Merge "Fix dst rect calculation"
diff --git a/apex/ld.config.txt b/apex/ld.config.txt
index a5937fd..af8ec06 100644
--- a/apex/ld.config.txt
+++ b/apex/ld.config.txt
@@ -37,9 +37,11 @@
namespace.platform.isolated = true
-namespace.platform.search.paths = /system/${LIB}
+namespace.platform.search.paths = /system/${LIB}
+namespace.platform.search.paths += /apex/com.android.runtime/${LIB}
namespace.platform.asan.search.paths = /data/asan/system/${LIB}
namespace.platform.asan.search.paths += /system/${LIB}
+namespace.platform.asan.search.paths += /apex/com.android.runtime/${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)
diff --git a/camera/cameraserver/Android.bp b/camera/cameraserver/Android.bp
index 334f879..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",
diff --git a/camera/ndk/impl/ACameraDevice.cpp b/camera/ndk/impl/ACameraDevice.cpp
index d24cb81..46a8dae 100644
--- a/camera/ndk/impl/ACameraDevice.cpp
+++ b/camera/ndk/impl/ACameraDevice.cpp
@@ -29,7 +29,7 @@
#include "ACameraCaptureSession.inc"
ACameraDevice::~ACameraDevice() {
- mDevice->stopLooper();
+ mDevice->stopLooperAndDisconnect();
}
namespace android {
@@ -112,19 +112,7 @@
}
}
-// Device close implementaiton
-CameraDevice::~CameraDevice() {
- sp<ACameraCaptureSession> session = mCurrentSession.promote();
- {
- Mutex::Autolock _l(mDeviceLock);
- if (!isClosed()) {
- disconnectLocked(session);
- }
- LOG_ALWAYS_FATAL_IF(mCbLooper != nullptr,
- "CameraDevice looper should've been stopped before ~CameraDevice");
- mCurrentSession = nullptr;
- }
-}
+CameraDevice::~CameraDevice() { }
void
CameraDevice::postSessionMsgAndCleanup(sp<AMessage>& msg) {
@@ -892,8 +880,14 @@
return;
}
-void CameraDevice::stopLooper() {
+void CameraDevice::stopLooperAndDisconnect() {
Mutex::Autolock _l(mDeviceLock);
+ sp<ACameraCaptureSession> session = mCurrentSession.promote();
+ if (!isClosed()) {
+ disconnectLocked(session);
+ }
+ mCurrentSession = nullptr;
+
if (mCbLooper != nullptr) {
mCbLooper->unregisterHandler(mHandler->id());
mCbLooper->stop();
diff --git a/camera/ndk/impl/ACameraDevice.h b/camera/ndk/impl/ACameraDevice.h
index 7a35bf0..6c2ceb3 100644
--- a/camera/ndk/impl/ACameraDevice.h
+++ b/camera/ndk/impl/ACameraDevice.h
@@ -40,6 +40,7 @@
#include <camera/NdkCameraManager.h>
#include <camera/NdkCameraCaptureSession.h>
+
#include "ACameraMetadata.h"
namespace android {
@@ -110,7 +111,7 @@
inline ACameraDevice* getWrapper() const { return mWrapper; };
// Stop the looper thread and unregister the handler
- void stopLooper();
+ void stopLooperAndDisconnect();
private:
friend ACameraCaptureSession;
diff --git a/camera/ndk/ndk_vendor/impl/ACameraDevice.cpp b/camera/ndk/ndk_vendor/impl/ACameraDevice.cpp
index 35c8355..e511a3f 100644
--- a/camera/ndk/ndk_vendor/impl/ACameraDevice.cpp
+++ b/camera/ndk/ndk_vendor/impl/ACameraDevice.cpp
@@ -45,7 +45,7 @@
using namespace android;
ACameraDevice::~ACameraDevice() {
- mDevice->stopLooper();
+ mDevice->stopLooperAndDisconnect();
}
namespace android {
@@ -125,19 +125,7 @@
}
}
-// Device close implementaiton
-CameraDevice::~CameraDevice() {
- sp<ACameraCaptureSession> session = mCurrentSession.promote();
- {
- Mutex::Autolock _l(mDeviceLock);
- if (!isClosed()) {
- disconnectLocked(session);
- }
- mCurrentSession = nullptr;
- LOG_ALWAYS_FATAL_IF(mCbLooper != nullptr,
- "CameraDevice looper should've been stopped before ~CameraDevice");
- }
-}
+CameraDevice::~CameraDevice() { }
void
CameraDevice::postSessionMsgAndCleanup(sp<AMessage>& msg) {
@@ -1388,6 +1376,7 @@
// before cbh goes out of scope and causing we call the session
// destructor while holding device lock
cbh.mSession.clear();
+
postSessionMsgAndCleanup(msg);
}
@@ -1400,8 +1389,13 @@
}
}
-void CameraDevice::stopLooper() {
+void CameraDevice::stopLooperAndDisconnect() {
Mutex::Autolock _l(mDeviceLock);
+ sp<ACameraCaptureSession> session = mCurrentSession.promote();
+ if (!isClosed()) {
+ disconnectLocked(session);
+ }
+ mCurrentSession = nullptr;
if (mCbLooper != nullptr) {
mCbLooper->unregisterHandler(mHandler->id());
mCbLooper->stop();
diff --git a/camera/ndk/ndk_vendor/impl/ACameraDevice.h b/camera/ndk/ndk_vendor/impl/ACameraDevice.h
index 3328a85..7fc699e 100644
--- a/camera/ndk/ndk_vendor/impl/ACameraDevice.h
+++ b/camera/ndk/ndk_vendor/impl/ACameraDevice.h
@@ -36,6 +36,7 @@
#include <camera/NdkCameraManager.h>
#include <camera/NdkCameraCaptureSession.h>
+
#include "ACameraMetadata.h"
#include "utils.h"
@@ -134,10 +135,11 @@
inline ACameraDevice* getWrapper() const { return mWrapper; };
// Stop the looper thread and unregister the handler
- void stopLooper();
+ void stopLooperAndDisconnect();
private:
friend ACameraCaptureSession;
+ friend ACameraDevice;
camera_status_t checkCameraClosedOrErrorLocked() const;
@@ -387,7 +389,6 @@
mDevice(new android::acam::CameraDevice(id, cb, std::move(chars), this)) {}
~ACameraDevice();
-
/*******************
* NDK public APIs *
*******************/
diff --git a/camera/ndk/ndk_vendor/tests/AImageReaderVendorTest.cpp b/camera/ndk/ndk_vendor/tests/AImageReaderVendorTest.cpp
index 37de30a..7ab0124 100644
--- a/camera/ndk/ndk_vendor/tests/AImageReaderVendorTest.cpp
+++ b/camera/ndk/ndk_vendor/tests/AImageReaderVendorTest.cpp
@@ -24,6 +24,7 @@
#include <algorithm>
#include <mutex>
#include <string>
+#include <variant>
#include <vector>
#include <stdio.h>
#include <stdio.h>
@@ -49,6 +50,7 @@
static constexpr int kTestImageFormat = AIMAGE_FORMAT_YUV_420_888;
using android::hardware::camera::common::V1_0::helper::VendorTagDescriptorCache;
+using ConfiguredWindows = std::set<native_handle_t *>;
class CameraHelper {
public:
@@ -60,9 +62,12 @@
const char* physicalCameraId;
native_handle_t* anw;
};
- int initCamera(native_handle_t* imgReaderAnw,
+
+ // Retaining the error code in case the caller needs to analyze it.
+ std::variant<int, ConfiguredWindows> initCamera(native_handle_t* imgReaderAnw,
const std::vector<PhysicalImgReaderInfo>& physicalImgReaders,
bool usePhysicalSettings) {
+ ConfiguredWindows configuredWindows;
if (imgReaderAnw == nullptr) {
ALOGE("Cannot initialize camera before image reader get initialized.");
return -1;
@@ -78,7 +83,7 @@
ret = ACameraManager_openCamera(mCameraManager, mCameraId, &mDeviceCb, &mDevice);
if (ret != AMEDIA_OK || mDevice == nullptr) {
ALOGE("Failed to open camera, ret=%d, mDevice=%p.", ret, mDevice);
- return -1;
+ return ret;
}
// Create capture session
@@ -97,8 +102,9 @@
ALOGE("ACaptureSessionOutputContainer_add failed, ret=%d", ret);
return ret;
}
-
+ configuredWindows.insert(mImgReaderAnw);
std::vector<const char*> idPointerList;
+ std::set<const native_handle_t*> physicalStreamMap;
for (auto& physicalStream : physicalImgReaders) {
ACaptureSessionOutput* sessionOutput = nullptr;
ret = ACaptureSessionPhysicalOutput_create(physicalStream.anw,
@@ -112,21 +118,25 @@
ALOGE("ACaptureSessionOutputContainer_add failed, ret=%d", ret);
return ret;
}
- mExtraOutputs.push_back(sessionOutput);
+ ret = ACameraDevice_isSessionConfigurationSupported(mDevice, mOutputs);
+ if (ret != ACAMERA_OK && ret != ACAMERA_ERROR_UNSUPPORTED_OPERATION) {
+ ALOGW("ACameraDevice_isSessionConfigurationSupported failed, ret=%d camera id %s",
+ ret, mCameraId);
+ ACaptureSessionOutputContainer_remove(mOutputs, sessionOutput);
+ ACaptureSessionOutput_free(sessionOutput);
+ continue;
+ }
+ configuredWindows.insert(physicalStream.anw);
// Assume that at most one physical stream per physical camera.
mPhysicalCameraIds.push_back(physicalStream.physicalCameraId);
idPointerList.push_back(physicalStream.physicalCameraId);
+ physicalStreamMap.insert(physicalStream.anw);
+ mSessionPhysicalOutputs.push_back(sessionOutput);
}
ACameraIdList cameraIdList;
cameraIdList.numCameras = idPointerList.size();
cameraIdList.cameraIds = idPointerList.data();
- ret = ACameraDevice_isSessionConfigurationSupported(mDevice, mOutputs);
- if (ret != ACAMERA_OK && ret != ACAMERA_ERROR_UNSUPPORTED_OPERATION) {
- ALOGE("ACameraDevice_isSessionConfigurationSupported failed, ret=%d", ret);
- return ret;
- }
-
ret = ACameraDevice_createCaptureSession(mDevice, mOutputs, &mSessionCb, &mSession);
if (ret != AMEDIA_OK) {
ALOGE("ACameraDevice_createCaptureSession failed, ret=%d", ret);
@@ -157,6 +167,10 @@
}
for (auto& physicalStream : physicalImgReaders) {
+ if (physicalStreamMap.find(physicalStream.anw) == physicalStreamMap.end()) {
+ ALOGI("Skipping physicalStream anw=%p", physicalStream.anw);
+ continue;
+ }
ACameraOutputTarget* outputTarget = nullptr;
ret = ACameraOutputTarget_create(physicalStream.anw, &outputTarget);
if (ret != AMEDIA_OK) {
@@ -168,11 +182,11 @@
ALOGE("ACaptureRequest_addTarget failed, ret=%d", ret);
return ret;
}
- mReqExtraOutputs.push_back(outputTarget);
+ mReqPhysicalOutputs.push_back(outputTarget);
}
mIsCameraReady = true;
- return 0;
+ return configuredWindows;
}
@@ -184,10 +198,10 @@
ACameraOutputTarget_free(mReqImgReaderOutput);
mReqImgReaderOutput = nullptr;
}
- for (auto& outputTarget : mReqExtraOutputs) {
+ for (auto& outputTarget : mReqPhysicalOutputs) {
ACameraOutputTarget_free(outputTarget);
}
- mReqExtraOutputs.clear();
+ mReqPhysicalOutputs.clear();
if (mStillRequest) {
ACaptureRequest_free(mStillRequest);
mStillRequest = nullptr;
@@ -201,10 +215,10 @@
ACaptureSessionOutput_free(mImgReaderOutput);
mImgReaderOutput = nullptr;
}
- for (auto& extraOutput : mExtraOutputs) {
+ for (auto& extraOutput : mSessionPhysicalOutputs) {
ACaptureSessionOutput_free(extraOutput);
}
- mExtraOutputs.clear();
+ mSessionPhysicalOutputs.clear();
if (mOutputs) {
ACaptureSessionOutputContainer_free(mOutputs);
mOutputs = nullptr;
@@ -262,13 +276,13 @@
// Capture session
ACaptureSessionOutputContainer* mOutputs = nullptr;
ACaptureSessionOutput* mImgReaderOutput = nullptr;
- std::vector<ACaptureSessionOutput*> mExtraOutputs;
+ std::vector<ACaptureSessionOutput*> mSessionPhysicalOutputs;
ACameraCaptureSession* mSession = nullptr;
// Capture request
ACaptureRequest* mStillRequest = nullptr;
ACameraOutputTarget* mReqImgReaderOutput = nullptr;
- std::vector<ACameraOutputTarget*> mReqExtraOutputs;
+ std::vector<ACameraOutputTarget*> mReqPhysicalOutputs;
bool mIsCameraReady = false;
const char* mCameraId;
@@ -581,9 +595,11 @@
}
CameraHelper cameraHelper(id, mCameraManager);
- ret = cameraHelper.initCamera(testCase.getNativeWindow(),
- {}/*physicalImageReaders*/, false/*usePhysicalSettings*/);
- if (ret < 0) {
+ std::variant<int, ConfiguredWindows> retInit =
+ cameraHelper.initCamera(testCase.getNativeWindow(), {}/*physicalImageReaders*/,
+ false/*usePhysicalSettings*/);
+ int *retp = std::get_if<int>(&retInit);
+ if (retp) {
ALOGE("Unable to initialize camera helper");
return false;
}
@@ -751,10 +767,15 @@
physicalImgReaderInfo.push_back({physicalCameraIds[0], testCases[1]->getNativeWindow()});
physicalImgReaderInfo.push_back({physicalCameraIds[1], testCases[2]->getNativeWindow()});
- int ret = cameraHelper.initCamera(testCases[0]->getNativeWindow(),
- physicalImgReaderInfo, usePhysicalSettings);
- ASSERT_EQ(ret, 0);
-
+ std::variant<int, ConfiguredWindows> retInit =
+ cameraHelper.initCamera(testCases[0]->getNativeWindow(), physicalImgReaderInfo,
+ usePhysicalSettings);
+ int *retp = std::get_if<int>(&retInit);
+ ASSERT_EQ(retp, nullptr);
+ ConfiguredWindows *configuredWindowsp = std::get_if<ConfiguredWindows>(&retInit);
+ ASSERT_NE(configuredWindowsp, nullptr);
+ ASSERT_LE(configuredWindowsp->size(), testCases.size());
+ int ret = 0;
if (!cameraHelper.isCameraReady()) {
ALOGW("Camera is not ready after successful initialization. It's either due to camera "
"on board lacks BACKWARDS_COMPATIBLE capability or the device does not have "
@@ -776,9 +797,15 @@
break;
}
}
- ASSERT_EQ(testCases[0]->getAcquiredImageCount(), pictureCount);
- ASSERT_EQ(testCases[1]->getAcquiredImageCount(), pictureCount);
- ASSERT_EQ(testCases[2]->getAcquiredImageCount(), pictureCount);
+ for(auto &testCase : testCases) {
+ auto it = configuredWindowsp->find(testCase->getNativeWindow());
+ if (it == configuredWindowsp->end()) {
+ continue;
+ }
+ ALOGI("Testing window %p", testCase->getNativeWindow());
+ ASSERT_EQ(testCase->getAcquiredImageCount(), pictureCount);
+ }
+
ASSERT_TRUE(cameraHelper.checkCallbacks(pictureCount));
ACameraMetadata_free(staticMetadata);
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 df28842..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"
diff --git a/cmds/stagefright/Android.mk b/cmds/stagefright/Android.mk
index 0c8d44a..defc94f 100644
--- a/cmds/stagefright/Android.mk
+++ b/cmds/stagefright/Android.mk
@@ -3,14 +3,15 @@
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:= \
@@ -31,14 +32,16 @@
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
@@ -56,12 +59,12 @@
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 \
@@ -82,12 +85,13 @@
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 \
@@ -110,7 +114,7 @@
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 \
@@ -132,6 +136,9 @@
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
@@ -158,17 +165,18 @@
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/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..9ba6ee1 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;
@@ -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 c29d004..84f2f6d 100644
--- a/drm/libmediadrm/Android.bp
+++ b/drm/libmediadrm/Android.bp
@@ -2,6 +2,15 @@
// libmediadrm
//
+cc_library_headers {
+ name: "libmediadrm_headers",
+
+ export_include_dirs: [
+ "interface"
+ ],
+
+}
+
cc_library_shared {
name: "libmediadrm",
@@ -17,6 +26,19 @@
"CryptoHal.cpp",
],
+ local_include_dirs: [
+ "include",
+ "interface"
+ ],
+
+ export_include_dirs: [
+ "include"
+ ],
+
+ header_libs: [
+ "libmedia_headers",
+ ],
+
shared_libs: [
"libbinder",
"libcutils",
@@ -49,10 +71,17 @@
"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",
@@ -80,10 +109,17 @@
"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..e79fd4b 100644
--- a/drm/libmediadrm/DrmHal.cpp
+++ b/drm/libmediadrm/DrmHal.cpp
@@ -895,9 +895,8 @@
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/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 100%
rename from media/libmedia/include/media/DrmHal.h
rename to drm/libmediadrm/include/mediadrm/DrmHal.h
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 100%
rename from media/libmedia/include/media/DrmSessionManager.h
rename to drm/libmediadrm/include/mediadrm/DrmSessionManager.h
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/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/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 99d3c13..0000000
--- a/include/media/ExtendedAudioBufferProvider.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/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/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/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/media/audioserver/Android.mk b/media/audioserver/Android.mk
index f5f021b..6697cb5 100644
--- a/media/audioserver/Android.mk
+++ b/media/audioserver/Android.mk
@@ -9,6 +9,7 @@
libaaudioservice \
libaudioflinger \
libaudiopolicyservice \
+ libaudioprocessing \
libbinder \
libcutils \
liblog \
diff --git a/media/bufferpool/1.0/AccessorImpl.cpp b/media/bufferpool/1.0/AccessorImpl.cpp
index fa17f15..a5366f6 100644
--- a/media/bufferpool/1.0/AccessorImpl.cpp
+++ b/media/bufferpool/1.0/AccessorImpl.cpp
@@ -151,6 +151,7 @@
newConnection->initialize(accessor, id);
*connection = newConnection;
*pConnectionId = id;
+ mBufferPool.mConnectionIds.insert(id);
++sSeqId;
}
}
@@ -305,7 +306,12 @@
found->second->mSenderValidated = true;
return true;
}
- // TODO: verify there is target connection Id
+ if (mConnectionIds.find(message.targetConnectionId) == mConnectionIds.end()) {
+ // N.B: it could be fake or receive connection already closed.
+ ALOGD("bufferpool %p receiver connection %lld is no longer valid",
+ this, (long long)message.targetConnectionId);
+ return false;
+ }
mStats.onBufferSent();
mTransactions.insert(std::make_pair(
message.transactionId,
@@ -450,6 +456,7 @@
}
}
}
+ mConnectionIds.erase(connectionId);
return true;
}
diff --git a/media/bufferpool/1.0/AccessorImpl.h b/media/bufferpool/1.0/AccessorImpl.h
index c04dbf3..84cb685 100644
--- a/media/bufferpool/1.0/AccessorImpl.h
+++ b/media/bufferpool/1.0/AccessorImpl.h
@@ -94,6 +94,7 @@
std::map<BufferId, std::unique_ptr<InternalBuffer>> mBuffers;
std::set<BufferId> mFreeBuffers;
+ std::set<ConnectionId> mConnectionIds;
/// Buffer pool statistics which tracks allocation and transfer statistics.
struct Stats {
diff --git a/media/bufferpool/2.0/AccessorImpl.cpp b/media/bufferpool/2.0/AccessorImpl.cpp
index 94cf006..cacd465 100644
--- a/media/bufferpool/2.0/AccessorImpl.cpp
+++ b/media/bufferpool/2.0/AccessorImpl.cpp
@@ -163,6 +163,7 @@
*connection = newConnection;
*pConnectionId = id;
*pMsgId = mBufferPool.mInvalidation.mInvalidationId;
+ mBufferPool.mConnectionIds.insert(id);
mBufferPool.mInvalidationChannel.getDesc(invDescPtr);
mBufferPool.mInvalidation.onConnect(id, observer);
++sSeqId;
@@ -474,7 +475,12 @@
found->second->mSenderValidated = true;
return true;
}
- // TODO: verify there is target connection Id
+ if (mConnectionIds.find(message.targetConnectionId) == mConnectionIds.end()) {
+ // N.B: it could be fake or receive connection already closed.
+ ALOGD("bufferpool2 %p receiver connection %lld is no longer valid",
+ this, (long long)message.targetConnectionId);
+ return false;
+ }
mStats.onBufferSent();
mTransactions.insert(std::make_pair(
message.transactionId,
@@ -644,6 +650,7 @@
}
}
}
+ mConnectionIds.erase(connectionId);
return true;
}
@@ -774,11 +781,19 @@
std::mutex &mutex,
std::condition_variable &cv,
bool &ready) {
+ constexpr uint32_t NUM_SPIN_TO_INCREASE_SLEEP = 1024;
+ constexpr uint32_t NUM_SPIN_TO_LOG = 1024*8;
+ constexpr useconds_t MAX_SLEEP_US = 10000;
+ uint32_t numSpin = 0;
+ useconds_t sleepUs = 1;
+
while(true) {
std::map<uint32_t, const std::weak_ptr<Accessor::Impl>> copied;
{
std::unique_lock<std::mutex> lock(mutex);
if (!ready) {
+ numSpin = 0;
+ sleepUs = 1;
cv.wait(lock);
}
copied.insert(accessors.begin(), accessors.end());
@@ -800,9 +815,20 @@
if (accessors.size() == 0) {
ready = false;
} else {
- // prevent draining cpu.
+ // TODO Use an efficient way to wait over FMQ.
