Merge "Fix hardware protected path for AImageReader."
diff --git a/apex/Android.bp b/apex/Android.bp
index 51e4c23..39997d2 100644
--- a/apex/Android.bp
+++ b/apex/Android.bp
@@ -15,6 +15,7 @@
apex {
name: "com.android.media",
manifest: "manifest.json",
+ java_libs: ["updatable-media"],
native_shared_libs: [
// Extractor plugins
"libaacextractor",
@@ -28,14 +29,13 @@
"liboggextractor",
"libwavextractor",
// MediaPlayer2
- "libmediaplayer2_jni",
+ "libmedia2_jni",
],
key: "com.android.media.key",
}
apex {
name: "com.android.media.swcodec",
- compile_multilib: "32",
manifest: "manifest_codec.json",
native_shared_libs: [
"libmedia_codecserviceregistrant",
diff --git a/camera/ndk/include/camera/NdkCameraMetadataTags.h b/camera/ndk/include/camera/NdkCameraMetadataTags.h
index 4bb74cb..641816f 100644
--- a/camera/ndk/include/camera/NdkCameraMetadataTags.h
+++ b/camera/ndk/include/camera/NdkCameraMetadataTags.h
@@ -5561,12 +5561,12 @@
* <li>ACameraMetadata from ACameraManager_getCameraCharacteristics</li>
* </ul></p>
*
- * <p>For a logical camera, this is concatenation of all underlying physical camera ids.
- * The null terminator for physical camera id must be preserved so that the whole string
- * can be tokenized using '\0' to generate list of physical camera ids.</p>
- * <p>For example, if the physical camera ids of the logical camera are "2" and "3", the
+ * <p>For a logical camera, this is concatenation of all underlying physical camera IDs.
+ * The null terminator for physical camera ID must be preserved so that the whole string
+ * can be tokenized using '\0' to generate list of physical camera IDs.</p>
+ * <p>For example, if the physical camera IDs of the logical camera are "2" and "3", the
* value of this tag will be ['2', '\0', '3', '\0'].</p>
- * <p>The number of physical camera ids must be no less than 2.</p>
+ * <p>The number of physical camera IDs must be no less than 2.</p>
*/
ACAMERA_LOGICAL_MULTI_CAMERA_PHYSICAL_IDS = // byte[n]
ACAMERA_LOGICAL_MULTI_CAMERA_START,
@@ -5591,6 +5591,28 @@
*/
ACAMERA_LOGICAL_MULTI_CAMERA_SENSOR_SYNC_TYPE = // byte (acamera_metadata_enum_android_logical_multi_camera_sensor_sync_type_t)
ACAMERA_LOGICAL_MULTI_CAMERA_START + 1,
+ /**
+ * <p>String containing the ID of the underlying active physical camera.</p>
+ *
+ * <p>Type: byte</p>
+ *
+ * <p>This tag may appear in:
+ * <ul>
+ * <li>ACameraMetadata from ACameraCaptureSession_captureCallback_result callbacks</li>
+ * </ul></p>
+ *
+ * <p>The ID of the active physical camera that's backing the logical camera. All camera
+ * streams and metadata that are not physical camera specific will be originating from this
+ * physical camera. This must be one of valid physical IDs advertised in the physicalIds
+ * static tag.</p>
+ * <p>For a logical camera made up of physical cameras where each camera's lenses have
+ * different characteristics, the camera device may choose to switch between the physical
+ * cameras when application changes FOCAL_LENGTH or SCALER_CROP_REGION.
+ * At the time of lens switch, this result metadata reflects the new active physical camera
+ * ID.</p>
+ */
+ ACAMERA_LOGICAL_MULTI_CAMERA_ACTIVE_PHYSICAL_ID = // byte
+ ACAMERA_LOGICAL_MULTI_CAMERA_START + 2,
ACAMERA_LOGICAL_MULTI_CAMERA_END,
/**
@@ -7162,6 +7184,10 @@
* <p>If this is supported, android.scaler.streamConfigurationMap will
* additionally return a min frame duration that is greater than
* zero for each supported size-format combination.</p>
+ * <p>For camera devices with LOGICAL_MULTI_CAMERA capability, when the underlying active
+ * physical camera switches, exposureTime, sensitivity, and lens properties may change
+ * even if AE/AF is locked. However, the overall auto exposure and auto focus experience
+ * for users will be consistent. Refer to LOGICAL_MULTI_CAMERA capability for details.</p>
*
* @see ACAMERA_BLACK_LEVEL_LOCK
* @see ACAMERA_CONTROL_AE_LOCK
@@ -7217,6 +7243,10 @@
* will accurately report the values applied by AWB in the result.</p>
* <p>A given camera device may also support additional post-processing
* controls, but this capability only covers the above list of controls.</p>
+ * <p>For camera devices with LOGICAL_MULTI_CAMERA capability, when underlying active
+ * physical camera switches, tonemap, white balance, and shading map may change even if
+ * awb is locked. However, the overall post-processing experience for users will be
+ * consistent. Refer to LOGICAL_MULTI_CAMERA capability for details.</p>
*
* @see ACAMERA_COLOR_CORRECTION_ABERRATION_MODE
* @see ACAMERA_COLOR_CORRECTION_AVAILABLE_ABERRATION_MODES
@@ -7396,7 +7426,7 @@
* </li>
* <li>The SENSOR_INFO_TIMESTAMP_SOURCE of the logical device and physical devices must be
* the same.</li>
- * <li>The logical camera device must be LIMITED or higher device.</li>
+ * <li>The logical camera must be LIMITED or higher device.</li>
* </ul>
* <p>Both the logical camera device and its underlying physical devices support the
* mandatory stream combinations required for their device levels.</p>
@@ -7416,13 +7446,84 @@
* <p>Using physical streams in place of a logical stream of the same size and format will
* not slow down the frame rate of the capture, as long as the minimum frame duration
* of the physical and logical streams are the same.</p>
+ * <p>A logical camera device's dynamic metadata may contain
+ * ACAMERA_LOGICAL_MULTI_CAMERA_ACTIVE_PHYSICAL_ID to notify the application of the current
+ * active physical camera Id. An active physical camera is the physical camera from which
+ * the logical camera's main image data outputs (YUV or RAW) and metadata come from.
+ * In addition, this serves as an indication which physical camera is used to output to
+ * a RAW stream, or in case only physical cameras support RAW, which physical RAW stream
+ * the application should request.</p>
+ * <p>Logical camera's static metadata tags below describe the default active physical
+ * camera. An active physical camera is default if it's used when application directly
+ * uses requests built from a template. All templates will default to the same active
+ * physical camera.</p>
+ * <ul>
+ * <li>ACAMERA_SENSOR_INFO_SENSITIVITY_RANGE</li>
+ * <li>ACAMERA_SENSOR_INFO_COLOR_FILTER_ARRANGEMENT</li>
+ * <li>ACAMERA_SENSOR_INFO_EXPOSURE_TIME_RANGE</li>
+ * <li>ACAMERA_SENSOR_INFO_MAX_FRAME_DURATION</li>
+ * <li>ACAMERA_SENSOR_INFO_PHYSICAL_SIZE</li>
+ * <li>ACAMERA_SENSOR_INFO_WHITE_LEVEL</li>
+ * <li>ACAMERA_SENSOR_INFO_LENS_SHADING_APPLIED</li>
+ * <li>ACAMERA_SENSOR_REFERENCE_ILLUMINANT1</li>
+ * <li>ACAMERA_SENSOR_REFERENCE_ILLUMINANT2</li>
+ * <li>ACAMERA_SENSOR_CALIBRATION_TRANSFORM1</li>
+ * <li>ACAMERA_SENSOR_CALIBRATION_TRANSFORM2</li>
+ * <li>ACAMERA_SENSOR_COLOR_TRANSFORM1</li>
+ * <li>ACAMERA_SENSOR_COLOR_TRANSFORM2</li>
+ * <li>ACAMERA_SENSOR_FORWARD_MATRIX1</li>
+ * <li>ACAMERA_SENSOR_FORWARD_MATRIX2</li>
+ * <li>ACAMERA_SENSOR_BLACK_LEVEL_PATTERN</li>
+ * <li>ACAMERA_SENSOR_MAX_ANALOG_SENSITIVITY</li>
+ * <li>ACAMERA_SENSOR_OPTICAL_BLACK_REGIONS</li>
+ * <li>ACAMERA_SENSOR_AVAILABLE_TEST_PATTERN_MODES</li>
+ * <li>ACAMERA_LENS_INFO_HYPERFOCAL_DISTANCE</li>
+ * <li>ACAMERA_LENS_INFO_MINIMUM_FOCUS_DISTANCE</li>
+ * <li>ACAMERA_LENS_INFO_FOCUS_DISTANCE_CALIBRATION</li>
+ * <li>ACAMERA_LENS_POSE_ROTATION</li>
+ * <li>ACAMERA_LENS_POSE_TRANSLATION</li>
+ * <li>ACAMERA_LENS_INTRINSIC_CALIBRATION</li>
+ * <li>ACAMERA_LENS_POSE_REFERENCE</li>
+ * <li>ACAMERA_LENS_DISTORTION</li>
+ * </ul>
+ * <p>To maintain backward compatibility, the capture request and result metadata tags
+ * required for basic camera functionalities will be solely based on the
+ * logical camera capabiltity. Other request and result metadata tags, on the other
+ * hand, will be based on current active physical camera. For example, the physical
+ * cameras' sensor sensitivity and lens capability could be different from each other.
+ * So when the application manually controls sensor exposure time/gain, or does manual
+ * focus control, it must checks the current active physical camera's exposure, gain,
+ * and focus distance range.</p>
*
* @see ACAMERA_LENS_DISTORTION
+ * @see ACAMERA_LENS_INFO_FOCUS_DISTANCE_CALIBRATION
+ * @see ACAMERA_LENS_INFO_HYPERFOCAL_DISTANCE
+ * @see ACAMERA_LENS_INFO_MINIMUM_FOCUS_DISTANCE
* @see ACAMERA_LENS_INTRINSIC_CALIBRATION
* @see ACAMERA_LENS_POSE_REFERENCE
* @see ACAMERA_LENS_POSE_ROTATION
* @see ACAMERA_LENS_POSE_TRANSLATION
+ * @see ACAMERA_LOGICAL_MULTI_CAMERA_ACTIVE_PHYSICAL_ID
* @see ACAMERA_LOGICAL_MULTI_CAMERA_SENSOR_SYNC_TYPE
+ * @see ACAMERA_SENSOR_AVAILABLE_TEST_PATTERN_MODES
+ * @see ACAMERA_SENSOR_BLACK_LEVEL_PATTERN
+ * @see ACAMERA_SENSOR_CALIBRATION_TRANSFORM1
+ * @see ACAMERA_SENSOR_CALIBRATION_TRANSFORM2
+ * @see ACAMERA_SENSOR_COLOR_TRANSFORM1
+ * @see ACAMERA_SENSOR_COLOR_TRANSFORM2
+ * @see ACAMERA_SENSOR_FORWARD_MATRIX1
+ * @see ACAMERA_SENSOR_FORWARD_MATRIX2
+ * @see ACAMERA_SENSOR_INFO_COLOR_FILTER_ARRANGEMENT
+ * @see ACAMERA_SENSOR_INFO_EXPOSURE_TIME_RANGE
+ * @see ACAMERA_SENSOR_INFO_LENS_SHADING_APPLIED
+ * @see ACAMERA_SENSOR_INFO_MAX_FRAME_DURATION
+ * @see ACAMERA_SENSOR_INFO_PHYSICAL_SIZE
+ * @see ACAMERA_SENSOR_INFO_SENSITIVITY_RANGE
+ * @see ACAMERA_SENSOR_INFO_WHITE_LEVEL
+ * @see ACAMERA_SENSOR_MAX_ANALOG_SENSITIVITY
+ * @see ACAMERA_SENSOR_OPTICAL_BLACK_REGIONS
+ * @see ACAMERA_SENSOR_REFERENCE_ILLUMINANT1
+ * @see ACAMERA_SENSOR_REFERENCE_ILLUMINANT2
*/
ACAMERA_REQUEST_AVAILABLE_CAPABILITIES_LOGICAL_MULTI_CAMERA = 11,
diff --git a/camera/ndk/ndk_vendor/impl/utils.cpp b/camera/ndk/ndk_vendor/impl/utils.cpp
index 7193006..5d2d47c 100644
--- a/camera/ndk/ndk_vendor/impl/utils.cpp
+++ b/camera/ndk/ndk_vendor/impl/utils.cpp
@@ -70,7 +70,6 @@
return;
}
size_t size = get_camera_metadata_size(src);
- ALOGE("Converting metadata size: %d", (int)size);
dst->setToExternal((uint8_t *) src, size);
return;
}
diff --git a/drm/drmserver/DrmManagerService.cpp b/drm/drmserver/DrmManagerService.cpp
index 2532275..2600a2c 100644
--- a/drm/drmserver/DrmManagerService.cpp
+++ b/drm/drmserver/DrmManagerService.cpp
@@ -58,22 +58,26 @@
return drm_perm_labels[index];
}
-bool DrmManagerService::selinuxIsProtectedCallAllowed(pid_t spid, drm_perm_t perm) {
+bool DrmManagerService::selinuxIsProtectedCallAllowed(pid_t spid, const char* ssid, drm_perm_t perm) {
if (selinux_enabled <= 0) {
return true;
}
- char *sctx;
+ char *sctx = NULL;
const char *selinux_class = "drmservice";
const char *str_perm = get_perm_label(perm);
- if (getpidcon(spid, &sctx) != 0) {
- ALOGE("SELinux: getpidcon(pid=%d) failed.\n", spid);
- return false;
+ if (ssid == NULL) {
+ android_errorWriteLog(0x534e4554, "121035042");
+
+ if (getpidcon(spid, &sctx) != 0) {
+ ALOGE("SELinux: getpidcon(pid=%d) failed.\n", spid);
+ return false;
+ }
}
- bool allowed = (selinux_check_access(sctx, drmserver_context, selinux_class,
- str_perm, NULL) == 0);
+ bool allowed = (selinux_check_access(ssid ? ssid : sctx, drmserver_context,
+ selinux_class, str_perm, NULL) == 0);
freecon(sctx);
return allowed;
@@ -86,10 +90,11 @@
IPCThreadState* ipcState = IPCThreadState::self();
uid_t uid = ipcState->getCallingUid();
pid_t spid = ipcState->getCallingPid();
+ const char* ssid = ipcState->getCallingSid();
for (unsigned int i = 0; i < trustedUids.size(); ++i) {
if (trustedUids[i] == uid) {
- return selinuxIsProtectedCallAllowed(spid, perm);
+ return selinuxIsProtectedCallAllowed(spid, ssid, perm);
}
}
return false;
@@ -97,7 +102,9 @@
void DrmManagerService::instantiate() {
ALOGV("instantiate");
- defaultServiceManager()->addService(String16("drm.drmManager"), new DrmManagerService());
+ sp<DrmManagerService> service = new DrmManagerService();
+ service->setRequestingSid(true);
+ defaultServiceManager()->addService(String16("drm.drmManager"), service);
if (0 >= trustedUids.size()) {
// TODO
diff --git a/drm/drmserver/DrmManagerService.h b/drm/drmserver/DrmManagerService.h
index 7aaeab5..2e27a3c 100644
--- a/drm/drmserver/DrmManagerService.h
+++ b/drm/drmserver/DrmManagerService.h
@@ -60,7 +60,7 @@
static const char *get_perm_label(drm_perm_t perm);
- static bool selinuxIsProtectedCallAllowed(pid_t spid, drm_perm_t perm);
+ static bool selinuxIsProtectedCallAllowed(pid_t spid, const char* ssid, drm_perm_t perm);
static bool isProtectedCallAllowed(drm_perm_t perm);
diff --git a/drm/libmediadrm/DrmHal.cpp b/drm/libmediadrm/DrmHal.cpp
index f4c0341..480c7cd 100644
--- a/drm/libmediadrm/DrmHal.cpp
+++ b/drm/libmediadrm/DrmHal.cpp
@@ -63,6 +63,7 @@
typedef drm::V1_1::KeyRequestType KeyRequestType_V1_1;
typedef drm::V1_2::Status Status_V1_2;
+typedef drm::V1_2::HdcpLevel HdcpLevel_V1_2;
namespace {
@@ -144,6 +145,23 @@
}
}
+static SecurityLevel toHidlSecurityLevel(DrmPlugin::SecurityLevel level) {
+ switch(level) {
+ case DrmPlugin::kSecurityLevelSwSecureCrypto:
+ return SecurityLevel::SW_SECURE_CRYPTO;
+ case DrmPlugin::kSecurityLevelSwSecureDecode:
+ return SecurityLevel::SW_SECURE_DECODE;
+ case DrmPlugin::kSecurityLevelHwSecureCrypto:
+ return SecurityLevel::HW_SECURE_CRYPTO;
+ case DrmPlugin::kSecurityLevelHwSecureDecode:
+ return SecurityLevel::HW_SECURE_DECODE;
+ case DrmPlugin::kSecurityLevelHwSecureAll:
+ return SecurityLevel::HW_SECURE_ALL;
+ default:
+ return SecurityLevel::UNKNOWN;
+ }
+}
+
static DrmPlugin::OfflineLicenseState toOfflineLicenseState(
OfflineLicenseState licenseState) {
switch(licenseState) {
@@ -156,26 +174,26 @@
}
}
-static DrmPlugin::HdcpLevel toHdcpLevel(HdcpLevel level) {
+static DrmPlugin::HdcpLevel toHdcpLevel(HdcpLevel_V1_2 level) {
switch(level) {
- case HdcpLevel::HDCP_NONE:
+ case HdcpLevel_V1_2::HDCP_NONE:
return DrmPlugin::kHdcpNone;
- case HdcpLevel::HDCP_V1:
+ case HdcpLevel_V1_2::HDCP_V1:
return DrmPlugin::kHdcpV1;
- case HdcpLevel::HDCP_V2:
+ case HdcpLevel_V1_2::HDCP_V2:
return DrmPlugin::kHdcpV2;
- case HdcpLevel::HDCP_V2_1:
+ case HdcpLevel_V1_2::HDCP_V2_1:
return DrmPlugin::kHdcpV2_1;
- case HdcpLevel::HDCP_V2_2:
+ case HdcpLevel_V1_2::HDCP_V2_2:
return DrmPlugin::kHdcpV2_2;
- case HdcpLevel::HDCP_NO_OUTPUT:
+ case HdcpLevel_V1_2::HDCP_V2_3:
+ return DrmPlugin::kHdcpV2_3;
+ case HdcpLevel_V1_2::HDCP_NO_OUTPUT:
return DrmPlugin::kHdcpNoOutput;
default:
return DrmPlugin::kHdcpLevelUnknown;
}
}
-
-
static ::KeyedVector toHidlKeyedVector(const KeyedVector<String8, String8>&
keyedVector) {
std::vector<KeyValue> stdKeyedVector;
@@ -568,16 +586,39 @@
return Void();
}
-bool DrmHal::isCryptoSchemeSupported(const uint8_t uuid[16], const String8 &mimeType) {
+bool DrmHal::matchMimeTypeAndSecurityLevel(sp<IDrmFactory> &factory,
+ const uint8_t uuid[16],
+ const String8 &mimeType,
+ DrmPlugin::SecurityLevel level) {
+ if (mimeType == "") {
+ return true;
+ } else if (!factory->isContentTypeSupported(mimeType.string())) {
+ return false;
+ }
+
+ if (level == DrmPlugin::kSecurityLevelUnknown) {
+ return true;
+ } else {
+ sp<drm::V1_2::IDrmFactory> factoryV1_2 = drm::V1_2::IDrmFactory::castFrom(factory);
+ if (factoryV1_2 == NULL) {
+ return true;
+ } else if (factoryV1_2->isCryptoSchemeSupported_1_2(uuid,
+ mimeType.string(), toHidlSecurityLevel(level))) {
+ return true;
+ }
+ }
+ return false;
+}
+
+bool DrmHal::isCryptoSchemeSupported(const uint8_t uuid[16],
+ const String8 &mimeType,
+ DrmPlugin::SecurityLevel level) {
Mutex::Autolock autoLock(mLock);
for (size_t i = 0; i < mFactories.size(); i++) {
- if (mFactories[i]->isCryptoSchemeSupported(uuid)) {
- if (mimeType != "") {
- if (mFactories[i]->isContentTypeSupported(mimeType.string())) {
- return true;
- }
- } else {
+ sp<IDrmFactory> factory = mFactories[i];
+ if (factory->isCryptoSchemeSupported(uuid)) {
+ if (matchMimeTypeAndSecurityLevel(factory, uuid, mimeType, level)) {
return true;
}
}
@@ -633,30 +674,15 @@
Mutex::Autolock autoLock(mLock);
INIT_CHECK();
- SecurityLevel hSecurityLevel;
+ SecurityLevel hSecurityLevel = toHidlSecurityLevel(level);
bool setSecurityLevel = true;
- switch(level) {
- case DrmPlugin::kSecurityLevelSwSecureCrypto:
- hSecurityLevel = SecurityLevel::SW_SECURE_CRYPTO;
- break;
- case DrmPlugin::kSecurityLevelSwSecureDecode:
- hSecurityLevel = SecurityLevel::SW_SECURE_DECODE;
- break;
- case DrmPlugin::kSecurityLevelHwSecureCrypto:
- hSecurityLevel = SecurityLevel::HW_SECURE_CRYPTO;
- break;
- case DrmPlugin::kSecurityLevelHwSecureDecode:
- hSecurityLevel = SecurityLevel::HW_SECURE_DECODE;
- break;
- case DrmPlugin::kSecurityLevelHwSecureAll:
- hSecurityLevel = SecurityLevel::HW_SECURE_ALL;
- break;
- case DrmPlugin::kSecurityLevelMax:
+ if (level == DrmPlugin::kSecurityLevelMax) {
setSecurityLevel = false;
- break;
- default:
- return ERROR_DRM_CANNOT_HANDLE;
+ } else {
+ if (hSecurityLevel == SecurityLevel::UNKNOWN) {
+ return ERROR_DRM_CANNOT_HANDLE;
+ }
}
status_t err = UNKNOWN_ERROR;
@@ -1093,22 +1119,31 @@
}
status_t err = UNKNOWN_ERROR;
- if (mPluginV1_1 == NULL) {
- return ERROR_DRM_CANNOT_HANDLE;
- }
-
*connected = DrmPlugin::kHdcpLevelUnknown;
*max = DrmPlugin::kHdcpLevelUnknown;
- Return<void> hResult = mPluginV1_1->getHdcpLevels(
- [&](Status status, const HdcpLevel& hConnected, const HdcpLevel& hMax) {
- if (status == Status::OK) {
- *connected = toHdcpLevel(hConnected);
- *max = toHdcpLevel(hMax);
- }
- err = toStatusT(status);
- }
- );
+ Return<void> hResult;
+ if (mPluginV1_2 != NULL) {
+ hResult = mPluginV1_2->getHdcpLevels_1_2(
+ [&](Status_V1_2 status, const HdcpLevel_V1_2& hConnected, const HdcpLevel_V1_2& hMax) {
+ if (status == Status_V1_2::OK) {
+ *connected = toHdcpLevel(hConnected);
+ *max = toHdcpLevel(hMax);
+ }
+ err = toStatusT_1_2(status);
+ });
+ } else if (mPluginV1_1 != NULL) {
+ hResult = mPluginV1_1->getHdcpLevels(
+ [&](Status status, const HdcpLevel& hConnected, const HdcpLevel& hMax) {
+ if (status == Status::OK) {
+ *connected = toHdcpLevel(static_cast<HdcpLevel_V1_2>(hConnected));
+ *max = toHdcpLevel(static_cast<HdcpLevel_V1_2>(hMax));
+ }
+ err = toStatusT(status);
+ });
+ } else {
+ return ERROR_DRM_CANNOT_HANDLE;
+ }
return hResult.isOk() ? err : DEAD_OBJECT;
}
diff --git a/drm/libmediadrm/IDrm.cpp b/drm/libmediadrm/IDrm.cpp
index 8c26317..0f34315 100644
--- a/drm/libmediadrm/IDrm.cpp
+++ b/drm/libmediadrm/IDrm.cpp
@@ -83,11 +83,14 @@
return reply.readInt32();
}
- virtual bool isCryptoSchemeSupported(const uint8_t uuid[16], const String8 &mimeType) {
+ virtual bool isCryptoSchemeSupported(const uint8_t uuid[16], const String8 &mimeType,
+ DrmPlugin::SecurityLevel level) {
Parcel data, reply;
data.writeInterfaceToken(IDrm::getInterfaceDescriptor());
data.write(uuid, 16);
data.writeString8(mimeType);
+ data.writeInt32(level);
+
status_t status = remote()->transact(IS_CRYPTO_SUPPORTED, data, &reply);
if (status != OK) {
ALOGE("isCryptoSchemeSupported: binder call failed: %d", status);
@@ -123,11 +126,11 @@
return reply.readInt32();
}
- virtual status_t openSession(DrmPlugin::SecurityLevel securityLevel,
+ virtual status_t openSession(DrmPlugin::SecurityLevel level,
Vector<uint8_t> &sessionId) {
Parcel data, reply;
data.writeInterfaceToken(IDrm::getInterfaceDescriptor());
- data.writeInt32(securityLevel);
+ data.writeInt32(level);
status_t status = remote()->transact(OPEN_SESSION, data, &reply);
if (status != OK) {
@@ -768,7 +771,9 @@
uint8_t uuid[16];
data.read(uuid, sizeof(uuid));
String8 mimeType = data.readString8();
- reply->writeInt32(isCryptoSchemeSupported(uuid, mimeType));
+ DrmPlugin::SecurityLevel level =
+ static_cast<DrmPlugin::SecurityLevel>(data.readInt32());
+ reply->writeInt32(isCryptoSchemeSupported(uuid, mimeType, level));
return OK;
}
diff --git a/drm/mediadrm/plugins/clearkey/hidl/DrmFactory.cpp b/drm/mediadrm/plugins/clearkey/hidl/DrmFactory.cpp
index 9d040a8..9fb5bbe 100644
--- a/drm/mediadrm/plugins/clearkey/hidl/DrmFactory.cpp
+++ b/drm/mediadrm/plugins/clearkey/hidl/DrmFactory.cpp
@@ -34,6 +34,7 @@
namespace clearkey {
using ::android::hardware::drm::V1_0::Status;
+using ::android::hardware::drm::V1_1::SecurityLevel;
using ::android::hardware::Void;
Return<bool> DrmFactory::isCryptoSchemeSupported(
@@ -41,6 +42,13 @@
return clearkeydrm::isClearKeyUUID(uuid.data());
}
+Return<bool> DrmFactory::isCryptoSchemeSupported_1_2(const hidl_array<uint8_t, 16>& uuid,
+ const hidl_string &mimeType,
+ SecurityLevel level) {
+ return isCryptoSchemeSupported(uuid) && isContentTypeSupported(mimeType) &&
+ level == SecurityLevel::SW_SECURE_CRYPTO;
+}
+
Return<bool> DrmFactory::isContentTypeSupported(const hidl_string &mimeType) {
// This should match the mimeTypes handed by InitDataParser.
return mimeType == kIsoBmffVideoMimeType ||
diff --git a/drm/mediadrm/plugins/clearkey/hidl/include/ClearKeyTypes.h b/drm/mediadrm/plugins/clearkey/hidl/include/ClearKeyTypes.h
index 2dafa36..03c434e 100644
--- a/drm/mediadrm/plugins/clearkey/hidl/include/ClearKeyTypes.h
+++ b/drm/mediadrm/plugins/clearkey/hidl/include/ClearKeyTypes.h
@@ -28,6 +28,7 @@
namespace clearkey {
using ::android::hardware::drm::V1_0::KeyValue;
+using ::android::hardware::drm::V1_1::SecurityLevel;
using ::android::hardware::hidl_vec;
const uint8_t kBlockSize = 16; //AES_BLOCK_SIZE;
diff --git a/drm/mediadrm/plugins/clearkey/hidl/include/DrmFactory.h b/drm/mediadrm/plugins/clearkey/hidl/include/DrmFactory.h
index ff715ea..4ca856d 100644
--- a/drm/mediadrm/plugins/clearkey/hidl/include/DrmFactory.h
+++ b/drm/mediadrm/plugins/clearkey/hidl/include/DrmFactory.h
@@ -39,6 +39,10 @@
Return<bool> isCryptoSchemeSupported(const hidl_array<uint8_t, 16>& uuid)
override;
+ Return<bool> isCryptoSchemeSupported_1_2(const hidl_array<uint8_t, 16>& uuid,
+ const hidl_string& mimeType,
+ SecurityLevel level) override;
+
Return<bool> isContentTypeSupported(const hidl_string &mimeType)
override;
diff --git a/drm/mediadrm/plugins/clearkey/hidl/include/DrmPlugin.h b/drm/mediadrm/plugins/clearkey/hidl/include/DrmPlugin.h
index a9b897b..ba5fa65 100644
--- a/drm/mediadrm/plugins/clearkey/hidl/include/DrmPlugin.h
+++ b/drm/mediadrm/plugins/clearkey/hidl/include/DrmPlugin.h
@@ -63,6 +63,7 @@
typedef drm::V1_1::KeyRequestType KeyRequestType_V1_1;
typedef drm::V1_2::IDrmPluginListener IDrmPluginListener_V1_2;
typedef drm::V1_2::Status Status_V1_2;
+typedef drm::V1_2::HdcpLevel HdcpLevel_V1_2;
struct DrmPlugin : public IDrmPlugin {
explicit DrmPlugin(SessionLibrary* sessionLibrary);
@@ -162,6 +163,13 @@
return Void();
}
+ Return<void> getHdcpLevels_1_2(getHdcpLevels_1_2_cb _hidl_cb) {
+ HdcpLevel_V1_2 connectedLevel = HdcpLevel_V1_2::HDCP_NONE;
+ HdcpLevel_V1_2 maxLevel = HdcpLevel_V1_2::HDCP_NO_OUTPUT;
+ _hidl_cb(Status_V1_2::OK, connectedLevel, maxLevel);
+ return Void();
+ }
+
Return<void> getNumberOfSessions(getNumberOfSessions_cb _hidl_cb) override;
Return<void> getSecurityLevel(const hidl_vec<uint8_t>& sessionId,
diff --git a/include/media/MediaExtractorPluginHelper.h b/include/media/MediaExtractorPluginHelper.h
index f4d4da6..b86f177 100644
--- a/include/media/MediaExtractorPluginHelper.h
+++ b/include/media/MediaExtractorPluginHelper.h
@@ -171,6 +171,9 @@
};
inline CMediaTrack *wrap(MediaTrackHelper *track) {
+ if (track == nullptr) {
+ return nullptr;
+ }
CMediaTrack *wrapper = (CMediaTrack*) malloc(sizeof(CMediaTrack));
wrapper->data = track;
wrapper->free = [](void *data) -> void {
diff --git a/include/media/MediaTrack.h b/include/media/MediaTrack.h
index e828a7f..493eba3 100644
--- a/include/media/MediaTrack.h
+++ b/include/media/MediaTrack.h
@@ -142,7 +142,7 @@
class MediaTrackCUnwrapper : public MediaTrack {
public:
- explicit MediaTrackCUnwrapper(CMediaTrack *wrapper);
+ static MediaTrackCUnwrapper *create(CMediaTrack *wrapper);
virtual status_t start();
virtual status_t stop();
@@ -155,6 +155,7 @@
virtual ~MediaTrackCUnwrapper();
private:
+ explicit MediaTrackCUnwrapper(CMediaTrack *wrapper);
CMediaTrack *wrapper;
MediaBufferGroup *bufferGroup;
};
diff --git a/media/codec2/components/opus/Android.bp b/media/codec2/components/opus/Android.bp
index 240cdb9..0ed141b 100644
--- a/media/codec2/components/opus/Android.bp
+++ b/media/codec2/components/opus/Android.bp
@@ -9,3 +9,14 @@
shared_libs: ["libopus"],
}
+cc_library_shared {
+ name: "libcodec2_soft_opusenc",
+ defaults: [
+ "libcodec2_soft-defaults",
+ "libcodec2_soft_sanitize_all-defaults",
+ ],
+
+ srcs: ["C2SoftOpusEnc.cpp"],
+
+ shared_libs: ["libopus"],
+}
diff --git a/media/codec2/components/opus/C2SoftOpusDec.cpp b/media/codec2/components/opus/C2SoftOpusDec.cpp
index 2439c3c..3ce1fd6 100644
--- a/media/codec2/components/opus/C2SoftOpusDec.cpp
+++ b/media/codec2/components/opus/C2SoftOpusDec.cpp
@@ -19,10 +19,9 @@
#include <log/log.h>
#include <media/stagefright/foundation/MediaDefs.h>
-
+#include <media/stagefright/foundation/OpusHeader.h>
#include <C2PlatformSupport.h>
#include <SimpleC2Interface.h>
-
#include "C2SoftOpusDec.h"
extern "C" {
@@ -188,16 +187,6 @@
work->workletsProcessed = 1u;
}
-static uint16_t ReadLE16(const uint8_t *data, size_t data_size,
- uint32_t read_offset) {
- if (read_offset + 1 > data_size)
- return 0;
- uint16_t val;
- val = data[read_offset];
- val |= data[read_offset + 1] << 8;
- return val;
-}
-
static const int kRate = 48000;
// Opus uses Vorbis channel mapping, and Vorbis channel mapping specifies
@@ -216,81 +205,6 @@
static const int kMaxChannelsWithDefaultLayout = 2;
static const uint8_t kDefaultOpusChannelLayout[kMaxChannelsWithDefaultLayout] = { 0, 1 };
-// Parses Opus Header. Header spec: http://wiki.xiph.org/OggOpus#ID_Header
-static bool ParseOpusHeader(const uint8_t *data, size_t data_size,
- OpusHeader* header) {
- // Size of the Opus header excluding optional mapping information.
- const size_t kOpusHeaderSize = 19;
-
- // Offset to the channel count byte in the Opus header.
- const size_t kOpusHeaderChannelsOffset = 9;
-
- // Offset to the pre-skip value in the Opus header.
- const size_t kOpusHeaderSkipSamplesOffset = 10;
-
- // Offset to the gain value in the Opus header.
- const size_t kOpusHeaderGainOffset = 16;
-
- // Offset to the channel mapping byte in the Opus header.
- const size_t kOpusHeaderChannelMappingOffset = 18;
-
- // Opus Header contains a stream map. The mapping values are in the header
- // beyond the always present |kOpusHeaderSize| bytes of data. The mapping
- // data contains stream count, coupling information, and per channel mapping
- // values:
- // - Byte 0: Number of streams.
- // - Byte 1: Number coupled.
- // - Byte 2: Starting at byte 2 are |header->channels| uint8 mapping
- // values.
- const size_t kOpusHeaderNumStreamsOffset = kOpusHeaderSize;
- const size_t kOpusHeaderNumCoupledOffset = kOpusHeaderNumStreamsOffset + 1;
- const size_t kOpusHeaderStreamMapOffset = kOpusHeaderNumStreamsOffset + 2;
-
- if (data_size < kOpusHeaderSize) {
- ALOGE("Header size is too small.");
- return false;
- }
- header->channels = *(data + kOpusHeaderChannelsOffset);
- if (header->channels <= 0 || header->channels > kMaxChannels) {
- ALOGE("Invalid Header, wrong channel count: %d", header->channels);
- return false;
- }
-
- header->skip_samples = ReadLE16(data,
- data_size,
- kOpusHeaderSkipSamplesOffset);
-
- header->gain_db = static_cast<int16_t>(ReadLE16(data,
- data_size,
- kOpusHeaderGainOffset));
-
- header->channel_mapping = *(data + kOpusHeaderChannelMappingOffset);
- if (!header->channel_mapping) {
- if (header->channels > kMaxChannelsWithDefaultLayout) {
- ALOGE("Invalid Header, missing stream map.");
- return false;
- }
- header->num_streams = 1;
- header->num_coupled = header->channels > 1;
- header->stream_map[0] = 0;
- header->stream_map[1] = 1;
- return true;
- }
- if (data_size < kOpusHeaderStreamMapOffset + header->channels) {
- ALOGE("Invalid stream map; insufficient data for current channel "
- "count: %d", header->channels);
- return false;
- }
- header->num_streams = *(data + kOpusHeaderNumStreamsOffset);
- header->num_coupled = *(data + kOpusHeaderNumCoupledOffset);
- if (header->num_streams + header->num_coupled != header->channels) {
- ALOGE("Inconsistent channel mapping.");
- return false;
- }
- for (int i = 0; i < header->channels; ++i)
- header->stream_map[i] = *(data + kOpusHeaderStreamMapOffset + i);
- return true;
-}
// Convert nanoseconds to number of samples.
