reset();
+ }
+
+ /**
+ * Write the benchmark logs for the given input file
+ *
+ * @param inputReference Name of the input file
+ * @param muxFormat Format of the muxed output
+ * @param clipDuration Duration of the given inputReference file
+ * @param statsFile The output file where the stats data is written
+ */
+ public void dumpStatistics(String inputReference, String muxFormat, long clipDuration,
+ String statsFile) throws IOException {
+ String operation = "mux";
+ mStats.dumpStatistics(inputReference, operation, muxFormat, "", clipDuration, statsFile);
+ }
+}
diff --git a/media/tests/benchmark/MediaBenchmarkTest/src/main/java/com/android/media/benchmark/library/Native.java b/media/tests/benchmark/MediaBenchmarkTest/src/main/java/com/android/media/benchmark/library/Native.java
new file mode 100644
index 0000000..38b608a
--- /dev/null
+++ b/media/tests/benchmark/MediaBenchmarkTest/src/main/java/com/android/media/benchmark/library/Native.java
@@ -0,0 +1,32 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package com.android.media.benchmark.library;
+
+public class Native {
+ static { System.loadLibrary("mediabenchmark_jni"); }
+
+ public native int Extract(String inputFilePath, String inputFileName, String statsFile);
+
+ public native int Mux(String inputFilePath, String inputFileName, String outputFilePath,
+ String statsFile, String format);
+
+ public native int Decode(String inputFilePath, String inputFileName, String statsFile,
+ String codecName, boolean asyncMode);
+
+ public native int Encode(String inputFilePath, String inputFileName, String outputFilePath,
+ String statsFile, String codecName);
+}
diff --git a/media/tests/benchmark/MediaBenchmarkTest/src/main/java/com/android/media/benchmark/library/Stats.java b/media/tests/benchmark/MediaBenchmarkTest/src/main/java/com/android/media/benchmark/library/Stats.java
new file mode 100644
index 0000000..7245a3a
--- /dev/null
+++ b/media/tests/benchmark/MediaBenchmarkTest/src/main/java/com/android/media/benchmark/library/Stats.java
@@ -0,0 +1,178 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package com.android.media.benchmark.library;
+
+import android.util.Log;
+
+import java.io.File;
+import java.io.FileDescriptor;
+import java.io.FileOutputStream;
+import java.io.IOException;
+import java.util.ArrayList;
+
+/**
+ * Measures Performance.
+ */
+public class Stats {
+ private static final String TAG = "Stats";
+ private long mInitTimeNs;
+ private long mDeInitTimeNs;
+ private long mStartTimeNs;
+ private ArrayList<Integer> mFrameSizes;
+ private ArrayList<Long> mInputTimer;
+ private ArrayList<Long> mOutputTimer;
+
+ public Stats() {
+ mFrameSizes = new ArrayList<>();
+ mInputTimer = new ArrayList<>();
+ mOutputTimer = new ArrayList<>();
+ mInitTimeNs = 0;
+ mDeInitTimeNs = 0;
+ }
+
+ public long getCurTime() { return System.nanoTime(); }
+
+ public void setInitTime(long initTime) { mInitTimeNs = initTime; }
+
+ public void setDeInitTime(long deInitTime) { mDeInitTimeNs = deInitTime; }
+
+ public void setStartTime() { mStartTimeNs = System.nanoTime(); }
+
+ public void addFrameSize(int size) { mFrameSizes.add(size); }
+
+ public void addInputTime() { mInputTimer.add(System.nanoTime()); }
+
+ public void addOutputTime() { mOutputTimer.add(System.nanoTime()); }
+
+ public void reset() {
+ if (mFrameSizes.size() != 0) {
+ mFrameSizes.clear();
+ }
+
+ if (mInputTimer.size() != 0) {
+ mInputTimer.clear();
+ }
+
+ if (mOutputTimer.size() != 0) {
+ mOutputTimer.clear();
+ }
+ }
+
+ public long getInitTime() { return mInitTimeNs; }
+
+ public long getDeInitTime() { return mDeInitTimeNs; }
+
+ public long getTimeDiff(long sTime, long eTime) { return (eTime - sTime); }
+
+ private long getTotalTime() {
+ if (mOutputTimer.size() == 0) {
+ return -1;
+ }
+ long lastTime = mOutputTimer.get(mOutputTimer.size() - 1);
+ return lastTime - mStartTimeNs;
+ }
+
+ private long getTotalSize() {
+ long totalSize = 0;
+ for (long size : mFrameSizes) {
+ totalSize += size;
+ }
+ return totalSize;
+ }
+
+ /**
+ * Writes the stats header to a file
+ * <p>
+ * \param statsFile file where the stats data is to be written
+ **/
+ public boolean writeStatsHeader(String statsFile) throws IOException {
+ File outputFile = new File(statsFile);
+ FileOutputStream out = new FileOutputStream(outputFile, true);
+ if (!outputFile.exists())
+ return false;
+ String statsHeader =
+ "currentTime, fileName, operation, componentName, NDK/SDK, sync/async, setupTime, "
+ + "destroyTime, minimumTime, maximumTime, "
+ + "averageTime, timeToProcess1SecContent, totalBytesProcessedPerSec, "
+ + "timeToFirstFrame, totalSizeInBytes, totalTime\n";
+ out.write(statsHeader.getBytes());
+ out.close();
+ return true;
+ }
+
+ /**
+ * Dumps the stats of the operation for a given input media.
+ * <p>
+ * \param inputReference input media
+ * \param operation describes the operation performed on the input media
+ * (i.e. extract/mux/decode/encode)
+ * \param componentName name of the codec/muxFormat/mime
+ * \param mode the operating mode: sync/async.
+ * \param durationUs is a duration of the input media in microseconds.
+ * \param statsFile the file where the stats data is to be written.
+ */
+ public void dumpStatistics(String inputReference, String operation, String componentName,
+ String mode, long durationUs, String statsFile) throws IOException {
+ if (mOutputTimer.size() == 0) {
+ Log.e(TAG, "No output produced");
+ return;
+ }
+ long totalTimeTakenNs = getTotalTime();
+ long timeTakenPerSec = (totalTimeTakenNs * 1000000) / durationUs;
+ long timeToFirstFrameNs = mOutputTimer.get(0) - mStartTimeNs;
+ long size = getTotalSize();
+ // get min and max output intervals.
+ long intervalNs;
+ long minTimeTakenNs = Long.MAX_VALUE;
+ long maxTimeTakenNs = 0;
+ long prevIntervalNs = mStartTimeNs;
+ for (int idx = 0; idx < mOutputTimer.size() - 1; idx++) {
+ intervalNs = mOutputTimer.get(idx) - prevIntervalNs;
+ prevIntervalNs = mOutputTimer.get(idx);
+ if (minTimeTakenNs > intervalNs) {
+ minTimeTakenNs = intervalNs;
+ } else if (maxTimeTakenNs < intervalNs) {
+ maxTimeTakenNs = intervalNs;
+ }
+ }
+
+ // Write the stats row data to file
+ String rowData = "";
+ rowData += System.nanoTime() + ", ";
+ rowData += inputReference + ", ";
+ rowData += operation + ", ";
+ rowData += componentName + ", ";
+ rowData += "SDK, ";
+ rowData += mode + ", ";
+ rowData += mInitTimeNs + ", ";
+ rowData += mDeInitTimeNs + ", ";
+ rowData += minTimeTakenNs + ", ";
+ rowData += maxTimeTakenNs + ", ";
+ rowData += totalTimeTakenNs / mOutputTimer.size() + ", ";
+ rowData += timeTakenPerSec + ", ";
+ rowData += (size * 1000000000) / totalTimeTakenNs + ", ";
+ rowData += timeToFirstFrameNs + ", ";
+ rowData += size + ", ";
+ rowData += totalTimeTakenNs + "\n";
+
+ File outputFile = new File(statsFile);
+ FileOutputStream out = new FileOutputStream(outputFile, true);
+ assert outputFile.exists() : "Failed to open the stats file for writing!";
+ out.write(rowData.getBytes());
+ out.close();
+ }
+}
diff --git a/media/tests/benchmark/README.md b/media/tests/benchmark/README.md
new file mode 100644
index 0000000..05fbe6f
--- /dev/null
+++ b/media/tests/benchmark/README.md
@@ -0,0 +1,156 @@
+# Benchmark tests
+
+Benchmark app analyses the time taken by MediaCodec, MediaExtractor and MediaMuxer for given set of inputs. It is used to benchmark these modules on android devices.
+Benchmark results are emitted to logcat.
+
+This page describes steps to run the NDK and SDK layer test.
+
+Run the following steps to build the test suite:
+```
+mmm frameworks/av/media/tests/benchmark/
+```
+
+# NDK
+
+To run the test suite for measuring performance of the native layer, follow the following steps:
+
+The binaries will be created in the following path : $OUT/data/nativetest64/
+
+adb push $OUT/data/nativetest64/* /data/local/tmp/
+
+Eg. adb push $OUT/data/nativetest64/extractorTest/extractorTest /data/local/tmp/
+
+To run the binary, follow the commands mentioned below under each module.
+
+The resource file for the tests is taken from [here](https://drive.google.com/open?id=1ghMr17BBJ7n0pqbm7oREiTN_MNemJUqy)
+
+Download the MediaBenchmark.zip file, unzip and push it to /data/local/tmp/ on the device.
+
+```
+unzip MediaBenchmark.zip
+adb push MediaBenchmark /data/local/tmp
+```
+
+## Extractor
+
+The test extracts elementary stream and benchmarks the extractors available in NDK.
+
+The resource files are assumed to be at /data/local/tmp/MediaBenchmark/res/. You can use a different location, but you have to modify the rest of the instructions to replace /data/local/tmp/MediaBenchmark/res/ with wherever you chose to put the files.
+
+The path to these files on the device is required to be given for the test.
+
+```
+adb shell /data/local/tmp/extractorTest -P /data/local/tmp/MediaBenchmark/res/
+```
+
+## Decoder
+
+The test decodes input stream and benchmarks the decoders available in NDK.
+
+Setup steps are same as extractor.
+
+```
+adb shell /data/local/tmp/decoderTest -P /data/local/tmp/MediaBenchmark/res/
+```
+
+## Muxer
+
+The test muxes elementary stream and benchmarks the muxers available in NDK.
+
+Setup steps are same as extractor.
+
+```
+adb shell /data/local/tmp/muxerTest -P /data/local/tmp/MediaBenchmark/res/
+```
+
+## Encoder
+
+The test encodes input stream and benchmarks the encoders available in NDK.
+
+Setup steps are same as extractor.
+
+```
+adb shell /data/local/tmp/encoderTest -P /data/local/tmp/MediaBenchmark/res/
+```
+
+# SDK
+
+To run the test suite for measuring performance of the SDK APIs, follow the following steps:
+
+The apk will be created at the following path:
+$OUT/testcases/MediaBenchmarkTest/arm64/
+
+To get the resorce files for the test follow instructions given in [NDK](#NDK)
+
+For installing the apk, run the command:
+```
+adb install -f -r $OUT/testcases/MediaBenchmarkTest/arm64/MediaBenchmarkTest.apk
+```
+
+For running all the tests, run the command:
+```
+adb shell am instrument -w -r -e package com.android.media.benchmark.tests com.android.media.benchmark/androidx.test.runner.AndroidJUnitRunner
+```
+
+## Extractor
+
+The test extracts elementary stream and benchmarks the extractors available in SDK.
+```
+adb shell am instrument -w -r -e class 'com.android.media.benchmark.tests.ExtractorTest' com.android.media.benchmark/androidx.test.runner.AndroidJUnitRunner
+```
+
+## Decoder
+
+The test decodes input stream and benchmarks the decoders available in SDK.
+```
+adb shell am instrument -w -r -e class 'com.android.media.benchmark.tests.DecoderTest' com.android.media.benchmark/androidx.test.runner.AndroidJUnitRunner
+```
+
+## Muxer
+
+The test muxes elementary stream and benchmarks different writers available in SDK.
+```
+adb shell am instrument -w -r -e class 'com.android.media.benchmark.tests.MuxerTest' com.android.media.benchmark/androidx.test.runner.AndroidJUnitRunner
+```
+
+## Encoder
+
+The test encodes input stream and benchmarks the encoders available in SDK.
+```
+adb shell am instrument -w -r -e class 'com.android.media.benchmark.tests.EncoderTest' com.android.media.benchmark/androidx.test.runner.AndroidJUnitRunner
+```
+
+# Codec2
+To run the test suite for measuring performance of the codec2 layer, follow the following steps:
+
+The 32-bit binaries will be created in the following path : ${OUT}/data/nativetest/
+The 64-bit binaries will be created in the following path : ${OUT}/data/nativetest64/
+
+To test 64-bit binary push binaries from nativetest64.
+adb push $(OUT)/data/nativetest64/* /data/local/tmp/
+Eg. adb push $(OUT)/data/nativetest64/C2DecoderTest/C2DecoderTest /data/local/tmp/
+
+To test 32-bit binary push binaries from nativetest.
+adb push $(OUT)/data/nativetest/* /data/local/tmp/
+Eg. adb push $(OUT)/data/nativetest/C2DecoderTest/C2DecoderTest /data/local/tmp/
+
+To get the resource files for the test follow instructions given in [NDK](#NDK)
+
+## C2 Decoder
+
+The test decodes input stream and benchmarks the codec2 decoders available in device.
+
+Setup steps are same as [extractor](#extractor).
+
+```
+adb shell /data/local/tmp/C2DecoderTest -P /data/local/tmp/MediaBenchmark/res/
+```
+## C2 Encoder
+
+The test encodes input stream and benchmarks the codec2 encoders available in device.
+
+Setup steps are same as [extractor](#extractor).
+
+```
+adb shell /data/local/tmp/C2EncoderTest -P /data/local/tmp/MediaBenchmark/res/
+```
diff --git a/media/tests/benchmark/src/native/common/Android.bp b/media/tests/benchmark/src/native/common/Android.bp
new file mode 100644
index 0000000..d4389da
--- /dev/null
+++ b/media/tests/benchmark/src/native/common/Android.bp
@@ -0,0 +1,117 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+cc_library_static {
+ name: "libmediabenchmark_common",
+ defaults: [
+ "libmediabenchmark-defaults",
+ "libmediabenchmark_soft_sanitize_all-defaults",
+ ],
+
+ srcs: [
+ "BenchmarkCommon.cpp",
+ "Stats.cpp",
+ "utils/Timers.cpp",
+ ],
+
+ export_include_dirs: ["."],
+
+ ldflags: ["-Wl,-Bsymbolic"],
+}
+
+cc_defaults {
+ name: "libmediabenchmark_common-defaults",
+
+ defaults: [
+ "libmediabenchmark-defaults",
+ ],
+
+ static_libs: [
+ "libmediabenchmark_common",
+ ],
+}
+
+cc_defaults {
+ name: "libmediabenchmark-defaults",
+ sdk_version: "current",
+ stl: "c++_shared",
+
+ shared_libs: [
+ "libmediandk",
+ "liblog",
+ ],
+
+ cflags: [
+ "-Wall",
+ "-Werror",
+ ],
+}
+
+cc_library_static {
+ name: "libmediabenchmark_codec2_common",
+ defaults: [
+ "libmediabenchmark_codec2_common-defaults",
+ ],
+
+ srcs: [
+ "BenchmarkC2Common.cpp",
+ "BenchmarkCommon.cpp",
+ "Stats.cpp",
+ "utils/Timers.cpp",
+ ],
+
+ export_include_dirs: ["."],
+
+ ldflags: ["-Wl,-Bsymbolic"],
+}
+
+cc_defaults {
+ name: "libmediabenchmark_codec2_common-defaults",
+
+ defaults: [
+ "libcodec2-hidl-client-defaults",
+ "libmediabenchmark_soft_sanitize_all-defaults",
+ ],
+
+ include_dirs: [
+ "frameworks/av/media/codec2/hidl/client/include",
+ ],
+
+ shared_libs: [
+ "libcodec2_client",
+ "libmediandk",
+ "liblog",
+ ],
+
+ cflags: [
+ "-Wall",
+ "-Werror",
+ ],
+}
+
+// public dependency for native implementation
+// to be used by code under media/benchmark/* only
+cc_defaults {
+ name: "libmediabenchmark_soft_sanitize_all-defaults",
+
+ sanitize: {
+ misc_undefined: [
+ "unsigned-integer-overflow",
+ "signed-integer-overflow",
+ ],
+ cfi: true,
+ },
+}
diff --git a/media/tests/benchmark/src/native/common/BenchmarkC2Common.cpp b/media/tests/benchmark/src/native/common/BenchmarkC2Common.cpp
new file mode 100644
index 0000000..e09f468
--- /dev/null
+++ b/media/tests/benchmark/src/native/common/BenchmarkC2Common.cpp
@@ -0,0 +1,113 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "BenchmarkC2Common"
+
+#include "BenchmarkC2Common.h"
+
+int32_t BenchmarkC2Common::setupCodec2() {
+ ALOGV("In %s", __func__);
+ mClient = android::Codec2Client::CreateFromService("default");
+ if (!mClient) {
+ mClient = android::Codec2Client::CreateFromService("software");
+ }
+ if (!mClient) return -1;
+
+ std::shared_ptr<C2AllocatorStore> store = android::GetCodec2PlatformAllocatorStore();
+ if (!store) return -1;
+
+ c2_status_t status = store->fetchAllocator(C2AllocatorStore::DEFAULT_LINEAR, &mLinearAllocator);
+ if (status != C2_OK) return status;
+
+ mLinearPool = std::make_shared<C2PooledBlockPool>(mLinearAllocator, mBlockPoolId++);
+ if (!mLinearPool) return -1;
+
+ status = store->fetchAllocator(C2AllocatorStore::DEFAULT_GRAPHIC, &mGraphicAllocator);
+ if (status != C2_OK) return status;
+
+ mGraphicPool = std::make_shared<C2PooledBlockPool>(mGraphicAllocator, mBlockPoolId++);
+ if (!mGraphicPool) return -1;
+
+ for (int i = 0; i < MAX_INPUT_BUFFERS; ++i) {
+ mWorkQueue.emplace_back(new C2Work);
+ }
+ if (!mStats) mStats = new Stats();
+
+ return status;
+}
+
+vector<string> BenchmarkC2Common::getSupportedComponentList(bool isEncoder) {
+ // Get List of components from all known services
+ vector<string> codecList;
+ const std::vector<C2Component::Traits> listTraits = mClient->ListComponents();
+ if (listTraits.size() == 0)
+ ALOGE("ComponentInfo list empty.");
+ else {
+ for (size_t i = 0; i < listTraits.size(); i++) {
+ if (isEncoder && C2Component::KIND_ENCODER == listTraits[i].kind) {
+ codecList.push_back(listTraits[i].name);
+ } else if (!isEncoder && C2Component::KIND_DECODER == listTraits[i].kind) {
+ codecList.push_back(listTraits[i].name);
+ }
+ }
+ }
+ return codecList;
+}
+
+void BenchmarkC2Common::waitOnInputConsumption() {
+ typedef std::unique_lock<std::mutex> ULock;
+ uint32_t queueSize;
+ uint32_t maxRetry = 0;
+ {
+ ULock l(mQueueLock);
+ queueSize = mWorkQueue.size();
+ }
+ while ((maxRetry < MAX_RETRY) && (queueSize < MAX_INPUT_BUFFERS)) {
+ ULock l(mQueueLock);
+ if (queueSize != mWorkQueue.size()) {
+ queueSize = mWorkQueue.size();
+ maxRetry = 0;
+ } else {
+ mQueueCondition.wait_for(l, TIME_OUT);
+ maxRetry++;
+ }
+ }
+}
+
+void BenchmarkC2Common::handleWorkDone(std::list<std::unique_ptr<C2Work>> &workItems) {
+ ALOGV("In %s", __func__);
+ mStats->addOutputTime();
+ for (std::unique_ptr<C2Work> &work : workItems) {
+ if (!work->worklets.empty()) {
+ if (work->worklets.front()->output.flags != C2FrameData::FLAG_INCOMPLETE) {
+ mEos = (work->worklets.front()->output.flags & C2FrameData::FLAG_END_OF_STREAM) !=
+ 0;
+ ALOGV("WorkDone: frameID received %d , mEos : %d",
+ (int)work->worklets.front()->output.ordinal.frameIndex.peeku(), mEos);
+ work->input.buffers.clear();
+ work->worklets.clear();
+ {
+ typedef std::unique_lock<std::mutex> ULock;
+ ULock l(mQueueLock);
+ mWorkQueue.push_back(std::move(work));
+ mQueueCondition.notify_all();
+ }
+ }
+ }
+ }
+}
+
diff --git a/media/tests/benchmark/src/native/common/BenchmarkC2Common.h b/media/tests/benchmark/src/native/common/BenchmarkC2Common.h
new file mode 100644
index 0000000..d67758a
--- /dev/null
+++ b/media/tests/benchmark/src/native/common/BenchmarkC2Common.h
@@ -0,0 +1,141 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __BENCHMARK_C2_COMMON_H__
+#define __BENCHMARK_C2_COMMON_H__
+
+#include "codec2/hidl/client.h"
+
+#include <C2Component.h>
+#include <C2Config.h>
+
+#include <hidl/HidlSupport.h>
+
+#include <C2AllocatorIon.h>
+#include <C2Buffer.h>
+#include <C2BufferPriv.h>
+
+#include "BenchmarkCommon.h"
+
+#define MAX_RETRY 20
+#define TIME_OUT 400ms
+#define MAX_INPUT_BUFFERS 8
+
+using android::C2AllocatorIon;
+
+class LinearBuffer : public C2Buffer {
+ public:
+ explicit LinearBuffer(const std::shared_ptr<C2LinearBlock> &block)
+ : C2Buffer({block->share(block->offset(), block->size(), ::C2Fence())}) {}
+
+ explicit LinearBuffer(const std::shared_ptr<C2LinearBlock> &block, size_t size)
+ : C2Buffer({block->share(block->offset(), size, ::C2Fence())}) {}
+};
+
+class GraphicBuffer : public C2Buffer {
+ public:
+ explicit GraphicBuffer(const std::shared_ptr<C2GraphicBlock> &block)
+ : C2Buffer({block->share(C2Rect(block->width(), block->height()), ::C2Fence())}) {}
+};
+
+/**
+ * Handle Callback functions onWorkDone(), onTripped(),
+ * onError(), onDeath(), onFramesRendered() for C2 Components
+ */
+struct CodecListener : public android::Codec2Client::Listener {
+ public:
+ CodecListener(
+ const std::function<void(std::list<std::unique_ptr<C2Work>> &workItems)> fn = nullptr)
+ : callBack(fn) {}
+ virtual void onWorkDone(const std::weak_ptr<android::Codec2Client::Component> &comp,
+ std::list<std::unique_ptr<C2Work>> &workItems) override {
+ ALOGV("onWorkDone called");
+ (void)comp;
+ if (callBack) callBack(workItems);
+ }
+
+ virtual void onTripped(
+ const std::weak_ptr<android::Codec2Client::Component> &comp,
+ const std::vector<std::shared_ptr<C2SettingResult>> &settingResults) override {
+ (void)comp;
+ (void)settingResults;
+ }
+
+ virtual void onError(const std::weak_ptr<android::Codec2Client::Component> &comp,
+ uint32_t errorCode) override {
+ (void)comp;
+ ALOGV("onError called");
+ if (errorCode != 0) ALOGE("Error : %u", errorCode);
+ }
+
+ virtual void onDeath(const std::weak_ptr<android::Codec2Client::Component> &comp) override {
+ (void)comp;
+ }
+
+ virtual void onInputBufferDone(uint64_t frameIndex, size_t arrayIndex) override {
+ (void)frameIndex;
+ (void)arrayIndex;
+ }
+
+ virtual void onFrameRendered(uint64_t bufferQueueId, int32_t slotId,
+ int64_t timestampNs) override {
+ (void)bufferQueueId;
+ (void)slotId;
+ (void)timestampNs;
+ }
+
+ std::function<void(std::list<std::unique_ptr<C2Work>> &workItems)> callBack;
+};
+
+class BenchmarkC2Common {
+ public:
+ BenchmarkC2Common()
+ : mEos(false),
+ mStats(nullptr),
+ mClient(nullptr),
+ mBlockPoolId(0),
+ mLinearPool(nullptr),
+ mGraphicPool(nullptr),
+ mLinearAllocator(nullptr),
+ mGraphicAllocator(nullptr) {}
+
+ int32_t setupCodec2();
+
+ vector<string> getSupportedComponentList(bool isEncoder);
+
+ void waitOnInputConsumption();
+
+ // callback function to process onWorkDone received by Listener
+ void handleWorkDone(std::list<std::unique_ptr<C2Work>> &workItems);
+
+ bool mEos;
+ protected:
+ Stats *mStats;
+
+ std::shared_ptr<android::Codec2Client> mClient;
+
+ C2BlockPool::local_id_t mBlockPoolId;
+ std::shared_ptr<C2BlockPool> mLinearPool;
+ std::shared_ptr<C2BlockPool> mGraphicPool;
+ std::shared_ptr<C2Allocator> mLinearAllocator;
+ std::shared_ptr<C2Allocator> mGraphicAllocator;
+
+ std::mutex mQueueLock;
+ std::condition_variable mQueueCondition;
+ std::list<std::unique_ptr<C2Work>> mWorkQueue;
+};
+
+#endif // __BENCHMARK_C2_COMMON_H__
diff --git a/media/tests/benchmark/src/native/common/BenchmarkCommon.cpp b/media/tests/benchmark/src/native/common/BenchmarkCommon.cpp
new file mode 100644
index 0000000..cb49b8e
--- /dev/null
+++ b/media/tests/benchmark/src/native/common/BenchmarkCommon.cpp
@@ -0,0 +1,103 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "BenchmarkCommon"
+
+#include "BenchmarkCommon.h"
+#include <iostream>
+
+void CallBackHandle::ioThread() {
+ ALOGV("In %s mIsDone : %d, mSawError : %d ", __func__, mIsDone, mSawError);
+ while (!mIsDone && !mSawError) {
+ auto task = mIOQueue.pop();
+ task();
+ }
+}
+
+void OnInputAvailableCB(AMediaCodec *codec, void *userdata, int32_t index) {
+ ALOGV("OnInputAvailableCB: index(%d)", index);
+ CallBackHandle *self = (CallBackHandle *)userdata;
+ self->getStats()->addInputTime();
+ self->mIOQueue.push([self, codec, index]() { self->onInputAvailable(codec, index); });
+}
+
+void OnOutputAvailableCB(AMediaCodec *codec, void *userdata, int32_t index,
+ AMediaCodecBufferInfo *bufferInfo) {
+ ALOGV("OnOutputAvailableCB: index(%d), (%d, %d, %lld, 0x%x)", index, bufferInfo->offset,
+ bufferInfo->size, (long long)bufferInfo->presentationTimeUs, bufferInfo->flags);
+ CallBackHandle *self = (CallBackHandle *)userdata;
+ self->getStats()->addOutputTime();
+ AMediaCodecBufferInfo bufferInfoCopy = *bufferInfo;
+ self->mIOQueue.push([self, codec, index, bufferInfoCopy]() {
+ AMediaCodecBufferInfo bc = bufferInfoCopy;
+ self->onOutputAvailable(codec, index, &bc);
+ });
+}
+
+void OnFormatChangedCB(AMediaCodec *codec, void *userdata, AMediaFormat *format) {
+ ALOGV("OnFormatChangedCB: format(%s)", AMediaFormat_toString(format));
+ CallBackHandle *self = (CallBackHandle *)userdata;
+ self->mIOQueue.push([self, codec, format]() { self->onFormatChanged(codec, format); });
+}
+
+void OnErrorCB(AMediaCodec *codec, void *userdata, media_status_t err, int32_t actionCode,
+ const char *detail) {
+ (void)codec;
+ ALOGE("OnErrorCB: err(%d), actionCode(%d), detail(%s)", err, actionCode, detail);
+ CallBackHandle *self = (CallBackHandle *)userdata;
+ self->mSawError = true;
+ self->mIOQueue.push([self, codec, err]() { self->onError(codec, err); });
+}
+
+AMediaCodec *createMediaCodec(AMediaFormat *format, const char *mime, string codecName,
+ bool isEncoder) {
+ ALOGV("In %s", __func__);
+ if (!mime) {
+ ALOGE("Please specify a mime type to create codec");
+ return nullptr;
+ }
+
+ AMediaCodec *codec;
+ if (!codecName.empty()) {
+ codec = AMediaCodec_createCodecByName(codecName.c_str());
+ if (!codec) {
+ ALOGE("Unable to create codec by name: %s", codecName.c_str());
+ return nullptr;
+ }
+ } else {
+ if (isEncoder) {
+ codec = AMediaCodec_createEncoderByType(mime);
+ } else {
+ codec = AMediaCodec_createDecoderByType(mime);
+ }
+ if (!codec) {
+ ALOGE("Unable to create codec by mime: %s", mime);
+ return nullptr;
+ }
+ }
+
+ /* Configure codec with the given format*/
+ const char *s = AMediaFormat_toString(format);
+ ALOGI("Input format: %s\n", s);
+
+ media_status_t status = AMediaCodec_configure(codec, format, nullptr, nullptr, isEncoder);
+ if (status != AMEDIA_OK) {
+ ALOGE("AMediaCodec_configure failed %d", status);
+ return nullptr;
+ }
+ return codec;
+}
diff --git a/media/tests/benchmark/src/native/common/BenchmarkCommon.h b/media/tests/benchmark/src/native/common/BenchmarkCommon.h
new file mode 100644
index 0000000..40a8c9e
--- /dev/null
+++ b/media/tests/benchmark/src/native/common/BenchmarkCommon.h
@@ -0,0 +1,135 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __BENCHMARK_COMMON_H__
+#define __BENCHMARK_COMMON_H__
+
+#include <sys/stat.h>
+#include <inttypes.h>
+#include <mutex>
+#include <queue>
+#include <thread>
+#include <iostream>
+
+#include <media/NdkMediaCodec.h>
+#include <media/NdkMediaError.h>
+
+#include "Stats.h"
+#define UNUSED(x) (void)(x)
+
+using namespace std;
+
+constexpr uint32_t kQueueDequeueTimeoutUs = 1000;
+constexpr uint32_t kMaxCSDStrlen = 16;
+constexpr uint32_t kMaxBufferSize = 1024 * 1024 * 16;
+// Change in kDefaultAudioEncodeFrameSize should also be taken to
+// AUDIO_ENCODE_DEFAULT_MAX_INPUT_SIZE present in Encoder.java
+constexpr uint32_t kDefaultAudioEncodeFrameSize = 4096;
+
+template <typename T>
+class CallBackQueue {
+ public:
+ CallBackQueue() {}
+ ~CallBackQueue() {}
+
+ void push(T elem) {
+ bool needsNotify = false;
+ {
+ lock_guard<mutex> lock(mMutex);
+ needsNotify = mQueue.empty();
+ mQueue.push(move(elem));
+ }
+ if (needsNotify) mQueueNotEmptyCondition.notify_one();
+ }
+
+ T pop() {
+ unique_lock<mutex> lock(mMutex);
+ if (mQueue.empty()) {
+ mQueueNotEmptyCondition.wait(lock, [this]() { return !mQueue.empty(); });
+ }
+ auto result = mQueue.front();
+ mQueue.pop();
+ return result;
+ }
+
+ private:
+ mutex mMutex;
+ queue<T> mQueue;
+ condition_variable mQueueNotEmptyCondition;
+};
+
+class CallBackHandle {
+ public:
+ CallBackHandle() : mSawError(false), mIsDone(false), mStats(nullptr) {
+ mStats = new Stats();
+ }
+
+ virtual ~CallBackHandle() {
+ if (mIOThread.joinable()) mIOThread.join();
+ if (mStats) delete mStats;
+ }
+
+ void ioThread();
+
+ // Implementation in child class (Decoder/Encoder)
+ virtual void onInputAvailable(AMediaCodec *codec, int32_t index) {
+ (void)codec;
+ (void)index;
+ }
+ virtual void onFormatChanged(AMediaCodec *codec, AMediaFormat *format) {
+ (void)codec;
+ (void)format;
+ }
+ virtual void onError(AMediaCodec *codec, media_status_t err) {
+ (void)codec;
+ (void)err;
+ }
+ virtual void onOutputAvailable(AMediaCodec *codec, int32_t index,
+ AMediaCodecBufferInfo *bufferInfo) {
+ (void)codec;
+ (void)index;
+ (void)bufferInfo;
+ }
+
+ Stats *getStats() { return mStats; }
+
+ // Keep a queue of all function callbacks.
+ typedef function<void()> IOTask;
+ CallBackQueue<IOTask> mIOQueue;
+ thread mIOThread;
+ bool mSawError;
+ bool mIsDone;
+
+ protected:
+ Stats *mStats;
+};
+
+// Async API's callback
+void OnInputAvailableCB(AMediaCodec *codec, void *userdata, int32_t index);
+
+void OnOutputAvailableCB(AMediaCodec *codec, void *userdata, int32_t index,
+ AMediaCodecBufferInfo *bufferInfo);
+
+void OnFormatChangedCB(AMediaCodec *codec, void *userdata, AMediaFormat *format);
+
+void OnErrorCB(AMediaCodec *codec, void * /* userdata */, media_status_t err, int32_t actionCode,
+ const char *detail);
+
+// Utility to create and configure AMediaCodec
+AMediaCodec *createMediaCodec(AMediaFormat *format, const char *mime, string codecName,
+ bool isEncoder);
+
+#endif // __BENCHMARK_COMMON_H__
diff --git a/media/tests/benchmark/src/native/common/Stats.cpp b/media/tests/benchmark/src/native/common/Stats.cpp
new file mode 100644
index 0000000..bfde125
--- /dev/null
+++ b/media/tests/benchmark/src/native/common/Stats.cpp
@@ -0,0 +1,89 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "Stats"
+
+#include <ctime>
+#include <iostream>
+#include <stdint.h>
+#include <fstream>
+
+#include "Stats.h"
+
+/**
+ * Dumps the stats of the operation for a given input media.
+ *
+ * \param operation describes the operation performed on the input media
+ * (i.e. extract/mux/decode/encode)
+ * \param inputReference input media
+ * \param durationUs is a duration of the input media in microseconds.
+ * \param componentName describes the codecName/muxFormat/mimeType.
+ * \param mode the operating mode: sync/async.
+ * \param statsFile the file where the stats data is to be written.
+ */
+void Stats::dumpStatistics(string operation, string inputReference, int64_t durationUs,
+ string componentName, string mode, string statsFile) {
+ ALOGV("In %s", __func__);
+ if (!mOutputTimer.size()) {
+ ALOGE("No output produced");
+ return;
+ }
+ nsecs_t totalTimeTakenNs = getTotalTime();
+ nsecs_t timeTakenPerSec = (totalTimeTakenNs * 1000000) / durationUs;
+ nsecs_t timeToFirstFrameNs = *mOutputTimer.begin() - mStartTimeNs;
+ int32_t size = std::accumulate(mFrameSizes.begin(), mFrameSizes.end(), 0);
+ // get min and max output intervals.
+ nsecs_t intervalNs;
+ nsecs_t minTimeTakenNs = INT64_MAX;
+ nsecs_t maxTimeTakenNs = 0;
+ nsecs_t prevIntervalNs = mStartTimeNs;
+ for (int32_t idx = 0; idx < mOutputTimer.size() - 1; idx++) {
+ intervalNs = mOutputTimer.at(idx) - prevIntervalNs;
+ prevIntervalNs = mOutputTimer.at(idx);
+ if (minTimeTakenNs > intervalNs) minTimeTakenNs = intervalNs;
+ else if (maxTimeTakenNs < intervalNs) maxTimeTakenNs = intervalNs;
+ }
+
+ // Write the stats data to file.
+ int64_t dataSize = size;
+ int64_t bytesPerSec = ((int64_t)dataSize * 1000000000) / totalTimeTakenNs;
+ string rowData = "";
+ rowData.append(to_string(systemTime(CLOCK_MONOTONIC)) + ", ");
+ rowData.append(inputReference + ", ");
+ rowData.append(operation + ", ");
+ rowData.append(componentName + ", ");
+ rowData.append("NDK, ");
+ rowData.append(mode + ", ");
+ rowData.append(to_string(mInitTimeNs) + ", ");
+ rowData.append(to_string(mDeInitTimeNs) + ", ");
+ rowData.append(to_string(minTimeTakenNs) + ", ");
+ rowData.append(to_string(maxTimeTakenNs) + ", ");
+ rowData.append(to_string(totalTimeTakenNs / mOutputTimer.size()) + ", ");
+ rowData.append(to_string(timeTakenPerSec) + ", ");
+ rowData.append(to_string(bytesPerSec) + ", ");
+ rowData.append(to_string(timeToFirstFrameNs) + ", ");
+ rowData.append(to_string(size) + ",");
+ rowData.append(to_string(totalTimeTakenNs) + ",\n");
+
+ ofstream out(statsFile, ios::out | ios::app);
+ if(out.bad()) {
+ ALOGE("Failed to open stats file for writing!");
+ return;
+ }
+ out << rowData;
+ out.close();
+}
diff --git a/media/tests/benchmark/src/native/common/Stats.h b/media/tests/benchmark/src/native/common/Stats.h
new file mode 100644
index 0000000..18e4b06
--- /dev/null
+++ b/media/tests/benchmark/src/native/common/Stats.h
@@ -0,0 +1,109 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __STATS_H__
+#define __STATS_H__
+
+#include <android/log.h>
+#include <inttypes.h>
+
+#ifndef ALOG
+#define ALOG(priority, tag, ...) ((void)__android_log_print(ANDROID_##priority, tag, __VA_ARGS__))
+
+#define ALOGI(...) ALOG(LOG_INFO, LOG_TAG, __VA_ARGS__)
+#define ALOGE(...) ALOG(LOG_ERROR, LOG_TAG, __VA_ARGS__)
+#define ALOGD(...) ALOG(LOG_DEBUG, LOG_TAG, __VA_ARGS__)
+#define ALOGW(...) ALOG(LOG_WARN, LOG_TAG, __VA_ARGS__)
+
+#ifndef LOG_NDEBUG
+#define LOG_NDEBUG 1
+#endif
+
+#if LOG_NDEBUG
+#define ALOGV(cond, ...) ((void)0)
+#else
+#define ALOGV(...) ALOG(LOG_VERBOSE, LOG_TAG, __VA_ARGS__)
+#endif
+#endif // ALOG
+
+#include <sys/time.h>
+#include <algorithm>
+#include <numeric>
+#include <vector>
+
+// Include local copy of Timers taken from system/core/libutils
+#include "utils/Timers.h"
+
+using namespace std;
+
+class Stats {
+ public:
+ Stats() {
+ mInitTimeNs = 0;
+ mDeInitTimeNs = 0;
+ }
+
+ ~Stats() {
+ reset();
+ }
+
+ private:
+ nsecs_t mInitTimeNs;
+ nsecs_t mDeInitTimeNs;
+ nsecs_t mStartTimeNs;
+ std::vector<int32_t> mFrameSizes;
+ std::vector<nsecs_t> mInputTimer;
+ std::vector<nsecs_t> mOutputTimer;
+
+ public:
+ nsecs_t getCurTime() { return systemTime(CLOCK_MONOTONIC); }
+
+ void setInitTime(nsecs_t initTime) { mInitTimeNs = initTime; }
+
+ void setDeInitTime(nsecs_t deInitTime) { mDeInitTimeNs = deInitTime; }
+
+ void setStartTime() { mStartTimeNs = systemTime(CLOCK_MONOTONIC); }
+
+ void addFrameSize(int32_t size) { mFrameSizes.push_back(size); }
+
+ void addInputTime() { mInputTimer.push_back(systemTime(CLOCK_MONOTONIC)); }
+
+ void addOutputTime() { mOutputTimer.push_back(systemTime(CLOCK_MONOTONIC)); }
+
+ void reset() {
+ if (!mFrameSizes.empty()) mFrameSizes.clear();
+ if (!mInputTimer.empty()) mInputTimer.clear();
+ if (!mOutputTimer.empty()) mOutputTimer.clear();
+ }
+
+ std::vector<nsecs_t> getOutputTimer() { return mOutputTimer; }
+
+ nsecs_t getInitTime() { return mInitTimeNs; }
+
+ nsecs_t getDeInitTime() { return mDeInitTimeNs; }
+
+ nsecs_t getTimeDiff(nsecs_t sTime, nsecs_t eTime) { return (eTime - sTime); }
+
+ nsecs_t getTotalTime() {
+ if (mOutputTimer.empty()) return -1;
+ return (*(mOutputTimer.end() - 1) - mStartTimeNs);
+ }
+
+ void dumpStatistics(string operation, string inputReference, int64_t duarationUs,
+ string codecName = "", string mode = "", string statsFile = "");
+};
+
+#endif // __STATS_H__
diff --git a/media/tests/benchmark/src/native/common/utils/Timers.cpp b/media/tests/benchmark/src/native/common/utils/Timers.cpp
new file mode 100644
index 0000000..1acbdb3
--- /dev/null
+++ b/media/tests/benchmark/src/native/common/utils/Timers.cpp
@@ -0,0 +1,62 @@
+/*
+ * Copyright (C) 2005 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//
+// Timer functions.
+//
+
+#define LOG_TAG "Timers"
+
+#include <limits.h>
+#include <time.h>
+
+#include "Timers.h"
+
+#if defined(__ANDROID__)
+nsecs_t systemTime(int clock) {
+ static const clockid_t clocks[] = {CLOCK_REALTIME, CLOCK_MONOTONIC, CLOCK_PROCESS_CPUTIME_ID,
+ CLOCK_THREAD_CPUTIME_ID, CLOCK_BOOTTIME};
+ struct timespec t;
+ t.tv_sec = t.tv_nsec = 0;
+ clock_gettime(clocks[clock], &t);
+ return nsecs_t(t.tv_sec) * 1000000000LL + t.tv_nsec;
+}
+#else
+nsecs_t systemTime(int /*clock*/) {
+ // Clock support varies widely across hosts. Mac OS doesn't support
+ // posix clocks, older glibcs don't support CLOCK_BOOTTIME and Windows
+ // is windows.
+ struct timeval t;
+ t.tv_sec = t.tv_usec = 0;
+ gettimeofday(&t, NULL);
+ return nsecs_t(t.tv_sec) * 1000000000LL + nsecs_t(t.tv_usec) * 1000LL;
+}
+#endif
+
+int toMillisecondTimeoutDelay(nsecs_t referenceTime, nsecs_t timeoutTime) {
+ nsecs_t timeoutDelayMillis;
+ if (timeoutTime > referenceTime) {
+ uint64_t timeoutDelay = uint64_t(timeoutTime - referenceTime);
+ if (timeoutDelay > uint64_t((INT_MAX - 1) * 1000000LL)) {
+ timeoutDelayMillis = -1;
+ } else {
+ timeoutDelayMillis = (timeoutDelay + 999999LL) / 1000000LL;
+ }
+ } else {
+ timeoutDelayMillis = 0;
+ }
+ return (int)timeoutDelayMillis;
+}
diff --git a/media/tests/benchmark/src/native/common/utils/Timers.h b/media/tests/benchmark/src/native/common/utils/Timers.h
new file mode 100644
index 0000000..d643dcd
--- /dev/null
+++ b/media/tests/benchmark/src/native/common/utils/Timers.h
@@ -0,0 +1,119 @@
+/*
+ * Copyright (C) 2005 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//
+// Timer functions.
+//
+
+#ifndef _LIBS_UTILS_TIMERS_H
+#define _LIBS_UTILS_TIMERS_H
+
+#include <stdint.h>
+#include <sys/time.h>
+#include <sys/types.h>
+
+// ------------------------------------------------------------------
+// C API
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+typedef int64_t nsecs_t; // nano-seconds
+
+static inline nsecs_t seconds_to_nanoseconds(nsecs_t secs) {
+ return secs * 1000000000;
+}
+
+static inline nsecs_t milliseconds_to_nanoseconds(nsecs_t secs) {
+ return secs * 1000000;
+}
+
+static inline nsecs_t microseconds_to_nanoseconds(nsecs_t secs) {
+ return secs * 1000;
+}
+
+static inline nsecs_t nanoseconds_to_seconds(nsecs_t secs) {
+ return secs / 1000000000;
+}
+
+static inline nsecs_t nanoseconds_to_milliseconds(nsecs_t secs) {
+ return secs / 1000000;
+}
+
+static inline nsecs_t nanoseconds_to_microseconds(nsecs_t secs) {
+ return secs / 1000;
+}
+
+static inline nsecs_t s2ns(nsecs_t v) {
+ return seconds_to_nanoseconds(v);
+}
+static inline nsecs_t ms2ns(nsecs_t v) {
+ return milliseconds_to_nanoseconds(v);
+}
+static inline nsecs_t us2ns(nsecs_t v) {
+ return microseconds_to_nanoseconds(v);
+}
+static inline nsecs_t ns2s(nsecs_t v) {
+ return nanoseconds_to_seconds(v);
+}
+static inline nsecs_t ns2ms(nsecs_t v) {
+ return nanoseconds_to_milliseconds(v);
+}
+static inline nsecs_t ns2us(nsecs_t v) {
+ return nanoseconds_to_microseconds(v);
+}
+
+static inline nsecs_t seconds(nsecs_t v) {
+ return s2ns(v);
+}
+static inline nsecs_t milliseconds(nsecs_t v) {
+ return ms2ns(v);
+}
+static inline nsecs_t microseconds(nsecs_t v) {
+ return us2ns(v);
+}
+
+enum {
+ SYSTEM_TIME_REALTIME = 0, // system-wide realtime clock
+ SYSTEM_TIME_MONOTONIC = 1, // monotonic time since unspecified starting point
+ SYSTEM_TIME_PROCESS = 2, // high-resolution per-process clock
+ SYSTEM_TIME_THREAD = 3, // high-resolution per-thread clock
+ SYSTEM_TIME_BOOTTIME = 4 // same as SYSTEM_TIME_MONOTONIC, but including CPU suspend time
+};
+
+// return the system-time according to the specified clock
+#ifdef __cplusplus
+nsecs_t systemTime(int clock = SYSTEM_TIME_MONOTONIC);
+#else
+nsecs_t systemTime(int clock);
+#endif // def __cplusplus
+
+/**
+ * Returns the number of milliseconds to wait between the reference time and the timeout time.
+ * If the timeout is in the past relative to the reference time, returns 0.
+ * If the timeout is more than INT_MAX milliseconds in the future relative to the reference time,
+ * such as when timeoutTime == LLONG_MAX, returns -1 to indicate an infinite timeout delay.
+ * Otherwise, returns the difference between the reference time and timeout time
+ * rounded up to the next millisecond.
+ */
+int toMillisecondTimeoutDelay(nsecs_t referenceTime, nsecs_t timeoutTime);
+
+#ifdef __cplusplus
+} // extern "C"
+#endif
+
+#endif // _LIBS_UTILS_TIMERS_H
diff --git a/media/tests/benchmark/src/native/decoder/Android.bp b/media/tests/benchmark/src/native/decoder/Android.bp
new file mode 100644
index 0000000..9791c11
--- /dev/null
+++ b/media/tests/benchmark/src/native/decoder/Android.bp
@@ -0,0 +1,52 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+cc_library_static {
+ name: "libmediabenchmark_decoder",
+ defaults: [
+ "libmediabenchmark_common-defaults",
+ "libmediabenchmark_soft_sanitize_all-defaults",
+ ],
+
+ srcs: ["Decoder.cpp"],
+
+ static_libs: ["libmediabenchmark_extractor"],
+
+ export_include_dirs: ["."],
+
+ ldflags: ["-Wl,-Bsymbolic"],
+}
+
+cc_library_static {
+ name: "libmediabenchmark_codec2_decoder",
+ defaults: [
+ "libmediabenchmark_codec2_common-defaults",
+ ],
+
+ srcs: [
+ "C2Decoder.cpp",
+ "Decoder.cpp",
+ ],
+
+ static_libs: [
+ "libmediabenchmark_codec2_common",
+ "libmediabenchmark_codec2_extractor",
+ ],
+
+ export_include_dirs: ["."],
+
+ ldflags: ["-Wl,-Bsymbolic"],
+}
diff --git a/media/tests/benchmark/src/native/decoder/C2Decoder.cpp b/media/tests/benchmark/src/native/decoder/C2Decoder.cpp
new file mode 100644
index 0000000..e88d011
--- /dev/null
+++ b/media/tests/benchmark/src/native/decoder/C2Decoder.cpp
@@ -0,0 +1,166 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "C2Decoder"
+
+#include "C2Decoder.h"
+
+int32_t C2Decoder::createCodec2Component(string compName, AMediaFormat *format) {
+ ALOGV("In %s", __func__);
+ mListener.reset(new CodecListener(
+ [this](std::list<std::unique_ptr<C2Work>> &workItems) { handleWorkDone(workItems); }));
+ if (!mListener) return -1;
+
+ const char *mime = nullptr;
+ AMediaFormat_getString(format, AMEDIAFORMAT_KEY_MIME, &mime);
+ if (!mime) {
+ ALOGE("Error in AMediaFormat_getString");
+ return -1;
+ }
+ // Configure the plugin with Input properties
+ std::vector<C2Param *> configParam;
+ if (!strncmp(mime, "audio/", 6)) {
+ int32_t sampleRate, numChannels;
+ AMediaFormat_getInt32(format, AMEDIAFORMAT_KEY_SAMPLE_RATE, &sampleRate);
+ AMediaFormat_getInt32(format, AMEDIAFORMAT_KEY_CHANNEL_COUNT, &numChannels);
+ C2StreamSampleRateInfo::output sampleRateInfo(0u, sampleRate);
+ C2StreamChannelCountInfo::output channelCountInfo(0u, numChannels);
+ configParam.push_back(&sampleRateInfo);
+ configParam.push_back(&channelCountInfo);
+
+ } else {
+ int32_t width, height;
+ AMediaFormat_getInt32(format, AMEDIAFORMAT_KEY_WIDTH, &width);
+ AMediaFormat_getInt32(format, AMEDIAFORMAT_KEY_HEIGHT, &height);
+ C2StreamPictureSizeInfo::input inputSize(0u, width, height);
+ configParam.push_back(&inputSize);
+ }
+
+ int64_t sTime = mStats->getCurTime();
+ mComponent = mClient->CreateComponentByName(compName.c_str(), mListener, &mClient);
+ if (mComponent == nullptr) {
+ ALOGE("Create component failed for %s", compName.c_str());
+ return -1;
+ }
+ std::vector<std::unique_ptr<C2SettingResult>> failures;
+ int32_t status = mComponent->config(configParam, C2_DONT_BLOCK, &failures);
+ if (failures.size() != 0) {
+ ALOGE("Invalid Configuration");
+ return -1;
+ }
+
+ status |= mComponent->start();
+ int64_t eTime = mStats->getCurTime();
+ int64_t timeTaken = mStats->getTimeDiff(sTime, eTime);
+ mStats->setInitTime(timeTaken);
+ return status;
+}
+
+int32_t C2Decoder::decodeFrames(uint8_t *inputBuffer, vector<AMediaCodecBufferInfo> &frameInfo) {
+ ALOGV("In %s", __func__);
+ typedef std::unique_lock<std::mutex> ULock;
+ c2_status_t status = C2_OK;
+ mStats->setStartTime();
+ while (1) {
+ if (mNumInputFrame == frameInfo.size()) break;
+ std::unique_ptr<C2Work> work;
+ // Prepare C2Work
+ {
+ ULock l(mQueueLock);
+ if (mWorkQueue.empty()) mQueueCondition.wait_for(l, MAX_RETRY * TIME_OUT);
+ if (!mWorkQueue.empty()) {
+ mStats->addInputTime();
+ work.swap(mWorkQueue.front());
+ mWorkQueue.pop_front();
+ } else {
+ cout << "Wait for generating C2Work exceeded timeout" << endl;
+ return -1;
+ }
+ }
+
+ uint32_t flags = frameInfo[mNumInputFrame].flags;
+ if (flags == AMEDIACODEC_BUFFER_FLAG_CODEC_CONFIG) {
+ flags = C2FrameData::FLAG_CODEC_CONFIG;
+ }
+ if (mNumInputFrame == (frameInfo.size() - 1)) {
+ flags |= C2FrameData::FLAG_END_OF_STREAM;
+ }
+ work->input.flags = (C2FrameData::flags_t)flags;
+ work->input.ordinal.timestamp = frameInfo[mNumInputFrame].presentationTimeUs;
+ work->input.ordinal.frameIndex = mNumInputFrame;
+ work->input.buffers.clear();
+ int size = frameInfo[mNumInputFrame].size;
+ int alignedSize = ALIGN(size, PAGE_SIZE);
+ if (size) {
+ std::shared_ptr<C2LinearBlock> block;
+ status = mLinearPool->fetchLinearBlock(
+ alignedSize, {C2MemoryUsage::CPU_READ, C2MemoryUsage::CPU_WRITE}, &block);
+ if (status != C2_OK || block == nullptr) {
+ cout << "C2LinearBlock::map() failed : " << status << endl;
+ return status;
+ }
+
+ C2WriteView view = block->map().get();
+ if (view.error() != C2_OK) {
+ cout << "C2LinearBlock::map() failed : " << view.error() << endl;
+ return view.error();
+ }
+ memcpy(view.base(), inputBuffer + mOffset, size);
+ work->input.buffers.emplace_back(new LinearBuffer(block, size));
+ mStats->addFrameSize(size);
+ }
+ work->worklets.clear();
+ work->worklets.emplace_back(new C2Worklet);
+
+ std::list<std::unique_ptr<C2Work>> items;
+ items.push_back(std::move(work));
+ // queue() invokes process() function of C2 Plugin.
+ status = mComponent->queue(&items);
+ if (status != C2_OK) {
+ ALOGE("queue failed");
+ return status;
+ }
+ ALOGV("Frame #%d size = %d queued", mNumInputFrame, size);
+ mNumInputFrame++;
+ mOffset += size;
+ }
+ return status;
+}
+
+void C2Decoder::deInitCodec() {
+ ALOGV("In %s", __func__);
+ if (!mComponent) return;
+
+ int64_t sTime = mStats->getCurTime();
+ mComponent->stop();
+ mComponent->release();
+ mComponent = nullptr;
+ int64_t eTime = mStats->getCurTime();
+ int64_t timeTaken = mStats->getTimeDiff(sTime, eTime);
+ mStats->setDeInitTime(timeTaken);
+}
+
+void C2Decoder::dumpStatistics(string inputReference, int64_t durationUs) {
+ string operation = "c2decode";
+ mStats->dumpStatistics(operation, inputReference, durationUs);
+}
+
+void C2Decoder::resetDecoder() {
+ mOffset = 0;
+ mNumInputFrame = 0;
+ if (mStats) mStats->reset();
+}
diff --git a/media/tests/benchmark/src/native/decoder/C2Decoder.h b/media/tests/benchmark/src/native/decoder/C2Decoder.h
new file mode 100644
index 0000000..0e79d51
--- /dev/null
+++ b/media/tests/benchmark/src/native/decoder/C2Decoder.h
@@ -0,0 +1,51 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __C2_DECODER_H__
+#define __C2_DECODER_H__
+
+#include <stdio.h>
+#include <algorithm>
+#include <fstream>
+
+#include "BenchmarkC2Common.h"
+
+#define ALIGN(_sz, _align) (((_sz) + ((_align) - 1)) & ~((_align) - 1))
+
+class C2Decoder : public BenchmarkC2Common {
+ public:
+ C2Decoder() : mOffset(0), mNumInputFrame(0), mComponent(nullptr) {}
+
+ int32_t createCodec2Component(string codecName, AMediaFormat *format);
+
+ int32_t decodeFrames(uint8_t *inputBuffer, vector<AMediaCodecBufferInfo> &frameInfo);
+
+ void deInitCodec();
+
+ void dumpStatistics(string inputReference, int64_t durationUs);
+
+ void resetDecoder();
+
+ private:
+ int32_t mOffset;
+ int32_t mNumInputFrame;
+ vector<AMediaCodecBufferInfo> mFrameMetaData;
+
+ std::shared_ptr<android::Codec2Client::Listener> mListener;
+ std::shared_ptr<android::Codec2Client::Component> mComponent;
+};
+
+#endif // __C2_DECODER_H__
diff --git a/media/tests/benchmark/src/native/decoder/Decoder.cpp b/media/tests/benchmark/src/native/decoder/Decoder.cpp
new file mode 100644
index 0000000..090f3e1
--- /dev/null
+++ b/media/tests/benchmark/src/native/decoder/Decoder.cpp
@@ -0,0 +1,257 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "decoder"
+
+#include <iostream>
+
+#include "Decoder.h"
+
+tuple<ssize_t, uint32_t, int64_t> readSampleData(uint8_t *inputBuffer, int32_t &offset,
+ vector<AMediaCodecBufferInfo> &frameInfo,
+ uint8_t *buf, int32_t frameID, size_t bufSize) {
+ ALOGV("In %s", __func__);
+ if (frameID == (int32_t)frameInfo.size()) {
+ return make_tuple(0, AMEDIACODEC_BUFFER_FLAG_END_OF_STREAM, 0);
+ }
+ uint32_t flags = frameInfo[frameID].flags;
+ int64_t timestamp = frameInfo[frameID].presentationTimeUs;
+ ssize_t bytesCount = frameInfo[frameID].size;
+ if (bufSize < bytesCount) {
+ ALOGE("Error : Buffer size is insufficient to read sample");
+ return make_tuple(0, AMEDIA_ERROR_MALFORMED, 0);
+ }
+
+ memcpy(buf, inputBuffer + offset, bytesCount);
+ offset += bytesCount;
+ return make_tuple(bytesCount, flags, timestamp);
+}
+
+void Decoder::onInputAvailable(AMediaCodec *mediaCodec, int32_t bufIdx) {
+ ALOGV("In %s", __func__);
+ if (mediaCodec == mCodec && mediaCodec) {
+ if (mSawInputEOS || bufIdx < 0) return;
+ if (mSignalledError) {
+ CallBackHandle::mSawError = true;
+ mDecoderDoneCondition.notify_one();
+ return;
+ }
+
+ size_t bufSize;
+ uint8_t *buf = AMediaCodec_getInputBuffer(mCodec, bufIdx, &bufSize);
+ if (!buf) {
+ mErrorCode = AMEDIA_ERROR_IO;
+ mSignalledError = true;
+ mDecoderDoneCondition.notify_one();
+ return;
+ }
+
+ ssize_t bytesRead = 0;
+ uint32_t flag = 0;
+ int64_t presentationTimeUs = 0;
+ tie(bytesRead, flag, presentationTimeUs) =
+ readSampleData(mInputBuffer, mOffset, mFrameMetaData, buf, mNumInputFrame, bufSize);
+ if (flag == AMEDIA_ERROR_MALFORMED) {
+ mErrorCode = (media_status_t)flag;
+ mSignalledError = true;
+ mDecoderDoneCondition.notify_one();
+ return;
+ }
+
+ if (flag == AMEDIACODEC_BUFFER_FLAG_END_OF_STREAM) mSawInputEOS = true;
+ ALOGV("%s bytesRead : %zd presentationTimeUs : %" PRId64 " mSawInputEOS : %s", __FUNCTION__,
+ bytesRead, presentationTimeUs, mSawInputEOS ? "TRUE" : "FALSE");
+
+ media_status_t status = AMediaCodec_queueInputBuffer(mCodec, bufIdx, 0 /* offset */,
+ bytesRead, presentationTimeUs, flag);
+ if (AMEDIA_OK != status) {
+ mErrorCode = status;
+ mSignalledError = true;
+ mDecoderDoneCondition.notify_one();
+ return;
+ }
+ mStats->addFrameSize(bytesRead);
+ mNumInputFrame++;
+ }
+}
+
+void Decoder::onOutputAvailable(AMediaCodec *mediaCodec, int32_t bufIdx,
+ AMediaCodecBufferInfo *bufferInfo) {
+ ALOGV("In %s", __func__);
+ if (mediaCodec == mCodec && mediaCodec) {
+ if (mSawOutputEOS || bufIdx < 0) return;
+ if (mSignalledError) {
+ CallBackHandle::mSawError = true;
+ mDecoderDoneCondition.notify_one();
+ return;
+ }
+
+ if (mOutFp != nullptr) {
+ size_t bufSize;
+ uint8_t *buf = AMediaCodec_getOutputBuffer(mCodec, bufIdx, &bufSize);
+ if (buf) {
+ fwrite(buf, sizeof(char), bufferInfo->size, mOutFp);
+ ALOGV("bytes written into file %d\n", bufferInfo->size);
+ }
+ }
+
+ AMediaCodec_releaseOutputBuffer(mCodec, bufIdx, false);
+ mSawOutputEOS = (0 != (bufferInfo->flags & AMEDIACODEC_BUFFER_FLAG_END_OF_STREAM));
+ mNumOutputFrame++;
+ ALOGV("%s index : %d mSawOutputEOS : %s count : %u", __FUNCTION__, bufIdx,
+ mSawOutputEOS ? "TRUE" : "FALSE", mNumOutputFrame);
+
+ if (mSawOutputEOS) {
+ CallBackHandle::mIsDone = true;
+ mDecoderDoneCondition.notify_one();
+ }
+ }
+}
+
+void Decoder::onFormatChanged(AMediaCodec *mediaCodec, AMediaFormat *format) {
+ ALOGV("In %s", __func__);
+ if (mediaCodec == mCodec && mediaCodec) {
+ ALOGV("%s { %s }", __FUNCTION__, AMediaFormat_toString(format));
+ mFormat = format;
+ }
+}
+
+void Decoder::onError(AMediaCodec *mediaCodec, media_status_t err) {
+ ALOGV("In %s", __func__);
+ if (mediaCodec == mCodec && mediaCodec) {
+ ALOGE("Received Error %d", err);
+ mErrorCode = err;
+ mSignalledError = true;
+ mDecoderDoneCondition.notify_one();
+ }
+}
+
+void Decoder::setupDecoder() {
+ if (!mFormat) mFormat = mExtractor->getFormat();
+}
+
+AMediaFormat *Decoder::getFormat() {
+ ALOGV("In %s", __func__);
+ return AMediaCodec_getOutputFormat(mCodec);
+}
+
+int32_t Decoder::decode(uint8_t *inputBuffer, vector<AMediaCodecBufferInfo> &frameInfo,
+ string &codecName, bool asyncMode, FILE *outFp) {
+ ALOGV("In %s", __func__);
+ mInputBuffer = inputBuffer;
+ mFrameMetaData = frameInfo;
+ mOffset = 0;
+ mOutFp = outFp;
+
+ const char *mime = nullptr;
+ AMediaFormat_getString(mFormat, AMEDIAFORMAT_KEY_MIME, &mime);
+ if (!mime) return AMEDIA_ERROR_INVALID_OBJECT;
+
+ int64_t sTime = mStats->getCurTime();
+ mCodec = createMediaCodec(mFormat, mime, codecName, false /*isEncoder*/);
+ if (!mCodec) return AMEDIA_ERROR_INVALID_OBJECT;
+
+ if (asyncMode) {
+ AMediaCodecOnAsyncNotifyCallback aCB = {OnInputAvailableCB, OnOutputAvailableCB,
+ OnFormatChangedCB, OnErrorCB};
+ AMediaCodec_setAsyncNotifyCallback(mCodec, aCB, this);
+
+ mIOThread = thread(&CallBackHandle::ioThread, this);
+ }
+
+ AMediaCodec_start(mCodec);
+ int64_t eTime = mStats->getCurTime();
+ int64_t timeTaken = mStats->getTimeDiff(sTime, eTime);
+ mStats->setInitTime(timeTaken);
+
+ mStats->setStartTime();
+ if (!asyncMode) {
+ while (!mSawOutputEOS && !mSignalledError) {
+ /* Queue input data */
+ if (!mSawInputEOS) {
+ ssize_t inIdx = AMediaCodec_dequeueInputBuffer(mCodec, kQueueDequeueTimeoutUs);
+ if (inIdx < 0 && inIdx != AMEDIACODEC_INFO_TRY_AGAIN_LATER) {
+ ALOGE("AMediaCodec_dequeueInputBuffer returned invalid index %zd\n", inIdx);
+ mErrorCode = (media_status_t)inIdx;
+ return mErrorCode;
+ } else if (inIdx >= 0) {
+ mStats->addInputTime();
+ onInputAvailable(mCodec, inIdx);
+ }
+ }
+
+ /* Dequeue output data */
+ AMediaCodecBufferInfo info;
+ ssize_t outIdx = AMediaCodec_dequeueOutputBuffer(mCodec, &info, kQueueDequeueTimeoutUs);
+ if (outIdx == AMEDIACODEC_INFO_OUTPUT_FORMAT_CHANGED) {
+ mFormat = AMediaCodec_getOutputFormat(mCodec);
+ const char *s = AMediaFormat_toString(mFormat);
+ ALOGI("Output format: %s\n", s);
+ } else if (outIdx >= 0) {
+ mStats->addOutputTime();
+ onOutputAvailable(mCodec, outIdx, &info);
+ } else if (!(outIdx == AMEDIACODEC_INFO_TRY_AGAIN_LATER ||
+ outIdx == AMEDIACODEC_INFO_OUTPUT_BUFFERS_CHANGED)) {
+ ALOGE("AMediaCodec_dequeueOutputBuffer returned invalid index %zd\n", outIdx);
+ mErrorCode = (media_status_t)outIdx;
+ return mErrorCode;
+ }
+ }
+ } else {
+ unique_lock<mutex> lock(mMutex);
+ mDecoderDoneCondition.wait(lock, [this]() { return (mSawOutputEOS || mSignalledError); });
+ }
+ if (mSignalledError) {
+ ALOGE("Received Error while Decoding");
+ return mErrorCode;
+ }
+
+ if (codecName.empty()) {
+ char *decName;
+ AMediaCodec_getName(mCodec, &decName);
+ codecName.assign(decName);
+ AMediaCodec_releaseName(mCodec, decName);
+ }
+ return AMEDIA_OK;
+}
+
+void Decoder::deInitCodec() {
+ if (mFormat) {
+ AMediaFormat_delete(mFormat);
+ mFormat = nullptr;
+ }
+ if (!mCodec) return;
+ int64_t sTime = mStats->getCurTime();
+ AMediaCodec_stop(mCodec);
+ AMediaCodec_delete(mCodec);
+ int64_t eTime = mStats->getCurTime();
+ int64_t timeTaken = mStats->getTimeDiff(sTime, eTime);
+ mStats->setDeInitTime(timeTaken);
+}
+
+void Decoder::dumpStatistics(string inputReference, string componentName, string mode,
+ string statsFile) {
+ int64_t durationUs = mExtractor->getClipDuration();
+ string operation = "decode";
+ mStats->dumpStatistics(operation, inputReference, durationUs, componentName, mode, statsFile);
+}
+
+void Decoder::resetDecoder() {
+ if (mStats) mStats->reset();
+ if (mInputBuffer) mInputBuffer = nullptr;
+ if (!mFrameMetaData.empty()) mFrameMetaData.clear();
+}
diff --git a/media/tests/benchmark/src/native/decoder/Decoder.h b/media/tests/benchmark/src/native/decoder/Decoder.h
new file mode 100644
index 0000000..e619cb4
--- /dev/null
+++ b/media/tests/benchmark/src/native/decoder/Decoder.h
@@ -0,0 +1,108 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __DECODER_H__
+#define __DECODER_H__
+
+#include <chrono>
+#include <condition_variable>
+#include <mutex>
+#include <queue>
+#include <thread>
+
+#include "BenchmarkCommon.h"
+#include "Extractor.h"
+#include "Stats.h"
+
+class Decoder : public CallBackHandle {
+ public:
+ Decoder()
+ : mCodec(nullptr),
+ mFormat(nullptr),
+ mExtractor(nullptr),
+ mNumInputFrame(0),
+ mNumOutputFrame(0),
+ mSawInputEOS(false),
+ mSawOutputEOS(false),
+ mSignalledError(false),
+ mErrorCode(AMEDIA_OK),
+ mInputBuffer(nullptr),
+ mOutFp(nullptr) {
+ mExtractor = new Extractor();
+ }
+
+ virtual ~Decoder() {
+ if (mExtractor) delete mExtractor;
+ }
+
+ Extractor *getExtractor() { return mExtractor; }
+
+ // Decoder related utilities
+ void setupDecoder();
+
+ void deInitCodec();
+
+ void resetDecoder();
+
+ AMediaFormat *getFormat();
+
+ // Async callback APIs
+ void onInputAvailable(AMediaCodec *codec, int32_t index) override;
+
+ void onFormatChanged(AMediaCodec *codec, AMediaFormat *format) override;
+
+ void onError(AMediaCodec *mediaCodec, media_status_t err) override;
+
+ void onOutputAvailable(AMediaCodec *codec, int32_t index,
+ AMediaCodecBufferInfo *bufferInfo) override;
+
+ // Process the frames and give decoded output
+ int32_t decode(uint8_t *inputBuffer, vector<AMediaCodecBufferInfo> &frameInfo,
+ string &codecName, bool asyncMode, FILE *outFp = nullptr);
+
+ void dumpStatistics(string inputReference, string componentName = "", string mode = "",
+ string statsFile = "");
+
+ private:
+ AMediaCodec *mCodec;
+ AMediaFormat *mFormat;
+
+ Extractor *mExtractor;
+
+ int32_t mNumInputFrame;
+ int32_t mNumOutputFrame;
+
+ bool mSawInputEOS;
+ bool mSawOutputEOS;
+ bool mSignalledError;
+ media_status_t mErrorCode;
+
+ int32_t mOffset;
+ uint8_t *mInputBuffer;
+ vector<AMediaCodecBufferInfo> mFrameMetaData;
+ FILE *mOutFp;
+
+ /* Asynchronous locks */
+ mutex mMutex;
+ condition_variable mDecoderDoneCondition;
+};
+
+// Read input samples
+tuple<ssize_t, uint32_t, int64_t> readSampleData(uint8_t *inputBuffer, int32_t &offset,
+ vector<AMediaCodecBufferInfo> &frameSizes,
+ uint8_t *buf, int32_t frameID, size_t bufSize);
+
+#endif // __DECODER_H__
diff --git a/media/tests/benchmark/src/native/encoder/Android.bp b/media/tests/benchmark/src/native/encoder/Android.bp
new file mode 100644
index 0000000..8de7823
--- /dev/null
+++ b/media/tests/benchmark/src/native/encoder/Android.bp
@@ -0,0 +1,53 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+cc_library_static {
+ name: "libmediabenchmark_encoder",
+ defaults: [
+ "libmediabenchmark_common-defaults",
+ "libmediabenchmark_soft_sanitize_all-defaults",
+ ],
+
+ srcs: ["Encoder.cpp"],
+
+ static_libs: [
+ "libmediabenchmark_extractor",
+ "libmediabenchmark_decoder",
+ ],
+
+ export_include_dirs: ["."],
+
+ ldflags: ["-Wl,-Bsymbolic"],
+}
+
+cc_library_static {
+ name: "libmediabenchmark_codec2_encoder",
+ defaults: [
+ "libmediabenchmark_codec2_common-defaults",
+ ],
+
+ srcs: ["C2Encoder.cpp"],
+
+ static_libs: [
+ "libmediabenchmark_codec2_common",
+ "libmediabenchmark_codec2_extractor",
+ "libmediabenchmark_codec2_decoder",
+ ],
+
+ export_include_dirs: ["."],
+
+ ldflags: ["-Wl,-Bsymbolic"],
+}
diff --git a/media/tests/benchmark/src/native/encoder/C2Encoder.cpp b/media/tests/benchmark/src/native/encoder/C2Encoder.cpp
new file mode 100644
index 0000000..33429ef
--- /dev/null
+++ b/media/tests/benchmark/src/native/encoder/C2Encoder.cpp
@@ -0,0 +1,264 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "C2Encoder"
+
+#include "C2Encoder.h"
+
+int32_t C2Encoder::createCodec2Component(string compName, AMediaFormat *format) {
+ ALOGV("In %s", __func__);
+ mListener.reset(new CodecListener(
+ [this](std::list<std::unique_ptr<C2Work>> &workItems) { handleWorkDone(workItems); }));
+ if (!mListener) return -1;
+
+ const char *mime = nullptr;
+ AMediaFormat_getString(format, AMEDIAFORMAT_KEY_MIME, &mime);
+ if (!mime) {
+ ALOGE("Error in AMediaFormat_getString");
+ return -1;
+ }
+ // Configure the plugin with Input properties
+ std::vector<C2Param *> configParam;
+ if (!strncmp(mime, "audio/", 6)) {
+ mIsAudioEncoder = true;
+ int32_t numChannels;
+ if (!AMediaFormat_getInt32(format, AMEDIAFORMAT_KEY_SAMPLE_RATE, &mSampleRate)) {
+ ALOGE("AMEDIAFORMAT_KEY_SAMPLE_RATE not set");
+ return -1;
+ }
+ if (!AMediaFormat_getInt32(format, AMEDIAFORMAT_KEY_CHANNEL_COUNT, &numChannels)) {
+ ALOGE("AMEDIAFORMAT_KEY_CHANNEL_COUNT not set");
+ return -1;
+ }
+ C2StreamSampleRateInfo::input sampleRateInfo(0u, mSampleRate);
+ C2StreamChannelCountInfo::input channelCountInfo(0u, numChannels);
+ configParam.push_back(&sampleRateInfo);
+ configParam.push_back(&channelCountInfo);
+ } else {
+ mIsAudioEncoder = false;
+ if (!AMediaFormat_getInt32(format, AMEDIAFORMAT_KEY_WIDTH, &mWidth)) {
+ ALOGE("AMEDIAFORMAT_KEY_WIDTH not set");
+ return -1;
+ }
+ if (!AMediaFormat_getInt32(format, AMEDIAFORMAT_KEY_HEIGHT, &mHeight)) {
+ ALOGE("AMEDIAFORMAT_KEY_HEIGHT not set");
+ return -1;
+ }
+ C2StreamPictureSizeInfo::input inputSize(0u, mWidth, mHeight);
+ configParam.push_back(&inputSize);
+
+ if (!AMediaFormat_getInt32(format, AMEDIAFORMAT_KEY_FRAME_RATE, &mFrameRate) ||
+ (mFrameRate <= 0)) {
+ mFrameRate = KDefaultFrameRate;
+ }
+ }
+
+ int64_t sTime = mStats->getCurTime();
+ mComponent = mClient->CreateComponentByName(compName.c_str(), mListener, &mClient);
+ if (mComponent == nullptr) {
+ ALOGE("Create component failed for %s", compName.c_str());
+ return -1;
+ }
+ std::vector<std::unique_ptr<C2SettingResult>> failures;
+ int32_t status = mComponent->config(configParam, C2_DONT_BLOCK, &failures);
+ if (failures.size() != 0) {
+ ALOGE("Invalid Configuration");
+ return -1;
+ }
+
+ status |= mComponent->start();
+ int64_t eTime = mStats->getCurTime();
+ int64_t timeTaken = mStats->getTimeDiff(sTime, eTime);
+ mStats->setInitTime(timeTaken);
+ return status;
+}
+
+// In encoder components, fetch the size of input buffer allocated
+int32_t C2Encoder::getInputMaxBufSize() {
+ int32_t bitStreamInfo[1] = {0};
+ std::vector<std::unique_ptr<C2Param>> inParams;
+ c2_status_t status = mComponent->query({}, {C2StreamMaxBufferSizeInfo::input::PARAM_TYPE},
+ C2_DONT_BLOCK, &inParams);
+ if (status != C2_OK && inParams.size() == 0) {
+ ALOGE("Query MaxBufferSizeInfo failed => %d", status);
+ return status;
+ } else {
+ size_t offset = sizeof(C2Param);
+ for (size_t i = 0; i < inParams.size(); ++i) {
+ C2Param *param = inParams[i].get();
+ bitStreamInfo[i] = *(int32_t *)((uint8_t *)param + offset);
+ }
+ }
+ mInputMaxBufSize = bitStreamInfo[0];
+ if (mInputMaxBufSize < 0) {
+ ALOGE("Invalid mInputMaxBufSize %d\n", mInputMaxBufSize);
+ return -1;
+ }
+ return status;
+}
+
+int32_t C2Encoder::encodeFrames(ifstream &eleStream, size_t inputBufferSize) {
+ ALOGV("In %s", __func__);
+ int32_t frameSize = 0;
+ if (!mIsAudioEncoder) {
+ frameSize = mWidth * mHeight * 3 / 2;
+ } else {
+ frameSize = DEFAULT_AUDIO_FRAME_SIZE;
+ if (getInputMaxBufSize() != 0) return -1;
+ if (frameSize > mInputMaxBufSize) {
+ frameSize = mInputMaxBufSize;
+ }
+ }
+ int32_t numFrames = (inputBufferSize + frameSize - 1) / frameSize;
+ // Temporary buffer to read data from the input file
+ char *data = (char *)malloc(frameSize);
+ if (!data) {
+ ALOGE("Insufficient memory to read from input file");
+ return -1;
+ }
+
+ typedef std::unique_lock<std::mutex> ULock;
+ uint64_t presentationTimeUs = 0;
+ size_t offset = 0;
+ c2_status_t status = C2_OK;
+
+ mStats->setStartTime();
+ while (numFrames > 0) {
+ std::unique_ptr<C2Work> work;
+ // Prepare C2Work
+ {
+ ULock l(mQueueLock);
+ if (mWorkQueue.empty()) mQueueCondition.wait_for(l, MAX_RETRY * TIME_OUT);
+ if (!mWorkQueue.empty()) {
+ mStats->addInputTime();
+ work.swap(mWorkQueue.front());
+ mWorkQueue.pop_front();
+ } else {
+ cout << "Wait for generating C2Work exceeded timeout" << endl;
+ return -1;
+ }
+ }
+
+ if (mIsAudioEncoder) {
+ presentationTimeUs = mNumInputFrame * frameSize * (1000000 / mSampleRate);
+ } else {
+ presentationTimeUs = mNumInputFrame * (1000000 / mFrameRate);
+ }
+ uint32_t flags = 0;
+ if (numFrames == 1) flags |= C2FrameData::FLAG_END_OF_STREAM;
+
+ work->input.flags = (C2FrameData::flags_t)flags;
+ work->input.ordinal.timestamp = presentationTimeUs;
+ work->input.ordinal.frameIndex = mNumInputFrame;
+ work->input.buffers.clear();
+
+ if (inputBufferSize - offset < frameSize) {
+ frameSize = inputBufferSize - offset;
+ }
+ eleStream.read(data, frameSize);
+ if (eleStream.gcount() != frameSize) {
+ ALOGE("read() from file failed. Incorrect bytes read");
+ return -1;
+ }
+ offset += frameSize;
+
+ if (frameSize) {
+ if (mIsAudioEncoder) {
+ std::shared_ptr<C2LinearBlock> block;
+ status = mLinearPool->fetchLinearBlock(
+ frameSize, {C2MemoryUsage::CPU_READ, C2MemoryUsage::CPU_WRITE}, &block);
+ if (status != C2_OK || !block) {
+ cout << "fetchLinearBlock failed : " << status << endl;
+ return status;
+ }
+ C2WriteView view = block->map().get();
+ if (view.error() != C2_OK) {
+ cout << "C2LinearBlock::map() failed : " << view.error() << endl;
+ return view.error();
+ }
+
+ memcpy(view.base(), data, frameSize);
+ work->input.buffers.emplace_back(new LinearBuffer(block));
+ } else {
+ std::shared_ptr<C2GraphicBlock> block;
+ status = mGraphicPool->fetchGraphicBlock(
+ mWidth, mHeight, HAL_PIXEL_FORMAT_YV12,
+ {C2MemoryUsage::CPU_READ, C2MemoryUsage::CPU_WRITE}, &block);
+ if (status != C2_OK || !block) {
+ cout << "fetchGraphicBlock failed : " << status << endl;
+ return status;
+ }
+ C2GraphicView view = block->map().get();
+ if (view.error() != C2_OK) {
+ cout << "C2GraphicBlock::map() failed : " << view.error() << endl;
+ return view.error();
+ }
+
+ uint8_t *pY = view.data()[C2PlanarLayout::PLANE_Y];
+ uint8_t *pU = view.data()[C2PlanarLayout::PLANE_U];
+ uint8_t *pV = view.data()[C2PlanarLayout::PLANE_V];
+ memcpy(pY, data, mWidth * mHeight);
+ memcpy(pU, data + mWidth * mHeight, (mWidth * mHeight >> 2));
+ memcpy(pV, data + (mWidth * mHeight * 5 >> 2), mWidth * mHeight >> 2);
+ work->input.buffers.emplace_back(new GraphicBuffer(block));
+ }
+ mStats->addFrameSize(frameSize);
+ }
+
+ work->worklets.clear();
+ work->worklets.emplace_back(new C2Worklet);
+
+ std::list<std::unique_ptr<C2Work>> items;
+ items.push_back(std::move(work));
+ // queue() invokes process() function of C2 Plugin.
+ status = mComponent->queue(&items);
+ if (status != C2_OK) {
+ ALOGE("queue failed");
+ return status;
+ }
+ ALOGV("Frame #%d size = %d queued", mNumInputFrame, frameSize);
+ numFrames--;
+ mNumInputFrame++;
+ }
+ free(data);
+ return status;
+}
+
+void C2Encoder::deInitCodec() {
+ ALOGV("In %s", __func__);
+ if (!mComponent) return;
+
+ int64_t sTime = mStats->getCurTime();
+ mComponent->stop();
+ mComponent->release();
+ mComponent = nullptr;
+ int64_t eTime = mStats->getCurTime();
+ int64_t timeTaken = mStats->getTimeDiff(sTime, eTime);
+ mStats->setDeInitTime(timeTaken);
+}
+
+void C2Encoder::dumpStatistics(string inputReference, int64_t durationUs) {
+ string operation = "c2encode";
+ mStats->dumpStatistics(operation, inputReference, durationUs);
+}
+
+void C2Encoder::resetEncoder() {
+ mIsAudioEncoder = false;
+ mNumInputFrame = 0;
+ mEos = false;
+ if (mStats) mStats->reset();
+}
diff --git a/media/tests/benchmark/src/native/encoder/C2Encoder.h b/media/tests/benchmark/src/native/encoder/C2Encoder.h
new file mode 100644
index 0000000..a4ca097
--- /dev/null
+++ b/media/tests/benchmark/src/native/encoder/C2Encoder.h
@@ -0,0 +1,66 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __C2_ENCODER_H__
+#define __C2_ENCODER_H__
+
+#include <stdio.h>
+#include <algorithm>
+#include <fstream>
+
+#include "BenchmarkC2Common.h"
+
+#define DEFAULT_AUDIO_FRAME_SIZE 4096
+
+constexpr int32_t KDefaultFrameRate = 25;
+
+class C2Encoder : public BenchmarkC2Common {
+ public:
+ C2Encoder()
+ : mIsAudioEncoder(false),
+ mWidth(0),
+ mHeight(0),
+ mNumInputFrame(0),
+ mComponent(nullptr) {}
+
+ int32_t createCodec2Component(string codecName, AMediaFormat *format);
+
+ int32_t encodeFrames(ifstream &eleStream, size_t inputBufferSize);
+
+ int32_t getInputMaxBufSize();
+
+ void deInitCodec();
+
+ void dumpStatistics(string inputReference, int64_t durationUs);
+
+ void resetEncoder();
+
+ private:
+ bool mIsAudioEncoder;
+
+ int32_t mWidth;
+ int32_t mHeight;
+ int32_t mFrameRate;
+ int32_t mSampleRate;
+
+ int32_t mNumInputFrame;
+ int32_t mInputMaxBufSize;
+
+ std::shared_ptr<android::Codec2Client::Listener> mListener;
+ std::shared_ptr<android::Codec2Client::Component> mComponent;
+};
+
+#endif // __C2_ENCODER_H__
diff --git a/media/tests/benchmark/src/native/encoder/Encoder.cpp b/media/tests/benchmark/src/native/encoder/Encoder.cpp
new file mode 100644
index 0000000..26fb1b9
--- /dev/null
+++ b/media/tests/benchmark/src/native/encoder/Encoder.cpp
@@ -0,0 +1,304 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "encoder"
+
+#include <fstream>
+
+#include "Encoder.h"
+
+void Encoder::onInputAvailable(AMediaCodec *mediaCodec, int32_t bufIdx) {
+ ALOGV("In %s", __func__);
+ if (mediaCodec == mCodec && mediaCodec) {
+ if (mSawInputEOS || bufIdx < 0) return;
+ if (mSignalledError) {
+ CallBackHandle::mSawError = true;
+ mEncoderDoneCondition.notify_one();
+ return;
+ }
+
+ size_t bufSize = 0;
+ char *buf = (char *)AMediaCodec_getInputBuffer(mCodec, bufIdx, &bufSize);
+ if (!buf) {
+ mErrorCode = AMEDIA_ERROR_IO;
+ mSignalledError = true;
+ mEncoderDoneCondition.notify_one();
+ return;
+ }
+
+ if (mInputBufferSize < mOffset) {
+ ALOGE("Out of bound access of input buffer\n");
+ mErrorCode = AMEDIA_ERROR_MALFORMED;
+ mSignalledError = true;
+ mEncoderDoneCondition.notify_one();
+ return;
+ }
+ size_t bytesToRead = mParams.frameSize;
+ if (mInputBufferSize - mOffset < mParams.frameSize) {
+ bytesToRead = mInputBufferSize - mOffset;
+ }
+ //b/148655275 - Update Frame size, as Format value may not be valid
+ if (bufSize < bytesToRead) {
+ if(mNumInputFrame == 0) {
+ mParams.frameSize = bufSize;
+ bytesToRead = bufSize;
+ mParams.numFrames = (mInputBufferSize + mParams.frameSize - 1) / mParams.frameSize;
+ } else {
+ ALOGE("bytes to read %zu bufSize %zu \n", bytesToRead, bufSize);
+ mErrorCode = AMEDIA_ERROR_MALFORMED;
+ mSignalledError = true;
+ mEncoderDoneCondition.notify_one();
+ return;
+ }
+ }
+ if (bytesToRead < mParams.frameSize && mNumInputFrame < mParams.numFrames - 1) {
+ ALOGE("Partial frame at frameID %d bytesToRead %zu frameSize %d total numFrames %d\n",
+ mNumInputFrame, bytesToRead, mParams.frameSize, mParams.numFrames);
+ mErrorCode = AMEDIA_ERROR_MALFORMED;
+ mSignalledError = true;
+ mEncoderDoneCondition.notify_one();
+ return;
+ }
+ mEleStream->read(buf, bytesToRead);
+ size_t bytesgcount = mEleStream->gcount();
+ if (bytesgcount != bytesToRead) {
+ ALOGE("bytes to read %zu actual bytes read %zu \n", bytesToRead, bytesgcount);
+ mErrorCode = AMEDIA_ERROR_MALFORMED;
+ mSignalledError = true;
+ mEncoderDoneCondition.notify_one();
+ return;
+ }
+
+ uint32_t flag = 0;
+ if (mNumInputFrame == mParams.numFrames - 1 || bytesToRead == 0) {
+ ALOGD("Sending EOS on input Last frame\n");
+ flag |= AMEDIACODEC_BUFFER_FLAG_END_OF_STREAM;
+ }
+
+ uint64_t presentationTimeUs;
+ if (!strncmp(mMime, "video/", 6)) {
+ presentationTimeUs = mNumInputFrame * (1000000 / mParams.frameRate);
+ } else {
+ presentationTimeUs =
+ (uint64_t)mNumInputFrame * mParams.frameSize * 1000000 / mParams.sampleRate;
+ }
+
+ if (flag == AMEDIACODEC_BUFFER_FLAG_END_OF_STREAM) mSawInputEOS = true;
+ ALOGV("%s bytesRead : %zd presentationTimeUs : %" PRIu64 " mSawInputEOS : %s", __FUNCTION__,
+ bytesToRead, presentationTimeUs, mSawInputEOS ? "TRUE" : "FALSE");
+
+ media_status_t status = AMediaCodec_queueInputBuffer(mCodec, bufIdx, 0 /* offset */,
+ bytesToRead, presentationTimeUs, flag);
+ if (AMEDIA_OK != status) {
+ mErrorCode = status;
+ mSignalledError = true;
+ mEncoderDoneCondition.notify_one();
+ return;
+ }
+ mNumInputFrame++;
+ mOffset += bytesToRead;
+ }
+}
+
+void Encoder::onOutputAvailable(AMediaCodec *mediaCodec, int32_t bufIdx,
+ AMediaCodecBufferInfo *bufferInfo) {
+ ALOGV("In %s", __func__);
+ if (mediaCodec == mCodec && mediaCodec) {
+ if (mSawOutputEOS || bufIdx < 0) return;
+ if (mSignalledError) {
+ CallBackHandle::mSawError = true;
+ mEncoderDoneCondition.notify_one();
+ return;
+ }
+
+ mStats->addFrameSize(bufferInfo->size);
+ AMediaCodec_releaseOutputBuffer(mCodec, bufIdx, false);
+ mSawOutputEOS = (0 != (bufferInfo->flags & AMEDIACODEC_BUFFER_FLAG_END_OF_STREAM));
+ mNumOutputFrame++;
+ ALOGV("%s index : %d mSawOutputEOS : %s count : %u", __FUNCTION__, bufIdx,
+ mSawOutputEOS ? "TRUE" : "FALSE", mNumOutputFrame);
+ if (mSawOutputEOS) {
+ CallBackHandle::mIsDone = true;
+ mEncoderDoneCondition.notify_one();
+ }
+ }
+}
+
+void Encoder::onFormatChanged(AMediaCodec *mediaCodec, AMediaFormat *format) {
+ ALOGV("In %s", __func__);
+ if (mediaCodec == mCodec && mediaCodec) {
+ ALOGV("%s { %s }", __FUNCTION__, AMediaFormat_toString(format));
+ mFormat = format;
+ }
+}
+
+void Encoder::onError(AMediaCodec *mediaCodec, media_status_t err) {
+ ALOGV("In %s", __func__);
+ if (mediaCodec == mCodec && mediaCodec) {
+ ALOGE("Received Error %d", err);
+ mErrorCode = err;
+ mSignalledError = true;
+ mEncoderDoneCondition.notify_one();
+ }
+}
+
+void Encoder::setupEncoder() {
+ if (!mFormat) mFormat = AMediaFormat_new();
+}
+
+void Encoder::deInitCodec() {
+ if (mFormat) {
+ AMediaFormat_delete(mFormat);
+ mFormat = nullptr;
+ }
+ if (!mCodec) return;
+ int64_t sTime = mStats->getCurTime();
+ AMediaCodec_stop(mCodec);
+ AMediaCodec_delete(mCodec);
+ int64_t eTime = mStats->getCurTime();
+ int64_t timeTaken = mStats->getTimeDiff(sTime, eTime);
+ mStats->setDeInitTime(timeTaken);
+}
+
+void Encoder::resetEncoder() {
+ if (mStats) mStats->reset();
+ if (mEleStream) mEleStream = nullptr;
+ if (mMime) mMime = nullptr;
+ mInputBufferSize = 0;
+ memset(&mParams, 0, sizeof mParams);
+}
+
+void Encoder::dumpStatistics(string inputReference, int64_t durationUs, string componentName,
+ string mode, string statsFile) {
+ string operation = "encode";
+ mStats->dumpStatistics(operation, inputReference, durationUs, componentName, mode, statsFile);
+}
+
+int32_t Encoder::encode(string &codecName, ifstream &eleStream, size_t eleSize, bool asyncMode,
+ encParameter encParams, char *mime) {
+ ALOGV("In %s", __func__);
+ mEleStream = &eleStream;
+ mInputBufferSize = eleSize;
+ mParams = encParams;
+ mOffset = 0;
+ mMime = mime;
+ AMediaFormat_setString(mFormat, AMEDIAFORMAT_KEY_MIME, mMime);
+
+ // Set Format
+ if (!strncmp(mMime, "video/", 6)) {
+ AMediaFormat_setInt32(mFormat, AMEDIAFORMAT_KEY_WIDTH, mParams.width);
+ AMediaFormat_setInt32(mFormat, AMEDIAFORMAT_KEY_HEIGHT, mParams.height);
+ AMediaFormat_setInt32(mFormat, AMEDIAFORMAT_KEY_FRAME_RATE, mParams.frameRate);
+ AMediaFormat_setInt32(mFormat, AMEDIAFORMAT_KEY_BIT_RATE, mParams.bitrate);
+ AMediaFormat_setInt32(mFormat, AMEDIAFORMAT_KEY_I_FRAME_INTERVAL, 1);
+ if (mParams.profile && mParams.level) {
+ AMediaFormat_setInt32(mFormat, AMEDIAFORMAT_KEY_PROFILE, mParams.profile);
+ AMediaFormat_setInt32(mFormat, AMEDIAFORMAT_KEY_LEVEL, mParams.level);
+ }
+ AMediaFormat_setInt32(mFormat, AMEDIAFORMAT_KEY_COLOR_FORMAT, mParams.colorFormat);
+ } else {
+ AMediaFormat_setInt32(mFormat, AMEDIAFORMAT_KEY_SAMPLE_RATE, mParams.sampleRate);
+ AMediaFormat_setInt32(mFormat, AMEDIAFORMAT_KEY_CHANNEL_COUNT, mParams.numChannels);
+ AMediaFormat_setInt32(mFormat, AMEDIAFORMAT_KEY_BIT_RATE, mParams.bitrate);
+ }
+ const char *s = AMediaFormat_toString(mFormat);
+ ALOGI("Input format: %s\n", s);
+
+ int64_t sTime = mStats->getCurTime();
+ mCodec = createMediaCodec(mFormat, mMime, codecName, true /*isEncoder*/);
+ if (!mCodec) return AMEDIA_ERROR_INVALID_OBJECT;
+ int64_t eTime = mStats->getCurTime();
+ int64_t timeTaken = mStats->getTimeDiff(sTime, eTime);
+
+ if (!strncmp(mMime, "video/", 6)) {
+ mParams.frameSize = mParams.width * mParams.height * 3 / 2;
+ } else {
+ mParams.frameSize = kDefaultAudioEncodeFrameSize;
+ // Get mInputMaxBufSize
+ AMediaFormat *inputFormat = AMediaCodec_getInputFormat(mCodec);
+ AMediaFormat_getInt32(inputFormat, AMEDIAFORMAT_KEY_MAX_INPUT_SIZE, &mParams.maxFrameSize);
+ if (mParams.maxFrameSize < 0) {
+ mParams.maxFrameSize = kDefaultAudioEncodeFrameSize;
+ }
+ if (mParams.frameSize > mParams.maxFrameSize) {
+ mParams.frameSize = mParams.maxFrameSize;
+ }
+ }
+ mParams.numFrames = (mInputBufferSize + mParams.frameSize - 1) / mParams.frameSize;
+
+ sTime = mStats->getCurTime();
+ if (asyncMode) {
+ AMediaCodecOnAsyncNotifyCallback aCB = {OnInputAvailableCB, OnOutputAvailableCB,
+ OnFormatChangedCB, OnErrorCB};
+ AMediaCodec_setAsyncNotifyCallback(mCodec, aCB, this);
+ mIOThread = thread(&CallBackHandle::ioThread, this);
+ }
+ AMediaCodec_start(mCodec);
+ eTime = mStats->getCurTime();
+ timeTaken += mStats->getTimeDiff(sTime, eTime);
+ mStats->setInitTime(timeTaken);
+
+ mStats->setStartTime();
+ if (!asyncMode) {
+ while (!mSawOutputEOS && !mSignalledError) {
+ // Queue input data
+ if (!mSawInputEOS) {
+ ssize_t inIdx = AMediaCodec_dequeueInputBuffer(mCodec, kQueueDequeueTimeoutUs);
+ if (inIdx < 0 && inIdx != AMEDIACODEC_INFO_TRY_AGAIN_LATER) {
+ ALOGE("AMediaCodec_dequeueInputBuffer returned invalid index %zd\n", inIdx);
+ mErrorCode = (media_status_t)inIdx;
+ return mErrorCode;
+ } else if (inIdx >= 0) {
+ mStats->addInputTime();
+ onInputAvailable(mCodec, inIdx);
+ }
+ }
+
+ // Dequeue output data
+ AMediaCodecBufferInfo info;
+ ssize_t outIdx = AMediaCodec_dequeueOutputBuffer(mCodec, &info, kQueueDequeueTimeoutUs);
+ if (outIdx == AMEDIACODEC_INFO_OUTPUT_FORMAT_CHANGED) {
+ mFormat = AMediaCodec_getOutputFormat(mCodec);
+ const char *s = AMediaFormat_toString(mFormat);
+ ALOGI("Output format: %s\n", s);
+ } else if (outIdx >= 0) {
+ mStats->addOutputTime();
+ onOutputAvailable(mCodec, outIdx, &info);
+ } else if (!(outIdx == AMEDIACODEC_INFO_TRY_AGAIN_LATER ||
+ outIdx == AMEDIACODEC_INFO_OUTPUT_BUFFERS_CHANGED)) {
+ ALOGE("AMediaCodec_dequeueOutputBuffer returned invalid index %zd\n", outIdx);
+ mErrorCode = (media_status_t)outIdx;
+ return mErrorCode;
+ }
+ }
+ } else {
+ unique_lock<mutex> lock(mMutex);
+ mEncoderDoneCondition.wait(lock, [this]() { return (mSawOutputEOS || mSignalledError); });
+ }
+ if (mSignalledError) {
+ ALOGE("Received Error while Encoding");
+ return mErrorCode;
+ }
+
+ if (codecName.empty()) {
+ char *encName;
+ AMediaCodec_getName(mCodec, &encName);
+ codecName.assign(encName);
+ AMediaCodec_releaseName(mCodec, encName);
+ }
+ return AMEDIA_OK;
+}
diff --git a/media/tests/benchmark/src/native/encoder/Encoder.h b/media/tests/benchmark/src/native/encoder/Encoder.h
new file mode 100644
index 0000000..5ad142b
--- /dev/null
+++ b/media/tests/benchmark/src/native/encoder/Encoder.h
@@ -0,0 +1,105 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __ENCODER_H__
+#define __ENCODER_H__
+
+#include <chrono>
+#include <condition_variable>
+#include <mutex>
+#include <queue>
+#include <thread>
+
+#include "media/NdkImage.h"
+#include "BenchmarkCommon.h"
+#include "Stats.h"
+
+
+struct encParameter {
+ int32_t bitrate = -1;
+ int32_t numFrames = -1;
+ int32_t frameSize = -1;
+ int32_t sampleRate = 0;
+ int32_t numChannels = 0;
+ int32_t maxFrameSize = -1;
+ int32_t width = 0;
+ int32_t height = 0;
+ int32_t frameRate = -1;
+ int32_t profile = 0;
+ int32_t level = 0;
+ int32_t colorFormat = AIMAGE_FORMAT_YUV_420_888;
+};
+
+class Encoder : public CallBackHandle {
+ public:
+ Encoder()
+ : mCodec(nullptr),
+ mFormat(nullptr),
+ mNumInputFrame(0),
+ mNumOutputFrame(0),
+ mSawInputEOS(false),
+ mSawOutputEOS(false),
+ mSignalledError(false),
+ mErrorCode(AMEDIA_OK) {}
+
+ virtual ~Encoder() {}
+
+ // Encoder related utilities
+ void setupEncoder();
+
+ void deInitCodec();
+
+ void resetEncoder();
+
+ // Async callback APIs
+ void onInputAvailable(AMediaCodec *codec, int32_t index) override;
+
+ void onFormatChanged(AMediaCodec *codec, AMediaFormat *format) override;
+
+ void onError(AMediaCodec *mediaCodec, media_status_t err) override;
+
+ void onOutputAvailable(AMediaCodec *codec, int32_t index,
+ AMediaCodecBufferInfo *bufferInfo) override;
+
+ // Process the frames and give encoded output
+ int32_t encode(std::string &codecName, std::ifstream &eleStream, size_t eleSize, bool asyncMode,
+ encParameter encParams, char *mime);
+
+ void dumpStatistics(string inputReference, int64_t durationUs, string codecName = "",
+ string mode = "", string statsFile = "");
+
+ private:
+ AMediaCodec *mCodec;
+ AMediaFormat *mFormat;
+
+ int32_t mNumInputFrame;
+ int32_t mNumOutputFrame;
+ bool mSawInputEOS;
+ bool mSawOutputEOS;
+ bool mSignalledError;
+ media_status_t mErrorCode;
+
+ char *mMime;
+ int32_t mOffset;
+ std::ifstream *mEleStream;
+ size_t mInputBufferSize;
+ encParameter mParams;
+
+ // Asynchronous locks
+ std::mutex mMutex;
+ std::condition_variable mEncoderDoneCondition;
+};
+#endif // __ENCODER_H__
diff --git a/media/tests/benchmark/src/native/extractor/Android.bp b/media/tests/benchmark/src/native/extractor/Android.bp
new file mode 100644
index 0000000..7ed9476
--- /dev/null
+++ b/media/tests/benchmark/src/native/extractor/Android.bp
@@ -0,0 +1,46 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+cc_library_static {
+ name: "libmediabenchmark_extractor",
+ defaults: [
+ "libmediabenchmark_common-defaults",
+ "libmediabenchmark_soft_sanitize_all-defaults",
+ ],
+
+ srcs: ["Extractor.cpp"],
+
+ export_include_dirs: ["."],
+
+ ldflags: ["-Wl,-Bsymbolic"]
+}
+
+cc_library_static {
+ name: "libmediabenchmark_codec2_extractor",
+ defaults: [
+ "libmediabenchmark_codec2_common-defaults",
+ ],
+
+ srcs: ["Extractor.cpp"],
+
+ static_libs: [
+ "libmediabenchmark_codec2_common",
+ ],
+
+ export_include_dirs: ["."],
+
+ ldflags: ["-Wl,-Bsymbolic"]
+}
diff --git a/media/tests/benchmark/src/native/extractor/Extractor.cpp b/media/tests/benchmark/src/native/extractor/Extractor.cpp
new file mode 100644
index 0000000..f0bb3b9
--- /dev/null
+++ b/media/tests/benchmark/src/native/extractor/Extractor.cpp
@@ -0,0 +1,135 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "extractor"
+
+#include <iostream>
+
+#include "Extractor.h"
+
+int32_t Extractor::initExtractor(int32_t fd, size_t fileSize) {
+ mStats = new Stats();
+
+ mFrameBuf = (uint8_t *)calloc(kMaxBufferSize, sizeof(uint8_t));
+ if (!mFrameBuf) return -1;
+
+ int64_t sTime = mStats->getCurTime();
+
+ mExtractor = AMediaExtractor_new();
+ if (!mExtractor) return AMEDIACODEC_ERROR_INSUFFICIENT_RESOURCE;
+ media_status_t status = AMediaExtractor_setDataSourceFd(mExtractor, fd, 0, fileSize);
+ if (status != AMEDIA_OK) return status;
+
+ int64_t eTime = mStats->getCurTime();
+ int64_t timeTaken = mStats->getTimeDiff(sTime, eTime);
+ mStats->setInitTime(timeTaken);
+
+ return AMediaExtractor_getTrackCount(mExtractor);
+}
+
+void *Extractor::getCSDSample(AMediaCodecBufferInfo &frameInfo, int32_t csdIndex) {
+ char csdName[kMaxCSDStrlen];
+ void *csdBuffer = nullptr;
+ frameInfo.presentationTimeUs = 0;
+ frameInfo.flags = AMEDIACODEC_BUFFER_FLAG_CODEC_CONFIG;
+ snprintf(csdName, sizeof(csdName), "csd-%d", csdIndex);
+
+ size_t size;
+ bool csdFound = AMediaFormat_getBuffer(mFormat, csdName, &csdBuffer, &size);
+ if (!csdFound) return nullptr;
+ frameInfo.size = (int32_t)size;
+ mStats->addFrameSize(frameInfo.size);
+
+ return csdBuffer;
+}
+
+int32_t Extractor::getFrameSample(AMediaCodecBufferInfo &frameInfo) {
+ int32_t size = AMediaExtractor_readSampleData(mExtractor, mFrameBuf, kMaxBufferSize);
+ if (size < 0) return -1;
+
+ frameInfo.flags = AMediaExtractor_getSampleFlags(mExtractor);
+ frameInfo.size = size;
+ mStats->addFrameSize(frameInfo.size);
+ frameInfo.presentationTimeUs = AMediaExtractor_getSampleTime(mExtractor);
+ AMediaExtractor_advance(mExtractor);
+
+ return 0;
+}
+
+int32_t Extractor::setupTrackFormat(int32_t trackId) {
+ AMediaExtractor_selectTrack(mExtractor, trackId);
+ mFormat = AMediaExtractor_getTrackFormat(mExtractor, trackId);
+ if (!mFormat) return AMEDIA_ERROR_INVALID_OBJECT;
+
+ bool durationFound = AMediaFormat_getInt64(mFormat, AMEDIAFORMAT_KEY_DURATION, &mDurationUs);
+ if (!durationFound) return AMEDIA_ERROR_INVALID_OBJECT;
+
+ return AMEDIA_OK;
+}
+
+int32_t Extractor::extract(int32_t trackId) {
+ int32_t status = setupTrackFormat(trackId);
+ if (status != AMEDIA_OK) return status;
+
+ int32_t idx = 0;
+ AMediaCodecBufferInfo frameInfo;
+ while (1) {
+ memset(&frameInfo, 0, sizeof(AMediaCodecBufferInfo));
+ void *csdBuffer = getCSDSample(frameInfo, idx);
+ if (!csdBuffer || !frameInfo.size) break;
+ idx++;
+ }
+
+ mStats->setStartTime();
+ while (1) {
+ int32_t status = getFrameSample(frameInfo);
+ if (status || !frameInfo.size) break;
+ mStats->addOutputTime();
+ }
+
+ if (mFormat) {
+ AMediaFormat_delete(mFormat);
+ mFormat = nullptr;
+ }
+
+ AMediaExtractor_unselectTrack(mExtractor, trackId);
+
+ return AMEDIA_OK;
+}
+
+void Extractor::dumpStatistics(string inputReference, string componentName, string statsFile) {
+ string operation = "extract";
+ mStats->dumpStatistics(operation, inputReference, mDurationUs, componentName, "", statsFile);
+}
+
+void Extractor::deInitExtractor() {
+ if (mFrameBuf) {
+ free(mFrameBuf);
+ mFrameBuf = nullptr;
+ }
+
+ int64_t sTime = mStats->getCurTime();
+ if (mExtractor) {
+ // TODO: (b/140128505) Multiple calls result in DoS.
+ // Uncomment call to AMediaExtractor_delete() once this is resolved
+ // AMediaExtractor_delete(mExtractor);
+ mExtractor = nullptr;
+ }
+ int64_t eTime = mStats->getCurTime();
+ int64_t deInitTime = mStats->getTimeDiff(sTime, eTime);
+ mStats->setDeInitTime(deInitTime);
+}
diff --git a/media/tests/benchmark/src/native/extractor/Extractor.h b/media/tests/benchmark/src/native/extractor/Extractor.h
new file mode 100644
index 0000000..1694fc7
--- /dev/null
+++ b/media/tests/benchmark/src/native/extractor/Extractor.h
@@ -0,0 +1,66 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __EXTRACTOR_H__
+#define __EXTRACTOR_H__
+
+#include <media/NdkMediaExtractor.h>
+
+#include "BenchmarkCommon.h"
+#include "Stats.h"
+
+class Extractor {
+ public:
+ Extractor()
+ : mFormat(nullptr),
+ mExtractor(nullptr),
+ mStats(nullptr),
+ mFrameBuf{nullptr},
+ mDurationUs{0} {}
+
+ ~Extractor() {
+ if (mStats) delete mStats;
+ }
+
+ int32_t initExtractor(int32_t fd, size_t fileSize);
+
+ int32_t setupTrackFormat(int32_t trackId);
+
+ void *getCSDSample(AMediaCodecBufferInfo &frameInfo, int32_t csdIndex);
+
+ int32_t getFrameSample(AMediaCodecBufferInfo &frameInfo);
+
+ int32_t extract(int32_t trackId);
+
+ void dumpStatistics(string inputReference, string componentName = "", string statsFile = "");
+
+ void deInitExtractor();
+
+ AMediaFormat *getFormat() { return mFormat; }
+
+ uint8_t *getFrameBuf() { return mFrameBuf; }
+
+ int64_t getClipDuration() { return mDurationUs; }
+
+ private:
+ AMediaFormat *mFormat;
+ AMediaExtractor *mExtractor;
+ Stats *mStats;
+ uint8_t *mFrameBuf;
+ int64_t mDurationUs;
+};
+
+#endif // __EXTRACTOR_H__
\ No newline at end of file
diff --git a/media/tests/benchmark/src/native/muxer/Android.bp b/media/tests/benchmark/src/native/muxer/Android.bp
new file mode 100644
index 0000000..f669d4a
--- /dev/null
+++ b/media/tests/benchmark/src/native/muxer/Android.bp
@@ -0,0 +1,31 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+cc_library_static {
+ name: "libmediabenchmark_muxer",
+ defaults: [
+ "libmediabenchmark_common-defaults",
+ "libmediabenchmark_soft_sanitize_all-defaults",
+ ],
+
+ srcs: ["Muxer.cpp"],
+
+ static_libs: ["libmediabenchmark_extractor"],
+
+ export_include_dirs: ["."],
+
+ ldflags: ["-Wl,-Bsymbolic"]
+}
diff --git a/media/tests/benchmark/src/native/muxer/Muxer.cpp b/media/tests/benchmark/src/native/muxer/Muxer.cpp
new file mode 100644
index 0000000..3e150ca
--- /dev/null
+++ b/media/tests/benchmark/src/native/muxer/Muxer.cpp
@@ -0,0 +1,91 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "muxer"
+
+#include <fstream>
+#include <iostream>
+
+#include "Muxer.h"
+
+int32_t Muxer::initMuxer(int32_t fd, MUXER_OUTPUT_T outputFormat) {
+ if (!mFormat) mFormat = mExtractor->getFormat();
+ if (!mStats) mStats = new Stats();
+
+ int64_t sTime = mStats->getCurTime();
+ mMuxer = AMediaMuxer_new(fd, (OutputFormat)outputFormat);
+ if (!mMuxer) {
+ ALOGV("Unable to create muxer");
+ return AMEDIA_ERROR_INVALID_OBJECT;
+ }
+ /*
+ * AMediaMuxer_addTrack returns the index of the new track or a negative value
+ * in case of failure, which can be interpreted as a media_status_t.
+ */
+ ssize_t index = AMediaMuxer_addTrack(mMuxer, mFormat);
+ if (index < 0) {
+ ALOGV("Format not supported");
+ return index;
+ }
+ AMediaMuxer_start(mMuxer);
+ int64_t eTime = mStats->getCurTime();
+ int64_t timeTaken = mStats->getTimeDiff(sTime, eTime);
+ mStats->setInitTime(timeTaken);
+ return AMEDIA_OK;
+}
+
+void Muxer::deInitMuxer() {
+ if (mFormat) {
+ AMediaFormat_delete(mFormat);
+ mFormat = nullptr;
+ }
+ if (!mMuxer) return;
+ int64_t sTime = mStats->getCurTime();
+ AMediaMuxer_stop(mMuxer);
+ AMediaMuxer_delete(mMuxer);
+ int64_t eTime = mStats->getCurTime();
+ int64_t timeTaken = mStats->getTimeDiff(sTime, eTime);
+ mStats->setDeInitTime(timeTaken);
+}
+
+void Muxer::resetMuxer() {
+ if (mStats) mStats->reset();
+}
+
+void Muxer::dumpStatistics(string inputReference, string componentName, string statsFile) {
+ string operation = "mux";
+ mStats->dumpStatistics(operation, inputReference, mExtractor->getClipDuration(), componentName,
+ "", statsFile);
+}
+
+int32_t Muxer::mux(uint8_t *inputBuffer, vector<AMediaCodecBufferInfo> &frameInfos) {
+ // Mux frame data
+ size_t frameIdx = 0;
+ mStats->setStartTime();
+ while (frameIdx < frameInfos.size()) {
+ AMediaCodecBufferInfo info = frameInfos.at(frameIdx);
+ media_status_t status = AMediaMuxer_writeSampleData(mMuxer, 0, inputBuffer, &info);
+ if (status != 0) {
+ ALOGE("Error in AMediaMuxer_writeSampleData");
+ return status;
+ }
+ mStats->addOutputTime();
+ mStats->addFrameSize(info.size);
+ frameIdx++;
+ }
+ return AMEDIA_OK;
+}
diff --git a/media/tests/benchmark/src/native/muxer/Muxer.h b/media/tests/benchmark/src/native/muxer/Muxer.h
new file mode 100644
index 0000000..860fdaf
--- /dev/null
+++ b/media/tests/benchmark/src/native/muxer/Muxer.h
@@ -0,0 +1,63 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __MUXER_H__
+#define __MUXER_H__
+
+#include <media/NdkMediaMuxer.h>
+
+#include "BenchmarkCommon.h"
+#include "Stats.h"
+#include "Extractor.h"
+
+typedef enum {
+ MUXER_OUTPUT_FORMAT_MPEG_4 = 0,
+ MUXER_OUTPUT_FORMAT_WEBM = 1,
+ MUXER_OUTPUT_FORMAT_3GPP = 2,
+ MUXER_OUTPUT_FORMAT_OGG = 4,
+ MUXER_OUTPUT_FORMAT_INVALID = 5,
+} MUXER_OUTPUT_T;
+
+class Muxer {
+ public:
+ Muxer() : mFormat(nullptr), mMuxer(nullptr), mStats(nullptr) { mExtractor = new Extractor(); }
+
+ virtual ~Muxer() {
+ if (mStats) delete mStats;
+ if (mExtractor) delete mExtractor;
+ }
+
+ Stats *getStats() { return mStats; }
+ Extractor *getExtractor() { return mExtractor; }
+
+ /* Muxer related utilities */
+ int32_t initMuxer(int32_t fd, MUXER_OUTPUT_T outputFormat);
+ void deInitMuxer();
+ void resetMuxer();
+
+ /* Process the frames and give Muxed output */
+ int32_t mux(uint8_t *inputBuffer, vector<AMediaCodecBufferInfo> &frameSizes);
+
+ void dumpStatistics(string inputReference, string codecName = "", string statsFile = "");
+
+ private:
+ AMediaFormat *mFormat;
+ AMediaMuxer *mMuxer;
+ Extractor *mExtractor;
+ Stats *mStats;
+};
+
+#endif // __MUXER_H__
diff --git a/media/tests/benchmark/tests/Android.bp b/media/tests/benchmark/tests/Android.bp
new file mode 100644
index 0000000..f46fa4a
--- /dev/null
+++ b/media/tests/benchmark/tests/Android.bp
@@ -0,0 +1,111 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+cc_test {
+ name: "extractorTest",
+ gtest: true,
+ defaults: [
+ "libmediabenchmark_common-defaults",
+ "libmediabenchmark_soft_sanitize_all-defaults",
+ ],
+
+ srcs: ["ExtractorTest.cpp"],
+
+ static_libs: ["libmediabenchmark_extractor"]
+}
+
+cc_test {
+ name: "decoderTest",
+ gtest: true,
+ defaults: [
+ "libmediabenchmark_common-defaults",
+ "libmediabenchmark_soft_sanitize_all-defaults",
+ ],
+
+ srcs: ["DecoderTest.cpp"],
+
+ static_libs: [
+ "libmediabenchmark_extractor",
+ "libmediabenchmark_decoder",
+ ],
+}
+
+cc_test {
+ name: "muxerTest",
+ gtest: true,
+ defaults: [
+ "libmediabenchmark_common-defaults",
+ "libmediabenchmark_soft_sanitize_all-defaults",
+ ],
+
+ srcs: ["MuxerTest.cpp"],
+
+ static_libs: [
+ "libmediabenchmark_extractor",
+ "libmediabenchmark_muxer",
+ ],
+}
+
+cc_test {
+ name: "encoderTest",
+ gtest: true,
+ defaults: [
+ "libmediabenchmark_common-defaults",
+ "libmediabenchmark_soft_sanitize_all-defaults",
+ ],
+
+ srcs: ["EncoderTest.cpp"],
+
+ static_libs: [
+ "libmediabenchmark_extractor",
+ "libmediabenchmark_decoder",
+ "libmediabenchmark_encoder",
+ ],
+}
+
+cc_test {
+ name: "C2DecoderTest",
+ gtest: true,
+ defaults: [
+ "libmediabenchmark_codec2_common-defaults",
+ "libmediabenchmark_soft_sanitize_all-defaults",
+ ],
+
+ srcs: ["C2DecoderTest.cpp"],
+
+ static_libs: [
+ "libmediabenchmark_codec2_extractor",
+ "libmediabenchmark_codec2_common",
+ "libmediabenchmark_codec2_decoder",
+ ],
+}
+
+cc_test {
+ name: "C2EncoderTest",
+ gtest: true,
+ defaults: [
+ "libmediabenchmark_codec2_common-defaults",
+ ],
+
+ srcs: ["C2EncoderTest.cpp"],
+
+ static_libs: [
+ "libmediabenchmark_codec2_extractor",
+ "libmediabenchmark_codec2_decoder",
+ "libmediabenchmark_codec2_common",
+ "libmediabenchmark_codec2_encoder",
+ ],
+}
diff --git a/media/tests/benchmark/tests/BenchmarkTestEnvironment.h b/media/tests/benchmark/tests/BenchmarkTestEnvironment.h
new file mode 100644
index 0000000..ae2eee1
--- /dev/null
+++ b/media/tests/benchmark/tests/BenchmarkTestEnvironment.h
@@ -0,0 +1,73 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __BENCHMARK_TEST_ENVIRONMENT_H__
+#define __BENCHMARK_TEST_ENVIRONMENT_H__
+
+#include <gtest/gtest.h>
+
+#include <getopt.h>
+
+using namespace std;
+
+class BenchmarkTestEnvironment : public ::testing::Environment {
+ public:
+ BenchmarkTestEnvironment() : res("/sdcard/media/") {}
+
+ // Parses the command line argument
+ int initFromOptions(int argc, char **argv);
+
+ void setRes(const char *_res) { res = _res; }
+
+ const string getRes() const { return res; }
+
+ private:
+ string res;
+};
+
+int BenchmarkTestEnvironment::initFromOptions(int argc, char **argv) {
+ static struct option options[] = {{"path", required_argument, 0, 'P'}, {0, 0, 0, 0}};
+
+ while (true) {
+ int index = 0;
+ int c = getopt_long(argc, argv, "P:", options, &index);
+ if (c == -1) {
+ break;
+ }
+
+ switch (c) {
+ case 'P': {
+ setRes(optarg);
+ break;
+ }
+ default:
+ break;
+ }
+ }
+
+ if (optind < argc) {
+ fprintf(stderr,
+ "unrecognized option: %s\n\n"
+ "usage: %s <gtest options> <test options>\n\n"
+ "test options are:\n\n"
+ "-P, --path: Resource files directory location\n",
+ argv[optind ?: 1], argv[0]);
+ return 2;
+ }
+ return 0;
+}
+
+#endif // __BENCHMARK_TEST_ENVIRONMENT_H__
diff --git a/media/tests/benchmark/tests/C2DecoderTest.cpp b/media/tests/benchmark/tests/C2DecoderTest.cpp
new file mode 100644
index 0000000..dedc743
--- /dev/null
+++ b/media/tests/benchmark/tests/C2DecoderTest.cpp
@@ -0,0 +1,185 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "C2DecoderTest"
+
+#include <fstream>
+#include <iostream>
+#include <limits>
+
+#include "BenchmarkTestEnvironment.h"
+#include "C2Decoder.h"
+#include "Extractor.h"
+
+static BenchmarkTestEnvironment *gEnv = nullptr;
+
+class C2DecoderTest : public ::testing::TestWithParam<pair<string, string>> {
+ public:
+ C2DecoderTest() : mDecoder(nullptr) {}
+
+ ~C2DecoderTest() {
+ if (!mCodecList.empty()) {
+ mCodecList.clear();
+ }
+ if (mDecoder) {
+ delete mDecoder;
+ mDecoder = nullptr;
+ }
+ }
+
+ virtual void SetUp() override { setupC2DecoderTest(); }
+
+ void setupC2DecoderTest();
+
+ vector<string> mCodecList;
+ C2Decoder *mDecoder;
+};
+
+void C2DecoderTest::setupC2DecoderTest() {
+ mDecoder = new C2Decoder();
+ ASSERT_NE(mDecoder, nullptr) << "C2Decoder creation failed";
+
+ int32_t status = mDecoder->setupCodec2();
+ ASSERT_EQ(status, 0) << "Codec2 setup failed";
+
+ mCodecList = mDecoder->getSupportedComponentList(false /* isEncoder*/);
+ ASSERT_GT(mCodecList.size(), 0) << "Codec2 client didn't recognise any component";
+}
+
+TEST_P(C2DecoderTest, Codec2Decode) {
+ ALOGV("Decode the samples given by extractor using codec2");
+ string inputFile = gEnv->getRes() + GetParam().first;
+ FILE *inputFp = fopen(inputFile.c_str(), "rb");
+ ASSERT_NE(inputFp, nullptr) << "Unable to open " << inputFile << " file for reading";
+
+ Extractor *extractor = new Extractor();
+ ASSERT_NE(extractor, nullptr) << "Extractor creation failed";
+
+ // Read file properties
+ struct stat buf;
+ stat(inputFile.c_str(), &buf);
+ size_t fileSize = buf.st_size;
+ int32_t fd = fileno(inputFp);
+
+ ASSERT_LE(fileSize, kMaxBufferSize)
+ << "Input file size is greater than the threshold memory dedicated to the test";
+
+ int32_t trackCount = extractor->initExtractor(fd, fileSize);
+ ASSERT_GT(trackCount, 0) << "initExtractor failed";
+
+ for (int32_t curTrack = 0; curTrack < trackCount; curTrack++) {
+ int32_t status = extractor->setupTrackFormat(curTrack);
+ ASSERT_EQ(status, 0) << "Track Format invalid";
+
+ uint8_t *inputBuffer = (uint8_t *)malloc(fileSize);
+ ASSERT_NE(inputBuffer, nullptr) << "Insufficient memory";
+
+ vector<AMediaCodecBufferInfo> frameInfo;
+ AMediaCodecBufferInfo info;
+ uint32_t inputBufferOffset = 0;
+ int32_t idx = 0;
+
+ // Get CSD data
+ while (1) {
+ void *csdBuffer = extractor->getCSDSample(info, idx);
+ if (!csdBuffer || !info.size) break;
+ // copy the meta data and buffer to be passed to decoder
+ ASSERT_LE(inputBufferOffset + info.size, fileSize) << "Memory allocated not sufficient";
+
+ memcpy(inputBuffer + inputBufferOffset, csdBuffer, info.size);
+ frameInfo.push_back(info);
+ inputBufferOffset += info.size;
+ idx++;
+ }
+
+ // Get frame data
+ while (1) {
+ status = extractor->getFrameSample(info);
+ if (status || !info.size) break;
+ // copy the meta data and buffer to be passed to decoder
+ ASSERT_LE(inputBufferOffset + info.size, fileSize) << "Memory allocated not sufficient";
+
+ memcpy(inputBuffer + inputBufferOffset, extractor->getFrameBuf(), info.size);
+ frameInfo.push_back(info);
+ inputBufferOffset += info.size;
+ }
+
+ AMediaFormat *format = extractor->getFormat();
+ // Decode the given input stream for all C2 codecs supported by device
+ for (string codecName : mCodecList) {
+ if (codecName.find(GetParam().second) != string::npos &&
+ codecName.find("secure") == string::npos) {
+ status = mDecoder->createCodec2Component(codecName, format);
+ ASSERT_EQ(status, 0) << "Create component failed for " << codecName;
+
+ // Send the inputs to C2 Decoder and wait till all buffers are returned.
+ status = mDecoder->decodeFrames(inputBuffer, frameInfo);
+ ASSERT_EQ(status, 0) << "Decoder failed for " << codecName;
+
+ mDecoder->waitOnInputConsumption();
+ ASSERT_TRUE(mDecoder->mEos) << "Test Failed. Didn't receive EOS \n";
+
+ mDecoder->deInitCodec();
+ int64_t durationUs = extractor->getClipDuration();
+ ALOGV("codec : %s", codecName.c_str());
+ mDecoder->dumpStatistics(GetParam().first, durationUs);
+ mDecoder->resetDecoder();
+ }
+ }
+ free(inputBuffer);
+ fclose(inputFp);
+ extractor->deInitExtractor();
+ delete extractor;
+ delete mDecoder;
+ mDecoder = nullptr;
+ }
+}
+
+// TODO: (b/140549596)
+// Add wav files
+INSTANTIATE_TEST_SUITE_P(
+ AudioDecoderTest, C2DecoderTest,
+ ::testing::Values(make_pair("bbb_44100hz_2ch_128kbps_aac_30sec.mp4", "aac"),
+ make_pair("bbb_44100hz_2ch_128kbps_mp3_30sec.mp3", "mp3"),
+ make_pair("bbb_8000hz_1ch_8kbps_amrnb_30sec.3gp", "amrnb"),
+ make_pair("bbb_16000hz_1ch_9kbps_amrwb_30sec.3gp", "amrnb"),
+ make_pair("bbb_44100hz_2ch_80kbps_vorbis_30sec.webm", "vorbis"),
+ make_pair("bbb_44100hz_2ch_600kbps_flac_30sec.mp4", "flac"),
+ make_pair("bbb_48000hz_2ch_100kbps_opus_30sec.webm", "opus")));
+
+INSTANTIATE_TEST_SUITE_P(
+ VideoDecoderTest, C2DecoderTest,
+ ::testing::Values(make_pair("crowd_1920x1080_25fps_4000kbps_vp9.webm", "vp9"),
+ make_pair("crowd_1920x1080_25fps_4000kbps_vp8.webm", "vp8"),
+ make_pair("crowd_1920x1080_25fps_4000kbps_av1.webm", "av1"),
+ make_pair("crowd_1920x1080_25fps_7300kbps_mpeg2.mp4", "mpeg2"),
+ make_pair("crowd_1920x1080_25fps_6000kbps_mpeg4.mp4", "mpeg4"),
+ make_pair("crowd_352x288_25fps_6000kbps_h263.3gp", "h263"),
+ make_pair("crowd_1920x1080_25fps_6700kbps_h264.ts", "avc"),
+ make_pair("crowd_1920x1080_25fps_4000kbps_h265.mkv", "hevc")));
+
+int main(int argc, char **argv) {
+ gEnv = new BenchmarkTestEnvironment();
+ ::testing::AddGlobalTestEnvironment(gEnv);
+ ::testing::InitGoogleTest(&argc, argv);
+ int status = gEnv->initFromOptions(argc, argv);
+ if (status == 0) {
+ status = RUN_ALL_TESTS();
+ ALOGV("C2 Decoder Test result = %d\n", status);
+ }
+ return status;
+}
diff --git a/media/tests/benchmark/tests/C2EncoderTest.cpp b/media/tests/benchmark/tests/C2EncoderTest.cpp
new file mode 100644
index 0000000..98eb17a
--- /dev/null
+++ b/media/tests/benchmark/tests/C2EncoderTest.cpp
@@ -0,0 +1,187 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "C2EncoderTest"
+
+#include <fstream>
+#include <iostream>
+#include <limits>
+
+#include "BenchmarkTestEnvironment.h"
+#include "C2Encoder.h"
+#include "Decoder.h"
+
+static BenchmarkTestEnvironment *gEnv = nullptr;
+
+class C2EncoderTest : public ::testing::TestWithParam<pair<string, string>> {
+ public:
+ C2EncoderTest() : mEncoder(nullptr) {}
+
+ ~C2EncoderTest() {
+ if (!mCodecList.empty()) {
+ mCodecList.clear();
+ }
+ if (mEncoder) {
+ delete mEncoder;
+ mEncoder = nullptr;
+ }
+ }
+
+ virtual void SetUp() override { setupC2EncoderTest(); }
+
+ void setupC2EncoderTest();
+
+ vector<string> mCodecList;
+ C2Encoder *mEncoder;
+};
+
+void C2EncoderTest::setupC2EncoderTest() {
+ mEncoder = new C2Encoder();
+ ASSERT_NE(mEncoder, nullptr) << "C2Encoder creation failed";
+
+ int32_t status = mEncoder->setupCodec2();
+ ASSERT_EQ(status, 0) << "Codec2 setup failed";
+
+ mCodecList = mEncoder->getSupportedComponentList(true /* isEncoder*/);
+ ASSERT_GT(mCodecList.size(), 0) << "Codec2 client didn't recognise any component";
+}
+
+TEST_P(C2EncoderTest, Codec2Encode) {
+ ALOGV("Encodes the input using codec2 framework");
+ string inputFile = gEnv->getRes() + GetParam().first;
+ FILE *inputFp = fopen(inputFile.c_str(), "rb");
+ ASSERT_NE(inputFp, nullptr) << "Unable to open input file for reading";
+
+ Decoder *decoder = new Decoder();
+ ASSERT_NE(decoder, nullptr) << "Decoder creation failed";
+
+ Extractor *extractor = decoder->getExtractor();
+ ASSERT_NE(extractor, nullptr) << "Extractor creation failed";
+
+ // Read file properties
+ struct stat buf;
+ stat(inputFile.c_str(), &buf);
+ size_t fileSize = buf.st_size;
+ int32_t fd = fileno(inputFp);
+
+ ASSERT_LE(fileSize, kMaxBufferSize)
+ << "Input file size is greater than the threshold memory dedicated to the test";
+
+ int32_t trackCount = extractor->initExtractor(fd, fileSize);
+ ASSERT_GT(trackCount, 0) << "initExtractor failed";
+
+ for (int curTrack = 0; curTrack < trackCount; curTrack++) {
+ int32_t status = extractor->setupTrackFormat(curTrack);
+ ASSERT_EQ(status, 0) << "Track Format invalid";
+
+ uint8_t *inputBuffer = (uint8_t *)malloc(fileSize);
+ ASSERT_NE(inputBuffer, nullptr) << "Insufficient memory";
+
+ vector<AMediaCodecBufferInfo> frameInfo;
+ AMediaCodecBufferInfo info;
+ uint32_t inputBufferOffset = 0;
+
+ // Get frame data
+ while (1) {
+ status = extractor->getFrameSample(info);
+ if (status || !info.size) break;
+ // copy the meta data and buffer to be passed to decoder
+ ASSERT_LE(inputBufferOffset + info.size, fileSize) << "Memory allocated not sufficient";
+
+ memcpy(inputBuffer + inputBufferOffset, extractor->getFrameBuf(), info.size);
+ frameInfo.push_back(info);
+ inputBufferOffset += info.size;
+ }
+
+ string decName = "";
+ string outputFileName = "decode.out";
+ FILE *outFp = fopen(outputFileName.c_str(), "wb");
+ ASSERT_NE(outFp, nullptr) << "Unable to open output file" << outputFileName
+ << " for dumping decoder's output";
+
+ decoder->setupDecoder();
+ status = decoder->decode(inputBuffer, frameInfo, decName, false /*asyncMode */, outFp);
+ ASSERT_EQ(status, AMEDIA_OK) << "Decode returned error : " << status;
+
+ // Encode the given input stream for all C2 codecs supported by device
+ AMediaFormat *format = extractor->getFormat();
+ ifstream eleStream;
+ eleStream.open(outputFileName.c_str(), ifstream::binary | ifstream::ate);
+ ASSERT_EQ(eleStream.is_open(), true) << outputFileName.c_str() << " - file not found";
+ size_t eleSize = eleStream.tellg();
+
+ for (string codecName : mCodecList) {
+ if (codecName.find(GetParam().second) != string::npos) {
+ status = mEncoder->createCodec2Component(codecName, format);
+ ASSERT_EQ(status, 0) << "Create component failed for " << codecName;
+
+ // Send the inputs to C2 Encoder and wait till all buffers are returned.