+ // N.B. Since there is not a efficient way to wait over FMQ,
+ // polling over the FMQ is the current way to prevent draining
+ // CPU.
lock.unlock();
- std::this_thread::yield();
+ ++numSpin;
+ if (numSpin % NUM_SPIN_TO_INCREASE_SLEEP == 0 &&
+ sleepUs < MAX_SLEEP_US) {
+ sleepUs *= 10;
+ }
+ if (numSpin % NUM_SPIN_TO_LOG == 0) {
+ ALOGW("invalidator thread spinning");
+ }
+ ::usleep(sleepUs);
}
}
}
diff --git a/media/bufferpool/2.0/AccessorImpl.h b/media/bufferpool/2.0/AccessorImpl.h
index eea72b9..807e0f1 100644
--- a/media/bufferpool/2.0/AccessorImpl.h
+++ b/media/bufferpool/2.0/AccessorImpl.h
@@ -111,6 +111,7 @@
std::map<BufferId, std::unique_ptr<InternalBuffer>> mBuffers;
std::set<BufferId> mFreeBuffers;
+ std::set<ConnectionId> mConnectionIds;
struct Invalidation {
static std::atomic<std::uint32_t> sInvSeqId;
diff --git a/media/codec2/components/aac/C2SoftAacEnc.cpp b/media/codec2/components/aac/C2SoftAacEnc.cpp
index 8e3852c..1dc676b 100644
--- a/media/codec2/components/aac/C2SoftAacEnc.cpp
+++ b/media/codec2/components/aac/C2SoftAacEnc.cpp
@@ -157,7 +157,7 @@
mSentCodecSpecificData(false),
mInputTimeSet(false),
mInputSize(0),
- mInputTimeUs(0),
+ mNextFrameTimestampUs(0),
mSignalledError(false),
mOutIndex(0u) {
}
@@ -183,7 +183,7 @@
mSentCodecSpecificData = false;
mInputTimeSet = false;
mInputSize = 0u;
- mInputTimeUs = 0;
+ mNextFrameTimestampUs = 0;
mSignalledError = false;
return C2_OK;
}
@@ -201,7 +201,7 @@
mSentCodecSpecificData = false;
mInputTimeSet = false;
mInputSize = 0u;
- mInputTimeUs = 0;
+ mNextFrameTimestampUs = 0;
return C2_OK;
}
@@ -365,17 +365,18 @@
capacity = view.capacity();
}
if (!mInputTimeSet && capacity > 0) {
- mInputTimeUs = work->input.ordinal.timestamp;
+ mNextFrameTimestampUs = work->input.ordinal.timestamp;
mInputTimeSet = true;
}
size_t numFrames = (capacity + mInputSize + (eos ? mNumBytesPerInputFrame - 1 : 0))
/ mNumBytesPerInputFrame;
- ALOGV("capacity = %zu; mInputSize = %zu; numFrames = %zu mNumBytesPerInputFrame = %u",
- capacity, mInputSize, numFrames, mNumBytesPerInputFrame);
+ ALOGV("capacity = %zu; mInputSize = %zu; numFrames = %zu "
+ "mNumBytesPerInputFrame = %u inputTS = %lld",
+ capacity, mInputSize, numFrames,
+ mNumBytesPerInputFrame, work->input.ordinal.timestamp.peekll());
std::shared_ptr<C2LinearBlock> block;
- std::shared_ptr<C2Buffer> buffer;
std::unique_ptr<C2WriteView> wView;
uint8_t *outPtr = temp;
size_t outAvailable = 0u;
@@ -442,7 +443,11 @@
const std::shared_ptr<C2Buffer> mBuffer;
};
- C2WorkOrdinalStruct outOrdinal = work->input.ordinal;
+ struct OutputBuffer {
+ std::shared_ptr<C2Buffer> buffer;
+ c2_cntr64_t timestampUs;
+ };
+ std::list<OutputBuffer> outputBuffers;
while (encoderErr == AACENC_OK && inargs.numInSamples > 0) {
if (numFrames && !block) {
@@ -473,29 +478,22 @@
&outargs);
if (encoderErr == AACENC_OK) {
- if (buffer) {
- outOrdinal.frameIndex = mOutIndex++;
- outOrdinal.timestamp = mInputTimeUs;
- cloneAndSend(
- inputIndex,
- work,
- FillWork(C2FrameData::FLAG_INCOMPLETE, outOrdinal, buffer));
- buffer.reset();
- }
-
if (outargs.numOutBytes > 0) {
mInputSize = 0;
int consumed = (capacity / sizeof(int16_t)) - inargs.numInSamples
+ outargs.numInSamples;
- mInputTimeUs = work->input.ordinal.timestamp
+ c2_cntr64_t currentFrameTimestampUs = mNextFrameTimestampUs;
+ mNextFrameTimestampUs = work->input.ordinal.timestamp
+ (consumed * 1000000ll / channelCount / sampleRate);
- buffer = createLinearBuffer(block, 0, outargs.numOutBytes);
+ std::shared_ptr<C2Buffer> buffer = createLinearBuffer(block, 0, outargs.numOutBytes);
#if defined(LOG_NDEBUG) && !LOG_NDEBUG
hexdump(outPtr, std::min(outargs.numOutBytes, 256));
#endif
outPtr = temp;
outAvailable = 0;
block.reset();
+
+ outputBuffers.push_back({buffer, currentFrameTimestampUs});
} else {
mInputSize += outargs.numInSamples * sizeof(int16_t);
}
@@ -506,8 +504,9 @@
inargs.numInSamples -= outargs.numInSamples;
}
}
- ALOGV("encoderErr = %d mInputSize = %zu inargs.numInSamples = %d, mInputTimeUs = %lld",
- encoderErr, mInputSize, inargs.numInSamples, mInputTimeUs.peekll());
+ ALOGV("encoderErr = %d mInputSize = %zu "
+ "inargs.numInSamples = %d, mNextFrameTimestampUs = %lld",
+ encoderErr, mInputSize, inargs.numInSamples, mNextFrameTimestampUs.peekll());
}
if (eos && inBufferSize[0] > 0) {
@@ -542,10 +541,27 @@
&outargs);
}
- outOrdinal.frameIndex = mOutIndex++;
- outOrdinal.timestamp = mInputTimeUs;
+ while (outputBuffers.size() > 1) {
+ const OutputBuffer& front = outputBuffers.front();
+ C2WorkOrdinalStruct ordinal = work->input.ordinal;
+ ordinal.frameIndex = mOutIndex++;
+ ordinal.timestamp = front.timestampUs;
+ cloneAndSend(
+ inputIndex,
+ work,
+ FillWork(C2FrameData::FLAG_INCOMPLETE, ordinal, front.buffer));
+ outputBuffers.pop_front();
+ }
+ std::shared_ptr<C2Buffer> buffer;
+ C2WorkOrdinalStruct ordinal = work->input.ordinal;
+ ordinal.frameIndex = mOutIndex++;
+ if (!outputBuffers.empty()) {
+ ordinal.timestamp = outputBuffers.front().timestampUs;
+ buffer = outputBuffers.front().buffer;
+ }
+ // Mark the end of frame
FillWork((C2FrameData::flags_t)(eos ? C2FrameData::FLAG_END_OF_STREAM : 0),
- outOrdinal, buffer)(work);
+ ordinal, buffer)(work);
}
c2_status_t C2SoftAacEnc::drain(
@@ -569,7 +585,7 @@
mSentCodecSpecificData = false;
mInputTimeSet = false;
mInputSize = 0u;
- mInputTimeUs = 0;
+ mNextFrameTimestampUs = 0;
// TODO: we don't have any pending work at this time to drain.
return C2_OK;
diff --git a/media/codec2/components/aac/C2SoftAacEnc.h b/media/codec2/components/aac/C2SoftAacEnc.h
index a38be19..2655039 100644
--- a/media/codec2/components/aac/C2SoftAacEnc.h
+++ b/media/codec2/components/aac/C2SoftAacEnc.h
@@ -56,7 +56,7 @@
bool mSentCodecSpecificData;
bool mInputTimeSet;
size_t mInputSize;
- c2_cntr64_t mInputTimeUs;
+ c2_cntr64_t mNextFrameTimestampUs;
bool mSignalledError;
std::atomic_uint64_t mOutIndex;
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/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 9d1cc60..8223273 100644
--- a/media/codec2/sfplugin/CCodec.cpp
+++ b/media/codec2/sfplugin/CCodec.cpp
@@ -375,7 +375,11 @@
// consumer usage is queried earlier.
- ALOGD("ISConfig%s", status.str().c_str());
+ if (status.str().empty()) {
+ ALOGD("ISConfig not changed");
+ } else {
+ ALOGD("ISConfig%s", status.str().c_str());
+ }
return err;
}
diff --git a/media/codec2/sfplugin/CCodecBufferChannel.cpp b/media/codec2/sfplugin/CCodecBufferChannel.cpp
index 8308292..0cbf62b 100644
--- a/media/codec2/sfplugin/CCodecBufferChannel.cpp
+++ b/media/codec2/sfplugin/CCodecBufferChannel.cpp
@@ -224,7 +224,7 @@
mFirstValidFrameIndex(0u),
mMetaMode(MODE_NONE),
mInputMetEos(false) {
- mOutputSurface.lock()->maxDequeueBuffers = kSmoothnessFactor + kRenderingDepth;
+ mOutputSurface.lock()->maxDequeueBuffers = 2 * kSmoothnessFactor + kRenderingDepth;
{
Mutexed<Input>::Locked input(mInput);
input->buffers.reset(new DummyInputBuffers(""));
@@ -948,7 +948,8 @@
uint32_t outputGeneration;
{
Mutexed<OutputSurface>::Locked output(mOutputSurface);
- output->maxDequeueBuffers = numOutputSlots + reorderDepth.value + kRenderingDepth;
+ output->maxDequeueBuffers = numOutputSlots + numInputSlots +
+ reorderDepth.value + kRenderingDepth;
outputSurface = output->surface ?
output->surface->getIGraphicBufferProducer() : nullptr;
if (outputSurface) {
@@ -1332,9 +1333,10 @@
ALOGV("[%s] onWorkDone: updated reorder depth to %u",
mName, reorderDepth.value);
size_t numOutputSlots = mOutput.lock()->numSlots;
+ size_t numInputSlots = mInput.lock()->numSlots;
Mutexed<OutputSurface>::Locked output(mOutputSurface);
- output->maxDequeueBuffers =
- numOutputSlots + reorderDepth.value + kRenderingDepth;
+ output->maxDequeueBuffers = numOutputSlots + numInputSlots +
+ reorderDepth.value + kRenderingDepth;
if (output->surface) {
output->surface->setMaxDequeuedBufferCount(output->maxDequeueBuffers);
}
@@ -1382,6 +1384,7 @@
bool outputBuffersChanged = false;
size_t numOutputSlots = 0;
+ size_t numInputSlots = mInput.lock()->numSlots;
{
Mutexed<Output>::Locked output(mOutput);
output->outputDelay = outputDelay.value;
@@ -1406,7 +1409,8 @@
uint32_t depth = mReorderStash.lock()->depth();
Mutexed<OutputSurface>::Locked output(mOutputSurface);
- output->maxDequeueBuffers = numOutputSlots + depth + kRenderingDepth;
+ output->maxDequeueBuffers = numOutputSlots + numInputSlots +
+ depth + kRenderingDepth;
if (output->surface) {
output->surface->setMaxDequeuedBufferCount(output->maxDequeueBuffers);
}
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/CCodecConfig.cpp b/media/codec2/sfplugin/CCodecConfig.cpp
index 1cfdc19..5adcd94 100644
--- a/media/codec2/sfplugin/CCodecConfig.cpp
+++ b/media/codec2/sfplugin/CCodecConfig.cpp
@@ -235,7 +235,10 @@
const std::vector<ConfigMapper> &getConfigMappersForSdkKey(std::string key) const {
auto it = mConfigMappers.find(key);
if (it == mConfigMappers.end()) {
- ALOGD("no c2 equivalents for %s", key.c_str());
+ if (mComplained.count(key) == 0) {
+ ALOGD("no c2 equivalents for %s", key.c_str());
+ mComplained.insert(key);
+ }
return NO_MAPPERS;
}
ALOGV("found %zu eqs for %s", it->second.size(), key.c_str());
@@ -304,6 +307,7 @@
private:
std::map<SdkKey, std::vector<ConfigMapper>> mConfigMappers;
+ mutable std::set<std::string> mComplained;
};
const std::vector<ConfigMapper> StandardParams::NO_MAPPERS;
@@ -508,7 +512,8 @@
.limitTo(D::ENCODER & D::VIDEO));
// convert to timestamp base
add(ConfigMapper(KEY_I_FRAME_INTERVAL, C2_PARAMKEY_SYNC_FRAME_INTERVAL, "value")
- .withMappers([](C2Value v) -> C2Value {
+ .limitTo(D::VIDEO & D::ENCODER & D::CONFIG)
+ .withMapper([](C2Value v) -> C2Value {
// convert from i32 to float
int32_t i32Value;
float fpValue;
@@ -518,12 +523,6 @@
return int64_t(c2_min(1000000 * fpValue + 0.5, (double)INT64_MAX));
}
return C2Value();
- }, [](C2Value v) -> C2Value {
- int64_t i64;
- if (v.get(&i64)) {
- return float(i64) / 1000000;
- }
- return C2Value();
}));
// remove when codecs switch to proper coding.gop (add support for calculating gop)
deprecated(ConfigMapper("i-frame-period", "coding.gop", "intra-period")
@@ -1033,7 +1032,25 @@
}
ReflectedParamUpdater::Dict reflected = mParamUpdater->getParams(paramPointers);
- ALOGD("c2 config is %s", reflected.debugString().c_str());
+ std::string config = reflected.debugString();
+ std::set<std::string> configLines;
+ std::string diff;
+ for (size_t start = 0; start != std::string::npos; ) {
+ size_t end = config.find('\n', start);
+ size_t count = (end == std::string::npos)
+ ? std::string::npos
+ : end - start + 1;
+ std::string line = config.substr(start, count);
+ configLines.insert(line);
+ if (mLastConfig.count(line) == 0) {
+ diff.append(line);
+ }
+ start = (end == std::string::npos) ? std::string::npos : end + 1;
+ }
+ if (!diff.empty()) {
+ ALOGD("c2 config diff is %s", diff.c_str());
+ }
+ mLastConfig.swap(configLines);
bool changed = false;
if (domain & mInputDomain) {
diff --git a/media/codec2/sfplugin/CCodecConfig.h b/media/codec2/sfplugin/CCodecConfig.h
index 3bafe3f..a61c8b7 100644
--- a/media/codec2/sfplugin/CCodecConfig.h
+++ b/media/codec2/sfplugin/CCodecConfig.h
@@ -134,6 +134,8 @@
/// For now support a validation function.
std::map<C2Param::Index, LocalParamValidator> mLocalParams;
+ std::set<std::string> mLastConfig;
+
CCodecConfig();
/// initializes the members required to manage the format: descriptors, reflector,
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/PipelineWatcher.cpp b/media/codec2/sfplugin/PipelineWatcher.cpp
index 74d14e8..0ee9056 100644
--- a/media/codec2/sfplugin/PipelineWatcher.cpp
+++ b/media/codec2/sfplugin/PipelineWatcher.cpp
@@ -146,7 +146,7 @@
std::chrono::duration_cast<std::chrono::milliseconds>(elapsed).count());
durations.push_back(elapsed);
}
- std::nth_element(durations.begin(), durations.end(), durations.begin() + n,
+ std::nth_element(durations.begin(), durations.begin() + n, durations.end(),
std::greater<Clock::duration>());
return durations[n];
}
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/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..d36cb49 100644
--- a/media/extractors/midi/Android.bp
+++ b/media/extractors/midi/Android.bp
@@ -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/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..1d9e1e6 100644
--- a/media/extractors/mpeg2/Android.bp
+++ b/media/extractors/mpeg2/Android.bp
@@ -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 731584d..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>
@@ -111,8 +110,10 @@
AMediaFormat *meta = AMediaFormat_new();
for (size_t i = mTracks.size(); i > 0;) {
i--;
- if (mTracks.valueAt(i)->getFormat(meta) != AMEDIA_OK) {
+ Track *track = mTracks.valueAt(i);
+ if (track->getFormat(meta) != AMEDIA_OK) {
mTracks.removeItemsAt(i);
+ delete track;
}
}
AMediaFormat_delete(meta);
@@ -122,6 +123,10 @@
MPEG2PSExtractor::~MPEG2PSExtractor() {
delete mDataSource;
+ for (size_t i = mTracks.size(); i > 0;) {
+ i--;
+ delete mTracks.valueAt(i);
+ }
}
size_t MPEG2PSExtractor::countTracks() {
@@ -793,7 +798,9 @@
}
media_status_t MPEG2PSExtractor::WrappedTrack::start() {
+ delete mTrack->mBufferGroup;
mTrack->mBufferGroup = mBufferGroup;
+ mBufferGroup = nullptr;
return mTrack->start();
}
diff --git a/media/libaaudio/src/Android.bp b/media/libaaudio/src/Android.bp
index 56c0170..850b1d0 100644
--- a/media/libaaudio/src/Android.bp
+++ b/media/libaaudio/src/Android.bp
@@ -10,7 +10,9 @@
"legacy",
"utility",
],
- header_libs: ["libaaudio_headers"],
+ header_libs: [
+ "libaaudio_headers",
+ ],
export_header_lib_headers: ["libaaudio_headers"],
version_script: "libaaudio.map.txt",
@@ -53,7 +55,10 @@
],
export_include_dirs: ["."],
- header_libs: ["libaaudio_headers"],
+ header_libs: [
+ "libaaudio_headers",
+ "libmedia_headers"
+ ],
export_header_lib_headers: ["libaaudio_headers"],
shared_libs: [
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/client/AudioStreamInternalCapture.cpp b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
index a6cc45b..366cc87 100644
--- a/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
@@ -89,7 +89,11 @@
if (mAudioEndpoint.isFreeRunning()) {
//ALOGD("AudioStreamInternalCapture::processDataNow() - update remote counter");
// Update data queue based on the timing model.
- int64_t estimatedRemoteCounter = mClockModel.convertTimeToPosition(currentNanoTime);
+ // Jitter in the DSP can cause late writes to the FIFO.
+ // This might be caused by resampling.
+ // We want to read the FIFO after the latest possible time
+ // that the DSP could have written the data.
+ int64_t estimatedRemoteCounter = mClockModel.convertLatestTimeToPosition(currentNanoTime);
// TODO refactor, maybe use setRemoteCounter()
mAudioEndpoint.setDataWriteCounter(estimatedRemoteCounter);
}
@@ -139,7 +143,7 @@
// the writeCounter might have just advanced in the background,
// causing us to sleep until a later burst.
int64_t nextPosition = mAudioEndpoint.getDataReadCounter() + mFramesPerBurst;
- wakeTime = mClockModel.convertPositionToTime(nextPosition);
+ wakeTime = mClockModel.convertPositionToLatestTime(nextPosition);
}
break;
default:
diff --git a/media/libaaudio/src/client/IsochronousClockModel.cpp b/media/libaaudio/src/client/IsochronousClockModel.cpp
index 747d0e1..9abdf53 100644
--- a/media/libaaudio/src/client/IsochronousClockModel.cpp
+++ b/media/libaaudio/src/client/IsochronousClockModel.cpp
@@ -19,12 +19,11 @@
#include <log/log.h>
#include <stdint.h>
+#include <algorithm>
#include "utility/AudioClock.h"
#include "IsochronousClockModel.h"
-#define MIN_LATENESS_NANOS (10 * AAUDIO_NANOS_PER_MICROSECOND)
-
using namespace aaudio;
IsochronousClockModel::IsochronousClockModel()
@@ -32,7 +31,7 @@
, mMarkerNanoTime(0)
, mSampleRate(48000)
, mFramesPerBurst(64)
- , mMaxLatenessInNanos(0)
+ , mMaxMeasuredLatenessNanos(0)
, mState(STATE_STOPPED)
{
}
@@ -41,8 +40,7 @@
}
void IsochronousClockModel::setPositionAndTime(int64_t framePosition, int64_t nanoTime) {
- ALOGV("setPositionAndTime(%lld, %lld)",
- (long long) framePosition, (long long) nanoTime);
+ ALOGV("setPositionAndTime, %lld, %lld", (long long) framePosition, (long long) nanoTime);
mMarkerFramePosition = framePosition;
mMarkerNanoTime = nanoTime;
}
@@ -54,7 +52,9 @@
}
void IsochronousClockModel::stop(int64_t nanoTime) {
- ALOGV("stop(nanos = %lld)\n", (long long) nanoTime);
+ ALOGD("stop(nanos = %lld) max lateness = %d micros\n",
+ (long long) nanoTime,
+ (int) (mMaxMeasuredLatenessNanos / 1000));
setPositionAndTime(convertTimeToPosition(nanoTime), nanoTime);
// TODO should we set position?
mState = STATE_STOPPED;
@@ -69,9 +69,10 @@
}
void IsochronousClockModel::processTimestamp(int64_t framePosition, int64_t nanoTime) {
-// ALOGD("processTimestamp() - framePosition = %lld at nanoTime %llu",
-// (long long)framePosition,
-// (long long)nanoTime);
+ mTimestampCount++;
+// Log position and time in CSV format so we can import it easily into spreadsheets.
+ //ALOGD("%s() CSV, %d, %lld, %lld", __func__,
+ //mTimestampCount, (long long)framePosition, (long long)nanoTime);
int64_t framesDelta = framePosition - mMarkerFramePosition;
int64_t nanosDelta = nanoTime - mMarkerNanoTime;
if (nanosDelta < 1000) {
@@ -108,17 +109,56 @@
case STATE_RUNNING:
if (nanosDelta < expectedNanosDelta) {
// Earlier than expected timestamp.
- // This data is probably more accurate so use it.
- // or we may be drifting due to a slow HW clock.
-// ALOGD("processTimestamp() - STATE_RUNNING - %d < %d micros - EARLY",
-// (int) (nanosDelta / 1000), (int)(expectedNanosDelta / 1000));
+ // This data is probably more accurate, so use it.
+ // Or we may be drifting due to a fast HW clock.
+ //int microsDelta = (int) (nanosDelta / 1000);
+ //int expectedMicrosDelta = (int) (expectedNanosDelta / 1000);
+ //ALOGD("%s() - STATE_RUNNING - #%d, %4d micros EARLY",
+ //__func__, mTimestampCount, expectedMicrosDelta - microsDelta);
+
setPositionAndTime(framePosition, nanoTime);
- } else if (nanosDelta > (expectedNanosDelta + mMaxLatenessInNanos)) {
- // Later than expected timestamp.
-// ALOGD("processTimestamp() - STATE_RUNNING - %d > %d + %d micros - LATE",
-// (int) (nanosDelta / 1000), (int)(expectedNanosDelta / 1000),
-// (int) (mMaxLatenessInNanos / 1000));
- setPositionAndTime(framePosition - mFramesPerBurst, nanoTime - mMaxLatenessInNanos);
+ } else if (nanosDelta > (expectedNanosDelta + (2 * mBurstPeriodNanos))) {
+ // In this case we do not update mMaxMeasuredLatenessNanos because it
+ // would force it too high.