static uint64_t ns_to_samples(uint64_t ns, int rate) {
@@ -338,7 +252,19 @@
const uint8_t *data = rView.data() + inOffset;
if (mInputBufferCount < 3) {
if (mInputBufferCount == 0) {
- if (!ParseOpusHeader(data, inSize, &mHeader)) {
+ size_t opusHeadSize = inSize;
+ size_t codecDelayBufSize = 0;
+ size_t seekPreRollBufSize = 0;
+ void *opusHeadBuf = (void *)data;
+ void *codecDelayBuf = NULL;
+ void *seekPreRollBuf = NULL;
+
+ GetOpusHeaderBuffers(data, inSize, &opusHeadBuf,
+ &opusHeadSize, &codecDelayBuf,
+ &codecDelayBufSize, &seekPreRollBuf,
+ &seekPreRollBufSize);
+
+ if (!ParseOpusHeader((uint8_t *)opusHeadBuf, opusHeadSize, &mHeader)) {
ALOGE("Encountered error while Parsing Opus Header.");
mSignalledError = true;
work->result = C2_CORRUPTED;
@@ -377,6 +303,20 @@
work->result = C2_CORRUPTED;
return;
}
+
+ if (codecDelayBuf && codecDelayBufSize == 8) {
+ uint64_t value;
+ memcpy(&value, codecDelayBuf, sizeof(uint64_t));
+ mCodecDelay = ns_to_samples(value, kRate);
+ mSamplesToDiscard = mCodecDelay;
+ ++mInputBufferCount;
+ }
+ if (seekPreRollBuf && seekPreRollBufSize == 8) {
+ uint64_t value;
+ memcpy(&value, codecDelayBuf, sizeof(uint64_t));
+ mSeekPreRoll = ns_to_samples(value, kRate);
+ ++mInputBufferCount;
+ }
} else {
if (inSize < 8) {
ALOGE("Input sample size is too small.");
@@ -392,29 +332,30 @@
}
else {
mSeekPreRoll = samples;
-
- ALOGI("Configuring decoder: %d Hz, %d channels",
- kRate, mHeader.channels);
- C2StreamSampleRateInfo::output sampleRateInfo(0u, kRate);
- C2StreamChannelCountInfo::output channelCountInfo(0u, mHeader.channels);
- std::vector<std::unique_ptr<C2SettingResult>> failures;
- c2_status_t err = mIntf->config(
- { &sampleRateInfo, &channelCountInfo },
- C2_MAY_BLOCK,
- &failures);
- if (err == OK) {
- work->worklets.front()->output.configUpdate.push_back(C2Param::Copy(sampleRateInfo));
- work->worklets.front()->output.configUpdate.push_back(C2Param::Copy(channelCountInfo));
- } else {
- ALOGE("Config Update failed");
- mSignalledError = true;
- work->result = C2_CORRUPTED;
- return;
- }
}
}
++mInputBufferCount;
+ if (mInputBufferCount == 3) {
+ ALOGI("Configuring decoder: %d Hz, %d channels",
+ kRate, mHeader.channels);
+ C2StreamSampleRateInfo::output sampleRateInfo(0u, kRate);
+ C2StreamChannelCountInfo::output channelCountInfo(0u, mHeader.channels);
+ std::vector<std::unique_ptr<C2SettingResult>> failures;
+ c2_status_t err = mIntf->config(
+ { &sampleRateInfo, &channelCountInfo },
+ C2_MAY_BLOCK,
+ &failures);
+ if (err == OK) {
+ work->worklets.front()->output.configUpdate.push_back(C2Param::Copy(sampleRateInfo));
+ work->worklets.front()->output.configUpdate.push_back(C2Param::Copy(channelCountInfo));
+ } else {
+ ALOGE("Config Update failed");
+ mSignalledError = true;
+ work->result = C2_CORRUPTED;
+ return;
+ }
+ }
fillEmptyWork(work);
if (eos) {
mSignalledOutputEos = true;
diff --git a/media/codec2/components/opus/C2SoftOpusDec.h b/media/codec2/components/opus/C2SoftOpusDec.h
index 92b7426..b0715ac 100644
--- a/media/codec2/components/opus/C2SoftOpusDec.h
+++ b/media/codec2/components/opus/C2SoftOpusDec.h
@@ -24,16 +24,6 @@
namespace android {
-struct OpusHeader {
- int channels;
- int skip_samples;
- int channel_mapping;
- int num_streams;
- int num_coupled;
- int16_t gain_db;
- uint8_t stream_map[8];
-};
-
struct C2SoftOpusDec : public SimpleC2Component {
class IntfImpl;
diff --git a/media/codec2/components/opus/C2SoftOpusEnc.cpp b/media/codec2/components/opus/C2SoftOpusEnc.cpp
new file mode 100644
index 0000000..d6ed5ff
--- /dev/null
+++ b/media/codec2/components/opus/C2SoftOpusEnc.cpp
@@ -0,0 +1,638 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "C2SoftOpusEnc"
+#include <utils/Log.h>
+
+#include <C2PlatformSupport.h>
+#include <SimpleC2Interface.h>
+#include <media/stagefright/foundation/MediaDefs.h>
+#include <media/stagefright/foundation/OpusHeader.h>
+#include "C2SoftOpusEnc.h"
+
+extern "C" {
+ #include <opus.h>
+ #include <opus_multistream.h>
+}
+
+#define DEFAULT_FRAME_DURATION_MS 20
+namespace android {
+
+constexpr char COMPONENT_NAME[] = "c2.android.opus.encoder";
+
+class C2SoftOpusEnc::IntfImpl : public C2InterfaceHelper {
+public:
+ explicit IntfImpl(const std::shared_ptr<C2ReflectorHelper> &helper)
+ : C2InterfaceHelper(helper) {
+
+ setDerivedInstance(this);
+
+ addParameter(
+ DefineParam(mInputFormat, C2_NAME_INPUT_STREAM_FORMAT_SETTING)
+ .withConstValue(new C2StreamFormatConfig::input(0u, C2FormatAudio))
+ .build());
+
+ addParameter(
+ DefineParam(mOutputFormat, C2_NAME_OUTPUT_STREAM_FORMAT_SETTING)
+ .withConstValue(new C2StreamFormatConfig::output(0u, C2FormatCompressed))
+ .build());
+
+ addParameter(
+ DefineParam(mInputMediaType, C2_NAME_INPUT_PORT_MIME_SETTING)
+ .withConstValue(AllocSharedString<C2PortMimeConfig::input>(
+ MEDIA_MIMETYPE_AUDIO_RAW))
+ .build());
+
+ addParameter(
+ DefineParam(mOutputMediaType, C2_NAME_OUTPUT_PORT_MIME_SETTING)
+ .withConstValue(AllocSharedString<C2PortMimeConfig::output>(
+ MEDIA_MIMETYPE_AUDIO_OPUS))
+ .build());
+
+ addParameter(
+ DefineParam(mSampleRate, C2_NAME_STREAM_SAMPLE_RATE_SETTING)
+ .withDefault(new C2StreamSampleRateInfo::input(0u, 48000))
+ .withFields({C2F(mSampleRate, value).oneOf({
+ 8000, 12000, 16000, 24000, 48000})})
+ .withSetter((Setter<decltype(*mSampleRate)>::StrictValueWithNoDeps))
+ .build());
+
+ addParameter(
+ DefineParam(mChannelCount, C2_NAME_STREAM_CHANNEL_COUNT_SETTING)
+ .withDefault(new C2StreamChannelCountInfo::input(0u, 1))
+ .withFields({C2F(mChannelCount, value).inRange(1, 8)})
+ .withSetter((Setter<decltype(*mChannelCount)>::StrictValueWithNoDeps))
+ .build());
+
+ addParameter(
+ DefineParam(mBitrate, C2_NAME_STREAM_BITRATE_SETTING)
+ .withDefault(new C2BitrateTuning::output(0u, 128000))
+ .withFields({C2F(mBitrate, value).inRange(500, 512000)})
+ .withSetter(Setter<decltype(*mBitrate)>::NonStrictValueWithNoDeps)
+ .build());
+
+ addParameter(
+ DefineParam(mComplexity, C2_PARAMKEY_COMPLEXITY)
+ .withDefault(new C2StreamComplexityTuning::output(0u, 10))
+ .withFields({C2F(mComplexity, value).inRange(1, 10)})
+ .withSetter(Setter<decltype(*mComplexity)>::NonStrictValueWithNoDeps)
+ .build());
+
+ addParameter(
+ DefineParam(mInputMaxBufSize, C2_PARAMKEY_INPUT_MAX_BUFFER_SIZE)
+ .withConstValue(new C2StreamMaxBufferSizeInfo::input(0u, 3840))
+ .build());
+ }
+
+ uint32_t getSampleRate() const { return mSampleRate->value; }
+ uint32_t getChannelCount() const { return mChannelCount->value; }
+ uint32_t getBitrate() const { return mBitrate->value; }
+ uint32_t getComplexity() const { return mComplexity->value; }
+
+private:
+ std::shared_ptr<C2StreamFormatConfig::input> mInputFormat;
+ std::shared_ptr<C2StreamFormatConfig::output> mOutputFormat;
+ std::shared_ptr<C2PortMimeConfig::input> mInputMediaType;
+ std::shared_ptr<C2PortMimeConfig::output> mOutputMediaType;
+ std::shared_ptr<C2StreamSampleRateInfo::input> mSampleRate;
+ std::shared_ptr<C2StreamChannelCountInfo::input> mChannelCount;
+ std::shared_ptr<C2BitrateTuning::output> mBitrate;
+ std::shared_ptr<C2StreamComplexityTuning::output> mComplexity;
+ std::shared_ptr<C2StreamMaxBufferSizeInfo::input> mInputMaxBufSize;
+};
+
+C2SoftOpusEnc::C2SoftOpusEnc(const char* name, c2_node_id_t id,
+ const std::shared_ptr<IntfImpl>& intfImpl)
+ : SimpleC2Component(
+ std::make_shared<SimpleInterface<IntfImpl>>(name, id, intfImpl)),
+ mIntf(intfImpl),
+ mOutputBlock(nullptr),
+ mEncoder(nullptr),
+ mInputBufferPcm16(nullptr),
+ mOutIndex(0u) {
+}
+
+C2SoftOpusEnc::~C2SoftOpusEnc() {
+ onRelease();
+}
+
+c2_status_t C2SoftOpusEnc::onInit() {
+ return initEncoder();
+}
+
+c2_status_t C2SoftOpusEnc::configureEncoder() {
+ unsigned char mono_mapping[256] = {0};
+ unsigned char stereo_mapping[256] = {0, 1};
+ unsigned char surround_mapping[256] = {0, 1, 255};
+ mSampleRate = mIntf->getSampleRate();
+ mChannelCount = mIntf->getChannelCount();
+ uint32_t bitrate = mIntf->getBitrate();
+ int complexity = mIntf->getComplexity();
+ mNumSamplesPerFrame = mSampleRate / (1000 / mFrameDurationMs);
+ mNumPcmBytesPerInputFrame =
+ mChannelCount * mNumSamplesPerFrame * sizeof(int16_t);
+ int err = C2_OK;
+
+ unsigned char* mapping;
+ if (mChannelCount < 2) {
+ mapping = mono_mapping;
+ } else if (mChannelCount == 2) {
+ mapping = stereo_mapping;
+ } else {
+ mapping = surround_mapping;
+ }
+
+ if (mEncoder != nullptr) {
+ opus_multistream_encoder_destroy(mEncoder);
+ }
+
+ mEncoder = opus_multistream_encoder_create(mSampleRate, mChannelCount,
+ 1, 1, mapping, OPUS_APPLICATION_AUDIO, &err);
+ if (err) {
+ ALOGE("Could not create libopus encoder. Error code: %i", err);
+ return C2_CORRUPTED;
+ }
+
+ // Complexity
+ if (opus_multistream_encoder_ctl(
+ mEncoder, OPUS_SET_COMPLEXITY(complexity)) != OPUS_OK) {
+ ALOGE("failed to set complexity");
+ return C2_BAD_VALUE;
+ }
+
+ // DTX
+ if (opus_multistream_encoder_ctl(mEncoder, OPUS_SET_DTX(0) != OPUS_OK)) {
+ ALOGE("failed to set dtx");
+ return C2_BAD_VALUE;
+ }
+
+ // Application
+ if (opus_multistream_encoder_ctl(mEncoder,
+ OPUS_SET_APPLICATION(OPUS_APPLICATION_AUDIO)) != OPUS_OK) {
+ ALOGE("failed to set application");
+ return C2_BAD_VALUE;
+ }
+
+ // Signal type
+ if (opus_multistream_encoder_ctl(mEncoder, OPUS_SET_SIGNAL(OPUS_AUTO)) !=
+ OPUS_OK) {
+ ALOGE("failed to set signal");
+ return C2_BAD_VALUE;
+ }
+
+ // Unconstrained VBR
+ if (opus_multistream_encoder_ctl(mEncoder, OPUS_SET_VBR(0) != OPUS_OK)) {
+ ALOGE("failed to set vbr type");
+ return C2_BAD_VALUE;
+ }
+ if (opus_multistream_encoder_ctl(mEncoder, OPUS_SET_VBR_CONSTRAINT(0) !=
+ OPUS_OK)) {
+ ALOGE("failed to set vbr constraint");
+ return C2_BAD_VALUE;
+ }
+
+ // Bitrate
+ if (opus_multistream_encoder_ctl(mEncoder, OPUS_SET_BITRATE(bitrate)) !=
+ OPUS_OK) {
+ ALOGE("failed to set bitrate");
+ return C2_BAD_VALUE;
+ }
+
+ // Get codecDelay
+ int32_t lookahead;
+ if (opus_multistream_encoder_ctl(mEncoder, OPUS_GET_LOOKAHEAD(&lookahead)) !=
+ OPUS_OK) {
+ ALOGE("failed to get lookahead");
+ return C2_BAD_VALUE;
+ }
+ mCodecDelay = lookahead * 1000000000ll / mSampleRate;
+
+ // Set seek preroll to 80 ms
+ mSeekPreRoll = 80000000;
+ return C2_OK;
+}
+
+c2_status_t C2SoftOpusEnc::initEncoder() {
+ mSignalledEos = false;
+ mSignalledError = false;
+ mHeaderGenerated = false;
+ mIsFirstFrame = true;
+ mEncoderFlushed = false;
+ mBufferAvailable = false;
+ mAnchorTimeStamp = 0ull;
+ mProcessedSamples = 0;
+ mFilledLen = 0;
+ mFrameDurationMs = DEFAULT_FRAME_DURATION_MS;
+ if (!mInputBufferPcm16) {
+ mInputBufferPcm16 =
+ (int16_t*)malloc(kFrameSize * kMaxNumChannels * sizeof(int16_t));
+ }
+ if (!mInputBufferPcm16) return C2_NO_MEMORY;
+
+ /* Default Configurations */
+ c2_status_t status = configureEncoder();
+ return status;
+}
+
+c2_status_t C2SoftOpusEnc::onStop() {
+ mSignalledEos = false;
+ mSignalledError = false;
+ mIsFirstFrame = true;
+ mEncoderFlushed = false;
+ mBufferAvailable = false;
+ mAnchorTimeStamp = 0ull;
+ mProcessedSamples = 0u;
+ mFilledLen = 0;
+ if (mEncoder) {
+ int status = opus_multistream_encoder_ctl(mEncoder, OPUS_RESET_STATE);
+ if (status != OPUS_OK) {
+ ALOGE("OPUS_RESET_STATE failed status = %s", opus_strerror(status));
+ mSignalledError = true;
+ return C2_CORRUPTED;
+ }
+ }
+ if (mOutputBlock) mOutputBlock.reset();
+ mOutputBlock = nullptr;
+
+ return C2_OK;
+}
+
+void C2SoftOpusEnc::onReset() {
+ (void)onStop();
+}
+
+void C2SoftOpusEnc::onRelease() {
+ (void)onStop();
+ if (mInputBufferPcm16) {
+ free(mInputBufferPcm16);
+ mInputBufferPcm16 = nullptr;
+ }
+ if (mEncoder) {
+ opus_multistream_encoder_destroy(mEncoder);
+ mEncoder = nullptr;
+ }
+}
+
+c2_status_t C2SoftOpusEnc::onFlush_sm() {
+ return onStop();
+}
+
+// Drain the encoder to get last frames (if any)
+int C2SoftOpusEnc::drainEncoder(uint8_t* outPtr) {
+ memset((uint8_t *)mInputBufferPcm16 + mFilledLen, 0,
+ (mNumPcmBytesPerInputFrame - mFilledLen));
+ int encodedBytes = opus_multistream_encode(
+ mEncoder, mInputBufferPcm16, mNumSamplesPerFrame, outPtr, kMaxPayload);
+ if (encodedBytes > mOutputBlock->capacity()) {
+ ALOGE("not enough space left to write encoded data, dropping %d bytes",
+ mBytesEncoded);
+ // a fatal error would stop the encoding
+ return -1;
+ }
+ ALOGV("encoded %i Opus bytes from %zu PCM bytes", encodedBytes,
+ mNumPcmBytesPerInputFrame);
+ mEncoderFlushed = true;
+ mFilledLen = 0;
+ return encodedBytes;
+}
+
+void C2SoftOpusEnc::process(const std::unique_ptr<C2Work>& work,
+ const std::shared_ptr<C2BlockPool>& pool) {
+ // Initialize output work
+ work->result = C2_OK;
+ work->workletsProcessed = 1u;
+ work->worklets.front()->output.flags = work->input.flags;
+
+ if (mSignalledError || mSignalledEos) {
+ work->result = C2_BAD_VALUE;
+ return;
+ }
+
+ bool eos = (work->input.flags & C2FrameData::FLAG_END_OF_STREAM) != 0;
+ C2ReadView rView = mDummyReadView;
+ size_t inOffset = 0u;
+ size_t inSize = 0u;
+ c2_status_t err = C2_OK;
+ if (!work->input.buffers.empty()) {
+ rView =
+ work->input.buffers[0]->data().linearBlocks().front().map().get();
+ inSize = rView.capacity();
+ if (inSize && rView.error()) {
+ ALOGE("read view map failed %d", rView.error());
+ work->result = C2_CORRUPTED;
+ return;
+ }
+ }
+
+ ALOGV("in buffer attr. size %zu timestamp %d frameindex %d, flags %x",
+ inSize, (int)work->input.ordinal.timestamp.peeku(),
+ (int)work->input.ordinal.frameIndex.peeku(), work->input.flags);
+
+ if (!mEncoder) {
+ if (initEncoder() != C2_OK) {
+ ALOGE("initEncoder failed with status %d", err);
+ work->result = err;
+ mSignalledError = true;
+ return;
+ }
+ }
+ if (mIsFirstFrame) {
+ mAnchorTimeStamp = work->input.ordinal.timestamp.peekull();
+ mIsFirstFrame = false;
+ }
+
+ C2MemoryUsage usage = {C2MemoryUsage::CPU_READ, C2MemoryUsage::CPU_WRITE};
+ err = pool->fetchLinearBlock(kMaxPayload, usage, &mOutputBlock);
+ if (err != C2_OK) {
+ ALOGE("fetchLinearBlock for Output failed with status %d", err);
+ work->result = C2_NO_MEMORY;
+ return;
+ }
+
+ C2WriteView wView = mOutputBlock->map().get();
+ if (wView.error()) {
+ ALOGE("write view map failed %d", wView.error());
+ work->result = C2_CORRUPTED;
+ mOutputBlock.reset();
+ return;
+ }
+
+ size_t inPos = 0;
+ size_t processSize = 0;
+ mBytesEncoded = 0;
+ uint64_t outTimeStamp = 0u;
+ std::shared_ptr<C2Buffer> buffer;
+ uint64_t inputIndex = work->input.ordinal.frameIndex.peeku();
+ const uint8_t* inPtr = rView.data() + inOffset;
+
+ class FillWork {
+ public:
+ FillWork(uint32_t flags, C2WorkOrdinalStruct ordinal,
+ const std::shared_ptr<C2Buffer> &buffer)
+ : mFlags(flags), mOrdinal(ordinal), mBuffer(buffer) {
+ }
+ ~FillWork() = default;
+
+ void operator()(const std::unique_ptr<C2Work>& work) {
+ work->worklets.front()->output.flags = (C2FrameData::flags_t)mFlags;
+ work->worklets.front()->output.buffers.clear();
+ work->worklets.front()->output.ordinal = mOrdinal;
+ work->workletsProcessed = 1u;
+ work->result = C2_OK;
+ if (mBuffer) {
+ work->worklets.front()->output.buffers.push_back(mBuffer);
+ }
+ ALOGV("timestamp = %lld, index = %lld, w/%s buffer",
+ mOrdinal.timestamp.peekll(),
+ mOrdinal.frameIndex.peekll(),
+ mBuffer ? "" : "o");
+ }
+
+ private:
+ const uint32_t mFlags;
+ const C2WorkOrdinalStruct mOrdinal;
+ const std::shared_ptr<C2Buffer> mBuffer;
+ };
+
+ C2WorkOrdinalStruct outOrdinal = work->input.ordinal;
+
+ if (!mHeaderGenerated) {
+ uint8_t header[AOPUS_UNIFIED_CSD_MAXSIZE];
+ memset(header, 0, sizeof(header));
+ OpusHeader opusHeader;
+ opusHeader.channels = mChannelCount;
+ opusHeader.num_streams = mChannelCount;
+ opusHeader.num_coupled = 0;
+ opusHeader.channel_mapping = ((mChannelCount > 8) ? 255 : (mChannelCount > 2));
+ opusHeader.gain_db = 0;
+ opusHeader.skip_samples = 0;
+ int headerLen = WriteOpusHeaders(opusHeader, mSampleRate, header,
+ sizeof(header), mCodecDelay, mSeekPreRoll);
+
+ std::unique_ptr<C2StreamCsdInfo::output> csd =
+ C2StreamCsdInfo::output::AllocUnique(headerLen, 0u);
+ if (!csd) {
+ ALOGE("CSD allocation failed");
+ mSignalledError = true;
+ work->result = C2_NO_MEMORY;
+ return;
+ }
+ ALOGV("put csd, %d bytes", headerLen);
+ memcpy(csd->m.value, header, headerLen);
+ work->worklets.front()->output.configUpdate.push_back(std::move(csd));
+ mHeaderGenerated = true;
+ }
+
+ /*
+ * For buffer size which is not a multiple of mNumPcmBytesPerInputFrame, we will
+ * accumulate the input and keep it. Once the input is filled with expected number
+ * of bytes, we will send it to encoder. mFilledLen manages the bytes of input yet
+ * to be processed. The next call will fill mNumPcmBytesPerInputFrame - mFilledLen
+ * bytes to input and send it to the encoder.