+ eleStream.seekg(0, ifstream::beg);
+ status = mEncoder->encodeFrames(eleStream, eleSize);
+ ASSERT_EQ(status, 0) << "Encoder failed for " << codecName;
+
+ mEncoder->waitOnInputConsumption();
+ ASSERT_TRUE(mEncoder->mEos) << "Test Failed. Didn't receive EOS \n";
+
+ mEncoder->deInitCodec();
+ int64_t durationUs = extractor->getClipDuration();
+ ALOGV("codec : %s", codecName.c_str());
+ mEncoder->dumpStatistics(GetParam().first, durationUs);
+ mEncoder->resetEncoder();
+ }
+ }
+
+ // Destroy the decoder for the given input
+ decoder->deInitCodec();
+ decoder->resetDecoder();
+ free(inputBuffer);
+ }
+ fclose(inputFp);
+ extractor->deInitExtractor();
+ delete decoder;
+ delete mEncoder;
+ mEncoder = nullptr;
+}
+
+INSTANTIATE_TEST_SUITE_P(
+ AudioEncoderTest, C2EncoderTest,
+ ::testing::Values(make_pair("bbb_44100hz_2ch_128kbps_aac_30sec.mp4", "aac"),
+ make_pair("bbb_8000hz_1ch_8kbps_amrnb_30sec.3gp", "amrnb"),
+ make_pair("bbb_16000hz_1ch_9kbps_amrwb_30sec.3gp", "amrwb"),
+ make_pair("bbb_44100hz_2ch_600kbps_flac_30sec.mp4", "flac"),
+ make_pair("bbb_48000hz_2ch_100kbps_opus_30sec.webm", "opus")));
+
+INSTANTIATE_TEST_SUITE_P(
+ VideoEncoderTest, C2EncoderTest,
+ ::testing::Values(make_pair("crowd_1920x1080_25fps_4000kbps_vp9.webm", "vp9"),
+ make_pair("crowd_1920x1080_25fps_4000kbps_vp8.webm", "vp8"),
+ make_pair("crowd_176x144_25fps_6000kbps_mpeg4.mp4", "mpeg4"),
+ make_pair("crowd_176x144_25fps_6000kbps_h263.3gp", "h263"),
+ make_pair("crowd_1920x1080_25fps_6700kbps_h264.ts", "avc"),
+ make_pair("crowd_1920x1080_25fps_4000kbps_h265.mkv", "hevc")));
+
+int main(int argc, char **argv) {
+ gEnv = new BenchmarkTestEnvironment();
+ ::testing::AddGlobalTestEnvironment(gEnv);
+ ::testing::InitGoogleTest(&argc, argv);
+ int status = gEnv->initFromOptions(argc, argv);
+ if (status == 0) {
+ status = RUN_ALL_TESTS();
+ ALOGV("C2 Encoder Test result = %d\n", status);
+ }
+ return status;
+}
diff --git a/media/tests/benchmark/tests/DecoderTest.cpp b/media/tests/benchmark/tests/DecoderTest.cpp
new file mode 100644
index 0000000..9f96d3b
--- /dev/null
+++ b/media/tests/benchmark/tests/DecoderTest.cpp
@@ -0,0 +1,186 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "decoderTest"
+
+#include <fstream>
+#include <iostream>
+#include <limits>
+
+#include "BenchmarkTestEnvironment.h"
+#include "Decoder.h"
+
+static BenchmarkTestEnvironment *gEnv = nullptr;
+
+class DecoderTest : public ::testing::TestWithParam<tuple<string, string, bool>> {};
+
+TEST_P(DecoderTest, Decode) {
+ ALOGV("Decode the samples given by extractor");
+ tuple<string /* InputFile */, string /* CodecName */, bool /* asyncMode */> params = GetParam();
+
+ string inputFile = gEnv->getRes() + get<0>(params);
+ FILE *inputFp = fopen(inputFile.c_str(), "rb");
+ ASSERT_NE(inputFp, nullptr) << "Unable to open " << inputFile << " file for reading";
+
+ Decoder *decoder = new Decoder();
+ ASSERT_NE(decoder, nullptr) << "Decoder creation failed";
+
+ Extractor *extractor = decoder->getExtractor();
+ ASSERT_NE(extractor, nullptr) << "Extractor creation failed";
+
+ // Read file properties
+ struct stat buf;
+ stat(inputFile.c_str(), &buf);
+ size_t fileSize = buf.st_size;
+ int32_t fd = fileno(inputFp);
+
+ int32_t trackCount = extractor->initExtractor(fd, fileSize);
+ ASSERT_GT(trackCount, 0) << "initExtractor failed";
+
+ for (int curTrack = 0; curTrack < trackCount; curTrack++) {
+ int32_t status = extractor->setupTrackFormat(curTrack);
+ ASSERT_EQ(status, 0) << "Track Format invalid";
+
+ uint8_t *inputBuffer = (uint8_t *)malloc(kMaxBufferSize);
+ ASSERT_NE(inputBuffer, nullptr) << "Insufficient memory";
+
+ vector<AMediaCodecBufferInfo> frameInfo;
+ AMediaCodecBufferInfo info;
+ uint32_t inputBufferOffset = 0;
+
+ // Get frame data
+ while (1) {
+ status = extractor->getFrameSample(info);
+ if (status || !info.size) break;
+ // copy the meta data and buffer to be passed to decoder
+ ASSERT_LE(inputBufferOffset + info.size, kMaxBufferSize)
+ << "Memory allocated not sufficient";
+
+ memcpy(inputBuffer + inputBufferOffset, extractor->getFrameBuf(), info.size);
+ frameInfo.push_back(info);
+ inputBufferOffset += info.size;
+ }
+
+ string codecName = get<1>(params);
+ bool asyncMode = get<2>(params);
+ decoder->setupDecoder();
+ status = decoder->decode(inputBuffer, frameInfo, codecName, asyncMode);
+ ASSERT_EQ(status, AMEDIA_OK) << "Decoder failed for " << codecName;
+
+ decoder->deInitCodec();
+ ALOGV("codec : %s", codecName.c_str());
+ string inputReference = get<0>(params);
+ decoder->dumpStatistics(inputReference);
+ free(inputBuffer);
+ decoder->resetDecoder();
+ }
+ fclose(inputFp);
+ extractor->deInitExtractor();
+ delete decoder;
+}
+
+// TODO: (b/140549596)
+// Add wav files
+INSTANTIATE_TEST_SUITE_P(
+ AudioDecoderSyncTest, DecoderTest,
+ ::testing::Values(make_tuple("bbb_44100hz_2ch_128kbps_aac_30sec.mp4", "", false),
+ make_tuple("bbb_44100hz_2ch_128kbps_mp3_30sec.mp3", "", false),
+ make_tuple("bbb_8000hz_1ch_8kbps_amrnb_30sec.3gp", "", false),
+ make_tuple("bbb_16000hz_1ch_9kbps_amrwb_30sec.3gp", "", false),
+ make_tuple("bbb_44100hz_2ch_80kbps_vorbis_30sec.webm", "", false),
+ make_tuple("bbb_44100hz_2ch_600kbps_flac_30sec.mp4", "", false),
+ make_tuple("bbb_48000hz_2ch_100kbps_opus_30sec.webm", "", false)));
+
+INSTANTIATE_TEST_SUITE_P(
+ AudioDecoderAsyncTest, DecoderTest,
+ ::testing::Values(make_tuple("bbb_44100hz_2ch_128kbps_aac_30sec.mp4", "", true),
+ make_tuple("bbb_44100hz_2ch_128kbps_mp3_30sec.mp3", "", true),
+ make_tuple("bbb_8000hz_1ch_8kbps_amrnb_30sec.3gp", "", true),
+ make_tuple("bbb_16000hz_1ch_9kbps_amrwb_30sec.3gp", "", true),
+ make_tuple("bbb_44100hz_2ch_80kbps_vorbis_30sec.webm", "", true),
+ make_tuple("bbb_44100hz_2ch_600kbps_flac_30sec.mp4", "", true),
+ make_tuple("bbb_48000hz_2ch_100kbps_opus_30sec.webm", "", true)));
+
+INSTANTIATE_TEST_SUITE_P(VideDecoderSyncTest, DecoderTest,
+ ::testing::Values(
+ // Hardware codecs
+ make_tuple("crowd_1920x1080_25fps_4000kbps_vp9.webm", "", false),
+ make_tuple("crowd_1920x1080_25fps_4000kbps_vp8.webm", "", false),
+ make_tuple("crowd_1920x1080_25fps_4000kbps_av1.webm", "", false),
+ make_tuple("crowd_1920x1080_25fps_7300kbps_mpeg2.mp4", "", false),
+ make_tuple("crowd_1920x1080_25fps_6000kbps_mpeg4.mp4", "", false),
+ make_tuple("crowd_352x288_25fps_6000kbps_h263.3gp", "", false),
+ make_tuple("crowd_1920x1080_25fps_6700kbps_h264.ts", "", false),
+ make_tuple("crowd_1920x1080_25fps_4000kbps_h265.mkv", "", false),
+ // Software codecs
+ make_tuple("crowd_1920x1080_25fps_4000kbps_vp9.webm",
+ "c2.android.vp9.decoder", false),
+ make_tuple("crowd_1920x1080_25fps_4000kbps_vp8.webm",
+ "c2.android.vp8.decoder", false),
+ make_tuple("crowd_1920x1080_25fps_4000kbps_av1.webm",
+ "c2.android.av1.decoder", false),
+ make_tuple("crowd_1920x1080_25fps_7300kbps_mpeg2.mp4",
+ "c2.android.mpeg2.decoder", false),
+ make_tuple("crowd_1920x1080_25fps_6000kbps_mpeg4.mp4",
+ "c2.android.mpeg4.decoder", false),
+ make_tuple("crowd_352x288_25fps_6000kbps_h263.3gp",
+ "c2.android.h263.decoder", false),
+ make_tuple("crowd_1920x1080_25fps_6700kbps_h264.ts",
+ "c2.android.avc.decoder", false),
+ make_tuple("crowd_1920x1080_25fps_4000kbps_h265.mkv",
+ "c2.android.hevc.decoder", false)));
+
+INSTANTIATE_TEST_SUITE_P(VideoDecoderAsyncTest, DecoderTest,
+ ::testing::Values(
+ // Hardware codecs
+ make_tuple("crowd_1920x1080_25fps_4000kbps_vp9.webm", "", true),
+ make_tuple("crowd_1920x1080_25fps_4000kbps_vp8.webm", "", true),
+ make_tuple("crowd_1920x1080_25fps_4000kbps_av1.webm", "", true),
+ make_tuple("crowd_1920x1080_25fps_7300kbps_mpeg2.mp4", "", true),
+ make_tuple("crowd_1920x1080_25fps_6000kbps_mpeg4.mp4", "", true),
+ make_tuple("crowd_352x288_25fps_6000kbps_h263.3gp", "", true),
+ make_tuple("crowd_1920x1080_25fps_6700kbps_h264.ts", "", true),
+ make_tuple("crowd_1920x1080_25fps_4000kbps_h265.mkv", "", true),
+ // Software codecs
+ make_tuple("crowd_1920x1080_25fps_4000kbps_vp9.webm",
+ "c2.android.vp9.decoder", true),
+ make_tuple("crowd_1920x1080_25fps_4000kbps_vp8.webm",
+ "c2.android.vp8.decoder", true),
+ make_tuple("crowd_1920x1080_25fps_4000kbps_av1.webm",
+ "c2.android.av1.decoder", true),
+ make_tuple("crowd_1920x1080_25fps_7300kbps_mpeg2.mp4",
+ "c2.android.mpeg2.decoder", true),
+ make_tuple("crowd_1920x1080_25fps_6000kbps_mpeg4.mp4",
+ "c2.android.mpeg4.decoder", true),
+ make_tuple("crowd_352x288_25fps_6000kbps_h263.3gp",
+ "c2.android.h263.decoder", true),
+ make_tuple("crowd_1920x1080_25fps_6700kbps_h264.ts",
+ "c2.android.avc.decoder", true),
+ make_tuple("crowd_1920x1080_25fps_4000kbps_h265.mkv",
+ "c2.android.hevc.decoder", true)));
+
+int main(int argc, char **argv) {
+ gEnv = new BenchmarkTestEnvironment();
+ ::testing::AddGlobalTestEnvironment(gEnv);
+ ::testing::InitGoogleTest(&argc, argv);
+ int status = gEnv->initFromOptions(argc, argv);
+ if (status == 0) {
+ status = RUN_ALL_TESTS();
+ ALOGD("Decoder Test result = %d\n", status);
+ }
+ return status;
+}
\ No newline at end of file
diff --git a/media/tests/benchmark/tests/EncoderTest.cpp b/media/tests/benchmark/tests/EncoderTest.cpp
new file mode 100644
index 0000000..dc2a2dd
--- /dev/null
+++ b/media/tests/benchmark/tests/EncoderTest.cpp
@@ -0,0 +1,221 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "encoderTest"
+
+#include <fstream>
+
+#include "BenchmarkTestEnvironment.h"
+#include "Decoder.h"
+#include "Encoder.h"
+
+static BenchmarkTestEnvironment *gEnv = nullptr;
+
+class EncoderTest : public ::testing::TestWithParam<tuple<string, string, bool>> {};
+
+TEST_P(EncoderTest, Encode) {
+ ALOGD("Encode test for all codecs");
+ tuple<string /* InputFile */, string /* CodecName */, bool /* asyncMode */> params = GetParam();
+
+ string inputFile = gEnv->getRes() + get<0>(params);
+ FILE *inputFp = fopen(inputFile.c_str(), "rb");
+ ASSERT_NE(inputFp, nullptr) << "Unable to open " << inputFile << " file for reading";
+
+ Decoder *decoder = new Decoder();
+ ASSERT_NE(decoder, nullptr) << "Decoder creation failed";
+
+ Extractor *extractor = decoder->getExtractor();
+ ASSERT_NE(extractor, nullptr) << "Extractor creation failed";
+
+ // Read file properties
+ struct stat buf;
+ stat(inputFile.c_str(), &buf);
+ size_t fileSize = buf.st_size;
+ int32_t fd = fileno(inputFp);
+
+ int32_t trackCount = extractor->initExtractor(fd, fileSize);
+ ASSERT_GT(trackCount, 0) << "initExtractor failed";
+
+ Encoder *encoder = new Encoder();
+ ASSERT_NE(encoder, nullptr) << "Decoder creation failed";
+
+ for (int curTrack = 0; curTrack < trackCount; curTrack++) {
+ int32_t status = extractor->setupTrackFormat(curTrack);
+ ASSERT_EQ(status, 0) << "Track Format invalid";
+
+ uint8_t *inputBuffer = (uint8_t *)malloc(kMaxBufferSize);
+ ASSERT_NE(inputBuffer, nullptr) << "Insufficient memory";
+
+ vector<AMediaCodecBufferInfo> frameInfo;
+ AMediaCodecBufferInfo info;
+ uint32_t inputBufferOffset = 0;
+
+ // Get frame data
+ while (1) {
+ status = extractor->getFrameSample(info);
+ if (status || !info.size) break;
+ // copy the meta data and buffer to be passed to decoder
+ ASSERT_LE(inputBufferOffset + info.size, kMaxBufferSize)
+ << "Memory allocated not sufficient";
+
+ memcpy(inputBuffer + inputBufferOffset, extractor->getFrameBuf(), info.size);
+ frameInfo.push_back(info);
+ inputBufferOffset += info.size;
+ }
+
+ string decName = "";
+ string outputFileName = "decode.out";
+ FILE *outFp = fopen(outputFileName.c_str(), "wb");
+ ASSERT_NE(outFp, nullptr) << "Unable to open output file" << outputFileName
+ << " for dumping decoder's output";
+
+ decoder->setupDecoder();
+ status = decoder->decode(inputBuffer, frameInfo, decName, false /*asyncMode */, outFp);
+ ASSERT_EQ(status, AMEDIA_OK) << "Decode returned error : " << status;
+
+ ifstream eleStream;
+ eleStream.open(outputFileName.c_str(), ifstream::binary | ifstream::ate);
+ ASSERT_EQ(eleStream.is_open(), true) << outputFileName.c_str() << " - file not found";
+ size_t eleSize = eleStream.tellg();
+ eleStream.seekg(0, ifstream::beg);
+
+ AMediaFormat *format = extractor->getFormat();
+ const char *mime = nullptr;
+ AMediaFormat_getString(format, AMEDIAFORMAT_KEY_MIME, &mime);
+ ASSERT_NE(mime, nullptr) << "Invalid mime type";
+
+ // Get encoder params
+ encParameter encParams;
+ if (!strncmp(mime, "video/", 6)) {
+ ASSERT_TRUE(AMediaFormat_getInt32(format, AMEDIAFORMAT_KEY_WIDTH, &encParams.width));
+ ASSERT_TRUE(AMediaFormat_getInt32(format, AMEDIAFORMAT_KEY_HEIGHT, &encParams.height));
+ AMediaFormat_getInt32(format, AMEDIAFORMAT_KEY_FRAME_RATE, &encParams.frameRate);
+ AMediaFormat_getInt32(format, AMEDIAFORMAT_KEY_BIT_RATE, &encParams.bitrate);
+ if (encParams.bitrate <= 0 || encParams.frameRate <= 0) {
+ encParams.frameRate = 25;
+ if (!strcmp(mime, "video/3gpp") || !strcmp(mime, "video/mp4v-es")) {
+ encParams.bitrate = 600000 /* 600 Kbps */;
+ } else {
+ encParams.bitrate = 8000000 /* 8 Mbps */;
+ }
+ }
+ AMediaFormat_getInt32(format, AMEDIAFORMAT_KEY_PROFILE, &encParams.profile);
+ AMediaFormat_getInt32(format, AMEDIAFORMAT_KEY_LEVEL, &encParams.level);
+ } else {
+ ASSERT_TRUE(AMediaFormat_getInt32(format, AMEDIAFORMAT_KEY_SAMPLE_RATE,
+ &encParams.sampleRate));
+ ASSERT_TRUE(AMediaFormat_getInt32(format, AMEDIAFORMAT_KEY_CHANNEL_COUNT,
+ &encParams.numChannels));
+ encParams.bitrate =
+ encParams.sampleRate * encParams.numChannels * 16 /* bitsPerSample */;
+ }
+
+ encoder->setupEncoder();
+ string codecName = get<1>(params);
+ bool asyncMode = get<2>(params);
+ status = encoder->encode(codecName, eleStream, eleSize, asyncMode, encParams, (char *)mime);
+ ASSERT_EQ(status, 0) << "Encoder failed for " << codecName;
+
+ encoder->deInitCodec();
+ ALOGV("codec : %s", codecName.c_str());
+ string inputReference = get<0>(params);
+ encoder->dumpStatistics(inputReference, extractor->getClipDuration());
+ eleStream.close();
+ if (outFp) fclose(outFp);
+
+ if (format) {
+ AMediaFormat_delete(format);
+ format = nullptr;
+ }
+ encoder->resetEncoder();
+ decoder->deInitCodec();
+ free(inputBuffer);
+ decoder->resetDecoder();
+ }
+ delete encoder;
+ fclose(inputFp);
+ extractor->deInitExtractor();
+ delete decoder;
+}
+
+INSTANTIATE_TEST_SUITE_P(
+ AudioEncoderSyncTest, EncoderTest,
+ ::testing::Values(make_tuple("bbb_44100hz_2ch_128kbps_aac_30sec.mp4", "", false),
+ make_tuple("bbb_8000hz_1ch_8kbps_amrnb_30sec.3gp", "", false),
+ make_tuple("bbb_16000hz_1ch_9kbps_amrwb_30sec.3gp", "", false),
+ make_tuple("bbb_44100hz_2ch_600kbps_flac_30sec.mp4", "", false),
+ make_tuple("bbb_48000hz_2ch_100kbps_opus_30sec.webm", "", false)));
+
+INSTANTIATE_TEST_SUITE_P(
+ AudioEncoderAsyncTest, EncoderTest,
+ ::testing::Values(make_tuple("bbb_44100hz_2ch_128kbps_aac_30sec.mp4", "", true),
+ make_tuple("bbb_8000hz_1ch_8kbps_amrnb_30sec.3gp", "", true),
+ make_tuple("bbb_16000hz_1ch_9kbps_amrwb_30sec.3gp", "", true),
+ make_tuple("bbb_44100hz_2ch_600kbps_flac_30sec.mp4", "", true),
+ make_tuple("bbb_48000hz_2ch_100kbps_opus_30sec.webm", "", true)));
+
+INSTANTIATE_TEST_SUITE_P(VideEncoderSyncTest, EncoderTest,
+ ::testing::Values(
+ // Hardware codecs
+ make_tuple("crowd_1920x1080_25fps_4000kbps_vp8.webm", "", false),
+ make_tuple("crowd_1920x1080_25fps_6700kbps_h264.ts", "", false),
+ make_tuple("crowd_1920x1080_25fps_4000kbps_h265.mkv", "", false),
+ // Software codecs
+ make_tuple("crowd_1920x1080_25fps_4000kbps_vp9.webm",
+ "c2.android.vp9.encoder", false),
+ make_tuple("crowd_1920x1080_25fps_4000kbps_vp8.webm",
+ "c2.android.vp8.encoder", false),
+ make_tuple("crowd_176x144_25fps_6000kbps_mpeg4.mp4",
+ "c2.android.mpeg4.encoder", false),
+ make_tuple("crowd_176x144_25fps_6000kbps_h263.3gp",
+ "c2.android.h263.encoder", false),
+ make_tuple("crowd_1920x1080_25fps_6700kbps_h264.ts",
+ "c2.android.avc.encoder", false),
+ make_tuple("crowd_1920x1080_25fps_4000kbps_h265.mkv",
+ "c2.android.hevc.encoder", false)));
+
+INSTANTIATE_TEST_SUITE_P(VideoEncoderAsyncTest, EncoderTest,
+ ::testing::Values(
+ // Hardware codecs
+ make_tuple("crowd_1920x1080_25fps_4000kbps_vp8.webm", "", true),
+ make_tuple("crowd_1920x1080_25fps_6700kbps_h264.ts", "", true),
+ make_tuple("crowd_1920x1080_25fps_4000kbps_h265.mkv", "", true),
+ // Software codecs
+ make_tuple("crowd_1920x1080_25fps_4000kbps_vp9.webm",
+ "c2.android.vp9.encoder", true),
+ make_tuple("crowd_1920x1080_25fps_4000kbps_vp8.webm",
+ "c2.android.vp8.encoder", true),
+ make_tuple("crowd_176x144_25fps_6000kbps_mpeg4.mp4",
+ "c2.android.mpeg4.encoder", true),
+ make_tuple("crowd_176x144_25fps_6000kbps_h263.3gp",
+ "c2.android.h263.encoder", true),
+ make_tuple("crowd_1920x1080_25fps_6700kbps_h264.ts",
+ "c2.android.avc.encoder", true),
+ make_tuple("crowd_1920x1080_25fps_4000kbps_h265.mkv",
+ "c2.android.hevc.encoder", true)));
+
+int main(int argc, char **argv) {
+ gEnv = new BenchmarkTestEnvironment();
+ ::testing::AddGlobalTestEnvironment(gEnv);
+ ::testing::InitGoogleTest(&argc, argv);
+ int status = gEnv->initFromOptions(argc, argv);
+ if (status == 0) {
+ status = RUN_ALL_TESTS();
+ ALOGD("Encoder Test result = %d\n", status);
+ }
+ return status;
+}
diff --git a/media/tests/benchmark/tests/ExtractorTest.cpp b/media/tests/benchmark/tests/ExtractorTest.cpp
new file mode 100644
index 0000000..ad8f1e6
--- /dev/null
+++ b/media/tests/benchmark/tests/ExtractorTest.cpp
@@ -0,0 +1,86 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "extractorTest"
+
+#include <gtest/gtest.h>
+
+#include "BenchmarkTestEnvironment.h"
+#include "Extractor.h"
+
+static BenchmarkTestEnvironment *gEnv = nullptr;
+
+class ExtractorTest : public ::testing::TestWithParam<pair<string, int32_t>> {};
+
+TEST_P(ExtractorTest, Extract) {
+ Extractor *extractObj = new Extractor();
+ ASSERT_NE(extractObj, nullptr) << "Extractor creation failed";
+
+ string inputFile = gEnv->getRes() + GetParam().first;
+ FILE *inputFp = fopen(inputFile.c_str(), "rb");
+ ASSERT_NE(inputFp, nullptr) << "Unable to open " << inputFile << " file for reading";
+
+ // Read file properties
+ struct stat buf;
+ stat(inputFile.c_str(), &buf);
+ size_t fileSize = buf.st_size;
+ int32_t fd = fileno(inputFp);
+
+ int32_t trackCount = extractObj->initExtractor(fd, fileSize);
+ ASSERT_GT(trackCount, 0) << "initExtractor failed";
+
+ int32_t trackID = GetParam().second;
+ int32_t status = extractObj->extract(trackID);
+ ASSERT_EQ(status, AMEDIA_OK) << "Extraction failed \n";
+
+ extractObj->deInitExtractor();
+
+ extractObj->dumpStatistics(GetParam().first);
+
+ fclose(inputFp);
+ delete extractObj;
+}
+
+INSTANTIATE_TEST_SUITE_P(ExtractorTestAll, ExtractorTest,
+ ::testing::Values(make_pair("crowd_1920x1080_25fps_4000kbps_vp9.webm", 0),
+ make_pair("crowd_1920x1080_25fps_6000kbps_h263.3gp", 0),
+ make_pair("crowd_1920x1080_25fps_6000kbps_mpeg4.mp4", 0),
+ make_pair("crowd_1920x1080_25fps_6700kbps_h264.ts", 0),
+ make_pair("crowd_1920x1080_25fps_7300kbps_mpeg2.mp4", 0),
+ make_pair("crowd_1920x1080_25fps_4000kbps_av1.webm", 0),
+ make_pair("crowd_1920x1080_25fps_4000kbps_h265.mkv", 0),
+ make_pair("crowd_1920x1080_25fps_4000kbps_vp8.webm", 0),
+ make_pair("bbb_44100hz_2ch_128kbps_aac_5mins.mp4", 0),
+ make_pair("bbb_44100hz_2ch_128kbps_mp3_5mins.mp3", 0),
+ make_pair("bbb_44100hz_2ch_600kbps_flac_5mins.flac", 0),
+ make_pair("bbb_8000hz_1ch_8kbps_amrnb_5mins.3gp", 0),
+ make_pair("bbb_16000hz_1ch_9kbps_amrwb_5mins.3gp", 0),
+ make_pair("bbb_44100hz_2ch_80kbps_vorbis_5mins.webm", 0),
+ make_pair("bbb_48000hz_2ch_100kbps_opus_5mins.webm",
+ 0)));
+
+int main(int argc, char **argv) {
+ gEnv = new BenchmarkTestEnvironment();
+ ::testing::AddGlobalTestEnvironment(gEnv);
+ ::testing::InitGoogleTest(&argc, argv);
+ int status = gEnv->initFromOptions(argc, argv);
+ if (status == 0) {
+ status = RUN_ALL_TESTS();
+ ALOGD(" Extractor Test result = %d\n", status);
+ }
+ return status;
+}
diff --git a/media/tests/benchmark/tests/MuxerTest.cpp b/media/tests/benchmark/tests/MuxerTest.cpp
new file mode 100644
index 0000000..fa2635d
--- /dev/null
+++ b/media/tests/benchmark/tests/MuxerTest.cpp
@@ -0,0 +1,158 @@
+
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "muxerTest"
+
+#include <fstream>
+#include <iostream>
+
+#include "BenchmarkTestEnvironment.h"
+#include "Muxer.h"
+
+#define OUTPUT_FILE_NAME "/data/local/tmp/mux.out"
+
+static BenchmarkTestEnvironment *gEnv = nullptr;
+
+class MuxerTest : public ::testing::TestWithParam<pair<string, string>> {};
+
+static MUXER_OUTPUT_T getMuxerOutFormat(string fmt) {
+ static const struct {
+ string name;
+ MUXER_OUTPUT_T value;
+ } kFormatMaps[] = {{"mp4", MUXER_OUTPUT_FORMAT_MPEG_4},
+ {"webm", MUXER_OUTPUT_FORMAT_WEBM},
+ {"3gpp", MUXER_OUTPUT_FORMAT_3GPP},
+ {"ogg", MUXER_OUTPUT_FORMAT_OGG}};
+
+ MUXER_OUTPUT_T format = MUXER_OUTPUT_FORMAT_INVALID;
+ for (size_t i = 0; i < sizeof(kFormatMaps) / sizeof(kFormatMaps[0]); ++i) {
+ if (!fmt.compare(kFormatMaps[i].name)) {
+ format = kFormatMaps[i].value;
+ break;
+ }
+ }
+ return format;
+}
+
+TEST_P(MuxerTest, Mux) {
+ ALOGV("Mux the samples given by extractor");
+ string inputFile = gEnv->getRes() + GetParam().first;
+ FILE *inputFp = fopen(inputFile.c_str(), "rb");
+ ASSERT_NE(inputFp, nullptr) << "Unable to open " << inputFile << " file for reading";
+
+ string fmt = GetParam().second;
+ MUXER_OUTPUT_T outputFormat = getMuxerOutFormat(fmt);
+ ASSERT_NE(outputFormat, MUXER_OUTPUT_FORMAT_INVALID) << "Invalid muxer output format";
+
+ Muxer *muxerObj = new Muxer();
+ ASSERT_NE(muxerObj, nullptr) << "Muxer creation failed";
+
+ Extractor *extractor = muxerObj->getExtractor();
+ ASSERT_NE(extractor, nullptr) << "Extractor creation failed";
+
+ // Read file properties
+ struct stat buf;
+ stat(inputFile.c_str(), &buf);
+ size_t fileSize = buf.st_size;
+ int32_t fd = fileno(inputFp);
+
+ int32_t trackCount = extractor->initExtractor(fd, fileSize);
+ ASSERT_GT(trackCount, 0) << "initExtractor failed";
+
+ for (int curTrack = 0; curTrack < trackCount; curTrack++) {
+ int32_t status = extractor->setupTrackFormat(curTrack);
+ ASSERT_EQ(status, 0) << "Track Format invalid";
+
+ uint8_t *inputBuffer = (uint8_t *)malloc(kMaxBufferSize);
+ ASSERT_NE(inputBuffer, nullptr) << "Insufficient memory";
+
+ // AMediaCodecBufferInfo : <size of frame> <flags> <presentationTimeUs> <offset>
+ vector<AMediaCodecBufferInfo> frameInfos;
+ AMediaCodecBufferInfo info;
+ uint32_t inputBufferOffset = 0;
+
+ // Get Frame Data
+ while (1) {
+ status = extractor->getFrameSample(info);
+ if (status || !info.size) break;
+ // copy the meta data and buffer to be passed to muxer
+ ASSERT_LE(inputBufferOffset + info.size, kMaxBufferSize)
+ << "Memory allocated not sufficient";
+
+ memcpy(inputBuffer + inputBufferOffset, extractor->getFrameBuf(), info.size);
+ info.offset = inputBufferOffset;
+ frameInfos.push_back(info);
+ inputBufferOffset += info.size;
+ }
+
+ string outputFileName = OUTPUT_FILE_NAME;
+ FILE *outputFp = fopen(outputFileName.c_str(), "w+b");
+ ASSERT_NE(outputFp, nullptr)
+ << "Unable to open output file" << outputFileName << " for writing";
+
+ int32_t fd = fileno(outputFp);
+ status = muxerObj->initMuxer(fd, outputFormat);
+ ASSERT_EQ(status, 0) << "initMuxer failed";
+
+ status = muxerObj->mux(inputBuffer, frameInfos);
+ ASSERT_EQ(status, 0) << "Mux failed";
+
+ muxerObj->deInitMuxer();
+ muxerObj->dumpStatistics(GetParam().first + "." + fmt.c_str());
+ free(inputBuffer);
+ fclose(outputFp);
+ muxerObj->resetMuxer();
+ }
+ fclose(inputFp);
+ extractor->deInitExtractor();
+ delete muxerObj;
+}
+
+INSTANTIATE_TEST_SUITE_P(
+ MuxerTestAll, MuxerTest,
+ ::testing::Values(make_pair("crowd_1920x1080_25fps_4000kbps_vp8.webm", "webm"),
+ make_pair("crowd_1920x1080_25fps_4000kbps_vp9.webm", "webm"),
+ make_pair("crowd_1920x1080_25fps_6000kbps_mpeg4.mp4", "mp4"),
+ make_pair("crowd_352x288_25fps_6000kbps_h263.3gp", "mp4"),
+ make_pair("crowd_1920x1080_25fps_6700kbps_h264.ts", "mp4"),
+ make_pair("crowd_1920x1080_25fps_4000kbps_h265.mkv", "mp4"),
+ make_pair("crowd_1920x1080_25fps_6000kbps_mpeg4.mp4", "3gpp"),
+ make_pair("crowd_352x288_25fps_6000kbps_h263.3gp", "3gpp"),
+ make_pair("crowd_1920x1080_25fps_6700kbps_h264.ts", "3gpp"),
+ make_pair("crowd_1920x1080_25fps_4000kbps_h265.mkv", "3gpp"),
+ make_pair("bbb_48000hz_2ch_100kbps_opus_5mins.webm", "ogg"),
+ make_pair("bbb_44100hz_2ch_80kbps_vorbis_5mins.webm", "webm"),
+ make_pair("bbb_48000hz_2ch_100kbps_opus_5mins.webm", "webm"),
+ make_pair("bbb_44100hz_2ch_128kbps_aac_5mins.mp4", "mp4"),
+ make_pair("bbb_8000hz_1ch_8kbps_amrnb_5mins.3gp", "mp4"),
+ make_pair("bbb_16000hz_1ch_9kbps_amrwb_5mins.3gp", "mp4"),
+ make_pair("bbb_44100hz_2ch_128kbps_aac_5mins.mp4", "3gpp"),
+ make_pair("bbb_8000hz_1ch_8kbps_amrnb_5mins.3gp", "3gpp"),
+ make_pair("bbb_16000hz_1ch_9kbps_amrwb_5mins.3gp", "3gpp")));
+
+int main(int argc, char **argv) {
+ gEnv = new BenchmarkTestEnvironment();
+ ::testing::AddGlobalTestEnvironment(gEnv);
+ ::testing::InitGoogleTest(&argc, argv);
+ int status = gEnv->initFromOptions(argc, argv);
+ if (status == 0) {
+ status = RUN_ALL_TESTS();
+ ALOGV("Test result = %d\n", status);
+ }
+ return status;
+}
diff --git a/media/utils/Android.bp b/media/utils/Android.bp
index e2cd4e3..5047b19 100644
--- a/media/utils/Android.bp
+++ b/media/utils/Android.bp
@@ -25,12 +25,14 @@
"ServiceUtilities.cpp",
"TimeCheck.cpp",
],
+ static_libs: [
+ "libc_malloc_debug_backtrace",
+ ],
shared_libs: [
"libbinder",
"libcutils",
"liblog",
"libutils",
- "libmemunreachable",
"libhidlbase",
"android.hardware.graphics.bufferqueue@1.0",
"android.hidl.token@1.0-utils",
@@ -44,15 +46,14 @@
"-Werror",
],
- product_variables: {
- product_is_iot: {
- cflags: ["-DTARGET_ANDROID_THINGS"],
- },
- },
+ header_libs: [
+ "bionic_libc_platform_headers",
+ "libmedia_headers",
+ ],
include_dirs: [
- // For android_mallopt definitions.
- "bionic/libc/private"
+ // For DEBUGGER_SIGNAL
+ "system/core/debuggerd/include",
],
local_include_dirs: ["include"],
export_include_dirs: ["include"],
diff --git a/media/utils/MemoryLeakTrackUtil.cpp b/media/utils/MemoryLeakTrackUtil.cpp
index 2988b52..6166859 100644
--- a/media/utils/MemoryLeakTrackUtil.cpp
+++ b/media/utils/MemoryLeakTrackUtil.cpp
@@ -22,7 +22,7 @@
#include "media/MemoryLeakTrackUtil.h"
#include <sstream>
-#include <bionic_malloc.h>
+#include <bionic/malloc.h>
/*
* The code here originally resided in MediaPlayerService.cpp
diff --git a/media/utils/ProcessInfo.cpp b/media/utils/ProcessInfo.cpp
index 27f1a79..113e4a7 100644
--- a/media/utils/ProcessInfo.cpp
+++ b/media/utils/ProcessInfo.cpp
@@ -23,6 +23,7 @@
#include <binder/IPCThreadState.h>
#include <binder/IProcessInfoService.h>
#include <binder/IServiceManager.h>
+#include <private/android_filesystem_config.h>
namespace android {
@@ -55,8 +56,9 @@
bool ProcessInfo::isValidPid(int pid) {
int callingPid = IPCThreadState::self()->getCallingPid();
+ int callingUid = IPCThreadState::self()->getCallingUid();
// Trust it if this is called from the same process otherwise pid has to match the calling pid.
- return (callingPid == getpid()) || (callingPid == pid);
+ return (callingPid == getpid()) || (callingPid == pid) || (callingUid == AID_MEDIA);
}
ProcessInfo::~ProcessInfo() {}
diff --git a/media/utils/ServiceUtilities.cpp b/media/utils/ServiceUtilities.cpp
index bc8fff6..a661470 100644
--- a/media/utils/ServiceUtilities.cpp
+++ b/media/utils/ServiceUtilities.cpp
@@ -174,23 +174,19 @@
}
bool modifyDefaultAudioEffectsAllowed() {
+ return modifyDefaultAudioEffectsAllowed(
+ IPCThreadState::self()->getCallingPid(), IPCThreadState::self()->getCallingUid());
+}
+
+bool modifyDefaultAudioEffectsAllowed(pid_t pid, uid_t uid) {
+ if (isAudioServerUid(IPCThreadState::self()->getCallingUid())) return true;
+
static const String16 sModifyDefaultAudioEffectsAllowed(
"android.permission.MODIFY_DEFAULT_AUDIO_EFFECTS");
// IMPORTANT: Use PermissionCache - not a runtime permission and may not change.
- bool ok = PermissionCache::checkCallingPermission(sModifyDefaultAudioEffectsAllowed);
-
-#ifdef TARGET_ANDROID_THINGS
- if (!ok) {
- // Use a secondary permission on Android Things to allow a more lenient level of protection.
- static const String16 sModifyDefaultAudioEffectsAndroidThingsAllowed(
- "com.google.android.things.permission.MODIFY_DEFAULT_AUDIO_EFFECTS");
- ok = PermissionCache::checkCallingPermission(
- sModifyDefaultAudioEffectsAndroidThingsAllowed);
- }
- if (!ok) ALOGE("com.google.android.things.permission.MODIFY_DEFAULT_AUDIO_EFFECTS");
-#else
- if (!ok) ALOGE("android.permission.MODIFY_DEFAULT_AUDIO_EFFECTS");
-#endif
+ bool ok = PermissionCache::checkPermission(sModifyDefaultAudioEffectsAllowed, pid, uid);
+ ALOGE_IF(!ok, "%s(): android.permission.MODIFY_DEFAULT_AUDIO_EFFECTS denied for uid %d",
+ __func__, uid);
return ok;
}
diff --git a/media/utils/TimeCheck.cpp b/media/utils/TimeCheck.cpp
index f16776f..4a3e470 100644
--- a/media/utils/TimeCheck.cpp
+++ b/media/utils/TimeCheck.cpp
@@ -14,13 +14,50 @@
* limitations under the License.
*/
+#define LOG_TAG "TimeCheck"
#include <utils/Log.h>
-#include <media/TimeCheck.h>
-#include <media/EventLog.h>
+#include <mediautils/TimeCheck.h>
+#include <mediautils/EventLog.h>
+#include "debuggerd/handler.h"
namespace android {
+// Audio HAL server pids vector used to generate audio HAL processes tombstone
+// when audioserver watchdog triggers.
+// We use a lockless storage to avoid potential deadlocks in the context of watchdog
+// trigger.
+// Protection again simultaneous writes is not needed given one update takes place
+// during AudioFlinger construction and other comes necessarily later once the IAudioFlinger
+// interface is available.
+// The use of an atomic index just guaranties that current vector is fully initialized
+// when read.
+/* static */
+void TimeCheck::accessAudioHalPids(std::vector<pid_t>* pids, bool update) {
+ static constexpr int kNumAudioHalPidsVectors = 3;
+ static std::vector<pid_t> audioHalPids[kNumAudioHalPidsVectors];
+ static std::atomic<int> curAudioHalPids = 0;
+
+ if (update) {
+ audioHalPids[(curAudioHalPids + 1) % kNumAudioHalPidsVectors] = *pids;
+ curAudioHalPids++;
+ } else {
+ *pids = audioHalPids[curAudioHalPids];
+ }
+}
+
+/* static */
+void TimeCheck::setAudioHalPids(const std::vector<pid_t>& pids) {
+ accessAudioHalPids(&(const_cast<std::vector<pid_t>&>(pids)), true);
+}
+
+/* static */
+std::vector<pid_t> TimeCheck::getAudioHalPids() {
+ std::vector<pid_t> pids;
+ accessAudioHalPids(&pids, false);
+ return pids;
+}
+
/* static */
sp<TimeCheck::TimeCheckThread> TimeCheck::getTimeCheckThread()
{
@@ -83,6 +120,18 @@
status = mCond.waitRelative(mMutex, waitTimeNs);
}
if (status != NO_ERROR) {
+ // Generate audio HAL processes tombstones and allow time to complete
+ // before forcing restart
+ std::vector<pid_t> pids = getAudioHalPids();
+ if (pids.size() != 0) {
+ for (const auto& pid : pids) {
+ ALOGI("requesting tombstone for pid: %d", pid);
+ sigqueue(pid, DEBUGGER_SIGNAL, {.sival_int = 0});
+ }
+ sleep(1);
+ } else {
+ ALOGI("No HAL process pid available, skipping tombstones");
+ }
LOG_EVENT_STRING(LOGTAG_AUDIO_BINDER_TIMEOUT, tag);
LOG_ALWAYS_FATAL("TimeCheck timeout for %s", tag);
}
diff --git a/media/utils/include/mediautils/ServiceUtilities.h b/media/utils/include/mediautils/ServiceUtilities.h
index e1089d5..f5768bd 100644
--- a/media/utils/include/mediautils/ServiceUtilities.h
+++ b/media/utils/include/mediautils/ServiceUtilities.h
@@ -85,6 +85,7 @@
bool settingsAllowed();
bool modifyAudioRoutingAllowed();
bool modifyDefaultAudioEffectsAllowed();
+bool modifyDefaultAudioEffectsAllowed(pid_t pid, uid_t uid);
bool dumpAllowed();
bool modifyPhoneStateAllowed(pid_t pid, uid_t uid);
bool bypassInterruptionPolicyAllowed(pid_t pid, uid_t uid);
diff --git a/media/utils/include/mediautils/TimeCheck.h b/media/utils/include/mediautils/TimeCheck.h
index 6c5f656..5ba6d7c 100644
--- a/media/utils/include/mediautils/TimeCheck.h
+++ b/media/utils/include/mediautils/TimeCheck.h
@@ -20,7 +20,7 @@
#include <utils/KeyedVector.h>
#include <utils/Thread.h>
-
+#include <vector>
namespace android {
@@ -35,6 +35,8 @@
TimeCheck(const char *tag, uint32_t timeoutMs = kDefaultTimeOutMs);
~TimeCheck();
+ static void setAudioHalPids(const std::vector<pid_t>& pids);
+ static std::vector<pid_t> getAudioHalPids();
private:
@@ -63,6 +65,7 @@
};
static sp<TimeCheckThread> getTimeCheckThread();
+ static void accessAudioHalPids(std::vector<pid_t>* pids, bool update);
const nsecs_t mEndTimeNs;
};
diff --git a/services/audioflinger/Android.bp b/services/audioflinger/Android.bp
index 96ad54b..c58360d 100644
--- a/services/audioflinger/Android.bp
+++ b/services/audioflinger/Android.bp
@@ -9,6 +9,7 @@
"AudioStreamOut.cpp",
"AudioWatchdog.cpp",
"BufLog.cpp",
+ "DeviceEffectManager.cpp",
"Effects.cpp",
"FastCapture.cpp",
"FastCaptureDumpState.cpp",
@@ -34,6 +35,7 @@
],
shared_libs: [
+ "libaudiofoundation",
"libaudiohal",
"libaudioprocessing",
"libaudiospdif",
@@ -60,6 +62,10 @@
"libsndfile",
],
+ header_libs: [
+ "libmedia_headers",
+ ],
+
cflags: [
"-DSTATE_QUEUE_INSTANTIATIONS=\"StateQueueInstantiations.cpp\"",
"-fvisibility=hidden",
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 8bbdc69..10a9c63 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -19,6 +19,9 @@
#define LOG_TAG "AudioFlinger"
//#define LOG_NDEBUG 0
+// Define AUDIO_ARRAYS_STATIC_CHECK to check all audio arrays are correct
+#define AUDIO_ARRAYS_STATIC_CHECK 1
+
#include "Configuration.h"
#include <dirent.h>
#include <math.h>
@@ -67,6 +70,7 @@
#include <media/nbaio/PipeReader.h>
#include <mediautils/BatteryNotifier.h>
#include <mediautils/ServiceUtilities.h>
+#include <mediautils/TimeCheck.h>
#include <private/android_filesystem_config.h>
//#define BUFLOG_NDEBUG 0
@@ -166,6 +170,7 @@
mClientSharedHeapSize(kMinimumClientSharedHeapSizeBytes),
mGlobalEffectEnableTime(0),
mPatchPanel(this),
+ mDeviceEffectManager(this),
mSystemReady(false)
{
// unsigned instead of audio_unique_id_use_t, because ++ operator is unavailable for enum
@@ -190,6 +195,9 @@
mEffectsFactoryHal = EffectsFactoryHalInterface::create();
mMediaLogNotifier->run("MediaLogNotifier");
+ std::vector<pid_t> halPids;
+ mDevicesFactoryHal->getHalPids(&halPids);
+ TimeCheck::setAudioHalPids(halPids);
}
void AudioFlinger::onFirstRef()
@@ -215,6 +223,11 @@
gAudioFlinger = this;
}
+status_t AudioFlinger::setAudioHalPids(const std::vector<pid_t>& pids) {
+ TimeCheck::setAudioHalPids(pids);
+ return NO_ERROR;
+}
+
AudioFlinger::~AudioFlinger()
{
while (!mRecordThreads.isEmpty()) {
@@ -370,6 +383,24 @@
}
}
+status_t AudioFlinger::addEffectToHal(audio_port_handle_t deviceId,
+ audio_module_handle_t hwModuleId, sp<EffectHalInterface> effect) {
+ AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(hwModuleId);
+ if (audioHwDevice == nullptr) {
+ return NO_INIT;
+ }
+ return audioHwDevice->hwDevice()->addDeviceEffect(deviceId, effect);
+}
+
+status_t AudioFlinger::removeEffectFromHal(audio_port_handle_t deviceId,
+ audio_module_handle_t hwModuleId, sp<EffectHalInterface> effect) {
+ AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(hwModuleId);
+ if (audioHwDevice == nullptr) {
+ return NO_INIT;
+ }
+ return audioHwDevice->hwDevice()->removeDeviceEffect(deviceId, effect);
+}
+
static const char * const audio_interfaces[] = {
AUDIO_HARDWARE_MODULE_ID_PRIMARY,
AUDIO_HARDWARE_MODULE_ID_A2DP,
@@ -378,7 +409,7 @@
AudioHwDevice* AudioFlinger::findSuitableHwDev_l(
audio_module_handle_t module,
- audio_devices_t devices)
+ audio_devices_t deviceType)
{
// if module is 0, the request comes from an old policy manager and we should load
// well known modules
@@ -393,7 +424,7 @@
sp<DeviceHalInterface> dev = audioHwDevice->hwDevice();
uint32_t supportedDevices;
if (dev->getSupportedDevices(&supportedDevices) == OK &&
- (supportedDevices & devices) == devices) {
+ (supportedDevices & deviceType) == deviceType) {
return audioHwDevice;
}
}
@@ -546,6 +577,8 @@
mPatchPanel.dump(fd);
+ mDeviceEffectManager.dump(fd);
+
// dump external setParameters
auto dumpLogger = [fd](SimpleLog& logger, const char* name) {
dprintf(fd, "\n%s setParameters:\n", name);
@@ -1354,6 +1387,13 @@
}
}
+void AudioFlinger::updateOutDevicesForRecordThreads_l(const DeviceDescriptorBaseVector& devices)
+{
+ for (size_t i = 0; i < mRecordThreads.size(); i++) {
+ mRecordThreads.valueAt(i)->updateOutDevices(devices);
+ }
+}
+
// forwardAudioHwSyncToDownstreamPatches_l() must be called with AudioFlinger::mLock held
void AudioFlinger::forwardParametersToDownstreamPatches_l(
audio_io_handle_t upStream, const String8& keyValuePairs,
@@ -1384,8 +1424,8 @@
String8(AudioParameter::keyFrameCount),
String8(AudioParameter::keyInputSource),
String8(AudioParameter::keyMonoOutput),
- String8(AudioParameter::keyStreamConnect),
- String8(AudioParameter::keyStreamDisconnect),
+ String8(AudioParameter::keyDeviceConnect),
+ String8(AudioParameter::keyDeviceDisconnect),
String8(AudioParameter::keyStreamSupportedFormats),
String8(AudioParameter::keyStreamSupportedChannels),
String8(AudioParameter::keyStreamSupportedSamplingRates),
@@ -1570,7 +1610,7 @@
proposed.format = format;
sp<DeviceHalInterface> dev = mPrimaryHardwareDev->hwDevice();
- size_t frames;
+ size_t frames = 0;
for (;;) {
// Note: config is currently a const parameter for get_input_buffer_size()
// but we use a copy from proposed in case config changes from the call.
@@ -2301,13 +2341,13 @@
sp<AudioFlinger::ThreadBase> AudioFlinger::openOutput_l(audio_module_handle_t module,
- audio_io_handle_t *output,
- audio_config_t *config,
- audio_devices_t devices,
- const String8& address,
- audio_output_flags_t flags)
+ audio_io_handle_t *output,
+ audio_config_t *config,
+ audio_devices_t deviceType,
+ const String8& address,
+ audio_output_flags_t flags)
{
- AudioHwDevice *outHwDev = findSuitableHwDev_l(module, devices);
+ AudioHwDevice *outHwDev = findSuitableHwDev_l(module, deviceType);
if (outHwDev == NULL) {
return 0;
}
@@ -2348,7 +2388,7 @@
status_t status = outHwDev->openOutputStream(
&outputStream,
*output,
- devices,
+ deviceType,
flags,
config,
address.string());
@@ -2358,8 +2398,7 @@
if (status == NO_ERROR) {
if (flags & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) {
sp<MmapPlaybackThread> thread =
- new MmapPlaybackThread(this, *output, outHwDev, outputStream,
- devices, AUDIO_DEVICE_NONE, mSystemReady);
+ new MmapPlaybackThread(this, *output, outHwDev, outputStream, mSystemReady);
mMmapThreads.add(*output, thread);
ALOGV("openOutput_l() created mmap playback thread: ID %d thread %p",
*output, thread.get());
@@ -2367,17 +2406,17 @@
} else {
sp<PlaybackThread> thread;
if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
- thread = new OffloadThread(this, outputStream, *output, devices, mSystemReady);
+ thread = new OffloadThread(this, outputStream, *output, mSystemReady);
ALOGV("openOutput_l() created offload output: ID %d thread %p",
*output, thread.get());
} else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT)
|| !isValidPcmSinkFormat(config->format)
|| !isValidPcmSinkChannelMask(config->channel_mask)) {
- thread = new DirectOutputThread(this, outputStream, *output, devices, mSystemReady);
+ thread = new DirectOutputThread(this, outputStream, *output, mSystemReady);
ALOGV("openOutput_l() created direct output: ID %d thread %p",
*output, thread.get());
} else {
- thread = new MixerThread(this, outputStream, *output, devices, mSystemReady);
+ thread = new MixerThread(this, outputStream, *output, mSystemReady);
ALOGV("openOutput_l() created mixer output: ID %d thread %p",
*output, thread.get());
}
@@ -2393,27 +2432,29 @@
status_t AudioFlinger::openOutput(audio_module_handle_t module,
audio_io_handle_t *output,
audio_config_t *config,
- audio_devices_t *devices,
- const String8& address,
+ const sp<DeviceDescriptorBase>& device,
uint32_t *latencyMs,
audio_output_flags_t flags)
{
- ALOGI("openOutput() this %p, module %d Device %#x, SamplingRate %d, Format %#08x, "
+ ALOGI("openOutput() this %p, module %d Device %s, SamplingRate %d, Format %#08x, "
"Channels %#x, flags %#x",
this, module,
- (devices != NULL) ? *devices : 0,
+ device->toString().c_str(),
config->sample_rate,
config->format,
config->channel_mask,
flags);
- if (devices == NULL || *devices == AUDIO_DEVICE_NONE) {
+ audio_devices_t deviceType = device->type();
+ const String8 address = String8(device->address().c_str());
+
+ if (deviceType == AUDIO_DEVICE_NONE) {
return BAD_VALUE;
}
Mutex::Autolock _l(mLock);
- sp<ThreadBase> thread = openOutput_l(module, output, config, *devices, address, flags);
+ sp<ThreadBase> thread = openOutput_l(module, output, config, deviceType, address, flags);
if (thread != 0) {
if ((flags & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) == 0) {
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
@@ -2683,9 +2724,7 @@
AudioStreamIn *inputStream = new AudioStreamIn(inHwDev, inStream, flags);
if ((flags & AUDIO_INPUT_FLAG_MMAP_NOIRQ) != 0) {
sp<MmapCaptureThread> thread =
- new MmapCaptureThread(this, *input,
- inHwDev, inputStream,
- primaryOutputDevice_l(), devices, mSystemReady);
+ new MmapCaptureThread(this, *input, inHwDev, inputStream, mSystemReady);
mMmapThreads.add(*input, thread);
ALOGV("openInput_l() created mmap capture thread: ID %d thread %p", *input,
thread.get());
@@ -2694,13 +2733,7 @@
// Start record thread
// RecordThread requires both input and output device indication to forward to audio
// pre processing modules
- sp<RecordThread> thread = new RecordThread(this,
- inputStream,
- *input,
- primaryOutputDevice_l(),
- devices,
- mSystemReady
- );
+ sp<RecordThread> thread = new RecordThread(this, inputStream, *input, mSystemReady);
mRecordThreads.add(*input, thread);
ALOGV("openInput_l() created record thread: ID %d thread %p", *input, thread.get());
return thread;
@@ -2932,7 +2965,7 @@
Mutex::Autolock _l(t->mLock);
for (size_t j = 0; j < t->mEffectChains.size(); j++) {
sp<EffectChain> ec = t->mEffectChains[j];
- if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) {
+ if (!audio_is_global_session(ec->sessionId())) {
chains.push(ec);
}
}
@@ -2959,7 +2992,7 @@
for (size_t i = 0; i < chains.size(); i++) {
sp<EffectChain> ec = chains[i];
int sessionid = ec->sessionId();
- sp<ThreadBase> t = ec->mThread.promote();
+ sp<ThreadBase> t = ec->thread().promote();
if (t == 0) {
continue;
}
@@ -2982,7 +3015,7 @@
effect->unPin();
t->removeEffect_l(effect, /*release*/ true);
if (effect->purgeHandles()) {
- t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId());
+ effect->checkSuspendOnEffectEnabled(false, true /*threadLocked*/);
}
removedEffects.push_back(effect);
}
@@ -3116,15 +3149,15 @@
return NULL;
}
-audio_devices_t AudioFlinger::primaryOutputDevice_l() const
+DeviceTypeSet AudioFlinger::primaryOutputDevice_l() const
{
PlaybackThread *thread = primaryPlaybackThread_l();
if (thread == NULL) {
- return 0;
+ return DeviceTypeSet();
}
- return thread->outDevice();
+ return thread->outDeviceTypes();
}
AudioFlinger::PlaybackThread *AudioFlinger::fastPlaybackThread_l() const
@@ -3282,6 +3315,7 @@
int32_t priority,
audio_io_handle_t io,
audio_session_t sessionId,
+ const AudioDeviceTypeAddr& device,
const String16& opPackageName,
pid_t pid,
status_t *status,
@@ -3334,6 +3368,17 @@
lStatus = BAD_VALUE;
goto Exit;
}
+ } else if (sessionId == AUDIO_SESSION_DEVICE) {
+ if (!modifyDefaultAudioEffectsAllowed(pid, callingUid)) {
+ ALOGE("%s: device effect permission denied for uid %d", __func__, callingUid);
+ lStatus = PERMISSION_DENIED;
+ goto Exit;
+ }
+ if (io != AUDIO_IO_HANDLE_NONE) {
+ ALOGE("%s: io handle should not be specified for device effect", __func__);
+ lStatus = BAD_VALUE;
+ goto Exit;
+ }
} else {
// general sessionId.
@@ -3369,7 +3414,7 @@
// check recording permission for visualizer
if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
// TODO: Do we need to start/stop op - i.e. is there recording being performed?
- !recordingAllowed(opPackageName, pid, IPCThreadState::self()->getCallingUid())) {
+ !recordingAllowed(opPackageName, pid, callingUid)) {
lStatus = PERMISSION_DENIED;
goto Exit;
}
@@ -3386,6 +3431,23 @@
Mutex::Autolock _l(mLock);
+ if (sessionId == AUDIO_SESSION_DEVICE) {
+ sp<Client> client = registerPid(pid);
+ ALOGV("%s device type %d address %s", __func__, device.mType, device.getAddress());
+ handle = mDeviceEffectManager.createEffect_l(
+ &desc, device, client, effectClient, mPatchPanel.patches_l(),
+ enabled, &lStatus);
+ if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) {
+ // remove local strong reference to Client with mClientLock held
+ Mutex::Autolock _cl(mClientLock);
+ client.clear();
+ } else {
+ // handle must be valid here, but check again to be safe.
+ if (handle.get() != nullptr && id != nullptr) *id = handle->id();
+ }
+ goto Register;
+ }
+
// If output is not specified try to find a matching audio session ID in one of the
// output threads.
// If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX
@@ -3467,7 +3529,7 @@
sp<Client> client = registerPid(pid);
// create effect on selected output thread
- bool pinned = (sessionId > AUDIO_SESSION_OUTPUT_MIX) && isSessionAcquired_l(sessionId);
+ bool pinned = !audio_is_global_session(sessionId) && isSessionAcquired_l(sessionId);
handle = thread->createEffect_l(client, effectClient, priority, sessionId,
&desc, enabled, &lStatus, pinned);
if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) {
@@ -3480,9 +3542,10 @@
}
}
+Register:
if (lStatus == NO_ERROR || lStatus == ALREADY_EXISTS) {
// Check CPU and memory usage
- sp<EffectModule> effect = handle->effect().promote();
+ sp<EffectBase> effect = handle->effect().promote();
if (effect != nullptr) {
status_t rStatus = effect->updatePolicyState();
if (rStatus != NO_ERROR) {
@@ -3592,7 +3655,7 @@
// if the move request is not received from audio policy manager, the effect must be
// re-registered with the new strategy and output
if (dstChain == 0) {
- dstChain = effect->chain().promote();
+ dstChain = effect->callback()->chain().promote();
if (dstChain == 0) {
ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
status = NO_INIT;
@@ -3642,7 +3705,7 @@
goto Exit;
}
- dstChain = effect->chain().promote();
+ dstChain = effect->callback()->chain().promote();
if (dstChain == 0) {
thread->addEffect_l(effect);
status = INVALID_OPERATION;
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 72e669a..a43a6dc 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -65,7 +65,10 @@
#include <media/audiohal/EffectBufferHalInterface.h>
#include <media/audiohal/StreamHalInterface.h>
#include <media/AudioBufferProvider.h>
+#include <media/AudioContainers.h>
+#include <media/AudioDeviceTypeAddr.h>
#include <media/AudioMixer.h>
+#include <media/DeviceDescriptorBase.h>
#include <media/ExtendedAudioBufferProvider.h>
#include <media/LinearMap.h>
#include <media/VolumeShaper.h>
@@ -175,8 +178,7 @@
virtual status_t openOutput(audio_module_handle_t module,
audio_io_handle_t *output,
audio_config_t *config,
- audio_devices_t *devices,
- const String8& address,
+ const sp<DeviceDescriptorBase>& device,
uint32_t *latencyMs,
audio_output_flags_t flags);
@@ -230,6 +232,7 @@
int32_t priority,
audio_io_handle_t io,
audio_session_t sessionId,
+ const AudioDeviceTypeAddr& device,
const String16& opPackageName,
pid_t pid,
status_t *status /*non-NULL*/,
@@ -279,6 +282,8 @@
virtual status_t getMicrophones(std::vector<media::MicrophoneInfo> *microphones);
+ virtual status_t setAudioHalPids(const std::vector<pid_t>& pids);
+
virtual status_t onTransact(
uint32_t code,
const Parcel& data,
@@ -303,6 +308,12 @@
static int onExternalVibrationStart(const sp<os::ExternalVibration>& externalVibration);
static void onExternalVibrationStop(const sp<os::ExternalVibration>& externalVibration);
+
+ status_t addEffectToHal(audio_port_handle_t deviceId,
+ audio_module_handle_t hwModuleId, sp<EffectHalInterface> effect);
+ status_t removeEffectFromHal(audio_port_handle_t deviceId,
+ audio_module_handle_t hwModuleId, sp<EffectHalInterface> effect);
+
private:
// FIXME The 400 is temporarily too high until a leak of writers in media.log is fixed.
static const size_t kLogMemorySize = 400 * 1024;
@@ -372,7 +383,7 @@
virtual void onFirstRef();
AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module,
- audio_devices_t devices);
+ audio_devices_t deviceType);
// Set kEnableExtendedChannels to true to enable greater than stereo output
// for the MixerThread and device sink. Number of channels allowed is
@@ -528,9 +539,14 @@
class AsyncCallbackThread;
class Track;
class RecordTrack;
+ class EffectBase;
class EffectModule;
class EffectHandle;
class EffectChain;
+ class DeviceEffectProxy;
+ class DeviceEffectManager;
+ class PatchPanel;
+ class DeviceEffectManagerCallback;
struct AudioStreamIn;
struct TeePatch;
@@ -547,6 +563,16 @@
bool mute;
};
+ // Abstraction for the Audio Source for the RecordThread (HAL or PassthruPatchRecord).
+ struct Source
+ {
+ virtual ~Source() = default;
+ // The following methods have the same signatures as in StreamHalInterface.
+ virtual status_t read(void *buffer, size_t bytes, size_t *read) = 0;
+ virtual status_t getCapturePosition(int64_t *frames, int64_t *time) = 0;
+ virtual status_t standby() = 0;
+ };
+
// --- PlaybackThread ---
#ifdef FLOAT_EFFECT_CHAIN
#define EFFECT_BUFFER_FORMAT AUDIO_FORMAT_PCM_FLOAT
@@ -558,9 +584,11 @@
#include "Threads.h"
+#include "PatchPanel.h"
+
#include "Effects.h"
-#include "PatchPanel.h"
+#include "DeviceEffectManager.h"
// Find io handle by session id.
// Preference is given to an io handle with a matching effect chain to session id.
@@ -668,11 +696,11 @@
audio_devices_t outputDevice,
const String8& outputDeviceAddress);
sp<ThreadBase> openOutput_l(audio_module_handle_t module,
- audio_io_handle_t *output,
- audio_config_t *config,
- audio_devices_t devices,
- const String8& address,
- audio_output_flags_t flags);
+ audio_io_handle_t *output,
+ audio_config_t *config,
+ audio_devices_t deviceType,
+ const String8& address,
+ audio_output_flags_t flags);
void closeOutputFinish(const sp<PlaybackThread>& thread);
void closeInputFinish(const sp<RecordThread>& thread);
@@ -707,7 +735,7 @@
// return thread associated with primary hardware device, or NULL
PlaybackThread *primaryPlaybackThread_l() const;
- audio_devices_t primaryOutputDevice_l() const;
+ DeviceTypeSet primaryOutputDevice_l() const;
// return the playback thread with smallest HAL buffer size, and prefer fast
PlaybackThread *fastPlaybackThread_l() const;
@@ -741,6 +769,7 @@
std::vector< sp<EffectModule> > purgeStaleEffects_l();
void broacastParametersToRecordThreads_l(const String8& keyValuePairs);
+ void updateOutDevicesForRecordThreads_l(const DeviceDescriptorBaseVector& devices);
void forwardParametersToDownstreamPatches_l(
audio_io_handle_t upStream, const String8& keyValuePairs,
std::function<bool(const sp<PlaybackThread>&)> useThread = nullptr);
@@ -749,7 +778,7 @@
// For emphasis, we could also make all pointers to them be "const *",
// but that would clutter the code unnecessarily.
- struct AudioStreamIn {
+ struct AudioStreamIn : public Source {
AudioHwDevice* const audioHwDev;
sp<StreamInHalInterface> stream;
audio_input_flags_t flags;
@@ -758,6 +787,13 @@
AudioStreamIn(AudioHwDevice *dev, sp<StreamInHalInterface> in, audio_input_flags_t flags) :
audioHwDev(dev), stream(in), flags(flags) {}
+ status_t read(void *buffer, size_t bytes, size_t *read) override {
+ return stream->read(buffer, bytes, read);
+ }
+ status_t getCapturePosition(int64_t *frames, int64_t *time) override {
+ return stream->getCapturePosition(frames, time);
+ }
+ status_t standby() override { return stream->standby(); }
};
struct TeePatch {
@@ -898,6 +934,8 @@
PatchPanel mPatchPanel;
sp<EffectsFactoryHalInterface> mEffectsFactoryHal;
+ DeviceEffectManager mDeviceEffectManager;
+
bool mSystemReady;
SimpleLog mRejectedSetParameterLog;
diff --git a/services/audioflinger/AudioHwDevice.cpp b/services/audioflinger/AudioHwDevice.cpp
index b109d06..dda164c 100644
--- a/services/audioflinger/AudioHwDevice.cpp
+++ b/services/audioflinger/AudioHwDevice.cpp
@@ -34,7 +34,7 @@
status_t AudioHwDevice::openOutputStream(
AudioStreamOut **ppStreamOut,
audio_io_handle_t handle,
- audio_devices_t devices,
+ audio_devices_t deviceType,
audio_output_flags_t flags,
struct audio_config *config,
const char *address)
@@ -50,7 +50,7 @@
config->sample_rate,
config->format,
config->channel_mask);
- status_t status = outputStream->open(handle, devices, config, address);
+ status_t status = outputStream->open(handle, deviceType, config, address);
if (status != NO_ERROR) {
delete outputStream;
@@ -75,7 +75,7 @@
if (wrapperNeeded) {
if (SPDIFEncoder::isFormatSupported(originalConfig.format)) {
outputStream = new SpdifStreamOut(this, flags, originalConfig.format);
- status = outputStream->open(handle, devices, &originalConfig, address);
+ status = outputStream->open(handle, deviceType, &originalConfig, address);
if (status != NO_ERROR) {
ALOGE("ERROR - openOutputStream(), SPDIF open returned %d",
status);
diff --git a/services/audioflinger/AudioHwDevice.h b/services/audioflinger/AudioHwDevice.h
index d4299b0..6709d17 100644
--- a/services/audioflinger/AudioHwDevice.h
+++ b/services/audioflinger/AudioHwDevice.h
@@ -76,7 +76,7 @@
status_t openOutputStream(
AudioStreamOut **ppStreamOut,
audio_io_handle_t handle,
- audio_devices_t devices,
+ audio_devices_t deviceType,
audio_output_flags_t flags,
struct audio_config *config,
const char *address);
diff --git a/services/audioflinger/AudioStreamOut.cpp b/services/audioflinger/AudioStreamOut.cpp
index a60a5f2..d13cb8f 100644
--- a/services/audioflinger/AudioStreamOut.cpp
+++ b/services/audioflinger/AudioStreamOut.cpp
@@ -118,7 +118,7 @@
status_t AudioStreamOut::open(
audio_io_handle_t handle,
- audio_devices_t devices,
+ audio_devices_t deviceType,
struct audio_config *config,
const char *address)
{
@@ -130,7 +130,7 @@
int status = hwDev()->openOutputStream(
handle,
- devices,
+ deviceType,
customFlags,
config,
address,
@@ -152,7 +152,7 @@
status = hwDev()->openOutputStream(
handle,
- devices,
+ deviceType,
customFlags,
&customConfig,
address,
diff --git a/services/audioflinger/AudioStreamOut.h b/services/audioflinger/AudioStreamOut.h
index b16b1af..16fbcf2 100644
--- a/services/audioflinger/AudioStreamOut.h
+++ b/services/audioflinger/AudioStreamOut.h
@@ -47,7 +47,7 @@
virtual status_t open(
audio_io_handle_t handle,
- audio_devices_t devices,
+ audio_devices_t deviceType,
struct audio_config *config,
const char *address);
diff --git a/services/audioflinger/DeviceEffectManager.cpp b/services/audioflinger/DeviceEffectManager.cpp
new file mode 100644
index 0000000..87a4c6e
--- /dev/null
+++ b/services/audioflinger/DeviceEffectManager.cpp
@@ -0,0 +1,277 @@
+/*
+**
+** Copyright 2019, The Android Open Source Project
+**
+** Licensed under the Apache License, Version 2.0 (the "License");
+** you may not use this file except in compliance with the License.
+** You may obtain a copy of the License at
+**
+** http://www.apache.org/licenses/LICENSE-2.0
+**
+** Unless required by applicable law or agreed to in writing, software
+** distributed under the License is distributed on an "AS IS" BASIS,
+** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+** See the License for the specific language governing permissions and
+** limitations under the License.
+*/
+
+
+#define LOG_TAG "AudioFlinger::DeviceEffectManager"
+//#define LOG_NDEBUG 0
+
+#include <utils/Log.h>
+#include <audio_utils/primitives.h>
+
+#include "AudioFlinger.h"
+#include <media/audiohal/EffectsFactoryHalInterface.h>
+
+// ----------------------------------------------------------------------------
+
+
+namespace android {
+
+void AudioFlinger::DeviceEffectManager::createAudioPatch(audio_patch_handle_t handle,
+ const PatchPanel::Patch& patch) {
+ ALOGV("%s handle %d mHalHandle %d num sinks %d device sink %08x",
+ __func__, handle, patch.mHalHandle,
+ patch.mAudioPatch.num_sinks,
+ patch.mAudioPatch.num_sinks > 0 ? patch.mAudioPatch.sinks[0].ext.device.type : 0);
+
+ mCommandThread->createAudioPatchCommand(handle, patch);
+}
+
+void AudioFlinger::DeviceEffectManager::onCreateAudioPatch(audio_patch_handle_t handle,
+ const PatchPanel::Patch& patch) {
+ ALOGV("%s handle %d mHalHandle %d device sink %08x",
+ __func__, handle, patch.mHalHandle,
+ patch.mAudioPatch.num_sinks > 0 ? patch.mAudioPatch.sinks[0].ext.device.type : 0);
+ Mutex::Autolock _l(mLock);
+ for (auto& effect : mDeviceEffects) {
+ status_t status = effect.second->onCreatePatch(handle, patch);
+ ALOGV("%s Effect onCreatePatch status %d", __func__, status);
+ ALOGW_IF(status == BAD_VALUE, "%s onCreatePatch error %d", __func__, status);
+ }
+}
+
+void AudioFlinger::DeviceEffectManager::releaseAudioPatch(audio_patch_handle_t handle) {
+ ALOGV("%s", __func__);
+ mCommandThread->releaseAudioPatchCommand(handle);
+}
+
+void AudioFlinger::DeviceEffectManager::onReleaseAudioPatch(audio_patch_handle_t handle) {
+ ALOGV("%s", __func__);
+ Mutex::Autolock _l(mLock);
+ for (auto& effect : mDeviceEffects) {
+ effect.second->onReleasePatch(handle);
+ }
+}
+
+// DeviceEffectManager::createEffect_l() must be called with AudioFlinger::mLock held
+sp<AudioFlinger::EffectHandle> AudioFlinger::DeviceEffectManager::createEffect_l(
+ effect_descriptor_t *descriptor,
+ const AudioDeviceTypeAddr& device,
+ const sp<AudioFlinger::Client>& client,
+ const sp<IEffectClient>& effectClient,
+ const std::map<audio_patch_handle_t, PatchPanel::Patch>& patches,
+ int *enabled,
+ status_t *status) {
+ sp<DeviceEffectProxy> effect;
+ sp<EffectHandle> handle;
+ status_t lStatus;
+
+ lStatus = checkEffectCompatibility(descriptor);
+ if (lStatus != NO_ERROR) {
+ *status = lStatus;
+ return handle;
+ }
+
+ {
+ Mutex::Autolock _l(mLock);
+ auto iter = mDeviceEffects.find(device);
+ if (iter != mDeviceEffects.end()) {
+ effect = iter->second;
+ } else {
+ effect = new DeviceEffectProxy(device, mMyCallback,
+ descriptor, mAudioFlinger.nextUniqueId(AUDIO_UNIQUE_ID_USE_EFFECT));
+ }
+ // create effect handle and connect it to effect module
+ handle = new EffectHandle(effect, client, effectClient, 0 /*priority*/);
+ lStatus = handle->initCheck();
+ if (lStatus == NO_ERROR) {
+ lStatus = effect->addHandle(handle.get());
+ if (lStatus == NO_ERROR) {
+ effect->init(patches);
+ mDeviceEffects.emplace(device, effect);
+ }
+ }
+ }
+ if (enabled != NULL) {
+ *enabled = (int)effect->isEnabled();
+ }
+ *status = lStatus;
+ return handle;
+}
+
+status_t AudioFlinger::DeviceEffectManager::checkEffectCompatibility(
+ const effect_descriptor_t *desc) {
+
+ if ((desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC
+ && (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_POST_PROC) {
+ ALOGW("%s() non pre/post processing device effect %s", __func__, desc->name);
+ return BAD_VALUE;
+ }
+
+ return NO_ERROR;
+}
+
+status_t AudioFlinger::DeviceEffectManager::createEffectHal(
+ const effect_uuid_t *pEffectUuid, int32_t sessionId, int32_t deviceId,
+ sp<EffectHalInterface> *effect) {
+ status_t status = NO_INIT;
+ sp<EffectsFactoryHalInterface> effectsFactory = mAudioFlinger.getEffectsFactory();
+ if (effectsFactory != 0) {
+ status = effectsFactory->createEffect(
+ pEffectUuid, sessionId, AUDIO_IO_HANDLE_NONE, deviceId, effect);
+ }
+ return status;
+}
+
+void AudioFlinger::DeviceEffectManager::dump(int fd) {
+ const bool locked = dumpTryLock(mLock);
+ if (!locked) {
+ String8 result("DeviceEffectManager may be deadlocked\n");
+ write(fd, result.string(), result.size());
+ }
+
+ write(fd, "\nDevice Effects:\n", sizeof("\nDevice Effects:\n"));
+ for (const auto& iter : mDeviceEffects) {
+ String8 outStr;
+ outStr.appendFormat("%*sEffect for device %s address %s:\n", 2, "",
+ ::android::toString(iter.first.mType).c_str(), iter.first.getAddress());
+ write(fd, outStr.string(), outStr.size());
+ iter.second->dump(fd, 4);
+ }
+
+ if (locked) {
+ mLock.unlock();
+ }
+}
+
+
+size_t AudioFlinger::DeviceEffectManager::removeEffect(const sp<DeviceEffectProxy>& effect)
+{
+ Mutex::Autolock _l(mLock);
+ mDeviceEffects.erase(effect->device());
+ return mDeviceEffects.size();
+}
+
+bool AudioFlinger::DeviceEffectManagerCallback::disconnectEffectHandle(
+ EffectHandle *handle, bool unpinIfLast) {
+ sp<EffectBase> effectBase = handle->effect().promote();
+ if (effectBase == nullptr) {
+ return false;
+ }
+
+ sp<DeviceEffectProxy> effect = effectBase->asDeviceEffectProxy();
+ if (effect == nullptr) {
+ return false;
+ }
+ // restore suspended effects if the disconnected handle was enabled and the last one.
+ bool remove = (effect->removeHandle(handle) == 0) && (!effect->isPinned() || unpinIfLast);
+ if (remove) {
+ mManager.removeEffect(effect);
+ if (handle->enabled()) {
+ effectBase->checkSuspendOnEffectEnabled(false, false /*threadLocked*/);
+ }
+ }
+ return true;
+}
+
+// ----------- DeviceEffectManager::CommandThread implementation ----------
+
+
+AudioFlinger::DeviceEffectManager::CommandThread::~CommandThread()
+{
+ Mutex::Autolock _l(mLock);
+ mCommands.clear();
+}
+
+void AudioFlinger::DeviceEffectManager::CommandThread::onFirstRef()
+{
+ run("DeviceEffectManage_CommandThread", ANDROID_PRIORITY_AUDIO);
+}
+
+bool AudioFlinger::DeviceEffectManager::CommandThread::threadLoop()
+{
+ mLock.lock();
+ while (!exitPending())
+ {
+ while (!mCommands.empty() && !exitPending()) {
+ sp<Command> command = mCommands.front();
+ mCommands.pop_front();
+ mLock.unlock();
+
+ switch (command->mCommand) {
+ case CREATE_AUDIO_PATCH: {
+ CreateAudioPatchData *data = (CreateAudioPatchData *)command->mData.get();
+ ALOGV("CommandThread() processing create audio patch handle %d", data->mHandle);
+ mManager.onCreateAudioPatch(data->mHandle, data->mPatch);
+ } break;
+ case RELEASE_AUDIO_PATCH: {
+ ReleaseAudioPatchData *data = (ReleaseAudioPatchData *)command->mData.get();
+ ALOGV("CommandThread() processing release audio patch handle %d", data->mHandle);
+ mManager.onReleaseAudioPatch(data->mHandle);
+ } break;
+ default:
+ ALOGW("CommandThread() unknown command %d", command->mCommand);
+ }
+ mLock.lock();
+ }
+
+ // At this stage we have either an empty command queue or the first command in the queue
+ // has a finite delay. So unless we are exiting it is safe to wait.
+ if (!exitPending()) {
+ ALOGV("CommandThread() going to sleep");
+ mWaitWorkCV.wait(mLock);
+ }
+ }
+ mLock.unlock();
+ return false;
+}
+
+void AudioFlinger::DeviceEffectManager::CommandThread::sendCommand(sp<Command> command) {
+ Mutex::Autolock _l(mLock);
+ mCommands.push_back(command);
+ mWaitWorkCV.signal();
+}
+
+void AudioFlinger::DeviceEffectManager::CommandThread::createAudioPatchCommand(
+ audio_patch_handle_t handle, const PatchPanel::Patch& patch)
+{
+ sp<Command> command = new Command(CREATE_AUDIO_PATCH, new CreateAudioPatchData(handle, patch));
+ ALOGV("CommandThread() adding create patch handle %d mHalHandle %d.", handle, patch.mHalHandle);
+ sendCommand(command);
+}
+
+void AudioFlinger::DeviceEffectManager::CommandThread::releaseAudioPatchCommand(
+ audio_patch_handle_t handle)
+{
+ sp<Command> command = new Command(RELEASE_AUDIO_PATCH, new ReleaseAudioPatchData(handle));
+ ALOGV("CommandThread() adding release patch");
+ sendCommand(command);
+}
+
+void AudioFlinger::DeviceEffectManager::CommandThread::exit()
+{
+ ALOGV("CommandThread::exit");
+ {
+ AutoMutex _l(mLock);
+ requestExit();
+ mWaitWorkCV.signal();
+ }
+ // Note that we can call it from the thread loop if all other references have been released
+ // but it will safely return WOULD_BLOCK in this case
+ requestExitAndWait();
+}
+
+} // namespace android
diff --git a/services/audioflinger/DeviceEffectManager.h b/services/audioflinger/DeviceEffectManager.h
new file mode 100644
index 0000000..14ff14d
--- /dev/null
+++ b/services/audioflinger/DeviceEffectManager.h
@@ -0,0 +1,203 @@
+/*
+**
+** Copyright 2019, The Android Open Source Project
+**
+** Licensed under the Apache License, Version 2.0 (the "License");
+** you may not use this file except in compliance with the License.
+** You may obtain a copy of the License at
+**
+** http://www.apache.org/licenses/LICENSE-2.0
+**
+** Unless required by applicable law or agreed to in writing, software
+** distributed under the License is distributed on an "AS IS" BASIS,
+** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+** See the License for the specific language governing permissions and
+** limitations under the License.
+*/
+
+#ifndef INCLUDING_FROM_AUDIOFLINGER_H
+ #error This header file should only be included from AudioFlinger.h
+#endif
+
+// DeviceEffectManager is concealed within AudioFlinger, their lifetimes are the same.
+class DeviceEffectManager {
+public:
+ explicit DeviceEffectManager(AudioFlinger* audioFlinger)
+ : mCommandThread(new CommandThread(*this)), mAudioFlinger(*audioFlinger),
+ mMyCallback(new DeviceEffectManagerCallback(this)) {}
+
+ ~DeviceEffectManager() {
+ mCommandThread->exit();
+ }
+
+ sp<EffectHandle> createEffect_l(effect_descriptor_t *descriptor,
+ const AudioDeviceTypeAddr& device,
+ const sp<AudioFlinger::Client>& client,
+ const sp<IEffectClient>& effectClient,
+ const std::map<audio_patch_handle_t, PatchPanel::Patch>& patches,
+ int *enabled,
+ status_t *status);
+ void createAudioPatch(audio_patch_handle_t handle, const PatchPanel::Patch& patch);
+ void releaseAudioPatch(audio_patch_handle_t handle);
+
+ size_t removeEffect(const sp<DeviceEffectProxy>& effect);
+ status_t createEffectHal(const effect_uuid_t *pEffectUuid,
+ int32_t sessionId, int32_t deviceId,
+ sp<EffectHalInterface> *effect);
+ status_t addEffectToHal(audio_port_handle_t deviceId, audio_module_handle_t hwModuleId,
+ sp<EffectHalInterface> effect) {
+ return mAudioFlinger.addEffectToHal(deviceId, hwModuleId, effect);
+ };
+ status_t removeEffectFromHal(audio_port_handle_t deviceId, audio_module_handle_t hwModuleId,
+ sp<EffectHalInterface> effect) {
+ return mAudioFlinger.removeEffectFromHal(deviceId, hwModuleId, effect);
+ };
+
+ AudioFlinger& audioFlinger() const { return mAudioFlinger; }
+
+ void dump(int fd);
+
+private:
+
+ // Thread to execute create and release patch commands asynchronously. This is needed because
+ // PatchPanel::createAudioPatch and releaseAudioPatch are executed from audio policy service
+ // with mutex locked and effect management requires to call back into audio policy service
+ class Command;
+ class CommandThread : public Thread {
+ public:
+
+ enum {
+ CREATE_AUDIO_PATCH,
+ RELEASE_AUDIO_PATCH,
+ };
+
+ CommandThread(DeviceEffectManager& manager)
+ : Thread(false), mManager(manager) {}
+ ~CommandThread() override;
+
+ // Thread virtuals
+ void onFirstRef() override;
+ bool threadLoop() override;
+
+ void exit();
+
+ void createAudioPatchCommand(audio_patch_handle_t handle,
+ const PatchPanel::Patch& patch);
+ void releaseAudioPatchCommand(audio_patch_handle_t handle);
+
+ private:
+ class CommandData;
+
+ // descriptor for requested tone playback event
+ class Command: public RefBase {
+ public:
+ Command() = default;
+ Command(int command, sp<CommandData> data)
+ : mCommand(command), mData(data) {}
+
+ int mCommand = -1;
+ sp<CommandData> mData;
+ };
+
+ class CommandData: public RefBase {
+ public:
+ virtual ~CommandData() = default;
+ };
+
+ class CreateAudioPatchData : public CommandData {
+ public:
+ CreateAudioPatchData(audio_patch_handle_t handle, const PatchPanel::Patch& patch)
+ : mHandle(handle), mPatch(patch) {}
+
+ audio_patch_handle_t mHandle;
+ const PatchPanel::Patch mPatch;
+ };
+
+ class ReleaseAudioPatchData : public CommandData {
+ public:
+ ReleaseAudioPatchData(audio_patch_handle_t handle)
+ : mHandle(handle) {}
+
+ audio_patch_handle_t mHandle;
+ };
+
+ void sendCommand(sp<Command> command);
+
+ Mutex mLock;
+ Condition mWaitWorkCV;
+ std::deque <sp<Command>> mCommands; // list of pending commands
+ DeviceEffectManager& mManager;
+ };
+
+ void onCreateAudioPatch(audio_patch_handle_t handle, const PatchPanel::Patch& patch);
+ void onReleaseAudioPatch(audio_patch_handle_t handle);
+
+ status_t checkEffectCompatibility(const effect_descriptor_t *desc);
+
+ Mutex mLock;
+ sp<CommandThread> mCommandThread;
+ AudioFlinger &mAudioFlinger;
+ const sp<DeviceEffectManagerCallback> mMyCallback;
+ std::map<AudioDeviceTypeAddr, sp<DeviceEffectProxy>> mDeviceEffects;
+};
+
+class DeviceEffectManagerCallback : public EffectCallbackInterface {
+public:
+ DeviceEffectManagerCallback(DeviceEffectManager *manager)
+ : mManager(*manager) {}
+
+ status_t createEffectHal(const effect_uuid_t *pEffectUuid,
+ int32_t sessionId, int32_t deviceId,
+ sp<EffectHalInterface> *effect) override {
+ return mManager.createEffectHal(pEffectUuid, sessionId, deviceId, effect);
+ }
+ status_t allocateHalBuffer(size_t size __unused,
+ sp<EffectBufferHalInterface>* buffer __unused) override { return NO_ERROR; }
+ bool updateOrphanEffectChains(const sp<EffectBase>& effect __unused) override { return false; }
+
+ audio_io_handle_t io() const override { return AUDIO_IO_HANDLE_NONE; }
+ bool isOutput() const override { return false; }
+ bool isOffload() const override { return false; }
+ bool isOffloadOrDirect() const override { return false; }
+ bool isOffloadOrMmap() const override { return false; }
+
+ uint32_t sampleRate() const override { return 0; }
+ audio_channel_mask_t channelMask() const override { return AUDIO_CHANNEL_NONE; }
+ uint32_t channelCount() const override { return 0; }
+ size_t frameCount() const override { return 0; }
+ uint32_t latency() const override { return 0; }
+
+ status_t addEffectToHal(sp<EffectHalInterface> effect __unused) override {
+ return NO_ERROR;
+ }
+ status_t removeEffectFromHal(sp<EffectHalInterface> effect __unused) override {
+ return NO_ERROR;
+ }
+
+ bool disconnectEffectHandle(EffectHandle *handle, bool unpinIfLast) override;
+ void setVolumeForOutput(float left __unused, float right __unused) const override {}
+
+ // check if effects should be suspended or restored when a given effect is enable or disabled
+ void checkSuspendOnEffectEnabled(const sp<EffectBase>& effect __unused,
+ bool enabled __unused, bool threadLocked __unused) override {}
+ void resetVolume() override {}
+ uint32_t strategy() const override { return 0; }
+ int32_t activeTrackCnt() const override { return 0; }
+ void onEffectEnable(const sp<EffectBase>& effect __unused) override {}
+ void onEffectDisable(const sp<EffectBase>& effect __unused) override {}
+
+ wp<EffectChain> chain() const override { return nullptr; }
+
+ int newEffectId() { return mManager.audioFlinger().nextUniqueId(AUDIO_UNIQUE_ID_USE_EFFECT); }
+
+ status_t addEffectToHal(audio_port_handle_t deviceId,
+ audio_module_handle_t hwModuleId, sp<EffectHalInterface> effect) {
+ return mManager.addEffectToHal(deviceId, hwModuleId, effect);
+ }
+ status_t removeEffectFromHal(audio_port_handle_t deviceId,
+ audio_module_handle_t hwModuleId, sp<EffectHalInterface> effect) {
+ return mManager.removeEffectFromHal(deviceId, hwModuleId, effect);
+ }
+private:
+ DeviceEffectManager& mManager;
+};
diff --git a/services/audioflinger/Effects.cpp b/services/audioflinger/Effects.cpp
index 13152d0..ca8fd2d 100644
--- a/services/audioflinger/Effects.cpp
+++ b/services/audioflinger/Effects.cpp
@@ -29,7 +29,9 @@
#include <system/audio_effects/effect_visualizer.h>
#include <audio_utils/channels.h>
#include <audio_utils/primitives.h>
+#include <media/AudioContainers.h>
#include <media/AudioEffect.h>
+#include <media/AudioDeviceTypeAddr.h>
#include <media/audiohal/EffectHalInterface.h>
#include <media/audiohal/EffectsFactoryHalInterface.h>
#include <mediautils/ServiceUtilities.h>
@@ -56,81 +58,115 @@
namespace android {
// ----------------------------------------------------------------------------
-// EffectModule implementation
+// EffectBase implementation
// ----------------------------------------------------------------------------
#undef LOG_TAG
-#define LOG_TAG "AudioFlinger::EffectModule"
+#define LOG_TAG "AudioFlinger::EffectBase"
-AudioFlinger::EffectModule::EffectModule(ThreadBase *thread,
- const wp<AudioFlinger::EffectChain>& chain,
+AudioFlinger::EffectBase::EffectBase(const sp<AudioFlinger::EffectCallbackInterface>& callback,
effect_descriptor_t *desc,
int id,
audio_session_t sessionId,
bool pinned)
: mPinned(pinned),
- mThread(thread), mChain(chain), mId(id), mSessionId(sessionId),
- mDescriptor(*desc),
- // clear mConfig to ensure consistent initial value of buffer framecount
- // in case buffers are associated by setInBuffer() or setOutBuffer()
- // prior to configure().
- mConfig{{}, {}},
- mStatus(NO_INIT), mState(IDLE),
- mMaxDisableWaitCnt(1), // set by configure(), should be >= 1
- mDisableWaitCnt(0), // set by process() and updateState()
- mSuspended(false),
- mOffloaded(false),
- mAudioFlinger(thread->mAudioFlinger)
-#ifdef FLOAT_EFFECT_CHAIN
- , mSupportsFloat(false)
-#endif
+ mCallback(callback), mId(id), mSessionId(sessionId),
+ mDescriptor(*desc)
{
- ALOGV("Constructor %p pinned %d", this, pinned);
- int lStatus;
+}
- // create effect engine from effect factory
- mStatus = -ENODEV;
- sp<AudioFlinger> audioFlinger = mAudioFlinger.promote();
- if (audioFlinger != 0) {
- sp<EffectsFactoryHalInterface> effectsFactory = audioFlinger->getEffectsFactory();
- if (effectsFactory != 0) {
- mStatus = effectsFactory->createEffect(
- &desc->uuid, sessionId, thread->id(), &mEffectInterface);
+// must be called with EffectModule::mLock held
+status_t AudioFlinger::EffectBase::setEnabled_l(bool enabled)
+{
+
+ ALOGV("setEnabled %p enabled %d", this, enabled);
+
+ if (enabled != isEnabled()) {
+ switch (mState) {
+ // going from disabled to enabled
+ case IDLE:
+ mState = STARTING;
+ break;
+ case STOPPED:
+ mState = RESTART;
+ break;
+ case STOPPING:
+ mState = ACTIVE;
+ break;
+
+ // going from enabled to disabled
+ case RESTART:
+ mState = STOPPED;
+ break;
+ case STARTING:
+ mState = IDLE;
+ break;
+ case ACTIVE:
+ mState = STOPPING;
+ break;
+ case DESTROYED:
+ return NO_ERROR; // simply ignore as we are being destroyed
+ }
+ for (size_t i = 1; i < mHandles.size(); i++) {
+ EffectHandle *h = mHandles[i];
+ if (h != NULL && !h->disconnected()) {
+ h->setEnabled(enabled);
+ }
}
}
-
- if (mStatus != NO_ERROR) {
- return;
- }
- lStatus = init();
- if (lStatus < 0) {
- mStatus = lStatus;
- goto Error;
- }
-
- setOffloaded(thread->type() == ThreadBase::OFFLOAD, thread->id());
- ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface.get());
-
- return;
-Error:
- mEffectInterface.clear();
- ALOGV("Constructor Error %d", mStatus);
+ return NO_ERROR;
}
-AudioFlinger::EffectModule::~EffectModule()
+status_t AudioFlinger::EffectBase::setEnabled(bool enabled, bool fromHandle)
{
- ALOGV("Destructor %p", this);
- if (mEffectInterface != 0) {
- char uuidStr[64];
- AudioEffect::guidToString(&mDescriptor.uuid, uuidStr, sizeof(uuidStr));
- ALOGW("EffectModule %p destructor called with unreleased interface, effect %s",
- this, uuidStr);
- release_l();
+ status_t status;
+ {
+ Mutex::Autolock _l(mLock);
+ status = setEnabled_l(enabled);
}
-
+ if (fromHandle) {
+ if (enabled) {
+ if (status != NO_ERROR) {
+ mCallback->checkSuspendOnEffectEnabled(this, false, false /*threadLocked*/);
+ } else {
+ mCallback->onEffectEnable(this);
+ }
+ } else {
+ mCallback->onEffectDisable(this);
+ }
+ }
+ return status;
}
-status_t AudioFlinger::EffectModule::addHandle(EffectHandle *handle)
+bool AudioFlinger::EffectBase::isEnabled() const
+{
+ switch (mState) {
+ case RESTART:
+ case STARTING:
+ case ACTIVE:
+ return true;
+ case IDLE:
+ case STOPPING:
+ case STOPPED:
+ case DESTROYED:
+ default:
+ return false;
+ }
+}
+
+void AudioFlinger::EffectBase::setSuspended(bool suspended)
+{
+ Mutex::Autolock _l(mLock);
+ mSuspended = suspended;
+}
+
+bool AudioFlinger::EffectBase::suspended() const
+{
+ Mutex::Autolock _l(mLock);
+ return mSuspended;
+}
+
+status_t AudioFlinger::EffectBase::addHandle(EffectHandle *handle)
{
status_t status;
@@ -169,7 +205,7 @@
return status;
}
-status_t AudioFlinger::EffectModule::updatePolicyState()
+status_t AudioFlinger::EffectBase::updatePolicyState()
{
status_t status = NO_ERROR;
bool doRegister = false;
@@ -186,14 +222,8 @@
doRegister = true;
mPolicyRegistered = mHandles.size() > 0;
if (mPolicyRegistered) {
- sp <EffectChain> chain = mChain.promote();
- sp <ThreadBase> thread = mThread.promote();
-
- if (thread == nullptr || chain == nullptr) {
- return INVALID_OPERATION;
- }
- io = thread->id();
- strategy = chain->strategy();
+ io = mCallback->io();
+ strategy = mCallback->strategy();
}
}
// enable effect when registered according to enable state requested by controlling handle
@@ -231,13 +261,13 @@
}
-ssize_t AudioFlinger::EffectModule::removeHandle(EffectHandle *handle)
+ssize_t AudioFlinger::EffectBase::removeHandle(EffectHandle *handle)
{
Mutex::Autolock _l(mLock);
return removeHandle_l(handle);
}
-ssize_t AudioFlinger::EffectModule::removeHandle_l(EffectHandle *handle)
+ssize_t AudioFlinger::EffectBase::removeHandle_l(EffectHandle *handle)
{
size_t size = mHandles.size();
size_t i;
@@ -261,19 +291,15 @@
}
}
- // Prevent calls to process() and other functions on effect interface from now on.
- // The effect engine will be released by the destructor when the last strong reference on
- // this object is released which can happen after next process is called.
if (mHandles.size() == 0 && !mPinned) {
mState = DESTROYED;
- mEffectInterface->close();
}
return mHandles.size();
}
// must be called with EffectModule::mLock held
-AudioFlinger::EffectHandle *AudioFlinger::EffectModule::controlHandle_l()
+AudioFlinger::EffectHandle *AudioFlinger::EffectBase::controlHandle_l()
{
// the first valid handle in the list has control over the module
for (size_t i = 0; i < mHandles.size(); i++) {
@@ -287,22 +313,271 @@
}
// unsafe method called when the effect parent thread has been destroyed
-ssize_t AudioFlinger::EffectModule::disconnectHandle(EffectHandle *handle, bool unpinIfLast)
+ssize_t AudioFlinger::EffectBase::disconnectHandle(EffectHandle *handle, bool unpinIfLast)
{
ALOGV("disconnect() %p handle %p", this, handle);
+ if (mCallback->disconnectEffectHandle(handle, unpinIfLast)) {
+ return mHandles.size();
+ }
+
Mutex::Autolock _l(mLock);
ssize_t numHandles = removeHandle_l(handle);
if ((numHandles == 0) && (!mPinned || unpinIfLast)) {
- sp<AudioFlinger> af = mAudioFlinger.promote();
- if (af != 0) {
- mLock.unlock();
- af->updateOrphanEffectChains(this);
- mLock.lock();
- }
+ mLock.unlock();
+ mCallback->updateOrphanEffectChains(this);
+ mLock.lock();
}
return numHandles;
}
+bool AudioFlinger::EffectBase::purgeHandles()
+{
+ bool enabled = false;
+ Mutex::Autolock _l(mLock);
+ EffectHandle *handle = controlHandle_l();
+ if (handle != NULL) {
+ enabled = handle->enabled();
+ }
+ mHandles.clear();
+ return enabled;
+}
+
+void AudioFlinger::EffectBase::checkSuspendOnEffectEnabled(bool enabled, bool threadLocked) {
+ mCallback->checkSuspendOnEffectEnabled(this, enabled, threadLocked);
+}
+
+static String8 effectFlagsToString(uint32_t flags) {
+ String8 s;
+
+ s.append("conn. mode: ");
+ switch (flags & EFFECT_FLAG_TYPE_MASK) {
+ case EFFECT_FLAG_TYPE_INSERT: s.append("insert"); break;
+ case EFFECT_FLAG_TYPE_AUXILIARY: s.append("auxiliary"); break;
+ case EFFECT_FLAG_TYPE_REPLACE: s.append("replace"); break;
+ case EFFECT_FLAG_TYPE_PRE_PROC: s.append("preproc"); break;
+ case EFFECT_FLAG_TYPE_POST_PROC: s.append("postproc"); break;
+ default: s.append("unknown/reserved"); break;
+ }
+ s.append(", ");
+
+ s.append("insert pref: ");
+ switch (flags & EFFECT_FLAG_INSERT_MASK) {
+ case EFFECT_FLAG_INSERT_ANY: s.append("any"); break;
+ case EFFECT_FLAG_INSERT_FIRST: s.append("first"); break;
+ case EFFECT_FLAG_INSERT_LAST: s.append("last"); break;
+ case EFFECT_FLAG_INSERT_EXCLUSIVE: s.append("exclusive"); break;
+ default: s.append("unknown/reserved"); break;
+ }
+ s.append(", ");
+
+ s.append("volume mgmt: ");
+ switch (flags & EFFECT_FLAG_VOLUME_MASK) {
+ case EFFECT_FLAG_VOLUME_NONE: s.append("none"); break;
+ case EFFECT_FLAG_VOLUME_CTRL: s.append("implements control"); break;
+ case EFFECT_FLAG_VOLUME_IND: s.append("requires indication"); break;
+ case EFFECT_FLAG_VOLUME_MONITOR: s.append("monitors volume"); break;
+ default: s.append("unknown/reserved"); break;
+ }
+ s.append(", ");
+
+ uint32_t devind = flags & EFFECT_FLAG_DEVICE_MASK;
+ if (devind) {
+ s.append("device indication: ");
+ switch (devind) {
+ case EFFECT_FLAG_DEVICE_IND: s.append("requires updates"); break;
+ default: s.append("unknown/reserved"); break;
+ }
+ s.append(", ");
+ }
+
+ s.append("input mode: ");
+ switch (flags & EFFECT_FLAG_INPUT_MASK) {
+ case EFFECT_FLAG_INPUT_DIRECT: s.append("direct"); break;
+ case EFFECT_FLAG_INPUT_PROVIDER: s.append("provider"); break;
+ case EFFECT_FLAG_INPUT_BOTH: s.append("direct+provider"); break;
+ default: s.append("not set"); break;
+ }
+ s.append(", ");
+
+ s.append("output mode: ");
+ switch (flags & EFFECT_FLAG_OUTPUT_MASK) {
+ case EFFECT_FLAG_OUTPUT_DIRECT: s.append("direct"); break;
+ case EFFECT_FLAG_OUTPUT_PROVIDER: s.append("provider"); break;
+ case EFFECT_FLAG_OUTPUT_BOTH: s.append("direct+provider"); break;
+ default: s.append("not set"); break;
+ }
+ s.append(", ");
+
+ uint32_t accel = flags & EFFECT_FLAG_HW_ACC_MASK;
+ if (accel) {
+ s.append("hardware acceleration: ");
+ switch (accel) {
+ case EFFECT_FLAG_HW_ACC_SIMPLE: s.append("non-tunneled"); break;
+ case EFFECT_FLAG_HW_ACC_TUNNEL: s.append("tunneled"); break;
+ default: s.append("unknown/reserved"); break;
+ }
+ s.append(", ");
+ }
+
+ uint32_t modeind = flags & EFFECT_FLAG_AUDIO_MODE_MASK;
+ if (modeind) {
+ s.append("mode indication: ");
+ switch (modeind) {
+ case EFFECT_FLAG_AUDIO_MODE_IND: s.append("required"); break;
+ default: s.append("unknown/reserved"); break;
+ }
+ s.append(", ");
+ }
+
+ uint32_t srcind = flags & EFFECT_FLAG_AUDIO_SOURCE_MASK;
+ if (srcind) {
+ s.append("source indication: ");
+ switch (srcind) {
+ case EFFECT_FLAG_AUDIO_SOURCE_IND: s.append("required"); break;
+ default: s.append("unknown/reserved"); break;
+ }
+ s.append(", ");
+ }
+
+ if (flags & EFFECT_FLAG_OFFLOAD_MASK) {
+ s.append("offloadable, ");
+ }
+
+ int len = s.length();
+ if (s.length() > 2) {
+ (void) s.lockBuffer(len);
+ s.unlockBuffer(len - 2);
+ }
+ return s;
+}
+
+void AudioFlinger::EffectBase::dump(int fd, const Vector<String16>& args __unused)
+{
+ String8 result;
+
+ result.appendFormat("\tEffect ID %d:\n", mId);
+
+ bool locked = AudioFlinger::dumpTryLock(mLock);
+ // failed to lock - AudioFlinger is probably deadlocked
+ if (!locked) {
+ result.append("\t\tCould not lock Fx mutex:\n");
+ }
+
+ result.append("\t\tSession State Registered Enabled Suspended:\n");
+ result.appendFormat("\t\t%05d %03d %s %s %s\n",
+ mSessionId, mState, mPolicyRegistered ? "y" : "n",
+ mPolicyEnabled ? "y" : "n", mSuspended ? "y" : "n");
+
+ result.append("\t\tDescriptor:\n");
+ char uuidStr[64];
+ AudioEffect::guidToString(&mDescriptor.uuid, uuidStr, sizeof(uuidStr));
+ result.appendFormat("\t\t- UUID: %s\n", uuidStr);
+ AudioEffect::guidToString(&mDescriptor.type, uuidStr, sizeof(uuidStr));
+ result.appendFormat("\t\t- TYPE: %s\n", uuidStr);
+ result.appendFormat("\t\t- apiVersion: %08X\n\t\t- flags: %08X (%s)\n",
+ mDescriptor.apiVersion,
+ mDescriptor.flags,
+ effectFlagsToString(mDescriptor.flags).string());
+ result.appendFormat("\t\t- name: %s\n",
+ mDescriptor.name);
+
+ result.appendFormat("\t\t- implementor: %s\n",
+ mDescriptor.implementor);
+
+ result.appendFormat("\t\t%zu Clients:\n", mHandles.size());
+ result.append("\t\t\t Pid Priority Ctrl Locked client server\n");
+ char buffer[256];
+ for (size_t i = 0; i < mHandles.size(); ++i) {
+ EffectHandle *handle = mHandles[i];
+ if (handle != NULL && !handle->disconnected()) {
+ handle->dumpToBuffer(buffer, sizeof(buffer));
+ result.append(buffer);
+ }
+ }
+ if (locked) {
+ mLock.unlock();
+ }
+
+ write(fd, result.string(), result.length());
+}
+
+// ----------------------------------------------------------------------------
+// EffectModule implementation
+// ----------------------------------------------------------------------------
+
+#undef LOG_TAG
+#define LOG_TAG "AudioFlinger::EffectModule"
+
+AudioFlinger::EffectModule::EffectModule(const sp<AudioFlinger::EffectCallbackInterface>& callback,
+ effect_descriptor_t *desc,
+ int id,
+ audio_session_t sessionId,
+ bool pinned,
+ audio_port_handle_t deviceId)
+ : EffectBase(callback, desc, id, sessionId, pinned),
+ // clear mConfig to ensure consistent initial value of buffer framecount
+ // in case buffers are associated by setInBuffer() or setOutBuffer()
+ // prior to configure().
+ mConfig{{}, {}},
+ mStatus(NO_INIT),
+ mMaxDisableWaitCnt(1), // set by configure(), should be >= 1
+ mDisableWaitCnt(0), // set by process() and updateState()
+ mOffloaded(false)
+#ifdef FLOAT_EFFECT_CHAIN
+ , mSupportsFloat(false)
+#endif
+{
+ ALOGV("Constructor %p pinned %d", this, pinned);
+ int lStatus;
+
+ // create effect engine from effect factory
+ mStatus = callback->createEffectHal(
+ &desc->uuid, sessionId, deviceId, &mEffectInterface);
+ if (mStatus != NO_ERROR) {
+ return;
+ }
+ lStatus = init();
+ if (lStatus < 0) {
+ mStatus = lStatus;
+ goto Error;
+ }
+
+ setOffloaded(callback->isOffload(), callback->io());
+ ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface.get());
+
+ return;
+Error:
+ mEffectInterface.clear();
+ ALOGV("Constructor Error %d", mStatus);
+}
+
+AudioFlinger::EffectModule::~EffectModule()
+{
+ ALOGV("Destructor %p", this);
+ if (mEffectInterface != 0) {
+ char uuidStr[64];
+ AudioEffect::guidToString(&mDescriptor.uuid, uuidStr, sizeof(uuidStr));
+ ALOGW("EffectModule %p destructor called with unreleased interface, effect %s",
+ this, uuidStr);
+ release_l();
+ }
+
+}
+
+ssize_t AudioFlinger::EffectModule::removeHandle_l(EffectHandle *handle)
+{
+ ssize_t status = EffectBase::removeHandle_l(handle);
+
+ // Prevent calls to process() and other functions on effect interface from now on.
+ // The effect engine will be released by the destructor when the last strong reference on
+ // this object is released which can happen after next process is called.
+ if (status == 0 && !mPinned) {
+ mEffectInterface->close();
+ }
+
+ return status;
+}
+
bool AudioFlinger::EffectModule::updateState() {
Mutex::Autolock _l(mLock);
@@ -540,8 +815,7 @@
mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
// If an insert effect is idle and input buffer is different from output buffer,
// accumulate input onto output
- sp<EffectChain> chain = mChain.promote();
- if (chain.get() != nullptr && chain->activeTrackCnt() != 0) {
+ if (mCallback->activeTrackCnt() != 0) {
// similar handling with data_bypass above.
if (mConfig.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE) {
accumulateInputToOutput();
@@ -564,7 +838,6 @@
{
ALOGVV("configure() started");
status_t status;
- sp<ThreadBase> thread;
uint32_t size;
audio_channel_mask_t channelMask;
@@ -573,17 +846,11 @@
goto exit;
}
- thread = mThread.promote();
- if (thread == 0) {
- status = DEAD_OBJECT;
- goto exit;
- }
-
// TODO: handle configuration of effects replacing track process
// TODO: handle configuration of input (record) SW effects above the HAL,
// similar to output EFFECT_FLAG_TYPE_INSERT/REPLACE,
// in which case input channel masks should be used here.
- channelMask = thread->channelMask();
+ channelMask = mCallback->channelMask();
mConfig.inputCfg.channels = channelMask;
mConfig.outputCfg.channels = channelMask;
@@ -620,11 +887,11 @@
mConfig.outputCfg.format = EFFECT_BUFFER_FORMAT;
// Don't use sample rate for thread if effect isn't offloadable.
- if ((thread->type() == ThreadBase::OFFLOAD) && !isOffloaded()) {
+ if (mCallback->isOffloadOrDirect() && !isOffloaded()) {
mConfig.inputCfg.samplingRate = DEFAULT_OUTPUT_SAMPLE_RATE;
ALOGV("Overriding effect input as 48kHz");
} else {
- mConfig.inputCfg.samplingRate = thread->sampleRate();
+ mConfig.inputCfg.samplingRate = mCallback->sampleRate();
}
mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate;
mConfig.inputCfg.bufferProvider.cookie = NULL;
@@ -635,7 +902,7 @@
mConfig.outputCfg.bufferProvider.releaseBuffer = NULL;
mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
// Insert effect:
- // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE,
+ // - in global sessions (e.g AUDIO_SESSION_OUTPUT_MIX),
// always overwrites output buffer: input buffer == output buffer
// - in other sessions:
// last effect in the chain accumulates in output buffer: input buffer != output buffer
@@ -650,11 +917,13 @@
}
mConfig.inputCfg.mask = EFFECT_CONFIG_ALL;
mConfig.outputCfg.mask = EFFECT_CONFIG_ALL;
- mConfig.inputCfg.buffer.frameCount = thread->frameCount();
+ mConfig.inputCfg.buffer.frameCount = mCallback->frameCount();
mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount;
- ALOGV("configure() %p thread %p buffer %p framecount %zu",
- this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount);
+ ALOGV("configure() %p chain %p buffer %p framecount %zu",
+ this, mCallback->chain().promote() != nullptr ? mCallback->chain().promote().get() :
+ nullptr,
+ mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount);
status_t cmdStatus;
size = sizeof(int);
@@ -669,7 +938,7 @@
#ifdef MULTICHANNEL_EFFECT_CHAIN
if (status != NO_ERROR &&
- thread->isOutput() &&
+ mCallback->isOutput() &&
(mConfig.inputCfg.channels != AUDIO_CHANNEL_OUT_STEREO
|| mConfig.outputCfg.channels != AUDIO_CHANNEL_OUT_STEREO)) {
// Older effects may require exact STEREO position mask.
@@ -736,11 +1005,7 @@
size = sizeof(int);
*(int32_t *)p->data = VISUALIZER_PARAM_LATENCY;
- uint32_t latency = 0;
- PlaybackThread *pbt = thread->mAudioFlinger->checkPlaybackThread_l(thread->mId);
- if (pbt != NULL) {
- latency = pbt->latency_l();
- }
+ uint32_t latency = mCallback->latency();
*((int32_t *)p->data + 1)= latency;
mEffectInterface->command(EFFECT_CMD_SET_PARAM,
@@ -787,31 +1052,20 @@
{
if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
(mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) {
- sp<ThreadBase> thread = mThread.promote();
- if (thread != 0) {
- sp<StreamHalInterface> stream = thread->stream();
- if (stream != 0) {
- status_t result = stream->addEffect(mEffectInterface);
- ALOGE_IF(result != OK, "Error when adding effect: %d", result);
- }
- }
+ (void)mCallback->addEffectToHal(mEffectInterface);
}
}
// start() must be called with PlaybackThread::mLock or EffectChain::mLock held
status_t AudioFlinger::EffectModule::start()
{
- sp<EffectChain> chain;
status_t status;
{
Mutex::Autolock _l(mLock);
status = start_l();
- if (status == NO_ERROR) {
- chain = mChain.promote();
- }
}
- if (chain != 0) {
- chain->resetVolume_l();
+ if (status == NO_ERROR) {
+ mCallback->resetVolume();
}
return status;
}
@@ -858,11 +1112,10 @@
uint32_t size = sizeof(status_t);
if (isVolumeControl() && isOffloadedOrDirect()) {
- sp<EffectChain>chain = mChain.promote();
// We have the EffectChain and EffectModule lock, permit a reentrant call to setVolume:
// resetVolume_l --> setVolume_l --> EffectModule::setVolume
mSetVolumeReentrantTid = gettid();
- chain->resetVolume_l();
+ mCallback->resetVolume();
mSetVolumeReentrantTid = INVALID_PID;
}
@@ -875,7 +1128,7 @@
status = cmdStatus;
}
if (status == NO_ERROR) {
- status = remove_effect_from_hal_l();
+ status = removeEffectFromHal_l();
}
return status;
}
@@ -884,25 +1137,18 @@
void AudioFlinger::EffectModule::release_l()
{
if (mEffectInterface != 0) {
- remove_effect_from_hal_l();
+ removeEffectFromHal_l();
// release effect engine
mEffectInterface->close();
mEffectInterface.clear();
}
}
-status_t AudioFlinger::EffectModule::remove_effect_from_hal_l()
+status_t AudioFlinger::EffectModule::removeEffectFromHal_l()
{
if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
(mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) {
- sp<ThreadBase> thread = mThread.promote();
- if (thread != 0) {
- sp<StreamHalInterface> stream = thread->stream();
- if (stream != 0) {
- status_t result = stream->removeEffect(mEffectInterface);
- ALOGE_IF(result != OK, "Error when removing effect: %d", result);
- }
- }
+ mCallback->removeEffectFromHal(mEffectInterface);
}
return NO_ERROR;
}
@@ -992,70 +1238,6 @@
return status;
}
-status_t AudioFlinger::EffectModule::setEnabled(bool enabled)
-{
- Mutex::Autolock _l(mLock);
- return setEnabled_l(enabled);
-}
-
-// must be called with EffectModule::mLock held
-status_t AudioFlinger::EffectModule::setEnabled_l(bool enabled)
-{
-
- ALOGV("setEnabled %p enabled %d", this, enabled);
-
- if (enabled != isEnabled()) {
- switch (mState) {
- // going from disabled to enabled
- case IDLE:
- mState = STARTING;
- break;
- case STOPPED:
- mState = RESTART;
- break;
- case STOPPING:
- mState = ACTIVE;
- break;
-
- // going from enabled to disabled
- case RESTART:
- mState = STOPPED;
- break;
- case STARTING:
- mState = IDLE;
- break;
- case ACTIVE:
- mState = STOPPING;
- break;
- case DESTROYED:
- return NO_ERROR; // simply ignore as we are being destroyed
- }
- for (size_t i = 1; i < mHandles.size(); i++) {
- EffectHandle *h = mHandles[i];
- if (h != NULL && !h->disconnected()) {
- h->setEnabled(enabled);
- }
- }
- }
- return NO_ERROR;
-}
-
-bool AudioFlinger::EffectModule::isEnabled() const
-{
- switch (mState) {
- case RESTART:
- case STARTING:
- case ACTIVE:
- return true;
- case IDLE:
- case STOPPING:
- case STOPPED:
- case DESTROYED:
- default:
- return false;
- }
-}
-
bool AudioFlinger::EffectModule::isProcessEnabled() const
{
if (mStatus != NO_ERROR) {
@@ -1078,7 +1260,7 @@
bool AudioFlinger::EffectModule::isOffloadedOrDirect() const
{
- return (mThreadType == ThreadBase::OFFLOAD || mThreadType == ThreadBase::DIRECT);
+ return mCallback->isOffloadOrDirect();
}
bool AudioFlinger::EffectModule::isVolumeControlEnabled() const
@@ -1122,9 +1304,7 @@
|| size > mInConversionBuffer->getSize())) {
mInConversionBuffer.clear();
ALOGV("%s: allocating mInConversionBuffer %zu", __func__, size);
- sp<AudioFlinger> audioFlinger = mAudioFlinger.promote();
- LOG_ALWAYS_FATAL_IF(audioFlinger == nullptr, "EM could not retrieved audioFlinger");
- (void)audioFlinger->mEffectsFactoryHal->allocateBuffer(size, &mInConversionBuffer);
+ (void)mCallback->allocateHalBuffer(size, &mInConversionBuffer);
}
if (mInConversionBuffer.get() != nullptr) {
mInConversionBuffer->setFrameCount(inFrameCount);
@@ -1168,9 +1348,7 @@
|| size > mOutConversionBuffer->getSize())) {
mOutConversionBuffer.clear();
ALOGV("%s: allocating mOutConversionBuffer %zu", __func__, size);
- sp<AudioFlinger> audioFlinger = mAudioFlinger.promote();
- LOG_ALWAYS_FATAL_IF(audioFlinger == nullptr, "EM could not retrieved audioFlinger");
- (void)audioFlinger->mEffectsFactoryHal->allocateBuffer(size, &mOutConversionBuffer);
+ (void)mCallback->allocateHalBuffer(size, &mOutConversionBuffer);
}
if (mOutConversionBuffer.get() != nullptr) {
mOutConversionBuffer->setFrameCount(outFrameCount);
@@ -1218,20 +1396,18 @@
void AudioFlinger::EffectChain::setVolumeForOutput_l(uint32_t left, uint32_t right)
{
- sp<ThreadBase> thread = mThread.promote();
- if (thread != 0 &&
- (thread->type() == ThreadBase::OFFLOAD || thread->type() == ThreadBase::DIRECT) &&
- !isNonOffloadableEnabled_l()) {
- PlaybackThread *t = (PlaybackThread *)thread.get();
+ if (mEffectCallback->isOffloadOrDirect() && !isNonOffloadableEnabled_l()) {
float vol_l = (float)left / (1 << 24);
float vol_r = (float)right / (1 << 24);
- t->setVolumeForOutput_l(vol_l, vol_r);
+ mEffectCallback->setVolumeForOutput(vol_l, vol_r);
}
}
-status_t AudioFlinger::EffectModule::setDevice(audio_devices_t device)
+status_t AudioFlinger::EffectModule::sendSetAudioDevicesCommand(
+ const AudioDeviceTypeAddrVector &devices, uint32_t cmdCode)
{
- if (device == AUDIO_DEVICE_NONE) {
+ audio_devices_t deviceType = deviceTypesToBitMask(getAudioDeviceTypes(devices));
+ if (deviceType == AUDIO_DEVICE_NONE) {
return NO_ERROR;
}
@@ -1243,17 +1419,26 @@
if ((mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) {
status_t cmdStatus;
uint32_t size = sizeof(status_t);
- uint32_t cmd = audio_is_output_devices(device) ? EFFECT_CMD_SET_DEVICE :
- EFFECT_CMD_SET_INPUT_DEVICE;
- status = mEffectInterface->command(cmd,
+ // FIXME: use audio device types and addresses when the hal interface is ready.
+ status = mEffectInterface->command(cmdCode,
sizeof(uint32_t),
- &device,
+ &deviceType,
&size,
&cmdStatus);
}
return status;
}
+status_t AudioFlinger::EffectModule::setDevices(const AudioDeviceTypeAddrVector &devices)
+{
+ return sendSetAudioDevicesCommand(devices, EFFECT_CMD_SET_DEVICE);
+}
+
+status_t AudioFlinger::EffectModule::setInputDevice(const AudioDeviceTypeAddr &device)
+{
+ return sendSetAudioDevicesCommand({device}, EFFECT_CMD_SET_INPUT_DEVICE);
+}
+
status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode)
{
Mutex::Autolock _l(mLock);
@@ -1294,30 +1479,6 @@
return status;
}
-void AudioFlinger::EffectModule::setSuspended(bool suspended)
-{
- Mutex::Autolock _l(mLock);
- mSuspended = suspended;
-}
-
-bool AudioFlinger::EffectModule::suspended() const
-{
- Mutex::Autolock _l(mLock);
- return mSuspended;
-}
-
-bool AudioFlinger::EffectModule::purgeHandles()
-{
- bool enabled = false;
- Mutex::Autolock _l(mLock);
- EffectHandle *handle = controlHandle_l();
- if (handle != NULL) {
- enabled = handle->enabled();
- }
- mHandles.clear();
- return enabled;
-}
-
status_t AudioFlinger::EffectModule::setOffloaded(bool offloaded, audio_io_handle_t io)
{
Mutex::Autolock _l(mLock);
@@ -1357,111 +1518,6 @@
return mOffloaded;
}
-String8 effectFlagsToString(uint32_t flags) {
- String8 s;
-
- s.append("conn. mode: ");
- switch (flags & EFFECT_FLAG_TYPE_MASK) {
- case EFFECT_FLAG_TYPE_INSERT: s.append("insert"); break;
- case EFFECT_FLAG_TYPE_AUXILIARY: s.append("auxiliary"); break;
- case EFFECT_FLAG_TYPE_REPLACE: s.append("replace"); break;
- case EFFECT_FLAG_TYPE_PRE_PROC: s.append("preproc"); break;
- case EFFECT_FLAG_TYPE_POST_PROC: s.append("postproc"); break;
- default: s.append("unknown/reserved"); break;
- }
- s.append(", ");
-
- s.append("insert pref: ");
- switch (flags & EFFECT_FLAG_INSERT_MASK) {
- case EFFECT_FLAG_INSERT_ANY: s.append("any"); break;
- case EFFECT_FLAG_INSERT_FIRST: s.append("first"); break;
- case EFFECT_FLAG_INSERT_LAST: s.append("last"); break;
- case EFFECT_FLAG_INSERT_EXCLUSIVE: s.append("exclusive"); break;
- default: s.append("unknown/reserved"); break;
- }
- s.append(", ");
-
- s.append("volume mgmt: ");
- switch (flags & EFFECT_FLAG_VOLUME_MASK) {
- case EFFECT_FLAG_VOLUME_NONE: s.append("none"); break;
- case EFFECT_FLAG_VOLUME_CTRL: s.append("implements control"); break;
- case EFFECT_FLAG_VOLUME_IND: s.append("requires indication"); break;
- case EFFECT_FLAG_VOLUME_MONITOR: s.append("monitors volume"); break;
- default: s.append("unknown/reserved"); break;
- }
- s.append(", ");
-
- uint32_t devind = flags & EFFECT_FLAG_DEVICE_MASK;
- if (devind) {
- s.append("device indication: ");
- switch (devind) {
- case EFFECT_FLAG_DEVICE_IND: s.append("requires updates"); break;
- default: s.append("unknown/reserved"); break;
- }
- s.append(", ");
- }
-
- s.append("input mode: ");
- switch (flags & EFFECT_FLAG_INPUT_MASK) {
- case EFFECT_FLAG_INPUT_DIRECT: s.append("direct"); break;
- case EFFECT_FLAG_INPUT_PROVIDER: s.append("provider"); break;
- case EFFECT_FLAG_INPUT_BOTH: s.append("direct+provider"); break;
- default: s.append("not set"); break;
- }
- s.append(", ");
-
- s.append("output mode: ");
- switch (flags & EFFECT_FLAG_OUTPUT_MASK) {
- case EFFECT_FLAG_OUTPUT_DIRECT: s.append("direct"); break;
- case EFFECT_FLAG_OUTPUT_PROVIDER: s.append("provider"); break;
- case EFFECT_FLAG_OUTPUT_BOTH: s.append("direct+provider"); break;
- default: s.append("not set"); break;
- }
- s.append(", ");
-
- uint32_t accel = flags & EFFECT_FLAG_HW_ACC_MASK;
- if (accel) {
- s.append("hardware acceleration: ");
- switch (accel) {
- case EFFECT_FLAG_HW_ACC_SIMPLE: s.append("non-tunneled"); break;
- case EFFECT_FLAG_HW_ACC_TUNNEL: s.append("tunneled"); break;
- default: s.append("unknown/reserved"); break;
- }
- s.append(", ");
- }
-
- uint32_t modeind = flags & EFFECT_FLAG_AUDIO_MODE_MASK;
- if (modeind) {
- s.append("mode indication: ");
- switch (modeind) {
- case EFFECT_FLAG_AUDIO_MODE_IND: s.append("required"); break;
- default: s.append("unknown/reserved"); break;
- }
- s.append(", ");
- }
-
- uint32_t srcind = flags & EFFECT_FLAG_AUDIO_SOURCE_MASK;
- if (srcind) {
- s.append("source indication: ");
- switch (srcind) {
- case EFFECT_FLAG_AUDIO_SOURCE_IND: s.append("required"); break;
- default: s.append("unknown/reserved"); break;
- }
- s.append(", ");
- }
-
- if (flags & EFFECT_FLAG_OFFLOAD_MASK) {
- s.append("offloadable, ");
- }
-
- int len = s.length();
- if (s.length() > 2) {
- (void) s.lockBuffer(len);
- s.unlockBuffer(len - 2);
- }
- return s;
-}
-
static std::string dumpInOutBuffer(bool isInput, const sp<EffectBufferHalInterface> &buffer) {
std::stringstream ss;
@@ -1477,38 +1533,16 @@
return ss.str();
}
-void AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args __unused)
+void AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args)
{
+ EffectBase::dump(fd, args);
+
String8 result;
-
- result.appendFormat("\tEffect ID %d:\n", mId);
-
bool locked = AudioFlinger::dumpTryLock(mLock);
- // failed to lock - AudioFlinger is probably deadlocked
- if (!locked) {
- result.append("\t\tCould not lock Fx mutex:\n");
- }
- result.append("\t\tSession Status State Registered Enabled Suspended Engine:\n");
- result.appendFormat("\t\t%05d %03d %03d %s %s %s %p\n",
- mSessionId, mStatus, mState, mPolicyRegistered ? "y" : "n", mPolicyEnabled ? "y" : "n",
- mSuspended ? "y" : "n", mEffectInterface.get());
-
- result.append("\t\tDescriptor:\n");
- char uuidStr[64];
- AudioEffect::guidToString(&mDescriptor.uuid, uuidStr, sizeof(uuidStr));
- result.appendFormat("\t\t- UUID: %s\n", uuidStr);
- AudioEffect::guidToString(&mDescriptor.type, uuidStr, sizeof(uuidStr));
- result.appendFormat("\t\t- TYPE: %s\n", uuidStr);
- result.appendFormat("\t\t- apiVersion: %08X\n\t\t- flags: %08X (%s)\n",
- mDescriptor.apiVersion,
- mDescriptor.flags,
- effectFlagsToString(mDescriptor.flags).string());
- result.appendFormat("\t\t- name: %s\n",
- mDescriptor.name);
-
- result.appendFormat("\t\t- implementor: %s\n",
- mDescriptor.implementor);
+ result.append("\t\tStatus Engine:\n");
+ result.appendFormat("\t\t%03d %p\n",
+ mStatus, mEffectInterface.get());
result.appendFormat("\t\t- data: %s\n", mSupportsFloat ? "float" : "int16");
@@ -1542,17 +1576,6 @@
dumpInOutBuffer(false /* isInput */, mOutConversionBuffer).c_str());
#endif
- result.appendFormat("\t\t%zu Clients:\n", mHandles.size());
- result.append("\t\t\t Pid Priority Ctrl Locked client server\n");
- char buffer[256];
- for (size_t i = 0; i < mHandles.size(); ++i) {
- EffectHandle *handle = mHandles[i];
- if (handle != NULL && !handle->disconnected()) {
- handle->dumpToBuffer(buffer, sizeof(buffer));
- result.append(buffer);
- }
- }
-
write(fd, result.string(), result.length());
if (mEffectInterface != 0) {
@@ -1572,7 +1595,7 @@
#undef LOG_TAG
#define LOG_TAG "AudioFlinger::EffectHandle"
-AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect,
+AudioFlinger::EffectHandle::EffectHandle(const sp<EffectBase>& effect,
const sp<AudioFlinger::Client>& client,
const sp<IEffectClient>& effectClient,
int32_t priority)
@@ -1580,7 +1603,7 @@
mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL),
mPriority(priority), mHasControl(false), mEnabled(false), mDisconnected(false)
{
- ALOGV("constructor %p", this);
+ ALOGV("constructor %p client %p", this, client.get());
if (client == 0) {
return;
@@ -1613,7 +1636,7 @@
{
AutoMutex _l(mLock);
ALOGV("enable %p", this);
- sp<EffectModule> effect = mEffect.promote();
+ sp<EffectBase> effect = mEffect.promote();
if (effect == 0 || mDisconnected) {
return DEAD_OBJECT;
}
@@ -1633,38 +1656,16 @@
return status;
}
- sp<ThreadBase> thread = effect->thread().promote();
- if (thread != 0) {
- thread->checkSuspendOnEffectEnabled(effect, true, effect->sessionId());
- }
+ effect->checkSuspendOnEffectEnabled(true, false /*threadLocked*/);
// checkSuspendOnEffectEnabled() can suspend this same effect when enabled
if (effect->suspended()) {
return NO_ERROR;
}
- status = effect->setEnabled(true);
+ status = effect->setEnabled(true, true /*fromHandle*/);
if (status != NO_ERROR) {
- if (thread != 0) {
- thread->checkSuspendOnEffectEnabled(effect, false, effect->sessionId());
- }
mEnabled = false;
- } else {
- if (thread != 0) {
- if (thread->type() == ThreadBase::OFFLOAD || thread->type() == ThreadBase::MMAP) {
- Mutex::Autolock _l(thread->mLock);
- thread->broadcast_l();
- }
- if (!effect->isOffloadable()) {
- if (thread->type() == ThreadBase::OFFLOAD) {
- PlaybackThread *t = (PlaybackThread *)thread.get();
- t->invalidateTracks(AUDIO_STREAM_MUSIC);
- }
- if (effect->sessionId() == AUDIO_SESSION_OUTPUT_MIX) {
- thread->mAudioFlinger->onNonOffloadableGlobalEffectEnable();
- }
- }
- }
}
return status;
}
@@ -1673,7 +1674,7 @@
{
ALOGV("disable %p", this);
AutoMutex _l(mLock);
- sp<EffectModule> effect = mEffect.promote();
+ sp<EffectBase> effect = mEffect.promote();
if (effect == 0 || mDisconnected) {
return DEAD_OBJECT;
}
@@ -1692,17 +1693,7 @@
return NO_ERROR;
}
- status_t status = effect->setEnabled(false);
-
- sp<ThreadBase> thread = effect->thread().promote();
- if (thread != 0) {
- thread->checkSuspendOnEffectEnabled(effect, false, effect->sessionId());
- if (thread->type() == ThreadBase::OFFLOAD || thread->type() == ThreadBase::MMAP) {
- Mutex::Autolock _l(thread->mLock);
- thread->broadcast_l();
- }
- }
-
+ status_t status = effect->setEnabled(false, true /*fromHandle*/);
return status;
}
@@ -1724,12 +1715,9 @@
}
mDisconnected = true;
{
- sp<EffectModule> effect = mEffect.promote();
+ sp<EffectBase> effect = mEffect.promote();
if (effect != 0) {
- sp<ThreadBase> thread = effect->thread().promote();
- if (thread != 0) {
- thread->disconnectEffectHandle(this, unpinIfLast);
- } else if (effect->disconnectHandle(this, unpinIfLast) > 0) {
+ if (effect->disconnectHandle(this, unpinIfLast) > 0) {
ALOGW("%s Effect handle %p disconnected after thread destruction",
__func__, this);
}
@@ -1795,7 +1783,7 @@
}
AutoMutex _l(mLock);
- sp<EffectModule> effect = mEffect.promote();
+ sp<EffectBase> effect = mEffect.promote();
if (effect == 0 || mDisconnected) {
return DEAD_OBJECT;
}
@@ -1803,12 +1791,13 @@
if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) {
return INVALID_OPERATION;
}
- if (mClient == 0) {
- return INVALID_OPERATION;
- }
// handle commands that are not forwarded transparently to effect engine
if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) {
+ if (mClient == 0) {
+ return INVALID_OPERATION;
+ }
+
if (*replySize < sizeof(int)) {
android_errorWriteLog(0x534e4554, "32095713");
return BAD_VALUE;
@@ -1843,12 +1832,13 @@
}
// copy to local memory in case of client corruption b/32220769
- param = (effect_param_t *)realloc(param, size);
- if (param == NULL) {
+ auto *newParam = (effect_param_t *)realloc(param, size);
+ if (newParam == NULL) {
ALOGW("command(): out of memory");
status = NO_MEMORY;
break;
}
+ param = newParam;
memcpy(param, p, size);
int reply = 0;
@@ -1949,12 +1939,13 @@
AudioFlinger::EffectChain::EffectChain(ThreadBase *thread,
audio_session_t sessionId)
- : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0),
+ : mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0),
mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX),
- mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX)
+ mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX),
+ mEffectCallback(new EffectCallback(this, thread, thread->mAudioFlinger.get()))
{
mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
- if (thread == NULL) {
+ if (thread == nullptr) {
return;
}
mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) /
@@ -2020,43 +2011,30 @@
void AudioFlinger::EffectChain::clearInputBuffer()
{
Mutex::Autolock _l(mLock);
- sp<ThreadBase> thread = mThread.promote();
- if (thread == 0) {
- ALOGW("clearInputBuffer(): cannot promote mixer thread");
- return;
- }
- clearInputBuffer_l(thread);
+ clearInputBuffer_l();
}
// Must be called with EffectChain::mLock locked
-void AudioFlinger::EffectChain::clearInputBuffer_l(const sp<ThreadBase>& thread)
+void AudioFlinger::EffectChain::clearInputBuffer_l()
{
if (mInBuffer == NULL) {
return;
}
const size_t frameSize =
- audio_bytes_per_sample(EFFECT_BUFFER_FORMAT) * thread->channelCount();
+ audio_bytes_per_sample(EFFECT_BUFFER_FORMAT) * mEffectCallback->channelCount();
- memset(mInBuffer->audioBuffer()->raw, 0, thread->frameCount() * frameSize);
+ memset(mInBuffer->audioBuffer()->raw, 0, mEffectCallback->frameCount() * frameSize);
mInBuffer->commit();
}
// Must be called with EffectChain::mLock locked
void AudioFlinger::EffectChain::process_l()
{
- sp<ThreadBase> thread = mThread.promote();
- if (thread == 0) {
- ALOGW("process_l(): cannot promote mixer thread");
- return;
- }
- bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) ||
- (mSessionId == AUDIO_SESSION_OUTPUT_STAGE);
// never process effects when:
// - on an OFFLOAD thread
// - no more tracks are on the session and the effect tail has been rendered
- bool doProcess = (thread->type() != ThreadBase::OFFLOAD)
- && (thread->type() != ThreadBase::MMAP);
- if (!isGlobalSession) {
+ bool doProcess = !mEffectCallback->isOffloadOrMmap();
+ if (!audio_is_global_session(mSessionId)) {
bool tracksOnSession = (trackCnt() != 0);
if (!tracksOnSession && mTailBufferCount == 0) {
@@ -2067,7 +2045,7 @@
// if no track is active and the effect tail has not been rendered,
// the input buffer must be cleared here as the mixer process will not do it
if (tracksOnSession || mTailBufferCount > 0) {
- clearInputBuffer_l(thread);
+ clearInputBuffer_l();
if (mTailBufferCount > 0) {
mTailBufferCount--;
}
@@ -2103,14 +2081,13 @@
// createEffect_l() must be called with ThreadBase::mLock held
status_t AudioFlinger::EffectChain::createEffect_l(sp<EffectModule>& effect,
- ThreadBase *thread,
effect_descriptor_t *desc,
int id,
audio_session_t sessionId,
bool pinned)
{
Mutex::Autolock _l(mLock);
- effect = new EffectModule(thread, this, desc, id, sessionId, pinned);
+ effect = new EffectModule(mEffectCallback, desc, id, sessionId, pinned, AUDIO_PORT_HANDLE_NONE);
status_t lStatus = effect->status();
if (lStatus == NO_ERROR) {
lStatus = addEffect_ll(effect);
@@ -2133,12 +2110,7 @@
effect_descriptor_t desc = effect->desc();
uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK;
- effect->setChain(this);
- sp<ThreadBase> thread = mThread.promote();
- if (thread == 0) {
- return NO_INIT;
- }
- effect->setThread(thread);
+ effect->setCallback(mEffectCallback);
if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
// Auxiliary effects are inserted at the beginning of mEffects vector as
@@ -2149,13 +2121,13 @@
// 32 bit format. This is to avoid saturation in AudoMixer
// accumulation stage. Saturation is done in EffectModule::process() before
// calling the process in effect engine
- size_t numSamples = thread->frameCount();
+ size_t numSamples = mEffectCallback->frameCount();
sp<EffectBufferHalInterface> halBuffer;
#ifdef FLOAT_EFFECT_CHAIN
- status_t result = thread->mAudioFlinger->mEffectsFactoryHal->allocateBuffer(
+ status_t result = mEffectCallback->allocateHalBuffer(
numSamples * sizeof(float), &halBuffer);
#else
- status_t result = thread->mAudioFlinger->mEffectsFactoryHal->allocateBuffer(
+ status_t result = mEffectCallback->allocateHalBuffer(
numSamples * sizeof(int32_t), &halBuffer);
#endif
if (result != OK) return result;
@@ -2288,12 +2260,21 @@
return mEffects.size();
}
-// setDevice_l() must be called with ThreadBase::mLock held
-void AudioFlinger::EffectChain::setDevice_l(audio_devices_t device)
+// setDevices_l() must be called with ThreadBase::mLock held
+void AudioFlinger::EffectChain::setDevices_l(const AudioDeviceTypeAddrVector &devices)
{
size_t size = mEffects.size();
for (size_t i = 0; i < size; i++) {
- mEffects[i]->setDevice(device);
+ mEffects[i]->setDevices(devices);
+ }
+}
+
+// setInputDevice_l() must be called with ThreadBase::mLock held
+void AudioFlinger::EffectChain::setInputDevice_l(const AudioDeviceTypeAddr &device)
+{
+ size_t size = mEffects.size();
+ for (size_t i = 0; i < size; i++) {
+ mEffects[i]->setInputDevice(device);
}
}
@@ -2464,7 +2445,7 @@
if (effect != 0) {
desc->mEffect = effect;
effect->setSuspended(true);
- effect->setEnabled(false);
+ effect->setEnabled(false, false /*fromHandle*/);
}
}
} else {
@@ -2622,7 +2603,7 @@
// if effect is requested to suspended but was not yet enabled, suspend it now.
if (desc->mEffect == 0) {
desc->mEffect = effect;
- effect->setEnabled(false);
+ effect->setEnabled(false, false /*fromHandle*/);
effect->setSuspended(true);
}
} else {
@@ -2657,10 +2638,7 @@
void AudioFlinger::EffectChain::setThread(const sp<ThreadBase>& thread)
{
Mutex::Autolock _l(mLock);
- mThread = thread;
- for (size_t i = 0; i < mEffects.size(); i++) {
- mEffects[i]->setThread(thread);
- }
+ mEffectCallback->setThread(thread.get());
}
void AudioFlinger::EffectChain::checkOutputFlagCompatibility(audio_output_flags_t *flags) const
@@ -2720,4 +2698,549 @@
return true;
}
+// EffectCallbackInterface implementation
+status_t AudioFlinger::EffectChain::EffectCallback::createEffectHal(
+ const effect_uuid_t *pEffectUuid, int32_t sessionId, int32_t deviceId,
+ sp<EffectHalInterface> *effect) {
+ status_t status = NO_INIT;
+ sp<AudioFlinger> af = mAudioFlinger.promote();
+ if (af == nullptr) {
+ return status;
+ }
+ sp<EffectsFactoryHalInterface> effectsFactory = af->getEffectsFactory();
+ if (effectsFactory != 0) {
+ status = effectsFactory->createEffect(pEffectUuid, sessionId, io(), deviceId, effect);
+ }
+ return status;
+}
+
+bool AudioFlinger::EffectChain::EffectCallback::updateOrphanEffectChains(
+ const sp<AudioFlinger::EffectBase>& effect) {
+ sp<AudioFlinger> af = mAudioFlinger.promote();
+ if (af == nullptr) {
+ return false;
+ }
+ // in EffectChain context, an EffectBase is always from an EffectModule so static cast is safe
+ return af->updateOrphanEffectChains(effect->asEffectModule());
+}
+
+status_t AudioFlinger::EffectChain::EffectCallback::allocateHalBuffer(
+ size_t size, sp<EffectBufferHalInterface>* buffer) {
+ sp<AudioFlinger> af = mAudioFlinger.promote();
+ LOG_ALWAYS_FATAL_IF(af == nullptr, "allocateHalBuffer() could not retrieved audio flinger");
+ return af->mEffectsFactoryHal->allocateBuffer(size, buffer);
+}
+
+status_t AudioFlinger::EffectChain::EffectCallback::addEffectToHal(
+ sp<EffectHalInterface> effect) {
+ status_t result = NO_INIT;
+ sp<ThreadBase> t = mThread.promote();
+ if (t == nullptr) {
+ return result;
+ }
+ sp <StreamHalInterface> st = t->stream();
+ if (st == nullptr) {
+ return result;
+ }
+ result = st->addEffect(effect);
+ ALOGE_IF(result != OK, "Error when adding effect: %d", result);
+ return result;
+}
+
+status_t AudioFlinger::EffectChain::EffectCallback::removeEffectFromHal(
+ sp<EffectHalInterface> effect) {
+ status_t result = NO_INIT;
+ sp<ThreadBase> t = mThread.promote();
+ if (t == nullptr) {
+ return result;
+ }
+ sp <StreamHalInterface> st = t->stream();
+ if (st == nullptr) {
+ return result;
+ }
+ result = st->removeEffect(effect);
+ ALOGE_IF(result != OK, "Error when removing effect: %d", result);
+ return result;
+}
+
+audio_io_handle_t AudioFlinger::EffectChain::EffectCallback::io() const {
+ sp<ThreadBase> t = mThread.promote();
+ if (t == nullptr) {
+ return AUDIO_IO_HANDLE_NONE;
+ }
+ return t->id();
+}
+
+bool AudioFlinger::EffectChain::EffectCallback::isOutput() const {
+ sp<ThreadBase> t = mThread.promote();
+ if (t == nullptr) {
+ return true;
+ }
+ return t->isOutput();
+}
+
+bool AudioFlinger::EffectChain::EffectCallback::isOffload() const {
+ sp<ThreadBase> t = mThread.promote();
+ if (t == nullptr) {
+ return false;
+ }
+ return t->type() == ThreadBase::OFFLOAD;
+}
+
+bool AudioFlinger::EffectChain::EffectCallback::isOffloadOrDirect() const {
+ sp<ThreadBase> t = mThread.promote();
+ if (t == nullptr) {
+ return false;
+ }
+ return t->type() == ThreadBase::OFFLOAD || t->type() == ThreadBase::DIRECT;
+}
+
+bool AudioFlinger::EffectChain::EffectCallback::isOffloadOrMmap() const {
+ sp<ThreadBase> t = mThread.promote();
+ if (t == nullptr) {
+ return false;
+ }
+ return t->type() == ThreadBase::OFFLOAD || t->type() == ThreadBase::MMAP;
+}
+
+uint32_t AudioFlinger::EffectChain::EffectCallback::sampleRate() const {
+ sp<ThreadBase> t = mThread.promote();
+ if (t == nullptr) {
+ return 0;
+ }
+ return t->sampleRate();
+}
+
+audio_channel_mask_t AudioFlinger::EffectChain::EffectCallback::channelMask() const {
+ sp<ThreadBase> t = mThread.promote();
+ if (t == nullptr) {
+ return AUDIO_CHANNEL_NONE;
+ }
+ return t->channelMask();
+}
+
+uint32_t AudioFlinger::EffectChain::EffectCallback::channelCount() const {
+ sp<ThreadBase> t = mThread.promote();
+ if (t == nullptr) {
+ return 0;
+ }
+ return t->channelCount();
+}
+
+size_t AudioFlinger::EffectChain::EffectCallback::frameCount() const {
+ sp<ThreadBase> t = mThread.promote();
+ if (t == nullptr) {
+ return 0;
+ }
+ return t->frameCount();
+}
+
+uint32_t AudioFlinger::EffectChain::EffectCallback::latency() const {
+ sp<ThreadBase> t = mThread.promote();
+ if (t == nullptr) {
+ return 0;
+ }
+ return t->latency_l();
+}
+
+void AudioFlinger::EffectChain::EffectCallback::setVolumeForOutput(float left, float right) const {
+ sp<ThreadBase> t = mThread.promote();
+ if (t == nullptr) {
+ return;
+ }
+ t->setVolumeForOutput_l(left, right);
+}
+
+void AudioFlinger::EffectChain::EffectCallback::checkSuspendOnEffectEnabled(
+ const sp<EffectBase>& effect, bool enabled, bool threadLocked) {
+ sp<ThreadBase> t = mThread.promote();
+ if (t == nullptr) {
+ return;
+ }
+ t->checkSuspendOnEffectEnabled(enabled, effect->sessionId(), threadLocked);
+
+ sp<EffectChain> c = mChain.promote();
+ if (c == nullptr) {
+ return;
+ }
+ // in EffectChain context, an EffectBase is always from an EffectModule so static cast is safe
+ c->checkSuspendOnEffectEnabled(effect->asEffectModule(), enabled);
+}
+
+void AudioFlinger::EffectChain::EffectCallback::onEffectEnable(const sp<EffectBase>& effect) {
+ sp<ThreadBase> t = mThread.promote();
+ if (t == nullptr) {
+ return;
+ }
+ // in EffectChain context, an EffectBase is always from an EffectModule so static cast is safe
+ t->onEffectEnable(effect->asEffectModule());
+}
+
+void AudioFlinger::EffectChain::EffectCallback::onEffectDisable(const sp<EffectBase>& effect) {
+ checkSuspendOnEffectEnabled(effect, false, false /*threadLocked*/);
+
+ sp<ThreadBase> t = mThread.promote();
+ if (t == nullptr) {
+ return;
+ }
+ t->onEffectDisable();
+}
+
+bool AudioFlinger::EffectChain::EffectCallback::disconnectEffectHandle(EffectHandle *handle,
+ bool unpinIfLast) {
+ sp<ThreadBase> t = mThread.promote();
+ if (t == nullptr) {
+ return false;
+ }
+ t->disconnectEffectHandle(handle, unpinIfLast);
+ return true;
+}
+
+void AudioFlinger::EffectChain::EffectCallback::resetVolume() {
+ sp<EffectChain> c = mChain.promote();
+ if (c == nullptr) {
+ return;
+ }
+ c->resetVolume_l();
+
+}
+
+uint32_t AudioFlinger::EffectChain::EffectCallback::strategy() const {
+ sp<EffectChain> c = mChain.promote();
+ if (c == nullptr) {
+ return PRODUCT_STRATEGY_NONE;
+ }
+ return c->strategy();
+}
+
+int32_t AudioFlinger::EffectChain::EffectCallback::activeTrackCnt() const {
+ sp<EffectChain> c = mChain.promote();
+ if (c == nullptr) {
+ return 0;
+ }
+ return c->activeTrackCnt();
+}
+
+
+#undef LOG_TAG
+#define LOG_TAG "AudioFlinger::DeviceEffectProxy"
+
+status_t AudioFlinger::DeviceEffectProxy::setEnabled(bool enabled, bool fromHandle)
+{
+ status_t status = EffectBase::setEnabled(enabled, fromHandle);
+ Mutex::Autolock _l(mProxyLock);
+ if (status == NO_ERROR) {
+ for (auto& handle : mEffectHandles) {
+ if (enabled) {
+ status = handle.second->enable();
+ } else {
+ status = handle.second->disable();
+ }
+ }
+ }
+ ALOGV("%s enable %d status %d", __func__, enabled, status);
+ return status;
+}
+
+status_t AudioFlinger::DeviceEffectProxy::init(
+ const std::map <audio_patch_handle_t, PatchPanel::Patch>& patches) {
+//For all audio patches
+//If src or sink device match
+//If the effect is HW accelerated
+// if no corresponding effect module
+// Create EffectModule: mHalEffect
+//Create and attach EffectHandle
+//If the effect is not HW accelerated and the patch sink or src is a mixer port
+// Create Effect on patch input or output thread on session -1
+//Add EffectHandle to EffectHandle map of Effect Proxy:
+ ALOGV("%s device type %d address %s", __func__, mDevice.mType, mDevice.getAddress());
+ status_t status = NO_ERROR;
+ for (auto &patch : patches) {
+ status = onCreatePatch(patch.first, patch.second);
+ ALOGV("%s onCreatePatch status %d", __func__, status);
+ if (status == BAD_VALUE) {
+ return status;
+ }
+ }
+ return status;
+}
+
+status_t AudioFlinger::DeviceEffectProxy::onCreatePatch(
+ audio_patch_handle_t patchHandle, const AudioFlinger::PatchPanel::Patch& patch) {
+ status_t status = NAME_NOT_FOUND;
+ sp<EffectHandle> handle;
+ // only consider source[0] as this is the only "true" source of a patch
+ status = checkPort(patch, &patch.mAudioPatch.sources[0], &handle);
+ ALOGV("%s source checkPort status %d", __func__, status);
+ for (uint32_t i = 0; i < patch.mAudioPatch.num_sinks && status == NAME_NOT_FOUND; i++) {
+ status = checkPort(patch, &patch.mAudioPatch.sinks[i], &handle);
+ ALOGV("%s sink %d checkPort status %d", __func__, i, status);
+ }
+ if (status == NO_ERROR || status == ALREADY_EXISTS) {
+ Mutex::Autolock _l(mProxyLock);
+ mEffectHandles.emplace(patchHandle, handle);
+ }
+ ALOGW_IF(status == BAD_VALUE,
+ "%s cannot attach effect %s on patch %d", __func__, mDescriptor.name, patchHandle);
+
+ return status;
+}
+
+status_t AudioFlinger::DeviceEffectProxy::checkPort(const PatchPanel::Patch& patch,
+ const struct audio_port_config *port, sp <EffectHandle> *handle) {
+
+ ALOGV("%s type %d device type %d address %s device ID %d patch.isSoftware() %d",
+ __func__, port->type, port->ext.device.type,
+ port->ext.device.address, port->id, patch.isSoftware());
+ if (port->type != AUDIO_PORT_TYPE_DEVICE || port->ext.device.type != mDevice.mType
+ || port->ext.device.address != mDevice.mAddress) {
+ return NAME_NOT_FOUND;
+ }
+ status_t status = NAME_NOT_FOUND;
+
+ if (mDescriptor.flags & EFFECT_FLAG_HW_ACC_TUNNEL) {
+ Mutex::Autolock _l(mProxyLock);
+ mDevicePort = *port;
+ mHalEffect = new EffectModule(mMyCallback,
+ const_cast<effect_descriptor_t *>(&mDescriptor),
+ mMyCallback->newEffectId(), AUDIO_SESSION_DEVICE,
+ false /* pinned */, port->id);
+ if (audio_is_input_device(mDevice.mType)) {
+ mHalEffect->setInputDevice(mDevice);
+ } else {
+ mHalEffect->setDevices({mDevice});
+ }
+ *handle = new EffectHandle(mHalEffect, nullptr, nullptr, 0 /*priority*/);
+ status = (*handle)->initCheck();
+ if (status == OK) {
+ status = mHalEffect->addHandle((*handle).get());
+ } else {
+ mHalEffect.clear();
+ mDevicePort.id = AUDIO_PORT_HANDLE_NONE;
+ }
+ } else if (patch.isSoftware() || patch.thread().promote() != nullptr) {
+ sp <ThreadBase> thread;
+ if (audio_port_config_has_input_direction(port)) {
+ if (patch.isSoftware()) {
+ thread = patch.mRecord.thread();
+ } else {
+ thread = patch.thread().promote();
+ }
+ } else {
+ if (patch.isSoftware()) {
+ thread = patch.mPlayback.thread();
+ } else {
+ thread = patch.thread().promote();
+ }
+ }
+ int enabled;
+ *handle = thread->createEffect_l(nullptr, nullptr, 0, AUDIO_SESSION_DEVICE,
+ const_cast<effect_descriptor_t *>(&mDescriptor),
+ &enabled, &status, false);
+ ALOGV("%s thread->createEffect_l status %d", __func__, status);
+ } else {
+ status = BAD_VALUE;
+ }
+ if (status == NO_ERROR || status == ALREADY_EXISTS) {
+ if (isEnabled()) {
+ (*handle)->enable();
+ } else {
+ (*handle)->disable();
+ }
+ }
+ return status;
+}
+
+void AudioFlinger::DeviceEffectProxy::onReleasePatch(audio_patch_handle_t patchHandle) {
+ Mutex::Autolock _l(mProxyLock);
+ mEffectHandles.erase(patchHandle);
+}
+
+
+size_t AudioFlinger::DeviceEffectProxy::removeEffect(const sp<EffectModule>& effect)
+{
+ Mutex::Autolock _l(mProxyLock);
+ if (effect == mHalEffect) {
+ mHalEffect.clear();
+ mDevicePort.id = AUDIO_PORT_HANDLE_NONE;
+ }
+ return mHalEffect == nullptr ? 0 : 1;
+}
+
+status_t AudioFlinger::DeviceEffectProxy::addEffectToHal(
+ sp<EffectHalInterface> effect) {
+ if (mHalEffect == nullptr) {
+ return NO_INIT;
+ }
+ return mManagerCallback->addEffectToHal(
+ mDevicePort.id, mDevicePort.ext.device.hw_module, effect);
+}
+
+status_t AudioFlinger::DeviceEffectProxy::removeEffectFromHal(
+ sp<EffectHalInterface> effect) {
+ if (mHalEffect == nullptr) {
+ return NO_INIT;
+ }
+ return mManagerCallback->removeEffectFromHal(
+ mDevicePort.id, mDevicePort.ext.device.hw_module, effect);
+}
+
+bool AudioFlinger::DeviceEffectProxy::isOutput() const {
+ if (mDevicePort.id != AUDIO_PORT_HANDLE_NONE) {
+ return mDevicePort.role == AUDIO_PORT_ROLE_SINK;
+ }
+ return true;
+}
+
+uint32_t AudioFlinger::DeviceEffectProxy::sampleRate() const {
+ if (mDevicePort.id != AUDIO_PORT_HANDLE_NONE &&
+ (mDevicePort.config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) != 0) {
+ return mDevicePort.sample_rate;
+ }
+ return DEFAULT_OUTPUT_SAMPLE_RATE;
+}
+
+audio_channel_mask_t AudioFlinger::DeviceEffectProxy::channelMask() const {
+ if (mDevicePort.id != AUDIO_PORT_HANDLE_NONE &&
+ (mDevicePort.config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) != 0) {
+ return mDevicePort.channel_mask;
+ }
+ return AUDIO_CHANNEL_OUT_STEREO;
+}
+
+uint32_t AudioFlinger::DeviceEffectProxy::channelCount() const {
+ if (isOutput()) {
+ return audio_channel_count_from_out_mask(channelMask());
+ }
+ return audio_channel_count_from_in_mask(channelMask());
+}
+
+void AudioFlinger::DeviceEffectProxy::dump(int fd, int spaces) {
+ const Vector<String16> args;
+ EffectBase::dump(fd, args);
+
+ const bool locked = dumpTryLock(mProxyLock);
+ if (!locked) {
+ String8 result("DeviceEffectProxy may be deadlocked\n");
+ write(fd, result.string(), result.size());
+ }
+
+ String8 outStr;
+ if (mHalEffect != nullptr) {
+ outStr.appendFormat("%*sHAL Effect Id: %d\n", spaces, "", mHalEffect->id());
+ } else {
+ outStr.appendFormat("%*sNO HAL Effect\n", spaces, "");
+ }
+ write(fd, outStr.string(), outStr.size());
+ outStr.clear();
+
+ outStr.appendFormat("%*sSub Effects:\n", spaces, "");
+ write(fd, outStr.string(), outStr.size());
+ outStr.clear();
+
+ for (const auto& iter : mEffectHandles) {
+ outStr.appendFormat("%*sEffect for patch handle %d:\n", spaces + 2, "", iter.first);
+ write(fd, outStr.string(), outStr.size());
+ outStr.clear();
+ sp<EffectBase> effect = iter.second->effect().promote();
+ if (effect != nullptr) {
+ effect->dump(fd, args);
+ }
+ }
+
+ if (locked) {
+ mLock.unlock();
+ }
+}
+
+#undef LOG_TAG
+#define LOG_TAG "AudioFlinger::DeviceEffectProxy::ProxyCallback"
+
+int AudioFlinger::DeviceEffectProxy::ProxyCallback::newEffectId() {
+ return mManagerCallback->newEffectId();
+}
+
+
+bool AudioFlinger::DeviceEffectProxy::ProxyCallback::disconnectEffectHandle(
+ EffectHandle *handle, bool unpinIfLast) {
+ sp<EffectBase> effectBase = handle->effect().promote();
+ if (effectBase == nullptr) {
+ return false;
+ }
+
+ sp<EffectModule> effect = effectBase->asEffectModule();
+ if (effect == nullptr) {
+ return false;
+ }
+
+ // restore suspended effects if the disconnected handle was enabled and the last one.
+ bool remove = (effect->removeHandle(handle) == 0) && (!effect->isPinned() || unpinIfLast);
+ if (remove) {
+ sp<DeviceEffectProxy> proxy = mProxy.promote();
+ if (proxy != nullptr) {
+ proxy->removeEffect(effect);
+ }
+ if (handle->enabled()) {
+ effectBase->checkSuspendOnEffectEnabled(false, false /*threadLocked*/);
+ }
+ }
+ return true;
+}
+
+status_t AudioFlinger::DeviceEffectProxy::ProxyCallback::createEffectHal(
+ const effect_uuid_t *pEffectUuid, int32_t sessionId, int32_t deviceId,
+ sp<EffectHalInterface> *effect) {
+ return mManagerCallback->createEffectHal(pEffectUuid, sessionId, deviceId, effect);
+}
+
+status_t AudioFlinger::DeviceEffectProxy::ProxyCallback::addEffectToHal(
+ sp<EffectHalInterface> effect) {
+ sp<DeviceEffectProxy> proxy = mProxy.promote();
+ if (proxy == nullptr) {
+ return NO_INIT;
+ }
+ return proxy->addEffectToHal(effect);
+}
+
+status_t AudioFlinger::DeviceEffectProxy::ProxyCallback::removeEffectFromHal(
+ sp<EffectHalInterface> effect) {
+ sp<DeviceEffectProxy> proxy = mProxy.promote();
+ if (proxy == nullptr) {
+ return NO_INIT;
+ }
+ return proxy->addEffectToHal(effect);
+}
+
+bool AudioFlinger::DeviceEffectProxy::ProxyCallback::isOutput() const {
+ sp<DeviceEffectProxy> proxy = mProxy.promote();
+ if (proxy == nullptr) {
+ return true;
+ }
+ return proxy->isOutput();
+}
+
+uint32_t AudioFlinger::DeviceEffectProxy::ProxyCallback::sampleRate() const {
+ sp<DeviceEffectProxy> proxy = mProxy.promote();
+ if (proxy == nullptr) {
+ return DEFAULT_OUTPUT_SAMPLE_RATE;
+ }
+ return proxy->sampleRate();
+}
+
+audio_channel_mask_t AudioFlinger::DeviceEffectProxy::ProxyCallback::channelMask() const {
+ sp<DeviceEffectProxy> proxy = mProxy.promote();
+ if (proxy == nullptr) {
+ return AUDIO_CHANNEL_OUT_STEREO;
+ }
+ return proxy->channelMask();
+}
+
+uint32_t AudioFlinger::DeviceEffectProxy::ProxyCallback::channelCount() const {
+ sp<DeviceEffectProxy> proxy = mProxy.promote();
+ if (proxy == nullptr) {
+ return 2;
+ }
+ return proxy->channelCount();
+}
+
} // namespace android
diff --git a/services/audioflinger/Effects.h b/services/audioflinger/Effects.h
index 220874d..40bb226 100644
--- a/services/audioflinger/Effects.h
+++ b/services/audioflinger/Effects.h
@@ -21,34 +21,78 @@
//--- Audio Effect Management
-// EffectModule and EffectChain classes both have their own mutex to protect
+// Interface implemented by the EffectModule parent or owner (e.g an EffectChain) to abstract
+// interactions between the EffectModule and the reset of the audio framework.