+ // mMaxMeasuredLatenessNanos should range from 1 to 2 * mBurstPeriodNanos
+ //int32_t measuredLatenessNanos = (int32_t)(nanosDelta - expectedNanosDelta);
+ //ALOGD("%s() - STATE_RUNNING - #%d, lateness %d - max %d = %4d micros VERY LATE",
+ //__func__,
+ //mTimestampCount,
+ //measuredLatenessNanos / 1000,
+ //mMaxMeasuredLatenessNanos / 1000,
+ //(measuredLatenessNanos - mMaxMeasuredLatenessNanos) / 1000
+ //);
+
+ // This typically happens when we are modelling a service instead of a DSP.
+ setPositionAndTime(framePosition, nanoTime - (2 * mBurstPeriodNanos));
+ } else if (nanosDelta > (expectedNanosDelta + mMaxMeasuredLatenessNanos)) {
+ //int32_t previousLatenessNanos = mMaxMeasuredLatenessNanos;
+ mMaxMeasuredLatenessNanos = (int32_t)(nanosDelta - expectedNanosDelta);
+
+ //ALOGD("%s() - STATE_RUNNING - #%d, newmax %d - oldmax %d = %4d micros LATE",
+ //__func__,
+ //mTimestampCount,
+ //mMaxMeasuredLatenessNanos / 1000,
+ //previousLatenessNanos / 1000,
+ //(mMaxMeasuredLatenessNanos - previousLatenessNanos) / 1000
+ //);
+
+ // When we are late, it may be because of preemption in the kernel,
+ // or timing jitter caused by resampling in the DSP,
+ // or we may be drifting due to a slow HW clock.
+ // We add slight drift value just in case there is actual long term drift
+ // forward caused by a slower clock.
+ // If the clock is faster than the model will get pushed earlier
+ // by the code in the preceding branch.
+ // The two opposing forces should allow the model to track the real clock
+ // over a long time.
+ int64_t driftingTime = mMarkerNanoTime + expectedNanosDelta + kDriftNanos;
+ setPositionAndTime(framePosition, driftingTime);
+ //ALOGD("%s() - #%d, max lateness = %d micros",
+ //__func__,
+ //mTimestampCount,
+ //(int) (mMaxMeasuredLatenessNanos / 1000));
}
break;
default:
@@ -138,9 +178,12 @@
update();
}
+// Update expected lateness based on sampleRate and framesPerBurst
void IsochronousClockModel::update() {
- int64_t nanosLate = convertDeltaPositionToTime(mFramesPerBurst); // uses mSampleRate
- mMaxLatenessInNanos = (nanosLate > MIN_LATENESS_NANOS) ? nanosLate : MIN_LATENESS_NANOS;
+ mBurstPeriodNanos = convertDeltaPositionToTime(mFramesPerBurst); // uses mSampleRate
+ // Timestamps may be late by up to a burst because we are randomly sampling the time period
+ // after the DSP position is actually updated.
+ mMaxMeasuredLatenessNanos = mBurstPeriodNanos;
}
int64_t IsochronousClockModel::convertDeltaPositionToTime(int64_t framesDelta) const {
@@ -183,11 +226,25 @@
return position;
}
+int32_t IsochronousClockModel::getLateTimeOffsetNanos() const {
+ // This will never be < 0 because mMaxLatenessNanos starts at
+ // mBurstPeriodNanos and only gets bigger.
+ return (mMaxMeasuredLatenessNanos - mBurstPeriodNanos) + kExtraLatenessNanos;
+}
+
+int64_t IsochronousClockModel::convertPositionToLatestTime(int64_t framePosition) const {
+ return convertPositionToTime(framePosition) + getLateTimeOffsetNanos();
+}
+
+int64_t IsochronousClockModel::convertLatestTimeToPosition(int64_t nanoTime) const {
+ return convertTimeToPosition(nanoTime - getLateTimeOffsetNanos());
+}
+
void IsochronousClockModel::dump() const {
ALOGD("mMarkerFramePosition = %lld", (long long) mMarkerFramePosition);
ALOGD("mMarkerNanoTime = %lld", (long long) mMarkerNanoTime);
ALOGD("mSampleRate = %6d", mSampleRate);
ALOGD("mFramesPerBurst = %6d", mFramesPerBurst);
- ALOGD("mMaxLatenessInNanos = %6d", mMaxLatenessInNanos);
+ ALOGD("mMaxMeasuredLatenessNanos = %6d", mMaxMeasuredLatenessNanos);
ALOGD("mState = %6d", mState);
}
diff --git a/media/libaaudio/src/client/IsochronousClockModel.h b/media/libaaudio/src/client/IsochronousClockModel.h
index 46ca48e..582bf4e 100644
--- a/media/libaaudio/src/client/IsochronousClockModel.h
+++ b/media/libaaudio/src/client/IsochronousClockModel.h
@@ -18,6 +18,7 @@
#define ANDROID_AAUDIO_ISOCHRONOUS_CLOCK_MODEL_H
#include <stdint.h>
+#include "utility/AudioClock.h"
namespace aaudio {
@@ -79,6 +80,15 @@
int64_t convertPositionToTime(int64_t framePosition) const;
/**
+ * Calculate the latest estimated time that the stream will be at that position.
+ * The more jittery the clock is then the later this will be.
+ *
+ * @param framePosition
+ * @return time in nanoseconds
+ */
+ int64_t convertPositionToLatestTime(int64_t framePosition) const;
+
+ /**
* Calculate an estimated position where the stream will be at the specified time.
*
* @param nanoTime time of interest
@@ -87,6 +97,18 @@
int64_t convertTimeToPosition(int64_t nanoTime) const;
/**
+ * Calculate the corresponding estimated position based on the specified time being
+ * the latest possible time.
+ *
+ * For the same nanoTime, this may return an earlier position than
+ * convertTimeToPosition().
+ *
+ * @param nanoTime
+ * @return position in frames
+ */
+ int64_t convertLatestTimeToPosition(int64_t nanoTime) const;
+
+ /**
* @param framesDelta difference in frames
* @return duration in nanoseconds
*/
@@ -101,6 +123,9 @@
void dump() const;
private:
+
+ int32_t getLateTimeOffsetNanos() const;
+
enum clock_model_state_t {
STATE_STOPPED,
STATE_STARTING,
@@ -108,13 +133,23 @@
STATE_RUNNING
};
+ // Amount of time to drift forward when we get a late timestamp.
+ // This value was calculated to allow tracking of a clock with 50 ppm error.
+ static constexpr int32_t kDriftNanos = 10 * 1000;
+ // TODO review value of kExtraLatenessNanos
+ static constexpr int32_t kExtraLatenessNanos = 100 * 1000;
+
int64_t mMarkerFramePosition;
int64_t mMarkerNanoTime;
int32_t mSampleRate;
int32_t mFramesPerBurst;
- int32_t mMaxLatenessInNanos;
+ int32_t mBurstPeriodNanos;
+ // Includes mBurstPeriodNanos because we sample randomly over time.
+ int32_t mMaxMeasuredLatenessNanos;
clock_model_state_t mState;
+ int32_t mTimestampCount = 0;
+
void update();
};
diff --git a/media/libaudioclient/Android.bp b/media/libaudioclient/Android.bp
index c516d20..32904bb 100644
--- a/media/libaudioclient/Android.bp
+++ b/media/libaudioclient/Android.bp
@@ -84,6 +84,7 @@
header_libs: [
"libaudioclient_headers",
"libbase_headers",
+ "libmedia_headers",
],
export_header_lib_headers: ["libaudioclient_headers"],
diff --git a/media/libaudioclient/IAudioFlinger.cpp b/media/libaudioclient/IAudioFlinger.cpp
index dd95e34..e3e64af 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 {
diff --git a/media/libaudioclient/IAudioPolicyService.cpp b/media/libaudioclient/IAudioPolicyService.cpp
index 64f0aca..7cc95e5 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 {
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/AudioParameter.h b/media/libaudioclient/include/media/AudioParameter.h
index 24837e3..3c190f2 100644
--- a/media/libaudioclient/include/media/AudioParameter.h
+++ b/media/libaudioclient/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/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/libaudiohal/impl/Android.bp b/media/libaudiohal/impl/Android.bp
index d4a4f41..a23d945 100644
--- a/media/libaudiohal/impl/Android.bp
+++ b/media/libaudiohal/impl/Android.bp
@@ -43,6 +43,7 @@
],
header_libs: [
"android.hardware.audio.common.util@all-versions",
+ "libaudioclient_headers",
"libaudiohal_headers"
],
diff --git a/media/libaudioprocessing/Android.bp b/media/libaudioprocessing/Android.bp
index cb78063..9b5d58c 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: [
@@ -33,18 +26,32 @@
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/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 99%
rename from media/libstagefright/NuCachedSource2.cpp
rename to media/libdatasource/NuCachedSource2.cpp
index 522c81d..7f5ae61 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>
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 100%
rename from media/libstagefright/include/NuCachedSource2.h
rename to media/libdatasource/include/datasource/NuCachedSource2.h
diff --git a/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp b/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
index 3fbbc09..10dda19 100644
--- a/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
+++ b/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
@@ -302,6 +302,8 @@
for (int i = 0; i < FIVEBAND_NUMBANDS; i++) {
pContext->pBundledContext->bandGaindB[i] = EQNB_5BandSoftPresets[i];
}
+ pContext->pBundledContext->effectProcessCalled = 0;
+ pContext->pBundledContext->effectInDrain = 0;
ALOGV("\tEffectCreate - Calling LvmBundle_init");
ret = LvmBundle_init(pContext);
@@ -394,6 +396,8 @@
// Clear the instantiated flag for the effect
// protect agains the case where an effect is un-instantiated without being disabled
+
+ int &effectInDrain = pContext->pBundledContext->effectInDrain;
if(pContext->EffectType == LVM_BASS_BOOST) {
ALOGV("\tEffectRelease LVM_BASS_BOOST Clearing global intstantiated flag");
pSessionContext->bBassInstantiated = LVM_FALSE;
@@ -418,12 +422,16 @@
} else if(pContext->EffectType == LVM_VOLUME) {
ALOGV("\tEffectRelease LVM_VOLUME Clearing global intstantiated flag");
pSessionContext->bVolumeInstantiated = LVM_FALSE;
- if (pContext->pBundledContext->bVolumeEnabled == LVM_TRUE){
+ // There is no samplesToExitCount for volume so we also use the drain flag to check
+ // if we should decrement the effects enabled.
+ if (pContext->pBundledContext->bVolumeEnabled == LVM_TRUE
+ || (effectInDrain & 1 << LVM_VOLUME) != 0) {
pContext->pBundledContext->NumberEffectsEnabled--;
}
} else {
ALOGV("\tLVM_ERROR : EffectRelease : Unsupported effect\n\n\n\n\n\n\n");
}
+ effectInDrain &= ~(1 << pContext->EffectType); // no need to drain if released
// Disable effect, in this case ignore errors (return codes)
// if an effect has already been disabled
@@ -3124,8 +3132,9 @@
int Effect_setEnabled(EffectContext *pContext, bool enabled)
{
- ALOGV("\tEffect_setEnabled() type %d, enabled %d", pContext->EffectType, enabled);
-
+ ALOGV("%s effectType %d, enabled %d, currently enabled %d", __func__,
+ pContext->EffectType, enabled, pContext->pBundledContext->NumberEffectsEnabled);
+ int &effectInDrain = pContext->pBundledContext->effectInDrain;
if (enabled) {
// Bass boost or Virtualizer can be temporarily disabled if playing over device speaker due
// to their nature.
@@ -3139,6 +3148,7 @@
if(pContext->pBundledContext->SamplesToExitCountBb <= 0){
pContext->pBundledContext->NumberEffectsEnabled++;
}
+ effectInDrain &= ~(1 << LVM_BASS_BOOST);
pContext->pBundledContext->SamplesToExitCountBb =
(LVM_INT32)(pContext->pBundledContext->SamplesPerSecond*0.1);
pContext->pBundledContext->bBassEnabled = LVM_TRUE;
@@ -3152,6 +3162,7 @@
if(pContext->pBundledContext->SamplesToExitCountEq <= 0){
pContext->pBundledContext->NumberEffectsEnabled++;
}
+ effectInDrain &= ~(1 << LVM_EQUALIZER);
pContext->pBundledContext->SamplesToExitCountEq =
(LVM_INT32)(pContext->pBundledContext->SamplesPerSecond*0.1);
pContext->pBundledContext->bEqualizerEnabled = LVM_TRUE;
@@ -3164,6 +3175,7 @@
if(pContext->pBundledContext->SamplesToExitCountVirt <= 0){
pContext->pBundledContext->NumberEffectsEnabled++;
}
+ effectInDrain &= ~(1 << LVM_VIRTUALIZER);
pContext->pBundledContext->SamplesToExitCountVirt =
(LVM_INT32)(pContext->pBundledContext->SamplesPerSecond*0.1);
pContext->pBundledContext->bVirtualizerEnabled = LVM_TRUE;
@@ -3174,7 +3186,10 @@
ALOGV("\tEffect_setEnabled() LVM_VOLUME is already enabled");
return -EINVAL;
}
- pContext->pBundledContext->NumberEffectsEnabled++;
+ if ((effectInDrain & 1 << LVM_VOLUME) == 0) {
+ pContext->pBundledContext->NumberEffectsEnabled++;
+ }
+ effectInDrain &= ~(1 << LVM_VOLUME);
pContext->pBundledContext->bVolumeEnabled = LVM_TRUE;
break;
default:
@@ -3192,6 +3207,7 @@
return -EINVAL;
}
pContext->pBundledContext->bBassEnabled = LVM_FALSE;
+ effectInDrain |= 1 << LVM_BASS_BOOST;
break;
case LVM_EQUALIZER:
if (pContext->pBundledContext->bEqualizerEnabled == LVM_FALSE) {
@@ -3199,6 +3215,7 @@
return -EINVAL;
}
pContext->pBundledContext->bEqualizerEnabled = LVM_FALSE;
+ effectInDrain |= 1 << LVM_EQUALIZER;
break;
case LVM_VIRTUALIZER:
if (pContext->pBundledContext->bVirtualizerEnabled == LVM_FALSE) {
@@ -3206,6 +3223,7 @@
return -EINVAL;
}
pContext->pBundledContext->bVirtualizerEnabled = LVM_FALSE;
+ effectInDrain |= 1 << LVM_VIRTUALIZER;
break;
case LVM_VOLUME:
if (pContext->pBundledContext->bVolumeEnabled == LVM_FALSE) {
@@ -3213,6 +3231,7 @@
return -EINVAL;
}
pContext->pBundledContext->bVolumeEnabled = LVM_FALSE;
+ effectInDrain |= 1 << LVM_VOLUME;
break;
default:
ALOGV("\tEffect_setEnabled() invalid effect type");
@@ -3283,6 +3302,38 @@
ALOGV("\tLVM_ERROR : Effect_process() ERROR NULL INPUT POINTER OR FRAME COUNT IS WRONG");
return -EINVAL;
}
+
+ int &effectProcessCalled = pContext->pBundledContext->effectProcessCalled;
+ int &effectInDrain = pContext->pBundledContext->effectInDrain;
+ if ((effectProcessCalled & 1 << pContext->EffectType) != 0) {
+ ALOGW("Effect %d already called", pContext->EffectType);
+ const int undrainedEffects = effectInDrain & ~effectProcessCalled;
+ if ((undrainedEffects & 1 << LVM_BASS_BOOST) != 0) {
+ ALOGW("Draining BASS_BOOST");
+ pContext->pBundledContext->SamplesToExitCountBb = 0;
+ --pContext->pBundledContext->NumberEffectsEnabled;
+ effectInDrain &= ~(1 << LVM_BASS_BOOST);
+ }
+ if ((undrainedEffects & 1 << LVM_EQUALIZER) != 0) {
+ ALOGW("Draining EQUALIZER");
+ pContext->pBundledContext->SamplesToExitCountEq = 0;
+ --pContext->pBundledContext->NumberEffectsEnabled;
+ effectInDrain &= ~(1 << LVM_EQUALIZER);
+ }
+ if ((undrainedEffects & 1 << LVM_VIRTUALIZER) != 0) {
+ ALOGW("Draining VIRTUALIZER");
+ pContext->pBundledContext->SamplesToExitCountVirt = 0;
+ --pContext->pBundledContext->NumberEffectsEnabled;
+ effectInDrain &= ~(1 << LVM_VIRTUALIZER);
+ }
+ if ((undrainedEffects & 1 << LVM_VOLUME) != 0) {
+ ALOGW("Draining VOLUME");
+ --pContext->pBundledContext->NumberEffectsEnabled;
+ effectInDrain &= ~(1 << LVM_VOLUME);
+ }
+ }
+ effectProcessCalled |= 1 << pContext->EffectType;
+
if ((pContext->pBundledContext->bBassEnabled == LVM_FALSE)&&
(pContext->EffectType == LVM_BASS_BOOST)){
//ALOGV("\tEffect_process() LVM_BASS_BOOST Effect is not enabled");
@@ -3291,9 +3342,12 @@
//ALOGV("\tEffect_process: Waiting to turn off BASS_BOOST, %d samples left",
// pContext->pBundledContext->SamplesToExitCountBb);
}
- if(pContext->pBundledContext->SamplesToExitCountBb <= 0) {
+ if (pContext->pBundledContext->SamplesToExitCountBb <= 0) {
status = -ENODATA;
- pContext->pBundledContext->NumberEffectsEnabled--;
+ if ((effectInDrain & 1 << LVM_BASS_BOOST) != 0) {
+ pContext->pBundledContext->NumberEffectsEnabled--;
+ effectInDrain &= ~(1 << LVM_BASS_BOOST);
+ }
ALOGV("\tEffect_process() this is the last frame for LVM_BASS_BOOST");
}
}
@@ -3301,7 +3355,10 @@
(pContext->EffectType == LVM_VOLUME)){
//ALOGV("\tEffect_process() LVM_VOLUME Effect is not enabled");
status = -ENODATA;
- pContext->pBundledContext->NumberEffectsEnabled--;
+ if ((effectInDrain & 1 << LVM_VOLUME) != 0) {
+ pContext->pBundledContext->NumberEffectsEnabled--;
+ effectInDrain &= ~(1 << LVM_VOLUME);
+ }
}
if ((pContext->pBundledContext->bEqualizerEnabled == LVM_FALSE)&&
(pContext->EffectType == LVM_EQUALIZER)){
@@ -3311,9 +3368,12 @@
//ALOGV("\tEffect_process: Waiting to turn off EQUALIZER, %d samples left",
// pContext->pBundledContext->SamplesToExitCountEq);
}
- if(pContext->pBundledContext->SamplesToExitCountEq <= 0) {
+ if (pContext->pBundledContext->SamplesToExitCountEq <= 0) {
status = -ENODATA;
- pContext->pBundledContext->NumberEffectsEnabled--;
+ if ((effectInDrain & 1 << LVM_EQUALIZER) != 0) {
+ pContext->pBundledContext->NumberEffectsEnabled--;
+ effectInDrain &= ~(1 << LVM_EQUALIZER);
+ }
ALOGV("\tEffect_process() this is the last frame for LVM_EQUALIZER");
}
}
@@ -3326,9 +3386,12 @@
//ALOGV("\tEffect_process: Waiting for to turn off VIRTUALIZER, %d samples left",
// pContext->pBundledContext->SamplesToExitCountVirt);
}
- if(pContext->pBundledContext->SamplesToExitCountVirt <= 0) {
+ if (pContext->pBundledContext->SamplesToExitCountVirt <= 0) {
status = -ENODATA;
- pContext->pBundledContext->NumberEffectsEnabled--;
+ if ((effectInDrain & 1 << LVM_VIRTUALIZER) != 0) {
+ pContext->pBundledContext->NumberEffectsEnabled--;
+ effectInDrain &= ~(1 << LVM_VIRTUALIZER);
+ }
ALOGV("\tEffect_process() this is the last frame for LVM_VIRTUALIZER");
}
}
@@ -3337,8 +3400,18 @@
pContext->pBundledContext->NumberEffectsCalled++;
}
- if(pContext->pBundledContext->NumberEffectsCalled ==
- pContext->pBundledContext->NumberEffectsEnabled){
+ if (pContext->pBundledContext->NumberEffectsCalled >=
+ pContext->pBundledContext->NumberEffectsEnabled) {
+
+ // We expect the # effects called to be equal to # effects enabled in sequence (including
+ // draining effects). Warn if this is not the case due to inconsistent calls.
+ ALOGW_IF(pContext->pBundledContext->NumberEffectsCalled >
+ pContext->pBundledContext->NumberEffectsEnabled,
+ "%s Number of effects called %d is greater than number of effects enabled %d",
+ __func__, pContext->pBundledContext->NumberEffectsCalled,
+ pContext->pBundledContext->NumberEffectsEnabled);
+ effectProcessCalled = 0; // reset our consistency check.
+
//ALOGV("\tEffect_process Calling process with %d effects enabled, %d called: Effect %d",
//pContext->pBundledContext->NumberEffectsEnabled,
//pContext->pBundledContext->NumberEffectsCalled, pContext->EffectType);
diff --git a/media/libeffects/lvm/wrapper/Bundle/EffectBundle.h b/media/libeffects/lvm/wrapper/Bundle/EffectBundle.h
index 6af4554..e4aacd0 100644
--- a/media/libeffects/lvm/wrapper/Bundle/EffectBundle.h
+++ b/media/libeffects/lvm/wrapper/Bundle/EffectBundle.h
@@ -110,6 +110,14 @@
#ifdef SUPPORT_MC
LVM_INT32 ChMask;
#endif
+
+ /* Bitmask whether drain is in progress due to disabling the effect.
+ The corresponding bit to an effect is set by 1 << lvm_effect_en. */
+ int effectInDrain;
+
+ /* Bitmask whether process() was called for a particular effect.