+ */
+ while (inPos < inSize) {
+ const uint8_t* pcmBytes = inPtr + inPos;
+ int filledSamples = mFilledLen / sizeof(int16_t);
+ if ((inPos + (mNumPcmBytesPerInputFrame - mFilledLen)) <= inSize) {
+ processSize = mNumPcmBytesPerInputFrame - mFilledLen;
+ mBufferAvailable = true;
+ } else {
+ processSize = inSize - inPos;
+ mBufferAvailable = false;
+ if (eos) {
+ memset(mInputBufferPcm16 + filledSamples, 0,
+ (mNumPcmBytesPerInputFrame - mFilledLen));
+ mBufferAvailable = true;
+ }
+ }
+ const unsigned nInputSamples = processSize / sizeof(int16_t);
+
+ for (unsigned i = 0; i < nInputSamples; i++) {
+ int32_t data = pcmBytes[2 * i + 1] << 8 | pcmBytes[2 * i];
+ data = ((data & 0xFFFF) ^ 0x8000) - 0x8000;
+ mInputBufferPcm16[i + filledSamples] = data;
+ }
+ inPos += processSize;
+ mFilledLen += processSize;
+ if (!mBufferAvailable) break;
+ uint8_t* outPtr = wView.data() + mBytesEncoded;
+ int encodedBytes =
+ opus_multistream_encode(mEncoder, mInputBufferPcm16,
+ mNumSamplesPerFrame, outPtr, kMaxPayload);
+ ALOGV("encoded %i Opus bytes from %zu PCM bytes", encodedBytes,
+ processSize);
+
+ if (encodedBytes < 0 || encodedBytes > kMaxPayload) {
+ ALOGE("opus_encode failed, encodedBytes : %d", encodedBytes);
+ mSignalledError = true;
+ work->result = C2_CORRUPTED;
+ return;
+ }
+ if (buffer) {
+ outOrdinal.frameIndex = mOutIndex++;
+ outOrdinal.timestamp = mAnchorTimeStamp + outTimeStamp;
+ cloneAndSend(
+ inputIndex, work,
+ FillWork(C2FrameData::FLAG_INCOMPLETE, outOrdinal, buffer));
+ buffer.reset();
+ }
+ if (encodedBytes > 0) {
+ buffer =
+ createLinearBuffer(mOutputBlock, mBytesEncoded, encodedBytes);
+ }
+ mBytesEncoded += encodedBytes;
+ mProcessedSamples += (filledSamples + nInputSamples);
+ outTimeStamp =
+ mProcessedSamples * 1000000ll / mChannelCount / mSampleRate;
+ if ((processSize + mFilledLen) < mNumPcmBytesPerInputFrame)
+ mEncoderFlushed = true;
+ mFilledLen = 0;
+ }
+
+ uint32_t flags = 0;
+ if (eos) {
+ ALOGV("signalled eos");
+ mSignalledEos = true;
+ if (!mEncoderFlushed) {
+ if (buffer) {
+ outOrdinal.frameIndex = mOutIndex++;
+ outOrdinal.timestamp = mAnchorTimeStamp + outTimeStamp;
+ cloneAndSend(
+ inputIndex, work,
+ FillWork(C2FrameData::FLAG_INCOMPLETE, outOrdinal, buffer));
+ buffer.reset();
+ }
+ // drain the encoder for last buffer
+ drainInternal(pool, work);
+ }
+ flags = C2FrameData::FLAG_END_OF_STREAM;
+ }
+ if (buffer) {
+ outOrdinal.frameIndex = mOutIndex++;
+ outOrdinal.timestamp = mAnchorTimeStamp + outTimeStamp;
+ FillWork((C2FrameData::flags_t)(flags), outOrdinal, buffer)(work);
+ buffer.reset();
+ }
+ mOutputBlock = nullptr;
+}
+
+c2_status_t C2SoftOpusEnc::drainInternal(
+ const std::shared_ptr<C2BlockPool>& pool,
+ const std::unique_ptr<C2Work>& work) {
+ mBytesEncoded = 0;
+ std::shared_ptr<C2Buffer> buffer = nullptr;
+ C2WorkOrdinalStruct outOrdinal = work->input.ordinal;
+ bool eos = (work->input.flags & C2FrameData::FLAG_END_OF_STREAM) != 0;
+
+ C2MemoryUsage usage = {C2MemoryUsage::CPU_READ, C2MemoryUsage::CPU_WRITE};
+ c2_status_t err = pool->fetchLinearBlock(kMaxPayload, usage, &mOutputBlock);
+ if (err != C2_OK) {
+ ALOGE("fetchLinearBlock for Output failed with status %d", err);
+ return C2_NO_MEMORY;
+ }
+
+ C2WriteView wView = mOutputBlock->map().get();
+ if (wView.error()) {
+ ALOGE("write view map failed %d", wView.error());
+ mOutputBlock.reset();
+ return C2_CORRUPTED;
+ }
+
+ int encBytes = drainEncoder(wView.data());
+ if (encBytes > 0) mBytesEncoded += encBytes;
+ if (mBytesEncoded > 0) {
+ buffer = createLinearBuffer(mOutputBlock, 0, mBytesEncoded);
+ mOutputBlock.reset();
+ }
+ mProcessedSamples += (mNumPcmBytesPerInputFrame / sizeof(int16_t));
+ uint64_t outTimeStamp =
+ mProcessedSamples * 1000000ll / mChannelCount / mSampleRate;
+ outOrdinal.frameIndex = mOutIndex++;
+ outOrdinal.timestamp = mAnchorTimeStamp + outTimeStamp;
+ work->worklets.front()->output.flags =
+ (C2FrameData::flags_t)(eos ? C2FrameData::FLAG_END_OF_STREAM : 0);
+ work->worklets.front()->output.buffers.clear();
+ work->worklets.front()->output.ordinal = outOrdinal;
+ work->workletsProcessed = 1u;
+ work->result = C2_OK;
+ if (buffer) {
+ work->worklets.front()->output.buffers.push_back(buffer);
+ }
+ mOutputBlock = nullptr;
+ return C2_OK;
+}
+
+c2_status_t C2SoftOpusEnc::drain(uint32_t drainMode,
+ const std::shared_ptr<C2BlockPool>& pool) {
+ if (drainMode == NO_DRAIN) {
+ ALOGW("drain with NO_DRAIN: no-op");
+ return C2_OK;
+ }
+ if (drainMode == DRAIN_CHAIN) {
+ ALOGW("DRAIN_CHAIN not supported");
+ return C2_OMITTED;
+ }
+ mIsFirstFrame = true;
+ mAnchorTimeStamp = 0ull;
+ mProcessedSamples = 0u;
+ return drainInternal(pool, nullptr);
+}
+
+class C2SoftOpusEncFactory : public C2ComponentFactory {
+public:
+ C2SoftOpusEncFactory()
+ : mHelper(std::static_pointer_cast<C2ReflectorHelper>(
+ GetCodec2PlatformComponentStore()->getParamReflector())) {}
+
+ virtual c2_status_t createComponent(
+ c2_node_id_t id, std::shared_ptr<C2Component>* const component,
+ std::function<void(C2Component*)> deleter) override {
+ *component = std::shared_ptr<C2Component>(
+ new C2SoftOpusEnc(
+ COMPONENT_NAME, id,
+ std::make_shared<C2SoftOpusEnc::IntfImpl>(mHelper)),
+ deleter);
+ return C2_OK;
+ }
+
+ virtual c2_status_t createInterface(
+ c2_node_id_t id, std::shared_ptr<C2ComponentInterface>* const interface,
+ std::function<void(C2ComponentInterface*)> deleter) override {
+ *interface = std::shared_ptr<C2ComponentInterface>(
+ new SimpleInterface<C2SoftOpusEnc::IntfImpl>(
+ COMPONENT_NAME, id,
+ std::make_shared<C2SoftOpusEnc::IntfImpl>(mHelper)),
+ deleter);
+ return C2_OK;
+ }
+
+ virtual ~C2SoftOpusEncFactory() override = default;
+private:
+ std::shared_ptr<C2ReflectorHelper> mHelper;
+};
+
+} // namespace android
+
+extern "C" ::C2ComponentFactory* CreateCodec2Factory() {
+ ALOGV("in %s", __func__);
+ return new ::android::C2SoftOpusEncFactory();
+}
+
+extern "C" void DestroyCodec2Factory(::C2ComponentFactory* factory) {
+ ALOGV("in %s", __func__);
+ delete factory;
+}
diff --git a/media/codec2/components/opus/C2SoftOpusEnc.h b/media/codec2/components/opus/C2SoftOpusEnc.h
new file mode 100644
index 0000000..69e5240
--- /dev/null
+++ b/media/codec2/components/opus/C2SoftOpusEnc.h
@@ -0,0 +1,90 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_C2_SOFT_OPUS_ENC_H_
+#define ANDROID_C2_SOFT_OPUS_ENC_H_
+
+#include <atomic>
+#include <SimpleC2Component.h>
+#define MIN(a, b) (((a) < (b)) ? (a) : (b))
+
+struct OpusMSEncoder;
+
+namespace android {
+
+struct C2SoftOpusEnc : public SimpleC2Component {
+ class IntfImpl;
+
+ C2SoftOpusEnc(const char *name, c2_node_id_t id,
+ const std::shared_ptr<IntfImpl> &intfImpl);
+ virtual ~C2SoftOpusEnc();
+
+ // From SimpleC2Component
+ c2_status_t onInit() override;
+ c2_status_t onStop() override;
+ void onReset() override;
+ void onRelease() override;
+ c2_status_t onFlush_sm() override;
+ void process(
+ const std::unique_ptr<C2Work> &work,
+ const std::shared_ptr<C2BlockPool> &pool) override;
+ c2_status_t drain(
+ uint32_t drainMode,
+ const std::shared_ptr<C2BlockPool> &pool) override;
+private:
+ /* OPUS_FRAMESIZE_20_MS */
+ const int kFrameSize = 960;
+ const int kMaxPayload = 4000;
+ const int kMaxNumChannels = 8;
+
+ std::shared_ptr<IntfImpl> mIntf;
+ std::shared_ptr<C2LinearBlock> mOutputBlock;
+
+ OpusMSEncoder* mEncoder;
+ int16_t* mInputBufferPcm16;
+
+ bool mHeaderGenerated;
+ bool mIsFirstFrame;
+ bool mEncoderFlushed;
+ bool mBufferAvailable;
+ bool mSignalledEos;
+ bool mSignalledError;
+ uint32_t mSampleRate;
+ uint32_t mChannelCount;
+ uint32_t mFrameDurationMs;
+ uint64_t mAnchorTimeStamp;
+ uint64_t mProcessedSamples;
+ // Codec delay in ns
+ uint64_t mCodecDelay;
+ // Seek pre-roll in ns
+ uint64_t mSeekPreRoll;
+ int mNumSamplesPerFrame;
+ int mBytesEncoded;
+ int32_t mFilledLen;
+ size_t mNumPcmBytesPerInputFrame;
+ std::atomic_uint64_t mOutIndex;
+ c2_status_t initEncoder();
+ c2_status_t configureEncoder();
+ int drainEncoder(uint8_t* outPtr);
+ c2_status_t drainInternal(const std::shared_ptr<C2BlockPool>& pool,
+ const std::unique_ptr<C2Work>& work);
+
+ C2_DO_NOT_COPY(C2SoftOpusEnc);
+};
+
+} // namespace android
+
+#endif // ANDROID_C2_SOFT_OPUS_ENC_H_
diff --git a/media/codec2/core/include/C2Component.h b/media/codec2/core/include/C2Component.h
index 8810725..ecf8d2e 100644
--- a/media/codec2/core/include/C2Component.h
+++ b/media/codec2/core/include/C2Component.h
@@ -409,12 +409,13 @@
kind_t kind; ///< component kind
rank_t rank; ///< component rank
C2String mediaType; ///< media type supported by the component
+ C2String owner; ///< name of the component store owning this component
/**
* name alias(es) for backward compatibility.
* \note Multiple components can have the same alias as long as their media-type differs.
*/
- std::vector<C2StringLiteral> aliases; ///< name aliases for backward compatibility
+ std::vector<C2String> aliases; ///< name aliases for backward compatibility
};
// METHODS AVAILABLE WHEN RUNNING
diff --git a/media/codec2/core/include/C2Config.h b/media/codec2/core/include/C2Config.h
index cf1f6cf..23939b5 100644
--- a/media/codec2/core/include/C2Config.h
+++ b/media/codec2/core/include/C2Config.h
@@ -638,7 +638,7 @@
LEVEL_VP9_6_1, ///< VP9 Level 6.1
LEVEL_VP9_6_2, ///< VP9 Level 6.2
- // Dolby Vision level
+ // Dolby Vision levels
LEVEL_DV_MAIN_HD_24 = _C2_PL_DV_BASE, ///< Dolby Vision main tier hd24
LEVEL_DV_MAIN_HD_30, ///< Dolby Vision main tier hd30
LEVEL_DV_MAIN_FHD_24, ///< Dolby Vision main tier fhd24
@@ -659,6 +659,7 @@
LEVEL_DV_HIGH_UHD_48, ///< Dolby Vision high tier uhd48
LEVEL_DV_HIGH_UHD_60, ///< Dolby Vision high tier uhd60
+ // AV1 levels
LEVEL_AV1_2 = _C2_PL_AV1_BASE , ///< AV1 Level 2
LEVEL_AV1_2_1, ///< AV1 Level 2.1
LEVEL_AV1_2_2, ///< AV1 Level 2.2
diff --git a/media/codec2/hidl/client/client.cpp b/media/codec2/hidl/client/client.cpp
index ff3e534..f5cc9ff 100644
--- a/media/codec2/hidl/client/client.cpp
+++ b/media/codec2/hidl/client/client.cpp
@@ -557,6 +557,7 @@
for (size_t i = 0; i < t.size(); ++i) {
c2_status_t status = objcpy(
&mTraitsList[i], &mAliasesBuffer[i], t[i]);
+ mTraitsList[i].owner = mInstanceName;
if (status != C2_OK) {
ALOGE("listComponents -- corrupted output.");
return;
diff --git a/media/codec2/sfplugin/C2OMXNode.cpp b/media/codec2/sfplugin/C2OMXNode.cpp
index 749fd7a..9500aed 100644
--- a/media/codec2/sfplugin/C2OMXNode.cpp
+++ b/media/codec2/sfplugin/C2OMXNode.cpp
@@ -225,14 +225,18 @@
if (omxBuf.mBufferType == OMXBuffer::kBufferTypeANWBuffer
&& omxBuf.mGraphicBuffer != nullptr) {
std::shared_ptr<C2GraphicAllocation> alloc;
+ native_handle_t *clonedHandle = native_handle_clone(omxBuf.mGraphicBuffer->handle);
handle = WrapNativeCodec2GrallocHandle(
- native_handle_clone(omxBuf.mGraphicBuffer->handle),
+ clonedHandle,
omxBuf.mGraphicBuffer->width,
omxBuf.mGraphicBuffer->height,
omxBuf.mGraphicBuffer->format,
omxBuf.mGraphicBuffer->usage,
omxBuf.mGraphicBuffer->stride);
c2_status_t err = mAllocator->priorGraphicAllocation(handle, &alloc);
+ if (clonedHandle) {
+ native_handle_delete(clonedHandle);
+ }
if (err != OK) {
return UNKNOWN_ERROR;
}
diff --git a/media/codec2/sfplugin/CCodecBufferChannel.cpp b/media/codec2/sfplugin/CCodecBufferChannel.cpp
index 55a97d8..b529cbc 100644
--- a/media/codec2/sfplugin/CCodecBufferChannel.cpp
+++ b/media/codec2/sfplugin/CCodecBufferChannel.cpp
@@ -271,12 +271,8 @@
namespace {
-// TODO: get this info from component
-const static size_t kMinInputBufferArraySize = 4;
-const static size_t kMaxPipelineCapacity = 18;
-const static size_t kChannelOutputDelay = 0;
-const static size_t kMinOutputBufferArraySize = kMaxPipelineCapacity +
- kChannelOutputDelay;
+const static size_t kSmoothnessFactor = 4;
+const static size_t kRenderingDepth = 3;
const static size_t kLinearBufferSize = 1048576;
// This can fit 4K RGBA frame, and most likely client won't need more than this.