+class EffectCallbackInterface : public RefBase {
+public:
+ ~EffectCallbackInterface() override = default;
+
+ // Trivial methods usually implemented with help from ThreadBase
+ virtual audio_io_handle_t io() const = 0;
+ virtual bool isOutput() const = 0;
+ virtual bool isOffload() const = 0;
+ virtual bool isOffloadOrDirect() const = 0;
+ virtual bool isOffloadOrMmap() const = 0;
+ virtual uint32_t sampleRate() const = 0;
+ virtual audio_channel_mask_t channelMask() const = 0;
+ virtual uint32_t channelCount() const = 0;
+ virtual size_t frameCount() const = 0;
+
+ // Non trivial methods usually implemented with help from ThreadBase:
+ // pay attention to mutex locking order
+ virtual uint32_t latency() const { return 0; }
+ virtual status_t addEffectToHal(sp<EffectHalInterface> effect) = 0;
+ virtual status_t removeEffectFromHal(sp<EffectHalInterface> effect) = 0;
+ virtual void setVolumeForOutput(float left, float right) const = 0;
+ virtual bool disconnectEffectHandle(EffectHandle *handle, bool unpinIfLast) = 0;
+ virtual void checkSuspendOnEffectEnabled(const sp<EffectBase>& effect,
+ bool enabled,
+ bool threadLocked) = 0;
+ virtual void onEffectEnable(const sp<EffectBase>& effect) = 0;
+ virtual void onEffectDisable(const sp<EffectBase>& effect) = 0;
+
+ // Methods usually implemented with help from AudioFlinger: pay attention to mutex locking order
+ virtual status_t createEffectHal(const effect_uuid_t *pEffectUuid,
+ int32_t sessionId, int32_t deviceId, sp<EffectHalInterface> *effect) = 0;
+ virtual status_t allocateHalBuffer(size_t size, sp<EffectBufferHalInterface>* buffer) = 0;
+ virtual bool updateOrphanEffectChains(const sp<EffectBase>& effect) = 0;
+
+ // Methods usually implemented with help from EffectChain: pay attention to mutex locking order
+ virtual uint32_t strategy() const = 0;
+ virtual int32_t activeTrackCnt() const = 0;
+ virtual void resetVolume() = 0;
+
+ virtual wp<EffectChain> chain() const = 0;
+};
+
+// EffectBase(EffectModule) and EffectChain classes both have their own mutex to protect
// state changes or resource modifications. Always respect the following order
// if multiple mutexes must be acquired to avoid cross deadlock:
-// AudioFlinger -> ThreadBase -> EffectChain -> EffectModule
-// AudioHandle -> ThreadBase -> EffectChain -> EffectModule
+// AudioFlinger -> ThreadBase -> EffectChain -> EffectBase(EffectModule)
+// AudioHandle -> ThreadBase -> EffectChain -> EffectBase(EffectModule)
+
+// NOTE: When implementing the EffectCallbackInterface, in an EffectChain or other, it is important
+// to pay attention to this locking order as some callback methods can be called from a state where
+// EffectModule and/or EffectChain mutexes are held.
+
// In addition, methods that lock the AudioPolicyService mutex (getOutputForEffect(),
// startOutput(), getInputForAttr(), releaseInput()...) should never be called with AudioFlinger or
// Threadbase mutex locked to avoid cross deadlock with other clients calling AudioPolicyService
// methods that in turn call AudioFlinger thus locking the same mutexes in the reverse order.
-// The EffectModule class is a wrapper object controlling the effect engine implementation
-// in the effect library. It prevents concurrent calls to process() and command() functions
-// from different client threads. It keeps a list of EffectHandle objects corresponding
-// to all client applications using this effect and notifies applications of effect state,
-// control or parameter changes. It manages the activation state machine to send appropriate
-// reset, enable, disable commands to effect engine and provide volume
-// ramping when effects are activated/deactivated.
-// When controlling an auxiliary effect, the EffectModule also provides an input buffer used by
-// the attached track(s) to accumulate their auxiliary channel.
-class EffectModule : public RefBase {
+
+// The EffectBase class contains common properties, state and behavior for and EffectModule or
+// other derived classes managing an audio effect instance within the effect framework.
+// It also contains the class mutex (see comment on locking order above).
+class EffectBase : public RefBase {
public:
- EffectModule(ThreadBase *thread,
- const wp<AudioFlinger::EffectChain>& chain,
- effect_descriptor_t *desc,
- int id,
- audio_session_t sessionId,
- bool pinned);
- virtual ~EffectModule();
+ EffectBase(const sp<EffectCallbackInterface>& callback,
+ effect_descriptor_t *desc,
+ int id,
+ audio_session_t sessionId,
+ bool pinned);
+
+ ~EffectBase() override = default;
enum effect_state {
IDLE,
@@ -60,71 +104,14 @@
DESTROYED
};
- int id() const { return mId; }
- void process();
- bool updateState();
- status_t command(uint32_t cmdCode,
- uint32_t cmdSize,
- void *pCmdData,
- uint32_t *replySize,
- void *pReplyData);
-
- void reset_l();
- status_t configure();
- status_t init();
+ int id() const { return mId; }
effect_state state() const {
return mState;
}
- uint32_t status() {
- return mStatus;
- }
audio_session_t sessionId() const {
return mSessionId;
}
- status_t setEnabled(bool enabled);
- status_t setEnabled_l(bool enabled);
- bool isEnabled() const;
- bool isProcessEnabled() const;
- bool isOffloadedOrDirect() const;
- bool isVolumeControlEnabled() const;
-
- void setInBuffer(const sp<EffectBufferHalInterface>& buffer);
- int16_t *inBuffer() const {
- return mInBuffer != 0 ? reinterpret_cast<int16_t*>(mInBuffer->ptr()) : NULL;
- }
- void setOutBuffer(const sp<EffectBufferHalInterface>& buffer);
- int16_t *outBuffer() const {
- return mOutBuffer != 0 ? reinterpret_cast<int16_t*>(mOutBuffer->ptr()) : NULL;
- }
- void setChain(const wp<EffectChain>& chain) { mChain = chain; }
- void setThread(const wp<ThreadBase>& thread)
- { mThread = thread; mThreadType = thread.promote()->type(); }
- const wp<ThreadBase>& thread() { return mThread; }
-
- status_t addHandle(EffectHandle *handle);
- ssize_t disconnectHandle(EffectHandle *handle, bool unpinIfLast);
- ssize_t removeHandle(EffectHandle *handle);
- ssize_t removeHandle_l(EffectHandle *handle);
-
const effect_descriptor_t& desc() const { return mDescriptor; }
- wp<EffectChain>& chain() { return mChain; }
-
- status_t setDevice(audio_devices_t device);
- status_t setVolume(uint32_t *left, uint32_t *right, bool controller);
- status_t setMode(audio_mode_t mode);
- status_t setAudioSource(audio_source_t source);
- status_t start();
- status_t stop();
- void setSuspended(bool suspended);
- bool suspended() const;
-
- EffectHandle* controlHandle_l();
-
- bool isPinned() const { return mPinned; }
- void unPin() { mPinned = false; }
- bool purgeHandles();
- void lock() { mLock.lock(); }
- void unlock() { mLock.unlock(); }
bool isOffloadable() const
{ return (mDescriptor.flags & EFFECT_FLAG_OFFLOAD_SUPPORTED) != 0; }
bool isImplementationSoftware() const
@@ -137,18 +124,143 @@
bool isVolumeMonitor() const
{ return (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK)
== EFFECT_FLAG_VOLUME_MONITOR; }
- status_t setOffloaded(bool offloaded, audio_io_handle_t io);
- bool isOffloaded() const;
- void addEffectToHal_l();
- void release_l();
+
+ virtual status_t setEnabled(bool enabled, bool fromHandle);
+ status_t setEnabled_l(bool enabled);
+ bool isEnabled() const;
+
+ void setSuspended(bool suspended);
+ bool suspended() const;
+
+ virtual status_t command(uint32_t cmdCode __unused,
+ uint32_t cmdSize __unused,
+ void *pCmdData __unused,
+ uint32_t *replySize __unused,
+ void *pReplyData __unused) { return NO_ERROR; };
+
+ void setCallback(const sp<EffectCallbackInterface>& callback) { mCallback = callback; }
+ sp<EffectCallbackInterface>& callback() { return mCallback; }
+
+ status_t addHandle(EffectHandle *handle);
+ ssize_t disconnectHandle(EffectHandle *handle, bool unpinIfLast);
+ ssize_t removeHandle(EffectHandle *handle);
+ virtual ssize_t removeHandle_l(EffectHandle *handle);
+ EffectHandle* controlHandle_l();
+ bool purgeHandles();
+
+ void checkSuspendOnEffectEnabled(bool enabled, bool threadLocked);
+
+ bool isPinned() const { return mPinned; }
+ void unPin() { mPinned = false; }
+
+ void lock() { mLock.lock(); }
+ void unlock() { mLock.unlock(); }
status_t updatePolicyState();
+ virtual sp<EffectModule> asEffectModule() { return nullptr; }
+ virtual sp<DeviceEffectProxy> asDeviceEffectProxy() { return nullptr; }
+
void dump(int fd, const Vector<String16>& args);
private:
friend class AudioFlinger; // for mHandles
- bool mPinned;
+ bool mPinned = false;
+
+ DISALLOW_COPY_AND_ASSIGN(EffectBase);
+
+mutable Mutex mLock; // mutex for process, commands and handles list protection
+ sp<EffectCallbackInterface> mCallback; // parent effect chain
+ const int mId; // this instance unique ID
+ const audio_session_t mSessionId; // audio session ID
+ const effect_descriptor_t mDescriptor;// effect descriptor received from effect engine
+ effect_state mState = IDLE; // current activation state
+ // effect is suspended: temporarily disabled by framework
+ bool mSuspended = false;
+
+ Vector<EffectHandle *> mHandles; // list of client handles
+ // First handle in mHandles has highest priority and controls the effect module
+
+ // Audio policy effect state management
+ // Mutex protecting transactions with audio policy manager as mLock cannot
+ // be held to avoid cross deadlocks with audio policy mutex
+ Mutex mPolicyLock;
+ // Effect is registered in APM or not
+ bool mPolicyRegistered = false;
+ // Effect enabled state communicated to APM. Enabled state corresponds to
+ // state requested by the EffectHandle with control
+ bool mPolicyEnabled = false;
+};
+
+// The EffectModule class is a wrapper object controlling the effect engine implementation
+// in the effect library. It prevents concurrent calls to process() and command() functions
+// from different client threads. It keeps a list of EffectHandle objects corresponding
+// to all client applications using this effect and notifies applications of effect state,
+// control or parameter changes. It manages the activation state machine to send appropriate
+// reset, enable, disable commands to effect engine and provide volume
+// ramping when effects are activated/deactivated.
+// When controlling an auxiliary effect, the EffectModule also provides an input buffer used by
+// the attached track(s) to accumulate their auxiliary channel.
+class EffectModule : public EffectBase {
+public:
+ EffectModule(const sp<EffectCallbackInterface>& callabck,
+ effect_descriptor_t *desc,
+ int id,
+ audio_session_t sessionId,
+ bool pinned,
+ audio_port_handle_t deviceId);
+ virtual ~EffectModule();
+
+ void process();
+ bool updateState();
+ status_t command(uint32_t cmdCode,
+ uint32_t cmdSize,
+ void *pCmdData,
+ uint32_t *replySize,
+ void *pReplyData) override;
+
+ void reset_l();
+ status_t configure();
+ status_t init();
+
+ uint32_t status() {
+ return mStatus;
+ }
+
+ bool isProcessEnabled() const;
+ bool isOffloadedOrDirect() const;
+ bool isVolumeControlEnabled() const;
+
+ void setInBuffer(const sp<EffectBufferHalInterface>& buffer);
+ int16_t *inBuffer() const {
+ return mInBuffer != 0 ? reinterpret_cast<int16_t*>(mInBuffer->ptr()) : NULL;
+ }
+ void setOutBuffer(const sp<EffectBufferHalInterface>& buffer);
+ int16_t *outBuffer() const {
+ return mOutBuffer != 0 ? reinterpret_cast<int16_t*>(mOutBuffer->ptr()) : NULL;
+ }
+
+ ssize_t removeHandle_l(EffectHandle *handle) override;
+
+ status_t setDevices(const AudioDeviceTypeAddrVector &devices);
+ status_t setInputDevice(const AudioDeviceTypeAddr &device);
+ status_t setVolume(uint32_t *left, uint32_t *right, bool controller);
+ status_t setMode(audio_mode_t mode);
+ status_t setAudioSource(audio_source_t source);
+ status_t start();
+ status_t stop();
+
+ status_t setOffloaded(bool offloaded, audio_io_handle_t io);
+ bool isOffloaded() const;
+ void addEffectToHal_l();
+ void release_l();
+
+ sp<EffectModule> asEffectModule() override { return this; }
+
+ void dump(int fd, const Vector<String16>& args);
+
+private:
+ friend class AudioFlinger; // for mHandles
// Maximum time allocated to effect engines to complete the turn off sequence
static const uint32_t MAX_DISABLE_TIME_MS = 10000;
@@ -157,29 +269,19 @@
status_t start_l();
status_t stop_l();
- status_t remove_effect_from_hal_l();
+ status_t removeEffectFromHal_l();
+ status_t sendSetAudioDevicesCommand(const AudioDeviceTypeAddrVector &devices, uint32_t cmdCode);
-mutable Mutex mLock; // mutex for process, commands and handles list protection
- wp<ThreadBase> mThread; // parent thread
- ThreadBase::type_t mThreadType; // parent thread type
- wp<EffectChain> mChain; // parent effect chain
- const int mId; // this instance unique ID
- const audio_session_t mSessionId; // audio session ID
- const effect_descriptor_t mDescriptor;// effect descriptor received from effect engine
effect_config_t mConfig; // input and output audio configuration
sp<EffectHalInterface> mEffectInterface; // Effect module HAL
sp<EffectBufferHalInterface> mInBuffer; // Buffers for interacting with HAL
sp<EffectBufferHalInterface> mOutBuffer;
status_t mStatus; // initialization status
- effect_state mState; // current activation state
- Vector<EffectHandle *> mHandles; // list of client handles
// First handle in mHandles has highest priority and controls the effect module
uint32_t mMaxDisableWaitCnt; // maximum grace period before forcing an effect off after
// sending disable command.
uint32_t mDisableWaitCnt; // current process() calls count during disable period.
- bool mSuspended; // effect is suspended: temporarily disabled by framework
bool mOffloaded; // effect is currently offloaded to the audio DSP
- wp<AudioFlinger> mAudioFlinger;
#ifdef FLOAT_EFFECT_CHAIN
bool mSupportsFloat; // effect supports float processing
@@ -206,16 +308,6 @@
static constexpr pid_t INVALID_PID = (pid_t)-1;
// this tid is allowed to call setVolume() without acquiring the mutex.
pid_t mSetVolumeReentrantTid = INVALID_PID;
-
- // Audio policy effect state management
- // Mutex protecting transactions with audio policy manager as mLock cannot
- // be held to avoid cross deadlocks with audio policy mutex
- Mutex mPolicyLock;
- // Effect is registered in APM or not
- bool mPolicyRegistered = false;
- // Effect enabled state communicated to APM. Enabled state corresponds to
- // state requested by the EffectHandle with control
- bool mPolicyEnabled = false;
};
// The EffectHandle class implements the IEffect interface. It provides resources
@@ -227,7 +319,7 @@
class EffectHandle: public android::BnEffect {
public:
- EffectHandle(const sp<EffectModule>& effect,
+ EffectHandle(const sp<EffectBase>& effect,
const sp<AudioFlinger::Client>& client,
const sp<IEffectClient>& effectClient,
int32_t priority);
@@ -265,9 +357,9 @@
bool enabled() const { return mEnabled; }
// Getters
- wp<EffectModule> effect() const { return mEffect; }
+ wp<EffectBase> effect() const { return mEffect; }
int id() const {
- sp<EffectModule> effect = mEffect.promote();
+ sp<EffectBase> effect = mEffect.promote();
if (effect == 0) {
return 0;
}
@@ -284,7 +376,7 @@
DISALLOW_COPY_AND_ASSIGN(EffectHandle);
Mutex mLock; // protects IEffect method calls
- wp<EffectModule> mEffect; // pointer to controlled EffectModule
+ wp<EffectBase> mEffect; // pointer to controlled EffectModule
sp<IEffectClient> mEffectClient; // callback interface for client notifications
/*const*/ sp<Client> mClient; // client for shared memory allocation, see disconnect()
sp<IMemory> mCblkMemory; // shared memory for control block
@@ -331,7 +423,6 @@
}
status_t createEffect_l(sp<EffectModule>& effect,
- ThreadBase *thread,
effect_descriptor_t *desc,
int id,
audio_session_t sessionId,
@@ -350,7 +441,8 @@
// FIXME use float to improve the dynamic range
bool setVolume_l(uint32_t *left, uint32_t *right, bool force = false);
void resetVolume_l();
- void setDevice_l(audio_devices_t device);
+ void setDevices_l(const AudioDeviceTypeAddrVector &devices);
+ void setInputDevice_l(const AudioDeviceTypeAddr &device);
void setMode_l(audio_mode_t mode);
void setAudioSource_l(audio_source_t source);
@@ -386,9 +478,8 @@
bool suspend);
// suspend all eligible effects
void setEffectSuspendedAll_l(bool suspend);
- // check if effects should be suspend or restored when a given effect is enable or disabled
- void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
- bool enabled);
+ // check if effects should be suspended or restored when a given effect is enable or disabled
+ void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, bool enabled);
void clearInputBuffer();
@@ -413,9 +504,60 @@
// isCompatibleWithThread_l() must be called with thread->mLock held
bool isCompatibleWithThread_l(const sp<ThreadBase>& thread) const;
+ sp<EffectCallbackInterface> effectCallback() const { return mEffectCallback; }
+ wp<ThreadBase> thread() const { return mEffectCallback->thread(); }
+
void dump(int fd, const Vector<String16>& args);
private:
+
+ class EffectCallback : public EffectCallbackInterface {
+ public:
+ EffectCallback(EffectChain *chain, ThreadBase *thread, AudioFlinger *audioFlinger)
+ : mChain(chain), mThread(thread), mAudioFlinger(audioFlinger) {}
+
+ status_t createEffectHal(const effect_uuid_t *pEffectUuid,
+ int32_t sessionId, int32_t deviceId, sp<EffectHalInterface> *effect) override;
+ status_t allocateHalBuffer(size_t size, sp<EffectBufferHalInterface>* buffer) override;
+ bool updateOrphanEffectChains(const sp<EffectBase>& effect) override;
+
+ audio_io_handle_t io() const override;
+ bool isOutput() const override;
+ bool isOffload() const override;
+ bool isOffloadOrDirect() const override;
+ bool isOffloadOrMmap() const override;
+
+ uint32_t sampleRate() const override;
+ audio_channel_mask_t channelMask() const override;
+ uint32_t channelCount() const override;
+ size_t frameCount() const override;
+ uint32_t latency() const override;
+
+ status_t addEffectToHal(sp<EffectHalInterface> effect) override;
+ status_t removeEffectFromHal(sp<EffectHalInterface> effect) override;
+ bool disconnectEffectHandle(EffectHandle *handle, bool unpinIfLast) override;
+ void setVolumeForOutput(float left, float right) const override;
+
+ // check if effects should be suspended/restored when a given effect is enable/disabled
+ void checkSuspendOnEffectEnabled(const sp<EffectBase>& effect,
+ bool enabled, bool threadLocked) override;
+ void resetVolume() override;
+ uint32_t strategy() const override;
+ int32_t activeTrackCnt() const override;
+ void onEffectEnable(const sp<EffectBase>& effect) override;
+ void onEffectDisable(const sp<EffectBase>& effect) override;
+
+ wp<EffectChain> chain() const override { return mChain; }
+
+ wp<ThreadBase> thread() { return mThread; }
+ void setThread(ThreadBase *thread) { mThread = thread; };
+
+ private:
+ wp<EffectChain> mChain;
+ wp<ThreadBase> mThread;
+ wp<AudioFlinger> mAudioFlinger;
+ };
+
friend class AudioFlinger; // for mThread, mEffects
DISALLOW_COPY_AND_ASSIGN(EffectChain);
@@ -441,13 +583,12 @@
static bool isEffectEligibleForBtNrecSuspend(const effect_uuid_t *type);
- void clearInputBuffer_l(const sp<ThreadBase>& thread);
+ void clearInputBuffer_l();
void setThread(const sp<ThreadBase>& thread);
void setVolumeForOutput_l(uint32_t left, uint32_t right);
- wp<ThreadBase> mThread; // parent mixer thread
mutable Mutex mLock; // mutex protecting effect list
Vector< sp<EffectModule> > mEffects; // list of effect modules
audio_session_t mSessionId; // audio session ID
@@ -471,4 +612,99 @@
// timeLow fields among effect type UUIDs.
// Updated by setEffectSuspended_l() and setEffectSuspendedAll_l() only.
KeyedVector< int, sp<SuspendedEffectDesc> > mSuspendedEffects;
+
+ const sp<EffectCallback> mEffectCallback;
+};
+
+class DeviceEffectProxy : public EffectBase {
+public:
+ DeviceEffectProxy (const AudioDeviceTypeAddr& device,
+ const sp<DeviceEffectManagerCallback>& callback,
+ effect_descriptor_t *desc, int id)
+ : EffectBase(callback, desc, id, AUDIO_SESSION_DEVICE, false),
+ mDevice(device), mManagerCallback(callback),
+ mMyCallback(new ProxyCallback(this, callback)) {}
+
+ status_t setEnabled(bool enabled, bool fromHandle) override;
+ sp<DeviceEffectProxy> asDeviceEffectProxy() override { return this; }
+
+ status_t init(const std::map<audio_patch_handle_t, PatchPanel::Patch>& patches);
+ status_t onCreatePatch(audio_patch_handle_t patchHandle, const PatchPanel::Patch& patch);
+ void onReleasePatch(audio_patch_handle_t patchHandle);
+
+ size_t removeEffect(const sp<EffectModule>& effect);
+
+ status_t addEffectToHal(sp<EffectHalInterface> effect);
+ status_t removeEffectFromHal(sp<EffectHalInterface> effect);
+
+ const AudioDeviceTypeAddr& device() { return mDevice; };
+ bool isOutput() const;
+ uint32_t sampleRate() const;
+ audio_channel_mask_t channelMask() const;
+ uint32_t channelCount() const;
+
+ void dump(int fd, int spaces);
+
+private:
+
+ class ProxyCallback : public EffectCallbackInterface {
+ public:
+ ProxyCallback(DeviceEffectProxy *proxy,
+ const sp<DeviceEffectManagerCallback>& callback)
+ : mProxy(proxy), mManagerCallback(callback) {}
+
+ status_t createEffectHal(const effect_uuid_t *pEffectUuid,
+ int32_t sessionId, int32_t deviceId, sp<EffectHalInterface> *effect) override;
+ status_t allocateHalBuffer(size_t size __unused,
+ sp<EffectBufferHalInterface>* buffer __unused) override { return NO_ERROR; }
+ bool updateOrphanEffectChains(const sp<EffectBase>& effect __unused) override {
+ return false;
+ }
+
+ audio_io_handle_t io() const override { return AUDIO_IO_HANDLE_NONE; }
+ bool isOutput() const override;
+ bool isOffload() const override { return false; }
+ bool isOffloadOrDirect() const override { return false; }
+ bool isOffloadOrMmap() const override { return false; }
+
+ uint32_t sampleRate() const override;
+ audio_channel_mask_t channelMask() const override;
+ uint32_t channelCount() const override;
+ size_t frameCount() const override { return 0; }
+ uint32_t latency() const override { return 0; }
+
+ status_t addEffectToHal(sp<EffectHalInterface> effect) override;
+ status_t removeEffectFromHal(sp<EffectHalInterface> effect) override;
+
+ bool disconnectEffectHandle(EffectHandle *handle, bool unpinIfLast) override;
+ void setVolumeForOutput(float left __unused, float right __unused) const override {}
+
+ void checkSuspendOnEffectEnabled(const sp<EffectBase>& effect __unused,
+ bool enabled __unused, bool threadLocked __unused) override {}
+ void resetVolume() override {}
+ uint32_t strategy() const override { return 0; }
+ int32_t activeTrackCnt() const override { return 0; }
+ void onEffectEnable(const sp<EffectBase>& effect __unused) override {}
+ void onEffectDisable(const sp<EffectBase>& effect __unused) override {}
+
+ wp<EffectChain> chain() const override { return nullptr; }
+
+ int newEffectId();
+
+ private:
+ const wp<DeviceEffectProxy> mProxy;
+ const sp<DeviceEffectManagerCallback> mManagerCallback;
+ };
+
+ status_t checkPort(const PatchPanel::Patch& patch, const struct audio_port_config *port,
+ sp<EffectHandle> *handle);
+
+ const AudioDeviceTypeAddr mDevice;
+ const sp<DeviceEffectManagerCallback> mManagerCallback;
+ const sp<ProxyCallback> mMyCallback;
+
+ Mutex mProxyLock;
+ std::map<audio_patch_handle_t, sp<EffectHandle>> mEffectHandles; // protected by mProxyLock
+ sp<EffectModule> mHalEffect; // protected by mProxyLock
+ struct audio_port_config mDevicePort = { .id = AUDIO_PORT_HANDLE_NONE };
};
diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp
index c5b9953..3eacc8c 100644
--- a/services/audioflinger/FastMixer.cpp
+++ b/services/audioflinger/FastMixer.cpp
@@ -105,13 +105,8 @@
return mSQ.poll();
}
-void FastMixer::setNBLogWriter(NBLog::Writer *logWriter)
+void FastMixer::setNBLogWriter(NBLog::Writer *logWriter __unused)
{
- // FIXME If mMixer is set or changed prior to this, we don't inform correctly.
- // Should cache logWriter and re-apply it at the assignment to mMixer.
- if (mMixer != NULL) {
- mMixer->setNBLogWriter(logWriter);
- }
}
void FastMixer::onIdle()
diff --git a/services/audioflinger/FastThread.cpp b/services/audioflinger/FastThread.cpp
index 04b32c2..8b7a124 100644
--- a/services/audioflinger/FastThread.cpp
+++ b/services/audioflinger/FastThread.cpp
@@ -124,7 +124,7 @@
mDumpState = next->mDumpState != NULL ? next->mDumpState : mDummyDumpState;
tlNBLogWriter = next->mNBLogWriter != NULL ?
next->mNBLogWriter : mDummyNBLogWriter.get();
- setNBLogWriter(tlNBLogWriter); // FastMixer informs its AudioMixer, FastCapture ignores
+ setNBLogWriter(tlNBLogWriter); // This is used for debugging only
// We want to always have a valid reference to the previous (non-idle) state.
// However, the state queue only guarantees access to current and previous states.
diff --git a/services/audioflinger/PatchPanel.cpp b/services/audioflinger/PatchPanel.cpp
index edb331d..786c279 100644
--- a/services/audioflinger/PatchPanel.cpp
+++ b/services/audioflinger/PatchPanel.cpp
@@ -25,6 +25,7 @@
#include "AudioFlinger.h"
#include <media/AudioParameter.h>
+#include <media/DeviceDescriptorBase.h>
#include <media/PatchBuilder.h>
#include <mediautils/ServiceUtilities.h>
@@ -168,8 +169,7 @@
hwDevice->releaseAudioPatch(removedPatch.mHalHandle);
}
}
- mPatches.erase(iter);
- removeSoftwarePatchFromInsertedModules(*handle);
+ erasePatch(*handle);
}
}
@@ -324,10 +324,14 @@
}
}
status = thread->sendCreateAudioPatchConfigEvent(patch, &halHandle);
+ if (status == NO_ERROR) {
+ newPatch.setThread(thread);
+ }
+
// remove stale audio patch with same input as sink if any
for (auto& iter : mPatches) {
if (iter.second.mAudioPatch.sinks[0].ext.mix.handle == thread->id()) {
- mPatches.erase(iter.first);
+ erasePatch(iter.first);
break;
}
}
@@ -351,7 +355,7 @@
goto exit;
}
// limit to connections between devices and output streams
- audio_devices_t type = AUDIO_DEVICE_NONE;
+ DeviceDescriptorBaseVector devices;
for (unsigned int i = 0; i < patch->num_sinks; i++) {
if (patch->sinks[i].type != AUDIO_PORT_TYPE_DEVICE) {
ALOGW("%s() invalid sink type %d for mix source",
@@ -364,7 +368,11 @@
status = BAD_VALUE;
goto exit;
}
- type |= patch->sinks[i].ext.device.type;
+ sp<DeviceDescriptorBase> device = new DeviceDescriptorBase(
+ patch->sinks[i].ext.device.type);
+ device->setAddress(patch->sinks[i].ext.device.address);
+ device->applyAudioPortConfig(&patch->sinks[i]);
+ devices.push_back(device);
}
sp<ThreadBase> thread =
mAudioFlinger.checkPlaybackThread_l(patch->sources[0].ext.mix.handle);
@@ -378,18 +386,18 @@
}
}
if (thread == mAudioFlinger.primaryPlaybackThread_l()) {
- AudioParameter param = AudioParameter();
- param.addInt(String8(AudioParameter::keyRouting), (int)type);
-
- mAudioFlinger.broacastParametersToRecordThreads_l(param.toString());
+ mAudioFlinger.updateOutDevicesForRecordThreads_l(devices);
}
status = thread->sendCreateAudioPatchConfigEvent(patch, &halHandle);
+ if (status == NO_ERROR) {
+ newPatch.setThread(thread);
+ }
// remove stale audio patch with same output as source if any
for (auto& iter : mPatches) {
if (iter.second.mAudioPatch.sources[0].ext.mix.handle == thread->id()) {
- mPatches.erase(iter.first);
+ erasePatch(iter.first);
break;
}
}
@@ -403,11 +411,11 @@
if (status == NO_ERROR) {
*handle = (audio_patch_handle_t) mAudioFlinger.nextUniqueId(AUDIO_UNIQUE_ID_USE_PATCH);
newPatch.mHalHandle = halHandle;
+ mAudioFlinger.mDeviceEffectManager.createAudioPatch(*handle, newPatch);
mPatches.insert(std::make_pair(*handle, std::move(newPatch)));
if (insertedModule != AUDIO_MODULE_HANDLE_NONE) {
addSoftwarePatchToInsertedModules(insertedModule, *handle);
}
- ALOGV("%s() added new patch handle %d halHandle %d", __func__, *handle, halHandle);
} else {
newPatch.clearConnections(this);
}
@@ -445,18 +453,6 @@
*mPlayback.handlePtr() = AUDIO_PATCH_HANDLE_NONE;
}
- // use a pseudo LCM between input and output framecount
- size_t playbackFrameCount = mPlayback.thread()->frameCount();
- int playbackShift = __builtin_ctz(playbackFrameCount);
- size_t recordFrameCount = mRecord.thread()->frameCount();
- int shift = __builtin_ctz(recordFrameCount);
- if (playbackShift < shift) {
- shift = playbackShift;
- }
- size_t frameCount = (playbackFrameCount * recordFrameCount) >> shift;
- ALOGV("%s() playframeCount %zu recordFrameCount %zu frameCount %zu",
- __func__, playbackFrameCount, recordFrameCount, frameCount);
-
// create a special record track to capture from record thread
uint32_t channelCount = mPlayback.thread()->channelCount();
audio_channel_mask_t inChannelMask = audio_channel_in_mask_from_count(channelCount);
@@ -483,19 +479,6 @@
// Fast mode is not available in this case.
inputFlags = (audio_input_flags_t) (inputFlags & ~AUDIO_INPUT_FLAG_FAST);
}
- sp<RecordThread::PatchRecord> tempRecordTrack = new (std::nothrow) RecordThread::PatchRecord(
- mRecord.thread().get(),
- sampleRate,
- inChannelMask,
- format,
- frameCount,
- NULL,
- (size_t)0 /* bufferSize */,
- inputFlags);
- status = mRecord.checkTrack(tempRecordTrack.get());
- if (status != NO_ERROR) {
- return status;
- }
audio_output_flags_t outputFlags = mAudioPatch.sinks[0].config_mask & AUDIO_PORT_CONFIG_FLAGS ?
mAudioPatch.sinks[0].flags.output : AUDIO_OUTPUT_FLAG_NONE;
@@ -512,9 +495,54 @@
outputFlags = (audio_output_flags_t) (outputFlags & ~AUDIO_OUTPUT_FLAG_FAST);
}
+ sp<RecordThread::PatchRecord> tempRecordTrack;
+ const bool usePassthruPatchRecord =
+ (inputFlags & AUDIO_INPUT_FLAG_DIRECT) && (outputFlags & AUDIO_OUTPUT_FLAG_DIRECT);
+ const size_t playbackFrameCount = mPlayback.thread()->frameCount();
+ const size_t recordFrameCount = mRecord.thread()->frameCount();
+ size_t frameCount = 0;
+ if (usePassthruPatchRecord) {
+ // PassthruPatchRecord producesBufferOnDemand, so use
+ // maximum of playback and record thread framecounts
+ frameCount = std::max(playbackFrameCount, recordFrameCount);
+ ALOGV("%s() playframeCount %zu recordFrameCount %zu frameCount %zu",
+ __func__, playbackFrameCount, recordFrameCount, frameCount);
+ tempRecordTrack = new RecordThread::PassthruPatchRecord(
+ mRecord.thread().get(),
+ sampleRate,
+ inChannelMask,
+ format,
+ frameCount,
+ inputFlags);
+ } else {
+ // use a pseudo LCM between input and output framecount
+ int playbackShift = __builtin_ctz(playbackFrameCount);
+ int shift = __builtin_ctz(recordFrameCount);
+ if (playbackShift < shift) {
+ shift = playbackShift;
+ }
+ frameCount = (playbackFrameCount * recordFrameCount) >> shift;
+ ALOGV("%s() playframeCount %zu recordFrameCount %zu frameCount %zu",
+ __func__, playbackFrameCount, recordFrameCount, frameCount);
+
+ tempRecordTrack = new RecordThread::PatchRecord(
+ mRecord.thread().get(),
+ sampleRate,
+ inChannelMask,
+ format,
+ frameCount,
+ nullptr,
+ (size_t)0 /* bufferSize */,
+ inputFlags);
+ }
+ status = mRecord.checkTrack(tempRecordTrack.get());
+ if (status != NO_ERROR) {
+ return status;
+ }
+
// create a special playback track to render to playback thread.
// this track is given the same buffer as the PatchRecord buffer
- sp<PlaybackThread::PatchTrack> tempPatchTrack = new (std::nothrow) PlaybackThread::PatchTrack(
+ sp<PlaybackThread::PatchTrack> tempPatchTrack = new PlaybackThread::PatchTrack(
mPlayback.thread().get(),
streamType,
sampleRate,
@@ -530,8 +558,14 @@
}
// tie playback and record tracks together
- mRecord.setTrackAndPeer(tempRecordTrack, tempPatchTrack);
- mPlayback.setTrackAndPeer(tempPatchTrack, tempRecordTrack);
+ // In the case of PassthruPatchRecord no I/O activity happens on RecordThread,
+ // everything is driven from PlaybackThread. Thus AudioBufferProvider methods
+ // of PassthruPatchRecord can only be called if the corresponding PatchTrack
+ // is alive. There is no need to hold a reference, and there is no need
+ // to clear it. In fact, since playback stopping is asynchronous, there is
+ // no proper time when clearing could be done.
+ mRecord.setTrackAndPeer(tempRecordTrack, tempPatchTrack, !usePassthruPatchRecord);
+ mPlayback.setTrackAndPeer(tempPatchTrack, tempRecordTrack, true /*holdReference*/);
// start capture and playback
mRecord.track()->start(AudioSystem::SYNC_EVENT_NONE, AUDIO_SESSION_NONE);
@@ -613,8 +647,21 @@
String8 AudioFlinger::PatchPanel::Patch::dump(audio_patch_handle_t myHandle) const
{
// TODO: Consider table dump form for patches, just like tracks.
- String8 result = String8::format("Patch %d: thread %p => thread %p",
- myHandle, mRecord.const_thread().get(), mPlayback.const_thread().get());
+ String8 result = String8::format("Patch %d: %s (thread %p => thread %p)",
+ myHandle, isSoftware() ? "Software bridge between" : "No software bridge",
+ mRecord.const_thread().get(), mPlayback.const_thread().get());
+
+ bool hasSinkDevice =
+ mAudioPatch.num_sinks > 0 && mAudioPatch.sinks[0].type == AUDIO_PORT_TYPE_DEVICE;
+ bool hasSourceDevice =
+ mAudioPatch.num_sources > 0 && mAudioPatch.sources[0].type == AUDIO_PORT_TYPE_DEVICE;
+ result.appendFormat(" thread %p %s (%d) first device type %08x", mThread.unsafe_get(),
+ hasSinkDevice ? "num sinks" :
+ (hasSourceDevice ? "num sources" : "no devices"),
+ hasSinkDevice ? mAudioPatch.num_sinks :
+ (hasSourceDevice ? mAudioPatch.num_sources : 0),
+ hasSinkDevice ? mAudioPatch.sinks[0].ext.device.type :
+ (hasSourceDevice ? mAudioPatch.sources[0].ext.device.type : 0));
// add latency if it exists
double latencyMs;
@@ -690,11 +737,16 @@
status = BAD_VALUE;
}
- mPatches.erase(iter);
- removeSoftwarePatchFromInsertedModules(handle);
+ erasePatch(handle);
return status;
}
+void AudioFlinger::PatchPanel::erasePatch(audio_patch_handle_t handle) {
+ mPatches.erase(handle);
+ removeSoftwarePatchFromInsertedModules(handle);
+ mAudioFlinger.mDeviceEffectManager.releaseAudioPatch(handle);
+}
+
/* List connected audio ports and they attributes */
status_t AudioFlinger::PatchPanel::listAudioPatches(unsigned int *num_patches __unused,
struct audio_patch *patches __unused)
@@ -778,16 +830,13 @@
String8 patchPanelDump;
const char *indent = " ";
- // Only dump software patches.
bool headerPrinted = false;
for (const auto& iter : mPatches) {
- if (iter.second.isSoftware()) {
- if (!headerPrinted) {
- patchPanelDump += "\nSoftware patches:\n";
- headerPrinted = true;
- }
- patchPanelDump.appendFormat("%s%s\n", indent, iter.second.dump(iter.first).string());
+ if (!headerPrinted) {
+ patchPanelDump += "\nPatches:\n";
+ headerPrinted = true;
}
+ patchPanelDump.appendFormat("%s%s\n", indent, iter.second.dump(iter.first).string());
}
headerPrinted = false;
diff --git a/services/audioflinger/PatchPanel.h b/services/audioflinger/PatchPanel.h
index 181e27c..89d4eb1 100644
--- a/services/audioflinger/PatchPanel.h
+++ b/services/audioflinger/PatchPanel.h
@@ -76,13 +76,18 @@
void dump(int fd) const;
-private:
template<typename ThreadType, typename TrackType>
- class Endpoint {
+ class Endpoint final {
public:
Endpoint() = default;
Endpoint(const Endpoint&) = delete;
- Endpoint& operator=(const Endpoint&) = delete;
+ Endpoint& operator=(const Endpoint& other) noexcept {
+ mThread = other.mThread;
+ mCloseThread = other.mCloseThread;
+ mHandle = other.mHandle;
+ mTrack = other.mTrack;
+ return *this;
+ }
Endpoint(Endpoint&& other) noexcept { swap(other); }
Endpoint& operator=(Endpoint&& other) noexcept {
swap(other);
@@ -98,8 +103,8 @@
return trackOrNull->initCheck();
}
audio_patch_handle_t handle() const { return mHandle; }
- sp<ThreadType> thread() { return mThread; }
- sp<TrackType> track() { return mTrack; }
+ sp<ThreadType> thread() const { return mThread; }
+ sp<TrackType> track() const { return mTrack; }
sp<const ThreadType> const_thread() const { return mThread; }
sp<const TrackType> const_track() const { return mTrack; }
@@ -123,18 +128,20 @@
mCloseThread = closeThread;
}
template <typename T>
- void setTrackAndPeer(const sp<TrackType>& track, const sp<T> &peer) {
+ void setTrackAndPeer(const sp<TrackType>& track, const sp<T> &peer, bool holdReference) {
mTrack = track;
mThread->addPatchTrack(mTrack);
- mTrack->setPeerProxy(peer, true /* holdReference */);
+ mTrack->setPeerProxy(peer, holdReference);
+ mClearPeerProxy = holdReference;
}
- void clearTrackPeer() { if (mTrack) mTrack->clearPeerProxy(); }
+ void clearTrackPeer() { if (mClearPeerProxy && mTrack) mTrack->clearPeerProxy(); }
void stopTrack() { if (mTrack) mTrack->stop(); }
void swap(Endpoint &other) noexcept {
using std::swap;
swap(mThread, other.mThread);
swap(mCloseThread, other.mCloseThread);
+ swap(mClearPeerProxy, other.mClearPeerProxy);
swap(mHandle, other.mHandle);
swap(mTrack, other.mTrack);
}
@@ -146,18 +153,41 @@
private:
sp<ThreadType> mThread;
bool mCloseThread = true;
+ bool mClearPeerProxy = true;
audio_patch_handle_t mHandle = AUDIO_PATCH_HANDLE_NONE;
sp<TrackType> mTrack;
};
- class Patch {
+ class Patch final {
public:
explicit Patch(const struct audio_patch &patch) : mAudioPatch(patch) {}
+ Patch() = default;
~Patch();
- Patch(const Patch&) = delete;
- Patch(Patch&&) = default;
- Patch& operator=(const Patch&) = delete;
- Patch& operator=(Patch&&) = default;
+ Patch(const Patch& other) noexcept {
+ mAudioPatch = other.mAudioPatch;
+ mHalHandle = other.mHalHandle;
+ mPlayback = other.mPlayback;
+ mRecord = other.mRecord;
+ mThread = other.mThread;
+ }
+ Patch(Patch&& other) noexcept { swap(other); }
+ Patch& operator=(Patch&& other) noexcept {
+ swap(other);
+ return *this;
+ }
+
+ void swap(Patch &other) noexcept {
+ using std::swap;
+ swap(mAudioPatch, other.mAudioPatch);
+ swap(mHalHandle, other.mHalHandle);
+ swap(mPlayback, other.mPlayback);
+ swap(mRecord, other.mRecord);
+ swap(mThread, other.mThread);
+ }
+
+ friend void swap(Patch &a, Patch &b) noexcept {
+ a.swap(b);
+ }
status_t createConnections(PatchPanel *panel);
void clearConnections(PatchPanel *panel);
@@ -165,6 +195,9 @@
return mRecord.handle() != AUDIO_PATCH_HANDLE_NONE ||
mPlayback.handle() != AUDIO_PATCH_HANDLE_NONE; }
+ void setThread(sp<ThreadBase> thread) { mThread = thread; }
+ wp<ThreadBase> thread() const { return mThread; }
+
// returns the latency of the patch (from record to playback).
status_t getLatencyMs(double *latencyMs) const;
@@ -182,13 +215,20 @@
Endpoint<PlaybackThread, PlaybackThread::PatchTrack> mPlayback;
// connects source device to record thread input
Endpoint<RecordThread, RecordThread::PatchRecord> mRecord;
+
+ wp<ThreadBase> mThread;
};
+ // Call with AudioFlinger mLock held
+ std::map<audio_patch_handle_t, Patch>& patches_l() { return mPatches; }
+
+private:
AudioHwDevice* findAudioHwDeviceByModule(audio_module_handle_t module);
sp<DeviceHalInterface> findHwDeviceByModule(audio_module_handle_t module);
void addSoftwarePatchToInsertedModules(
audio_module_handle_t module, audio_patch_handle_t handle);
void removeSoftwarePatchFromInsertedModules(audio_patch_handle_t handle);
+ void erasePatch(audio_patch_handle_t handle);
AudioFlinger &mAudioFlinger;
std::map<audio_patch_handle_t, Patch> mPatches;
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index 17adba5..1ff03c4 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -271,8 +271,6 @@
private:
void interceptBuffer(const AudioBufferProvider::Buffer& buffer);
- /** Write the source data in the buffer provider. @return written frame count. */
- size_t writeFrames(AudioBufferProvider* dest, const void* src, size_t frameCount);
template <class F>
void forEachTeePatchTrack(F f) {
for (auto& tp : mTeePatches) { f(tp.patchTrack); }
@@ -396,6 +394,8 @@
* even if it might glitch. */);
virtual ~PatchTrack();
+ size_t framesReady() const override;
+
virtual status_t start(AudioSystem::sync_event_t event =
AudioSystem::SYNC_EVENT_NONE,
audio_session_t triggerSession = AUDIO_SESSION_NONE);
diff --git a/services/audioflinger/RecordTracks.h b/services/audioflinger/RecordTracks.h
index c8397cd..d5257bd 100644
--- a/services/audioflinger/RecordTracks.h
+++ b/services/audioflinger/RecordTracks.h
@@ -167,6 +167,8 @@
const Timeout& timeout = {});
virtual ~PatchRecord();
+ virtual Source* getSource() { return nullptr; }
+
// AudioBufferProvider interface
virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
@@ -175,4 +177,71 @@
virtual status_t obtainBuffer(Proxy::Buffer *buffer,
const struct timespec *timeOut = NULL);
virtual void releaseBuffer(Proxy::Buffer *buffer);
+
+ size_t writeFrames(const void* src, size_t frameCount, size_t frameSize) {
+ return writeFrames(this, src, frameCount, frameSize);
+ }
+
+protected:
+ /** Write the source data into the buffer provider. @return written frame count. */
+ static size_t writeFrames(AudioBufferProvider* dest, const void* src,
+ size_t frameCount, size_t frameSize);
+
}; // end of PatchRecord
+
+class PassthruPatchRecord : public PatchRecord, public Source {
+public:
+ PassthruPatchRecord(RecordThread *recordThread,
+ uint32_t sampleRate,
+ audio_channel_mask_t channelMask,
+ audio_format_t format,
+ size_t frameCount,
+ audio_input_flags_t flags);
+
+ Source* getSource() override { return static_cast<Source*>(this); }
+
+ // Source interface
+ status_t read(void *buffer, size_t bytes, size_t *read) override;
+ status_t getCapturePosition(int64_t *frames, int64_t *time) override;
+ status_t standby() override;
+
+ // AudioBufferProvider interface
+ // This interface is used by RecordThread to pass the data obtained
+ // from HAL or other source to the client. PassthruPatchRecord receives
+ // the data in 'obtainBuffer' so these calls are stubbed out.
+ status_t getNextBuffer(AudioBufferProvider::Buffer* buffer) override;
+ void releaseBuffer(AudioBufferProvider::Buffer* buffer) override;
+
+ // PatchProxyBufferProvider interface
+ // This interface is used from DirectOutputThread to acquire data from HAL.
+ bool producesBufferOnDemand() const override { return true; }
+ status_t obtainBuffer(Proxy::Buffer *buffer, const struct timespec *timeOut = nullptr) override;
+ void releaseBuffer(Proxy::Buffer *buffer) override;
+
+private:
+ // This is to use with PatchRecord::writeFrames
+ struct PatchRecordAudioBufferProvider : public AudioBufferProvider {
+ explicit PatchRecordAudioBufferProvider(PassthruPatchRecord& passthru) :
+ mPassthru(passthru) {}
+ status_t getNextBuffer(AudioBufferProvider::Buffer* buffer) override {
+ return mPassthru.PatchRecord::getNextBuffer(buffer);
+ }
+ void releaseBuffer(AudioBufferProvider::Buffer* buffer) override {
+ return mPassthru.PatchRecord::releaseBuffer(buffer);
+ }
+ private:
+ PassthruPatchRecord& mPassthru;
+ };
+
+ sp<StreamInHalInterface> obtainStream(sp<ThreadBase>* thread);
+
+ PatchRecordAudioBufferProvider mPatchRecordAudioBufferProvider;
+ std::unique_ptr<void, decltype(free)*> mSinkBuffer; // frame size aligned continuous buffer
+ std::unique_ptr<void, decltype(free)*> mStubBuffer; // buffer used for AudioBufferProvider
+ size_t mUnconsumedFrames = 0;
+ std::mutex mReadLock;
+ std::condition_variable mReadCV;
+ size_t mReadBytes = 0; // GUARDED_BY(mReadLock)
+ status_t mReadError = NO_ERROR; // GUARDED_BY(mReadLock)
+ int64_t mLastReadFrames = 0; // accessed on RecordThread only
+};
diff --git a/services/audioflinger/SpdifStreamOut.cpp b/services/audioflinger/SpdifStreamOut.cpp
index a44ab2a..c7aba79 100644
--- a/services/audioflinger/SpdifStreamOut.cpp
+++ b/services/audioflinger/SpdifStreamOut.cpp
@@ -59,6 +59,7 @@
// TODO Move this into the audio_utils as a static method.
switch(config->format) {
case AUDIO_FORMAT_E_AC3:
+ case AUDIO_FORMAT_E_AC3_JOC:
mRateMultiplier = 4;
break;
case AUDIO_FORMAT_AC3:
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index bd9bf7b..59d0ad9 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -29,6 +29,8 @@
#include <sys/stat.h>
#include <sys/syscall.h>
#include <cutils/properties.h>
+#include <media/AudioContainers.h>
+#include <media/AudioDeviceTypeAddr.h>
#include <media/AudioParameter.h>
#include <media/AudioResamplerPublic.h>
#include <media/RecordBufferConverter.h>
@@ -460,7 +462,7 @@
}
AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
- audio_devices_t outDevice, audio_devices_t inDevice, type_t type, bool systemReady)
+ type_t type, bool systemReady)
: Thread(false /*canCallJava*/),
mType(type),
mAudioFlinger(audioFlinger),
@@ -468,8 +470,7 @@
// are set by PlaybackThread::readOutputParameters_l() or
// RecordThread::readInputParameters_l()
//FIXME: mStandby should be true here. Is this some kind of hack?
- mStandby(false), mOutDevice(outDevice), mInDevice(inDevice),
- mPrevOutDevice(AUDIO_DEVICE_NONE), mPrevInDevice(AUDIO_DEVICE_NONE),
+ mStandby(false),
mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id),
// mName will be set by concrete (non-virtual) subclass
mDeathRecipient(new PMDeathRecipient(this)),
@@ -646,6 +647,18 @@
return sendConfigEvent_l(configEvent);
}
+status_t AudioFlinger::ThreadBase::sendUpdateOutDeviceConfigEvent(
+ const DeviceDescriptorBaseVector& outDevices)
+{
+ if (type() != RECORD) {
+ // The update out device operation is only for record thread.
+ return INVALID_OPERATION;
+ }
+ Mutex::Autolock _l(mLock);
+ sp<ConfigEvent> configEvent = (ConfigEvent *)new UpdateOutDevicesConfigEvent(outDevices);
+ return sendConfigEvent_l(configEvent);
+}
+
// post condition: mConfigEvents.isEmpty()
void AudioFlinger::ThreadBase::processConfigEvents_l()
@@ -680,24 +693,29 @@
}
} break;
case CFG_EVENT_CREATE_AUDIO_PATCH: {
- const audio_devices_t oldDevice = getDevice();
+ const DeviceTypeSet oldDevices = getDeviceTypes();
CreateAudioPatchConfigEventData *data =
(CreateAudioPatchConfigEventData *)event->mData.get();
event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle);
- const audio_devices_t newDevice = getDevice();
- mLocalLog.log("CFG_EVENT_CREATE_AUDIO_PATCH: old device %#x (%s) new device %#x (%s)",
- (unsigned)oldDevice, toString(oldDevice).c_str(),
- (unsigned)newDevice, toString(newDevice).c_str());
+ const DeviceTypeSet newDevices = getDeviceTypes();
+ mLocalLog.log("CFG_EVENT_CREATE_AUDIO_PATCH: old device %s (%s) new device %s (%s)",
+ dumpDeviceTypes(oldDevices).c_str(), toString(oldDevices).c_str(),
+ dumpDeviceTypes(newDevices).c_str(), toString(newDevices).c_str());
} break;
case CFG_EVENT_RELEASE_AUDIO_PATCH: {
- const audio_devices_t oldDevice = getDevice();
+ const DeviceTypeSet oldDevices = getDeviceTypes();
ReleaseAudioPatchConfigEventData *data =
(ReleaseAudioPatchConfigEventData *)event->mData.get();
event->mStatus = releaseAudioPatch_l(data->mHandle);
- const audio_devices_t newDevice = getDevice();
- mLocalLog.log("CFG_EVENT_RELEASE_AUDIO_PATCH: old device %#x (%s) new device %#x (%s)",
- (unsigned)oldDevice, toString(oldDevice).c_str(),
- (unsigned)newDevice, toString(newDevice).c_str());
+ const DeviceTypeSet newDevices = getDeviceTypes();
+ mLocalLog.log("CFG_EVENT_RELEASE_AUDIO_PATCH: old device %s (%s) new device %s (%s)",
+ dumpDeviceTypes(oldDevices).c_str(), toString(oldDevices).c_str(),
+ dumpDeviceTypes(newDevices).c_str(), toString(newDevices).c_str());
+ } break;
+ case CFG_EVENT_UPDATE_OUT_DEVICE: {
+ UpdateOutDevicesConfigEventData *data =
+ (UpdateOutDevicesConfigEventData *)event->mData.get();
+ updateOutDevices(data->mOutDevices);
} break;
default:
ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType);
@@ -840,8 +858,10 @@
dprintf(fd, " none\n");
}
// Note: output device may be used by capture threads for effects such as AEC.
- dprintf(fd, " Output device: %#x (%s)\n", mOutDevice, toString(mOutDevice).c_str());
- dprintf(fd, " Input device: %#x (%s)\n", mInDevice, toString(mInDevice).c_str());
+ dprintf(fd, " Output devices: %s (%s)\n",
+ dumpDeviceTypes(outDeviceTypes()).c_str(), toString(outDeviceTypes()).c_str());
+ dprintf(fd, " Input device: %#x (%s)\n",
+ inDeviceType(), toString(inDeviceType()).c_str());
dprintf(fd, " Audio source: %d (%s)\n", mAudioSource, toString(mAudioSource).c_str());
// Dump timestamp statistics for the Thread types that support it.
@@ -1011,6 +1031,12 @@
mPowerManager.clear();
}
+void AudioFlinger::ThreadBase::updateOutDevices(
+ const DeviceDescriptorBaseVector& outDevices __unused)
+{
+ ALOGE("%s should only be called in RecordThread", __func__);
+}
+
void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused)
{
sp<ThreadBase> thread = mThread.promote();
@@ -1120,32 +1146,26 @@
}
}
-void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
- bool enabled,
- audio_session_t sessionId)
-{
- Mutex::Autolock _l(mLock);
- checkSuspendOnEffectEnabled_l(effect, enabled, sessionId);
-}
+void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(bool enabled,
+ audio_session_t sessionId,
+ bool threadLocked) {
+ if (!threadLocked) {
+ mLock.lock();
+ }
-void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect,
- bool enabled,
- audio_session_t sessionId)
-{
if (mType != RECORD) {
// suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on
// another session. This gives the priority to well behaved effect control panels
// and applications not using global effects.
// Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect
// global effects
- if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) {
+ if (!audio_is_global_session(sessionId)) {
setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX);
}
}
- sp<EffectChain> chain = getEffectChain_l(sessionId);
- if (chain != 0) {
- chain->checkSuspendOnEffectEnabled(effect, enabled);
+ if (!threadLocked) {
+ mLock.unlock();
}
}
@@ -1153,8 +1173,9 @@
status_t AudioFlinger::RecordThread::checkEffectCompatibility_l(
const effect_descriptor_t *desc, audio_session_t sessionId)
{
- // No global effect sessions on record threads
- if (sessionId == AUDIO_SESSION_OUTPUT_MIX || sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
+ // No global output effect sessions on record threads
+ if (sessionId == AUDIO_SESSION_OUTPUT_MIX
+ || sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
ALOGW("checkEffectCompatibility_l(): global effect %s on record thread %s",
desc->name, mThreadName);
return BAD_VALUE;
@@ -1228,6 +1249,13 @@
" on output stage session", desc->name);
return BAD_VALUE;
}
+ } else if (sessionId == AUDIO_SESSION_DEVICE) {
+ // only post processing on output stage session
+ if ((desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_POST_PROC) {
+ ALOGW("checkEffectCompatibility_l(): non post processing effect %s not allowed"
+ " on device session", desc->name);
+ return BAD_VALUE;
+ }
} else {
// no restriction on effects applied on non fast tracks
if ((hasAudioSession_l(sessionId) & ThreadBase::FAST_SESSION) == 0) {
@@ -1269,7 +1297,7 @@
return BAD_VALUE;
}
#endif
- if ((sessionId == AUDIO_SESSION_OUTPUT_STAGE) || (sessionId == AUDIO_SESSION_OUTPUT_MIX)) {
+ if (audio_is_global_session(sessionId)) {
ALOGW("checkEffectCompatibility_l(): global effect %s on DUPLICATING"
" thread %s", desc->name, mThreadName);
return BAD_VALUE;
@@ -1345,14 +1373,15 @@
if (effect == 0) {
effectId = mAudioFlinger->nextUniqueId(AUDIO_UNIQUE_ID_USE_EFFECT);
// create a new effect module if none present in the chain
- lStatus = chain->createEffect_l(effect, this, desc, effectId, sessionId, pinned);
+ lStatus = chain->createEffect_l(effect, desc, effectId, sessionId, pinned);
if (lStatus != NO_ERROR) {
goto Exit;
}
effectCreated = true;
- effect->setDevice(mOutDevice);
- effect->setDevice(mInDevice);
+ // FIXME: use vector of device and address when effect interface is ready.
+ effect->setDevices(outDeviceTypeAddrs());
+ effect->setInputDevice(inDeviceTypeAddr());
effect->setMode(mAudioFlinger->getMode());
effect->setAudioSource(mAudioSource);
}
@@ -1390,9 +1419,12 @@
sp<EffectModule> effect;
{
Mutex::Autolock _l(mLock);
-
- effect = handle->effect().promote();
- if (effect == 0) {
+ sp<EffectBase> effectBase = handle->effect().promote();
+ if (effectBase == nullptr) {
+ return;
+ }
+ effect = effectBase->asEffectModule();
+ if (effect == nullptr) {
return;
}
// restore suspended effects if the disconnected handle was enabled and the last one.
@@ -1404,11 +1436,34 @@
if (remove) {
mAudioFlinger->updateOrphanEffectChains(effect);
if (handle->enabled()) {
- checkSuspendOnEffectEnabled(effect, false, effect->sessionId());
+ effect->checkSuspendOnEffectEnabled(false, false /*threadLocked*/);
}
}
}
+void AudioFlinger::ThreadBase::onEffectEnable(const sp<EffectModule>& effect) {
+ if (mType == OFFLOAD || mType == MMAP) {
+ Mutex::Autolock _l(mLock);
+ broadcast_l();
+ }
+ if (!effect->isOffloadable()) {
+ if (mType == ThreadBase::OFFLOAD) {
+ PlaybackThread *t = (PlaybackThread *)this;
+ t->invalidateTracks(AUDIO_STREAM_MUSIC);
+ }
+ if (effect->sessionId() == AUDIO_SESSION_OUTPUT_MIX) {
+ mAudioFlinger->onNonOffloadableGlobalEffectEnable();
+ }
+ }
+}
+
+void AudioFlinger::ThreadBase::onEffectDisable() {
+ if (mType == OFFLOAD || mType == MMAP) {
+ Mutex::Autolock _l(mLock);
+ broadcast_l();
+ }
+}
+
sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(audio_session_t sessionId,
int effectId)
{
@@ -1468,8 +1523,8 @@
return status;
}
- effect->setDevice(mOutDevice);
- effect->setDevice(mInDevice);
+ effect->setDevices(outDeviceTypeAddrs());
+ effect->setInputDevice(inDeviceTypeAddr());
effect->setMode(mAudioFlinger->getMode());
effect->setAudioSource(mAudioSource);
@@ -1484,7 +1539,7 @@
detachAuxEffect_l(effect->id());
}
- sp<EffectChain> chain = effect->chain().promote();
+ sp<EffectChain> chain = effect->callback()->chain().promote();
if (chain != 0) {
// remove effect chain if removing last effect
if (chain->removeEffect_l(effect, release) == 0) {
@@ -1702,8 +1757,8 @@
item->setInt64(MM_PREFIX "channelMask", (int64_t)mChannelMask);
item->setCString(MM_PREFIX "encoding", toString(mFormat).c_str());
item->setInt32(MM_PREFIX "frameCount", (int32_t)mFrameCount);
- item->setCString(MM_PREFIX "outDevice", toString(mOutDevice).c_str());
- item->setCString(MM_PREFIX "inDevice", toString(mInDevice).c_str());
+ item->setCString(MM_PREFIX "outDevice", toString(outDeviceTypes()).c_str());
+ item->setCString(MM_PREFIX "inDevice", toString(inDeviceType()).c_str());
// thread statistics
if (mIoJitterMs.getN() > 0) {
@@ -1734,10 +1789,9 @@
AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger,
AudioStreamOut* output,
audio_io_handle_t id,
- audio_devices_t device,
type_t type,
bool systemReady)
- : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type, systemReady),
+ : ThreadBase(audioFlinger, id, type, systemReady),
mNormalFrameCount(0), mSinkBuffer(NULL),
mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision),
mMixerBuffer(NULL),
@@ -1799,9 +1853,10 @@
// TODO: We may also match on address as well as device type for
// AUDIO_DEVICE_OUT_BUS, AUDIO_DEVICE_OUT_ALL_A2DP, AUDIO_DEVICE_OUT_REMOTE_SUBMIX
- if (type == MIXER || type == DIRECT) {
- mTimestampCorrectedDevices = (audio_devices_t)property_get_int64(
- "audio.timestamp.corrected_output_devices",
+ if (type == MIXER || type == DIRECT || type == OFFLOAD) {
+ // TODO: This property should be ensure that only contains one single device type.
+ mTimestampCorrectedDevice = (audio_devices_t)property_get_int64(
+ "audio.timestamp.corrected_output_device",
(int64_t)(mIsMsdDevice ? AUDIO_DEVICE_OUT_BUS // turn on by default for MSD
: AUDIO_DEVICE_NONE));
}
@@ -2024,6 +2079,7 @@
{ // scope for mLock
Mutex::Autolock _l(mLock);
for (audio_session_t session : {
+ AUDIO_SESSION_DEVICE,
AUDIO_SESSION_OUTPUT_STAGE,
AUDIO_SESSION_OUTPUT_MIX,
sessionId,
@@ -2891,7 +2947,7 @@
{
if (!mMasterMute) {
char value[PROPERTY_VALUE_MAX];
- if (mOutDevice == AUDIO_DEVICE_OUT_REMOTE_SUBMIX) {
+ if (isSingleDeviceType(outDeviceTypes(), AUDIO_DEVICE_OUT_REMOTE_SUBMIX)) {
ALOGD("ro.audio.silent will be ignored for threads on AUDIO_DEVICE_OUT_REMOTE_SUBMIX");
return;
}
@@ -2953,9 +3009,11 @@
ALOG_ASSERT(mCallbackThread != 0);
mCallbackThread->setWriteBlocked(mWriteAckSequence);
}
+ ATRACE_BEGIN("write");
// FIXME We should have an implementation of timestamps for direct output threads.
// They are used e.g for multichannel PCM playback over HDMI.
bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining);
+ ATRACE_END();
if (mUseAsyncWrite &&
((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) {
@@ -3033,7 +3091,7 @@
// make sure standby delay is not too short when connected to an A2DP sink to avoid
// truncating audio when going to standby.
mStandbyDelayNs = AudioFlinger::mStandbyTimeInNsecs;
- if ((mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != 0) {
+ if (!Intersection(outDeviceTypes(), getAudioDeviceOutAllA2dpSet()).empty()) {
if (mStandbyDelayNs < kDefaultStandbyTimeInNsecs) {
mStandbyDelayNs = kDefaultStandbyTimeInNsecs;
}
@@ -3074,7 +3132,7 @@
halOutBuffer = halInBuffer;
effect_buffer_t *buffer = reinterpret_cast<effect_buffer_t*>(halInBuffer->externalData());
ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session);
- if (session > AUDIO_SESSION_OUTPUT_MIX) {
+ if (!audio_is_global_session(session)) {
// Only one effect chain can be present in direct output thread and it uses
// the sink buffer as input
if (mType != DIRECT) {
@@ -3114,8 +3172,11 @@
chain->setThread(this);
chain->setInBuffer(halInBuffer);
chain->setOutBuffer(halOutBuffer);
- // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect
- // chains list in order to be processed last as it contains output stage effects.
+ // Effect chain for session AUDIO_SESSION_DEVICE is inserted at end of effect
+ // chains list in order to be processed last as it contains output device effects.
+ // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted just before to apply post
+ // processing effects specific to an output stream before effects applied to all streams
+ // routed to a given device.
// Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before
// session AUDIO_SESSION_OUTPUT_STAGE to be processed
// after track specific effects and before output stage.
@@ -3125,7 +3186,8 @@
// chains list to be processed before output mix effects. Relative order between other
// sessions is not important.
static_assert(AUDIO_SESSION_OUTPUT_MIX == 0 &&
- AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX,
+ AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX &&
+ AUDIO_SESSION_DEVICE < AUDIO_SESSION_OUTPUT_STAGE,
"audio_session_t constants misdefined");
size_t size = mEffectChains.size();
size_t i = 0;
@@ -3281,8 +3343,8 @@
// If the device is AUDIO_DEVICE_OUT_BUS, check for downstream latency.
//
- // Note: we access outDevice() outside of mLock.
- if (isMsdDevice() && (outDevice() & AUDIO_DEVICE_OUT_BUS) != 0) {
+ // Note: we access outDeviceTypes() outside of mLock.
+ if (isMsdDevice() && outDeviceTypes().count(AUDIO_DEVICE_OUT_BUS) != 0) {
// Here, we try for the AF lock, but do not block on it as the latency
// is more informational.
if (mAudioFlinger->mLock.tryLock() == NO_ERROR) {
@@ -3818,8 +3880,10 @@
if (diff > 0) {
// notify of throttle end on debug log
// but prevent spamming for bluetooth
- ALOGD_IF(!audio_is_a2dp_out_device(outDevice()) &&
- !audio_is_hearing_aid_out_device(outDevice()),
+ ALOGD_IF(!isSingleDeviceType(
+ outDeviceTypes(), audio_is_a2dp_out_device) &&
+ !isSingleDeviceType(
+ outDeviceTypes(), audio_is_hearing_aid_out_device),
"mixer(%p) throttle end: throttle time(%u)", this, diff);
mThreadThrottleEndMs = mThreadThrottleTimeMs;
}
@@ -4004,25 +4068,31 @@
// store new device and send to effects
audio_devices_t type = AUDIO_DEVICE_NONE;
+ AudioDeviceTypeAddrVector deviceTypeAddrs;
for (unsigned int i = 0; i < patch->num_sinks; i++) {
+ LOG_ALWAYS_FATAL_IF(popcount(patch->sinks[i].ext.device.type) > 1
+ && !mOutput->audioHwDev->supportsAudioPatches(),
+ "Enumerated device type(%#x) must not be used "
+ "as it does not support audio patches",
+ patch->sinks[i].ext.device.type);
type |= patch->sinks[i].ext.device.type;
+ deviceTypeAddrs.push_back(AudioDeviceTypeAddr(patch->sinks[i].ext.device.type,
+ patch->sinks[i].ext.device.address));
}
audio_port_handle_t sinkPortId = patch->sinks[0].id;
#ifdef ADD_BATTERY_DATA
// when changing the audio output device, call addBatteryData to notify
// the change
- if (mOutDevice != type) {
+ if (outDeviceTypes() != deviceTypes) {
uint32_t params = 0;
// check whether speaker is on
- if (type & AUDIO_DEVICE_OUT_SPEAKER) {
+ if (deviceTypes.count(AUDIO_DEVICE_OUT_SPEAKER) > 0) {
params |= IMediaPlayerService::kBatteryDataSpeakerOn;
}
- audio_devices_t deviceWithoutSpeaker
- = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER;
// check if any other device (except speaker) is on
- if (type & deviceWithoutSpeaker) {
+ if (!isSingleDeviceType(deviceTypes, AUDIO_DEVICE_OUT_SPEAKER)) {
params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn;
}
@@ -4033,14 +4103,15 @@
#endif
for (size_t i = 0; i < mEffectChains.size(); i++) {
- mEffectChains[i]->setDevice_l(type);
+ mEffectChains[i]->setDevices_l(deviceTypeAddrs);
}
- // mPrevOutDevice is the latest device set by createAudioPatch_l(). It is not set when
- // the thread is created so that the first patch creation triggers an ioConfigChanged callback
- bool configChanged = (mPrevOutDevice != type) || (mDeviceId != sinkPortId);
- mOutDevice = type;
+ // mPatch.num_sinks is not set when the thread is created so that
+ // the first patch creation triggers an ioConfigChanged callback
+ bool configChanged = (mPatch.num_sinks == 0) ||
+ (mPatch.sinks[0].id != sinkPortId);
mPatch = *patch;
+ mOutDeviceTypeAddrs = deviceTypeAddrs;
if (mOutput->audioHwDev->supportsAudioPatches()) {
sp<DeviceHalInterface> hwDevice = mOutput->audioHwDev->hwDevice();
@@ -4066,8 +4137,6 @@
*handle = AUDIO_PATCH_HANDLE_NONE;
}
if (configChanged) {
- mPrevOutDevice = type;
- mDeviceId = sinkPortId;
sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED);
}
return status;
@@ -4091,7 +4160,8 @@
{
status_t status = NO_ERROR;
- mOutDevice = AUDIO_DEVICE_NONE;
+ mPatch = audio_patch{};
+ mOutDeviceTypeAddrs.clear();
if (mOutput->audioHwDev->supportsAudioPatches()) {
sp<DeviceHalInterface> hwDevice = mOutput->audioHwDev->hwDevice();
@@ -4131,8 +4201,8 @@
// ----------------------------------------------------------------------------
AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
- audio_io_handle_t id, audio_devices_t device, bool systemReady, type_t type)
- : PlaybackThread(audioFlinger, output, id, device, type, systemReady),
+ audio_io_handle_t id, bool systemReady, type_t type)
+ : PlaybackThread(audioFlinger, output, id, type, systemReady),
// mAudioMixer below
// mFastMixer below
mFastMixerFutex(0),
@@ -4142,7 +4212,7 @@
// mNormalSink below
{
setMasterBalance(audioFlinger->getMasterBalance_l());
- ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type);
+ ALOGV("MixerThread() id=%d type=%d", id, type);
ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%#x, mFrameSize=%zu, "
"mFrameCount=%zu, mNormalFrameCount=%zu",
mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount,
@@ -4184,7 +4254,7 @@
// scheduled reliably with CFS. However, the BT A2DP HAL is
// bursty (does not pull at a regular rate) and so cannot operate with FastMixer.
initFastMixer = mFrameCount < mNormalFrameCount
- && (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) == 0;
+ && Intersection(outDeviceTypes(), getAudioDeviceOutAllA2dpSet()).empty();
break;
}
ALOGW_IF(initFastMixer == false && mFrameCount < mNormalFrameCount,
@@ -5301,11 +5371,11 @@
return false;
}
// Check validity as we don't call AudioMixer::create() here.
- if (!AudioMixer::isValidFormat(format)) {
+ if (!mAudioMixer->isValidFormat(format)) {
ALOGW("%s: invalid format: %#x", __func__, format);
return false;
}
- if (!AudioMixer::isValidChannelMask(channelMask)) {
+ if (!mAudioMixer->isValidChannelMask(channelMask)) {
ALOGW("%s: invalid channelMask: %#x", __func__, channelMask);
return false;
}
@@ -5355,39 +5425,7 @@
}
}
if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
-#ifdef ADD_BATTERY_DATA
- // when changing the audio output device, call addBatteryData to notify
- // the change
- if (mOutDevice != value) {
- uint32_t params = 0;
- // check whether speaker is on
- if (value & AUDIO_DEVICE_OUT_SPEAKER) {
- params |= IMediaPlayerService::kBatteryDataSpeakerOn;
- }
-
- audio_devices_t deviceWithoutSpeaker
- = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER;
- // check if any other device (except speaker) is on
- if (value & deviceWithoutSpeaker) {
- params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn;
- }
-
- if (params != 0) {
- addBatteryData(params);
- }
- }
-#endif
-
- // forward device change to effects that have requested to be
- // aware of attached audio device.
- if (value != AUDIO_DEVICE_NONE) {
- a2dpDeviceChanged =
- (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP);
- mOutDevice = value;
- for (size_t i = 0; i < mEffectChains.size(); i++) {
- mEffectChains[i]->setDevice_l(mOutDevice);
- }
- }
+ LOG_FATAL("Should not set routing device in MixerThread");
}
if (status == NO_ERROR) {
@@ -5488,9 +5526,8 @@
// ----------------------------------------------------------------------------
AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger,
- AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device,
- ThreadBase::type_t type, bool systemReady)
- : PlaybackThread(audioFlinger, output, id, device, type, systemReady)
+ AudioStreamOut* output, audio_io_handle_t id, ThreadBase::type_t type, bool systemReady)
+ : PlaybackThread(audioFlinger, output, id, type, systemReady)
{
setMasterBalance(audioFlinger->getMasterBalance_l());
}
@@ -5658,10 +5695,17 @@
minFrames = 1;
}
- if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() &&
+ const size_t framesReady = track->framesReady();
+ const int trackId = track->id();
+ if (ATRACE_ENABLED()) {
+ std::string traceName("nRdy");
+ traceName += std::to_string(trackId);
+ ATRACE_INT(traceName.c_str(), framesReady);
+ }
+ if ((framesReady >= minFrames) && track->isReady() && !track->isPaused() &&
!track->isStopping_2() && !track->isStopped())
{
- ALOGVV("track(%d) s=%08x [OK]", track->id(), cblk->mServer);
+ ALOGVV("track(%d) s=%08x [OK]", trackId, cblk->mServer);
if (track->mFillingUpStatus == Track::FS_FILLED) {
track->mFillingUpStatus = Track::FS_ACTIVE;
@@ -5724,7 +5768,7 @@
int64_t framesWritten = mBytesWritten / mFrameSize;
if (mStandby || !last ||
track->presentationComplete(framesWritten, audioHALFrames) ||
- track->isPaused()) {
+ track->isPaused() || mHwPaused) {
if (track->isStopping_2()) {
track->mState = TrackBase::STOPPED;
}
@@ -5738,7 +5782,7 @@
// fill a buffer, then remove it from active list.
// Only consider last track started for mixer state control
if (--(track->mRetryCount) <= 0) {
- ALOGV("BUFFER TIMEOUT: remove track(%d) from active list", track->id());
+ ALOGV("BUFFER TIMEOUT: remove track(%d) from active list", trackId);
tracksToRemove->add(track);
// indicate to client process that the track was disabled because of underrun;
// it will then automatically call start() when data is available
@@ -5746,7 +5790,7 @@
} else if (last) {
ALOGW("pause because of UNDERRUN, framesReady = %zu,"
"minFrames = %u, mFormat = %#x",
- track->framesReady(), minFrames, mFormat);
+ framesReady, minFrames, mFormat);
mixerStatus = MIXER_TRACKS_ENABLED;
if (mHwSupportsPause && !mHwPaused && !mStandby) {
doHwPause = true;
@@ -5885,16 +5929,7 @@
AudioParameter param = AudioParameter(keyValuePair);
int value;
if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
- // forward device change to effects that have requested to be
- // aware of attached audio device.
- if (value != AUDIO_DEVICE_NONE) {
- a2dpDeviceChanged =
- (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP);
- mOutDevice = value;
- for (size_t i = 0; i < mEffectChains.size(); i++) {
- mEffectChains[i]->setDevice_l(mOutDevice);
- }
- }
+ LOG_FATAL("Should not set routing device in DirectOutputThread");
}
if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
// do not accept frame count changes if tracks are open as the track buffer
@@ -5978,6 +6013,7 @@
mHwPaused = false;
mFlushPending = false;
mTimestampVerifier.discontinuity(); // DIRECT and OFFLOADED flush resets frame count.
+ mTimestamp.clear();
}
int64_t AudioFlinger::DirectOutputThread::computeWaitTimeNs_l() const {
@@ -6106,8 +6142,8 @@
// ----------------------------------------------------------------------------
AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger,
- AudioStreamOut* output, audio_io_handle_t id, uint32_t device, bool systemReady)
- : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD, systemReady),
+ AudioStreamOut* output, audio_io_handle_t id, bool systemReady)
+ : DirectOutputThread(audioFlinger, output, id, OFFLOAD, systemReady),
mPausedWriteLength(0), mPausedBytesRemaining(0), mKeepWakeLock(true),
mOffloadUnderrunPosition(~0LL)
{
@@ -6354,7 +6390,9 @@
}
}
// compute volume for this track
- processVolume_l(track, last);
+ if (track->isReady()) { // check ready to prevent premature start.
+ processVolume_l(track, last);
+ }
}
// make sure the pause/flush/resume sequence is executed in the right order.
@@ -6430,7 +6468,7 @@
AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger,
AudioFlinger::MixerThread* mainThread, audio_io_handle_t id, bool systemReady)
- : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(),
+ : MixerThread(audioFlinger, mainThread->getOutput(), id,
systemReady, DUPLICATING),
mWaitTimeMs(UINT_MAX)
{
@@ -6662,12 +6700,11 @@
AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger,
AudioStreamIn *input,
audio_io_handle_t id,
- audio_devices_t outDevice,
- audio_devices_t inDevice,
bool systemReady
) :
- ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD, systemReady),
+ ThreadBase(audioFlinger, id, RECORD, systemReady),
mInput(input),
+ mSource(mInput),
mActiveTracks(&this->mLocalLog),
mRsmpInBuffer(NULL),
// mRsmpInFrames, mRsmpInFramesP2, and mRsmpInFramesOA are set by readInputParameters_l()
@@ -6697,8 +6734,9 @@
// TODO: We may also match on address as well as device type for
// AUDIO_DEVICE_IN_BUS, AUDIO_DEVICE_IN_BLUETOOTH_A2DP, AUDIO_DEVICE_IN_REMOTE_SUBMIX
- mTimestampCorrectedDevices = (audio_devices_t)property_get_int64(
- "audio.timestamp.corrected_input_devices",
+ // TODO: This property should be ensure that only contains one single device type.
+ mTimestampCorrectedDevice = (audio_devices_t)property_get_int64(
+ "audio.timestamp.corrected_input_device",
(int64_t)(mIsMsdDevice ? AUDIO_DEVICE_IN_BUS // turn on by default for MSD
: AUDIO_DEVICE_NONE));
@@ -7120,7 +7158,7 @@
} else {
ATRACE_BEGIN("read");
size_t bytesRead;
- status_t result = mInput->stream->read(
+ status_t result = mSource->read(
(uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize, &bytesRead);
ATRACE_END();
if (result < 0) {
@@ -7142,7 +7180,7 @@
int64_t position, time;
if (mStandby) {
mTimestampVerifier.discontinuity();
- } else if (mInput->stream->getCapturePosition(&position, &time) == NO_ERROR
+ } else if (mSource->getCapturePosition(&position, &time) == NO_ERROR
&& time > mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL]) {
mTimestampVerifier.add(position, time, mSampleRate);
@@ -7423,7 +7461,7 @@
sq->end(false /*didModify*/);
}
}
- status_t result = mInput->stream->standby();
+ status_t result = mSource->standby();
ALOGE_IF(result != OK, "Error when putting input stream into standby: %d", result);
// If going into standby, flush the pipe source.
@@ -8042,7 +8080,7 @@
{
// disable AEC and NS if the device is a BT SCO headset supporting those
// pre processings
- bool suspend = audio_is_bluetooth_sco_device(mInDevice) &&
+ bool suspend = audio_is_bluetooth_sco_device(inDeviceType()) &&
mAudioFlinger->btNrecIsOff();
if (mBtNrecSuspended.exchange(suspend) != suspend) {
for (size_t i = 0; i < mEffectChains.size(); i++) {
@@ -8107,34 +8145,11 @@
}
}
if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
- // forward device change to effects that have requested to be
- // aware of attached audio device.
- for (size_t i = 0; i < mEffectChains.size(); i++) {
- mEffectChains[i]->setDevice_l(value);
- }
-
- // store input device and output device but do not forward output device to audio HAL.
- // Note that status is ignored by the caller for output device
- // (see AudioFlinger::setParameters()
- if (audio_is_output_devices(value)) {
- mOutDevice = value;
- status = BAD_VALUE;
- } else {
- mInDevice = value;
- if (value != AUDIO_DEVICE_NONE) {
- mPrevInDevice = value;
- }
- checkBtNrec_l();
- }
+ LOG_FATAL("Should not set routing device in RecordThread");
}
if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR &&
mAudioSource != (audio_source_t)value) {
- // forward device change to effects that have requested to be
- // aware of attached audio device.
- for (size_t i = 0; i < mEffectChains.size(); i++) {
- mEffectChains[i]->setAudioSource_l((audio_source_t)value);
- }
- mAudioSource = (audio_source_t)value;
+ LOG_FATAL("Should not set audio source in RecordThread");
}
if (status == NO_ERROR) {
@@ -8336,11 +8351,11 @@
status_t status = NO_ERROR;
// store new device and send to effects
- mInDevice = patch->sources[0].ext.device.type;
+ mInDeviceTypeAddr.mType = patch->sources[0].ext.device.type;
+ mInDeviceTypeAddr.mAddress = patch->sources[0].ext.device.address;
audio_port_handle_t deviceId = patch->sources[0].id;
- mPatch = *patch;
for (size_t i = 0; i < mEffectChains.size(); i++) {
- mEffectChains[i]->setDevice_l(mInDevice);
+ mEffectChains[i]->setInputDevice_l(inDeviceTypeAddr());
}
checkBtNrec_l();
@@ -8379,10 +8394,9 @@
*handle = AUDIO_PATCH_HANDLE_NONE;
}
- if ((mInDevice != mPrevInDevice) || (mDeviceId != deviceId)) {
+ if ((mPatch.num_sources == 0) || (mPatch.sources[0].id != deviceId)) {
sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED);
- mPrevInDevice = mInDevice;
- mDeviceId = deviceId;
+ mPatch = *patch;
}
return status;
@@ -8392,7 +8406,8 @@
{
status_t status = NO_ERROR;
- mInDevice = AUDIO_DEVICE_NONE;
+ mPatch = audio_patch{};
+ mInDeviceTypeAddr.reset();
if (mInput->audioHwDev->supportsAudioPatches()) {
sp<DeviceHalInterface> hwDevice = mInput->audioHwDev->hwDevice();
@@ -8405,15 +8420,30 @@
return status;
}
+void AudioFlinger::RecordThread::updateOutDevices(const DeviceDescriptorBaseVector& outDevices)
+{
+ mOutDevices = outDevices;
+ mOutDeviceTypeAddrs = deviceTypeAddrsFromDescriptors(mOutDevices);
+ for (size_t i = 0; i < mEffectChains.size(); i++) {
+ mEffectChains[i]->setDevices_l(outDeviceTypeAddrs());
+ }
+}
+
void AudioFlinger::RecordThread::addPatchTrack(const sp<PatchRecord>& record)
{
Mutex::Autolock _l(mLock);
mTracks.add(record);
+ if (record->getSource()) {
+ mSource = record->getSource();
+ }
}
void AudioFlinger::RecordThread::deletePatchTrack(const sp<PatchRecord>& record)
{
Mutex::Autolock _l(mLock);
+ if (mSource == record->getSource()) {
+ mSource = mInput;
+ }
destroyTrack_l(record);
}
@@ -8475,9 +8505,8 @@
AudioFlinger::MmapThread::MmapThread(
const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
- AudioHwDevice *hwDev, sp<StreamHalInterface> stream,
- audio_devices_t outDevice, audio_devices_t inDevice, bool systemReady)
- : ThreadBase(audioFlinger, id, outDevice, inDevice, MMAP, systemReady),
+ AudioHwDevice *hwDev, sp<StreamHalInterface> stream, bool systemReady)
+ : ThreadBase(audioFlinger, id, MMAP, systemReady),
mSessionId(AUDIO_SESSION_NONE),
mPortId(AUDIO_PORT_HANDLE_NONE),
mHalStream(stream), mHalDevice(hwDev->hwDevice()), mAudioHwDev(hwDev),
@@ -8843,26 +8872,7 @@
int value;
bool sendToHal = true;
if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
- audio_devices_t device = (audio_devices_t)value;
- // forward device change to effects that have requested to be
- // aware of attached audio device.
- if (device != AUDIO_DEVICE_NONE) {
- for (size_t i = 0; i < mEffectChains.size(); i++) {
- mEffectChains[i]->setDevice_l(device);
- }
- }
- if (audio_is_output_devices(device)) {
- mOutDevice = device;
- if (!isOutput()) {
- sendToHal = false;
- }
- } else {
- mInDevice = device;
- if (device != AUDIO_DEVICE_NONE) {
- mPrevInDevice = value;
- }
- // TODO: implement and call checkBtNrec_l();
- }
+ LOG_FATAL("Should not happen set routing device in MmapThread");
}
if (sendToHal) {
status = mHalStream->setParameters(keyValuePair);
@@ -8921,24 +8931,39 @@
// store new device and send to effects
audio_devices_t type = AUDIO_DEVICE_NONE;
audio_port_handle_t deviceId;
+ AudioDeviceTypeAddrVector sinkDeviceTypeAddrs;
+ AudioDeviceTypeAddr sourceDeviceTypeAddr;
+ uint32_t numDevices = 0;
if (isOutput()) {
for (unsigned int i = 0; i < patch->num_sinks; i++) {
+ LOG_ALWAYS_FATAL_IF(popcount(patch->sinks[i].ext.device.type) > 1
+ && !mAudioHwDev->supportsAudioPatches(),
+ "Enumerated device type(%#x) must not be used "
+ "as it does not support audio patches",
+ patch->sinks[i].ext.device.type);
type |= patch->sinks[i].ext.device.type;
+ sinkDeviceTypeAddrs.push_back(AudioDeviceTypeAddr(patch->sinks[i].ext.device.type,
+ patch->sinks[i].ext.device.address));
}
deviceId = patch->sinks[0].id;
+ numDevices = mPatch.num_sinks;
} else {
type = patch->sources[0].ext.device.type;
deviceId = patch->sources[0].id;
+ numDevices = mPatch.num_sources;
+ sourceDeviceTypeAddr.mType = patch->sources[0].ext.device.type;
+ sourceDeviceTypeAddr.mAddress = patch->sources[0].ext.device.address;
}
for (size_t i = 0; i < mEffectChains.size(); i++) {
- mEffectChains[i]->setDevice_l(type);
+ if (isOutput()) {
+ mEffectChains[i]->setDevices_l(sinkDeviceTypeAddrs);
+ } else {
+ mEffectChains[i]->setInputDevice_l(sourceDeviceTypeAddr);
+ }
}
- if (isOutput()) {
- mOutDevice = type;
- } else {
- mInDevice = type;
+ if (!isOutput()) {
// store new source and send to effects
if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) {
mAudioSource = patch->sinks[0].ext.mix.usecase.source;
@@ -8975,26 +9000,21 @@
*handle = AUDIO_PATCH_HANDLE_NONE;
}
- if (isOutput() && (mPrevOutDevice != mOutDevice || mDeviceId != deviceId)) {
- mPrevOutDevice = type;
- sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED);
+ if (numDevices == 0 || mDeviceId != deviceId) {
+ if (isOutput()) {
+ sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED);
+ mOutDeviceTypeAddrs = sinkDeviceTypeAddrs;
+ } else {
+ sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED);
+ mInDeviceTypeAddr = sourceDeviceTypeAddr;
+ }
sp<MmapStreamCallback> callback = mCallback.promote();
if (mDeviceId != deviceId && callback != 0) {
mLock.unlock();
callback->onRoutingChanged(deviceId);
mLock.lock();
}
- mDeviceId = deviceId;
- }
- if (!isOutput() && (mPrevInDevice != mInDevice || mDeviceId != deviceId)) {
- mPrevInDevice = type;
- sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED);
- sp<MmapStreamCallback> callback = mCallback.promote();
- if (mDeviceId != deviceId && callback != 0) {
- mLock.unlock();
- callback->onRoutingChanged(deviceId);
- mLock.lock();
- }
+ mPatch = *patch;
mDeviceId = deviceId;
}
return status;
@@ -9004,7 +9024,9 @@
{
status_t status = NO_ERROR;
- mInDevice = AUDIO_DEVICE_NONE;
+ mPatch = audio_patch{};
+ mOutDeviceTypeAddrs.clear();
+ mInDeviceTypeAddr.reset();
bool supportsAudioPatches = mHalDevice->supportsAudioPatches(&supportsAudioPatches) == OK ?
supportsAudioPatches : false;
@@ -9105,8 +9127,8 @@
const effect_descriptor_t *desc, audio_session_t sessionId)
{
// No global effect sessions on mmap threads
- if (sessionId == AUDIO_SESSION_OUTPUT_MIX || sessionId == AUDIO_SESSION_OUTPUT_STAGE) {
- ALOGW("checkEffectCompatibility_l(): global effect %s on record thread %s",
+ if (audio_is_global_session(sessionId)) {
+ ALOGW("checkEffectCompatibility_l(): global effect %s on MMAP thread %s",
desc->name, mThreadName);
return BAD_VALUE;
}
@@ -9128,7 +9150,6 @@
}
return NO_ERROR;
-
}
void AudioFlinger::MmapThread::checkInvalidTracks_l()
@@ -9180,9 +9201,8 @@
AudioFlinger::MmapPlaybackThread::MmapPlaybackThread(
const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
- AudioHwDevice *hwDev, AudioStreamOut *output,
- audio_devices_t outDevice, audio_devices_t inDevice, bool systemReady)
- : MmapThread(audioFlinger, id, hwDev, output->stream, outDevice, inDevice, systemReady),
+ AudioHwDevice *hwDev, AudioStreamOut *output, bool systemReady)
+ : MmapThread(audioFlinger, id, hwDev, output->stream, systemReady),
mStreamType(AUDIO_STREAM_MUSIC),
mStreamVolume(1.0),
mStreamMute(false),
@@ -9392,9 +9412,8 @@
AudioFlinger::MmapCaptureThread::MmapCaptureThread(
const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
- AudioHwDevice *hwDev, AudioStreamIn *input,
- audio_devices_t outDevice, audio_devices_t inDevice, bool systemReady)
- : MmapThread(audioFlinger, id, hwDev, input->stream, outDevice, inDevice, systemReady),
+ AudioHwDevice *hwDev, AudioStreamIn *input, bool systemReady)
+ : MmapThread(audioFlinger, id, hwDev, input->stream, systemReady),
mInput(input)
{
snprintf(mThreadName, kThreadNameLength, "AudioMmapIn_%X", id);
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 87bebf3..4c53e28 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -37,8 +37,7 @@
static const char *threadTypeToString(type_t type);
ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
- audio_devices_t outDevice, audio_devices_t inDevice, type_t type,
- bool systemReady);
+ type_t type, bool systemReady);
virtual ~ThreadBase();
virtual status_t readyToRun();
@@ -52,6 +51,7 @@
CFG_EVENT_SET_PARAMETER,
CFG_EVENT_CREATE_AUDIO_PATCH,
CFG_EVENT_RELEASE_AUDIO_PATCH,
+ CFG_EVENT_UPDATE_OUT_DEVICE,
};
class ConfigEventData: public RefBase {
@@ -219,6 +219,28 @@
virtual ~ReleaseAudioPatchConfigEvent() {}
};
+ class UpdateOutDevicesConfigEventData : public ConfigEventData {
+ public:
+ explicit UpdateOutDevicesConfigEventData(const DeviceDescriptorBaseVector& outDevices) :
+ mOutDevices(outDevices) {}
+
+ virtual void dump(char *buffer, size_t size) {
+ snprintf(buffer, size, "Devices: %s", android::toString(mOutDevices).c_str());
+ }
+
+ DeviceDescriptorBaseVector mOutDevices;
+ };
+
+ class UpdateOutDevicesConfigEvent : public ConfigEvent {
+ public:
+ explicit UpdateOutDevicesConfigEvent(const DeviceDescriptorBaseVector& outDevices) :
+ ConfigEvent(CFG_EVENT_UPDATE_OUT_DEVICE) {
+ mData = new UpdateOutDevicesConfigEventData(outDevices);
+ }
+
+ virtual ~UpdateOutDevicesConfigEvent();
+ };
+
class PMDeathRecipient : public IBinder::DeathRecipient {
public:
explicit PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {}
@@ -249,6 +271,8 @@
// Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects,
// and returns the [normal mix] buffer's frame count.
virtual size_t frameCount() const = 0;
+ virtual uint32_t latency_l() const { return 0; }
+ virtual void setVolumeForOutput_l(float left __unused, float right __unused) const {}
// Return's the HAL's frame count i.e. fast mixer buffer size.
size_t frameCountHAL() const { return mFrameCount; }
@@ -278,19 +302,33 @@
status_t sendCreateAudioPatchConfigEvent(const struct audio_patch *patch,
audio_patch_handle_t *handle);
status_t sendReleaseAudioPatchConfigEvent(audio_patch_handle_t handle);
+ status_t sendUpdateOutDeviceConfigEvent(
+ const DeviceDescriptorBaseVector& outDevices);
void processConfigEvents_l();
virtual void cacheParameters_l() = 0;
virtual status_t createAudioPatch_l(const struct audio_patch *patch,
audio_patch_handle_t *handle) = 0;
virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle) = 0;
+ virtual void updateOutDevices(const DeviceDescriptorBaseVector& outDevices);
virtual void toAudioPortConfig(struct audio_port_config *config) = 0;
// see note at declaration of mStandby, mOutDevice and mInDevice
bool standby() const { return mStandby; }
- audio_devices_t outDevice() const { return mOutDevice; }
- audio_devices_t inDevice() const { return mInDevice; }
- audio_devices_t getDevice() const { return isOutput() ? mOutDevice : mInDevice; }
+ const DeviceTypeSet outDeviceTypes() const {
+ return getAudioDeviceTypes(mOutDeviceTypeAddrs);
+ }
+ audio_devices_t inDeviceType() const { return mInDeviceTypeAddr.mType; }
+ DeviceTypeSet getDeviceTypes() const {
+ return isOutput() ? outDeviceTypes() : DeviceTypeSet({inDeviceType()});
+ }
+
+ const AudioDeviceTypeAddrVector& outDeviceTypeAddrs() const {
+ return mOutDeviceTypeAddrs;
+ }
+ const AudioDeviceTypeAddr& inDeviceTypeAddr() const {
+ return mInDeviceTypeAddr;
+ }
virtual bool isOutput() const = 0;
@@ -388,14 +426,9 @@
// check if some effects must be suspended/restored when an effect is enabled
// or disabled
- void checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
- bool enabled,
- audio_session_t sessionId =
- AUDIO_SESSION_OUTPUT_MIX);
- void checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect,
- bool enabled,
- audio_session_t sessionId =
- AUDIO_SESSION_OUTPUT_MIX);
+ void checkSuspendOnEffectEnabled(bool enabled,
+ audio_session_t sessionId,
+ bool threadLocked);
virtual status_t setSyncEvent(const sp<SyncEvent>& event) = 0;
virtual bool isValidSyncEvent(const sp<SyncEvent>& event) const = 0;
@@ -429,6 +462,9 @@
mutable Mutex mLock;
+ void onEffectEnable(const sp<EffectModule>& effect);
+ void onEffectDisable();
+
protected:
// entry describing an effect being suspended in mSuspendedSessions keyed vector
@@ -502,26 +538,21 @@
// HAL format if Fastmixer is used.
audio_format_t mHALFormat;
size_t mBufferSize; // HAL buffer size for read() or write()
-
+ AudioDeviceTypeAddrVector mOutDeviceTypeAddrs; // output device types and addresses
+ AudioDeviceTypeAddr mInDeviceTypeAddr; // input device type and address
Vector< sp<ConfigEvent> > mConfigEvents;
Vector< sp<ConfigEvent> > mPendingConfigEvents; // events awaiting system ready
// These fields are written and read by thread itself without lock or barrier,
- // and read by other threads without lock or barrier via standby(), outDevice()
- // and inDevice().
+ // and read by other threads without lock or barrier via standby(), outDeviceTypes()
+ // and inDeviceType().
// Because of the absence of a lock or barrier, any other thread that reads
// these fields must use the information in isolation, or be prepared to deal
// with possibility that it might be inconsistent with other information.
bool mStandby; // Whether thread is currently in standby.
- audio_devices_t mOutDevice; // output device
- audio_devices_t mInDevice; // input device
- audio_devices_t mPrevOutDevice; // previous output device
- audio_devices_t mPrevInDevice; // previous input device
+
struct audio_patch mPatch;
- /**
- * @brief mDeviceId current device port unique identifier
- */
- audio_port_handle_t mDeviceId = AUDIO_PORT_HANDLE_NONE;
+
audio_source_t mAudioSource;
const audio_io_handle_t mId;
@@ -544,7 +575,8 @@
ExtendedTimestamp mTimestamp;
TimestampVerifier< // For timestamp statistics.
int64_t /* frame count */, int64_t /* time ns */> mTimestampVerifier;
- audio_devices_t mTimestampCorrectedDevices = AUDIO_DEVICE_NONE;
+ // Timestamp corrected device should be a single device.
+ audio_devices_t mTimestampCorrectedDevice = AUDIO_DEVICE_NONE;
// ThreadLoop statistics per iteration.
int64_t mLastIoBeginNs = -1;
@@ -719,7 +751,7 @@
static const nsecs_t kMaxNextBufferDelayNs = 100000000;
PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
- audio_io_handle_t id, audio_devices_t device, type_t type, bool systemReady);
+ audio_io_handle_t id, type_t type, bool systemReady);
virtual ~PlaybackThread();
// Thread virtuals
@@ -782,7 +814,7 @@
// return estimated latency in milliseconds, as reported by HAL
uint32_t latency() const;
// same, but lock must already be held
- uint32_t latency_l() const;
+ uint32_t latency_l() const override;
// VolumeInterface
virtual void setMasterVolume(float value);
@@ -792,7 +824,7 @@
virtual void setStreamMute(audio_stream_type_t stream, bool muted);
virtual float streamVolume(audio_stream_type_t stream) const;
- void setVolumeForOutput_l(float left, float right) const;
+ void setVolumeForOutput_l(float left, float right) const override;
sp<Track> createTrack_l(
const sp<AudioFlinger::Client>& client,
@@ -886,10 +918,10 @@
}
bool isTimestampCorrectionEnabled() const override {
- const audio_devices_t device =
- mOutDevice & mTimestampCorrectedDevices;
- return audio_is_output_devices(device) && popcount(device) > 0;
+ return audio_is_output_devices(mTimestampCorrectedDevice)
+ && outDeviceTypes().count(mTimestampCorrectedDevice) != 0;
}
+
protected:
// updated by readOutputParameters_l()
size_t mNormalFrameCount; // normal mixer and effects
@@ -1171,7 +1203,6 @@
MixerThread(const sp<AudioFlinger>& audioFlinger,
AudioStreamOut* output,
audio_io_handle_t id,
- audio_devices_t device,
bool systemReady,
type_t type = MIXER);
virtual ~MixerThread();
@@ -1269,8 +1300,8 @@
public:
DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
- audio_io_handle_t id, audio_devices_t device, bool systemReady)
- : DirectOutputThread(audioFlinger, output, id, device, DIRECT, systemReady) { }
+ audio_io_handle_t id, bool systemReady)
+ : DirectOutputThread(audioFlinger, output, id, DIRECT, systemReady) { }
virtual ~DirectOutputThread();
@@ -1305,8 +1336,7 @@
bool mVolumeShaperActive = false;
DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
- audio_io_handle_t id, audio_devices_t device, ThreadBase::type_t type,
- bool systemReady);
+ audio_io_handle_t id, ThreadBase::type_t type, bool systemReady);
void processVolume_l(Track *track, bool lastTrack);
// prepareTracks_l() tells threadLoop_mix() the name of the single active track
@@ -1345,7 +1375,7 @@
public:
OffloadThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
- audio_io_handle_t id, uint32_t device, bool systemReady);
+ audio_io_handle_t id, bool systemReady);
virtual ~OffloadThread() {};
virtual void flushHw_l();
@@ -1516,8 +1546,6 @@
RecordThread(const sp<AudioFlinger>& audioFlinger,
AudioStreamIn *input,
audio_io_handle_t id,
- audio_devices_t outDevice,
- audio_devices_t inDevice,
bool systemReady
);
virtual ~RecordThread();
@@ -1577,6 +1605,7 @@
virtual status_t createAudioPatch_l(const struct audio_patch *patch,
audio_patch_handle_t *handle);
virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle);
+ void updateOutDevices(const DeviceDescriptorBaseVector& outDevices) override;
void addPatchTrack(const sp<PatchRecord>& record);
void deletePatchTrack(const sp<PatchRecord>& record);
@@ -1629,8 +1658,8 @@
bool isTimestampCorrectionEnabled() const override {
// checks popcount for exactly one device.
- return audio_is_input_device(
- mInDevice & mTimestampCorrectedDevices);
+ return audio_is_input_device(mTimestampCorrectedDevice)
+ && inDeviceType() == mTimestampCorrectedDevice;
}
protected:
@@ -1647,6 +1676,7 @@
void checkBtNrec_l();
AudioStreamIn *mInput;
+ Source *mSource;
SortedVector < sp<RecordTrack> > mTracks;
// mActiveTracks has dual roles: it indicates the current active track(s), and
// is used together with mStartStopCond to indicate start()/stop() progress
@@ -1708,6 +1738,8 @@
std::atomic_bool mBtNrecSuspended;
int64_t mFramesRead = 0; // continuous running counter.
+
+ DeviceDescriptorBaseVector mOutDevices;
};
class MmapThread : public ThreadBase
@@ -1717,8 +1749,7 @@
#include "MmapTracks.h"
MmapThread(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
- AudioHwDevice *hwDev, sp<StreamHalInterface> stream,
- audio_devices_t outDevice, audio_devices_t inDevice, bool systemReady);
+ AudioHwDevice *hwDev, sp<StreamHalInterface> stream, bool systemReady);
virtual ~MmapThread();
virtual void configure(const audio_attributes_t *attr,
@@ -1791,6 +1822,11 @@
void dumpInternals_l(int fd, const Vector<String16>& args) override;
void dumpTracks_l(int fd, const Vector<String16>& args) override;
+ /**
+ * @brief mDeviceId current device port unique identifier
+ */
+ audio_port_handle_t mDeviceId = AUDIO_PORT_HANDLE_NONE;
+
audio_attributes_t mAttr;
audio_session_t mSessionId;
audio_port_handle_t mPortId;
@@ -1811,8 +1847,7 @@
public:
MmapPlaybackThread(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
- AudioHwDevice *hwDev, AudioStreamOut *output,
- audio_devices_t outDevice, audio_devices_t inDevice, bool systemReady);
+ AudioHwDevice *hwDev, AudioStreamOut *output, bool systemReady);
virtual ~MmapPlaybackThread() {}
virtual void configure(const audio_attributes_t *attr,
@@ -1861,8 +1896,7 @@
public:
MmapCaptureThread(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
- AudioHwDevice *hwDev, AudioStreamIn *input,
- audio_devices_t outDevice, audio_devices_t inDevice, bool systemReady);
+ AudioHwDevice *hwDev, AudioStreamIn *input, bool systemReady);
virtual ~MmapCaptureThread() {}
AudioStreamIn* clearInput();
diff --git a/services/audioflinger/TrackBase.h b/services/audioflinger/TrackBase.h
index 65589e2..52e7d59 100644
--- a/services/audioflinger/TrackBase.h
+++ b/services/audioflinger/TrackBase.h
@@ -327,6 +327,7 @@
virtual ~PatchProxyBufferProvider() {}
+ virtual bool producesBufferOnDemand() const = 0;
virtual status_t obtainBuffer(Proxy::Buffer* buffer,
const struct timespec *requested = NULL) = 0;
virtual void releaseBuffer(Proxy::Buffer* buffer) = 0;
@@ -349,6 +350,8 @@
mPeerProxy = nullptr;
}
+ bool producesBufferOnDemand() const override { return false; }
+
protected:
const sp<ClientProxy> mProxy;
sp<RefBase> mPeerReferenceHold; // keeps mPeerProxy alive during access.
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 51e57b5..e4402bd 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -18,12 +18,14 @@
#define LOG_TAG "AudioFlinger"
//#define LOG_NDEBUG 0
+#define ATRACE_TAG ATRACE_TAG_AUDIO
#include "Configuration.h"
#include <linux/futex.h>
#include <math.h>
#include <sys/syscall.h>
#include <utils/Log.h>
+#include <utils/Trace.h>
#include <private/media/AudioTrackShared.h>
@@ -822,16 +824,9 @@
}
for (auto& teePatch : mTeePatches) {
RecordThread::PatchRecord* patchRecord = teePatch.patchRecord.get();
-
- size_t framesWritten = writeFrames(patchRecord, sourceBuffer.i8, frameCount);
- // On buffer wrap, the buffer frame count will be less than requested,
- // when this happens a second buffer needs to be used to write the leftover audio
- size_t framesLeft = frameCount - framesWritten;
- if (framesWritten != 0 && framesLeft != 0) {
- framesWritten +=
- writeFrames(patchRecord, sourceBuffer.i8 + framesWritten * mFrameSize, framesLeft);
- framesLeft = frameCount - framesWritten;
- }
+ const size_t framesWritten = patchRecord->writeFrames(
+ sourceBuffer.i8, frameCount, mFrameSize);
+ const size_t framesLeft = frameCount - framesWritten;
ALOGW_IF(framesLeft != 0, "%s(%d) PatchRecord %d can not provide big enough "
"buffer %zu/%zu, dropping %zu frames", __func__, mId, patchRecord->mId,
framesWritten, frameCount, framesLeft);
@@ -843,26 +838,6 @@
spent.count(), mTeePatches.size());
}
-size_t AudioFlinger::PlaybackThread::Track::writeFrames(AudioBufferProvider* dest,
- const void* src,
- size_t frameCount) {
- AudioBufferProvider::Buffer patchBuffer;
- patchBuffer.frameCount = frameCount;
- auto status = dest->getNextBuffer(&patchBuffer);
- if (status != NO_ERROR) {
- ALOGW("%s PathRecord getNextBuffer failed with error %d: %s",
- __func__, status, strerror(-status));
- return 0;
- }
- ALOG_ASSERT(patchBuffer.frameCount <= frameCount);
- memcpy(patchBuffer.raw, src, patchBuffer.frameCount * mFrameSize);
- auto framesWritten = patchBuffer.frameCount;
- dest->releaseBuffer(&patchBuffer);
- return framesWritten;
-}
-
-// releaseBuffer() is not overridden
-
// ExtendedAudioBufferProvider interface
// framesReady() may return an approximation of the number of frames if called
@@ -1819,6 +1794,15 @@
ALOGV("%s(%d)", __func__, mId);
}
+size_t AudioFlinger::PlaybackThread::PatchTrack::framesReady() const
+{
+ if (mPeerProxy && mPeerProxy->producesBufferOnDemand()) {
+ return std::numeric_limits<size_t>::max();
+ } else {
+ return Track::framesReady();
+ }
+}
+
status_t AudioFlinger::PlaybackThread::PatchTrack::start(AudioSystem::sync_event_t event,
audio_session_t triggerSession)
{
@@ -1837,9 +1821,19 @@
ALOG_ASSERT(mPeerProxy != 0, "%s(%d): called without peer proxy", __func__, mId);
Proxy::Buffer buf;
buf.mFrameCount = buffer->frameCount;
+ if (ATRACE_ENABLED()) {
+ std::string traceName("PTnReq");
+ traceName += std::to_string(id());
+ ATRACE_INT(traceName.c_str(), buf.mFrameCount);
+ }
status_t status = mPeerProxy->obtainBuffer(&buf, &mPeerTimeout);
ALOGV_IF(status != NO_ERROR, "%s(%d): getNextBuffer status %d", __func__, mId, status);
buffer->frameCount = buf.mFrameCount;
+ if (ATRACE_ENABLED()) {
+ std::string traceName("PTnObt");
+ traceName += std::to_string(id());
+ ATRACE_INT(traceName.c_str(), buf.mFrameCount);
+ }
if (buf.mFrameCount == 0) {
return WOULD_BLOCK;
}
@@ -2400,6 +2394,39 @@
ALOGV("%s(%d)", __func__, mId);
}
+static size_t writeFramesHelper(
+ AudioBufferProvider* dest, const void* src, size_t frameCount, size_t frameSize)
+{
+ AudioBufferProvider::Buffer patchBuffer;
+ patchBuffer.frameCount = frameCount;
+ auto status = dest->getNextBuffer(&patchBuffer);
+ if (status != NO_ERROR) {
+ ALOGW("%s PathRecord getNextBuffer failed with error %d: %s",
+ __func__, status, strerror(-status));
+ return 0;
+ }
+ ALOG_ASSERT(patchBuffer.frameCount <= frameCount);
+ memcpy(patchBuffer.raw, src, patchBuffer.frameCount * frameSize);
+ size_t framesWritten = patchBuffer.frameCount;
+ dest->releaseBuffer(&patchBuffer);
+ return framesWritten;
+}
+
+// static
+size_t AudioFlinger::RecordThread::PatchRecord::writeFrames(
+ AudioBufferProvider* dest, const void* src, size_t frameCount, size_t frameSize)
+{
+ size_t framesWritten = writeFramesHelper(dest, src, frameCount, frameSize);
+ // On buffer wrap, the buffer frame count will be less than requested,
+ // when this happens a second buffer needs to be used to write the leftover audio
+ const size_t framesLeft = frameCount - framesWritten;
+ if (framesWritten != 0 && framesLeft != 0) {
+ framesWritten += writeFramesHelper(dest, (const char*)src + framesWritten * frameSize,
+ framesLeft, frameSize);
+ }
+ return framesWritten;
+}
+
// AudioBufferProvider interface
status_t AudioFlinger::RecordThread::PatchRecord::getNextBuffer(
AudioBufferProvider::Buffer* buffer)
@@ -2411,6 +2438,11 @@
ALOGV_IF(status != NO_ERROR,
"%s(%d): mPeerProxy->obtainBuffer status %d", __func__, mId, status);
buffer->frameCount = buf.mFrameCount;
+ if (ATRACE_ENABLED()) {
+ std::string traceName("PRnObt");
+ traceName += std::to_string(id());
+ ATRACE_INT(traceName.c_str(), buf.mFrameCount);
+ }
if (buf.mFrameCount == 0) {
return WOULD_BLOCK;
}
@@ -2439,6 +2471,180 @@
mProxy->releaseBuffer(buffer);
}
+#undef LOG_TAG
+#define LOG_TAG "AF::PthrPatchRecord"
+
+static std::unique_ptr<void, decltype(free)*> allocAligned(size_t alignment, size_t size)
+{
+ void *ptr = nullptr;
+ (void)posix_memalign(&ptr, alignment, size);
+ return std::unique_ptr<void, decltype(free)*>(ptr, free);
+}
+
+AudioFlinger::RecordThread::PassthruPatchRecord::PassthruPatchRecord(
+ RecordThread *recordThread,
+ uint32_t sampleRate,
+ audio_channel_mask_t channelMask,
+ audio_format_t format,
+ size_t frameCount,
+ audio_input_flags_t flags)
+ : PatchRecord(recordThread, sampleRate, channelMask, format, frameCount,
+ nullptr /*buffer*/, 0 /*bufferSize*/, flags),
+ mPatchRecordAudioBufferProvider(*this),
+ mSinkBuffer(allocAligned(32, mFrameCount * mFrameSize)),
+ mStubBuffer(allocAligned(32, mFrameCount * mFrameSize))
+{
+ memset(mStubBuffer.get(), 0, mFrameCount * mFrameSize);
+}
+
+sp<StreamInHalInterface> AudioFlinger::RecordThread::PassthruPatchRecord::obtainStream(
+ sp<ThreadBase>* thread)
+{
+ *thread = mThread.promote();
+ if (!*thread) return nullptr;
+ RecordThread *recordThread = static_cast<RecordThread*>((*thread).get());
+ Mutex::Autolock _l(recordThread->mLock);
+ return recordThread->mInput ? recordThread->mInput->stream : nullptr;
+}
+
+// PatchProxyBufferProvider methods are called on DirectOutputThread
+status_t AudioFlinger::RecordThread::PassthruPatchRecord::obtainBuffer(
+ Proxy::Buffer* buffer, const struct timespec* timeOut)
+{
+ if (mUnconsumedFrames) {
+ buffer->mFrameCount = std::min(buffer->mFrameCount, mUnconsumedFrames);
+ // mUnconsumedFrames is decreased in releaseBuffer to use actual frame consumption figure.
+ return PatchRecord::obtainBuffer(buffer, timeOut);
+ }
+
+ // Otherwise, execute a read from HAL and write into the buffer.
+ nsecs_t startTimeNs = 0;
+ if (timeOut && (timeOut->tv_sec != 0 || timeOut->tv_nsec != 0) && timeOut->tv_sec != INT_MAX) {
+ // Will need to correct timeOut by elapsed time.
+ startTimeNs = systemTime();
+ }
+ const size_t framesToRead = std::min(buffer->mFrameCount, mFrameCount);
+ buffer->mFrameCount = 0;
+ buffer->mRaw = nullptr;
+ sp<ThreadBase> thread;
+ sp<StreamInHalInterface> stream = obtainStream(&thread);
+ if (!stream) return NO_INIT; // If there is no stream, RecordThread is not reading.
+
+ status_t result = NO_ERROR;
+ size_t bytesRead = 0;
+ {
+ ATRACE_NAME("read");
+ result = stream->read(mSinkBuffer.get(), framesToRead * mFrameSize, &bytesRead);
+ if (result != NO_ERROR) goto stream_error;
+ if (bytesRead == 0) return NO_ERROR;
+ }
+
+ {
+ std::lock_guard<std::mutex> lock(mReadLock);
+ mReadBytes += bytesRead;
+ mReadError = NO_ERROR;
+ }
+ mReadCV.notify_one();
+ // writeFrames handles wraparound and should write all the provided frames.
+ // If it couldn't, there is something wrong with the client/server buffer of the software patch.
+ buffer->mFrameCount = writeFrames(
+ &mPatchRecordAudioBufferProvider,
+ mSinkBuffer.get(), bytesRead / mFrameSize, mFrameSize);
+ ALOGW_IF(buffer->mFrameCount < bytesRead / mFrameSize,
+ "Lost %zu frames obtained from HAL", bytesRead / mFrameSize - buffer->mFrameCount);
+ mUnconsumedFrames = buffer->mFrameCount;
+ struct timespec newTimeOut;
+ if (startTimeNs) {
+ // Correct the timeout by elapsed time.
+ nsecs_t newTimeOutNs = audio_utils_ns_from_timespec(timeOut) - (systemTime() - startTimeNs);
+ if (newTimeOutNs < 0) newTimeOutNs = 0;
+ newTimeOut.tv_sec = newTimeOutNs / NANOS_PER_SECOND;
+ newTimeOut.tv_nsec = newTimeOutNs - newTimeOut.tv_sec * NANOS_PER_SECOND;
+ timeOut = &newTimeOut;
+ }
+ return PatchRecord::obtainBuffer(buffer, timeOut);
+
+stream_error:
+ stream->standby();
+ {
+ std::lock_guard<std::mutex> lock(mReadLock);
+ mReadError = result;
+ }
+ mReadCV.notify_one();
+ return result;
+}
+
+void AudioFlinger::RecordThread::PassthruPatchRecord::releaseBuffer(Proxy::Buffer* buffer)
+{
+ if (buffer->mFrameCount <= mUnconsumedFrames) {
+ mUnconsumedFrames -= buffer->mFrameCount;
+ } else {
+ ALOGW("Write side has consumed more frames than we had: %zu > %zu",
+ buffer->mFrameCount, mUnconsumedFrames);
+ mUnconsumedFrames = 0;
+ }
+ PatchRecord::releaseBuffer(buffer);
+}
+
+// AudioBufferProvider and Source methods are called on RecordThread
+// 'read' emulates actual audio data with 0's. This is OK as 'getNextBuffer'
+// and 'releaseBuffer' are stubbed out and ignore their input.
+// It's not possible to retrieve actual data here w/o blocking 'obtainBuffer'
+// until we copy it.
+status_t AudioFlinger::RecordThread::PassthruPatchRecord::read(
+ void* buffer, size_t bytes, size_t* read)
+{
+ bytes = std::min(bytes, mFrameCount * mFrameSize);
+ {
+ std::unique_lock<std::mutex> lock(mReadLock);
+ mReadCV.wait(lock, [&]{ return mReadError != NO_ERROR || mReadBytes != 0; });
+ if (mReadError != NO_ERROR) {
+ mLastReadFrames = 0;
+ return mReadError;
+ }
+ *read = std::min(bytes, mReadBytes);
+ mReadBytes -= *read;
+ }
+ mLastReadFrames = *read / mFrameSize;
+ memset(buffer, 0, *read);
+ return 0;
+}
+
+status_t AudioFlinger::RecordThread::PassthruPatchRecord::getCapturePosition(
+ int64_t* frames, int64_t* time)
+{
+ sp<ThreadBase> thread;
+ sp<StreamInHalInterface> stream = obtainStream(&thread);
+ return stream ? stream->getCapturePosition(frames, time) : NO_INIT;
+}
+
+status_t AudioFlinger::RecordThread::PassthruPatchRecord::standby()
+{
+ // RecordThread issues 'standby' command in two major cases:
+ // 1. Error on read--this case is handled in 'obtainBuffer'.
+ // 2. Track is stopping--as PassthruPatchRecord assumes continuous
+ // output, this can only happen when the software patch
+ // is being torn down. In this case, the RecordThread
+ // will terminate and close the HAL stream.
+ return 0;
+}
+
+// As the buffer gets filled in obtainBuffer, here we only simulate data consumption.
+status_t AudioFlinger::RecordThread::PassthruPatchRecord::getNextBuffer(
+ AudioBufferProvider::Buffer* buffer)
+{
+ buffer->frameCount = mLastReadFrames;
+ buffer->raw = buffer->frameCount != 0 ? mStubBuffer.get() : nullptr;
+ return NO_ERROR;
+}
+
+void AudioFlinger::RecordThread::PassthruPatchRecord::releaseBuffer(
+ AudioBufferProvider::Buffer* buffer)
+{
+ buffer->frameCount = 0;
+ buffer->raw = nullptr;
+}
+
// ----------------------------------------------------------------------------
#undef LOG_TAG
#define LOG_TAG "AF::MmapTrack"
diff --git a/services/audiopolicy/AudioPolicyInterface.h b/services/audiopolicy/AudioPolicyInterface.h
index 30f29d6..1fe60d4 100644
--- a/services/audiopolicy/AudioPolicyInterface.h
+++ b/services/audiopolicy/AudioPolicyInterface.h
@@ -17,8 +17,10 @@
#ifndef ANDROID_AUDIOPOLICY_INTERFACE_H
#define ANDROID_AUDIOPOLICY_INTERFACE_H
+#include <media/AudioDeviceTypeAddr.h>
#include <media/AudioSystem.h>
#include <media/AudioPolicy.h>
+#include <media/DeviceDescriptorBase.h>
#include <utils/String8.h>
namespace android {
@@ -269,6 +271,14 @@
virtual status_t getVolumeGroupFromAudioAttributes(const AudioAttributes &aa,
volume_group_t &volumeGroup) = 0;
+
+ virtual status_t setPreferredDeviceForStrategy(product_strategy_t strategy,
+ const AudioDeviceTypeAddr &device) = 0;
+
+ virtual status_t removePreferredDeviceForStrategy(product_strategy_t strategy) = 0;
+
+ virtual status_t getPreferredDeviceForStrategy(product_strategy_t strategy,
+ AudioDeviceTypeAddr &device) = 0;
};
@@ -296,8 +306,7 @@
virtual status_t openOutput(audio_module_handle_t module,
audio_io_handle_t *output,
audio_config_t *config,
- audio_devices_t *devices,
- const String8& address,
+ const sp<DeviceDescriptorBase>& device,
uint32_t *latencyMs,
audio_output_flags_t flags) = 0;
// creates a special output that is duplicated to the two outputs passed as arguments. The duplication is performed by
diff --git a/services/audiopolicy/TEST_MAPPING b/services/audiopolicy/TEST_MAPPING
index a94fd87..eb6c19e 100644
--- a/services/audiopolicy/TEST_MAPPING
+++ b/services/audiopolicy/TEST_MAPPING
@@ -2,9 +2,6 @@
"presubmit": [
{
"name": "audiopolicy_tests"
- },
- {
- "name": "systemaudio_tests"
}
]
}
diff --git a/services/audiopolicy/audio_policy.conf b/services/audiopolicy/audio_policy.conf
deleted file mode 100644
index 9b83fef..0000000
--- a/services/audiopolicy/audio_policy.conf
+++ /dev/null
@@ -1,145 +0,0 @@
-#
-# Template audio policy configuration file
-#
-
-# Global configuration section:
-# - before audio HAL version 3.0:
-# lists input and output devices always present on the device
-# as well as the output device selected by default.
-# Devices are designated by a string that corresponds to the enum in audio.h
-#
-# global_configuration {
-# attached_output_devices AUDIO_DEVICE_OUT_SPEAKER
-# default_output_device AUDIO_DEVICE_OUT_SPEAKER
-# attached_input_devices AUDIO_DEVICE_IN_BUILTIN_MIC|AUDIO_DEVICE_IN_REMOTE_SUBMIX
-# }
-#
-# - after and including audio HAL 3.0 the global_configuration section is included in each
-# hardware module section.
-# it also includes the audio HAL version of this hw module:
-# global_configuration {
-# ...
-# audio_hal_version <major.minor> # audio HAL version in e.g. 3.0
-# }
-# other attributes (attached devices, default device) have to be included in the
-# global_configuration section of each hardware module
-
-
-# audio hardware module section: contains descriptors for all audio hw modules present on the
-# device. Each hw module node is named after the corresponding hw module library base name.
-# For instance, "primary" corresponds to audio.primary.<device>.so.
-# The "primary" module is mandatory and must include at least one output with
-# AUDIO_OUTPUT_FLAG_PRIMARY flag.
-# Each module descriptor contains one or more output profile descriptors and zero or more
-# input profile descriptors. Each profile lists all the parameters supported by a given output
-# or input stream category.
-# The "channel_masks", "formats", "devices" and "flags" are specified using strings corresponding
-# to enums in audio.h and audio_policy.h. They are concatenated by use of "|" without space or "\n".
-#
-# For audio HAL version posterior to 3.0 the following sections or sub sections can be present in
-# a hw module section:
-# - A "global_configuration" section: see above
-# - Optionally a "devices" section:
-# This section contains descriptors for audio devices with attributes like an address or a
-# gain controller. The syntax for the devices section and device descriptor is as follows:
-# devices {
-# <device name> { # <device name>: any string without space
-# type <device type> # <device type> e.g. AUDIO_DEVICE_OUT_SPEAKER
-# address <address> # optional: device address, char string less than 64 in length
-# }
-# }
-# - one or more "gains" sections can be present in a device descriptor section.
-# If present, they describe the capabilities of gain controllers attached to this input or
-# output device. e.g. :
-# <device name> { # <device name>: any string without space
-# type <device type> # <device type> e.g. AUDIO_DEVICE_OUT_SPEAKER
-# address <address> # optional: device address, char string less than 64 in length
-# gains {
-# <gain name> {
-# mode <gain modes supported> # e.g. AUDIO_GAIN_MODE_CHANNELS
-# channel_mask <controlled channels> # needed if mode AUDIO_GAIN_MODE_CHANNELS
-# min_value_mB <min value in millibel>
-# max_value_mB <max value in millibel>
-# default_value_mB <default value in millibel>
-# step_value_mB <step value in millibel>
-# min_ramp_ms <min duration in ms> # needed if mode AUDIO_GAIN_MODE_RAMP
-# max_ramp_ms <max duration ms> # needed if mode AUDIO_GAIN_MODE_RAMP
-# }
-# }
-# }
-# - when a device descriptor is present, output and input profiles can refer to this device by
-# its name in their "devices" section instead of specifying a device type. e.g. :
-# outputs {
-# primary {
-# sampling_rates 44100
-# channel_masks AUDIO_CHANNEL_OUT_STEREO
-# formats AUDIO_FORMAT_PCM_16_BIT
-# devices <device name>
-# flags AUDIO_OUTPUT_FLAG_PRIMARY
-# }
-# }
-# sample audio_policy.conf file below
-
-audio_hw_modules {
- primary {
- global_configuration {
- attached_output_devices AUDIO_DEVICE_OUT_SPEAKER
- default_output_device AUDIO_DEVICE_OUT_SPEAKER
- attached_input_devices AUDIO_DEVICE_IN_BUILTIN_MIC
- audio_hal_version 3.0
- }
- devices {
- speaker {
- type AUDIO_DEVICE_OUT_SPEAKER
- gains {
- gain_1 {
- mode AUDIO_GAIN_MODE_JOINT
- min_value_mB -8400
- max_value_mB 4000
- default_value_mB 0
- step_value_mB 100
- }
- }
- }
- }
- outputs {
- primary {
- sampling_rates 48000
- channel_masks AUDIO_CHANNEL_OUT_STEREO
- formats AUDIO_FORMAT_PCM_16_BIT
- devices speaker
- flags AUDIO_OUTPUT_FLAG_PRIMARY
- }
- }
- inputs {
- primary {
- sampling_rates 8000|16000
- channel_masks AUDIO_CHANNEL_IN_MONO
- formats AUDIO_FORMAT_PCM_16_BIT
- devices AUDIO_DEVICE_IN_BUILTIN_MIC
- }
- }
- }
- r_submix {
- global_configuration {
- attached_input_devices AUDIO_DEVICE_IN_REMOTE_SUBMIX
- audio_hal_version 2.0
- }
- outputs {
- submix {
- sampling_rates 48000
- channel_masks AUDIO_CHANNEL_OUT_STEREO
- formats AUDIO_FORMAT_PCM_16_BIT
- devices AUDIO_DEVICE_OUT_REMOTE_SUBMIX
- }
- }
- inputs {
- submix {
- sampling_rates 48000
- channel_masks AUDIO_CHANNEL_IN_STEREO
- formats AUDIO_FORMAT_PCM_16_BIT
- devices AUDIO_DEVICE_IN_REMOTE_SUBMIX
- }
- }
- }
-}
diff --git a/services/audiopolicy/common/Android.bp b/services/audiopolicy/common/Android.bp
index a925b9a..6e0d2f6 100644
--- a/services/audiopolicy/common/Android.bp
+++ b/services/audiopolicy/common/Android.bp
@@ -1,4 +1,7 @@
cc_library_headers {
name: "libaudiopolicycommon",
+ header_libs: [
+ "libaudiofoundation_headers",
+ ],
export_include_dirs: ["include"],
}
diff --git a/services/audiopolicy/common/include/Volume.h b/services/audiopolicy/common/include/Volume.h
index 1dbd1eb..7c8ce83 100644
--- a/services/audiopolicy/common/include/Volume.h
+++ b/services/audiopolicy/common/include/Volume.h
@@ -17,10 +17,13 @@
#pragma once
#include <media/AudioCommonTypes.h>
+#include <media/AudioContainers.h>
#include <system/audio.h>
#include <utils/Log.h>
#include <math.h>
+#include "policy.h"
+
namespace android {
/**
@@ -82,43 +85,26 @@
*
* @return subset of device required to limit the number of volume category per device
*/
- static audio_devices_t getDeviceForVolume(audio_devices_t device)
+ static audio_devices_t getDeviceForVolume(const android::DeviceTypeSet& deviceTypes)
{
- if (device == AUDIO_DEVICE_NONE) {
+ if (deviceTypes.empty()) {
// this happens when forcing a route update and no track is active on an output.
// In this case the returned category is not important.
- device = AUDIO_DEVICE_OUT_SPEAKER;
- } else if (popcount(device) > 1) {
- // Multiple device selection is either:
- // - speaker + one other device: give priority to speaker in this case.
- // - one A2DP device + another device: happens with duplicated output. In this case
- // retain the device on the A2DP output as the other must not correspond to an active
- // selection if not the speaker.
- // - HDMI-CEC system audio mode only output: give priority to available item in order.
- if (device & AUDIO_DEVICE_OUT_SPEAKER) {
- device = AUDIO_DEVICE_OUT_SPEAKER;
- } else if (device & AUDIO_DEVICE_OUT_SPEAKER_SAFE) {
- device = AUDIO_DEVICE_OUT_SPEAKER_SAFE;
- } else if (device & AUDIO_DEVICE_OUT_HDMI_ARC) {
- device = AUDIO_DEVICE_OUT_HDMI_ARC;
- } else if (device & AUDIO_DEVICE_OUT_AUX_LINE) {
- device = AUDIO_DEVICE_OUT_AUX_LINE;
- } else if (device & AUDIO_DEVICE_OUT_SPDIF) {
- device = AUDIO_DEVICE_OUT_SPDIF;
- } else {
- device = (audio_devices_t)(device & AUDIO_DEVICE_OUT_ALL_A2DP);
- }
+ return AUDIO_DEVICE_OUT_SPEAKER;
}
+ audio_devices_t deviceType = apm_extract_one_audio_device(deviceTypes);
+
/*SPEAKER_SAFE is an alias of SPEAKER for purposes of volume control*/
- if (device == AUDIO_DEVICE_OUT_SPEAKER_SAFE)
- device = AUDIO_DEVICE_OUT_SPEAKER;
+ if (deviceType == AUDIO_DEVICE_OUT_SPEAKER_SAFE) {
+ deviceType = AUDIO_DEVICE_OUT_SPEAKER;
+ }
- ALOGW_IF(popcount(device) != 1,
- "getDeviceForVolume() invalid device combination: %08x",
- device);
+ ALOGW_IF(deviceType == AUDIO_DEVICE_NONE,
+ "getDeviceForVolume() invalid device combination: %s, returning AUDIO_DEVICE_NONE",
+ android::dumpDeviceTypes(deviceTypes).c_str());
- return device;
+ return deviceType;
}
/**
@@ -128,9 +114,9 @@
*
* @return device category.
*/
- static device_category getDeviceCategory(audio_devices_t device)
+ static device_category getDeviceCategory(const android::DeviceTypeSet& deviceTypes)
{
- switch(getDeviceForVolume(device)) {
+ switch(getDeviceForVolume(deviceTypes)) {
case AUDIO_DEVICE_OUT_EARPIECE:
return DEVICE_CATEGORY_EARPIECE;
case AUDIO_DEVICE_OUT_WIRED_HEADSET:
diff --git a/services/audiopolicy/common/include/policy.h b/services/audiopolicy/common/include/policy.h
index 605fc1c..0537365 100644
--- a/services/audiopolicy/common/include/policy.h
+++ b/services/audiopolicy/common/include/policy.h
@@ -19,6 +19,8 @@
#include <system/audio.h>
#include <vector>
+#include <media/AudioContainers.h>
+
namespace android {
using StreamTypeVector = std::vector<audio_stream_type_t>;
@@ -43,14 +45,6 @@
#define MAX_MIXER_CHANNEL_COUNT FCC_8
/**
- * A device mask for all audio input and output devices where matching inputs/outputs on device
- * type alone is not enough: the address must match too
- */
-#define APM_AUDIO_DEVICE_OUT_MATCH_ADDRESS_ALL (AUDIO_DEVICE_OUT_REMOTE_SUBMIX|AUDIO_DEVICE_OUT_BUS)
-
-#define APM_AUDIO_DEVICE_IN_MATCH_ADDRESS_ALL (AUDIO_DEVICE_IN_REMOTE_SUBMIX|AUDIO_DEVICE_IN_BUS)
-
-/**
* Alias to AUDIO_DEVICE_OUT_DEFAULT defined for clarification when this value is used by volume
* control APIs (e.g setStreamVolumeIndex().
*/
@@ -71,6 +65,34 @@
}
/**
+ * Check whether the output device type is one
+ * where addresses are used to distinguish between one connected device and another
+ *
+ * @param[in] device to consider
+ *
+ * @return true if the device needs distinguish on address, false otherwise..
+ */
+static inline bool apm_audio_out_device_distinguishes_on_address(audio_devices_t device)
+{
+ return device == AUDIO_DEVICE_OUT_REMOTE_SUBMIX ||
+ device == AUDIO_DEVICE_OUT_BUS;
+}
+
+/**
+ * Check whether the input device type is one
+ * where addresses are used to distinguish between one connected device and another
+ *
+ * @param[in] device to consider
+ *
+ * @return true if the device needs distinguish on address, false otherwise..
+ */
+static inline bool apm_audio_in_device_distinguishes_on_address(audio_devices_t device)
+{
+ return device == AUDIO_DEVICE_IN_REMOTE_SUBMIX ||
+ device == AUDIO_DEVICE_IN_BUS;
+}
+
+/**
* Check whether the device type is one
* where addresses are used to distinguish between one connected device and another
*
@@ -80,10 +102,8 @@
*/
static inline bool device_distinguishes_on_address(audio_devices_t device)
{
- return (((device & AUDIO_DEVICE_BIT_IN) != 0) &&
- ((~AUDIO_DEVICE_BIT_IN & device & APM_AUDIO_DEVICE_IN_MATCH_ADDRESS_ALL) != 0)) ||
- (((device & AUDIO_DEVICE_BIT_IN) == 0) &&
- ((device & APM_AUDIO_DEVICE_OUT_MATCH_ADDRESS_ALL) != 0));
+ return apm_audio_in_device_distinguishes_on_address(device) ||
+ apm_audio_out_device_distinguishes_on_address(device);
}
/**
@@ -95,10 +115,7 @@
*/
static inline bool device_has_encoding_capability(audio_devices_t device)
{
- if (device & AUDIO_DEVICE_OUT_ALL_A2DP) {
- return true;
- }
- return false;
+ return audio_is_a2dp_out_device(device);
}
/**
@@ -184,3 +201,43 @@
{
return hasStream(streams, AUDIO_STREAM_VOICE_CALL);
}
+
+/**
+ * @brief extract one device relevant from multiple device selection
+ * @param deviceTypes collection of audio device type
+ * @return the device type that is selected
+ */
+static inline audio_devices_t apm_extract_one_audio_device(
+ const android::DeviceTypeSet& deviceTypes) {
+ if (deviceTypes.empty()) {
+ return AUDIO_DEVICE_NONE;
+ } else if (deviceTypes.size() == 1) {
+ return *(deviceTypes.begin());
+ } else {
+ // Multiple device selection is either:
+ // - speaker + one other device: give priority to speaker in this case.
+ // - one A2DP device + another device: happens with duplicated output. In this case
+ // retain the device on the A2DP output as the other must not correspond to an active
+ // selection if not the speaker.
+ // - HDMI-CEC system audio mode only output: give priority to available item in order.
+ if (deviceTypes.count(AUDIO_DEVICE_OUT_SPEAKER) != 0) {
+ return AUDIO_DEVICE_OUT_SPEAKER;
+ } else if (deviceTypes.count(AUDIO_DEVICE_OUT_SPEAKER_SAFE) != 0) {
+ return AUDIO_DEVICE_OUT_SPEAKER_SAFE;
+ } else if (deviceTypes.count(AUDIO_DEVICE_OUT_HDMI_ARC) != 0) {
+ return AUDIO_DEVICE_OUT_HDMI_ARC;
+ } else if (deviceTypes.count(AUDIO_DEVICE_OUT_AUX_LINE) != 0) {
+ return AUDIO_DEVICE_OUT_AUX_LINE;
+ } else if (deviceTypes.count(AUDIO_DEVICE_OUT_SPDIF) != 0) {
+ return AUDIO_DEVICE_OUT_SPDIF;
+ } else {
+ std::vector<audio_devices_t> a2dpDevices = android::Intersection(
+ deviceTypes, android::getAudioDeviceOutAllA2dpSet());
+ if (a2dpDevices.empty() || a2dpDevices.size() > 1) {
+ ALOGW("%s invalid device combination: %s",
+ __func__, android::dumpDeviceTypes(deviceTypes).c_str());
+ }
+ return a2dpDevices.empty() ? AUDIO_DEVICE_NONE : a2dpDevices[0];
+ }
+ }
+}
\ No newline at end of file
diff --git a/services/audiopolicy/common/managerdefinitions/Android.bp b/services/audiopolicy/common/managerdefinitions/Android.bp
index f02f3cf..fad3c5b 100644
--- a/services/audiopolicy/common/managerdefinitions/Android.bp
+++ b/services/audiopolicy/common/managerdefinitions/Android.bp
@@ -3,24 +3,24 @@
srcs: [
"src/AudioCollections.cpp",
- "src/AudioGain.cpp",
"src/AudioInputDescriptor.cpp",
"src/AudioOutputDescriptor.cpp",
"src/AudioPatch.cpp",
"src/AudioPolicyMix.cpp",
- "src/AudioPort.cpp",
- "src/AudioProfile.cpp",
+ "src/AudioProfileVectorHelper.cpp",
"src/AudioRoute.cpp",
"src/ClientDescriptor.cpp",
"src/DeviceDescriptor.cpp",
"src/EffectDescriptor.cpp",
"src/HwModule.cpp",
"src/IOProfile.cpp",
+ "src/PolicyAudioPort.cpp",
"src/Serializer.cpp",
"src/SoundTriggerSession.cpp",
"src/TypeConverter.cpp",
],
shared_libs: [
+ "libaudiofoundation",
"libcutils",
"libhidlbase",
"liblog",
@@ -28,7 +28,10 @@
"libutils",
"libxml2",
],
- export_shared_lib_headers: ["libmedia"],
+ export_shared_lib_headers: [
+ "libaudiofoundation",
+ "libmedia",
+ ],
static_libs: [
"libaudioutils",
],
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioCollections.h b/services/audiopolicy/common/managerdefinitions/include/AudioCollections.h
index a948ea9..b692592 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioCollections.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioCollections.h
@@ -25,20 +25,15 @@
namespace android {
-class AudioPort;
+class PolicyAudioPort;
class AudioRoute;
-class AudioPortVector : public Vector<sp<AudioPort> >
-{
-public:
- sp<AudioPort> findByTagName(const String8 &tagName) const;
-};
+using PolicyAudioPortVector = Vector<sp<PolicyAudioPort>>;
+using AudioRouteVector = Vector<sp<AudioRoute>>;
+sp<PolicyAudioPort> findByTagName(const PolicyAudioPortVector& policyAudioPortVector,
+ const std::string &tagName);
-class AudioRouteVector : public Vector<sp<AudioRoute> >
-{
-public:
- void dump(String8 *dst, int spaces) const;
-};
+void dumpAudioRouteVector(const AudioRouteVector& audioRouteVector, String8 *dst, int spaces);
} // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h
index 37f9d14..ec82873 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h
@@ -21,11 +21,11 @@
#include <utils/SortedVector.h>
#include <utils/KeyedVector.h>
#include "AudioIODescriptorInterface.h"
-#include "AudioPort.h"
#include "ClientDescriptor.h"
#include "DeviceDescriptor.h"
#include "EffectDescriptor.h"
#include "IOProfile.h"
+#include "PolicyAudioPort.h"
namespace android {
@@ -34,13 +34,17 @@
// descriptor for audio inputs. Used to maintain current configuration of each opened audio input
// and keep track of the usage of this input.
-class AudioInputDescriptor: public AudioPortConfig, public AudioIODescriptorInterface
- , public ClientMapHandler<RecordClientDescriptor>
+class AudioInputDescriptor: public AudioPortConfig,
+ public PolicyAudioPortConfig,
+ public AudioIODescriptorInterface,
+ public ClientMapHandler<RecordClientDescriptor>
{
public:
- explicit AudioInputDescriptor(const sp<IOProfile>& profile,
- AudioPolicyClientInterface *clientInterface);
- audio_port_handle_t getId() const;
+ AudioInputDescriptor(const sp<IOProfile>& profile,
+ AudioPolicyClientInterface *clientInterface);
+
+ virtual ~AudioInputDescriptor() = default;
+
audio_module_handle_t getModuleHandle() const;
audio_devices_t getDeviceType() const { return (mDevice != nullptr) ?
@@ -56,9 +60,18 @@
wp<AudioPolicyMix> mPolicyMix; // non NULL when used by a dynamic policy
const sp<IOProfile> mProfile; // I/O profile this output derives from
+ // PolicyAudioPortConfig
+ virtual sp<PolicyAudioPort> getPolicyAudioPort() const {
+ return mProfile;
+ }
+
+ // AudioPortConfig
+ virtual status_t applyAudioPortConfig(const struct audio_port_config *config,
+ struct audio_port_config *backupConfig = NULL);
virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
const struct audio_port_config *srcConfig = NULL) const;
virtual sp<AudioPort> getAudioPort() const { return mProfile; }
+
void toAudioPort(struct audio_port *port) const;
void setPreemptedSessions(const SortedVector<audio_session_t>& sessions);
SortedVector<audio_session_t> getPreemptedSessions() const;
@@ -111,7 +124,6 @@
void updateClientRecordingConfiguration(int event, const sp<RecordClientDescriptor>& client);
audio_patch_handle_t mPatchHandle = AUDIO_PATCH_HANDLE_NONE;
- audio_port_handle_t mId = AUDIO_PORT_HANDLE_NONE;
sp<DeviceDescriptor> mDevice = nullptr; /**< current device this input is routed to */
// Because a preemptible capture session can preempt another one, we end up in an endless loop
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
index cd54085..41f7dfc 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
@@ -21,14 +21,15 @@
#include <sys/types.h>
+#include <media/AudioContainers.h>
#include <utils/Errors.h>
#include <utils/Timers.h>
#include <utils/KeyedVector.h>
#include <system/audio.h>
#include "AudioIODescriptorInterface.h"
-#include "AudioPort.h"
#include "ClientDescriptor.h"
#include "DeviceDescriptor.h"
+#include "PolicyAudioPort.h"
#include <vector>
namespace android {
@@ -138,27 +139,28 @@
// descriptor for audio outputs. Used to maintain current configuration of each opened audio output
// and keep track of the usage of this output by each audio stream type.
-class AudioOutputDescriptor: public AudioPortConfig, public AudioIODescriptorInterface
- , public ClientMapHandler<TrackClientDescriptor>
+class AudioOutputDescriptor: public AudioPortConfig,
+ public PolicyAudioPortConfig,
+ public AudioIODescriptorInterface,
+ public ClientMapHandler<TrackClientDescriptor>
{
public:
- AudioOutputDescriptor(const sp<AudioPort>& port,
+ AudioOutputDescriptor(const sp<PolicyAudioPort>& policyAudioPort,
AudioPolicyClientInterface *clientInterface);
virtual ~AudioOutputDescriptor() {}
void dump(String8 *dst) const override;
void log(const char* indent);
- audio_port_handle_t getId() const;
virtual DeviceVector devices() const { return mDevices; }
bool sharesHwModuleWith(const sp<AudioOutputDescriptor>& outputDesc);
virtual DeviceVector supportedDevices() const { return mDevices; }
virtual bool isDuplicated() const { return false; }
virtual uint32_t latency() { return 0; }
- virtual bool isFixedVolume(audio_devices_t device);
+ virtual bool isFixedVolume(const DeviceTypeSet& deviceTypes);
virtual bool setVolume(float volumeDb,
VolumeSource volumeSource, const StreamTypeVector &streams,
- audio_devices_t device,
+ const DeviceTypeSet& deviceTypes,
uint32_t delayMs,
bool force);
@@ -245,9 +247,19 @@
mRoutingActivities[ps].setMutedByDevice(isMuted);
}
+ // PolicyAudioPortConfig
+ virtual sp<PolicyAudioPort> getPolicyAudioPort() const
+ {
+ return mPolicyAudioPort;
+ }
+
+ // AudioPortConfig
+ virtual status_t applyAudioPortConfig(const struct audio_port_config *config,
+ struct audio_port_config *backupConfig = NULL);
virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
const struct audio_port_config *srcConfig = NULL) const;
- virtual sp<AudioPort> getAudioPort() const { return mPort; }
+ virtual sp<AudioPort> getAudioPort() const { return mPolicyAudioPort->asAudioPort(); }
+
virtual void toAudioPort(struct audio_port *port) const;
audio_module_handle_t getModuleHandle() const;
@@ -289,11 +301,10 @@
wp<AudioPolicyMix> mPolicyMix; // non NULL when used by a dynamic policy
protected:
- const sp<AudioPort> mPort;
+ const sp<PolicyAudioPort> mPolicyAudioPort;
AudioPolicyClientInterface * const mClientInterface;
uint32_t mGlobalActiveCount = 0; // non-client-specific active count
audio_patch_handle_t mPatchHandle = AUDIO_PATCH_HANDLE_NONE;
- audio_port_handle_t mId = AUDIO_PORT_HANDLE_NONE;
// The ActiveClients shows the clients that contribute to the @VolumeSource counts
// and may include upstream clients from a duplicating thread.