+ The corresponding bit to an effect is set by 1 << lvm_effect_en. */
+ int effectProcessCalled;
};
/* SessionContext : One session */
diff --git a/media/libmedia/Android.bp b/media/libmedia/Android.bp
index 8de6381..b49df9e 100644
--- a/media/libmedia/Android.bp
+++ b/media/libmedia/Android.bp
@@ -60,10 +60,8 @@
srcs: [
":libmedia_omx_aidl",
- "IMediaCodecList.cpp",
"IOMX.cpp",
"MediaCodecBuffer.cpp",
- "MediaCodecInfo.cpp",
"OMXBuffer.cpp",
"omx/1.0/WGraphicBufferSource.cpp",
"omx/1.0/WOmxBufferSource.cpp",
@@ -75,7 +73,7 @@
local_include_dirs: ["aidl"],
export_aidl_headers: true,
},
-
+
local_include_dirs: [
"include",
],
@@ -199,6 +197,7 @@
],
header_libs: [
+ "libmedia_headers",
"media_ndk_headers",
],
@@ -217,6 +216,49 @@
},
}
+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",
@@ -244,8 +286,6 @@
"mediarecorder.cpp",
"IMediaMetadataRetriever.cpp",
"mediametadataretriever.cpp",
- "MidiDeviceInfo.cpp",
- "JetPlayer.cpp",
"MediaScanner.cpp",
"MediaScannerClient.cpp",
"CharacterEncodingDetector.cpp",
@@ -253,7 +293,6 @@
"MediaProfiles.cpp",
"MediaResource.cpp",
"MediaResourcePolicy.cpp",
- "Visualizer.cpp",
"StringArray.cpp",
"NdkMediaFormatPriv.cpp",
"NdkMediaErrorPriv.cpp",
@@ -289,8 +328,8 @@
"libstagefright_foundation",
"libgui",
"libdl",
- "libaudioutils",
"libaudioclient",
+ "libmedia_codeclist",
"libmedia_omx",
],
@@ -304,7 +343,6 @@
static_libs: [
"libc_malloc_debug_backtrace", // for memory heap analysis
- "libmedia_midiiowrapper",
],
export_include_dirs: [
@@ -327,64 +365,3 @@
cfi: true,
},
}
-
-cc_library_static {
- name: "libmedia_player2_util",
-
- 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/AudioParameter.cpp b/media/libmedia/AudioParameter.cpp
index 1c95e27..9f34035 100644
--- a/media/libmedia/AudioParameter.cpp
+++ b/media/libmedia/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/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/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/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..6d46363 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 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 cb8d375..0000000
--- a/media/libmedia/Visualizer.cpp
+++ /dev/null
@@ -1,442 +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()) {
- if (t->requestExitAndWait() == WOULD_BLOCK) {
- ALOGE("Visualizer::enable() called from thread");
- return INVALID_OPERATION;
- }
- }
- }
- 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/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/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/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/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 84%
rename from media/libstagefright/FileSource.cpp
rename to media/libmediaplayerservice/datasource/PlayerServiceFileSource.cpp
index aee7fd8..1580891 100644
--- a/media/libstagefright/FileSource.cpp
+++ b/media/libmediaplayerservice/datasource/PlayerServiceFileSource.cpp
@@ -15,18 +15,17 @@
*/
//#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),
@@ -34,8 +33,8 @@
mDrmBuf(NULL){
}
-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),
@@ -43,7 +42,7 @@
mDrmBuf(NULL) {
}
-FileSource::~FileSource() {
+PlayerServiceFileSource::~PlayerServiceFileSource() {
if (mDrmBuf != NULL) {
delete[] mDrmBuf;
mDrmBuf = NULL;
@@ -62,7 +61,7 @@
}
}
-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;
}
@@ -87,7 +86,7 @@
}
}
-sp<DecryptHandle> FileSource::DrmInitialization(const char *mime) {
+sp<DecryptHandle> PlayerServiceFileSource::DrmInitialization(const char *mime) {
if (getuid() == AID_MEDIA_EX) return nullptr; // no DRM in media extractor
if (mDrmManagerClient == NULL) {
mDrmManagerClient = new DrmManagerClient();
@@ -110,7 +109,7 @@
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 +140,7 @@
}
/* 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 78%
rename from media/libstagefright/http/MediaHTTP.cpp
rename to media/libmediaplayerservice/datasource/PlayerServiceMediaHTTP.cpp
index 0fba3dc..0124720 100644
--- a/media/libstagefright/http/MediaHTTP.cpp
+++ b/media/libmediaplayerservice/datasource/PlayerServiceMediaHTTP.cpp
@@ -15,32 +15,32 @@
*/
//#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) {
}
-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 +62,7 @@
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 65%
rename from media/libstagefright/include/media/stagefright/FileSource.h
rename to media/libmediaplayerservice/datasource/include/datasource/PlayerServiceFileSource.h
index b610eef..08a013e 100644
--- a/media/libstagefright/include/media/stagefright/FileSource.h
+++ b/media/libmediaplayerservice/datasource/include/datasource/PlayerServiceFileSource.h
@@ -14,24 +14,26 @@
* 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);
@@ -40,7 +42,7 @@
static bool requiresDrm(int fd, int64_t offset, int64_t length, const char *mime);
protected:
- virtual ~FileSource();
+ virtual ~PlayerServiceFileSource();
private:
/*for DRM*/
@@ -52,11 +54,11 @@
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 67%
rename from media/libstagefright/include/media/stagefright/MediaHTTP.h
rename to media/libmediaplayerservice/datasource/include/datasource/PlayerServiceMediaHTTP.h
index acaa6c4..0032cd7 100644
--- a/media/libstagefright/include/media/stagefright/MediaHTTP.h
+++ b/media/libmediaplayerservice/datasource/include/datasource/PlayerServiceMediaHTTP.h
@@ -14,22 +14,24 @@
* 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 ~PlayerServiceMediaHTTP();
virtual sp<DecryptHandle> DrmInitialization(const char* mime);
@@ -39,9 +41,9 @@
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.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/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index 3d67c91..d198d39 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -4505,22 +4505,38 @@
status_t ACodec::configureImageGrid(
const sp<AMessage> &msg, sp<AMessage> &outputFormat) {
int32_t tileWidth, tileHeight, gridRows, gridCols;
- if (!msg->findInt32("tile-width", &tileWidth) ||
- !msg->findInt32("tile-height", &tileHeight) ||
- !msg->findInt32("grid-rows", &gridRows) ||
- !msg->findInt32("grid-cols", &gridCols)) {
+ OMX_BOOL useGrid = OMX_FALSE;
+ if (msg->findInt32("tile-width", &tileWidth) &&
+ msg->findInt32("tile-height", &tileHeight) &&
+ msg->findInt32("grid-rows", &gridRows) &&
+ msg->findInt32("grid-cols", &gridCols)) {
+ useGrid = OMX_TRUE;
+ } else {
+ // when bEnabled is false, the tile info is not used,
+ // but clear out these too.
+ tileWidth = tileHeight = gridRows = gridCols = 0;
+ }
+
+ if (!mIsImage && !useGrid) {
return OK;
}
OMX_VIDEO_PARAM_ANDROID_IMAGEGRIDTYPE gridType;
InitOMXParams(&gridType);
gridType.nPortIndex = kPortIndexOutput;
- gridType.bEnabled = OMX_TRUE;
+ gridType.bEnabled = useGrid;
gridType.nTileWidth = tileWidth;
gridType.nTileHeight = tileHeight;
gridType.nGridRows = gridRows;
gridType.nGridCols = gridCols;
+ ALOGV("sending image grid info to component: bEnabled %d, tile %dx%d, grid %dx%d",
+ gridType.bEnabled,
+ gridType.nTileWidth,
+ gridType.nTileHeight,
+ gridType.nGridRows,
+ gridType.nGridCols);
+
status_t err = mOMXNode->setParameter(
(OMX_INDEXTYPE)OMX_IndexParamVideoAndroidImageGrid,
&gridType, sizeof(gridType));
@@ -4541,6 +4557,13 @@
(OMX_INDEXTYPE)OMX_IndexParamVideoAndroidImageGrid,
&gridType, sizeof(gridType));
+ ALOGV("received image grid info from component: bEnabled %d, tile %dx%d, grid %dx%d",
+ gridType.bEnabled,
+ gridType.nTileWidth,
+ gridType.nTileHeight,
+ gridType.nGridRows,
+ gridType.nGridCols);
+
if (err == OK && gridType.bEnabled) {
outputFormat->setInt32("tile-width", gridType.nTileWidth);
outputFormat->setInt32("tile-height", gridType.nTileHeight);
diff --git a/media/libstagefright/Android.bp b/media/libstagefright/Android.bp
index bb7f2a5..59cc24b 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",
@@ -121,7 +127,6 @@
"ACodecBufferChannel.cpp",
"AHierarchicalStateMachine.cpp",
"AMRWriter.cpp",
- "AudioPlayer.cpp",
"AudioSource.cpp",
"BufferImpl.cpp",
"CallbackDataSource.cpp",
@@ -129,13 +134,7 @@
"CameraSource.cpp",
"CameraSourceTimeLapse.cpp",
"DataConverter.cpp",
- "DataSourceBase.cpp",
- "DataSourceFactory.cpp",
- "DataURISource.cpp",
- "ClearFileSource.cpp",
- "FileSource.cpp",
"FrameDecoder.cpp",
- "HTTPBase.cpp",
"HevcUtils.cpp",
"InterfaceUtils.cpp",
"JPEGSource.cpp",
@@ -152,10 +151,7 @@
"MediaSource.cpp",
"MediaSync.cpp",
"MediaTrack.cpp",
- "http/ClearMediaHTTP.cpp",
- "http/MediaHTTP.cpp",
"MediaMuxer.cpp",
- "NuCachedSource2.cpp",
"NuMediaExtractor.cpp",
"OggWriter.cpp",
"OMXClient.cpp",
@@ -165,11 +161,10 @@
"SimpleDecodingSource.cpp",
"SkipCutBuffer.cpp",
"StagefrightMediaScanner.cpp",
- "StagefrightMetadataRetriever.cpp",
"StagefrightPluginLoader.cpp",
"SurfaceUtils.cpp",
- "Utils.cpp",
"ThrottledSource.cpp",
+ "Utils.cpp",
"VideoFrameSchedulerBase.cpp",
"VideoFrameScheduler.cpp",
],
@@ -180,12 +175,13 @@
"libbinder",
"libcamera_client",
"libcutils",
+ "libdatasource",
"libdl",
"libdl_android",
- "libdrmframework",
"libgui",
"liblog",
"libmedia",
+ "libmedia_codeclist",
"libmedia_omx",
"libmedia_omx_client",
"libaudioclient",
@@ -207,6 +203,7 @@
],
static_libs: [
+ "libstagefright_esds",
"libstagefright_color_conversion",
"libyuv_static",
"libstagefright_mediafilter",
@@ -214,12 +211,11 @@
"libstagefright_timedtext",
"libogg",
"libwebm",
- "libstagefright_esds",
"libstagefright_id3",
- "libFLAC",
],
header_libs:[
+ "libmediadrm_headers",
"libnativeloader-headers",
"libstagefright_xmlparser_headers",
"media_ndk_headers",
@@ -260,4 +256,3 @@
],
},
}
-
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/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/DataSourceBase.cpp b/media/libstagefright/DataSourceBase.cpp
deleted file mode 100644
index 8f47ee5..0000000
--- a/media/libstagefright/DataSourceBase.cpp
+++ /dev/null
@@ -1,130 +0,0 @@
-/*
- * 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 "DataSourceBase"
-
-#include <media/DataSourceBase.h>
-#include <media/stagefright/foundation/ByteUtils.h>
-#include <media/stagefright/MediaErrors.h>
-#include <utils/String8.h>
-
-namespace android {
-
-bool DataSourceBase::getUInt16(off64_t offset, uint16_t *x) {
- *x = 0;
-
- uint8_t byte[2];
- if (readAt(offset, byte, 2) != 2) {
- return false;
- }
-
- *x = (byte[0] << 8) | byte[1];
-
- return true;
-}
-
-bool DataSourceBase::getUInt24(off64_t offset, uint32_t *x) {
- *x = 0;
-
- uint8_t byte[3];
- if (readAt(offset, byte, 3) != 3) {
- return false;
- }
-
- *x = (byte[0] << 16) | (byte[1] << 8) | byte[2];
-
- return true;
-}
-
-bool DataSourceBase::getUInt32(off64_t offset, uint32_t *x) {
- *x = 0;
-
- uint32_t tmp;
- if (readAt(offset, &tmp, 4) != 4) {
- return false;
- }
-
- *x = ntohl(tmp);
-
- return true;
-}
-
-bool DataSourceBase::getUInt64(off64_t offset, uint64_t *x) {
- *x = 0;
-
- uint64_t tmp;
- if (readAt(offset, &tmp, 8) != 8) {
- return false;
- }
-
- *x = ntoh64(tmp);
-
- return true;
-}
-
-bool DataSourceBase::getUInt16Var(off64_t offset, uint16_t *x, size_t size) {
- if (size == 2) {
- return getUInt16(offset, x);
- }
- if (size == 1) {
- uint8_t tmp;
- if (readAt(offset, &tmp, 1) == 1) {
- *x = tmp;
- return true;
- }
- }
- return false;
-}
-
-bool DataSourceBase::getUInt32Var(off64_t offset, uint32_t *x, size_t size) {
- if (size == 4) {
- return getUInt32(offset, x);
- }
- if (size == 2) {
- uint16_t tmp;
- if (getUInt16(offset, &tmp)) {
- *x = tmp;
- return true;
- }
- }
- return false;
-}
-
-bool DataSourceBase::getUInt64Var(off64_t offset, uint64_t *x, size_t size) {
- if (size == 8) {
- return getUInt64(offset, x);
- }
- if (size == 4) {
- uint32_t tmp;
- if (getUInt32(offset, &tmp)) {
- *x = tmp;
- return true;
- }
- }
- return false;
-}
-
-status_t DataSourceBase::getSize(off64_t *size) {
- *size = 0;
-
- return ERROR_UNSUPPORTED;
-}
-
-bool DataSourceBase::getUri(char *uriString __unused, size_t bufferSize __unused) {
- return false;
-}
-
-} // namespace android
diff --git a/media/libstagefright/FrameDecoder.cpp b/media/libstagefright/FrameDecoder.cpp
index cebce3a..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>
diff --git a/media/libstagefright/MPEG4Writer.cpp b/media/libstagefright/MPEG4Writer.cpp
index 2f13dc9..f130c9b 100644
--- a/media/libstagefright/MPEG4Writer.cpp
+++ b/media/libstagefright/MPEG4Writer.cpp
@@ -1635,8 +1635,13 @@
return BAD_VALUE;
}
+ // Increase moovExtraSize once only irrespective of how many times
+ // setCaptureRate is called.
+ bool containsCaptureFps = mMetaKeys->contains(kMetaKey_CaptureFps);
mMetaKeys->setFloat(kMetaKey_CaptureFps, captureFps);
- mMoovExtraSize += sizeof(kMetaKey_CaptureFps) + 4 + 32;
+ if (!containsCaptureFps) {
+ mMoovExtraSize += sizeof(kMetaKey_CaptureFps) + 4 + 32;
+ }
return OK;
}
diff --git a/media/libstagefright/MediaCodec.cpp b/media/libstagefright/MediaCodec.cpp
index f579e9d..161c178 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>
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/MediaMuxer.cpp b/media/libstagefright/MediaMuxer.cpp
index 9ba2add..7ebdb1a 100644
--- a/media/libstagefright/MediaMuxer.cpp
+++ b/media/libstagefright/MediaMuxer.cpp
@@ -96,10 +96,18 @@
sp<MediaAdapter> newTrack = new MediaAdapter(trackMeta);
status_t result = mWriter->addSource(newTrack);
- if (result == OK) {
- return mTrackList.add(newTrack);
+ if (result != OK) {
+ return -1;
}
- return -1;
+ float captureFps = -1.0;
+ if (format->findAsFloat("time-lapse-fps", &captureFps)) {
+ ALOGV("addTrack() time-lapse-fps: %f", captureFps);
+ result = mWriter->setCaptureRate(captureFps);
+ if (result != OK) {
+ ALOGW("addTrack() setCaptureRate failed :%d", result);
+ }
+ }
+ return mTrackList.add(newTrack);
}
status_t MediaMuxer::setOrientationHint(int degrees) {
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..bda6053 100644
--- a/media/libstagefright/Utils.cpp
+++ b/media/libstagefright/Utils.cpp
@@ -1895,22 +1895,6 @@
#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 +2083,6 @@
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 +2181,4 @@
}
}
-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/codecs/m4v_h263/dec/src/vop.cpp b/media/libstagefright/codecs/m4v_h263/dec/src/vop.cpp
index f18f789..679b091 100644
--- a/media/libstagefright/codecs/m4v_h263/dec/src/vop.cpp
+++ b/media/libstagefright/codecs/m4v_h263/dec/src/vop.cpp
@@ -1355,6 +1355,14 @@
int tmpHeight = (tmpDisplayHeight + 15) & -16;
int tmpWidth = (tmpDisplayWidth + 15) & -16;
+ if (tmpWidth > video->width)
+ {
+ // while allowed by the spec, this decoder does not actually
+ // support an increase in size.
+ ALOGE("width increase not supported");
+ status = PV_FAIL;
+ goto return_point;
+ }
if (tmpHeight * tmpWidth > video->size)
{
// This is just possibly "b/37079296".
diff --git a/media/libstagefright/filters/Android.bp b/media/libstagefright/filters/Android.bp
index b1f62c7..88f30c4 100644
--- a/media/libstagefright/filters/Android.bp
+++ b/media/libstagefright/filters/Android.bp
@@ -23,6 +23,10 @@
"-Wall",
],
+ header_libs: [
+ "libmediadrm_headers",
+ ],
+
shared_libs: [
"libgui",
"libmedia",
diff --git a/media/libstagefright/foundation/Android.bp b/media/libstagefright/foundation/Android.bp
index 533cd72..b95f054 100644
--- a/media/libstagefright/foundation/Android.bp
+++ b/media/libstagefright/foundation/Android.bp
@@ -65,6 +65,7 @@
"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/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/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/media/stagefright/DataSourceBase.h b/media/libstagefright/include/media/stagefright/DataSourceBase.h
index af5b83d..c607c91 100644
--- a/media/libstagefright/include/media/stagefright/DataSourceBase.h
+++ b/media/libstagefright/include/media/stagefright/DataSourceBase.h
@@ -18,6 +18,8 @@
#define DATA_SOURCE_BASE_H_
+#include <media/stagefright/foundation/ByteUtils.h>
+#include <media/stagefright/MediaErrors.h>
#include <sys/types.h>
#include <utils/Errors.h>
@@ -45,20 +47,106 @@
virtual ssize_t readAt(off64_t offset, void *data, size_t size) = 0;
// Convenience methods:
- bool getUInt16(off64_t offset, uint16_t *x);
- bool getUInt24(off64_t offset, uint32_t *x); // 3 byte int, returned as a 32-bit int
- bool getUInt32(off64_t offset, uint32_t *x);
- bool getUInt64(off64_t offset, uint64_t *x);
+ bool getUInt16(off64_t offset, uint16_t *x) {
+ *x = 0;
+
+ uint8_t byte[2];
+ if (readAt(offset, byte, 2) != 2) {
+ return false;
+ }
+
+ *x = (byte[0] << 8) | byte[1];
+
+ return true;
+ }
+ // 3 byte int, returned as a 32-bit int
+ bool getUInt24(off64_t offset, uint32_t *x) {
+ *x = 0;
+
+ uint8_t byte[3];
+ if (readAt(offset, byte, 3) != 3) {
+ return false;
+ }
+
+ *x = (byte[0] << 16) | (byte[1] << 8) | byte[2];
+
+ return true;
+ }
+ bool getUInt32(off64_t offset, uint32_t *x) {
+ *x = 0;
+
+ uint32_t tmp;
+ if (readAt(offset, &tmp, 4) != 4) {
+ return false;
+ }
+
+ *x = ntohl(tmp);
+
+ return true;
+ }
+ bool getUInt64(off64_t offset, uint64_t *x) {
+ *x = 0;
+
+ uint64_t tmp;
+ if (readAt(offset, &tmp, 8) != 8) {
+ return false;
+ }
+
+ *x = ntoh64(tmp);
+
+ return true;
+ }
// read either int<N> or int<2N> into a uint<2N>_t, size is the int size in bytes.
- bool getUInt16Var(off64_t offset, uint16_t *x, size_t size);
- bool getUInt32Var(off64_t offset, uint32_t *x, size_t size);
- bool getUInt64Var(off64_t offset, uint64_t *x, size_t size);
+ bool getUInt16Var(off64_t offset, uint16_t *x, size_t size) {
+ if (size == 2) {
+ return getUInt16(offset, x);
+ }
+ if (size == 1) {
+ uint8_t tmp;
+ if (readAt(offset, &tmp, 1) == 1) {
+ *x = tmp;
+ return true;
+ }
+ }
+ return false;
+ }
+ bool getUInt32Var(off64_t offset, uint32_t *x, size_t size) {
+ if (size == 4) {
+ return getUInt32(offset, x);
+ }
+ if (size == 2) {
+ uint16_t tmp;
+ if (getUInt16(offset, &tmp)) {
+ *x = tmp;
+ return true;
+ }
+ }
+ return false;
+ }
+ bool getUInt64Var(off64_t offset, uint64_t *x, size_t size) {
+ if (size == 8) {
+ return getUInt64(offset, x);
+ }
+ if (size == 4) {
+ uint32_t tmp;
+ if (getUInt32(offset, &tmp)) {
+ *x = tmp;
+ return true;
+ }
+ }
+ return false;
+ }
// May return ERROR_UNSUPPORTED.
- virtual status_t getSize(off64_t *size);
+ virtual status_t getSize(off64_t *size) {
+ *size = 0;
+ return ERROR_UNSUPPORTED;
+ }
- virtual bool getUri(char *uriString, size_t bufferSize);
+ virtual bool getUri(char * /*uriString*/, size_t /*bufferSize*/) {
+ return false;
+ }
virtual uint32_t flags() {
return 0;
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/MediaWriter.h b/media/libstagefright/include/media/stagefright/MediaWriter.h
index 2c12a87..972ae1d 100644
--- a/media/libstagefright/include/media/stagefright/MediaWriter.h
+++ b/media/libstagefright/include/media/stagefright/MediaWriter.h
@@ -35,6 +35,10 @@
virtual status_t start(MetaData *params = NULL) = 0;
virtual status_t stop() = 0;
virtual status_t pause() = 0;
+ virtual status_t setCaptureRate(float /* captureFps */) {
+ ALOGW("setCaptureRate unsupported");
+ return ERROR_UNSUPPORTED;
+ }
virtual void setMaxFileSize(int64_t bytes) { mMaxFileSizeLimitBytes = bytes; }
virtual void setMaxFileDuration(int64_t durationUs) { mMaxFileDurationLimitUs = durationUs; }
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/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..cac1af9 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 {
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/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/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/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/ndk/Android.bp b/media/ndk/Android.bp
index 78c8df5..aac6d71 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",
@@ -145,6 +149,10 @@
"-Wall",
],
+ header_libs: [
+ "libmedia_headers",
+ ],
+
shared_libs: [
],
diff --git a/media/ndk/NdkImage.cpp b/media/ndk/NdkImage.cpp
index 1883f63..1145b7b 100644
--- a/media/ndk/NdkImage.cpp
+++ b/media/ndk/NdkImage.cpp
@@ -35,6 +35,7 @@
int64_t timestamp, int32_t width, int32_t height, int32_t numPlanes) :
mReader(reader), mFormat(format), mUsage(usage), mBuffer(buffer), mLockedBuffer(nullptr),
mTimestamp(timestamp), mWidth(width), mHeight(height), mNumPlanes(numPlanes) {
+ LOG_FATAL_IF(reader == nullptr, "AImageReader shouldn't be null while creating AImage");
}
AImage::~AImage() {
@@ -57,14 +58,9 @@
if (mIsClosed) {
return;
}
- sp<AImageReader> reader = mReader.promote();
- if (reader != nullptr) {
- reader->releaseImageLocked(this, releaseFenceFd);
- } else if (mBuffer != nullptr) {
- LOG_ALWAYS_FATAL("%s: parent AImageReader closed without releasing image %p",
- __FUNCTION__, this);
+ if (!mReader->mIsClosed) {
+ mReader->releaseImageLocked(this, releaseFenceFd);
}
-
// Should have been set to nullptr in releaseImageLocked
// Set to nullptr here for extra safety only
mBuffer = nullptr;
@@ -83,22 +79,12 @@
void
AImage::lockReader() const {
- sp<AImageReader> reader = mReader.promote();
- if (reader == nullptr) {
- // Reader has been closed
- return;
- }
- reader->mLock.lock();
+ mReader->mLock.lock();
}
void
AImage::unlockReader() const {
- sp<AImageReader> reader = mReader.promote();
- if (reader == nullptr) {
- // Reader has been closed
- return;
- }
- reader->mLock.unlock();
+ mReader->mLock.unlock();
}
media_status_t
diff --git a/media/ndk/NdkImagePriv.h b/media/ndk/NdkImagePriv.h
index e0f16da..0e8cbcb 100644
--- a/media/ndk/NdkImagePriv.h
+++ b/media/ndk/NdkImagePriv.h
@@ -72,7 +72,7 @@
uint32_t getJpegSize() const;
// When reader is close, AImage will only accept close API call
- wp<AImageReader> mReader;
+ const sp<AImageReader> mReader;
const int32_t mFormat;
const uint64_t mUsage; // AHARDWAREBUFFER_USAGE_* flags.