const static size_t kMaxLinearBufferSize = 3840 * 2160 * 4;
@@ -829,6 +825,7 @@
const sp<ICrypto> &crypto,
int32_t heapSeqNum,
size_t capacity,
+ size_t numInputSlots,
const char *componentName, const char *name = "EncryptedInput")
: LinearInputBuffers(componentName, name),
mUsage({0, 0}),
@@ -840,7 +837,7 @@
} else {
mUsage = { C2MemoryUsage::CPU_READ, C2MemoryUsage::CPU_WRITE };
}
- for (size_t i = 0; i < kMinInputBufferArraySize; ++i) {
+ for (size_t i = 0; i < numInputSlots; ++i) {
sp<IMemory> memory = mDealer->allocate(capacity);
if (memory == nullptr) {
ALOGD("[%s] Failed to allocate memory from dealer: only %zu slots allocated", mName, i);
@@ -951,11 +948,12 @@
class GraphicInputBuffers : public CCodecBufferChannel::InputBuffers {
public:
- GraphicInputBuffers(const char *componentName, const char *name = "2D-BB-Input")
+ GraphicInputBuffers(
+ size_t numInputSlots, const char *componentName, const char *name = "2D-BB-Input")
: InputBuffers(componentName, name),
mImpl(mName),
mLocalBufferPool(LocalBufferPool::Create(
- kMaxLinearBufferSize * kMinInputBufferArraySize)) { }
+ kMaxLinearBufferSize * numInputSlots)) { }
~GraphicInputBuffers() override = default;
bool requestNewBuffer(size_t *index, sp<MediaCodecBuffer> *buffer) override {
@@ -1291,10 +1289,11 @@
class RawGraphicOutputBuffers : public FlexOutputBuffers {
public:
- RawGraphicOutputBuffers(const char *componentName, const char *name = "2D-BB-Output")
+ RawGraphicOutputBuffers(
+ size_t numOutputSlots, const char *componentName, const char *name = "2D-BB-Output")
: FlexOutputBuffers(componentName, name),
mLocalBufferPool(LocalBufferPool::Create(
- kMaxLinearBufferSize * kMinOutputBufferArraySize)) { }
+ kMaxLinearBufferSize * numOutputSlots)) { }
~RawGraphicOutputBuffers() override = default;
sp<Codec2Buffer> wrap(const std::shared_ptr<C2Buffer> &buffer) override {
@@ -1545,6 +1544,8 @@
const std::shared_ptr<CCodecCallback> &callback)
: mHeapSeqNum(-1),
mCCodecCallback(callback),
+ mNumInputSlots(kSmoothnessFactor),
+ mNumOutputSlots(kSmoothnessFactor),
mFrameIndex(0u),
mFirstValidFrameIndex(0u),
mMetaMode(MODE_NONE),
@@ -2006,7 +2007,7 @@
Mutexed<std::unique_ptr<InputBuffers>>::Locked buffers(mInputBuffers);
if (!(*buffers)->isArrayMode()) {
- *buffers = (*buffers)->toArrayMode(kMinInputBufferArraySize);
+ *buffers = (*buffers)->toArrayMode(mNumInputSlots);
}
(*buffers)->getArray(array);
@@ -2017,7 +2018,7 @@
Mutexed<std::unique_ptr<OutputBuffers>>::Locked buffers(mOutputBuffers);
if (!(*buffers)->isArrayMode()) {
- *buffers = (*buffers)->toArrayMode(kMinOutputBufferArraySize);
+ *buffers = (*buffers)->toArrayMode(mNumOutputSlots);
}
(*buffers)->getArray(array);
@@ -2029,12 +2030,19 @@
C2StreamBufferTypeSetting::output oStreamFormat(0u);
C2PortReorderBufferDepthTuning::output reorderDepth;
C2PortReorderKeySetting::output reorderKey;
+ C2PortActualDelayTuning::input inputDelay(0);
+ C2PortActualDelayTuning::output outputDelay(0);
+ C2ActualPipelineDelayTuning pipelineDelay(0);
+
c2_status_t err = mComponent->query(
{
&iStreamFormat,
&oStreamFormat,
&reorderDepth,
&reorderKey,
+ &inputDelay,
+ &pipelineDelay,
+ &outputDelay,
},
{},
C2_DONT_BLOCK,
@@ -2057,6 +2065,13 @@
reorder->setKey(reorderKey.value);
}
}
+
+ mNumInputSlots =
+ (inputDelay ? inputDelay.value : 0) +
+ (pipelineDelay ? pipelineDelay.value : 0) +
+ kSmoothnessFactor;
+ mNumOutputSlots = (outputDelay ? outputDelay.value : 0) + kSmoothnessFactor;
+
// TODO: get this from input format
bool secure = mComponent->getName().find(".secure") != std::string::npos;
@@ -2127,6 +2142,7 @@
pools->inputPool = pool;
}
+ bool forceArrayMode = false;
Mutexed<std::unique_ptr<InputBuffers>>::Locked buffers(mInputBuffers);
if (graphic) {
if (mInputSurface) {
@@ -2134,7 +2150,7 @@
} else if (mMetaMode == MODE_ANW) {
buffers->reset(new GraphicMetadataInputBuffers(mName));
} else {
- buffers->reset(new GraphicInputBuffers(mName));
+ buffers->reset(new GraphicInputBuffers(mNumInputSlots, mName));
}
} else {
if (hasCryptoOrDescrambler()) {
@@ -2147,7 +2163,7 @@
if (mDealer == nullptr) {
mDealer = new MemoryDealer(
align(capacity, MemoryDealer::getAllocationAlignment())
- * (kMinInputBufferArraySize + 1),
+ * (mNumInputSlots + 1),
"EncryptedLinearInputBuffers");
mDecryptDestination = mDealer->allocate((size_t)capacity);
}
@@ -2157,7 +2173,9 @@
mHeapSeqNum = -1;
}
buffers->reset(new EncryptedLinearInputBuffers(
- secure, mDealer, mCrypto, mHeapSeqNum, (size_t)capacity, mName));
+ secure, mDealer, mCrypto, mHeapSeqNum, (size_t)capacity,
+ mNumInputSlots, mName));
+ forceArrayMode = true;
} else {
buffers->reset(new LinearInputBuffers(mName));
}
@@ -2169,6 +2187,10 @@
} else {
// TODO: error
}
+
+ if (forceArrayMode) {
+ *buffers = (*buffers)->toArrayMode(mNumInputSlots);
+ }
}
if (outputFormat != nullptr) {
@@ -2286,7 +2308,7 @@
if (outputSurface) {
buffers->reset(new GraphicOutputBuffers(mName));
} else {
- buffers->reset(new RawGraphicOutputBuffers(mName));
+ buffers->reset(new RawGraphicOutputBuffers(mNumOutputSlots, mName));
}
} else {
buffers->reset(new LinearOutputBuffers(mName));
@@ -2307,7 +2329,7 @@
// WORKAROUND: if we're using early CSD workaround we convert to
// array mode, to appease apps assuming the output
// buffers to be of the same size.
- (*buffers) = (*buffers)->toArrayMode(kMinOutputBufferArraySize);
+ (*buffers) = (*buffers)->toArrayMode(mNumOutputSlots);
int32_t channelCount;
int32_t sampleRate;
@@ -2335,27 +2357,10 @@
// about buffers from the previous generation do not interfere with the
// newly initialized pipeline capacity.
- // Query delays
- C2PortRequestedDelayTuning::input inputDelay;
- C2PortRequestedDelayTuning::output outputDelay;
- C2RequestedPipelineDelayTuning pipelineDelay;
-#if 0
- err = mComponent->query(
- { &inputDelay, &pipelineDelay, &outputDelay },
- {},
- C2_DONT_BLOCK,
- nullptr);
mAvailablePipelineCapacity.initialize(
- inputDelay,
- inputDelay + pipelineDelay,
- inputDelay + pipelineDelay + outputDelay,
+ mNumInputSlots,
+ mNumInputSlots + mNumOutputSlots,
mName);
-#else
- mAvailablePipelineCapacity.initialize(
- kMinInputBufferArraySize,
- kMaxPipelineCapacity,
- mName);
-#endif
mInputMetEos = false;
mSync.start();
@@ -2374,7 +2379,7 @@
}
std::vector<sp<MediaCodecBuffer>> toBeQueued;
// TODO: use proper buffer depth instead of this random value
- for (size_t i = 0; i < kMinInputBufferArraySize; ++i) {
+ for (size_t i = 0; i < mNumInputSlots; ++i) {
size_t index;
sp<MediaCodecBuffer> buffer;
{
@@ -2731,7 +2736,7 @@
sp<IGraphicBufferProducer> producer;
if (newSurface) {
newSurface->setScalingMode(NATIVE_WINDOW_SCALING_MODE_SCALE_TO_WINDOW);
- newSurface->setMaxDequeuedBufferCount(kMinOutputBufferArraySize);
+ newSurface->setMaxDequeuedBufferCount(mNumOutputSlots + kRenderingDepth);
producer = newSurface->getIGraphicBufferProducer();
producer->setGenerationNumber(generation);
} else {
diff --git a/media/codec2/sfplugin/CCodecBufferChannel.h b/media/codec2/sfplugin/CCodecBufferChannel.h
index 431baaa..fd806b7 100644
--- a/media/codec2/sfplugin/CCodecBufferChannel.h
+++ b/media/codec2/sfplugin/CCodecBufferChannel.h
@@ -37,6 +37,8 @@
namespace android {
+class MemoryDealer;
+
class CCodecCallback {
public:
virtual ~CCodecCallback() = default;
@@ -233,6 +235,9 @@
QueueSync mQueueSync;
std::vector<std::unique_ptr<C2Param>> mParamsToBeSet;
+ size_t mNumInputSlots;
+ size_t mNumOutputSlots;
+
Mutexed<std::unique_ptr<InputBuffers>> mInputBuffers;
Mutexed<std::list<sp<ABuffer>>> mFlushedConfigs;
Mutexed<std::unique_ptr<OutputBuffers>> mOutputBuffers;
diff --git a/media/codec2/sfplugin/Codec2InfoBuilder.cpp b/media/codec2/sfplugin/Codec2InfoBuilder.cpp
index 5d0ccd2..a8cc62d 100644
--- a/media/codec2/sfplugin/Codec2InfoBuilder.cpp
+++ b/media/codec2/sfplugin/Codec2InfoBuilder.cpp
@@ -73,10 +73,10 @@
constexpr OMX_U32 kMaxIndicesToCheck = 32;
status_t queryOmxCapabilities(
- const char* name, const char* mime, bool isEncoder,
+ const char* name, const char* mediaType, bool isEncoder,
MediaCodecInfo::CapabilitiesWriter* caps) {
- const char *role = GetComponentRole(isEncoder, mime);
+ const char *role = GetComponentRole(isEncoder, mediaType);
if (role == nullptr) {
return BAD_VALUE;
}
@@ -128,8 +128,8 @@
return err;
}
- bool isVideo = hasPrefix(mime, "video/") == 0;
- bool isImage = hasPrefix(mime, "image/") == 0;
+ bool isVideo = hasPrefix(mediaType, "video/") == 0;
+ bool isImage = hasPrefix(mediaType, "image/") == 0;
if (isVideo || isImage) {
OMX_VIDEO_PARAM_PROFILELEVELTYPE param;
@@ -149,7 +149,7 @@
// AVC components may not list the constrained profiles explicitly, but
// decoders that support a profile also support its constrained version.
// Encoders must explicitly support constrained profiles.
- if (!isEncoder && strcasecmp(mime, MEDIA_MIMETYPE_VIDEO_AVC) == 0) {
+ if (!isEncoder && strcasecmp(mediaType, MEDIA_MIMETYPE_VIDEO_AVC) == 0) {
if (param.eProfile == OMX_VIDEO_AVCProfileHigh) {
caps->addProfileLevel(OMX_VIDEO_AVCProfileConstrainedHigh, param.eLevel);
} else if (param.eProfile == OMX_VIDEO_AVCProfileBaseline) {
@@ -193,7 +193,7 @@
asString(portFormat.eColorFormat), portFormat.eColorFormat);
}
}
- } else if (strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_AAC) == 0) {
+ } else if (strcasecmp(mediaType, MEDIA_MIMETYPE_AUDIO_AAC) == 0) {
// More audio codecs if they have profiles.
OMX_AUDIO_PARAM_ANDROID_PROFILETYPE param;
InitOMXParams(¶m);
@@ -228,14 +228,17 @@
if (omxNode->configureVideoTunnelMode(
kPortIndexOutput, OMX_TRUE, 0, &sidebandHandle) == OK) {
// tunneled playback includes adaptive playback
- caps->addFlags(MediaCodecInfo::Capabilities::kFlagSupportsAdaptivePlayback
- | MediaCodecInfo::Capabilities::kFlagSupportsTunneledPlayback);
- } else if (omxNode->setPortMode(
- kPortIndexOutput, IOMX::kPortModeDynamicANWBuffer) == OK ||
- omxNode->prepareForAdaptivePlayback(
- kPortIndexOutput, OMX_TRUE,
- 1280 /* width */, 720 /* height */) == OK) {
- caps->addFlags(MediaCodecInfo::Capabilities::kFlagSupportsAdaptivePlayback);
+ } else {
+ // tunneled playback is not supported
+ caps->removeDetail(MediaCodecInfo::Capabilities::FEATURE_TUNNELED_PLAYBACK);
+ if (omxNode->setPortMode(
+ kPortIndexOutput, IOMX::kPortModeDynamicANWBuffer) == OK ||
+ omxNode->prepareForAdaptivePlayback(
+ kPortIndexOutput, OMX_TRUE,
+ 1280 /* width */, 720 /* height */) != OK) {
+ // adaptive playback is not supported
+ caps->removeDetail(MediaCodecInfo::Capabilities::FEATURE_ADAPTIVE_PLAYBACK);
+ }
}
}
@@ -243,11 +246,20 @@
OMX_VIDEO_CONFIG_ANDROID_INTRAREFRESHTYPE params;
InitOMXParams(¶ms);
params.nPortIndex = kPortIndexOutput;
- // TODO: should we verify if fallback is supported?
+
+ OMX_VIDEO_PARAM_INTRAREFRESHTYPE fallbackParams;
+ InitOMXParams(&fallbackParams);
+ fallbackParams.nPortIndex = kPortIndexOutput;
+ fallbackParams.eRefreshMode = OMX_VIDEO_IntraRefreshCyclic;
+
if (omxNode->getConfig(
(OMX_INDEXTYPE)OMX_IndexConfigAndroidIntraRefresh,
- ¶ms, sizeof(params)) == OK) {
- caps->addFlags(MediaCodecInfo::Capabilities::kFlagSupportsIntraRefresh);
+ ¶ms, sizeof(params)) != OK &&
+ omxNode->getParameter(
+ OMX_IndexParamVideoIntraRefresh, &fallbackParams,
+ sizeof(fallbackParams)) != OK) {
+ // intra refresh is not supported
+ caps->removeDetail(MediaCodecInfo::Capabilities::FEATURE_INTRA_REFRESH);
}
}
@@ -270,12 +282,26 @@
writer->addMediaCodecInfo();
info->setName(name.c_str());
info->setOwner("default");
- info->setEncoder(encoder);
+ typename std::underlying_type<MediaCodecInfo::Attributes>::type attrs = 0;
+ if (encoder) {
+ attrs |= MediaCodecInfo::kFlagIsEncoder;
+ }
+ // NOTE: we don't support software-only codecs in OMX
+ if (!hasPrefix(name, "OMX.google.")) {
+ attrs |= MediaCodecInfo::kFlagIsVendor;
+ if (properties.quirkSet.find("attribute::software-codec")
+ == properties.quirkSet.end()) {
+ attrs |= MediaCodecInfo::kFlagIsHardwareAccelerated;
+ }
+ }
+ info->setAttributes(attrs);
info->setRank(omxRank);
- for (const MediaCodecsXmlParser::Type& type : properties.typeMap) {
- const std::string &mime = type.first;
+ // OMX components don't have aliases
+ for (const MediaCodecsXmlParser::Type &type : properties.typeMap) {
+ const std::string &mediaType = type.first;
+
std::unique_ptr<MediaCodecInfo::CapabilitiesWriter> caps =
- info->addMime(mime.c_str());
+ info->addMediaType(mediaType.c_str());
const MediaCodecsXmlParser::AttributeMap &attrMap = type.second;
for (const MediaCodecsXmlParser::Attribute& attr : attrMap) {
const std::string &key = attr.first;
@@ -289,13 +315,13 @@
}
status_t err = queryOmxCapabilities(
name.c_str(),
- mime.c_str(),
+ mediaType.c_str(),
encoder,
caps.get());
if (err != OK) {
- ALOGE("Failed to query capabilities for %s (mime: %s). Error: %d",
+ ALOGI("Failed to query capabilities for %s (media type: %s). Error: %d",
name.c_str(),
- mime.c_str(),
+ mediaType.c_str(),
static_cast<int>(err));
}
}
@@ -407,20 +433,40 @@
break;
}
+ ALOGV("canonName = %s", canonName.c_str());
std::unique_ptr<MediaCodecInfoWriter> codecInfo = writer->addMediaCodecInfo();
codecInfo->setName(trait.name.c_str());
- codecInfo->setOwner("codec2");
+ codecInfo->setOwner(("codec2::" + trait.owner).c_str());
+ const MediaCodecsXmlParser::CodecProperties &codec = parser.getCodecMap().at(canonName);
+
bool encoder = trait.kind == C2Component::KIND_ENCODER;
- codecInfo->setEncoder(encoder);
+ typename std::underlying_type<MediaCodecInfo::Attributes>::type attrs = 0;
+
+ if (encoder) {
+ attrs |= MediaCodecInfo::kFlagIsEncoder;
+ }
+ if (trait.owner == "software") {
+ attrs |= MediaCodecInfo::kFlagIsSoftwareOnly;
+ } else {
+ attrs |= MediaCodecInfo::kFlagIsVendor;
+ if (trait.owner == "vendor-software") {
+ attrs |= MediaCodecInfo::kFlagIsSoftwareOnly;
+ } else if (codec.quirkSet.find("attribute::software-codec") == codec.quirkSet.end()) {
+ attrs |= MediaCodecInfo::kFlagIsHardwareAccelerated;
+ }
+ }
+ codecInfo->setAttributes(attrs);
codecInfo->setRank(rank);
- const MediaCodecsXmlParser::CodecProperties &codec =
- parser.getCodecMap().at(canonName);
+
+ for (const std::string &alias : codec.aliases) {
+ codecInfo->addAlias(alias.c_str());
+ }
for (auto typeIt = codec.typeMap.begin(); typeIt != codec.typeMap.end(); ++typeIt) {
const std::string &mediaType = typeIt->first;
const MediaCodecsXmlParser::AttributeMap &attrMap = typeIt->second;
std::unique_ptr<MediaCodecInfo::CapabilitiesWriter> caps =
- codecInfo->addMime(mediaType.c_str());
+ codecInfo->addMediaType(mediaType.c_str());
for (auto attrIt = attrMap.begin(); attrIt != attrMap.end(); ++attrIt) {
std::string key, value;
std::tie(key, value) = *attrIt;
@@ -450,6 +496,23 @@
asString(err), asString(profileQuery[0].status));
if (err == C2_OK && profileQuery[0].status == C2_OK) {
if (profileQuery[0].values.type == C2FieldSupportedValues::VALUES) {
+ std::vector<std::shared_ptr<C2ParamDescriptor>> paramDescs;
+ c2_status_t err1 = intf->querySupportedParams(¶mDescs);
+ bool isHdr = false, isHdr10Plus = false;
+ if (err1 == C2_OK) {
+ for (const std::shared_ptr<C2ParamDescriptor> &desc : paramDescs) {
+ if ((uint32_t)desc->index() ==
+ C2StreamHdr10PlusInfo::output::PARAM_TYPE) {
+ isHdr10Plus = true;
+ } else if ((uint32_t)desc->index() ==
+ C2StreamHdrStaticInfo::output::PARAM_TYPE) {
+ isHdr = true;
+ }
+ }
+ }
+ // For VP9, the static info is always propagated by framework.