@@ -319,10 +330,10 @@
void setDevices(const DeviceVector &devices) { mDevices = devices; }
bool sharesHwModuleWith(const sp<SwAudioOutputDescriptor>& outputDesc);
virtual DeviceVector supportedDevices() const;
- virtual bool deviceSupportsEncodedFormats(audio_devices_t device);
+ virtual bool devicesSupportEncodedFormats(const DeviceTypeSet& deviceTypes);
virtual uint32_t latency();
virtual bool isDuplicated() const { return (mOutput1 != NULL && mOutput2 != NULL); }
- virtual bool isFixedVolume(audio_devices_t device);
+ virtual bool isFixedVolume(const DeviceTypeSet& deviceTypes);
sp<SwAudioOutputDescriptor> subOutput1() { return mOutput1; }
sp<SwAudioOutputDescriptor> subOutput2() { return mOutput2; }
void setClientActive(const sp<TrackClientDescriptor>& client, bool active) override;
@@ -334,7 +345,7 @@
}
virtual bool setVolume(float volumeDb,
VolumeSource volumeSource, const StreamTypeVector &streams,
- audio_devices_t device,
+ const DeviceTypeSet& device,
uint32_t delayMs,
bool force);
@@ -408,7 +419,7 @@
virtual bool setVolume(float volumeDb,
VolumeSource volumeSource, const StreamTypeVector &streams,
- audio_devices_t device,
+ const DeviceTypeSet& deviceTypes,
uint32_t delayMs,
bool force);
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPatch.h b/services/audiopolicy/common/managerdefinitions/include/AudioPatch.h
index 0843fea..a5de655 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioPatch.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioPatch.h
@@ -31,12 +31,24 @@
public:
AudioPatch(const struct audio_patch *patch, uid_t uid);
+ audio_patch_handle_t getHandle() const { return mHandle; }
+
+ audio_patch_handle_t getAfHandle() const { return mAfPatchHandle; }
+
+ void setAfHandle(audio_patch_handle_t afHandle) { mAfPatchHandle = afHandle; }
+
+ uid_t getUid() const { return mUid; }
+
+ void setUid(uid_t uid) { mUid = uid; }
+
void dump(String8 *dst, int spaces, int index) const;
- audio_patch_handle_t mHandle;
struct audio_patch mPatch;
+
+private:
+ const audio_patch_handle_t mHandle;
uid_t mUid;
- audio_patch_handle_t mAfPatchHandle;
+ audio_patch_handle_t mAfPatchHandle = AUDIO_PATCH_HANDLE_NONE;
};
class AudioPatchCollection : public DefaultKeyedVector<audio_patch_handle_t, sp<AudioPatch> >
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h
index 2264d8f..56596f5 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h
@@ -19,17 +19,17 @@
#include <unordered_map>
#include <unordered_set>
-#include <AudioGain.h>
-#include <AudioPort.h>
#include <AudioPatch.h>
#include <DeviceDescriptor.h>
#include <IOProfile.h>
#include <HwModule.h>
+#include <PolicyAudioPort.h>
#include <AudioInputDescriptor.h>
#include <AudioOutputDescriptor.h>
#include <AudioPolicyMix.h>
#include <EffectDescriptor.h>
#include <SoundTriggerSession.h>
+#include <media/AudioProfile.h>
namespace android {
@@ -40,7 +40,8 @@
DeviceVector &availableOutputDevices,
DeviceVector &availableInputDevices,
sp<DeviceDescriptor> &defaultOutputDevice)
- : mHwModules(hwModules),
+ : mEngineLibraryNameSuffix(kDefaultEngineLibraryNameSuffix),
+ mHwModules(hwModules),
mAvailableOutputDevices(availableOutputDevices),
mAvailableInputDevices(availableInputDevices),
mDefaultOutputDevice(defaultOutputDevice),
@@ -55,6 +56,14 @@
mSource = file;
}
+ const std::string& getEngineLibraryNameSuffix() const {
+ return mEngineLibraryNameSuffix;
+ }
+
+ void setEngineLibraryNameSuffix(const std::string& suffix) {
+ mEngineLibraryNameSuffix = suffix;
+ }
+
void setHwModules(const HwModuleCollection &hwModules)
{
mHwModules = hwModules;
@@ -108,10 +117,11 @@
void setDefault(void)
{
mSource = "AudioPolicyConfig::setDefault";
+ mEngineLibraryNameSuffix = kDefaultEngineLibraryNameSuffix;
mDefaultOutputDevice = new DeviceDescriptor(AUDIO_DEVICE_OUT_SPEAKER);
- mDefaultOutputDevice->addAudioProfile(AudioProfile::createFullDynamic());
+ mDefaultOutputDevice->addAudioProfile(AudioProfile::createFullDynamic(gDynamicFormat));
sp<DeviceDescriptor> defaultInputDevice = new DeviceDescriptor(AUDIO_DEVICE_IN_BUILTIN_MIC);
- defaultInputDevice->addAudioProfile(AudioProfile::createFullDynamic());
+ defaultInputDevice->addAudioProfile(AudioProfile::createFullDynamic(gDynamicFormat));
sp<AudioProfile> micProfile = new AudioProfile(
AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_IN_MONO, 8000);
defaultInputDevice->addAudioProfile(micProfile);
@@ -123,14 +133,14 @@
mDefaultOutputDevice->attach(module);
defaultInputDevice->attach(module);
- sp<OutputProfile> outProfile = new OutputProfile(String8("primary"));
+ sp<OutputProfile> outProfile = new OutputProfile("primary");
outProfile->addAudioProfile(
new AudioProfile(AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO, 44100));
outProfile->addSupportedDevice(mDefaultOutputDevice);
outProfile->setFlags(AUDIO_OUTPUT_FLAG_PRIMARY);
module->addOutputProfile(outProfile);
- sp<InputProfile> inProfile = new InputProfile(String8("primary"));
+ sp<InputProfile> inProfile = new InputProfile("primary");
inProfile->addAudioProfile(micProfile);
inProfile->addSupportedDevice(defaultInputDevice);
module->addInputProfile(inProfile);
@@ -167,7 +177,10 @@
}
private:
+ static const constexpr char* const kDefaultEngineLibraryNameSuffix = "default";
+
std::string mSource;
+ std::string mEngineLibraryNameSuffix;
HwModuleCollection &mHwModules; /**< Collection of Module, with Profiles, i.e. Mix Ports. */
DeviceVector &mAvailableOutputDevices;
DeviceVector &mAvailableInputDevices;
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h
index 094f506..fc79ab1 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h
@@ -18,6 +18,7 @@
#include "DeviceDescriptor.h"
#include <utils/RefBase.h>
+#include <media/AudioDeviceTypeAddr.h>
#include <media/AudioPolicy.h>
#include <utils/Vector.h>
#include <system/audio.h>
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPort.h b/services/audiopolicy/common/managerdefinitions/include/AudioPort.h
deleted file mode 100644
index d906f11..0000000
--- a/services/audiopolicy/common/managerdefinitions/include/AudioPort.h
+++ /dev/null
@@ -1,179 +0,0 @@
-/*
- * Copyright (C) 2015 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#pragma once
-
-#include "AudioCollections.h"
-#include "AudioProfile.h"
-#include "AudioGain.h"
-#include "HandleGenerator.h"
-#include <utils/String8.h>
-#include <utils/Vector.h>
-#include <utils/RefBase.h>
-#include <utils/Errors.h>
-#include <system/audio.h>
-#include <cutils/config_utils.h>
-
-namespace android {
-
-class HwModule;
-class AudioRoute;
-
-class AudioPort : public virtual RefBase, private HandleGenerator<audio_port_handle_t>
-{
-public:
- AudioPort(const String8& name, audio_port_type_t type, audio_port_role_t role) :
- mName(name), mType(type), mRole(role), mFlags(AUDIO_OUTPUT_FLAG_NONE) {}
-
- virtual ~AudioPort() {}
-
- void setName(const String8 &name) { mName = name; }
- const String8 &getName() const { return mName; }
-
- audio_port_type_t getType() const { return mType; }
- audio_port_role_t getRole() const { return mRole; }
-
- virtual const String8 getTagName() const = 0;
-
- void setGains(const AudioGains &gains) { mGains = gains; }
- const AudioGains &getGains() const { return mGains; }
-
- virtual void setFlags(uint32_t flags)
- {
- //force direct flag if offload flag is set: offloading implies a direct output stream
- // and all common behaviors are driven by checking only the direct flag
- // this should normally be set appropriately in the policy configuration file
- if (mRole == AUDIO_PORT_ROLE_SOURCE && (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
- flags |= AUDIO_OUTPUT_FLAG_DIRECT;
- }
- mFlags = flags;
- }
- uint32_t getFlags() const { return mFlags; }
-
- virtual void attach(const sp<HwModule>& module);
- virtual void detach();
- bool isAttached() { return mModule != 0; }
-
- // Audio port IDs are in a different namespace than AudioFlinger unique IDs
- static audio_port_handle_t getNextUniqueId();
-
- virtual void toAudioPort(struct audio_port *port) const;
-
- virtual void importAudioPort(const sp<AudioPort>& port, bool force = false);
-
- void addAudioProfile(const sp<AudioProfile> &profile) { mProfiles.add(profile); }
-
- void setAudioProfiles(const AudioProfileVector &profiles) { mProfiles = profiles; }
- AudioProfileVector &getAudioProfiles() { return mProfiles; }
-
- bool hasValidAudioProfile() const { return mProfiles.hasValidProfile(); }
-
- bool hasDynamicAudioProfile() const { return mProfiles.hasDynamicProfile(); }
-
- // searches for an exact match
- virtual status_t checkExactAudioProfile(const struct audio_port_config *config) const;
-
- // searches for a compatible match, currently implemented for input
- // parameters are input|output, returned value is the best match.
- status_t checkCompatibleAudioProfile(uint32_t &samplingRate,
- audio_channel_mask_t &channelMask,
- audio_format_t &format) const
- {
- return mProfiles.checkCompatibleProfile(samplingRate, channelMask, format, mType, mRole);
- }
-
- void clearAudioProfiles() { return mProfiles.clearProfiles(); }
-
- status_t checkGain(const struct audio_gain_config *gainConfig, int index) const;
-
- void pickAudioProfile(uint32_t &samplingRate,
- audio_channel_mask_t &channelMask,
- audio_format_t &format) const;
-
- static const audio_format_t sPcmFormatCompareTable[];
-
- static int compareFormats(audio_format_t format1, audio_format_t format2);
-
- // Used to select an audio HAL output stream with a sample format providing the
- // less degradation for a given AudioTrack sample format.
- static bool isBetterFormatMatch(audio_format_t newFormat,
- audio_format_t currentFormat,
- audio_format_t targetFormat);
- static uint32_t formatDistance(audio_format_t format1,
- audio_format_t format2);
- static const uint32_t kFormatDistanceMax = 4;
-
- audio_module_handle_t getModuleHandle() const;
- uint32_t getModuleVersionMajor() const;
- const char *getModuleName() const;
- sp<HwModule> getModule() const { return mModule; }
-
- bool useInputChannelMask() const
- {
- return ((mType == AUDIO_PORT_TYPE_DEVICE) && (mRole == AUDIO_PORT_ROLE_SOURCE)) ||
- ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SINK));
- }
-
- inline bool isDirectOutput() const
- {
- return (mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) &&
- (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD));
- }
-
- void addRoute(const sp<AudioRoute> &route) { mRoutes.add(route); }
- const AudioRouteVector &getRoutes() const { return mRoutes; }
-
- void dump(String8 *dst, int spaces, bool verbose = true) const;
-
- void log(const char* indent) const;
-
- AudioGains mGains; // gain controllers
-
-private:
- void pickChannelMask(audio_channel_mask_t &channelMask, const ChannelsVector &channelMasks) const;
- void pickSamplingRate(uint32_t &rate,const SampleRateVector &samplingRates) const;
-
- sp<HwModule> mModule; // audio HW module exposing this I/O stream
- String8 mName;
- audio_port_type_t mType;
- audio_port_role_t mRole;
- uint32_t mFlags; // attribute flags mask (e.g primary output, direct output...).
- AudioProfileVector mProfiles; // AudioProfiles supported by this port (format, Rates, Channels)
- AudioRouteVector mRoutes; // Routes involving this port
-};
-
-class AudioPortConfig : public virtual RefBase
-{
-public:
- status_t applyAudioPortConfig(const struct audio_port_config *config,
- struct audio_port_config *backupConfig = NULL);
- virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig = NULL) const = 0;
- virtual sp<AudioPort> getAudioPort() const = 0;
- virtual bool hasSameHwModuleAs(const sp<AudioPortConfig>& other) const {
- return (other != 0) && (other->getAudioPort() != 0) && (getAudioPort() != 0) &&
- (other->getAudioPort()->getModuleHandle() == getAudioPort()->getModuleHandle());
- }
- bool hasGainController(bool canUseForVolume = false) const;
-
- unsigned int mSamplingRate = 0u;
- audio_format_t mFormat = AUDIO_FORMAT_INVALID;
- audio_channel_mask_t mChannelMask = AUDIO_CHANNEL_NONE;
- struct audio_gain_config mGain = { .index = -1 };
- union audio_io_flags mFlags = { AUDIO_INPUT_FLAG_NONE };
-};
-
-} // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioProfile.h b/services/audiopolicy/common/managerdefinitions/include/AudioProfile.h
deleted file mode 100644
index b588d57..0000000
--- a/services/audiopolicy/common/managerdefinitions/include/AudioProfile.h
+++ /dev/null
@@ -1,180 +0,0 @@
-/*
- * Copyright (C) 2015 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#pragma once
-
-#include <vector>
-
-#include <system/audio.h>
-#include <utils/RefBase.h>
-#include <utils/SortedVector.h>
-#include <utils/String8.h>
-
-#include "policy.h"
-
-namespace android {
-
-typedef SortedVector<uint32_t> SampleRateVector;
-typedef Vector<audio_format_t> FormatVector;
-
-template <typename T>
-bool operator== (const SortedVector<T> &left, const SortedVector<T> &right)
-{
- if (left.size() != right.size()) {
- return false;
- }
- for (size_t index = 0; index < right.size(); index++) {
- if (left[index] != right[index]) {
- return false;
- }
- }
- return true;
-}
-
-template <typename T>
-bool operator!= (const SortedVector<T> &left, const SortedVector<T> &right)
-{
- return !(left == right);
-}
-
-class ChannelsVector : public SortedVector<audio_channel_mask_t>
-{
-public:
- ChannelsVector() = default;
- ChannelsVector(const ChannelsVector&) = default;
- ChannelsVector(const SortedVector<audio_channel_mask_t>& sv) :
- SortedVector<audio_channel_mask_t>(sv) {}
- ChannelsVector& operator=(const ChannelsVector&) = default;
-
- // Applies audio_channel_mask_out_to_in to all elements and returns the result.
- ChannelsVector asInMask() const;
- // Applies audio_channel_mask_in_to_out to all elements and returns the result.
- ChannelsVector asOutMask() const;
-};
-
-class AudioProfile : public virtual RefBase
-{
-public:
- static sp<AudioProfile> createFullDynamic();
-
- AudioProfile(audio_format_t format, audio_channel_mask_t channelMasks, uint32_t samplingRate);
- AudioProfile(audio_format_t format,
- const ChannelsVector &channelMasks,
- const SampleRateVector &samplingRateCollection);
-
- audio_format_t getFormat() const { return mFormat; }
- const ChannelsVector &getChannels() const { return mChannelMasks; }
- const SampleRateVector &getSampleRates() const { return mSamplingRates; }
- void setChannels(const ChannelsVector &channelMasks);
- void setSampleRates(const SampleRateVector &sampleRates);
-
- void clear();
- bool isValid() const { return hasValidFormat() && hasValidRates() && hasValidChannels(); }
- bool supportsChannels(audio_channel_mask_t channels) const
- {
- return mChannelMasks.indexOf(channels) >= 0;
- }
- bool supportsRate(uint32_t rate) const { return mSamplingRates.indexOf(rate) >= 0; }
-
- status_t checkExact(uint32_t rate, audio_channel_mask_t channels, audio_format_t format) const;
- status_t checkCompatibleChannelMask(audio_channel_mask_t channelMask,
- audio_channel_mask_t &updatedChannelMask,
- audio_port_type_t portType,
- audio_port_role_t portRole) const;
- status_t checkCompatibleSamplingRate(uint32_t samplingRate,
- uint32_t &updatedSamplingRate) const;
-
- bool hasValidFormat() const { return mFormat != AUDIO_FORMAT_DEFAULT; }
- bool hasValidRates() const { return !mSamplingRates.isEmpty(); }
- bool hasValidChannels() const { return !mChannelMasks.isEmpty(); }
-
- void setDynamicChannels(bool dynamic) { mIsDynamicChannels = dynamic; }
- bool isDynamicChannels() const { return mIsDynamicChannels; }
-
- void setDynamicRate(bool dynamic) { mIsDynamicRate = dynamic; }
- bool isDynamicRate() const { return mIsDynamicRate; }
-
- void setDynamicFormat(bool dynamic) { mIsDynamicFormat = dynamic; }
- bool isDynamicFormat() const { return mIsDynamicFormat; }
-
- bool isDynamic() { return mIsDynamicFormat || mIsDynamicChannels || mIsDynamicRate; }
-
- void dump(String8 *dst, int spaces) const;
-
-private:
- String8 mName;
- audio_format_t mFormat;
- ChannelsVector mChannelMasks;
- SampleRateVector mSamplingRates;
-
- bool mIsDynamicFormat = false;
- bool mIsDynamicChannels = false;
- bool mIsDynamicRate = false;
-};
-
-
-class AudioProfileVector : public Vector<sp<AudioProfile> >
-{
-public:
- ssize_t add(const sp<AudioProfile> &profile);
- // This API is intended to be used by the policy manager once retrieving capabilities
- // for a profile with dynamic format, rate and channels attributes
- ssize_t addProfileFromHal(const sp<AudioProfile> &profileToAdd);
-
- status_t checkExactProfile(uint32_t samplingRate, audio_channel_mask_t channelMask,
- audio_format_t format) const;
- status_t checkCompatibleProfile(uint32_t &samplingRate, audio_channel_mask_t &channelMask,
- audio_format_t &format,
- audio_port_type_t portType,
- audio_port_role_t portRole) const;
- void clearProfiles();
- // Assuming that this profile vector contains input profiles,
- // find the best matching config from 'outputProfiles', according to
- // the given preferences for audio formats and channel masks.
- // Note: std::vectors are used because specialized containers for formats
- // and channels can be sorted and use their own ordering.
- status_t findBestMatchingOutputConfig(const AudioProfileVector& outputProfiles,
- const std::vector<audio_format_t>& preferredFormats, // order: most pref -> least pref
- const std::vector<audio_channel_mask_t>& preferredOutputChannels,
- bool preferHigherSamplingRates,
- audio_config_base *bestOutputConfig) const;
-
- sp<AudioProfile> getFirstValidProfile() const;
- sp<AudioProfile> getFirstValidProfileFor(audio_format_t format) const;
- bool hasValidProfile() const { return getFirstValidProfile() != 0; }
-
- FormatVector getSupportedFormats() const;
- bool hasDynamicChannelsFor(audio_format_t format) const;
- bool hasDynamicFormat() const { return getProfileFor(gDynamicFormat) != 0; }
- bool hasDynamicProfile() const;
- bool hasDynamicRateFor(audio_format_t format) const;
-
- // One audio profile will be added for each format supported by Audio HAL
- void setFormats(const FormatVector &formats);
-
- void dump(String8 *dst, int spaces) const;
-
-private:
- sp<AudioProfile> getProfileFor(audio_format_t format) const;
- void setSampleRatesFor(const SampleRateVector &sampleRates, audio_format_t format);
- void setChannelsFor(const ChannelsVector &channelMasks, audio_format_t format);
-
- static int compareFormats(const sp<AudioProfile> *profile1, const sp<AudioProfile> *profile2);
-};
-
-bool operator == (const AudioProfile &left, const AudioProfile &right);
-
-} // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioProfileVectorHelper.h b/services/audiopolicy/common/managerdefinitions/include/AudioProfileVectorHelper.h
new file mode 100644
index 0000000..f84bda7
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioProfileVectorHelper.h
@@ -0,0 +1,67 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#pragma once
+
+#include <media/AudioProfile.h>
+#include <system/audio.h>
+
+namespace android {
+
+void sortAudioProfiles(AudioProfileVector &audioProfileVector);
+
+ssize_t addAudioProfileAndSort(AudioProfileVector &audioProfileVector,
+ const sp<AudioProfile> &profile);
+
+// One audio profile will be added for each format supported by Audio HAL
+void addProfilesForFormats(AudioProfileVector &audioProfileVector,
+ const FormatVector &formatVector);
+
+// This API is intended to be used by the policy manager once retrieving capabilities
+// for a profile with dynamic format, rate and channels attributes
+void addDynamicAudioProfileAndSort(AudioProfileVector &audioProfileVector,
+ const sp<AudioProfile> &profileToAdd);
+
+void appendAudioProfiles(AudioProfileVector &audioProfileVector,
+ const AudioProfileVector &audioProfileVectorToAppend);
+
+status_t checkExactProfile(const AudioProfileVector &audioProfileVector,
+ const uint32_t samplingRate,
+ audio_channel_mask_t channelMask,
+ audio_format_t format);
+
+status_t checkCompatibleProfile(const AudioProfileVector &audioProfileVector,
+ uint32_t &samplingRate,
+ audio_channel_mask_t &channelMask,
+ audio_format_t &format,
+ audio_port_type_t portType,
+ audio_port_role_t portRole);
+
+// Assuming that this profile vector contains input profiles,
+// find the best matching config from 'outputProfiles', according to
+// the given preferences for audio formats and channel masks.
+// Note: std::vectors are used because specialized containers for formats
+// and channels can be sorted and use their own ordering.
+status_t findBestMatchingOutputConfig(
+ const AudioProfileVector &audioProfileVector,
+ const AudioProfileVector &outputProfileVector,
+ const std::vector<audio_format_t> &preferredFormatVector, // order: most pref -> least pref
+ const std::vector<audio_channel_mask_t> &preferredOutputChannelVector,
+ bool preferHigherSamplingRates,
+ audio_config_base &bestOutputConfig);
+
+
+} // namespace android
\ No newline at end of file
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioRoute.h b/services/audiopolicy/common/managerdefinitions/include/AudioRoute.h
index 0357ff4..a7def3e 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioRoute.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioRoute.h
@@ -25,7 +25,7 @@
namespace android
{
-class AudioPort;
+class PolicyAudioPort;
class DeviceDescriptor;
typedef enum {
@@ -38,11 +38,11 @@
public:
explicit AudioRoute(audio_route_type_t type) : mType(type) {}
- void setSources(const AudioPortVector &sources) { mSources = sources; }
- const AudioPortVector &getSources() const { return mSources; }
+ void setSources(const PolicyAudioPortVector &sources) { mSources = sources; }
+ const PolicyAudioPortVector &getSources() const { return mSources; }
- void setSink(const sp<AudioPort> &sink) { mSink = sink; }
- const sp<AudioPort> &getSink() const { return mSink; }
+ void setSink(const sp<PolicyAudioPort> &sink) { mSink = sink; }
+ const sp<PolicyAudioPort> &getSink() const { return mSink; }
audio_route_type_t getType() const { return mType; }
@@ -57,13 +57,14 @@
* @return true if the audio route supports the connection between the sink and the source,
* false otherwise
*/
- bool supportsPatch(const sp<AudioPort> &srcPort, const sp<AudioPort> &dstPort) const;
+ bool supportsPatch(const sp<PolicyAudioPort> &srcPort,
+ const sp<PolicyAudioPort> &dstPort) const;
void dump(String8 *dst, int spaces) const;
private:
- AudioPortVector mSources;
- sp<AudioPort> mSink;
+ PolicyAudioPortVector mSources;
+ sp<PolicyAudioPort> mSink;
audio_route_type_t mType;
};
diff --git a/services/audiopolicy/common/managerdefinitions/include/ClientDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/ClientDescriptor.h
index 0d05a63..0c5d1d0 100644
--- a/services/audiopolicy/common/managerdefinitions/include/ClientDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/ClientDescriptor.h
@@ -183,13 +183,17 @@
{
public:
SourceClientDescriptor(audio_port_handle_t portId, uid_t uid, audio_attributes_t attributes,
- const sp<AudioPatch>& patchDesc, const sp<DeviceDescriptor>& srcDevice,
+ const struct audio_port_config &config,
+ const sp<DeviceDescriptor>& srcDevice,
audio_stream_type_t stream, product_strategy_t strategy,
VolumeSource volumeSource);
+
~SourceClientDescriptor() override = default;
- sp<AudioPatch> patchDesc() const { return mPatchDesc; }
- sp<DeviceDescriptor> srcDevice() const { return mSrcDevice; };
+ audio_patch_handle_t getPatchHandle() const { return mPatchHandle; }
+ void setPatchHandle(audio_patch_handle_t patchHandle) { mPatchHandle = patchHandle; }
+
+ sp<DeviceDescriptor> srcDevice() const { return mSrcDevice; }
wp<SwAudioOutputDescriptor> swOutput() const { return mSwOutput; }
void setSwOutput(const sp<SwAudioOutputDescriptor>& swOutput);
wp<HwAudioOutputDescriptor> hwOutput() const { return mHwOutput; }
@@ -199,7 +203,7 @@
void dump(String8 *dst, int spaces, int index) const override;
private:
- const sp<AudioPatch> mPatchDesc;
+ audio_patch_handle_t mPatchHandle = AUDIO_PATCH_HANDLE_NONE;
const sp<DeviceDescriptor> mSrcDevice;
wp<SwAudioOutputDescriptor> mSwOutput;
wp<HwAudioOutputDescriptor> mHwOutput;
diff --git a/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
index 33e506f..a6562d7 100644
--- a/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
@@ -16,7 +16,9 @@
#pragma once
-#include "AudioPort.h"
+#include "PolicyAudioPort.h"
+#include <media/AudioContainers.h>
+#include <media/DeviceDescriptorBase.h>
#include <utils/Errors.h>
#include <utils/String8.h>
#include <utils/SortedVector.h>
@@ -26,21 +28,26 @@
namespace android {
-class DeviceDescriptor : public AudioPort, public AudioPortConfig
+class DeviceDescriptor : public DeviceDescriptorBase,
+ public PolicyAudioPort, public PolicyAudioPortConfig
{
public:
// Note that empty name refers by convention to a generic device.
- explicit DeviceDescriptor(audio_devices_t type, const String8 &tagName = String8(""));
- DeviceDescriptor(audio_devices_t type, const FormatVector &encodedFormats,
- const String8 &tagName = String8(""));
+ explicit DeviceDescriptor(audio_devices_t type);
+ DeviceDescriptor(audio_devices_t type, const std::string &tagName,
+ const FormatVector &encodedFormats = FormatVector{});
+ DeviceDescriptor(audio_devices_t type, const std::string &tagName,
+ const std::string &address, const FormatVector &encodedFormats = FormatVector{});
+ DeviceDescriptor(const AudioDeviceTypeAddr &deviceTypeAddr, const std::string &tagName = "",
+ const FormatVector &encodedFormats = FormatVector{});
virtual ~DeviceDescriptor() {}
- virtual const String8 getTagName() const { return mTagName; }
+ virtual void addAudioProfile(const sp<AudioProfile> &profile) {
+ addAudioProfileAndSort(mProfiles, profile);
+ }
- audio_devices_t type() const { return mDeviceType; }
- String8 address() const { return mAddress; }
- void setAddress(const String8 &address) { mAddress = address; }
+ virtual const std::string getTagName() const { return mTagName; }
const FormatVector& encodedFormats() const { return mEncodedFormats; }
@@ -56,36 +63,42 @@
bool supportsFormat(audio_format_t format);
+ // PolicyAudioPortConfig
+ virtual sp<PolicyAudioPort> getPolicyAudioPort() const {
+ return static_cast<PolicyAudioPort*>(const_cast<DeviceDescriptor*>(this));
+ }
+
// AudioPortConfig
- virtual sp<AudioPort> getAudioPort() const { return (AudioPort*) this; }
+ virtual status_t applyAudioPortConfig(const struct audio_port_config *config,
+ struct audio_port_config *backupConfig = NULL);
virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
const struct audio_port_config *srcConfig = NULL) const;
- // AudioPort
+ // PolicyAudioPort
+ virtual sp<AudioPort> asAudioPort() const {
+ return static_cast<AudioPort*>(const_cast<DeviceDescriptor*>(this));
+ }
virtual void attach(const sp<HwModule>& module);
virtual void detach();
+ // AudioPort
virtual void toAudioPort(struct audio_port *port) const;
- virtual void importAudioPort(const sp<AudioPort>& port, bool force = false);
- audio_port_handle_t getId() const;
+ void importAudioPortAndPickAudioProfile(const sp<PolicyAudioPort>& policyPort,
+ bool force = false);
+
void dump(String8 *dst, int spaces, int index, bool verbose = true) const;
- void log() const;
- std::string toString() const;
private:
- String8 mAddress{""};
- String8 mTagName; // Unique human readable identifier for a device port found in conf file.
- audio_devices_t mDeviceType;
+ std::string mTagName; // Unique human readable identifier for a device port found in conf file.
FormatVector mEncodedFormats;
- audio_port_handle_t mId = AUDIO_PORT_HANDLE_NONE;
audio_format_t mCurrentEncodedFormat;
};
class DeviceVector : public SortedVector<sp<DeviceDescriptor> >
{
public:
- DeviceVector() : SortedVector(), mDeviceTypes(AUDIO_DEVICE_NONE) {}
+ DeviceVector() : SortedVector() {}
explicit DeviceVector(const sp<DeviceDescriptor>& item) : DeviceVector()
{
add(item);
@@ -97,13 +110,16 @@
void remove(const DeviceVector &devices);
ssize_t indexOf(const sp<DeviceDescriptor>& item) const;
- audio_devices_t types() const { return mDeviceTypes; }
+ DeviceTypeSet types() const { return mDeviceTypes; }
// If 'address' is empty and 'codec' is AUDIO_FORMAT_DEFAULT, a device with a non-empty
// address may be returned if there is no device with the specified 'type' and empty address.
sp<DeviceDescriptor> getDevice(audio_devices_t type, const String8 &address,
audio_format_t codec) const;
- DeviceVector getDevicesFromTypeMask(audio_devices_t types) const;
+ DeviceVector getDevicesFromTypes(const DeviceTypeSet& types) const;
+ DeviceVector getDevicesFromType(audio_devices_t type) const {
+ return getDevicesFromTypes({type});
+ }
/**
* @brief getDeviceFromId
@@ -112,9 +128,35 @@
* equal to AUDIO_PORT_HANDLE_NONE, it also returns a nullptr.
*/
sp<DeviceDescriptor> getDeviceFromId(audio_port_handle_t id) const;
- sp<DeviceDescriptor> getDeviceFromTagName(const String8 &tagName) const;
+ sp<DeviceDescriptor> getDeviceFromTagName(const std::string &tagName) const;
DeviceVector getDevicesFromHwModule(audio_module_handle_t moduleHandle) const;
- audio_devices_t getDeviceTypesFromHwModule(audio_module_handle_t moduleHandle) const;
+
+ DeviceVector getFirstDevicesFromTypes(std::vector<audio_devices_t> orderedTypes) const;
+ sp<DeviceDescriptor> getFirstExistingDevice(std::vector<audio_devices_t> orderedTypes) const;
+
+ // Return device descriptor that is used to open an input/output stream.
+ // Null pointer will be returned if
+ // 1) this collection is empty
+ // 2) the device descriptors are not the same category(input or output)
+ // 3) there are more than one device type for input case
+ // 4) the combination of all devices is invalid for selection
+ sp<DeviceDescriptor> getDeviceForOpening() const;
+
+ // If there are devices with the given type and the devices to add is not empty,
+ // remove all the devices with the given type and add all the devices to add.
+ void replaceDevicesByType(audio_devices_t typeToRemove, const DeviceVector &devicesToAdd);
+
+ bool containsDeviceAmongTypes(const DeviceTypeSet& deviceTypes) const {
+ return !Intersection(mDeviceTypes, deviceTypes).empty();
+ }
+
+ bool containsDeviceWithType(audio_devices_t deviceType) const {
+ return containsDeviceAmongTypes({deviceType});
+ }
+
+ bool onlyContainsDevicesWithType(audio_devices_t deviceType) const {
+ return isSingleDeviceType(mDeviceTypes, deviceType);
+ }
bool contains(const sp<DeviceDescriptor>& item) const { return indexOf(item) >= 0; }
@@ -196,7 +238,7 @@
{
for (const auto &device : *this) {
if (device->address() != "") {
- return device->address();
+ return String8(device->address().c_str());
}
}
return String8("");
@@ -208,7 +250,7 @@
private:
void refreshTypes();
- audio_devices_t mDeviceTypes;
+ DeviceTypeSet mDeviceTypes;
};
} // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/HwModule.h b/services/audiopolicy/common/managerdefinitions/include/HwModule.h
index eb34da4..23f0c9a 100644
--- a/services/audiopolicy/common/managerdefinitions/include/HwModule.h
+++ b/services/audiopolicy/common/managerdefinitions/include/HwModule.h
@@ -82,19 +82,19 @@
status_t addInputProfile(const sp<IOProfile> &profile);
status_t addProfile(const sp<IOProfile> &profile);
- status_t addOutputProfile(const String8& name, const audio_config_t *config,
+ status_t addOutputProfile(const std::string& name, const audio_config_t *config,
audio_devices_t device, const String8& address);
- status_t removeOutputProfile(const String8& name);
- status_t addInputProfile(const String8& name, const audio_config_t *config,
+ status_t removeOutputProfile(const std::string& name);
+ status_t addInputProfile(const std::string& name, const audio_config_t *config,
audio_devices_t device, const String8& address);
- status_t removeInputProfile(const String8& name);
+ status_t removeInputProfile(const std::string& name);
audio_module_handle_t getHandle() const { return mHandle; }
void setHandle(audio_module_handle_t handle);
- sp<AudioPort> findPortByTagName(const String8 &tagName) const
+ sp<PolicyAudioPort> findPortByTagName(const std::string &tagName) const
{
- return mPorts.findByTagName(tagName);
+ return findByTagName(mPorts, tagName);
}
/**
@@ -106,7 +106,8 @@
* @return true if the HwModule supports the connection between the sink and the source,
* false otherwise
*/
- bool supportsPatch(const sp<AudioPort> &srcPort, const sp<AudioPort> &dstPort) const;
+ bool supportsPatch(const sp<PolicyAudioPort> &srcPort,
+ const sp<PolicyAudioPort> &dstPort) const;
// TODO remove from here (split serialization)
void dump(String8 *dst) const;
@@ -122,7 +123,7 @@
DeviceVector mDeclaredDevices; // devices declared in audio_policy configuration file.
DeviceVector mDynamicDevices; /**< devices that can be added/removed at runtime (e.g. rsbumix)*/
AudioRouteVector mRoutes;
- AudioPortVector mPorts;
+ PolicyAudioPortVector mPorts;
};
class HwModuleCollection : public Vector<sp<HwModule> >
@@ -130,8 +131,8 @@
public:
sp<HwModule> getModuleFromName(const char *name) const;
- sp<HwModule> getModuleForDeviceTypes(audio_devices_t device,
- audio_format_t encodedFormat) const;
+ sp<HwModule> getModuleForDeviceType(audio_devices_t device,
+ audio_format_t encodedFormat) const;
sp<HwModule> getModuleForDevice(const sp<DeviceDescriptor> &device,
audio_format_t encodedFormat) const;
diff --git a/services/audiopolicy/common/managerdefinitions/include/IOProfile.h b/services/audiopolicy/common/managerdefinitions/include/IOProfile.h
index e0b56d4..2044863 100644
--- a/services/audiopolicy/common/managerdefinitions/include/IOProfile.h
+++ b/services/audiopolicy/common/managerdefinitions/include/IOProfile.h
@@ -16,8 +16,10 @@
#pragma once
-#include "AudioPort.h"
#include "DeviceDescriptor.h"
+#include "PolicyAudioPort.h"
+#include "policy.h"
+#include <media/AudioContainers.h>
#include <utils/String8.h>
#include <system/audio.h>
@@ -30,18 +32,28 @@
// It is used by the policy manager to determine if an output or input is suitable for
// a given use case, open/close it accordingly and connect/disconnect audio tracks
// to/from it.
-class IOProfile : public AudioPort
+class IOProfile : public AudioPort, public PolicyAudioPort
{
public:
- IOProfile(const String8 &name, audio_port_role_t role)
+ IOProfile(const std::string &name, audio_port_role_t role)
: AudioPort(name, AUDIO_PORT_TYPE_MIX, role),
maxOpenCount(1),
curOpenCount(0),
maxActiveCount(1),
curActiveCount(0) {}
+ virtual ~IOProfile() = default;
+
// For a Profile aka MixPort, tag name and name are equivalent.
- virtual const String8 getTagName() const { return getName(); }
+ virtual const std::string getTagName() const { return getName(); }
+
+ virtual void addAudioProfile(const sp<AudioProfile> &profile) {
+ addAudioProfileAndSort(mProfiles, profile);
+ }
+
+ virtual sp<AudioPort> asAudioPort() const {
+ return static_cast<AudioPort*>(const_cast<IOProfile*>(this));
+ }
// FIXME: this is needed because shared MMAP stream clients use the same audio session.
// Once capture clients are tracked individually and not per session this can be removed
@@ -51,7 +63,7 @@
// flags are parsed before maxActiveCount by the serializer.
void setFlags(uint32_t flags) override
{
- AudioPort::setFlags(flags);
+ PolicyAudioPort::setFlags(flags);
if (getRole() == AUDIO_PORT_ROLE_SINK && (flags & AUDIO_INPUT_FLAG_MMAP_NOIRQ) != 0) {
maxActiveCount = 0;
}
@@ -91,15 +103,12 @@
bool hasSupportedDevices() const { return !mSupportedDevices.isEmpty(); }
- bool supportsDeviceTypes(audio_devices_t device) const
+ bool supportsDeviceTypes(const DeviceTypeSet& deviceTypes) const
{
- if (audio_is_output_devices(device)) {
- if (deviceSupportsEncodedFormats(device)) {
- return mSupportedDevices.types() & device;
- }
- return false;
- }
- return mSupportedDevices.types() & (device & ~AUDIO_DEVICE_BIT_IN);
+ const bool areOutputDevices = Intersection(deviceTypes, getAudioDeviceInAllSet()).empty();
+ const bool devicesSupported = !mSupportedDevices.getDevicesFromTypes(deviceTypes).empty();
+ return devicesSupported &&
+ (!areOutputDevices || devicesSupportEncodedFormats(deviceTypes));
}
/**
@@ -114,18 +123,18 @@
bool supportsDevice(const sp<DeviceDescriptor> &device, bool forceCheckOnAddress = false) const
{
if (!device_distinguishes_on_address(device->type()) && !forceCheckOnAddress) {
- return supportsDeviceTypes(device->type());
+ return supportsDeviceTypes(DeviceTypeSet({device->type()}));
}
return mSupportedDevices.contains(device);
}
- bool deviceSupportsEncodedFormats(audio_devices_t device) const
+ bool devicesSupportEncodedFormats(DeviceTypeSet deviceTypes) const
{
- if (device == AUDIO_DEVICE_NONE) {
+ if (deviceTypes.empty()) {
return true; // required for isOffloadSupported() check
}
DeviceVector deviceList =
- mSupportedDevices.getDevicesFromTypeMask(device);
+ mSupportedDevices.getDevicesFromTypes(deviceTypes);
if (!deviceList.empty()) {
return deviceList.itemAt(0)->hasCurrentEncodedFormat();
}
@@ -183,13 +192,13 @@
class InputProfile : public IOProfile
{
public:
- explicit InputProfile(const String8 &name) : IOProfile(name, AUDIO_PORT_ROLE_SINK) {}
+ explicit InputProfile(const std::string &name) : IOProfile(name, AUDIO_PORT_ROLE_SINK) {}
};
class OutputProfile : public IOProfile
{
public:
- explicit OutputProfile(const String8 &name) : IOProfile(name, AUDIO_PORT_ROLE_SOURCE) {}
+ explicit OutputProfile(const std::string &name) : IOProfile(name, AUDIO_PORT_ROLE_SOURCE) {}
};
} // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/IVolumeCurves.h b/services/audiopolicy/common/managerdefinitions/include/IVolumeCurves.h
index d408446..fd8b81a 100644
--- a/services/audiopolicy/common/managerdefinitions/include/IVolumeCurves.h
+++ b/services/audiopolicy/common/managerdefinitions/include/IVolumeCurves.h
@@ -16,8 +16,9 @@
#pragma once
-#include <system/audio.h>
#include <Volume.h>
+#include <media/AudioContainers.h>
+#include <system/audio.h>
#include <utils/Errors.h>
#include <utils/String8.h>
#include <vector>
@@ -33,7 +34,7 @@
virtual void addCurrentVolumeIndex(audio_devices_t device, int index) = 0;
virtual bool canBeMuted() const = 0;
virtual int getVolumeIndexMin() const = 0;
- virtual int getVolumeIndex(audio_devices_t device) const = 0;
+ virtual int getVolumeIndex(const DeviceTypeSet& device) const = 0;
virtual int getVolumeIndexMax() const = 0;
virtual float volIndexToDb(device_category device, int indexInUi) const = 0;
virtual bool hasVolumeIndexForDevice(audio_devices_t device) const = 0;
diff --git a/services/audiopolicy/common/managerdefinitions/include/PolicyAudioPort.h b/services/audiopolicy/common/managerdefinitions/include/PolicyAudioPort.h
new file mode 100644
index 0000000..99df3c0
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/include/PolicyAudioPort.h
@@ -0,0 +1,153 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#pragma once
+
+#include "AudioCollections.h"
+#include "AudioProfileVectorHelper.h"
+#include "HandleGenerator.h"
+#include <media/AudioGain.h>
+#include <media/AudioPort.h>
+#include <utils/String8.h>
+#include <utils/Vector.h>
+#include <utils/RefBase.h>
+#include <utils/Errors.h>
+#include <system/audio.h>
+#include <cutils/config_utils.h>
+
+namespace android {
+
+class HwModule;
+class AudioRoute;
+
+class PolicyAudioPort : public virtual RefBase, private HandleGenerator<audio_port_handle_t>
+{
+public:
+ PolicyAudioPort() : mFlags(AUDIO_OUTPUT_FLAG_NONE) {}
+
+ virtual ~PolicyAudioPort() = default;
+
+ virtual const std::string getTagName() const = 0;
+
+ virtual sp<AudioPort> asAudioPort() const = 0;
+
+ virtual void setFlags(uint32_t flags)
+ {
+ //force direct flag if offload flag is set: offloading implies a direct output stream
+ // and all common behaviors are driven by checking only the direct flag
+ // this should normally be set appropriately in the policy configuration file
+ if (asAudioPort()->getRole() == AUDIO_PORT_ROLE_SOURCE &&
+ (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
+ flags |= AUDIO_OUTPUT_FLAG_DIRECT;
+ }
+ mFlags = flags;
+ }
+ uint32_t getFlags() const { return mFlags; }
+
+ virtual void attach(const sp<HwModule>& module);
+ virtual void detach();
+ bool isAttached() { return mModule != 0; }
+
+ // Audio port IDs are in a different namespace than AudioFlinger unique IDs
+ static audio_port_handle_t getNextUniqueId();
+
+ // searches for an exact match
+ virtual status_t checkExactAudioProfile(const struct audio_port_config *config) const;
+
+ // searches for a compatible match, currently implemented for input
+ // parameters are input|output, returned value is the best match.
+ status_t checkCompatibleAudioProfile(uint32_t &samplingRate,
+ audio_channel_mask_t &channelMask,
+ audio_format_t &format) const
+ {
+ return checkCompatibleProfile(
+ asAudioPort()->getAudioProfiles(), samplingRate, channelMask, format,
+ asAudioPort()->getType(), asAudioPort()->getRole());
+ }
+
+ void pickAudioProfile(uint32_t &samplingRate,
+ audio_channel_mask_t &channelMask,
+ audio_format_t &format) const;
+
+ static const audio_format_t sPcmFormatCompareTable[];
+
+ static int compareFormats(audio_format_t format1, audio_format_t format2);
+
+ // Used to select an audio HAL output stream with a sample format providing the
+ // less degradation for a given AudioTrack sample format.
+ static bool isBetterFormatMatch(audio_format_t newFormat,
+ audio_format_t currentFormat,
+ audio_format_t targetFormat);
+ static uint32_t formatDistance(audio_format_t format1,
+ audio_format_t format2);
+ static const uint32_t kFormatDistanceMax = 4;
+
+ audio_module_handle_t getModuleHandle() const;
+ uint32_t getModuleVersionMajor() const;
+ const char *getModuleName() const;
+ sp<HwModule> getModule() const { return mModule; }
+
+ inline bool isDirectOutput() const
+ {
+ return (asAudioPort()->getType() == AUDIO_PORT_TYPE_MIX) &&
+ (asAudioPort()->getRole() == AUDIO_PORT_ROLE_SOURCE) &&
+ (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD));
+ }
+
+ void addRoute(const sp<AudioRoute> &route) { mRoutes.add(route); }
+ const AudioRouteVector &getRoutes() const { return mRoutes; }
+
+private:
+ void pickChannelMask(audio_channel_mask_t &channelMask,
+ const ChannelMaskSet &channelMasks) const;
+ void pickSamplingRate(uint32_t &rate, const SampleRateSet &samplingRates) const;
+
+ uint32_t mFlags; // attribute flags mask (e.g primary output, direct output...).
+ sp<HwModule> mModule; // audio HW module exposing this I/O stream
+ AudioRouteVector mRoutes; // Routes involving this port
+};
+
+class PolicyAudioPortConfig : public virtual RefBase
+{
+public:
+ virtual ~PolicyAudioPortConfig() = default;
+
+ virtual sp<PolicyAudioPort> getPolicyAudioPort() const = 0;
+
+ status_t validationBeforeApplyConfig(const struct audio_port_config *config) const;
+
+ void applyPolicyAudioPortConfig(const struct audio_port_config *config) {
+ if (config->config_mask & AUDIO_PORT_CONFIG_FLAGS) {
+ mFlags = config->flags;
+ }
+ }
+
+ void toPolicyAudioPortConfig(
+ struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig = NULL) const;
+
+
+ virtual bool hasSameHwModuleAs(const sp<PolicyAudioPortConfig>& other) const {
+ return (other.get() != nullptr) && (other->getPolicyAudioPort().get() != nullptr) &&
+ (getPolicyAudioPort().get() != nullptr) &&
+ (other->getPolicyAudioPort()->getModuleHandle() ==
+ getPolicyAudioPort()->getModuleHandle());
+ }
+
+ union audio_io_flags mFlags = { AUDIO_INPUT_FLAG_NONE };
+};
+
+} // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/audio_policy_conf.h b/services/audiopolicy/common/managerdefinitions/include/audio_policy_conf.h
deleted file mode 100644
index 0a27947..0000000
--- a/services/audiopolicy/common/managerdefinitions/include/audio_policy_conf.h
+++ /dev/null
@@ -1,71 +0,0 @@
-/*
- * Copyright (C) 2012 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#pragma once
-
-
-/////////////////////////////////////////////////
-// Definitions for audio policy configuration file (audio_policy.conf)
-/////////////////////////////////////////////////
-
-#define AUDIO_HARDWARE_MODULE_ID_MAX_LEN 32
-
-#define AUDIO_POLICY_CONFIG_FILE "/system/etc/audio_policy.conf"
-#define AUDIO_POLICY_VENDOR_CONFIG_FILE "/vendor/etc/audio_policy.conf"
-
-// global configuration
-#define GLOBAL_CONFIG_TAG "global_configuration"
-
-#define ATTACHED_OUTPUT_DEVICES_TAG "attached_output_devices"
-#define DEFAULT_OUTPUT_DEVICE_TAG "default_output_device"
-#define ATTACHED_INPUT_DEVICES_TAG "attached_input_devices"
-#define SPEAKER_DRC_ENABLED_TAG "speaker_drc_enabled"
-#define AUDIO_HAL_VERSION_TAG "audio_hal_version"
-
-// hw modules descriptions
-#define AUDIO_HW_MODULE_TAG "audio_hw_modules"
-
-#define OUTPUTS_TAG "outputs"
-#define INPUTS_TAG "inputs"
-
-#define SAMPLING_RATES_TAG "sampling_rates"
-#define FORMATS_TAG "formats"
-#define CHANNELS_TAG "channel_masks"
-#define DEVICES_TAG "devices"
-#define FLAGS_TAG "flags"
-
-#define APM_DEVICES_TAG "devices"
-#define APM_DEVICE_TYPE "type"
-#define APM_DEVICE_ADDRESS "address"
-
-#define MIXERS_TAG "mixers"
-#define MIXER_TYPE "type"
-#define MIXER_TYPE_MUX "mux"
-#define MIXER_TYPE_MIX "mix"
-
-#define GAINS_TAG "gains"
-#define GAIN_MODE "mode"
-#define GAIN_CHANNELS "channel_mask"
-#define GAIN_MIN_VALUE "min_value_mB"
-#define GAIN_MAX_VALUE "max_value_mB"
-#define GAIN_DEFAULT_VALUE "default_value_mB"
-#define GAIN_STEP_VALUE "step_value_mB"
-#define GAIN_MIN_RAMP_MS "min_ramp_ms"
-#define GAIN_MAX_RAMP_MS "max_ramp_ms"
-
-#define DYNAMIC_VALUE_TAG "dynamic" // special value for "channel_masks", "sampling_rates" and
- // "formats" in outputs descriptors indicating that supported
- // values should be queried after opening the output.
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioCollections.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioCollections.cpp
index c90a582..cd10010 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioCollections.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioCollections.cpp
@@ -18,16 +18,16 @@
//#define LOG_NDEBUG 0
#include "AudioCollections.h"
-#include "AudioPort.h"
#include "AudioRoute.h"
#include "HwModule.h"
-#include "AudioGain.h"
+#include "PolicyAudioPort.h"
namespace android {
-sp<AudioPort> AudioPortVector::findByTagName(const String8 &tagName) const
+sp<PolicyAudioPort> findByTagName(const PolicyAudioPortVector& policyAudioPortVector,
+ const std::string &tagName)
{
- for (const auto& port : *this) {
+ for (const auto& port : policyAudioPortVector) {
if (port->getTagName() == tagName) {
return port;
}
@@ -35,15 +35,15 @@
return nullptr;
}
-void AudioRouteVector::dump(String8 *dst, int spaces) const
+void dumpAudioRouteVector(const AudioRouteVector& audioRouteVector, String8 *dst, int spaces)
{
- if (isEmpty()) {
+ if (audioRouteVector.isEmpty()) {
return;
}
- dst->appendFormat("\n%*sAudio Routes (%zu):\n", spaces, "", size());
- for (size_t i = 0; i < size(); i++) {
+ dst->appendFormat("\n%*sAudio Routes (%zu):\n", spaces, "", audioRouteVector.size());
+ for (size_t i = 0; i < audioRouteVector.size(); i++) {
dst->appendFormat("%*s- Route %zu:\n", spaces, "", i + 1);
- itemAt(i)->dump(dst, 4);
+ audioRouteVector.itemAt(i)->dump(dst, 4);
}
}
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioGain.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioGain.cpp
deleted file mode 100644
index 2725870..0000000
--- a/services/audiopolicy/common/managerdefinitions/src/AudioGain.cpp
+++ /dev/null
@@ -1,114 +0,0 @@
-/*
- * Copyright (C) 2015 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#define LOG_TAG "APM::AudioGain"
-//#define LOG_NDEBUG 0
-
-//#define VERY_VERBOSE_LOGGING
-#ifdef VERY_VERBOSE_LOGGING
-#define ALOGVV ALOGV
-#else
-#define ALOGVV(a...) do { } while(0)
-#endif
-
-#include "AudioGain.h"
-#include <utils/Log.h>
-#include <utils/String8.h>
-
-#include <math.h>
-
-namespace android {
-
-AudioGain::AudioGain(int index, bool useInChannelMask)
-{
- mIndex = index;
- mUseInChannelMask = useInChannelMask;
- memset(&mGain, 0, sizeof(struct audio_gain));
-}
-
-void AudioGain::getDefaultConfig(struct audio_gain_config *config)
-{
- config->index = mIndex;
- config->mode = mGain.mode;
- config->channel_mask = mGain.channel_mask;
- if ((mGain.mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
- config->values[0] = mGain.default_value;
- } else {
- uint32_t numValues;
- if (mUseInChannelMask) {
- numValues = audio_channel_count_from_in_mask(mGain.channel_mask);
- } else {
- numValues = audio_channel_count_from_out_mask(mGain.channel_mask);
- }
- for (size_t i = 0; i < numValues; i++) {
- config->values[i] = mGain.default_value;
- }
- }
- if ((mGain.mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
- config->ramp_duration_ms = mGain.min_ramp_ms;
- }
-}
-
-status_t AudioGain::checkConfig(const struct audio_gain_config *config)
-{
- if ((config->mode & ~mGain.mode) != 0) {
- return BAD_VALUE;
- }
- if ((config->mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
- if ((config->values[0] < mGain.min_value) ||
- (config->values[0] > mGain.max_value)) {
- return BAD_VALUE;
- }
- } else {
- if ((config->channel_mask & ~mGain.channel_mask) != 0) {
- return BAD_VALUE;
- }
- uint32_t numValues;
- if (mUseInChannelMask) {
- numValues = audio_channel_count_from_in_mask(config->channel_mask);
- } else {
- numValues = audio_channel_count_from_out_mask(config->channel_mask);
- }
- for (size_t i = 0; i < numValues; i++) {
- if ((config->values[i] < mGain.min_value) ||
- (config->values[i] > mGain.max_value)) {
- return BAD_VALUE;
- }
- }
- }
- if ((config->mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
- if ((config->ramp_duration_ms < mGain.min_ramp_ms) ||
- (config->ramp_duration_ms > mGain.max_ramp_ms)) {
- return BAD_VALUE;
- }
- }
- return NO_ERROR;
-}
-
-void AudioGain::dump(String8 *dst, int spaces, int index) const
-{
- dst->appendFormat("%*sGain %d:\n", spaces, "", index+1);
- dst->appendFormat("%*s- mode: %08x\n", spaces, "", mGain.mode);
- dst->appendFormat("%*s- channel_mask: %08x\n", spaces, "", mGain.channel_mask);
- dst->appendFormat("%*s- min_value: %d mB\n", spaces, "", mGain.min_value);
- dst->appendFormat("%*s- max_value: %d mB\n", spaces, "", mGain.max_value);
- dst->appendFormat("%*s- default_value: %d mB\n", spaces, "", mGain.default_value);
- dst->appendFormat("%*s- step_value: %d mB\n", spaces, "", mGain.step_value);
- dst->appendFormat("%*s- min_ramp_ms: %d ms\n", spaces, "", mGain.min_ramp_ms);
- dst->appendFormat("%*s- max_ramp_ms: %d ms\n", spaces, "", mGain.max_ramp_ms);
-}
-
-} // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp
index a096e8f..cb3c953 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp
@@ -22,7 +22,6 @@
#include <policy.h>
#include <AudioPolicyInterface.h>
#include "AudioInputDescriptor.h"
-#include "AudioGain.h"
#include "AudioPolicyMix.h"
#include "HwModule.h"
@@ -35,8 +34,8 @@
{
if (profile != NULL) {
profile->pickAudioProfile(mSamplingRate, mChannelMask, mFormat);
- if (profile->mGains.size() > 0) {
- profile->mGains[0]->getDefaultConfig(&mGain);
+ if (profile->getGains().size() > 0) {
+ profile->getGains()[0]->getDefaultConfig(&mGain);
}
}
}
@@ -49,16 +48,29 @@
return mProfile->getModuleHandle();
}
-audio_port_handle_t AudioInputDescriptor::getId() const
-{
- return mId;
-}
-
audio_source_t AudioInputDescriptor::source() const
{
return getHighestPriorityAttributes().source;
}
+status_t AudioInputDescriptor::applyAudioPortConfig(const struct audio_port_config *config,
+ audio_port_config *backupConfig)
+{
+ struct audio_port_config localBackupConfig = { .config_mask = config->config_mask };
+ status_t status = NO_ERROR;
+
+ toAudioPortConfig(&localBackupConfig);
+ if ((status = validationBeforeApplyConfig(config)) == NO_ERROR) {
+ AudioPortConfig::applyAudioPortConfig(config, backupConfig);
+ applyPolicyAudioPortConfig(config);
+ }
+
+ if (backupConfig != NULL) {
+ *backupConfig = localBackupConfig;
+ }
+ return status;
+}
+
void AudioInputDescriptor::toAudioPortConfig(struct audio_port_config *dstConfig,
const struct audio_port_config *srcConfig) const
{
@@ -71,8 +83,8 @@
}
AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
+ toPolicyAudioPortConfig(dstConfig, srcConfig);
- dstConfig->id = mId;
dstConfig->role = AUDIO_PORT_ROLE_SINK;
dstConfig->type = AUDIO_PORT_TYPE_MIX;
dstConfig->ext.mix.hw_module = getModuleHandle();
@@ -213,7 +225,7 @@
mDevice = device;
ALOGV("opening input for device %s profile %p name %s",
- mDevice->toString().c_str(), mProfile.get(), mProfile->getName().string());
+ mDevice->toString().c_str(), mProfile.get(), mProfile->getName().c_str());
audio_devices_t deviceType = mDevice->type();
@@ -221,7 +233,7 @@
input,
&lConfig,
&deviceType,
- mDevice->address(),
+ String8(mDevice->address().c_str()),
source,
flags);
LOG_ALWAYS_FATAL_IF(mDevice->type() != deviceType,
@@ -235,7 +247,7 @@
mSamplingRate = lConfig.sample_rate;
mChannelMask = lConfig.channel_mask;
mFormat = lConfig.format;
- mId = AudioPort::getNextUniqueId();
+ mId = PolicyAudioPort::getNextUniqueId();
mIoHandle = *input;
mProfile->curOpenCount++;
}
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
index 8a60cf2..dd51658 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
@@ -21,27 +21,28 @@
#include "AudioOutputDescriptor.h"
#include "AudioPolicyMix.h"
#include "IOProfile.h"
-#include "AudioGain.h"
#include "Volume.h"
#include "HwModule.h"
#include "TypeConverter.h"
+#include <media/AudioGain.h>
#include <media/AudioParameter.h>
#include <media/AudioPolicy.h>
// A device mask for all audio output devices that are considered "remote" when evaluating
// active output devices in isStreamActiveRemotely()
-#define APM_AUDIO_OUT_DEVICE_REMOTE_ALL AUDIO_DEVICE_OUT_REMOTE_SUBMIX
namespace android {
-AudioOutputDescriptor::AudioOutputDescriptor(const sp<AudioPort>& port,
+DeviceTypeSet APM_AUDIO_OUT_DEVICE_REMOTE_ALL = {AUDIO_DEVICE_OUT_REMOTE_SUBMIX};
+
+AudioOutputDescriptor::AudioOutputDescriptor(const sp<PolicyAudioPort>& policyAudioPort,
AudioPolicyClientInterface *clientInterface)
- : mPort(port), mClientInterface(clientInterface)
+ : mPolicyAudioPort(policyAudioPort), mClientInterface(clientInterface)
{
- if (mPort.get() != nullptr) {
- mPort->pickAudioProfile(mSamplingRate, mChannelMask, mFormat);
- if (mPort->mGains.size() > 0) {
- mPort->mGains[0]->getDefaultConfig(&mGain);
+ if (mPolicyAudioPort.get() != nullptr) {
+ mPolicyAudioPort->pickAudioProfile(mSamplingRate, mChannelMask, mFormat);
+ if (mPolicyAudioPort->asAudioPort()->getGains().size() > 0) {
+ mPolicyAudioPort->asAudioPort()->getGains()[0]->getDefaultConfig(&mGain);
}
}
}
@@ -55,7 +56,8 @@
audio_module_handle_t AudioOutputDescriptor::getModuleHandle() const
{
- return mPort.get() != nullptr ? mPort->getModuleHandle() : AUDIO_MODULE_HANDLE_NONE;
+ return mPolicyAudioPort.get() != nullptr ?
+ mPolicyAudioPort->getModuleHandle() : AUDIO_MODULE_HANDLE_NONE;
}
audio_patch_handle_t AudioOutputDescriptor::getPatchHandle() const
@@ -68,11 +70,6 @@
mPatchHandle = handle;
}
-audio_port_handle_t AudioOutputDescriptor::getId() const
-{
- return mId;
-}
-
bool AudioOutputDescriptor::sharesHwModuleWith(
const sp<AudioOutputDescriptor>& outputDesc)
{
@@ -144,7 +141,7 @@
return false;
}
-bool AudioOutputDescriptor::isFixedVolume(audio_devices_t device __unused)
+bool AudioOutputDescriptor::isFixedVolume(const DeviceTypeSet& deviceTypes __unused)
{
return false;
}
@@ -152,7 +149,7 @@
bool AudioOutputDescriptor::setVolume(float volumeDb,
VolumeSource volumeSource,
const StreamTypeVector &/*streams*/,
- audio_devices_t /*device*/,
+ const DeviceTypeSet& /*deviceTypes*/,
uint32_t delayMs,
bool force)
{
@@ -167,9 +164,27 @@
return false;
}
-void AudioOutputDescriptor::toAudioPortConfig(
- struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig) const
+status_t AudioOutputDescriptor::applyAudioPortConfig(const struct audio_port_config *config,
+ audio_port_config *backupConfig)
+{
+ struct audio_port_config localBackupConfig = { .config_mask = config->config_mask };
+ status_t status = NO_ERROR;
+
+ toAudioPortConfig(&localBackupConfig);
+ if ((status = validationBeforeApplyConfig(config)) == NO_ERROR) {
+ AudioPortConfig::applyAudioPortConfig(config, backupConfig);
+ applyPolicyAudioPortConfig(config);
+ }
+
+ if (backupConfig != NULL) {
+ *backupConfig = localBackupConfig;
+ }
+ return status;
+}
+
+
+void AudioOutputDescriptor::toAudioPortConfig(struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig) const
{
dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
AUDIO_PORT_CONFIG_FORMAT|AUDIO_PORT_CONFIG_GAIN;
@@ -177,8 +192,8 @@
dstConfig->config_mask |= srcConfig->config_mask;
}
AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
+ toPolicyAudioPortConfig(dstConfig, srcConfig);
- dstConfig->id = mId;
dstConfig->role = AUDIO_PORT_ROLE_SOURCE;
dstConfig->type = AUDIO_PORT_TYPE_MIX;
dstConfig->ext.mix.hw_module = getModuleHandle();
@@ -188,7 +203,7 @@
void AudioOutputDescriptor::toAudioPort(struct audio_port *port) const
{
// Should not be called for duplicated ports, see SwAudioOutputDescriptor::toAudioPortConfig.
- mPort->toAudioPort(port);
+ mPolicyAudioPort->asAudioPort()->toAudioPort(port);
port->id = mId;
port->ext.mix.hw_module = getModuleHandle();
}
@@ -320,13 +335,13 @@
return filteredDevices.filter(devices);
}
-bool SwAudioOutputDescriptor::deviceSupportsEncodedFormats(audio_devices_t device)
+bool SwAudioOutputDescriptor::devicesSupportEncodedFormats(const DeviceTypeSet& deviceTypes)
{
if (isDuplicated()) {
- return (mOutput1->deviceSupportsEncodedFormats(device)
- || mOutput2->deviceSupportsEncodedFormats(device));
+ return (mOutput1->devicesSupportEncodedFormats(deviceTypes)
+ || mOutput2->devicesSupportEncodedFormats(deviceTypes));
} else {
- return mProfile->deviceSupportsEncodedFormats(device);
+ return mProfile->devicesSupportEncodedFormats(deviceTypes);
}
}
@@ -349,16 +364,16 @@
AudioOutputDescriptor::setClientActive(client, active);
}
-bool SwAudioOutputDescriptor::isFixedVolume(audio_devices_t device)
+bool SwAudioOutputDescriptor::isFixedVolume(const DeviceTypeSet& deviceTypes)
{
// unit gain if rerouting to external policy
- if (device == AUDIO_DEVICE_OUT_REMOTE_SUBMIX) {
+ if (isSingleDeviceType(deviceTypes, AUDIO_DEVICE_OUT_REMOTE_SUBMIX)) {
if (mPolicyMix != NULL) {
ALOGV("max gain when rerouting for output=%d", mIoHandle);
return true;
}
}
- if (device == AUDIO_DEVICE_OUT_TELEPHONY_TX) {
+ if (isSingleDeviceType(deviceTypes, AUDIO_DEVICE_OUT_TELEPHONY_TX)) {
ALOGV("max gain when output device is telephony tx");
return true;
}
@@ -391,12 +406,12 @@
bool SwAudioOutputDescriptor::setVolume(float volumeDb,
VolumeSource vs, const StreamTypeVector &streamTypes,
- audio_devices_t device,
+ const DeviceTypeSet& deviceTypes,
uint32_t delayMs,
bool force)
{
StreamTypeVector streams = streamTypes;
- if (!AudioOutputDescriptor::setVolume(volumeDb, vs, streamTypes, device, delayMs, force)) {
+ if (!AudioOutputDescriptor::setVolume(volumeDb, vs, streamTypes, deviceTypes, delayMs, force)) {
return false;
}
if (streams.empty()) {
@@ -406,7 +421,7 @@
// APM loops on all group, so filter on active group to set the port gain,
// let the other groups set the stream volume as per legacy
// TODO: Pass in the device address and check against it.
- if (device == devicePort->type() &&
+ if (isSingleDeviceType(deviceTypes, devicePort->type()) &&
devicePort->hasGainController(true) && isActive(vs)) {
ALOGV("%s: device %s has gain controller", __func__, devicePort->toString().c_str());
// @todo: here we might be in trouble if the SwOutput has several active clients with
@@ -452,8 +467,11 @@
audio_io_handle_t *output)
{
mDevices = devices;
- const String8& address = devices.getFirstValidAddress();
- audio_devices_t device = devices.types();
+ sp<DeviceDescriptor> device = devices.getDeviceForOpening();
+ LOG_ALWAYS_FATAL_IF(device == nullptr,
+ "%s failed to get device descriptor for opening "
+ "with the requested devices, all device types: %s",
+ __func__, dumpDeviceTypes(devices.types()).c_str());
audio_config_t lConfig;
if (config == nullptr) {
@@ -483,27 +501,25 @@
mFlags = (audio_output_flags_t)(mFlags | flags);
ALOGV("opening output for device %s profile %p name %s",
- mDevices.toString().c_str(), mProfile.get(), mProfile->getName().string());
+ mDevices.toString().c_str(), mProfile.get(), mProfile->getName().c_str());
status_t status = mClientInterface->openOutput(mProfile->getModuleHandle(),
output,
&lConfig,
- &device,
- address,
+ device,
&mLatency,
mFlags);
- LOG_ALWAYS_FATAL_IF(mDevices.types() != device,
- "%s openOutput returned device %08x when given device %08x",
- __FUNCTION__, mDevices.types(), device);
if (status == NO_ERROR) {
LOG_ALWAYS_FATAL_IF(*output == AUDIO_IO_HANDLE_NONE,
- "%s openOutput returned output handle %d for device %08x",
- __FUNCTION__, *output, device);
+ "%s openOutput returned output handle %d for device %s, "
+ "selected device %s for opening",
+ __FUNCTION__, *output, devices.toString().c_str(),
+ device->toString().c_str());
mSamplingRate = lConfig.sample_rate;
mChannelMask = lConfig.channel_mask;
mFormat = lConfig.format;
- mId = AudioPort::getNextUniqueId();
+ mId = PolicyAudioPort::getNextUniqueId();
mIoHandle = *output;
mProfile->curOpenCount++;
}
@@ -589,7 +605,7 @@
return INVALID_OPERATION;
}
- mId = AudioPort::getNextUniqueId();
+ mId = PolicyAudioPort::getNextUniqueId();
mIoHandle = *ioHandle;
mOutput1 = output1;
mOutput2 = output2;
@@ -632,12 +648,12 @@
bool HwAudioOutputDescriptor::setVolume(float volumeDb,
VolumeSource volumeSource, const StreamTypeVector &streams,
- audio_devices_t device,
+ const DeviceTypeSet& deviceTypes,
uint32_t delayMs,
bool force)
{
- bool changed =
- AudioOutputDescriptor::setVolume(volumeDb, volumeSource, streams, device, delayMs, force);
+ bool changed = AudioOutputDescriptor::setVolume(
+ volumeDb, volumeSource, streams, deviceTypes, delayMs, force);
if (changed) {
// TODO: use gain controller on source device if any to adjust volume
@@ -664,7 +680,8 @@
for (size_t i = 0; i < this->size(); i++) {
const sp<SwAudioOutputDescriptor> outputDesc = this->valueAt(i);
if (outputDesc->isActive(volumeSource, inPastMs, sysTime)
- && ((outputDesc->devices().types() & APM_AUDIO_OUT_DEVICE_REMOTE_ALL) == 0)) {
+ && (!(outputDesc->devices()
+ .containsDeviceAmongTypes(APM_AUDIO_OUT_DEVICE_REMOTE_ALL)))) {
return true;
}
}
@@ -676,7 +693,7 @@
nsecs_t sysTime = systemTime();
for (size_t i = 0; i < size(); i++) {
const sp<SwAudioOutputDescriptor> outputDesc = valueAt(i);
- if (((outputDesc->devices().types() & APM_AUDIO_OUT_DEVICE_REMOTE_ALL) != 0) &&
+ if (outputDesc->devices().containsDeviceAmongTypes(APM_AUDIO_OUT_DEVICE_REMOTE_ALL) &&
outputDesc->isActive(volumeSource, inPastMs, sysTime)) {
// do not consider re routing (when the output is going to a dynamic policy)
// as "remote playback"
@@ -707,9 +724,8 @@
for (size_t i = 0; i < size(); i++) {
sp<SwAudioOutputDescriptor> outputDesc = valueAt(i);
if (!outputDesc->isDuplicated() &&
- outputDesc->devices().types() & AUDIO_DEVICE_OUT_ALL_A2DP &&
- outputDesc->deviceSupportsEncodedFormats(
- AUDIO_DEVICE_OUT_BLUETOOTH_A2DP)) {
+ outputDesc->devices().containsDeviceAmongTypes(getAudioDeviceOutAllA2dpSet()) &&
+ outputDesc->devicesSupportEncodedFormats(getAudioDeviceOutAllA2dpSet())) {
return this->keyAt(i);
}
}
@@ -725,7 +741,7 @@
sp<HwModule> primaryHwModule = primaryOutput->mProfile->getModule();
for (const auto &outputProfile : primaryHwModule->getOutputProfiles()) {
- if (outputProfile->supportsDeviceTypes(AUDIO_DEVICE_OUT_ALL_A2DP)) {
+ if (outputProfile->supportsDeviceTypes(getAudioDeviceOutAllA2dpSet())) {
return true;
}
}
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp
index 3a4db90..d79110a 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp
@@ -18,7 +18,6 @@
//#define LOG_NDEBUG 0
#include "AudioPatch.h"
-#include "AudioGain.h"
#include "TypeConverter.h"
#include <log/log.h>
@@ -27,10 +26,9 @@
namespace android {
AudioPatch::AudioPatch(const struct audio_patch *patch, uid_t uid) :
- mHandle(HandleGenerator<audio_patch_handle_t>::getNextHandle()),
mPatch(*patch),
- mUid(uid),
- mAfPatchHandle(AUDIO_PATCH_HANDLE_NONE)
+ mHandle(HandleGenerator<audio_patch_handle_t>::getNextHandle()),
+ mUid(uid)
{
}
@@ -69,7 +67,7 @@
add(handle, patch);
ALOGV("addAudioPatch() handle %d af handle %d num_sources %d num_sinks %d source handle %d"
"sink handle %d",
- handle, patch->mAfPatchHandle, patch->mPatch.num_sources, patch->mPatch.num_sinks,
+ handle, patch->getAfHandle(), patch->mPatch.num_sources, patch->mPatch.num_sinks,
patch->mPatch.sources[0].id, patch->mPatch.sinks[0].id);
return NO_ERROR;
}
@@ -82,7 +80,7 @@
ALOGW("removeAudioPatch() patch %d not in", handle);
return ALREADY_EXISTS;
}
- ALOGV("removeAudioPatch() handle %d af handle %d", handle, valueAt(index)->mAfPatchHandle);
+ ALOGV("removeAudioPatch() handle %d af handle %d", handle, valueAt(index)->getAfHandle());
removeItemsAt(index);
return NO_ERROR;
}
@@ -124,7 +122,7 @@
}
if (patchesWritten < patchesMax) {
patches[patchesWritten] = patch->mPatch;
- patches[patchesWritten++].id = patch->mHandle;
+ patches[patchesWritten++].id = patch->getHandle();
}
(*num_patches)++;
ALOGV("listAudioPatches() patch %zu num_sources %d num_sinks %d",
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp
index c42923a..20c0a24 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp
@@ -20,9 +20,8 @@
#include "AudioPolicyMix.h"
#include "TypeConverter.h"
#include "HwModule.h"
-#include "AudioPort.h"
+#include "PolicyAudioPort.h"
#include "IOProfile.h"
-#include "AudioGain.h"
#include <AudioOutputDescriptor.h>
namespace android {
@@ -456,9 +455,9 @@
}
// check if this mix goes to a device in the list of devices
bool deviceMatch = false;
+ const AudioDeviceTypeAddr mixDevice(mix->mDeviceType, mix->mDeviceAddress.string());
for (size_t j = 0; j < devices.size(); j++) {
- if (devices[j].mType == mix->mDeviceType
- && devices[j].mAddress == mix->mDeviceAddress) {
+ if (mixDevice.equals(devices[j])) {
deviceMatch = true;
break;
}
@@ -523,7 +522,7 @@
}
}
if (ruleAllowsUid) {
- devices.add(AudioDeviceTypeAddr(mix->mDeviceType, mix->mDeviceAddress));
+ devices.add(AudioDeviceTypeAddr(mix->mDeviceType, mix->mDeviceAddress.string()));
}
}
return NO_ERROR;
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp
deleted file mode 100644
index c11490a..0000000
--- a/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp
+++ /dev/null
@@ -1,487 +0,0 @@
-/*
- * Copyright (C) 2015 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#define LOG_TAG "APM::AudioPort"
-//#define LOG_NDEBUG 0
-#include "TypeConverter.h"
-#include "AudioPort.h"
-#include "HwModule.h"
-#include "AudioGain.h"
-#include <policy.h>
-
-#ifndef ARRAY_SIZE
-#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
-#endif
-
-namespace android {
-
-// --- AudioPort class implementation
-void AudioPort::attach(const sp<HwModule>& module)
-{
- ALOGV("%s: attaching module %s to port %s", __FUNCTION__, getModuleName(), mName.string());
- mModule = module;
-}
-
-void AudioPort::detach()
-{
- mModule = nullptr;
-}
-
-// Note that is a different namespace than AudioFlinger unique IDs
-audio_port_handle_t AudioPort::getNextUniqueId()
-{
- return getNextHandle();
-}
-
-audio_module_handle_t AudioPort::getModuleHandle() const
-{
- return mModule != 0 ? mModule->getHandle() : AUDIO_MODULE_HANDLE_NONE;
-}
-
-uint32_t AudioPort::getModuleVersionMajor() const
-{
- return mModule != 0 ? mModule->getHalVersionMajor() : 0;
-}
-
-const char *AudioPort::getModuleName() const
-{
- return mModule != 0 ? mModule->getName() : "invalid module";
-}
-
-void AudioPort::toAudioPort(struct audio_port *port) const
-{
- // TODO: update this function once audio_port structure reflects the new profile definition.
- // For compatibility reason: flatening the AudioProfile into audio_port structure.
- SortedVector<audio_format_t> flatenedFormats;
- SampleRateVector flatenedRates;
- ChannelsVector flatenedChannels;
- for (const auto& profile : mProfiles) {
- if (profile->isValid()) {
- audio_format_t formatToExport = profile->getFormat();
- const SampleRateVector &ratesToExport = profile->getSampleRates();
- const ChannelsVector &channelsToExport = profile->getChannels();
-
- if (flatenedFormats.indexOf(formatToExport) < 0) {
- flatenedFormats.add(formatToExport);
- }
- for (size_t rateIndex = 0; rateIndex < ratesToExport.size(); rateIndex++) {
- uint32_t rate = ratesToExport[rateIndex];
- if (flatenedRates.indexOf(rate) < 0) {
- flatenedRates.add(rate);
- }
- }
- for (size_t chanIndex = 0; chanIndex < channelsToExport.size(); chanIndex++) {
- audio_channel_mask_t channels = channelsToExport[chanIndex];
- if (flatenedChannels.indexOf(channels) < 0) {
- flatenedChannels.add(channels);
- }
- }
- if (flatenedRates.size() > AUDIO_PORT_MAX_SAMPLING_RATES ||
- flatenedChannels.size() > AUDIO_PORT_MAX_CHANNEL_MASKS ||
- flatenedFormats.size() > AUDIO_PORT_MAX_FORMATS) {
- ALOGE("%s: bailing out: cannot export profiles to port config", __FUNCTION__);
- return;
- }
- }
- }
- port->role = mRole;
- port->type = mType;
- strlcpy(port->name, mName, AUDIO_PORT_MAX_NAME_LEN);
- port->num_sample_rates = flatenedRates.size();
- port->num_channel_masks = flatenedChannels.size();
- port->num_formats = flatenedFormats.size();
- for (size_t i = 0; i < flatenedRates.size(); i++) {
- port->sample_rates[i] = flatenedRates[i];
- }
- for (size_t i = 0; i < flatenedChannels.size(); i++) {
- port->channel_masks[i] = flatenedChannels[i];
- }
- for (size_t i = 0; i < flatenedFormats.size(); i++) {
- port->formats[i] = flatenedFormats[i];
- }
-
- ALOGV("AudioPort::toAudioPort() num gains %zu", mGains.size());
-
- uint32_t i;
- for (i = 0; i < mGains.size() && i < AUDIO_PORT_MAX_GAINS; i++) {
- port->gains[i] = mGains[i]->getGain();
- }
- port->num_gains = i;
-}
-
-void AudioPort::importAudioPort(const sp<AudioPort>& port, bool force __unused)
-{
- for (const auto& profileToImport : port->mProfiles) {
- if (profileToImport->isValid()) {
- // Import only valid port, i.e. valid format, non empty rates and channels masks
- bool hasSameProfile = false;
- for (const auto& profile : mProfiles) {
- if (*profile == *profileToImport) {
- // never import a profile twice
- hasSameProfile = true;
- break;
- }
- }
- if (hasSameProfile) { // never import a same profile twice
- continue;
- }
- addAudioProfile(profileToImport);
- }
- }
-}
-
-status_t AudioPort::checkExactAudioProfile(const struct audio_port_config *config) const
-{
- status_t status = NO_ERROR;
- auto config_mask = config->config_mask;
- if (config_mask & AUDIO_PORT_CONFIG_GAIN) {
- config_mask &= ~AUDIO_PORT_CONFIG_GAIN;
- status = checkGain(&config->gain, config->gain.index);
- if (status != NO_ERROR) {
- return status;
- }
- }
- if (config_mask != 0) {
- // TODO should we check sample_rate / channel_mask / format separately?
- status = mProfiles.checkExactProfile(config->sample_rate,
- config->channel_mask,
- config->format);
- }
- return status;
-}
-
-void AudioPort::pickSamplingRate(uint32_t &pickedRate,const SampleRateVector &samplingRates) const
-{
- pickedRate = 0;
- // For direct outputs, pick minimum sampling rate: this helps ensuring that the
- // channel count / sampling rate combination chosen will be supported by the connected
- // sink
- if (isDirectOutput()) {
- uint32_t samplingRate = UINT_MAX;
- for (size_t i = 0; i < samplingRates.size(); i ++) {
- if ((samplingRates[i] < samplingRate) && (samplingRates[i] > 0)) {
- samplingRate = samplingRates[i];
- }
- }
- pickedRate = (samplingRate == UINT_MAX) ? 0 : samplingRate;
- } else {
- uint32_t maxRate = SAMPLE_RATE_HZ_MAX;
-
- // For mixed output and inputs, use max mixer sampling rates. Do not
- // limit sampling rate otherwise
- // For inputs, also see checkCompatibleSamplingRate().
- if (mType != AUDIO_PORT_TYPE_MIX) {
- maxRate = UINT_MAX;
- }
- // TODO: should mSamplingRates[] be ordered in terms of our preference
- // and we return the first (and hence most preferred) match? This is of concern if
- // we want to choose 96kHz over 192kHz for USB driver stability or resource constraints.
- for (size_t i = 0; i < samplingRates.size(); i ++) {
- if ((samplingRates[i] > pickedRate) && (samplingRates[i] <= maxRate)) {
- pickedRate = samplingRates[i];
- }
- }
- }
-}
-
-void AudioPort::pickChannelMask(audio_channel_mask_t &pickedChannelMask,
- const ChannelsVector &channelMasks) const
-{
- pickedChannelMask = AUDIO_CHANNEL_NONE;
- // For direct outputs, pick minimum channel count: this helps ensuring that the
- // channel count / sampling rate combination chosen will be supported by the connected
- // sink
- if (isDirectOutput()) {
- uint32_t channelCount = UINT_MAX;
- for (size_t i = 0; i < channelMasks.size(); i ++) {
- uint32_t cnlCount;
- if (useInputChannelMask()) {
- cnlCount = audio_channel_count_from_in_mask(channelMasks[i]);
- } else {
- cnlCount = audio_channel_count_from_out_mask(channelMasks[i]);
- }
- if ((cnlCount < channelCount) && (cnlCount > 0)) {
- pickedChannelMask = channelMasks[i];
- channelCount = cnlCount;
- }
- }
- } else {
- uint32_t channelCount = 0;
- uint32_t maxCount = MAX_MIXER_CHANNEL_COUNT;
-
- // For mixed output and inputs, use max mixer channel count. Do not
- // limit channel count otherwise
- if (mType != AUDIO_PORT_TYPE_MIX) {
- maxCount = UINT_MAX;
- }
- for (size_t i = 0; i < channelMasks.size(); i ++) {
- uint32_t cnlCount;
- if (useInputChannelMask()) {
- cnlCount = audio_channel_count_from_in_mask(channelMasks[i]);
- } else {
- cnlCount = audio_channel_count_from_out_mask(channelMasks[i]);
- }
- if ((cnlCount > channelCount) && (cnlCount <= maxCount)) {
- pickedChannelMask = channelMasks[i];
- channelCount = cnlCount;
- }
- }
- }
-}
-
-/* format in order of increasing preference */
-const audio_format_t AudioPort::sPcmFormatCompareTable[] = {
- AUDIO_FORMAT_DEFAULT,
- AUDIO_FORMAT_PCM_16_BIT,
- AUDIO_FORMAT_PCM_8_24_BIT,
- AUDIO_FORMAT_PCM_24_BIT_PACKED,
- AUDIO_FORMAT_PCM_32_BIT,
- AUDIO_FORMAT_PCM_FLOAT,
-};
-
-int AudioPort::compareFormats(audio_format_t format1, audio_format_t format2)
-{
- // NOTE: AUDIO_FORMAT_INVALID is also considered not PCM and will be compared equal to any
- // compressed format and better than any PCM format. This is by design of pickFormat()
- if (!audio_is_linear_pcm(format1)) {
- if (!audio_is_linear_pcm(format2)) {
- return 0;
- }
- return 1;
- }
- if (!audio_is_linear_pcm(format2)) {
- return -1;
- }
-
- int index1 = -1, index2 = -1;
- for (size_t i = 0;
- (i < ARRAY_SIZE(sPcmFormatCompareTable)) && ((index1 == -1) || (index2 == -1));
- i ++) {
- if (sPcmFormatCompareTable[i] == format1) {
- index1 = i;
- }
- if (sPcmFormatCompareTable[i] == format2) {
- index2 = i;
- }
- }
- // format1 not found => index1 < 0 => format2 > format1
- // format2 not found => index2 < 0 => format2 < format1
- return index1 - index2;
-}
-
-uint32_t AudioPort::formatDistance(audio_format_t format1, audio_format_t format2)
-{
- if (format1 == format2) {
- return 0;
- }
- if (format1 == AUDIO_FORMAT_INVALID || format2 == AUDIO_FORMAT_INVALID) {
- return kFormatDistanceMax;
- }
- int diffBytes = (int)audio_bytes_per_sample(format1) -
- audio_bytes_per_sample(format2);
-
- return abs(diffBytes);
-}
-
-bool AudioPort::isBetterFormatMatch(audio_format_t newFormat,
- audio_format_t currentFormat,
- audio_format_t targetFormat)
-{
- return formatDistance(newFormat, targetFormat) < formatDistance(currentFormat, targetFormat);
-}
-
-void AudioPort::pickAudioProfile(uint32_t &samplingRate,
- audio_channel_mask_t &channelMask,
- audio_format_t &format) const
-{
- format = AUDIO_FORMAT_DEFAULT;
- samplingRate = 0;
- channelMask = AUDIO_CHANNEL_NONE;
-
- // special case for uninitialized dynamic profile
- if (!mProfiles.hasValidProfile()) {
- return;
- }
- audio_format_t bestFormat = sPcmFormatCompareTable[ARRAY_SIZE(sPcmFormatCompareTable) - 1];
- // For mixed output and inputs, use best mixer output format.
- // Do not limit format otherwise
- if ((mType != AUDIO_PORT_TYPE_MIX) || isDirectOutput()) {
- bestFormat = AUDIO_FORMAT_INVALID;
- }
-
- for (size_t i = 0; i < mProfiles.size(); i ++) {
- if (!mProfiles[i]->isValid()) {
- continue;
- }
- audio_format_t formatToCompare = mProfiles[i]->getFormat();
- if ((compareFormats(formatToCompare, format) > 0) &&
- (compareFormats(formatToCompare, bestFormat) <= 0)) {
- uint32_t pickedSamplingRate = 0;
- audio_channel_mask_t pickedChannelMask = AUDIO_CHANNEL_NONE;
- pickChannelMask(pickedChannelMask, mProfiles[i]->getChannels());
- pickSamplingRate(pickedSamplingRate, mProfiles[i]->getSampleRates());
-
- if (formatToCompare != AUDIO_FORMAT_DEFAULT && pickedChannelMask != AUDIO_CHANNEL_NONE
- && pickedSamplingRate != 0) {
- format = formatToCompare;
- channelMask = pickedChannelMask;
- samplingRate = pickedSamplingRate;
- // TODO: shall we return on the first one or still trying to pick a better Profile?
- }
- }
- }
- ALOGV("%s Port[nm:%s] profile rate=%d, format=%d, channels=%d", __FUNCTION__, mName.string(),
- samplingRate, channelMask, format);
-}
-
-status_t AudioPort::checkGain(const struct audio_gain_config *gainConfig, int index) const
-{
- if (index < 0 || (size_t)index >= mGains.size()) {
- return BAD_VALUE;
- }
- return mGains[index]->checkConfig(gainConfig);
-}
-
-void AudioPort::dump(String8 *dst, int spaces, bool verbose) const
-{
- if (!mName.isEmpty()) {
- dst->appendFormat("%*s- name: %s\n", spaces, "", mName.string());
- }
- if (verbose) {
- mProfiles.dump(dst, spaces);
-
- if (mGains.size() != 0) {
- dst->appendFormat("%*s- gains:\n", spaces, "");
- for (size_t i = 0; i < mGains.size(); i++) {
- mGains[i]->dump(dst, spaces + 2, i);
- }
- }
- }
-}
-
-void AudioPort::log(const char* indent) const
-{
- ALOGI("%s Port[nm:%s, type:%d, role:%d]", indent, mName.string(), mType, mRole);
-}
-
-// --- AudioPortConfig class implementation
-
-status_t AudioPortConfig::applyAudioPortConfig(const struct audio_port_config *config,
- struct audio_port_config *backupConfig)
-{
- struct audio_port_config localBackupConfig = { .config_mask = config->config_mask };
- status_t status = NO_ERROR;
-
- toAudioPortConfig(&localBackupConfig);
-
- sp<AudioPort> audioport = getAudioPort();
- if (audioport == 0) {
- status = NO_INIT;
- goto exit;
- }
- status = audioport->checkExactAudioProfile(config);
- if (status != NO_ERROR) {
- goto exit;
- }
- if (config->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
- mSamplingRate = config->sample_rate;
- }
- if (config->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
- mChannelMask = config->channel_mask;
- }
- if (config->config_mask & AUDIO_PORT_CONFIG_FORMAT) {
- mFormat = config->format;
- }
- if (config->config_mask & AUDIO_PORT_CONFIG_GAIN) {
- mGain = config->gain;
- }
- if (config->config_mask & AUDIO_PORT_CONFIG_FLAGS) {
- mFlags = config->flags;
- }
-
-exit:
- if (status != NO_ERROR) {
- applyAudioPortConfig(&localBackupConfig);
- }
- if (backupConfig != NULL) {
- *backupConfig = localBackupConfig;
- }
- return status;
-}
-
-namespace {
-
-template<typename T>
-void updateField(
- const T& portConfigField, T audio_port_config::*port_config_field,
- struct audio_port_config *dstConfig, const struct audio_port_config *srcConfig,
- unsigned int configMask, T defaultValue)
-{
- if (dstConfig->config_mask & configMask) {
- if ((srcConfig != nullptr) && (srcConfig->config_mask & configMask)) {
- dstConfig->*port_config_field = srcConfig->*port_config_field;
- } else {
- dstConfig->*port_config_field = portConfigField;
- }
- } else {
- dstConfig->*port_config_field = defaultValue;
- }
-}
-
-} // namespace
-
-void AudioPortConfig::toAudioPortConfig(struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig) const
-{
- updateField(mSamplingRate, &audio_port_config::sample_rate,
- dstConfig, srcConfig, AUDIO_PORT_CONFIG_SAMPLE_RATE, 0u);
- updateField(mChannelMask, &audio_port_config::channel_mask,
- dstConfig, srcConfig, AUDIO_PORT_CONFIG_CHANNEL_MASK,
- (audio_channel_mask_t)AUDIO_CHANNEL_NONE);
- updateField(mFormat, &audio_port_config::format,
- dstConfig, srcConfig, AUDIO_PORT_CONFIG_FORMAT, AUDIO_FORMAT_INVALID);
-
- sp<AudioPort> audioport = getAudioPort();
- if ((dstConfig->config_mask & AUDIO_PORT_CONFIG_GAIN) && audioport != NULL) {
- dstConfig->gain = mGain;
- if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_GAIN)
- && audioport->checkGain(&srcConfig->gain, srcConfig->gain.index) == OK) {
- dstConfig->gain = srcConfig->gain;
- }
- } else {
- dstConfig->gain.index = -1;
- }
- if (dstConfig->gain.index != -1) {
- dstConfig->config_mask |= AUDIO_PORT_CONFIG_GAIN;
- } else {
- dstConfig->config_mask &= ~AUDIO_PORT_CONFIG_GAIN;
- }
-
- updateField(mFlags, &audio_port_config::flags,
- dstConfig, srcConfig, AUDIO_PORT_CONFIG_FLAGS, { AUDIO_INPUT_FLAG_NONE });
-}
-
-bool AudioPortConfig::hasGainController(bool canUseForVolume) const
-{
- sp<AudioPort> audioport = getAudioPort();
- if (audioport == nullptr) {
- return false;
- }
- return canUseForVolume ? audioport->getGains().canUseForVolume()
- : audioport->getGains().size() > 0;
-}
-
-} // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioProfile.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioProfile.cpp
deleted file mode 100644
index 69d6b0c..0000000
--- a/services/audiopolicy/common/managerdefinitions/src/AudioProfile.cpp
+++ /dev/null
@@ -1,621 +0,0 @@
-/*
- * Copyright (C) 2015 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include <algorithm>
-#include <set>
-#include <string>
-
-#define LOG_TAG "APM::AudioProfile"
-//#define LOG_NDEBUG 0
-
-#include <media/AudioResamplerPublic.h>
-#include <utils/Errors.h>
-
-#include "AudioGain.h"
-#include "AudioPort.h"
-#include "AudioProfile.h"
-#include "HwModule.h"
-#include "TypeConverter.h"
-
-namespace android {
-
-ChannelsVector ChannelsVector::asInMask() const
-{
- ChannelsVector inMaskVector;
- for (const auto& channel : *this) {
- if (audio_channel_mask_out_to_in(channel) != AUDIO_CHANNEL_INVALID) {
- inMaskVector.add(audio_channel_mask_out_to_in(channel));
- }
- }
- return inMaskVector;
-}
-
-ChannelsVector ChannelsVector::asOutMask() const
-{
- ChannelsVector outMaskVector;
- for (const auto& channel : *this) {
- if (audio_channel_mask_in_to_out(channel) != AUDIO_CHANNEL_INVALID) {
- outMaskVector.add(audio_channel_mask_in_to_out(channel));
- }
- }
- return outMaskVector;
-}
-
-bool operator == (const AudioProfile &left, const AudioProfile &compareTo)
-{
- return (left.getFormat() == compareTo.getFormat()) &&
- (left.getChannels() == compareTo.getChannels()) &&
- (left.getSampleRates() == compareTo.getSampleRates());
-}
-
-static AudioProfile* createFullDynamicImpl()
-{
- AudioProfile* dynamicProfile = new AudioProfile(gDynamicFormat,
- ChannelsVector(), SampleRateVector());
- dynamicProfile->setDynamicFormat(true);
- dynamicProfile->setDynamicChannels(true);
- dynamicProfile->setDynamicRate(true);
- return dynamicProfile;
-}
-
-// static
-sp<AudioProfile> AudioProfile::createFullDynamic()
-{
- static sp<AudioProfile> dynamicProfile = createFullDynamicImpl();
- return dynamicProfile;
-}
-
-AudioProfile::AudioProfile(audio_format_t format,
- audio_channel_mask_t channelMasks,
- uint32_t samplingRate) :
- mName(String8("")),
- mFormat(format)
-{
- mChannelMasks.add(channelMasks);
- mSamplingRates.add(samplingRate);
-}
-
-AudioProfile::AudioProfile(audio_format_t format,
- const ChannelsVector &channelMasks,
- const SampleRateVector &samplingRateCollection) :
- mName(String8("")),
- mFormat(format),
- mChannelMasks(channelMasks),
- mSamplingRates(samplingRateCollection) {}
-
-void AudioProfile::setChannels(const ChannelsVector &channelMasks)
-{
- if (mIsDynamicChannels) {
- mChannelMasks = channelMasks;
- }
-}
-
-void AudioProfile::setSampleRates(const SampleRateVector &sampleRates)
-{
- if (mIsDynamicRate) {
- mSamplingRates = sampleRates;
- }
-}
-
-void AudioProfile::clear()
-{
- if (mIsDynamicChannels) {
- mChannelMasks.clear();
- }
- if (mIsDynamicRate) {
- mSamplingRates.clear();
- }
-}
-
-status_t AudioProfile::checkExact(uint32_t samplingRate, audio_channel_mask_t channelMask,
- audio_format_t format) const
-{
- if (audio_formats_match(format, mFormat) &&
- supportsChannels(channelMask) &&
- supportsRate(samplingRate)) {
- return NO_ERROR;
- }
- return BAD_VALUE;
-}
-
-status_t AudioProfile::checkCompatibleSamplingRate(uint32_t samplingRate,
- uint32_t &updatedSamplingRate) const
-{
- ALOG_ASSERT(samplingRate > 0);
-
- if (mSamplingRates.isEmpty()) {
- updatedSamplingRate = samplingRate;
- return NO_ERROR;
- }
-
- // Search for the closest supported sampling rate that is above (preferred)
- // or below (acceptable) the desired sampling rate, within a permitted ratio.
- // The sampling rates are sorted in ascending order.
- size_t orderOfDesiredRate = mSamplingRates.orderOf(samplingRate);
-
- // Prefer to down-sample from a higher sampling rate, as we get the desired frequency spectrum.
- if (orderOfDesiredRate < mSamplingRates.size()) {
- uint32_t candidate = mSamplingRates[orderOfDesiredRate];
- if (candidate / AUDIO_RESAMPLER_DOWN_RATIO_MAX <= samplingRate) {
- updatedSamplingRate = candidate;
- return NO_ERROR;
- }
- }
- // But if we have to up-sample from a lower sampling rate, that's OK.
- if (orderOfDesiredRate != 0) {
- uint32_t candidate = mSamplingRates[orderOfDesiredRate - 1];
- if (candidate * AUDIO_RESAMPLER_UP_RATIO_MAX >= samplingRate) {
- updatedSamplingRate = candidate;
- return NO_ERROR;
- }
- }
- // leave updatedSamplingRate unmodified
- return BAD_VALUE;
-}
-
-status_t AudioProfile::checkCompatibleChannelMask(audio_channel_mask_t channelMask,
- audio_channel_mask_t &updatedChannelMask,
- audio_port_type_t portType,
- audio_port_role_t portRole) const
-{
- if (mChannelMasks.isEmpty()) {
- updatedChannelMask = channelMask;
- return NO_ERROR;
- }
- const bool isRecordThread = portType == AUDIO_PORT_TYPE_MIX && portRole == AUDIO_PORT_ROLE_SINK;
- const bool isIndex = audio_channel_mask_get_representation(channelMask)
- == AUDIO_CHANNEL_REPRESENTATION_INDEX;
- const uint32_t channelCount = audio_channel_count_from_in_mask(channelMask);
- int bestMatch = 0;
- for (size_t i = 0; i < mChannelMasks.size(); i ++) {
- audio_channel_mask_t supported = mChannelMasks[i];
- if (supported == channelMask) {
- // Exact matches always taken.
- updatedChannelMask = channelMask;
- return NO_ERROR;
- }
-
- // AUDIO_CHANNEL_NONE (value: 0) is used for dynamic channel support
- if (isRecordThread && supported != AUDIO_CHANNEL_NONE) {
- // Approximate (best) match:
- // The match score measures how well the supported channel mask matches the
- // desired mask, where increasing-is-better.
- //
- // TODO: Some tweaks may be needed.
- // Should be a static function of the data processing library.
- //
- // In priority:
- // match score = 1000 if legacy channel conversion equivalent (always prefer this)
- // OR
- // match score += 100 if the channel mask representations match
- // match score += number of channels matched.
- // match score += 100 if the channel mask representations DO NOT match
- // but the profile has positional channel mask and less than 2 channels.
- // This is for audio HAL convention to not list index masks for less than 2 channels
- //
- // If there are no matched channels, the mask may still be accepted
- // but the playback or record will be silent.
- const bool isSupportedIndex = (audio_channel_mask_get_representation(supported)
- == AUDIO_CHANNEL_REPRESENTATION_INDEX);
- const uint32_t supportedChannelCount = audio_channel_count_from_in_mask(supported);
- int match;
- if (isIndex && isSupportedIndex) {
- // index equivalence
- match = 100 + __builtin_popcount(
- audio_channel_mask_get_bits(channelMask)
- & audio_channel_mask_get_bits(supported));
- } else if (isIndex && !isSupportedIndex) {
- const uint32_t equivalentBits = (1 << supportedChannelCount) - 1 ;
- match = __builtin_popcount(
- audio_channel_mask_get_bits(channelMask) & equivalentBits);
- if (supportedChannelCount <= FCC_2) {
- match += 100;
- }
- } else if (!isIndex && isSupportedIndex) {
- const uint32_t equivalentBits = (1 << channelCount) - 1;
- match = __builtin_popcount(
- equivalentBits & audio_channel_mask_get_bits(supported));
- } else {
- // positional equivalence
- match = 100 + __builtin_popcount(
- audio_channel_mask_get_bits(channelMask)
- & audio_channel_mask_get_bits(supported));
- switch (supported) {
- case AUDIO_CHANNEL_IN_FRONT_BACK:
- case AUDIO_CHANNEL_IN_STEREO:
- if (channelMask == AUDIO_CHANNEL_IN_MONO) {
- match = 1000;
- }
- break;
- case AUDIO_CHANNEL_IN_MONO:
- if (channelMask == AUDIO_CHANNEL_IN_FRONT_BACK
- || channelMask == AUDIO_CHANNEL_IN_STEREO) {
- match = 1000;
- }
- break;
- default:
- break;
- }
- }
- if (match > bestMatch) {
- bestMatch = match;
- updatedChannelMask = supported;
- }
- }
- }
- return bestMatch > 0 ? NO_ERROR : BAD_VALUE;
-}
-
-void AudioProfile::dump(String8 *dst, int spaces) const
-{
- dst->appendFormat("%s%s%s\n", mIsDynamicFormat ? "[dynamic format]" : "",
- mIsDynamicChannels ? "[dynamic channels]" : "",
- mIsDynamicRate ? "[dynamic rates]" : "");
- if (mName.length() != 0) {
- dst->appendFormat("%*s- name: %s\n", spaces, "", mName.string());
- }
- std::string formatLiteral;
- if (FormatConverter::toString(mFormat, formatLiteral)) {
- dst->appendFormat("%*s- format: %s\n", spaces, "", formatLiteral.c_str());
- }
- if (!mSamplingRates.isEmpty()) {
- dst->appendFormat("%*s- sampling rates:", spaces, "");
- for (size_t i = 0; i < mSamplingRates.size(); i++) {
- dst->appendFormat("%d", mSamplingRates[i]);
- dst->append(i == (mSamplingRates.size() - 1) ? "" : ", ");
- }
- dst->append("\n");
- }
-
- if (!mChannelMasks.isEmpty()) {
- dst->appendFormat("%*s- channel masks:", spaces, "");
- for (size_t i = 0; i < mChannelMasks.size(); i++) {
- dst->appendFormat("0x%04x", mChannelMasks[i]);
- dst->append(i == (mChannelMasks.size() - 1) ? "" : ", ");
- }
- dst->append("\n");
- }
-}
-
-ssize_t AudioProfileVector::add(const sp<AudioProfile> &profile)
-{
- ssize_t index = Vector::add(profile);
- // we sort from worst to best, so that AUDIO_FORMAT_DEFAULT is always the first entry.
- // TODO: compareFormats could be a lambda to convert between pointer-to-format to format:
- // [](const audio_format_t *format1, const audio_format_t *format2) {
- // return compareFormats(*format1, *format2);
- // }
- sort(compareFormats);
- return index;
-}
-
-ssize_t AudioProfileVector::addProfileFromHal(const sp<AudioProfile> &profileToAdd)
-{
- // Check valid profile to add:
- if (!profileToAdd->hasValidFormat()) {
- return -1;
- }
- if (!profileToAdd->hasValidChannels() && !profileToAdd->hasValidRates()) {
- FormatVector formats;
- formats.add(profileToAdd->getFormat());
- setFormats(FormatVector(formats));
- return 0;
- }
- if (!profileToAdd->hasValidChannels() && profileToAdd->hasValidRates()) {
- setSampleRatesFor(profileToAdd->getSampleRates(), profileToAdd->getFormat());
- return 0;
- }
- if (profileToAdd->hasValidChannels() && !profileToAdd->hasValidRates()) {
- setChannelsFor(profileToAdd->getChannels(), profileToAdd->getFormat());
- return 0;
- }
- // Go through the list of profile to avoid duplicates
- for (size_t profileIndex = 0; profileIndex < size(); profileIndex++) {
- const sp<AudioProfile> &profile = itemAt(profileIndex);
- if (profile->isValid() && profile == profileToAdd) {
- // Nothing to do
- return profileIndex;
- }
- }
- profileToAdd->setDynamicFormat(true); // set the format as dynamic to allow removal
- return add(profileToAdd);
-}
-
-status_t AudioProfileVector::checkExactProfile(uint32_t samplingRate,
- audio_channel_mask_t channelMask,
- audio_format_t format) const
-{
- if (isEmpty()) {
- return NO_ERROR;
- }
-
- for (const auto& profile : *this) {
- if (profile->checkExact(samplingRate, channelMask, format) == NO_ERROR) {
- return NO_ERROR;
- }
- }
- return BAD_VALUE;
-}
-
-status_t AudioProfileVector::checkCompatibleProfile(uint32_t &samplingRate,
- audio_channel_mask_t &channelMask,
- audio_format_t &format,
- audio_port_type_t portType,
- audio_port_role_t portRole) const
-{
- if (isEmpty()) {
- return NO_ERROR;
- }
-
- const bool checkInexact = // when port is input and format is linear pcm
- portType == AUDIO_PORT_TYPE_MIX && portRole == AUDIO_PORT_ROLE_SINK
- && audio_is_linear_pcm(format);
-
- // iterate from best format to worst format (reverse order)
- for (ssize_t i = size() - 1; i >= 0 ; --i) {
- const sp<AudioProfile> profile = itemAt(i);
- audio_format_t formatToCompare = profile->getFormat();
- if (formatToCompare == format ||
- (checkInexact
- && formatToCompare != AUDIO_FORMAT_DEFAULT
- && audio_is_linear_pcm(formatToCompare))) {
- // Compatible profile has been found, checks if this profile has compatible
- // rate and channels as well
- audio_channel_mask_t updatedChannels;
- uint32_t updatedRate;
- if (profile->checkCompatibleChannelMask(channelMask, updatedChannels,
- portType, portRole) == NO_ERROR &&
- profile->checkCompatibleSamplingRate(samplingRate, updatedRate) == NO_ERROR) {
- // for inexact checks we take the first linear pcm format due to sorting.
- format = formatToCompare;
- channelMask = updatedChannels;
- samplingRate = updatedRate;
- return NO_ERROR;
- }
- }
- }
- return BAD_VALUE;
-}
-
-void AudioProfileVector::clearProfiles()
-{
- for (size_t i = size(); i != 0; ) {
- sp<AudioProfile> profile = itemAt(--i);
- if (profile->isDynamicFormat() && profile->hasValidFormat()) {
- removeAt(i);
- continue;
- }
- profile->clear();
- }
-}
-
-// Returns an intersection between two possibly unsorted vectors and the contents of 'order'.
-// The result is ordered according to 'order'.
-template<typename T, typename Order>
-std::vector<typename T::value_type> intersectFilterAndOrder(
- const T& input1, const T& input2, const Order& order)
-{
- std::set<typename T::value_type> set1{input1.begin(), input1.end()};
- std::set<typename T::value_type> set2{input2.begin(), input2.end()};
- std::set<typename T::value_type> common;
- std::set_intersection(set1.begin(), set1.end(), set2.begin(), set2.end(),
- std::inserter(common, common.begin()));
- std::vector<typename T::value_type> result;
- for (const auto& e : order) {
- if (common.find(e) != common.end()) result.push_back(e);
- }
- return result;
-}
-
-// Intersect two possibly unsorted vectors, return common elements according to 'comp' ordering.
-// 'comp' is a comparator function.
-template<typename T, typename Compare>
-std::vector<typename T::value_type> intersectAndOrder(
- const T& input1, const T& input2, Compare comp)
-{
- std::set<typename T::value_type, Compare> set1{input1.begin(), input1.end(), comp};
- std::set<typename T::value_type, Compare> set2{input2.begin(), input2.end(), comp};
- std::vector<typename T::value_type> result;
- std::set_intersection(set1.begin(), set1.end(), set2.begin(), set2.end(),
- std::back_inserter(result), comp);
- return result;
-}
-
-status_t AudioProfileVector::findBestMatchingOutputConfig(const AudioProfileVector& outputProfiles,
- const std::vector<audio_format_t>& preferredFormats,
- const std::vector<audio_channel_mask_t>& preferredOutputChannels,
- bool preferHigherSamplingRates,
- audio_config_base *bestOutputConfig) const
-{
- auto formats = intersectFilterAndOrder(getSupportedFormats(),
- outputProfiles.getSupportedFormats(), preferredFormats);
- // Pick the best compatible profile.
- for (const auto& f : formats) {
- sp<AudioProfile> inputProfile = getFirstValidProfileFor(f);
- sp<AudioProfile> outputProfile = outputProfiles.getFirstValidProfileFor(f);
- if (inputProfile == nullptr || outputProfile == nullptr) {
- continue;
- }
- auto channels = intersectFilterAndOrder(inputProfile->getChannels().asOutMask(),
- outputProfile->getChannels(), preferredOutputChannels);
- if (channels.empty()) {
- continue;
- }
- auto sampleRates = preferHigherSamplingRates ?
- intersectAndOrder(inputProfile->getSampleRates(), outputProfile->getSampleRates(),
- std::greater<typename SampleRateVector::value_type>()) :
- intersectAndOrder(inputProfile->getSampleRates(), outputProfile->getSampleRates(),
- std::less<typename SampleRateVector::value_type>());
- if (sampleRates.empty()) {
- continue;
- }
- ALOGD("%s() found channel mask %#x and sample rate %d for format %#x.",
- __func__, *channels.begin(), *sampleRates.begin(), f);
- bestOutputConfig->format = f;
- bestOutputConfig->sample_rate = *sampleRates.begin();
- bestOutputConfig->channel_mask = *channels.begin();
- return NO_ERROR;
- }
- return BAD_VALUE;
-}
-
-sp<AudioProfile> AudioProfileVector::getFirstValidProfile() const
-{
- for (size_t i = 0; i < size(); i++) {
- if (itemAt(i)->isValid()) {
- return itemAt(i);
- }
- }
- return 0;
-}
-
-sp<AudioProfile> AudioProfileVector::getFirstValidProfileFor(audio_format_t format) const
-{
- for (size_t i = 0; i < size(); i++) {
- if (itemAt(i)->isValid() && itemAt(i)->getFormat() == format) {
- return itemAt(i);
- }
- }
- return 0;
-}
-
-FormatVector AudioProfileVector::getSupportedFormats() const
-{
- FormatVector supportedFormats;
- for (size_t i = 0; i < size(); i++) {
- if (itemAt(i)->hasValidFormat()) {
- supportedFormats.add(itemAt(i)->getFormat());
- }
- }
- return supportedFormats;
-}
-
-bool AudioProfileVector::hasDynamicChannelsFor(audio_format_t format) const
-{
- for (size_t i = 0; i < size(); i++) {
- sp<AudioProfile> profile = itemAt(i);
- if (profile->getFormat() == format && profile->isDynamicChannels()) {
- return true;
- }
- }
- return false;
-}
-
-bool AudioProfileVector::hasDynamicProfile() const
-{
- for (size_t i = 0; i < size(); i++) {
- if (itemAt(i)->isDynamic()) {
- return true;
- }
- }
- return false;
-}
-
-bool AudioProfileVector::hasDynamicRateFor(audio_format_t format) const
-{
- for (size_t i = 0; i < size(); i++) {
- sp<AudioProfile> profile = itemAt(i);
- if (profile->getFormat() == format && profile->isDynamicRate()) {
- return true;
- }
- }
- return false;
-}
-
-void AudioProfileVector::setFormats(const FormatVector &formats)
-{
- // Only allow to change the format of dynamic profile
- sp<AudioProfile> dynamicFormatProfile = getProfileFor(gDynamicFormat);
- if (dynamicFormatProfile == 0) {
- return;
- }
- for (size_t i = 0; i < formats.size(); i++) {
- sp<AudioProfile> profile = new AudioProfile(formats[i],
- dynamicFormatProfile->getChannels(),
- dynamicFormatProfile->getSampleRates());
- profile->setDynamicFormat(true);
- profile->setDynamicChannels(dynamicFormatProfile->isDynamicChannels());
- profile->setDynamicRate(dynamicFormatProfile->isDynamicRate());
- add(profile);
- }
-}
-
-void AudioProfileVector::dump(String8 *dst, int spaces) const
-{
- dst->appendFormat("%*s- Profiles:\n", spaces, "");
- for (size_t i = 0; i < size(); i++) {
- dst->appendFormat("%*sProfile %zu:", spaces + 4, "", i);
- itemAt(i)->dump(dst, spaces + 8);
- }
-}
-
-sp<AudioProfile> AudioProfileVector::getProfileFor(audio_format_t format) const
-{
- for (size_t i = 0; i < size(); i++) {
- if (itemAt(i)->getFormat() == format) {
- return itemAt(i);
- }
- }
- return 0;
-}
-
-void AudioProfileVector::setSampleRatesFor(
- const SampleRateVector &sampleRates, audio_format_t format)
-{
- for (size_t i = 0; i < size(); i++) {
- sp<AudioProfile> profile = itemAt(i);
- if (profile->getFormat() == format && profile->isDynamicRate()) {
- if (profile->hasValidRates()) {
- // Need to create a new profile with same format
- sp<AudioProfile> profileToAdd = new AudioProfile(format, profile->getChannels(),
- sampleRates);
- profileToAdd->setDynamicFormat(true); // need to set to allow cleaning
- add(profileToAdd);
- } else {
- profile->setSampleRates(sampleRates);
- }
- return;
- }
- }
-}
-
-void AudioProfileVector::setChannelsFor(const ChannelsVector &channelMasks, audio_format_t format)
-{
- for (size_t i = 0; i < size(); i++) {
- sp<AudioProfile> profile = itemAt(i);
- if (profile->getFormat() == format && profile->isDynamicChannels()) {
- if (profile->hasValidChannels()) {
- // Need to create a new profile with same format
- sp<AudioProfile> profileToAdd = new AudioProfile(format, channelMasks,
- profile->getSampleRates());
- profileToAdd->setDynamicFormat(true); // need to set to allow cleaning
- add(profileToAdd);
- } else {
- profile->setChannels(channelMasks);
- }
- return;
- }
- }
-}
-
-// static
-int AudioProfileVector::compareFormats(const sp<AudioProfile> *profile1,
- const sp<AudioProfile> *profile2)
-{
- return AudioPort::compareFormats((*profile1)->getFormat(), (*profile2)->getFormat());
-}
-
-} // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioProfileVectorHelper.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioProfileVectorHelper.cpp
new file mode 100644
index 0000000..8ccb8b9
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioProfileVectorHelper.cpp
@@ -0,0 +1,439 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <algorithm>
+#include <set>
+#include <string>
+
+#define LOG_TAG "APM::AudioProfileVectorHelper"
+//#define LOG_NDEBUG 0
+
+#include <media/AudioContainers.h>
+#include <media/AudioResamplerPublic.h>
+#include <utils/Errors.h>
+
+#include "AudioProfileVectorHelper.h"
+#include "HwModule.h"
+#include "PolicyAudioPort.h"
+#include "policy.h"
+
+namespace android {
+
+void sortAudioProfiles(AudioProfileVector &audioProfileVector) {
+ std::sort(audioProfileVector.begin(), audioProfileVector.end(),
+ [](const sp<AudioProfile> & a, const sp<AudioProfile> & b)
+ {
+ return PolicyAudioPort::compareFormats(a->getFormat(), b->getFormat()) < 0;
+ });
+}
+
+ssize_t addAudioProfileAndSort(AudioProfileVector &audioProfileVector,
+ const sp<AudioProfile> &profile)
+{
+ ssize_t ret = audioProfileVector.add(profile);
+ // we sort from worst to best, so that AUDIO_FORMAT_DEFAULT is always the first entry.
+ sortAudioProfiles(audioProfileVector);
+ return ret;
+}
+
+sp<AudioProfile> getAudioProfileForFormat(const AudioProfileVector &audioProfileVector,
+ audio_format_t format)
+{
+ for (const auto &profile : audioProfileVector) {
+ if (profile->getFormat() == format) {
+ return profile;
+ }
+ }
+ return nullptr;
+}
+
+void setSampleRatesForAudioProfiles(AudioProfileVector &audioProfileVector,
+ const SampleRateSet &sampleRateSet,
+ audio_format_t format)
+{
+ for (const auto &profile : audioProfileVector) {
+ if (profile->getFormat() == format && profile->isDynamicRate()) {
+ if (profile->hasValidRates()) {
+ // Need to create a new profile with same format
+ sp<AudioProfile> profileToAdd = new AudioProfile(
+ format, profile->getChannels(), sampleRateSet);
+ profileToAdd->setDynamicFormat(true); // need to set to allow cleaning
+ addAudioProfileAndSort(audioProfileVector, profileToAdd);
+ } else {
+ profile->setSampleRates(sampleRateSet);
+ }
+ return;
+ }
+ }
+}
+
+void setChannelsForAudioProfiles(AudioProfileVector &audioProfileVector,
+ const ChannelMaskSet &channelMaskSet,
+ audio_format_t format)
+{
+ for (const auto &profile : audioProfileVector) {
+ if (profile->getFormat() == format && profile->isDynamicChannels()) {
+ if (profile->hasValidChannels()) {
+ // Need to create a new profile with same format
+ sp<AudioProfile> profileToAdd = new AudioProfile(format, channelMaskSet,
+ profile->getSampleRates());
+ profileToAdd->setDynamicFormat(true); // need to set to allow cleaning
+ addAudioProfileAndSort(audioProfileVector, profileToAdd);
+ } else {
+ profile->setChannels(channelMaskSet);
+ }
+ return;
+ }
+ }
+}
+
+void addProfilesForFormats(AudioProfileVector &audioProfileVector, const FormatVector &formatVector)
+{
+ // Only allow to change the format of dynamic profile
+ sp<AudioProfile> dynamicFormatProfile = getAudioProfileForFormat(
+ audioProfileVector, gDynamicFormat);
+ if (!dynamicFormatProfile) {
+ return;
+ }
+ for (const auto &format : formatVector) {
+ sp<AudioProfile> profile = new AudioProfile(format,
+ dynamicFormatProfile->getChannels(),
+ dynamicFormatProfile->getSampleRates());
+ profile->setDynamicFormat(true);
+ profile->setDynamicChannels(dynamicFormatProfile->isDynamicChannels());
+ profile->setDynamicRate(dynamicFormatProfile->isDynamicRate());
+ addAudioProfileAndSort(audioProfileVector, profile);
+ }
+}
+
+void addDynamicAudioProfileAndSort(AudioProfileVector &audioProfileVector,
+ const sp<AudioProfile> &profileToAdd)
+{
+ // Check valid profile to add:
+ if (!profileToAdd->hasValidFormat()) {
+ ALOGW("Adding dynamic audio profile without valid format");
+ return;
+ }
+ if (!profileToAdd->hasValidChannels() && !profileToAdd->hasValidRates()) {
+ FormatVector formats;
+ formats.push_back(profileToAdd->getFormat());
+ addProfilesForFormats(audioProfileVector, FormatVector(formats));
+ return;
+ }
+ if (!profileToAdd->hasValidChannels() && profileToAdd->hasValidRates()) {
+ setSampleRatesForAudioProfiles(
+ audioProfileVector, profileToAdd->getSampleRates(), profileToAdd->getFormat());
+ return;
+ }
+ if (profileToAdd->hasValidChannels() && !profileToAdd->hasValidRates()) {
+ setChannelsForAudioProfiles(
+ audioProfileVector, profileToAdd->getChannels(), profileToAdd->getFormat());
+ return;
+ }
+ // Go through the list of profile to avoid duplicates
+ for (size_t profileIndex = 0; profileIndex < audioProfileVector.size(); profileIndex++) {
+ const sp<AudioProfile> &profile = audioProfileVector.at(profileIndex);
+ if (profile->isValid() && profile == profileToAdd) {
+ // Nothing to do
+ return;
+ }
+ }
+ profileToAdd->setDynamicFormat(true); // set the format as dynamic to allow removal
+ addAudioProfileAndSort(audioProfileVector, profileToAdd);
+}
+
+void appendAudioProfiles(AudioProfileVector &audioProfileVector,
+ const AudioProfileVector &audioProfileVectorToAppend)
+{
+ audioProfileVector.insert(audioProfileVector.end(),
+ audioProfileVectorToAppend.begin(),
+ audioProfileVectorToAppend.end());
+}
+
+status_t checkExact(const sp<AudioProfile> &audioProfile,
+ uint32_t samplingRate,
+ audio_channel_mask_t channelMask,
+ audio_format_t format)
+{
+ if (audio_formats_match(format, audioProfile->getFormat()) &&
+ audioProfile->supportsChannels(channelMask) &&
+ audioProfile->supportsRate(samplingRate)) {
+ return NO_ERROR;
+ }
+ return BAD_VALUE;
+}
+
+status_t checkCompatibleSamplingRate(const sp<AudioProfile> &audioProfile,
+ uint32_t samplingRate,
+ uint32_t &updatedSamplingRate)
+{
+ ALOG_ASSERT(samplingRate > 0);
+
+ const SampleRateSet sampleRates = audioProfile->getSampleRates();
+ if (sampleRates.empty()) {
+ updatedSamplingRate = samplingRate;
+ return NO_ERROR;
+ }
+
+ // Search for the closest supported sampling rate that is above (preferred)
+ // or below (acceptable) the desired sampling rate, within a permitted ratio.
+ // The sampling rates are sorted in ascending order.
+ auto desiredRate = sampleRates.lower_bound(samplingRate);
+
+ // Prefer to down-sample from a higher sampling rate, as we get the desired frequency spectrum.
+ if (desiredRate != sampleRates.end()) {
+ if (*desiredRate / AUDIO_RESAMPLER_DOWN_RATIO_MAX <= samplingRate) {
+ updatedSamplingRate = *desiredRate;
+ return NO_ERROR;
+ }
+ }
+ // But if we have to up-sample from a lower sampling rate, that's OK.
+ if (desiredRate != sampleRates.begin()) {
+ uint32_t candidate = *(--desiredRate);
+ if (candidate * AUDIO_RESAMPLER_UP_RATIO_MAX >= samplingRate) {
+ updatedSamplingRate = candidate;
+ return NO_ERROR;
+ }
+ }
+ // leave updatedSamplingRate unmodified
+ return BAD_VALUE;
+}
+
+status_t checkCompatibleChannelMask(const sp<AudioProfile> &audioProfile,
+ audio_channel_mask_t channelMask,
+ audio_channel_mask_t &updatedChannelMask,
+ audio_port_type_t portType,
+ audio_port_role_t portRole)
+{
+ const ChannelMaskSet channelMasks = audioProfile->getChannels();
+ if (channelMasks.empty()) {
+ updatedChannelMask = channelMask;
+ return NO_ERROR;
+ }
+ const bool isRecordThread = portType == AUDIO_PORT_TYPE_MIX && portRole == AUDIO_PORT_ROLE_SINK;
+ const bool isIndex = audio_channel_mask_get_representation(channelMask)
+ == AUDIO_CHANNEL_REPRESENTATION_INDEX;
+ const uint32_t channelCount = audio_channel_count_from_in_mask(channelMask);
+ int bestMatch = 0;
+ for (const auto &supported : channelMasks) {
+ if (supported == channelMask) {
+ // Exact matches always taken.
+ updatedChannelMask = channelMask;
+ return NO_ERROR;
+ }
+
+ // AUDIO_CHANNEL_NONE (value: 0) is used for dynamic channel support
+ if (isRecordThread && supported != AUDIO_CHANNEL_NONE) {
+ // Approximate (best) match:
+ // The match score measures how well the supported channel mask matches the
+ // desired mask, where increasing-is-better.
+ //
+ // TODO: Some tweaks may be needed.
+ // Should be a static function of the data processing library.
+ //
+ // In priority:
+ // match score = 1000 if legacy channel conversion equivalent (always prefer this)
+ // OR
+ // match score += 100 if the channel mask representations match
+ // match score += number of channels matched.
+ // match score += 100 if the channel mask representations DO NOT match
+ // but the profile has positional channel mask and less than 2 channels.
+ // This is for audio HAL convention to not list index masks for less than 2 channels
+ //
+ // If there are no matched channels, the mask may still be accepted
+ // but the playback or record will be silent.
+ const bool isSupportedIndex = (audio_channel_mask_get_representation(supported)
+ == AUDIO_CHANNEL_REPRESENTATION_INDEX);
+ const uint32_t supportedChannelCount = audio_channel_count_from_in_mask(supported);
+ int match;
+ if (isIndex && isSupportedIndex) {
+ // index equivalence
+ match = 100 + __builtin_popcount(
+ audio_channel_mask_get_bits(channelMask)
+ & audio_channel_mask_get_bits(supported));
+ } else if (isIndex && !isSupportedIndex) {
+ const uint32_t equivalentBits = (1 << supportedChannelCount) - 1 ;
+ match = __builtin_popcount(
+ audio_channel_mask_get_bits(channelMask) & equivalentBits);
+ if (supportedChannelCount <= FCC_2) {
+ match += 100;
+ }
+ } else if (!isIndex && isSupportedIndex) {
+ const uint32_t equivalentBits = (1 << channelCount) - 1;
+ match = __builtin_popcount(
+ equivalentBits & audio_channel_mask_get_bits(supported));
+ } else {
+ // positional equivalence
+ match = 100 + __builtin_popcount(
+ audio_channel_mask_get_bits(channelMask)
+ & audio_channel_mask_get_bits(supported));
+ switch (supported) {
+ case AUDIO_CHANNEL_IN_FRONT_BACK:
+ case AUDIO_CHANNEL_IN_STEREO:
+ if (channelMask == AUDIO_CHANNEL_IN_MONO) {
+ match = 1000;
+ }
+ break;
+ case AUDIO_CHANNEL_IN_MONO:
+ if (channelMask == AUDIO_CHANNEL_IN_FRONT_BACK
+ || channelMask == AUDIO_CHANNEL_IN_STEREO) {
+ match = 1000;
+ }
+ break;
+ default:
+ break;
+ }
+ }
+ if (match > bestMatch) {
+ bestMatch = match;
+ updatedChannelMask = supported;
+ }
+ }
+ }
+ return bestMatch > 0 ? NO_ERROR : BAD_VALUE;
+}
+
+status_t checkExactProfile(const AudioProfileVector& audioProfileVector,
+ const uint32_t samplingRate,
+ audio_channel_mask_t channelMask,
+ audio_format_t format)
+{
+ if (audioProfileVector.empty()) {
+ return NO_ERROR;
+ }
+
+ for (const auto& profile : audioProfileVector) {
+ if (checkExact(profile, samplingRate, channelMask, format) == NO_ERROR) {
+ return NO_ERROR;
+ }
+ }
+ return BAD_VALUE;
+}
+
+status_t checkCompatibleProfile(const AudioProfileVector &audioProfileVector,
+ uint32_t &samplingRate,
+ audio_channel_mask_t &channelMask,
+ audio_format_t &format,
+ audio_port_type_t portType,
+ audio_port_role_t portRole)
+{
+ if (audioProfileVector.empty()) {
+ return NO_ERROR;
+ }
+
+ const bool checkInexact = // when port is input and format is linear pcm
+ portType == AUDIO_PORT_TYPE_MIX && portRole == AUDIO_PORT_ROLE_SINK
+ && audio_is_linear_pcm(format);
+
+ // iterate from best format to worst format (reverse order)
+ for (ssize_t i = audioProfileVector.size() - 1; i >= 0 ; --i) {
+ const sp<AudioProfile> profile = audioProfileVector.at(i);
+ audio_format_t formatToCompare = profile->getFormat();
+ if (formatToCompare == format ||
+ (checkInexact
+ && formatToCompare != AUDIO_FORMAT_DEFAULT
+ && audio_is_linear_pcm(formatToCompare))) {
+ // Compatible profile has been found, checks if this profile has compatible
+ // rate and channels as well
+ audio_channel_mask_t updatedChannels;
+ uint32_t updatedRate;
+ if (checkCompatibleChannelMask(profile, channelMask, updatedChannels,
+ portType, portRole) == NO_ERROR &&
+ checkCompatibleSamplingRate(profile, samplingRate, updatedRate) == NO_ERROR) {
+ // for inexact checks we take the first linear pcm format due to sorting.
+ format = formatToCompare;
+ channelMask = updatedChannels;
+ samplingRate = updatedRate;
+ return NO_ERROR;
+ }
+ }
+ }
+ return BAD_VALUE;
+}
+
+// Returns an intersection between two possibly unsorted vectors and the contents of 'order'.
+// The result is ordered according to 'order'.
+template<typename T, typename Order>
+std::vector<typename T::value_type> intersectFilterAndOrder(
+ const T& input1, const T& input2, const Order& order)
+{
+ std::set<typename T::value_type> set1{input1.begin(), input1.end()};
+ std::set<typename T::value_type> set2{input2.begin(), input2.end()};
+ std::set<typename T::value_type> common;
+ std::set_intersection(set1.begin(), set1.end(), set2.begin(), set2.end(),
+ std::inserter(common, common.begin()));
+ std::vector<typename T::value_type> result;
+ for (const auto& e : order) {
+ if (common.find(e) != common.end()) result.push_back(e);
+ }
+ return result;
+}
+
+// Intersect two possibly unsorted vectors, return common elements according to 'comp' ordering.
+// 'comp' is a comparator function.
+template<typename T, typename Compare>
+std::vector<typename T::value_type> intersectAndOrder(
+ const T& input1, const T& input2, Compare comp)
+{
+ std::set<typename T::value_type, Compare> set1{input1.begin(), input1.end(), comp};
+ std::set<typename T::value_type, Compare> set2{input2.begin(), input2.end(), comp};
+ std::vector<typename T::value_type> result;
+ std::set_intersection(set1.begin(), set1.end(), set2.begin(), set2.end(),
+ std::back_inserter(result), comp);
+ return result;
+}
+
+status_t findBestMatchingOutputConfig(
+ const AudioProfileVector &audioProfileVector,
+ const AudioProfileVector &outputProfileVector,
+ const std::vector<audio_format_t> &preferredFormatVector, // order: most pref -> least pref
+ const std::vector<audio_channel_mask_t> &preferredOutputChannelVector,
+ bool preferHigherSamplingRates,
+ audio_config_base &bestOutputConfig)
+{
+ auto formats = intersectFilterAndOrder(audioProfileVector.getSupportedFormats(),
+ outputProfileVector.getSupportedFormats(), preferredFormatVector);
+ // Pick the best compatible profile.
+ for (const auto& f : formats) {
+ sp<AudioProfile> inputProfile = audioProfileVector.getFirstValidProfileFor(f);
+ sp<AudioProfile> outputProfile = outputProfileVector.getFirstValidProfileFor(f);
+ if (inputProfile == nullptr || outputProfile == nullptr) {
+ continue;
+ }
+ auto channels = intersectFilterAndOrder(asOutMask(inputProfile->getChannels()),
+ outputProfile->getChannels(), preferredOutputChannelVector);
+ if (channels.empty()) {
+ continue;
+ }
+ auto sampleRates = preferHigherSamplingRates ?
+ intersectAndOrder(inputProfile->getSampleRates(), outputProfile->getSampleRates(),
+ std::greater<typename SampleRateSet::value_type>()) :
+ intersectAndOrder(inputProfile->getSampleRates(), outputProfile->getSampleRates(),
+ std::less<typename SampleRateSet::value_type>());
+ if (sampleRates.empty()) {
+ continue;
+ }
+ ALOGD("%s() found channel mask %#x and sample rate %d for format %#x.",
+ __func__, *channels.begin(), *sampleRates.begin(), f);
+ bestOutputConfig.format = f;
+ bestOutputConfig.sample_rate = *sampleRates.begin();
+ bestOutputConfig.channel_mask = *channels.begin();
+ return NO_ERROR;
+ }
+ return BAD_VALUE;
+}
+
+} // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioRoute.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioRoute.cpp
index 79f0919..2a18f19 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioRoute.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioRoute.cpp
@@ -19,7 +19,6 @@
#include "AudioRoute.h"
#include "HwModule.h"
-#include "AudioGain.h"
namespace android
{
@@ -27,25 +26,26 @@
void AudioRoute::dump(String8 *dst, int spaces) const
{
dst->appendFormat("%*s- Type: %s\n", spaces, "", mType == AUDIO_ROUTE_MUX ? "Mux" : "Mix");
- dst->appendFormat("%*s- Sink: %s\n", spaces, "", mSink->getTagName().string());
+ dst->appendFormat("%*s- Sink: %s\n", spaces, "", mSink->getTagName().c_str());
if (mSources.size() != 0) {
dst->appendFormat("%*s- Sources: \n", spaces, "");
for (size_t i = 0; i < mSources.size(); i++) {
- dst->appendFormat("%*s%s \n", spaces + 4, "", mSources[i]->getTagName().string());
+ dst->appendFormat("%*s%s \n", spaces + 4, "", mSources[i]->getTagName().c_str());
}
}
dst->append("\n");
}
-bool AudioRoute::supportsPatch(const sp<AudioPort> &srcPort, const sp<AudioPort> &dstPort) const
+bool AudioRoute::supportsPatch(const sp<PolicyAudioPort> &srcPort,
+ const sp<PolicyAudioPort> &dstPort) const
{
if (mSink == 0 || dstPort == 0 || dstPort != mSink) {
return false;
}
- ALOGV("%s: sinks %s matching", __FUNCTION__, mSink->getTagName().string());
+ ALOGV("%s: sinks %s matching", __FUNCTION__, mSink->getTagName().c_str());
for (const auto &sourcePort : mSources) {
if (sourcePort == srcPort) {
- ALOGV("%s: sources %s matching", __FUNCTION__, sourcePort->getTagName().string());
+ ALOGV("%s: sources %s matching", __FUNCTION__, sourcePort->getTagName().c_str());
return true;
}
}
diff --git a/services/audiopolicy/common/managerdefinitions/src/ClientDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/ClientDescriptor.cpp
index ad07ab1..95822b9 100644
--- a/services/audiopolicy/common/managerdefinitions/src/ClientDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/ClientDescriptor.cpp
@@ -21,7 +21,6 @@
#include <utils/Log.h>
#include <utils/String8.h>
#include <TypeConverter.h>
-#include "AudioGain.h"
#include "AudioOutputDescriptor.h"
#include "AudioPatch.h"
#include "ClientDescriptor.h"
@@ -83,14 +82,13 @@
}
SourceClientDescriptor::SourceClientDescriptor(audio_port_handle_t portId, uid_t uid,
- audio_attributes_t attributes, const sp<AudioPatch>& patchDesc,
+ audio_attributes_t attributes, const struct audio_port_config &config,
const sp<DeviceDescriptor>& srcDevice, audio_stream_type_t stream,
product_strategy_t strategy, VolumeSource volumeSource) :
TrackClientDescriptor::TrackClientDescriptor(portId, uid, AUDIO_SESSION_NONE, attributes,
- AUDIO_CONFIG_BASE_INITIALIZER, AUDIO_PORT_HANDLE_NONE,
+ {config.sample_rate, config.channel_mask, config.format}, AUDIO_PORT_HANDLE_NONE,
stream, strategy, volumeSource, AUDIO_OUTPUT_FLAG_NONE, false,
- {} /* Sources do not support secondary outputs*/),
- mPatchDesc(patchDesc), mSrcDevice(srcDevice)
+ {} /* Sources do not support secondary outputs*/), mSrcDevice(srcDevice)
{
}
diff --git a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
index ecd5b34..86dbba8 100644
--- a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
@@ -22,52 +22,56 @@
#include <set>
#include "DeviceDescriptor.h"
#include "TypeConverter.h"
-#include "AudioGain.h"
#include "HwModule.h"
namespace android {
-DeviceDescriptor::DeviceDescriptor(audio_devices_t type, const String8 &tagName) :
- DeviceDescriptor(type, FormatVector{}, tagName)
+DeviceDescriptor::DeviceDescriptor(audio_devices_t type) :
+ DeviceDescriptor(type, "" /*tagName*/)
{
}
-DeviceDescriptor::DeviceDescriptor(audio_devices_t type, const FormatVector &encodedFormats,
- const String8 &tagName) :
- AudioPort(String8(""), AUDIO_PORT_TYPE_DEVICE,
- audio_is_output_device(type) ? AUDIO_PORT_ROLE_SINK :
- AUDIO_PORT_ROLE_SOURCE),
- mTagName(tagName), mDeviceType(type), mEncodedFormats(encodedFormats)
+DeviceDescriptor::DeviceDescriptor(audio_devices_t type,
+ const std::string &tagName,
+ const FormatVector &encodedFormats) :
+ DeviceDescriptor(type, tagName, "" /*address*/, encodedFormats)
+{
+}
+
+DeviceDescriptor::DeviceDescriptor(audio_devices_t type,
+ const std::string &tagName,
+ const std::string &address,
+ const FormatVector &encodedFormats) :
+ DeviceDescriptor(AudioDeviceTypeAddr(type, address), tagName, encodedFormats)
+{
+}
+
+DeviceDescriptor::DeviceDescriptor(const AudioDeviceTypeAddr &deviceTypeAddr,
+ const std::string &tagName,
+ const FormatVector &encodedFormats) :
+ DeviceDescriptorBase(deviceTypeAddr), mTagName(tagName), mEncodedFormats(encodedFormats)
{
mCurrentEncodedFormat = AUDIO_FORMAT_DEFAULT;
- if (audio_is_remote_submix_device(type)) {
- mAddress = String8("0");
- }
/* If framework runs against a pre 5.0 Audio HAL, encoded formats are absent from the config.
* FIXME: APM should know the version of the HAL and don't add the formats for V5.0.
* For now, the workaround to remove AC3 and IEC61937 support on HDMI is to declare
* something like 'encodedFormats="AUDIO_FORMAT_PCM_16_BIT"' on the HDMI devicePort.
*/
- if (type == AUDIO_DEVICE_OUT_HDMI && mEncodedFormats.isEmpty()) {
- mEncodedFormats.add(AUDIO_FORMAT_AC3);
- mEncodedFormats.add(AUDIO_FORMAT_IEC61937);
+ if (mDeviceTypeAddr.mType == AUDIO_DEVICE_OUT_HDMI && mEncodedFormats.empty()) {
+ mEncodedFormats.push_back(AUDIO_FORMAT_AC3);
+ mEncodedFormats.push_back(AUDIO_FORMAT_IEC61937);
}
}
-audio_port_handle_t DeviceDescriptor::getId() const
-{
- return mId;
-}
-
void DeviceDescriptor::attach(const sp<HwModule>& module)
{
- AudioPort::attach(module);
+ PolicyAudioPort::attach(module);
mId = getNextUniqueId();
}
void DeviceDescriptor::detach() {
mId = AUDIO_PORT_HANDLE_NONE;
- AudioPort::detach();
+ PolicyAudioPort::detach();
}
template<typename T>
@@ -88,7 +92,7 @@
return false;
}
- return (mDeviceType == other->mDeviceType) && (mAddress == other->mAddress) &&
+ return mDeviceTypeAddr.equals(other->mDeviceTypeAddr) &&
checkEqual(mEncodedFormats, other->mEncodedFormats);
}
@@ -97,7 +101,7 @@
if (!device_has_encoding_capability(type())) {
return true;
}
- if (mEncodedFormats.isEmpty()) {
+ if (mEncodedFormats.empty()) {
return true;
}
@@ -106,7 +110,7 @@
bool DeviceDescriptor::supportsFormat(audio_format_t format)
{
- if (mEncodedFormats.isEmpty()) {
+ if (mEncodedFormats.empty()) {
return true;
}
@@ -118,13 +122,69 @@
return false;
}
+status_t DeviceDescriptor::applyAudioPortConfig(const struct audio_port_config *config,
+ audio_port_config *backupConfig)
+{
+ struct audio_port_config localBackupConfig = { .config_mask = config->config_mask };
+ status_t status = NO_ERROR;
+
+ toAudioPortConfig(&localBackupConfig);
+ if ((status = validationBeforeApplyConfig(config)) == NO_ERROR) {
+ AudioPortConfig::applyAudioPortConfig(config, backupConfig);
+ applyPolicyAudioPortConfig(config);
+ }
+
+ if (backupConfig != NULL) {
+ *backupConfig = localBackupConfig;
+ }
+ return status;
+}
+
+void DeviceDescriptor::toAudioPortConfig(struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig) const
+{
+ DeviceDescriptorBase::toAudioPortConfig(dstConfig, srcConfig);
+ toPolicyAudioPortConfig(dstConfig, srcConfig);
+
+ dstConfig->ext.device.hw_module = getModuleHandle();
+}
+
+void DeviceDescriptor::toAudioPort(struct audio_port *port) const
+{
+ ALOGV("DeviceDescriptor::toAudioPort() handle %d type %08x", mId, mDeviceTypeAddr.mType);
+ DeviceDescriptorBase::toAudioPort(port);
+ port->ext.device.hw_module = getModuleHandle();
+}
+
+void DeviceDescriptor::importAudioPortAndPickAudioProfile(
+ const sp<PolicyAudioPort>& policyPort, bool force) {
+ if (!force && !policyPort->asAudioPort()->hasDynamicAudioProfile()) {
+ return;
+ }
+ AudioPort::importAudioPort(policyPort->asAudioPort());
+ policyPort->pickAudioProfile(mSamplingRate, mChannelMask, mFormat);
+}
+
+void DeviceDescriptor::dump(String8 *dst, int spaces, int index, bool verbose) const
+{
+ String8 extraInfo;
+ if (!mTagName.empty()) {
+ extraInfo.appendFormat("%*s- tag name: %s\n", spaces, "", mTagName.c_str());
+ }
+
+ std::string descBaseDumpStr;
+ DeviceDescriptorBase::dump(&descBaseDumpStr, spaces, index, extraInfo.string(), verbose);
+ dst->append(descBaseDumpStr.c_str());
+}
+
+
void DeviceVector::refreshTypes()
{
- mDeviceTypes = AUDIO_DEVICE_NONE;
+ mDeviceTypes.clear();
for (size_t i = 0; i < size(); i++) {
- mDeviceTypes |= itemAt(i)->type();
+ mDeviceTypes.insert(itemAt(i)->type());
}
- ALOGV("DeviceVector::refreshTypes() mDeviceTypes %08x", mDeviceTypes);
+ ALOGV("DeviceVector::refreshTypes() mDeviceTypes %s", dumpDeviceTypes(mDeviceTypes).c_str());
}
ssize_t DeviceVector::indexOf(const sp<DeviceDescriptor>& item) const
@@ -199,17 +259,6 @@
return devices;
}
-audio_devices_t DeviceVector::getDeviceTypesFromHwModule(audio_module_handle_t moduleHandle) const
-{
- audio_devices_t deviceTypes = AUDIO_DEVICE_NONE;
- for (const auto& device : *this) {
- if (device->getModuleHandle() == moduleHandle) {
- deviceTypes |= device->type();
- }
- }
- return deviceTypes;
-}
-
sp<DeviceDescriptor> DeviceVector::getDevice(audio_devices_t type, const String8& address,
audio_format_t format) const
{
@@ -219,11 +268,11 @@
// If format is specified, match it and ignore address
// Otherwise if address is specified match it
// Otherwise always match
- if (((address == "" || itemAt(i)->address() == address) &&
+ if (((address == "" || (itemAt(i)->address().compare(address.c_str()) == 0)) &&
format == AUDIO_FORMAT_DEFAULT) ||
(itemAt(i)->supportsFormat(format) && format != AUDIO_FORMAT_DEFAULT)) {
device = itemAt(i);
- if (itemAt(i)->address() == address) {
+ if (itemAt(i)->address().compare(address.c_str()) == 0) {
break;
}
}
@@ -246,17 +295,15 @@
return nullptr;
}
-DeviceVector DeviceVector::getDevicesFromTypeMask(audio_devices_t type) const
+DeviceVector DeviceVector::getDevicesFromTypes(const DeviceTypeSet& types) const
{
DeviceVector devices;
- bool isOutput = audio_is_output_devices(type);
- type &= ~AUDIO_DEVICE_BIT_IN;
- for (size_t i = 0; (i < size()) && (type != AUDIO_DEVICE_NONE); i++) {
- bool curIsOutput = audio_is_output_devices(itemAt(i)->type());
- audio_devices_t curType = itemAt(i)->type() & ~AUDIO_DEVICE_BIT_IN;
- if ((isOutput == curIsOutput) && ((type & curType) != 0)) {
+ if (types.empty()) {
+ return devices;
+ }
+ for (size_t i = 0; i < size(); i++) {
+ if (types.count(itemAt(i)->type()) != 0) {
devices.add(itemAt(i));
- type &= ~curType;
ALOGV("DeviceVector::%s() for type %08x found %p",
__func__, itemAt(i)->type(), itemAt(i).get());
}
@@ -264,7 +311,7 @@
return devices;
}
-sp<DeviceDescriptor> DeviceVector::getDeviceFromTagName(const String8 &tagName) const
+sp<DeviceDescriptor> DeviceVector::getDeviceFromTagName(const std::string &tagName) const
{
for (const auto& device : *this) {
if (device->getTagName() == tagName) {
@@ -274,6 +321,56 @@
return nullptr;
}
+DeviceVector DeviceVector::getFirstDevicesFromTypes(
+ std::vector<audio_devices_t> orderedTypes) const
+{
+ DeviceVector devices;
+ for (auto deviceType : orderedTypes) {
+ if (!(devices = getDevicesFromType(deviceType)).isEmpty()) {
+ break;
+ }
+ }
+ return devices;
+}
+
+sp<DeviceDescriptor> DeviceVector::getFirstExistingDevice(
+ std::vector<audio_devices_t> orderedTypes) const {
+ sp<DeviceDescriptor> device;
+ for (auto deviceType : orderedTypes) {
+ if ((device = getDevice(deviceType, String8(""), AUDIO_FORMAT_DEFAULT)) != nullptr) {
+ break;
+ }
+ }
+ return device;
+}
+
+sp<DeviceDescriptor> DeviceVector::getDeviceForOpening() const
+{
+ if (isEmpty()) {
+ // Return nullptr if this collection is empty.
+ return nullptr;
+ } else if (areAllOfSameDeviceType(types(), audio_is_input_device)) {
+ // For input case, return the first one when there is only one device.
+ return size() > 1 ? nullptr : *begin();
+ } else if (areAllOfSameDeviceType(types(), audio_is_output_device)) {
+ // For output case, return the device descriptor according to apm strategy.
+ audio_devices_t deviceType = apm_extract_one_audio_device(types());
+ return deviceType == AUDIO_DEVICE_NONE ? nullptr :
+ getDevice(deviceType, String8(""), AUDIO_FORMAT_DEFAULT);
+ }
+ // Return null pointer if the devices are not all input/output device.
+ return nullptr;
+}
+
+void DeviceVector::replaceDevicesByType(
+ audio_devices_t typeToRemove, const DeviceVector &devicesToAdd) {
+ DeviceVector devicesToRemove = getDevicesFromType(typeToRemove);
+ if (!devicesToRemove.isEmpty() && !devicesToAdd.isEmpty()) {
+ remove(devicesToRemove);
+ add(devicesToAdd);
+ }
+}
+
void DeviceVector::dump(String8 *dst, const String8 &tag, int spaces, bool verbose) const
{
if (isEmpty()) {
@@ -285,84 +382,6 @@
}
}
-void DeviceDescriptor::toAudioPortConfig(struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig) const
-{
- dstConfig->config_mask = AUDIO_PORT_CONFIG_GAIN;
- if (mSamplingRate != 0) {
- dstConfig->config_mask |= AUDIO_PORT_CONFIG_SAMPLE_RATE;
- }
- if (mChannelMask != AUDIO_CHANNEL_NONE) {
- dstConfig->config_mask |= AUDIO_PORT_CONFIG_CHANNEL_MASK;
- }
- if (mFormat != AUDIO_FORMAT_INVALID) {
- dstConfig->config_mask |= AUDIO_PORT_CONFIG_FORMAT;
- }
-
- if (srcConfig != NULL) {
- dstConfig->config_mask |= srcConfig->config_mask;
- }
-
- AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
-
- dstConfig->id = mId;
- dstConfig->role = audio_is_output_device(mDeviceType) ?