BufferItem* mBuffer;
diff --git a/media/ndk/NdkImageReader.cpp b/media/ndk/NdkImageReader.cpp
index baa4fc7..c0ceb3d 100644
--- a/media/ndk/NdkImageReader.cpp
+++ b/media/ndk/NdkImageReader.cpp
@@ -113,12 +113,12 @@
void
AImageReader::FrameListener::onFrameAvailable(const BufferItem& /*item*/) {
- Mutex::Autolock _l(mLock);
sp<AImageReader> reader = mReader.promote();
if (reader == nullptr) {
ALOGW("A frame is available after AImageReader closed!");
return; // reader has been closed
}
+ Mutex::Autolock _l(mLock);
if (mListener.onImageAvailable == nullptr) {
return; // No callback registered
}
@@ -143,12 +143,12 @@
void
AImageReader::BufferRemovedListener::onBufferFreed(const wp<GraphicBuffer>& graphicBuffer) {
- Mutex::Autolock _l(mLock);
sp<AImageReader> reader = mReader.promote();
if (reader == nullptr) {
ALOGW("A frame is available after AImageReader closed!");
return; // reader has been closed
}
+ Mutex::Autolock _l(mLock);
if (mListener.onBufferRemoved == nullptr) {
return; // No callback registered
}
@@ -272,6 +272,11 @@
mFrameListener(new FrameListener(this)),
mBufferRemovedListener(new BufferRemovedListener(this)) {}
+AImageReader::~AImageReader() {
+ Mutex::Autolock _l(mLock);
+ LOG_FATAL_IF("AImageReader not closed before destruction", mIsClosed != true);
+}
+
media_status_t
AImageReader::init() {
PublicFormat publicFormat = static_cast<PublicFormat>(mFormat);
@@ -347,8 +352,12 @@
return AMEDIA_OK;
}
-AImageReader::~AImageReader() {
+void AImageReader::close() {
Mutex::Autolock _l(mLock);
+ if (mIsClosed) {
+ return;
+ }
+ mIsClosed = true;
AImageReader_ImageListener nullListener = {nullptr, nullptr};
setImageListenerLocked(&nullListener);
@@ -741,6 +750,7 @@
void AImageReader_delete(AImageReader* reader) {
ALOGV("%s", __FUNCTION__);
if (reader != nullptr) {
+ reader->close();
reader->decStrong((void*) AImageReader_delete);
}
return;
diff --git a/media/ndk/NdkImageReaderPriv.h b/media/ndk/NdkImageReaderPriv.h
index e328cb1..0779a71 100644
--- a/media/ndk/NdkImageReaderPriv.h
+++ b/media/ndk/NdkImageReaderPriv.h
@@ -76,6 +76,7 @@
int32_t getHeight() const { return mHeight; };
int32_t getFormat() const { return mFormat; };
int32_t getMaxImages() const { return mMaxImages; };
+ void close();
private:
@@ -134,7 +135,7 @@
private:
AImageReader_ImageListener mListener = {nullptr, nullptr};
- wp<AImageReader> mReader;
+ const wp<AImageReader> mReader;
Mutex mLock;
};
sp<FrameListener> mFrameListener;
@@ -149,7 +150,7 @@
private:
AImageReader_BufferRemovedListener mListener = {nullptr, nullptr};
- wp<AImageReader> mReader;
+ const wp<AImageReader> mReader;
Mutex mLock;
};
sp<BufferRemovedListener> mBufferRemovedListener;
@@ -165,6 +166,7 @@
native_handle_t* mWindowHandle = nullptr;
List<AImage*> mAcquiredImages;
+ bool mIsClosed = false;
Mutex mLock;
};
diff --git a/media/ndk/NdkMediaCodec.cpp b/media/ndk/NdkMediaCodec.cpp
index c23f19b..51b516a 100644
--- a/media/ndk/NdkMediaCodec.cpp
+++ b/media/ndk/NdkMediaCodec.cpp
@@ -244,8 +244,8 @@
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 e567613..c1d4686 100644
--- a/media/ndk/NdkMediaDataSource.cpp
+++ b/media/ndk/NdkMediaDataSource.cpp
@@ -26,16 +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 <utils/Log.h>
+#include <utils/StrongPointer.h>
-#include "../../libstagefright/include/HTTPBase.h"
-#include "../../libstagefright/include/NuCachedSource2.h"
#include "NdkMediaDataSourceCallbacksPriv.h"
@@ -202,7 +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/NdkMediaDrm.cpp b/media/ndk/NdkMediaDrm.cpp
index 85dbffe..60f3e8e 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/tests/benchmark/Android.bp b/media/tests/benchmark/Android.bp
index 8a7a59f..de408dd 100644
--- a/media/tests/benchmark/Android.bp
+++ b/media/tests/benchmark/Android.bp
@@ -17,4 +17,5 @@
subdirs = [
"src",
"tests",
-]
\ No newline at end of file
+ "MediaBenchmarkTest",
+]
diff --git a/media/tests/benchmark/MediaBenchmarkTest/Android.bp b/media/tests/benchmark/MediaBenchmarkTest/Android.bp
new file mode 100644
index 0000000..91b03f1
--- /dev/null
+++ b/media/tests/benchmark/MediaBenchmarkTest/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.
+ */
+
+android_test {
+ name: "MediaBenchmarkTest",
+
+ // Include all the test code
+ srcs: ["src/androidTest/**/*.java"],
+
+ sdk_version: "system_current",
+
+ resource_dirs: ["res"],
+
+ libs: [
+ "android.test.runner",
+ "android.test.base",
+ ],
+
+ static_libs: [
+ "libMediaBenchmark",
+ "junit",
+ "androidx.test.runner",
+ ],
+}
+
+android_library {
+ name: "libMediaBenchmark",
+
+ // Include all the libraries
+ srcs: ["src/main/**/*.java"],
+
+ sdk_version: "system_current",
+
+ static_libs: [
+ "androidx.test.core",
+ ],
+}
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..89d6ce2
--- /dev/null
+++ b/media/tests/benchmark/MediaBenchmarkTest/AndroidTest.xml
@@ -0,0 +1,27 @@
+<?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.TestAppInstallSetup">
+ <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..b0ee692
--- /dev/null
+++ b/media/tests/benchmark/MediaBenchmarkTest/build.gradle
@@ -0,0 +1,64 @@
+/*
+ * 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 21
+ 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'
+ }
+ }
+}
+
+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..be2633d
--- /dev/null
+++ b/media/tests/benchmark/MediaBenchmarkTest/src/androidTest/java/com/android/media/benchmark/tests/DecoderTest.java
@@ -0,0 +1,197 @@
+/*
+ * 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 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;
+
+@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 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.mp4", 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.mp4", 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}});
+ }
+
+ @Test(timeout = PER_TEST_TIMEOUT_MS)
+ public void testDecoder() throws IOException {
+ File inputFile = new File(mInputFilePath + mInputFile);
+ if (inputFile.exists()) {
+ FileInputStream fileInput = new FileInputStream(inputFile);
+ FileDescriptor fileDescriptor = fileInput.getFD();
+ Extractor extractor = new Extractor();
+ int trackCount = extractor.setUpExtractor(fileDescriptor);
+ ArrayList<ByteBuffer> inputBuffer = new ArrayList<>();
+ ArrayList<MediaCodec.BufferInfo> frameInfo = new ArrayList<>();
+ if (trackCount <= 0) {
+ Log.e(TAG, "Extraction failed. No tracks for file: " + mInputFile);
+ return;
+ }
+ 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);
+ if (mediaCodecs.size() <= 0) {
+ Log.e(TAG,
+ "No suitable codecs found for file: " + mInputFile
+ + " track : " + currentTrack + " mime: " + mime);
+ continue;
+ }
+ // 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()) {
+ if (!outFile.delete()) {
+ Log.e(TAG, " Unable to delete existing file" + outFile.toString());
+ }
+ }
+ if (outFile.createNewFile()) {
+ decodeOutputStream = new FileOutputStream(outFile);
+ } else {
+ Log.e(TAG, "Unable to create file: " + outFile.toString());
+ }
+ }
+ Decoder decoder = new Decoder();
+ decoder.setupDecoder(decodeOutputStream);
+ int status =
+ decoder.decode(inputBuffer, frameInfo, mAsyncMode, format, codecName);
+ decoder.deInitCodec();
+ if (status == 0) {
+ decoder.dumpStatistics(
+ mInputFile + " " + codecName, extractor.getClipDuration());
+ Log.i(TAG,
+ "Decoding Successful for file: " + mInputFile
+ + " with codec: " + codecName);
+ } else {
+ Log.e(TAG,
+ "Decoder returned error " + status + " for file: " + mInputFile
+ + " with codec: " + codecName);
+ }
+ decoder.resetDecoder();
+ if (decodeOutputStream != null) {
+ decodeOutputStream.close();
+ }
+ }
+ extractor.unselectExtractorTrack(currentTrack);
+ inputBuffer.clear();
+ frameInfo.clear();
+ }
+ extractor.deinitExtractor();
+ fileInput.close();
+ } else {
+ Log.w(TAG,
+ "Warning: Test Skipped. Cannot find " + mInputFile + " in directory "
+ + mInputFilePath);
+ }
+ }
+}
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..9db9c84
--- /dev/null
+++ b/media/tests/benchmark/MediaBenchmarkTest/src/androidTest/java/com/android/media/benchmark/tests/EncoderTest.java
@@ -0,0 +1,272 @@
+/*
+ * 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.Encoder;
+import com.android.media.benchmark.library.Extractor;
+
+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.nio.ByteBuffer;
+
+import java.util.ArrayList;
+import java.util.Arrays;
+import java.util.Collection;
+
+@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 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 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;
+ }
+
+ @Test(timeout = PER_TEST_TIMEOUT_MS)
+ public void sampleEncoderTest() 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);
+ if (inputFile.exists()) {
+ FileInputStream fileInput = new FileInputStream(inputFile);
+ FileDescriptor fileDescriptor = fileInput.getFD();
+ Extractor extractor = new Extractor();
+ int trackCount = extractor.setUpExtractor(fileDescriptor);
+ if (trackCount <= 0) {
+ Log.e(TAG, "Extraction failed. No tracks for file: " + mInputFile);
+ return;
+ }
+ 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);
+ // 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, "");
+ if (status == 0) {
+ Log.i(TAG, "Decoding complete.");
+ } else {
+ Log.e(TAG, "Decode returned error. Encoding did not take place." + status);
+ return;
+ }
+ 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);
+ if (mediaCodecs.size() <= 0) {
+ Log.e(TAG, "No suitable codecs found for file: " + mInputFile + " track : " +
+ currentTrack + " mime: " + mime);
+ return;
+ }
+ 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()) {
+ if (!outEncodeFile.delete()) {
+ Log.e(TAG, "Unable to delete existing file" +
+ decodedFile.toString());
+ }
+ }
+ if (outEncodeFile.createNewFile()) {
+ encodeOutputStream = new FileOutputStream(outEncodeFile);
+ } else {
+ Log.e(TAG, "Unable to create file to write encoder output: " +
+ outEncodeFile.toString());
+ }
+ }
+ File rawFile =
+ new File(mContext.getFilesDir() + "/decoder_" + tid + ".out");
+ if (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);
+ }
+ } else {
+ sampleRate = format.getInteger(MediaFormat.KEY_SAMPLE_RATE);
+ numChannels = format.getInteger(MediaFormat.KEY_CHANNEL_COUNT);
+ bitRate = sampleRate * numChannels * 16;
+ }
+ /*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);
+ } 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();
+ if (status == 0) {
+ encoder.dumpStatistics(mInputFile + "with " + codecName + " for " +
+ "aSyncMode = " + asyncMode, extractor.getClipDuration());
+ Log.i(TAG, "Encoding complete for file: " + mInputFile +
+ " with codec: " + codecName + " for aSyncMode = " +
+ asyncMode);
+ } else {
+ Log.e(TAG,
+ codecName + " encoder returned error " + status + " for " +
+ "file:" + " " + mInputFile);
+ }
+ encoder.resetEncoder();
+ eleStream.close();
+ if (encodeOutputStream != null) {
+ encodeOutputStream.close();
+ }
+ }
+ }
+ }
+ //Cleanup temporary input file
+ if (decodedFile.exists()) {
+ if (decodedFile.delete()) {
+ Log.i(TAG, "Successfully deleted decoded file");
+ } else {
+ Log.e(TAG, "Unable to delete decoded file");
+ }
+ }
+ }
+ extractor.deinitExtractor();
+ fileInput.close();
+ } else {
+ Log.w(TAG, "Warning: Test Skipped. Cannot find " + mInputFile + " in directory " +
+ mInputFilePath);
+ }
+ }
+}
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..a02011c
--- /dev/null
+++ b/media/tests/benchmark/MediaBenchmarkTest/src/androidTest/java/com/android/media/benchmark/tests/ExtractorTest.java
@@ -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.
+ */
+
+package com.android.media.benchmark.tests;
+
+import com.android.media.benchmark.R;
+import com.android.media.benchmark.library.Extractor;
+
+import android.content.Context;
+import android.util.Log;
+
+import androidx.test.platform.app.InstrumentationRegistry;
+
+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.IOException;
+import java.util.Arrays;
+import java.util.Collection;
+
+import static org.hamcrest.CoreMatchers.equalTo;
+import static org.hamcrest.CoreMatchers.is;
+import static org.junit.Assert.assertThat;
+
+@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 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.mp4", 0},
+ {"bbb_48000hz_2ch_100kbps_opus_5mins.webm", 0}});
+ }
+
+ public ExtractorTest(String filename, int track) {
+ this.mInputFileName = filename;
+ this.mTrackId = track;
+ }
+
+ @Test
+ public void sampleExtractTest() throws IOException {
+ int status = -1;
+ File inputFile = new File(mInputFilePath + mInputFileName);
+ if (inputFile.exists()) {
+ FileInputStream fileInput = new FileInputStream(inputFile);
+ FileDescriptor fileDescriptor = fileInput.getFD();
+ Extractor extractor = new Extractor();
+ extractor.setUpExtractor(fileDescriptor);
+ status = extractor.extractSample(mTrackId);
+ extractor.deinitExtractor();
+ extractor.dumpStatistics(mInputFileName);
+ fileInput.close();
+ } else {
+ Log.e(TAG, "Cannot find " + mInputFileName + " in directory " + mInputFilePath);
+ }
+ assertThat(status, is(equalTo(0)));
+ }
+}
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..8c3080c
--- /dev/null
+++ b/media/tests/benchmark/MediaBenchmarkTest/src/androidTest/java/com/android/media/benchmark/tests/MuxerTest.java
@@ -0,0 +1,152 @@
+/*
+ * 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 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.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.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.hamcrest.CoreMatchers.equalTo;
+import static org.hamcrest.CoreMatchers.is;
+import static org.junit.Assert.assertThat;
+
+@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 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.mp4", "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;
+ }
+
+ @Test
+ public void sampleMuxerTest() throws IOException {
+ int status = -1;
+ File inputFile = new File(mInputFilePath + mInputFileName);
+ if (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);
+ if (outputFormat != -1) {
+ Muxer muxer = new Muxer();
+ int trackIndex = muxer.setUpMuxer(mContext, outputFormat, format);
+ status = muxer.mux(trackIndex, inputBuffer, inputBufferInfo);
+ if (status != 0) {
+ Log.e(TAG, "Cannot perform write operation for " + mInputFileName);
+ }
+ muxer.deInitMuxer();
+ muxer.dumpStatistics(mInputFileName, extractor.getClipDuration());
+ muxer.resetMuxer();
+ extractor.unselectExtractorTrack(currentTrack);
+ inputBufferInfo.clear();
+ inputBuffer.clear();
+ } else {
+ Log.e(TAG, "Test failed for " + mInputFileName + ". Returned invalid " +
+ "output format for given " + mFormat + " format.");
+ }
+ }
+ extractor.deinitExtractor();
+ fileInput.close();
+ } else {
+ Log.w(TAG, "Warning: Test Skipped. Cannot find " + mInputFileName + " in directory " +
+ mInputFilePath);
+ }
+ assertThat(status, is(equalTo(0)));
+ }
+}
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..2cd27c2
--- /dev/null
+++ b/media/tests/benchmark/MediaBenchmarkTest/src/main/java/com/android/media/benchmark/library/Decoder.java
@@ -0,0 +1,301 @@
+/*
+ * 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) {
+ Log.i(TAG, "Saw output EOS");
+ 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);
+ }
+ if (outputBufferInfo.flags == MediaCodec.BUFFER_FLAG_END_OF_STREAM) {
+ Log.i(TAG, "Saw output EOS");
+ }
+ }
+ }
+ 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 durationUs Duration of the clip in microseconds
+ */
+ public void dumpStatistics(String inputReference, long durationUs) {
+ String operation = "decode";
+ mStats.dumpStatistics(operation, inputReference, durationUs);
+ }
+
+ /**
+ * Resets the stats
+ */
+ public void resetDecoder() { mStats.reset(); }
+
+ 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++;
+ if (bufInfo.flags == MediaCodec.BUFFER_FLAG_END_OF_STREAM) {
+ mSawInputEOS = true;
+ 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);
+ }
+}
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..03db294
--- /dev/null
+++ b/media/tests/benchmark/MediaBenchmarkTest/src/main/java/com/android/media/benchmark/library/Encoder.java
@@ -0,0 +1,349 @@
+/*
+ * 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 {
+ private static final int ENCODE_DEFAULT_MAX_INPUT_SIZE = 3840;
+ 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 = 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);
+ }
+
+ private void onInputAvailable(MediaCodec mediaCodec, int inputBufferId) throws IOException {
+ if (mSawOutputEOS || inputBufferId < 0) {
+ if (mSawOutputEOS) {
+ 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 bytesRead = mFrameSize;
+ if (mInputBufferSize - mOffset < mFrameSize) {
+ bytesRead = (int) (mInputBufferSize - mOffset);
+ }
+ if (bufSize < bytesRead) {
+ mSignalledError = true;
+ return;
+ }
+ byte[] inputArray = new byte[bytesRead];
+ mInputStream.read(inputArray, 0, bytesRead);
+ inputBuffer.put(inputArray);
+ int flag = 0;
+ if (mNumInputFrame >= mNumFrames - 1 || bytesRead == 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, bytesRead, presentationTimeUs, flag);
+ mNumInputFrame++;
+ mOffset += bytesRead;
+ }
+
+ /**
+ * 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 encode
+ * @param durationUs Duration of the clip in microseconds
+ */
+ public void dumpStatistics(String inputReference, long durationUs) {
+ String operation = "encode";
+ mStats.dumpStatistics(operation, inputReference, durationUs);
+ }
+
+ /**
+ * 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..459e2a9
--- /dev/null
+++ b/media/tests/benchmark/MediaBenchmarkTest/src/main/java/com/android/media/benchmark/library/Extractor.java
@@ -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.
+ */
+
+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
+ */
+ public void dumpStatistics(String inputReference) {
+ String operation = "extract";
+ mStats.dumpStatistics(operation, inputReference, mDurationUs);
+ }
+}
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..49eaa1c
--- /dev/null
+++ b/media/tests/benchmark/MediaBenchmarkTest/src/main/java/com/android/media/benchmark/library/Muxer.java
@@ -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.
+ */
+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 clipDuration Duration of the given inputReference file
+ */
+ public void dumpStatistics(String inputReference, long clipDuration) {
+ String operation = "mux";
+ mStats.dumpStatistics(operation, inputReference, clipDuration);
+ }
+}
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..18ab5be
--- /dev/null
+++ b/media/tests/benchmark/MediaBenchmarkTest/src/main/java/com/android/media/benchmark/library/Stats.java
@@ -0,0 +1,138 @@
+/*
+ * 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.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;
+ }
+
+ /**
+ * Dumps the stats of the operation for a given input media.
+ * <p>
+ * \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.
+ */
+ public void dumpStatistics(String operation, String inputReference, long durationUs) {
+ 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;
+ }
+ }
+ // Print the Stats
+ Log.i(TAG, "Input Reference : " + inputReference);
+ Log.i(TAG, "Setup Time in nano sec : " + mInitTimeNs);
+ Log.i(TAG, "Average Time in nano sec : " + totalTimeTakenNs / mOutputTimer.size());
+ Log.i(TAG, "Time to first frame in nano sec : " + timeToFirstFrameNs);
+ Log.i(TAG, "Time taken (in nano sec) to " + operation + " 1 sec of content : " +
+ timeTakenPerSec);
+ Log.i(TAG, "Total bytes " + operation + "ed : " + size);
+ Log.i(TAG, "Number of bytes " + operation + "ed per second : " +
+ (size * 1000000000) / totalTimeTakenNs);
+ Log.i(TAG, "Minimum Time in nano sec : " + minTimeTakenNs);
+ Log.i(TAG, "Maximum Time in nano sec : " + maxTimeTakenNs);
+ Log.i(TAG, "Destroy Time in nano sec : " + mDeInitTimeNs);
+ }
+}
\ No newline at end of file
diff --git a/media/tests/benchmark/README.md b/media/tests/benchmark/README.md
index 8db3fd3..487ddb8 100644
--- a/media/tests/benchmark/README.md
+++ b/media/tests/benchmark/README.md
@@ -1,10 +1,19 @@
# 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/
@@ -13,20 +22,25 @@
To run the binary, follow the commands mentioned below under each module.