+ isHdr |= (mediaType == MIMETYPE_VIDEO_VP9);
+
for (C2Value::Primitive profile : profileQuery[0].values.values) {
pl.profile = (C2Config::profile_t)profile.ref<uint32_t>();
std::vector<std::unique_ptr<C2SettingResult>> failures;
@@ -473,6 +536,26 @@
caps->addProfileLevel(
(uint32_t)sdkProfile, (uint32_t)sdkLevel);
gotProfileLevels = true;
+ if (isHdr) {
+ auto hdrMapper = C2Mapper::GetHdrProfileLevelMapper(
+ trait.mediaType);
+ if (hdrMapper && hdrMapper->mapProfile(
+ pl.profile, &sdkProfile)) {
+ caps->addProfileLevel(
+ (uint32_t)sdkProfile,
+ (uint32_t)sdkLevel);
+ }
+ if (isHdr10Plus) {
+ hdrMapper = C2Mapper::GetHdrProfileLevelMapper(
+ trait.mediaType, true /*isHdr10Plus*/);
+ if (hdrMapper && hdrMapper->mapProfile(
+ pl.profile, &sdkProfile)) {
+ caps->addProfileLevel(
+ (uint32_t)sdkProfile,
+ (uint32_t)sdkLevel);
+ }
+ }
+ }
} else if (!mapper) {
caps->addProfileLevel(pl.profile, pl.level);
gotProfileLevels = true;
diff --git a/media/codec2/sfplugin/utils/Codec2Mapper.cpp b/media/codec2/sfplugin/utils/Codec2Mapper.cpp
index b1b33e1..c369e16 100644
--- a/media/codec2/sfplugin/utils/Codec2Mapper.cpp
+++ b/media/codec2/sfplugin/utils/Codec2Mapper.cpp
@@ -253,6 +253,14 @@
{ C2Config::PROFILE_HEVC_MAIN_10_INTRA, HEVCProfileMain10 },
};
+ALookup<C2Config::profile_t, int32_t> sHevcHdrProfiles = {
+ { C2Config::PROFILE_HEVC_MAIN_10, HEVCProfileMain10HDR10 },
+};
+
+ALookup<C2Config::profile_t, int32_t> sHevcHdr10PlusProfiles = {
+ { C2Config::PROFILE_HEVC_MAIN_10, HEVCProfileMain10HDR10Plus },
+};
+
ALookup<C2Config::level_t, int32_t> sMpeg2Levels = {
{ C2Config::LEVEL_MP2V_LOW, MPEG2LevelLL },
{ C2Config::LEVEL_MP2V_MAIN, MPEG2LevelML },
@@ -324,6 +332,20 @@
{ C2Config::PROFILE_VP9_1, VP9Profile1 },
{ C2Config::PROFILE_VP9_2, VP9Profile2 },
{ C2Config::PROFILE_VP9_3, VP9Profile3 },
+ { C2Config::PROFILE_VP9_2, VP9Profile2HDR },
+ { C2Config::PROFILE_VP9_3, VP9Profile3HDR },
+ { C2Config::PROFILE_VP9_2, VP9Profile2HDR10Plus },
+ { C2Config::PROFILE_VP9_3, VP9Profile3HDR10Plus },
+};
+
+ALookup<C2Config::profile_t, int32_t> sVp9HdrProfiles = {
+ { C2Config::PROFILE_VP9_2, VP9Profile2HDR },
+ { C2Config::PROFILE_VP9_3, VP9Profile3HDR },
+};
+
+ALookup<C2Config::profile_t, int32_t> sVp9Hdr10PlusProfiles = {
+ { C2Config::PROFILE_VP9_2, VP9Profile2HDR10Plus },
+ { C2Config::PROFILE_VP9_3, VP9Profile3HDR10Plus },
};
ALookup<C2Config::level_t, int32_t> sAv1Levels = {
@@ -461,6 +483,10 @@
};
struct HevcProfileLevelMapper : ProfileLevelMapperHelper {
+ HevcProfileLevelMapper(bool isHdr = false, bool isHdr10Plus = false) :
+ ProfileLevelMapperHelper(),
+ mIsHdr(isHdr), mIsHdr10Plus(isHdr10Plus) {}
+
virtual bool simpleMap(C2Config::level_t from, int32_t *to) {
return sHevcLevels.map(from, to);
}
@@ -468,11 +494,19 @@
return sHevcLevels.map(from, to);
}
virtual bool simpleMap(C2Config::profile_t from, int32_t *to) {
- return sHevcProfiles.map(from, to);
+ return mIsHdr10Plus ? sHevcHdr10PlusProfiles.map(from, to) :
+ mIsHdr ? sHevcHdrProfiles.map(from, to) :
+ sHevcProfiles.map(from, to);
}
virtual bool simpleMap(int32_t from, C2Config::profile_t *to) {
- return sHevcProfiles.map(from, to);
+ return mIsHdr10Plus ? sHevcHdr10PlusProfiles.map(from, to) :
+ mIsHdr ? sHevcHdrProfiles.map(from, to) :
+ sHevcProfiles.map(from, to);
}
+
+private:
+ bool mIsHdr;
+ bool mIsHdr10Plus;
};
struct Mpeg2ProfileLevelMapper : ProfileLevelMapperHelper {
@@ -527,6 +561,10 @@
};
struct Vp9ProfileLevelMapper : ProfileLevelMapperHelper {
+ Vp9ProfileLevelMapper(bool isHdr = false, bool isHdr10Plus = false) :
+ ProfileLevelMapperHelper(),
+ mIsHdr(isHdr), mIsHdr10Plus(isHdr10Plus) {}
+
virtual bool simpleMap(C2Config::level_t from, int32_t *to) {
return sVp9Levels.map(from, to);
}
@@ -534,11 +572,19 @@
return sVp9Levels.map(from, to);
}
virtual bool simpleMap(C2Config::profile_t from, int32_t *to) {
- return sVp9Profiles.map(from, to);
+ return mIsHdr10Plus ? sVp9Hdr10PlusProfiles.map(from, to) :
+ mIsHdr ? sVp9HdrProfiles.map(from, to) :
+ sVp9Profiles.map(from, to);
}
virtual bool simpleMap(int32_t from, C2Config::profile_t *to) {
- return sVp9Profiles.map(from, to);
+ return mIsHdr10Plus ? sVp9Hdr10PlusProfiles.map(from, to) :
+ mIsHdr ? sVp9HdrProfiles.map(from, to) :
+ sVp9Profiles.map(from, to);
}
+
+private:
+ bool mIsHdr;
+ bool mIsHdr10Plus;
};
} // namespace
@@ -570,6 +616,18 @@
}
// static
+std::shared_ptr<C2Mapper::ProfileLevelMapper>
+C2Mapper::GetHdrProfileLevelMapper(std::string mediaType, bool isHdr10Plus) {
+ std::transform(mediaType.begin(), mediaType.begin(), mediaType.end(), ::tolower);
+ if (mediaType == MIMETYPE_VIDEO_HEVC) {
+ return std::make_shared<HevcProfileLevelMapper>(true, isHdr10Plus);
+ } else if (mediaType == MIMETYPE_VIDEO_VP9) {
+ return std::make_shared<Vp9ProfileLevelMapper>(true, isHdr10Plus);
+ }
+ return nullptr;
+}
+
+// static
bool C2Mapper::map(C2Config::bitrate_mode_t from, int32_t *to) {
return sBitrateModes.map(from, to);
}
diff --git a/media/codec2/sfplugin/utils/Codec2Mapper.h b/media/codec2/sfplugin/utils/Codec2Mapper.h
index 1eeb92e..cec6f07 100644
--- a/media/codec2/sfplugin/utils/Codec2Mapper.h
+++ b/media/codec2/sfplugin/utils/Codec2Mapper.h
@@ -40,6 +40,9 @@
static std::shared_ptr<ProfileLevelMapper>
GetProfileLevelMapper(std::string mediaType);
+ static std::shared_ptr<ProfileLevelMapper>
+ GetHdrProfileLevelMapper(std::string mediaType, bool isHdr10Plus = false);
+
// convert between bitrates
static bool map(C2Config::bitrate_mode_t, int32_t*);
static bool map(int32_t, C2Config::bitrate_mode_t*);
diff --git a/media/codec2/vndk/C2AllocatorGralloc.cpp b/media/codec2/vndk/C2AllocatorGralloc.cpp
index 4878974..18f2430 100644
--- a/media/codec2/vndk/C2AllocatorGralloc.cpp
+++ b/media/codec2/vndk/C2AllocatorGralloc.cpp
@@ -304,17 +304,23 @@
}
C2AllocationGralloc::~C2AllocationGralloc() {
- if (!mBuffer) {
- return;
- }
- if (mLocked) {
+ if (mBuffer && mLocked) {
// implementation ignores addresss and rect
uint8_t* addr[C2PlanarLayout::MAX_NUM_PLANES] = {};
unmap(addr, C2Rect(), nullptr);
}
- mMapper->freeBuffer(const_cast<native_handle_t *>(mBuffer));
- native_handle_delete(const_cast<native_handle_t*>(
- reinterpret_cast<const native_handle_t*>(mHandle)));
+ if (mBuffer) {
+ mMapper->freeBuffer(const_cast<native_handle_t *>(mBuffer));
+ }
+ if (mHandle) {
+ native_handle_delete(
+ const_cast<native_handle_t *>(reinterpret_cast<const native_handle_t *>(mHandle)));
+ }
+ if (mLockedHandle) {
+ native_handle_delete(
+ const_cast<native_handle_t *>(
+ reinterpret_cast<const native_handle_t *>(mLockedHandle)));
+ }
}
c2_status_t C2AllocationGralloc::map(
diff --git a/media/codec2/vndk/C2Config.cpp b/media/codec2/vndk/C2Config.cpp
index 782bec5..8a27088 100644
--- a/media/codec2/vndk/C2Config.cpp
+++ b/media/codec2/vndk/C2Config.cpp
@@ -221,6 +221,30 @@
{ "vp9-6", C2Config::LEVEL_VP9_6 },
{ "vp9-6.1", C2Config::LEVEL_VP9_6_1 },
{ "vp9-6.2", C2Config::LEVEL_VP9_6_2 },
+ { "av1-2", C2Config::LEVEL_AV1_2 },
+ { "av1-2.1", C2Config::LEVEL_AV1_2_1 },
+ { "av1-2.2", C2Config::LEVEL_AV1_2_2 },
+ { "av1-2.3", C2Config::LEVEL_AV1_2_3 },
+ { "av1-3", C2Config::LEVEL_AV1_3 },
+ { "av1-3.1", C2Config::LEVEL_AV1_3_1 },
+ { "av1-3.2", C2Config::LEVEL_AV1_3_2 },
+ { "av1-3.3", C2Config::LEVEL_AV1_3_3 },
+ { "av1-4", C2Config::LEVEL_AV1_4 },
+ { "av1-4.1", C2Config::LEVEL_AV1_4_1 },
+ { "av1-4.2", C2Config::LEVEL_AV1_4_2 },
+ { "av1-4.3", C2Config::LEVEL_AV1_4_3 },
+ { "av1-5", C2Config::LEVEL_AV1_5 },
+ { "av1-5.1", C2Config::LEVEL_AV1_5_1 },
+ { "av1-5.2", C2Config::LEVEL_AV1_5_2 },
+ { "av1-5.3", C2Config::LEVEL_AV1_5_3 },
+ { "av1-6", C2Config::LEVEL_AV1_6 },
+ { "av1-6.1", C2Config::LEVEL_AV1_6_1 },
+ { "av1-6.2", C2Config::LEVEL_AV1_6_2 },
+ { "av1-6.3", C2Config::LEVEL_AV1_6_3 },
+ { "av1-7", C2Config::LEVEL_AV1_7 },
+ { "av1-7.1", C2Config::LEVEL_AV1_7_1 },
+ { "av1-7.2", C2Config::LEVEL_AV1_7_2 },
+ { "av1-7.3", C2Config::LEVEL_AV1_7_3 },
}))
DEFINE_C2_ENUM_VALUE_CUSTOM_HELPER(C2BufferData::type_t, ({
diff --git a/media/codec2/vndk/C2Store.cpp b/media/codec2/vndk/C2Store.cpp
index a5dd203..dc7e89c 100644
--- a/media/codec2/vndk/C2Store.cpp
+++ b/media/codec2/vndk/C2Store.cpp
@@ -817,6 +817,7 @@
emplace("c2.android.mp3.decoder", "libcodec2_soft_mp3dec.so");
emplace("c2.android.vorbis.decoder", "libcodec2_soft_vorbisdec.so");
emplace("c2.android.opus.decoder", "libcodec2_soft_opusdec.so");
+ emplace("c2.android.opus.encoder", "libcodec2_soft_opusenc.so");
emplace("c2.android.vp8.decoder", "libcodec2_soft_vp8dec.so");
emplace("c2.android.vp9.decoder", "libcodec2_soft_vp9dec.so");
emplace("c2.android.vp8.encoder", "libcodec2_soft_vp8enc.so");
diff --git a/media/extractors/flac/FLACExtractor.cpp b/media/extractors/flac/FLACExtractor.cpp
index dcda6bf..84fbcee 100644
--- a/media/extractors/flac/FLACExtractor.cpp
+++ b/media/extractors/flac/FLACExtractor.cpp
@@ -806,7 +806,32 @@
bool SniffFLAC(DataSourceHelper *source, float *confidence)
{
- // FLAC header.
+ // Skip ID3 tags
+ off64_t pos = 0;
+ uint8_t header[10];
+ for (;;) {
+ if (source->readAt(pos, header, sizeof(header)) != sizeof(header)) {
+ return false; // no more file to read.
+ }
+
+ // check for ID3 tag
+ if (memcmp("ID3", header, 3) != 0) {
+ break; // not an ID3 tag.
+ }
+
+ // skip the ID3v2 data and check again
+ const unsigned id3Len = 10 +
+ (((header[6] & 0x7f) << 21)
+ | ((header[7] & 0x7f) << 14)
+ | ((header[8] & 0x7f) << 7)
+ | (header[9] & 0x7f));
+ pos += id3Len;
+
+ ALOGV("skipped ID3 tag of len %u new starting offset is %#016llx",
+ id3Len, (long long)pos);
+ }
+
+ // Check FLAC header.
// https://xiph.org/flac/format.html#stream
//
// Note: content stored big endian.
@@ -815,12 +840,8 @@
// 4 8 metadata type STREAMINFO (0) (note: OR with 0x80 if last metadata)
// 5 24 size of metadata, for STREAMINFO (0x22).
- // Android is LE, so express header as little endian int64 constant.
- constexpr int64_t flacHeader = (0x22LL << 56) | 'CaLf';
- constexpr int64_t flacHeader2 = flacHeader | (0x80LL << 32); // alternate form (last metadata)
- int64_t header;
- if (source->readAt(0, &header, sizeof(header)) != sizeof(header)
- || (header != flacHeader && header != flacHeader2)) {
+ if (memcmp("fLaC\x00\x00\x00\x22", header, 8) != 0 &&
+ memcmp("fLaC\x80\x00\x00\x22", header, 8) != 0) {
return false;
}
diff --git a/media/extractors/mp4/ItemTable.cpp b/media/extractors/mp4/ItemTable.cpp
index 55a0c47..eb6602c 100644
--- a/media/extractors/mp4/ItemTable.cpp
+++ b/media/extractors/mp4/ItemTable.cpp
@@ -48,7 +48,7 @@
offset(0), size(0), nextTileIndex(0) {}
bool isGrid() const {
- return type == FOURCC('g', 'r', 'i', 'd');
+ return type == FOURCC("grid");
}
status_t getNextTileItemId(uint32_t *nextTileItemId, bool reset) {
@@ -223,7 +223,7 @@
struct PitmBox : public FullBox {
PitmBox(DataSourceHelper *source) :
- FullBox(source, FOURCC('p', 'i', 't', 'm')) {}
+ FullBox(source, FOURCC("pitm")) {}
status_t parse(off64_t offset, size_t size, uint32_t *primaryItemId);
};
@@ -303,7 +303,7 @@
struct IlocBox : public FullBox {
IlocBox(DataSourceHelper *source, KeyedVector<uint32_t, ItemLoc> *itemLocs) :
- FullBox(source, FOURCC('i', 'l', 'o', 'c')),
+ FullBox(source, FOURCC("iloc")),
mItemLocs(itemLocs), mHasConstructMethod1(false) {}
status_t parse(off64_t offset, size_t size);
@@ -497,7 +497,7 @@
ALOGV("attach reference type 0x%x to item id %d)", type(), mItemId);
switch(type()) {
- case FOURCC('d', 'i', 'm', 'g'): {
+ case FOURCC("dimg"): {
ssize_t itemIndex = itemIdToItemMap.indexOfKey(mItemId);
// ignore non-image items
@@ -525,7 +525,7 @@
}
break;
}
- case FOURCC('t', 'h', 'm', 'b'): {
+ case FOURCC("thmb"): {
ssize_t itemIndex = itemIdToItemMap.indexOfKey(mItemId);
// ignore non-image items
@@ -554,7 +554,7 @@
}
break;
}
- case FOURCC('c', 'd', 's', 'c'): {
+ case FOURCC("cdsc"): {
ssize_t itemIndex = itemIdToExifMap.indexOfKey(mItemId);
// ignore non-exif block items
@@ -575,7 +575,7 @@
}
break;
}
- case FOURCC('a', 'u', 'x', 'l'): {
+ case FOURCC("auxl"): {
ssize_t itemIndex = itemIdToItemMap.indexOfKey(mItemId);
// ignore non-image items
@@ -628,7 +628,7 @@
struct IrefBox : public FullBox {
IrefBox(DataSourceHelper *source, Vector<sp<ItemReference> > *itemRefs) :
- FullBox(source, FOURCC('i', 'r', 'e', 'f')), mRefIdSize(0), mItemRefs(itemRefs) {}
+ FullBox(source, FOURCC("iref")), mRefIdSize(0), mItemRefs(itemRefs) {}
status_t parse(off64_t offset, size_t size);
@@ -690,7 +690,7 @@
struct IspeBox : public FullBox, public ItemProperty {
IspeBox(DataSourceHelper *source) :
- FullBox(source, FOURCC('i', 's', 'p', 'e')), mWidth(0), mHeight(0) {}
+ FullBox(source, FOURCC("ispe")), mWidth(0), mHeight(0) {}
status_t parse(off64_t offset, size_t size) override;
@@ -726,7 +726,7 @@
struct HvccBox : public Box, public ItemProperty {
HvccBox(DataSourceHelper *source) :
- Box(source, FOURCC('h', 'v', 'c', 'C')) {}
+ Box(source, FOURCC("hvcC")) {}
status_t parse(off64_t offset, size_t size) override;
@@ -759,7 +759,7 @@
struct IrotBox : public Box, public ItemProperty {
IrotBox(DataSourceHelper *source) :
- Box(source, FOURCC('i', 'r', 'o', 't')), mAngle(0) {}
+ Box(source, FOURCC("irot")), mAngle(0) {}
status_t parse(off64_t offset, size_t size) override;
@@ -788,7 +788,7 @@
struct ColrBox : public Box, public ItemProperty {
ColrBox(DataSourceHelper *source) :
- Box(source, FOURCC('c', 'o', 'l', 'r')) {}
+ Box(source, FOURCC("colr")) {}
status_t parse(off64_t offset, size_t size) override;
@@ -812,11 +812,11 @@
}
offset += 4;
size -= 4;
- if (colour_type == FOURCC('n', 'c', 'l', 'x')) {
+ if (colour_type == FOURCC("nclx")) {
return OK;
}
- if ((colour_type != FOURCC('r', 'I', 'C', 'C')) &&
- (colour_type != FOURCC('p', 'r', 'o', 'f'))) {
+ if ((colour_type != FOURCC("rICC")) &&
+ (colour_type != FOURCC("prof"))) {
return ERROR_MALFORMED;
}
@@ -836,7 +836,7 @@
struct IpmaBox : public FullBox {
IpmaBox(DataSourceHelper *source, Vector<AssociationEntry> *associations) :
- FullBox(source, FOURCC('i', 'p', 'm', 'a')), mAssociations(associations) {}
+ FullBox(source, FOURCC("ipma")), mAssociations(associations) {}
status_t parse(off64_t offset, size_t size);
private:
@@ -910,7 +910,7 @@
struct IpcoBox : public Box {
IpcoBox(DataSourceHelper *source, Vector<sp<ItemProperty> > *properties) :
- Box(source, FOURCC('i', 'p', 'c', 'o')), mItemProperties(properties) {}
+ Box(source, FOURCC("ipco")), mItemProperties(properties) {}
status_t parse(off64_t offset, size_t size);
protected:
@@ -930,22 +930,22 @@
status_t IpcoBox::onChunkData(uint32_t type, off64_t offset, size_t size) {
sp<ItemProperty> itemProperty;
switch(type) {
- case FOURCC('h', 'v', 'c', 'C'):
+ case FOURCC("hvcC"):
{
itemProperty = new HvccBox(source());
break;
}
- case FOURCC('i', 's', 'p', 'e'):
+ case FOURCC("ispe"):
{
itemProperty = new IspeBox(source());
break;
}
- case FOURCC('i', 'r', 'o', 't'):
+ case FOURCC("irot"):
{
itemProperty = new IrotBox(source());
break;
}
- case FOURCC('c', 'o', 'l', 'r'):
+ case FOURCC("colr"):
{
itemProperty = new ColrBox(source());
break;
@@ -969,7 +969,7 @@
IprpBox(DataSourceHelper *source,
Vector<sp<ItemProperty> > *properties,
Vector<AssociationEntry> *associations) :
- Box(source, FOURCC('i', 'p', 'r', 'p')),
+ Box(source, FOURCC("iprp")),
mProperties(properties), mAssociations(associations) {}
status_t parse(off64_t offset, size_t size);
@@ -993,12 +993,12 @@
status_t IprpBox::onChunkData(uint32_t type, off64_t offset, size_t size) {
switch(type) {
- case FOURCC('i', 'p', 'c', 'o'):
+ case FOURCC("ipco"):
{
IpcoBox ipcoBox(source(), mProperties);
return ipcoBox.parse(offset, size);
}
- case FOURCC('i', 'p', 'm', 'a'):
+ case FOURCC("ipma"):
{
IpmaBox ipmaBox(source(), mAssociations);
return ipmaBox.parse(offset, size);
@@ -1024,7 +1024,7 @@
struct InfeBox : public FullBox {
InfeBox(DataSourceHelper *source) :
- FullBox(source, FOURCC('i', 'n', 'f', 'e')) {}
+ FullBox(source, FOURCC("infe")) {}
status_t parse(off64_t offset, size_t size, ItemInfo *itemInfo);
@@ -1104,7 +1104,7 @@
}
ALOGV("item_name %s", item_name.c_str());
- if (item_type == FOURCC('m', 'i', 'm', 'e')) {
+ if (item_type == FOURCC("mime")) {
String8 content_type;
if (!parseNullTerminatedString(&offset, &size, &content_type)) {
return ERROR_MALFORMED;
@@ -1117,7 +1117,7 @@
return ERROR_MALFORMED;
}
}
- } else if (item_type == FOURCC('u', 'r', 'i', ' ')) {
+ } else if (item_type == FOURCC("uri ")) {
String8 item_uri_type;
if (!parseNullTerminatedString(&offset, &size, &item_uri_type)) {
return ERROR_MALFORMED;
@@ -1129,7 +1129,7 @@
struct IinfBox : public FullBox {
IinfBox(DataSourceHelper *source, Vector<ItemInfo> *itemInfos) :
- FullBox(source, FOURCC('i', 'i', 'n', 'f')),
+ FullBox(source, FOURCC("iinf")),
mItemInfos(itemInfos), mHasGrids(false) {}
status_t parse(off64_t offset, size_t size);
@@ -1179,7 +1179,7 @@
}
status_t IinfBox::onChunkData(uint32_t type, off64_t offset, size_t size) {
- if (type != FOURCC('i', 'n', 'f', 'e')) {
+ if (type != FOURCC("infe")) {
return OK;
}
@@ -1188,7 +1188,7 @@
status_t err = infeBox.parse(offset, size, &itemInfo);
if (err == OK) {
mItemInfos->push_back(itemInfo);
- mHasGrids |= (itemInfo.itemType == FOURCC('g', 'r', 'i', 'd'));
+ mHasGrids |= (itemInfo.itemType == FOURCC("grid"));
}
// InfeBox parse returns ERROR_UNSUPPORTED if the box if an unsupported
// version. Ignore this error as it's not fatal.