- AUDIO_PORT_ROLE_SINK : AUDIO_PORT_ROLE_SOURCE;
- dstConfig->type = AUDIO_PORT_TYPE_DEVICE;
- dstConfig->ext.device.type = mDeviceType;
-
- //TODO Understand why this test is necessary. i.e. why at boot time does it crash
- // without the test?
- // This has been demonstrated to NOT be true (at start up)
- // ALOG_ASSERT(mModule != NULL);
- dstConfig->ext.device.hw_module = getModuleHandle();
- (void)audio_utils_strlcpy_zerofill(dstConfig->ext.device.address, mAddress.string());
-}
-
-void DeviceDescriptor::toAudioPort(struct audio_port *port) const
-{
- ALOGV("DeviceDescriptor::toAudioPort() handle %d type %08x", mId, mDeviceType);
- AudioPort::toAudioPort(port);
- port->id = mId;
- toAudioPortConfig(&port->active_config);
- port->ext.device.type = mDeviceType;
- port->ext.device.hw_module = getModuleHandle();
- (void)audio_utils_strlcpy_zerofill(port->ext.device.address, mAddress.string());
-}
-
-void DeviceDescriptor::importAudioPort(const sp<AudioPort>& port, bool force) {
- if (!force && !port->hasDynamicAudioProfile()) {
- return;
- }
- AudioPort::importAudioPort(port);
- port->pickAudioProfile(mSamplingRate, mChannelMask, mFormat);
-}
-
-void DeviceDescriptor::dump(String8 *dst, int spaces, int index, bool verbose) const
-{
- dst->appendFormat("%*sDevice %d:\n", spaces, "", index + 1);
- if (mId != 0) {
- dst->appendFormat("%*s- id: %2d\n", spaces, "", mId);
- }
- if (!mTagName.isEmpty()) {
- dst->appendFormat("%*s- tag name: %s\n", spaces, "", mTagName.string());
- }
-
- dst->appendFormat("%*s- type: %-48s\n", spaces, "", ::android::toString(mDeviceType).c_str());
-
- if (mAddress.size() != 0) {
- dst->appendFormat("%*s- address: %-32s\n", spaces, "", mAddress.string());
- }
- AudioPort::dump(dst, spaces, verbose);
-}
-
-std::string DeviceDescriptor::toString() const
-{
- std::stringstream sstream;
- sstream << "type:0x" << std::hex << type() << ",@:" << mAddress;
- return sstream.str();
-}
-
std::string DeviceVector::toString() const
{
if (isEmpty()) {
@@ -411,13 +430,4 @@
return filteredDevices;
}
-void DeviceDescriptor::log() const
-{
- ALOGI("Device id:%d type:0x%08X:%s, addr:%s", mId, mDeviceType,
- ::android::toString(mDeviceType).c_str(),
- mAddress.string());
-
- AudioPort::log(" ");
-}
-
} // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp b/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
index 1f9b725..886e4c9 100644
--- a/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
@@ -19,7 +19,6 @@
#include "HwModule.h"
#include "IOProfile.h"
-#include "AudioGain.h"
#include <policy.h>
#include <system/audio.h>
@@ -42,7 +41,7 @@
}
}
-status_t HwModule::addOutputProfile(const String8& name, const audio_config_t *config,
+status_t HwModule::addOutputProfile(const std::string& name, const audio_config_t *config,
audio_devices_t device, const String8& address)
{
sp<IOProfile> profile = new OutputProfile(name);
@@ -50,8 +49,7 @@
profile->addAudioProfile(new AudioProfile(config->format, config->channel_mask,
config->sample_rate));
- sp<DeviceDescriptor> devDesc = new DeviceDescriptor(device);
- devDesc->setAddress(address);
+ sp<DeviceDescriptor> devDesc = new DeviceDescriptor(device, "" /*tagName*/, address.string());
addDynamicDevice(devDesc);
// Reciprocally attach the device to the module
devDesc->attach(this);
@@ -96,7 +94,7 @@
}
}
-status_t HwModule::removeOutputProfile(const String8& name)
+status_t HwModule::removeOutputProfile(const std::string& name)
{
for (size_t i = 0; i < mOutputProfiles.size(); i++) {
if (mOutputProfiles[i]->getName() == name) {
@@ -111,27 +109,26 @@
return NO_ERROR;
}
-status_t HwModule::addInputProfile(const String8& name, const audio_config_t *config,
+status_t HwModule::addInputProfile(const std::string& name, const audio_config_t *config,
audio_devices_t device, const String8& address)
{
sp<IOProfile> profile = new InputProfile(name);
profile->addAudioProfile(new AudioProfile(config->format, config->channel_mask,
config->sample_rate));
- sp<DeviceDescriptor> devDesc = new DeviceDescriptor(device);
- devDesc->setAddress(address);
+ sp<DeviceDescriptor> devDesc = new DeviceDescriptor(device, "" /*tagName*/, address.string());
addDynamicDevice(devDesc);
// Reciprocally attach the device to the module
devDesc->attach(this);
profile->addSupportedDevice(devDesc);
ALOGV("addInputProfile() name %s rate %d mask 0x%08x",
- name.string(), config->sample_rate, config->channel_mask);
+ name.c_str(), config->sample_rate, config->channel_mask);
return addInputProfile(profile);
}
-status_t HwModule::removeInputProfile(const String8& name)
+status_t HwModule::removeInputProfile(const std::string& name)
{
for (size_t i = 0; i < mInputProfiles.size(); i++) {
if (mInputProfiles[i]->getName() == name) {
@@ -157,7 +154,7 @@
sp<DeviceDescriptor> HwModule::getRouteSinkDevice(const sp<AudioRoute> &route) const
{
sp<DeviceDescriptor> sinkDevice = 0;
- if (route->getSink()->getType() == AUDIO_PORT_TYPE_DEVICE) {
+ if (route->getSink()->asAudioPort()->getType() == AUDIO_PORT_TYPE_DEVICE) {
sinkDevice = mDeclaredDevices.getDeviceFromTagName(route->getSink()->getTagName());
}
return sinkDevice;
@@ -167,7 +164,7 @@
{
DeviceVector sourceDevices;
for (const auto& source : route->getSources()) {
- if (source->getType() == AUDIO_PORT_TYPE_DEVICE) {
+ if (source->asAudioPort()->getType() == AUDIO_PORT_TYPE_DEVICE) {
sourceDevices.add(mDeclaredDevices.getDeviceFromTagName(source->getTagName()));
}
}
@@ -187,20 +184,20 @@
for (const auto& stream : mInputProfiles) {
DeviceVector sourceDevices;
for (const auto& route : stream->getRoutes()) {
- sp<AudioPort> sink = route->getSink();
+ sp<PolicyAudioPort> sink = route->getSink();
if (sink == 0 || stream != sink) {
ALOGE("%s: Invalid route attached to input stream", __FUNCTION__);
continue;
}
DeviceVector sourceDevicesForRoute = getRouteSourceDevices(route);
if (sourceDevicesForRoute.isEmpty()) {
- ALOGE("%s: invalid source devices for %s", __FUNCTION__, stream->getName().string());
+ ALOGE("%s: invalid source devices for %s", __FUNCTION__, stream->getName().c_str());
continue;
}
sourceDevices.add(sourceDevicesForRoute);
}
if (sourceDevices.isEmpty()) {
- ALOGE("%s: invalid source devices for %s", __FUNCTION__, stream->getName().string());
+ ALOGE("%s: invalid source devices for %s", __FUNCTION__, stream->getName().c_str());
continue;
}
stream->setSupportedDevices(sourceDevices);
@@ -208,14 +205,14 @@
for (const auto& stream : mOutputProfiles) {
DeviceVector sinkDevices;
for (const auto& route : stream->getRoutes()) {
- sp<AudioPort> source = route->getSources().findByTagName(stream->getTagName());
+ sp<PolicyAudioPort> source = findByTagName(route->getSources(), stream->getTagName());
if (source == 0 || stream != source) {
ALOGE("%s: Invalid route attached to output stream", __FUNCTION__);
continue;
}
sp<DeviceDescriptor> sinkDevice = getRouteSinkDevice(route);
if (sinkDevice == 0) {
- ALOGE("%s: invalid sink device for %s", __FUNCTION__, stream->getName().string());
+ ALOGE("%s: invalid sink device for %s", __FUNCTION__, stream->getName().c_str());
continue;
}
sinkDevices.add(sinkDevice);
@@ -230,7 +227,8 @@
mHandle = handle;
}
-bool HwModule::supportsPatch(const sp<AudioPort> &srcPort, const sp<AudioPort> &dstPort) const {
+bool HwModule::supportsPatch(const sp<PolicyAudioPort> &srcPort,
+ const sp<PolicyAudioPort> &dstPort) const {
for (const auto &route : mRoutes) {
if (route->supportsPatch(srcPort, dstPort)) {
return true;
@@ -260,7 +258,7 @@
}
mDeclaredDevices.dump(dst, String8("Declared"), 2, true);
mDynamicDevices.dump(dst, String8("Dynamic"), 2, true);
- mRoutes.dump(dst, 2);
+ dumpAudioRouteVector(mRoutes, dst, 2);
}
sp <HwModule> HwModuleCollection::getModuleFromName(const char *name) const
@@ -273,14 +271,14 @@
return nullptr;
}
-sp <HwModule> HwModuleCollection::getModuleForDeviceTypes(audio_devices_t type,
- audio_format_t encodedFormat) const
+sp <HwModule> HwModuleCollection::getModuleForDeviceType(audio_devices_t type,
+ audio_format_t encodedFormat) const
{
for (const auto& module : *this) {
const auto& profiles = audio_is_output_device(type) ?
module->getOutputProfiles() : module->getInputProfiles();
for (const auto& profile : profiles) {
- if (profile->supportsDeviceTypes(type)) {
+ if (profile->supportsDeviceTypes({type})) {
if (encodedFormat != AUDIO_FORMAT_DEFAULT) {
DeviceVector declaredDevices = module->getDeclaredDevices();
sp <DeviceDescriptor> deviceDesc =
@@ -300,7 +298,7 @@
sp<HwModule> HwModuleCollection::getModuleForDevice(const sp<DeviceDescriptor> &device,
audio_format_t encodedFormat) const
{
- return getModuleForDeviceTypes(device->type(), encodedFormat);
+ return getModuleForDeviceType(device->type(), encodedFormat);
}
DeviceVector HwModuleCollection::getAvailableDevicesFromModuleName(
@@ -335,8 +333,8 @@
}
if (allowToCreate) {
moduleDevice->attach(hwModule);
- moduleDevice->setAddress(devAddress);
- moduleDevice->setName(String8(name));
+ moduleDevice->setAddress(devAddress.string());
+ moduleDevice->setName(name);
}
return moduleDevice;
}
@@ -354,15 +352,15 @@
const char *name,
const audio_format_t encodedFormat) const
{
- sp<HwModule> hwModule = getModuleForDeviceTypes(type, encodedFormat);
+ sp<HwModule> hwModule = getModuleForDeviceType(type, encodedFormat);
if (hwModule == 0) {
ALOGE("%s: could not find HW module for device %04x address %s", __FUNCTION__, type,
address);
return nullptr;
}
- sp<DeviceDescriptor> device = new DeviceDescriptor(type, String8(name));
- device->setName(String8(name));
- device->setAddress(String8(address));
+
+ sp<DeviceDescriptor> device = new DeviceDescriptor(type, name, address);
+ device->setName(name);
device->setEncodedFormat(encodedFormat);
// Add the device to the list of dynamic devices
@@ -382,7 +380,7 @@
// @todo quid of audio profile? import the profile from device of the same type?
const auto &isoTypeDeviceForProfile =
profile->getSupportedDevices().getDevice(type, String8(), AUDIO_FORMAT_DEFAULT);
- device->importAudioPort(isoTypeDeviceForProfile, true /* force */);
+ device->importAudioPortAndPickAudioProfile(isoTypeDeviceForProfile, true /* force */);
ALOGV("%s: adding device %s to profile %s", __FUNCTION__,
device->toString().c_str(), profile->getTagName().c_str());
diff --git a/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp b/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp
index fe2eaee..bf1a0f7 100644
--- a/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp
@@ -20,7 +20,6 @@
#include <system/audio-base.h>
#include "IOProfile.h"
#include "HwModule.h"
-#include "AudioGain.h"
#include "TypeConverter.h"
namespace android {
@@ -79,7 +78,10 @@
}
}
- if (isPlaybackThread && (getFlags() & flags) != flags) {
+ const uint32_t mustMatchOutputFlags =
+ AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_HW_AV_SYNC|AUDIO_OUTPUT_FLAG_MMAP_NOIRQ;
+ if (isPlaybackThread && (((getFlags() ^ flags) & mustMatchOutputFlags)
+ || (getFlags() & flags) != flags)) {
return false;
}
// The only input flag that is allowed to be different is the fast flag.
@@ -105,7 +107,9 @@
void IOProfile::dump(String8 *dst) const
{
- AudioPort::dump(dst, 4);
+ std::string portStr;
+ AudioPort::dump(&portStr, 4);
+ dst->append(portStr.c_str());
dst->appendFormat(" - flags: 0x%04x", getFlags());
std::string flagsLiteral;
diff --git a/services/audiopolicy/common/managerdefinitions/src/PolicyAudioPort.cpp b/services/audiopolicy/common/managerdefinitions/src/PolicyAudioPort.cpp
new file mode 100644
index 0000000..8c61b90
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/src/PolicyAudioPort.cpp
@@ -0,0 +1,294 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "APM::PolicyAudioPort"
+//#define LOG_NDEBUG 0
+#include "TypeConverter.h"
+#include "PolicyAudioPort.h"
+#include "HwModule.h"
+#include <policy.h>
+
+#ifndef ARRAY_SIZE
+#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
+#endif
+
+namespace android {
+
+// --- PolicyAudioPort class implementation
+void PolicyAudioPort::attach(const sp<HwModule>& module)
+{
+ ALOGV("%s: attaching module %s to port %s",
+ __FUNCTION__, getModuleName(), asAudioPort()->getName().c_str());
+ mModule = module;
+}
+
+void PolicyAudioPort::detach()
+{
+ mModule = nullptr;
+}
+
+// Note that is a different namespace than AudioFlinger unique IDs
+audio_port_handle_t PolicyAudioPort::getNextUniqueId()
+{
+ return getNextHandle();
+}
+
+audio_module_handle_t PolicyAudioPort::getModuleHandle() const
+{
+ return mModule != 0 ? mModule->getHandle() : AUDIO_MODULE_HANDLE_NONE;
+}
+
+uint32_t PolicyAudioPort::getModuleVersionMajor() const
+{
+ return mModule != 0 ? mModule->getHalVersionMajor() : 0;
+}
+
+const char *PolicyAudioPort::getModuleName() const
+{
+ return mModule != 0 ? mModule->getName() : "invalid module";
+}
+
+status_t PolicyAudioPort::checkExactAudioProfile(const struct audio_port_config *config) const
+{
+ status_t status = NO_ERROR;
+ auto config_mask = config->config_mask;
+ if (config_mask & AUDIO_PORT_CONFIG_GAIN) {
+ config_mask &= ~AUDIO_PORT_CONFIG_GAIN;
+ status = asAudioPort()->checkGain(&config->gain, config->gain.index);
+ if (status != NO_ERROR) {
+ return status;
+ }
+ }
+ if (config_mask != 0) {
+ // TODO should we check sample_rate / channel_mask / format separately?
+ status = checkExactProfile(asAudioPort()->getAudioProfiles(), config->sample_rate,
+ config->channel_mask, config->format);
+ }
+ return status;
+}
+
+void PolicyAudioPort::pickSamplingRate(uint32_t &pickedRate,
+ const SampleRateSet &samplingRates) const
+{
+ pickedRate = 0;
+ // For direct outputs, pick minimum sampling rate: this helps ensuring that the
+ // channel count / sampling rate combination chosen will be supported by the connected
+ // sink
+ if (isDirectOutput()) {
+ uint32_t samplingRate = UINT_MAX;
+ for (const auto rate : samplingRates) {
+ if ((rate < samplingRate) && (rate > 0)) {
+ samplingRate = rate;
+ }
+ }
+ pickedRate = (samplingRate == UINT_MAX) ? 0 : samplingRate;
+ } else {
+ uint32_t maxRate = SAMPLE_RATE_HZ_MAX;
+
+ // For mixed output and inputs, use max mixer sampling rates. Do not
+ // limit sampling rate otherwise
+ // For inputs, also see checkCompatibleSamplingRate().
+ if (asAudioPort()->getType() == AUDIO_PORT_TYPE_MIX) {
+ maxRate = UINT_MAX;
+ }
+ // TODO: should mSamplingRates[] be ordered in terms of our preference
+ // and we return the first (and hence most preferred) match? This is of concern if
+ // we want to choose 96kHz over 192kHz for USB driver stability or resource constraints.
+ for (const auto rate : samplingRates) {
+ if ((rate > pickedRate) && (rate <= maxRate)) {
+ pickedRate = rate;
+ }
+ }
+ }
+}
+
+void PolicyAudioPort::pickChannelMask(audio_channel_mask_t &pickedChannelMask,
+ const ChannelMaskSet &channelMasks) const
+{
+ pickedChannelMask = AUDIO_CHANNEL_NONE;
+ // For direct outputs, pick minimum channel count: this helps ensuring that the
+ // channel count / sampling rate combination chosen will be supported by the connected
+ // sink
+ if (isDirectOutput()) {
+ uint32_t channelCount = UINT_MAX;
+ for (const auto channelMask : channelMasks) {
+ uint32_t cnlCount;
+ if (asAudioPort()->useInputChannelMask()) {
+ cnlCount = audio_channel_count_from_in_mask(channelMask);
+ } else {
+ cnlCount = audio_channel_count_from_out_mask(channelMask);
+ }
+ if ((cnlCount < channelCount) && (cnlCount > 0)) {
+ pickedChannelMask = channelMask;
+ channelCount = cnlCount;
+ }
+ }
+ } else {
+ uint32_t channelCount = 0;
+ uint32_t maxCount = MAX_MIXER_CHANNEL_COUNT;
+
+ // For mixed output and inputs, use max mixer channel count. Do not
+ // limit channel count otherwise
+ if (asAudioPort()->getType() != AUDIO_PORT_TYPE_MIX) {
+ maxCount = UINT_MAX;
+ }
+ for (const auto channelMask : channelMasks) {
+ uint32_t cnlCount;
+ if (asAudioPort()->useInputChannelMask()) {
+ cnlCount = audio_channel_count_from_in_mask(channelMask);
+ } else {
+ cnlCount = audio_channel_count_from_out_mask(channelMask);
+ }
+ if ((cnlCount > channelCount) && (cnlCount <= maxCount)) {
+ pickedChannelMask = channelMask;
+ channelCount = cnlCount;
+ }
+ }
+ }
+}
+
+/* format in order of increasing preference */
+const audio_format_t PolicyAudioPort::sPcmFormatCompareTable[] = {
+ AUDIO_FORMAT_DEFAULT,
+ AUDIO_FORMAT_PCM_16_BIT,
+ AUDIO_FORMAT_PCM_8_24_BIT,
+ AUDIO_FORMAT_PCM_24_BIT_PACKED,
+ AUDIO_FORMAT_PCM_32_BIT,
+ AUDIO_FORMAT_PCM_FLOAT,
+};
+
+int PolicyAudioPort::compareFormats(audio_format_t format1, audio_format_t format2)
+{
+ // NOTE: AUDIO_FORMAT_INVALID is also considered not PCM and will be compared equal to any
+ // compressed format and better than any PCM format. This is by design of pickFormat()
+ if (!audio_is_linear_pcm(format1)) {
+ if (!audio_is_linear_pcm(format2)) {
+ return 0;
+ }
+ return 1;
+ }
+ if (!audio_is_linear_pcm(format2)) {
+ return -1;
+ }
+
+ int index1 = -1, index2 = -1;
+ for (size_t i = 0;
+ (i < ARRAY_SIZE(sPcmFormatCompareTable)) && ((index1 == -1) || (index2 == -1));
+ i ++) {
+ if (sPcmFormatCompareTable[i] == format1) {
+ index1 = i;
+ }
+ if (sPcmFormatCompareTable[i] == format2) {
+ index2 = i;
+ }
+ }
+ // format1 not found => index1 < 0 => format2 > format1
+ // format2 not found => index2 < 0 => format2 < format1
+ return index1 - index2;
+}
+
+uint32_t PolicyAudioPort::formatDistance(audio_format_t format1, audio_format_t format2)
+{
+ if (format1 == format2) {
+ return 0;
+ }
+ if (format1 == AUDIO_FORMAT_INVALID || format2 == AUDIO_FORMAT_INVALID) {
+ return kFormatDistanceMax;
+ }
+ int diffBytes = (int)audio_bytes_per_sample(format1) -
+ audio_bytes_per_sample(format2);
+
+ return abs(diffBytes);
+}
+
+bool PolicyAudioPort::isBetterFormatMatch(audio_format_t newFormat,
+ audio_format_t currentFormat,
+ audio_format_t targetFormat)
+{
+ return formatDistance(newFormat, targetFormat) < formatDistance(currentFormat, targetFormat);
+}
+
+void PolicyAudioPort::pickAudioProfile(uint32_t &samplingRate,
+ audio_channel_mask_t &channelMask,
+ audio_format_t &format) const
+{
+ format = AUDIO_FORMAT_DEFAULT;
+ samplingRate = 0;
+ channelMask = AUDIO_CHANNEL_NONE;
+
+ // special case for uninitialized dynamic profile
+ if (!asAudioPort()->hasValidAudioProfile()) {
+ return;
+ }
+ audio_format_t bestFormat = sPcmFormatCompareTable[ARRAY_SIZE(sPcmFormatCompareTable) - 1];
+ // For mixed output and inputs, use best mixer output format.
+ // Do not limit format otherwise
+ if ((asAudioPort()->getType() != AUDIO_PORT_TYPE_MIX) || isDirectOutput()) {
+ bestFormat = AUDIO_FORMAT_INVALID;
+ }
+
+ const AudioProfileVector& audioProfiles = asAudioPort()->getAudioProfiles();
+ for (size_t i = 0; i < audioProfiles.size(); i ++) {
+ if (!audioProfiles[i]->isValid()) {
+ continue;
+ }
+ audio_format_t formatToCompare = audioProfiles[i]->getFormat();
+ if ((compareFormats(formatToCompare, format) > 0) &&
+ (compareFormats(formatToCompare, bestFormat) <= 0)) {
+ uint32_t pickedSamplingRate = 0;
+ audio_channel_mask_t pickedChannelMask = AUDIO_CHANNEL_NONE;
+ pickChannelMask(pickedChannelMask, audioProfiles[i]->getChannels());
+ pickSamplingRate(pickedSamplingRate, audioProfiles[i]->getSampleRates());
+
+ if (formatToCompare != AUDIO_FORMAT_DEFAULT && pickedChannelMask != AUDIO_CHANNEL_NONE
+ && pickedSamplingRate != 0) {
+ format = formatToCompare;
+ channelMask = pickedChannelMask;
+ samplingRate = pickedSamplingRate;
+ // TODO: shall we return on the first one or still trying to pick a better Profile?
+ }
+ }
+ }
+ ALOGV("%s Port[nm:%s] profile rate=%d, format=%d, channels=%d", __FUNCTION__,
+ asAudioPort()->getName().c_str(), samplingRate, channelMask, format);
+}
+
+// --- PolicyAudioPortConfig class implementation
+
+status_t PolicyAudioPortConfig::validationBeforeApplyConfig(
+ const struct audio_port_config *config) const
+{
+ sp<PolicyAudioPort> policyAudioPort = getPolicyAudioPort();
+ return policyAudioPort ? policyAudioPort->checkExactAudioProfile(config) : NO_INIT;
+}
+
+void PolicyAudioPortConfig::toPolicyAudioPortConfig(struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig) const
+{
+ if (dstConfig->config_mask & AUDIO_PORT_CONFIG_FLAGS) {
+ if ((srcConfig != nullptr) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_FLAGS)) {
+ dstConfig->flags = srcConfig->flags;
+ } else {
+ dstConfig->flags = mFlags;
+ }
+ } else {
+ dstConfig->flags = { AUDIO_INPUT_FLAG_NONE };
+ }
+}
+
+
+
+} // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp b/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp
index 5f820c2..f0bb28e 100644
--- a/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp
@@ -199,6 +199,7 @@
struct Attributes
{
static constexpr const char *speakerDrcEnabled = "speaker_drc_enabled";
+ static constexpr const char *engineLibrarySuffix = "engine_library";
};
static status_t deserialize(const xmlNode *root, AudioPolicyConfig *config);
@@ -406,8 +407,8 @@
samplingRatesFromString(samplingRates, ","));
profile->setDynamicFormat(profile->getFormat() == gDynamicFormat);
- profile->setDynamicChannels(profile->getChannels().isEmpty());
- profile->setDynamicRate(profile->getSampleRates().isEmpty());
+ profile->setDynamicChannels(profile->getChannels().empty());
+ profile->setDynamicRate(profile->getSampleRates().empty());
return profile;
}
@@ -430,16 +431,19 @@
audio_port_role_t portRole = (role == Attributes::roleSource) ?
AUDIO_PORT_ROLE_SOURCE : AUDIO_PORT_ROLE_SINK;
- Element mixPort = new IOProfile(String8(name.c_str()), portRole);
+ Element mixPort = new IOProfile(name, portRole);
AudioProfileTraits::Collection profiles;
status_t status = deserializeCollection<AudioProfileTraits>(child, &profiles, NULL);
if (status != NO_ERROR) {
return Status::fromStatusT(status);
}
- if (profiles.isEmpty()) {
- profiles.add(AudioProfile::createFullDynamic());
+ if (profiles.empty()) {
+ profiles.add(AudioProfile::createFullDynamic(gDynamicFormat));
}
+ // The audio profiles are in order of listed in audio policy configuration file.
+ // Sort audio profiles accroding to the format.
+ sortAudioProfiles(profiles);
mixPort->setAudioProfiles(profiles);
std::string flags = getXmlAttribute(child, Attributes::flags);
@@ -508,22 +512,20 @@
if (!encodedFormatsLiteral.empty()) {
encodedFormats = formatsFromString(encodedFormatsLiteral, " ");
}
- Element deviceDesc = new DeviceDescriptor(type, encodedFormats, String8(name.c_str()));
-
std::string address = getXmlAttribute(cur, Attributes::address);
- if (!address.empty()) {
- ALOGV("%s: address=%s for %s", __func__, address.c_str(), name.c_str());
- deviceDesc->setAddress(String8(address.c_str()));
- }
+ Element deviceDesc = new DeviceDescriptor(type, name, address, encodedFormats);
AudioProfileTraits::Collection profiles;
status_t status = deserializeCollection<AudioProfileTraits>(cur, &profiles, NULL);
if (status != NO_ERROR) {
return Status::fromStatusT(status);
}
- if (profiles.isEmpty()) {
- profiles.add(AudioProfile::createFullDynamic());
+ if (profiles.empty()) {
+ profiles.add(AudioProfile::createFullDynamic(gDynamicFormat));
}
+ // The audio profiles are in order of listed in audio policy configuration file.
+ // Sort audio profiles accroding to the format.
+ sortAudioProfiles(profiles);
deviceDesc->setAudioProfiles(profiles);
// Deserialize AudioGain children
@@ -532,7 +534,7 @@
return Status::fromStatusT(status);
}
ALOGV("%s: adding device tag %s type %08x address %s", __func__,
- deviceDesc->getName().string(), type, deviceDesc->address().string());
+ deviceDesc->getName().c_str(), type, deviceDesc->address().c_str());
return deviceDesc;
}
@@ -555,7 +557,7 @@
return Status::fromStatusT(BAD_VALUE);
}
// Convert Sink name to port pointer
- sp<AudioPort> sink = ctx->findPortByTagName(String8(sinkAttr.c_str()));
+ sp<PolicyAudioPort> sink = ctx->findPortByTagName(sinkAttr);
if (sink == NULL) {
ALOGE("%s: no sink found with name=%s", __func__, sinkAttr.c_str());
return Status::fromStatusT(BAD_VALUE);
@@ -568,13 +570,13 @@
return Status::fromStatusT(BAD_VALUE);
}
// Tokenize and Convert Sources name to port pointer
- AudioPortVector sources;
+ PolicyAudioPortVector sources;
std::unique_ptr<char[]> sourcesLiteral{strndup(
sourcesAttr.c_str(), strlen(sourcesAttr.c_str()))};
char *devTag = strtok(sourcesLiteral.get(), ",");
while (devTag != NULL) {
if (strlen(devTag) != 0) {
- sp<AudioPort> source = ctx->findPortByTagName(String8(devTag));
+ sp<PolicyAudioPort> source = ctx->findPortByTagName(devTag);
if (source == NULL) {
ALOGE("%s: no source found with name=%s", __func__, devTag);
return Status::fromStatusT(BAD_VALUE);
@@ -586,7 +588,7 @@
sink->addRoute(route);
for (size_t i = 0; i < sources.size(); i++) {
- sp<AudioPort> source = sources.itemAt(i);
+ sp<PolicyAudioPort> source = sources.itemAt(i);
source->addRoute(route);
}
route->setSources(sources);
@@ -648,7 +650,7 @@
ALOGV("%s: %s %s=%s", __func__, tag, childAttachedDeviceTag,
reinterpret_cast<const char*>(attachedDevice.get()));
sp<DeviceDescriptor> device = module->getDeclaredDevices().
- getDeviceFromTagName(String8(reinterpret_cast<const char*>(
+ getDeviceFromTagName(std::string(reinterpret_cast<const char*>(
attachedDevice.get())));
ctx->addAvailableDevice(device);
}
@@ -663,7 +665,7 @@
ALOGV("%s: %s %s=%s", __func__, tag, childDefaultOutputDeviceTag,
reinterpret_cast<const char*>(defaultOutputDevice.get()));
sp<DeviceDescriptor> device = module->getDeclaredDevices().getDeviceFromTagName(
- String8(reinterpret_cast<const char*>(defaultOutputDevice.get())));
+ std::string(reinterpret_cast<const char*>(defaultOutputDevice.get())));
if (device != 0 && ctx->getDefaultOutputDevice() == 0) {
ctx->setDefaultOutputDevice(device);
ALOGV("%s: default is %08x",
@@ -686,6 +688,10 @@
convertTo<std::string, bool>(speakerDrcEnabled, isSpeakerDrcEnabled)) {
config->setSpeakerDrcEnabled(isSpeakerDrcEnabled);
}
+ std::string engineLibrarySuffix = getXmlAttribute(cur, Attributes::engineLibrarySuffix);
+ if (!engineLibrarySuffix.empty()) {
+ config->setEngineLibraryNameSuffix(engineLibrarySuffix);
+ }
return NO_ERROR;
}
}
diff --git a/services/audiopolicy/config/Android.bp b/services/audiopolicy/config/Android.bp
new file mode 100644
index 0000000..f4610bb
--- /dev/null
+++ b/services/audiopolicy/config/Android.bp
@@ -0,0 +1,105 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+soong_namespace {
+}
+
+prebuilt_etc {
+ name: "a2dp_in_audio_policy_configuration.xml",
+ vendor: true,
+ src: ":a2dp_in_audio_policy_configuration",
+}
+prebuilt_etc {
+ name: "a2dp_audio_policy_configuration.xml",
+ vendor: true,
+ src: ":a2dp_audio_policy_configuration",
+}
+prebuilt_etc {
+ name: "audio_policy_configuration.xml",
+ vendor: true,
+ src: ":audio_policy_configuration_generic",
+}
+prebuilt_etc {
+ name: "r_submix_audio_policy_configuration.xml",
+ vendor: true,
+ src: ":r_submix_audio_policy_configuration",
+}
+prebuilt_etc {
+ name: "audio_policy_volumes.xml",
+ vendor: true,
+ src: ":audio_policy_volumes",
+}
+prebuilt_etc {
+ name: "default_volume_tables.xml",
+ vendor: true,
+ src: ":default_volume_tables",
+}
+prebuilt_etc {
+ name: "surround_sound_configuration_5_0.xml",
+ vendor: true,
+ src: ":surround_sound_configuration_5_0",
+}
+prebuilt_etc {
+ name: "usb_audio_policy_configuration.xml",
+ vendor: true,
+ src: ":usb_audio_policy_configuration",
+}
+prebuilt_etc {
+ name: "primary_audio_policy_configuration.xml",
+ src: ":primary_audio_policy_configuration",
+ vendor: true,
+}
+
+filegroup {
+ name: "a2dp_in_audio_policy_configuration",
+ srcs: ["a2dp_in_audio_policy_configuration.xml"],
+}
+filegroup {
+ name: "a2dp_audio_policy_configuration",
+ srcs: ["a2dp_audio_policy_configuration.xml"],
+}
+filegroup {
+ name: "primary_audio_policy_configuration",
+ srcs: ["primary_audio_policy_configuration.xml"],
+}
+filegroup {
+ name: "surround_sound_configuration_5_0",
+ srcs: ["surround_sound_configuration_5_0.xml"],
+}
+filegroup {
+ name: "default_volume_tables",
+ srcs: ["default_volume_tables.xml"],
+}
+filegroup {
+ name: "audio_policy_volumes",
+ srcs: ["audio_policy_volumes.xml"],
+}
+filegroup {
+ name: "audio_policy_configuration_generic",
+ srcs: ["audio_policy_configuration_generic.xml"],
+}
+filegroup {
+ name: "audio_policy_configuration_generic_configurable",
+ srcs: ["audio_policy_configuration_generic_configurable.xml"],
+}
+filegroup {
+ name: "usb_audio_policy_configuration",
+ srcs: ["usb_audio_policy_configuration.xml"],
+}
+filegroup {
+ name: "r_submix_audio_policy_configuration",
+ srcs: ["r_submix_audio_policy_configuration.xml"],
+}
diff --git a/services/audiopolicy/config/audio_policy_configuration_generic_configurable.xml b/services/audiopolicy/config/audio_policy_configuration_generic_configurable.xml
new file mode 100644
index 0000000..fbe4f7f
--- /dev/null
+++ b/services/audiopolicy/config/audio_policy_configuration_generic_configurable.xml
@@ -0,0 +1,50 @@
+<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
+<!-- Copyright (C) 2020 The Android Open Source Project
+
+ Licensed under the Apache License, Version 2.0 (the "License");
+ you may not use this file except in compliance with the License.
+ You may obtain a copy of the License at
+
+ http://www.apache.org/licenses/LICENSE-2.0
+
+ Unless required by applicable law or agreed to in writing, software
+ distributed under the License is distributed on an "AS IS" BASIS,
+ WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ See the License for the specific language governing permissions and
+ limitations under the License.
+-->
+
+<audioPolicyConfiguration version="1.0" xmlns:xi="http://www.w3.org/2001/XInclude">
+ <!-- version section contains a “version” tag in the form “major.minor” e.g version=”1.0” -->
+
+ <!-- Global configuration Decalaration -->
+ <globalConfiguration speaker_drc_enabled="false" engine_library="configurable"/>
+
+ <modules>
+ <!-- Primary Audio HAL -->
+ <xi:include href="primary_audio_policy_configuration.xml"/>
+
+ <!-- Remote Submix Audio HAL -->
+ <xi:include href="r_submix_audio_policy_configuration.xml"/>
+
+ </modules>
+ <!-- End of Modules section -->
+
+ <!-- Volume section:
+ IMPORTANT NOTE: Volume tables have been moved to engine configuration.
+ Keep it here for legacy.
+ Engine will fallback on these files if none are provided by engine.
+ -->
+
+ <xi:include href="audio_policy_volumes.xml"/>
+ <xi:include href="default_volume_tables.xml"/>
+
+ <!-- End of Volume section -->
+
+ <!-- Surround Sound configuration -->
+
+ <xi:include href="surround_sound_configuration_5_0.xml"/>
+
+ <!-- End of Surround Sound configuration -->
+
+</audioPolicyConfiguration>
diff --git a/services/audiopolicy/config/audio_policy_volumes.xml b/services/audiopolicy/config/audio_policy_volumes.xml
index ec64a7c..ddd031a 100644
--- a/services/audiopolicy/config/audio_policy_volumes.xml
+++ b/services/audiopolicy/config/audio_policy_volumes.xml
@@ -181,6 +181,16 @@
ref="DEFAULT_NON_MUTABLE_VOLUME_CURVE"/>
<volume stream="AUDIO_STREAM_ACCESSIBILITY" deviceCategory="DEVICE_CATEGORY_HEARING_AID"
ref="DEFAULT_NON_MUTABLE_HEARING_AID_VOLUME_CURVE"/>
+ <volume stream="AUDIO_STREAM_ASSISTANT" deviceCategory="DEVICE_CATEGORY_HEADSET"
+ ref="DEFAULT_MEDIA_VOLUME_CURVE"/>
+ <volume stream="AUDIO_STREAM_ASSISTANT" deviceCategory="DEVICE_CATEGORY_SPEAKER"
+ ref="DEFAULT_DEVICE_CATEGORY_SPEAKER_VOLUME_CURVE"/>
+ <volume stream="AUDIO_STREAM_ASSISTANT" deviceCategory="DEVICE_CATEGORY_EARPIECE"
+ ref="DEFAULT_MEDIA_VOLUME_CURVE"/>
+ <volume stream="AUDIO_STREAM_ASSISTANT" deviceCategory="DEVICE_CATEGORY_EXT_MEDIA"
+ ref="DEFAULT_MEDIA_VOLUME_CURVE"/>
+ <volume stream="AUDIO_STREAM_ASSISTANT" deviceCategory="DEVICE_CATEGORY_HEARING_AID"
+ ref="DEFAULT_HEARING_AID_VOLUME_CURVE"/>
<volume stream="AUDIO_STREAM_REROUTING" deviceCategory="DEVICE_CATEGORY_HEADSET"
ref="FULL_SCALE_VOLUME_CURVE"/>
<volume stream="AUDIO_STREAM_REROUTING" deviceCategory="DEVICE_CATEGORY_SPEAKER"
diff --git a/services/audiopolicy/engine/common/Android.bp b/services/audiopolicy/engine/common/Android.bp
old mode 100644
new mode 100755
index d0775ad..a1c69f2
--- a/services/audiopolicy/engine/common/Android.bp
+++ b/services/audiopolicy/engine/common/Android.bp
@@ -25,6 +25,7 @@
"src/ProductStrategy.cpp",
"src/VolumeCurve.cpp",
"src/VolumeGroup.cpp",
+ "src/LastRemovableMediaDevices.cpp",
],
cflags: [
"-Wall",
@@ -44,4 +45,7 @@
"libaudiopolicycomponents",
"libaudiopolicyengine_config",
],
+ shared_libs: [
+ "libaudiofoundation",
+ ],
}
diff --git a/services/audiopolicy/engine/common/include/EngineBase.h b/services/audiopolicy/engine/common/include/EngineBase.h
old mode 100644
new mode 100755
index cedc78f..7f339dc
--- a/services/audiopolicy/engine/common/include/EngineBase.h
+++ b/services/audiopolicy/engine/common/include/EngineBase.h
@@ -17,18 +17,19 @@
#pragma once
#include <EngineConfig.h>
-#include <AudioPolicyManagerInterface.h>
+#include <EngineInterface.h>
#include <ProductStrategy.h>
#include <VolumeGroup.h>
+#include <LastRemovableMediaDevices.h>
namespace android {
namespace audio_policy {
-class EngineBase : public AudioPolicyManagerInterface
+class EngineBase : public EngineInterface
{
public:
///
- /// from AudioPolicyManagerInterface
+ /// from EngineInterface
///
android::status_t initCheck() override;
@@ -49,10 +50,8 @@
return mForceUse[usage];
}
android::status_t setDeviceConnectionState(const sp<DeviceDescriptor> /*devDesc*/,
- audio_policy_dev_state_t /*state*/) override
- {
- return NO_ERROR;
- }
+ audio_policy_dev_state_t /*state*/) override;
+
product_strategy_t getProductStrategyForAttributes(
const audio_attributes_t &attr) const override;
@@ -86,8 +85,21 @@
status_t listAudioVolumeGroups(AudioVolumeGroupVector &groups) const override;
+ std::vector<audio_devices_t> getLastRemovableMediaDevices(
+ device_out_group_t group = GROUP_NONE) const
+ {
+ return mLastRemovableMediaDevices.getLastRemovableMediaDevices(group);
+ }
+
void dump(String8 *dst) const override;
+ status_t setPreferredDeviceForStrategy(product_strategy_t strategy,
+ const AudioDeviceTypeAddr &device) override;
+
+ status_t removePreferredDeviceForStrategy(product_strategy_t strategy) override;
+
+ status_t getPreferredDeviceForStrategy(product_strategy_t strategy,
+ AudioDeviceTypeAddr &device) const override;
engineConfig::ParsingResult loadAudioPolicyEngineConfig();
@@ -115,11 +127,13 @@
status_t restoreOriginVolumeCurve(audio_stream_type_t stream);
- private:
+private:
AudioPolicyManagerObserver *mApmObserver = nullptr;
ProductStrategyMap mProductStrategies;
+ ProductStrategyPreferredRoutingMap mProductStrategyPreferredDevices;
VolumeGroupMap mVolumeGroups;
+ LastRemovableMediaDevices mLastRemovableMediaDevices;
audio_mode_t mPhoneState = AUDIO_MODE_NORMAL; /**< current phone state. */
/** current forced use configuration. */
diff --git a/services/audiopolicy/engine/common/include/LastRemovableMediaDevices.h b/services/audiopolicy/engine/common/include/LastRemovableMediaDevices.h
new file mode 100755
index 0000000..a3053a4
--- /dev/null
+++ b/services/audiopolicy/engine/common/include/LastRemovableMediaDevices.h
@@ -0,0 +1,52 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_LAST_REMOVABLE_MEDIA_DEVICES_H
+#define ANDROID_LAST_REMOVABLE_MEDIA_DEVICES_H
+
+#include <vector>
+#include <HwModule.h>
+#include <system/audio_policy.h>
+
+namespace android {
+
+typedef enum {
+ GROUP_NONE = -1,
+ GROUP_WIRED,
+ GROUP_BT_A2DP,
+ NUM_GROUP
+} device_out_group_t;
+
+class LastRemovableMediaDevices
+{
+public:
+ void setRemovableMediaDevices(sp<DeviceDescriptor> desc, audio_policy_dev_state_t state);
+ std::vector<audio_devices_t> getLastRemovableMediaDevices(
+ device_out_group_t group = GROUP_NONE) const;
+
+private:
+ struct DeviceGroupDescriptor {
+ sp<DeviceDescriptor> desc;
+ device_out_group_t group;
+ };
+ std::vector<DeviceGroupDescriptor> mMediaDevices;
+
+ device_out_group_t getDeviceOutGroup(audio_devices_t device) const;
+};
+
+} // namespace android
+
+#endif // ANDROID_LAST_REMOVABLE_MEDIA_DEVICES_H
diff --git a/services/audiopolicy/engine/common/include/ProductStrategy.h b/services/audiopolicy/engine/common/include/ProductStrategy.h
index 1a2a198..3ebe7d1 100644
--- a/services/audiopolicy/engine/common/include/ProductStrategy.h
+++ b/services/audiopolicy/engine/common/include/ProductStrategy.h
@@ -19,7 +19,6 @@
#include "VolumeGroup.h"
#include <system/audio.h>
-#include <AudioPolicyManagerInterface.h>
#include <utils/RefBase.h>
#include <HandleGenerator.h>
#include <string>
@@ -27,6 +26,9 @@
#include <map>
#include <utils/Errors.h>
#include <utils/String8.h>
+#include <media/AudioAttributes.h>
+#include <media/AudioContainers.h>
+#include <media/AudioPolicy.h>
namespace android {
@@ -77,12 +79,12 @@
std::string getDeviceAddress() const { return mDeviceAddress; }
- void setDeviceTypes(audio_devices_t devices)
+ void setDeviceTypes(const DeviceTypeSet& devices)
{
mApplicableDevices = devices;
}
- audio_devices_t getDeviceTypes() const { return mApplicableDevices; }
+ DeviceTypeSet getDeviceTypes() const { return mApplicableDevices; }
audio_attributes_t getAttributesForStreamType(audio_stream_type_t stream) const;
audio_stream_type_t getStreamTypeForAttributes(const audio_attributes_t &attr) const;
@@ -109,7 +111,7 @@
/**
* Applicable device(s) type mask for this strategy.
*/
- audio_devices_t mApplicableDevices = AUDIO_DEVICE_NONE;
+ DeviceTypeSet mApplicableDevices;
};
class ProductStrategyMap : public std::map<product_strategy_t, sp<ProductStrategy> >
@@ -144,7 +146,7 @@
*/
audio_attributes_t getAttributesForProductStrategy(product_strategy_t strategy) const;
- audio_devices_t getDeviceTypesForProductStrategy(product_strategy_t strategy) const;
+ DeviceTypeSet getDeviceTypesForProductStrategy(product_strategy_t strategy) const;
std::string getDeviceAddressForProductStrategy(product_strategy_t strategy) const;
@@ -162,4 +164,10 @@
product_strategy_t mDefaultStrategy = PRODUCT_STRATEGY_NONE;
};
+class ProductStrategyPreferredRoutingMap : public std::map<product_strategy_t, AudioDeviceTypeAddr>
+{
+public:
+ void dump(String8 *dst, int spaces = 0) const;
+};
+
} // namespace android
diff --git a/services/audiopolicy/engine/common/include/VolumeCurve.h b/services/audiopolicy/engine/common/include/VolumeCurve.h
index 54314e3..2e75ff1 100644
--- a/services/audiopolicy/engine/common/include/VolumeCurve.h
+++ b/services/audiopolicy/engine/common/include/VolumeCurve.h
@@ -18,7 +18,6 @@
#include "IVolumeCurves.h"
#include <policy.h>
-#include <AudioPolicyManagerInterface.h>
#include <utils/RefBase.h>
#include <HandleGenerator.h>
#include <utils/String8.h>
@@ -92,9 +91,9 @@
return valueFor(device);
}
- virtual int getVolumeIndex(audio_devices_t device) const
+ virtual int getVolumeIndex(const DeviceTypeSet& deviceTypes) const
{
- device = Volume::getDeviceForVolume(device);
+ audio_devices_t device = Volume::getDeviceForVolume(deviceTypes);
// there is always a valid entry for AUDIO_DEVICE_OUT_DEFAULT_FOR_VOLUME
if (mIndexCur.find(device) == end(mIndexCur)) {
device = AUDIO_DEVICE_OUT_DEFAULT_FOR_VOLUME;
@@ -115,7 +114,7 @@
bool hasVolumeIndexForDevice(audio_devices_t device) const
{
- device = Volume::getDeviceForVolume(device);
+ device = Volume::getDeviceForVolume({device});
return mIndexCur.find(device) != end(mIndexCur);
}
diff --git a/services/audiopolicy/engine/common/include/VolumeGroup.h b/services/audiopolicy/engine/common/include/VolumeGroup.h
index c34b406..5378f64 100644
--- a/services/audiopolicy/engine/common/include/VolumeGroup.h
+++ b/services/audiopolicy/engine/common/include/VolumeGroup.h
@@ -16,7 +16,6 @@
#pragma once
-#include <AudioPolicyManagerInterface.h>
#include <VolumeCurve.h>
#include <system/audio.h>
#include <utils/RefBase.h>
diff --git a/services/audiopolicy/engine/common/src/EngineBase.cpp b/services/audiopolicy/engine/common/src/EngineBase.cpp
index 07a7e65..b46a50a 100644
--- a/services/audiopolicy/engine/common/src/EngineBase.cpp
+++ b/services/audiopolicy/engine/common/src/EngineBase.cpp
@@ -39,7 +39,7 @@
{
ALOGV("setPhoneState() state %d", state);
- if (state < 0 || state >= AUDIO_MODE_CNT) {
+ if (state < 0 || uint32_t(state) >= AUDIO_MODE_CNT) {
ALOGW("setPhoneState() invalid state %d", state);
return BAD_VALUE;
}
@@ -63,6 +63,17 @@
return NO_ERROR;
}
+status_t EngineBase::setDeviceConnectionState(const sp<DeviceDescriptor> devDesc,
+ audio_policy_dev_state_t state)
+{
+ audio_devices_t deviceType = devDesc->type();
+ if ((deviceType != AUDIO_DEVICE_NONE) && audio_is_output_device(deviceType)) {
+ mLastRemovableMediaDevices.setRemovableMediaDevices(devDesc, state);
+ }
+
+ return NO_ERROR;
+}
+
product_strategy_t EngineBase::getProductStrategyForAttributes(const audio_attributes_t &attr) const
{
return mProductStrategies.getProductStrategyForAttributes(attr);
@@ -95,48 +106,49 @@
engineConfig::ParsingResult EngineBase::loadAudioPolicyEngineConfig()
{
- auto loadProductStrategies =
- [](auto& strategyConfigs, auto& productStrategies, auto& volumeGroups) {
- for (auto& strategyConfig : strategyConfigs) {
- sp<ProductStrategy> strategy = new ProductStrategy(strategyConfig.name);
- for (const auto &group : strategyConfig.attributesGroups) {
- const auto &iter = std::find_if(begin(volumeGroups), end(volumeGroups),
- [&group](const auto &volumeGroup) {
- return group.volumeGroup == volumeGroup.second->getName(); });
- ALOG_ASSERT(iter != end(volumeGroups), "Invalid Volume Group Name %s",
- group.volumeGroup.c_str());
- if (group.stream != AUDIO_STREAM_DEFAULT) {
- iter->second->addSupportedStream(group.stream);
- }
- for (const auto &attr : group.attributesVect) {
- strategy->addAttributes({group.stream, iter->second->getId(), attr});
- iter->second->addSupportedAttributes(attr);
- }
- }
- product_strategy_t strategyId = strategy->getId();
- productStrategies[strategyId] = strategy;
- }
- };
- auto loadVolumeGroups = [](auto &volumeConfigs, auto &volumeGroups) {
- for (auto &volumeConfig : volumeConfigs) {
- sp<VolumeGroup> volumeGroup = new VolumeGroup(volumeConfig.name, volumeConfig.indexMin,
- volumeConfig.indexMax);
- volumeGroups[volumeGroup->getId()] = volumeGroup;
+ auto loadVolumeConfig = [](auto &volumeGroups, auto &volumeConfig) {
+ // Ensure name unicity to prevent duplicate
+ const auto &iter = std::find_if(std::begin(volumeGroups), std::end(volumeGroups),
+ [&volumeConfig](const auto &volumeGroup) {
+ return volumeConfig.name == volumeGroup.second->getName(); });
+ LOG_ALWAYS_FATAL_IF(iter != std::end(volumeGroups),
+ "group name %s defined twice, review the configuration",
+ volumeConfig.name.c_str());
- for (auto &configCurve : volumeConfig.volumeCurves) {
- device_category deviceCat = DEVICE_CATEGORY_SPEAKER;
- if (!DeviceCategoryConverter::fromString(configCurve.deviceCategory, deviceCat)) {
- ALOGE("%s: Invalid %s", __FUNCTION__, configCurve.deviceCategory.c_str());
- continue;
- }
- sp<VolumeCurve> curve = new VolumeCurve(deviceCat);
- for (auto &point : configCurve.curvePoints) {
- curve->add({point.index, point.attenuationInMb});
- }
- volumeGroup->add(curve);
+ sp<VolumeGroup> volumeGroup = new VolumeGroup(volumeConfig.name, volumeConfig.indexMin,
+ volumeConfig.indexMax);
+ volumeGroups[volumeGroup->getId()] = volumeGroup;
+
+ for (auto &configCurve : volumeConfig.volumeCurves) {
+ device_category deviceCat = DEVICE_CATEGORY_SPEAKER;
+ if (!DeviceCategoryConverter::fromString(configCurve.deviceCategory, deviceCat)) {
+ ALOGE("%s: Invalid %s", __FUNCTION__, configCurve.deviceCategory.c_str());
+ continue;
}
+ sp<VolumeCurve> curve = new VolumeCurve(deviceCat);
+ for (auto &point : configCurve.curvePoints) {
+ curve->add({point.index, point.attenuationInMb});
+ }
+ volumeGroup->add(curve);
+ }
+ return volumeGroup;
+ };
+ auto addSupportedAttributesToGroup = [](auto &group, auto &volumeGroup, auto &strategy) {
+ for (const auto &attr : group.attributesVect) {
+ strategy->addAttributes({group.stream, volumeGroup->getId(), attr});
+ volumeGroup->addSupportedAttributes(attr);
}
};
+ auto checkStreamForGroups = [](auto streamType, const auto &volumeGroups) {
+ const auto &iter = std::find_if(std::begin(volumeGroups), std::end(volumeGroups),
+ [&streamType](const auto &volumeGroup) {
+ const auto& streams = volumeGroup.second->getStreamTypes();
+ return std::find(std::begin(streams), std::end(streams), streamType) !=
+ std::end(streams);
+ });
+ return iter != end(volumeGroups);
+ };
+
auto result = engineConfig::parse();
if (result.parsedConfig == nullptr) {
ALOGW("%s: No configuration found, using default matching phone experience.", __FUNCTION__);
@@ -144,11 +156,67 @@
android::status_t ret = engineConfig::parseLegacyVolumes(config.volumeGroups);
result = {std::make_unique<engineConfig::Config>(config),
static_cast<size_t>(ret == NO_ERROR ? 0 : 1)};
+ } else {
+ // Append for internal use only volume groups (e.g. rerouting/patch)
+ result.parsedConfig->volumeGroups.insert(
+ std::end(result.parsedConfig->volumeGroups),
+ std::begin(gSystemVolumeGroups), std::end(gSystemVolumeGroups));
}
+ // Append for internal use only strategies (e.g. rerouting/patch)
+ result.parsedConfig->productStrategies.insert(
+ std::end(result.parsedConfig->productStrategies),
+ std::begin(gOrderedSystemStrategies), std::end(gOrderedSystemStrategies));
+
+
ALOGE_IF(result.nbSkippedElement != 0, "skipped %zu elements", result.nbSkippedElement);
- loadVolumeGroups(result.parsedConfig->volumeGroups, mVolumeGroups);
- loadProductStrategies(result.parsedConfig->productStrategies, mProductStrategies,
- mVolumeGroups);
+
+ engineConfig::VolumeGroup defaultVolumeConfig;
+ engineConfig::VolumeGroup defaultSystemVolumeConfig;
+ for (auto &volumeConfig : result.parsedConfig->volumeGroups) {
+ // save default volume config for streams not defined in configuration
+ if (volumeConfig.name.compare("AUDIO_STREAM_MUSIC") == 0) {
+ defaultVolumeConfig = volumeConfig;
+ }
+ if (volumeConfig.name.compare("AUDIO_STREAM_PATCH") == 0) {
+ defaultSystemVolumeConfig = volumeConfig;
+ }
+ loadVolumeConfig(mVolumeGroups, volumeConfig);
+ }
+ for (auto& strategyConfig : result.parsedConfig->productStrategies) {
+ sp<ProductStrategy> strategy = new ProductStrategy(strategyConfig.name);
+ for (const auto &group : strategyConfig.attributesGroups) {
+ const auto &iter = std::find_if(begin(mVolumeGroups), end(mVolumeGroups),
+ [&group](const auto &volumeGroup) {
+ return group.volumeGroup == volumeGroup.second->getName(); });
+ sp<VolumeGroup> volumeGroup = nullptr;
+ // If no volume group provided for this strategy, creates a new one using
+ // Music Volume Group configuration (considered as the default)
+ if (iter == end(mVolumeGroups)) {
+ engineConfig::VolumeGroup volumeConfig;
+ if (group.stream >= AUDIO_STREAM_PUBLIC_CNT) {
+ volumeConfig = defaultSystemVolumeConfig;
+ } else {
+ volumeConfig = defaultVolumeConfig;
+ }
+ ALOGW("%s: No configuration of %s found, using default volume configuration"
+ , __FUNCTION__, group.volumeGroup.c_str());
+ volumeConfig.name = group.volumeGroup;
+ volumeGroup = loadVolumeConfig(mVolumeGroups, volumeConfig);
+ } else {
+ volumeGroup = iter->second;
+ }
+ if (group.stream != AUDIO_STREAM_DEFAULT) {
+ // A legacy stream can be assigned once to a volume group
+ LOG_ALWAYS_FATAL_IF(checkStreamForGroups(group.stream, mVolumeGroups),
+ "stream %s already assigned to a volume group, "
+ "review the configuration", toString(group.stream).c_str());
+ volumeGroup->addSupportedStream(group.stream);
+ }
+ addSupportedAttributesToGroup(group, volumeGroup, strategy);
+ }
+ product_strategy_t strategyId = strategy->getId();
+ mProductStrategies[strategyId] = strategy;
+ }
mProductStrategies.initialize();
return result;
}
@@ -272,9 +340,57 @@
return NO_ERROR;
}
+status_t EngineBase::setPreferredDeviceForStrategy(product_strategy_t strategy,
+ const AudioDeviceTypeAddr &device)
+{
+ // verify strategy exists
+ if (mProductStrategies.find(strategy) == mProductStrategies.end()) {
+ ALOGE("%s invalid strategy %u", __func__, strategy);
+ return BAD_VALUE;
+ }
+
+ mProductStrategyPreferredDevices[strategy] = device;
+ return NO_ERROR;
+}
+
+status_t EngineBase::removePreferredDeviceForStrategy(product_strategy_t strategy)
+{
+ // verify strategy exists
+ if (mProductStrategies.find(strategy) == mProductStrategies.end()) {
+ ALOGE("%s invalid strategy %u", __func__, strategy);
+ return BAD_VALUE;
+ }
+
+ if (mProductStrategyPreferredDevices.erase(strategy) == 0) {
+ // no preferred device was set
+ return NAME_NOT_FOUND;
+ }
+ return NO_ERROR;
+}
+
+status_t EngineBase::getPreferredDeviceForStrategy(product_strategy_t strategy,
+ AudioDeviceTypeAddr &device) const
+{
+ // verify strategy exists
+ if (mProductStrategies.find(strategy) == mProductStrategies.end()) {
+ ALOGE("%s unknown strategy %u", __func__, strategy);
+ return BAD_VALUE;
+ }
+ // preferred device for this strategy?
+ auto devIt = mProductStrategyPreferredDevices.find(strategy);
+ if (devIt == mProductStrategyPreferredDevices.end()) {
+ ALOGV("%s no preferred device for strategy %u", __func__, strategy);
+ return NAME_NOT_FOUND;
+ }
+
+ device = devIt->second;
+ return NO_ERROR;
+}
+
void EngineBase::dump(String8 *dst) const
{
mProductStrategies.dump(dst, 2);
+ mProductStrategyPreferredDevices.dump(dst, 2);
mVolumeGroups.dump(dst, 2);
}
diff --git a/services/audiopolicy/engine/common/src/EngineDefaultConfig.h b/services/audiopolicy/engine/common/src/EngineDefaultConfig.h
index fede0d9..a3071d7 100644
--- a/services/audiopolicy/engine/common/src/EngineDefaultConfig.h
+++ b/services/audiopolicy/engine/common/src/EngineDefaultConfig.h
@@ -81,6 +81,10 @@
},
{"STRATEGY_MEDIA",
{
+ {"assistant", AUDIO_STREAM_ASSISTANT, "AUDIO_STREAM_ASSISTANT",
+ {{AUDIO_CONTENT_TYPE_SPEECH, AUDIO_USAGE_ASSISTANT,
+ AUDIO_SOURCE_DEFAULT, 0, ""}}
+ },
{"music", AUDIO_STREAM_MUSIC, "AUDIO_STREAM_MUSIC",
{
{AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_MEDIA, AUDIO_SOURCE_DEFAULT, 0, ""},
@@ -114,15 +118,22 @@
AUDIO_FLAG_BEACON, ""}}
}
},
- },
- {"STRATEGY_REROUTING",
+ }
+};
+
+/**
+ * For Internal use of respectively audio policy and audioflinger
+ * For compatibility reason why apm volume config file, volume group name is the stream type.
+ */
+const engineConfig::ProductStrategies gOrderedSystemStrategies = {
+ {"rerouting",
{
{"", AUDIO_STREAM_REROUTING, "AUDIO_STREAM_REROUTING",
{{AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN, AUDIO_SOURCE_DEFAULT, 0, ""}}
}
},
},
- {"STRATEGY_PATCH",
+ {"patch",
{
{"", AUDIO_STREAM_PATCH, "AUDIO_STREAM_PATCH",
{{AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN, AUDIO_SOURCE_DEFAULT, 0, ""}}
@@ -130,6 +141,28 @@
},
}
};
+const engineConfig::VolumeGroups gSystemVolumeGroups = {
+ {"AUDIO_STREAM_REROUTING", 0, 1,
+ {
+ {"DEVICE_CATEGORY_SPEAKER", {{0,0}, {100, 0}}},
+ {"DEVICE_CATEGORY_HEADSET", {{0,0}, {100, 0}}},
+ {"DEVICE_CATEGORY_EARPIECE", {{0,0}, {100, 0}}},
+ {"DEVICE_CATEGORY_EXT_MEDIA", {{0,0}, {100, 0}}},
+ {"DEVICE_CATEGORY_HEARING_AID", {{0,0}, {100, 0}}},
+
+ }
+ },
+ {"AUDIO_STREAM_PATCH", 0, 1,
+ {
+ {"DEVICE_CATEGORY_SPEAKER", {{0,0}, {100, 0}}},
+ {"DEVICE_CATEGORY_HEADSET", {{0,0}, {100, 0}}},
+ {"DEVICE_CATEGORY_EARPIECE", {{0,0}, {100, 0}}},
+ {"DEVICE_CATEGORY_EXT_MEDIA", {{0,0}, {100, 0}}},
+ {"DEVICE_CATEGORY_HEARING_AID", {{0,0}, {100, 0}}},
+
+ }
+ }
+};
const engineConfig::Config gDefaultEngineConfig = {
1.0,
diff --git a/services/audiopolicy/engine/common/src/LastRemovableMediaDevices.cpp b/services/audiopolicy/engine/common/src/LastRemovableMediaDevices.cpp
new file mode 100755
index 0000000..87b6aaf
--- /dev/null
+++ b/services/audiopolicy/engine/common/src/LastRemovableMediaDevices.cpp
@@ -0,0 +1,78 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "APM::AudioPolicyEngine/LastRemovableMediaDevices"
+//#define LOG_NDEBUG 0
+
+#include "LastRemovableMediaDevices.h"
+#include <log/log.h>
+
+namespace android {
+
+void LastRemovableMediaDevices::setRemovableMediaDevices(sp<DeviceDescriptor> desc,
+ audio_policy_dev_state_t state)
+{
+ if (desc == nullptr) {
+ return;
+ } else {
+ if ((state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) &&
+ (getDeviceOutGroup(desc->type()) != GROUP_NONE)) {
+ setRemovableMediaDevices(desc, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE);
+ mMediaDevices.insert(mMediaDevices.begin(), {desc, getDeviceOutGroup(desc->type())});
+ } else if (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) {
+ for (auto iter = mMediaDevices.begin(); iter != mMediaDevices.end(); ++iter) {
+ if ((iter->desc)->equals(desc)) {
+ mMediaDevices.erase(iter);
+ break;
+ }
+ }
+ }
+ }
+}
+
+std::vector<audio_devices_t> LastRemovableMediaDevices::getLastRemovableMediaDevices(
+ device_out_group_t group) const
+{
+ std::vector<audio_devices_t> ret;
+ for (auto iter = mMediaDevices.begin(); iter != mMediaDevices.end(); ++iter) {
+ if ((group == GROUP_NONE) || (group == getDeviceOutGroup((iter->desc)->type()))) {
+ ret.push_back((iter->desc)->type());
+ }
+ }
+ return ret;
+}
+
+device_out_group_t LastRemovableMediaDevices::getDeviceOutGroup(audio_devices_t device) const
+{
+ switch (device) {
+ case AUDIO_DEVICE_OUT_WIRED_HEADPHONE:
+ case AUDIO_DEVICE_OUT_LINE:
+ case AUDIO_DEVICE_OUT_WIRED_HEADSET:
+ case AUDIO_DEVICE_OUT_USB_HEADSET:
+ case AUDIO_DEVICE_OUT_USB_ACCESSORY:
+ case AUDIO_DEVICE_OUT_USB_DEVICE:
+ case AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET:
+ return GROUP_WIRED;
+ case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER:
+ return GROUP_BT_A2DP;
+ default:
+ return GROUP_NONE;
+ }
+}
+
+} // namespace android
diff --git a/services/audiopolicy/engine/common/src/ProductStrategy.cpp b/services/audiopolicy/engine/common/src/ProductStrategy.cpp
index f74f190..fe15ff6 100644
--- a/services/audiopolicy/engine/common/src/ProductStrategy.cpp
+++ b/services/audiopolicy/engine/common/src/ProductStrategy.cpp
@@ -19,6 +19,7 @@
#include "ProductStrategy.h"
+#include <media/AudioProductStrategy.h>
#include <media/TypeConverter.h>
#include <utils/String8.h>
#include <cstdint>
@@ -142,8 +143,9 @@
{
dst->appendFormat("\n%*s-%s (id: %d)\n", spaces, "", mName.c_str(), mId);
std::string deviceLiteral;
- if (!OutputDeviceConverter::toString(mApplicableDevices, deviceLiteral)) {
- ALOGE("%s: failed to convert device %d", __FUNCTION__, mApplicableDevices);
+ if (!deviceTypesToString(mApplicableDevices, deviceLiteral)) {
+ ALOGE("%s: failed to convert device %s",
+ __FUNCTION__, dumpDeviceTypes(mApplicableDevices).c_str());
}
dst->appendFormat("%*sSelected Device: {type:%s, @:%s}\n", spaces + 2, "",
deviceLiteral.c_str(), mDeviceAddress.c_str());
@@ -235,14 +237,14 @@
}
-audio_devices_t ProductStrategyMap::getDeviceTypesForProductStrategy(
+DeviceTypeSet ProductStrategyMap::getDeviceTypesForProductStrategy(
product_strategy_t strategy) const
{
if (find(strategy) == end()) {
ALOGE("Invalid %d strategy requested, returning device for default strategy", strategy);
product_strategy_t defaultStrategy = getDefault();
if (defaultStrategy == PRODUCT_STRATEGY_NONE) {
- return AUDIO_DEVICE_NONE;
+ return {AUDIO_DEVICE_NONE};
}
return at(getDefault())->getDeviceTypes();
}
@@ -308,5 +310,15 @@
}
}
+void ProductStrategyPreferredRoutingMap::dump(android::String8* dst, int spaces) const {
+ dst->appendFormat("\n%*sPreferred devices per product strategy dump:", spaces, "");
+ for (const auto& iter : *this) {
+ dst->appendFormat("\n%*sStrategy %u dev:%08x addr:%s",
+ spaces + 2, "",
+ (uint32_t) iter.first,
+ iter.second.mType, iter.second.mAddress.c_str());
+ }
+ dst->appendFormat("\n");
+}
}
diff --git a/services/audiopolicy/engine/config/Android.bp b/services/audiopolicy/engine/config/Android.bp
index 6e72f2a..ff840f9 100644
--- a/services/audiopolicy/engine/config/Android.bp
+++ b/services/audiopolicy/engine/config/Android.bp
@@ -1,9 +1,8 @@
-cc_library_static {
+cc_library {
name: "libaudiopolicyengine_config",
export_include_dirs: ["include"],
include_dirs: [
"external/libxml2/include",
- "external/icu/icu4c/source/common",
],
srcs: [
"src/EngineConfig.cpp",
@@ -15,17 +14,14 @@
],
shared_libs: [
"libmedia_helper",
- "libandroidicu",
"libxml2",
"libutils",
"liblog",
"libcutils",
],
- static_libs: [
- "libaudiopolicycomponents",
- ],
header_libs: [
"libaudio_system_headers",
- "libaudiopolicycommon",
+ "libmedia_headers",
+ "libaudioclient_headers",
],
}
diff --git a/services/audiopolicy/engine/config/src/EngineConfig.cpp b/services/audiopolicy/engine/config/src/EngineConfig.cpp
index 1ad7739..7f8cdd9 100644
--- a/services/audiopolicy/engine/config/src/EngineConfig.cpp
+++ b/services/audiopolicy/engine/config/src/EngineConfig.cpp
@@ -18,7 +18,6 @@
//#define LOG_NDEBUG 0
#include "EngineConfig.h"
-#include <policy.h>
#include <cutils/properties.h>
#include <media/TypeConverter.h>
#include <media/convert.h>
@@ -32,9 +31,9 @@
#include <istream>
#include <cstdint>
+#include <stdarg.h>
#include <string>
-
namespace android {
using utilities::convertTo;
@@ -603,7 +602,39 @@
return NO_ERROR;
}
+namespace {
+
+class XmlErrorHandler {
+public:
+ XmlErrorHandler() {
+ xmlSetGenericErrorFunc(this, &xmlErrorHandler);
+ }
+ XmlErrorHandler(const XmlErrorHandler&) = delete;
+ XmlErrorHandler(XmlErrorHandler&&) = delete;
+ XmlErrorHandler& operator=(const XmlErrorHandler&) = delete;
+ XmlErrorHandler& operator=(XmlErrorHandler&&) = delete;
+ ~XmlErrorHandler() {
+ xmlSetGenericErrorFunc(NULL, NULL);
+ if (!mErrorMessage.empty()) {
+ ALOG(LOG_ERROR, "libxml2", "%s", mErrorMessage.c_str());
+ }
+ }
+ static void xmlErrorHandler(void* ctx, const char* msg, ...) {
+ char buffer[256];
+ va_list args;
+ va_start(args, msg);
+ vsnprintf(buffer, sizeof(buffer), msg, args);
+ va_end(args);
+ static_cast<XmlErrorHandler*>(ctx)->mErrorMessage += buffer;
+ }
+private:
+ std::string mErrorMessage;
+};
+
+} // namespace
+
ParsingResult parse(const char* path) {
+ XmlErrorHandler errorHandler;
xmlDocPtr doc;
doc = xmlParseFile(path);
if (doc == NULL) {
@@ -641,6 +672,7 @@
}
android::status_t parseLegacyVolumeFile(const char* path, VolumeGroups &volumeGroups) {
+ XmlErrorHandler errorHandler;
xmlDocPtr doc;
doc = xmlParseFile(path);
if (doc == NULL) {
diff --git a/services/audiopolicy/engine/interface/AudioPolicyManagerObserver.h b/services/audiopolicy/engine/interface/AudioPolicyManagerObserver.h
index ebd82a7..349f969 100644
--- a/services/audiopolicy/engine/interface/AudioPolicyManagerObserver.h
+++ b/services/audiopolicy/engine/interface/AudioPolicyManagerObserver.h
@@ -16,8 +16,7 @@
#pragma once
-#include <AudioGain.h>
-#include <AudioPort.h>
+#include <PolicyAudioPort.h>
#include <AudioPatch.h>
#include <IOProfile.h>
#include <DeviceDescriptor.h>
diff --git a/services/audiopolicy/engine/interface/AudioPolicyManagerInterface.h b/services/audiopolicy/engine/interface/EngineInterface.h
similarity index 87%
rename from services/audiopolicy/engine/interface/AudioPolicyManagerInterface.h
rename to services/audiopolicy/engine/interface/EngineInterface.h
index b7fd031..dfb20b5 100644
--- a/services/audiopolicy/engine/interface/AudioPolicyManagerInterface.h
+++ b/services/audiopolicy/engine/interface/EngineInterface.h
@@ -38,7 +38,7 @@
/**
* This interface is dedicated to the policy manager that a Policy Engine shall implement.
*/
-class AudioPolicyManagerInterface
+class EngineInterface
{
public:
/**
@@ -292,10 +292,49 @@
*/
virtual status_t listAudioVolumeGroups(AudioVolumeGroupVector &groups) const = 0;
+ /**
+ * @brief setPreferredDeviceForStrategy sets the default device to be used for a
+ * strategy when available
+ * @param strategy the audio strategy whose routing will be affected
+ * @param device the audio device to route to when available
+ * @return BAD_VALUE if the strategy is invalid,
+ * or NO_ERROR if the preferred device was set
+ */
+ virtual status_t setPreferredDeviceForStrategy(product_strategy_t strategy,
+ const AudioDeviceTypeAddr &device) = 0;
+
+ /**
+ * @brief removePreferredDeviceForStrategy removes the preferred device previously set
+ * for the given strategy
+ * @param strategy the audio strategy whose routing will be affected
+ * @return BAD_VALUE if the strategy is invalid,
+ * or NO_ERROR if the preferred device was removed
+ */
+ virtual status_t removePreferredDeviceForStrategy(product_strategy_t strategy) = 0;
+
+ /**
+ * @brief getPreferredDeviceForStrategy queries which device is set as the
+ * preferred device for the given strategy
+ * @param strategy the strategy to query
+ * @param device returns configured as the preferred device if one was set
+ * @return BAD_VALUE if the strategy is invalid,
+ * or NAME_NOT_FOUND if no preferred device was set
+ * or NO_ERROR if the device parameter was initialized to the preferred device
+ */
+ virtual status_t getPreferredDeviceForStrategy(product_strategy_t strategy,
+ AudioDeviceTypeAddr &device) const = 0;
+
+
virtual void dump(String8 *dst) const = 0;
protected:
- virtual ~AudioPolicyManagerInterface() {}
+ virtual ~EngineInterface() {}
};
+__attribute__((visibility("default")))
+extern "C" EngineInterface* createEngineInstance();
+
+__attribute__((visibility("default")))
+extern "C" void destroyEngineInstance(EngineInterface *engine);
+
} // namespace android
diff --git a/services/audiopolicy/engineconfigurable/Android.bp b/services/audiopolicy/engineconfigurable/Android.bp
index c27dc88..8f522f0 100644
--- a/services/audiopolicy/engineconfigurable/Android.bp
+++ b/services/audiopolicy/engineconfigurable/Android.bp
@@ -33,6 +33,7 @@
],
shared_libs: [
+ "libaudiofoundation",
"liblog",
"libcutils",
"libutils",
diff --git a/services/audiopolicy/engineconfigurable/config/Android.bp b/services/audiopolicy/engineconfigurable/config/Android.bp
new file mode 100644
index 0000000..fe3eae0
--- /dev/null
+++ b/services/audiopolicy/engineconfigurable/config/Android.bp
@@ -0,0 +1,31 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+// Root soong_namespace for common components
+
+prebuilt_etc {
+ name: "audio_policy_engine_criteria.xml",
+ vendor: true,
+ src: ":audio_policy_engine_criteria",
+}
+filegroup {
+ name: "audio_policy_engine_criterion_types_template",
+ srcs: ["example/common/audio_policy_engine_criterion_types.xml.in"],
+}
+filegroup {
+ name: "audio_policy_engine_criteria",
+ srcs: ["example/common/audio_policy_engine_criteria.xml"],
+}
diff --git a/services/audiopolicy/engineconfigurable/config/example/Android.mk b/services/audiopolicy/engineconfigurable/config/example/Android.mk
deleted file mode 100644
index a0f1a90..0000000
--- a/services/audiopolicy/engineconfigurable/config/example/Android.mk
+++ /dev/null
@@ -1,151 +0,0 @@
-LOCAL_PATH := $(call my-dir)
-
-ifdef BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION
-
-TOOLS := frameworks/av/services/audiopolicy/engineconfigurable/tools
-PROVISION_CRITERION_TYPES := $(TOOLS)/provision_criterion_types_from_android_headers.mk
-
-##################################################################
-# CONFIGURATION TOP FILE
-##################################################################
-
-ifeq ($(BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION), phone_configurable)
-
-include $(CLEAR_VARS)
-LOCAL_MODULE := audio_policy_engine_configuration.xml
-
-LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE_CLASS := ETC
-LOCAL_VENDOR_MODULE := true
-LOCAL_SRC_FILES := phone/$(LOCAL_MODULE)
-
-LOCAL_REQUIRED_MODULES := \
- audio_policy_engine_product_strategies.xml \
- audio_policy_engine_stream_volumes.xml \
- audio_policy_engine_default_stream_volumes.xml \
- audio_policy_engine_criteria.xml \
- audio_policy_engine_criterion_types.xml
-
-include $(BUILD_PREBUILT)
-
-include $(CLEAR_VARS)
-LOCAL_MODULE := audio_policy_engine_product_strategies.xml
-LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE_CLASS := ETC
-LOCAL_VENDOR_MODULE := true
-LOCAL_SRC_FILES := phone/$(LOCAL_MODULE)
-include $(BUILD_PREBUILT)
-
-include $(CLEAR_VARS)
-LOCAL_MODULE := audio_policy_engine_stream_volumes.xml
-LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE_CLASS := ETC
-LOCAL_VENDOR_MODULE := true
-LOCAL_SRC_FILES := phone/$(LOCAL_MODULE)
-include $(BUILD_PREBUILT)
-
-include $(CLEAR_VARS)
-LOCAL_MODULE := audio_policy_engine_default_stream_volumes.xml
-LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE_CLASS := ETC
-LOCAL_VENDOR_MODULE := true
-LOCAL_SRC_FILES := phone/$(LOCAL_MODULE)
-include $(BUILD_PREBUILT)
-
-endif # ifeq ($(BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION), phone_configurable)
-
-
-ifeq ($(BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION),$(filter $(BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION),automotive_configurable caremu_configurable))
-
-##################################################################
-# AUTOMOTIVE CONFIGURATION TOP FILE
-##################################################################
-include $(CLEAR_VARS)
-LOCAL_MODULE := audio_policy_engine_configuration.xml
-LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE_CLASS := ETC
-LOCAL_VENDOR_MODULE := true
-LOCAL_SRC_FILES := automotive/$(LOCAL_MODULE)
-
-LOCAL_REQUIRED_MODULES := \
- audio_policy_engine_product_strategies.xml \
- audio_policy_engine_criteria.xml \
- audio_policy_engine_criterion_types.xml \
- audio_policy_engine_volumes.xml
-
-include $(BUILD_PREBUILT)
-
-endif #ifeq ($(BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION),$(filter $(BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION),automotive_configurable caremu_configurable))
-
-ifeq ($(BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION), automotive_configurable)
-
-##################################################################
-# CONFIGURATION FILES
-##################################################################
-
-include $(CLEAR_VARS)
-LOCAL_MODULE := audio_policy_engine_product_strategies.xml
-LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE_CLASS := ETC
-LOCAL_VENDOR_MODULE := true
-LOCAL_SRC_FILES := automotive/$(LOCAL_MODULE)
-include $(BUILD_PREBUILT)
-
-include $(CLEAR_VARS)
-LOCAL_MODULE := audio_policy_engine_volumes.xml
-LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE_CLASS := ETC
-LOCAL_VENDOR_MODULE := true
-LOCAL_SRC_FILES := automotive/$(LOCAL_MODULE)
-include $(BUILD_PREBUILT)
-
-endif #ifeq ($(BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION), automotive_configurable)
-
-ifeq ($(BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION), caremu_configurable)
-
-##################################################################
-# CONFIGURATION FILES
-##################################################################
-
-include $(CLEAR_VARS)
-LOCAL_MODULE := audio_policy_engine_product_strategies.xml
-LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE_CLASS := ETC
-LOCAL_VENDOR_MODULE := true
-LOCAL_SRC_FILES := caremu/$(LOCAL_MODULE)
-include $(BUILD_PREBUILT)
-
-include $(CLEAR_VARS)
-LOCAL_MODULE := audio_policy_engine_volumes.xml
-LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE_CLASS := ETC
-LOCAL_VENDOR_MODULE := true
-LOCAL_SRC_FILES := caremu/$(LOCAL_MODULE)
-include $(BUILD_PREBUILT)
-
-endif #ifeq ($(BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION), caremu_configurable)
-
-ifeq ($(BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION),$(filter $(BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION),phone_configurable automotive_configurable caremu_configurable))
-
-include $(CLEAR_VARS)
-LOCAL_MODULE := audio_policy_engine_criteria.xml
-LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE_CLASS := ETC
-LOCAL_VENDOR_MODULE := true
-LOCAL_SRC_FILES := common/$(LOCAL_MODULE)
-include $(BUILD_PREBUILT)
-
-include $(CLEAR_VARS)
-LOCAL_MODULE := audio_policy_engine_criterion_types.xml
-LOCAL_MODULE_CLASS := ETC
-LOCAL_VENDOR_MODULE := true
-LOCAL_ADDITIONAL_DEPENDENCIES := $(TARGET_OUT_VENDOR_ETC)/primary_audio_policy_configuration.xml
-ANDROID_AUDIO_BASE_HEADER_FILE := system/media/audio/include/system/audio-base.h
-AUDIO_POLICY_CONFIGURATION_FILE := $(TARGET_OUT_VENDOR_ETC)/audio_policy_configuration.xml
-CRITERION_TYPES_FILE := $(LOCAL_PATH)/common/$(LOCAL_MODULE).in
-
-include $(PROVISION_CRITERION_TYPES)
-
-endif #ifeq ($(BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION),$(filter $(BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION),phone_configurable automotive_configurable caremu_configurable))
-
-endif #ifdef BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION
diff --git a/services/audiopolicy/engineconfigurable/config/example/automotive/Android.bp b/services/audiopolicy/engineconfigurable/config/example/automotive/Android.bp
new file mode 100644
index 0000000..f913a14
--- /dev/null
+++ b/services/audiopolicy/engineconfigurable/config/example/automotive/Android.bp
@@ -0,0 +1,94 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+// Import this namespace in order to use AOSP Automotive configuration example
+
+soong_namespace {
+ imports: [
+ "frameworks/av/services/audiopolicy/config",
+ ],
+}
+
+prebuilt_etc {
+ name: "audio_policy_engine_configuration.xml",
+ vendor: true,
+ src: ":audio_policy_engine_configuration",
+ required: [
+ ":audio_policy_engine_criterion_types.xml",
+ ":audio_policy_engine_criteria.xml",
+ ":audio_policy_engine_product_strategies.xml",
+ ":audio_policy_engine_volumes.xml",
+ ],
+}
+prebuilt_etc {
+ name: "audio_policy_engine_product_strategies.xml",
+ vendor: true,
+ src: "audio_policy_engine_product_strategies.xml",
+}
+prebuilt_etc {
+ name: "audio_policy_engine_volumes.xml",
+ vendor: true,
+ src: ":audio_policy_engine_volumes",
+}
+prebuilt_etc {
+ name: "audio_policy_engine_criterion_types.xml",
+ vendor: true,
+ src: ":audio_policy_engine_criterion_types",
+}
+
+//
+// Generate audio_policy_engine criterion type file => provides device addresses criterion type
+//
+genrule {
+ name: "audio_policy_engine_criterion_types",
+ defaults: ["buildpolicycriteriontypesrule"],
+ srcs: [
+ ":audio_policy_configuration_top_file",
+ ":audio_policy_configuration_files",
+ ],
+}
+filegroup {
+ name: "audio_policy_configuration_files",
+ srcs: [
+ ":r_submix_audio_policy_configuration",
+ ":default_volume_tables",
+ ":audio_policy_volumes",
+ ":surround_sound_configuration_5_0",
+ ":primary_audio_policy_configuration",
+ ],
+}
+filegroup {
+ name : "audio_policy_configuration_top_file",
+ srcs: [":audio_policy_configuration_generic"],
+}
+filegroup {
+ name: "audio_policy_engine_configuration",
+ srcs: ["audio_policy_engine_configuration.xml"],
+}
+filegroup {
+ name: "audio_policy_engine_volumes",
+ srcs: ["audio_policy_engine_volumes.xml"],
+}
+filegroup {
+ name: "audio_policy_engine_configuration_files",
+ srcs: [
+ ":audio_policy_engine_configuration",
+ "audio_policy_engine_product_strategies.xml",
+ ":audio_policy_engine_volumes",
+ ":audio_policy_engine_criterion_types",
+ ":audio_policy_engine_criteria",
+ ],
+}
diff --git a/services/audiopolicy/engineconfigurable/config/example/automotive/audio_policy_engine_product_strategies.xml b/services/audiopolicy/engineconfigurable/config/example/automotive/audio_policy_engine_product_strategies.xml
index 0ee83a2..f598cf2 100644
--- a/services/audiopolicy/engineconfigurable/config/example/automotive/audio_policy_engine_product_strategies.xml
+++ b/services/audiopolicy/engineconfigurable/config/example/automotive/audio_policy_engine_product_strategies.xml
@@ -84,7 +84,7 @@
</AttributesGroup>
</ProductStrategy>
<ProductStrategy name="voice_command">
- <AttributesGroup volumeGroup="speech">
+ <AttributesGroup volumeGroup="speech" streamType="AUDIO_STREAM_ASSISTANT">
<Attributes>
<ContentType value="AUDIO_CONTENT_TYPE_SPEECH"/>
<Usage value="AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE"/>
@@ -147,10 +147,6 @@
<ProductStrategy name="notification">
<AttributesGroup streamType="AUDIO_STREAM_NOTIFICATION" volumeGroup="ring">
<Attributes> <Usage value="AUDIO_USAGE_NOTIFICATION"/> </Attributes>
- <Attributes> <Usage value="AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT"/> </Attributes>
- <Attributes> <Usage value="AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED"/> </Attributes>
- <Attributes> <Usage value="AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST"/> </Attributes>
- <Attributes> <Usage value="AUDIO_USAGE_NOTIFICATION_EVENT"/> </Attributes>
</AttributesGroup>
</ProductStrategy>
<ProductStrategy name="system">
@@ -167,19 +163,5 @@
</AttributesGroup>
</ProductStrategy>
- <!-- Routing Strategy rerouting may be removed as following media??? -->
- <ProductStrategy name="rerouting">
- <AttributesGroup streamType="AUDIO_STREAM_REROUTING" volumeGroup="rerouting">
- <Attributes></Attributes>
- </AttributesGroup>
- </ProductStrategy>
-
- <!-- Patch stream needs full scale volume, define it otherwise switch to default... -->
- <ProductStrategy name="patch">
- <AttributesGroup streamType="AUDIO_STREAM_PATCH" volumeGroup="patch">
- <Attributes></Attributes>
- </AttributesGroup>
- </ProductStrategy>
-
</ProductStrategies>
diff --git a/services/audiopolicy/engineconfigurable/config/example/automotive/audio_policy_engine_volumes.xml b/services/audiopolicy/engineconfigurable/config/example/automotive/audio_policy_engine_volumes.xml
index 6e72dc5..97a25a8 100644
--- a/services/audiopolicy/engineconfigurable/config/example/automotive/audio_policy_engine_volumes.xml
+++ b/services/audiopolicy/engineconfigurable/config/example/automotive/audio_policy_engine_volumes.xml
@@ -189,25 +189,5 @@
</volume>
</volumeGroup>
- <volumeGroup>
- <name>rerouting</name>
- <indexMin>0</indexMin>
- <indexMax>1</indexMax>
- <volume deviceCategory="DEVICE_CATEGORY_HEADSET">
- <point>0,0</point>
- <point>100,0</point>
- </volume>
- </volumeGroup>
-
- <volumeGroup>
- <name>patch</name>
- <indexMin>0</indexMin>
- <indexMax>1</indexMax>
- <volume deviceCategory="DEVICE_CATEGORY_HEADSET">
- <point>0,0</point>
- <point>100,0</point>
- </volume>
- </volumeGroup>
-
</volumeGroups>
diff --git a/services/audiopolicy/engineconfigurable/config/example/caremu/Android.bp b/services/audiopolicy/engineconfigurable/config/example/caremu/Android.bp
new file mode 100644
index 0000000..fae6b7b
--- /dev/null
+++ b/services/audiopolicy/engineconfigurable/config/example/caremu/Android.bp
@@ -0,0 +1,82 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+// Import this namespace in order to use AOSP Car Emulator configuration example
+
+soong_namespace {
+ imports: [
+ "frameworks/av/services/audiopolicy/engineconfigurable/config/example/automotive",
+ "frameworks/av/services/audiopolicy/config",
+ ],
+}
+
+prebuilt_etc {
+ name: "audio_policy_engine_configuration.xml",
+ vendor: true,
+ src: ":audio_policy_engine_configuration",
+ required: [
+ "audio_policy_engine_criterion_types.xml",
+ "audio_policy_engine_criteria.xml",
+ "audio_policy_engine_product_strategies.xml",
+ ":audio_policy_engine_volumes.xml",
+ ],
+}
+prebuilt_etc {
+ name: "audio_policy_engine_product_strategies.xml",
+ vendor: true,
+ src: "audio_policy_engine_product_strategies.xml",
+}
+prebuilt_etc {
+ name: "audio_policy_engine_criterion_types.xml",
+ vendor: true,
+ src: ":audio_policy_engine_criterion_types",
+}
+
+//
+// Generate audio_policy_engine criterion type file => provides device addresses criterion type
+//
+genrule {
+ name: "audio_policy_engine_criterion_types",
+ defaults: ["buildpolicycriteriontypesrule"],
+ srcs: [
+ ":audio_policy_configuration_top_file",
+ ":audio_policy_configuration_files",
+ ],
+}
+filegroup {
+ name: "audio_policy_configuration_files",
+ srcs: [
+ ":r_submix_audio_policy_configuration",
+ ":default_volume_tables",
+ ":audio_policy_volumes",
+ ":surround_sound_configuration_5_0",
+ ":primary_audio_policy_configuration",
+ ],
+}
+filegroup {
+ name : "audio_policy_configuration_top_file",
+ srcs: [":audio_policy_configuration_generic"],
+}
+filegroup {
+ name: "audio_policy_engine_configuration_files",
+ srcs: [
+ ":audio_policy_engine_configuration",
+ "audio_policy_engine_product_strategies.xml",
+ ":audio_policy_engine_volumes",
+ ":audio_policy_engine_criterion_types",
+ ":audio_policy_engine_criteria",
+ ],
+}
diff --git a/services/audiopolicy/engineconfigurable/config/example/caremu/audio_policy_engine_product_strategies.xml b/services/audiopolicy/engineconfigurable/config/example/caremu/audio_policy_engine_product_strategies.xml
index adcbd83..f598cf2 100644
--- a/services/audiopolicy/engineconfigurable/config/example/caremu/audio_policy_engine_product_strategies.xml
+++ b/services/audiopolicy/engineconfigurable/config/example/caremu/audio_policy_engine_product_strategies.xml
@@ -84,7 +84,7 @@
</AttributesGroup>
</ProductStrategy>
<ProductStrategy name="voice_command">
- <AttributesGroup volumeGroup="speech">
+ <AttributesGroup volumeGroup="speech" streamType="AUDIO_STREAM_ASSISTANT">
<Attributes>
<ContentType value="AUDIO_CONTENT_TYPE_SPEECH"/>
<Usage value="AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE"/>
@@ -147,10 +147,6 @@
<ProductStrategy name="notification">
<AttributesGroup streamType="AUDIO_STREAM_NOTIFICATION" volumeGroup="ring">
<Attributes> <Usage value="AUDIO_USAGE_NOTIFICATION"/> </Attributes>
- <Attributes> <Usage value="AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT"/> </Attributes>
- <Attributes> <Usage value="AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED"/> </Attributes>
- <Attributes> <Usage value="AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST"/> </Attributes>
- <Attributes> <Usage value="AUDIO_USAGE_NOTIFICATION_EVENT"/> </Attributes>
</AttributesGroup>
</ProductStrategy>
<ProductStrategy name="system">
@@ -167,18 +163,5 @@
</AttributesGroup>
</ProductStrategy>
- <!-- Routing Strategy rerouting may be removed as following media??? -->
- <ProductStrategy name="rerouting">
- <AttributesGroup streamType="AUDIO_STREAM_REROUTING" volumeGroup="rerouting">
- <Attributes></Attributes>
- </AttributesGroup>
- </ProductStrategy>
-
- <!-- Patch stream needs full scale volume, define it otherwise switch to default... -->
- <ProductStrategy name="patch">
- <AttributesGroup streamType="AUDIO_STREAM_PATCH" volumeGroup="patch">
- <Attributes></Attributes>
- </AttributesGroup>
- </ProductStrategy>
</ProductStrategies>
diff --git a/services/audiopolicy/engineconfigurable/config/example/caremu/audio_policy_engine_volumes.xml b/services/audiopolicy/engineconfigurable/config/example/caremu/audio_policy_engine_volumes.xml
index 6e72dc5..97a25a8 100644
--- a/services/audiopolicy/engineconfigurable/config/example/caremu/audio_policy_engine_volumes.xml
+++ b/services/audiopolicy/engineconfigurable/config/example/caremu/audio_policy_engine_volumes.xml
@@ -189,25 +189,5 @@
</volume>
</volumeGroup>
- <volumeGroup>
- <name>rerouting</name>
- <indexMin>0</indexMin>
- <indexMax>1</indexMax>
- <volume deviceCategory="DEVICE_CATEGORY_HEADSET">
- <point>0,0</point>
- <point>100,0</point>
- </volume>
- </volumeGroup>
-
- <volumeGroup>
- <name>patch</name>
- <indexMin>0</indexMin>
- <indexMax>1</indexMax>
- <volume deviceCategory="DEVICE_CATEGORY_HEADSET">
- <point>0,0</point>
- <point>100,0</point>
- </volume>
- </volumeGroup>
-
</volumeGroups>
diff --git a/services/audiopolicy/engineconfigurable/config/example/common/audio_policy_engine_criterion_types.xml.in b/services/audiopolicy/engineconfigurable/config/example/common/audio_policy_engine_criterion_types.xml.in
index fe17369..e134c42 100644
--- a/services/audiopolicy/engineconfigurable/config/example/common/audio_policy_engine_criterion_types.xml.in
+++ b/services/audiopolicy/engineconfigurable/config/example/common/audio_policy_engine_criterion_types.xml.in
@@ -22,7 +22,12 @@
<value literal="0" numerical="1"/>
</values>
</criterion_type>
- <criterion_type name="InputDevicesAddressesType" type="inclusive"/>
+ <criterion_type name="InputDevicesAddressesType" type="inclusive">
+ <values>
+ <!-- legacy remote submix -->
+ <value literal="0" numerical="1"/>
+ </values>
+ </criterion_type>
<criterion_type name="AndroidModeType" type="exclusive"/>
<criterion_type name="BooleanType" type="exclusive">
<values>
diff --git a/services/audiopolicy/engineconfigurable/config/example/phone/Android.bp b/services/audiopolicy/engineconfigurable/config/example/phone/Android.bp
new file mode 100644
index 0000000..94d33bd
--- /dev/null
+++ b/services/audiopolicy/engineconfigurable/config/example/phone/Android.bp
@@ -0,0 +1,104 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+// Import this namespace in order to use AOSP Phone configuration example
+
+soong_namespace {
+ imports: [
+ "frameworks/av/services/audiopolicy/config",
+ ],
+}
+
+prebuilt_etc {
+ name: "audio_policy_engine_configuration.xml",
+ vendor: true,
+ src: ":audio_policy_engine_configuration",
+ required: [
+ ":audio_policy_engine_criterion_types.xml",
+ ":audio_policy_engine_criteria.xml",
+ ":audio_policy_engine_product_strategies.xml",
+ ":audio_policy_engine_volumes.xml",
+ ],
+}
+prebuilt_etc {
+ name: "audio_policy_engine_product_strategies.xml",
+ vendor: true,
+ src: "audio_policy_engine_product_strategies.xml",
+}
+prebuilt_etc {
+ name: "audio_policy_engine_stream_volumes.xml",
+ vendor: true,
+ src: ":audio_policy_engine_stream_volumes",
+}
+prebuilt_etc {
+ name: "audio_policy_engine_default_stream_volumes.xml",
+ vendor: true,
+ src: ":audio_policy_engine_default_stream_volumes",
+}
+prebuilt_etc {
+ name: "audio_policy_engine_criterion_types.xml",
+ vendor: true,
+ src: ":audio_policy_engine_criterion_types",
+}
+
+//
+// Generate audio_policy_engine criterion type file => provides device addresses criterion type
+//
+genrule {
+ name: "audio_policy_engine_criterion_types",
+ defaults: ["buildpolicycriteriontypesrule"],
+ srcs: [
+ ":audio_policy_configuration_top_file",
+ ":audio_policy_configuration_files",
+ ],
+}
+filegroup {
+ name: "audio_policy_configuration_files",
+ srcs: [
+ ":r_submix_audio_policy_configuration",
+ ":default_volume_tables",
+ ":audio_policy_volumes",
+ ":surround_sound_configuration_5_0",
+ ":primary_audio_policy_configuration",
+ ],
+}
+filegroup {
+ name : "audio_policy_configuration_top_file",
+ srcs: [":audio_policy_configuration_generic"],
+}
+filegroup {
+ name: "audio_policy_engine_configuration",
+ srcs: ["audio_policy_engine_configuration.xml"],
+}
+filegroup {
+ name: "audio_policy_engine_stream_volumes",
+ srcs: ["audio_policy_engine_stream_volumes.xml"],
+}
+filegroup {
+ name: "audio_policy_engine_default_stream_volumes",
+ srcs: ["audio_policy_engine_default_stream_volumes.xml"],
+}
+filegroup {
+ name: "audio_policy_engine_configuration_files",
+ srcs: [
+ ":audio_policy_engine_configuration",
+ "audio_policy_engine_product_strategies.xml",
+ ":audio_policy_engine_stream_volumes",
+ ":audio_policy_engine_default_stream_volumes",
+ ":audio_policy_engine_criterion_types",
+ ":audio_policy_engine_criteria",
+ ],
+}
diff --git a/services/audiopolicy/engineconfigurable/config/example/phone/audio_policy_engine_product_strategies.xml b/services/audiopolicy/engineconfigurable/config/example/phone/audio_policy_engine_product_strategies.xml
index 9398743..a7388da 100644
--- a/services/audiopolicy/engineconfigurable/config/example/phone/audio_policy_engine_product_strategies.xml
+++ b/services/audiopolicy/engineconfigurable/config/example/phone/audio_policy_engine_product_strategies.xml
@@ -72,6 +72,12 @@
<Attributes> <Usage value="AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE"/> </Attributes>
<Attributes></Attributes>
</AttributesGroup>
+ <AttributesGroup streamType="AUDIO_STREAM_ASSISTANT" volumeGroup="assistant">
+ <Attributes>
+ <ContentType value="AUDIO_CONTENT_TYPE_SPEECH"/>
+ <Usage value="AUDIO_USAGE_ASSISTANT"/>
+ </Attributes>
+ </AttributesGroup>
<AttributesGroup streamType="AUDIO_STREAM_SYSTEM" volumeGroup="system">
<Attributes> <Usage value="AUDIO_USAGE_ASSISTANCE_SONIFICATION"/> </Attributes>
</AttributesGroup>
@@ -91,20 +97,5 @@
</AttributesGroup>
</ProductStrategy>
- <!-- Routing Strategy rerouting may be removed as following media??? -->
- <ProductStrategy name="STRATEGY_REROUTING">
- <AttributesGroup streamType="AUDIO_STREAM_REROUTING" volumeGroup="rerouting">
- <Attributes></Attributes>
- </AttributesGroup>
- </ProductStrategy>
-
- <!-- Default product strategy has empty attributes -->
- <ProductStrategy name="STRATEGY_PATCH">
- <AttributesGroup streamType="AUDIO_STREAM_PATCH" volumeGroup="patch">
- <Attributes></Attributes>
- </AttributesGroup>
- </ProductStrategy>
-
-
</ProductStrategies>
diff --git a/services/audiopolicy/engineconfigurable/config/example/phone/audio_policy_engine_stream_volumes.xml b/services/audiopolicy/engineconfigurable/config/example/phone/audio_policy_engine_stream_volumes.xml
index 707a184..8aa71ca 100644
--- a/services/audiopolicy/engineconfigurable/config/example/phone/audio_policy_engine_stream_volumes.xml
+++ b/services/audiopolicy/engineconfigurable/config/example/phone/audio_policy_engine_stream_volumes.xml
@@ -205,27 +205,16 @@
<volume deviceCategory="DEVICE_CATEGORY_EXT_MEDIA" ref="DEFAULT_NON_MUTABLE_VOLUME_CURVE"/>
<volume deviceCategory="DEVICE_CATEGORY_HEARING_AID" ref="DEFAULT_NON_MUTABLE_HEARING_AID_VOLUME_CURVE"/>
</volumeGroup>
-
<volumeGroup>
- <name>rerouting</name>
+ <name>assistant</name>
<indexMin>0</indexMin>
- <indexMax>1</indexMax>
- <volume deviceCategory="DEVICE_CATEGORY_HEADSET" ref="FULL_SCALE_VOLUME_CURVE"/>
- <volume deviceCategory="DEVICE_CATEGORY_SPEAKER" ref="FULL_SCALE_VOLUME_CURVE"/>
- <volume deviceCategory="DEVICE_CATEGORY_EARPIECE" ref="FULL_SCALE_VOLUME_CURVE"/>
- <volume deviceCategory="DEVICE_CATEGORY_EXT_MEDIA" ref="FULL_SCALE_VOLUME_CURVE"/>
- <volume deviceCategory="DEVICE_CATEGORY_HEARING_AID" ref="FULL_SCALE_VOLUME_CURVE"/>
+ <indexMax>15</indexMax>
+ <volume deviceCategory="DEVICE_CATEGORY_HEADSET" ref="DEFAULT_MEDIA_VOLUME_CURVE"/>
+ <volume deviceCategory="DEVICE_CATEGORY_SPEAKER" ref="DEFAULT_DEVICE_CATEGORY_SPEAKER_VOLUME_CURVE"/>
+ <volume deviceCategory="DEVICE_CATEGORY_EARPIECE" ref="DEFAULT_MEDIA_VOLUME_CURVE"/>
+ <volume deviceCategory="DEVICE_CATEGORY_EXT_MEDIA" ref="DEFAULT_MEDIA_VOLUME_CURVE"/>
+ <volume deviceCategory="DEVICE_CATEGORY_HEARING_AID" ref="DEFAULT_HEARING_AID_VOLUME_CURVE"/>
</volumeGroup>
- <volumeGroup>
- <name>patch</name>
- <indexMin>0</indexMin>
- <indexMax>1</indexMax>
- <volume deviceCategory="DEVICE_CATEGORY_HEADSET" ref="FULL_SCALE_VOLUME_CURVE"/>
- <volume deviceCategory="DEVICE_CATEGORY_SPEAKER" ref="FULL_SCALE_VOLUME_CURVE"/>
- <volume deviceCategory="DEVICE_CATEGORY_EARPIECE" ref="FULL_SCALE_VOLUME_CURVE"/>
- <volume deviceCategory="DEVICE_CATEGORY_EXT_MEDIA" ref="FULL_SCALE_VOLUME_CURVE"/>
- <volume deviceCategory="DEVICE_CATEGORY_HEARING_AID" ref="FULL_SCALE_VOLUME_CURVE"/>
- </volumeGroup>
</volumeGroups>
diff --git a/services/audiopolicy/engineconfigurable/include/AudioPolicyEngineInstance.h b/services/audiopolicy/engineconfigurable/include/AudioPolicyEngineInstance.h
index efc69da..f52de21 100644
--- a/services/audiopolicy/engineconfigurable/include/AudioPolicyEngineInstance.h
+++ b/services/audiopolicy/engineconfigurable/include/AudioPolicyEngineInstance.h
@@ -16,7 +16,7 @@
#pragma once
-class AudioPolicyManagerInterface;
+class EngineInterface;
class AudioPolicyPluginInterface;
namespace android {
@@ -69,7 +69,7 @@
* Compile time error will claim if invalid interface is requested.
*/
template <>
-AudioPolicyManagerInterface *EngineInstance::queryInterface() const;
+EngineInterface *EngineInstance::queryInterface() const;
template <>
AudioPolicyPluginInterface *EngineInstance::queryInterface() const;
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/Android.bp b/services/audiopolicy/engineconfigurable/parameter-framework/Android.bp
new file mode 100644
index 0000000..90ebffd
--- /dev/null
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/Android.bp
@@ -0,0 +1,77 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+// Root soong_namespace for common components
+
+prebuilt_etc {
+ name: "PolicyClass.xml",
+ vendor: true,
+ src: ":PolicyClass",
+ sub_dir: "parameter-framework/Structure/Policy",
+}
+prebuilt_etc {
+ name: "PolicySubsystem.xml",
+ vendor: true,
+ src: ":PolicySubsystem",
+ sub_dir: "parameter-framework/Structure/Policy",
+}
+prebuilt_etc {
+ name: "PolicySubsystem-CommonTypes.xml",
+ vendor: true,
+ src: ":buildcommontypesstructure_gen",
+ sub_dir: "parameter-framework/Structure/Policy",
+}
+genrule {
+ name: "buildcommontypesstructure_gen",
+ defaults: ["buildcommontypesstructurerule"],
+}
+
+filegroup {
+ name: "product_strategies_structure_template",
+ srcs: ["examples/common/Structure/ProductStrategies.xml.in"],
+}
+filegroup {
+ name: "PolicySubsystem",
+ srcs: ["examples/common/Structure/PolicySubsystem.xml"],
+}
+filegroup {
+ name: "PolicySubsystem-no-strategy",
+ srcs: ["examples/common/Structure/PolicySubsystem-no-strategy.xml"],
+}
+filegroup {
+ name: "common_types_structure_template",
+ srcs: ["examples/common/Structure/PolicySubsystem-CommonTypes.xml.in"],
+}
+filegroup {
+ name: "PolicyClass",
+ srcs: ["examples/common/Structure/PolicyClass.xml"],
+}
+filegroup {
+ name: "volumes.pfw",
+ srcs: ["examples/Settings/volumes.pfw"],
+}
+filegroup {
+ name: "device_for_input_source.pfw",
+ srcs: ["examples/Settings/device_for_input_source.pfw"],
+}
+filegroup {
+ name: "ParameterFrameworkConfigurationPolicy.userdebug.xml",
+ srcs: ["examples/ParameterFrameworkConfigurationPolicy.userdebug.xml"],
+}
+filegroup {
+ name: "ParameterFrameworkConfigurationPolicy.user.xml",
+ srcs: ["examples/ParameterFrameworkConfigurationPolicy.user.xml"],
+}
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Android.mk b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Android.mk
deleted file mode 100644
index 19f93b3..0000000
--- a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Android.mk
+++ /dev/null
@@ -1,187 +0,0 @@
-################################################################################################
-#
-# @NOTE:
-# Audio Policy Engine configurable example for generic device build
-#
-# Any vendor shall have its own configuration within the corresponding device folder
-#
-################################################################################################
-
-LOCAL_PATH := $(call my-dir)
-
-ifdef BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION
-
-ifeq ($(BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION),$(filter $(BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION),phone_configurable automotive_configurable caremu_configurable no-output_configurable no-input_configurable))
-
-PFW_CORE := external/parameter-framework
-#@TODO: upstream new domain generator
-#BUILD_PFW_SETTINGS := $(PFW_CORE)/support/android/build_pfw_settings.mk
-PFW_DEFAULT_SCHEMAS_DIR := $(PFW_CORE)/upstream/schemas
-PFW_SCHEMAS_DIR := $(PFW_DEFAULT_SCHEMAS_DIR)
-
-TOOLS := frameworks/av/services/audiopolicy/engineconfigurable/tools
-BUILD_PFW_SETTINGS := $(TOOLS)/build_audio_pfw_settings.mk
-
-PROVISION_STRATEGIES_STRUCTURE := $(TOOLS)/provision_strategies_structure.mk
-
-endif
-
-##################################################################
-# CONFIGURATION FILES
-##################################################################
-######### Policy PFW top level file #########
-
-ifeq ($(BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION),$(filter $(BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION),phone_configurable automotive_configurable caremu_configurable))
-
-include $(CLEAR_VARS)
-LOCAL_MODULE := ParameterFrameworkConfigurationPolicy.xml
-LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE_CLASS := ETC
-LOCAL_VENDOR_MODULE := true
-LOCAL_MODULE_RELATIVE_PATH := parameter-framework
-LOCAL_SRC_FILES := $(LOCAL_MODULE).in
-LOCAL_REQUIRED_MODULES := \
- PolicySubsystem.xml \
- PolicyClass.xml
-
-# external/parameter-framework prevents from using debug interface
-AUDIO_PATTERN = @TUNING_ALLOWED@
-ifeq ($(TARGET_BUILD_VARIANT),user)
-AUDIO_VALUE = false
-else
-AUDIO_VALUE = true
-endif
-
-LOCAL_POST_INSTALL_CMD := $(hide) sed -i -e 's|$(AUDIO_PATTERN)|$(AUDIO_VALUE)|g' $(TARGET_OUT_VENDOR_ETC)/$(LOCAL_MODULE_RELATIVE_PATH)/$(LOCAL_MODULE)
-
-include $(BUILD_PREBUILT)
-
-########## Policy PFW Common Structures #########
-
-include $(CLEAR_VARS)
-LOCAL_MODULE := PolicySubsystem.xml
-LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE_CLASS := ETC
-LOCAL_VENDOR_MODULE := true
-LOCAL_REQUIRED_MODULES := \
- PolicySubsystem-CommonTypes.xml \
- ProductStrategies.xml
-
-LOCAL_MODULE_RELATIVE_PATH := parameter-framework/Structure/Policy
-LOCAL_SRC_FILES := common/Structure/$(LOCAL_MODULE)
-include $(BUILD_PREBUILT)
-
-include $(CLEAR_VARS)
-LOCAL_MODULE := PolicySubsystem-CommonTypes.xml
-LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE_CLASS := ETC
-LOCAL_VENDOR_MODULE := true
-LOCAL_MODULE_RELATIVE_PATH := parameter-framework/Structure/Policy
-LOCAL_SRC_FILES := common/Structure/$(LOCAL_MODULE)
-include $(BUILD_PREBUILT)
-
-include $(CLEAR_VARS)
-LOCAL_MODULE := PolicyClass.xml
-LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE_CLASS := ETC
-LOCAL_VENDOR_MODULE := true
-LOCAL_MODULE_RELATIVE_PATH := parameter-framework/Structure/Policy
-LOCAL_SRC_FILES := common/Structure/$(LOCAL_MODULE)
-include $(BUILD_PREBUILT)
-
-
-include $(CLEAR_VARS)
-LOCAL_MODULE := ProductStrategies.xml
-LOCAL_MODULE_CLASS := ETC
-LOCAL_VENDOR_MODULE := true
-LOCAL_MODULE_RELATIVE_PATH := parameter-framework/Structure/Policy
-
-AUDIO_POLICY_ENGINE_CONFIGURATION_FILE := \
- $(TARGET_OUT_VENDOR_ETC)/audio_policy_engine_configuration.xml
-STRATEGIES_STRUCTURE_FILE := $(LOCAL_PATH)/common/Structure/$(LOCAL_MODULE).in
-
-include $(PROVISION_STRATEGIES_STRUCTURE)
-
-endif #ifeq ($(BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION),$(filter $(BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION),phone_configurable automotive_configurable caremu_configurable))
-
-########## Policy PFW Example Structures #########
-ifeq ($(BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION),$(filter $(BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION),no-output_configurable no-input_configurable))
-
-include $(CLEAR_VARS)
-LOCAL_MODULE := PolicySubsystem.xml
-LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE_CLASS := ETC
-LOCAL_VENDOR_MODULE := true
-LOCAL_REQUIRED_MODULES := PolicySubsystem-CommonTypes.xml
-
-LOCAL_MODULE_RELATIVE_PATH := parameter-framework/Structure/Policy
-LOCAL_SRC_FILES := common/Structure/$(LOCAL_MODULE)
-include $(BUILD_PREBUILT)
-
-include $(CLEAR_VARS)
-LOCAL_MODULE := ParameterFrameworkConfigurationPolicy-no-strategy.xml
-LOCAL_MODULE_STEM := ParameterFrameworkConfigurationPolicy.xml
-LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE_CLASS := ETC
-LOCAL_VENDOR_MODULE := true
-LOCAL_MODULE_RELATIVE_PATH := parameter-framework
-LOCAL_SRC_FILES := $(LOCAL_MODULE).in
-LOCAL_REQUIRED_MODULES := \
- PolicySubsystem.xml \
- PolicyClass.xml
-AUDIO_VALUE = false
-LOCAL_POST_INSTALL_CMD := $(hide) sed -i -e 's|$(AUDIO_PATTERN)|$(AUDIO_VALUE)|g' $(TARGET_OUT_VENDOR_ETC)/$(LOCAL_MODULE_RELATIVE_PATH)/$(LOCAL_MODULE)
-
-include $(BUILD_PREBUILT)
-
-endif # ifeq ($(BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION),$(filter $(BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION),no-output_configurable no-input_configurable))
-
-######### Policy PFW Settings - No Output #########
-ifeq ($(BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION),no-output_configurable)
-
-include $(CLEAR_VARS)
-LOCAL_MODULE := parameter-framework.policy
-LOCAL_MODULE_STEM := PolicyConfigurableDomains-NoOutputDevice.xml
-LOCAL_MODULE_CLASS := ETC
-LOCAL_VENDOR_MODULE := true
-LOCAL_MODULE_RELATIVE_PATH := parameter-framework/Settings/Policy
-
-PFW_TOPLEVEL_FILE := $(TARGET_OUT_VENDOR_ETC)/parameter-framework/ParameterFrameworkConfigurationPolicy.xml
-PFW_CRITERION_TYPES_FILE := $(TARGET_OUT_VENDOR_ETC)/audio_policy_engine_criterion_types.xml
-PFW_CRITERIA_FILE := $(TARGET_OUT_VENDOR_ETC)/audio_policy_engine_criteria.xml
-PFW_EDD_FILES := \
- $(LOCAL_PATH)/SettingsNoOutput/device_for_strategies.pfw \
- $(LOCAL_PATH)/Settings/device_for_input_source.pfw \
- $(LOCAL_PATH)/Settings/volumes.pfw
-LOCAL_REQUIRED_MODULES := libpolicy-subsystem
-include $(BUILD_PFW_SETTINGS)
-
-endif # ifeq ($(BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION),no-output_configurable)
-######### Policy PFW Settings - No Input #########
-ifeq ($(BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION),no-input_configurable)
-
-include $(CLEAR_VARS)
-LOCAL_MODULE := parameter-framework.policy
-LOCAL_MODULE_STEM := PolicyConfigurableDomains-NoInputDevice.xml
-LOCAL_MODULE_CLASS := ETC
-LOCAL_VENDOR_MODULE := true
-LOCAL_MODULE_RELATIVE_PATH := parameter-framework/Settings/Policy
-
-PFW_TOPLEVEL_FILE := $(TARGET_OUT_VENDOR_ETC)/parameter-framework/ParameterFrameworkConfigurationPolicy.xml
-PFW_CRITERION_TYPES_FILE := $(TARGET_OUT_VENDOR_ETC)/audio_policy_engine_criterion_types.xml
-PFW_CRITERIA_FILE := $(TARGET_OUT_VENDOR_ETC)/audio_policy_engine_criteria.xml
-PFW_EDD_FILES := \
- $(LOCAL_PATH)/SettingsNoInput/device_for_input_source.pfw \
- $(LOCAL_PATH)/Settings/volumes.pfw
-LOCAL_REQUIRED_MODULES := libpolicy-subsystem
-include $(BUILD_PFW_SETTINGS)
-
-endif #ifeq ($(BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION),no-input_configurable)
-#######################################################################
-# Recursive call sub-folder Android.mk
-#######################################################################
-
-include $(call all-makefiles-under,$(LOCAL_PATH))
-
-endif #ifdef BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION
-
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Car/Android.bp b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Car/Android.bp
new file mode 100644
index 0000000..82b1b6d
--- /dev/null
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Car/Android.bp
@@ -0,0 +1,91 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+// Import this namespace in order to use AOSP Automotive configuration example
+
+soong_namespace {
+ imports: [
+ "frameworks/av/services/audiopolicy/engineconfigurable/config/example/automotive",
+ "frameworks/av/services/audiopolicy/config",
+ ],
+}
+
+//
+// Generate Audio Policy Parameter Framework Product Strategies Structure file from template
+//
+prebuilt_etc {
+ name: "ProductStrategies.xml",
+ vendor: true,
+ src: ":buildstrategiesstructure_gen",
+ sub_dir: "parameter-framework/Structure/Policy",
+ required: ["libpolicy-subsystem"],
+}
+genrule {
+ name: "buildstrategiesstructure_gen",
+ defaults: ["buildstrategiesstructurerule"],
+ srcs: [
+ ":audio_policy_engine_configuration_files",
+ ],
+}
+
+//
+// Generate Audio Policy Parameter Framework Configurable Domains
+//
+prebuilt_etc {
+ name: "parameter-framework.policy",
+ filename_from_src: true,
+ vendor: true,
+ src: ":domaingeneratorpolicyrule_gen",
+ sub_dir: "parameter-framework/Settings/Policy",
+ required: [
+ "ProductStrategies.xml",
+ "PolicyClass.xml",
+ "PolicySubsystem.xml",
+ "PolicySubsystem-CommonTypes.xml",
+ ],
+}
+genrule {
+ name: "domaingeneratorpolicyrule_gen",
+ defaults: ["domaingeneratorpolicyrule"],
+ srcs: [
+ ":audio_policy_pfw_toplevel",
+ ":audio_policy_pfw_structure_files",
+ ":audio_policy_engine_criterion_types",
+ ":edd_files",
+ ],
+}
+filegroup {
+ name: "edd_files",
+ srcs: [
+ ":device_for_input_source.pfw",
+ ":volumes.pfw",
+ "Settings/device_for_product_strategies.pfw",
+ ],
+}
+// This is for Settings generation, must use socket port, so userdebug version is required
+filegroup {
+ name: "audio_policy_pfw_toplevel",
+ srcs: [":ParameterFrameworkConfigurationPolicy.userdebug.xml"],
+}
+filegroup {
+ name: "audio_policy_pfw_structure_files",
+ srcs: [
+ ":PolicyClass",
+ ":PolicySubsystem",
+ ":buildcommontypesstructure_gen",
+ ":buildstrategiesstructure_gen",
+ ],
+}
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Car/Android.mk b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Car/Android.mk
deleted file mode 100644
index 7304ec2..0000000
--- a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Car/Android.mk
+++ /dev/null
@@ -1,47 +0,0 @@
-################################################################################################
-#
-# @NOTE:
-# Audio Policy Engine configurable example for generic device build
-#
-# Any vendor shall have its own configuration within the corresponding device folder
-#
-################################################################################################
-
-ifeq ($(BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION), automotive_configurable)
-LOCAL_PATH := $(call my-dir)
-
-PFW_CORE := external/parameter-framework
-PFW_DEFAULT_SCHEMAS_DIR := $(PFW_CORE)/upstream/schemas
-PFW_SCHEMAS_DIR := $(PFW_DEFAULT_SCHEMAS_DIR)
-
-TOOLS := frameworks/av/services/audiopolicy/engineconfigurable/tools
-BUILD_PFW_SETTINGS := $(TOOLS)/build_audio_pfw_settings.mk
-
-
-##################################################################
-# CONFIGURATION FILES
-##################################################################
-
-########## Policy PFW Structures #########
-######### Policy PFW Settings #########
-include $(CLEAR_VARS)
-LOCAL_MODULE := parameter-framework.policy
-LOCAL_MODULE_STEM := PolicyConfigurableDomains.xml
-LOCAL_MODULE_CLASS := ETC
-LOCAL_VENDOR_MODULE := true
-LOCAL_MODULE_RELATIVE_PATH := parameter-framework/Settings/Policy
-LOCAL_REQUIRED_MODULES := libpolicy-subsystem
-
-PFW_EDD_FILES := \
- $(LOCAL_PATH)/Settings/device_for_product_strategies.pfw \
- $(LOCAL_PATH)/../Settings/device_for_input_source.pfw \
- $(LOCAL_PATH)/../Settings/volumes.pfw
-
-PFW_CRITERION_TYPES_FILE := $(TARGET_OUT_VENDOR_ETC)/audio_policy_engine_criterion_types.xml
-PFW_CRITERIA_FILE := $(TARGET_OUT_VENDOR_ETC)/audio_policy_engine_criteria.xml
-PFW_TOPLEVEL_FILE := $(TARGET_OUT_VENDOR_ETC)/parameter-framework/ParameterFrameworkConfigurationPolicy.xml
-PFW_SCHEMAS_DIR := $(PFW_DEFAULT_SCHEMAS_DIR)
-
-include $(BUILD_PFW_SETTINGS)
-
-endif #ifeq ($(BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION), automotive_configurable)
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Car/Settings/device_for_product_strategies.pfw b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Car/Settings/device_for_product_strategies.pfw
index 57ad592..ddae356 100644
--- a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Car/Settings/device_for_product_strategies.pfw
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Car/Settings/device_for_product_strategies.pfw
@@ -14,7 +14,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -59,7 +59,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -106,7 +106,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -152,7 +152,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -205,7 +205,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -251,7 +251,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -304,7 +304,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -357,7 +357,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -411,7 +411,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -464,7 +464,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -517,7 +517,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -570,7 +570,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -623,7 +623,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -676,7 +676,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -729,7 +729,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/CarEmu/Android.bp b/services/audiopolicy/engineconfigurable/parameter-framework/examples/CarEmu/Android.bp
new file mode 100644
index 0000000..e4605b2
--- /dev/null
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/examples/CarEmu/Android.bp
@@ -0,0 +1,92 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+// Import this namespace in order to use AOSP Car Emulator configuration example
+
+soong_namespace {
+ imports: [
+ "frameworks/av/services/audiopolicy/engineconfigurable/config/example/caremu",
+ "frameworks/av/services/audiopolicy/engineconfigurable/parameter-framework/examples/Car",
+ "frameworks/av/services/audiopolicy/config",
+ ],
+}
+
+//
+// Generate Audio Policy Parameter Framework Product Strategies Structure file from template
+//
+prebuilt_etc {
+ name: "ProductStrategies.xml",
+ vendor: true,
+ src: ":buildstrategiesstructure_gen",
+ sub_dir: "parameter-framework/Structure/Policy",
+ required: ["libpolicy-subsystem"],
+}
+genrule {
+ name: "buildstrategiesstructure_gen",
+ defaults: ["buildstrategiesstructurerule"],
+ srcs: [
+ ":audio_policy_engine_configuration_files",
+ ],
+}
+
+//
+// Generate Audio Policy Parameter Framework Configurable Domains
+//
+prebuilt_etc {
+ name: "parameter-framework.policy",
+ filename_from_src: true,
+ vendor: true,
+ src: ":domaingeneratorpolicyrule_gen",
+ sub_dir: "parameter-framework/Settings/Policy",
+ required: [
+ "ProductStrategies.xml",
+ "PolicyClass.xml",
+ "PolicySubsystem.xml",
+ "PolicySubsystem-CommonTypes.xml",
+ ],
+}
+genrule {
+ name: "domaingeneratorpolicyrule_gen",
+ defaults: ["domaingeneratorpolicyrule"],
+ srcs: [
+ ":audio_policy_pfw_toplevel",
+ ":audio_policy_pfw_structure_files",
+ ":audio_policy_engine_criterion_types",
+ ":edd_files",
+ ],
+}
+filegroup {
+ name: "edd_files",
+ srcs: [
+ ":device_for_input_source.pfw",
+ ":volumes.pfw",
+ "Settings/device_for_product_strategies.pfw",
+ ],
+}
+// This is for Settings generation, must use socket port, so userdebug version is required
+filegroup {
+ name: "audio_policy_pfw_toplevel",
+ srcs: [":ParameterFrameworkConfigurationPolicy.userdebug.xml"],
+}
+filegroup {
+ name: "audio_policy_pfw_structure_files",
+ srcs: [
+ ":PolicyClass",
+ ":PolicySubsystem",
+ ":buildcommontypesstructure_gen",
+ ":buildstrategiesstructure_gen",
+ ],
+}
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/CarEmu/Android.mk b/services/audiopolicy/engineconfigurable/parameter-framework/examples/CarEmu/Android.mk
deleted file mode 100644
index f5eb7d1..0000000
--- a/services/audiopolicy/engineconfigurable/parameter-framework/examples/CarEmu/Android.mk
+++ /dev/null
@@ -1,46 +0,0 @@
-################################################################################################
-#
-# @NOTE:
-# Audio Policy Engine configurable example for generic device build
-#
-# Any vendor shall have its own configuration within the corresponding device folder
-#
-################################################################################################
-
-ifeq ($(BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION), caremu_configurable)
-LOCAL_PATH := $(call my-dir)
-
-PFW_CORE := external/parameter-framework
-PFW_DEFAULT_SCHEMAS_DIR := $(PFW_CORE)/upstream/schemas
-PFW_SCHEMAS_DIR := $(PFW_DEFAULT_SCHEMAS_DIR)
-
-TOOLS := frameworks/av/services/audiopolicy/engineconfigurable/tools
-BUILD_PFW_SETTINGS := $(TOOLS)/build_audio_pfw_settings.mk
-
-
-##################################################################
-# CONFIGURATION FILES
-##################################################################
-
-########## Policy PFW Structures #########
-######### Policy PFW Settings #########
-include $(CLEAR_VARS)
-LOCAL_MODULE := parameter-framework.policy
-LOCAL_MODULE_STEM := PolicyConfigurableDomains.xml
-LOCAL_MODULE_CLASS := ETC
-LOCAL_VENDOR_MODULE := true
-LOCAL_MODULE_RELATIVE_PATH := parameter-framework/Settings/Policy
-LOCAL_REQUIRED_MODULES := libpolicy-subsystem
-
-PFW_EDD_FILES := \
- $(LOCAL_PATH)/Settings/device_for_product_strategies.pfw \
- $(LOCAL_PATH)/../Settings/device_for_input_source.pfw \
- $(LOCAL_PATH)/../Settings/volumes.pfw
-PFW_CRITERION_TYPES_FILE := $(TARGET_OUT_VENDOR_ETC)/audio_policy_engine_criterion_types.xml
-PFW_CRITERIA_FILE := $(TARGET_OUT_VENDOR_ETC)/audio_policy_engine_criteria.xml
-PFW_TOPLEVEL_FILE := $(TARGET_OUT_VENDOR_ETC)/parameter-framework/ParameterFrameworkConfigurationPolicy.xml
-PFW_SCHEMAS_DIR := $(PFW_DEFAULT_SCHEMAS_DIR)
-
-include $(BUILD_PFW_SETTINGS)
-
-endif #ifeq ($(BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION), caremu_configurable)
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/CarEmu/Settings/device_for_product_strategies.pfw b/services/audiopolicy/engineconfigurable/parameter-framework/examples/CarEmu/Settings/device_for_product_strategies.pfw
index ca3464f..cc778df 100644
--- a/services/audiopolicy/engineconfigurable/parameter-framework/examples/CarEmu/Settings/device_for_product_strategies.pfw
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/examples/CarEmu/Settings/device_for_product_strategies.pfw
@@ -14,7 +14,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -59,7 +59,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -106,7 +106,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -153,7 +153,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -198,7 +198,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -245,7 +245,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -291,7 +291,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -337,7 +337,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -384,7 +384,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -430,7 +430,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -476,7 +476,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -522,7 +522,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -568,7 +568,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -614,7 +614,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -659,7 +659,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/ParameterFrameworkConfigurationPolicy.xml.in b/services/audiopolicy/engineconfigurable/parameter-framework/examples/ParameterFrameworkConfigurationPolicy.user.xml
similarity index 81%
copy from services/audiopolicy/engineconfigurable/parameter-framework/examples/ParameterFrameworkConfigurationPolicy.xml.in
copy to services/audiopolicy/engineconfigurable/parameter-framework/examples/ParameterFrameworkConfigurationPolicy.user.xml
index 1be67dd..c5960cb 100644
--- a/services/audiopolicy/engineconfigurable/parameter-framework/examples/ParameterFrameworkConfigurationPolicy.xml.in
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/examples/ParameterFrameworkConfigurationPolicy.user.xml
@@ -1,7 +1,6 @@
<?xml version="1.0" encoding="UTF-8"?>
<ParameterFrameworkConfiguration xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance"
- SystemClassName="Policy" ServerPort="unix:///dev/socket/audioserver/policy_debug"
- TuningAllowed="@TUNING_ALLOWED@">
+ SystemClassName="Policy" TuningAllowed="false">
<SubsystemPlugins>
<Location Folder="">
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/ParameterFrameworkConfigurationPolicy.xml.in b/services/audiopolicy/engineconfigurable/parameter-framework/examples/ParameterFrameworkConfigurationPolicy.userdebug.xml
similarity index 93%
rename from services/audiopolicy/engineconfigurable/parameter-framework/examples/ParameterFrameworkConfigurationPolicy.xml.in
rename to services/audiopolicy/engineconfigurable/parameter-framework/examples/ParameterFrameworkConfigurationPolicy.userdebug.xml
index 1be67dd..1b7d7d8 100644
--- a/services/audiopolicy/engineconfigurable/parameter-framework/examples/ParameterFrameworkConfigurationPolicy.xml.in
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/examples/ParameterFrameworkConfigurationPolicy.userdebug.xml
@@ -1,7 +1,7 @@
<?xml version="1.0" encoding="UTF-8"?>
<ParameterFrameworkConfiguration xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance"
SystemClassName="Policy" ServerPort="unix:///dev/socket/audioserver/policy_debug"
- TuningAllowed="@TUNING_ALLOWED@">
+ TuningAllowed="true">
<SubsystemPlugins>
<Location Folder="">
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Android.bp b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Android.bp
new file mode 100644
index 0000000..61b54cf
--- /dev/null
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Android.bp
@@ -0,0 +1,100 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+// Import this namespace in order to use AOSP Phone configuration example
+
+soong_namespace {
+ imports: [
+ "frameworks/av/services/audiopolicy/engineconfigurable/config/example/phone",
+ "frameworks/av/services/audiopolicy/config",
+ ],
+}
+
+//
+// Generate Audio Policy Parameter Framework Product Strategies Structure file from template
+//
+prebuilt_etc {
+ name: "ProductStrategies.xml",
+ vendor: true,
+ src: ":buildstrategiesstructure_gen",
+ sub_dir: "parameter-framework/Structure/Policy",
+ required: ["libpolicy-subsystem"],
+}
+genrule {
+ name: "buildstrategiesstructure_gen",
+ defaults: ["buildstrategiesstructurerule"],
+ srcs: [
+ ":audio_policy_engine_configuration_files",
+ ],
+}
+
+//
+// Generate Audio Policy Parameter Framework Configurable Domains
+//
+prebuilt_etc {
+ name: "parameter-framework.policy",
+ filename_from_src: true,
+ vendor: true,
+ src: ":domaingeneratorpolicyrule_gen",
+ sub_dir: "parameter-framework/Settings/Policy",
+ required: [
+ "ProductStrategies.xml",
+ "PolicyClass.xml",
+ "PolicySubsystem.xml",
+ "PolicySubsystem-CommonTypes.xml",
+ ],
+}
+genrule {
+ name: "domaingeneratorpolicyrule_gen",
+ defaults: ["domaingeneratorpolicyrule"],
+ srcs: [
+ ":audio_policy_pfw_toplevel",
+ ":audio_policy_pfw_structure_files",
+ ":audio_policy_engine_criterion_types",
+ ":edd_files",
+ ],
+}
+filegroup {
+ name: "edd_files",
+ srcs: [
+ ":device_for_input_source.pfw",
+ ":volumes.pfw",
+ "Settings/device_for_product_strategy_media.pfw",
+ "Settings/device_for_product_strategy_accessibility.pfw",
+ "Settings/device_for_product_strategy_dtmf.pfw",
+ "Settings/device_for_product_strategy_enforced_audible.pfw",
+ "Settings/device_for_product_strategy_phone.pfw",
+ "Settings/device_for_product_strategy_sonification.pfw",
+ "Settings/device_for_product_strategy_sonification_respectful.pfw",
+ "Settings/device_for_product_strategy_transmitted_through_speaker.pfw",
+ "Settings/device_for_product_strategy_rerouting.pfw",
+ "Settings/device_for_product_strategy_patch.pfw",
+ ],
+}
+// This is for Settings generation, must use socket port, so userdebug version is required
+filegroup {
+ name: "audio_policy_pfw_toplevel",
+ srcs: [":ParameterFrameworkConfigurationPolicy.userdebug.xml"],
+}
+filegroup {
+ name: "audio_policy_pfw_structure_files",
+ srcs: [
+ ":PolicyClass",
+ ":PolicySubsystem",
+ ":buildcommontypesstructure_gen",
+ ":buildstrategiesstructure_gen",
+ ],
+}
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Android.mk b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Android.mk
deleted file mode 100644
index 0b20781..0000000
--- a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Android.mk
+++ /dev/null
@@ -1,54 +0,0 @@
-################################################################################################
-#
-# @NOTE:
-# Audio Policy Engine configurable example for generic device build
-#
-# Any vendor shall have its own configuration within the corresponding device folder
-#
-################################################################################################
-
-ifeq ($(BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION), phone_configurable)
-
-LOCAL_PATH := $(call my-dir)
-
-PFW_CORE := external/parameter-framework
-PFW_DEFAULT_SCHEMAS_DIR := $(PFW_CORE)/upstream/schemas
-PFW_SCHEMAS_DIR := $(PFW_DEFAULT_SCHEMAS_DIR)
-
-TOOLS := frameworks/av/services/audiopolicy/engineconfigurable/tools
-BUILD_PFW_SETTINGS := $(TOOLS)/build_audio_pfw_settings.mk
-
-##################################################################
-# CONFIGURATION FILES
-##################################################################
-########## Policy PFW Structures #########
-######### Policy PFW Settings #########
-include $(CLEAR_VARS)
-LOCAL_MODULE := parameter-framework.policy
-LOCAL_MODULE_STEM := PolicyConfigurableDomains.xml
-LOCAL_MODULE_CLASS := ETC
-LOCAL_VENDOR_MODULE := true
-LOCAL_MODULE_RELATIVE_PATH := parameter-framework/Settings/Policy
-LOCAL_REQUIRED_MODULES := libpolicy-subsystem
-
-PFW_EDD_FILES := \
- $(LOCAL_PATH)/../Settings/device_for_input_source.pfw \
- $(LOCAL_PATH)/../Settings/volumes.pfw \
- $(LOCAL_PATH)/Settings/device_for_product_strategy_media.pfw \
- $(LOCAL_PATH)/Settings/device_for_product_strategy_accessibility.pfw \
- $(LOCAL_PATH)/Settings/device_for_product_strategy_dtmf.pfw \
- $(LOCAL_PATH)/Settings/device_for_product_strategy_enforced_audible.pfw \
- $(LOCAL_PATH)/Settings/device_for_product_strategy_phone.pfw \
- $(LOCAL_PATH)/Settings/device_for_product_strategy_sonification.pfw \
- $(LOCAL_PATH)/Settings/device_for_product_strategy_sonification_respectful.pfw \
- $(LOCAL_PATH)/Settings/device_for_product_strategy_transmitted_through_speaker.pfw \
- $(LOCAL_PATH)/Settings/device_for_product_strategy_rerouting.pfw \
- $(LOCAL_PATH)/Settings/device_for_product_strategy_patch.pfw
-PFW_CRITERION_TYPES_FILE := $(TARGET_OUT_VENDOR_ETC)/audio_policy_engine_criterion_types.xml
-PFW_CRITERIA_FILE := $(TARGET_OUT_VENDOR_ETC)/audio_policy_engine_criteria.xml
-PFW_TOPLEVEL_FILE := $(TARGET_OUT_VENDOR_ETC)/parameter-framework/ParameterFrameworkConfigurationPolicy.xml
-PFW_SCHEMAS_DIR := $(PFW_DEFAULT_SCHEMAS_DIR)
-
-include $(BUILD_PFW_SETTINGS)
-
-endif #ifeq ($(BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION), phone_configurable)
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_accessibility.pfw b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_accessibility.pfw
index 53e93de..d16a904 100644
--- a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_accessibility.pfw
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_accessibility.pfw
@@ -45,7 +45,7 @@
wired_headset = 0
wired_headphone = 0
line = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -73,7 +73,7 @@
wired_headset = 0
wired_headphone = 0
line = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -101,7 +101,7 @@
wired_headset = 0
wired_headphone = 0
line = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -129,7 +129,7 @@
wired_headset = 0
wired_headphone = 0
line = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -157,7 +157,7 @@
wired_headset = 0
wired_headphone = 0
line = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -186,7 +186,7 @@
wired_headset = 0
wired_headphone = 0
line = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -215,7 +215,7 @@
wired_headset = 0
wired_headphone = 0
line = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -244,7 +244,7 @@
wired_headset = 0
wired_headphone = 0
line = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -281,7 +281,7 @@
wired_headset = 0
wired_headphone = 1
line = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -317,7 +317,7 @@
wired_headset = 0
wired_headphone = 0
line = 1
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -354,7 +354,7 @@
wired_headset = 1
wired_headphone = 0
line = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -394,7 +394,7 @@
wired_headset = 0
wired_headphone = 0
line = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 1
@@ -425,7 +425,7 @@
wired_headset = 0
wired_headphone = 0
line = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 1
usb_device = 0
@@ -455,7 +455,7 @@
wired_headset = 0
wired_headphone = 0
line = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 1
usb_accessory = 0
usb_device = 0
@@ -485,7 +485,7 @@
wired_headset = 0
wired_headphone = 0
line = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -517,7 +517,7 @@
wired_headset = 0
wired_headphone = 0
line = 0
- angl_dock_headset = 1
+ anlg_dock_headset = 1
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -546,7 +546,7 @@
wired_headset = 0
wired_headphone = 0
line = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -568,7 +568,7 @@
wired_headset = 0
wired_headphone = 0
line = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -588,7 +588,7 @@
wired_headset = 0
wired_headphone = 0
line = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_dtmf.pfw b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_dtmf.pfw
index b8426c6..414445d 100644
--- a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_dtmf.pfw
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_dtmf.pfw
@@ -34,7 +34,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -62,7 +62,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -90,7 +90,7 @@
bluetooth_a2dp_headphones = 1
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -118,7 +118,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 1
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -147,7 +147,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -176,7 +176,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -205,7 +205,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -242,7 +242,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -281,7 +281,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -318,7 +318,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -358,7 +358,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 1
@@ -389,7 +389,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 1
usb_device = 0
@@ -419,7 +419,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 1
usb_accessory = 0
usb_device = 0
@@ -449,7 +449,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 1
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -481,7 +481,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 1
+ anlg_dock_headset = 1
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -510,7 +510,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -548,7 +548,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -568,7 +568,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_enforced_audible.pfw b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_enforced_audible.pfw
index 2daa9ac..36b8f3c 100644
--- a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_enforced_audible.pfw
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_enforced_audible.pfw
@@ -77,7 +77,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -100,7 +100,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -123,7 +123,7 @@
bluetooth_a2dp_headphones = 1
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -146,7 +146,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 1
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -169,7 +169,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -192,7 +192,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -215,7 +215,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -238,7 +238,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 1
usb_device = 0
@@ -261,7 +261,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 1
@@ -284,7 +284,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 1
usb_accessory = 0
usb_device = 0
@@ -307,7 +307,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 1
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -331,7 +331,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 1
+ anlg_dock_headset = 1
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -351,7 +351,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_media.pfw b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_media.pfw
index d6d355c..6210a57 100644
--- a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_media.pfw
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_media.pfw
@@ -26,7 +26,7 @@
speaker = 0
hdmi = 0
dgtl_dock_headset = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
usb_device = 0
usb_accessory = 0
wired_headset = 0
@@ -46,7 +46,7 @@
speaker = 0
hdmi = 0
dgtl_dock_headset = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
usb_device = 0
usb_accessory = 0
wired_headset = 0
@@ -66,7 +66,7 @@
speaker = 0
hdmi = 0
dgtl_dock_headset = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
usb_device = 0
usb_accessory = 0
wired_headset = 0
@@ -86,7 +86,7 @@
speaker = 0
hdmi = 0
dgtl_dock_headset = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
usb_device = 0
usb_accessory = 0
wired_headset = 0
@@ -109,7 +109,7 @@
speaker = 1
hdmi = 0
dgtl_dock_headset = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
usb_device = 0
usb_accessory = 0
wired_headset = 0
@@ -127,7 +127,7 @@
speaker = 0
hdmi = 0
dgtl_dock_headset = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
usb_device = 0
usb_accessory = 0
wired_headset = 0
@@ -145,7 +145,7 @@
speaker = 0
hdmi = 0
dgtl_dock_headset = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
usb_device = 0
usb_accessory = 0
wired_headset = 0
@@ -163,7 +163,7 @@
speaker = 0
hdmi = 0
dgtl_dock_headset = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
usb_device = 0
usb_accessory = 0
wired_headset = 1
@@ -181,7 +181,7 @@
speaker = 0
hdmi = 0
dgtl_dock_headset = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
usb_device = 0
usb_accessory = 1
wired_headset = 0
@@ -199,7 +199,7 @@
speaker = 0
hdmi = 0
dgtl_dock_headset = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
usb_device = 1
usb_accessory = 0
wired_headset = 0
@@ -217,7 +217,7 @@
speaker = 0
hdmi = 0
dgtl_dock_headset = 1
- angl_dock_headset = 0
+ anlg_dock_headset = 0
usb_device = 0
usb_accessory = 0
wired_headset = 0
@@ -235,7 +235,7 @@
speaker = 0
hdmi = 1
dgtl_dock_headset = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
usb_device = 0
usb_accessory = 0
wired_headset = 0
@@ -254,7 +254,7 @@
speaker = 0
hdmi = 0
dgtl_dock_headset = 0
- angl_dock_headset = 1
+ anlg_dock_headset = 1
usb_device = 0
usb_accessory = 0
wired_headset = 0
@@ -277,7 +277,7 @@
speaker = 1
hdmi = 0
dgtl_dock_headset = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
usb_device = 0
usb_accessory = 0
wired_headset = 0
@@ -293,7 +293,7 @@
speaker = 0
hdmi = 0
dgtl_dock_headset = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
usb_device = 0
usb_accessory = 0
wired_headset = 0
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_patch.pfw b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_patch.pfw
index d2cc090..feeeec6 100644
--- a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_patch.pfw
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_patch.pfw
@@ -14,7 +14,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_phone.pfw b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_phone.pfw
index 5693d4e..da2fc9b 100644
--- a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_phone.pfw
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_phone.pfw
@@ -32,7 +32,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -55,7 +55,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -78,7 +78,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -108,7 +108,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -138,7 +138,7 @@
bluetooth_a2dp_headphones = 1
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -168,7 +168,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 1
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -195,7 +195,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -222,7 +222,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -245,7 +245,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -283,7 +283,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 1
@@ -311,7 +311,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 1
@@ -339,7 +339,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 1
usb_accessory = 0
usb_device = 0
@@ -367,7 +367,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 1
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -395,7 +395,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 1
+ anlg_dock_headset = 1
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -422,7 +422,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -449,7 +449,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -472,7 +472,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_rerouting.pfw b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_rerouting.pfw
index 10f8814..3275cdf 100644
--- a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_rerouting.pfw
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_rerouting.pfw
@@ -14,7 +14,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_sonification.pfw b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_sonification.pfw
index c4edeeb..a60445b 100644
--- a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_sonification.pfw
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_sonification.pfw
@@ -69,7 +69,7 @@
bluetooth_a2dp = 1
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -95,7 +95,7 @@
bluetooth_a2dp = 0
bluetooth_a2dp_headphones = 1
bluetooth_a2dp_speaker = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -121,7 +121,7 @@
bluetooth_a2dp = 0
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 1
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -148,7 +148,7 @@
bluetooth_a2dp = 0
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -175,7 +175,7 @@
bluetooth_a2dp = 0
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -202,7 +202,7 @@
bluetooth_a2dp = 0
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -238,7 +238,7 @@
bluetooth_a2dp = 0
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -276,7 +276,7 @@
bluetooth_a2dp = 0
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -312,7 +312,7 @@
bluetooth_a2dp = 0
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -349,7 +349,7 @@
bluetooth_a2dp = 0
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 1
@@ -378,7 +378,7 @@
bluetooth_a2dp = 0
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 1
usb_device = 0
@@ -407,7 +407,7 @@
bluetooth_a2dp = 0
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 1
usb_accessory = 0
usb_device = 0
@@ -437,7 +437,7 @@
bluetooth_a2dp = 0
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
- angl_dock_headset = 1
+ anlg_dock_headset = 1
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -464,7 +464,7 @@
bluetooth_a2dp = 0
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -482,7 +482,7 @@
bluetooth_a2dp = 0
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_sonification_respectful.pfw b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_sonification_respectful.pfw
index 0a3dd5f..6b11e23 100644
--- a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_sonification_respectful.pfw
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_sonification_respectful.pfw
@@ -92,7 +92,7 @@
wired_headset = 0
wired_headphone = 0
line = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -119,7 +119,7 @@
wired_headset = 0
wired_headphone = 0
line = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -146,7 +146,7 @@
wired_headset = 0
wired_headphone = 0
line = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -173,7 +173,7 @@
wired_headset = 0
wired_headphone = 0
line = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -200,7 +200,7 @@
wired_headset = 0
wired_headphone = 0
line = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -227,7 +227,7 @@
wired_headset = 0
wired_headphone = 0
line = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -264,7 +264,7 @@
wired_headset = 0
wired_headphone = 1
line = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -305,7 +305,7 @@
wired_headset = 0
wired_headphone = 0
line = 1
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -342,7 +342,7 @@
wired_headset = 1
wired_headphone = 0
line = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -380,7 +380,7 @@
wired_headset = 0
wired_headphone = 0
line = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 1
@@ -410,7 +410,7 @@
wired_headset = 0
wired_headphone = 0
line = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 1
usb_device = 0
@@ -440,7 +440,7 @@
wired_headset = 0
wired_headphone = 0
line = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 1
usb_accessory = 0
usb_device = 0
@@ -470,7 +470,7 @@
wired_headset = 0
wired_headphone = 0
line = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -501,7 +501,7 @@
wired_headset = 0
wired_headphone = 0
line = 0
- angl_dock_headset = 1
+ anlg_dock_headset = 1
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -528,7 +528,7 @@
wired_headset = 0
wired_headphone = 0
line = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_transmitted_through_speaker.pfw b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_transmitted_through_speaker.pfw
index 3fc7670..418f3cc 100644
--- a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_transmitted_through_speaker.pfw
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Phone/Settings/device_for_product_strategy_transmitted_through_speaker.pfw
@@ -19,7 +19,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/PolicyConfigurableDomains.xml b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/PolicyConfigurableDomains.xml
index 0710441..baffa81 100644
--- a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/PolicyConfigurableDomains.xml
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/PolicyConfigurableDomains.xml
@@ -145,7 +145,7 @@
<ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/speaker"/>
<ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/hdmi"/>
<ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/dgtl_dock_headset"/>
- <ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/angl_dock_headset"/>
+ <ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/anlg_dock_headset"/>
<ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/usb_device"/>
<ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/usb_accessory"/>
<ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/wired_headset"/>
@@ -167,8 +167,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/usb_device">
<BitParameter Name="usb_device">0</BitParameter>
@@ -208,8 +208,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/usb_device">
<BitParameter Name="usb_device">0</BitParameter>
@@ -249,8 +249,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/usb_device">
<BitParameter Name="usb_device">0</BitParameter>
@@ -290,8 +290,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/usb_device">
<BitParameter Name="usb_device">0</BitParameter>
@@ -331,8 +331,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/usb_device">
<BitParameter Name="usb_device">0</BitParameter>
@@ -372,8 +372,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/usb_device">
<BitParameter Name="usb_device">0</BitParameter>
@@ -413,8 +413,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/usb_device">
<BitParameter Name="usb_device">0</BitParameter>
@@ -454,8 +454,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/usb_device">
<BitParameter Name="usb_device">0</BitParameter>
@@ -495,8 +495,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/usb_device">
<BitParameter Name="usb_device">0</BitParameter>
@@ -536,8 +536,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/usb_device">
<BitParameter Name="usb_device">1</BitParameter>
@@ -577,8 +577,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">1</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/usb_device">
<BitParameter Name="usb_device">0</BitParameter>
@@ -618,8 +618,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/usb_device">
<BitParameter Name="usb_device">0</BitParameter>
@@ -659,8 +659,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">1</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">1</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/usb_device">
<BitParameter Name="usb_device">0</BitParameter>
@@ -700,8 +700,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/usb_device">
<BitParameter Name="usb_device">0</BitParameter>
@@ -741,8 +741,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/media/selected_output_devices/mask/usb_device">
<BitParameter Name="usb_device">0</BitParameter>
@@ -1039,7 +1039,7 @@
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/bluetooth_a2dp_headphones"/>
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/bluetooth_a2dp_speaker"/>
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/hdmi"/>
- <ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/angl_dock_headset"/>
+ <ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/anlg_dock_headset"/>
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/dgtl_dock_headset"/>
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/usb_accessory"/>
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/usb_device"/>
@@ -1079,8 +1079,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -1132,8 +1132,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -1185,8 +1185,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -1238,8 +1238,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -1291,8 +1291,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -1344,8 +1344,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -1397,8 +1397,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -1450,8 +1450,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -1503,8 +1503,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -1556,8 +1556,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -1609,8 +1609,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -1662,8 +1662,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">1</BitParameter>
@@ -1715,8 +1715,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">1</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -1768,8 +1768,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">1</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">1</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -1821,8 +1821,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -1874,8 +1874,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -1927,8 +1927,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/phone/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -2228,7 +2228,7 @@
<ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/bluetooth_a2dp"/>
<ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/bluetooth_a2dp_headphones"/>
<ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/bluetooth_a2dp_speaker"/>
- <ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/angl_dock_headset"/>
+ <ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/anlg_dock_headset"/>
<ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/dgtl_dock_headset"/>
<ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/usb_accessory"/>
<ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/usb_device"/>
@@ -2264,8 +2264,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/bluetooth_a2dp_speaker">
<BitParameter Name="bluetooth_a2dp_speaker">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -2311,8 +2311,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/bluetooth_a2dp_speaker">
<BitParameter Name="bluetooth_a2dp_speaker">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -2358,8 +2358,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/bluetooth_a2dp_speaker">
<BitParameter Name="bluetooth_a2dp_speaker">1</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -2405,8 +2405,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/bluetooth_a2dp_speaker">
<BitParameter Name="bluetooth_a2dp_speaker">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -2452,8 +2452,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/bluetooth_a2dp_speaker">
<BitParameter Name="bluetooth_a2dp_speaker">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -2499,8 +2499,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/bluetooth_a2dp_speaker">
<BitParameter Name="bluetooth_a2dp_speaker">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -2546,8 +2546,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/bluetooth_a2dp_speaker">
<BitParameter Name="bluetooth_a2dp_speaker">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -2593,8 +2593,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/bluetooth_a2dp_speaker">
<BitParameter Name="bluetooth_a2dp_speaker">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -2640,8 +2640,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/bluetooth_a2dp_speaker">
<BitParameter Name="bluetooth_a2dp_speaker">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -2687,8 +2687,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/bluetooth_a2dp_speaker">
<BitParameter Name="bluetooth_a2dp_speaker">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -2734,8 +2734,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/bluetooth_a2dp_speaker">
<BitParameter Name="bluetooth_a2dp_speaker">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -2781,8 +2781,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/bluetooth_a2dp_speaker">
<BitParameter Name="bluetooth_a2dp_speaker">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">1</BitParameter>
@@ -2828,8 +2828,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/bluetooth_a2dp_speaker">
<BitParameter Name="bluetooth_a2dp_speaker">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">1</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">1</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -2875,8 +2875,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/bluetooth_a2dp_speaker">
<BitParameter Name="bluetooth_a2dp_speaker">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -2922,8 +2922,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/bluetooth_a2dp_speaker">
<BitParameter Name="bluetooth_a2dp_speaker">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/sonification/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -3246,7 +3246,7 @@
<ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/wired_headset"/>
<ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/wired_headphone"/>
<ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/line"/>
- <ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/angl_dock_headset"/>
+ <ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/anlg_dock_headset"/>
<ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/dgtl_dock_headset"/>
<ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/usb_accessory"/>
<ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/usb_device"/>
@@ -3284,8 +3284,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -3331,8 +3331,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -3378,8 +3378,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -3425,8 +3425,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -3472,8 +3472,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -3519,8 +3519,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -3566,8 +3566,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -3613,8 +3613,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/line">
<BitParameter Name="line">1</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -3660,8 +3660,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -3707,8 +3707,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -3754,8 +3754,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -3801,8 +3801,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">1</BitParameter>
@@ -3848,8 +3848,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -3895,8 +3895,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">1</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">1</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -3942,8 +3942,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/sonification_respectful/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -4207,7 +4207,7 @@
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/bluetooth_a2dp_headphones"/>
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/bluetooth_a2dp_speaker"/>
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/hdmi"/>
- <ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/angl_dock_headset"/>
+ <ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/anlg_dock_headset"/>
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/dgtl_dock_headset"/>
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/usb_accessory"/>
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/usb_device"/>
@@ -4247,8 +4247,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -4300,8 +4300,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -4353,8 +4353,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -4406,8 +4406,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -4459,8 +4459,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -4512,8 +4512,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -4565,8 +4565,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -4618,8 +4618,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -4671,8 +4671,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -4724,8 +4724,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -4777,8 +4777,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -4830,8 +4830,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -4883,8 +4883,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">1</BitParameter>
@@ -4936,8 +4936,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">1</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -4989,8 +4989,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">1</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">1</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -5042,8 +5042,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -5095,8 +5095,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -5148,8 +5148,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/dtmf/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -5448,7 +5448,7 @@
<ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/bluetooth_a2dp_headphones"/>
<ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/bluetooth_a2dp_speaker"/>
<ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/hdmi"/>
- <ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/angl_dock_headset"/>
+ <ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/anlg_dock_headset"/>
<ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/dgtl_dock_headset"/>
<ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/usb_accessory"/>
<ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/usb_device"/>
@@ -5490,8 +5490,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -5543,8 +5543,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -5596,8 +5596,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -5649,8 +5649,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -5702,8 +5702,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -5755,8 +5755,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -5808,8 +5808,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -5861,8 +5861,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -5914,8 +5914,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -5967,8 +5967,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">1</BitParameter>
@@ -6020,8 +6020,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">1</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -6073,8 +6073,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">1</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">1</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -6126,8 +6126,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/enforced_audible/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -6170,7 +6170,7 @@
<ConfigurableElement Path="/Policy/policy/strategies/transmitted_through_speaker/selected_output_devices/mask/bluetooth_a2dp_headphones"/>
<ConfigurableElement Path="/Policy/policy/strategies/transmitted_through_speaker/selected_output_devices/mask/bluetooth_a2dp_speaker"/>
<ConfigurableElement Path="/Policy/policy/strategies/transmitted_through_speaker/selected_output_devices/mask/hdmi"/>
- <ConfigurableElement Path="/Policy/policy/strategies/transmitted_through_speaker/selected_output_devices/mask/angl_dock_headset"/>
+ <ConfigurableElement Path="/Policy/policy/strategies/transmitted_through_speaker/selected_output_devices/mask/anlg_dock_headset"/>
<ConfigurableElement Path="/Policy/policy/strategies/transmitted_through_speaker/selected_output_devices/mask/dgtl_dock_headset"/>
<ConfigurableElement Path="/Policy/policy/strategies/transmitted_through_speaker/selected_output_devices/mask/usb_accessory"/>
<ConfigurableElement Path="/Policy/policy/strategies/transmitted_through_speaker/selected_output_devices/mask/usb_device"/>
@@ -6230,8 +6230,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/transmitted_through_speaker/selected_output_devices/mask/hdmi">
<BitParameter Name="hdmi">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/transmitted_through_speaker/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/transmitted_through_speaker/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/transmitted_through_speaker/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -6539,7 +6539,7 @@
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/wired_headset"/>
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/wired_headphone"/>
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/line"/>
- <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/angl_dock_headset"/>
+ <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/anlg_dock_headset"/>
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/dgtl_dock_headset"/>
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/usb_accessory"/>
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/usb_device"/>
@@ -6583,8 +6583,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -6636,8 +6636,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -6689,8 +6689,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -6742,8 +6742,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -6795,8 +6795,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -6848,8 +6848,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -6901,8 +6901,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -6954,8 +6954,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -7007,8 +7007,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -7060,8 +7060,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/line">
<BitParameter Name="line">1</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -7113,8 +7113,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -7166,8 +7166,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -7219,8 +7219,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -7272,8 +7272,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">1</BitParameter>
@@ -7325,8 +7325,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -7378,8 +7378,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">1</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">1</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -7431,8 +7431,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -7484,8 +7484,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -7537,8 +7537,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/accessibility/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -7710,7 +7710,7 @@
<ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/wired_headset"/>
<ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/wired_headphone"/>
<ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/line"/>
- <ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/angl_dock_headset"/>
+ <ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/anlg_dock_headset"/>
<ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/dgtl_dock_headset"/>
<ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/usb_accessory"/>
<ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/usb_device"/>
@@ -7742,8 +7742,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -7783,8 +7783,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -7824,8 +7824,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -7865,8 +7865,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -7906,8 +7906,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -7947,8 +7947,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -7988,8 +7988,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/line">
<BitParameter Name="line">1</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -8029,8 +8029,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -8070,8 +8070,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -8111,8 +8111,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -8152,8 +8152,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">1</BitParameter>
@@ -8193,8 +8193,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -8234,8 +8234,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">1</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">1</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -8275,8 +8275,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
@@ -8316,8 +8316,8 @@
<ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/line">
<BitParameter Name="line">0</BitParameter>
</ConfigurableElement>
- <ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/angl_dock_headset">
- <BitParameter Name="angl_dock_headset">0</BitParameter>
+ <ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/anlg_dock_headset">
+ <BitParameter Name="anlg_dock_headset">0</BitParameter>
</ConfigurableElement>
<ConfigurableElement Path="/Policy/policy/strategies/rerouting/selected_output_devices/mask/dgtl_dock_headset">
<BitParameter Name="dgtl_dock_headset">0</BitParameter>
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/device_for_input_source.pfw b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/device_for_input_source.pfw
index a990879..9e0957c 100644
--- a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/device_for_input_source.pfw
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/device_for_input_source.pfw
@@ -18,7 +18,6 @@
loopback = 0
ip = 0
bus = 0
- stub = 0
component: /Policy/policy/input_sources/mic/applicable_input_device/mask
communication = 0
ambient = 0
@@ -36,7 +35,6 @@
loopback = 0
ip = 0
bus = 0
- stub = 0
component: /Policy/policy/input_sources/voice_downlink/applicable_input_device/mask
communication = 0
ambient = 0
@@ -58,7 +56,6 @@
loopback = 0
ip = 0
bus = 0
- stub = 0
component: /Policy/policy/input_sources/voice_call/applicable_input_device/mask
communication = 0
ambient = 0
@@ -80,7 +77,6 @@
loopback = 0
ip = 0
bus = 0
- stub = 0
component: /Policy/policy/input_sources/voice_uplink/applicable_input_device/mask
communication = 0
ambient = 0
@@ -102,7 +98,6 @@
loopback = 0
ip = 0
bus = 0
- stub = 0
component: /Policy/policy/input_sources/camcorder/applicable_input_device/mask
communication = 0
ambient = 0
@@ -123,7 +118,6 @@
loopback = 0
ip = 0
bus = 0
- stub = 0
component: /Policy/policy/input_sources/voice_recognition/applicable_input_device/mask
communication = 0
ambient = 0
@@ -142,7 +136,6 @@
loopback = 0
ip = 0
bus = 0
- stub = 0
component: /Policy/policy/input_sources/voice_communication/applicable_input_device/mask
communication = 0
ambient = 0
@@ -160,7 +153,6 @@
loopback = 0
ip = 0
bus = 0
- stub = 0
component: /Policy/policy/input_sources/remote_submix/applicable_input_device/mask
communication = 0
ambient = 0
@@ -182,7 +174,6 @@
loopback = 0
ip = 0
bus = 0
- stub = 0
component: /Policy/policy/input_sources/hotword/applicable_input_device/mask
communication = 0
ambient = 0
@@ -201,7 +192,6 @@
loopback = 0
ip = 0
bus = 0
- stub = 0
component: /Policy/policy/input_sources/unprocessed/applicable_input_device/mask
communication = 0
ambient = 0
@@ -220,7 +210,6 @@
loopback = 0
ip = 0
bus = 0
- stub = 0
component: /Policy/policy/input_sources/fm_tuner/applicable_input_device/mask
communication = 0
ambient = 0
@@ -242,7 +231,6 @@
loopback = 0
ip = 0
bus = 0
- stub = 0
domain: DefaultAndMic
conf: A2dp
@@ -255,12 +243,14 @@
usb_device = 0
builtin_mic = 0
bluetooth_sco_headset = 0
+ stub = 0
component: mic/applicable_input_device/mask/
bluetooth_a2dp = 1
wired_headset = 0
usb_device = 0
builtin_mic = 0
bluetooth_sco_headset = 0
+ stub = 0
conf: Sco
AvailableInputDevices Includes BluetoothScoHeadset
@@ -273,12 +263,14 @@
usb_device = 0
builtin_mic = 0
bluetooth_sco_headset = 1
+ stub = 0
component: mic/applicable_input_device/mask/
bluetooth_a2dp = 0
wired_headset = 0
usb_device = 0
builtin_mic = 0
bluetooth_sco_headset = 1
+ stub = 0
conf: WiredHeadset
AvailableInputDevices Includes WiredHeadset
@@ -290,12 +282,14 @@
usb_device = 0
builtin_mic = 0
bluetooth_sco_headset = 0
+ stub = 0
component: mic/applicable_input_device/mask/
bluetooth_a2dp = 0
wired_headset = 1
usb_device = 0
builtin_mic = 0
bluetooth_sco_headset = 0
+ stub = 0
conf: UsbDevice
AvailableInputDevices Includes UsbDevice
@@ -307,12 +301,14 @@
usb_device = 1
builtin_mic = 0
bluetooth_sco_headset = 0
+ stub = 0
component: mic/applicable_input_device/mask/
bluetooth_a2dp = 0
wired_headset = 0
usb_device = 1
builtin_mic = 0
bluetooth_sco_headset = 0
+ stub = 0
conf: BuiltinMic
AvailableInputDevices Includes BuiltinMic
@@ -324,12 +320,33 @@
usb_device = 0
builtin_mic = 1
bluetooth_sco_headset = 0
+ stub = 0
component: mic/applicable_input_device/mask/
bluetooth_a2dp = 0
wired_headset = 0
usb_device = 0
builtin_mic = 1
bluetooth_sco_headset = 0
+ stub = 0
+
+ conf: Stub
+ AvailableInputDevices Includes Default
+
+ component: /Policy/policy/input_sources
+ component: default/applicable_input_device/mask/
+ bluetooth_a2dp = 0
+ wired_headset = 0
+ usb_device = 0
+ builtin_mic = 0
+ bluetooth_sco_headset = 0
+ stub = 1
+ component: mic/applicable_input_device/mask/
+ bluetooth_a2dp = 0
+ wired_headset = 0
+ usb_device = 0
+ builtin_mic = 0
+ bluetooth_sco_headset = 0
+ stub = 1
conf: Default
component: /Policy/policy/input_sources
@@ -339,12 +356,14 @@
usb_device = 0
builtin_mic = 0
bluetooth_sco_headset = 0
+ stub = 0
component: mic/applicable_input_device/mask/
bluetooth_a2dp = 0
wired_headset = 0
usb_device = 0
builtin_mic = 0
bluetooth_sco_headset = 0
+ stub = 0
domain: VoiceUplinkAndVoiceDownlinkAndVoiceCall
conf: VoiceCall
@@ -354,12 +373,29 @@
voice_downlink/applicable_input_device/mask/telephony_rx = 1
voice_call/applicable_input_device/mask/telephony_rx = 1
voice_uplink/applicable_input_device/mask/telephony_rx = 1
+ voice_downlink/applicable_input_device/mask/stub = 0
+ voice_call/applicable_input_device/mask/stub = 0
+ voice_uplink/applicable_input_device/mask/stub = 0
+
+ conf: Stub
+ AvailableInputDevices Includes Default
+
+ component: /Policy/policy/input_sources
+ voice_downlink/applicable_input_device/mask/telephony_rx = 0
+ voice_call/applicable_input_device/mask/telephony_rx = 0
+ voice_uplink/applicable_input_device/mask/telephony_rx = 0
+ voice_downlink/applicable_input_device/mask/stub = 1
+ voice_call/applicable_input_device/mask/stub = 1
+ voice_uplink/applicable_input_device/mask/stub = 1
conf: Default
component: /Policy/policy/input_sources
voice_downlink/applicable_input_device/mask/telephony_rx = 0
voice_call/applicable_input_device/mask/telephony_rx = 0
voice_uplink/applicable_input_device/mask/telephony_rx = 0
+ voice_downlink/applicable_input_device/mask/stub = 0
+ voice_call/applicable_input_device/mask/stub = 0
+ voice_uplink/applicable_input_device/mask/stub = 0
domain: Camcorder
conf: BackMic
@@ -368,6 +404,7 @@
component: /Policy/policy/input_sources/camcorder/applicable_input_device/mask
back_mic = 1
builtin_mic = 0
+ stub = 0
conf: BuiltinMic
AvailableInputDevices Includes BuiltinMic
@@ -375,11 +412,21 @@
component: /Policy/policy/input_sources/camcorder/applicable_input_device/mask
back_mic = 0
builtin_mic = 1
+ stub = 0
+
+ conf: Stub
+ AvailableInputDevices Includes Default
+
+ component: /Policy/policy/input_sources/camcorder/applicable_input_device/mask
+ back_mic = 0
+ builtin_mic = 0
+ stub = 1
conf: Default
component: /Policy/policy/input_sources/camcorder/applicable_input_device/mask
back_mic = 0
builtin_mic = 0
+ stub = 0
domain: VoiceRecognitionAndUnprocessedAndHotword
conf: ScoHeadset
@@ -392,16 +439,19 @@
wired_headset = 0
usb_device = 0
builtin_mic = 0
+ stub = 0
component: unprocessed/applicable_input_device/mask
bluetooth_sco_headset = 1
wired_headset = 0
usb_device = 0
builtin_mic = 0
+ stub = 0
component: hotword/applicable_input_device/mask
bluetooth_sco_headset = 1
wired_headset = 0
usb_device = 0
builtin_mic = 0
+ stub = 0
conf: WiredHeadset
AvailableInputDevices Includes WiredHeadset
@@ -411,17 +461,20 @@
bluetooth_sco_headset = 0
wired_headset = 1
usb_device = 0
+ stub = 0
builtin_mic = 0
component: unprocessed/applicable_input_device/mask
bluetooth_sco_headset = 0
wired_headset = 1
usb_device = 0
builtin_mic = 0
+ stub = 0
component: hotword/applicable_input_device/mask
bluetooth_sco_headset = 0
wired_headset = 1
usb_device = 0
builtin_mic = 0
+ stub = 0
conf: UsbDevice
AvailableInputDevices Includes UsbDevice
@@ -432,16 +485,19 @@
wired_headset = 0
usb_device = 1
builtin_mic = 0
+ stub = 0
component: unprocessed/applicable_input_device/mask
bluetooth_sco_headset = 0
wired_headset = 0
usb_device = 1
builtin_mic = 0
+ stub = 0
component: hotword/applicable_input_device/mask
bluetooth_sco_headset = 0
wired_headset = 0
usb_device = 1
builtin_mic = 0
+ stub = 0
conf: BuiltinMic
AvailableInputDevices Includes BuiltinMic
@@ -452,17 +508,42 @@
wired_headset = 0
usb_device = 0
builtin_mic = 1
+ stub = 0
component: unprocessed/applicable_input_device/mask
bluetooth_sco_headset = 0
wired_headset = 0
usb_device = 0
builtin_mic = 1
+ stub = 0
component: hotword/applicable_input_device/mask
bluetooth_sco_headset = 0
wired_headset = 0
usb_device = 0
builtin_mic = 1
+ stub = 0
+ conf: Stub
+ AvailableInputDevices Includes Default
+
+ component: /Policy/policy/input_sources
+ component: voice_recognition/applicable_input_device/mask
+ bluetooth_sco_headset = 0
+ wired_headset = 0
+ usb_device = 0
+ builtin_mic = 0
+ stub = 1
+ component: unprocessed/applicable_input_device/mask
+ bluetooth_sco_headset = 0
+ wired_headset = 0
+ usb_device = 0
+ builtin_mic = 0
+ stub = 1
+ component: hotword/applicable_input_device/mask
+ bluetooth_sco_headset = 0
+ wired_headset = 0
+ usb_device = 0
+ builtin_mic = 0
+ stub = 1
conf: Default
component: /Policy/policy/input_sources
component: voice_recognition/applicable_input_device/mask
@@ -470,16 +551,19 @@
wired_headset = 0
usb_device = 0
builtin_mic = 0
+ stub = 0
component: unprocessed/applicable_input_device/mask
bluetooth_sco_headset = 0
wired_headset = 0
usb_device = 0
builtin_mic = 0
+ stub = 0
component: hotword/applicable_input_device/mask
bluetooth_sco_headset = 0
wired_headset = 0
usb_device = 0
builtin_mic = 0
+ stub = 0
domain: VoiceCommunication
conf: ScoHeadset
@@ -495,6 +579,7 @@
usb_device = 0
builtin_mic = 0
back_mic = 0
+ stub = 0
conf: WiredHeadset
ForceUseForCommunication Is ForceNone
@@ -506,6 +591,7 @@
usb_device = 0
builtin_mic = 0
back_mic = 0
+ stub = 0
conf: UsbDevice
ForceUseForCommunication Is ForceNone
@@ -517,6 +603,7 @@
usb_device = 1
builtin_mic = 0
back_mic = 0
+ stub = 0
conf: BuiltinMic
AvailableInputDevices Includes BuiltinMic
@@ -532,6 +619,7 @@
usb_device = 0
builtin_mic = 1
back_mic = 0
+ stub = 0
conf: BackMic
ForceUseForCommunication Is ForceSpeaker
@@ -543,6 +631,7 @@
usb_device = 0
builtin_mic = 0
back_mic = 1
+ stub = 0
conf: Default
#
@@ -554,6 +643,7 @@
usb_device = 0
builtin_mic = 1
back_mic = 0
+ stub = 0
domain: RemoteSubmix
conf: RemoteSubmix
@@ -561,10 +651,19 @@
component: /Policy/policy/input_sources/remote_submix/applicable_input_device/mask
remote_submix = 1
+ stub = 0
+
+ conf: Stub
+ AvailableInputDevices Includes Default
+
+ component: /Policy/policy/input_sources/remote_submix/applicable_input_device/mask
+ remote_submix = 0
+ stub = 1
conf: Default
component: /Policy/policy/input_sources/remote_submix/applicable_input_device/mask
remote_submix = 0
+ stub = 0
domain: FmTuner
conf: FmTuner
@@ -572,8 +671,29 @@
component: /Policy/policy/input_sources/fm_tuner/applicable_input_device/mask
fm_tuner = 1
+ stub = 0
+
+ conf: Stub
+ AvailableInputDevices Includes Default
+
+ component: /Policy/policy/input_sources/fm_tuner/applicable_input_device/mask
+ fm_tuner = 0
+ stub = 1
conf: Default
component: /Policy/policy/input_sources/fm_tuner/applicable_input_device/mask
fm_tuner = 0
+ stub = 0
+
+ domain: Voice
+ conf: Stub
+ AvailableInputDevices Includes Default
+
+ /Policy/policy/input_sources/echo_reference/applicable_input_device/mask/stub = 1
+ /Policy/policy/input_sources/voice_performance/applicable_input_device/mask/stub = 1
+
+ conf: Default
+ /Policy/policy/input_sources/echo_reference/applicable_input_device/mask/stub = 0
+ /Policy/policy/input_sources/voice_performance/applicable_input_device/mask/stub = 0
+
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/volumes.pfw b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/volumes.pfw
index 7db4537..cf1857e 100644
--- a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/volumes.pfw
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/volumes.pfw
@@ -11,6 +11,7 @@
/Policy/policy/streams/enforced_audible/applicable_volume_profile/volume_profile = enforced_audible
/Policy/policy/streams/tts/applicable_volume_profile/volume_profile = tts
/Policy/policy/streams/accessibility/applicable_volume_profile/volume_profile = accessibility
+ /Policy/policy/streams/assistant/applicable_volume_profile/volume_profile = assistant
/Policy/policy/streams/rerouting/applicable_volume_profile/volume_profile = rerouting
/Policy/policy/streams/patch/applicable_volume_profile/volume_profile = patch
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/SettingsNoInput/Android.bp b/services/audiopolicy/engineconfigurable/parameter-framework/examples/SettingsNoInput/Android.bp
new file mode 100644
index 0000000..9abcb70
--- /dev/null
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/examples/SettingsNoInput/Android.bp
@@ -0,0 +1,73 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+// Import this namespace in order to use AOSP No Input configuration example
+
+soong_namespace {
+ imports: [
+ "frameworks/av/services/audiopolicy/engineconfigurable/config/example/phone",
+ "frameworks/av/services/audiopolicy/config",
+ ],
+}
+
+prebuilt_etc {
+ name: "parameter-framework.policy",
+ filename_from_src: true,
+ vendor: true,
+ src: ":domaingeneratorpolicyrule_gen",
+ sub_dir: "parameter-framework/Settings/Policy",
+ required: [
+ "PolicyClass.xml",
+ "PolicySubsystem.xml",
+ "PolicySubsystem-CommonTypes.xml",
+ ],
+}
+
+genrule {
+ name: "domaingeneratorpolicyrule_gen",
+ defaults: ["domaingeneratorpolicyrule"],
+ srcs: [
+ ":audio_policy_pfw_toplevel",
+ ":audio_policy_pfw_structure_files",
+ ":audio_policy_engine_criterion_types",
+ ":edd_files",
+ ],
+}
+filegroup {
+ name: "audio_policy_pfw_toplevel",
+ srcs: [":ParameterFrameworkConfigurationPolicy.userdebug.xml"],
+}
+filegroup {
+ name: "audio_policy_pfw_structure_files",
+ srcs: [
+ ":PolicyClass",
+ ":PolicySubsystem",
+ ":buildcommontypesstructure_gen",
+ ],
+}
+filegroup {
+ name: "edd_files",
+ srcs: [
+ "device_for_input_source.pfw",
+ ":volumes.pfw",
+ ],
+}
+prebuilt_etc {
+ name: "PolicySubsystem.xml",
+ vendor: true,
+ src: ":PolicySubsystem-no-strategy",
+ sub_dir: "parameter-framework/Structure/Policy",
+}
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/SettingsNoOutput/Android.bp b/services/audiopolicy/engineconfigurable/parameter-framework/examples/SettingsNoOutput/Android.bp
new file mode 100644
index 0000000..27172a4
--- /dev/null
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/examples/SettingsNoOutput/Android.bp
@@ -0,0 +1,73 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+// Import this namespace in order to use AOSP No output configuration example
+
+soong_namespace {
+ imports: [
+ "frameworks/av/services/audiopolicy/engineconfigurable/config/example/phone",
+ "frameworks/av/services/audiopolicy/config",
+ ],
+}
+
+prebuilt_etc {
+ name: "parameter-framework.policy",
+ filename_from_src: true,
+ vendor: true,
+ src: ":domaingeneratorpolicyrule_gen",
+ sub_dir: "parameter-framework/Settings/Policy",
+ required: [
+ "PolicyClass.xml",
+ "PolicySubsystem.xml",
+ "PolicySubsystem-CommonTypes.xml",
+ ],
+}
+genrule {
+ name: "domaingeneratorpolicyrule_gen",
+ defaults: ["domaingeneratorpolicyrule"],
+ srcs: [
+ ":audio_policy_pfw_toplevel",
+ ":audio_policy_pfw_structure_files",
+ ":audio_policy_engine_criterion_types",
+ ":edd_files",
+ ],
+}
+filegroup {
+ name: "audio_policy_pfw_toplevel",
+ srcs: [":ParameterFrameworkConfigurationPolicy.userdebug.xml"],
+}
+filegroup {
+ name: "audio_policy_pfw_structure_files",
+ srcs: [
+ ":PolicyClass",
+ ":PolicySubsystem",
+ ":buildcommontypesstructure_gen",
+ ],
+}
+filegroup {
+ name: "edd_files",
+ srcs: [
+ "device_for_strategies.pfw",
+ ":volumes.pfw",
+ ":device_for_input_source.pfw",
+ ],
+}
+prebuilt_etc {
+ name: "PolicySubsystem.xml",
+ vendor: true,
+ src: ":PolicySubsystem-no-strategy",
+ sub_dir: "parameter-framework/Structure/Policy",
+}
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/SettingsNoOutput/device_for_strategies.pfw b/services/audiopolicy/engineconfigurable/parameter-framework/examples/SettingsNoOutput/device_for_strategies.pfw
index f923610..e259c00 100644
--- a/services/audiopolicy/engineconfigurable/parameter-framework/examples/SettingsNoOutput/device_for_strategies.pfw
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/examples/SettingsNoOutput/device_for_strategies.pfw
@@ -13,7 +13,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -41,7 +41,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -69,7 +69,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -97,7 +97,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -125,7 +125,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -153,7 +153,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -181,7 +181,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -209,7 +209,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
@@ -237,7 +237,7 @@
bluetooth_a2dp_headphones = 0
bluetooth_a2dp_speaker = 0
hdmi = 0
- angl_dock_headset = 0
+ anlg_dock_headset = 0
dgtl_dock_headset = 0
usb_accessory = 0
usb_device = 0
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/common/Structure/PolicySubsystem-CommonTypes.xml b/services/audiopolicy/engineconfigurable/parameter-framework/examples/common/Structure/PolicySubsystem-CommonTypes.xml
deleted file mode 100644
index d17c021..0000000
--- a/services/audiopolicy/engineconfigurable/parameter-framework/examples/common/Structure/PolicySubsystem-CommonTypes.xml
+++ /dev/null
@@ -1,186 +0,0 @@
-<?xml version="1.0" encoding="UTF-8"?>
-<ComponentTypeSet xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance"
- xmlns:xi="http://www.w3.org/2001/XInclude"
- xsi:noNamespaceSchemaLocation="Schemas/ComponentTypeSet.xsd">
- <!-- Output devices definition as a bitfield for the supported devices per output
- profile. It must match with the output device enum parameter.