-The resource files for the tests are taken from [here](https://drive.google.com/open?id=1ghMr17BBJ7n0pqbm7oREiTN_MNemJUqy)
+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.
-Push the resource files to /sdcard/res on the device.
-
-You can use a different location, but you have to modify the rest of the instructions to replace /sdcard/res with wherever you chose to put the files.
+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 /sdcard/res/
+adb shell /data/local/tmp/extractorTest -P /data/local/tmp/MediaBenchmark/res/
```
## Decoder
@@ -36,7 +50,7 @@
Setup steps are same as extractor.
```
-adb shell /data/local/tmp/decoderTest -P /sdcard/res/
+adb shell /data/local/tmp/decoderTest -P /data/local/tmp/MediaBenchmark/res/
```
## Muxer
@@ -46,7 +60,7 @@
Setup steps are same as extractor.
```
-adb shell /data/local/tmp/muxerTest -P /sdcard/res/
+adb shell /data/local/tmp/muxerTest -P /data/local/tmp/MediaBenchmark/res/
```
## Encoder
@@ -56,5 +70,53 @@
Setup steps are same as extractor.
```
-adb shell /data/local/tmp/encoderTest -P /sdcard/res/
+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/MediaBenchmarkApp/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/MediaBenchmarkApp/arm64/MediaBenchmarkApp.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
+
```
diff --git a/media/tests/benchmark/src/native/common/Android.bp b/media/tests/benchmark/src/native/common/Android.bp
index 527f588..1da0102 100644
--- a/media/tests/benchmark/src/native/common/Android.bp
+++ b/media/tests/benchmark/src/native/common/Android.bp
@@ -15,10 +15,10 @@
*/
cc_library_static {
- name: "libbenchmark_common",
+ name: "libmediabenchmark_common",
defaults: [
- "libbenchmark-defaults",
- "libbenchmark_soft_sanitize_all-defaults",
+ "libmediabenchmark-defaults",
+ "libmediabenchmark_soft_sanitize_all-defaults",
],
srcs: [
@@ -32,23 +32,19 @@
}
cc_defaults {
- name: "libbenchmark_common-defaults",
+ name: "libmediabenchmark_common-defaults",
defaults: [
- "libbenchmark-defaults",
+ "libmediabenchmark-defaults",
],
static_libs: [
- "libbenchmark_common",
+ "libmediabenchmark_common",
],
}
cc_defaults {
- name: "libbenchmark-defaults",
-
- header_libs: [
- "media_ndk_headers",
- ],
+ name: "libmediabenchmark-defaults",
shared_libs: [
"libmediandk",
@@ -65,7 +61,7 @@
// public dependency for native implementation
// to be used by code under media/benchmark/* only
cc_defaults {
- name: "libbenchmark_soft_sanitize_all-defaults",
+ name: "libmediabenchmark_soft_sanitize_all-defaults",
sanitize: {
misc_undefined: [
@@ -73,6 +69,5 @@
"signed-integer-overflow",
],
cfi: true,
- address: true,
}
}
diff --git a/media/tests/benchmark/src/native/decoder/Android.bp b/media/tests/benchmark/src/native/decoder/Android.bp
index f2d3db5..b6286d4 100644
--- a/media/tests/benchmark/src/native/decoder/Android.bp
+++ b/media/tests/benchmark/src/native/decoder/Android.bp
@@ -15,15 +15,15 @@
*/
cc_library_static {
- name: "libbenchmark_decoder",
+ name: "libmediabenchmark_decoder",
defaults: [
- "libbenchmark_common-defaults",
- "libbenchmark_soft_sanitize_all-defaults",
+ "libmediabenchmark_common-defaults",
+ "libmediabenchmark_soft_sanitize_all-defaults",
],
srcs: ["Decoder.cpp"],
- static_libs: ["libbenchmark_extractor"],
+ static_libs: ["libmediabenchmark_extractor"],
export_include_dirs: ["."],
diff --git a/media/tests/benchmark/src/native/encoder/Android.bp b/media/tests/benchmark/src/native/encoder/Android.bp
index c14c319..239f378 100644
--- a/media/tests/benchmark/src/native/encoder/Android.bp
+++ b/media/tests/benchmark/src/native/encoder/Android.bp
@@ -15,16 +15,16 @@
*/
cc_library_static {
- name: "libbenchmark_encoder",
+ name: "libmediabenchmark_encoder",
defaults: [
- "libbenchmark_common-defaults",
- "libbenchmark_soft_sanitize_all-defaults",
+ "libmediabenchmark_common-defaults",
+ "libmediabenchmark_soft_sanitize_all-defaults",
],
srcs: ["Encoder.cpp"],
- static_libs: ["libbenchmark_extractor",
- "libbenchmark_decoder",
+ static_libs: ["libmediabenchmark_extractor",
+ "libmediabenchmark_decoder",
],
export_include_dirs: ["."],
diff --git a/media/tests/benchmark/src/native/extractor/Android.bp b/media/tests/benchmark/src/native/extractor/Android.bp
index 2fbe4e8..dfd0d49 100644
--- a/media/tests/benchmark/src/native/extractor/Android.bp
+++ b/media/tests/benchmark/src/native/extractor/Android.bp
@@ -15,10 +15,10 @@
*/
cc_library_static {
- name: "libbenchmark_extractor",
+ name: "libmediabenchmark_extractor",
defaults: [
- "libbenchmark_common-defaults",
- "libbenchmark_soft_sanitize_all-defaults",
+ "libmediabenchmark_common-defaults",
+ "libmediabenchmark_soft_sanitize_all-defaults",
],
srcs: ["Extractor.cpp"],
diff --git a/media/tests/benchmark/src/native/muxer/Android.bp b/media/tests/benchmark/src/native/muxer/Android.bp
index 6ef2a2e..f669d4a 100644
--- a/media/tests/benchmark/src/native/muxer/Android.bp
+++ b/media/tests/benchmark/src/native/muxer/Android.bp
@@ -15,15 +15,15 @@
*/
cc_library_static {
- name: "libbenchmark_muxer",
+ name: "libmediabenchmark_muxer",
defaults: [
- "libbenchmark_common-defaults",
- "libbenchmark_soft_sanitize_all-defaults",
+ "libmediabenchmark_common-defaults",
+ "libmediabenchmark_soft_sanitize_all-defaults",
],
srcs: ["Muxer.cpp"],
- static_libs: ["libbenchmark_extractor"],
+ static_libs: ["libmediabenchmark_extractor"],
export_include_dirs: ["."],
diff --git a/media/tests/benchmark/tests/Android.bp b/media/tests/benchmark/tests/Android.bp
index fc21ef7..24fd68c 100644
--- a/media/tests/benchmark/tests/Android.bp
+++ b/media/tests/benchmark/tests/Android.bp
@@ -18,28 +18,28 @@
name: "extractorTest",
gtest: true,
defaults: [
- "libbenchmark_common-defaults",
- "libbenchmark_soft_sanitize_all-defaults",
+ "libmediabenchmark_common-defaults",
+ "libmediabenchmark_soft_sanitize_all-defaults",
],
srcs: ["ExtractorTest.cpp"],
- static_libs: ["libbenchmark_extractor"]
+ static_libs: ["libmediabenchmark_extractor"]
}
cc_test {
name: "decoderTest",
gtest: true,
defaults: [
- "libbenchmark_common-defaults",
- "libbenchmark_soft_sanitize_all-defaults",
+ "libmediabenchmark_common-defaults",
+ "libmediabenchmark_soft_sanitize_all-defaults",
],
srcs: ["DecoderTest.cpp"],
static_libs: [
- "libbenchmark_extractor",
- "libbenchmark_decoder",
+ "libmediabenchmark_extractor",
+ "libmediabenchmark_decoder",
],
}
@@ -47,15 +47,15 @@
name: "muxerTest",
gtest: true,
defaults: [
- "libbenchmark_common-defaults",
- "libbenchmark_soft_sanitize_all-defaults",
+ "libmediabenchmark_common-defaults",
+ "libmediabenchmark_soft_sanitize_all-defaults",
],
srcs: ["MuxerTest.cpp"],
static_libs: [
- "libbenchmark_extractor",
- "libbenchmark_muxer",
+ "libmediabenchmark_extractor",
+ "libmediabenchmark_muxer",
],
}
@@ -63,15 +63,15 @@
name: "encoderTest",
gtest: true,
defaults: [
- "libbenchmark_common-defaults",
- "libbenchmark_soft_sanitize_all-defaults",
+ "libmediabenchmark_common-defaults",
+ "libmediabenchmark_soft_sanitize_all-defaults",
],
srcs: ["EncoderTest.cpp"],
static_libs: [
- "libbenchmark_extractor",
- "libbenchmark_decoder",
- "libbenchmark_encoder",
+ "libmediabenchmark_extractor",
+ "libmediabenchmark_decoder",
+ "libmediabenchmark_encoder",
],
}
diff --git a/media/tests/benchmark/tests/DecoderTest.cpp b/media/tests/benchmark/tests/DecoderTest.cpp
index 6cb42d6..242178f 100644
--- a/media/tests/benchmark/tests/DecoderTest.cpp
+++ b/media/tests/benchmark/tests/DecoderTest.cpp
@@ -72,24 +72,7 @@
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
- if (inputBufferOffset + info.size > kMaxBufferSize) {
- cout << "[ WARN ] Test Skipped. Memory allocated not sufficient\n";
- free(inputBuffer);
- return;
- }
- memcpy(inputBuffer + inputBufferOffset, csdBuffer, info.size);
- frameInfo.push_back(info);
- inputBufferOffset += info.size;
- idx++;
- }
// Get frame data
while (1) {
status = extractor->getFrameSample(info);
@@ -135,6 +118,7 @@
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.mp4", "", 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(
@@ -144,6 +128,7 @@
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.mp4", "", 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,
diff --git a/media/tests/benchmark/tests/EncoderTest.cpp b/media/tests/benchmark/tests/EncoderTest.cpp
index 574083d..9f42c64 100644
--- a/media/tests/benchmark/tests/EncoderTest.cpp
+++ b/media/tests/benchmark/tests/EncoderTest.cpp
@@ -72,24 +72,6 @@
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
- if (inputBufferOffset + info.size > kMaxBufferSize) {
- cout << "[ WARN ] Test Skipped. Memory allocated not sufficient\n";
- free(inputBuffer);
- return;
- }
- memcpy(inputBuffer + inputBufferOffset, csdBuffer, info.size);
- frameInfo.push_back(info);
- inputBufferOffset += info.size;
- idx++;
- }
// Get frame data
while (1) {
@@ -189,6 +171,7 @@
::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(
@@ -196,6 +179,7 @@
::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,
diff --git a/media/utils/Android.bp b/media/utils/Android.bp
index 2dbc476..63437da 100644
--- a/media/utils/Android.bp
+++ b/media/utils/Android.bp
@@ -47,6 +47,7 @@
header_libs: [
"bionic_libc_platform_headers",
+ "libmedia_headers",
],
local_include_dirs: ["include"],
diff --git a/media/utils/TimeCheck.cpp b/media/utils/TimeCheck.cpp
index 59cf4ef..265a232 100644
--- a/media/utils/TimeCheck.cpp
+++ b/media/utils/TimeCheck.cpp
@@ -16,8 +16,8 @@
#include <utils/Log.h>
-#include <media/TimeCheck.h>
-#include <media/EventLog.h>
+#include <mediautils/TimeCheck.h>
+#include <mediautils/EventLog.h>
namespace android {
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 0b745ac..9b0872e 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -810,7 +810,33 @@
continue;
}
- size_t frameCount = std::lcm(thread->frameCount(), secondaryThread->frameCount());
+ size_t sourceFrameCount = thread->frameCount() * output.sampleRate
+ / thread->sampleRate();
+ size_t sinkFrameCount = secondaryThread->frameCount() * output.sampleRate
+ / secondaryThread->sampleRate();
+ // If the secondary output has just been opened, the first secondaryThread write
+ // will not block as it will fill the empty startup buffer of the HAL,
+ // so a second sink buffer needs to be ready for the immediate next blocking write.
+ // Additionally, have a margin of one main thread buffer as the scheduling jitter
+ // can reorder the writes (eg if thread A&B have the same write intervale,
+ // the scheduler could schedule AB...BA)
+ size_t frameCountToBeReady = 2 * sinkFrameCount + sourceFrameCount;
+ // Total secondary output buffer must be at least as the read frames plus
+ // the margin of a few buffers on both sides in case the
+ // threads scheduling has some jitter.
+ // That value should not impact latency as the secondary track is started before
+ // its buffer is full, see frameCountToBeReady.
+ size_t frameCount = frameCountToBeReady + 2 * (sourceFrameCount + sinkFrameCount);
+ // The frameCount should also not be smaller than the secondary thread min frame
+ // count
+ size_t minFrameCount = AudioSystem::calculateMinFrameCount(
+ [&] { Mutex::Autolock _l(secondaryThread->mLock);
+ return secondaryThread->latency_l(); }(),
+ secondaryThread->mNormalFrameCount,
+ secondaryThread->mSampleRate,
+ output.sampleRate,
+ input.speed);
+ frameCount = std::max(frameCount, minFrameCount);
using namespace std::chrono_literals;
auto inChannelMask = audio_channel_mask_out_to_in(input.config.channel_mask);
@@ -843,7 +869,8 @@
patchRecord->buffer(),
patchRecord->bufferSize(),
outputFlags,
- 0ns /* timeout */);
+ 0ns /* timeout */,
+ frameCountToBeReady);
status = patchTrack->initCheck();
if (status != NO_ERROR) {
ALOGE("Secondary output patchTrack init failed: %d", status);
@@ -1357,8 +1384,8 @@
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),
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 72e669a..d639f26 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -547,6 +547,16 @@
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
@@ -749,7 +759,7 @@
// 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 +768,13 @@
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 {
diff --git a/services/audioflinger/Effects.cpp b/services/audioflinger/Effects.cpp
index 3c4fbba..13152d0 100644
--- a/services/audioflinger/Effects.cpp
+++ b/services/audioflinger/Effects.cpp
@@ -24,6 +24,7 @@
#include "Configuration.h"
#include <utils/Log.h>
#include <system/audio_effects/effect_aec.h>
+#include <system/audio_effects/effect_dynamicsprocessing.h>
#include <system/audio_effects/effect_ns.h>
#include <system/audio_effects/effect_visualizer.h>
#include <audio_utils/channels.h>
@@ -2569,7 +2570,8 @@
if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) &&
(((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) ||
(memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) ||
- (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) {
+ (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0) ||
+ (memcmp(&desc.type, SL_IID_DYNAMICSPROCESSING, sizeof(effect_uuid_t)) == 0))) {
return false;
}
return true;
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..18cb53b 100644
--- a/services/audioflinger/PatchPanel.cpp
+++ b/services/audioflinger/PatchPanel.cpp
@@ -483,19 +483,6 @@
// 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 +499,34 @@
outputFlags = (audio_output_flags_t) (outputFlags & ~AUDIO_OUTPUT_FLAG_FAST);
}
+ sp<RecordThread::PatchRecord> tempRecordTrack;
+ if ((inputFlags & AUDIO_INPUT_FLAG_DIRECT) && (outputFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
+ tempRecordTrack = new RecordThread::PassthruPatchRecord(
+ mRecord.thread().get(),
+ sampleRate,
+ inChannelMask,
+ format,
+ frameCount,
+ inputFlags);
+ } else {
+ 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,
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index a093893..1ff03c4 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -74,7 +74,10 @@
uid_t uid,
audio_output_flags_t flags,
track_type type,
- audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE);
+ audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE,
+ /** default behaviour is to start when there are as many frames
+ * ready as possible (aka. Buffer is full). */
+ size_t frameCountToBeReady = SIZE_MAX);
virtual ~Track();
virtual status_t initCheck() const;
@@ -263,11 +266,11 @@
};
sp<AudioVibrationController> mAudioVibrationController;
sp<os::ExternalVibration> mExternalVibration;
+ /** How many frames should be in the buffer before the track is considered ready */
+ const size_t mFrameCountToBeReady;
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); }
@@ -384,9 +387,15 @@
void *buffer,
size_t bufferSize,
audio_output_flags_t flags,
- const Timeout& timeout = {});
+ const Timeout& timeout = {},
+ size_t frameCountToBeReady = 1 /** Default behaviour is to start
+ * as soon as possible to have
+ * the lowest possible latency
+ * 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);
@@ -402,5 +411,4 @@
private:
void restartIfDisabled();
-
}; // end of PatchTrack
diff --git a/services/audioflinger/RecordTracks.h b/services/audioflinger/RecordTracks.h
index 08660dd..da05dac 100644
--- a/services/audioflinger/RecordTracks.h
+++ b/services/audioflinger/RecordTracks.h
@@ -128,6 +128,8 @@
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);
@@ -136,4 +138,71 @@
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 2e6037b0..73292d3 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -2953,9 +2953,11 @@
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))) {
@@ -3956,6 +3958,32 @@
return INVALID_OPERATION;
}
+// For dedicated VoIP outputs, let the HAL apply the stream volume. Track volume is
+// still applied by the mixer.
+// All tracks attached to a mixer with flag VOIP_RX are tied to the same
+// stream type STREAM_VOICE_CALL so this will only change the HAL volume once even
+// if more than one track are active
+status_t AudioFlinger::PlaybackThread::handleVoipVolume_l(float *volume)
+{
+ status_t result = NO_ERROR;
+ if ((mOutput->flags & AUDIO_OUTPUT_FLAG_VOIP_RX) != 0) {
+ if (*volume != mLeftVolFloat) {
+ result = mOutput->stream->setVolume(*volume, *volume);
+ ALOGE_IF(result != OK,
+ "Error when setting output stream volume: %d", result);
+ if (result == NO_ERROR) {
+ mLeftVolFloat = *volume;
+ }
+ }
+ // if stream volume was successfully sent to the HAL, mLeftVolFloat == v here and we
+ // remove stream volume contribution from software volume.
+ if (mLeftVolFloat == *volume) {
+ *volume = 1.0f;
+ }
+ }
+ return result;
+}
+
status_t AudioFlinger::MixerThread::createAudioPatch_l(const struct audio_patch *patch,
audio_patch_handle_t *handle)
{
@@ -4758,22 +4786,25 @@
// no acknowledgement required for newly active tracks
}
sp<AudioTrackServerProxy> proxy = track->mAudioTrackServerProxy;
+ float volume;
+ if (track->isPlaybackRestricted() || mStreamTypes[track->streamType()].mute) {
+ volume = 0.f;
+ } else {
+ volume = masterVolume * mStreamTypes[track->streamType()].volume;
+ }
+
+ handleVoipVolume_l(&volume);
+
// cache the combined master volume and stream type volume for fast mixer; this
// lacks any synchronization or barrier so VolumeProvider may read a stale value
const float vh = track->getVolumeHandler()->getVolume(
- proxy->framesReleased()).first;
- float volume;
- if (track->isPlaybackRestricted()) {
- volume = 0.f;
- } else {
- volume = masterVolume
- * mStreamTypes[track->streamType()].volume
- * vh;
- }
+ proxy->framesReleased()).first;
+ volume *= vh;
track->mCachedVolume = volume;
gain_minifloat_packed_t vlr = proxy->getVolumeLR();
float vlf = volume * float_from_gain(gain_minifloat_unpack_left(vlr));
float vrf = volume * float_from_gain(gain_minifloat_unpack_right(vlr));
+
track->setFinalVolume((vlf + vrf) / 2.f);
++fastTracks;
} else {
@@ -4916,20 +4947,22 @@
uint32_t vl, vr; // in U8.24 integer format
float vlf, vrf, vaf; // in [0.0, 1.0] float format
// read original volumes with volume control
- float typeVolume = mStreamTypes[track->streamType()].volume;
- float v = masterVolume * typeVolume;
+ float v = masterVolume * mStreamTypes[track->streamType()].volume;
// Always fetch volumeshaper volume to ensure state is updated.
const sp<AudioTrackServerProxy> proxy = track->mAudioTrackServerProxy;
const float vh = track->getVolumeHandler()->getVolume(
track->mAudioTrackServerProxy->framesReleased()).first;
- if (track->isPausing() || mStreamTypes[track->streamType()].mute
- || track->isPlaybackRestricted()) {
+ if (mStreamTypes[track->streamType()].mute || track->isPlaybackRestricted()) {
+ v = 0;
+ }
+
+ handleVoipVolume_l(&v);
+
+ if (track->isPausing()) {
vl = vr = 0;
vlf = vrf = vaf = 0.;
- if (track->isPausing()) {
- track->setPaused();
- }
+ track->setPaused();
} else {
gain_minifloat_packed_t vlr = proxy->getVolumeLR();
vlf = float_from_gain(gain_minifloat_unpack_left(vlr));
@@ -4981,25 +5014,6 @@
track->mHasVolumeController = false;
}
- // For dedicated VoIP outputs, let the HAL apply the stream volume. Track volume is
- // still applied by the mixer.
- if ((mOutput->flags & AUDIO_OUTPUT_FLAG_VOIP_RX) != 0) {
- v = mStreamTypes[track->streamType()].mute ? 0.0f : v;
- if (v != mLeftVolFloat) {
- status_t result = mOutput->stream->setVolume(v, v);
- ALOGE_IF(result != OK, "Error when setting output stream volume: %d", result);
- if (result == OK) {
- mLeftVolFloat = v;
- }
- }
- // if stream volume was successfully sent to the HAL, mLeftVolFloat == v here and we
- // remove stream volume contribution from software volume.
- if (v != 0.0f && mLeftVolFloat == v) {
- vlf = min(1.0f, vlf / v);
- vrf = min(1.0f, vrf / v);
- vaf = min(1.0f, vaf / v);
- }
- }
// XXX: these things DON'T need to be done each time
mAudioMixer->setBufferProvider(trackId, track);
mAudioMixer->enable(trackId);
@@ -5289,11 +5303,11 @@
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;
}
@@ -5646,10 +5660,17 @@
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;
@@ -5726,7 +5747,7 @@
// 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
@@ -5734,7 +5755,7 @@
} 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;
@@ -6658,6 +6679,7 @@
) :
ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD, systemReady),
mInput(input),
+ mSource(mInput),
mActiveTracks(&this->mLocalLog),
mRsmpInBuffer(NULL),
// mRsmpInFrames, mRsmpInFramesP2, and mRsmpInFramesOA are set by readInputParameters_l()
@@ -7110,7 +7132,7 @@
} 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) {
@@ -7132,7 +7154,7 @@
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);
@@ -7413,7 +7435,7 @@
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.