@@ -1214,31 +1214,31 @@
status_t ItemTable::parse(uint32_t type, off64_t data_offset, size_t chunk_data_size) {
switch(type) {
- case FOURCC('i', 'l', 'o', 'c'):
+ case FOURCC("iloc"):
{
return parseIlocBox(data_offset, chunk_data_size);
}
- case FOURCC('i', 'i', 'n', 'f'):
+ case FOURCC("iinf"):
{
return parseIinfBox(data_offset, chunk_data_size);
}
- case FOURCC('i', 'p', 'r', 'p'):
+ case FOURCC("iprp"):
{
return parseIprpBox(data_offset, chunk_data_size);
}
- case FOURCC('p', 'i', 't', 'm'):
+ case FOURCC("pitm"):
{
return parsePitmBox(data_offset, chunk_data_size);
}
- case FOURCC('i', 'd', 'a', 't'):
+ case FOURCC("idat"):
{
return parseIdatBox(data_offset, chunk_data_size);
}
- case FOURCC('i', 'r', 'e', 'f'):
+ case FOURCC("iref"):
{
return parseIrefBox(data_offset, chunk_data_size);
}
- case FOURCC('i', 'p', 'r', 'o'):
+ case FOURCC("ipro"):
{
ALOGW("ipro box not supported!");
break;
@@ -1355,9 +1355,9 @@
// 'grid': derived image from tiles
// 'hvc1': coded image (or tile)
// 'Exif': EXIF metadata
- if (info.itemType != FOURCC('g', 'r', 'i', 'd') &&
- info.itemType != FOURCC('h', 'v', 'c', '1') &&
- info.itemType != FOURCC('E', 'x', 'i', 'f')) {
+ if (info.itemType != FOURCC("grid") &&
+ info.itemType != FOURCC("hvc1") &&
+ info.itemType != FOURCC("Exif")) {
continue;
}
@@ -1380,7 +1380,7 @@
return ERROR_MALFORMED;
}
- if (info.itemType == FOURCC('E', 'x', 'i', 'f')) {
+ if (info.itemType == FOURCC("Exif")) {
// Only add if the Exif data is non-empty. The first 4 bytes contain
// the offset to TIFF header, which the Exif parser doesn't use.
if (size > 4) {
@@ -1687,8 +1687,31 @@
}
// skip the first 4-byte of the offset to TIFF header
- *offset = mItemIdToExifMap[exifIndex].offset + 4;
- *size = mItemIdToExifMap[exifIndex].size - 4;
+ uint32_t tiffOffset;
+ if (!mDataSource->readAt(
+ mItemIdToExifMap[exifIndex].offset, &tiffOffset, 4)) {
+ return ERROR_IO;
+ }
+
+ // We need 'Exif\0\0' before the tiff header
+ tiffOffset = ntohl(tiffOffset);
+ if (tiffOffset < 6) {
+ return ERROR_MALFORMED;
+ }
+ // The first 4-byte of the item is the offset of the tiff header within the
+ // exif data. The size of the item should be > 4 for a non-empty exif (this
+ // was already checked when the item was added). Also check that the tiff
+ // header offset is valid.
+ if (mItemIdToExifMap[exifIndex].size <= 4 ||
+ tiffOffset > mItemIdToExifMap[exifIndex].size - 4) {
+ return ERROR_MALFORMED;
+ }
+
+ // Offset of 'Exif\0\0' relative to the beginning of 'Exif' item
+ // (first 4-byte is the tiff header offset)
+ uint32_t exifOffset = 4 + tiffOffset - 6;
+ *offset = mItemIdToExifMap[exifIndex].offset + exifOffset;
+ *size = mItemIdToExifMap[exifIndex].size - exifOffset;
return OK;
}
diff --git a/media/extractors/mp4/MPEG4Extractor.cpp b/media/extractors/mp4/MPEG4Extractor.cpp
index 0441359..d0efddd 100644
--- a/media/extractors/mp4/MPEG4Extractor.cpp
+++ b/media/extractors/mp4/MPEG4Extractor.cpp
@@ -308,42 +308,42 @@
static const char *FourCC2MIME(uint32_t fourcc) {
switch (fourcc) {
- case FOURCC('m', 'p', '4', 'a'):
+ case FOURCC("mp4a"):
return MEDIA_MIMETYPE_AUDIO_AAC;
- case FOURCC('s', 'a', 'm', 'r'):
+ case FOURCC("samr"):
return MEDIA_MIMETYPE_AUDIO_AMR_NB;
- case FOURCC('s', 'a', 'w', 'b'):
+ case FOURCC("sawb"):
return MEDIA_MIMETYPE_AUDIO_AMR_WB;
- case FOURCC('e', 'c', '-', '3'):
+ case FOURCC("ec-3"):
return MEDIA_MIMETYPE_AUDIO_EAC3;
- case FOURCC('m', 'p', '4', 'v'):
+ case FOURCC("mp4v"):
return MEDIA_MIMETYPE_VIDEO_MPEG4;
- case FOURCC('s', '2', '6', '3'):
- case FOURCC('h', '2', '6', '3'):
- case FOURCC('H', '2', '6', '3'):
+ case FOURCC("s263"):
+ case FOURCC("h263"):
+ case FOURCC("H263"):
return MEDIA_MIMETYPE_VIDEO_H263;
- case FOURCC('a', 'v', 'c', '1'):
+ case FOURCC("avc1"):
return MEDIA_MIMETYPE_VIDEO_AVC;
- case FOURCC('h', 'v', 'c', '1'):
- case FOURCC('h', 'e', 'v', '1'):
+ case FOURCC("hvc1"):
+ case FOURCC("hev1"):
return MEDIA_MIMETYPE_VIDEO_HEVC;
- case FOURCC('a', 'c', '-', '4'):
+ case FOURCC("ac-4"):
return MEDIA_MIMETYPE_AUDIO_AC4;
- case FOURCC('t', 'w', 'o', 's'):
- case FOURCC('s', 'o', 'w', 't'):
+ case FOURCC("twos"):
+ case FOURCC("sowt"):
return MEDIA_MIMETYPE_AUDIO_RAW;
- case FOURCC('a', 'l', 'a', 'c'):
+ case FOURCC("alac"):
return MEDIA_MIMETYPE_AUDIO_ALAC;
- case FOURCC('a', 'v', '0', '1'):
+ case FOURCC("av01"):
return MEDIA_MIMETYPE_VIDEO_AV1;
default:
ALOGW("Unknown fourcc: %c%c%c%c",
@@ -594,7 +594,7 @@
}
} else {
uint32_t sampleIndex;
- uint32_t sampleTime;
+ uint64_t sampleTime;
if (track->timescale != 0 &&
track->sampleTable->findThumbnailSample(&sampleIndex) == OK
&& track->sampleTable->getMetaDataForSample(
@@ -749,21 +749,21 @@
static bool underMetaDataPath(const Vector<uint32_t> &path) {
return path.size() >= 5
- && path[0] == FOURCC('m', 'o', 'o', 'v')
- && path[1] == FOURCC('u', 'd', 't', 'a')
- && path[2] == FOURCC('m', 'e', 't', 'a')
- && path[3] == FOURCC('i', 'l', 's', 't');
+ && path[0] == FOURCC("moov")
+ && path[1] == FOURCC("udta")
+ && path[2] == FOURCC("meta")
+ && path[3] == FOURCC("ilst");
}
static bool underQTMetaPath(const Vector<uint32_t> &path, int32_t depth) {
return path.size() >= 2
- && path[0] == FOURCC('m', 'o', 'o', 'v')
- && path[1] == FOURCC('m', 'e', 't', 'a')
+ && path[0] == FOURCC("moov")
+ && path[1] == FOURCC("meta")
&& (depth == 2
|| (depth == 3
- && (path[2] == FOURCC('h', 'd', 'l', 'r')
- || path[2] == FOURCC('i', 'l', 's', 't')
- || path[2] == FOURCC('k', 'e', 'y', 's'))));
+ && (path[2] == FOURCC("hdlr")
+ || path[2] == FOURCC("ilst")
+ || path[2] == FOURCC("keys"))));
}
// Given a time in seconds since Jan 1 1904, produce a human-readable string.
@@ -867,7 +867,7 @@
ALOGE("b/23540914");
return ERROR_MALFORMED;
}
- if (chunk_type != FOURCC('m', 'd', 'a', 't') && chunk_data_size > kMaxAtomSize) {
+ if (chunk_type != FOURCC("mdat") && chunk_data_size > kMaxAtomSize) {
char errMsg[100];
sprintf(errMsg, "%s atom has size %" PRId64, chunk, chunk_data_size);
ALOGE("%s (b/28615448)", errMsg);
@@ -875,8 +875,8 @@
return ERROR_MALFORMED;
}
- if (chunk_type != FOURCC('c', 'p', 'r', 't')
- && chunk_type != FOURCC('c', 'o', 'v', 'r')
+ if (chunk_type != FOURCC("cprt")
+ && chunk_type != FOURCC("covr")
&& mPath.size() == 5 && underMetaDataPath(mPath)) {
off64_t stop_offset = *offset + chunk_size;
*offset = data_offset;
@@ -895,40 +895,40 @@
}
switch(chunk_type) {
- case FOURCC('m', 'o', 'o', 'v'):
- case FOURCC('t', 'r', 'a', 'k'):
- case FOURCC('m', 'd', 'i', 'a'):
- case FOURCC('m', 'i', 'n', 'f'):
- case FOURCC('d', 'i', 'n', 'f'):
- case FOURCC('s', 't', 'b', 'l'):
- case FOURCC('m', 'v', 'e', 'x'):
- case FOURCC('m', 'o', 'o', 'f'):
- case FOURCC('t', 'r', 'a', 'f'):
- case FOURCC('m', 'f', 'r', 'a'):
- case FOURCC('u', 'd', 't', 'a'):
- case FOURCC('i', 'l', 's', 't'):
- case FOURCC('s', 'i', 'n', 'f'):
- case FOURCC('s', 'c', 'h', 'i'):
- case FOURCC('e', 'd', 't', 's'):
- case FOURCC('w', 'a', 'v', 'e'):
+ case FOURCC("moov"):
+ case FOURCC("trak"):
+ case FOURCC("mdia"):
+ case FOURCC("minf"):
+ case FOURCC("dinf"):
+ case FOURCC("stbl"):
+ case FOURCC("mvex"):
+ case FOURCC("moof"):
+ case FOURCC("traf"):
+ case FOURCC("mfra"):
+ case FOURCC("udta"):
+ case FOURCC("ilst"):
+ case FOURCC("sinf"):
+ case FOURCC("schi"):
+ case FOURCC("edts"):
+ case FOURCC("wave"):
{
- if (chunk_type == FOURCC('m', 'o', 'o', 'v') && depth != 0) {
+ if (chunk_type == FOURCC("moov") && depth != 0) {
ALOGE("moov: depth %d", depth);
return ERROR_MALFORMED;
}
- if (chunk_type == FOURCC('m', 'o', 'o', 'v') && mInitCheck == OK) {
+ if (chunk_type == FOURCC("moov") && mInitCheck == OK) {
ALOGE("duplicate moov");
return ERROR_MALFORMED;
}
- if (chunk_type == FOURCC('m', 'o', 'o', 'f') && !mMoofFound) {
+ if (chunk_type == FOURCC("moof") && !mMoofFound) {
// store the offset of the first segment
mMoofFound = true;
mMoofOffset = *offset;
}
- if (chunk_type == FOURCC('s', 't', 'b', 'l')) {
+ if (chunk_type == FOURCC("stbl")) {
ALOGV("sampleTable chunk is %" PRIu64 " bytes long.", chunk_size);
if (mDataSource->flags()
@@ -954,7 +954,7 @@
}
bool isTrack = false;
- if (chunk_type == FOURCC('t', 'r', 'a', 'k')) {
+ if (chunk_type == FOURCC("trak")) {
if (depth != 1) {
ALOGE("trak: depth %d", depth);
return ERROR_MALFORMED;
@@ -985,6 +985,22 @@
off64_t stop_offset = *offset + chunk_size;
*offset = data_offset;
while (*offset < stop_offset) {
+
+ // pass udata terminate
+ if (mIsQT && stop_offset - *offset == 4 && chunk_type == FOURCC("udta")) {
+ // handle the case that udta terminates with terminate code x00000000
+ // note that 0 terminator is optional and we just handle this case.
+ uint32_t terminate_code = 1;
+ mDataSource->readAt(*offset, &terminate_code, 4);
+ if (0 == terminate_code) {
+ *offset += 4;
+ ALOGD("Terminal code for udta");
+ continue;
+ } else {
+ ALOGW("invalid udta Terminal code");
+ }
+ }
+
status_t err = parseChunk(offset, depth + 1);
if (err != OK) {
if (isTrack) {
@@ -1033,7 +1049,7 @@
return OK;
}
- } else if (chunk_type == FOURCC('m', 'o', 'o', 'v')) {
+ } else if (chunk_type == FOURCC("moov")) {
mInitCheck = OK;
return UNKNOWN_ERROR; // Return a dummy error.
@@ -1041,7 +1057,7 @@
break;
}
- case FOURCC('s', 'c', 'h', 'm'):
+ case FOURCC("schm"):
{
*offset += chunk_size;
@@ -1056,23 +1072,23 @@
scheme_type = ntohl(scheme_type);
int32_t mode = kCryptoModeUnencrypted;
switch(scheme_type) {
- case FOURCC('c', 'b', 'c', '1'):
+ case FOURCC("cbc1"):
{
mode = kCryptoModeAesCbc;
break;
}
- case FOURCC('c', 'b', 'c', 's'):
+ case FOURCC("cbcs"):
{
mode = kCryptoModeAesCbc;
mLastTrack->subsample_encryption = true;
break;
}
- case FOURCC('c', 'e', 'n', 'c'):
+ case FOURCC("cenc"):
{
mode = kCryptoModeAesCtr;
break;
}
- case FOURCC('c', 'e', 'n', 's'):
+ case FOURCC("cens"):
{
mode = kCryptoModeAesCtr;
mLastTrack->subsample_encryption = true;
@@ -1086,7 +1102,7 @@
}
- case FOURCC('e', 'l', 's', 't'):
+ case FOURCC("elst"):
{
*offset += chunk_size;
@@ -1142,7 +1158,7 @@
break;
}
- case FOURCC('f', 'r', 'm', 'a'):
+ case FOURCC("frma"):
{
*offset += chunk_size;
@@ -1171,7 +1187,7 @@
// If format type is 'alac', it is necessary to get the parameters
// from a alac atom spreading behind the frma atom.
// See 'external/alac/ALACMagicCookieDescription.txt'.