- -->
- <!--#################### GLOBAL COMPONENTS BEGIN ####################-->
-
- <!--#################### GLOBAL COMPONENTS END ####################-->
-
- <ComponentType Name="OutputDevicesMask" Description="32th bit is not allowed as dedicated
- for input devices detection">
- <BitParameterBlock Name="mask" Size="32">
- <BitParameter Name="earpiece" Size="1" Pos="0"/>
- <BitParameter Name="speaker" Size="1" Pos="1"/>
- <BitParameter Name="wired_headset" Size="1" Pos="2"/>
- <BitParameter Name="wired_headphone" Size="1" Pos="3"/>
- <BitParameter Name="bluetooth_sco" Size="1" Pos="4"/>
- <BitParameter Name="bluetooth_sco_headset" Size="1" Pos="5"/>
- <BitParameter Name="bluetooth_sco_carkit" Size="1" Pos="6"/>
- <BitParameter Name="bluetooth_a2dp" Size="1" Pos="7"/>
- <BitParameter Name="bluetooth_a2dp_headphones" Size="1" Pos="8"/>
- <BitParameter Name="bluetooth_a2dp_speaker" Size="1" Pos="9"/>
- <BitParameter Name="hdmi" Size="1" Pos="10"/>
- <BitParameter Name="angl_dock_headset" Size="1" Pos="11"/>
- <BitParameter Name="dgtl_dock_headset" Size="1" Pos="12"/>
- <BitParameter Name="usb_accessory" Size="1" Pos="13"/>
- <BitParameter Name="usb_device" Size="1" Pos="14"/>
- <BitParameter Name="remote_submix" Size="1" Pos="15"/>
- <BitParameter Name="telephony_tx" Size="1" Pos="16"/>
- <BitParameter Name="line" Size="1" Pos="17"/>
- <BitParameter Name="hdmi_arc" Size="1" Pos="18"/>
- <BitParameter Name="spdif" Size="1" Pos="19"/>
- <BitParameter Name="fm" Size="1" Pos="20"/>
- <BitParameter Name="aux_line" Size="1" Pos="21"/>
- <BitParameter Name="speaker_safe" Size="1" Pos="22"/>
- <BitParameter Name="ip" Size="1" Pos="23"/>
- <BitParameter Name="bus" Size="1" Pos="24"/>
- <BitParameter Name="proxy" Size="1" Pos="25"/>
- <BitParameter Name="usb_headset" Size="1" Pos="26"/>
- <BitParameter Name="hearing_aid" Size="1" Pos="27"/>
- <BitParameter Name="echo_canceller" Size="1" Pos="28"/>
- <BitParameter Name="stub" Size="1" Pos="30"/>
- </BitParameterBlock>
- </ComponentType>
-
- <!-- Input devices definition as a bitfield for the supported devices per Input
- profile. It must match with the Input device enum parameter.
- -->
- <ComponentType Name="InputDevicesMask">
- <BitParameterBlock Name="mask" Size="32">
- <BitParameter Name="communication" Size="1" Pos="0"/>
- <BitParameter Name="ambient" Size="1" Pos="1"/>
- <BitParameter Name="builtin_mic" Size="1" Pos="2"/>
- <BitParameter Name="bluetooth_sco_headset" Size="1" Pos="3"/>
- <BitParameter Name="wired_headset" Size="1" Pos="4"/>
- <BitParameter Name="hdmi" Size="1" Pos="5"/>
- <BitParameter Name="telephony_rx" Size="1" Pos="6"/>
- <BitParameter Name="back_mic" Size="1" Pos="7"/>
- <BitParameter Name="remote_submix" Size="1" Pos="8"/>
- <BitParameter Name="anlg_dock_headset" Size="1" Pos="9"/>
- <BitParameter Name="dgtl_dock_headset" Size="1" Pos="10"/>
- <BitParameter Name="usb_accessory" Size="1" Pos="11"/>
- <BitParameter Name="usb_device" Size="1" Pos="12"/>
- <BitParameter Name="fm_tuner" Size="1" Pos="13"/>
- <BitParameter Name="tv_tuner" Size="1" Pos="14"/>
- <BitParameter Name="line" Size="1" Pos="15"/>
- <BitParameter Name="spdif" Size="1" Pos="16"/>
- <BitParameter Name="bluetooth_a2dp" Size="1" Pos="17"/>
- <BitParameter Name="loopback" Size="1" Pos="18"/>
- <BitParameter Name="ip" Size="1" Pos="19"/>
- <BitParameter Name="bus" Size="1" Pos="20"/>
- <BitParameter Name="proxy" Size="1" Pos="21"/>
- <BitParameter Name="usb_headset" Size="1" Pos="22"/>
- <BitParameter Name="bluetooth_ble" Size="1" Pos="23"/>
- <BitParameter Name="hdmi_arc" Size="1" Pos="24"/>
- <BitParameter Name="echo_reference" Size="1" Pos="25"/>
- <BitParameter Name="stub" Size="1" Pos="30"/>
- </BitParameterBlock>
- </ComponentType>
-
- <ComponentType Name="OutputFlags"
- Description="the audio output flags serve two purposes:
- - when an AudioTrack is created they indicate a wish to be connected to an
- output stream with attributes corresponding to the specified flags.
- - when present in an output profile descriptor listed for a particular audio
- hardware module, they indicate that an output stream can be opened that
- supports the attributes indicated by the flags.
- The audio policy manager will try to match the flags in the request
- (when getOuput() is called) to an available output stream.">
- <BitParameterBlock Name="mask" Size="32">
- <BitParameter Name="direct" Size="1" Pos="0"/>
- <BitParameter Name="primary" Size="1" Pos="1"/>
- <BitParameter Name="fast" Size="1" Pos="2"/>
- <BitParameter Name="deep_buffer" Size="1" Pos="3"/>
- <BitParameter Name="compress_offload" Size="1" Pos="4"/>
- <BitParameter Name="non_blocking" Size="1" Pos="5"/>
- <BitParameter Name="hw_av_sync" Size="1" Pos="6"/>
- <BitParameter Name="tts" Size="1" Pos="7"/>
- <BitParameter Name="raw" Size="1" Pos="8"/>
- <BitParameter Name="sync" Size="1" Pos="9"/>
- <BitParameter Name="iec958_nonaudio" Size="1" Pos="10"/>
- </BitParameterBlock>
- </ComponentType>
-
- <ComponentType Name="InputFlags"
- Description="The audio input flags are analogous to audio output flags.
- Currently they are used only when an AudioRecord is created,
- to indicate a preference to be connected to an input stream with
- attributes corresponding to the specified flags.">
- <BitParameterBlock Name="mask" Size="32">
- <BitParameter Name="fast" Size="1" Pos="0"/>
- <BitParameter Name="hw_hotword" Size="1" Pos="2"/>
- <BitParameter Name="raw" Size="1" Pos="3"/>
- <BitParameter Name="sync" Size="1" Pos="4"/>
- </BitParameterBlock>
- </ComponentType>
-
- <ComponentType Name="InputSourcesMask" Description="The audio input source is also known
- as the use case.">
- <BitParameterBlock Name="mask" Size="32">
- <BitParameter Name="default" Size="1" Pos="0"/>
- <BitParameter Name="mic" Size="1" Pos="1"/>
- <BitParameter Name="voice_uplink" Size="1" Pos="2"/>
- <BitParameter Name="voice_downlink" Size="1" Pos="3"/>
- <BitParameter Name="voice_call" Size="1" Pos="4"/>
- <BitParameter Name="camcorder" Size="1" Pos="5"/>
- <BitParameter Name="voice_recognition" Size="1" Pos="6"/>
- <BitParameter Name="voice_communication" Size="1" Pos="7"/>
- <BitParameter Name="remote_submix" Size="1" Pos="8"/>
- <BitParameter Name="unprocessed" Size="1" Pos="9"/>
- <BitParameter Name="voice_performance" Size="1" Pos="10"/>
- <BitParameter Name="echo_reference" Size="1" Pos="11"/>
- <BitParameter Name="fm_tuner" Size="1" Pos="12"/>
- <BitParameter Name="hotword" Size="1" Pos="13"/>
- </BitParameterBlock>
- </ComponentType>
-
- <!--#################### STREAM COMMON TYPES BEGIN ####################-->
-
- <ComponentType Name="VolumeProfileType">
- <EnumParameter Name="volume_profile" Size="32">
- <ValuePair Literal="voice_call" Numerical="0"/>
- <ValuePair Literal="system" Numerical="1"/>
- <ValuePair Literal="ring" Numerical="2"/>
- <ValuePair Literal="music" Numerical="3"/>
- <ValuePair Literal="alarm" Numerical="4"/>
- <ValuePair Literal="notification" Numerical="5"/>
- <ValuePair Literal="bluetooth_sco" Numerical="6"/>
- <ValuePair Literal="enforced_audible" Numerical="7"/>
- <ValuePair Literal="dtmf" Numerical="8"/>
- <ValuePair Literal="tts" Numerical="9"/>
- <ValuePair Literal="accessibility" Numerical="10"/>
- <ValuePair Literal="rerouting" Numerical="11"/>
- <ValuePair Literal="patch" Numerical="12"/>
- </EnumParameter>
- </ComponentType>
-
- <ComponentType Name="Stream" Mapping="Stream">
- <Component Name="applicable_volume_profile" Type="VolumeProfileType"
- Description="Volume profile followed by a given stream type."/>
- </ComponentType>
-
- <!--#################### STREAM COMMON TYPES END ####################-->
-
- <!--#################### INPUT SOURCE COMMON TYPES BEGIN ####################-->
-
- <ComponentType Name="InputSource">
- <Component Name="applicable_input_device" Type="InputDevicesMask"
- Mapping="InputSource" Description="Selected Input device"/>
- </ComponentType>
-
- <!--#################### INPUT SOURCE COMMON TYPES END ####################-->
-
- <!--#################### PRODUCT STRATEGY COMMON TYPES BEGIN ####################-->
-
- <ComponentType Name="ProductStrategy" Mapping="ProductStrategy">
- <Component Name="selected_output_devices" Type="OutputDevicesMask"/>
- <StringParameter Name="device_address" MaxLength="256"
- Description="if any, device address associated"/>
- </ComponentType>
-
- <!--#################### PRODUCT STRATEGY COMMON TYPES END ####################-->
-
-</ComponentTypeSet>
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/common/Structure/PolicySubsystem-CommonTypes.xml.in b/services/audiopolicy/engineconfigurable/parameter-framework/examples/common/Structure/PolicySubsystem-CommonTypes.xml.in
new file mode 100644
index 0000000..2e9f37e
--- /dev/null
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/examples/common/Structure/PolicySubsystem-CommonTypes.xml.in
@@ -0,0 +1,60 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<ComponentTypeSet xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance"
+ xmlns:xi="http://www.w3.org/2001/XInclude"
+ xsi:noNamespaceSchemaLocation="Schemas/ComponentTypeSet.xsd">
+ <!-- Output devices definition as a bitfield for the supported devices per output
+ profile. It must match with the output device enum parameter.
+ -->
+ <!--#################### GLOBAL COMPONENTS BEGIN ####################-->
+
+ <!--#################### GLOBAL COMPONENTS END ####################-->
+
+ <!-- Automatically filled from audio-base.h file -->
+ <ComponentType Name="OutputDevicesMask" Description="32th bit is not allowed as dedicated for input devices detection">
+ <BitParameterBlock Name="mask" Size="32">
+ </BitParameterBlock>
+ </ComponentType>
+
+ <!-- Input devices definition as a bitfield for the supported devices per Input
+ profile. It must match with the Input device enum parameter.
+ -->
+ <!-- Automatically filled from audio-base.h file -->
+ <ComponentType Name="InputDevicesMask">
+ <BitParameterBlock Name="mask" Size="32">
+ </BitParameterBlock>
+ </ComponentType>
+
+ <!--#################### STREAM COMMON TYPES BEGIN ####################-->
+ <!-- Automatically filled from audio-base.h file. VolumeProfileType is associated to stream type -->
+ <ComponentType Name="VolumeProfileType">
+ <EnumParameter Name="volume_profile" Size="32">
+ </EnumParameter>
+ </ComponentType>
+
+ <ComponentType Name="Stream" Mapping="Stream">
+ <Component Name="applicable_volume_profile" Type="VolumeProfileType"
+ Description="Volume profile followed by a given stream type."/>
+ </ComponentType>
+
+ <!--#################### STREAM COMMON TYPES END ####################-->
+
+ <!--#################### INPUT SOURCE COMMON TYPES BEGIN ####################-->
+
+ <ComponentType Name="InputSource">
+ <Component Name="applicable_input_device" Type="InputDevicesMask"
+ Mapping="InputSource" Description="Selected Input device"/>
+ </ComponentType>
+
+ <!--#################### INPUT SOURCE COMMON TYPES END ####################-->
+
+ <!--#################### PRODUCT STRATEGY COMMON TYPES BEGIN ####################-->
+
+ <ComponentType Name="ProductStrategy" Mapping="ProductStrategy">
+ <Component Name="selected_output_devices" Type="OutputDevicesMask"/>
+ <StringParameter Name="device_address" MaxLength="256"
+ Description="if any, device address associated"/>
+ </ComponentType>
+
+ <!--#################### PRODUCT STRATEGY COMMON TYPES END ####################-->
+
+</ComponentTypeSet>
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/common/Structure/PolicySubsystem-no-strategy.xml b/services/audiopolicy/engineconfigurable/parameter-framework/examples/common/Structure/PolicySubsystem-no-strategy.xml
index a4e7537..ed349c8 100644
--- a/services/audiopolicy/engineconfigurable/parameter-framework/examples/common/Structure/PolicySubsystem-no-strategy.xml
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/examples/common/Structure/PolicySubsystem-no-strategy.xml
@@ -28,6 +28,8 @@
Description="Transmitted Through Speaker. Plays over speaker only, silent on other devices"/>
<Component Name="accessibility" Type="Stream" Mapping="Name:AUDIO_STREAM_ACCESSIBILITY"
Description="For accessibility talk back prompts"/>
+ <Component Name="assistant" Type="Stream" Mapping="Name:AUDIO_STREAM_ASSISTANT"
+ Description="used by a virtual assistant like Google Assistant, Bixby, etc."/>
<Component Name="rerouting" Type="Stream" Mapping="Name:AUDIO_STREAM_REROUTING"
Description="For dynamic policy output mixes"/>
<Component Name="patch" Type="Stream" Mapping="Name:AUDIO_STREAM_PATCH"
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/common/Structure/PolicySubsystem.xml b/services/audiopolicy/engineconfigurable/parameter-framework/examples/common/Structure/PolicySubsystem.xml
index b55ce2c..7bbb57a 100644
--- a/services/audiopolicy/engineconfigurable/parameter-framework/examples/common/Structure/PolicySubsystem.xml
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/examples/common/Structure/PolicySubsystem.xml
@@ -44,6 +44,8 @@
Description="Transmitted Through Speaker. Plays over speaker only, silent on other devices"/>
<Component Name="accessibility" Type="Stream" Mapping="Name:AUDIO_STREAM_ACCESSIBILITY"
Description="For accessibility talk back prompts"/>
+ <Component Name="assistant" Type="Stream" Mapping="Name:AUDIO_STREAM_ASSISTANT"
+ Description="used by a virtual assistant like Google Assistant, Bixby, etc."/>
<Component Name="rerouting" Type="Stream" Mapping="Name:AUDIO_STREAM_REROUTING"
Description="For dynamic policy output mixes"/>
<Component Name="patch" Type="Stream" Mapping="Name:AUDIO_STREAM_PATCH"
@@ -73,10 +75,13 @@
Mapping="Name:AUDIO_SOURCE_REMOTE_SUBMIX"/>
<Component Name="unprocessed" Type="InputSource"
Mapping="Name:AUDIO_SOURCE_UNPROCESSED"/>
+ <Component Name="voice_performance" Type="InputSource"
+ Mapping="Name:AUDIO_SOURCE_VOICE_PERFORMANCE"/>
+ <Component Name="echo_reference" Type="InputSource"
+ Mapping="Name:AUDIO_SOURCE_ECHO_REFERENCE"/>
<Component Name="fm_tuner" Type="InputSource" Mapping="Name:AUDIO_SOURCE_FM_TUNER"/>
<Component Name="hotword" Type="InputSource" Mapping="Name:AUDIO_SOURCE_HOTWORD"/>
</ComponentType>
-
<!--#################### INPUT SOURCE END ####################-->
</ComponentLibrary>
diff --git a/services/audiopolicy/engineconfigurable/src/Engine.cpp b/services/audiopolicy/engineconfigurable/src/Engine.cpp
index cb45fcf..6d42fcf 100644
--- a/services/audiopolicy/engineconfigurable/src/Engine.cpp
+++ b/services/audiopolicy/engineconfigurable/src/Engine.cpp
@@ -32,6 +32,9 @@
#include <policy.h>
#include <AudioIODescriptorInterface.h>
#include <ParameterManagerWrapper.h>
+#include <media/AudioContainers.h>
+
+#include <media/TypeConverter.h>
using std::string;
using std::map;
@@ -77,8 +80,9 @@
status_t Engine::initCheck()
{
- if (mPolicyParameterMgr == nullptr || mPolicyParameterMgr->start() != NO_ERROR) {
- ALOGE("%s: could not start Policy PFW", __FUNCTION__);
+ std::string error;
+ if (mPolicyParameterMgr == nullptr || mPolicyParameterMgr->start(error) != NO_ERROR) {
+ ALOGE("%s: could not start Policy PFW: %s", __FUNCTION__, error.c_str());
return NO_INIT;
}
return EngineBase::initCheck();
@@ -125,7 +129,7 @@
Element<Key> *element = getFromCollection<Key>(key);
if (element == NULL) {
ALOGE("%s: Element not found within collection", __FUNCTION__);
- return BAD_VALUE;
+ return false;
}
return element->template set<Property>(property) == NO_ERROR;
}
@@ -159,19 +163,21 @@
return mPolicyParameterMgr->getForceUse(usage);
}
-status_t Engine::setDeviceConnectionState(const sp<DeviceDescriptor> devDesc,
+status_t Engine::setDeviceConnectionState(const sp<DeviceDescriptor> device,
audio_policy_dev_state_t state)
{
- mPolicyParameterMgr->setDeviceConnectionState(devDesc, state);
-
- if (audio_is_output_device(devDesc->type())) {
+ mPolicyParameterMgr->setDeviceConnectionState(
+ device->type(), device->address().c_str(), state);
+ if (audio_is_output_device(device->type())) {
+ // FIXME: Use DeviceTypeSet when the interface is ready
return mPolicyParameterMgr->setAvailableOutputDevices(
- getApmObserver()->getAvailableOutputDevices().types());
- } else if (audio_is_input_device(devDesc->type())) {
+ deviceTypesToBitMask(getApmObserver()->getAvailableOutputDevices().types()));
+ } else if (audio_is_input_device(device->type())) {
+ // FIXME: Use DeviceTypeSet when the interface is ready
return mPolicyParameterMgr->setAvailableInputDevices(
- getApmObserver()->getAvailableInputDevices().types());
+ deviceTypesToBitMask(getApmObserver()->getAvailableInputDevices().types()));
}
- return BAD_TYPE;
+ return EngineBase::setDeviceConnectionState(device, state);
}
status_t Engine::loadAudioPolicyEngineConfig()
@@ -209,7 +215,7 @@
}
const DeviceVector availableOutputDevices = getApmObserver()->getAvailableOutputDevices();
const SwAudioOutputCollection &outputs = getApmObserver()->getOutputs();
- uint32_t availableOutputDevicesType = availableOutputDevices.types();
+ DeviceTypeSet availableOutputDevicesTypes = availableOutputDevices.types();
/** This is the only case handled programmatically because the PFW is unable to know the
* activity of streams.
@@ -221,7 +227,7 @@
*
* -When media is not playing anymore, fall back on the sonification behavior
*/
- audio_devices_t devices = AUDIO_DEVICE_NONE;
+ DeviceTypeSet deviceTypes;
if (ps == getProductStrategyForStream(AUDIO_STREAM_NOTIFICATION) &&
!is_state_in_call(getPhoneState()) &&
!outputs.isActiveRemotely(toVolumeSource(AUDIO_STREAM_MUSIC),
@@ -230,7 +236,7 @@
SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) {
product_strategy_t strategyForMedia =
getProductStrategyForStream(AUDIO_STREAM_MUSIC);
- devices = productStrategies.getDeviceTypesForProductStrategy(strategyForMedia);
+ deviceTypes = productStrategies.getDeviceTypesForProductStrategy(strategyForMedia);
} else if (ps == getProductStrategyForStream(AUDIO_STREAM_ACCESSIBILITY) &&
(outputs.isActive(toVolumeSource(AUDIO_STREAM_RING)) ||
outputs.isActive(toVolumeSource(AUDIO_STREAM_ALARM)))) {
@@ -238,28 +244,37 @@
// compressed format as they would likely not be mixed and dropped.
// Device For Sonification conf file has HDMI, SPDIF and HDMI ARC unreacheable.
product_strategy_t strategyNotification = getProductStrategyForStream(AUDIO_STREAM_RING);
- devices = productStrategies.getDeviceTypesForProductStrategy(strategyNotification);
+ deviceTypes = productStrategies.getDeviceTypesForProductStrategy(strategyNotification);
} else {
- devices = productStrategies.getDeviceTypesForProductStrategy(ps);
+ deviceTypes = productStrategies.getDeviceTypesForProductStrategy(ps);
}
- if (devices == AUDIO_DEVICE_NONE ||
- (devices & availableOutputDevicesType) == AUDIO_DEVICE_NONE) {
- devices = getApmObserver()->getDefaultOutputDevice()->type();
- ALOGE_IF(devices == AUDIO_DEVICE_NONE, "%s: no valid default device defined", __FUNCTION__);
- return DeviceVector(getApmObserver()->getDefaultOutputDevice());
+ if (deviceTypes.empty() ||
+ Intersection(deviceTypes, availableOutputDevicesTypes).empty()) {
+ auto defaultDevice = getApmObserver()->getDefaultOutputDevice();
+ ALOG_ASSERT(defaultDevice != nullptr, "no valid default device defined");
+ return DeviceVector(defaultDevice);
}
- if (/*device_distinguishes_on_address(devices)*/ devices == AUDIO_DEVICE_OUT_BUS) {
+ if (/*device_distinguishes_on_address(*deviceTypes.begin())*/ isSingleDeviceType(
+ deviceTypes, AUDIO_DEVICE_OUT_BUS)) {
// We do expect only one device for these types of devices
// Criterion device address garantee this one is available
// If this criterion is not wished, need to ensure this device is available
const String8 address(productStrategies.getDeviceAddressForProductStrategy(ps).c_str());
- ALOGV("%s:device 0x%x %s %d", __FUNCTION__, devices, address.c_str(), ps);
- return DeviceVector(availableOutputDevices.getDevice(devices,
- address,
- AUDIO_FORMAT_DEFAULT));
+ ALOGV("%s:device %s %s %d",
+ __FUNCTION__, dumpDeviceTypes(deviceTypes).c_str(), address.c_str(), ps);
+ auto busDevice = availableOutputDevices.getDevice(
+ *deviceTypes.begin(), address, AUDIO_FORMAT_DEFAULT);
+ if (busDevice == nullptr) {
+ ALOGE("%s:unavailable device %s %s, fallback on default", __func__,
+ dumpDeviceTypes(deviceTypes).c_str(), address.c_str());
+ auto defaultDevice = getApmObserver()->getDefaultOutputDevice();
+ ALOG_ASSERT(defaultDevice != nullptr, "Default Output Device NOT available");
+ return DeviceVector(defaultDevice);
+ }
+ return DeviceVector(busDevice);
}
- ALOGV("%s:device 0x%x %d", __FUNCTION__, devices, ps);
- return availableOutputDevices.getDevicesFromTypeMask(devices);
+ ALOGV("%s:device %s %d", __FUNCTION__, dumpDeviceTypes(deviceTypes).c_str(), ps);
+ return availableOutputDevices.getDevicesFromTypes(deviceTypes);
}
DeviceVector Engine::getOutputDevicesForAttributes(const audio_attributes_t &attributes,
@@ -356,12 +371,13 @@
ALOGE("%s: set device %d on invalid strategy %d", __FUNCTION__, devices, strategy);
return false;
}
- getProductStrategies().at(strategy)->setDeviceTypes(devices);
+ // FIXME: stop using deviceTypesFromBitMask when the interface is ready
+ getProductStrategies().at(strategy)->setDeviceTypes(deviceTypesFromBitMask(devices));
return true;
}
template <>
-AudioPolicyManagerInterface *Engine::queryInterface()
+EngineInterface *Engine::queryInterface()
{
return this;
}
diff --git a/services/audiopolicy/engineconfigurable/src/Engine.h b/services/audiopolicy/engineconfigurable/src/Engine.h
index 4662e7e..3b371d8 100644
--- a/services/audiopolicy/engineconfigurable/src/Engine.h
+++ b/services/audiopolicy/engineconfigurable/src/Engine.h
@@ -17,7 +17,7 @@
#pragma once
#include "EngineBase.h"
-#include <AudioPolicyManagerInterface.h>
+#include <EngineInterface.h>
#include <AudioPolicyPluginInterface.h>
#include "Collection.h"
diff --git a/services/audiopolicy/engineconfigurable/src/EngineInstance.cpp b/services/audiopolicy/engineconfigurable/src/EngineInstance.cpp
index 2442590..b127796 100644
--- a/services/audiopolicy/engineconfigurable/src/EngineInstance.cpp
+++ b/services/audiopolicy/engineconfigurable/src/EngineInstance.cpp
@@ -14,7 +14,7 @@
* limitations under the License.
*/
-#include <AudioPolicyManagerInterface.h>
+#include <EngineInterface.h>
#include <AudioPolicyPluginInterface.h>
#include "AudioPolicyEngineInstance.h"
#include "Engine.h"
@@ -45,9 +45,9 @@
}
template <>
-AudioPolicyManagerInterface *EngineInstance::queryInterface() const
+EngineInterface *EngineInstance::queryInterface() const
{
- return getEngine()->queryInterface<AudioPolicyManagerInterface>();
+ return getEngine()->queryInterface<EngineInterface>();
}
template <>
@@ -57,5 +57,16 @@
}
} // namespace audio_policy
+
+extern "C" EngineInterface* createEngineInstance()
+{
+ return audio_policy::EngineInstance::getInstance()->queryInterface<EngineInterface>();
+}
+
+extern "C" void destroyEngineInstance(EngineInterface*)
+{
+ // The engine is a singleton.
+}
+
} // namespace android
diff --git a/services/audiopolicy/engineconfigurable/src/InputSource.cpp b/services/audiopolicy/engineconfigurable/src/InputSource.cpp
index d252d3f..aa06ae3 100644
--- a/services/audiopolicy/engineconfigurable/src/InputSource.cpp
+++ b/services/audiopolicy/engineconfigurable/src/InputSource.cpp
@@ -30,7 +30,7 @@
return BAD_VALUE;
}
mIdentifier = identifier;
- ALOGD("%s: InputSource %s identifier 0x%X", __FUNCTION__, getName().c_str(), identifier);
+ ALOGV("%s: InputSource %s identifier 0x%X", __FUNCTION__, getName().c_str(), identifier);
return NO_ERROR;
}
@@ -46,15 +46,18 @@
template <>
status_t Element<audio_source_t>::set(audio_devices_t devices)
{
- if (devices != AUDIO_DEVICE_NONE) {
- devices |= AUDIO_DEVICE_BIT_IN;
+ if (devices == AUDIO_DEVICE_NONE) {
+ // Reset
+ mApplicableDevices = devices;
+ return NO_ERROR;
}
+ devices |= AUDIO_DEVICE_BIT_IN;
if (!audio_is_input_device(devices)) {
ALOGE("%s: trying to set an invalid device 0x%X for input source %s",
__FUNCTION__, devices, getName().c_str());
return BAD_VALUE;
}
- ALOGD("%s: 0x%X for input source %s", __FUNCTION__, devices, getName().c_str());
+ ALOGV("%s: 0x%X for input source %s", __FUNCTION__, devices, getName().c_str());
mApplicableDevices = devices;
return NO_ERROR;
}
diff --git a/services/audiopolicy/engineconfigurable/src/InputSource.h b/services/audiopolicy/engineconfigurable/src/InputSource.h
index e1865cc..d64a60a 100644
--- a/services/audiopolicy/engineconfigurable/src/InputSource.h
+++ b/services/audiopolicy/engineconfigurable/src/InputSource.h
@@ -73,10 +73,11 @@
Element(const Element &object);
Element &operator=(const Element &object);
- std::string mName; /**< Unique literal Identifier of a policy base element*/
- audio_source_t mIdentifier; /**< Unique numerical Identifier of a policy base element*/
-
- audio_devices_t mApplicableDevices; /**< Applicable input device for this input source. */
+ const std::string mName; /**< Unique literal Identifier of a policy base element*/
+ /** Unique numerical Identifier of a policy base element */
+ audio_source_t mIdentifier = AUDIO_SOURCE_DEFAULT;
+ /** Applicable input device for this input source. */
+ audio_devices_t mApplicableDevices = AUDIO_DEVICE_NONE;
};
typedef Element<audio_source_t> InputSource;
diff --git a/services/audiopolicy/engineconfigurable/src/Stream.cpp b/services/audiopolicy/engineconfigurable/src/Stream.cpp
index 297eb02..e64ba4b 100644
--- a/services/audiopolicy/engineconfigurable/src/Stream.cpp
+++ b/services/audiopolicy/engineconfigurable/src/Stream.cpp
@@ -30,7 +30,7 @@
return BAD_VALUE;
}
mIdentifier = identifier;
- ALOGD("%s: Stream %s identifier 0x%X", __FUNCTION__, getName().c_str(), identifier);
+ ALOGV("%s: Stream %s identifier 0x%X", __FUNCTION__, getName().c_str(), identifier);
return NO_ERROR;
}
@@ -41,7 +41,7 @@
return BAD_VALUE;
}
mVolumeProfile = volumeProfile;
- ALOGD("%s: 0x%X for Stream %s", __FUNCTION__, mVolumeProfile, getName().c_str());
+ ALOGV("%s: 0x%X for Stream %s", __FUNCTION__, mVolumeProfile, getName().c_str());
return NO_ERROR;
}
diff --git a/services/audiopolicy/engineconfigurable/tools/Android.bp b/services/audiopolicy/engineconfigurable/tools/Android.bp
index 8c16972..3e47324 100644
--- a/services/audiopolicy/engineconfigurable/tools/Android.bp
+++ b/services/audiopolicy/engineconfigurable/tools/Android.bp
@@ -16,14 +16,17 @@
name: "tools_default",
version: {
py2: {
- enabled: true,
+ enabled: false,
},
py3: {
- enabled: false,
+ enabled: true,
},
},
}
+//##################################################################################################
+// Tools for audio policy engine criterion type configuration file
+//
python_binary_host {
name: "buildPolicyCriterionTypes.py",
main: "buildPolicyCriterionTypes.py",
@@ -33,6 +36,30 @@
defaults: ["tools_default"],
}
+genrule_defaults {
+ name: "buildpolicycriteriontypesrule",
+ tools: ["buildPolicyCriterionTypes.py"],
+ cmd: "cp $(locations :audio_policy_configuration_files) $(genDir)/. && " +
+ "cp $(location :audio_policy_configuration_top_file) $(genDir)/audio_policy_configuration.xml && " +
+ "$(location buildPolicyCriterionTypes.py) " +
+ // @todo update if 1428659 is merged "--androidaudiobaseheader $(location :android_audio_base_header_file) " +
+ " --androidaudiobaseheader system/media/audio/include/system/audio-base.h " +
+ "--audiopolicyconfigurationfile $(genDir)/audio_policy_configuration.xml " +
+ "--criteriontypes $(location :audio_policy_engine_criterion_types_template) " +
+ "--outputfile $(out)",
+ srcs: [
+ // The commented inputs must be provided to use this genrule_defaults
+ // @todo uncomment if 1428659 is merged":android_audio_base_header_file",
+ ":audio_policy_engine_criterion_types_template",
+ // ":audio_policy_configuration_top_file",
+ // ":audio_policy_configuration_files",
+ ],
+ out: ["audio_policy_engine_criterion_types.xml"],
+}
+
+//##################################################################################################
+// Tools for audio policy engine parameter framework configurable domains
+//
python_binary_host {
name: "domainGeneratorPolicy.py",
main: "domainGeneratorPolicy.py",
@@ -50,6 +77,38 @@
],
}
+genrule_defaults {
+ name: "domaingeneratorpolicyrule",
+ tools: [
+ "domainGeneratorPolicy.py",
+ "domainGeneratorConnector",
+ ],
+ cmd: "mkdir -p $(genDir)/Structure/Policy && " +
+ "cp $(locations :audio_policy_pfw_structure_files) $(genDir)/Structure/Policy && " +
+ "cp $(location :audio_policy_pfw_toplevel) $(genDir)/top_level && " +
+ "$(location domainGeneratorPolicy.py) " +
+ "--validate " +
+ "--domain-generator-tool $(location domainGeneratorConnector) " +
+ "--toplevel-config $(genDir)/top_level " +
+ "--criteria $(location :audio_policy_engine_criteria) " +
+ "--criteriontypes $(location :audio_policy_engine_criterion_types) " +
+ "--add-edds $(locations :edd_files) " +
+ "--schemas-dir external/parameter-framework/upstream/schemas " +
+ " > $(out)",
+ srcs: [
+ // The commented inputs must be provided to use this genrule_defaults
+ // ":audio_policy_pfw_toplevel",
+ // ":audio_policy_pfw_structure_files",
+ ":audio_policy_engine_criteria",
+ // ":audio_policy_engine_criterion_types",
+ // ":edd_files",
+ ],
+ out: ["PolicyConfigurableDomains.xml"],
+}
+
+//##################################################################################################
+// Tools for policy parameter-framework product strategies structure file generation
+//
python_binary_host {
name: "buildStrategiesStructureFile.py",
main: "buildStrategiesStructureFile.py",
@@ -58,3 +117,45 @@
],
defaults: ["tools_default"],
}
+
+genrule_defaults {
+ name: "buildstrategiesstructurerule",
+ tools: ["buildStrategiesStructureFile.py"],
+ cmd: "cp $(locations :audio_policy_engine_configuration_files) $(genDir) && ls -l $(genDir) &&"+
+ "$(location buildStrategiesStructureFile.py) " +
+ "--audiopolicyengineconfigurationfile $(genDir)/audio_policy_engine_configuration.xml "+
+ "--productstrategiesstructurefile $(location :product_strategies_structure_template) " +
+ "--outputfile $(out)",
+ srcs: [
+ // The commented inputs must be provided to use this genrule_defaults
+ // ":audio_policy_engine_configuration_files",
+ ":product_strategies_structure_template",
+ ],
+ out: ["ProductStrategies.xml"],
+}
+
+//##################################################################################################
+// Tools for policy parameter-framework common type structure file generation
+//
+python_binary_host {
+ name: "buildCommonTypesStructureFile.py",
+ main: "buildCommonTypesStructureFile.py",
+ srcs: [
+ "buildCommonTypesStructureFile.py",
+ ],
+ defaults: ["tools_default"],
+}
+
+genrule_defaults {
+ name: "buildcommontypesstructurerule",
+ tools: ["buildCommonTypesStructureFile.py"],
+ cmd: "$(location buildCommonTypesStructureFile.py) " +
+ "--androidaudiobaseheader $(location :libaudio_system_audio_base) " +
+ "--commontypesstructure $(location :common_types_structure_template) " +
+ "--outputfile $(out)",
+ srcs: [
+ ":common_types_structure_template",
+ ":libaudio_system_audio_base",
+ ],
+ out: ["PolicySubsystem-CommonTypes.xml"],
+}
diff --git a/services/audiopolicy/engineconfigurable/tools/buildCommonTypesStructureFile.py b/services/audiopolicy/engineconfigurable/tools/buildCommonTypesStructureFile.py
new file mode 100755
index 0000000..9a7fa8f
--- /dev/null
+++ b/services/audiopolicy/engineconfigurable/tools/buildCommonTypesStructureFile.py
@@ -0,0 +1,184 @@
+#! /usr/bin/python3
+#
+# pylint: disable=line-too-long, missing-docstring, logging-format-interpolation, invalid-name
+
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+# http://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+
+import argparse
+import re
+import sys
+import os
+import logging
+import xml.etree.ElementTree as ET
+from collections import OrderedDict
+import xml.dom.minidom as MINIDOM
+
+def parseArgs():
+ argparser = argparse.ArgumentParser(description="Parameter-Framework XML \
+ structure file generator.\n\
+ Exit with the number of (recoverable or not) error that occured.")
+ argparser.add_argument('--androidaudiobaseheader',
+ help="Android Audio Base C header file, Mandatory.",
+ metavar="ANDROID_AUDIO_BASE_HEADER",
+ type=argparse.FileType('r'),
+ required=True)
+ argparser.add_argument('--commontypesstructure',
+ help="Structure XML base file. Mandatory.",
+ metavar="STRUCTURE_FILE_IN",
+ type=argparse.FileType('r'),
+ required=True)
+ argparser.add_argument('--outputfile',
+ help="Structure XML file. Mandatory.",
+ metavar="STRUCTURE_FILE_OUT",
+ type=argparse.FileType('w'),
+ required=True)
+ argparser.add_argument('--verbose',
+ action='store_true')
+
+ return argparser.parse_args()
+
+
+def findBitPos(decimal):
+ pos = 0
+ i = 1
+ while i != decimal:
+ i = i << 1
+ pos = pos + 1
+ if pos == 32:
+ return -1
+ return pos
+
+
+def generateXmlStructureFile(componentTypeDict, structureTypesFile, outputFile):
+
+ logging.info("Importing structureTypesFile {}".format(structureTypesFile))
+ component_types_in_tree = ET.parse(structureTypesFile)
+
+ component_types_root = component_types_in_tree.getroot()
+
+ for component_types_name, values_dict in componentTypeDict.items():
+ for component_type in component_types_root.findall('ComponentType'):
+ if component_type.get('Name') == component_types_name:
+ bitparameters_node = component_type.find("BitParameterBlock")
+ if bitparameters_node is not None:
+ ordered_values = OrderedDict(sorted(values_dict.items(), key=lambda x: x[1]))
+ for key, value in ordered_values.items():
+ value_node = ET.SubElement(bitparameters_node, "BitParameter")
+ value_node.set('Name', key)
+ value_node.set('Size', "1")
+ value_node.set('Pos', str(findBitPos(value)))
+
+ enum_parameter_node = component_type.find("EnumParameter")
+ if enum_parameter_node is not None:
+ ordered_values = OrderedDict(sorted(values_dict.items(), key=lambda x: x[1]))
+ for key, value in ordered_values.items():
+ value_node = ET.SubElement(enum_parameter_node, "ValuePair")
+ value_node.set('Literal', key)
+ value_node.set('Numerical', str(value))
+
+ xmlstr = ET.tostring(component_types_root, encoding='utf8', method='xml')
+ reparsed = MINIDOM.parseString(xmlstr)
+ prettyXmlStr = reparsed.toprettyxml(indent=" ", newl='\n')
+ prettyXmlStr = os.linesep.join([s for s in prettyXmlStr.splitlines() if s.strip()])
+ outputFile.write(prettyXmlStr)
+
+
+def capitalizeLine(line):
+ return ' '.join((w.capitalize() for w in line.split(' ')))
+
+def parseAndroidAudioFile(androidaudiobaseheaderFile):
+ #
+ # Adaptation table between Android Enumeration prefix and Audio PFW Criterion type names
+ #
+ component_type_mapping_table = {
+ 'AUDIO_STREAM' : "VolumeProfileType",
+ 'AUDIO_DEVICE_OUT' : "OutputDevicesMask",
+ 'AUDIO_DEVICE_IN' : "InputDevicesMask"}
+
+ all_component_types = {
+ 'VolumeProfileType' : {},
+ 'OutputDevicesMask' : {},
+ 'InputDevicesMask' : {}
+ }
+
+ #
+ # _CNT, _MAX, _ALL and _NONE are prohibited values as ther are just helpers for enum users.
+ #
+ ignored_values = ['CNT', 'MAX', 'ALL', 'NONE']
+
+ criteria_pattern = re.compile(
+ r"\s*(?P<type>(?:"+'|'.join(component_type_mapping_table.keys()) + "))_" \
+ r"(?P<literal>(?!" + '|'.join(ignored_values) + ")\w*)\s*=\s*" \
+ r"(?P<values>(?:0[xX])?[0-9a-fA-F]+)")
+
+ logging.info("Checking Android Header file {}".format(androidaudiobaseheaderFile))
+
+ for line_number, line in enumerate(androidaudiobaseheaderFile):
+ match = criteria_pattern.match(line)
+ if match:
+ logging.debug("The following line is VALID: {}:{}\n{}".format(
+ androidaudiobaseheaderFile.name, line_number, line))
+
+ component_type_name = component_type_mapping_table[match.groupdict()['type']]
+ component_type_literal = match.groupdict()['literal'].lower()
+
+ component_type_numerical_value = match.groupdict()['values']
+
+ # for AUDIO_DEVICE_IN: need to remove sign bit / rename default to stub
+ if component_type_name == "InputDevicesMask":
+ component_type_numerical_value = str(int(component_type_numerical_value, 0) & ~2147483648)
+ if component_type_literal == "default":
+ component_type_literal = "stub"
+
+ if component_type_name == "OutputDevicesMask":
+ if component_type_literal == "default":
+ component_type_literal = "stub"
+
+ # Remove duplicated numerical values
+ if int(component_type_numerical_value, 0) in all_component_types[component_type_name].values():
+ logging.info("The value {}:{} is duplicated for criterion {}, KEEPING LATEST".format(component_type_numerical_value, component_type_literal, component_type_name))
+ for key in list(all_component_types[component_type_name]):
+ if all_component_types[component_type_name][key] == int(component_type_numerical_value, 0):
+ del all_component_types[component_type_name][key]
+
+ all_component_types[component_type_name][component_type_literal] = int(component_type_numerical_value, 0)
+
+ logging.debug("type:{}, literal:{}, values:{}.".format(component_type_name, component_type_literal, component_type_numerical_value))
+
+ # Transform input source in inclusive criterion
+ shift = len(all_component_types['OutputDevicesMask'])
+ if shift > 32:
+ logging.critical("OutputDevicesMask incompatible with criterion representation on 32 bits")
+ logging.info("EXIT ON FAILURE")
+ exit(1)
+
+ for component_types in all_component_types:
+ values = ','.join('{}:{}'.format(value, key) for key, value in all_component_types[component_types].items())
+ logging.info("{}: <{}>".format(component_types, values))
+
+ return all_component_types
+
+
+def main():
+ logging.root.setLevel(logging.INFO)
+ args = parseArgs()
+ route_criteria = 0
+
+ all_component_types = parseAndroidAudioFile(args.androidaudiobaseheader)
+
+ generateXmlStructureFile(all_component_types, args.commontypesstructure, args.outputfile)
+
+# If this file is directly executed
+if __name__ == "__main__":
+ sys.exit(main())
diff --git a/services/audiopolicy/engineconfigurable/tools/buildPolicyCriterionTypes.py b/services/audiopolicy/engineconfigurable/tools/buildPolicyCriterionTypes.py
index a63c858..b8b60c1 100755
--- a/services/audiopolicy/engineconfigurable/tools/buildPolicyCriterionTypes.py
+++ b/services/audiopolicy/engineconfigurable/tools/buildPolicyCriterionTypes.py
@@ -1,4 +1,4 @@
-#!/usr/bin/python
+#!/usr/bin/python3
#
# Copyright 2018, The Android Open Source Project
@@ -19,10 +19,8 @@
import argparse
import re
import sys
-import tempfile
import os
import logging
-import subprocess
import xml.etree.ElementTree as ET
import xml.etree.ElementInclude as EI
import xml.dom.minidom as MINIDOM
@@ -49,33 +47,35 @@
def parseArgs():
argparser = argparse.ArgumentParser(description="Parameter-Framework XML \
- audio criterion type file generator.\n\
- Exit with the number of (recoverable or not) error that occured.")
+ audio criterion type file generator.\n\
+ Exit with the number of (recoverable or not) \
+ error that occured.")
argparser.add_argument('--androidaudiobaseheader',
- help="Android Audio Base C header file, Mandatory.",
- metavar="ANDROID_AUDIO_BASE_HEADER",
- type=argparse.FileType('r'),
- required=True)
+ help="Android Audio Base C header file, Mandatory.",
+ metavar="ANDROID_AUDIO_BASE_HEADER",
+ type=argparse.FileType('r'),
+ required=True)
argparser.add_argument('--audiopolicyconfigurationfile',
- help="Android Audio Policy Configuration file, Mandatory.",
- metavar="(AUDIO_POLICY_CONFIGURATION_FILE)",
- type=argparse.FileType('r'),
- required=True)
+ help="Android Audio Policy Configuration file, Mandatory.",
+ metavar="(AUDIO_POLICY_CONFIGURATION_FILE)",
+ type=argparse.FileType('r'),
+ required=True)
argparser.add_argument('--criteriontypes',
- help="Criterion types XML base file, in \
- '<criterion_types> \
- <criterion_type name="" type=<inclusive|exclusive> values=<value1,value2,...>/>' \
- format. Mandatory.",
- metavar="CRITERION_TYPE_FILE",
- type=argparse.FileType('r'),
- required=True)
+ help="Criterion types XML base file, in \
+ '<criterion_types> \
+ <criterion_type name="" type=<inclusive|exclusive> \
+ values=<value1,value2,...>/>' \
+ format. Mandatory.",
+ metavar="CRITERION_TYPE_FILE",
+ type=argparse.FileType('r'),
+ required=True)
argparser.add_argument('--outputfile',
- help="Criterion types outputfile file. Mandatory.",
- metavar="CRITERION_TYPE_OUTPUT_FILE",
- type=argparse.FileType('w'),
- required=True)
+ help="Criterion types outputfile file. Mandatory.",
+ metavar="CRITERION_TYPE_OUTPUT_FILE",
+ type=argparse.FileType('w'),
+ required=True)
argparser.add_argument('--verbose',
- action='store_true')
+ action='store_true')
return argparser.parse_args()
@@ -120,7 +120,7 @@
reparsed = MINIDOM.parseString(xmlstr)
prettyXmlStr = reparsed.toprettyxml(newl='\r\n')
prettyXmlStr = os.linesep.join([s for s in prettyXmlStr.splitlines() if s.strip()])
- outputFile.write(prettyXmlStr.encode('utf-8'))
+ outputFile.write(prettyXmlStr)
def capitalizeLine(line):
return ' '.join((w.capitalize() for w in line.split(' ')))
@@ -137,30 +137,30 @@
#
address_criteria_mapping_table = {
'sink' : "OutputDevicesAddressesType",
- 'source' : "InputDevicesAddressesType" }
+ 'source' : "InputDevicesAddressesType"}
address_criteria = {
'OutputDevicesAddressesType' : [],
- 'InputDevicesAddressesType' : [] }
+ 'InputDevicesAddressesType' : []}
- oldWorkingDir = os.getcwd()
- print "Current working directory %s" % oldWorkingDir
+ old_working_dir = os.getcwd()
+ print("Current working directory %s" % old_working_dir)
- newDir = os.path.join(oldWorkingDir , audiopolicyconfigurationfile.name)
+ new_dir = os.path.join(old_working_dir, audiopolicyconfigurationfile.name)
policy_in_tree = ET.parse(audiopolicyconfigurationfile)
- os.chdir(os.path.dirname(os.path.normpath(newDir)))
+ os.chdir(os.path.dirname(os.path.normpath(new_dir)))
- print "new working directory %s" % os.getcwd()
+ print("new working directory %s" % os.getcwd())
policy_root = policy_in_tree.getroot()
EI.include(policy_root)
- os.chdir(oldWorkingDir)
+ os.chdir(old_working_dir)
for device in policy_root.iter('devicePort'):
for key in address_criteria_mapping_table.keys():
- if device.get('role') == key and device.get('address') :
+ if device.get('role') == key and device.get('address'):
logging.info("{}: <{}>".format(key, device.get('address')))
address_criteria[address_criteria_mapping_table[key]].append(device.get('address'))
@@ -188,15 +188,15 @@
all_criteria = {
'AndroidModeType' : {},
'OutputDevicesMaskType' : {},
- 'InputDevicesMaskType' : {} }
+ 'InputDevicesMaskType' : {}}
#
# _CNT, _MAX, _ALL and _NONE are prohibited values as ther are just helpers for enum users.
#
- ignored_values = [ 'CNT', 'MAX', 'ALL', 'NONE' ]
+ ignored_values = ['CNT', 'MAX', 'ALL', 'NONE']
criteria_pattern = re.compile(
- r"\s*(?P<type>(?:"+'|'.join(criterion_mapping_table.keys()) + "))\_" \
+ r"\s*(?P<type>(?:"+'|'.join(criterion_mapping_table.keys()) + "))_" \
r"(?P<literal>(?!" + '|'.join(ignored_values) + ")\w*)\s*=\s*" \
r"(?P<values>(?:0[xX])?[0-9a-fA-F]+)")
@@ -221,7 +221,7 @@
logging.info("criterion {} duplicated values:".format(criterion_name))
logging.info("{}:{}".format(numerical_value, literal))
logging.info("KEEPING LATEST")
- for key in all_criteria[criterion_name].keys():
+ for key in list(all_criteria[criterion_name]):
if all_criteria[criterion_name][key] == int(numerical_value, 0):
del all_criteria[criterion_name][key]
diff --git a/services/audiopolicy/engineconfigurable/tools/buildStrategiesStructureFile.py b/services/audiopolicy/engineconfigurable/tools/buildStrategiesStructureFile.py
index af40602..f69d346 100755
--- a/services/audiopolicy/engineconfigurable/tools/buildStrategiesStructureFile.py
+++ b/services/audiopolicy/engineconfigurable/tools/buildStrategiesStructureFile.py
@@ -1,4 +1,4 @@
-#!/usr/bin/python
+#!/usr/bin/python3
#
# Copyright 2019, The Android Open Source Project
@@ -17,16 +17,12 @@
#
import argparse
-import re
import sys
-import tempfile
import os
import logging
-import subprocess
import xml.etree.ElementTree as ET
import xml.etree.ElementInclude as EI
import xml.dom.minidom as MINIDOM
-from collections import OrderedDict
#
# Helper script that helps to feed at build time the XML Product Strategies Structure file file used
@@ -46,33 +42,34 @@
def parseArgs():
argparser = argparse.ArgumentParser(description="Parameter-Framework XML \
- product strategies structure file generator.\n\
- Exit with the number of (recoverable or not) error that occured.")
+ product strategies structure file generator.\n\
+ Exit with the number of (recoverable or not) \
+ error that occured.")
argparser.add_argument('--audiopolicyengineconfigurationfile',
- help="Android Audio Policy Engine Configuration file, Mandatory.",
- metavar="(AUDIO_POLICY_ENGINE_CONFIGURATION_FILE)",
- type=argparse.FileType('r'),
- required=True)
+ help="Android Audio Policy Engine Configuration file, Mandatory.",
+ metavar="(AUDIO_POLICY_ENGINE_CONFIGURATION_FILE)",
+ type=argparse.FileType('r'),
+ required=True)
argparser.add_argument('--productstrategiesstructurefile',
- help="Product Strategies Structure XML base file, Mandatory.",
- metavar="STRATEGIES_STRUCTURE_FILE",
- type=argparse.FileType('r'),
- required=True)
+ help="Product Strategies Structure XML base file, Mandatory.",
+ metavar="STRATEGIES_STRUCTURE_FILE",
+ type=argparse.FileType('r'),
+ required=True)
argparser.add_argument('--outputfile',
- help="Product Strategies Structure output file, Mandatory.",
- metavar="STRATEGIES_STRUCTURE_OUTPUT_FILE",
- type=argparse.FileType('w'),
- required=True)
+ help="Product Strategies Structure output file, Mandatory.",
+ metavar="STRATEGIES_STRUCTURE_OUTPUT_FILE",
+ type=argparse.FileType('w'),
+ required=True)
argparser.add_argument('--verbose',
- action='store_true')
+ action='store_true')
return argparser.parse_args()
-def generateXmlStructureFile(strategies, strategyStructureInFile, outputFile):
+def generateXmlStructureFile(strategies, strategy_structure_in_file, output_file):
- logging.info("Importing strategyStructureInFile {}".format(strategyStructureInFile))
- strategies_in_tree = ET.parse(strategyStructureInFile)
+ logging.info("Importing strategy_structure_in_file {}".format(strategy_structure_in_file))
+ strategies_in_tree = ET.parse(strategy_structure_in_file)
strategies_root = strategies_in_tree.getroot()
strategy_components = strategies_root.find('ComponentType')
@@ -80,13 +77,15 @@
for strategy_name in strategies:
context_mapping = "".join(map(str, ["Name:", strategy_name]))
strategy_pfw_name = strategy_name.replace('STRATEGY_', '').lower()
- strategy_component_node = ET.SubElement(strategy_components, "Component", Name=strategy_pfw_name, Type="ProductStrategy", Mapping=context_mapping)
+ ET.SubElement(strategy_components, "Component",
+ Name=strategy_pfw_name, Type="ProductStrategy",
+ Mapping=context_mapping)
xmlstr = ET.tostring(strategies_root, encoding='utf8', method='xml')
reparsed = MINIDOM.parseString(xmlstr)
prettyXmlStr = reparsed.toprettyxml(newl='\r\n')
prettyXmlStr = os.linesep.join([s for s in prettyXmlStr.splitlines() if s.strip()])
- outputFile.write(prettyXmlStr.encode('utf-8'))
+ output_file.write(prettyXmlStr)
def capitalizeLine(line):
return ' '.join((w.capitalize() for w in line.split(' ')))
@@ -97,26 +96,27 @@
#
def parseAndroidAudioPolicyEngineConfigurationFile(audiopolicyengineconfigurationfile):
- logging.info("Checking Audio Policy Engine Configuration file {}".format(audiopolicyengineconfigurationfile))
+ logging.info("Checking Audio Policy Engine Configuration file {}".format(
+ audiopolicyengineconfigurationfile))
#
# extract all product strategies name from audio policy engine configuration file
#
strategy_names = []
- oldWorkingDir = os.getcwd()
- print "Current working directory %s" % oldWorkingDir
+ old_working_dir = os.getcwd()
+ print("Current working directory %s" % old_working_dir)
- newDir = os.path.join(oldWorkingDir , audiopolicyengineconfigurationfile.name)
+ new_dir = os.path.join(old_working_dir, audiopolicyengineconfigurationfile.name)
policy_engine_in_tree = ET.parse(audiopolicyengineconfigurationfile)
- os.chdir(os.path.dirname(os.path.normpath(newDir)))
+ os.chdir(os.path.dirname(os.path.normpath(new_dir)))
- print "new working directory %s" % os.getcwd()
+ print("new working directory %s" % os.getcwd())
policy_engine_root = policy_engine_in_tree.getroot()
EI.include(policy_engine_root)
- os.chdir(oldWorkingDir)
+ os.chdir(old_working_dir)
for strategy in policy_engine_root.iter('ProductStrategy'):
strategy_names.append(strategy.get('name'))
@@ -128,7 +128,8 @@
logging.root.setLevel(logging.INFO)
args = parseArgs()
- strategies = parseAndroidAudioPolicyEngineConfigurationFile(args.audiopolicyengineconfigurationfile)
+ strategies = parseAndroidAudioPolicyEngineConfigurationFile(
+ args.audiopolicyengineconfigurationfile)
product_strategies_structure = args.productstrategiesstructurefile
diff --git a/services/audiopolicy/engineconfigurable/tools/build_audio_pfw_settings.mk b/services/audiopolicy/engineconfigurable/tools/build_audio_pfw_settings.mk
deleted file mode 100644
index ac60ef7..0000000
--- a/services/audiopolicy/engineconfigurable/tools/build_audio_pfw_settings.mk
+++ /dev/null
@@ -1,38 +0,0 @@
-LOCAL_MODULE_CLASS := ETC
-LOCAL_MODULE_TAGS := optional
-LOCAL_ADDITIONAL_DEPENDENCIES += \
- $(HOST_OUT_EXECUTABLES)/domainGeneratorPolicy.py \
- $(PFW_TOPLEVEL_FILE) $(PFW_CRITERIA_FILE) $(PFW_CRITERION_TYPES_FILE)
-
-include $(BUILD_SYSTEM)/base_rules.mk
-
-$(LOCAL_BUILT_MODULE): MY_CRITERION_TYPES_FILE := $(PFW_CRITERION_TYPES_FILE)
-$(LOCAL_BUILT_MODULE): MY_TOOL := $(HOST_OUT_EXECUTABLES)/domainGeneratorPolicy.py
-$(LOCAL_BUILT_MODULE): MY_TOPLEVEL_FILE := $(PFW_TOPLEVEL_FILE)
-$(LOCAL_BUILT_MODULE): MY_CRITERIA_FILE := $(PFW_CRITERIA_FILE)
-$(LOCAL_BUILT_MODULE): MY_TUNING_FILE := $(PFW_TUNING_FILE)
-$(LOCAL_BUILT_MODULE): MY_EDD_FILES := $(PFW_EDD_FILES)
-$(LOCAL_BUILT_MODULE): MY_DOMAIN_FILES := $(PFW_DOMAIN_FILES)
-$(LOCAL_BUILT_MODULE): MY_SCHEMAS_DIR := $(PFW_SCHEMAS_DIR)
-$(LOCAL_BUILT_MODULE): MY_CRITERION_TYPES_FILE := $(PFW_CRITERION_TYPES_FILE)
-$(LOCAL_BUILT_MODULE): $(LOCAL_ADDITIONAL_DEPENDENCIES)
-
- "$(MY_TOOL)" --validate \
- --toplevel-config "$(MY_TOPLEVEL_FILE)" \
- --criteria "$(MY_CRITERIA_FILE)" \
- --criteriontypes "$(MY_CRITERION_TYPES_FILE)" \
- --initial-settings $(MY_TUNING_FILE) \
- --add-edds $(MY_EDD_FILES) \
- --add-domains $(MY_DOMAIN_FILES) \
- --schemas-dir $(MY_SCHEMAS_DIR) > "$@"
-
-
-# Clear variables for further use
-PFW_TOPLEVEL_FILE :=
-PFW_STRUCTURE_FILES :=
-PFW_CRITERIA_FILE :=
-PFW_CRITERION_TYPES_FILE :=
-PFW_TUNING_FILE :=
-PFW_EDD_FILES :=
-PFW_DOMAIN_FILES :=
-PFW_SCHEMAS_DIR := $(PFW_DEFAULT_SCHEMAS_DIR)
diff --git a/services/audiopolicy/engineconfigurable/tools/domainGeneratorPolicy.py b/services/audiopolicy/engineconfigurable/tools/domainGeneratorPolicy.py
index 4dec9a2..b0c4b66 100755
--- a/services/audiopolicy/engineconfigurable/tools/domainGeneratorPolicy.py
+++ b/services/audiopolicy/engineconfigurable/tools/domainGeneratorPolicy.py
@@ -1,4 +1,4 @@
-#!/usr/bin/python
+#!/usr/bin/python3
#
# Copyright 2018, The Android Open Source Project
@@ -16,12 +16,7 @@
# limitations under the License.
#
-import EddParser
-from PFWScriptGenerator import PfwScriptTranslator
-import hostConfig
-
import argparse
-import re
import sys
import tempfile
import os
@@ -29,6 +24,10 @@
import subprocess
import xml.etree.ElementTree as ET
+import EddParser
+from PFWScriptGenerator import PfwScriptTranslator
+import hostConfig
+
#
# In order to build the XML Settings file at build time, an instance of the parameter-framework
# shall be started and fed with all the criterion types/criteria that will be used by
@@ -39,61 +38,67 @@
def parseArgs():
argparser = argparse.ArgumentParser(description="Parameter-Framework XML \
- Settings file generator.\n\
- Exit with the number of (recoverable or not) error that occured.")
+ Settings file generator.\n\
+ Exit with the number of (recoverable or not) \
+ error that occured.")
+ argparser.add_argument('--domain-generator-tool',
+ help="ParameterFramework domain generator tool. Mandatory.",
+ metavar="PFW_DOMAIN_GENERATOR_TOOL",
+ required=True)
argparser.add_argument('--toplevel-config',
- help="Top-level parameter-framework configuration file. Mandatory.",
- metavar="TOPLEVEL_CONFIG_FILE",
- required=True)
+ help="Top-level parameter-framework configuration file. Mandatory.",
+ metavar="TOPLEVEL_CONFIG_FILE",
+ required=True)
argparser.add_argument('--criteria',
- help="Criteria file, in XML format: \
- in '<criteria> \
- <criterion name="" type=""/> \
- </criteria>' \
- format. Mandatory.",
- metavar="CRITERIA_FILE",
- type=argparse.FileType('r'),
- required=True)
+ help="Criteria file, in XML format: \
+ in '<criteria> \
+ <criterion name="" type=""/> \
+ </criteria>' \
+ format. Mandatory.",
+ metavar="CRITERIA_FILE",
+ type=argparse.FileType('r'),
+ required=True)
argparser.add_argument('--criteriontypes',
- help="Criterion types XML file, in \
- '<criterion_types> \
- <criterion_type name="" type=<inclusive|exclusive> values=<value1,value2,...>/> \
- </criterion_types>' \
- format. Mandatory.",
- metavar="CRITERION_TYPE_FILE",
- type=argparse.FileType('r'),
- required=False)
+ help="Criterion types XML file, in \
+ '<criterion_types> \
+ <criterion_type name="" type=<inclusive|exclusive> \
+ values=<value1,value2,...>/> \
+ </criterion_types>' \
+ format. Mandatory.",
+ metavar="CRITERION_TYPE_FILE",
+ type=argparse.FileType('r'),
+ required=False)
argparser.add_argument('--initial-settings',
- help="Initial XML settings file (containing a \
- <ConfigurableDomains> tag",
- nargs='?',
- default=None,
- metavar="XML_SETTINGS_FILE")
+ help="Initial XML settings file (containing a \
+ <ConfigurableDomains> tag",
+ nargs='?',
+ default=None,
+ metavar="XML_SETTINGS_FILE")
argparser.add_argument('--add-domains',
- help="List of single domain files (each containing a single \
- <ConfigurableDomain> tag",
- metavar="XML_DOMAIN_FILE",
- nargs='*',
- dest='xml_domain_files',
- default=[])
+ help="List of single domain files (each containing a single \
+ <ConfigurableDomain> tag",
+ metavar="XML_DOMAIN_FILE",
+ nargs='*',
+ dest='xml_domain_files',
+ default=[])
argparser.add_argument('--add-edds',
- help="List of files in EDD syntax (aka \".pfw\" files)",
- metavar="EDD_FILE",
- type=argparse.FileType('r'),
- nargs='*',
- default=[],
- dest='edd_files')
+ help="List of files in EDD syntax (aka \".pfw\" files)",
+ metavar="EDD_FILE",
+ type=argparse.FileType('r'),
+ nargs='*',
+ default=[],
+ dest='edd_files')
argparser.add_argument('--schemas-dir',
- help="Directory of parameter-framework XML Schemas for generation \
- validation",
- default=None)
+ help="Directory of parameter-framework XML Schemas for generation \
+ validation",
+ default=None)
argparser.add_argument('--target-schemas-dir',
- help="Ignored. Kept for retro-compatibility")
+ help="Ignored. Kept for retro-compatibility")
argparser.add_argument('--validate',
- help="Validate the settings against XML schemas",
- action='store_true')
+ help="Validate the settings against XML schemas",
+ action='store_true')
argparser.add_argument('--verbose',
- action='store_true')
+ action='store_true')
return argparser.parse_args()
@@ -112,7 +117,6 @@
logging.info("Importing criterionTypesFile {}".format(criterionTypesFile))
criteria_root = criteria_tree.getroot()
- criterion_types_root = criterion_types_tree.getroot()
all_criteria = []
for criterion in criteria_root.findall('criterion'):
@@ -165,7 +169,7 @@
try:
root.propagate()
- except EddParser.MyPropagationError, ex :
+ except EddParser.MyPropagationError as ex:
logging.critical(str(ex))
logging.info("EXIT ON FAILURE")
exit(1)
@@ -179,32 +183,32 @@
# It takes as input the collection of criteria, the domains and the simplified settings read from
# pfw.
#
-def generateDomainCommands(logging, all_criteria, initial_settings, xml_domain_files, parsed_edds):
- # create and inject all the criteria
- logging.info("Creating all criteria")
- for criterion in all_criteria:
- yield ["createSelectionCriterion", criterion['inclusive'],
- criterion['name']] + criterion['values']
+def generateDomainCommands(logger, all_criteria, initial_settings, xml_domain_files, parsed_edds):
+ # create and inject all the criteria
+ logger.info("Creating all criteria")
+ for criterion in all_criteria:
+ yield ["createSelectionCriterion", criterion['inclusive'],
+ criterion['name']] + criterion['values']
- yield ["start"]
+ yield ["start"]
- # Import initial settings file
- if initial_settings:
- logging.info("Importing initial settings file {}".format(initial_settings))
- yield ["importDomainsWithSettingsXML", initial_settings]
+ # Import initial settings file
+ if initial_settings:
+ logger.info("Importing initial settings file {}".format(initial_settings))
+ yield ["importDomainsWithSettingsXML", initial_settings]
- # Import each standalone domain files
- for domain_file in xml_domain_files:
- logging.info("Importing single domain file {}".format(domain_file))
- yield ["importDomainWithSettingsXML", domain_file]
+ # Import each standalone domain files
+ for domain_file in xml_domain_files:
+ logger.info("Importing single domain file {}".format(domain_file))
+ yield ["importDomainWithSettingsXML", domain_file]
- # Generate the script for each EDD file
- for filename, parsed_edd in parsed_edds:
- logging.info("Translating and injecting EDD file {}".format(filename))
- translator = PfwScriptTranslator()
- parsed_edd.translate(translator)
- for command in translator.getScript():
- yield command
+ # Generate the script for each EDD file
+ for filename, parsed_edd in parsed_edds:
+ logger.info("Translating and injecting EDD file {}".format(filename))
+ translator = PfwScriptTranslator()
+ parsed_edd.translate(translator)
+ for command in translator.getScript():
+ yield command
#
# Entry point of the domain generator.
@@ -232,30 +236,29 @@
prefix="TMPdomainGeneratorPFConfig_")
install_path = os.path.dirname(os.path.realpath(args.toplevel_config))
- hostConfig.configure(
- infile=args.toplevel_config,
- outfile=fake_toplevel_config,
- structPath=install_path)
+ hostConfig.configure(infile=args.toplevel_config,
+ outfile=fake_toplevel_config,
+ structPath=install_path)
fake_toplevel_config.close()
# Create the connector. Pipe its input to us in order to write commands;
# connect its output to stdout in order to have it dump the domains
# there; connect its error output to stderr.
- connector = subprocess.Popen(["domainGeneratorConnector",
- fake_toplevel_config.name,
- 'verbose' if args.verbose else 'no-verbose',
- 'validate' if args.validate else 'no-validate',
- args.schemas_dir],
- stdout=sys.stdout, stdin=subprocess.PIPE, stderr=sys.stderr)
+ connector = subprocess.Popen([args.domain_generator_tool,
+ fake_toplevel_config.name,
+ 'verbose' if args.verbose else 'no-verbose',
+ 'validate' if args.validate else 'no-validate',
+ args.schemas_dir],
+ stdout=sys.stdout, stdin=subprocess.PIPE, stderr=sys.stderr)
initial_settings = None
if args.initial_settings:
initial_settings = os.path.realpath(args.initial_settings)
for command in generateDomainCommands(logging, all_criteria, initial_settings,
- args.xml_domain_files, parsed_edds):
- connector.stdin.write('\0'.join(command))
- connector.stdin.write("\n")
+ args.xml_domain_files, parsed_edds):
+ connector.stdin.write('\0'.join(command).encode('utf-8'))
+ connector.stdin.write("\n".encode('utf-8'))
# Closing the connector's input triggers the domain generation
connector.stdin.close()
diff --git a/services/audiopolicy/engineconfigurable/tools/provision_criterion_types_from_android_headers.mk b/services/audiopolicy/engineconfigurable/tools/provision_criterion_types_from_android_headers.mk
deleted file mode 100644
index dab5a0f..0000000
--- a/services/audiopolicy/engineconfigurable/tools/provision_criterion_types_from_android_headers.mk
+++ /dev/null
@@ -1,25 +0,0 @@
-LOCAL_MODULE_CLASS := ETC
-LOCAL_MODULE_TAGS := optional
-LOCAL_ADDITIONAL_DEPENDENCIES += \
- $(HOST_OUT_EXECUTABLES)/buildPolicyCriterionTypes.py \
- $(CRITERION_TYPES_FILE) $(AUDIO_POLICY_CONFIGURATION_FILE) \
- $(ANDROID_AUDIO_BASE_HEADER_FILE)
-
-include $(BUILD_SYSTEM)/base_rules.mk
-
-$(LOCAL_BUILT_MODULE): MY_CRITERION_TYPES_FILE := $(CRITERION_TYPES_FILE)
-$(LOCAL_BUILT_MODULE): MY_ANDROID_AUDIO_BASE_HEADER_FILE := $(ANDROID_AUDIO_BASE_HEADER_FILE)
-$(LOCAL_BUILT_MODULE): MY_AUDIO_POLICY_CONFIGURATION_FILE := $(AUDIO_POLICY_CONFIGURATION_FILE)
-$(LOCAL_BUILT_MODULE): MY_CRITERION_TOOL := $(HOST_OUT_EXECUTABLES)/buildPolicyCriterionTypes.py
-$(LOCAL_BUILT_MODULE): $(LOCAL_ADDITIONAL_DEPENDENCIES)
-
- "$(MY_CRITERION_TOOL)" \
- --androidaudiobaseheader "$(MY_ANDROID_AUDIO_BASE_HEADER_FILE)" \
- --audiopolicyconfigurationfile "$(MY_AUDIO_POLICY_CONFIGURATION_FILE)" \
- --criteriontypes "$(MY_CRITERION_TYPES_FILE)" \
- --outputfile "$(@)"
-
-# Clear variables for further use
-CRITERION_TYPES_FILE :=
-ANDROID_AUDIO_BASE_HEADER_FILE :=
-AUDIO_POLICY_CONFIGURATION_FILE :=
diff --git a/services/audiopolicy/engineconfigurable/tools/provision_strategies_structure.mk b/services/audiopolicy/engineconfigurable/tools/provision_strategies_structure.mk
deleted file mode 100644
index f2b1a19..0000000
--- a/services/audiopolicy/engineconfigurable/tools/provision_strategies_structure.mk
+++ /dev/null
@@ -1,21 +0,0 @@
-LOCAL_MODULE_CLASS := ETC
-LOCAL_MODULE_TAGS := optional
-LOCAL_ADDITIONAL_DEPENDENCIES += \
- $(HOST_OUT_EXECUTABLES)/buildStrategiesStructureFile.py \
- $(STRATEGIES_STRUCTURE_FILE) $(AUDIO_POLICY_ENGINE_CONFIGURATION_FILE)
-
-include $(BUILD_SYSTEM)/base_rules.mk
-
-$(LOCAL_BUILT_MODULE): MY_STRATEGIES_STRUCTURE_FILE := $(STRATEGIES_STRUCTURE_FILE)
-$(LOCAL_BUILT_MODULE): MY_AUDIO_POLICY_ENGINE_CONFIGURATION_FILE := $(AUDIO_POLICY_ENGINE_CONFIGURATION_FILE)
-$(LOCAL_BUILT_MODULE): MY_PROVISION_TOOL := $(HOST_OUT_EXECUTABLES)/buildStrategiesStructureFile.py
-$(LOCAL_BUILT_MODULE): $(LOCAL_ADDITIONAL_DEPENDENCIES)
-
- "$(MY_PROVISION_TOOL)" \
- --audiopolicyengineconfigurationfile "$(MY_AUDIO_POLICY_ENGINE_CONFIGURATION_FILE)" \
- --productstrategiesstructurefile "$(MY_STRATEGIES_STRUCTURE_FILE)" \
- --outputfile "$(@)"
-
-# Clear variables for further use
-STRATEGIES_STRUCTURE_FILE :=
-AUDIO_POLICY_ENGINE_CONFIGURATION_FILE :=
diff --git a/services/audiopolicy/engineconfigurable/wrapper/Android.bp b/services/audiopolicy/engineconfigurable/wrapper/Android.bp
index 6f59487..301ecc0 100644
--- a/services/audiopolicy/engineconfigurable/wrapper/Android.bp
+++ b/services/audiopolicy/engineconfigurable/wrapper/Android.bp
@@ -11,7 +11,6 @@
"libbase_headers",
"libaudiopolicycommon",
],
- static_libs: ["libaudiopolicycomponents"],
shared_libs: [
"liblog",
"libutils",
diff --git a/services/audiopolicy/engineconfigurable/wrapper/ParameterManagerWrapper.cpp b/services/audiopolicy/engineconfigurable/wrapper/ParameterManagerWrapper.cpp
index 4b57444..63990ac 100644
--- a/services/audiopolicy/engineconfigurable/wrapper/ParameterManagerWrapper.cpp
+++ b/services/audiopolicy/engineconfigurable/wrapper/ParameterManagerWrapper.cpp
@@ -92,7 +92,8 @@
template <>
struct ParameterManagerWrapper::parameterManagerElementSupported<ISelectionCriterionTypeInterface> {};
-ParameterManagerWrapper::ParameterManagerWrapper()
+ParameterManagerWrapper::ParameterManagerWrapper(bool enableSchemaVerification,
+ const std::string &schemaUri)
: mPfwConnectorLogger(new ParameterMgrPlatformConnectorLogger)
{
// Connector
@@ -104,6 +105,15 @@
// Logger
mPfwConnector->setLogger(mPfwConnectorLogger);
+
+ // Schema validation
+ std::string error;
+ bool ret = mPfwConnector->setValidateSchemasOnStart(enableSchemaVerification, error);
+ ALOGE_IF(!ret, "Failed to activate schema validation: %s", error.c_str());
+ if (enableSchemaVerification && ret && !schemaUri.empty()) {
+ ALOGE("Schema verification activated with schema URI: %s", schemaUri.c_str());
+ mPfwConnector->setSchemaUri(schemaUri);
+ }
}
status_t ParameterManagerWrapper::addCriterion(const std::string &name, bool isInclusive,
@@ -145,11 +155,10 @@
delete mPfwConnector;
}
-status_t ParameterManagerWrapper::start()
+status_t ParameterManagerWrapper::start(std::string &error)
{
ALOGD("%s: in", __FUNCTION__);
/// Start PFW
- std::string error;
if (!mPfwConnector->start(error)) {
ALOGE("%s: Policy PFW start error: %s", __FUNCTION__, error.c_str());
return NO_INIT;
@@ -253,13 +262,13 @@
return interface->getLiteralValue(valueToCheck, literalValue);
}
-status_t ParameterManagerWrapper::setDeviceConnectionState(const sp<DeviceDescriptor> devDesc,
- audio_policy_dev_state_t state)
+status_t ParameterManagerWrapper::setDeviceConnectionState(
+ audio_devices_t type, const std::string address, audio_policy_dev_state_t state)
{
- std::string criterionName = audio_is_output_device(devDesc->type()) ?
+ std::string criterionName = audio_is_output_device(type) ?
gOutputDeviceAddressCriterionName : gInputDeviceAddressCriterionName;
- ALOGV("%s: device with address %s %s", __FUNCTION__, devDesc->address().string(),
+ ALOGV("%s: device with address %s %s", __FUNCTION__, address.c_str(),
state != AUDIO_POLICY_DEVICE_STATE_AVAILABLE? "disconnected" : "connected");
ISelectionCriterionInterface *criterion =
getElement<ISelectionCriterionInterface>(criterionName, mPolicyCriteria);
@@ -271,8 +280,9 @@
auto criterionType = criterion->getCriterionType();
int deviceAddressId;
- if (not criterionType->getNumericalValue(devDesc->address().string(), deviceAddressId)) {
- ALOGW("%s: unknown device address reported (%s)", __FUNCTION__, devDesc->address().c_str());
+ if (not criterionType->getNumericalValue(address.c_str(), deviceAddressId)) {
+ ALOGW("%s: unknown device address reported (%s) for criterion %s", __FUNCTION__,
+ address.c_str(), criterionName.c_str());
return BAD_TYPE;
}
int currentValueMask = criterion->getCriterionState();
diff --git a/services/audiopolicy/engineconfigurable/wrapper/include/ParameterManagerWrapper.h b/services/audiopolicy/engineconfigurable/wrapper/include/ParameterManagerWrapper.h
index 5bfad29..62b129a 100644
--- a/services/audiopolicy/engineconfigurable/wrapper/include/ParameterManagerWrapper.h
+++ b/services/audiopolicy/engineconfigurable/wrapper/include/ParameterManagerWrapper.h
@@ -16,10 +16,6 @@
#pragma once
-#include <AudioGain.h>
-#include <AudioPort.h>
-#include <HwModule.h>
-#include <DeviceDescriptor.h>
#include <system/audio.h>
#include <system/audio_policy.h>
#include <utils/Errors.h>
@@ -48,16 +44,18 @@
using Criteria = std::map<std::string, ISelectionCriterionInterface *>;
public:
- ParameterManagerWrapper();
+ ParameterManagerWrapper(bool enableSchemaVerification = false,
+ const std::string &schemaUri = {});
~ParameterManagerWrapper();
/**
* Starts the platform state service.
* It starts the parameter framework policy instance.
+ * @param[out] contains human readable error if starts failed
*
- * @return NO_ERROR if success, error code otherwise.
+ * @return NO_ERROR if success, error code otherwise, and error is set to human readable string.
*/
- status_t start();
+ status_t start(std::string &error);
/**
* The following API wrap policy action to criteria
@@ -118,7 +116,15 @@
*/
status_t setAvailableOutputDevices(audio_devices_t outputDevices);
- status_t setDeviceConnectionState(const sp<DeviceDescriptor> devDesc,
+ /**
+ * @brief setDeviceConnectionState propagates a state event on a given device(s)
+ * @param type bit mask of the device whose state has changed
+ * @param address of the device whose state has changed
+ * @param state new state of the given device
+ * @return NO_ERROR if new state corretly propagated to Engine Parameter-Framework, error
+ * code otherwise.
+ */
+ status_t setDeviceConnectionState(audio_devices_t type, const std::string address,
audio_policy_dev_state_t state);
/**
diff --git a/services/audiopolicy/enginedefault/Android.bp b/services/audiopolicy/enginedefault/Android.bp
index 7b42c6a..aaf4158 100644
--- a/services/audiopolicy/enginedefault/Android.bp
+++ b/services/audiopolicy/enginedefault/Android.bp
@@ -1,16 +1,15 @@
cc_library_shared {
name: "libaudiopolicyenginedefault",
- export_include_dirs: ["include"],
srcs: [
"src/Engine.cpp",
"src/EngineInstance.cpp",
],
cflags: [
+ "-fvisibility=hidden",
"-Wall",
"-Werror",
"-Wextra",
],
- local_include_dirs: ["include"],
header_libs: [
"libbase_headers",
"libaudiopolicycommon",
@@ -22,6 +21,7 @@
"libaudiopolicyengine_config",
],
shared_libs: [
+ "libaudiofoundation",
"liblog",
"libcutils",
"libutils",
diff --git a/services/audiopolicy/enginedefault/config/example/Android.bp b/services/audiopolicy/enginedefault/config/example/Android.bp
new file mode 100644
index 0000000..0bfcaa1
--- /dev/null
+++ b/services/audiopolicy/enginedefault/config/example/Android.bp
@@ -0,0 +1,46 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+// Import this namespace in order to use AOSP Phone with Default Engine configuration example
+
+soong_namespace {
+}
+
+prebuilt_etc {
+ name: "audio_policy_engine_configuration.xml",
+ vendor: true,
+ src: "phone/audio_policy_engine_configuration.xml",
+ required: [
+ ":audio_policy_engine_stream_volumes.xml",
+ ":audio_policy_engine_default_stream_volumes.xml",
+ ":audio_policy_engine_product_strategies.xml",
+ ],
+}
+prebuilt_etc {
+ name: "audio_policy_engine_product_strategies.xml",
+ vendor: true,
+ src: "phone/audio_policy_engine_product_strategies.xml",
+}
+prebuilt_etc {
+ name: "audio_policy_engine_stream_volumes.xml",
+ vendor: true,
+ src: "phone/audio_policy_engine_stream_volumes.xml",
+}
+prebuilt_etc {
+ name: "audio_policy_engine_default_stream_volumes.xml",
+ vendor: true,
+ src: "phone/audio_policy_engine_default_stream_volumes.xml",
+}
diff --git a/services/audiopolicy/enginedefault/config/example/Android.mk b/services/audiopolicy/enginedefault/config/example/Android.mk
deleted file mode 100644
index 0badac8..0000000
--- a/services/audiopolicy/enginedefault/config/example/Android.mk
+++ /dev/null
@@ -1,48 +0,0 @@
-LOCAL_PATH := $(call my-dir)
-
-##################################################################
-# CONFIGURATION TOP FILE
-##################################################################
-
-ifeq ($(BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION), phone_default)
-
-include $(CLEAR_VARS)
-LOCAL_MODULE := audio_policy_engine_configuration.xml
-
-LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE_CLASS := ETC
-LOCAL_VENDOR_MODULE := true
-LOCAL_SRC_FILES := phone/$(LOCAL_MODULE)
-
-LOCAL_REQUIRED_MODULES := \
- audio_policy_engine_product_strategies.xml \
- audio_policy_engine_stream_volumes.xml \
- audio_policy_engine_default_stream_volumes.xml
-
-include $(BUILD_PREBUILT)
-
-include $(CLEAR_VARS)
-LOCAL_MODULE := audio_policy_engine_product_strategies.xml
-LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE_CLASS := ETC
-LOCAL_VENDOR_MODULE := true
-LOCAL_SRC_FILES := phone/$(LOCAL_MODULE)
-include $(BUILD_PREBUILT)
-
-include $(CLEAR_VARS)
-LOCAL_MODULE := audio_policy_engine_stream_volumes.xml
-LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE_CLASS := ETC
-LOCAL_VENDOR_MODULE := true
-LOCAL_SRC_FILES := phone/$(LOCAL_MODULE)
-include $(BUILD_PREBUILT)
-
-include $(CLEAR_VARS)
-LOCAL_MODULE := audio_policy_engine_default_stream_volumes.xml
-LOCAL_MODULE_TAGS := optional
-LOCAL_MODULE_CLASS := ETC
-LOCAL_VENDOR_MODULE := true
-LOCAL_SRC_FILES := phone/$(LOCAL_MODULE)
-include $(BUILD_PREBUILT)
-
-endif # ifeq ($(BUILD_AUDIO_POLICY_EXAMPLE_CONFIGURATION), phone_default)
diff --git a/services/audiopolicy/enginedefault/config/example/phone/audio_policy_engine_product_strategies.xml b/services/audiopolicy/enginedefault/config/example/phone/audio_policy_engine_product_strategies.xml
index 9398743..a7388da 100644
--- a/services/audiopolicy/enginedefault/config/example/phone/audio_policy_engine_product_strategies.xml
+++ b/services/audiopolicy/enginedefault/config/example/phone/audio_policy_engine_product_strategies.xml
@@ -72,6 +72,12 @@
<Attributes> <Usage value="AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE"/> </Attributes>
<Attributes></Attributes>
</AttributesGroup>
+ <AttributesGroup streamType="AUDIO_STREAM_ASSISTANT" volumeGroup="assistant">
+ <Attributes>
+ <ContentType value="AUDIO_CONTENT_TYPE_SPEECH"/>
+ <Usage value="AUDIO_USAGE_ASSISTANT"/>
+ </Attributes>
+ </AttributesGroup>
<AttributesGroup streamType="AUDIO_STREAM_SYSTEM" volumeGroup="system">
<Attributes> <Usage value="AUDIO_USAGE_ASSISTANCE_SONIFICATION"/> </Attributes>
</AttributesGroup>
@@ -91,20 +97,5 @@
</AttributesGroup>
</ProductStrategy>
- <!-- Routing Strategy rerouting may be removed as following media??? -->
- <ProductStrategy name="STRATEGY_REROUTING">
- <AttributesGroup streamType="AUDIO_STREAM_REROUTING" volumeGroup="rerouting">
- <Attributes></Attributes>
- </AttributesGroup>
- </ProductStrategy>
-
- <!-- Default product strategy has empty attributes -->
- <ProductStrategy name="STRATEGY_PATCH">
- <AttributesGroup streamType="AUDIO_STREAM_PATCH" volumeGroup="patch">
- <Attributes></Attributes>
- </AttributesGroup>
- </ProductStrategy>
-
-
</ProductStrategies>
diff --git a/services/audiopolicy/enginedefault/config/example/phone/audio_policy_engine_stream_volumes.xml b/services/audiopolicy/enginedefault/config/example/phone/audio_policy_engine_stream_volumes.xml
index 707a184..d5c3896 100644
--- a/services/audiopolicy/enginedefault/config/example/phone/audio_policy_engine_stream_volumes.xml
+++ b/services/audiopolicy/enginedefault/config/example/phone/audio_policy_engine_stream_volumes.xml
@@ -207,25 +207,15 @@
</volumeGroup>
<volumeGroup>
- <name>rerouting</name>
+ <name>assistant</name>
<indexMin>0</indexMin>
- <indexMax>1</indexMax>
- <volume deviceCategory="DEVICE_CATEGORY_HEADSET" ref="FULL_SCALE_VOLUME_CURVE"/>
- <volume deviceCategory="DEVICE_CATEGORY_SPEAKER" ref="FULL_SCALE_VOLUME_CURVE"/>
- <volume deviceCategory="DEVICE_CATEGORY_EARPIECE" ref="FULL_SCALE_VOLUME_CURVE"/>
- <volume deviceCategory="DEVICE_CATEGORY_EXT_MEDIA" ref="FULL_SCALE_VOLUME_CURVE"/>
- <volume deviceCategory="DEVICE_CATEGORY_HEARING_AID" ref="FULL_SCALE_VOLUME_CURVE"/>
+ <indexMax>15</indexMax>
+ <volume deviceCategory="DEVICE_CATEGORY_HEADSET" ref="DEFAULT_MEDIA_VOLUME_CURVE"/>
+ <volume deviceCategory="DEVICE_CATEGORY_SPEAKER" ref="DEFAULT_DEVICE_CATEGORY_SPEAKER_VOLUME_CURVE"/>
+ <volume deviceCategory="DEVICE_CATEGORY_EARPIECE" ref="DEFAULT_MEDIA_VOLUME_CURVE"/>
+ <volume deviceCategory="DEVICE_CATEGORY_EXT_MEDIA" ref="DEFAULT_MEDIA_VOLUME_CURVE"/>
+ <volume deviceCategory="DEVICE_CATEGORY_HEARING_AID" ref="DEFAULT_HEARING_AID_VOLUME_CURVE"/>
</volumeGroup>
- <volumeGroup>
- <name>patch</name>
- <indexMin>0</indexMin>
- <indexMax>1</indexMax>
- <volume deviceCategory="DEVICE_CATEGORY_HEADSET" ref="FULL_SCALE_VOLUME_CURVE"/>
- <volume deviceCategory="DEVICE_CATEGORY_SPEAKER" ref="FULL_SCALE_VOLUME_CURVE"/>
- <volume deviceCategory="DEVICE_CATEGORY_EARPIECE" ref="FULL_SCALE_VOLUME_CURVE"/>
- <volume deviceCategory="DEVICE_CATEGORY_EXT_MEDIA" ref="FULL_SCALE_VOLUME_CURVE"/>
- <volume deviceCategory="DEVICE_CATEGORY_HEARING_AID" ref="FULL_SCALE_VOLUME_CURVE"/>
- </volumeGroup>
</volumeGroups>
diff --git a/services/audiopolicy/enginedefault/include/AudioPolicyEngineInstance.h b/services/audiopolicy/enginedefault/include/AudioPolicyEngineInstance.h
deleted file mode 100644
index 1e329f0..0000000
--- a/services/audiopolicy/enginedefault/include/AudioPolicyEngineInstance.h
+++ /dev/null
@@ -1,76 +0,0 @@
-/*
- * Copyright (C) 2015 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#pragma once
-
-class AudioPolicyManagerInterface;
-
-namespace android
-{
-namespace audio_policy
-{
-
-class Engine;
-
-class EngineInstance
-{
-protected:
- EngineInstance();
-
-public:
- virtual ~EngineInstance();
-
- /**
- * Get Audio Policy Engine instance.
- *
- * @return pointer to Route Manager Instance object.
- */
- static EngineInstance *getInstance();
-
- /**
- * Interface query.
- * The first client of an interface of the policy engine will start the singleton.
- *
- * @tparam RequestedInterface: interface that the client is wishing to retrieve.
- *
- * @return interface handle.
- */
- template <class RequestedInterface>
- RequestedInterface *queryInterface() const;
-
-protected:
- /**
- * Get Audio Policy Engine instance.
- *
- * @return Audio Policy Engine singleton.
- */
- Engine *getEngine() const;
-
-private:
- /* Copy facilities are put private to disable copy. */
- EngineInstance(const EngineInstance &object);
- EngineInstance &operator=(const EngineInstance &object);
-};
-
-/**
- * Limit template instantation to supported type interfaces.
- * Compile time error will claim if invalid interface is requested.
- */
-template <>
-AudioPolicyManagerInterface *EngineInstance::queryInterface() const;
-
-} // namespace audio_policy
-} // namespace android
diff --git a/services/audiopolicy/enginedefault/src/Engine.cpp b/services/audiopolicy/enginedefault/src/Engine.cpp
old mode 100644
new mode 100755
index 04170ac..02b99d0
--- a/services/audiopolicy/enginedefault/src/Engine.cpp
+++ b/services/audiopolicy/enginedefault/src/Engine.cpp
@@ -27,10 +27,11 @@
#include "Engine.h"
#include <android-base/macros.h>
#include <AudioPolicyManagerObserver.h>
-#include <AudioPort.h>
+#include <PolicyAudioPort.h>
#include <IOProfile.h>
#include <AudioIODescriptorInterface.h>
#include <policy.h>
+#include <media/AudioContainers.h>
#include <utils/String8.h>
#include <utils/Log.h>
@@ -136,27 +137,23 @@
return EngineBase::setForceUse(usage, config);
}
-audio_devices_t Engine::getDeviceForStrategyInt(legacy_strategy strategy,
- DeviceVector availableOutputDevices,
- DeviceVector availableInputDevices,
- const SwAudioOutputCollection &outputs,
- uint32_t outputDeviceTypesToIgnore) const
+DeviceVector Engine::getDevicesForStrategyInt(legacy_strategy strategy,
+ DeviceVector availableOutputDevices,
+ DeviceVector availableInputDevices,
+ const SwAudioOutputCollection &outputs) const
{
- uint32_t device = AUDIO_DEVICE_NONE;
- uint32_t availableOutputDevicesType =
- availableOutputDevices.types() & ~outputDeviceTypesToIgnore;
+ DeviceVector devices;
switch (strategy) {
case STRATEGY_TRANSMITTED_THROUGH_SPEAKER:
- device = availableOutputDevicesType & AUDIO_DEVICE_OUT_SPEAKER;
+ devices = availableOutputDevices.getDevicesFromType(AUDIO_DEVICE_OUT_SPEAKER);
break;
case STRATEGY_SONIFICATION_RESPECTFUL:
if (isInCall() || outputs.isActiveLocally(toVolumeSource(AUDIO_STREAM_VOICE_CALL))) {
- device = getDeviceForStrategyInt(
- STRATEGY_SONIFICATION, availableOutputDevices, availableInputDevices, outputs,
- outputDeviceTypesToIgnore);
+ devices = getDevicesForStrategyInt(
+ STRATEGY_SONIFICATION, availableOutputDevices, availableInputDevices, outputs);
} else {
bool media_active_locally =
outputs.isActiveLocally(toVolumeSource(AUDIO_STREAM_MUSIC),
@@ -165,17 +162,18 @@
toVolumeSource(AUDIO_STREAM_ACCESSIBILITY),
SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY);
// routing is same as media without the "remote" device
- device = getDeviceForStrategyInt(STRATEGY_MEDIA,
+ availableOutputDevices.remove(availableOutputDevices.getDevicesFromType(
+ AUDIO_DEVICE_OUT_REMOTE_SUBMIX));
+ devices = getDevicesForStrategyInt(STRATEGY_MEDIA,
availableOutputDevices,
- availableInputDevices, outputs,
- AUDIO_DEVICE_OUT_REMOTE_SUBMIX | outputDeviceTypesToIgnore);
+ availableInputDevices, outputs);
// if no media is playing on the device, check for mandatory use of "safe" speaker
// when media would have played on speaker, and the safe speaker path is available
- if (!media_active_locally
- && (device & AUDIO_DEVICE_OUT_SPEAKER)
- && (availableOutputDevicesType & AUDIO_DEVICE_OUT_SPEAKER_SAFE)) {
- device |= AUDIO_DEVICE_OUT_SPEAKER_SAFE;
- device &= ~AUDIO_DEVICE_OUT_SPEAKER;
+ if (!media_active_locally) {
+ devices.replaceDevicesByType(
+ AUDIO_DEVICE_OUT_SPEAKER,
+ availableOutputDevices.getDevicesFromType(
+ AUDIO_DEVICE_OUT_SPEAKER_SAFE));
}
}
break;
@@ -183,9 +181,8 @@
case STRATEGY_DTMF:
if (!isInCall()) {
// when off call, DTMF strategy follows the same rules as MEDIA strategy
- device = getDeviceForStrategyInt(
- STRATEGY_MEDIA, availableOutputDevices, availableInputDevices, outputs,
- outputDeviceTypesToIgnore);
+ devices = getDevicesForStrategyInt(
+ STRATEGY_MEDIA, availableOutputDevices, availableInputDevices, outputs);
break;
}
// when in call, DTMF and PHONE strategies follow the same rules
@@ -197,24 +194,27 @@
// - cannot route from voice call RX OR
// - audio HAL version is < 3.0 and TX device is on the primary HW module
if (getPhoneState() == AUDIO_MODE_IN_CALL) {
- audio_devices_t txDevice = getDeviceForInputSource(AUDIO_SOURCE_VOICE_COMMUNICATION);
+ audio_devices_t txDevice = getDeviceForInputSource(
+ AUDIO_SOURCE_VOICE_COMMUNICATION)->type();
sp<AudioOutputDescriptor> primaryOutput = outputs.getPrimaryOutput();
- audio_devices_t availPrimaryInputDevices =
- availableInputDevices.getDeviceTypesFromHwModule(primaryOutput->getModuleHandle());
+ DeviceVector availPrimaryInputDevices =
+ availableInputDevices.getDevicesFromHwModule(primaryOutput->getModuleHandle());
// TODO: getPrimaryOutput return only devices from first module in
// audio_policy_configuration.xml, hearing aid is not there, but it's
// a primary device
// FIXME: this is not the right way of solving this problem
- audio_devices_t availPrimaryOutputDevices =
- (primaryOutput->supportedDevices().types() | AUDIO_DEVICE_OUT_HEARING_AID) &
- availableOutputDevices.types();
+ DeviceVector availPrimaryOutputDevices = availableOutputDevices.getDevicesFromTypes(
+ primaryOutput->supportedDevices().types());
+ availPrimaryOutputDevices.add(
+ availableOutputDevices.getDevicesFromType(AUDIO_DEVICE_OUT_HEARING_AID));
- if (((availableInputDevices.types() &
- AUDIO_DEVICE_IN_TELEPHONY_RX & ~AUDIO_DEVICE_BIT_IN) == 0) ||
- (((txDevice & availPrimaryInputDevices & ~AUDIO_DEVICE_BIT_IN) != 0) &&
- (primaryOutput->getAudioPort()->getModuleVersionMajor() < 3))) {
- availableOutputDevicesType = availPrimaryOutputDevices;
+ if ((availableInputDevices.getDevice(AUDIO_DEVICE_IN_TELEPHONY_RX,
+ String8(""), AUDIO_FORMAT_DEFAULT) == nullptr) ||
+ ((availPrimaryInputDevices.getDevice(
+ txDevice, String8(""), AUDIO_FORMAT_DEFAULT) != nullptr) &&
+ (primaryOutput->getPolicyAudioPort()->getModuleVersionMajor() < 3))) {
+ availableOutputDevices = availPrimaryOutputDevices;
}
}
// for phone strategy, we first consider the forced use and then the available devices by
@@ -222,49 +222,40 @@
switch (getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION)) {
case AUDIO_POLICY_FORCE_BT_SCO:
if (!isInCall() || strategy != STRATEGY_DTMF) {
- device = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
- if (device) break;
+ devices = availableOutputDevices.getDevicesFromType(
+ AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT);
+ if (!devices.isEmpty()) break;
}
- device = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET;
- if (device) break;
- device = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_SCO;
- if (device) break;
+ devices = availableOutputDevices.getFirstDevicesFromTypes({
+ AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET, AUDIO_DEVICE_OUT_BLUETOOTH_SCO});
+ if (!devices.isEmpty()) break;
// if SCO device is requested but no SCO device is available, fall back to default case
FALLTHROUGH_INTENDED;
default: // FORCE_NONE
- device = availableOutputDevicesType & AUDIO_DEVICE_OUT_HEARING_AID;
- if (device) break;
+ devices = availableOutputDevices.getDevicesFromType(AUDIO_DEVICE_OUT_HEARING_AID);
+ if (!devices.isEmpty()) break;
// when not in a phone call, phone strategy should route STREAM_VOICE_CALL to A2DP
if (!isInCall() &&
(getForceUse(AUDIO_POLICY_FORCE_FOR_MEDIA) != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
outputs.isA2dpSupported()) {
- device = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
- if (device) break;
- device = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
- if (device) break;
+ devices = availableOutputDevices.getFirstDevicesFromTypes({
+ AUDIO_DEVICE_OUT_BLUETOOTH_A2DP,
+ AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES});
+ if (!devices.isEmpty()) break;
}
- device = availableOutputDevicesType & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
- if (device) break;
- device = availableOutputDevicesType & AUDIO_DEVICE_OUT_WIRED_HEADSET;
- if (device) break;
- device = availableOutputDevicesType & AUDIO_DEVICE_OUT_LINE;
- if (device) break;
- device = availableOutputDevicesType & AUDIO_DEVICE_OUT_USB_HEADSET;
- if (device) break;
- device = availableOutputDevicesType & AUDIO_DEVICE_OUT_USB_DEVICE;
- if (device) break;
+ devices = availableOutputDevices.getFirstDevicesFromTypes({
+ AUDIO_DEVICE_OUT_WIRED_HEADPHONE, AUDIO_DEVICE_OUT_WIRED_HEADSET,
+ AUDIO_DEVICE_OUT_LINE, AUDIO_DEVICE_OUT_USB_HEADSET,
+ AUDIO_DEVICE_OUT_USB_DEVICE});
+ if (!devices.isEmpty()) break;
if (!isInCall()) {
- device = availableOutputDevicesType & AUDIO_DEVICE_OUT_USB_ACCESSORY;
- if (device) break;
- device = availableOutputDevicesType & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
- if (device) break;
- device = availableOutputDevicesType & AUDIO_DEVICE_OUT_AUX_DIGITAL;
- if (device) break;
- device = availableOutputDevicesType & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
- if (device) break;
+ devices = availableOutputDevices.getFirstDevicesFromTypes({
+ AUDIO_DEVICE_OUT_USB_ACCESSORY, AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET,
+ AUDIO_DEVICE_OUT_AUX_DIGITAL, AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET});
+ if (!devices.isEmpty()) break;
}
- device = availableOutputDevicesType & AUDIO_DEVICE_OUT_EARPIECE;
+ devices = availableOutputDevices.getDevicesFromType(AUDIO_DEVICE_OUT_EARPIECE);
break;
case AUDIO_POLICY_FORCE_SPEAKER:
@@ -273,22 +264,18 @@
if (!isInCall() &&
(getForceUse(AUDIO_POLICY_FORCE_FOR_MEDIA) != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
outputs.isA2dpSupported()) {
- device = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
- if (device) break;
+ devices = availableOutputDevices.getDevicesFromType(
+ AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER);
+ if (!devices.isEmpty()) break;
}
if (!isInCall()) {
- device = availableOutputDevicesType & AUDIO_DEVICE_OUT_USB_ACCESSORY;
- if (device) break;
- device = availableOutputDevicesType & AUDIO_DEVICE_OUT_USB_DEVICE;
- if (device) break;
- device = availableOutputDevicesType & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
- if (device) break;
- device = availableOutputDevicesType & AUDIO_DEVICE_OUT_AUX_DIGITAL;
- if (device) break;
- device = availableOutputDevicesType & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
- if (device) break;
+ devices = availableOutputDevices.getFirstDevicesFromTypes({
+ AUDIO_DEVICE_OUT_USB_ACCESSORY, AUDIO_DEVICE_OUT_USB_DEVICE,
+ AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET, AUDIO_DEVICE_OUT_AUX_DIGITAL,
+ AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET});
+ if (!devices.isEmpty()) break;
}
- device = availableOutputDevicesType & AUDIO_DEVICE_OUT_SPEAKER;
+ devices = availableOutputDevices.getDevicesFromType(AUDIO_DEVICE_OUT_SPEAKER);
break;
}
break;
@@ -298,9 +285,8 @@
// If incall, just select the STRATEGY_PHONE device
if (isInCall() ||
outputs.isActiveLocally(toVolumeSource(AUDIO_STREAM_VOICE_CALL))) {
- device = getDeviceForStrategyInt(
- STRATEGY_PHONE, availableOutputDevices, availableInputDevices, outputs,
- outputDeviceTypesToIgnore);
+ devices = getDevicesForStrategyInt(
+ STRATEGY_PHONE, availableOutputDevices, availableInputDevices, outputs);
break;
}
FALLTHROUGH_INTENDED;
@@ -313,41 +299,37 @@
if ((strategy == STRATEGY_SONIFICATION) ||
(getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED)) {
- device = availableOutputDevicesType & AUDIO_DEVICE_OUT_SPEAKER;
+ devices = availableOutputDevices.getDevicesFromType(AUDIO_DEVICE_OUT_SPEAKER);
}
// if SCO headset is connected and we are told to use it, play ringtone over
// speaker and BT SCO
- if ((availableOutputDevicesType & AUDIO_DEVICE_OUT_ALL_SCO) != 0) {
- uint32_t device2 = AUDIO_DEVICE_NONE;
- device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
- if (device2 == AUDIO_DEVICE_NONE) {
- device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET;
- }
- if (device2 == AUDIO_DEVICE_NONE) {
- device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_SCO;
- }
+ if (!availableOutputDevices.getDevicesFromTypes(getAudioDeviceOutAllScoSet()).isEmpty()) {
+ DeviceVector devices2;
+ devices2 = availableOutputDevices.getFirstDevicesFromTypes({
+ AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT, AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET,
+ AUDIO_DEVICE_OUT_BLUETOOTH_SCO});
// Use ONLY Bluetooth SCO output when ringing in vibration mode
if (!((getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED)
&& (strategy == STRATEGY_ENFORCED_AUDIBLE))) {
if (getForceUse(AUDIO_POLICY_FORCE_FOR_VIBRATE_RINGING)
== AUDIO_POLICY_FORCE_BT_SCO) {
- if (device2 != AUDIO_DEVICE_NONE) {
- device = device2;
+ if (!devices2.isEmpty()) {
+ devices = devices2;
break;
}
}
}
// Use both Bluetooth SCO and phone default output when ringing in normal mode
if (getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION) == AUDIO_POLICY_FORCE_BT_SCO) {
- if ((strategy == STRATEGY_SONIFICATION) &&
- (device & AUDIO_DEVICE_OUT_SPEAKER) &&
- (availableOutputDevicesType & AUDIO_DEVICE_OUT_SPEAKER_SAFE)) {
- device |= AUDIO_DEVICE_OUT_SPEAKER_SAFE;
- device &= ~AUDIO_DEVICE_OUT_SPEAKER;
+ if (strategy == STRATEGY_SONIFICATION) {
+ devices.replaceDevicesByType(
+ AUDIO_DEVICE_OUT_SPEAKER,
+ availableOutputDevices.getDevicesFromType(
+ AUDIO_DEVICE_OUT_SPEAKER_SAFE));
}
- if (device2 != AUDIO_DEVICE_NONE) {
- device |= device2;
+ if (!devices2.isEmpty()) {
+ devices.add(devices2);
break;
}
}
@@ -361,25 +343,20 @@
// compressed format as they would likely not be mixed and dropped.
for (size_t i = 0; i < outputs.size(); i++) {
sp<AudioOutputDescriptor> desc = outputs.valueAt(i);
- audio_devices_t devices = desc->devices().types() &
- (AUDIO_DEVICE_OUT_HDMI | AUDIO_DEVICE_OUT_SPDIF | AUDIO_DEVICE_OUT_HDMI_ARC);
- if (desc->isActive() && !audio_is_linear_pcm(desc->mFormat) &&
- devices != AUDIO_DEVICE_NONE) {
- availableOutputDevicesType = availableOutputDevices.types() & ~devices;
+ if (desc->isActive() && !audio_is_linear_pcm(desc->getFormat())) {
+ availableOutputDevices.remove(desc->devices().getDevicesFromTypes({
+ AUDIO_DEVICE_OUT_HDMI, AUDIO_DEVICE_OUT_SPDIF,
+ AUDIO_DEVICE_OUT_HDMI_ARC}));
}
}
- availableOutputDevices =
- availableOutputDevices.getDevicesFromTypeMask(availableOutputDevicesType);
if (outputs.isActive(toVolumeSource(AUDIO_STREAM_RING)) ||
outputs.isActive(toVolumeSource(AUDIO_STREAM_ALARM))) {
- return getDeviceForStrategyInt(
- STRATEGY_SONIFICATION, availableOutputDevices, availableInputDevices, outputs,
- outputDeviceTypesToIgnore);
+ return getDevicesForStrategyInt(
+ STRATEGY_SONIFICATION, availableOutputDevices, availableInputDevices, outputs);
}
if (isInCall()) {
- return getDeviceForStrategyInt(
- STRATEGY_PHONE, availableOutputDevices, availableInputDevices, outputs,
- outputDeviceTypesToIgnore);
+ return getDevicesForStrategyInt(
+ STRATEGY_PHONE, availableOutputDevices, availableInputDevices, outputs);
}
}
// For other cases, STRATEGY_ACCESSIBILITY behaves like STRATEGY_MEDIA
@@ -388,128 +365,114 @@
// FIXME: STRATEGY_REROUTING follow STRATEGY_MEDIA for now
case STRATEGY_REROUTING:
case STRATEGY_MEDIA: {
- uint32_t device2 = AUDIO_DEVICE_NONE;
+ DeviceVector devices2;
if (strategy != STRATEGY_SONIFICATION) {
// no sonification on remote submix (e.g. WFD)
- if (availableOutputDevices.getDevice(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
- String8("0"), AUDIO_FORMAT_DEFAULT) != 0) {
- device2 = availableOutputDevices.types() & AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
+ sp<DeviceDescriptor> remoteSubmix;
+ if ((remoteSubmix = availableOutputDevices.getDevice(
+ AUDIO_DEVICE_OUT_REMOTE_SUBMIX, String8("0"),
+ AUDIO_FORMAT_DEFAULT)) != nullptr) {
+ devices2.add(remoteSubmix);
}
}
if (isInCall() && (strategy == STRATEGY_MEDIA)) {
- device = getDeviceForStrategyInt(
- STRATEGY_PHONE, availableOutputDevices, availableInputDevices, outputs,
- outputDeviceTypesToIgnore);
+ devices = getDevicesForStrategyInt(
+ STRATEGY_PHONE, availableOutputDevices, availableInputDevices, outputs);
break;
}
// FIXME: Find a better solution to prevent routing to BT hearing aid(b/122931261).
- if ((device2 == AUDIO_DEVICE_NONE) &&
+ if ((devices2.isEmpty()) &&
(getForceUse(AUDIO_POLICY_FORCE_FOR_MEDIA) != AUDIO_POLICY_FORCE_NO_BT_A2DP)) {
- device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_HEARING_AID;
+ devices2 = availableOutputDevices.getDevicesFromType(AUDIO_DEVICE_OUT_HEARING_AID);
}
- if ((device2 == AUDIO_DEVICE_NONE) &&
- (getForceUse(AUDIO_POLICY_FORCE_FOR_MEDIA) != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
- outputs.isA2dpSupported()) {
- device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
- if (device2 == AUDIO_DEVICE_NONE) {
- device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
- }
- if (device2 == AUDIO_DEVICE_NONE) {
- device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
- }
- }
- if ((device2 == AUDIO_DEVICE_NONE) &&
+ if ((devices2.isEmpty()) &&
(getForceUse(AUDIO_POLICY_FORCE_FOR_MEDIA) == AUDIO_POLICY_FORCE_SPEAKER)) {
- device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_SPEAKER;
+ devices2 = availableOutputDevices.getDevicesFromType(AUDIO_DEVICE_OUT_SPEAKER);
}
- if (device2 == AUDIO_DEVICE_NONE) {
- device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
+ if (devices2.isEmpty() && (getLastRemovableMediaDevices().size() > 0)) {
+ if ((getForceUse(AUDIO_POLICY_FORCE_FOR_MEDIA) != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
+ outputs.isA2dpSupported()) {
+ // Get the last connected device of wired and bluetooth a2dp
+ devices2 = availableOutputDevices.getFirstDevicesFromTypes(
+ getLastRemovableMediaDevices());
+ } else {
+ // Get the last connected device of wired except bluetooth a2dp
+ devices2 = availableOutputDevices.getFirstDevicesFromTypes(
+ getLastRemovableMediaDevices(GROUP_WIRED));
+ }
}
- if (device2 == AUDIO_DEVICE_NONE) {
- device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_LINE;
- }
- if (device2 == AUDIO_DEVICE_NONE) {
- device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_WIRED_HEADSET;
- }
- if (device2 == AUDIO_DEVICE_NONE) {
- device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_USB_HEADSET;
- }
- if (device2 == AUDIO_DEVICE_NONE) {
- device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_USB_ACCESSORY;
- }
- if (device2 == AUDIO_DEVICE_NONE) {
- device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_USB_DEVICE;
- }
- if (device2 == AUDIO_DEVICE_NONE) {
- device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
- }
- if ((device2 == AUDIO_DEVICE_NONE) && (strategy != STRATEGY_SONIFICATION)) {
+ if ((devices2.isEmpty()) && (strategy != STRATEGY_SONIFICATION)) {
// no sonification on aux digital (e.g. HDMI)
- device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_AUX_DIGITAL;
+ devices2 = availableOutputDevices.getDevicesFromType(AUDIO_DEVICE_OUT_AUX_DIGITAL);
}
- if ((device2 == AUDIO_DEVICE_NONE) &&
+ if ((devices2.isEmpty()) &&
(getForceUse(AUDIO_POLICY_FORCE_FOR_DOCK) == AUDIO_POLICY_FORCE_ANALOG_DOCK)) {
- device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
+ devices2 = availableOutputDevices.getDevicesFromType(
+ AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET);
}
- if (device2 == AUDIO_DEVICE_NONE) {
- device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_SPEAKER;
+ if (devices2.isEmpty()) {
+ devices2 = availableOutputDevices.getDevicesFromType(AUDIO_DEVICE_OUT_SPEAKER);
}
- int device3 = AUDIO_DEVICE_NONE;
+ DeviceVector devices3;
if (strategy == STRATEGY_MEDIA) {
// ARC, SPDIF and AUX_LINE can co-exist with others.