@@ -8398,11 +8420,17 @@
{
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);
}
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 336c2b4..acb1370 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -747,6 +747,7 @@
// is safe to do so. That will drop the final ref count and destroy the tracks.
virtual mixer_state prepareTracks_l(Vector< sp<Track> > *tracksToRemove) = 0;
void removeTracks_l(const Vector< sp<Track> >& tracksToRemove);
+ status_t handleVoipVolume_l(float *volume);
// StreamOutHalInterfaceCallback implementation
virtual void onWriteReady();
@@ -1645,6 +1646,7 @@
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
diff --git a/services/audioflinger/TrackBase.h b/services/audioflinger/TrackBase.h
index 8f720b5..7a3bb0d 100644
--- a/services/audioflinger/TrackBase.h
+++ b/services/audioflinger/TrackBase.h
@@ -325,6 +325,7 @@
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;
@@ -347,6 +348,8 @@
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 78db80c..16a8a84 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>
@@ -511,7 +513,8 @@
uid_t uid,
audio_output_flags_t flags,
track_type type,
- audio_port_handle_t portId)
+ audio_port_handle_t portId,
+ size_t frameCountToBeReady)
: TrackBase(thread, client, attr, sampleRate, format, channelMask, frameCount,
(sharedBuffer != 0) ? sharedBuffer->pointer() : buffer,
(sharedBuffer != 0) ? sharedBuffer->size() : bufferSize,
@@ -530,6 +533,7 @@
mVolumeHandler(new media::VolumeHandler(sampleRate)),
mOpPlayAudioMonitor(OpPlayAudioMonitor::createIfNeeded(uid, attr, id(), streamType)),
// mSinkTimestamp
+ mFrameCountToBeReady(frameCountToBeReady),
mFastIndex(-1),
mCachedVolume(1.0),
/* The track might not play immediately after being active, similarly as if its volume was 0.
@@ -820,16 +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);
@@ -837,30 +834,10 @@
auto spent = ceil<std::chrono::microseconds>(std::chrono::steady_clock::now() - start);
using namespace std::chrono_literals;
// Average is ~20us per track, this should virtually never be logged (Logging takes >200us)
- ALOGD_IF(spent > 200us, "%s: took %lldus to intercept %zu tracks", __func__,
+ ALOGD_IF(spent > 500us, "%s: took %lldus to intercept %zu tracks", __func__,
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
@@ -910,8 +887,12 @@
return true;
}
- if (framesReady() >= mServerProxy->getBufferSizeInFrames() ||
- (mCblk->mFlags & CBLK_FORCEREADY)) {
+ size_t bufferSizeInFrames = mServerProxy->getBufferSizeInFrames();
+ size_t framesToBeReady = std::min(mFrameCountToBeReady, bufferSizeInFrames);
+
+ if (framesReady() >= framesToBeReady || (mCblk->mFlags & CBLK_FORCEREADY)) {
+ ALOGV("%s(%d): consider track ready with %zu/%zu, target was %zu)",
+ __func__, mId, framesReady(), bufferSizeInFrames, framesToBeReady);
mFillingUpStatus = FS_FILLED;
android_atomic_and(~CBLK_FORCEREADY, &mCblk->mFlags);
return true;
@@ -1413,6 +1394,7 @@
void AudioFlinger::PlaybackThread::Track::disable()
{
+ // TODO(b/142394888): the filling status should also be reset to filling
signalClientFlag(CBLK_DISABLED);
}
@@ -1790,12 +1772,14 @@
void *buffer,
size_t bufferSize,
audio_output_flags_t flags,
- const Timeout& timeout)
+ const Timeout& timeout,
+ size_t frameCountToBeReady)
: Track(playbackThread, NULL, streamType,
audio_attributes_t{} /* currently unused for patch track */,
sampleRate, format, channelMask, frameCount,
buffer, bufferSize, nullptr /* sharedBuffer */,
- AUDIO_SESSION_NONE, getpid(), AID_AUDIOSERVER, flags, TYPE_PATCH),
+ AUDIO_SESSION_NONE, getpid(), AID_AUDIOSERVER, flags, TYPE_PATCH,
+ AUDIO_PORT_HANDLE_NONE, frameCountToBeReady),
PatchTrackBase(new ClientProxy(mCblk, mBuffer, frameCount, mFrameSize, true, true),
*playbackThread, timeout)
{
@@ -1810,6 +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)
{
@@ -1828,9 +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;
}
@@ -1869,7 +1872,6 @@
{
mProxy->releaseBuffer(buffer);
restartIfDisabled();
- android_atomic_or(CBLK_FORCEREADY, &mCblk->mFlags);
}
void AudioFlinger::PlaybackThread::PatchTrack::restartIfDisabled()
@@ -2283,6 +2285,39 @@
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)
@@ -2294,6 +2329,11 @@
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;
}
@@ -2322,6 +2362,180 @@
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/common/managerdefinitions/include/AudioPolicyConfig.h b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h
index 2264d8f..0776a8d 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h
@@ -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;
@@ -108,6 +117,7 @@
void setDefault(void)
{
mSource = "AudioPolicyConfig::setDefault";
+ mEngineLibraryNameSuffix = kDefaultEngineLibraryNameSuffix;
mDefaultOutputDevice = new DeviceDescriptor(AUDIO_DEVICE_OUT_SPEAKER);
mDefaultOutputDevice->addAudioProfile(AudioProfile::createFullDynamic());
sp<DeviceDescriptor> defaultInputDevice = new DeviceDescriptor(AUDIO_DEVICE_IN_BUILTIN_MIC);
@@ -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/src/HwModule.cpp b/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
index 96a8337..1f9b725 100644
--- a/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
@@ -333,9 +333,10 @@
if (encodedFormat != AUDIO_FORMAT_DEFAULT) {
moduleDevice->setEncodedFormat(encodedFormat);
}
- moduleDevice->setAddress(devAddress);
if (allowToCreate) {
moduleDevice->attach(hwModule);
+ moduleDevice->setAddress(devAddress);
+ moduleDevice->setName(String8(name));
}
return moduleDevice;
}
diff --git a/services/audiopolicy/config/Android.bp b/services/audiopolicy/config/Android.bp
new file mode 100644
index 0000000..4b5e788
--- /dev/null
+++ b/services/audiopolicy/config/Android.bp
@@ -0,0 +1,101 @@
+/*
+ * 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: "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/engine/common/include/EngineBase.h b/services/audiopolicy/engine/common/include/EngineBase.h
index cedc78f..fca9a60 100644
--- a/services/audiopolicy/engine/common/include/EngineBase.h
+++ b/services/audiopolicy/engine/common/include/EngineBase.h
@@ -17,18 +17,18 @@
#pragma once
#include <EngineConfig.h>
-#include <AudioPolicyManagerInterface.h>
+#include <EngineInterface.h>
#include <ProductStrategy.h>
#include <VolumeGroup.h>
namespace android {
namespace audio_policy {
-class EngineBase : public AudioPolicyManagerInterface
+class EngineBase : public EngineInterface
{
public:
///
- /// from AudioPolicyManagerInterface
+ /// from EngineInterface
///
android::status_t initCheck() override;
diff --git a/services/audiopolicy/engine/common/include/ProductStrategy.h b/services/audiopolicy/engine/common/include/ProductStrategy.h
index 1a2a198..c538f52 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 +26,7 @@
#include <map>
#include <utils/Errors.h>
#include <utils/String8.h>
+#include <media/AudioAttributes.h>
namespace android {
diff --git a/services/audiopolicy/engine/common/include/VolumeCurve.h b/services/audiopolicy/engine/common/include/VolumeCurve.h
index 54314e3..d3d0904 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>
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..840eb34 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;
}
diff --git a/services/audiopolicy/engine/common/src/ProductStrategy.cpp b/services/audiopolicy/engine/common/src/ProductStrategy.cpp
index f74f190..ac3e462 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>
diff --git a/services/audiopolicy/engine/config/src/EngineConfig.cpp b/services/audiopolicy/engine/config/src/EngineConfig.cpp
index 1ad7739..d47fbd2 100644
--- a/services/audiopolicy/engine/config/src/EngineConfig.cpp
+++ b/services/audiopolicy/engine/config/src/EngineConfig.cpp
@@ -32,9 +32,9 @@
#include <istream>
#include <cstdint>
+#include <stdarg.h>
#include <string>
-
namespace android {
using utilities::convertTo;
@@ -603,7 +603,39 @@
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 +673,7 @@
}
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/AudioPolicyManagerInterface.h b/services/audiopolicy/engine/interface/EngineInterface.h
similarity index 97%
rename from services/audiopolicy/engine/interface/AudioPolicyManagerInterface.h
rename to services/audiopolicy/engine/interface/EngineInterface.h
index b7fd031..0c58a7c 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:
/**
@@ -295,7 +295,13 @@
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/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/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/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/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..a0b874a
--- /dev/null
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/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.
+ */
+
+// 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: ":PolicySubsystem-CommonTypes",
+ sub_dir: "parameter-framework/Structure/Policy",
+}
+
+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: "PolicySubsystem-CommonTypes",
+ srcs: ["examples/common/Structure/PolicySubsystem-CommonTypes.xml"],
+}
+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..5078268
--- /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",
+ ":PolicySubsystem-CommonTypes",
+ ":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/CarEmu/Android.bp b/services/audiopolicy/engineconfigurable/parameter-framework/examples/CarEmu/Android.bp
new file mode 100644
index 0000000..0917440
--- /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",
+ ":PolicySubsystem-CommonTypes",
+ ":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/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..11e220b
--- /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",
+ ":PolicySubsystem-CommonTypes",
+ ":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/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/SettingsNoInput/Android.bp b/services/audiopolicy/engineconfigurable/parameter-framework/examples/SettingsNoInput/Android.bp
new file mode 100644
index 0000000..ffd494e
--- /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",
+ ":PolicySubsystem-CommonTypes",
+ ],
+}
+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..6fca048
--- /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",
+ ":PolicySubsystem-CommonTypes",
+ ],
+}
+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/common/Structure/PolicySubsystem.xml b/services/audiopolicy/engineconfigurable/parameter-framework/examples/common/Structure/PolicySubsystem.xml
index b55ce2c..585ce87 100644
--- a/services/audiopolicy/engineconfigurable/parameter-framework/examples/common/Structure/PolicySubsystem.xml
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/examples/common/Structure/PolicySubsystem.xml
@@ -73,10 +73,13 @@
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..3987294 100644
--- a/services/audiopolicy/engineconfigurable/src/Engine.cpp
+++ b/services/audiopolicy/engineconfigurable/src/Engine.cpp
@@ -33,6 +33,8 @@
#include <AudioIODescriptorInterface.h>
#include <ParameterManagerWrapper.h>
+#include <media/TypeConverter.h>
+
using std::string;
using std::map;
@@ -244,9 +246,9 @@
}
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());
+ 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) {
// We do expect only one device for these types of devices
@@ -254,6 +256,14 @@
// 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);
+ auto busDevice = availableOutputDevices.getDevice(devices, address, AUDIO_FORMAT_DEFAULT);
+ if (busDevice == nullptr) {
+ ALOGE("%s:unavailable device 0x%x %s, fallback on default", __func__, devices,
+ address.c_str());
+ auto defaultDevice = getApmObserver()->getDefaultOutputDevice();
+ ALOG_ASSERT(defaultDevice != nullptr, "Default Output Device NOT available");
+ return DeviceVector(defaultDevice);
+ }
return DeviceVector(availableOutputDevices.getDevice(devices,
address,
AUDIO_FORMAT_DEFAULT));
@@ -361,7 +371,7 @@
}
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/tools/Android.bp b/services/audiopolicy/engineconfigurable/tools/Android.bp
index 8c16972..d9e97af 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 +117,19 @@
],
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"],
+}
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/enginedefault/Android.bp b/services/audiopolicy/enginedefault/Android.bp
index 7b42c6a..2b9cf09 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",
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/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
index 04170ac..c602f3a 100644
--- a/services/audiopolicy/enginedefault/src/Engine.cpp
+++ b/services/audiopolicy/enginedefault/src/Engine.cpp
@@ -762,12 +762,6 @@
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..62938cf 100644
--- a/services/audiopolicy/enginedefault/src/Engine.h
+++ b/services/audiopolicy/enginedefault/src/Engine.h
@@ -17,7 +17,7 @@
#pragma once
#include "EngineBase.h"
-#include "AudioPolicyManagerInterface.h"
+#include "EngineInterface.h"
#include <AudioGain.h>
#include <policy.h>
@@ -48,12 +48,9 @@
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;
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/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..8fbeff9
--- /dev/null
+++ b/services/audiopolicy/managerdefault/Android.bp
@@ -0,0 +1,43 @@
+cc_library_shared {
+ name: "libaudiopolicymanagerdefault",
+
+ srcs: [
+ "AudioPolicyManager.cpp",
+ "EngineLibrary.cpp",
+ ],
+
+ export_include_dirs: ["."],
+
+ shared_libs: [
+ "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 c5921c3..0000000
--- a/services/audiopolicy/managerdefault/Android.mk
+++ /dev/null
@@ -1,48 +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)
-
-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
-
-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 a8fd9cd..83ae35e 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -42,8 +42,6 @@
#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>
@@ -96,7 +94,7 @@
{
AudioParameter param(device->address());
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());
}
@@ -474,6 +472,10 @@
std::unordered_set<audio_format_t> formatSet;
sp<HwModule> primaryModule =
mHwModules.getModuleFromName(AUDIO_HARDWARE_MODULE_ID_PRIMARY);
+ if (primaryModule == nullptr) {
+ ALOGE("%s() unable to get primary module", __func__);
+ return NO_INIT;
+ }
DeviceVector declaredDevices = primaryModule->getDeclaredDevices().getDevicesFromTypeMask(
AUDIO_DEVICE_OUT_ALL_A2DP);
for (const auto& device : declaredDevices) {
@@ -838,7 +840,7 @@
// 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);
}
@@ -2238,16 +2240,22 @@
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)
@@ -2278,11 +2286,16 @@
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)
@@ -4293,7 +4306,6 @@
: AudioPolicyManager(clientInterface, false /*forTesting*/)
{
loadConfig();
- initialize();
}
void AudioPolicyManager::loadConfig() {
@@ -4304,17 +4316,18 @@
}
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();
@@ -5681,8 +5694,9 @@
const auto ringVolumeSrc = toVolumeSource(AUDIO_STREAM_RING);
const auto musicVolumeSrc = toVolumeSource(AUDIO_STREAM_MUSIC);
const auto alarmVolumeSrc = toVolumeSource(AUDIO_STREAM_ALARM);
+ const auto a11yVolumeSrc = toVolumeSource(AUDIO_STREAM_ACCESSIBILITY);
- if (volumeSource == toVolumeSource(AUDIO_STREAM_ACCESSIBILITY)
+ if (volumeSource == a11yVolumeSrc
&& (AUDIO_MODE_RINGTONE == mEngine->getPhoneState()) &&
mOutputs.isActive(ringVolumeSrc, 0)) {
auto &ringCurves = getVolumeCurves(AUDIO_STREAM_RING);
@@ -5699,7 +5713,7 @@
volumeSource == toVolumeSource(AUDIO_STREAM_NOTIFICATION) ||
volumeSource == toVolumeSource(AUDIO_STREAM_ENFORCED_AUDIBLE) ||
volumeSource == toVolumeSource(AUDIO_STREAM_DTMF) ||
- volumeSource == toVolumeSource(AUDIO_STREAM_ACCESSIBILITY))) {
+ volumeSource == a11yVolumeSrc)) {
auto &voiceCurves = getVolumeCurves(callVolumeSrc);
int voiceVolumeIndex = voiceCurves.getVolumeIndex(device);
const float maxVoiceVolDb =
@@ -5711,7 +5725,9 @@
// VOICE_CALL stream has minVolumeIndex > 0 : Users cannot set the volume of voice calls to
// 0. We don't want to cap volume when the system has programmatically muted the voice call
// stream. See setVolumeCurveIndex() for more information.
- bool exemptFromCapping = (volumeSource == ringVolumeSrc) && (voiceVolumeIndex == 0);
+ bool exemptFromCapping =
+ ((volumeSource == ringVolumeSrc) || (volumeSource == a11yVolumeSrc))
+ && (voiceVolumeIndex == 0);
ALOGV_IF(exemptFromCapping, "%s volume source %d at vol=%f not capped", __func__,
volumeSource, volumeDb);
if ((volumeDb > maxVoiceVolDb) && !exemptFromCapping) {
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
index 612bd8f..5f651cc 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -34,7 +34,6 @@
#include <media/PatchBuilder.h>
#include "AudioPolicyInterface.h"
-#include <AudioPolicyManagerInterface.h>
#include <AudioPolicyManagerObserver.h>
#include <AudioGain.h>
#include <AudioPolicyConfig.h>
@@ -49,6 +48,7 @@
#include <AudioPolicyMix.h>
#include <EffectDescriptor.h>
#include <SoundTriggerSession.h>
+#include "EngineLibrary.h"
#include "TypeConverter.h"
namespace android {
@@ -307,6 +307,8 @@
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 +323,6 @@
// - initialize.
AudioPolicyConfig& getConfig() { return mConfig; }
void loadConfig();
- status_t initialize();
// From AudioPolicyManagerObserver
virtual const AudioPatchCollection &getAudioPatches() const
@@ -752,7 +753,7 @@
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.
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/AudioPolicyService.cpp b/services/audiopolicy/service/AudioPolicyService.cpp
index 85ea94f..d1b59c1 100644
--- a/services/audiopolicy/service/AudioPolicyService.cpp
+++ b/services/audiopolicy/service/AudioPolicyService.cpp
@@ -409,12 +409,17 @@
// Another client in the same UID has already been allowed to capture
// OR The client is the assistant
// AND an accessibility service is on TOP or a RTT call is active
-// AND the source is VOICE_RECOGNITION or HOTWORD
-// OR uses VOICE_RECOGNITION AND is on TOP
-// OR uses HOTWORD
+// AND the source is VOICE_RECOGNITION or HOTWORD
+// OR uses VOICE_RECOGNITION AND is on TOP
+// OR uses HOTWORD
// AND there is no active privacy sensitive capture or call
// OR client has CAPTURE_AUDIO_OUTPUT privileged permission
// OR The client is an accessibility service
+// AND Is on TOP
+// AND the source is VOICE_RECOGNITION or HOTWORD
+// OR The assistant is not on TOP
+// AND there is no active privacy sensitive capture or call
+// OR client has CAPTURE_AUDIO_OUTPUT privileged permission
// AND is on TOP
// AND the source is VOICE_RECOGNITION or HOTWORD
// OR the client source is virtual (remote submix, call audio TX or RX...)
@@ -422,7 +427,7 @@
// AND The assistant is not on TOP
// AND is on TOP or latest started
// AND there is no active privacy sensitive capture or call
-// OR client has CAPTURE_AUDIO_OUTPUT privileged permission
+// OR client has CAPTURE_AUDIO_OUTPUT privileged permission
sp<AudioRecordClient> topActive;
sp<AudioRecordClient> latestActive;
@@ -458,16 +463,24 @@
continue;
}
- if (appState == APP_STATE_TOP) {
+ bool isAssistant = mUidPolicy->isAssistantUid(current->uid);
+ bool isAccessibility = mUidPolicy->isA11yUid(current->uid);
+ if (appState == APP_STATE_TOP && !isAccessibility) {
if (current->startTimeNs > topStartNs) {
topActive = current;
topStartNs = current->startTimeNs;
}
- if (mUidPolicy->isAssistantUid(current->uid)) {
+ if (isAssistant) {
isAssistantOnTop = true;
}
}
- if (current->startTimeNs > latestStartNs) {
+ // Assistant capturing for HOTWORD or Accessibility services not considered
+ // for latest active to avoid masking regular clients started before
+ if (current->startTimeNs > latestStartNs
+ && !((current->attributes.source == AUDIO_SOURCE_HOTWORD
+ || isA11yOnTop || rttCallActive)
+ && isAssistant)
+ && !isAccessibility) {
latestActive = current;
latestStartNs = current->startTimeNs;
}
@@ -540,10 +553,20 @@
} else if (mUidPolicy->isA11yUid(current->uid)) {
// For accessibility service allow capture if:
// Is on TOP
- // AND the source is VOICE_RECOGNITION or HOTWORD
- if (isA11yOnTop &&
- (source == AUDIO_SOURCE_VOICE_RECOGNITION || source == AUDIO_SOURCE_HOTWORD)) {
- allowCapture = true;
+ // AND the source is VOICE_RECOGNITION or HOTWORD
+ // Or
+ // The assistant is not on TOP
+ // AND there is no active privacy sensitive capture or call
+ // OR client has CAPTURE_AUDIO_OUTPUT privileged permission
+ if (isA11yOnTop) {
+ if (source == AUDIO_SOURCE_VOICE_RECOGNITION || source == AUDIO_SOURCE_HOTWORD) {
+ allowCapture = true;
+ }
+ } else {
+ if (!isAssistantOnTop
+ && (!(isSensitiveActive || isInCall) || current->canCaptureOutput)) {
+ allowCapture = true;
+ }
}
}
setAppState_l(current->uid,
diff --git a/services/audiopolicy/tests/audiopolicymanager_tests.cpp b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
index de5670c..e10a716 100644
--- a/services/audiopolicy/tests/audiopolicymanager_tests.cpp
+++ b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
@@ -30,7 +30,16 @@
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();
diff --git a/services/camera/libcameraservice/Android.bp b/services/camera/libcameraservice/Android.bp
index b26398e..c50a3c6 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",
@@ -88,6 +92,7 @@
"libhardware",
"libhidlbase",
"libjpeg",
+ "libmedia_codeclist",
"libmedia_omx",
"libmemunreachable",
"libsensorprivacy",
diff --git a/services/camera/libcameraservice/CameraService.cpp b/services/camera/libcameraservice/CameraService.cpp
index e663485..c70513c 100644
--- a/services/camera/libcameraservice/CameraService.cpp
+++ b/services/camera/libcameraservice/CameraService.cpp
@@ -253,6 +253,15 @@
enumerateProviders();
}
+bool CameraService::isPublicallyHiddenSecureCamera(const String8& cameraId) {
+ auto state = getCameraState(cameraId);
+ if (state != nullptr) {
+ return state->isPublicallyHiddenSecureCamera();
+ }
+ // Hidden physical camera ids won't have CameraState
+ return mCameraProviderManager->isPublicallyHiddenSecureCamera(cameraId.c_str());
+}
+
void CameraService::updateCameraNumAndIds() {
Mutex::Autolock l(mServiceLock);
mNumberOfCameras = mCameraProviderManager->getCameraCount();
@@ -268,6 +277,8 @@
ALOGE("Failed to query device resource cost: %s (%d)", strerror(-res), res);
return;
}
+ bool isPublicallyHiddenSecureCamera =
+ mCameraProviderManager->isPublicallyHiddenSecureCamera(id.string());
std::set<String8> conflicting;
for (size_t i = 0; i < cost.conflictingDevices.size(); i++) {
conflicting.emplace(String8(cost.conflictingDevices[i].c_str()));
@@ -276,7 +287,8 @@
{
Mutex::Autolock lock(mCameraStatesLock);
mCameraStates.emplace(id, std::make_shared<CameraState>(id, cost.resourceCost,
- conflicting));
+ conflicting,
+ isPublicallyHiddenSecureCamera));
}
if (mFlashlight->hasFlashUnit(id)) {
@@ -514,8 +526,16 @@
"Camera subsystem is not available");;
}
- Status ret{};
+ if (shouldRejectHiddenCameraConnection(String8(cameraId))) {
+ ALOGW("Attempting to retrieve characteristics for system-only camera id %s, rejected",
+ String8(cameraId).string());
+ return STATUS_ERROR_FMT(ERROR_DISCONNECTED,
+ "No camera device with ID \"%s\" currently available",
+ String8(cameraId).string());
+ }
+
+ Status ret{};
status_t res = mCameraProviderManager->getCameraCharacteristics(
String8(cameraId).string(), cameraInfo);
if (res != OK) {
@@ -1149,6 +1169,8 @@
clientPid,
states[states.size() - 1]);
+ resource_policy::ClientPriority clientPriority = clientDescriptor->getPriority();
+
// Find clients that would be evicted
auto evicted = mActiveClientManager.wouldEvict(clientDescriptor);
@@ -1166,8 +1188,7 @@
String8 msg = String8::format("%s : DENIED connect device %s client for package %s "
"(PID %d, score %d state %d) due to eviction policy", curTime.string(),
cameraId.string(), packageName.string(), clientPid,
- priorityScores[priorityScores.size() - 1],
- states[states.size() - 1]);
+ clientPriority.getScore(), clientPriority.getState());
for (auto& i : incompatibleClients) {
msg.appendFormat("\n - Blocked by existing device %s client for package %s"
@@ -1212,9 +1233,8 @@
i->getKey().string(), String8{clientSp->getPackageName()}.string(),
i->getOwnerId(), i->getPriority().getScore(),
i->getPriority().getState(), cameraId.string(),
- packageName.string(), clientPid,
- priorityScores[priorityScores.size() - 1],
- states[states.size() - 1]));
+ packageName.string(), clientPid, clientPriority.getScore(),
+ clientPriority.getState()));
// Notify the client of disconnection
clientSp->notifyError(hardware::camera2::ICameraDeviceCallbacks::ERROR_CAMERA_DISCONNECTED,
@@ -1330,7 +1350,7 @@
// publically hidden, we should reject the connection.