- device3 = availableOutputDevicesType & AUDIO_DEVICE_OUT_HDMI_ARC;
- device3 |= (availableOutputDevicesType & AUDIO_DEVICE_OUT_SPDIF);
- device3 |= (availableOutputDevicesType & AUDIO_DEVICE_OUT_AUX_LINE);
+ devices3 = availableOutputDevices.getDevicesFromTypes({
+ AUDIO_DEVICE_OUT_HDMI_ARC, AUDIO_DEVICE_OUT_SPDIF, AUDIO_DEVICE_OUT_AUX_LINE});
}
- device2 |= device3;
+ devices2.add(devices3);
// device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION or
// STRATEGY_ENFORCED_AUDIBLE, AUDIO_DEVICE_NONE otherwise
- device |= device2;
+ devices.add(devices2);
// If hdmi system audio mode is on, remove speaker out of output list.
if ((strategy == STRATEGY_MEDIA) &&
(getForceUse(AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO) ==
AUDIO_POLICY_FORCE_HDMI_SYSTEM_AUDIO_ENFORCED)) {
- device &= ~AUDIO_DEVICE_OUT_SPEAKER;
+ devices.remove(devices.getDevicesFromType(AUDIO_DEVICE_OUT_SPEAKER));
}
// for STRATEGY_SONIFICATION:
// if SPEAKER was selected, and SPEAKER_SAFE is available, use SPEAKER_SAFE instead
- if ((strategy == STRATEGY_SONIFICATION) &&
- (device & AUDIO_DEVICE_OUT_SPEAKER) &&
- (availableOutputDevicesType & AUDIO_DEVICE_OUT_SPEAKER_SAFE)) {
- device |= AUDIO_DEVICE_OUT_SPEAKER_SAFE;
- device &= ~AUDIO_DEVICE_OUT_SPEAKER;
+ if (strategy == STRATEGY_SONIFICATION) {
+ devices.replaceDevicesByType(
+ AUDIO_DEVICE_OUT_SPEAKER,
+ availableOutputDevices.getDevicesFromType(
+ AUDIO_DEVICE_OUT_SPEAKER_SAFE));
}
} break;
default:
- ALOGW("getDeviceForStrategy() unknown strategy: %d", strategy);
+ ALOGW("getDevicesForStrategy() unknown strategy: %d", strategy);
break;
}
- if (device == AUDIO_DEVICE_NONE) {
- ALOGV("getDeviceForStrategy() no device found for strategy %d", strategy);
- device = getApmObserver()->getDefaultOutputDevice()->type();
- ALOGE_IF(device == AUDIO_DEVICE_NONE,
- "getDeviceForStrategy() no default device defined");
+ if (devices.isEmpty()) {
+ ALOGV("getDevicesForStrategy() no device found for strategy %d", strategy);
+ sp<DeviceDescriptor> defaultOutputDevice = getApmObserver()->getDefaultOutputDevice();
+ if (defaultOutputDevice != nullptr) {
+ devices.add(defaultOutputDevice);
+ }
+ ALOGE_IF(devices.isEmpty(),
+ "getDevicesForStrategy() no default device defined");
}
- ALOGVV("getDeviceForStrategy() strategy %d, device %x", strategy, device);
- return device;
+
+ ALOGVV("getDevices"
+ "ForStrategy() strategy %d, device %x", strategy, devices.types());
+ return devices;
}
-audio_devices_t Engine::getDeviceForInputSource(audio_source_t inputSource) const
+sp<DeviceDescriptor> Engine::getDeviceForInputSource(audio_source_t inputSource) const
{
const DeviceVector availableOutputDevices = getApmObserver()->getAvailableOutputDevices();
const DeviceVector availableInputDevices = getApmObserver()->getAvailableInputDevices();
const SwAudioOutputCollection &outputs = getApmObserver()->getOutputs();
- audio_devices_t availableDeviceTypes = availableInputDevices.types() & ~AUDIO_DEVICE_BIT_IN;
+ DeviceVector availableDevices = availableInputDevices;
sp<AudioOutputDescriptor> primaryOutput = outputs.getPrimaryOutput();
- audio_devices_t availablePrimaryDeviceTypes = availableInputDevices.getDeviceTypesFromHwModule(
- primaryOutput->getModuleHandle()) & ~AUDIO_DEVICE_BIT_IN;
- uint32_t device = AUDIO_DEVICE_NONE;
+ DeviceVector availablePrimaryDevices = availableInputDevices.getDevicesFromHwModule(
+ primaryOutput->getModuleHandle());
+ sp<DeviceDescriptor> device;
// when a call is active, force device selection to match source VOICE_COMMUNICATION
// for most other input sources to avoid rerouting call TX audio
@@ -532,57 +495,47 @@
switch (inputSource) {
case AUDIO_SOURCE_DEFAULT:
case AUDIO_SOURCE_MIC:
- if (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_A2DP) {
- device = AUDIO_DEVICE_IN_BLUETOOTH_A2DP;
- } else if ((getForceUse(AUDIO_POLICY_FORCE_FOR_RECORD) == AUDIO_POLICY_FORCE_BT_SCO) &&
- (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET)) {
- device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
- } else if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) {
- device = AUDIO_DEVICE_IN_WIRED_HEADSET;
- } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_HEADSET) {
- device = AUDIO_DEVICE_IN_USB_HEADSET;
- } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) {
- device = AUDIO_DEVICE_IN_USB_DEVICE;
- } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
- device = AUDIO_DEVICE_IN_BUILTIN_MIC;
- }
- break;
+ device = availableDevices.getDevice(
+ AUDIO_DEVICE_IN_BLUETOOTH_A2DP, String8(""), AUDIO_FORMAT_DEFAULT);
+ if (device != nullptr) break;
+ if (getForceUse(AUDIO_POLICY_FORCE_FOR_RECORD) == AUDIO_POLICY_FORCE_BT_SCO) {
+ device = availableDevices.getDevice(
+ AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET, String8(""), AUDIO_FORMAT_DEFAULT);
+ if (device != nullptr) break;
+ }
+ device = availableDevices.getFirstExistingDevice({
+ AUDIO_DEVICE_IN_WIRED_HEADSET, AUDIO_DEVICE_IN_USB_HEADSET,
+ AUDIO_DEVICE_IN_USB_DEVICE, AUDIO_DEVICE_IN_BUILTIN_MIC});
+ break;
case AUDIO_SOURCE_VOICE_COMMUNICATION:
// Allow only use of devices on primary input if in call and HAL does not support routing
// to voice call path.
if ((getPhoneState() == AUDIO_MODE_IN_CALL) &&
- (availableOutputDevices.types() & AUDIO_DEVICE_OUT_TELEPHONY_TX) == 0) {
- availableDeviceTypes = availablePrimaryDeviceTypes;
+ (availableOutputDevices.getDevice(AUDIO_DEVICE_OUT_TELEPHONY_TX,
+ String8(""), AUDIO_FORMAT_DEFAULT)) == nullptr) {
+ availableDevices = availablePrimaryDevices;
}
switch (getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION)) {
case AUDIO_POLICY_FORCE_BT_SCO:
// if SCO device is requested but no SCO device is available, fall back to default case
- if (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
- device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
+ device = availableDevices.getDevice(
+ AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET, String8(""), AUDIO_FORMAT_DEFAULT);
+ if (device != nullptr) {
break;
}
FALLTHROUGH_INTENDED;
default: // FORCE_NONE
- if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) {
- device = AUDIO_DEVICE_IN_WIRED_HEADSET;
- } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_HEADSET) {
- device = AUDIO_DEVICE_IN_USB_HEADSET;
- } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) {
- device = AUDIO_DEVICE_IN_USB_DEVICE;
- } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
- device = AUDIO_DEVICE_IN_BUILTIN_MIC;
- }
+ device = availableDevices.getFirstExistingDevice({
+ AUDIO_DEVICE_IN_WIRED_HEADSET, AUDIO_DEVICE_IN_USB_HEADSET,
+ AUDIO_DEVICE_IN_USB_DEVICE, AUDIO_DEVICE_IN_BUILTIN_MIC});
break;
case AUDIO_POLICY_FORCE_SPEAKER:
- if (availableDeviceTypes & AUDIO_DEVICE_IN_BACK_MIC) {
- device = AUDIO_DEVICE_IN_BACK_MIC;
- } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
- device = AUDIO_DEVICE_IN_BUILTIN_MIC;
- }
+ device = availableDevices.getFirstExistingDevice({
+ AUDIO_DEVICE_IN_BACK_MIC, AUDIO_DEVICE_IN_BUILTIN_MIC});
break;
}
break;
@@ -591,84 +544,67 @@
case AUDIO_SOURCE_UNPROCESSED:
case AUDIO_SOURCE_HOTWORD:
if (inputSource == AUDIO_SOURCE_HOTWORD) {
- availableDeviceTypes = availablePrimaryDeviceTypes;
+ availableDevices = availablePrimaryDevices;
}
- if (getForceUse(AUDIO_POLICY_FORCE_FOR_RECORD) == AUDIO_POLICY_FORCE_BT_SCO &&
- availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
- device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
- } else if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) {
- device = AUDIO_DEVICE_IN_WIRED_HEADSET;
- } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_HEADSET) {
- device = AUDIO_DEVICE_IN_USB_HEADSET;
- } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) {
- device = AUDIO_DEVICE_IN_USB_DEVICE;
- } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
- device = AUDIO_DEVICE_IN_BUILTIN_MIC;
+ if (getForceUse(AUDIO_POLICY_FORCE_FOR_RECORD) == AUDIO_POLICY_FORCE_BT_SCO) {
+ device = availableDevices.getDevice(
+ AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET, String8(""), AUDIO_FORMAT_DEFAULT);
+ if (device != nullptr) break;
}
+ device = availableDevices.getFirstExistingDevice({
+ AUDIO_DEVICE_IN_WIRED_HEADSET, AUDIO_DEVICE_IN_USB_HEADSET,
+ AUDIO_DEVICE_IN_USB_DEVICE, AUDIO_DEVICE_IN_BUILTIN_MIC});
break;
case AUDIO_SOURCE_CAMCORDER:
- if (availableDeviceTypes & AUDIO_DEVICE_IN_BACK_MIC) {
- device = AUDIO_DEVICE_IN_BACK_MIC;
- } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
- device = AUDIO_DEVICE_IN_BUILTIN_MIC;
- } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) {
- // This is specifically for a device without built-in mic
- device = AUDIO_DEVICE_IN_USB_DEVICE;
- }
+ // For a device without built-in mic, adding usb device
+ device = availableDevices.getFirstExistingDevice({
+ AUDIO_DEVICE_IN_BACK_MIC, AUDIO_DEVICE_IN_BUILTIN_MIC,
+ AUDIO_DEVICE_IN_USB_DEVICE});
break;
case AUDIO_SOURCE_VOICE_DOWNLINK:
case AUDIO_SOURCE_VOICE_CALL:
case AUDIO_SOURCE_VOICE_UPLINK:
- if (availableDeviceTypes & AUDIO_DEVICE_IN_VOICE_CALL) {
- device = AUDIO_DEVICE_IN_VOICE_CALL;
- }
+ device = availableDevices.getDevice(
+ AUDIO_DEVICE_IN_VOICE_CALL, String8(""), AUDIO_FORMAT_DEFAULT);
break;
case AUDIO_SOURCE_VOICE_PERFORMANCE:
- if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) {
- device = AUDIO_DEVICE_IN_WIRED_HEADSET;
- } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_HEADSET) {
- device = AUDIO_DEVICE_IN_USB_HEADSET;
- } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) {
- device = AUDIO_DEVICE_IN_USB_DEVICE;
- } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
- device = AUDIO_DEVICE_IN_BUILTIN_MIC;
- }
+ device = availableDevices.getFirstExistingDevice({
+ AUDIO_DEVICE_IN_WIRED_HEADSET, AUDIO_DEVICE_IN_USB_HEADSET,
+ AUDIO_DEVICE_IN_USB_DEVICE, AUDIO_DEVICE_IN_BUILTIN_MIC});
break;
case AUDIO_SOURCE_REMOTE_SUBMIX:
- if (availableDeviceTypes & AUDIO_DEVICE_IN_REMOTE_SUBMIX) {
- device = AUDIO_DEVICE_IN_REMOTE_SUBMIX;
- }
+ device = availableDevices.getDevice(
+ AUDIO_DEVICE_IN_REMOTE_SUBMIX, String8(""), AUDIO_FORMAT_DEFAULT);
break;
case AUDIO_SOURCE_FM_TUNER:
- if (availableDeviceTypes & AUDIO_DEVICE_IN_FM_TUNER) {
- device = AUDIO_DEVICE_IN_FM_TUNER;
- }
+ device = availableDevices.getDevice(
+ AUDIO_DEVICE_IN_FM_TUNER, String8(""), AUDIO_FORMAT_DEFAULT);
break;
case AUDIO_SOURCE_ECHO_REFERENCE:
- if (availableDeviceTypes & AUDIO_DEVICE_IN_ECHO_REFERENCE) {
- device = AUDIO_DEVICE_IN_ECHO_REFERENCE;
- }
+ device = availableDevices.getDevice(
+ AUDIO_DEVICE_IN_ECHO_REFERENCE, String8(""), AUDIO_FORMAT_DEFAULT);
break;
default:
ALOGW("getDeviceForInputSource() invalid input source %d", inputSource);
break;
}
- if (device == AUDIO_DEVICE_NONE) {
+ if (device == nullptr) {
ALOGV("getDeviceForInputSource() no device found for source %d", inputSource);
- if (availableDeviceTypes & AUDIO_DEVICE_IN_STUB) {
- device = AUDIO_DEVICE_IN_STUB;
- }
- ALOGE_IF(device == AUDIO_DEVICE_NONE,
+ device = availableDevices.getDevice(
+ AUDIO_DEVICE_IN_STUB, String8(""), AUDIO_FORMAT_DEFAULT);
+ ALOGE_IF(device == nullptr,
"getDeviceForInputSource() no default device defined");
}
- ALOGV("getDeviceForInputSource()input source %d, device %08x", inputSource, device);
+ ALOGV_IF(device != nullptr,
+ "getDeviceForInputSource()input source %d, device %08x",
+ inputSource, device->type());
return device;
}
void Engine::updateDeviceSelectionCache()
{
for (const auto &iter : getProductStrategies()) {
- const auto &strategy = iter.second;
+ const auto& strategy = iter.second;
auto devices = getDevicesForProductStrategy(strategy->getId());
mDevicesForStrategies[strategy->getId()] = devices;
strategy->setDeviceTypes(devices.types());
@@ -676,19 +612,33 @@
}
}
-DeviceVector Engine::getDevicesForProductStrategy(product_strategy_t strategy) const
-{
+DeviceVector Engine::getDevicesForProductStrategy(product_strategy_t strategy) const {
DeviceVector availableOutputDevices = getApmObserver()->getAvailableOutputDevices();
- DeviceVector availableInputDevices = getApmObserver()->getAvailableInputDevices();
- const SwAudioOutputCollection &outputs = getApmObserver()->getOutputs();
+ // check if this strategy has a preferred device that is available,
+ // if yes, give priority to it
+ AudioDeviceTypeAddr preferredStrategyDevice;
+ const status_t status = getPreferredDeviceForStrategy(strategy, preferredStrategyDevice);
+ if (status == NO_ERROR) {
+ // there is a preferred device, is it available?
+ sp<DeviceDescriptor> preferredAvailableDevDescr = availableOutputDevices.getDevice(
+ preferredStrategyDevice.mType,
+ String8(preferredStrategyDevice.mAddress.c_str()),
+ AUDIO_FORMAT_DEFAULT);
+ if (preferredAvailableDevDescr != nullptr) {
+ ALOGVV("%s using pref device 0x%08x/%s for strategy %u", __FUNCTION__,
+ preferredStrategyDevice.mType, preferredStrategyDevice.mAddress, strategy);
+ return DeviceVector(preferredAvailableDevDescr);
+ }
+ }
+
+ DeviceVector availableInputDevices = getApmObserver()->getAvailableInputDevices();
+ const SwAudioOutputCollection& outputs = getApmObserver()->getOutputs();
auto legacyStrategy = mLegacyStrategyMap.find(strategy) != end(mLegacyStrategyMap) ?
- mLegacyStrategyMap.at(strategy) : STRATEGY_NONE;
- audio_devices_t devices = getDeviceForStrategyInt(legacyStrategy,
- availableOutputDevices,
- availableInputDevices, outputs,
- (uint32_t)AUDIO_DEVICE_NONE);
- return availableOutputDevices.getDevicesFromTypeMask(devices);
+ mLegacyStrategyMap.at(strategy) : STRATEGY_NONE;
+ return getDevicesForStrategyInt(legacyStrategy,
+ availableOutputDevices,
+ availableInputDevices, outputs);
}
DeviceVector Engine::getOutputDevicesForAttributes(const audio_attributes_t &attributes,
@@ -747,27 +697,25 @@
if (device != nullptr) {
return device;
}
- audio_devices_t deviceType = getDeviceForInputSource(attr.source);
- if (audio_is_remote_submix_device(deviceType)) {
- address = "0";
- std::size_t pos;
- std::string tags { attr.tags };
- if ((pos = tags.find("addr=")) != std::string::npos) {
- address = tags.substr(pos + std::strlen("addr="));
- }
+ device = getDeviceForInputSource(attr.source);
+ if (device == nullptr || !audio_is_remote_submix_device(device->type())) {
+ // Return immediately if the device is null or it is not a remote submix device.
+ return device;
}
- return availableInputDevices.getDevice(deviceType,
+
+ // For remote submix device, try to find the device by address.
+ address = "0";
+ std::size_t pos;
+ std::string tags { attr.tags };
+ if ((pos = tags.find("addr=")) != std::string::npos) {
+ address = tags.substr(pos + std::strlen("addr="));
+ }
+ return availableInputDevices.getDevice(device->type(),
String8(address.c_str()),
AUDIO_FORMAT_DEFAULT);
}
-template <>
-AudioPolicyManagerInterface *Engine::queryInterface()
-{
- return this;
-}
-
} // namespace audio_policy
} // namespace android
diff --git a/services/audiopolicy/enginedefault/src/Engine.h b/services/audiopolicy/enginedefault/src/Engine.h
index d5dfacc..4360c6f 100644
--- a/services/audiopolicy/enginedefault/src/Engine.h
+++ b/services/audiopolicy/enginedefault/src/Engine.h
@@ -17,8 +17,7 @@
#pragma once
#include "EngineBase.h"
-#include "AudioPolicyManagerInterface.h"
-#include <AudioGain.h>
+#include "EngineInterface.h"
#include <policy.h>
namespace android
@@ -48,12 +47,9 @@
Engine();
virtual ~Engine() = default;
- template <class RequestedInterface>
- RequestedInterface *queryInterface();
-
private:
///
- /// from EngineBase, so from AudioPolicyManagerInterface
+ /// from EngineBase, so from EngineInterface
///
status_t setForceUse(audio_policy_force_use_t usage,
audio_policy_forced_cfg_t config) override;
@@ -77,15 +73,14 @@
status_t setDefaultDevice(audio_devices_t device);
- audio_devices_t getDeviceForStrategyInt(legacy_strategy strategy,
- DeviceVector availableOutputDevices,
- DeviceVector availableInputDevices,
- const SwAudioOutputCollection &outputs,
- uint32_t outputDeviceTypesToIgnore) const;
+ DeviceVector getDevicesForStrategyInt(legacy_strategy strategy,
+ DeviceVector availableOutputDevices,
+ DeviceVector availableInputDevices,
+ const SwAudioOutputCollection &outputs) const;
DeviceVector getDevicesForProductStrategy(product_strategy_t strategy) const;
- audio_devices_t getDeviceForInputSource(audio_source_t inputSource) const;
+ sp<DeviceDescriptor> getDeviceForInputSource(audio_source_t inputSource) const;
DeviceStrategyMap mDevicesForStrategies;
diff --git a/services/audiopolicy/enginedefault/src/EngineInstance.cpp b/services/audiopolicy/enginedefault/src/EngineInstance.cpp
index 17e9832..eeb3758 100644
--- a/services/audiopolicy/enginedefault/src/EngineInstance.cpp
+++ b/services/audiopolicy/enginedefault/src/EngineInstance.cpp
@@ -14,41 +14,21 @@
* limitations under the License.
*/
-#include <AudioPolicyManagerInterface.h>
-#include "AudioPolicyEngineInstance.h"
+#include <EngineInterface.h>
#include "Engine.h"
-namespace android
-{
-namespace audio_policy
-{
+namespace android {
+namespace audio_policy {
-EngineInstance::EngineInstance()
+extern "C" EngineInterface* createEngineInstance()
{
+ return new (std::nothrow) Engine();
}
-EngineInstance *EngineInstance::getInstance()
+extern "C" void destroyEngineInstance(EngineInterface *engine)
{
- static EngineInstance instance;
- return &instance;
-}
-
-EngineInstance::~EngineInstance()
-{
-}
-
-Engine *EngineInstance::getEngine() const
-{
- static Engine engine;
- return &engine;
-}
-
-template <>
-AudioPolicyManagerInterface *EngineInstance::queryInterface() const
-{
- return getEngine()->queryInterface<AudioPolicyManagerInterface>();
+ delete static_cast<Engine*>(engine);
}
} // namespace audio_policy
} // namespace android
-
diff --git a/services/audiopolicy/manager/Android.mk b/services/audiopolicy/manager/Android.mk
index d6ca2f2..cae6cfa 100644
--- a/services/audiopolicy/manager/Android.mk
+++ b/services/audiopolicy/manager/Android.mk
@@ -23,8 +23,6 @@
LOCAL_CFLAGS := -Wall -Werror
-LOCAL_MULTILIB := $(AUDIOSERVER_MULTILIB)
-
LOCAL_MODULE:= libaudiopolicymanager
include $(BUILD_SHARED_LIBRARY)
diff --git a/services/audiopolicy/manager/AudioPolicyFactory.cpp b/services/audiopolicy/manager/AudioPolicyFactory.cpp
index 7aff6a9..476a1ec 100644
--- a/services/audiopolicy/manager/AudioPolicyFactory.cpp
+++ b/services/audiopolicy/manager/AudioPolicyFactory.cpp
@@ -21,7 +21,13 @@
extern "C" AudioPolicyInterface* createAudioPolicyManager(
AudioPolicyClientInterface *clientInterface)
{
- return new AudioPolicyManager(clientInterface);
+ AudioPolicyManager *apm = new AudioPolicyManager(clientInterface);
+ status_t status = apm->initialize();
+ if (status != NO_ERROR) {
+ delete apm;
+ apm = nullptr;
+ }
+ return apm;
}
extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface)
diff --git a/services/audiopolicy/managerdefault/Android.bp b/services/audiopolicy/managerdefault/Android.bp
new file mode 100644
index 0000000..1fa0d19
--- /dev/null
+++ b/services/audiopolicy/managerdefault/Android.bp
@@ -0,0 +1,44 @@
+cc_library_shared {
+ name: "libaudiopolicymanagerdefault",
+
+ srcs: [
+ "AudioPolicyManager.cpp",
+ "EngineLibrary.cpp",
+ ],
+
+ export_include_dirs: ["."],
+
+ shared_libs: [
+ "libaudiofoundation",
+ "libcutils",
+ "libdl",
+ "libutils",
+ "liblog",
+ "libaudiopolicy",
+ "libsoundtrigger",
+ "libmedia_helper",
+ "libmediametrics",
+ "libbinder",
+ "libhidlbase",
+ "libxml2",
+ // The default audio policy engine is always present in the system image.
+ // libaudiopolicyengineconfigurable can be built in addition by specifying
+ // a dependency on it in the device makefile. There will be no build time
+ // conflict with libaudiopolicyenginedefault.
+ "libaudiopolicyenginedefault",
+ ],
+
+ header_libs: [
+ "libaudiopolicycommon",
+ "libaudiopolicyengine_interface_headers",
+ "libaudiopolicymanager_interface_headers",
+ ],
+
+ static_libs: ["libaudiopolicycomponents"],
+
+ cflags: [
+ "-Wall",
+ "-Werror",
+ ],
+
+}
diff --git a/services/audiopolicy/managerdefault/Android.mk b/services/audiopolicy/managerdefault/Android.mk
deleted file mode 100644
index 684fc9f..0000000
--- a/services/audiopolicy/managerdefault/Android.mk
+++ /dev/null
@@ -1,56 +0,0 @@
-LOCAL_PATH:= $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES:= AudioPolicyManager.cpp
-
-LOCAL_EXPORT_C_INCLUDE_DIRS := $(LOCAL_PATH)
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libutils \
- liblog \
- libaudiopolicy \
- libsoundtrigger
-
-ifeq ($(USE_CONFIGURABLE_AUDIO_POLICY), 1)
-
-ifneq ($(USE_XML_AUDIO_POLICY_CONF), 1)
-$(error Configurable policy does not support legacy conf file)
-endif #ifneq ($(USE_XML_AUDIO_POLICY_CONF), 1)
-
-LOCAL_SHARED_LIBRARIES += libaudiopolicyengineconfigurable
-
-else
-
-LOCAL_SHARED_LIBRARIES += libaudiopolicyenginedefault
-
-endif # ifeq ($(USE_CONFIGURABLE_AUDIO_POLICY), 1)
-
-LOCAL_C_INCLUDES += \
- $(call include-path-for, audio-utils)
-
-LOCAL_HEADER_LIBRARIES := \
- libaudiopolicycommon \
- libaudiopolicyengine_interface_headers \
- libaudiopolicymanager_interface_headers
-
-LOCAL_STATIC_LIBRARIES := \
- libaudiopolicycomponents
-
-LOCAL_SHARED_LIBRARIES += libmedia_helper
-LOCAL_SHARED_LIBRARIES += libmediametrics
-
-LOCAL_SHARED_LIBRARIES += libbinder libhidlbase libxml2
-
-ifeq ($(USE_XML_AUDIO_POLICY_CONF), 1)
-LOCAL_CFLAGS += -DUSE_XML_AUDIO_POLICY_CONF
-endif #ifeq ($(USE_XML_AUDIO_POLICY_CONF), 1)
-
-LOCAL_CFLAGS += -Wall -Werror
-
-LOCAL_MULTILIB := $(AUDIOSERVER_MULTILIB)
-
-LOCAL_MODULE:= libaudiopolicymanagerdefault
-
-include $(BUILD_SHARED_LIBRARY)
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index c048de3..b747dd6 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -42,15 +42,12 @@
#include <set>
#include <unordered_set>
#include <vector>
-#include <AudioPolicyManagerInterface.h>
-#include <AudioPolicyEngineInstance.h>
#include <cutils/properties.h>
#include <utils/Log.h>
#include <media/AudioParameter.h>
#include <private/android_filesystem_config.h>
#include <soundtrigger/SoundTrigger.h>
#include <system/audio.h>
-#include <audio_policy_conf.h>
#include "AudioPolicyManager.h"
#include <Serializer.h>
#include "TypeConverter.h"
@@ -76,6 +73,26 @@
AUDIO_CHANNEL_OUT_2POINT1POINT2, AUDIO_CHANNEL_OUT_2POINT0POINT2,
AUDIO_CHANNEL_OUT_5POINT1, AUDIO_CHANNEL_OUT_STEREO }};
+template <typename T>
+bool operator== (const SortedVector<T> &left, const SortedVector<T> &right)
+{
+ if (left.size() != right.size()) {
+ return false;
+ }
+ for (size_t index = 0; index < right.size(); index++) {
+ if (left[index] != right[index]) {
+ return false;
+ }
+ }
+ return true;
+}
+
+template <typename T>
+bool operator!= (const SortedVector<T> &left, const SortedVector<T> &right)
+{
+ return !(left == right);
+}
+
// ----------------------------------------------------------------------------
// AudioPolicyInterface implementation
// ----------------------------------------------------------------------------
@@ -95,9 +112,9 @@
void AudioPolicyManager::broadcastDeviceConnectionState(const sp<DeviceDescriptor> &device,
audio_policy_dev_state_t state)
{
- AudioParameter param(device->address());
+ AudioParameter param(String8(device->address().c_str()));
const String8 key(state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE ?
- AudioParameter::keyStreamConnect : AudioParameter::keyStreamDisconnect);
+ AudioParameter::keyDeviceConnect : AudioParameter::keyDeviceDisconnect);
param.addInt(key, device->type());
mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
}
@@ -408,7 +425,7 @@
if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE;
// Check if the device is currently connected
- DeviceVector deviceList = mAvailableOutputDevices.getDevicesFromTypeMask(device);
+ DeviceVector deviceList = mAvailableOutputDevices.getDevicesFromType(device);
if (deviceList.empty()) {
// Nothing to do: device is not connected
return NO_ERROR;
@@ -422,8 +439,8 @@
// Case 1: A2DP active device switches from primary to primary
// module
// Case 2: A2DP device config changes on primary module.
- if (device & AUDIO_DEVICE_OUT_ALL_A2DP) {
- sp<HwModule> module = mHwModules.getModuleForDeviceTypes(device, encodedFormat);
+ if (audio_is_a2dp_out_device(device)) {
+ sp<HwModule> module = mHwModules.getModuleForDeviceType(device, encodedFormat);
audio_module_handle_t primaryHandle = mPrimaryOutput->getModuleHandle();
if (availablePrimaryOutputDevices().contains(devDesc) &&
(module != 0 && module->getHandle() == primaryHandle)) {
@@ -475,8 +492,12 @@
std::unordered_set<audio_format_t> formatSet;
sp<HwModule> primaryModule =
mHwModules.getModuleFromName(AUDIO_HARDWARE_MODULE_ID_PRIMARY);
- DeviceVector declaredDevices = primaryModule->getDeclaredDevices().getDevicesFromTypeMask(
- AUDIO_DEVICE_OUT_ALL_A2DP);
+ if (primaryModule == nullptr) {
+ ALOGE("%s() unable to get primary module", __func__);
+ return NO_INIT;
+ }
+ DeviceVector declaredDevices = primaryModule->getDeclaredDevices().getDevicesFromTypes(
+ getAudioDeviceOutAllA2dpSet());
for (const auto& device : declaredDevices) {
formatSet.insert(device->encodedFormats().begin(), device->encodedFormats().end());
}
@@ -490,7 +511,8 @@
bool createRxPatch = false;
uint32_t muteWaitMs = 0;
- if(!hasPrimaryOutput() || mPrimaryOutput->devices().types() == AUDIO_DEVICE_OUT_STUB) {
+ if(!hasPrimaryOutput() ||
+ mPrimaryOutput->devices().onlyContainsDevicesWithType(AUDIO_DEVICE_OUT_STUB)) {
return muteWaitMs;
}
ALOG_ASSERT(!rxDevices.isEmpty(), "updateCallRouting() no selected output device");
@@ -504,19 +526,19 @@
// release existing RX patch if any
if (mCallRxPatch != 0) {
- mpClientInterface->releaseAudioPatch(mCallRxPatch->mAfPatchHandle, 0);
+ releaseAudioPatchInternal(mCallRxPatch->getHandle());
mCallRxPatch.clear();
}
// release TX patch if any
if (mCallTxPatch != 0) {
- mpClientInterface->releaseAudioPatch(mCallTxPatch->mAfPatchHandle, 0);
+ releaseAudioPatchInternal(mCallTxPatch->getHandle());
mCallTxPatch.clear();
}
auto telephonyRxModule =
- mHwModules.getModuleForDeviceTypes(AUDIO_DEVICE_IN_TELEPHONY_RX, AUDIO_FORMAT_DEFAULT);
+ mHwModules.getModuleForDeviceType(AUDIO_DEVICE_IN_TELEPHONY_RX, AUDIO_FORMAT_DEFAULT);
auto telephonyTxModule =
- mHwModules.getModuleForDeviceTypes(AUDIO_DEVICE_OUT_TELEPHONY_TX, AUDIO_FORMAT_DEFAULT);
+ mHwModules.getModuleForDeviceType(AUDIO_DEVICE_OUT_TELEPHONY_TX, AUDIO_FORMAT_DEFAULT);
// retrieve Rx Source and Tx Sink device descriptors
sp<DeviceDescriptor> rxSourceDevice =
mAvailableInputDevices.getDevice(AUDIO_DEVICE_IN_TELEPHONY_RX,
@@ -535,11 +557,9 @@
ALOGE("updateCallRouting() no telephony Tx and/or RX device");
return muteWaitMs;
}
- // do not create a patch (aka Sw Bridging) if Primary HW module has declared supporting a
- // route between telephony RX to Sink device and Source device to telephony TX
- const auto &primaryModule = telephonyRxModule;
- createRxPatch = !primaryModule->supportsPatch(rxSourceDevice, rxDevices.itemAt(0));
- createTxPatch = !primaryModule->supportsPatch(txSourceDevice, txSinkDevice);
+ // createAudioPatchInternal now supports both HW / SW bridging
+ createRxPatch = true;
+ createTxPatch = true;
} else {
// If the RX device is on the primary HW module, then use legacy routing method for
// voice calls via setOutputDevice() on primary output.
@@ -563,6 +583,15 @@
// assuming the device uses audio HAL V5.0 and above
}
if (createTxPatch) { // create TX path audio patch
+ // terminate active capture if on the same HW module as the call TX source device
+ // FIXME: would be better to refine to only inputs whose profile connects to the
+ // call TX device but this information is not in the audio patch and logic here must be
+ // symmetric to the one in startInput()
+ for (const auto& activeDesc : mInputs.getActiveInputs()) {
+ if (activeDesc->hasSameHwModuleAs(txSourceDevice)) {
+ closeActiveClients(activeDesc);
+ }
+ }
mCallTxPatch = createTelephonyPatch(false /*isRx*/, txSourceDevice, delayMs);
}
@@ -576,6 +605,8 @@
if (device == nullptr) {
return nullptr;
}
+
+ // @TODO: still ignoring the address, or not dealing platform with mutliple telephony devices
if (isRx) {
patchBuilder.addSink(device).
addSource(mAvailableInputDevices.getDevice(
@@ -586,59 +617,15 @@
AUDIO_DEVICE_OUT_TELEPHONY_TX, String8(), AUDIO_FORMAT_DEFAULT));
}
- // @TODO: still ignoring the address, or not dealing platform with mutliple telephonydevices
- const sp<DeviceDescriptor> outputDevice = isRx ?
- device : mAvailableOutputDevices.getDevice(
- AUDIO_DEVICE_OUT_TELEPHONY_TX, String8(), AUDIO_FORMAT_DEFAULT);
- SortedVector<audio_io_handle_t> outputs =
- getOutputsForDevices(DeviceVector(outputDevice), mOutputs);
- const audio_io_handle_t output = selectOutput(outputs);
- // request to reuse existing output stream if one is already opened to reach the target device
- if (output != AUDIO_IO_HANDLE_NONE) {
- sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
- ALOG_ASSERT(!outputDesc->isDuplicated(), "%s() %s device output %d is duplicated", __func__,
- outputDevice->toString().c_str(), output);
- patchBuilder.addSource(outputDesc, { .stream = AUDIO_STREAM_PATCH });
+ audio_patch_handle_t patchHandle = AUDIO_PATCH_HANDLE_NONE;
+ status_t status =
+ createAudioPatchInternal(patchBuilder.patch(), &patchHandle, mUidCached, delayMs);
+ ssize_t index = mAudioPatches.indexOfKey(patchHandle);
+ if (status != NO_ERROR || index < 0) {
+ ALOGW("%s() error %d creating %s audio patch", __func__, status, isRx ? "RX" : "TX");
+ return nullptr;
}
-
- if (!isRx) {
- // terminate active capture if on the same HW module as the call TX source device
- // FIXME: would be better to refine to only inputs whose profile connects to the
- // call TX device but this information is not in the audio patch and logic here must be
- // symmetric to the one in startInput()
- for (const auto& activeDesc : mInputs.getActiveInputs()) {
- if (activeDesc->hasSameHwModuleAs(device)) {
- closeActiveClients(activeDesc);
- }
- }
- }
-
- audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
- status_t status = mpClientInterface->createAudioPatch(
- patchBuilder.patch(), &afPatchHandle, delayMs);
- ALOGW_IF(status != NO_ERROR,
- "%s() error %d creating %s audio patch", __func__, status, isRx ? "RX" : "TX");
- sp<AudioPatch> audioPatch;
- if (status == NO_ERROR) {
- audioPatch = new AudioPatch(patchBuilder.patch(), mUidCached);
- audioPatch->mAfPatchHandle = afPatchHandle;
- audioPatch->mUid = mUidCached;
- }
- return audioPatch;
-}
-
-sp<DeviceDescriptor> AudioPolicyManager::findDevice(
- const DeviceVector& devices, audio_devices_t device) const {
- DeviceVector deviceList = devices.getDevicesFromTypeMask(device);
- ALOG_ASSERT(!deviceList.isEmpty(),
- "%s() selected device type %#x is not in devices list", __func__, device);
- return deviceList.itemAt(0);
-}
-
-audio_devices_t AudioPolicyManager::getModuleDeviceTypes(
- const DeviceVector& devices, const char *moduleId) const {
- sp<HwModule> mod = mHwModules.getModuleFromName(moduleId);
- return mod != 0 ? devices.getDeviceTypesFromHwModule(mod->getHandle()) : AUDIO_DEVICE_NONE;
+ return mAudioPatches.valueAt(index);
}
bool AudioPolicyManager::isDeviceOfModule(
@@ -721,11 +708,11 @@
updateCallRouting(rxDevices, delayMs);
} else if (oldState == AUDIO_MODE_IN_CALL) {
if (mCallRxPatch != 0) {
- mpClientInterface->releaseAudioPatch(mCallRxPatch->mAfPatchHandle, 0);
+ releaseAudioPatchInternal(mCallRxPatch->getHandle());
mCallRxPatch.clear();
}
if (mCallTxPatch != 0) {
- mpClientInterface->releaseAudioPatch(mCallTxPatch->mAfPatchHandle, 0);
+ releaseAudioPatchInternal(mCallTxPatch->getHandle());
mCallTxPatch.clear();
}
setOutputDevices(mPrimaryOutput, rxDevices, force, 0);
@@ -786,27 +773,11 @@
//FIXME: workaround for truncated touch sounds
// to be removed when the problem is handled by system UI
uint32_t delayMs = 0;
- uint32_t waitMs = 0;
if (usage == AUDIO_POLICY_FORCE_FOR_COMMUNICATION) {
delayMs = TOUCH_SOUND_FIXED_DELAY_MS;
}
- if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
- DeviceVector newDevices = getNewOutputDevices(mPrimaryOutput, true /*fromCache*/);
- waitMs = updateCallRouting(newDevices, delayMs);
- }
- for (size_t i = 0; i < mOutputs.size(); i++) {
- sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
- DeviceVector newDevices = getNewOutputDevices(outputDesc, true /*fromCache*/);
- if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (outputDesc != mPrimaryOutput)) {
- // As done in setDeviceConnectionState, we could also fix default device issue by
- // preventing the force re-routing in case of default dev that distinguishes on address.
- // Let's give back to engine full device choice decision however.
- waitMs = setOutputDevices(outputDesc, newDevices, !newDevices.isEmpty(), delayMs);
- }
- if (forceVolumeReeval && !newDevices.isEmpty()) {
- applyStreamVolumes(outputDesc, newDevices.types(), waitMs, true);
- }
- }
+
+ updateCallAndOutputRouting(forceVolumeReeval, delayMs);
for (const auto& activeDesc : mInputs.getActiveInputs()) {
auto newDevice = getNewInputDevice(activeDesc);
@@ -839,7 +810,7 @@
// if explicitly requested
static const uint32_t kRelevantFlags =
(AUDIO_OUTPUT_FLAG_HW_AV_SYNC | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD |
- AUDIO_OUTPUT_FLAG_VOIP_RX);
+ AUDIO_OUTPUT_FLAG_VOIP_RX | AUDIO_OUTPUT_FLAG_MMAP_NOIRQ);
flags =
(audio_output_flags_t)((flags & kRelevantFlags) | AUDIO_OUTPUT_FLAG_DIRECT);
}
@@ -860,7 +831,7 @@
continue;
}
// reject profiles if connected device does not support codec
- if (!curProfile->deviceSupportsEncodedFormats(devices.types())) {
+ if (!curProfile->devicesSupportEncodedFormats(devices.types())) {
continue;
}
if (!directOnly) return curProfile;
@@ -1004,7 +975,7 @@
// FIXME: provide a more generic approach which is not device specific and move this back
// to getOutputForDevice.
// TODO: Remove check of AUDIO_STREAM_MUSIC once migration is completed on the app side.
- if (outputDevices.types() == AUDIO_DEVICE_OUT_TELEPHONY_TX &&
+ if (outputDevices.onlyContainsDevicesWithType(AUDIO_DEVICE_OUT_TELEPHONY_TX) &&
(*stream == AUDIO_STREAM_MUSIC || resultAttr->usage == AUDIO_USAGE_VOICE_COMMUNICATION) &&
audio_is_linear_pcm(config->format) &&
isInCall()) {
@@ -1085,9 +1056,10 @@
}
audio_config_base_t clientConfig = {.sample_rate = config->sample_rate,
+ .channel_mask = config->channel_mask,
.format = config->format,
- .channel_mask = config->channel_mask };
- *portId = AudioPort::getNextUniqueId();
+ };
+ *portId = PolicyAudioPort::getNextUniqueId();
sp<TrackClientDescriptor> clientDesc =
new TrackClientDescriptor(*portId, uid, session, resultAttr, clientConfig,
@@ -1184,9 +1156,9 @@
if (!desc->isDuplicated() && (profile == desc->mProfile)) {
// reuse direct output if currently open by the same client
// and configured with same parameters
- if ((config->sample_rate == desc->mSamplingRate) &&
- (config->format == desc->mFormat) &&
- (channelMask == desc->mChannelMask) &&
+ if ((config->sample_rate == desc->getSamplingRate()) &&
+ (config->format == desc->getFormat()) &&
+ (channelMask == desc->getChannelMask()) &&
(session == desc->mDirectClientSession)) {
desc->mDirectOpenCount++;
ALOGI("%s reusing direct output %d for session %d", __func__,
@@ -1213,10 +1185,10 @@
for (size_t j = 0; j < patch->mPatch.num_sinks; ++j) {
const struct audio_port_config *sink = &patch->mPatch.sinks[j];
if (sink->type == AUDIO_PORT_TYPE_DEVICE &&
- (sink->ext.device.type & devices.types()) != AUDIO_DEVICE_NONE &&
+ devices.containsDeviceWithType(sink->ext.device.type) &&
(address.isEmpty() || strncmp(sink->ext.device.address, address.string(),
AUDIO_DEVICE_MAX_ADDRESS_LEN) == 0)) {
- releaseAudioPatch(patch->mHandle, mUidCached);
+ releaseAudioPatch(patch->getHandle(), mUidCached);
break;
}
}
@@ -1226,13 +1198,13 @@
// only accept an output with the requested parameters
if (status != NO_ERROR ||
- (config->sample_rate != 0 && config->sample_rate != outputDesc->mSamplingRate) ||
- (config->format != AUDIO_FORMAT_DEFAULT && config->format != outputDesc->mFormat) ||
- (channelMask != 0 && channelMask != outputDesc->mChannelMask)) {
+ (config->sample_rate != 0 && config->sample_rate != outputDesc->getSamplingRate()) ||
+ (config->format != AUDIO_FORMAT_DEFAULT && config->format != outputDesc->getFormat()) ||
+ (channelMask != 0 && channelMask != outputDesc->getChannelMask())) {
ALOGV("%s failed opening direct output: output %d sample rate %d %d,"
"format %d %d, channel mask %04x %04x", __func__, output, config->sample_rate,
- outputDesc->mSamplingRate, config->format, outputDesc->mFormat,
- channelMask, outputDesc->mChannelMask);
+ outputDesc->getSamplingRate(), config->format, outputDesc->getFormat(),
+ channelMask, outputDesc->getChannelMask());
if (output != AUDIO_IO_HANDLE_NONE) {
outputDesc->close();
}
@@ -1303,7 +1275,7 @@
const struct audio_port_config *source = &patch->mPatch.sources[j];
if (source->type == AUDIO_PORT_TYPE_DEVICE &&
source->ext.device.hw_module == msdModule->getHandle()) {
- msdPatches.addAudioPatch(patch->mHandle, patch);
+ msdPatches.addAudioPatch(patch->getHandle(), patch);
}
}
}
@@ -1338,19 +1310,19 @@
// Each IOProfile represents a MixPort from audio_policy_configuration.xml
for (const auto &inProfile : inputProfiles) {
if (hwAvSync == ((inProfile->getFlags() & AUDIO_INPUT_FLAG_HW_AV_SYNC) != 0)) {
- msdProfiles.appendVector(inProfile->getAudioProfiles());
+ appendAudioProfiles(msdProfiles, inProfile->getAudioProfiles());
}
}
AudioProfileVector deviceProfiles;
for (const auto &outProfile : outputProfiles) {
if (hwAvSync == ((outProfile->getFlags() & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0)) {
- deviceProfiles.appendVector(outProfile->getAudioProfiles());
+ appendAudioProfiles(deviceProfiles, outProfile->getAudioProfiles());
}
}
struct audio_config_base bestSinkConfig;
- status_t result = msdProfiles.findBestMatchingOutputConfig(deviceProfiles,
+ status_t result = findBestMatchingOutputConfig(msdProfiles, deviceProfiles,
compressedFormatsOrder, surroundChannelMasksOrder, true /*preferHigherSamplingRates*/,
- &bestSinkConfig);
+ bestSinkConfig);
if (result != NO_ERROR) {
ALOGD("%s() no matching profiles found for device: %s, hwAvSync: %d",
__func__, outputDevice->toString().c_str(), hwAvSync);
@@ -1362,6 +1334,14 @@
// For encoded streams force direct flag to prevent downstream mixing.
sinkConfig->flags.output = static_cast<audio_output_flags_t>(
sinkConfig->flags.output | AUDIO_OUTPUT_FLAG_DIRECT);
+ if (audio_is_iec61937_compatible(sinkConfig->format)) {
+ // For formats compatible with IEC61937 encapsulation, assume that
+ // the record thread input from MSD is IEC61937 framed (for proportional buffer sizing).
+ // Add the AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO flag so downstream HAL can distinguish between
+ // raw and IEC61937 framed streams.
+ sinkConfig->flags.output = static_cast<audio_output_flags_t>(
+ sinkConfig->flags.output | AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO);
+ }
sourceConfig->sample_rate = bestSinkConfig.sample_rate;
// Specify exact channel mask to prevent guessing by bit count in PatchPanel.
sourceConfig->channel_mask = audio_channel_mask_out_to_in(bestSinkConfig.channel_mask);
@@ -1425,7 +1405,7 @@
if (audio_patches_are_equal(¤tPatch->mPatch, patch)) {
return NO_ERROR;
}
- releaseAudioPatch(currentPatch->mHandle, mUidCached);
+ releaseAudioPatch(currentPatch->getHandle(), mUidCached);
}
status_t status = installPatch(__func__, -1 /*index*/, nullptr /*patchHandle*/,
patch, 0 /*delayMs*/, mUidCached, nullptr /*patchDescPtr*/);
@@ -1503,13 +1483,13 @@
// If haptic channel is specified, use the haptic output if present.
// When using haptic output, same audio format and sample rate are required.
const uint32_t outputHapticChannelCount = audio_channel_count_from_out_mask(
- outputDesc->mChannelMask & AUDIO_CHANNEL_HAPTIC_ALL);
+ outputDesc->getChannelMask() & AUDIO_CHANNEL_HAPTIC_ALL);
if ((hapticChannelCount == 0) != (outputHapticChannelCount == 0)) {
continue;
}
if (outputHapticChannelCount >= hapticChannelCount
- && format == outputDesc->mFormat
- && samplingRate == outputDesc->mSamplingRate) {
+ && format == outputDesc->getFormat()
+ && samplingRate == outputDesc->getSamplingRate()) {
currentMatchCriteria[0] = outputHapticChannelCount;
}
@@ -1517,12 +1497,13 @@
currentMatchCriteria[1] = popcount(outputDesc->mFlags & functionalFlags);
// channel mask and channel count match
- uint32_t outputChannelCount = audio_channel_count_from_out_mask(outputDesc->mChannelMask);
+ uint32_t outputChannelCount = audio_channel_count_from_out_mask(
+ outputDesc->getChannelMask());
if (channelMask != AUDIO_CHANNEL_NONE && channelCount > 2 &&
channelCount <= outputChannelCount) {
if ((audio_channel_mask_get_representation(channelMask) ==
- audio_channel_mask_get_representation(outputDesc->mChannelMask)) &&
- ((channelMask & outputDesc->mChannelMask) == channelMask)) {
+ audio_channel_mask_get_representation(outputDesc->getChannelMask())) &&
+ ((channelMask & outputDesc->getChannelMask()) == channelMask)) {
currentMatchCriteria[2] = outputChannelCount;
}
currentMatchCriteria[3] = outputChannelCount;
@@ -1530,8 +1511,8 @@
// sampling rate match
if (samplingRate > SAMPLE_RATE_HZ_DEFAULT &&
- samplingRate <= outputDesc->mSamplingRate) {
- currentMatchCriteria[4] = outputDesc->mSamplingRate;
+ samplingRate <= outputDesc->getSamplingRate()) {
+ currentMatchCriteria[4] = outputDesc->getSamplingRate();
}
// performance flags match
@@ -1540,8 +1521,8 @@
// format match
if (format != AUDIO_FORMAT_INVALID) {
currentMatchCriteria[6] =
- AudioPort::kFormatDistanceMax -
- AudioPort::formatDistance(format, outputDesc->mFormat);
+ PolicyAudioPort::kFormatDistanceMax -
+ PolicyAudioPort::formatDistance(format, outputDesc->getFormat());
}
// primary output match
@@ -1755,14 +1736,15 @@
}
if (stream == AUDIO_STREAM_ENFORCED_AUDIBLE &&
- mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) {
+ mEngine->getForceUse(
+ AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) {
setStrategyMute(streamToStrategy(AUDIO_STREAM_ALARM), true, outputDesc);
}
// Automatically enable the remote submix input when output is started on a re routing mix
// of type MIX_TYPE_RECORDERS
- if (audio_is_remote_submix_device(devices.types()) && policyMix != NULL &&
- policyMix->mMixType == MIX_TYPE_RECORDERS) {
+ if (isSingleDeviceType(devices.types(), &audio_is_remote_submix_device) &&
+ policyMix != NULL && policyMix->mMixType == MIX_TYPE_RECORDERS) {
setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
address,
@@ -1809,7 +1791,8 @@
// Automatically disable the remote submix input when output is stopped on a
// re routing mix of type MIX_TYPE_RECORDERS
sp<AudioPolicyMix> policyMix = outputDesc->mPolicyMix.promote();
- if (audio_is_remote_submix_device(outputDesc->devices().types()) &&
+ if (isSingleDeviceType(
+ outputDesc->devices().types(), &audio_is_remote_submix_device) &&
policyMix != NULL &&
policyMix->mMixType == MIX_TYPE_RECORDERS) {
setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
@@ -2065,7 +2048,7 @@
isSoundTrigger = attributes.source == AUDIO_SOURCE_HOTWORD &&
mSoundTriggerSessions.indexOfKey(session) >= 0;
- *portId = AudioPort::getNextUniqueId();
+ *portId = PolicyAudioPort::getNextUniqueId();
clientDesc = new RecordClientDescriptor(*portId, riid, uid, session, attributes, *config,
requestedDeviceId, attributes.source, flags,
@@ -2239,16 +2222,22 @@
return status;
}
- // increment activity count before calling getNewInputDevice() below as only active sessions
+ // increment activity count before calling getNewInputDevice() below as only active sessions
// are considered for device selection
inputDesc->setClientActive(client, true);
// indicate active capture to sound trigger service if starting capture from a mic on
// primary HW module
sp<DeviceDescriptor> device = getNewInputDevice(inputDesc);
- setInputDevice(input, device, true /* force */);
+ if (device != nullptr) {
+ status = setInputDevice(input, device, true /* force */);
+ } else {
+ ALOGW("%s no new input device can be found for descriptor %d",
+ __FUNCTION__, inputDesc->getId());
+ status = BAD_VALUE;
+ }
- if (inputDesc->activeCount() == 1) {
+ if (status == NO_ERROR && inputDesc->activeCount() == 1) {
sp<AudioPolicyMix> policyMix = inputDesc->mPolicyMix.promote();
// if input maps to a dynamic policy with an activity listener, notify of state change
if ((policyMix != NULL)
@@ -2279,11 +2268,16 @@
address, "remote-submix", AUDIO_FORMAT_DEFAULT);
}
}
+ } else if (status != NO_ERROR) {
+ // Restore client activity state.
+ inputDesc->setClientActive(client, false);
+ inputDesc->stop();
}
- ALOGV("%s input %d source = %d exit", __FUNCTION__, input, client->source());
+ ALOGV("%s input %d source = %d status = %d exit",
+ __FUNCTION__, input, client->source(), status);
- return NO_ERROR;
+ return status;
}
status_t AudioPolicyManager::stopInput(audio_port_handle_t portId)
@@ -2397,7 +2391,8 @@
const sp<AudioInputDescriptor> input = mInputs.valueAt(i);
if (input->clientsList().size() == 0
|| !mAvailableInputDevices.containsAtLeastOne(input->supportedDevices())
- || (input->getAudioPort()->getFlags() & AUDIO_INPUT_FLAG_MMAP_NOIRQ) != 0) {
+ || (input->getPolicyAudioPort()->getFlags()
+ & AUDIO_INPUT_FLAG_MMAP_NOIRQ) != 0) {
inputsToClose.push_back(mInputs.keyAt(i));
} else {
bool close = false;
@@ -2457,10 +2452,12 @@
{
// if device is AUDIO_DEVICE_OUT_DEFAULT_FOR_VOLUME, return volume for device selected for this
// stream by the engine.
+ DeviceTypeSet deviceTypes = {device};
if (device == AUDIO_DEVICE_OUT_DEFAULT_FOR_VOLUME) {
- device = mEngine->getOutputDevicesForStream(stream, true /*fromCache*/).types();
+ deviceTypes = mEngine->getOutputDevicesForStream(
+ stream, true /*fromCache*/).types();
}
- return getVolumeIndex(getVolumeCurves(stream), *index, device);
+ return getVolumeIndex(getVolumeCurves(stream), *index, deviceTypes);
}
status_t AudioPolicyManager::setVolumeIndexForAttributes(const audio_attributes_t &attributes,
@@ -2485,19 +2482,20 @@
return status;
}
- audio_devices_t curSrcDevice;
+ DeviceTypeSet curSrcDevices;
auto curCurvAttrs = curves.getAttributes();
if (!curCurvAttrs.empty() && curCurvAttrs.front() != defaultAttr) {
auto attr = curCurvAttrs.front();
- curSrcDevice = mEngine->getOutputDevicesForAttributes(attr, nullptr, false).types();
+ curSrcDevices = mEngine->getOutputDevicesForAttributes(attr, nullptr, false).types();
} else if (!curves.getStreamTypes().empty()) {
auto stream = curves.getStreamTypes().front();
- curSrcDevice = mEngine->getOutputDevicesForStream(stream, false).types();
+ curSrcDevices = mEngine->getOutputDevicesForStream(stream, false).types();
} else {
ALOGE("%s: Invalid src %d: no valid attributes nor stream",__func__, vs);
return BAD_VALUE;
}
- curSrcDevice = Volume::getDeviceForVolume(curSrcDevice);
+ audio_devices_t curSrcDevice = Volume::getDeviceForVolume(curSrcDevices);
+ resetDeviceTypes(curSrcDevices, curSrcDevice);
// update volume on all outputs and streams matching the following:
// - The requested stream (or a stream matching for volume control) is active on the output
@@ -2509,21 +2507,34 @@
// no specific device volume value exists for currently selected device.
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
- audio_devices_t curDevice = desc->devices().types();
+ DeviceTypeSet curDevices = desc->devices().types();
- if (curDevice & AUDIO_DEVICE_OUT_SPEAKER_SAFE) {
- curDevice |= AUDIO_DEVICE_OUT_SPEAKER;
- curDevice &= ~AUDIO_DEVICE_OUT_SPEAKER_SAFE;
+ if (curDevices.erase(AUDIO_DEVICE_OUT_SPEAKER_SAFE)) {
+ curDevices.insert(AUDIO_DEVICE_OUT_SPEAKER);
}
-
+ if (!(desc->isActive(vs) || isInCall())) {
+ continue;
+ }
+ if (device != AUDIO_DEVICE_OUT_DEFAULT_FOR_VOLUME &&
+ curDevices.find(device) == curDevices.end()) {
+ continue;
+ }
+ bool applyVolume = false;
+ if (device != AUDIO_DEVICE_OUT_DEFAULT_FOR_VOLUME) {
+ curSrcDevices.insert(device);
+ applyVolume = (curSrcDevices.find(
+ Volume::getDeviceForVolume(curDevices)) != curSrcDevices.end());
+ } else {
+ applyVolume = !curves.hasVolumeIndexForDevice(curSrcDevice);
+ }
+ if (!applyVolume) {
+ continue; // next output
+ }
// Inter / intra volume group priority management: Loop on strategies arranged by priority
// If a higher priority strategy is active, and the output is routed to a device with a
// HW Gain management, do not change the volume
- bool applyVolume = false;
if (desc->useHwGain()) {
- if (!(desc->isActive(toVolumeSource(group)) || isInCall())) {
- continue;
- }
+ applyVolume = false;
for (const auto &productStrategy : mEngine->getOrderedProductStrategies()) {
auto activeClients = desc->clientsList(true /*activeOnly*/, productStrategy,
false /*preferredDevice*/);
@@ -2557,37 +2568,16 @@
if (!applyVolume) {
continue; // next output
}
- status_t volStatus = checkAndSetVolume(curves, vs, index, desc, curDevice,
- (vs == toVolumeSource(AUDIO_STREAM_SYSTEM)?
- TOUCH_SOUND_FIXED_DELAY_MS : 0));
- if (volStatus != NO_ERROR) {
- status = volStatus;
- }
- continue;
}
- if (!(desc->isActive(vs) || isInCall())) {
- continue;
- }
- if ((device != AUDIO_DEVICE_OUT_DEFAULT_FOR_VOLUME) && ((curDevice & device) == 0)) {
- continue;
- }
- if (device != AUDIO_DEVICE_OUT_DEFAULT_FOR_VOLUME) {
- curSrcDevice |= device;
- applyVolume = (Volume::getDeviceForVolume(curDevice) & curSrcDevice) != 0;
- } else {
- applyVolume = !curves.hasVolumeIndexForDevice(curSrcDevice);
- }
- if (applyVolume) {
- //FIXME: workaround for truncated touch sounds
- // delayed volume change for system stream to be removed when the problem is
- // handled by system UI
- status_t volStatus = checkAndSetVolume(
- curves, vs, index, desc, curDevice,
- ((vs == toVolumeSource(AUDIO_STREAM_SYSTEM))?
- TOUCH_SOUND_FIXED_DELAY_MS : 0));
- if (volStatus != NO_ERROR) {
- status = volStatus;
- }
+ //FIXME: workaround for truncated touch sounds
+ // delayed volume change for system stream to be removed when the problem is
+ // handled by system UI
+ status_t volStatus = checkAndSetVolume(
+ curves, vs, index, desc, curDevices,
+ ((vs == toVolumeSource(AUDIO_STREAM_SYSTEM))?
+ TOUCH_SOUND_FIXED_DELAY_MS : 0));
+ if (volStatus != NO_ERROR) {
+ status = volStatus;
}
}
mpClientInterface->onAudioVolumeGroupChanged(group, 0 /*flags*/);
@@ -2625,22 +2615,23 @@
{
// if device is AUDIO_DEVICE_OUT_DEFAULT_FOR_VOLUME, return volume for device selected for this
// stream by the engine.
+ DeviceTypeSet deviceTypes = {device};
if (device == AUDIO_DEVICE_OUT_DEFAULT_FOR_VOLUME) {
- device = mEngine->getOutputDevicesForAttributes(attr, nullptr, true /*fromCache*/).types();
+ DeviceTypeSet deviceTypes = mEngine->getOutputDevicesForAttributes(
+ attr, nullptr, true /*fromCache*/).types();
}
- return getVolumeIndex(getVolumeCurves(attr), index, device);
+ return getVolumeIndex(getVolumeCurves(attr), index, deviceTypes);
}
status_t AudioPolicyManager::getVolumeIndex(const IVolumeCurves &curves,
int &index,
- audio_devices_t device) const
+ const DeviceTypeSet& deviceTypes) const
{
- if (!audio_is_output_device(device)) {
+ if (isSingleDeviceType(deviceTypes, audio_is_output_device)) {
return BAD_VALUE;
}
- device = Volume::getDeviceForVolume(device);
- index = curves.getVolumeIndex(device);
- ALOGV("%s: device %08x index %d", __FUNCTION__, device, index);
+ index = curves.getVolumeIndex(deviceTypes);
+ ALOGV("%s: device %s index %d", __FUNCTION__, dumpDeviceTypes(deviceTypes).c_str(), index);
return NO_ERROR;
}
@@ -2735,12 +2726,14 @@
int session,
int id)
{
- ssize_t index = mOutputs.indexOfKey(io);
- if (index < 0) {
- index = mInputs.indexOfKey(io);
+ if (session != AUDIO_SESSION_DEVICE) {
+ ssize_t index = mOutputs.indexOfKey(io);
if (index < 0) {
- ALOGW("registerEffect() unknown io %d", io);
- return INVALID_OPERATION;
+ index = mInputs.indexOfKey(io);
+ if (index < 0) {
+ ALOGW("registerEffect() unknown io %d", io);
+ return INVALID_OPERATION;
+ }
}
}
return mEffects.registerEffect(desc, io, session, id,
@@ -2889,9 +2882,9 @@
// stereo and let audio flinger do the channel conversion if needed.
outputConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
inputConfig.channel_mask = AUDIO_CHANNEL_IN_STEREO;
- rSubmixModule->addOutputProfile(address, &outputConfig,
+ rSubmixModule->addOutputProfile(address.c_str(), &outputConfig,
AUDIO_DEVICE_OUT_REMOTE_SUBMIX, address);
- rSubmixModule->addInputProfile(address, &inputConfig,
+ rSubmixModule->addInputProfile(address.c_str(), &inputConfig,
AUDIO_DEVICE_IN_REMOTE_SUBMIX, address);
if ((res = setDeviceConnectionStateInt(deviceTypeToMakeAvailable,
@@ -2987,8 +2980,8 @@
}
}
}
- rSubmixModule->removeOutputProfile(address);
- rSubmixModule->removeInputProfile(address);
+ rSubmixModule->removeOutputProfile(address.c_str());
+ rSubmixModule->removeInputProfile(address.c_str());
} else if ((mix.mRouteFlags & MIX_ROUTE_FLAG_RENDER) == MIX_ROUTE_FLAG_RENDER) {
if (mPolicyMixes.unregisterMix(mix) != NO_ERROR) {
@@ -3029,13 +3022,13 @@
// reevaluate outputs for all given devices
for (size_t i = 0; i < devices.size(); i++) {
sp<DeviceDescriptor> devDesc = mHwModules.getDeviceDescriptor(
- devices[i].mType, devices[i].mAddress, String8(),
+ devices[i].mType, devices[i].mAddress.c_str(), String8(),
AUDIO_FORMAT_DEFAULT);
SortedVector<audio_io_handle_t> outputs;
if (checkOutputsForDevice(devDesc, AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
outputs) != NO_ERROR) {
ALOGE("setUidDeviceAffinities() error in checkOutputsForDevice for device=%08x"
- " addr=%s", devices[i].mType, devices[i].mAddress.string());
+ " addr=%s", devices[i].mType, devices[i].mAddress.c_str());
return INVALID_OPERATION;
}
}
@@ -3055,6 +3048,72 @@
return res;
}
+status_t AudioPolicyManager::setPreferredDeviceForStrategy(product_strategy_t strategy,
+ const AudioDeviceTypeAddr &device) {
+ ALOGI("%s() strategy=%d device=%08x addr=%s", __FUNCTION__,
+ strategy, device.mType, device.mAddress.c_str());
+ // strategy preferred device is only for output devices
+ if (!audio_is_output_device(device.mType)) {
+ ALOGE("%s() device=%08x is NOT an output device", __FUNCTION__, device.mType);
+ return BAD_VALUE;
+ }
+
+ status_t status = mEngine->setPreferredDeviceForStrategy(strategy, device);
+ if (status != NO_ERROR) {
+ ALOGW("Engine could not set preferred device %08x %s for strategy %d",
+ device.mType, device.mAddress.c_str(), strategy);
+ return status;
+ }
+
+ checkForDeviceAndOutputChanges();
+ updateCallAndOutputRouting();
+
+ return NO_ERROR;
+}
+
+void AudioPolicyManager::updateCallAndOutputRouting(bool forceVolumeReeval, uint32_t delayMs)
+{
+ uint32_t waitMs = 0;
+ if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
+ DeviceVector newDevices = getNewOutputDevices(mPrimaryOutput, true /*fromCache*/);
+ waitMs = updateCallRouting(newDevices, delayMs);
+ }
+ for (size_t i = 0; i < mOutputs.size(); i++) {
+ sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
+ DeviceVector newDevices = getNewOutputDevices(outputDesc, true /*fromCache*/);
+ if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (outputDesc != mPrimaryOutput)) {
+ // As done in setDeviceConnectionState, we could also fix default device issue by
+ // preventing the force re-routing in case of default dev that distinguishes on address.
+ // Let's give back to engine full device choice decision however.
+ waitMs = setOutputDevices(outputDesc, newDevices, !newDevices.isEmpty(), delayMs);
+ }
+ if (forceVolumeReeval && !newDevices.isEmpty()) {
+ applyStreamVolumes(outputDesc, newDevices.types(), waitMs, true);
+ }
+ }
+}
+
+status_t AudioPolicyManager::removePreferredDeviceForStrategy(product_strategy_t strategy)
+{
+ ALOGI("%s() strategy=%d", __FUNCTION__, strategy);
+
+ status_t status = mEngine->removePreferredDeviceForStrategy(strategy);
+ if (status != NO_ERROR) {
+ ALOGW("Engine could not remove preferred device for strategy %d", strategy);
+ return status;
+ }
+
+ checkForDeviceAndOutputChanges();
+ updateCallAndOutputRouting();
+
+ return NO_ERROR;
+}
+
+status_t AudioPolicyManager::getPreferredDeviceForStrategy(product_strategy_t strategy,
+ AudioDeviceTypeAddr &device) {
+ return mEngine->getPreferredDeviceForStrategy(strategy, device);
+}
+
void AudioPolicyManager::dump(String8 *dst) const
{
dst->appendFormat("\nAudioPolicyManager Dump: %p\n", this);
@@ -3200,7 +3259,7 @@
ALOGV("%s() profile %sfound with name: %s, "
"sample rate: %u, format: 0x%x, channel_mask: 0x%x, output flags: 0x%x",
__FUNCTION__, profile != 0 ? "" : "NOT ",
- (profile != 0 ? profile->getTagName().string() : "null"),
+ (profile != 0 ? profile->getTagName().c_str() : "null"),
config.sample_rate, config.format, config.channel_mask, output_flags);
return (profile != 0);
}
@@ -3302,16 +3361,16 @@
return BAD_VALUE;
}
-status_t AudioPolicyManager::createAudioPatch(const struct audio_patch *patch,
- audio_patch_handle_t *handle,
- uid_t uid)
+status_t AudioPolicyManager::createAudioPatchInternal(const struct audio_patch *patch,
+ audio_patch_handle_t *handle,
+ uid_t uid, uint32_t delayMs,
+ const sp<SourceClientDescriptor>& sourceDesc)
{
- ALOGV("createAudioPatch()");
-
+ ALOGV("%s", __func__);
if (handle == NULL || patch == NULL) {
return BAD_VALUE;
}
- ALOGV("createAudioPatch() num sources %d num sinks %d", patch->num_sources, patch->num_sinks);
+ ALOGV("%s num sources %d num sinks %d", __func__, patch->num_sources, patch->num_sinks);
if (!audio_patch_is_valid(patch)) {
return BAD_VALUE;
@@ -3333,22 +3392,22 @@
sp<AudioPatch> patchDesc;
ssize_t index = mAudioPatches.indexOfKey(*handle);
- ALOGV("createAudioPatch source id %d role %d type %d", patch->sources[0].id,
- patch->sources[0].role,
- patch->sources[0].type);
+ ALOGV("%s source id %d role %d type %d", __func__, patch->sources[0].id,
+ patch->sources[0].role,
+ patch->sources[0].type);
#if LOG_NDEBUG == 0
for (size_t i = 0; i < patch->num_sinks; i++) {
- ALOGV("createAudioPatch sink %zu: id %d role %d type %d", i, patch->sinks[i].id,
- patch->sinks[i].role,
- patch->sinks[i].type);
+ ALOGV("%s sink %zu: id %d role %d type %d", __func__ ,i, patch->sinks[i].id,
+ patch->sinks[i].role,
+ patch->sinks[i].type);
}
#endif
if (index >= 0) {
patchDesc = mAudioPatches.valueAt(index);
- ALOGV("createAudioPatch() mUidCached %d patchDesc->mUid %d uid %d",
- mUidCached, patchDesc->mUid, uid);
- if (patchDesc->mUid != mUidCached && uid != patchDesc->mUid) {
+ ALOGV("%s mUidCached %d patchDesc->mUid %d uid %d",
+ __func__, mUidCached, patchDesc->getUid(), uid);
+ if (patchDesc->getUid() != mUidCached && uid != patchDesc->getUid()) {
return INVALID_OPERATION;
}
} else {
@@ -3358,15 +3417,15 @@
if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) {
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.getOutputFromId(patch->sources[0].id);
if (outputDesc == NULL) {
- ALOGV("createAudioPatch() output not found for id %d", patch->sources[0].id);
+ ALOGV("%s output not found for id %d", __func__, patch->sources[0].id);
return BAD_VALUE;
}
ALOG_ASSERT(!outputDesc->isDuplicated(),"duplicated output %d in source in ports",
outputDesc->mIoHandle);
if (patchDesc != 0) {
if (patchDesc->mPatch.sources[0].id != patch->sources[0].id) {
- ALOGV("createAudioPatch() source id differs for patch current id %d new id %d",
- patchDesc->mPatch.sources[0].id, patch->sources[0].id);
+ ALOGV("%s source id differs for patch current id %d new id %d",
+ __func__, patchDesc->mPatch.sources[0].id, patch->sources[0].id);
return BAD_VALUE;
}
}
@@ -3375,13 +3434,13 @@
// Only support mix to devices connection
// TODO add support for mix to mix connection
if (patch->sinks[i].type != AUDIO_PORT_TYPE_DEVICE) {
- ALOGV("createAudioPatch() source mix but sink is not a device");
+ ALOGV("%s source mix but sink is not a device", __func__);
return INVALID_OPERATION;
}
sp<DeviceDescriptor> devDesc =
mAvailableOutputDevices.getDeviceFromId(patch->sinks[i].id);
if (devDesc == 0) {
- ALOGV("createAudioPatch() out device not found for id %d", patch->sinks[i].id);
+ ALOGV("%s out device not found for id %d", __func__, patch->sinks[i].id);
return BAD_VALUE;
}
@@ -3393,8 +3452,7 @@
patch->sources[0].channel_mask,
NULL, // updatedChannelMask
AUDIO_OUTPUT_FLAG_NONE /*FIXME*/)) {
- ALOGV("createAudioPatch() profile not supported for device %08x",
- devDesc->type());
+ ALOGV("%s profile not supported for device %08x", __func__, devDesc->type());
return INVALID_OPERATION;
}
devices.add(devDesc);
@@ -3404,19 +3462,19 @@
}
// TODO: reconfigure output format and channels here
- ALOGV("createAudioPatch() setting device %08x on output %d",
- devices.types(), outputDesc->mIoHandle);
+ ALOGV("%s setting device %s on output %d",
+ __func__, dumpDeviceTypes(devices.types()).c_str(), outputDesc->mIoHandle);
setOutputDevices(outputDesc, devices, true, 0, handle);
index = mAudioPatches.indexOfKey(*handle);
if (index >= 0) {
if (patchDesc != 0 && patchDesc != mAudioPatches.valueAt(index)) {
- ALOGW("createAudioPatch() setOutputDevice() did not reuse the patch provided");
+ ALOGW("%s setOutputDevice() did not reuse the patch provided", __func__);
}
patchDesc = mAudioPatches.valueAt(index);
- patchDesc->mUid = uid;
- ALOGV("createAudioPatch() success");
+ patchDesc->setUid(uid);
+ ALOGV("%s success", __func__);
} else {
- ALOGW("createAudioPatch() setOutputDevice() failed to create a patch");
+ ALOGW("%s setOutputDevice() failed to create a patch", __func__);
return INVALID_OPERATION;
}
} else if (patch->sources[0].type == AUDIO_PORT_TYPE_DEVICE) {
@@ -3455,19 +3513,19 @@
return INVALID_OPERATION;
}
// TODO: reconfigure output format and channels here
- ALOGV("%s() setting device %s on output %d", __func__,
+ ALOGV("%s setting device %s on output %d", __func__,
device->toString().c_str(), inputDesc->mIoHandle);
setInputDevice(inputDesc->mIoHandle, device, true, handle);
index = mAudioPatches.indexOfKey(*handle);
if (index >= 0) {
if (patchDesc != 0 && patchDesc != mAudioPatches.valueAt(index)) {
- ALOGW("createAudioPatch() setInputDevice() did not reuse the patch provided");
+ ALOGW("%s setInputDevice() did not reuse the patch provided", __func__);
}
patchDesc = mAudioPatches.valueAt(index);
- patchDesc->mUid = uid;
- ALOGV("createAudioPatch() success");
+ patchDesc->setUid(uid);
+ ALOGV("%s success", __func__);
} else {
- ALOGW("createAudioPatch() setInputDevice() failed to create a patch");
+ ALOGW("%s setInputDevice() failed to create a patch", __func__);
return INVALID_OPERATION;
}
} else if (patch->sinks[0].type == AUDIO_PORT_TYPE_DEVICE) {
@@ -3485,55 +3543,96 @@
//update source and sink with our own data as the data passed in the patch may
// be incomplete.
- struct audio_patch newPatch = *patch;
- srcDevice->toAudioPortConfig(&newPatch.sources[0], &patch->sources[0]);
+ PatchBuilder patchBuilder;
+ audio_port_config sourcePortConfig = {};
+ srcDevice->toAudioPortConfig(&sourcePortConfig, &patch->sources[0]);
+ patchBuilder.addSource(sourcePortConfig);
for (size_t i = 0; i < patch->num_sinks; i++) {
if (patch->sinks[i].type != AUDIO_PORT_TYPE_DEVICE) {
- ALOGV("createAudioPatch() source device but one sink is not a device");
+ ALOGV("%s source device but one sink is not a device", __func__);
return INVALID_OPERATION;
}
-
sp<DeviceDescriptor> sinkDevice =
mAvailableOutputDevices.getDeviceFromId(patch->sinks[i].id);
if (sinkDevice == 0) {
return BAD_VALUE;
}
- sinkDevice->toAudioPortConfig(&newPatch.sinks[i], &patch->sinks[i]);
+ audio_port_config sinkPortConfig = {};
+ sinkDevice->toAudioPortConfig(&sinkPortConfig, &patch->sinks[i]);
+ patchBuilder.addSink(sinkPortConfig);
// create a software bridge in PatchPanel if:
// - source and sink devices are on different HW modules OR
// - audio HAL version is < 3.0
// - audio HAL version is >= 3.0 but no route has been declared between devices
+ // - called from startAudioSource (aka sourceDesc != nullptr) and source device does
+ // not have a gain controller
if (!srcDevice->hasSameHwModuleAs(sinkDevice) ||
(srcDevice->getModuleVersionMajor() < 3) ||
- !srcDevice->getModule()->supportsPatch(srcDevice, sinkDevice)) {
+ !srcDevice->getModule()->supportsPatch(srcDevice, sinkDevice) ||
+ (sourceDesc != nullptr &&
+ srcDevice->getAudioPort()->getGains().size() == 0)) {
// support only one sink device for now to simplify output selection logic
if (patch->num_sinks > 1) {
return INVALID_OPERATION;
}
- SortedVector<audio_io_handle_t> outputs =
- getOutputsForDevices(DeviceVector(sinkDevice), mOutputs);
- // if the sink device is reachable via an opened output stream, request to go via
- // this output stream by adding a second source to the patch description
- const audio_io_handle_t output = selectOutput(outputs);
- if (output != AUDIO_IO_HANDLE_NONE) {
- sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
- if (outputDesc->isDuplicated()) {
+ audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
+ if (sourceDesc != nullptr) {
+ // take care of dynamic routing for SwOutput selection,
+ audio_attributes_t attributes = sourceDesc->attributes();
+ audio_stream_type_t stream = sourceDesc->stream();
+ audio_attributes_t resultAttr;
+ audio_config_t config = AUDIO_CONFIG_INITIALIZER;
+ config.sample_rate = sourceDesc->config().sample_rate;
+ config.channel_mask = sourceDesc->config().channel_mask;
+ config.format = sourceDesc->config().format;
+ audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE;
+ audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
+ bool isRequestedDeviceForExclusiveUse = false;
+ std::vector<sp<SwAudioOutputDescriptor>> secondaryOutputs;
+ getOutputForAttrInt(&resultAttr, &output, AUDIO_SESSION_NONE, &attributes,
+ &stream, sourceDesc->uid(), &config, &flags,
+ &selectedDeviceId, &isRequestedDeviceForExclusiveUse,
+ &secondaryOutputs);
+ if (output == AUDIO_IO_HANDLE_NONE) {
+ ALOGV("%s no output for device %s",
+ __FUNCTION__, sinkDevice->toString().c_str());
return INVALID_OPERATION;
}
- outputDesc->toAudioPortConfig(&newPatch.sources[1], &patch->sources[0]);
- newPatch.sources[1].ext.mix.usecase.stream = AUDIO_STREAM_PATCH;
- newPatch.num_sources = 2;
+ } else {
+ SortedVector<audio_io_handle_t> outputs =
+ getOutputsForDevices(DeviceVector(sinkDevice), mOutputs);
+ // if the sink device is reachable via an opened output stream, request to
+ // go via this output stream by adding a second source to the patch
+ // description
+ output = selectOutput(outputs);
+ }
+ if (output != AUDIO_IO_HANDLE_NONE) {
+ sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
+ if (outputDesc->isDuplicated()) {
+ ALOGV("%s output for device %s is duplicated",
+ __FUNCTION__, sinkDevice->toString().c_str());
+ return INVALID_OPERATION;
+ }
+ audio_port_config srcMixPortConfig = {};
+ outputDesc->toAudioPortConfig(&srcMixPortConfig, &patch->sources[0]);
+ if (sourceDesc != nullptr) {
+ sourceDesc->setSwOutput(outputDesc);
+ }
+ // for volume control, we may need a valid stream
+ srcMixPortConfig.ext.mix.usecase.stream = sourceDesc != nullptr ?
+ sourceDesc->stream() : AUDIO_STREAM_PATCH;
+ patchBuilder.addSource(srcMixPortConfig);
}
}
}
// TODO: check from routing capabilities in config file and other conflicting patches
- status_t status = installPatch(__func__, index, handle, &newPatch, 0, uid, &patchDesc);
+ status_t status = installPatch(
+ __func__, index, handle, patchBuilder.patch(), delayMs, uid, &patchDesc);
if (status != NO_ERROR) {
- ALOGW("createAudioPatch() patch panel could not connect device patch, error %d",
- status);
+ ALOGW("%s patch panel could not connect device patch, error %d", __func__, status);
return INVALID_OPERATION;
}
} else {
@@ -3556,18 +3655,29 @@
return BAD_VALUE;
}
sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
- ALOGV("releaseAudioPatch() mUidCached %d patchDesc->mUid %d uid %d",
- mUidCached, patchDesc->mUid, uid);
- if (patchDesc->mUid != mUidCached && uid != patchDesc->mUid) {
+ ALOGV("%s() mUidCached %d patchDesc->mUid %d uid %d",
+ __func__, mUidCached, patchDesc->getUid(), uid);
+ if (patchDesc->getUid() != mUidCached && uid != patchDesc->getUid()) {
return INVALID_OPERATION;
}
+ return releaseAudioPatchInternal(handle);
+}
+status_t AudioPolicyManager::releaseAudioPatchInternal(audio_patch_handle_t handle,
+ uint32_t delayMs)
+{
+ ALOGV("%s patch %d", __func__, handle);
+ if (mAudioPatches.indexOfKey(handle) < 0) {
+ ALOGE("%s: no patch found with handle=%d", __func__, handle);
+ return BAD_VALUE;
+ }
+ sp<AudioPatch> patchDesc = mAudioPatches.valueFor(handle);
struct audio_patch *patch = &patchDesc->mPatch;
- patchDesc->mUid = mUidCached;
+ patchDesc->setUid(mUidCached);
if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) {
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.getOutputFromId(patch->sources[0].id);
if (outputDesc == NULL) {
- ALOGV("releaseAudioPatch() output not found for id %d", patch->sources[0].id);
+ ALOGV("%s output not found for id %d", __func__, patch->sources[0].id);
return BAD_VALUE;
}
@@ -3580,7 +3690,7 @@
if (patch->sinks[0].type == AUDIO_PORT_TYPE_MIX) {
sp<AudioInputDescriptor> inputDesc = mInputs.getInputFromId(patch->sinks[0].id);
if (inputDesc == NULL) {
- ALOGV("releaseAudioPatch() input not found for id %d", patch->sinks[0].id);
+ ALOGV("%s input not found for id %d", __func__, patch->sinks[0].id);
return BAD_VALUE;
}
setInputDevice(inputDesc->mIoHandle,
@@ -3588,10 +3698,11 @@
true,
NULL);
} else if (patch->sinks[0].type == AUDIO_PORT_TYPE_DEVICE) {
- status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
- ALOGV("releaseAudioPatch() patch panel returned %d patchHandle %d",
- status, patchDesc->mAfPatchHandle);
- removeAudioPatch(patchDesc->mHandle);
+ status_t status =
+ mpClientInterface->releaseAudioPatch(patchDesc->getAfHandle(), delayMs);
+ ALOGV("%s patch panel returned %d patchHandle %d",
+ __func__, status, patchDesc->getAfHandle());
+ removeAudioPatch(patchDesc->getHandle());
nextAudioPortGeneration();
mpClientInterface->onAudioPatchListUpdate();
} else {
@@ -3689,7 +3800,7 @@
{
for (ssize_t i = (ssize_t)mAudioPatches.size() - 1; i >= 0; i--) {
sp<AudioPatch> patchDesc = mAudioPatches.valueAt(i);
- if (patchDesc->mUid == uid) {
+ if (patchDesc->getUid() == uid) {
releaseAudioPatch(mAudioPatches.keyAt(i), uid);
}
}
@@ -3823,13 +3934,10 @@
return BAD_VALUE;
}
- *portId = AudioPort::getNextUniqueId();
-
- struct audio_patch dummyPatch = {};
- sp<AudioPatch> patchDesc = new AudioPatch(&dummyPatch, uid);
+ *portId = PolicyAudioPort::getNextUniqueId();
sp<SourceClientDescriptor> sourceDesc =
- new SourceClientDescriptor(*portId, uid, *attributes, patchDesc, srcDevice,
+ new SourceClientDescriptor(*portId, uid, *attributes, *source, srcDevice,
mEngine->getStreamTypeForAttributes(*attributes),
mEngine->getProductStrategyForAttributes(*attributes),
toVolumeSource(*attributes));
@@ -3849,7 +3957,6 @@
disconnectAudioSource(sourceDesc);
audio_attributes_t attributes = sourceDesc->attributes();
- audio_stream_type_t stream = sourceDesc->stream();
sp<DeviceDescriptor> srcDevice = sourceDesc->srcDevice();
DeviceVector sinkDevices =
@@ -3858,90 +3965,55 @@
sp<DeviceDescriptor> sinkDevice = sinkDevices.itemAt(0);
ALOG_ASSERT(mAvailableOutputDevices.contains(sinkDevice), "%s: Device %s not available",
__FUNCTION__, sinkDevice->toString().c_str());
-
- audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
-
- if (srcDevice->hasSameHwModuleAs(sinkDevice) &&
- srcDevice->getModuleVersionMajor() >= 3 &&
- sinkDevice->getModule()->supportsPatch(srcDevice, sinkDevice) &&
- srcDevice->getAudioPort()->mGains.size() > 0) {
- ALOGV("%s Device to Device route supported by >=3.0 HAL", __FUNCTION__);
- // TODO: may explicitly specify whether we should use HW or SW patch
- // create patch between src device and output device
- // create Hwoutput and add to mHwOutputs
- } else {
- audio_attributes_t resultAttr;
- audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
- audio_config_t config = AUDIO_CONFIG_INITIALIZER;
- config.sample_rate = sourceDesc->config().sample_rate;
- config.channel_mask = sourceDesc->config().channel_mask;
- config.format = sourceDesc->config().format;
- audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE;
- audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
- bool isRequestedDeviceForExclusiveUse = false;
- std::vector<sp<SwAudioOutputDescriptor>> secondaryOutputs;
- getOutputForAttrInt(&resultAttr, &output, AUDIO_SESSION_NONE,
- &attributes, &stream, sourceDesc->uid(), &config, &flags,
- &selectedDeviceId, &isRequestedDeviceForExclusiveUse,
- &secondaryOutputs);
- if (output == AUDIO_IO_HANDLE_NONE) {
- ALOGV("%s no output for device %08x", __FUNCTION__, sinkDevices.types());
- return INVALID_OPERATION;
- }
- sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
- if (outputDesc->isDuplicated()) {
- ALOGV("%s output for device %08x is duplicated", __FUNCTION__, sinkDevices.types());
- return INVALID_OPERATION;
- }
- status_t status = outputDesc->start();
+ PatchBuilder patchBuilder;
+ patchBuilder.addSink(sinkDevice).addSource(srcDevice);
+ audio_patch_handle_t handle = AUDIO_PATCH_HANDLE_NONE;
+ status_t status =
+ createAudioPatchInternal(patchBuilder.patch(), &handle, mUidCached, 0, sourceDesc);
+ if (status != NO_ERROR || mAudioPatches.indexOfKey(handle) < 0) {
+ ALOGW("%s patch panel could not connect device patch, error %d", __func__, status);
+ return INVALID_OPERATION;
+ }
+ sourceDesc->setPatchHandle(handle);
+ // SW Bridge? (@todo: HW bridge, keep track of HwOutput for device selection "reconsideration")
+ sp<SwAudioOutputDescriptor> swOutput = sourceDesc->swOutput().promote();
+ if (swOutput != 0) {
+ status = swOutput->start();
if (status != NO_ERROR) {
- return status;
+ goto FailureSourceAdded;
}
-
- // create a special patch with no sink and two sources:
- // - the second source indicates to PatchPanel through which output mix this patch should
- // be connected as well as the stream type for volume control
- // - the sink is defined by whatever output device is currently selected for the output
- // though which this patch is routed.
- PatchBuilder patchBuilder;
- patchBuilder.addSource(srcDevice).addSource(outputDesc, { .stream = stream });
- status = mpClientInterface->createAudioPatch(patchBuilder.patch(),
- &afPatchHandle,
- 0);
- ALOGV("%s patch panel returned %d patchHandle %d", __FUNCTION__,
- status, afPatchHandle);
- sourceDesc->patchDesc()->mPatch = *patchBuilder.patch();
- if (status != NO_ERROR) {
- ALOGW("%s patch panel could not connect device patch, error %d",
- __FUNCTION__, status);
- return INVALID_OPERATION;
- }
-
- if (outputDesc->getClient(sourceDesc->portId()) != nullptr) {
+ if (swOutput->getClient(sourceDesc->portId()) != nullptr) {
ALOGW("%s source portId has already been attached to outputDesc", __func__);
- return INVALID_OPERATION;
+ goto FailureReleasePatch;
}
- outputDesc->addClient(sourceDesc);
-
+ swOutput->addClient(sourceDesc);
uint32_t delayMs = 0;
- status = startSource(outputDesc, sourceDesc, &delayMs);
-
+ status = startSource(swOutput, sourceDesc, &delayMs);
if (status != NO_ERROR) {
- mpClientInterface->releaseAudioPatch(sourceDesc->patchDesc()->mAfPatchHandle, 0);
- outputDesc->removeClient(sourceDesc->portId());
- outputDesc->stop();
- return status;
+ ALOGW("%s failed to start source, error %d", __FUNCTION__, status);
+ goto FailureSourceActive;
}
- sourceDesc->setSwOutput(outputDesc);
if (delayMs != 0) {
usleep(delayMs * 1000);
}
+ } else {
+ sp<HwAudioOutputDescriptor> hwOutputDesc = sourceDesc->hwOutput().promote();
+ if (hwOutputDesc != 0) {
+ // create Hwoutput and add to mHwOutputs
+ } else {
+ ALOGW("%s source has neither SW nor HW output", __FUNCTION__);
+ }
}
-
- sourceDesc->patchDesc()->mAfPatchHandle = afPatchHandle;
- addAudioPatch(sourceDesc->patchDesc()->mHandle, sourceDesc->patchDesc());
-
return NO_ERROR;
+
+FailureSourceActive:
+ swOutput->stop();
+ releaseOutput(sourceDesc->portId());
+FailureSourceAdded:
+ sourceDesc->setSwOutput(nullptr);
+FailureReleasePatch:
+ releaseAudioPatchInternal(handle);
+ return INVALID_OPERATION;
}
status_t AudioPolicyManager::stopAudioSource(audio_port_handle_t portId)
@@ -3999,7 +4071,7 @@
float AudioPolicyManager::getStreamVolumeDB(
audio_stream_type_t stream, int index, audio_devices_t device)
{
- return computeVolume(getVolumeCurves(stream), toVolumeSource(stream), index, device);
+ return computeVolume(getVolumeCurves(stream), toVolumeSource(stream), index, {device});
}
status_t AudioPolicyManager::getSurroundFormats(unsigned int *numSurroundFormats,
@@ -4099,12 +4171,12 @@
sp<SwAudioOutputDescriptor> outputDesc;
bool profileUpdated = false;
- DeviceVector hdmiOutputDevices = mAvailableOutputDevices.getDevicesFromTypeMask(
- AUDIO_DEVICE_OUT_HDMI);
+ DeviceVector hdmiOutputDevices = mAvailableOutputDevices.getDevicesFromType(
+ AUDIO_DEVICE_OUT_HDMI);
for (size_t i = 0; i < hdmiOutputDevices.size(); i++) {
// Simulate reconnection to update enabled surround sound formats.
- String8 address = hdmiOutputDevices[i]->address();
- String8 name = hdmiOutputDevices[i]->getName();
+ String8 address = String8(hdmiOutputDevices[i]->address().c_str());
+ std::string name = hdmiOutputDevices[i]->getName();
status_t status = setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_HDMI,
AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
address.c_str(),
@@ -4121,12 +4193,12 @@
profileUpdated |= (status == NO_ERROR);
}
// FIXME: Why doing this for input HDMI devices if we don't augment their reported formats?
- DeviceVector hdmiInputDevices = mAvailableInputDevices.getDevicesFromTypeMask(
+ DeviceVector hdmiInputDevices = mAvailableInputDevices.getDevicesFromType(
AUDIO_DEVICE_IN_HDMI);
for (size_t i = 0; i < hdmiInputDevices.size(); i++) {
// Simulate reconnection to update enabled surround sound formats.
- String8 address = hdmiInputDevices[i]->address();
- String8 name = hdmiInputDevices[i]->getName();
+ String8 address = String8(hdmiInputDevices[i]->address().c_str());
+ std::string name = hdmiInputDevices[i]->getName();
status_t status = setDeviceConnectionStateInt(AUDIO_DEVICE_IN_HDMI,
AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
address.c_str(),
@@ -4179,33 +4251,22 @@
status_t AudioPolicyManager::disconnectAudioSource(const sp<SourceClientDescriptor>& sourceDesc)
{
ALOGV("%s port Id %d", __FUNCTION__, sourceDesc->portId());
-
- sp<AudioPatch> patchDesc = mAudioPatches.valueFor(sourceDesc->patchDesc()->mHandle);
- if (patchDesc == 0) {
- ALOGW("%s source has no patch with handle %d", __FUNCTION__,
- sourceDesc->patchDesc()->mHandle);
- return BAD_VALUE;
- }
- removeAudioPatch(sourceDesc->patchDesc()->mHandle);
-
- sp<SwAudioOutputDescriptor> swOutputDesc = sourceDesc->swOutput().promote();
- if (swOutputDesc != 0) {
- status_t status = stopSource(swOutputDesc, sourceDesc);
+ sp<SwAudioOutputDescriptor> swOutput = sourceDesc->swOutput().promote();
+ if (swOutput != 0) {
+ status_t status = stopSource(swOutput, sourceDesc);
if (status == NO_ERROR) {
- swOutputDesc->stop();
+ swOutput->stop();
}
- swOutputDesc->removeClient(sourceDesc->portId());
- mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
+ releaseOutput(sourceDesc->portId());
} else {
sp<HwAudioOutputDescriptor> hwOutputDesc = sourceDesc->hwOutput().promote();
if (hwOutputDesc != 0) {
- // release patch between src device and output device
// close Hwoutput and remove from mHwOutputs
} else {
ALOGW("%s source has neither SW nor HW output", __FUNCTION__);
}
}
- return NO_ERROR;
+ return releaseAudioPatchInternal(sourceDesc->getPatchHandle());
}
sp<SourceClientDescriptor> AudioPolicyManager::getSourceForAttributesOnOutput(
@@ -4294,17 +4355,8 @@
: AudioPolicyManager(clientInterface, false /*forTesting*/)
{
loadConfig();
- initialize();
}
-// This check is to catch any legacy platform updating to Q without having
-// switched to XML since its deprecation on O.
-// TODO: after Q release, remove this check and flag as XML is now the only
-// option and all legacy platform should have transitioned to XML.
-#ifndef USE_XML_AUDIO_POLICY_CONF
-#error Audio policy no longer supports legacy .conf configuration format
-#endif
-
void AudioPolicyManager::loadConfig() {
if (deserializeAudioPolicyXmlConfig(getConfig()) != NO_ERROR) {
ALOGE("could not load audio policy configuration file, setting defaults");
@@ -4313,17 +4365,18 @@
}
status_t AudioPolicyManager::initialize() {
- // Once policy config has been parsed, retrieve an instance of the engine and initialize it.
- audio_policy::EngineInstance *engineInstance = audio_policy::EngineInstance::getInstance();
- if (!engineInstance) {
- ALOGE("%s: Could not get an instance of policy engine", __FUNCTION__);
- return NO_INIT;
- }
- // Retrieve the Policy Manager Interface
- mEngine = engineInstance->queryInterface<AudioPolicyManagerInterface>();
- if (mEngine == NULL) {
- ALOGE("%s: Failed to get Policy Engine Interface", __FUNCTION__);
- return NO_INIT;
+ {
+ auto engLib = EngineLibrary::load(
+ "libaudiopolicyengine" + getConfig().getEngineLibraryNameSuffix() + ".so");
+ if (!engLib) {
+ ALOGE("%s: Failed to load the engine library", __FUNCTION__);
+ return NO_INIT;
+ }
+ mEngine = engLib->createEngine();
+ if (mEngine == nullptr) {
+ ALOGE("%s: Failed to instantiate the APM engine", __FUNCTION__);
+ return NO_INIT;
+ }
}
mEngine->setObserver(this);
status_t status = mEngine->initCheck();
@@ -4445,7 +4498,7 @@
// give a valid ID to an attached device once confirmed it is reachable
if (!device->isAttached()) {
device->attach(hwModule);
- device->importAudioPort(inProfile, true);
+ device->importAudioPortAndPickAudioProfile(inProfile, true);
}
}
inputDesc->close();
@@ -4476,11 +4529,11 @@
}
// If microphones address is empty, set it according to device type
for (size_t i = 0; i < mAvailableInputDevices.size(); i++) {
- if (mAvailableInputDevices[i]->address().isEmpty()) {
+ if (mAvailableInputDevices[i]->address().empty()) {
if (mAvailableInputDevices[i]->type() == AUDIO_DEVICE_IN_BUILTIN_MIC) {
- mAvailableInputDevices[i]->setAddress(String8(AUDIO_BOTTOM_MICROPHONE_ADDRESS));
+ mAvailableInputDevices[i]->setAddress(AUDIO_BOTTOM_MICROPHONE_ADDRESS);
} else if (mAvailableInputDevices[i]->type() == AUDIO_DEVICE_IN_BACK_MIC) {
- mAvailableInputDevices[i]->setAddress(String8(AUDIO_BACK_MICROPHONE_ADDRESS));
+ mAvailableInputDevices[i]->setAddress(AUDIO_BACK_MICROPHONE_ADDRESS);
}
}
}
@@ -4525,7 +4578,7 @@
const sp<SwAudioOutputDescriptor>& outputDesc)
{
mOutputs.add(output, outputDesc);
- applyStreamVolumes(outputDesc, AUDIO_DEVICE_NONE, 0 /* delayMs */, true /* force */);
+ applyStreamVolumes(outputDesc, DeviceTypeSet(), 0 /* delayMs */, true /* force */);
updateMono(output); // update mono status when adding to output list
selectOutputForMusicEffects();
nextAudioPortGeneration();
@@ -4549,7 +4602,7 @@
SortedVector<audio_io_handle_t>& outputs)
{
audio_devices_t deviceType = device->type();
- const String8 &address = device->address();
+ const String8 &address = String8(device->address().c_str());
sp<SwAudioOutputDescriptor> desc;
if (audio_device_is_digital(deviceType)) {
@@ -4562,7 +4615,7 @@
for (size_t i = 0; i < mOutputs.size(); i++) {
desc = mOutputs.valueAt(i);
if (!desc->isDuplicated() && desc->supportsDevice(device)
- && desc->deviceSupportsEncodedFormats(deviceType)) {
+ && desc->devicesSupportEncodedFormats({deviceType})) {
ALOGV("checkOutputsForDevice(): adding opened output %d on device %s",
mOutputs.keyAt(i), device->toString().c_str());
outputs.add(mOutputs.keyAt(i));
@@ -4601,7 +4654,7 @@
// matching profile: save the sample rates, format and channel masks supported
// by the profile in our device descriptor
if (audio_device_is_digital(deviceType)) {
- device->importAudioPort(profile);
+ device->importAudioPortAndPickAudioProfile(profile);
}
break;
}
@@ -4617,7 +4670,7 @@
}
ALOGV("opening output for device %08x with params %s profile %p name %s",
- deviceType, address.string(), profile.get(), profile->getName().string());
+ deviceType, address.string(), profile.get(), profile->getName().c_str());
desc = new SwAudioOutputDescriptor(profile, mpClientInterface);
audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
status_t status = desc->open(nullptr, DeviceVector(device),
@@ -4703,7 +4756,7 @@
outputs.add(output);
// Load digital format info only for digital devices
if (audio_device_is_digital(deviceType)) {
- device->importAudioPort(profile);
+ device->importAudioPortAndPickAudioProfile(profile);
}
if (device_distinguishes_on_address(deviceType)) {
@@ -4727,7 +4780,7 @@
if (!desc->isDuplicated()) {
// exact match on device
if (device_distinguishes_on_address(deviceType) && desc->supportsDevice(device)
- && desc->deviceSupportsEncodedFormats(deviceType)) {
+ && desc->devicesSupportEncodedFormats({deviceType})) {
outputs.add(mOutputs.keyAt(i));
} else if (!mAvailableOutputDevices.containsAtLeastOne(desc->supportedDevices())) {
ALOGV("checkOutputsForDevice(): disconnecting adding output %d",
@@ -4797,7 +4850,7 @@
desc = mInputs.valueAt(input_index);
if (desc->mProfile == profile) {
if (audio_device_is_digital(device->type())) {
- device->importAudioPort(profile);
+ device->importAudioPortAndPickAudioProfile(profile);
}
break;
}
@@ -4821,7 +4874,7 @@
&input);
if (status == NO_ERROR) {
- const String8& address = device->address();
+ const String8& address = String8(device->address().c_str());
if (!address.isEmpty()) {
char *param = audio_device_address_to_parameter(device->type(), address);
mpClientInterface->setParameters(input, String8(param));
@@ -4846,7 +4899,7 @@
profile_index--;
} else {
if (audio_device_is_digital(device->type())) {
- device->importAudioPort(profile);
+ device->importAudioPortAndPickAudioProfile(profile);
}
ALOGV("checkInputsForDevice(): adding input %d", input);
}
@@ -4921,7 +4974,8 @@
ssize_t index = mAudioPatches.indexOfKey(closingOutput->getPatchHandle());
if (index >= 0) {
sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
- (void) /*status_t status*/ mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
+ (void) /*status_t status*/ mpClientInterface->releaseAudioPatch(
+ patchDesc->getAfHandle(), 0);
mAudioPatches.removeItemsAt(index);
mpClientInterface->onAudioPatchListUpdate();
}
@@ -4969,7 +5023,8 @@
ssize_t index = mAudioPatches.indexOfKey(inputDesc->getPatchHandle());
if (index >= 0) {
sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
- (void) /*status_t status*/ mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
+ (void) /*status_t status*/ mpClientInterface->releaseAudioPatch(
+ patchDesc->getAfHandle(), 0);
mAudioPatches.removeItemsAt(index);
mpClientInterface->onAudioPatchListUpdate();
}
@@ -4998,7 +5053,7 @@
i, openOutputs.valueAt(i)->isDuplicated(),
openOutputs.valueAt(i)->supportedDevices().toString().c_str());
if (openOutputs.valueAt(i)->supportsAllDevices(devices)
- && openOutputs.valueAt(i)->deviceSupportsEncodedFormats(devices.types())) {
+ && openOutputs.valueAt(i)->devicesSupportEncodedFormats(devices.types())) {
ALOGVV("%s() found output %d", __func__, openOutputs.keyAt(i));
outputs.add(openOutputs.keyAt(i));
}
@@ -5033,6 +5088,7 @@
DeviceVector oldDevices = mEngine->getOutputDevicesForAttributes(attr, 0, true /*fromCache*/);
DeviceVector newDevices = mEngine->getOutputDevicesForAttributes(attr, 0, false /*fromCache*/);
+
SortedVector<audio_io_handle_t> srcOutputs = getOutputsForDevices(oldDevices, mPreviousOutputs);
SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevices(newDevices, mOutputs);
@@ -5133,9 +5189,8 @@
}
bool isScoConnected =
- ((mAvailableInputDevices.types() & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET &
- ~AUDIO_DEVICE_BIT_IN) != 0) ||
- ((mAvailableOutputDevices.types() & AUDIO_DEVICE_OUT_ALL_SCO) != 0);
+ (mAvailableInputDevices.types().count(AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) != 0 ||
+ !Intersection(mAvailableOutputDevices.types(), getAudioDeviceOutAllScoSet()).empty());
// if suspended, restore A2DP output if:
// ((SCO device is NOT connected) ||
@@ -5182,7 +5237,7 @@
ssize_t index = mAudioPatches.indexOfKey(outputDesc->getPatchHandle());
if (index >= 0) {
sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
- if (patchDesc->mUid != mUidCached) {
+ if (patchDesc->getUid() != mUidCached) {
ALOGV("%s device %s forced by patch %d", __func__,
outputDesc->devices().toString().c_str(), outputDesc->getPatchHandle());
return outputDesc->devices();
@@ -5210,7 +5265,8 @@
auto attr = mEngine->getAllAttributesForProductStrategy(productStrategy).front();
if ((hasVoiceStream(streams) &&
- (isInCall() || mOutputs.isStrategyActiveOnSameModule(productStrategy, outputDesc))) ||
+ (isInCall() || mOutputs.isStrategyActiveOnSameModule(productStrategy, outputDesc)) &&
+ !isStreamActive(AUDIO_STREAM_ENFORCED_AUDIBLE, 0)) ||
((hasStream(streams, AUDIO_STREAM_ALARM) || hasStream(streams, AUDIO_STREAM_ENFORCED_AUDIBLE)) &&
mOutputs.isStrategyActiveOnSameModule(productStrategy, outputDesc)) ||
outputDesc->isStrategyActive(productStrategy)) {
@@ -5232,7 +5288,7 @@
ssize_t index = mAudioPatches.indexOfKey(inputDesc->getPatchHandle());
if (index >= 0) {
sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
- if (patchDesc->mUid != mUidCached) {
+ if (patchDesc->getUid() != mUidCached) {
ALOGV("getNewInputDevice() device %s forced by patch %d",
inputDesc->getDevice()->toString().c_str(), inputDesc->getPatchHandle());
return inputDesc->getDevice();
@@ -5297,12 +5353,13 @@
}
/*Filter SPEAKER_SAFE out of results, as AudioService doesn't know about it
and doesn't really need to.*/
- DeviceVector speakerSafeDevices = devices.getDevicesFromTypeMask(AUDIO_DEVICE_OUT_SPEAKER_SAFE);
+ DeviceVector speakerSafeDevices = devices.getDevicesFromType(AUDIO_DEVICE_OUT_SPEAKER_SAFE);
if (!speakerSafeDevices.isEmpty()) {
- devices.merge(mAvailableOutputDevices.getDevicesFromTypeMask(AUDIO_DEVICE_OUT_SPEAKER));
+ devices.merge(mAvailableOutputDevices.getDevicesFromType(AUDIO_DEVICE_OUT_SPEAKER));
devices.remove(speakerSafeDevices);
}
- return devices.types();
+ // FIXME: use DeviceTypeSet when Java layer is ready for it.
+ return deviceTypesToBitMask(devices.types());
}
void AudioPolicyManager::handleNotificationRoutingForStream(audio_stream_type_t stream) {
@@ -5362,7 +5419,7 @@
auto ttsVolumeSource = toVolumeSource(AUDIO_STREAM_TTS);
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
- setVolumeSourceMute(ttsVolumeSource, mute/*on*/, desc, 0 /*delay*/, AUDIO_DEVICE_NONE);
+ setVolumeSourceMute(ttsVolumeSource, mute/*on*/, desc, 0 /*delay*/, DeviceTypeSet());
const uint32_t latency = desc->latency() * 2;
if (latency > maxLatency) {
maxLatency = latency;
@@ -5558,10 +5615,10 @@
return INVALID_OPERATION;
}
sp< AudioPatch> patchDesc = mAudioPatches.valueAt(index);
- status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, delayMs);
+ status_t status = mpClientInterface->releaseAudioPatch(patchDesc->getAfHandle(), delayMs);
ALOGV("resetOutputDevice() releaseAudioPatch returned %d", status);
outputDesc->setPatchHandle(AUDIO_PATCH_HANDLE_NONE);
- removeAudioPatch(patchDesc->mHandle);
+ removeAudioPatch(patchDesc->getHandle());
nextAudioPortGeneration();
mpClientInterface->onAudioPatchListUpdate();
return status;
@@ -5611,10 +5668,10 @@
return INVALID_OPERATION;
}
sp< AudioPatch> patchDesc = mAudioPatches.valueAt(index);
- status_t status = mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
+ status_t status = mpClientInterface->releaseAudioPatch(patchDesc->getAfHandle(), 0);
ALOGV("resetInputDevice() releaseAudioPatch returned %d", status);
inputDesc->setPatchHandle(AUDIO_PATCH_HANDLE_NONE);
- removeAudioPatch(patchDesc->mHandle);
+ removeAudioPatch(patchDesc->getHandle());
nextAudioPortGeneration();
mpClientInterface->onAudioPatchListUpdate();
return status;
@@ -5678,9 +5735,9 @@
float AudioPolicyManager::computeVolume(IVolumeCurves &curves,
VolumeSource volumeSource,
int index,
- audio_devices_t device)
+ const DeviceTypeSet& deviceTypes)
{
- float volumeDb = curves.volIndexToDb(Volume::getDeviceCategory(device), index);
+ float volumeDb = curves.volIndexToDb(Volume::getDeviceCategory(deviceTypes), index);
// handle the case of accessibility active while a ringtone is playing: if the ringtone is much
// louder than the accessibility prompt, the prompt cannot be heard, thus masking the touch
@@ -5696,7 +5753,7 @@
&& (AUDIO_MODE_RINGTONE == mEngine->getPhoneState()) &&
mOutputs.isActive(ringVolumeSrc, 0)) {
auto &ringCurves = getVolumeCurves(AUDIO_STREAM_RING);
- const float ringVolumeDb = computeVolume(ringCurves, ringVolumeSrc, index, device);
+ const float ringVolumeDb = computeVolume(ringCurves, ringVolumeSrc, index, deviceTypes);
return ringVolumeDb - 4 > volumeDb ? ringVolumeDb - 4 : volumeDb;
}
@@ -5711,9 +5768,9 @@
volumeSource == toVolumeSource(AUDIO_STREAM_DTMF) ||
volumeSource == a11yVolumeSrc)) {
auto &voiceCurves = getVolumeCurves(callVolumeSrc);
- int voiceVolumeIndex = voiceCurves.getVolumeIndex(device);
+ int voiceVolumeIndex = voiceCurves.getVolumeIndex(deviceTypes);
const float maxVoiceVolDb =
- computeVolume(voiceCurves, callVolumeSrc, voiceVolumeIndex, device)
+ computeVolume(voiceCurves, callVolumeSrc, voiceVolumeIndex, deviceTypes)
+ IN_CALL_EARPIECE_HEADROOM_DB;
// FIXME: Workaround for call screening applications until a proper audio mode is defined
// to support this scenario : Exempt the RING stream from the audio cap if the audio was
@@ -5739,9 +5796,10 @@
// speaker is part of the select devices
// - if music is playing, always limit the volume to current music volume,
// with a minimum threshold at -36dB so that notification is always perceived.
- if ((device & (AUDIO_DEVICE_OUT_BLUETOOTH_A2DP | AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
- AUDIO_DEVICE_OUT_WIRED_HEADSET | AUDIO_DEVICE_OUT_WIRED_HEADPHONE |
- AUDIO_DEVICE_OUT_USB_HEADSET | AUDIO_DEVICE_OUT_HEARING_AID)) &&
+ if (!Intersection(deviceTypes,
+ {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP, AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES,
+ AUDIO_DEVICE_OUT_WIRED_HEADSET, AUDIO_DEVICE_OUT_WIRED_HEADPHONE,
+ AUDIO_DEVICE_OUT_USB_HEADSET, AUDIO_DEVICE_OUT_HEARING_AID}).empty() &&
((volumeSource == alarmVolumeSrc ||
volumeSource == ringVolumeSrc) ||
(volumeSource == toVolumeSource(AUDIO_STREAM_NOTIFICATION)) ||
@@ -5756,31 +5814,33 @@
if (isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY) ||
mLimitRingtoneVolume) {
volumeDb += SONIFICATION_HEADSET_VOLUME_FACTOR_DB;
- audio_devices_t musicDevice =
+ DeviceTypeSet musicDevice =
mEngine->getOutputDevicesForAttributes(attributes_initializer(AUDIO_USAGE_MEDIA),
nullptr, true /*fromCache*/).types();
auto &musicCurves = getVolumeCurves(AUDIO_STREAM_MUSIC);
- float musicVolDb = computeVolume(musicCurves, musicVolumeSrc,
- musicCurves.getVolumeIndex(musicDevice), musicDevice);
+ float musicVolDb = computeVolume(musicCurves,
+ musicVolumeSrc,
+ musicCurves.getVolumeIndex(musicDevice),
+ musicDevice);
float minVolDb = (musicVolDb > SONIFICATION_HEADSET_VOLUME_MIN_DB) ?
musicVolDb : SONIFICATION_HEADSET_VOLUME_MIN_DB;
if (volumeDb > minVolDb) {
volumeDb = minVolDb;
ALOGV("computeVolume limiting volume to %f musicVol %f", minVolDb, musicVolDb);
}
- if (device & (AUDIO_DEVICE_OUT_BLUETOOTH_A2DP |
- AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES)) {
+ if (!Intersection(deviceTypes, {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP,
+ AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES}).empty()) {
// on A2DP, also ensure notification volume is not too low compared to media when
// intended to be played
if ((volumeDb > -96.0f) &&
(musicVolDb - SONIFICATION_A2DP_MAX_MEDIA_DIFF_DB > volumeDb)) {
- ALOGV("%s increasing volume for volume source=%d device=0x%X from %f to %f",
- __func__, volumeSource, device, volumeDb,
+ ALOGV("%s increasing volume for volume source=%d device=%s from %f to %f",
+ __func__, volumeSource, dumpDeviceTypes(deviceTypes).c_str(), volumeDb,
musicVolDb - SONIFICATION_A2DP_MAX_MEDIA_DIFF_DB);
volumeDb = musicVolDb - SONIFICATION_A2DP_MAX_MEDIA_DIFF_DB;
}
}
- } else if ((Volume::getDeviceForVolume(device) != AUDIO_DEVICE_OUT_SPEAKER) ||
+ } else if ((Volume::getDeviceForVolume(deviceTypes) != AUDIO_DEVICE_OUT_SPEAKER) ||
(!(volumeSource == alarmVolumeSrc || volumeSource == ringVolumeSrc))) {
volumeDb += SONIFICATION_HEADSET_VOLUME_FACTOR_DB;
}
@@ -5819,7 +5879,7 @@
VolumeSource volumeSource,
int index,
const sp<AudioOutputDescriptor>& outputDesc,
- audio_devices_t device,
+ DeviceTypeSet deviceTypes,
int delayMs,
bool force)
{
@@ -5845,17 +5905,20 @@
volumeSource, forceUseForComm);
return INVALID_OPERATION;
}
- if (device == AUDIO_DEVICE_NONE) {
- device = outputDesc->devices().types();
+ if (deviceTypes.empty()) {
+ deviceTypes = outputDesc->devices().types();
}
- float volumeDb = computeVolume(curves, volumeSource, index, device);
- if (outputDesc->isFixedVolume(device) ||
+ float volumeDb = computeVolume(curves, volumeSource, index, deviceTypes);
+ if (outputDesc->isFixedVolume(deviceTypes) ||
// Force VoIP volume to max for bluetooth SCO
- ((isVoiceVolSrc || isBtScoVolSrc) && (device & AUDIO_DEVICE_OUT_ALL_SCO) != 0)) {
+
+ ((isVoiceVolSrc || isBtScoVolSrc) &&
+ isSingleDeviceType(deviceTypes, audio_is_bluetooth_out_sco_device))) {
volumeDb = 0.0f;
}
- outputDesc->setVolume(volumeDb, volumeSource, curves.getStreamTypes(), device, delayMs, force);
+ outputDesc->setVolume(
+ volumeDb, volumeSource, curves.getStreamTypes(), deviceTypes, delayMs, force);
if (isVoiceVolSrc || isBtScoVolSrc) {
float voiceVolume;
@@ -5874,15 +5937,16 @@
}
void AudioPolicyManager::applyStreamVolumes(const sp<AudioOutputDescriptor>& outputDesc,
- audio_devices_t device,
- int delayMs,
- bool force)
+ const DeviceTypeSet& deviceTypes,
+ int delayMs,
+ bool force)
{
ALOGVV("applyStreamVolumes() for device %08x", device);
for (const auto &volumeGroup : mEngine->getVolumeGroups()) {
auto &curves = getVolumeCurves(toVolumeSource(volumeGroup));
checkAndSetVolume(curves, toVolumeSource(volumeGroup),
- curves.getVolumeIndex(device), outputDesc, device, delayMs, force);
+ curves.getVolumeIndex(deviceTypes),
+ outputDesc, deviceTypes, delayMs, force);
}
}
@@ -5890,7 +5954,7 @@
bool on,
const sp<AudioOutputDescriptor>& outputDesc,
int delayMs,
- audio_devices_t device)
+ DeviceTypeSet deviceTypes)
{
std::vector<VolumeSource> sourcesToMute;
for (auto attributes: mEngine->getAllAttributesForProductStrategy(strategy)) {
@@ -5902,7 +5966,7 @@
}
}
for (auto source : sourcesToMute) {
- setVolumeSourceMute(source, on, outputDesc, delayMs, device);
+ setVolumeSourceMute(source, on, outputDesc, delayMs, deviceTypes);
}
}
@@ -5911,10 +5975,10 @@
bool on,
const sp<AudioOutputDescriptor>& outputDesc,
int delayMs,
- audio_devices_t device)
+ DeviceTypeSet deviceTypes)
{
- if (device == AUDIO_DEVICE_NONE) {
- device = outputDesc->devices().types();
+ if (deviceTypes.empty()) {
+ deviceTypes = outputDesc->devices().types();
}
auto &curves = getVolumeCurves(volumeSource);
if (on) {
@@ -5923,7 +5987,7 @@
(volumeSource != toVolumeSource(AUDIO_STREAM_ENFORCED_AUDIBLE) ||
(mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) ==
AUDIO_POLICY_FORCE_NONE))) {
- checkAndSetVolume(curves, volumeSource, 0, outputDesc, device, delayMs);
+ checkAndSetVolume(curves, volumeSource, 0, outputDesc, deviceTypes, delayMs);
}
}
// increment mMuteCount after calling checkAndSetVolume() so that volume change is not
@@ -5936,9 +6000,9 @@
}
if (outputDesc->decMuteCount(volumeSource) == 0) {
checkAndSetVolume(curves, volumeSource,
- curves.getVolumeIndex(device),
+ curves.getVolumeIndex(deviceTypes),
outputDesc,
- device,
+ deviceTypes,
delayMs);
}
}
@@ -6020,8 +6084,8 @@
}
}
if (release) {
- ALOGV("%s releasing patch %u", __FUNCTION__, patchDesc->mHandle);
- releaseAudioPatch(patchDesc->mHandle, patchDesc->mUid);
+ ALOGV("%s releasing patch %u", __FUNCTION__, patchDesc->getHandle());
+ releaseAudioPatch(patchDesc->getHandle(), patchDesc->getUid());
}
}
@@ -6080,24 +6144,24 @@
formatSet.insert(enforcedSurround.begin(), enforcedSurround.end());
}
for (const auto& format : formatSet) {
- formatsPtr->push(format);
+ formatsPtr->push_back(format);
}
}
-void AudioPolicyManager::modifySurroundChannelMasks(ChannelsVector *channelMasksPtr) {
- ChannelsVector &channelMasks = *channelMasksPtr;
+void AudioPolicyManager::modifySurroundChannelMasks(ChannelMaskSet *channelMasksPtr) {
+ ChannelMaskSet &channelMasks = *channelMasksPtr;
audio_policy_forced_cfg_t forceUse = mEngine->getForceUse(
AUDIO_POLICY_FORCE_FOR_ENCODED_SURROUND);
// If NEVER, then remove support for channelMasks > stereo.
if (forceUse == AUDIO_POLICY_FORCE_ENCODED_SURROUND_NEVER) {
- for (size_t maskIndex = 0; maskIndex < channelMasks.size(); ) {
- audio_channel_mask_t channelMask = channelMasks[maskIndex];
+ for (auto it = channelMasks.begin(); it != channelMasks.end();) {
+ audio_channel_mask_t channelMask = *it;
if (channelMask & ~AUDIO_CHANNEL_OUT_STEREO) {
ALOGI("%s: force NEVER, so remove channelMask 0x%08x", __FUNCTION__, channelMask);
- channelMasks.removeAt(maskIndex);
+ it = channelMasks.erase(it);
} else {
- maskIndex++;
+ ++it;
}
}
// If ALWAYS or MANUAL, then make sure we at least support 5.1
@@ -6113,7 +6177,7 @@
}
// If not then add 5.1 support.
if (!supports5dot1) {
- channelMasks.add(AUDIO_CHANNEL_OUT_5POINT1);
+ channelMasks.insert(AUDIO_CHANNEL_OUT_5POINT1);
ALOGI("%s: force MANUAL or ALWAYS, so adding channelMask for 5.1 surround", __func__);
}
}
@@ -6142,12 +6206,12 @@
|| isDeviceOfModule(devDesc, AUDIO_HARDWARE_MODULE_ID_MSD)) {
modifySurroundFormats(devDesc, &formats);
}
- profiles.setFormats(formats);
+ addProfilesForFormats(profiles, formats);
}
for (audio_format_t format : profiles.getSupportedFormats()) {
- ChannelsVector channelMasks;
- SampleRateVector samplingRates;
+ ChannelMaskSet channelMasks;
+ SampleRateSet samplingRates;
AudioParameter requestedParameters;
requestedParameters.addInt(String8(AudioParameter::keyFormat), format);
@@ -6178,7 +6242,8 @@
}
}
}
- profiles.addProfileFromHal(new AudioProfile(format, channelMasks, samplingRates));
+ addDynamicAudioProfileAndSort(
+ profiles, new AudioProfile(format, channelMasks, samplingRates));
}
}
@@ -6195,7 +6260,7 @@
status_t status = installPatch(
caller, index, patchHandle, patch, delayMs, mUidCached, &patchDesc);
if (status == NO_ERROR) {
- ioDescriptor->setPatchHandle(patchDesc->mHandle);
+ ioDescriptor->setPatchHandle(patchDesc->getHandle());
}
return status;
}
@@ -6212,7 +6277,7 @@
audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
if (index >= 0) {
patchDesc = mAudioPatches.valueAt(index);
- afPatchHandle = patchDesc->mAfPatchHandle;
+ afPatchHandle = patchDesc->getAfHandle();
}
status_t status = mpClientInterface->createAudioPatch(patch, &afPatchHandle, delayMs);
@@ -6221,13 +6286,13 @@
if (status == NO_ERROR) {
if (index < 0) {
patchDesc = new AudioPatch(patch, uid);
- addAudioPatch(patchDesc->mHandle, patchDesc);
+ addAudioPatch(patchDesc->getHandle(), patchDesc);
} else {
patchDesc->mPatch = *patch;
}
- patchDesc->mAfPatchHandle = afPatchHandle;
+ patchDesc->setAfHandle(afPatchHandle);
if (patchHandle) {
- *patchHandle = patchDesc->mHandle;
+ *patchHandle = patchDesc->getHandle();
}
nextAudioPortGeneration();
mpClientInterface->onAudioPatchListUpdate();
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
index dc548d6..9e08b0b 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -31,16 +31,14 @@
#include <utils/SortedVector.h>
#include <media/AudioParameter.h>
#include <media/AudioPolicy.h>
+#include <media/AudioProfile.h>
#include <media/PatchBuilder.h>
#include "AudioPolicyInterface.h"
-#include <AudioPolicyManagerInterface.h>
#include <AudioPolicyManagerObserver.h>
-#include <AudioGain.h>
#include <AudioPolicyConfig.h>
-#include <AudioPort.h>
+#include <PolicyAudioPort.h>
#include <AudioPatch.h>
-#include <AudioProfile.h>
#include <DeviceDescriptor.h>
#include <IOProfile.h>
#include <HwModule.h>
@@ -49,6 +47,7 @@
#include <AudioPolicyMix.h>
#include <EffectDescriptor.h>
#include <SoundTriggerSession.h>
+#include "EngineLibrary.h"
#include "TypeConverter.h"
namespace android {
@@ -177,7 +176,7 @@
IVolumeCurves &volumeCurves);
status_t getVolumeIndex(const IVolumeCurves &curves, int &index,
- audio_devices_t device) const;
+ const DeviceTypeSet& deviceTypes) const;
// return the strategy corresponding to a given stream type
virtual uint32_t getStrategyForStream(audio_stream_type_t stream)
@@ -234,7 +233,10 @@
virtual status_t getAudioPort(struct audio_port *port);
virtual status_t createAudioPatch(const struct audio_patch *patch,
audio_patch_handle_t *handle,
- uid_t uid);
+ uid_t uid) {
+ return createAudioPatchInternal(patch, handle, uid);
+ }
+
virtual status_t releaseAudioPatch(audio_patch_handle_t handle,
uid_t uid);
virtual status_t listAudioPatches(unsigned int *num_patches,
@@ -259,6 +261,12 @@
const Vector<AudioDeviceTypeAddr>& devices);
virtual status_t removeUidDeviceAffinities(uid_t uid);
+ virtual status_t setPreferredDeviceForStrategy(product_strategy_t strategy,
+ const AudioDeviceTypeAddr &device);
+ virtual status_t removePreferredDeviceForStrategy(product_strategy_t strategy);
+ virtual status_t getPreferredDeviceForStrategy(product_strategy_t strategy,
+ AudioDeviceTypeAddr &device);
+
virtual status_t startAudioSource(const struct audio_port_config *source,
const audio_attributes_t *attributes,
audio_port_handle_t *portId,
@@ -307,6 +315,8 @@
return volumeGroup != VOLUME_GROUP_NONE ? NO_ERROR : BAD_VALUE;
}
+ status_t initialize();
+
protected:
// A constructor that allows more fine-grained control over initialization process,
// used in automatic tests.
@@ -321,7 +331,6 @@
// - initialize.
AudioPolicyConfig& getConfig() { return mConfig; }
void loadConfig();
- status_t initialize();
// From AudioPolicyManagerObserver
virtual const AudioPatchCollection &getAudioPatches() const
@@ -422,7 +431,7 @@
virtual float computeVolume(IVolumeCurves &curves,
VolumeSource volumeSource,
int index,
- audio_devices_t device);
+ const DeviceTypeSet& deviceTypes);
// rescale volume index from srcStream within range of dstStream
int rescaleVolumeIndex(int srcIndex,
@@ -432,12 +441,13 @@
virtual status_t checkAndSetVolume(IVolumeCurves &curves,
VolumeSource volumeSource, int index,
const sp<AudioOutputDescriptor>& outputDesc,
- audio_devices_t device,
+ DeviceTypeSet deviceTypes,
int delayMs = 0, bool force = false);
// apply all stream volumes to the specified output and device
void applyStreamVolumes(const sp<AudioOutputDescriptor>& outputDesc,
- audio_devices_t device, int delayMs = 0, bool force = false);
+ const DeviceTypeSet& deviceTypes,
+ int delayMs = 0, bool force = false);
/**
* @brief setStrategyMute Mute or unmute all active clients on the considered output
@@ -452,7 +462,7 @@
bool on,
const sp<AudioOutputDescriptor>& outputDesc,
int delayMs = 0,
- audio_devices_t device = AUDIO_DEVICE_NONE);
+ DeviceTypeSet deviceTypes = DeviceTypeSet());
/**
* @brief setVolumeSourceMute Mute or unmute the volume source on the specified output
@@ -467,7 +477,7 @@
bool on,
const sp<AudioOutputDescriptor>& outputDesc,
int delayMs = 0,
- audio_devices_t device = AUDIO_DEVICE_NONE);
+ DeviceTypeSet deviceTypes = DeviceTypeSet());
audio_mode_t getPhoneState();
@@ -495,13 +505,19 @@
// close an input.
void closeInput(audio_io_handle_t input);
- // runs all the checks required for accomodating changes in devices and outputs
+ // runs all the checks required for accommodating changes in devices and outputs
// if 'onOutputsChecked' callback is provided, it is executed after the outputs
// check via 'checkOutputForAllStrategies'. If the callback returns 'true',
// A2DP suspend status is rechecked.
void checkForDeviceAndOutputChanges(std::function<bool()> onOutputsChecked = nullptr);
/**
+ * @brief updates routing for all outputs (including call if call in progress).
+ * @param delayMs delay for unmuting if required
+ */
+ void updateCallAndOutputRouting(bool forceVolumeReeval = true, uint32_t delayMs = 0);
+
+ /**
* @brief checkOutputForAttributes checks and if necessary changes outputs used for the
* given audio attributes.
* must be called every time a condition that affects the output choice for a given
@@ -646,16 +662,13 @@
}
String8 getFirstDeviceAddress(const DeviceVector &devices) const
{
- return (devices.size() > 0) ? devices.itemAt(0)->address() : String8("");
+ return (devices.size() > 0) ?
+ String8(devices.itemAt(0)->address().c_str()) : String8("");
}
uint32_t updateCallRouting(const DeviceVector &rxDevices, uint32_t delayMs = 0);
sp<AudioPatch> createTelephonyPatch(bool isRx, const sp<DeviceDescriptor> &device,
uint32_t delayMs);
- sp<DeviceDescriptor> findDevice(
- const DeviceVector& devices, audio_devices_t device) const;
- audio_devices_t getModuleDeviceTypes(
- const DeviceVector& devices, const char *moduleId) const;
bool isDeviceOfModule(const sp<DeviceDescriptor>& devDesc, const char *moduleId) const;
status_t startSource(const sp<SwAudioOutputDescriptor>& outputDesc,
@@ -752,7 +765,7 @@
uint32_t nextAudioPortGeneration();
// Audio Policy Engine Interface.
- AudioPolicyManagerInterface *mEngine;
+ EngineInstance mEngine;
// Surround formats that are enabled manually. Taken into account when
// "encoded surround" is forced into "manual" mode.
@@ -762,7 +775,7 @@
private:
// Add or remove AC3 DTS encodings based on user preferences.
void modifySurroundFormats(const sp<DeviceDescriptor>& devDesc, FormatVector *formatsPtr);
- void modifySurroundChannelMasks(ChannelsVector *channelMasksPtr);
+ void modifySurroundChannelMasks(ChannelMaskSet *channelMasksPtr);
// Support for Multi-Stream Decoder (MSD) module
sp<DeviceDescriptor> getMsdAudioInDevice() const;
@@ -858,6 +871,29 @@
param.addInt(String8(AudioParameter::keyMonoOutput), (int)mMasterMono);
mpClientInterface->setParameters(output, param.toString());
}
+
+ /**
+ * @brief createAudioPatchInternal internal function to manage audio patch creation
+ * @param[in] patch structure containing sink and source ports configuration
+ * @param[out] handle patch handle to be provided if patch installed correctly
+ * @param[in] uid of the client
+ * @param[in] delayMs if required
+ * @param[in] sourceDesc [optional] in case of external source, source client to be
+ * configured by the patch, i.e. assigning an Output (HW or SW)
+ * @return NO_ERROR if patch installed correclty, error code otherwise.
+ */
+ status_t createAudioPatchInternal(const struct audio_patch *patch,
+ audio_patch_handle_t *handle,
+ uid_t uid, uint32_t delayMs = 0,
+ const sp<SourceClientDescriptor>& sourceDesc = nullptr);
+ /**
+ * @brief releaseAudioPatchInternal internal function to remove an audio patch
+ * @param[in] handle of the patch to be removed
+ * @param[in] delayMs if required
+ * @return NO_ERROR if patch removed correclty, error code otherwise.
+ */
+ status_t releaseAudioPatchInternal(audio_patch_handle_t handle, uint32_t delayMs = 0);
+
status_t installPatch(const char *caller,
audio_patch_handle_t *patchHandle,
AudioIODescriptorInterface *ioDescriptor,
diff --git a/services/audiopolicy/managerdefault/EngineLibrary.cpp b/services/audiopolicy/managerdefault/EngineLibrary.cpp
new file mode 100644
index 0000000..ef699aa
--- /dev/null
+++ b/services/audiopolicy/managerdefault/EngineLibrary.cpp
@@ -0,0 +1,78 @@
+/*
+ * Copyright 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "APM_EngineLoader"
+
+#include <dlfcn.h>
+#include <utils/Log.h>
+
+#include "EngineLibrary.h"
+
+namespace android {
+
+// static
+std::shared_ptr<EngineLibrary> EngineLibrary::load(std::string libraryPath)
+{
+ std::shared_ptr<EngineLibrary> engLib(new EngineLibrary());
+ return engLib->init(std::move(libraryPath)) ? engLib : nullptr;
+}
+
+EngineLibrary::~EngineLibrary()
+{
+ close();
+}
+
+bool EngineLibrary::init(std::string libraryPath)
+{
+ mLibraryHandle = dlopen(libraryPath.c_str(), 0);
+ if (mLibraryHandle == nullptr) {
+ ALOGE("Could not dlopen %s: %s", libraryPath.c_str(), dlerror());
+ return false;
+ }
+ mCreateEngineInstance = (EngineInterface* (*)())dlsym(mLibraryHandle, "createEngineInstance");
+ mDestroyEngineInstance = (void (*)(EngineInterface*))dlsym(
+ mLibraryHandle, "destroyEngineInstance");
+ if (mCreateEngineInstance == nullptr || mDestroyEngineInstance == nullptr) {
+ ALOGE("Could not find engine interface functions in %s", libraryPath.c_str());
+ close();
+ return false;
+ }
+ ALOGD("Loaded engine from %s", libraryPath.c_str());
+ return true;
+}
+
+EngineInstance EngineLibrary::createEngine()
+{
+ if (mCreateEngineInstance == nullptr || mDestroyEngineInstance == nullptr) {
+ return EngineInstance();
+ }
+ return EngineInstance(mCreateEngineInstance(),
+ [lib = shared_from_this(), destroy = mDestroyEngineInstance] (EngineInterface* e) {
+ destroy(e);
+ });
+}
+
+void EngineLibrary::close()
+{
+ if (mLibraryHandle != nullptr) {
+ dlclose(mLibraryHandle);
+ }
+ mLibraryHandle = nullptr;
+ mCreateEngineInstance = nullptr;
+ mDestroyEngineInstance = nullptr;
+}
+
+} // namespace android
diff --git a/services/audiopolicy/managerdefault/EngineLibrary.h b/services/audiopolicy/managerdefault/EngineLibrary.h
new file mode 100644
index 0000000..f143916
--- /dev/null
+++ b/services/audiopolicy/managerdefault/EngineLibrary.h
@@ -0,0 +1,51 @@
+/*
+ * Copyright 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#pragma once
+
+#include <functional>
+#include <memory>
+#include <string>
+
+#include <EngineInterface.h>
+
+namespace android {
+
+using EngineInstance = std::unique_ptr<EngineInterface, std::function<void (EngineInterface*)>>;
+
+class EngineLibrary : public std::enable_shared_from_this<EngineLibrary> {
+public:
+ static std::shared_ptr<EngineLibrary> load(std::string libraryPath);
+ ~EngineLibrary();
+
+ EngineLibrary(const EngineLibrary&) = delete;
+ EngineLibrary(EngineLibrary&&) = delete;
+ EngineLibrary& operator=(const EngineLibrary&) = delete;
+ EngineLibrary& operator=(EngineLibrary&&) = delete;
+
+ EngineInstance createEngine();
+
+private:
+ EngineLibrary() = default;
+ bool init(std::string libraryPath);
+ void close();
+
+ void *mLibraryHandle = nullptr;
+ EngineInterface* (*mCreateEngineInstance)() = nullptr;
+ void (*mDestroyEngineInstance)(EngineInterface*) = nullptr;
+};
+
+} // namespace android
diff --git a/services/audiopolicy/service/Android.mk b/services/audiopolicy/service/Android.mk
index c4f4c56..fdf3eae 100644
--- a/services/audiopolicy/service/Android.mk
+++ b/services/audiopolicy/service/Android.mk
@@ -24,6 +24,7 @@
libbinder \
libaudioclient \
libaudioutils \
+ libaudiofoundation \
libhardware_legacy \
libaudiopolicymanager \
libmedia_helper \
@@ -38,8 +39,6 @@
LOCAL_STATIC_LIBRARIES := \
libaudiopolicycomponents
-LOCAL_MULTILIB := $(AUDIOSERVER_MULTILIB)
-
LOCAL_MODULE:= libaudiopolicyservice
LOCAL_CFLAGS += -fvisibility=hidden
diff --git a/services/audiopolicy/service/AudioPolicyClientImpl.cpp b/services/audiopolicy/service/AudioPolicyClientImpl.cpp
index d51cc6e..6de0c80 100644
--- a/services/audiopolicy/service/AudioPolicyClientImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyClientImpl.cpp
@@ -39,8 +39,7 @@
status_t AudioPolicyService::AudioPolicyClient::openOutput(audio_module_handle_t module,
audio_io_handle_t *output,
audio_config_t *config,
- audio_devices_t *devices,
- const String8& address,
+ const sp<DeviceDescriptorBase>& device,
uint32_t *latencyMs,
audio_output_flags_t flags)
{
@@ -49,7 +48,7 @@
ALOGW("%s: could not get AudioFlinger", __func__);
return PERMISSION_DENIED;
}
- return af->openOutput(module, output, config, devices, address, latencyMs, flags);
+ return af->openOutput(module, output, config, device, latencyMs, flags);
}
audio_io_handle_t AudioPolicyService::AudioPolicyClient::openDuplicateOutput(
diff --git a/services/audiopolicy/service/AudioPolicyEffects.cpp b/services/audiopolicy/service/AudioPolicyEffects.cpp
index 4947714..738a279 100644
--- a/services/audiopolicy/service/AudioPolicyEffects.cpp
+++ b/services/audiopolicy/service/AudioPolicyEffects.cpp
@@ -42,7 +42,10 @@
AudioPolicyEffects::AudioPolicyEffects()
{
status_t loadResult = loadAudioEffectXmlConfig();
- if (loadResult < 0) {
+ if (loadResult == NO_ERROR) {
+ mDefaultDeviceEffectFuture = std::async(
+ std::launch::async, &AudioPolicyEffects::initDefaultDeviceEffects, this);
+ } else if (loadResult < 0) {
ALOGW("Failed to load XML effect configuration, fallback to .conf");
// load automatic audio effect modules
if (access(AUDIO_EFFECT_VENDOR_CONFIG_FILE, R_OK) == 0) {
@@ -562,7 +565,8 @@
AUDIO_STREAM_BLUETOOTH_SCO_TAG,
AUDIO_STREAM_ENFORCED_AUDIBLE_TAG,
AUDIO_STREAM_DTMF_TAG,
- AUDIO_STREAM_TTS_TAG
+ AUDIO_STREAM_TTS_TAG,
+ AUDIO_STREAM_ASSISTANT_TAG
};
// returns the audio_stream_t enum corresponding to the output stream name or
@@ -907,8 +911,24 @@
streams.add(stream.type, effectDescs.release());
}
};
+
+ auto loadDeviceProcessingChain = [](auto &processingChain, auto& devicesEffects) {
+ for (auto& deviceProcess : processingChain) {
+
+ auto effectDescs = std::make_unique<EffectDescVector>();
+ for (auto& effect : deviceProcess.effects) {
+ effectDescs->mEffects.add(
+ new EffectDesc{effect.get().name.c_str(), effect.get().uuid});
+ }
+ auto deviceEffects = std::make_unique<DeviceEffects>(
+ std::move(effectDescs), deviceProcess.type, deviceProcess.address);
+ devicesEffects.emplace(deviceProcess.address, std::move(deviceEffects));
+ }
+ };
+
loadProcessingChain(result.parsedConfig->preprocess, mInputSources);
loadProcessingChain(result.parsedConfig->postprocess, mOutputStreams);
+ loadDeviceProcessingChain(result.parsedConfig->deviceprocess, mDeviceEffects);
// Casting from ssize_t to status_t is probably safe, there should not be more than 2^31 errors
return result.nbSkippedElement;
}
@@ -941,5 +961,32 @@
return NO_ERROR;
}
+void AudioPolicyEffects::initDefaultDeviceEffects()
+{
+ Mutex::Autolock _l(mLock);
+ for (const auto& deviceEffectsIter : mDeviceEffects) {
+ const auto& deviceEffects = deviceEffectsIter.second;
+ for (const auto& effectDesc : deviceEffects->mEffectDescriptors->mEffects) {
+ auto fx = std::make_unique<AudioEffect>(
+ EFFECT_UUID_NULL, String16("android"), &effectDesc->mUuid, 0, nullptr,
+ nullptr, AUDIO_SESSION_DEVICE, AUDIO_IO_HANDLE_NONE,
+ AudioDeviceTypeAddr{deviceEffects->getDeviceType(),
+ deviceEffects->getDeviceAddress()});
+ status_t status = fx->initCheck();
+ if (status != NO_ERROR && status != ALREADY_EXISTS) {
+ ALOGE("%s(): failed to create Fx %s on port type=%d address=%s", __func__,
+ effectDesc->mName, deviceEffects->getDeviceType(),
+ deviceEffects->getDeviceAddress().c_str());
+ // fx goes out of scope and strong ref on AudioEffect is released
+ continue;
+ }
+ fx->setEnabled(true);
+ ALOGV("%s(): create Fx %s added on port type=%d address=%s", __func__,
+ effectDesc->mName, deviceEffects->getDeviceType(),
+ deviceEffects->getDeviceAddress().c_str());
+ deviceEffects->mEffects.push_back(std::move(fx));
+ }
+ }
+}
} // namespace android
diff --git a/services/audiopolicy/service/AudioPolicyEffects.h b/services/audiopolicy/service/AudioPolicyEffects.h
index dcf093b..88be1ad 100644
--- a/services/audiopolicy/service/AudioPolicyEffects.h
+++ b/services/audiopolicy/service/AudioPolicyEffects.h
@@ -25,6 +25,9 @@
#include <system/audio.h>
#include <utils/Vector.h>
#include <utils/SortedVector.h>
+#include <android-base/thread_annotations.h>
+
+#include <future>
namespace android {
@@ -104,6 +107,7 @@
status_t removeStreamDefaultEffect(audio_unique_id_t id);
private:
+ void initDefaultDeviceEffects();
// class to store the description of an effects and its parameters
// as defined in audio_effects.conf
@@ -192,6 +196,28 @@
Vector< sp<AudioEffect> >mEffects;
};
+ /**
+ * @brief The DeviceEffects class stores the effects associated to a given Device Port.
+ */
+ class DeviceEffects {
+ public:
+ explicit DeviceEffects(std::unique_ptr<EffectDescVector> effectDescriptors,
+ audio_devices_t device, const std::string& address) :
+ mEffectDescriptors(std::move(effectDescriptors)),
+ mDeviceType(device), mDeviceAddress(address) {}
+ /*virtual*/ ~DeviceEffects() = default;
+
+ std::vector<std::unique_ptr<AudioEffect>> mEffects;
+ audio_devices_t getDeviceType() const { return mDeviceType; }
+ std::string getDeviceAddress() const { return mDeviceAddress; }
+ const std::unique_ptr<EffectDescVector> mEffectDescriptors;
+
+ private:
+ const audio_devices_t mDeviceType;
+ const std::string mDeviceAddress;
+
+ };
+
static const char * const kInputSourceNames[AUDIO_SOURCE_CNT -1];
static audio_source_t inputSourceNameToEnum(const char *name);
@@ -237,6 +263,19 @@
KeyedVector< audio_stream_type_t, EffectDescVector* > mOutputStreams;
// Automatic output effects are unique for audiosession ID
KeyedVector< audio_session_t, EffectVector* > mOutputSessions;
+
+ /**
+ * @brief mDeviceEffects map of device effects indexed by the device address
+ */
+ std::map<std::string, std::unique_ptr<DeviceEffects>> mDeviceEffects GUARDED_BY(mLock);
+
+ /**
+ * Device Effect initialization must be asynchronous: the audio_policy service parses and init
+ * effect on first reference. AudioFlinger will handle effect creation and register these
+ * effect on audio_policy service.
+ * We must store the reference of the furture garantee real asynchronous operation.