if (!hardware::IPCThreadState::self()->isServingCall() &&
CameraThreadState::getCallingPid() != getpid() &&
- mCameraProviderManager->isPublicallyHiddenSecureCamera(cameraId.c_str())) {
+ isPublicallyHiddenSecureCamera(cameraId)) {
return true;
}
return false;
@@ -1348,14 +1368,19 @@
Status ret = Status::ok();
String8 id = String8(cameraId);
sp<CameraDeviceClient> client = nullptr;
-
+ String16 clientPackageNameAdj = clientPackageName;
+ if (hardware::IPCThreadState::self()->isServingCall()) {
+ std::string vendorClient =
+ StringPrintf("vendor.client.pid<%d>", CameraThreadState::getCallingPid());
+ clientPackageNameAdj = String16(vendorClient.c_str());
+ }
ret = connectHelper<hardware::camera2::ICameraDeviceCallbacks,CameraDeviceClient>(cameraCb, id,
/*api1CameraId*/-1,
- CAMERA_HAL_API_VERSION_UNSPECIFIED, clientPackageName,
+ CAMERA_HAL_API_VERSION_UNSPECIFIED, clientPackageNameAdj,
clientUid, USE_CALLING_PID, API_2, /*shimUpdateOnly*/ false, /*out*/client);
if(!ret.isOk()) {
- logRejected(id, CameraThreadState::getCallingPid(), String8(clientPackageName),
+ logRejected(id, CameraThreadState::getCallingPid(), String8(clientPackageNameAdj),
ret.toString8());
return ret;
}
@@ -1799,16 +1824,25 @@
{
Mutex::Autolock lock(mCameraStatesLock);
for (auto& i : mCameraStates) {
- if (!isVendorListener &&
- mCameraProviderManager->isPublicallyHiddenSecureCamera(i.first.c_str())) {
- ALOGV("Cannot add public listener for hidden system-only %s for pid %d",
- i.first.c_str(), CameraThreadState::getCallingPid());
- continue;
- }
cameraStatuses->emplace_back(i.first, mapToInterface(i.second->getStatus()));
}
}
+ // Remove the camera statuses that should be hidden from the client, we do
+ // this after collecting the states in order to avoid holding
+ // mCameraStatesLock and mInterfaceLock (held in
+ // isPublicallyHiddenSecureCamera()) at the same time.
+ cameraStatuses->erase(std::remove_if(cameraStatuses->begin(), cameraStatuses->end(),
+ [this, &isVendorListener](const hardware::CameraStatus& s) {
+ bool ret = !isVendorListener && isPublicallyHiddenSecureCamera(s.cameraId);
+ if (ret) {
+ ALOGV("Cannot add public listener for hidden system-only %s for pid %d",
+ s.cameraId.c_str(), CameraThreadState::getCallingPid());
+ }
+ return ret;
+ }),
+ cameraStatuses->end());
+
/*
* Immediately signal current torch status to this listener only
* This may be a subset of all the devices, so don't include it in the response directly
@@ -2368,11 +2402,7 @@
}
mClientPackageName = packages[0];
}
- if (hardware::IPCThreadState::self()->isServingCall()) {
- std::string vendorClient =
- StringPrintf("vendor.client.pid<%d>", CameraThreadState::getCallingPid());
- mClientPackageName = String16(vendorClient.c_str());
- } else {
+ if (!hardware::IPCThreadState::self()->isServingCall()) {
mAppOpsManager = std::make_unique<AppOpsManager>();
}
}
@@ -2870,8 +2900,9 @@
// ----------------------------------------------------------------------------
CameraService::CameraState::CameraState(const String8& id, int cost,
- const std::set<String8>& conflicting) : mId(id),
- mStatus(StatusInternal::NOT_PRESENT), mCost(cost), mConflicting(conflicting) {}
+ const std::set<String8>& conflicting, bool isHidden) : mId(id),
+ mStatus(StatusInternal::NOT_PRESENT), mCost(cost), mConflicting(conflicting),
+ mIsPublicallyHiddenSecureCamera(isHidden) {}
CameraService::CameraState::~CameraState() {}
@@ -2900,6 +2931,10 @@
return mId;
}
+bool CameraService::CameraState::isPublicallyHiddenSecureCamera() const {
+ return mIsPublicallyHiddenSecureCamera;
+}
+
// ----------------------------------------------------------------------------
// ClientEventListener
// ----------------------------------------------------------------------------
@@ -3235,10 +3270,10 @@
cameraId.string());
return;
}
-
+ bool isHidden = isPublicallyHiddenSecureCamera(cameraId);
// 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]
+ state->updateStatus(status, cameraId, rejectSourceStates, [this,&isHidden]
(const String8& cameraId, StatusInternal status) {
if (status != StatusInternal::ENUMERATING) {
@@ -3260,8 +3295,7 @@
Mutex::Autolock lock(mStatusListenerLock);
for (auto& listener : mListenerList) {
- if (!listener.first &&
- mCameraProviderManager->isPublicallyHiddenSecureCamera(cameraId.c_str())) {
+ if (!listener.first && isHidden) {
ALOGV("Skipping camera discovery callback for system-only camera %s",
cameraId.c_str());
continue;
diff --git a/services/camera/libcameraservice/CameraService.h b/services/camera/libcameraservice/CameraService.h
index cf93a41..8bb78cd 100644
--- a/services/camera/libcameraservice/CameraService.h
+++ b/services/camera/libcameraservice/CameraService.h
@@ -471,7 +471,8 @@
* Make a new CameraState and set the ID, cost, and conflicting devices using the values
* returned in the HAL's camera_info struct for each device.
*/
- CameraState(const String8& id, int cost, const std::set<String8>& conflicting);
+ CameraState(const String8& id, int cost, const std::set<String8>& conflicting,
+ bool isHidden);
virtual ~CameraState();
/**
@@ -523,6 +524,11 @@
*/
String8 getId() const;
+ /**
+ * Return if the camera device is a publically hidden secure camera
+ */
+ bool isPublicallyHiddenSecureCamera() const;
+
private:
const String8 mId;
StatusInternal mStatus; // protected by mStatusLock
@@ -530,6 +536,7 @@
std::set<String8> mConflicting;
mutable Mutex mStatusLock;
CameraParameters mShimParams;
+ const bool mIsPublicallyHiddenSecureCamera;
}; // class CameraState
// Observer for UID lifecycle enforcing that UIDs in idle
@@ -635,7 +642,9 @@
// Should an operation attempt on a cameraId be rejected, if the camera id is
// advertised as a publically hidden secure camera, by the camera HAL ?
- bool shouldRejectHiddenCameraConnection(const String8 & cameraId);
+ bool shouldRejectHiddenCameraConnection(const String8& cameraId);
+
+ bool isPublicallyHiddenSecureCamera(const String8& cameraId);
// Single implementation shared between the various connect calls
template<class CALLBACK, class CLIENT>
diff --git a/services/camera/libcameraservice/api2/HeicCompositeStream.cpp b/services/camera/libcameraservice/api2/HeicCompositeStream.cpp
index 5a87134..3d1235e 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/AMessage.h>
diff --git a/services/camera/libcameraservice/common/CameraProviderManager.cpp b/services/camera/libcameraservice/common/CameraProviderManager.cpp
index 09638d0..cb2c324 100644
--- a/services/camera/libcameraservice/common/CameraProviderManager.cpp
+++ b/services/camera/libcameraservice/common/CameraProviderManager.cpp
@@ -97,7 +97,13 @@
std::lock_guard<std::mutex> lock(mInterfaceMutex);
int count = 0;
for (auto& provider : mProviders) {
- count += provider->mUniqueCameraIds.size();
+ for (auto& id : provider->mUniqueCameraIds) {
+ // Hidden secure camera ids are not to be exposed to camera1 api.
+ if (isPublicallyHiddenSecureCameraLocked(id)) {
+ continue;
+ }
+ count++;
+ }
}
return count;
}
@@ -123,7 +129,11 @@
// for each camera facing, only take the first id advertised by HAL in
// all [logical, physical1, physical2, ...] id combos, and filter out the rest.
filterLogicalCameraIdsLocked(providerDeviceIds);
-
+ // Hidden secure camera ids are not to be exposed to camera1 api.
+ providerDeviceIds.erase(std::remove_if(providerDeviceIds.begin(), providerDeviceIds.end(),
+ [this](const std::string& s) {
+ return this->isPublicallyHiddenSecureCameraLocked(s);}),
+ providerDeviceIds.end());
deviceIds.insert(deviceIds.end(), providerDeviceIds.begin(), providerDeviceIds.end());
}
@@ -1035,23 +1045,42 @@
return deviceInfo->mIsLogicalCamera;
}
-bool CameraProviderManager::isPublicallyHiddenSecureCamera(const std::string& id) {
+bool CameraProviderManager::isPublicallyHiddenSecureCamera(const std::string& id) const {
std::lock_guard<std::mutex> lock(mInterfaceMutex);
-
- auto deviceInfo = findDeviceInfoLocked(id);
- if (deviceInfo == nullptr) {
- return false;
- }
- return deviceInfo->mIsPublicallyHiddenSecureCamera;
+ return isPublicallyHiddenSecureCameraLocked(id);
}
-bool CameraProviderManager::isHiddenPhysicalCamera(const std::string& cameraId) {
+bool CameraProviderManager::isPublicallyHiddenSecureCameraLocked(const std::string& id) const {
+ auto deviceInfo = findDeviceInfoLocked(id);
+ if (deviceInfo != nullptr) {
+ return deviceInfo->mIsPublicallyHiddenSecureCamera;
+ }
+ // If this is a hidden physical camera, we should return what kind of
+ // camera the enclosing logical camera is.
+ auto isHiddenAndParent = isHiddenPhysicalCameraInternal(id);
+ if (isHiddenAndParent.first) {
+ LOG_ALWAYS_FATAL_IF(id == isHiddenAndParent.second->mId,
+ "%s: hidden physical camera id %s and enclosing logical camera id %s are the same",
+ __FUNCTION__, id.c_str(), isHiddenAndParent.second->mId.c_str());
+ return isPublicallyHiddenSecureCameraLocked(isHiddenAndParent.second->mId);
+ }
+ // Invalid camera id
+ return true;
+}
+
+bool CameraProviderManager::isHiddenPhysicalCamera(const std::string& cameraId) const {
+ return isHiddenPhysicalCameraInternal(cameraId).first;
+}
+
+std::pair<bool, CameraProviderManager::ProviderInfo::DeviceInfo *>
+CameraProviderManager::isHiddenPhysicalCameraInternal(const std::string& cameraId) const {
+ auto falseRet = std::make_pair(false, nullptr);
for (auto& provider : mProviders) {
for (auto& deviceInfo : provider->mDevices) {
if (deviceInfo->mId == cameraId) {
// cameraId is found in public camera IDs advertised by the
// provider.
- return false;
+ return falseRet;
}
}
}
@@ -1063,7 +1092,7 @@
if (res != OK) {
ALOGE("%s: Failed to getCameraCharacteristics for id %s", __FUNCTION__,
deviceInfo->mId.c_str());
- return false;
+ return falseRet;
}
std::vector<std::string> physicalIds;
@@ -1075,16 +1104,16 @@
if (deviceVersion < CAMERA_DEVICE_API_VERSION_3_5) {
ALOGE("%s: Wrong deviceVersion %x for hiddenPhysicalCameraId %s",
__FUNCTION__, deviceVersion, cameraId.c_str());
- return false;
+ return falseRet;
} else {
- return true;
+ return std::make_pair(true, deviceInfo.get());
}
}
}
}
}
- return false;
+ return falseRet;
}
status_t CameraProviderManager::addProviderLocked(const std::string& newProvider, bool expected) {
@@ -2058,6 +2087,13 @@
return OK;
}
bool CameraProviderManager::ProviderInfo::DeviceInfo3::isAPI1Compatible() const {
+ // Do not advertise NIR cameras to API1 camera app.
+ camera_metadata_ro_entry cfa = mCameraCharacteristics.find(
+ ANDROID_SENSOR_INFO_COLOR_FILTER_ARRANGEMENT);
+ if (cfa.count == 1 && cfa.data.u8[0] == ANDROID_SENSOR_INFO_COLOR_FILTER_ARRANGEMENT_NIR) {
+ return false;
+ }
+
bool isBackwardCompatible = false;
camera_metadata_ro_entry_t caps = mCameraCharacteristics.find(
ANDROID_REQUEST_AVAILABLE_CAPABILITIES);
diff --git a/services/camera/libcameraservice/common/CameraProviderManager.h b/services/camera/libcameraservice/common/CameraProviderManager.h
index a42fb4d..cd283b3 100644
--- a/services/camera/libcameraservice/common/CameraProviderManager.h
+++ b/services/camera/libcameraservice/common/CameraProviderManager.h
@@ -269,8 +269,8 @@
*/
bool isLogicalCamera(const std::string& id, std::vector<std::string>* physicalCameraIds);
- bool isPublicallyHiddenSecureCamera(const std::string& id);
- bool isHiddenPhysicalCamera(const std::string& cameraId);
+ bool isPublicallyHiddenSecureCamera(const std::string& id) const;
+ bool isHiddenPhysicalCamera(const std::string& cameraId) const;
static const float kDepthARTolerance;
private:
@@ -591,7 +591,15 @@
status_t getCameraCharacteristicsLocked(const std::string &id,
CameraMetadata* characteristics) const;
+
+ bool isPublicallyHiddenSecureCameraLocked(const std::string& id) const;
+
void filterLogicalCameraIdsLocked(std::vector<std::string>& deviceIds) const;
+
+ bool isPublicallyHiddenSecureCameraLocked(const std::string& id);
+
+ std::pair<bool, CameraProviderManager::ProviderInfo::DeviceInfo *>
+ isHiddenPhysicalCameraInternal(const std::string& cameraId) const;
};
} // namespace android
diff --git a/services/camera/libcameraservice/common/DepthPhotoProcessor.cpp b/services/camera/libcameraservice/common/DepthPhotoProcessor.cpp
index 3c90de0..94541d8 100644
--- a/services/camera/libcameraservice/common/DepthPhotoProcessor.cpp
+++ b/services/camera/libcameraservice/common/DepthPhotoProcessor.cpp
@@ -419,7 +419,7 @@
std::vector<std::unique_ptr<Item>> items;
std::vector<std::unique_ptr<Camera>> cameraList;
- auto image = Image::FromDataForPrimaryImage("android/mainimage", &items);
+ auto image = Image::FromDataForPrimaryImage("image/jpeg", &items);
std::unique_ptr<CameraParams> cameraParams(new CameraParams(std::move(image)));
if (cameraParams == nullptr) {
ALOGE("%s: Failed to initialize camera parameters", __FUNCTION__);
diff --git a/services/camera/libcameraservice/device3/Camera3Device.cpp b/services/camera/libcameraservice/device3/Camera3Device.cpp
index a8e80fa..4227a3b 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.cpp
+++ b/services/camera/libcameraservice/device3/Camera3Device.cpp
@@ -29,6 +29,9 @@
#define CLOGE(fmt, ...) ALOGE("Camera %s: %s: " fmt, mId.string(), __FUNCTION__, \
##__VA_ARGS__)
+#define CLOGW(fmt, ...) ALOGW("Camera %s: %s: " fmt, mId.string(), __FUNCTION__, \
+ ##__VA_ARGS__)
+
// Convenience macros for transitioning to the error state
#define SET_ERR(fmt, ...) setErrorState( \
"%s: " fmt, __FUNCTION__, \
@@ -3267,14 +3270,19 @@
ALOGVV("%s: removed frame %d from InFlightMap", __FUNCTION__, frameNumber);
}
- // Sanity check - if we have too many in-flight frames, something has
- // likely gone wrong
- if (!mIsConstrainedHighSpeedConfiguration && mInFlightMap.size() > kInFlightWarnLimit) {
- CLOGE("In-flight list too large: %zu", mInFlightMap.size());
- } else if (mIsConstrainedHighSpeedConfiguration && mInFlightMap.size() >
- kInFlightWarnLimitHighSpeed) {
- CLOGE("In-flight list too large for high speed configuration: %zu",
- mInFlightMap.size());
+ // Sanity check - if we have too many in-flight frames with long total inflight duration,
+ // something has likely gone wrong. This might still be legit only if application send in
+ // a long burst of long exposure requests.
+ if (mExpectedInflightDuration > kMinWarnInflightDuration) {
+ if (!mIsConstrainedHighSpeedConfiguration && mInFlightMap.size() > kInFlightWarnLimit) {
+ CLOGW("In-flight list too large: %zu, total inflight duration %" PRIu64,
+ mInFlightMap.size(), mExpectedInflightDuration);
+ } else if (mIsConstrainedHighSpeedConfiguration && mInFlightMap.size() >
+ kInFlightWarnLimitHighSpeed) {
+ CLOGW("In-flight list too large for high speed configuration: %zu,"
+ "total inflight duration %" PRIu64,
+ mInFlightMap.size(), mExpectedInflightDuration);
+ }
}
}
diff --git a/services/camera/libcameraservice/device3/Camera3Device.h b/services/camera/libcameraservice/device3/Camera3Device.h
index 6e8ac84..cae34ce 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.h
+++ b/services/camera/libcameraservice/device3/Camera3Device.h
@@ -227,6 +227,7 @@
static const size_t kDumpLockAttempts = 10;
static const size_t kDumpSleepDuration = 100000; // 0.10 sec
static const nsecs_t kActiveTimeout = 500000000; // 500 ms
+ static const nsecs_t kMinWarnInflightDuration = 5000000000; // 5 s
static const size_t kInFlightWarnLimit = 30;
static const size_t kInFlightWarnLimitHighSpeed = 256; // batch size 32 * pipe depth 8
static const nsecs_t kDefaultExpectedDuration = 100000000; // 100 ms
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/main_codecservice.cpp b/services/mediacodec/main_codecservice.cpp
index f668c33..6a82b1b 100644
--- a/services/mediacodec/main_codecservice.cpp
+++ b/services/mediacodec/main_codecservice.cpp
@@ -21,6 +21,7 @@
#include "minijail.h"
#include <binder/ProcessState.h>
+#include <cutils/properties.h>
#include <hidl/HidlTransportSupport.h>
#include <media/stagefright/omx/1.0/Omx.h>
#include <media/stagefright/omx/1.0/OmxStore.h>
@@ -57,7 +58,8 @@
} else {
LOG(INFO) << "IOmx HAL service created.";
}
- sp<IOmxStore> omxStore = new implementation::OmxStore(omx);
+ sp<IOmxStore> omxStore = new implementation::OmxStore(
+ property_get_int64("vendor.media.omx", 1) ? omx : nullptr);
if (omxStore == nullptr) {
LOG(ERROR) << "Cannot create IOmxStore HAL service.";
} else if (omxStore->registerAsService() != OK) {
diff --git a/services/mediadrm/Android.mk b/services/mediadrm/Android.mk
index 3e94596..9f03964 100644
--- a/services/mediadrm/Android.mk
+++ b/services/mediadrm/Android.mk
@@ -20,6 +20,10 @@
MediaDrmService.cpp \
main_mediadrmserver.cpp
+LOCAL_HEADER_LIBRARIES:= \
+ libmedia_headers \
+ libmediadrm_headers
+
LOCAL_SHARED_LIBRARIES:= \
libbinder \
liblog \
diff --git a/services/mediaextractor/Android.bp b/services/mediaextractor/Android.bp
index b812244..e906500 100644
--- a/services/mediaextractor/Android.bp
+++ b/services/mediaextractor/Android.bp
@@ -8,6 +8,7 @@
srcs: ["MediaExtractorService.cpp"],
shared_libs: [
+ "libdatasource",
"libmedia",
"libstagefright",
"libbinder",
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/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",