+ */
+ std::future<void> mDefaultDeviceEffectFuture;
};
} // namespace android
diff --git a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
index 20e1c9e..c1190be 100644
--- a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
@@ -351,12 +351,17 @@
return NO_INIT;
}
+ audio_source_t inputSource = attr->source;
+ if (inputSource == AUDIO_SOURCE_DEFAULT) {
+ inputSource = AUDIO_SOURCE_MIC;
+ }
+
// already checked by client, but double-check in case the client wrapper is bypassed
- if ((attr->source < AUDIO_SOURCE_DEFAULT)
- || (attr->source >= AUDIO_SOURCE_CNT
- && attr->source != AUDIO_SOURCE_HOTWORD
- && attr->source != AUDIO_SOURCE_FM_TUNER
- && attr->source != AUDIO_SOURCE_ECHO_REFERENCE)) {
+ if ((inputSource < AUDIO_SOURCE_DEFAULT)
+ || (inputSource >= AUDIO_SOURCE_CNT
+ && inputSource != AUDIO_SOURCE_HOTWORD
+ && inputSource != AUDIO_SOURCE_FM_TUNER
+ && inputSource != AUDIO_SOURCE_ECHO_REFERENCE)) {
return BAD_VALUE;
}
@@ -385,16 +390,16 @@
}
bool canCaptureOutput = captureAudioOutputAllowed(pid, uid);
- if ((attr->source == AUDIO_SOURCE_VOICE_UPLINK ||
- attr->source == AUDIO_SOURCE_VOICE_DOWNLINK ||
- attr->source == AUDIO_SOURCE_VOICE_CALL ||
- attr->source == AUDIO_SOURCE_ECHO_REFERENCE) &&
+ if ((inputSource == AUDIO_SOURCE_VOICE_UPLINK ||
+ inputSource == AUDIO_SOURCE_VOICE_DOWNLINK ||
+ inputSource == AUDIO_SOURCE_VOICE_CALL ||
+ inputSource == AUDIO_SOURCE_ECHO_REFERENCE) &&
!canCaptureOutput) {
return PERMISSION_DENIED;
}
bool canCaptureHotword = captureHotwordAllowed(opPackageName, pid, uid);
- if ((attr->source == AUDIO_SOURCE_HOTWORD) && !canCaptureHotword) {
+ if ((inputSource == AUDIO_SOURCE_HOTWORD) && !canCaptureHotword) {
return BAD_VALUE;
}
@@ -459,7 +464,7 @@
if (audioPolicyEffects != 0) {
// create audio pre processors according to input source
- status_t status = audioPolicyEffects->addInputEffects(*input, attr->source, session);
+ status_t status = audioPolicyEffects->addInputEffects(*input, inputSource, session);
if (status != NO_ERROR && status != ALREADY_EXISTS) {
ALOGW("Failed to add effects on input %d", *input);
}
@@ -976,6 +981,11 @@
ALOGV("%s() mAudioPolicyManager == NULL", __func__);
return NO_INIT;
}
+ uint_t callingUid = IPCThreadState::self()->getCallingUid();
+ if (uid != callingUid) {
+ ALOGD("%s() uid invalid %d != %d", __func__, uid, callingUid);
+ return PERMISSION_DENIED;
+ }
return mAudioPolicyManager->setAllowedCapturePolicy(uid, capturePolicy);
}
@@ -1320,4 +1330,33 @@
return NO_ERROR;
}
+status_t AudioPolicyService::setPreferredDeviceForStrategy(product_strategy_t strategy,
+ const AudioDeviceTypeAddr &device)
+{
+ if (mAudioPolicyManager == NULL) {
+ return NO_INIT;
+ }
+ Mutex::Autolock _l(mLock);
+ return mAudioPolicyManager->setPreferredDeviceForStrategy(strategy, device);
+}
+
+status_t AudioPolicyService::removePreferredDeviceForStrategy(product_strategy_t strategy)
+{
+ if (mAudioPolicyManager == NULL) {
+ return NO_INIT;
+ }
+ Mutex::Autolock _l(mLock);
+ return mAudioPolicyManager->removePreferredDeviceForStrategy(strategy);
+}
+
+status_t AudioPolicyService::getPreferredDeviceForStrategy(product_strategy_t strategy,
+ AudioDeviceTypeAddr &device)
+{
+ if (mAudioPolicyManager == NULL) {
+ return NO_INIT;
+ }
+ Mutex::Autolock _l(mLock);
+ return mAudioPolicyManager->getPreferredDeviceForStrategy(strategy, device);
+}
+
} // namespace android
diff --git a/services/audiopolicy/service/AudioPolicyService.cpp b/services/audiopolicy/service/AudioPolicyService.cpp
index a6cda20..90939ce 100644
--- a/services/audiopolicy/service/AudioPolicyService.cpp
+++ b/services/audiopolicy/service/AudioPolicyService.cpp
@@ -29,7 +29,6 @@
#include <utils/Log.h>
#include <cutils/properties.h>
#include <binder/IPCThreadState.h>
-#include <binder/ActivityManager.h>
#include <binder/PermissionController.h>
#include <binder/IResultReceiver.h>
#include <utils/String16.h>
@@ -844,28 +843,26 @@
// ----------- AudioPolicyService::UidPolicy implementation ----------
void AudioPolicyService::UidPolicy::registerSelf() {
- ActivityManager am;
- am.registerUidObserver(this, ActivityManager::UID_OBSERVER_GONE
+ status_t res = mAm.linkToDeath(this);
+ mAm.registerUidObserver(this, ActivityManager::UID_OBSERVER_GONE
| ActivityManager::UID_OBSERVER_IDLE
| ActivityManager::UID_OBSERVER_ACTIVE
| ActivityManager::UID_OBSERVER_PROCSTATE,
ActivityManager::PROCESS_STATE_UNKNOWN,
String16("audioserver"));
- status_t res = am.linkToDeath(this);
if (!res) {
Mutex::Autolock _l(mLock);
mObserverRegistered = true;
} else {
ALOGE("UidPolicy::registerSelf linkToDeath failed: %d", res);
- am.unregisterUidObserver(this);
+ mAm.unregisterUidObserver(this);
}
}
void AudioPolicyService::UidPolicy::unregisterSelf() {
- ActivityManager am;
- am.unlinkToDeath(this);
- am.unregisterUidObserver(this);
+ mAm.unlinkToDeath(this);
+ mAm.unregisterUidObserver(this);
Mutex::Autolock _l(mLock);
mObserverRegistered = false;
}
diff --git a/services/audiopolicy/service/AudioPolicyService.h b/services/audiopolicy/service/AudioPolicyService.h
index e467f70..7b72dc1 100644
--- a/services/audiopolicy/service/AudioPolicyService.h
+++ b/services/audiopolicy/service/AudioPolicyService.h
@@ -23,6 +23,7 @@
#include <utils/String8.h>
#include <utils/Vector.h>
#include <utils/SortedVector.h>
+#include <binder/ActivityManager.h>
#include <binder/BinderService.h>
#include <binder/IUidObserver.h>
#include <system/audio.h>
@@ -223,6 +224,15 @@
virtual status_t removeUidDeviceAffinities(uid_t uid);
+ virtual status_t setPreferredDeviceForStrategy(product_strategy_t strategy,
+ const AudioDeviceTypeAddr &device);
+
+ virtual status_t removePreferredDeviceForStrategy(product_strategy_t strategy);
+
+
+ virtual status_t getPreferredDeviceForStrategy(product_strategy_t strategy,
+ AudioDeviceTypeAddr &device);
+
virtual status_t startAudioSource(const struct audio_port_config *source,
const audio_attributes_t *attributes,
audio_port_handle_t *portId);
@@ -387,6 +397,7 @@
wp<AudioPolicyService> mService;
Mutex mLock;
+ ActivityManager mAm;
bool mObserverRegistered;
std::unordered_map<uid_t, std::pair<bool, int>> mOverrideUids;
std::unordered_map<uid_t, std::pair<bool, int>> mCachedUids;
@@ -620,8 +631,7 @@
virtual status_t openOutput(audio_module_handle_t module,
audio_io_handle_t *output,
audio_config_t *config,
- audio_devices_t *devices,
- const String8& address,
+ const sp<DeviceDescriptorBase>& device,
uint32_t *latencyMs,
audio_output_flags_t flags);
// creates a special output that is duplicated to the two outputs passed as arguments. The duplication is performed by
diff --git a/services/audiopolicy/tests/Android.bp b/services/audiopolicy/tests/Android.bp
new file mode 100644
index 0000000..efdb241
--- /dev/null
+++ b/services/audiopolicy/tests/Android.bp
@@ -0,0 +1,71 @@
+cc_test {
+ name: "audiopolicy_tests",
+
+ include_dirs: [
+ "frameworks/av/services/audiopolicy",
+ ],
+
+ shared_libs: [
+ "libaudioclient",
+ "libaudiofoundation",
+ "libaudiopolicy",
+ "libaudiopolicymanagerdefault",
+ "libbase",
+ "libhidlbase",
+ "liblog",
+ "libmedia_helper",
+ "libutils",
+ "libxml2",
+ ],
+
+ static_libs: ["libaudiopolicycomponents"],
+
+ header_libs: [
+ "libaudiopolicycommon",
+ "libaudiopolicyengine_interface_headers",
+ "libaudiopolicymanager_interface_headers",
+ ],
+
+ srcs: ["audiopolicymanager_tests.cpp"],
+
+ data: [":audiopolicytest_configuration_files",],
+
+ cflags: [
+ "-Werror",
+ "-Wall",
+ ],
+
+ test_suites: ["device-tests"],
+
+}
+
+
+cc_test {
+ name: "audio_health_tests",
+
+ shared_libs: [
+ "libaudiofoundation",
+ "libaudioclient",
+ "libaudiopolicymanagerdefault",
+ "liblog",
+ "libmedia_helper",
+ "libutils",
+ ],
+
+ static_libs: ["libaudiopolicycomponents"],
+
+ header_libs: [
+ "libaudiopolicyengine_interface_headers",
+ "libaudiopolicymanager_interface_headers",
+ ],
+
+ srcs: ["audio_health_tests.cpp"],
+
+ cflags: [
+ "-Werror",
+ "-Wall",
+ ],
+
+ test_suites: ["device-tests"],
+
+}
diff --git a/services/audiopolicy/tests/Android.mk b/services/audiopolicy/tests/Android.mk
deleted file mode 100644
index ab9f78b..0000000
--- a/services/audiopolicy/tests/Android.mk
+++ /dev/null
@@ -1,65 +0,0 @@
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_C_INCLUDES := \
- frameworks/av/services/audiopolicy \
- $(call include-path-for, audio-utils) \
-
-LOCAL_SHARED_LIBRARIES := \
- libaudiopolicymanagerdefault \
- libbase \
- liblog \
- libmedia_helper \
- libutils \
-
-LOCAL_STATIC_LIBRARIES := \
- libaudiopolicycomponents \
-
-LOCAL_HEADER_LIBRARIES := \
- libaudiopolicycommon \
- libaudiopolicyengine_interface_headers \
- libaudiopolicymanager_interface_headers
-
-LOCAL_SRC_FILES := \
- audiopolicymanager_tests.cpp \
-
-LOCAL_MODULE := audiopolicy_tests
-
-LOCAL_MODULE_TAGS := tests
-
-LOCAL_CFLAGS := -Werror -Wall
-
-LOCAL_MULTILIB := $(AUDIOSERVER_MULTILIB)
-
-LOCAL_COMPATIBILITY_SUITE := device-tests
-
-include $(BUILD_NATIVE_TEST)
-
-# system/audio.h utilities test
-
-include $(CLEAR_VARS)
-
-LOCAL_SHARED_LIBRARIES := \
- libbase \
- liblog \
- libmedia_helper \
- libutils
-
-LOCAL_HEADER_LIBRARIES := \
- libmedia_headers
-
-LOCAL_SRC_FILES := \
- systemaudio_tests.cpp \
-
-LOCAL_MODULE := systemaudio_tests
-
-LOCAL_MODULE_TAGS := tests
-
-LOCAL_CFLAGS := -Werror -Wall
-
-LOCAL_MULTILIB := $(AUDIOSERVER_MULTILIB)
-
-LOCAL_COMPATIBILITY_SUITE := device-tests
-
-include $(BUILD_NATIVE_TEST)
diff --git a/services/audiopolicy/tests/AudioPolicyManagerTestClient.h b/services/audiopolicy/tests/AudioPolicyManagerTestClient.h
new file mode 100644
index 0000000..c2a92d7
--- /dev/null
+++ b/services/audiopolicy/tests/AudioPolicyManagerTestClient.h
@@ -0,0 +1,111 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <map>
+
+#include <system/audio.h>
+#include <utils/Log.h>
+#include <utils/String8.h>
+
+#include "AudioPolicyTestClient.h"
+
+namespace android {
+
+class AudioPolicyManagerTestClient : public AudioPolicyTestClient {
+public:
+ // AudioPolicyClientInterface implementation
+ audio_module_handle_t loadHwModule(const char * /*name*/) override {
+ return mNextModuleHandle++;
+ }
+
+ status_t openOutput(audio_module_handle_t module,
+ audio_io_handle_t *output,
+ audio_config_t * /*config*/,
+ const sp<DeviceDescriptorBase>& /*device*/,
+ uint32_t * /*latencyMs*/,
+ audio_output_flags_t /*flags*/) override {
+ if (module >= mNextModuleHandle) {
+ ALOGE("%s: Module handle %d has not been allocated yet (next is %d)",
+ __func__, module, mNextModuleHandle);
+ return BAD_VALUE;
+ }
+ *output = mNextIoHandle++;
+ return NO_ERROR;
+ }
+
+ audio_io_handle_t openDuplicateOutput(audio_io_handle_t /*output1*/,
+ audio_io_handle_t /*output2*/) override {
+ audio_io_handle_t id = mNextIoHandle++;
+ return id;
+ }
+
+ status_t openInput(audio_module_handle_t module,
+ audio_io_handle_t *input,
+ audio_config_t * /*config*/,
+ audio_devices_t * /*device*/,
+ const String8 & /*address*/,
+ audio_source_t /*source*/,
+ audio_input_flags_t /*flags*/) override {
+ if (module >= mNextModuleHandle) {
+ ALOGE("%s: Module handle %d has not been allocated yet (next is %d)",
+ __func__, module, mNextModuleHandle);
+ return BAD_VALUE;
+ }
+ *input = mNextIoHandle++;
+ return NO_ERROR;
+ }
+
+ status_t createAudioPatch(const struct audio_patch *patch,
+ audio_patch_handle_t *handle,
+ int /*delayMs*/) override {
+ *handle = mNextPatchHandle++;
+ mActivePatches.insert(std::make_pair(*handle, *patch));
+ return NO_ERROR;
+ }
+
+ status_t releaseAudioPatch(audio_patch_handle_t handle,
+ int /*delayMs*/) override {
+ if (mActivePatches.erase(handle) != 1) {
+ if (handle >= mNextPatchHandle) {
+ ALOGE("%s: Patch handle %d has not been allocated yet (next is %d)",
+ __func__, handle, mNextPatchHandle);
+ } else {
+ ALOGE("%s: Attempt to release patch %d twice", __func__, handle);
+ }
+ return BAD_VALUE;
+ }
+ return NO_ERROR;
+ }
+
+ // Helper methods for tests
+ size_t getActivePatchesCount() const { return mActivePatches.size(); }
+
+ const struct audio_patch *getLastAddedPatch() const {
+ if (mActivePatches.empty()) {
+ return nullptr;
+ }
+ auto it = --mActivePatches.end();
+ return &it->second;
+ };
+
+private:
+ audio_module_handle_t mNextModuleHandle = AUDIO_MODULE_HANDLE_NONE + 1;
+ audio_io_handle_t mNextIoHandle = AUDIO_IO_HANDLE_NONE + 1;
+ audio_patch_handle_t mNextPatchHandle = AUDIO_PATCH_HANDLE_NONE + 1;
+ std::map<audio_patch_handle_t, struct audio_patch> mActivePatches;
+};
+
+} // namespace android
diff --git a/services/audiopolicy/tests/AudioPolicyTestClient.h b/services/audiopolicy/tests/AudioPolicyTestClient.h
index e4c64e5..b92a2e6 100644
--- a/services/audiopolicy/tests/AudioPolicyTestClient.h
+++ b/services/audiopolicy/tests/AudioPolicyTestClient.h
@@ -31,8 +31,7 @@
status_t openOutput(audio_module_handle_t /*module*/,
audio_io_handle_t* /*output*/,
audio_config_t* /*config*/,
- audio_devices_t* /*devices*/,
- const String8& /*address*/,
+ const sp<DeviceDescriptorBase>& /*device*/,
uint32_t* /*latencyMs*/,
audio_output_flags_t /*flags*/) override { return NO_INIT; }
audio_io_handle_t openDuplicateOutput(audio_io_handle_t /*output1*/,
diff --git a/services/audiopolicy/tests/AudioPolicyTestManager.h b/services/audiopolicy/tests/AudioPolicyTestManager.h
index fe543a6..c77dcdc 100644
--- a/services/audiopolicy/tests/AudioPolicyTestManager.h
+++ b/services/audiopolicy/tests/AudioPolicyTestManager.h
@@ -24,6 +24,7 @@
explicit AudioPolicyTestManager(AudioPolicyClientInterface *clientInterface)
: AudioPolicyManager(clientInterface, true /*forTesting*/) { }
using AudioPolicyManager::getConfig;
+ using AudioPolicyManager::loadConfig;
using AudioPolicyManager::initialize;
};
diff --git a/services/audiopolicy/tests/audio_health_tests.cpp b/services/audiopolicy/tests/audio_health_tests.cpp
new file mode 100644
index 0000000..8736cf1
--- /dev/null
+++ b/services/audiopolicy/tests/audio_health_tests.cpp
@@ -0,0 +1,76 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "AudioPolicy_Boot_Test"
+
+#include <unordered_set>
+
+#include <gtest/gtest.h>
+
+#include <media/AudioSystem.h>
+#include <system/audio.h>
+#include <utils/Log.h>
+
+#include "AudioPolicyManagerTestClient.h"
+#include "AudioPolicyTestManager.h"
+
+using namespace android;
+
+TEST(AudioHealthTest, AttachedDeviceFound) {
+ unsigned int numPorts;
+ unsigned int generation1;
+ unsigned int generation;
+ struct audio_port *audioPorts = NULL;
+ int attempts = 10;
+ do {
+ if (attempts-- < 0) {
+ free(audioPorts);
+ GTEST_FAIL() << "Query audio ports time out";
+ }
+ numPorts = 0;
+ ASSERT_EQ(NO_ERROR, AudioSystem::listAudioPorts(
+ AUDIO_PORT_ROLE_NONE, AUDIO_PORT_TYPE_DEVICE, &numPorts, NULL, &generation1));
+ if (numPorts == 0) {
+ free(audioPorts);
+ GTEST_FAIL() << "Number of audio ports should not be zero";
+ }
+
+ audioPorts = (struct audio_port *)realloc(audioPorts, numPorts * sizeof(struct audio_port));
+ status_t status = AudioSystem::listAudioPorts(
+ AUDIO_PORT_ROLE_NONE, AUDIO_PORT_TYPE_DEVICE, &numPorts, audioPorts, &generation);
+ if (status != NO_ERROR) {
+ free(audioPorts);
+ GTEST_FAIL() << "Query audio ports failed";
+ }
+ } while (generation1 != generation);
+ std::unordered_set<audio_devices_t> attachedDevices;
+ for (int i = 0 ; i < numPorts; i++) {
+ attachedDevices.insert(audioPorts[i].ext.device.type);
+ }
+ free(audioPorts);
+
+ AudioPolicyManagerTestClient client;
+ AudioPolicyTestManager manager(&client);
+ manager.loadConfig();
+ ASSERT_NE("AudioPolicyConfig::setDefault", manager.getConfig().getSource());
+
+ for (auto desc : manager.getConfig().getAvailableInputDevices()) {
+ ASSERT_NE(attachedDevices.end(), attachedDevices.find(desc->type()));
+ }
+ for (auto desc : manager.getConfig().getAvailableOutputDevices()) {
+ ASSERT_NE(attachedDevices.end(), attachedDevices.find(desc->type()));
+ }
+}
diff --git a/services/audiopolicy/tests/audiopolicymanager_tests.cpp b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
index de5670c..0263597 100644
--- a/services/audiopolicy/tests/audiopolicymanager_tests.cpp
+++ b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
@@ -15,22 +15,38 @@
*/
#include <memory>
-#include <set>
+#include <string>
#include <sys/wait.h>
#include <unistd.h>
#include <gtest/gtest.h>
#define LOG_TAG "APM_Test"
-#include <log/log.h>
+#include <Serializer.h>
+#include <android-base/file.h>
+#include <media/AudioPolicy.h>
#include <media/PatchBuilder.h>
+#include <media/RecordingActivityTracker.h>
+#include <utils/Log.h>
+#include <utils/Vector.h>
+#include "AudioPolicyInterface.h"
+#include "AudioPolicyManagerTestClient.h"
#include "AudioPolicyTestClient.h"
#include "AudioPolicyTestManager.h"
using namespace android;
-TEST(AudioPolicyManagerTestInit, Failure) {
+TEST(AudioPolicyManagerTestInit, EngineFailure) {
+ AudioPolicyTestClient client;
+ AudioPolicyTestManager manager(&client);
+ manager.getConfig().setDefault();
+ manager.getConfig().setEngineLibraryNameSuffix("non-existent");
+ ASSERT_EQ(NO_INIT, manager.initialize());
+ ASSERT_EQ(NO_INIT, manager.initCheck());
+}
+
+TEST(AudioPolicyManagerTestInit, ClientFailure) {
AudioPolicyTestClient client;
AudioPolicyTestManager manager(&client);
manager.getConfig().setDefault();
@@ -41,77 +57,6 @@
}
-class AudioPolicyManagerTestClient : public AudioPolicyTestClient {
- public:
- // AudioPolicyClientInterface implementation
- audio_module_handle_t loadHwModule(const char* /*name*/) override {
- return mNextModuleHandle++;
- }
-
- status_t openOutput(audio_module_handle_t module,
- audio_io_handle_t* output,
- audio_config_t* /*config*/,
- audio_devices_t* /*devices*/,
- const String8& /*address*/,
- uint32_t* /*latencyMs*/,
- audio_output_flags_t /*flags*/) override {
- if (module >= mNextModuleHandle) {
- ALOGE("%s: Module handle %d has not been allocated yet (next is %d)",
- __func__, module, mNextModuleHandle);
- return BAD_VALUE;
- }
- *output = mNextIoHandle++;
- return NO_ERROR;
- }
-
- status_t openInput(audio_module_handle_t module,
- audio_io_handle_t* input,
- audio_config_t* /*config*/,
- audio_devices_t* /*device*/,
- const String8& /*address*/,
- audio_source_t /*source*/,
- audio_input_flags_t /*flags*/) override {
- if (module >= mNextModuleHandle) {
- ALOGE("%s: Module handle %d has not been allocated yet (next is %d)",
- __func__, module, mNextModuleHandle);
- return BAD_VALUE;
- }
- *input = mNextIoHandle++;
- return NO_ERROR;
- }
-
- status_t createAudioPatch(const struct audio_patch* /*patch*/,
- audio_patch_handle_t* handle,
- int /*delayMs*/) override {
- *handle = mNextPatchHandle++;
- mActivePatches.insert(*handle);
- return NO_ERROR;
- }
-
- status_t releaseAudioPatch(audio_patch_handle_t handle,
- int /*delayMs*/) override {
- if (mActivePatches.erase(handle) != 1) {
- if (handle >= mNextPatchHandle) {
- ALOGE("%s: Patch handle %d has not been allocated yet (next is %d)",
- __func__, handle, mNextPatchHandle);
- } else {
- ALOGE("%s: Attempt to release patch %d twice", __func__, handle);
- }
- return BAD_VALUE;
- }
- return NO_ERROR;
- }
-
- // Helper methods for tests
- size_t getActivePatchesCount() const { return mActivePatches.size(); }
-
- private:
- audio_module_handle_t mNextModuleHandle = AUDIO_MODULE_HANDLE_NONE + 1;
- audio_io_handle_t mNextIoHandle = AUDIO_IO_HANDLE_NONE + 1;
- audio_patch_handle_t mNextPatchHandle = AUDIO_PATCH_HANDLE_NONE + 1;
- std::set<audio_patch_handle_t> mActivePatches;
-};
-
class PatchCountCheck {
public:
explicit PatchCountCheck(AudioPolicyManagerTestClient *client)
@@ -134,18 +79,34 @@
protected:
void SetUp() override;
void TearDown() override;
- virtual void SetUpConfig(AudioPolicyConfig *config) { (void)config; }
+ virtual void SetUpManagerConfig();
void dumpToLog();
+ // When explicitly routing is needed, selectedDeviceId need to be set as the wanted port
+ // id. Otherwise, selectedDeviceId need to be initialized as AUDIO_PORT_HANDLE_NONE.
void getOutputForAttr(
audio_port_handle_t *selectedDeviceId,
audio_format_t format,
int channelMask,
int sampleRate,
audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
+ audio_port_handle_t *portId = nullptr,
+ audio_attributes_t attr = {});
+ void getInputForAttr(
+ const audio_attributes_t &attr,
+ audio_unique_id_t riid,
+ audio_port_handle_t *selectedDeviceId,
+ audio_format_t format,
+ int channelMask,
+ int sampleRate,
+ audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE,
audio_port_handle_t *portId = nullptr);
PatchCountCheck snapshotPatchCount() { return PatchCountCheck(mClient.get()); }
+ void findDevicePort(audio_port_role_t role, audio_devices_t deviceType,
+ const std::string &address, audio_port &foundPort);
+ static audio_port_handle_t getDeviceIdFromPatch(const struct audio_patch* patch);
+
std::unique_ptr<AudioPolicyManagerTestClient> mClient;
std::unique_ptr<AudioPolicyTestManager> mManager;
};
@@ -153,8 +114,7 @@
void AudioPolicyManagerTest::SetUp() {
mClient.reset(new AudioPolicyManagerTestClient);
mManager.reset(new AudioPolicyTestManager(mClient.get()));
- mManager->getConfig().setDefault();
- SetUpConfig(&mManager->getConfig()); // Subclasses may want to customize the config.
+ SetUpManagerConfig(); // Subclasses may want to customize the config.
ASSERT_EQ(NO_ERROR, mManager->initialize());
ASSERT_EQ(NO_ERROR, mManager->initCheck());
}
@@ -164,6 +124,10 @@
mClient.reset();
}
+void AudioPolicyManagerTest::SetUpManagerConfig() {
+ mManager->getConfig().setDefault();
+}
+
void AudioPolicyManagerTest::dumpToLog() {
int pipefd[2];
ASSERT_NE(-1, pipe(pipefd));
@@ -200,15 +164,14 @@
int channelMask,
int sampleRate,
audio_output_flags_t flags,
- audio_port_handle_t *portId) {
- audio_attributes_t attr = {};
+ audio_port_handle_t *portId,
+ audio_attributes_t attr) {
audio_io_handle_t output = AUDIO_PORT_HANDLE_NONE;
audio_stream_type_t stream = AUDIO_STREAM_DEFAULT;
audio_config_t config = AUDIO_CONFIG_INITIALIZER;
config.sample_rate = sampleRate;
config.channel_mask = channelMask;
config.format = format;
- *selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
audio_port_handle_t localPortId;
if (!portId) portId = &localPortId;
*portId = AUDIO_PORT_HANDLE_NONE;
@@ -218,6 +181,71 @@
ASSERT_NE(AUDIO_PORT_HANDLE_NONE, *portId);
}
+void AudioPolicyManagerTest::getInputForAttr(
+ const audio_attributes_t &attr,
+ audio_unique_id_t riid,
+ audio_port_handle_t *selectedDeviceId,
+ audio_format_t format,
+ int channelMask,
+ int sampleRate,
+ audio_input_flags_t flags,
+ audio_port_handle_t *portId) {
+ audio_io_handle_t input = AUDIO_PORT_HANDLE_NONE;
+ audio_config_base_t config = AUDIO_CONFIG_BASE_INITIALIZER;
+ config.sample_rate = sampleRate;
+ config.channel_mask = channelMask;
+ config.format = format;
+ audio_port_handle_t localPortId;
+ if (!portId) portId = &localPortId;
+ *portId = AUDIO_PORT_HANDLE_NONE;
+ AudioPolicyInterface::input_type_t inputType;
+ ASSERT_EQ(OK, mManager->getInputForAttr(
+ &attr, &input, riid, AUDIO_SESSION_NONE, 0 /*uid*/, &config, flags,
+ selectedDeviceId, &inputType, portId));
+ ASSERT_NE(AUDIO_PORT_HANDLE_NONE, *portId);
+}
+
+void AudioPolicyManagerTest::findDevicePort(audio_port_role_t role,
+ audio_devices_t deviceType, const std::string &address, audio_port &foundPort) {
+ uint32_t numPorts = 0;
+ uint32_t generation1;
+ status_t ret;
+
+ ret = mManager->listAudioPorts(role, AUDIO_PORT_TYPE_DEVICE, &numPorts, nullptr, &generation1);
+ ASSERT_EQ(NO_ERROR, ret);
+
+ uint32_t generation2;
+ struct audio_port ports[numPorts];
+ ret = mManager->listAudioPorts(role, AUDIO_PORT_TYPE_DEVICE, &numPorts, ports, &generation2);
+ ASSERT_EQ(NO_ERROR, ret);
+ ASSERT_EQ(generation1, generation2);
+
+ for (const auto &port : ports) {
+ if (port.role == role && port.ext.device.type == deviceType &&
+ (strncmp(port.ext.device.address, address.c_str(),
+ AUDIO_DEVICE_MAX_ADDRESS_LEN) == 0)) {
+ foundPort = port;
+ return;
+ }
+ }
+ GTEST_FAIL();
+}
+
+audio_port_handle_t AudioPolicyManagerTest::getDeviceIdFromPatch(
+ const struct audio_patch* patch) {
+ // The logic here is the same as the one in AudioIoDescriptor.
+ // Note this function is aim to get routed device id for test.
+ // In that case, device to device patch is not expected here.
+ if (patch->num_sources != 0 && patch->num_sinks != 0) {
+ if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) {
+ return patch->sinks[0].id;
+ } else {
+ return patch->sources[0].id;
+ }
+ }
+ return AUDIO_PORT_HANDLE_NONE;
+}
+
TEST_F(AudioPolicyManagerTest, InitSuccess) {
// SetUp must finish with no assertions.
@@ -277,15 +305,17 @@
class AudioPolicyManagerTestMsd : public AudioPolicyManagerTest {
protected:
- void SetUpConfig(AudioPolicyConfig *config) override;
+ void SetUpManagerConfig() override;
void TearDown() override;
sp<DeviceDescriptor> mMsdOutputDevice;
sp<DeviceDescriptor> mMsdInputDevice;
};
-void AudioPolicyManagerTestMsd::SetUpConfig(AudioPolicyConfig *config) {
+void AudioPolicyManagerTestMsd::SetUpManagerConfig() {
// TODO: Consider using Serializer to load part of the config from a string.
+ AudioPolicyManagerTest::SetUpManagerConfig();
+ AudioPolicyConfig& config = mManager->getConfig();
mMsdOutputDevice = new DeviceDescriptor(AUDIO_DEVICE_OUT_BUS);
sp<AudioProfile> pcmOutputProfile = new AudioProfile(
AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO, 48000);
@@ -298,22 +328,21 @@
sp<AudioProfile> pcmInputProfile = new AudioProfile(
AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_IN_STEREO, 44100);
mMsdInputDevice->addAudioProfile(pcmInputProfile);
- config->addAvailableDevice(mMsdOutputDevice);
- config->addAvailableDevice(mMsdInputDevice);
+ config.addAvailableDevice(mMsdOutputDevice);
+ config.addAvailableDevice(mMsdInputDevice);
sp<HwModule> msdModule = new HwModule(AUDIO_HARDWARE_MODULE_ID_MSD, 2 /*halVersionMajor*/);
- HwModuleCollection modules = config->getHwModules();
+ HwModuleCollection modules = config.getHwModules();
modules.add(msdModule);
- config->setHwModules(modules);
+ config.setHwModules(modules);
mMsdOutputDevice->attach(msdModule);
mMsdInputDevice->attach(msdModule);
- sp<OutputProfile> msdOutputProfile = new OutputProfile(String8("msd input"));
+ sp<OutputProfile> msdOutputProfile = new OutputProfile("msd input");
msdOutputProfile->addAudioProfile(pcmOutputProfile);
msdOutputProfile->addSupportedDevice(mMsdOutputDevice);
msdModule->addOutputProfile(msdOutputProfile);
- sp<OutputProfile> msdCompressedOutputProfile =
- new OutputProfile(String8("msd compressed input"));
+ sp<OutputProfile> msdCompressedOutputProfile = new OutputProfile("msd compressed input");
msdCompressedOutputProfile->addAudioProfile(ac3OutputProfile);
msdCompressedOutputProfile->setFlags(
AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD |
@@ -321,7 +350,7 @@
msdCompressedOutputProfile->addSupportedDevice(mMsdOutputDevice);
msdModule->addOutputProfile(msdCompressedOutputProfile);
- sp<InputProfile> msdInputProfile = new InputProfile(String8("msd output"));
+ sp<InputProfile> msdInputProfile = new InputProfile("msd output");
msdInputProfile->addAudioProfile(pcmInputProfile);
msdInputProfile->addSupportedDevice(mMsdInputDevice);
msdModule->addInputProfile(msdInputProfile);
@@ -330,12 +359,12 @@
// of streams that are not supported by MSD.
sp<AudioProfile> dtsOutputProfile = new AudioProfile(
AUDIO_FORMAT_DTS, AUDIO_CHANNEL_OUT_5POINT1, 48000);
- config->getDefaultOutputDevice()->addAudioProfile(dtsOutputProfile);
- sp<OutputProfile> primaryEncodedOutputProfile = new OutputProfile(String8("encoded"));
+ config.getDefaultOutputDevice()->addAudioProfile(dtsOutputProfile);
+ sp<OutputProfile> primaryEncodedOutputProfile = new OutputProfile("encoded");
primaryEncodedOutputProfile->addAudioProfile(dtsOutputProfile);
primaryEncodedOutputProfile->setFlags(AUDIO_OUTPUT_FLAG_DIRECT);
- primaryEncodedOutputProfile->addSupportedDevice(config->getDefaultOutputDevice());
- config->getHwModules().getModuleFromName(AUDIO_HARDWARE_MODULE_ID_PRIMARY)->
+ primaryEncodedOutputProfile->addSupportedDevice(config.getDefaultOutputDevice());
+ config.getHwModules().getModuleFromName(AUDIO_HARDWARE_MODULE_ID_PRIMARY)->
addOutputProfile(primaryEncodedOutputProfile);
}
@@ -363,7 +392,7 @@
TEST_F(AudioPolicyManagerTestMsd, GetOutputForAttrEncodedRoutesToMsd) {
const PatchCountCheck patchCount = snapshotPatchCount();
- audio_port_handle_t selectedDeviceId;
+ audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
getOutputForAttr(&selectedDeviceId,
AUDIO_FORMAT_AC3, AUDIO_CHANNEL_OUT_5POINT1, 48000, AUDIO_OUTPUT_FLAG_DIRECT);
ASSERT_EQ(selectedDeviceId, mMsdOutputDevice->getId());
@@ -372,7 +401,7 @@
TEST_F(AudioPolicyManagerTestMsd, GetOutputForAttrPcmRoutesToMsd) {
const PatchCountCheck patchCount = snapshotPatchCount();
- audio_port_handle_t selectedDeviceId;
+ audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
getOutputForAttr(&selectedDeviceId,
AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO, 48000);
ASSERT_EQ(selectedDeviceId, mMsdOutputDevice->getId());
@@ -381,7 +410,7 @@
TEST_F(AudioPolicyManagerTestMsd, GetOutputForAttrEncodedPlusPcmRoutesToMsd) {
const PatchCountCheck patchCount = snapshotPatchCount();
- audio_port_handle_t selectedDeviceId;
+ audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
getOutputForAttr(&selectedDeviceId,
AUDIO_FORMAT_AC3, AUDIO_CHANNEL_OUT_5POINT1, 48000, AUDIO_OUTPUT_FLAG_DIRECT);
ASSERT_EQ(selectedDeviceId, mMsdOutputDevice->getId());
@@ -394,7 +423,7 @@
TEST_F(AudioPolicyManagerTestMsd, GetOutputForAttrUnsupportedFormatBypassesMsd) {
const PatchCountCheck patchCount = snapshotPatchCount();
- audio_port_handle_t selectedDeviceId;
+ audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
getOutputForAttr(&selectedDeviceId,
AUDIO_FORMAT_DTS, AUDIO_CHANNEL_OUT_5POINT1, 48000, AUDIO_OUTPUT_FLAG_DIRECT);
ASSERT_NE(selectedDeviceId, mMsdOutputDevice->getId());
@@ -405,7 +434,8 @@
// Switch between formats that are supported and not supported by MSD.
{
const PatchCountCheck patchCount = snapshotPatchCount();
- audio_port_handle_t selectedDeviceId, portId;
+ audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
+ audio_port_handle_t portId;
getOutputForAttr(&selectedDeviceId,
AUDIO_FORMAT_AC3, AUDIO_CHANNEL_OUT_5POINT1, 48000, AUDIO_OUTPUT_FLAG_DIRECT,
&portId);
@@ -416,7 +446,8 @@
}
{
const PatchCountCheck patchCount = snapshotPatchCount();
- audio_port_handle_t selectedDeviceId, portId;
+ audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
+ audio_port_handle_t portId;
getOutputForAttr(&selectedDeviceId,
AUDIO_FORMAT_DTS, AUDIO_CHANNEL_OUT_5POINT1, 48000, AUDIO_OUTPUT_FLAG_DIRECT,
&portId);
@@ -427,10 +458,603 @@
}
{
const PatchCountCheck patchCount = snapshotPatchCount();
- audio_port_handle_t selectedDeviceId;
+ audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
getOutputForAttr(&selectedDeviceId,
AUDIO_FORMAT_AC3, AUDIO_CHANNEL_OUT_5POINT1, 48000, AUDIO_OUTPUT_FLAG_DIRECT);
ASSERT_EQ(selectedDeviceId, mMsdOutputDevice->getId());
ASSERT_EQ(0, patchCount.deltaFromSnapshot());
}
}
+
+class AudioPolicyManagerTestWithConfigurationFile : public AudioPolicyManagerTest {
+protected:
+ void SetUpManagerConfig() override;
+ virtual std::string getConfigFile() { return sDefaultConfig; }
+
+ static const std::string sExecutableDir;
+ static const std::string sDefaultConfig;
+};
+
+const std::string AudioPolicyManagerTestWithConfigurationFile::sExecutableDir =
+ base::GetExecutableDirectory() + "/";
+
+const std::string AudioPolicyManagerTestWithConfigurationFile::sDefaultConfig =
+ sExecutableDir + "test_audio_policy_configuration.xml";
+
+void AudioPolicyManagerTestWithConfigurationFile::SetUpManagerConfig() {
+ status_t status = deserializeAudioPolicyFile(getConfigFile().c_str(), &mManager->getConfig());
+ ASSERT_EQ(NO_ERROR, status);
+}
+
+TEST_F(AudioPolicyManagerTestWithConfigurationFile, InitSuccess) {
+ // SetUp must finish with no assertions.
+}
+
+TEST_F(AudioPolicyManagerTestWithConfigurationFile, Dump) {
+ dumpToLog();
+}
+
+using PolicyMixTuple = std::tuple<audio_usage_t, audio_source_t, uint32_t>;
+
+class AudioPolicyManagerTestDynamicPolicy : public AudioPolicyManagerTestWithConfigurationFile {
+protected:
+ void TearDown() override;
+
+ status_t addPolicyMix(int mixType, int mixFlag, audio_devices_t deviceType,
+ std::string mixAddress, const audio_config_t& audioConfig,
+ const std::vector<PolicyMixTuple>& rules);
+ void clearPolicyMix();
+
+ Vector<AudioMix> mAudioMixes;
+ const std::string mMixAddress = "remote_submix_media";
+};
+
+void AudioPolicyManagerTestDynamicPolicy::TearDown() {
+ mManager->unregisterPolicyMixes(mAudioMixes);
+ AudioPolicyManagerTestWithConfigurationFile::TearDown();
+}
+
+status_t AudioPolicyManagerTestDynamicPolicy::addPolicyMix(int mixType, int mixFlag,
+ audio_devices_t deviceType, std::string mixAddress, const audio_config_t& audioConfig,
+ const std::vector<PolicyMixTuple>& rules) {
+ Vector<AudioMixMatchCriterion> myMixMatchCriteria;
+
+ for(const auto &rule: rules) {
+ myMixMatchCriteria.add(AudioMixMatchCriterion(
+ std::get<0>(rule), std::get<1>(rule), std::get<2>(rule)));
+ }
+
+ AudioMix myAudioMix(myMixMatchCriteria, mixType, audioConfig, mixFlag,
+ String8(mixAddress.c_str()), 0);
+ myAudioMix.mDeviceType = deviceType;
+ // Clear mAudioMix before add new one to make sure we don't add already exist mixes.
+ mAudioMixes.clear();
+ mAudioMixes.add(myAudioMix);
+
+ // As the policy mixes registration may fail at some case,
+ // caller need to check the returned status.
+ status_t ret = mManager->registerPolicyMixes(mAudioMixes);
+ return ret;
+}
+
+void AudioPolicyManagerTestDynamicPolicy::clearPolicyMix() {
+ if (mManager != nullptr) {
+ mManager->unregisterPolicyMixes(mAudioMixes);
+ }
+ mAudioMixes.clear();
+}
+
+TEST_F(AudioPolicyManagerTestDynamicPolicy, InitSuccess) {
+ // SetUp must finish with no assertions
+}
+
+TEST_F(AudioPolicyManagerTestDynamicPolicy, Dump) {
+ dumpToLog();
+}
+
+TEST_F(AudioPolicyManagerTestDynamicPolicy, RegisterPolicyMixes) {
+ status_t ret;
+ audio_config_t audioConfig = AUDIO_CONFIG_INITIALIZER;
+
+ // Only capture of playback is allowed in LOOP_BACK &RENDER mode
+ ret = addPolicyMix(MIX_TYPE_RECORDERS, MIX_ROUTE_FLAG_LOOP_BACK_AND_RENDER,
+ AUDIO_DEVICE_OUT_REMOTE_SUBMIX, "", audioConfig, std::vector<PolicyMixTuple>());
+ ASSERT_EQ(INVALID_OPERATION, ret);
+
+ // Fail due to the device is already connected.
+ clearPolicyMix();
+ ret = addPolicyMix(MIX_TYPE_PLAYERS, MIX_ROUTE_FLAG_LOOP_BACK,
+ AUDIO_DEVICE_OUT_REMOTE_SUBMIX, "", audioConfig, std::vector<PolicyMixTuple>());
+ ASSERT_EQ(INVALID_OPERATION, ret);
+
+ // The first time to register policy mixes with valid parameter should succeed.
+ clearPolicyMix();
+ audioConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+ audioConfig.format = AUDIO_FORMAT_PCM_16_BIT;
+ audioConfig.sample_rate = 48000;
+ ret = addPolicyMix(MIX_TYPE_PLAYERS, MIX_ROUTE_FLAG_LOOP_BACK,
+ AUDIO_DEVICE_OUT_REMOTE_SUBMIX, mMixAddress, audioConfig,
+ std::vector<PolicyMixTuple>());
+ ASSERT_EQ(NO_ERROR, ret);
+ // Registering the same policy mixes should fail.
+ ret = mManager->registerPolicyMixes(mAudioMixes);
+ ASSERT_EQ(INVALID_OPERATION, ret);
+
+ // Registration should fail due to device not found.
+ // Note that earpiece is not present in the test configuration file.
+ // This will need to be updated if earpiece is added in the test configuration file.
+ clearPolicyMix();
+ ret = addPolicyMix(MIX_TYPE_PLAYERS, MIX_ROUTE_FLAG_RENDER,
+ AUDIO_DEVICE_OUT_EARPIECE, "", audioConfig, std::vector<PolicyMixTuple>());
+ ASSERT_EQ(INVALID_OPERATION, ret);
+
+ // Registration should fail due to output not found.
+ clearPolicyMix();
+ ret = addPolicyMix(MIX_TYPE_PLAYERS, MIX_ROUTE_FLAG_RENDER,
+ AUDIO_DEVICE_OUT_REMOTE_SUBMIX, "", audioConfig, std::vector<PolicyMixTuple>());
+ ASSERT_EQ(INVALID_OPERATION, ret);
+
+ // The first time to register valid policy mixes should succeed.
+ clearPolicyMix();
+ ret = addPolicyMix(MIX_TYPE_PLAYERS, MIX_ROUTE_FLAG_RENDER,
+ AUDIO_DEVICE_OUT_SPEAKER, "", audioConfig, std::vector<PolicyMixTuple>());
+ ASSERT_EQ(NO_ERROR, ret);
+ // Registering the same policy mixes should fail.
+ ret = mManager->registerPolicyMixes(mAudioMixes);
+ ASSERT_EQ(INVALID_OPERATION, ret);
+}
+
+TEST_F(AudioPolicyManagerTestDynamicPolicy, UnregisterPolicyMixes) {
+ status_t ret;
+ audio_config_t audioConfig = AUDIO_CONFIG_INITIALIZER;
+
+ audioConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+ audioConfig.format = AUDIO_FORMAT_PCM_16_BIT;
+ audioConfig.sample_rate = 48000;
+ ret = addPolicyMix(MIX_TYPE_PLAYERS, MIX_ROUTE_FLAG_LOOP_BACK,
+ AUDIO_DEVICE_OUT_REMOTE_SUBMIX, mMixAddress, audioConfig,
+ std::vector<PolicyMixTuple>());
+ ASSERT_EQ(NO_ERROR, ret);
+
+ // After successfully registering policy mixes, it should be able to unregister.
+ ret = mManager->unregisterPolicyMixes(mAudioMixes);
+ ASSERT_EQ(NO_ERROR, ret);
+
+ // After unregistering policy mixes successfully, it should fail unregistering
+ // the same policy mixes as they are not registered.
+ ret = mManager->unregisterPolicyMixes(mAudioMixes);
+ ASSERT_EQ(INVALID_OPERATION, ret);
+}
+
+class AudioPolicyManagerTestDPNoRemoteSubmixModule : public AudioPolicyManagerTestDynamicPolicy {
+protected:
+ std::string getConfigFile() override { return sPrimaryOnlyConfig; }
+
+ static const std::string sPrimaryOnlyConfig;
+};
+
+const std::string AudioPolicyManagerTestDPNoRemoteSubmixModule::sPrimaryOnlyConfig =
+ sExecutableDir + "test_audio_policy_primary_only_configuration.xml";
+
+TEST_F(AudioPolicyManagerTestDPNoRemoteSubmixModule, InitSuccess) {
+ // SetUp must finish with no assertions.
+}
+
+TEST_F(AudioPolicyManagerTestDPNoRemoteSubmixModule, Dump) {
+ dumpToLog();
+}
+
+TEST_F(AudioPolicyManagerTestDPNoRemoteSubmixModule, RegistrationFailure) {
+ // Registration/Unregistration should fail due to module for remote submix not found.
+ status_t ret;
+ audio_config_t audioConfig = AUDIO_CONFIG_INITIALIZER;
+ audioConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+ audioConfig.format = AUDIO_FORMAT_PCM_16_BIT;
+ audioConfig.sample_rate = 48000;
+ ret = addPolicyMix(MIX_TYPE_PLAYERS, MIX_ROUTE_FLAG_LOOP_BACK,
+ AUDIO_DEVICE_OUT_REMOTE_SUBMIX, "", audioConfig, std::vector<PolicyMixTuple>());
+ ASSERT_EQ(INVALID_OPERATION, ret);
+
+ ret = mManager->unregisterPolicyMixes(mAudioMixes);
+ ASSERT_EQ(INVALID_OPERATION, ret);
+}
+
+class AudioPolicyManagerTestDPPlaybackReRouting : public AudioPolicyManagerTestDynamicPolicy,
+ public testing::WithParamInterface<audio_attributes_t> {
+protected:
+ void SetUp() override;
+ void TearDown() override;
+
+ std::unique_ptr<RecordingActivityTracker> mTracker;
+
+ std::vector<PolicyMixTuple> mUsageRules = {
+ {AUDIO_USAGE_MEDIA, AUDIO_SOURCE_DEFAULT, RULE_MATCH_ATTRIBUTE_USAGE},
+ {AUDIO_USAGE_ALARM, AUDIO_SOURCE_DEFAULT, RULE_MATCH_ATTRIBUTE_USAGE}
+ };
+
+ struct audio_port mInjectionPort;
+ audio_port_handle_t mPortId = AUDIO_PORT_HANDLE_NONE;
+};
+
+void AudioPolicyManagerTestDPPlaybackReRouting::SetUp() {
+ AudioPolicyManagerTestDynamicPolicy::SetUp();
+
+ mTracker.reset(new RecordingActivityTracker());
+
+ audio_config_t audioConfig = AUDIO_CONFIG_INITIALIZER;
+ audioConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+ audioConfig.format = AUDIO_FORMAT_PCM_16_BIT;
+ audioConfig.sample_rate = 48000;
+ status_t ret = addPolicyMix(MIX_TYPE_PLAYERS, MIX_ROUTE_FLAG_LOOP_BACK,
+ AUDIO_DEVICE_OUT_REMOTE_SUBMIX, mMixAddress, audioConfig, mUsageRules);
+ ASSERT_EQ(NO_ERROR, ret);
+
+ struct audio_port extractionPort;
+ findDevicePort(AUDIO_PORT_ROLE_SOURCE, AUDIO_DEVICE_IN_REMOTE_SUBMIX,
+ mMixAddress, extractionPort);
+
+ audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
+ audio_source_t source = AUDIO_SOURCE_REMOTE_SUBMIX;
+ audio_attributes_t attr = {AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN, source, 0, ""};
+ std::string tags = "addr=" + mMixAddress;
+ strncpy(attr.tags, tags.c_str(), AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - 1);
+ getInputForAttr(attr, mTracker->getRiid(), &selectedDeviceId, AUDIO_FORMAT_PCM_16_BIT,
+ AUDIO_CHANNEL_IN_STEREO, 48000 /*sampleRate*/, AUDIO_INPUT_FLAG_NONE, &mPortId);
+ ASSERT_EQ(NO_ERROR, mManager->startInput(mPortId));
+ ASSERT_EQ(extractionPort.id, selectedDeviceId);
+
+ findDevicePort(AUDIO_PORT_ROLE_SINK, AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
+ mMixAddress, mInjectionPort);
+}
+
+void AudioPolicyManagerTestDPPlaybackReRouting::TearDown() {
+ mManager->stopInput(mPortId);
+ AudioPolicyManagerTestDynamicPolicy::TearDown();
+}
+
+TEST_F(AudioPolicyManagerTestDPPlaybackReRouting, InitSuccess) {
+ // SetUp must finish with no assertions
+}
+
+TEST_F(AudioPolicyManagerTestDPPlaybackReRouting, Dump) {
+ dumpToLog();
+}
+
+TEST_P(AudioPolicyManagerTestDPPlaybackReRouting, PlaybackReRouting) {
+ const audio_attributes_t attr = GetParam();
+ const audio_usage_t usage = attr.usage;
+
+ audio_port_handle_t playbackRoutedPortId = AUDIO_PORT_HANDLE_NONE;
+ audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE;
+ getOutputForAttr(&playbackRoutedPortId, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
+ 48000 /*sampleRate*/, AUDIO_OUTPUT_FLAG_NONE, &portId, attr);
+ if (std::find_if(begin(mUsageRules), end(mUsageRules), [&usage](const auto &usageRule) {
+ return (std::get<0>(usageRule) == usage) &&
+ (std::get<2>(usageRule) == RULE_MATCH_ATTRIBUTE_USAGE);}) != end(mUsageRules) ||
+ (strncmp(attr.tags, "addr=", strlen("addr=")) == 0 &&
+ strncmp(attr.tags + strlen("addr="), mMixAddress.c_str(),
+ AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - strlen("addr=") - 1) == 0)) {
+ EXPECT_EQ(mInjectionPort.id, playbackRoutedPortId);
+ } else {
+ EXPECT_NE(mInjectionPort.id, playbackRoutedPortId);
+ }
+}
+
+INSTANTIATE_TEST_CASE_P(
+ PlaybackReroutingUsageMatch,
+ AudioPolicyManagerTestDPPlaybackReRouting,
+ testing::Values(
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_MEDIA,
+ AUDIO_SOURCE_DEFAULT, 0, ""},
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_ALARM,
+ AUDIO_SOURCE_DEFAULT, 0, ""}
+ )
+ );
+
+INSTANTIATE_TEST_CASE_P(
+ PlaybackReroutingAddressPriorityMatch,
+ AudioPolicyManagerTestDPPlaybackReRouting,
+ testing::Values(
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_MEDIA,
+ AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_VOICE_COMMUNICATION,
+ AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
+ AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING,
+ AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_ALARM,
+ AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_NOTIFICATION,
+ AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
+ AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE,
+ AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
+ AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST,
+ AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
+ AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT,
+ AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
+ AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED,
+ AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_NOTIFICATION_EVENT,
+ AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
+ AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY,
+ AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
+ AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE,
+ AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
+ AUDIO_USAGE_ASSISTANCE_SONIFICATION,
+ AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_GAME,
+ AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_VIRTUAL_SOURCE,
+ AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_ASSISTANT,
+ AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_SPEECH, AUDIO_USAGE_ASSISTANT,
+ AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"}
+ )
+ );
+
+INSTANTIATE_TEST_CASE_P(
+ PlaybackReroutingUnHandledUsages,
+ AudioPolicyManagerTestDPPlaybackReRouting,
+ testing::Values(
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_VOICE_COMMUNICATION,
+ AUDIO_SOURCE_DEFAULT, 0, ""},
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
+ AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING,
+ AUDIO_SOURCE_DEFAULT, 0, ""},
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_NOTIFICATION,
+ AUDIO_SOURCE_DEFAULT, 0, ""},
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
+ AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE,
+ AUDIO_SOURCE_DEFAULT, 0, ""},
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
+ AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST,
+ AUDIO_SOURCE_DEFAULT, 0, ""},
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
+ AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT,
+ AUDIO_SOURCE_DEFAULT, 0, ""},
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
+ AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED,
+ AUDIO_SOURCE_DEFAULT, 0, ""},
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_NOTIFICATION_EVENT,
+ AUDIO_SOURCE_DEFAULT, 0, ""},
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
+ AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY,
+ AUDIO_SOURCE_DEFAULT, 0, ""},
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
+ AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE,
+ AUDIO_SOURCE_DEFAULT, 0, ""},
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
+ AUDIO_USAGE_ASSISTANCE_SONIFICATION,
+ AUDIO_SOURCE_DEFAULT, 0, ""},
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_GAME,
+ AUDIO_SOURCE_DEFAULT, 0, ""},
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_ASSISTANT,
+ AUDIO_SOURCE_DEFAULT, 0, ""},
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_SPEECH, AUDIO_USAGE_ASSISTANT,
+ AUDIO_SOURCE_DEFAULT, 0, ""}
+ )
+ );
+
+class AudioPolicyManagerTestDPMixRecordInjection : public AudioPolicyManagerTestDynamicPolicy,
+ public testing::WithParamInterface<audio_attributes_t> {
+protected:
+ void SetUp() override;
+ void TearDown() override;
+
+ std::unique_ptr<RecordingActivityTracker> mTracker;
+
+ std::vector<PolicyMixTuple> mSourceRules = {
+ {AUDIO_USAGE_UNKNOWN, AUDIO_SOURCE_CAMCORDER, RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET},
+ {AUDIO_USAGE_UNKNOWN, AUDIO_SOURCE_MIC, RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET},
+ {AUDIO_USAGE_UNKNOWN, AUDIO_SOURCE_VOICE_COMMUNICATION, RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET}
+ };
+
+ struct audio_port mExtractionPort;
+ audio_port_handle_t mPortId = AUDIO_PORT_HANDLE_NONE;
+};
+
+void AudioPolicyManagerTestDPMixRecordInjection::SetUp() {
+ AudioPolicyManagerTestDynamicPolicy::SetUp();
+
+ mTracker.reset(new RecordingActivityTracker());
+
+ audio_config_t audioConfig = AUDIO_CONFIG_INITIALIZER;
+ audioConfig.channel_mask = AUDIO_CHANNEL_IN_STEREO;
+ audioConfig.format = AUDIO_FORMAT_PCM_16_BIT;
+ audioConfig.sample_rate = 48000;
+ status_t ret = addPolicyMix(MIX_TYPE_RECORDERS, MIX_ROUTE_FLAG_LOOP_BACK,
+ AUDIO_DEVICE_IN_REMOTE_SUBMIX, mMixAddress, audioConfig, mSourceRules);
+ ASSERT_EQ(NO_ERROR, ret);
+
+ struct audio_port injectionPort;
+ findDevicePort(AUDIO_PORT_ROLE_SINK, AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
+ mMixAddress, injectionPort);
+
+ audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
+ audio_usage_t usage = AUDIO_USAGE_VIRTUAL_SOURCE;
+ audio_attributes_t attr = {AUDIO_CONTENT_TYPE_UNKNOWN, usage, AUDIO_SOURCE_DEFAULT, 0, ""};
+ std::string tags = std::string("addr=") + mMixAddress;
+ strncpy(attr.tags, tags.c_str(), AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - 1);
+ getOutputForAttr(&selectedDeviceId, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
+ 48000 /*sampleRate*/, AUDIO_OUTPUT_FLAG_NONE, &mPortId, attr);
+ ASSERT_EQ(NO_ERROR, mManager->startOutput(mPortId));
+ ASSERT_EQ(injectionPort.id, getDeviceIdFromPatch(mClient->getLastAddedPatch()));
+
+ findDevicePort(AUDIO_PORT_ROLE_SOURCE, AUDIO_DEVICE_IN_REMOTE_SUBMIX,
+ mMixAddress, mExtractionPort);
+}
+
+void AudioPolicyManagerTestDPMixRecordInjection::TearDown() {
+ mManager->stopOutput(mPortId);
+ AudioPolicyManagerTestDynamicPolicy::TearDown();
+}
+
+TEST_F(AudioPolicyManagerTestDPMixRecordInjection, InitSuccess) {
+ // SetUp mush finish with no assertions.
+}
+
+TEST_F(AudioPolicyManagerTestDPMixRecordInjection, Dump) {
+ dumpToLog();
+}
+
+TEST_P(AudioPolicyManagerTestDPMixRecordInjection, RecordingInjection) {
+ const audio_attributes_t attr = GetParam();
+ const audio_source_t source = attr.source;
+
+ audio_port_handle_t captureRoutedPortId = AUDIO_PORT_HANDLE_NONE;
+ audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE;
+ getInputForAttr(attr, mTracker->getRiid(), &captureRoutedPortId, AUDIO_FORMAT_PCM_16_BIT,
+ AUDIO_CHANNEL_IN_STEREO, 48000 /*sampleRate*/, AUDIO_INPUT_FLAG_NONE, &portId);
+ if (std::find_if(begin(mSourceRules), end(mSourceRules), [&source](const auto &sourceRule) {
+ return (std::get<1>(sourceRule) == source) &&
+ (std::get<2>(sourceRule) == RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET);})
+ != end(mSourceRules)) {
+ EXPECT_EQ(mExtractionPort.id, captureRoutedPortId);
+ } else {
+ EXPECT_NE(mExtractionPort.id, captureRoutedPortId);
+ }
+}
+
+// No address priority rule for remote recording, address is a "don't care"
+INSTANTIATE_TEST_CASE_P(
+ RecordInjectionSourceMatch,
+ AudioPolicyManagerTestDPMixRecordInjection,
+ testing::Values(
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN,
+ AUDIO_SOURCE_CAMCORDER, 0, ""},
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN,
+ AUDIO_SOURCE_CAMCORDER, 0, "addr=remote_submix_media"},
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN,
+ AUDIO_SOURCE_MIC, 0, "addr=remote_submix_media"},
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN,
+ AUDIO_SOURCE_MIC, 0, ""},
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN,
+ AUDIO_SOURCE_VOICE_COMMUNICATION, 0, ""},
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN,
+ AUDIO_SOURCE_VOICE_COMMUNICATION, 0,
+ "addr=remote_submix_media"}
+ )
+ );
+
+// No address priority rule for remote recording
+INSTANTIATE_TEST_CASE_P(
+ RecordInjectionSourceNotMatch,
+ AudioPolicyManagerTestDPMixRecordInjection,
+ testing::Values(
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN,
+ AUDIO_SOURCE_VOICE_RECOGNITION, 0, ""},
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN,
+ AUDIO_SOURCE_HOTWORD, 0, ""},
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN,
+ AUDIO_SOURCE_VOICE_RECOGNITION, 0,
+ "addr=remote_submix_media"},
+ (audio_attributes_t){AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN,
+ AUDIO_SOURCE_HOTWORD, 0, "addr=remote_submix_media"}
+ )
+ );
+
+using DeviceConnectionTestParams =
+ std::tuple<audio_devices_t /*type*/, std::string /*name*/, std::string /*address*/>;
+
+class AudioPolicyManagerTestDeviceConnection : public AudioPolicyManagerTestWithConfigurationFile,
+ public testing::WithParamInterface<DeviceConnectionTestParams> {
+};
+
+TEST_F(AudioPolicyManagerTestDeviceConnection, InitSuccess) {
+ // SetUp must finish with no assertions.
+}
+
+TEST_F(AudioPolicyManagerTestDeviceConnection, Dump) {
+ dumpToLog();
+}
+
+TEST_P(AudioPolicyManagerTestDeviceConnection, SetDeviceConnectionState) {
+ const audio_devices_t type = std::get<0>(GetParam());
+ const std::string name = std::get<1>(GetParam());
+ const std::string address = std::get<2>(GetParam());
+
+ if (type == AUDIO_DEVICE_OUT_HDMI) {
+ // Set device connection state failed due to no device descriptor found
+ // For HDMI case, it is easier to simulate device descriptor not found error
+ // by using a undeclared encoded format.
+ ASSERT_EQ(INVALID_OPERATION, mManager->setDeviceConnectionState(
+ type, AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
+ address.c_str(), name.c_str(), AUDIO_FORMAT_MAT_2_1));
+ }
+ // Connect with valid parameters should succeed
+ ASSERT_EQ(NO_ERROR, mManager->setDeviceConnectionState(
+ type, AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
+ address.c_str(), name.c_str(), AUDIO_FORMAT_DEFAULT));
+ // Try to connect with the same device again should fail
+ ASSERT_EQ(INVALID_OPERATION, mManager->setDeviceConnectionState(
+ type, AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
+ address.c_str(), name.c_str(), AUDIO_FORMAT_DEFAULT));
+ // Disconnect the connected device should succeed
+ ASSERT_EQ(NO_ERROR, mManager->setDeviceConnectionState(
+ type, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
+ address.c_str(), name.c_str(), AUDIO_FORMAT_DEFAULT));
+ // Disconnect device that is not connected should fail
+ ASSERT_EQ(INVALID_OPERATION, mManager->setDeviceConnectionState(
+ type, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
+ address.c_str(), name.c_str(), AUDIO_FORMAT_DEFAULT));
+ // Try to set device connection state with a invalid connection state should fail
+ ASSERT_EQ(BAD_VALUE, mManager->setDeviceConnectionState(
+ type, AUDIO_POLICY_DEVICE_STATE_CNT,
+ "", "", AUDIO_FORMAT_DEFAULT));
+}
+
+TEST_P(AudioPolicyManagerTestDeviceConnection, ExplicitlyRoutingAfterConnection) {
+ const audio_devices_t type = std::get<0>(GetParam());
+ const std::string name = std::get<1>(GetParam());
+ const std::string address = std::get<2>(GetParam());
+
+ // Connect device to do explicitly routing test
+ ASSERT_EQ(NO_ERROR, mManager->setDeviceConnectionState(
+ type, AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
+ address.c_str(), name.c_str(), AUDIO_FORMAT_DEFAULT));
+
+ audio_port devicePort;
+ const audio_port_role_t role = audio_is_output_device(type)
+ ? AUDIO_PORT_ROLE_SINK : AUDIO_PORT_ROLE_SOURCE;
+ findDevicePort(role, type, address, devicePort);
+
+ audio_port_handle_t routedPortId = devicePort.id;
+ audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE;
+ // Try start input or output according to the device type
+ if (audio_is_output_devices(type)) {
+ getOutputForAttr(&routedPortId, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
+ 48000 /*sampleRate*/, AUDIO_OUTPUT_FLAG_NONE, &portId);
+ } else if (audio_is_input_device(type)) {
+ RecordingActivityTracker tracker;
+ getInputForAttr({}, tracker.getRiid(), &routedPortId, AUDIO_FORMAT_PCM_16_BIT,
+ AUDIO_CHANNEL_IN_STEREO, 48000 /*sampleRate*/, AUDIO_INPUT_FLAG_NONE, &portId);
+ }
+ ASSERT_EQ(devicePort.id, routedPortId);
+
+ ASSERT_EQ(NO_ERROR, mManager->setDeviceConnectionState(
+ type, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
+ address.c_str(), name.c_str(), AUDIO_FORMAT_DEFAULT));
+}
+
+INSTANTIATE_TEST_CASE_P(
+ DeviceConnectionState,
+ AudioPolicyManagerTestDeviceConnection,
+ testing::Values(
+ DeviceConnectionTestParams({AUDIO_DEVICE_IN_HDMI, "test_in_hdmi",
+ "audio_policy_test_in_hdmi"}),
+ DeviceConnectionTestParams({AUDIO_DEVICE_OUT_HDMI, "test_out_hdmi",
+ "audio_policy_test_out_hdmi"}),
+ DeviceConnectionTestParams({AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET, "bt_hfp_in",
+ "hfp_client_in"}),
+ DeviceConnectionTestParams({AUDIO_DEVICE_OUT_BLUETOOTH_SCO, "bt_hfp_out",
+ "hfp_client_out"})
+ )
+ );
diff --git a/services/audiopolicy/tests/resources/Android.bp b/services/audiopolicy/tests/resources/Android.bp
new file mode 100644
index 0000000..41f5ee1
--- /dev/null
+++ b/services/audiopolicy/tests/resources/Android.bp
@@ -0,0 +1,7 @@
+filegroup {
+ name: "audiopolicytest_configuration_files",
+ srcs: [
+ "test_audio_policy_configuration.xml",
+ "test_audio_policy_primary_only_configuration.xml",
+ ],
+}
diff --git a/services/audiopolicy/tests/resources/test_audio_policy_configuration.xml b/services/audiopolicy/tests/resources/test_audio_policy_configuration.xml
new file mode 100644
index 0000000..87f0ab9
--- /dev/null
+++ b/services/audiopolicy/tests/resources/test_audio_policy_configuration.xml
@@ -0,0 +1,111 @@
+<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
+<!-- Copyright (C) 2019 The Android Open Source Project
+
+ Licensed under the Apache License, Version 2.0 (the "License");
+ you may not use this file except in compliance with the License.
+ You may obtain a copy of the License at
+
+ http://www.apache.org/licenses/LICENSE-2.0
+
+ Unless required by applicable law or agreed to in writing, software
+ distributed under the License is distributed on an "AS IS" BASIS,
+ WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ See the License for the specific language governing permissions and
+ limitations under the License.
+-->
+
+<audioPolicyConfiguration version="1.0" xmlns:xi="http://www.w3.org/2001/XInclude">
+ <globalConfiguration speaker_drc_enabled="true"/>
+
+ <modules>
+ <!-- Primary module -->
+ <module name="primary" halVersion="2.0">
+ <attachedDevices>
+ <item>Speaker</item>
+ <item>Built-In Mic</item>
+ </attachedDevices>
+ <defaultOutputDevice>Speaker</defaultOutputDevice>
+ <mixPorts>
+ <mixPort name="primary output" role="source" flags="AUDIO_OUTPUT_FLAG_PRIMARY">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="primary input" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000"
+ channelMasks="AUDIO_CHANNEL_IN_STEREO"/>
+ </mixPort>
+ <mixPort name="mixport_bt_hfp_output" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="mixport_bt_hfp_input" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="8000,11025,16000,44100,48000"
+ channelMasks="AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_MONO"/>
+ </mixPort>
+ </mixPorts>
+ <devicePorts>
+ <devicePort tagName="Speaker" type="AUDIO_DEVICE_OUT_SPEAKER" role="sink">
+ </devicePort>
+ <devicePort tagName="Built-In Mic" type="AUDIO_DEVICE_IN_BUILTIN_MIC" role="source">
+ </devicePort>
+ <devicePort tagName="Hdmi" type="AUDIO_DEVICE_OUT_HDMI" role="sink">
+ </devicePort>
+ <devicePort tagName="Hdmi-In Mic" type="AUDIO_DEVICE_IN_HDMI" role="source">
+ </devicePort>
+ <devicePort tagName="BT SCO" type="AUDIO_DEVICE_OUT_BLUETOOTH_SCO"
+ role="sink" address="hfp_client_out">
+ </devicePort>
+ <devicePort tagName="BT SCO Headset Mic" type="AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET"
+ role="source" address="hfp_client_in">
+ </devicePort>
+ </devicePorts>
+ <routes>
+ <route type="mix" sink="Speaker"
+ sources="primary output"/>
+ <route type="mix" sink="primary input"
+ sources="Built-In Mic,Hdmi-In Mic"/>
+ <route type="mix" sink="Hdmi"
+ sources="primary output"/>
+ <route type="mix" sink="BT SCO"
+ sources="mixport_bt_hfp_output"/>
+ <route type="mix" sink="mixport_bt_hfp_input"
+ sources="BT SCO Headset Mic"/>
+ </routes>
+ </module>
+
+ <!-- Remote Submix module -->
+ <module name="r_submix" halVersion="2.0">
+ <attachedDevices>
+ <item>Remote Submix In</item>
+ </attachedDevices>
+ <mixPorts>
+ <mixPort name="r_submix output" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="r_submix input" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_IN_STEREO"/>
+ </mixPort>
+ </mixPorts>
+ <devicePorts>
+ <devicePort tagName="Remote Submix Out" type="AUDIO_DEVICE_OUT_REMOTE_SUBMIX" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </devicePort>
+ <devicePort tagName="Remote Submix In" type="AUDIO_DEVICE_IN_REMOTE_SUBMIX" role="source">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_IN_STEREO"/>
+ </devicePort>
+ </devicePorts>
+ <routes>
+ <route type="mix" sink="Remote Submix Out"
+ sources="r_submix output"/>
+ <route type="mix" sink="r_submix input"
+ sources="Remote Submix In"/>
+ </routes>
+ </module>
+ </modules>
+</audioPolicyConfiguration>
diff --git a/services/audiopolicy/tests/resources/test_audio_policy_primary_only_configuration.xml b/services/audiopolicy/tests/resources/test_audio_policy_primary_only_configuration.xml
new file mode 100644
index 0000000..edc0adb
--- /dev/null
+++ b/services/audiopolicy/tests/resources/test_audio_policy_primary_only_configuration.xml
@@ -0,0 +1,53 @@
+<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
+<!-- Copyright (C) 2019 The Android Open Source Project
+
+ Licensed under the Apache License, Version 2.0 (the "License");
+ you may not use this file except in compliance with the License.
+ You may obtain a copy of the License at
+
+ http://www.apache.org/licenses/LICENSE-2.0
+
+ Unless required by applicable law or agreed to in writing, software
+ distributed under the License is distributed on an "AS IS" BASIS,
+ WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ See the License for the specific language governing permissions and
+ limitations under the License.
+-->
+
+<audioPolicyConfiguration version="1.0" xmlns:xi="http://www.w3.org/2001/XInclude">
+ <globalConfiguration speaker_drc_enabled="true"/>
+
+ <modules>
+ <!-- Primary module -->
+ <module name="primary" halVersion="2.0">
+ <attachedDevices>
+ <item>Speaker</item>
+ <item>Built-In Mic</item>
+ </attachedDevices>
+ <defaultOutputDevice>Speaker</defaultOutputDevice>
+ <mixPorts>
+ <mixPort name="primary output" role="source" flags="AUDIO_OUTPUT_FLAG_PRIMARY">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="primary input" role="sink">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000"
+ channelMasks="AUDIO_CHANNEL_IN_STEREO"/>
+ </mixPort>
+ </mixPorts>
+ <devicePorts>
+ <devicePort tagName="Speaker" type="AUDIO_DEVICE_OUT_SPEAKER" role="sink">
+ </devicePort>
+ <devicePort tagName="Built-In Mic" type="AUDIO_DEVICE_IN_BUILTIN_MIC" role="source">
+ </devicePort>
+ </devicePorts>
+ <routes>
+ <route type="mix" sink="Speaker"
+ sources="primary output"/>
+ <route type="mix" sink="primary input"
+ sources="Built-In Mic"/>
+ </routes>
+ </module>
+ </modules>
+</audioPolicyConfiguration>
diff --git a/services/audiopolicy/tests/systemaudio_tests.cpp b/services/audiopolicy/tests/systemaudio_tests.cpp
deleted file mode 100644
index abaae52..0000000
--- a/services/audiopolicy/tests/systemaudio_tests.cpp
+++ /dev/null
@@ -1,117 +0,0 @@
-/*
- * Copyright (C) 2018 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include <gtest/gtest.h>
-
-#define LOG_TAG "SysAudio_Test"
-#include <log/log.h>
-#include <media/PatchBuilder.h>
-#include <system/audio.h>
-
-using namespace android;
-
-TEST(SystemAudioTest, PatchInvalid) {
- audio_patch patch{};
- ASSERT_FALSE(audio_patch_is_valid(&patch));
- patch.num_sources = AUDIO_PATCH_PORTS_MAX + 1;
- patch.num_sinks = 1;
- ASSERT_FALSE(audio_patch_is_valid(&patch));
- patch.num_sources = 1;
- patch.num_sinks = AUDIO_PATCH_PORTS_MAX + 1;
- ASSERT_FALSE(audio_patch_is_valid(&patch));
- patch.num_sources = 0;
- patch.num_sinks = 1;
- ASSERT_FALSE(audio_patch_is_valid(&patch));
-}
-
-TEST(SystemAudioTest, PatchValid) {
- const audio_port_config src = {
- .id = 1, .role = AUDIO_PORT_ROLE_SOURCE, .type = AUDIO_PORT_TYPE_DEVICE };
- // It's OK not to have sinks.
- ASSERT_TRUE(audio_patch_is_valid((PatchBuilder{}).addSource(src).patch()));
- const audio_port_config sink = {
- .id = 2, .role = AUDIO_PORT_ROLE_SINK, .type = AUDIO_PORT_TYPE_DEVICE };
- ASSERT_TRUE(audio_patch_is_valid((PatchBuilder{}).addSource(src).addSink(sink).patch()));
- ASSERT_TRUE(audio_patch_is_valid(
- (PatchBuilder{}).addSource(src).addSource(src).addSink(sink).patch()));
- ASSERT_TRUE(audio_patch_is_valid(
- (PatchBuilder{}).addSource(src).addSink(sink).addSink(sink).patch()));
- ASSERT_TRUE(audio_patch_is_valid(
- (PatchBuilder{}).addSource(src).addSource(src).
- addSink(sink).addSink(sink).patch()));
-}
-
-TEST(SystemAudioTest, PatchHwAvSync) {
- audio_port_config device_src_cfg = {
- .id = 1, .role = AUDIO_PORT_ROLE_SOURCE, .type = AUDIO_PORT_TYPE_DEVICE };
- ASSERT_FALSE(audio_port_config_has_hw_av_sync(&device_src_cfg));
- device_src_cfg.config_mask |= AUDIO_PORT_CONFIG_FLAGS;
- ASSERT_FALSE(audio_port_config_has_hw_av_sync(&device_src_cfg));
- device_src_cfg.flags.input = AUDIO_INPUT_FLAG_HW_AV_SYNC;
- ASSERT_TRUE(audio_port_config_has_hw_av_sync(&device_src_cfg));
-
- audio_port_config device_sink_cfg = {
- .id = 1, .role = AUDIO_PORT_ROLE_SINK, .type = AUDIO_PORT_TYPE_DEVICE };
- ASSERT_FALSE(audio_port_config_has_hw_av_sync(&device_sink_cfg));
- device_sink_cfg.config_mask |= AUDIO_PORT_CONFIG_FLAGS;
- ASSERT_FALSE(audio_port_config_has_hw_av_sync(&device_sink_cfg));
- device_sink_cfg.flags.output = AUDIO_OUTPUT_FLAG_HW_AV_SYNC;
- ASSERT_TRUE(audio_port_config_has_hw_av_sync(&device_sink_cfg));
-
- audio_port_config mix_sink_cfg = {
- .id = 1, .role = AUDIO_PORT_ROLE_SINK, .type = AUDIO_PORT_TYPE_MIX };
- ASSERT_FALSE(audio_port_config_has_hw_av_sync(&mix_sink_cfg));
- mix_sink_cfg.config_mask |= AUDIO_PORT_CONFIG_FLAGS;
- ASSERT_FALSE(audio_port_config_has_hw_av_sync(&mix_sink_cfg));
- mix_sink_cfg.flags.input = AUDIO_INPUT_FLAG_HW_AV_SYNC;
- ASSERT_TRUE(audio_port_config_has_hw_av_sync(&mix_sink_cfg));
-
- audio_port_config mix_src_cfg = {
- .id = 1, .role = AUDIO_PORT_ROLE_SOURCE, .type = AUDIO_PORT_TYPE_MIX };
- ASSERT_FALSE(audio_port_config_has_hw_av_sync(&mix_src_cfg));
- mix_src_cfg.config_mask |= AUDIO_PORT_CONFIG_FLAGS;
- ASSERT_FALSE(audio_port_config_has_hw_av_sync(&mix_src_cfg));
- mix_src_cfg.flags.output = AUDIO_OUTPUT_FLAG_HW_AV_SYNC;
- ASSERT_TRUE(audio_port_config_has_hw_av_sync(&mix_src_cfg));
-}
-
-TEST(SystemAudioTest, PatchEqual) {
- const audio_patch patch1{}, patch2{};
- // Invalid patches are not equal.
- ASSERT_FALSE(audio_patches_are_equal(&patch1, &patch2));
- const audio_port_config src = {
- .id = 1, .role = AUDIO_PORT_ROLE_SOURCE, .type = AUDIO_PORT_TYPE_DEVICE };
- const audio_port_config sink = {
- .id = 2, .role = AUDIO_PORT_ROLE_SINK, .type = AUDIO_PORT_TYPE_DEVICE };
- ASSERT_FALSE(audio_patches_are_equal(
- (PatchBuilder{}).addSource(src).patch(),
- (PatchBuilder{}).addSource(src).addSink(sink).patch()));
- ASSERT_TRUE(audio_patches_are_equal(
- (PatchBuilder{}).addSource(src).addSink(sink).patch(),
- (PatchBuilder{}).addSource(src).addSink(sink).patch()));
- ASSERT_FALSE(audio_patches_are_equal(
- (PatchBuilder{}).addSource(src).addSink(sink).patch(),
- (PatchBuilder{}).addSource(src).addSource(src).addSink(sink).patch()));
- audio_port_config sink_hw_av_sync = sink;
- sink_hw_av_sync.config_mask |= AUDIO_PORT_CONFIG_FLAGS;
- sink_hw_av_sync.flags.output = AUDIO_OUTPUT_FLAG_HW_AV_SYNC;
- ASSERT_FALSE(audio_patches_are_equal(
- (PatchBuilder{}).addSource(src).addSink(sink).patch(),
- (PatchBuilder{}).addSource(src).addSink(sink_hw_av_sync).patch()));
- ASSERT_TRUE(audio_patches_are_equal(
- (PatchBuilder{}).addSource(src).addSink(sink_hw_av_sync).patch(),
- (PatchBuilder{}).addSource(src).addSink(sink_hw_av_sync).patch()));
-}
diff --git a/services/camera/libcameraservice/Android.bp b/services/camera/libcameraservice/Android.bp
index 1c1f5e6..072afd2 100644
--- a/services/camera/libcameraservice/Android.bp
+++ b/services/camera/libcameraservice/Android.bp
@@ -69,6 +69,10 @@
"utils/LatencyHistogram.cpp",
],
+ header_libs: [
+ "libmediadrm_headers"
+ ],
+
shared_libs: [
"libbase",
"libdl",
@@ -86,10 +90,9 @@
"libfmq",
"libgui",
"libhardware",
- "libhwbinder",
"libhidlbase",
- "libhidltransport",
"libjpeg",
+ "libmedia_codeclist",
"libmedia_omx",
"libmemunreachable",
"libsensorprivacy",
@@ -109,6 +112,10 @@
"android.hardware.camera.device@3.5",
],
+ static_libs: [
+ "libbinderthreadstateutils",
+ ],
+
export_shared_lib_headers: [
"libbinder",
"libcamera_client",
diff --git a/services/camera/libcameraservice/CameraService.cpp b/services/camera/libcameraservice/CameraService.cpp
index b20c9a4..c566485 100644
--- a/services/camera/libcameraservice/CameraService.cpp
+++ b/services/camera/libcameraservice/CameraService.cpp
@@ -43,6 +43,7 @@
#include <binder/PermissionController.h>
#include <binder/ProcessInfoService.h>
#include <binder/IResultReceiver.h>
+#include <binderthreadstate/CallerUtils.h>
#include <cutils/atomic.h>
#include <cutils/properties.h>
#include <cutils/misc.h>
@@ -56,7 +57,6 @@
#include <media/IMediaHTTPService.h>
#include <media/mediaplayer.h>
#include <mediautils/BatteryNotifier.h>
-#include <sensorprivacy/SensorPrivacyManager.h>
#include <utils/Errors.h>
#include <utils/Log.h>
#include <utils/String16.h>
@@ -1030,7 +1030,7 @@
// Only allow clients who are being used by the current foreground device user, unless calling
// from our own process OR the caller is using the cameraserver's HIDL interface.
- if (!hardware::IPCThreadState::self()->isServingCall() && callingPid != getpid() &&
+ if (getCurrentServingCall() != BinderCallType::HWBINDER && callingPid != getpid() &&
(mAllowedUsers.find(clientUserId) == mAllowedUsers.end())) {
ALOGE("CameraService::connect X (PID %d) rejected (cannot connect from "
"device user %d, currently allowed device users: %s)", callingPid, clientUserId,
@@ -1351,7 +1351,7 @@
// If the thread serving this call is not a hwbinder thread and the caller
// isn't the cameraserver itself, and the camera id being requested is to be
// publically hidden, we should reject the connection.
- if (!hardware::IPCThreadState::self()->isServingCall() &&
+ if (getCurrentServingCall() != BinderCallType::HWBINDER &&
CameraThreadState::getCallingPid() != getpid() &&
isPublicallyHiddenSecureCamera(cameraId)) {
return true;
@@ -1372,7 +1372,8 @@
String8 id = String8(cameraId);
sp<CameraDeviceClient> client = nullptr;
String16 clientPackageNameAdj = clientPackageName;
- if (hardware::IPCThreadState::self()->isServingCall()) {
+
+ if (getCurrentServingCall() == BinderCallType::HWBINDER) {
std::string vendorClient =
StringPrintf("vendor.client.pid<%d>", CameraThreadState::getCallingPid());
clientPackageNameAdj = String16(vendorClient.c_str());
@@ -2405,7 +2406,7 @@
}
mClientPackageName = packages[0];
}
- if (!hardware::IPCThreadState::self()->isServingCall()) {
+ if (getCurrentServingCall() != BinderCallType::HWBINDER) {
mAppOpsManager = std::make_unique<AppOpsManager>();
}
}
@@ -2640,14 +2641,13 @@
void CameraService::UidPolicy::registerSelf() {
Mutex::Autolock _l(mUidLock);
- ActivityManager am;
if (mRegistered) return;
- am.registerUidObserver(this, ActivityManager::UID_OBSERVER_GONE
+ status_t res = mAm.linkToDeath(this);
+ mAm.registerUidObserver(this, ActivityManager::UID_OBSERVER_GONE
| ActivityManager::UID_OBSERVER_IDLE
| ActivityManager::UID_OBSERVER_ACTIVE | ActivityManager::UID_OBSERVER_PROCSTATE,
ActivityManager::PROCESS_STATE_UNKNOWN,
String16("cameraserver"));
- status_t res = am.linkToDeath(this);
if (res == OK) {
mRegistered = true;
ALOGV("UidPolicy: Registered with ActivityManager");
@@ -2657,9 +2657,8 @@
void CameraService::UidPolicy::unregisterSelf() {
Mutex::Autolock _l(mUidLock);
- ActivityManager am;
- am.unregisterUidObserver(this);
- am.unlinkToDeath(this);
+ mAm.unregisterUidObserver(this);
+ mAm.unlinkToDeath(this);
mRegistered = false;
mActiveUids.clear();
ALOGV("UidPolicy: Unregistered with ActivityManager");
@@ -2855,10 +2854,9 @@
if (mRegistered) {
return;
}
- SensorPrivacyManager spm;
- spm.addSensorPrivacyListener(this);
- mSensorPrivacyEnabled = spm.isSensorPrivacyEnabled();
- status_t res = spm.linkToDeath(this);
+ mSpm.addSensorPrivacyListener(this);
+ mSensorPrivacyEnabled = mSpm.isSensorPrivacyEnabled();
+ status_t res = mSpm.linkToDeath(this);
if (res == OK) {
mRegistered = true;
ALOGV("SensorPrivacyPolicy: Registered with SensorPrivacyManager");
@@ -2867,9 +2865,8 @@
void CameraService::SensorPrivacyPolicy::unregisterSelf() {
Mutex::Autolock _l(mSensorPrivacyLock);
- SensorPrivacyManager spm;
- spm.removeSensorPrivacyListener(this);
- spm.unlinkToDeath(this);
+ mSpm.removeSensorPrivacyListener(this);
+ mSpm.unlinkToDeath(this);
mRegistered = false;
ALOGV("SensorPrivacyPolicy: Unregistered with SensorPrivacyManager");
}
@@ -3032,7 +3029,7 @@
const std::set<String8>& conflictingKeys, int32_t score, int32_t ownerId,
int32_t state) {
- bool isVendorClient = hardware::IPCThreadState::self()->isServingCall();
+ bool isVendorClient = getCurrentServingCall() == BinderCallType::HWBINDER;
int32_t score_adj = isVendorClient ? kVendorClientScore : score;
int32_t state_adj = isVendorClient ? kVendorClientState: state;
@@ -3276,9 +3273,21 @@
return;
}
bool isHidden = isPublicallyHiddenSecureCamera(cameraId);
+ bool supportsHAL3 = false;
+ // supportsCameraApi also holds mInterfaceMutex, we can't call it in the
+ // HIDL onStatusChanged wrapper call (we'll hold mStatusListenerLock and
+ // mInterfaceMutex together, which can lead to deadlocks)
+ binder::Status sRet =
+ supportsCameraApi(String16(cameraId), hardware::ICameraService::API_VERSION_2,
+ &supportsHAL3);
+ if (!sRet.isOk()) {
+ ALOGW("%s: Failed to determine if device supports HAL3 %s, supportsCameraApi call failed",
+ __FUNCTION__, cameraId.string());
+ return;
+ }
// Update the status for this camera state, then send the onStatusChangedCallbacks to each
// of the listeners with both the mStatusStatus and mStatusListenerLock held
- state->updateStatus(status, cameraId, rejectSourceStates, [this,&isHidden]
+ state->updateStatus(status, cameraId, rejectSourceStates, [this, &isHidden, &supportsHAL3]
(const String8& cameraId, StatusInternal status) {
if (status != StatusInternal::ENUMERATING) {
@@ -3300,8 +3309,8 @@
Mutex::Autolock lock(mStatusListenerLock);
for (auto& listener : mListenerList) {
- if (!listener.first && isHidden) {
- ALOGV("Skipping camera discovery callback for system-only camera %s",
+ if (!listener.first && (isHidden || !supportsHAL3)) {
+ ALOGV("Skipping camera discovery callback for system-only / HAL1 camera %s",
cameraId.c_str());
continue;
}
diff --git a/services/camera/libcameraservice/CameraService.h b/services/camera/libcameraservice/CameraService.h
index 22842a1..4e04f0e 100644
--- a/services/camera/libcameraservice/CameraService.h
+++ b/services/camera/libcameraservice/CameraService.h
@@ -25,11 +25,13 @@
#include <cutils/multiuser.h>
#include <utils/Vector.h>
#include <utils/KeyedVector.h>
+#include <binder/ActivityManager.h>
#include <binder/AppOpsManager.h>
#include <binder/BinderService.h>
#include <binder/IAppOpsCallback.h>
#include <binder/IUidObserver.h>
#include <hardware/camera.h>
+#include <sensorprivacy/SensorPrivacyManager.h>
#include <android/hardware/camera/common/1.0/types.h>
@@ -48,6 +50,7 @@
#include <string>
#include <map>
#include <memory>
+#include <optional>
#include <utility>
#include <unordered_map>
#include <unordered_set>
@@ -571,6 +574,7 @@
Mutex mUidLock;
bool mRegistered;
+ ActivityManager mAm;
wp<CameraService> mService;
std::unordered_set<uid_t> mActiveUids;
// Monitored uid map to cached procState and refCount pair
@@ -597,6 +601,7 @@
virtual void binderDied(const wp<IBinder> &who);
private:
+ SensorPrivacyManager mSpm;
wp<CameraService> mService;
Mutex mSensorPrivacyLock;
bool mSensorPrivacyEnabled;
diff --git a/services/camera/libcameraservice/api2/HeicCompositeStream.cpp b/services/camera/libcameraservice/api2/HeicCompositeStream.cpp
index 9f15be0..9790e32 100644
--- a/services/camera/libcameraservice/api2/HeicCompositeStream.cpp
+++ b/services/camera/libcameraservice/api2/HeicCompositeStream.cpp
@@ -28,7 +28,7 @@
#include <utils/Log.h>
#include <utils/Trace.h>
-#include <media/ICrypto.h>
+#include <mediadrm/ICrypto.h>
#include <media/MediaCodecBuffer.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/MediaDefs.h>
diff --git a/services/camera/libcameraservice/common/CameraProviderManager.h b/services/camera/libcameraservice/common/CameraProviderManager.h
index 3a4655c..954c0d9 100644
--- a/services/camera/libcameraservice/common/CameraProviderManager.h
+++ b/services/camera/libcameraservice/common/CameraProviderManager.h
@@ -273,6 +273,7 @@
bool isLogicalCamera(const std::string& id, std::vector<std::string>* physicalCameraIds);
bool isPublicallyHiddenSecureCamera(const std::string& id) const;
+
bool isHiddenPhysicalCamera(const std::string& cameraId) const;
static const float kDepthARTolerance;
diff --git a/services/camera/libcameraservice/device3/Camera3Device.cpp b/services/camera/libcameraservice/device3/Camera3Device.cpp
index 4227a3b..bda35f3 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.cpp
+++ b/services/camera/libcameraservice/device3/Camera3Device.cpp
@@ -2165,7 +2165,9 @@
}
void Camera3Device::pauseStateNotify(bool enable) {
- Mutex::Autolock il(mInterfaceLock);
+ // We must not hold mInterfaceLock here since this function is called from
+ // RequestThread::threadLoop and holding mInterfaceLock could lead to
+ // deadlocks (http://b/143513518)
Mutex::Autolock l(mLock);
mPauseStateNotify = enable;
@@ -2742,7 +2744,9 @@
ATRACE_CALL();
bool ret = false;
- Mutex::Autolock il(mInterfaceLock);
+ // We must not hold mInterfaceLock here since this function is called from
+ // RequestThread::threadLoop and holding mInterfaceLock could lead to
+ // deadlocks (http://b/143513518)
nsecs_t maxExpectedDuration = getExpectedInFlightDuration();
Mutex::Autolock l(mLock);
@@ -5371,6 +5375,9 @@
bool Camera3Device::RequestThread::threadLoop() {
ATRACE_CALL();
status_t res;
+ // Any function called from threadLoop() must not hold mInterfaceLock since
+ // it could lead to deadlocks (disconnect() -> hold mInterfaceMutex -> wait for request thread
+ // to finish -> request thread waits on mInterfaceMutex) http://b/143513518
// Handle paused state.
if (waitIfPaused()) {
diff --git a/services/camera/libcameraservice/hidl/HidlCameraService.cpp b/services/camera/libcameraservice/hidl/HidlCameraService.cpp
index 74cfe42..1daa035 100644
--- a/services/camera/libcameraservice/hidl/HidlCameraService.cpp
+++ b/services/camera/libcameraservice/hidl/HidlCameraService.cpp
@@ -191,6 +191,14 @@
_hidl_cb(status, {});
return Void();
}
+ cameraStatusAndIds.erase(std::remove_if(cameraStatusAndIds.begin(), cameraStatusAndIds.end(),
+ [this](const hardware::CameraStatus& s) {
+ bool supportsHAL3 = false;
+ binder::Status sRet =
+ mAidlICameraService->supportsCameraApi(String16(s.cameraId),
+ hardware::ICameraService::API_VERSION_2, &supportsHAL3);
+ return !sRet.isOk() || !supportsHAL3;
+ }), cameraStatusAndIds.end());
hidl_vec<HCameraStatusAndId> hCameraStatusAndIds;
//Convert cameraStatusAndIds to HIDL and call callback
convertToHidl(cameraStatusAndIds, &hCameraStatusAndIds);
diff --git a/services/camera/libcameraservice/tests/Android.mk b/services/camera/libcameraservice/tests/Android.mk
index b4e7c32..ec5e876 100644
--- a/services/camera/libcameraservice/tests/Android.mk
+++ b/services/camera/libcameraservice/tests/Android.mk
@@ -23,7 +23,6 @@
libcameraservice \
libhidlbase \
liblog \
- libhidltransport \
libcamera_client \
libcamera_metadata \
libutils \
diff --git a/services/camera/libcameraservice/utils/CameraThreadState.cpp b/services/camera/libcameraservice/utils/CameraThreadState.cpp
index b9e344b..2352b80 100644
--- a/services/camera/libcameraservice/utils/CameraThreadState.cpp
+++ b/services/camera/libcameraservice/utils/CameraThreadState.cpp
@@ -17,33 +17,34 @@
#include "CameraThreadState.h"
#include <binder/IPCThreadState.h>
#include <hwbinder/IPCThreadState.h>
+#include <binderthreadstate/CallerUtils.h>
#include <unistd.h>
namespace android {
int CameraThreadState::getCallingUid() {
- if (hardware::IPCThreadState::self()->isServingCall()) {
+ if (getCurrentServingCall() == BinderCallType::HWBINDER) {
return hardware::IPCThreadState::self()->getCallingUid();
}
return IPCThreadState::self()->getCallingUid();
}
int CameraThreadState::getCallingPid() {
- if (hardware::IPCThreadState::self()->isServingCall()) {
+ if (getCurrentServingCall() == BinderCallType::HWBINDER) {
return hardware::IPCThreadState::self()->getCallingPid();
}
return IPCThreadState::self()->getCallingPid();
}
int64_t CameraThreadState::clearCallingIdentity() {
- if (hardware::IPCThreadState::self()->isServingCall()) {
+ if (getCurrentServingCall() == BinderCallType::HWBINDER) {
return hardware::IPCThreadState::self()->clearCallingIdentity();
}
return IPCThreadState::self()->clearCallingIdentity();
}
void CameraThreadState::restoreCallingIdentity(int64_t token) {
- if (hardware::IPCThreadState::self()->isServingCall()) {
+ if (getCurrentServingCall() == BinderCallType::HWBINDER) {
hardware::IPCThreadState::self()->restoreCallingIdentity(token);
} else {
IPCThreadState::self()->restoreCallingIdentity(token);
diff --git a/services/camera/libcameraservice/utils/ClientManager.h b/services/camera/libcameraservice/utils/ClientManager.h
index ec6f01c..35d25bf 100644
--- a/services/camera/libcameraservice/utils/ClientManager.h
+++ b/services/camera/libcameraservice/utils/ClientManager.h
@@ -35,7 +35,7 @@
public:
/**
* Choosing to set mIsVendorClient through a parameter instead of calling
- * hardware::IPCThreadState::self()->isServingCall() to protect against the
+ * getCurrentServingCall() == BinderCallType::HWBINDER to protect against the
* case where the construction is offloaded to another thread which isn't a
* hwbinder thread.
*/
@@ -237,7 +237,7 @@
// We don't use the usual copy constructor here since we want to remember
// whether a client is a vendor client or not. This could have been wiped
// off in the incoming priority argument since an AIDL thread might have
- // called hardware::IPCThreadState::self()->isServingCall() after refreshing
+ // called getCurrentServingCall() == BinderCallType::HWBINDER after refreshing
// priorities for old clients through ProcessInfoService::getProcessStatesScoresFromPids().
mPriority.setScore(priority.getScore());
mPriority.setState(priority.getState());
diff --git a/services/mediaanalytics/Android.bp b/services/mediaanalytics/Android.bp
index 72f4b52..c27aced 100644
--- a/services/mediaanalytics/Android.bp
+++ b/services/mediaanalytics/Android.bp
@@ -50,7 +50,7 @@
"frameworks/av/media/libstagefright/rtsp",
"frameworks/av/media/libstagefright/webm",
"frameworks/av/include/media",
- "frameworks/av/include/camera",
+ "frameworks/av/camera/include/camera",
"frameworks/native/include/media/openmax",
"frameworks/native/include/media/hardware",
"external/tremolo/Tremolo",
diff --git a/services/mediaanalytics/statsd_audiopolicy.cpp b/services/mediaanalytics/statsd_audiopolicy.cpp
index 06c4dde..95cb274 100644
--- a/services/mediaanalytics/statsd_audiopolicy.cpp
+++ b/services/mediaanalytics/statsd_audiopolicy.cpp
@@ -60,14 +60,14 @@
metrics_proto.set_status(status);
}
//string char kAudioPolicyRqstSrc[] = "android.media.audiopolicy.rqst.src";
- char *rqst_src = NULL;
- if (item->getCString("android.media.audiopolicy.rqst.src", &rqst_src)) {
- metrics_proto.set_request_source(rqst_src);
+ std::string rqst_src;
+ if (item->getString("android.media.audiopolicy.rqst.src", &rqst_src)) {
+ metrics_proto.set_request_source(std::move(rqst_src));
}
//string char kAudioPolicyRqstPkg[] = "android.media.audiopolicy.rqst.pkg";
- char *rqst_pkg = NULL;
- if (item->getCString("android.media.audiopolicy.rqst.pkg", &rqst_pkg)) {
- metrics_proto.set_request_package(rqst_pkg);
+ std::string rqst_pkg;
+ if (item->getString("android.media.audiopolicy.rqst.pkg", &rqst_pkg)) {
+ metrics_proto.set_request_package(std::move(rqst_pkg));
}
//int32 char kAudioPolicyRqstSession[] = "android.media.audiopolicy.rqst.session";
int32_t rqst_session = -1;
@@ -75,20 +75,20 @@
metrics_proto.set_request_session(rqst_session);
}
//string char kAudioPolicyRqstDevice[] = "android.media.audiopolicy.rqst.device";
- char *rqst_device = NULL;
- if (item->getCString("android.media.audiopolicy.rqst.device", &rqst_device)) {
- metrics_proto.set_request_device(rqst_device);
+ std::string rqst_device;
+ if (item->getString("android.media.audiopolicy.rqst.device", &rqst_device)) {
+ metrics_proto.set_request_device(std::move(rqst_device));
}
//string char kAudioPolicyActiveSrc[] = "android.media.audiopolicy.active.src";
- char *active_src = NULL;
- if (item->getCString("android.media.audiopolicy.active.src", &active_src)) {
- metrics_proto.set_active_source(active_src);
+ std::string active_src;
+ if (item->getString("android.media.audiopolicy.active.src", &active_src)) {
+ metrics_proto.set_active_source(std::move(active_src));
}
//string char kAudioPolicyActivePkg[] = "android.media.audiopolicy.active.pkg";
- char *active_pkg = NULL;
- if (item->getCString("android.media.audiopolicy.active.pkg", &active_pkg)) {
- metrics_proto.set_active_package(active_pkg);
+ std::string active_pkg;
+ if (item->getString("android.media.audiopolicy.active.pkg", &active_pkg)) {
+ metrics_proto.set_active_package(std::move(active_pkg));
}
//int32 char kAudioPolicyActiveSession[] = "android.media.audiopolicy.active.session";
int32_t active_session = -1;
@@ -96,9 +96,9 @@
metrics_proto.set_active_session(active_session);
}
//string char kAudioPolicyActiveDevice[] = "android.media.audiopolicy.active.device";
- char *active_device = NULL;
- if (item->getCString("android.media.audiopolicy.active.device", &active_device)) {
- metrics_proto.set_active_device(active_device);
+ std::string active_device;
+ if (item->getString("android.media.audiopolicy.active.device", &active_device)) {
+ metrics_proto.set_active_device(std::move(active_device));
}
@@ -119,14 +119,6 @@
ALOGV("NOT sending: private data (len=%zu)", strlen(serialized.c_str()));
}
- // must free the strings that we were given
- free(rqst_src);
- free(rqst_pkg);
- free(rqst_device);
- free(active_src);
- free(active_pkg);
- free(active_device);
-
return true;
}
diff --git a/services/mediaanalytics/statsd_audiorecord.cpp b/services/mediaanalytics/statsd_audiorecord.cpp
index c9edb27..7c7a62c 100644
--- a/services/mediaanalytics/statsd_audiorecord.cpp
+++ b/services/mediaanalytics/statsd_audiorecord.cpp
@@ -54,14 +54,14 @@
// flesh out the protobuf we'll hand off with our data
//
- char *encoding = NULL;
- if (item->getCString("android.media.audiorecord.encoding", &encoding)) {
- metrics_proto.set_encoding(encoding);
+ std::string encoding;
+ if (item->getString("android.media.audiorecord.encoding", &encoding)) {
+ metrics_proto.set_encoding(std::move(encoding));
}
- char *source = NULL;
- if (item->getCString("android.media.audiorecord.source", &source)) {
- metrics_proto.set_source(source);
+ std::string source;
+ if (item->getString("android.media.audiorecord.source", &source)) {
+ metrics_proto.set_source(std::move(source));
}
int32_t latency = -1;
@@ -101,11 +101,11 @@
metrics_proto.set_error_code(errcode);
}
- char *errfunc = NULL;
- if (item->getCString("android.media.audiorecord.errfunc", &errfunc)) {
- metrics_proto.set_error_function(errfunc);
- } else if (item->getCString("android.media.audiorecord.lastError.at", &errfunc)) {
- metrics_proto.set_error_function(errfunc);
+ std::string errfunc;
+ if (item->getString("android.media.audiorecord.errfunc", &errfunc)) {
+ metrics_proto.set_error_function(std::move(errfunc));
+ } else if (item->getString("android.media.audiorecord.lastError.at", &errfunc)) {
+ metrics_proto.set_error_function(std::move(errfunc));
}
// portId (int32)
@@ -119,9 +119,9 @@
metrics_proto.set_frame_count(frameCount);
}
// attributes (string)
- char *attributes = NULL;
- if (item->getCString("android.media.audiorecord.attributes", &attributes)) {
- metrics_proto.set_attributes(attributes);
+ std::string attributes;
+ if (item->getString("android.media.audiorecord.attributes", &attributes)) {
+ metrics_proto.set_attributes(std::move(attributes));
}
// channelMask (int64)
int64_t channelMask = -1;
@@ -152,12 +152,6 @@
ALOGV("NOT sending: private data (len=%zu)", strlen(serialized.c_str()));
}
- // must free the strings that we were given
- free(encoding);
- free(source);
- free(errfunc);
- free(attributes);
-
return true;
}
diff --git a/services/mediaanalytics/statsd_audiothread.cpp b/services/mediaanalytics/statsd_audiothread.cpp
index 8232424..e9d6b17 100644
--- a/services/mediaanalytics/statsd_audiothread.cpp
+++ b/services/mediaanalytics/statsd_audiothread.cpp
@@ -56,9 +56,9 @@
// flesh out the protobuf we'll hand off with our data
//
- char *mytype = NULL;
- if (item->getCString(MM_PREFIX "type", &mytype)) {
- metrics_proto.set_type(mytype);
+ std::string mytype;
+ if (item->getString(MM_PREFIX "type", &mytype)) {
+ metrics_proto.set_type(std::move(mytype));
}
int32_t framecount = -1;
if (item->getInt32(MM_PREFIX "framecount", &framecount)) {
@@ -68,17 +68,17 @@
if (item->getInt32(MM_PREFIX "samplerate", &samplerate)) {
metrics_proto.set_samplerate(samplerate);
}
- char *workhist = NULL;
- if (item->getCString(MM_PREFIX "workMs.hist", &workhist)) {
- metrics_proto.set_work_millis_hist(workhist);
+ std::string workhist;
+ if (item->getString(MM_PREFIX "workMs.hist", &workhist)) {
+ metrics_proto.set_work_millis_hist(std::move(workhist));
}
- char *latencyhist = NULL;
- if (item->getCString(MM_PREFIX "latencyMs.hist", &latencyhist)) {
- metrics_proto.set_latency_millis_hist(latencyhist);
+ std::string latencyhist;
+ if (item->getString(MM_PREFIX "latencyMs.hist", &latencyhist)) {
+ metrics_proto.set_latency_millis_hist(std::move(latencyhist));
}
- char *warmuphist = NULL;
- if (item->getCString(MM_PREFIX "warmupMs.hist", &warmuphist)) {
- metrics_proto.set_warmup_millis_hist(warmuphist);
+ std::string warmuphist;
+ if (item->getString(MM_PREFIX "warmupMs.hist", &warmuphist)) {
+ metrics_proto.set_warmup_millis_hist(std::move(warmuphist));
}
int64_t underruns = -1;
if (item->getInt64(MM_PREFIX "underruns", &underruns)) {
@@ -108,9 +108,9 @@
metrics_proto.set_port_id(port_id);
}
// item->setCString(MM_PREFIX "type", threadTypeToString(mType));
- char *type = NULL;
- if (item->getCString(MM_PREFIX "type", &type)) {
- metrics_proto.set_type(type);
+ std::string type;
+ if (item->getString(MM_PREFIX "type", &type)) {
+ metrics_proto.set_type(std::move(type));
}
// item->setInt32(MM_PREFIX "sampleRate", (int32_t)mSampleRate);
int32_t sample_rate = -1;
@@ -123,9 +123,9 @@
metrics_proto.set_channel_mask(channel_mask);
}
// item->setCString(MM_PREFIX "encoding", toString(mFormat).c_str());
- char *encoding = NULL;
- if (item->getCString(MM_PREFIX "encoding", &encoding)) {
- metrics_proto.set_encoding(encoding);
+ std::string encoding;
+ if (item->getString(MM_PREFIX "encoding", &encoding)) {
+ metrics_proto.set_encoding(std::move(encoding));
}
// item->setInt32(MM_PREFIX "frameCount", (int32_t)mFrameCount);
int32_t frame_count = -1;
@@ -133,14 +133,14 @@
metrics_proto.set_frame_count(frame_count);
}
// item->setCString(MM_PREFIX "outDevice", toString(mOutDevice).c_str());
- char *outDevice = NULL;
- if (item->getCString(MM_PREFIX "outDevice", &outDevice)) {
- metrics_proto.set_output_device(outDevice);
+ std::string outDevice;
+ if (item->getString(MM_PREFIX "outDevice", &outDevice)) {
+ metrics_proto.set_output_device(std::move(outDevice));
}
// item->setCString(MM_PREFIX "inDevice", toString(mInDevice).c_str());
- char *inDevice = NULL;
- if (item->getCString(MM_PREFIX "inDevice", &inDevice)) {
- metrics_proto.set_input_device(inDevice);
+ std::string inDevice;
+ if (item->getString(MM_PREFIX "inDevice", &inDevice)) {
+ metrics_proto.set_input_device(std::move(inDevice));
}
// item->setDouble(MM_PREFIX "ioJitterMs.mean", mIoJitterMs.getMean());
double iojitters_ms_mean = -1;
@@ -201,16 +201,6 @@
ALOGV("NOT sending: private data (len=%zu)", strlen(serialized.c_str()));
}
- // must free the strings that we were given
- free(mytype);
- free(workhist);
- free(latencyhist);
- free(warmuphist);
- free(type);
- free(encoding);
- free(inDevice);
- free(outDevice);
-
return true;
}
diff --git a/services/mediaanalytics/statsd_audiotrack.cpp b/services/mediaanalytics/statsd_audiotrack.cpp
index f250ced..57cda99 100644
--- a/services/mediaanalytics/statsd_audiotrack.cpp
+++ b/services/mediaanalytics/statsd_audiotrack.cpp
@@ -57,23 +57,23 @@
// static constexpr char kAudioTrackStreamType[] = "android.media.audiotrack.streamtype";
// optional string streamType;
- char *streamtype = NULL;
- if (item->getCString("android.media.audiotrack.streamtype", &streamtype)) {
- metrics_proto.set_stream_type(streamtype);
+ std::string streamtype;
+ if (item->getString("android.media.audiotrack.streamtype", &streamtype)) {
+ metrics_proto.set_stream_type(std::move(streamtype));
}
// static constexpr char kAudioTrackContentType[] = "android.media.audiotrack.type";
// optional string contentType;
- char *contenttype = NULL;
- if (item->getCString("android.media.audiotrack.type", &contenttype)) {
- metrics_proto.set_content_type(contenttype);
+ std::string contenttype;
+ if (item->getString("android.media.audiotrack.type", &contenttype)) {
+ metrics_proto.set_content_type(std::move(contenttype));
}
// static constexpr char kAudioTrackUsage[] = "android.media.audiotrack.usage";
// optional string trackUsage;
- char *trackusage = NULL;
- if (item->getCString("android.media.audiotrack.usage", &trackusage)) {
- metrics_proto.set_track_usage(trackusage);
+ std::string trackusage;
+ if (item->getString("android.media.audiotrack.usage", &trackusage)) {
+ metrics_proto.set_track_usage(std::move(trackusage));
}
// static constexpr char kAudioTrackSampleRate[] = "android.media.audiotrack.samplerate";
@@ -111,9 +111,9 @@
metrics_proto.set_port_id(port_id);
}
// encoding (string)
- char *encoding = NULL;
- if (item->getCString("android.media.audiotrack.encoding", &encoding)) {
- metrics_proto.set_encoding(encoding);
+ std::string encoding;
+ if (item->getString("android.media.audiotrack.encoding", &encoding)) {
+ metrics_proto.set_encoding(std::move(encoding));
}
// frameCount (int32)
int32_t frame_count = -1;
@@ -121,9 +121,9 @@
metrics_proto.set_frame_count(frame_count);
}
// attributes (string)
- char *attributes = NULL;
- if (item->getCString("android.media.audiotrack.attributes", &attributes)) {
- metrics_proto.set_attributes(attributes);
+ std::string attributes;
+ if (item->getString("android.media.audiotrack.attributes", &attributes)) {
+ metrics_proto.set_attributes(std::move(attributes));
}
std::string serialized;
@@ -143,13 +143,6 @@
ALOGV("NOT sending: private data (len=%zu)", strlen(serialized.c_str()));
}
- // must free the strings that we were given
- free(streamtype);
- free(contenttype);
- free(trackusage);
- free(encoding);
- free(attributes);
-
return true;
}
diff --git a/services/mediaanalytics/statsd_codec.cpp b/services/mediaanalytics/statsd_codec.cpp
index dc8e4ef..bf82e50 100644
--- a/services/mediaanalytics/statsd_codec.cpp
+++ b/services/mediaanalytics/statsd_codec.cpp
@@ -55,19 +55,19 @@
// flesh out the protobuf we'll hand off with our data
//
// android.media.mediacodec.codec string
- char *codec = NULL;
- if (item->getCString("android.media.mediacodec.codec", &codec)) {
- metrics_proto.set_codec(codec);
+ std::string codec;
+ if (item->getString("android.media.mediacodec.codec", &codec)) {
+ metrics_proto.set_codec(std::move(codec));
}
// android.media.mediacodec.mime string
- char *mime = NULL;
- if (item->getCString("android.media.mediacodec.mime", &mime)) {
- metrics_proto.set_mime(mime);
+ std::string mime;
+ if (item->getString("android.media.mediacodec.mime", &mime)) {
+ metrics_proto.set_mime(std::move(mime));
}
// android.media.mediacodec.mode string
- char *mode = NULL;
- if ( item->getCString("android.media.mediacodec.mode", &mode)) {
- metrics_proto.set_mode(mode);
+ std::string mode;
+ if ( item->getString("android.media.mediacodec.mode", &mode)) {
+ metrics_proto.set_mode(std::move(mode));
}
// android.media.mediacodec.encoder int32
int32_t encoder = -1;
@@ -125,9 +125,9 @@
metrics_proto.set_error_code(errcode);
}
// android.media.mediacodec.errstate string
- char *errstate = NULL;
- if ( item->getCString("android.media.mediacodec.errstate", &errstate)) {
- metrics_proto.set_error_state(errstate);
+ std::string errstate;
+ if ( item->getString("android.media.mediacodec.errstate", &errstate)) {
+ metrics_proto.set_error_state(std::move(errstate));
}
// android.media.mediacodec.latency.max int64
int64_t latency_max = -1;
@@ -173,12 +173,6 @@
ALOGV("NOT sending: private data (len=%zu)", strlen(serialized.c_str()));
}
- // must free the strings that we were given
- free(codec);
- free(mime);
- free(mode);
- free(errstate);
-
return true;
}
diff --git a/services/mediaanalytics/statsd_extractor.cpp b/services/mediaanalytics/statsd_extractor.cpp
index 395c912..d84930c 100644
--- a/services/mediaanalytics/statsd_extractor.cpp
+++ b/services/mediaanalytics/statsd_extractor.cpp
@@ -56,14 +56,14 @@
//
// android.media.mediaextractor.fmt string
- char *fmt = NULL;
- if (item->getCString("android.media.mediaextractor.fmt", &fmt)) {
- metrics_proto.set_format(fmt);
+ std::string fmt;
+ if (item->getString("android.media.mediaextractor.fmt", &fmt)) {
+ metrics_proto.set_format(std::move(fmt));
}
// android.media.mediaextractor.mime string
- char *mime = NULL;
- if (item->getCString("android.media.mediaextractor.mime", &mime)) {
- metrics_proto.set_mime(mime);
+ std::string mime;
+ if (item->getString("android.media.mediaextractor.mime", &mime)) {
+ metrics_proto.set_mime(std::move(mime));
}
// android.media.mediaextractor.ntrk int32
int32_t ntrk = -1;
@@ -88,10 +88,6 @@
ALOGV("NOT sending: private data (len=%zu)", strlen(serialized.c_str()));
}
- // must free the strings that we were given
- free(fmt);
- free(mime);
-
return true;
}
diff --git a/services/mediaanalytics/statsd_nuplayer.cpp b/services/mediaanalytics/statsd_nuplayer.cpp
index 5ec118a..e6e0f2c 100644
--- a/services/mediaanalytics/statsd_nuplayer.cpp
+++ b/services/mediaanalytics/statsd_nuplayer.cpp
@@ -62,13 +62,13 @@
// differentiate between nuplayer and nuplayer2
metrics_proto.set_whichplayer(item->getKey().c_str());
- char *video_mime = NULL;
- if (item->getCString("android.media.mediaplayer.video.mime", &video_mime)) {
- metrics_proto.set_video_mime(video_mime);
+ std::string video_mime;
+ if (item->getString("android.media.mediaplayer.video.mime", &video_mime)) {
+ metrics_proto.set_video_mime(std::move(video_mime));
}
- char *video_codec = NULL;
- if (item->getCString("android.media.mediaplayer.video.codec", &video_codec)) {
- metrics_proto.set_video_codec(video_codec);
+ std::string video_codec;
+ if (item->getString("android.media.mediaplayer.video.codec", &video_codec)) {
+ metrics_proto.set_video_codec(std::move(video_codec));
}
int32_t width = -1;
@@ -97,13 +97,13 @@
metrics_proto.set_framerate(fps);
}
- char *audio_mime = NULL;
- if (item->getCString("android.media.mediaplayer.audio.mime", &audio_mime)) {
- metrics_proto.set_audio_mime(audio_mime);
+ std::string audio_mime;
+ if (item->getString("android.media.mediaplayer.audio.mime", &audio_mime)) {
+ metrics_proto.set_audio_mime(std::move(audio_mime));
}
- char *audio_codec = NULL;
- if (item->getCString("android.media.mediaplayer.audio.codec", &audio_codec)) {
- metrics_proto.set_audio_codec(audio_codec);
+ std::string audio_codec;
+ if (item->getString("android.media.mediaplayer.audio.codec", &audio_codec)) {
+ metrics_proto.set_audio_codec(std::move(audio_codec));
}
int64_t duration_ms = -1;
@@ -123,14 +123,14 @@
if (item->getInt32("android.media.mediaplayer.errcode", &error_code)) {
metrics_proto.set_error_code(error_code);
}
- char *error_state = NULL;
- if (item->getCString("android.media.mediaplayer.errstate", &error_state)) {
- metrics_proto.set_error_state(error_state);
+ std::string error_state;
+ if (item->getString("android.media.mediaplayer.errstate", &error_state)) {
+ metrics_proto.set_error_state(std::move(error_state));
}
- char *data_source_type = NULL;
- if (item->getCString("android.media.mediaplayer.dataSource", &data_source_type)) {
- metrics_proto.set_data_source_type(data_source_type);
+ std::string data_source_type;
+ if (item->getString("android.media.mediaplayer.dataSource", &data_source_type)) {
+ metrics_proto.set_data_source_type(std::move(data_source_type));
}
int64_t rebufferingMs = -1;
@@ -164,14 +164,6 @@
ALOGV("NOT sending: private data (len=%zu)", strlen(serialized.c_str()));
}
- // must free the strings that we were given
- free(video_mime);
- free(video_codec);
- free(audio_mime);
- free(audio_codec);
- free(error_state);
- free(data_source_type);
-
return true;
}
diff --git a/services/mediaanalytics/statsd_recorder.cpp b/services/mediaanalytics/statsd_recorder.cpp
index 4d981b4..d286f00 100644
--- a/services/mediaanalytics/statsd_recorder.cpp
+++ b/services/mediaanalytics/statsd_recorder.cpp
@@ -56,14 +56,14 @@
//
// string kRecorderAudioMime = "android.media.mediarecorder.audio.mime";
- char *audio_mime = NULL;
- if (item->getCString("android.media.mediarecorder.audio.mime", &audio_mime)) {
- metrics_proto.set_audio_mime(audio_mime);
+ std::string audio_mime;
+ if (item->getString("android.media.mediarecorder.audio.mime", &audio_mime)) {
+ metrics_proto.set_audio_mime(std::move(audio_mime));
}
// string kRecorderVideoMime = "android.media.mediarecorder.video.mime";
- char *video_mime = NULL;
- if (item->getCString("android.media.mediarecorder.video.mime", &video_mime)) {
- metrics_proto.set_video_mime(video_mime);
+ std::string video_mime;
+ if (item->getString("android.media.mediarecorder.video.mime", &video_mime)) {
+ metrics_proto.set_video_mime(std::move(video_mime));
}
// int32 kRecorderVideoProfile = "android.media.mediarecorder.video-encoder-profile";
int32_t videoProfile = -1;
@@ -183,10 +183,6 @@
ALOGV("NOT sending: private data (len=%zu)", strlen(serialized.c_str()));
}
- // must free the strings that we were given
- free(audio_mime);
- free(video_mime);
-
return true;
}
diff --git a/services/mediacodec/Android.bp b/services/mediacodec/Android.bp
index 2f3cad9..5811068 100644
--- a/services/mediacodec/Android.bp
+++ b/services/mediacodec/Android.bp
@@ -10,17 +10,11 @@
"libavservices_minijail",
"libbase",
"libhidlbase",
- "libhidltransport",
- "libhwbinder",
"liblog",
"libmedia_codecserviceregistrant",
],
target: {
- vendor: {
- exclude_shared_libs: ["libavservices_minijail"],
- shared_libs: ["libavservices_minijail_vendor"],
- },
android: {
product_variables: {
malloc_not_svelte: {
diff --git a/services/mediacodec/Android.mk b/services/mediacodec/Android.mk
index 15bc503..88a79e7 100644
--- a/services/mediacodec/Android.mk
+++ b/services/mediacodec/Android.mk
@@ -37,11 +37,9 @@
libutils \
liblog \
libbase \
- libavservices_minijail_vendor \
+ libavservices_minijail \
libcutils \
- libhwbinder \
libhidlbase \
- libhidltransport \
libstagefright_omx \
libstagefright_xmlparser \
android.hardware.media.omx@1.0 \
diff --git a/services/mediacodec/registrant/Android.bp b/services/mediacodec/registrant/Android.bp
index 765ac99..fa5bc4a 100644
--- a/services/mediacodec/registrant/Android.bp
+++ b/services/mediacodec/registrant/Android.bp
@@ -14,7 +14,6 @@
"libbase",
"libcodec2_hidl@1.0",
"libcodec2_vndk",
- "libhidlbase",
"libutils",
],
diff --git a/services/mediacodec/registrant/CodecServiceRegistrant.cpp b/services/mediacodec/registrant/CodecServiceRegistrant.cpp
index 7f7ceca..706ebee 100644
--- a/services/mediacodec/registrant/CodecServiceRegistrant.cpp
+++ b/services/mediacodec/registrant/CodecServiceRegistrant.cpp
@@ -18,411 +18,21 @@
#define LOG_TAG "CodecServiceRegistrant"
#include <android-base/logging.h>
-#include <android-base/properties.h>
-#include <C2Component.h>
#include <C2PlatformSupport.h>
#include <codec2/hidl/1.0/ComponentStore.h>
-#include <codec2/hidl/1.0/Configurable.h>
-#include <codec2/hidl/1.0/types.h>
-#include <hidl/HidlSupport.h>
#include <media/CodecServiceRegistrant.h>
-namespace /* unnamed */ {
-
-using ::android::hardware::hidl_vec;
-using ::android::hardware::hidl_string;
-using ::android::hardware::Return;
-using ::android::hardware::Void;
-using ::android::sp;
-using namespace ::android::hardware::media::c2::V1_0;
-using namespace ::android::hardware::media::c2::V1_0::utils;
-
-constexpr c2_status_t C2_TRANSACTION_FAILED = C2_CORRUPTED;
-
-// Converter from IComponentStore to C2ComponentStore.
-class H2C2ComponentStore : public C2ComponentStore {
-protected:
- sp<IComponentStore> mStore;
- sp<IConfigurable> mConfigurable;
-public:
- explicit H2C2ComponentStore(sp<IComponentStore> const& store)
- : mStore{store},
- mConfigurable{[store]() -> sp<IConfigurable>{
- if (!store) {
- return nullptr;
- }
- Return<sp<IConfigurable>> transResult =
- store->getConfigurable();
- return transResult.isOk() ?
- static_cast<sp<IConfigurable>>(transResult) :
- nullptr;
- }()} {
- if (!mConfigurable) {
- LOG(ERROR) << "Preferred store is corrupted.";
- }
- }
-
- virtual ~H2C2ComponentStore() override = default;
-
- virtual c2_status_t config_sm(
- std::vector<C2Param*> const ¶ms,
- std::vector<std::unique_ptr<C2SettingResult>>* const failures
- ) override {
- Params hidlParams;
- if (!createParamsBlob(&hidlParams, params)) {
- LOG(ERROR) << "config -- bad input.";
- return C2_TRANSACTION_FAILED;
- }
- c2_status_t status{};
- Return<void> transResult = mConfigurable->config(
- hidlParams,
- true,
- [&status, ¶ms, failures](
- Status s,
- const hidl_vec<SettingResult> f,
- const Params& o) {
- status = static_cast<c2_status_t>(s);
- if (status != C2_OK && status != C2_BAD_INDEX) {
- LOG(DEBUG) << "config -- call failed: "
- << status << ".";
- }
- size_t i = failures->size();
- failures->resize(i + f.size());
- for (const SettingResult& sf : f) {
- if (!objcpy(&(*failures)[i++], sf)) {
- LOG(ERROR) << "config -- "
- << "invalid SettingResult returned.";
- return;
- }
- }
- if (!updateParamsFromBlob(params, o)) {
- LOG(ERROR) << "config -- "
- << "failed to parse returned params.";
- status = C2_CORRUPTED;
- }
- });
- if (!transResult.isOk()) {
- LOG(ERROR) << "config -- transaction failed.";
- return C2_TRANSACTION_FAILED;
- }
- return status;
- };
-
- virtual c2_status_t copyBuffer(
- std::shared_ptr<C2GraphicBuffer>,
- std::shared_ptr<C2GraphicBuffer>) override {
- LOG(ERROR) << "copyBuffer -- not supported.";
- return C2_OMITTED;
- }
-
- virtual c2_status_t createComponent(
- C2String, std::shared_ptr<C2Component> *const component) override {
- component->reset();
- LOG(ERROR) << "createComponent -- not supported.";
- return C2_OMITTED;
- }
-
- virtual c2_status_t createInterface(
- C2String, std::shared_ptr<C2ComponentInterface> *const interface) {
- interface->reset();
- LOG(ERROR) << "createInterface -- not supported.";
- return C2_OMITTED;
- }
-
- virtual c2_status_t query_sm(
- const std::vector<C2Param *> &stackParams,
- const std::vector<C2Param::Index> &heapParamIndices,
- std::vector<std::unique_ptr<C2Param>> *const heapParams) const
- override {
- hidl_vec<ParamIndex> indices(
- stackParams.size() + heapParamIndices.size());
- size_t numIndices = 0;
- for (C2Param* const& stackParam : stackParams) {
- if (!stackParam) {
- LOG(WARNING) << "query -- null stack param encountered.";
- continue;
- }
- indices[numIndices++] = static_cast<ParamIndex>(stackParam->index());
- }
- size_t numStackIndices = numIndices;
- for (const C2Param::Index& index : heapParamIndices) {
- indices[numIndices++] =
- static_cast<ParamIndex>(static_cast<uint32_t>(index));
- }
- indices.resize(numIndices);
- if (heapParams) {
- heapParams->reserve(heapParams->size() + numIndices);
- }
- c2_status_t status;
- Return<void> transResult = mConfigurable->query(
- indices,
- true,
- [&status, &numStackIndices, &stackParams, heapParams](
- Status s, const Params& p) {
- status = static_cast<c2_status_t>(s);
- if (status != C2_OK && status != C2_BAD_INDEX) {
- LOG(DEBUG) << "query -- call failed: "
- << status << ".";
- return;
- }
- std::vector<C2Param*> paramPointers;
- if (!parseParamsBlob(¶mPointers, p)) {
- LOG(ERROR) << "query -- error while parsing params.";
- status = C2_CORRUPTED;
- return;
- }
- size_t i = 0;
- for (auto it = paramPointers.begin();
- it != paramPointers.end(); ) {
- C2Param* paramPointer = *it;
- if (numStackIndices > 0) {
- --numStackIndices;
- if (!paramPointer) {
- LOG(WARNING) << "query -- null stack param.";
- ++it;
- continue;
- }
- for (; i < stackParams.size() && !stackParams[i]; ) {
- ++i;
- }
- if (i >= stackParams.size()) {
- LOG(ERROR) << "query -- unexpected error.";
- status = C2_CORRUPTED;
- return;
- }
- if (stackParams[i]->index() != paramPointer->index()) {
- LOG(WARNING) << "query -- param skipped: "
- "index = "
- << stackParams[i]->index() << ".";
- stackParams[i++]->invalidate();
- continue;
- }
- if (!stackParams[i++]->updateFrom(*paramPointer)) {
- LOG(WARNING) << "query -- param update failed: "
- "index = "
- << paramPointer->index() << ".";
- }
- } else {
- if (!paramPointer) {
- LOG(WARNING) << "query -- null heap param.";
- ++it;
- continue;
- }
- if (!heapParams) {
- LOG(WARNING) << "query -- "
- "unexpected extra stack param.";
- } else {
- heapParams->emplace_back(
- C2Param::Copy(*paramPointer));
- }
- }
- ++it;
- }
- });
- if (!transResult.isOk()) {
- LOG(ERROR) << "query -- transaction failed.";
- return C2_TRANSACTION_FAILED;
- }
- return status;
- }
-
- virtual c2_status_t querySupportedParams_nb(
- std::vector<std::shared_ptr<C2ParamDescriptor>> *const params) const {
- c2_status_t status;
- Return<void> transResult = mConfigurable->querySupportedParams(
- std::numeric_limits<uint32_t>::min(),
- std::numeric_limits<uint32_t>::max(),
- [&status, params](
- Status s,
- const hidl_vec<ParamDescriptor>& p) {
- status = static_cast<c2_status_t>(s);
- if (status != C2_OK) {
- LOG(DEBUG) << "querySupportedParams -- call failed: "
- << status << ".";
- return;
- }
- size_t i = params->size();
- params->resize(i + p.size());
- for (const ParamDescriptor& sp : p) {
- if (!objcpy(&(*params)[i++], sp)) {
- LOG(ERROR) << "querySupportedParams -- "
- << "invalid returned ParamDescriptor.";
- return;
- }
- }
- });
- if (!transResult.isOk()) {
- LOG(ERROR) << "querySupportedParams -- transaction failed.";
- return C2_TRANSACTION_FAILED;
- }
- return status;
- }
-
- virtual c2_status_t querySupportedValues_sm(
- std::vector<C2FieldSupportedValuesQuery> &fields) const {
- hidl_vec<FieldSupportedValuesQuery> inFields(fields.size());
- for (size_t i = 0; i < fields.size(); ++i) {
- if (!objcpy(&inFields[i], fields[i])) {
- LOG(ERROR) << "querySupportedValues -- bad input";
- return C2_TRANSACTION_FAILED;
- }
- }
-
- c2_status_t status;
- Return<void> transResult = mConfigurable->querySupportedValues(
- inFields,
- true,
- [&status, &inFields, &fields](
- Status s,
- const hidl_vec<FieldSupportedValuesQueryResult>& r) {
- status = static_cast<c2_status_t>(s);
- if (status != C2_OK) {
- LOG(DEBUG) << "querySupportedValues -- call failed: "
- << status << ".";
- return;
- }
- if (r.size() != fields.size()) {
- LOG(ERROR) << "querySupportedValues -- "
- "input and output lists "
- "have different sizes.";
- status = C2_CORRUPTED;
- return;
- }
- for (size_t i = 0; i < fields.size(); ++i) {
- if (!objcpy(&fields[i], inFields[i], r[i])) {
- LOG(ERROR) << "querySupportedValues -- "
- "invalid returned value.";
- status = C2_CORRUPTED;
- return;
- }
- }
- });
- if (!transResult.isOk()) {
- LOG(ERROR) << "querySupportedValues -- transaction failed.";
- return C2_TRANSACTION_FAILED;
- }
- return status;
- }
-
- virtual C2String getName() const {
- C2String outName;
- Return<void> transResult = mConfigurable->getName(
- [&outName](const hidl_string& name) {
- outName = name.c_str();
- });
- if (!transResult.isOk()) {
- LOG(ERROR) << "getName -- transaction failed.";
- }
- return outName;
- }
-
- virtual std::shared_ptr<C2ParamReflector> getParamReflector() const
- override {
- struct SimpleParamReflector : public C2ParamReflector {
- virtual std::unique_ptr<C2StructDescriptor> describe(
- C2Param::CoreIndex coreIndex) const {
- hidl_vec<ParamIndex> indices(1);
- indices[0] = static_cast<ParamIndex>(coreIndex.coreIndex());
- std::unique_ptr<C2StructDescriptor> descriptor;
- Return<void> transResult = mBase->getStructDescriptors(
- indices,
- [&descriptor](
- Status s,
- const hidl_vec<StructDescriptor>& sd) {
- c2_status_t status = static_cast<c2_status_t>(s);
- if (status != C2_OK) {
- LOG(DEBUG) << "SimpleParamReflector -- "
- "getStructDescriptors() failed: "
- << status << ".";
- descriptor.reset();
- return;
- }
- if (sd.size() != 1) {
- LOG(DEBUG) << "SimpleParamReflector -- "
- "getStructDescriptors() "
- "returned vector of size "
- << sd.size() << ". "
- "It should be 1.";
- descriptor.reset();
- return;
- }
- if (!objcpy(&descriptor, sd[0])) {
- LOG(DEBUG) << "SimpleParamReflector -- "
- "getStructDescriptors() returned "
- "corrupted data.";
- descriptor.reset();
- return;
- }
- });
- return descriptor;
- }
-
- explicit SimpleParamReflector(sp<IComponentStore> base)
- : mBase(base) { }
-
- sp<IComponentStore> mBase;
- };
-
- return std::make_shared<SimpleParamReflector>(mStore);
- }
-
- virtual std::vector<std::shared_ptr<const C2Component::Traits>>
- listComponents() override {
- LOG(ERROR) << "listComponents -- not supported.";
- return {};
- }
-};
-
-bool ionPropertiesDefined() {
- using namespace ::android::base;
- std::string heapMask =
- GetProperty("ro.com.android.media.swcodec.ion.heapmask", "undefined");
- std::string flags =
- GetProperty("ro.com.android.media.swcodec.ion.flags", "undefined");
- std::string align =
- GetProperty("ro.com.android.media.swcodec.ion.align", "undefined");
- if (heapMask != "undefined" ||
- flags != "undefined" ||
- align != "undefined") {
- LOG(INFO)
- << "Some system properties for mediaswcodec ION usage are set: "
- << "heapmask = " << heapMask << ", "
- << "flags = " << flags << ", "
- << "align = " << align << ". "
- << "Preferred Codec2 store is defaulted to \"software\".";
- return true;
- }
- return false;
-}
-
-} // unnamed namespace
-
extern "C" void RegisterCodecServices() {
- using ComponentStore = ::android::hardware::media::c2::V1_0::utils::
- ComponentStore;
+ using namespace ::android::hardware::media::c2::V1_0;
LOG(INFO) << "Creating software Codec2 service...";
- sp<ComponentStore> store =
- new ComponentStore(::android::GetCodec2PlatformComponentStore());
+ android::sp<IComponentStore> store =
+ new utils::ComponentStore(
+ android::GetCodec2PlatformComponentStore());
if (store == nullptr) {
LOG(ERROR) <<
"Cannot create software Codec2 service.";
} else {
- if (!ionPropertiesDefined()) {
- std::string preferredStoreName = "default";
- sp<IComponentStore> preferredStore =
- IComponentStore::getService(preferredStoreName.c_str());
- if (preferredStore) {
- ::android::SetPreferredCodec2ComponentStore(
- std::make_shared<H2C2ComponentStore>(preferredStore));
- LOG(INFO) <<
- "Preferred Codec2 store is set to \"" <<
- preferredStoreName << "\".";
- } else {
- LOG(INFO) <<
- "Preferred Codec2 store is defaulted to \"software\".";
- }
- }
if (store->registerAsService("software") != android::OK) {
LOG(ERROR) <<
"Cannot register software Codec2 service.";
diff --git a/services/mediadrm/Android.mk b/services/mediadrm/Android.mk
index 227a29d..72d42ae 100644
--- a/services/mediadrm/Android.mk
+++ b/services/mediadrm/Android.mk
@@ -20,14 +20,18 @@
MediaDrmService.cpp \
main_mediadrmserver.cpp
+LOCAL_HEADER_LIBRARIES:= \
+ libmedia_headers \
+ libmediadrm_headers
+
LOCAL_SHARED_LIBRARIES:= \
libbinder \
liblog \
+ libmedia \
libmediadrm \
libutils \
libhidlbase \
libhidlmemory \
- libhidltransport \
android.hardware.drm@1.0 \
android.hardware.drm@1.1 \
android.hardware.drm@1.2
diff --git a/services/mediaextractor/Android.bp b/services/mediaextractor/Android.bp
index 0c701d7..dac53a5 100644
--- a/services/mediaextractor/Android.bp
+++ b/services/mediaextractor/Android.bp
@@ -8,6 +8,7 @@
srcs: ["MediaExtractorService.cpp"],
shared_libs: [
+ "libdatasource",
"libmedia",
"libstagefright",
"libbinder",
@@ -28,6 +29,9 @@
"liblog",
"libavservices_minijail",
],
+ header_libs: [
+ "bionic_libc_platform_headers",
+ ],
target: {
android: {
product_variables: {
diff --git a/services/mediaextractor/MediaExtractorService.cpp b/services/mediaextractor/MediaExtractorService.cpp
index 36e084b..6239fb2 100644
--- a/services/mediaextractor/MediaExtractorService.cpp
+++ b/services/mediaextractor/MediaExtractorService.cpp
@@ -20,8 +20,8 @@
#include <utils/Vector.h>
+#include <datasource/DataSourceFactory.h>
#include <media/DataSource.h>
-#include <media/stagefright/DataSourceFactory.h>
#include <media/stagefright/InterfaceUtils.h>
#include <media/stagefright/MediaExtractorFactory.h>
#include <media/stagefright/RemoteDataSource.h>
@@ -55,7 +55,7 @@
sp<IDataSource> MediaExtractorService::makeIDataSource(int fd, int64_t offset, int64_t length)
{
- sp<DataSource> source = DataSourceFactory::CreateFromFd(fd, offset, length);
+ sp<DataSource> source = DataSourceFactory::getInstance()->CreateFromFd(fd, offset, length);
return CreateIDataSourceFromDataSource(source);
}
diff --git a/services/mediaextractor/main_extractorservice.cpp b/services/mediaextractor/main_extractorservice.cpp
index 3c4125b..afb7692 100644
--- a/services/mediaextractor/main_extractorservice.cpp
+++ b/services/mediaextractor/main_extractorservice.cpp
@@ -28,6 +28,8 @@
#include <android-base/properties.h>
#include <utils/misc.h>
+#include <bionic/reserved_signals.h>
+
// from LOCAL_C_INCLUDES
#include "MediaExtractorService.h"
#include "MediaUtils.h"
@@ -49,6 +51,10 @@
signal(SIGPIPE, SIG_IGN);
+ // Do not assist platform profilers (relevant only on debug builds).
+ // Otherwise, the signal handler can violate the seccomp policy.
+ signal(BIONIC_SIGNAL_PROFILER, SIG_IGN);
+
//b/62255959: this forces libutis.so to dlopen vendor version of libutils.so
//before minijail is on. This is dirty but required since some syscalls such
//as pread64 are used by linker but aren't allowed in the minijail. By
diff --git a/services/medialog/Android.bp b/services/medialog/Android.bp
index bee5d25..74b63d5 100644
--- a/services/medialog/Android.bp
+++ b/services/medialog/Android.bp
@@ -6,6 +6,10 @@
"MediaLogService.cpp",
],
+ header_libs: [
+ "libmedia_headers",
+ ],
+
shared_libs: [
"libaudioutils",
"libbinder",
diff --git a/services/mediaresourcemanager/Android.bp b/services/mediaresourcemanager/Android.bp
index f3339a0..d468406 100644
--- a/services/mediaresourcemanager/Android.bp
+++ b/services/mediaresourcemanager/Android.bp
@@ -23,4 +23,6 @@
"-Wall",
],
+ export_include_dirs: ["."],
+
}
diff --git a/services/mediaresourcemanager/ResourceManagerService.cpp b/services/mediaresourcemanager/ResourceManagerService.cpp
index bdcd5e4..45eea0f 100644
--- a/services/mediaresourcemanager/ResourceManagerService.cpp
+++ b/services/mediaresourcemanager/ResourceManagerService.cpp
@@ -290,6 +290,18 @@
}
}
+void ResourceManagerService::mergeResources(
+ MediaResource& r1, const MediaResource& r2) {
+ if (r1.mType == MediaResource::kDrmSession) {
+ // This means we are using a session. Each session's mValue is initialized to UINT64_MAX.
+ // The oftener a session is used the lower it's mValue. During reclaim the session with
+ // the highest mValue/lowest usage would be closed.
+ r1.mValue -= (r1.mValue == 0 ? 0 : 1);
+ } else {
+ r1.mValue += r2.mValue;
+ }
+}
+
void ResourceManagerService::addResource(
int pid,
int uid,
@@ -309,15 +321,16 @@
ResourceInfo& info = getResourceInfoForEdit(uid, clientId, client, infos);
for (size_t i = 0; i < resources.size(); ++i) {
- const auto resType = std::make_pair(resources[i].mType, resources[i].mSubType);
+ const auto &res = resources[i];
+ const auto resType = std::tuple(res.mType, res.mSubType, res.mId);
if (info.resources.find(resType) == info.resources.end()) {
- onFirstAdded(resources[i], info);
- info.resources[resType] = resources[i];
+ onFirstAdded(res, info);
+ info.resources[resType] = res;
} else {
- info.resources[resType].mValue += resources[i].mValue;
+ mergeResources(info.resources[resType], res);
}
}
- if (info.deathNotifier == nullptr) {
+ if (info.deathNotifier == nullptr && client != nullptr) {
info.deathNotifier = new DeathNotifier(this, pid, clientId);
IInterface::asBinder(client)->linkToDeath(info.deathNotifier);
}
@@ -351,14 +364,17 @@
ResourceInfo &info = infos.editValueAt(index);
for (size_t i = 0; i < resources.size(); ++i) {
- const auto resType = std::make_pair(resources[i].mType, resources[i].mSubType);
+ const auto &res = resources[i];
+ const auto resType = std::tuple(res.mType, res.mSubType, res.mId);
// ignore if we don't have it
if (info.resources.find(resType) != info.resources.end()) {
MediaResource &resource = info.resources[resType];
- if (resource.mValue > resources[i].mValue) {
- resource.mValue -= resources[i].mValue;
+ if (resource.mValue > res.mValue) {
+ resource.mValue -= res.mValue;
} else {
- onLastRemoved(resources[i], info);
+ // drm sessions always take this branch because res.mValue is set
+ // to UINT64_MAX
+ onLastRemoved(res, info);
info.resources.erase(resType);
}
}
@@ -430,6 +446,7 @@
const MediaResource *secureCodec = NULL;
const MediaResource *nonSecureCodec = NULL;
const MediaResource *graphicMemory = NULL;
+ const MediaResource *drmSession = NULL;
for (size_t i = 0; i < resources.size(); ++i) {
MediaResource::Type type = resources[i].mType;
if (resources[i].mType == MediaResource::kSecureCodec) {
@@ -438,6 +455,8 @@
nonSecureCodec = &resources[i];
} else if (type == MediaResource::kGraphicMemory) {
graphicMemory = &resources[i];
+ } else if (type == MediaResource::kDrmSession) {
+ drmSession = &resources[i];
}
}
@@ -461,6 +480,12 @@
}
}
}
+ if (drmSession != NULL) {
+ getClientForResource_l(callingPid, drmSession, &clients);
+ if (clients.size() == 0) {
+ return false;
+ }
+ }
if (clients.size() == 0) {
// if no secure/non-secure codec conflict, run second pass to handle other resources.
diff --git a/services/mediaresourcemanager/ResourceManagerService.h b/services/mediaresourcemanager/ResourceManagerService.h
index f086dc3..44d0c28 100644
--- a/services/mediaresourcemanager/ResourceManagerService.h
+++ b/services/mediaresourcemanager/ResourceManagerService.h
@@ -33,7 +33,7 @@
class ServiceLog;
struct ProcessInfoInterface;
-typedef std::map<std::pair<MediaResource::Type, MediaResource::SubType>, MediaResource> ResourceList;
+typedef std::map<std::tuple<MediaResource::Type, MediaResource::SubType, std::vector<uint8_t>>, MediaResource> ResourceList;
struct ResourceInfo {
int64_t clientId;
uid_t uid;
@@ -126,6 +126,9 @@
void onFirstAdded(const MediaResource& res, const ResourceInfo& clientInfo);
void onLastRemoved(const MediaResource& res, const ResourceInfo& clientInfo);
+ // Merge r2 into r1
+ void mergeResources(MediaResource& r1, const MediaResource& r2);
+
mutable Mutex mLock;
sp<ProcessInfoInterface> mProcessInfo;
sp<SystemCallbackInterface> mSystemCB;
diff --git a/services/mediaresourcemanager/test/ResourceManagerService_test.cpp b/services/mediaresourcemanager/test/ResourceManagerService_test.cpp
index ae97ec8..9e14151 100644
--- a/services/mediaresourcemanager/test/ResourceManagerService_test.cpp
+++ b/services/mediaresourcemanager/test/ResourceManagerService_test.cpp
@@ -173,8 +173,9 @@
// convert resource1 to ResourceList
ResourceList r1;
for (size_t i = 0; i < resources1.size(); ++i) {
- const auto resType = std::make_pair(resources1[i].mType, resources1[i].mSubType);
- r1[resType] = resources1[i];
+ const auto &res = resources1[i];
+ const auto resType = std::tuple(res.mType, res.mSubType, res.mId);
+ r1[resType] = res;
}
return r1 == resources2;
}
diff --git a/services/minijail/Android.bp b/services/minijail/Android.bp
index 07a94cc..0713a87 100644
--- a/services/minijail/Android.bp
+++ b/services/minijail/Android.bp
@@ -17,10 +17,14 @@
cc_library_shared {
name: "libavservices_minijail",
defaults: ["libavservices_minijail_defaults"],
+ vendor_available: true,
export_include_dirs: ["."],
}
-// Small library for media.extractor and media.codec sandboxing.
+// By adding "vendor_available: true" to "libavservices_minijail", we don't
+// need to have "libavservices_minijail_vendor" any longer.
+// "libavservices_minijail_vendor" will be removed, once we replace it with
+// "libavservices_minijail" in all vendor modules. (b/146313710)
cc_library_shared {
name: "libavservices_minijail_vendor",
vendor: true,
diff --git a/services/oboeservice/AAudioClientTracker.cpp b/services/oboeservice/AAudioClientTracker.cpp
index 8572561..6e14434 100644
--- a/services/oboeservice/AAudioClientTracker.cpp
+++ b/services/oboeservice/AAudioClientTracker.cpp
@@ -75,10 +75,10 @@
std::lock_guard<std::mutex> lock(mLock);
if (mNotificationClients.count(pid) == 0) {
- sp<NotificationClient> notificationClient = new NotificationClient(pid);
+ sp<IBinder> binder = IInterface::asBinder(client);
+ sp<NotificationClient> notificationClient = new NotificationClient(pid, binder);
mNotificationClients[pid] = notificationClient;
- sp<IBinder> binder = IInterface::asBinder(client);
status_t status = binder->linkToDeath(notificationClient);
ALOGW_IF(status != NO_ERROR, "registerClient() linkToDeath = %d\n", status);
return AAudioConvert_androidToAAudioResult(status);
@@ -113,7 +113,7 @@
if (notificationClient == 0) {
// This will get called the first time the audio server registers an internal stream.
ALOGV("registerClientStream(%d,) unrecognized pid\n", pid);
- notificationClient = new NotificationClient(pid);
+ notificationClient = new NotificationClient(pid, nullptr);
mNotificationClients[pid] = notificationClient;
}
notificationClient->registerClientStream(serviceStream);
@@ -136,8 +136,8 @@
return AAUDIO_OK;
}
-AAudioClientTracker::NotificationClient::NotificationClient(pid_t pid)
- : mProcessId(pid) {
+AAudioClientTracker::NotificationClient::NotificationClient(pid_t pid, const sp<IBinder>& binder)
+ : mProcessId(pid), mBinder(binder) {
}
AAudioClientTracker::NotificationClient::~NotificationClient() {
diff --git a/services/oboeservice/AAudioClientTracker.h b/services/oboeservice/AAudioClientTracker.h
index accf1a7..00ff467 100644
--- a/services/oboeservice/AAudioClientTracker.h
+++ b/services/oboeservice/AAudioClientTracker.h
@@ -73,7 +73,7 @@
*/
class NotificationClient : public IBinder::DeathRecipient {
public:
- NotificationClient(pid_t pid);
+ NotificationClient(pid_t pid, const android::sp<IBinder>& binder);
virtual ~NotificationClient();
int32_t getStreamCount();
@@ -91,6 +91,8 @@
mutable std::mutex mLock;
const pid_t mProcessId;
std::set<android::sp<AAudioServiceStreamBase>> mStreams;
+ // hold onto binder to receive death notifications
+ android::sp<IBinder> mBinder;
};
mutable std::mutex mLock;
diff --git a/services/oboeservice/Android.mk b/services/oboeservice/Android.mk
index 3d5f140..96ccebc 100644
--- a/services/oboeservice/Android.mk
+++ b/services/oboeservice/Android.mk
@@ -14,7 +14,6 @@
$(call include-path-for, audio-utils) \
frameworks/native/include \
system/core/base/include \
- $(TOP)/frameworks/native/media/libaaudio/include/include \
$(TOP)/frameworks/av/media/libaaudio/include \
$(TOP)/frameworks/av/media/utils/include \
frameworks/native/include \
@@ -40,14 +39,12 @@
TimestampScheduler.cpp \
AAudioThread.cpp
-LOCAL_MULTILIB := $(AUDIOSERVER_MULTILIB)
-
# LOCAL_CFLAGS += -fvisibility=hidden
LOCAL_CFLAGS += -Wno-unused-parameter
LOCAL_CFLAGS += -Wall -Werror
LOCAL_SHARED_LIBRARIES := \
- libaaudio \
+ libaaudio_internal \
libaudioflinger \
libaudioclient \
libbinder \
diff --git a/services/soundtrigger/Android.bp b/services/soundtrigger/Android.bp
index 3f02f48..1bbd591 100644
--- a/services/soundtrigger/Android.bp
+++ b/services/soundtrigger/Android.bp
@@ -31,10 +31,8 @@
"libaudioutils",
"libmediautils",
- "libhwbinder",
"libhidlbase",
"libhidlmemory",
- "libhidltransport",
"libbase",
"libaudiohal",
"libaudiohal_deathhandler",
diff --git a/tools/resampler_tools/Android.bp b/tools/resampler_tools/Android.bp
new file mode 100644
index 0000000..7549359
--- /dev/null
+++ b/tools/resampler_tools/Android.bp
@@ -0,0 +1,15 @@
+// Copyright 2005 The Android Open Source Project
+//
+// Android.mk for resampler_tools
+//
+
+cc_binary_host {
+ name: "fir",
+
+ srcs: ["fir.cpp"],
+
+ cflags: [
+ "-Werror",
+ "-Wall",
+ ],
+}
diff --git a/tools/resampler_tools/Android.mk b/tools/resampler_tools/Android.mk
deleted file mode 100644
index bba5199..0000000
--- a/tools/resampler_tools/Android.mk
+++ /dev/null
@@ -1,17 +0,0 @@
-# Copyright 2005 The Android Open Source Project
-#
-# Android.mk for resampler_tools
-#
-
-
-LOCAL_PATH:= $(call my-dir)
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES := \
- fir.cpp
-
-LOCAL_MODULE := fir
-
-LOCAL_CFLAGS := -Werror -Wall
-
-include $(BUILD_HOST_EXECUTABLE)