- if (original_fourcc == FOURCC('a', 'l', 'a', 'c')) {
+ if (original_fourcc == FOURCC("alac")) {
// Store ALAC magic cookie (decoder needs it).
uint8_t alacInfo[12];
data_offset = *offset;
@@ -1181,7 +1197,7 @@
}
uint32_t size = U32_AT(&alacInfo[0]);
if ((size != ALAC_SPECIFIC_INFO_SIZE) ||
- (U32_AT(&alacInfo[4]) != FOURCC('a', 'l', 'a', 'c')) ||
+ (U32_AT(&alacInfo[4]) != FOURCC("alac")) ||
(U32_AT(&alacInfo[8]) != 0)) {
return ERROR_MALFORMED;
}
@@ -1210,7 +1226,7 @@
break;
}
- case FOURCC('t', 'e', 'n', 'c'):
+ case FOURCC("tenc"):
{
*offset += chunk_size;
@@ -1323,7 +1339,7 @@
break;
}
- case FOURCC('t', 'k', 'h', 'd'):
+ case FOURCC("tkhd"):
{
*offset += chunk_size;
@@ -1335,7 +1351,7 @@
break;
}
- case FOURCC('t', 'r', 'e', 'f'):
+ case FOURCC("tref"):
{
off64_t stop_offset = *offset + chunk_size;
*offset = data_offset;
@@ -1351,7 +1367,7 @@
break;
}
- case FOURCC('t', 'h', 'm', 'b'):
+ case FOURCC("thmb"):
{
*offset += chunk_size;
@@ -1368,7 +1384,7 @@
break;
}
- case FOURCC('p', 's', 's', 'h'):
+ case FOURCC("pssh"):
{
*offset += chunk_size;
@@ -1404,7 +1420,7 @@
break;
}
- case FOURCC('m', 'd', 'h', 'd'):
+ case FOURCC("mdhd"):
{
*offset += chunk_size;
@@ -1500,7 +1516,7 @@
break;
}
- case FOURCC('s', 't', 's', 'd'):
+ case FOURCC("stsd"):
{
uint8_t buffer[8];
if (chunk_data_size < (off64_t)sizeof(buffer)) {
@@ -1552,7 +1568,7 @@
}
break;
}
- case FOURCC('m', 'e', 't', 't'):
+ case FOURCC("mett"):
{
*offset += chunk_size;
@@ -1606,16 +1622,16 @@
break;
}
- case FOURCC('m', 'p', '4', 'a'):
- case FOURCC('e', 'n', 'c', 'a'):
- case FOURCC('s', 'a', 'm', 'r'):
- case FOURCC('s', 'a', 'w', 'b'):
- case FOURCC('t', 'w', 'o', 's'):
- case FOURCC('s', 'o', 'w', 't'):
- case FOURCC('a', 'l', 'a', 'c'):
+ case FOURCC("mp4a"):
+ case FOURCC("enca"):
+ case FOURCC("samr"):
+ case FOURCC("sawb"):
+ case FOURCC("twos"):
+ case FOURCC("sowt"):
+ case FOURCC("alac"):
{
- if (mIsQT && chunk_type == FOURCC('m', 'p', '4', 'a')
- && depth >= 1 && mPath[depth - 1] == FOURCC('w', 'a', 'v', 'e')) {
+ if (mIsQT && chunk_type == FOURCC("mp4a")
+ && depth >= 1 && mPath[depth - 1] == FOURCC("wave")) {
// Ignore mp4a embedded in QT wave atom
*offset += chunk_size;
break;
@@ -1645,7 +1661,7 @@
off64_t stop_offset = *offset + chunk_size;
*offset = data_offset + sizeof(buffer);
- if (mIsQT && chunk_type == FOURCC('m', 'p', '4', 'a')) {
+ if (mIsQT && chunk_type == FOURCC("mp4a")) {
if (version == 1) {
if (mDataSource->readAt(*offset, buffer, 16) < 16) {
return ERROR_IO;
@@ -1678,7 +1694,7 @@
}
}
- if (chunk_type != FOURCC('e', 'n', 'c', 'a')) {
+ if (chunk_type != FOURCC("enca")) {
// if the chunk type is enca, we'll get the type from the frma box later
AMediaFormat_setString(mLastTrack->meta,
AMEDIAFORMAT_KEY_MIME, FourCC2MIME(chunk_type));
@@ -1687,7 +1703,7 @@
if (!strcasecmp(MEDIA_MIMETYPE_AUDIO_RAW, FourCC2MIME(chunk_type))) {
AMediaFormat_setInt32(mLastTrack->meta,
AMEDIAFORMAT_KEY_BITS_PER_SAMPLE, sample_size);
- if (chunk_type == FOURCC('t', 'w', 'o', 's')) {
+ if (chunk_type == FOURCC("twos")) {
AMediaFormat_setInt32(mLastTrack->meta,
AMEDIAFORMAT_KEY_PCM_BIG_ENDIAN, 1);
}
@@ -1698,7 +1714,7 @@
AMediaFormat_setInt32(mLastTrack->meta, AMEDIAFORMAT_KEY_CHANNEL_COUNT, num_channels);
AMediaFormat_setInt32(mLastTrack->meta, AMEDIAFORMAT_KEY_SAMPLE_RATE, sample_rate);
- if (chunk_type == FOURCC('a', 'l', 'a', 'c')) {
+ if (chunk_type == FOURCC("alac")) {
// See 'external/alac/ALACMagicCookieDescription.txt for the detail'.
// Store ALAC magic cookie (decoder needs it).
@@ -1710,7 +1726,7 @@
}
uint32_t size = U32_AT(&alacInfo[0]);
if ((size != ALAC_SPECIFIC_INFO_SIZE) ||
- (U32_AT(&alacInfo[4]) != FOURCC('a', 'l', 'a', 'c')) ||
+ (U32_AT(&alacInfo[4]) != FOURCC("alac")) ||
(U32_AT(&alacInfo[8]) != 0)) {
return ERROR_MALFORMED;
}
@@ -1748,15 +1764,15 @@
break;
}
- case FOURCC('m', 'p', '4', 'v'):
- case FOURCC('e', 'n', 'c', 'v'):
- case FOURCC('s', '2', '6', '3'):
- case FOURCC('H', '2', '6', '3'):
- case FOURCC('h', '2', '6', '3'):
- case FOURCC('a', 'v', 'c', '1'):
- case FOURCC('h', 'v', 'c', '1'):
- case FOURCC('h', 'e', 'v', '1'):
- case FOURCC('a', 'v', '0', '1'):
+ case FOURCC("mp4v"):
+ case FOURCC("encv"):
+ case FOURCC("s263"):
+ case FOURCC("H263"):
+ case FOURCC("h263"):
+ case FOURCC("avc1"):
+ case FOURCC("hvc1"):
+ case FOURCC("hev1"):
+ case FOURCC("av01"):
{
uint8_t buffer[78];
if (chunk_data_size < (ssize_t)sizeof(buffer)) {
@@ -1786,7 +1802,7 @@
if (mLastTrack == NULL)
return ERROR_MALFORMED;
- if (chunk_type != FOURCC('e', 'n', 'c', 'v')) {
+ if (chunk_type != FOURCC("encv")) {
// if the chunk type is encv, we'll get the type from the frma box later
AMediaFormat_setString(mLastTrack->meta,
AMEDIAFORMAT_KEY_MIME, FourCC2MIME(chunk_type));
@@ -1809,8 +1825,8 @@
break;
}
- case FOURCC('s', 't', 'c', 'o'):
- case FOURCC('c', 'o', '6', '4'):
+ case FOURCC("stco"):
+ case FOURCC("co64"):
{
if ((mLastTrack == NULL) || (mLastTrack->sampleTable == NULL)) {
return ERROR_MALFORMED;
@@ -1829,7 +1845,7 @@
break;
}
- case FOURCC('s', 't', 's', 'c'):
+ case FOURCC("stsc"):
{
if ((mLastTrack == NULL) || (mLastTrack->sampleTable == NULL))
return ERROR_MALFORMED;
@@ -1847,8 +1863,8 @@
break;
}
- case FOURCC('s', 't', 's', 'z'):
- case FOURCC('s', 't', 'z', '2'):
+ case FOURCC("stsz"):
+ case FOURCC("stz2"):
{
if ((mLastTrack == NULL) || (mLastTrack->sampleTable == NULL)) {
return ERROR_MALFORMED;
@@ -1967,7 +1983,7 @@
break;
}
- case FOURCC('s', 't', 't', 's'):
+ case FOURCC("stts"):
{
if ((mLastTrack == NULL) || (mLastTrack->sampleTable == NULL))
return ERROR_MALFORMED;
@@ -1985,7 +2001,7 @@
break;
}
- case FOURCC('c', 't', 't', 's'):
+ case FOURCC("ctts"):
{
if ((mLastTrack == NULL) || (mLastTrack->sampleTable == NULL))
return ERROR_MALFORMED;
@@ -2003,7 +2019,7 @@
break;
}
- case FOURCC('s', 't', 's', 's'):
+ case FOURCC("stss"):
{
if ((mLastTrack == NULL) || (mLastTrack->sampleTable == NULL))
return ERROR_MALFORMED;
@@ -2022,7 +2038,7 @@
}
// \xA9xyz
- case FOURCC(0xA9, 'x', 'y', 'z'):
+ case FOURCC("\251xyz"):
{
*offset += chunk_size;
@@ -2072,7 +2088,7 @@
break;
}
- case FOURCC('e', 's', 'd', 's'):
+ case FOURCC("esds"):
{
*offset += chunk_size;
@@ -2102,7 +2118,7 @@
AMEDIAFORMAT_KEY_ESDS, &buffer[4], chunk_data_size - 4);
if (mPath.size() >= 2
- && mPath[mPath.size() - 2] == FOURCC('m', 'p', '4', 'a')) {
+ && mPath[mPath.size() - 2] == FOURCC("mp4a")) {
// Information from the ESDS must be relied on for proper
// setup of sample rate and channel count for MPEG4 Audio.
// The generic header appears to only contain generic
@@ -2116,7 +2132,7 @@
}
}
if (mPath.size() >= 2
- && mPath[mPath.size() - 2] == FOURCC('m', 'p', '4', 'v')) {
+ && mPath[mPath.size() - 2] == FOURCC("mp4v")) {
// Check if the video is MPEG2
ESDS esds(&buffer[4], chunk_data_size - 4);
@@ -2131,7 +2147,7 @@
break;
}
- case FOURCC('b', 't', 'r', 't'):
+ case FOURCC("btrt"):
{
*offset += chunk_size;
if (mLastTrack == NULL) {
@@ -2161,7 +2177,7 @@
break;
}
- case FOURCC('a', 'v', 'c', 'C'):
+ case FOURCC("avcC"):
{
*offset += chunk_size;
@@ -2185,7 +2201,7 @@
break;
}
- case FOURCC('h', 'v', 'c', 'C'):
+ case FOURCC("hvcC"):
{
auto buffer = heapbuffer<uint8_t>(chunk_data_size);
@@ -2209,7 +2225,7 @@
break;
}
- case FOURCC('d', '2', '6', '3'):
+ case FOURCC("d263"):
{
*offset += chunk_size;
/*
@@ -2244,7 +2260,7 @@
break;
}
- case FOURCC('m', 'e', 't', 'a'):
+ case FOURCC("meta"):
{
off64_t stop_offset = *offset + chunk_size;
*offset = data_offset;
@@ -2288,13 +2304,13 @@
break;
}
- case FOURCC('i', 'l', 'o', 'c'):
- case FOURCC('i', 'i', 'n', 'f'):
- case FOURCC('i', 'p', 'r', 'p'):
- case FOURCC('p', 'i', 't', 'm'):
- case FOURCC('i', 'd', 'a', 't'):
- case FOURCC('i', 'r', 'e', 'f'):
- case FOURCC('i', 'p', 'r', 'o'):
+ case FOURCC("iloc"):
+ case FOURCC("iinf"):
+ case FOURCC("iprp"):
+ case FOURCC("pitm"):
+ case FOURCC("idat"):
+ case FOURCC("iref"):
+ case FOURCC("ipro"):
{
if (mIsHeif) {
if (mItemTable == NULL) {
@@ -2310,9 +2326,9 @@
break;
}
- case FOURCC('m', 'e', 'a', 'n'):
- case FOURCC('n', 'a', 'm', 'e'):
- case FOURCC('d', 'a', 't', 'a'):
+ case FOURCC("mean"):
+ case FOURCC("name"):
+ case FOURCC("data"):
{
*offset += chunk_size;
@@ -2327,7 +2343,7 @@
break;
}
- case FOURCC('m', 'v', 'h', 'd'):
+ case FOURCC("mvhd"):
{
*offset += chunk_size;
@@ -2379,7 +2395,7 @@
break;
}
- case FOURCC('m', 'e', 'h', 'd'):
+ case FOURCC("mehd"):
{
*offset += chunk_size;
@@ -2424,7 +2440,7 @@
break;
}
- case FOURCC('m', 'd', 'a', 't'):
+ case FOURCC("mdat"):
{
mMdatFound = true;
@@ -2432,7 +2448,7 @@
break;
}
- case FOURCC('h', 'd', 'l', 'r'):
+ case FOURCC("hdlr"):
{
*offset += chunk_size;
@@ -2450,7 +2466,7 @@
// For the 3GPP file format, the handler-type within the 'hdlr' box
// shall be 'text'. We also want to support 'sbtl' handler type
// for a practical reason as various MPEG4 containers use it.
- if (type == FOURCC('t', 'e', 'x', 't') || type == FOURCC('s', 'b', 't', 'l')) {
+ if (type == FOURCC("text") || type == FOURCC("sbtl")) {
if (mLastTrack != NULL) {
AMediaFormat_setString(mLastTrack->meta,
AMEDIAFORMAT_KEY_MIME, MEDIA_MIMETYPE_TEXT_3GPP);
@@ -2460,7 +2476,7 @@
break;
}
- case FOURCC('k', 'e', 'y', 's'):
+ case FOURCC("keys"):
{
*offset += chunk_size;
@@ -2473,7 +2489,7 @@
break;
}
- case FOURCC('t', 'r', 'e', 'x'):
+ case FOURCC("trex"):
{
*offset += chunk_size;
@@ -2492,7 +2508,7 @@
break;
}
- case FOURCC('t', 'x', '3', 'g'):
+ case FOURCC("tx3g"):
{
if (mLastTrack == NULL)
return ERROR_MALFORMED;
@@ -2536,7 +2552,7 @@
break;
}
- case FOURCC('c', 'o', 'v', 'r'):
+ case FOURCC("covr"):
{
*offset += chunk_size;
@@ -2567,12 +2583,12 @@
break;
}
- case FOURCC('c', 'o', 'l', 'r'):
+ case FOURCC("colr"):
{
*offset += chunk_size;
// this must be in a VisualSampleEntry box under the Sample Description Box ('stsd')
// ignore otherwise
- if (depth >= 2 && mPath[depth - 2] == FOURCC('s', 't', 's', 'd')) {
+ if (depth >= 2 && mPath[depth - 2] == FOURCC("stsd")) {
status_t err = parseColorInfo(data_offset, chunk_data_size);
if (err != OK) {
return err;
@@ -2582,12 +2598,12 @@
break;
}
- case FOURCC('t', 'i', 't', 'l'):
- case FOURCC('p', 'e', 'r', 'f'):
- case FOURCC('a', 'u', 't', 'h'):
- case FOURCC('g', 'n', 'r', 'e'):
- case FOURCC('a', 'l', 'b', 'm'):
- case FOURCC('y', 'r', 'r', 'c'):
+ case FOURCC("titl"):
+ case FOURCC("perf"):
+ case FOURCC("auth"):
+ case FOURCC("gnre"):
+ case FOURCC("albm"):
+ case FOURCC("yrrc"):
{
*offset += chunk_size;
@@ -2600,7 +2616,7 @@
break;
}
- case FOURCC('I', 'D', '3', '2'):
+ case FOURCC("ID32"):
{
*offset += chunk_size;
@@ -2613,7 +2629,7 @@
break;
}
- case FOURCC('-', '-', '-', '-'):
+ case FOURCC("----"):
{
mLastCommentMean.clear();
mLastCommentName.clear();
@@ -2622,7 +2638,7 @@
break;
}
- case FOURCC('s', 'i', 'd', 'x'):
+ case FOURCC("sidx"):
{
status_t err = parseSegmentIndex(data_offset, chunk_data_size);
if (err != OK) {
@@ -2632,25 +2648,25 @@
return UNKNOWN_ERROR; // stop parsing after sidx
}
- case FOURCC('a', 'c', '-', '3'):
+ case FOURCC("ac-3"):
{
*offset += chunk_size;
return parseAC3SpecificBox(data_offset);
}
- case FOURCC('e', 'c', '-', '3'):
+ case FOURCC("ec-3"):
{
*offset += chunk_size;
return parseEAC3SpecificBox(data_offset);
}
- case FOURCC('a', 'c', '-', '4'):
+ case FOURCC("ac-4"):
{
*offset += chunk_size;
return parseAC4SpecificBox(data_offset);
}
- case FOURCC('f', 't', 'y', 'p'):
+ case FOURCC("ftyp"):
{
if (chunk_data_size < 8 || depth != 0) {
return ERROR_MALFORMED;
@@ -2675,16 +2691,16 @@
brandSet.insert(brand);
}
- if (brandSet.count(FOURCC('q', 't', ' ', ' ')) > 0) {
+ if (brandSet.count(FOURCC("qt ")) > 0) {
mIsQT = true;
} else {
- if (brandSet.count(FOURCC('m', 'i', 'f', '1')) > 0
- && brandSet.count(FOURCC('h', 'e', 'i', 'c')) > 0) {
+ if (brandSet.count(FOURCC("mif1")) > 0
+ && brandSet.count(FOURCC("heic")) > 0) {
ALOGV("identified HEIF image");
mIsHeif = true;
- brandSet.erase(FOURCC('m', 'i', 'f', '1'));
- brandSet.erase(FOURCC('h', 'e', 'i', 'c'));
+ brandSet.erase(FOURCC("mif1"));
+ brandSet.erase(FOURCC("heic"));
}
if (!brandSet.empty()) {
@@ -2771,7 +2787,7 @@
// + 4-byte size
offset += 4;
uint32_t type;
- if (!mDataSource->getUInt32(offset, &type) || type != FOURCC('d', 'a', 'c', '4')) {
+ if (!mDataSource->getUInt32(offset, &type) || type != FOURCC("dac4")) {
ALOGE("MPEG4Extractor: error while reading ac-4 specific block: header not dac4");
return ERROR_MALFORMED;
}
@@ -2898,7 +2914,7 @@
offset += 4;
uint32_t type;
- if (!mDataSource->getUInt32(offset, &type) || type != FOURCC('d', 'e', 'c', '3')) {
+ if (!mDataSource->getUInt32(offset, &type) || type != FOURCC("dec3")) {
ALOGE("MPEG4Extractor: error while reading eac-3 specific block: header not dec3");
return ERROR_MALFORMED;
}
@@ -3055,7 +3071,7 @@
offset += 4;
uint32_t type;
- if (!mDataSource->getUInt32(offset, &type) || type != FOURCC('d', 'a', 'c', '3')) {
+ if (!mDataSource->getUInt32(offset, &type) || type != FOURCC("dac3")) {
ALOGE("MPEG4Extractor: error while reading ac-3 specific block: header not dac3");
return ERROR_MALFORMED;
}
@@ -3257,7 +3273,7 @@
uint32_t type;
if (!mDataSource->getUInt32(keyOffset + 4, &type)
- || type != FOURCC('m', 'd', 't', 'a')) {
+ || type != FOURCC("mdta")) {
return ERROR_MALFORMED;
}
@@ -3299,7 +3315,7 @@
}
uint32_t atomFourCC;
if (!mDataSource->getUInt32(offset + 4, &atomFourCC)
- || atomFourCC != FOURCC('d', 'a', 't', 'a')) {
+ || atomFourCC != FOURCC("data")) {
return ERROR_MALFORMED;
}
uint32_t dataType;
@@ -3460,48 +3476,48 @@
MakeFourCCString(mPath[4], chunk);
ALOGV("meta: %s @ %lld", chunk, (long long)offset);
switch ((int32_t)mPath[4]) {
- case FOURCC(0xa9, 'a', 'l', 'b'):
+ case FOURCC("\251alb"):
{
metadataKey = "album";
break;
}
- case FOURCC(0xa9, 'A', 'R', 'T'):
+ case FOURCC("\251ART"):
{
metadataKey = "artist";
break;
}
- case FOURCC('a', 'A', 'R', 'T'):
+ case FOURCC("aART"):
{
metadataKey = "albumartist";
break;
}
- case FOURCC(0xa9, 'd', 'a', 'y'):
+ case FOURCC("\251day"):
{
metadataKey = "year";
break;
}
- case FOURCC(0xa9, 'n', 'a', 'm'):
+ case FOURCC("\251nam"):
{
metadataKey = "title";
break;
}
- case FOURCC(0xa9, 'w', 'r', 't'):
+ case FOURCC("\251wrt"):
{
metadataKey = "writer";
break;
}
- case FOURCC('c', 'o', 'v', 'r'):
+ case FOURCC("covr"):
{
metadataKey = "albumart";
break;
}
- case FOURCC('g', 'n', 'r', 'e'):
- case FOURCC(0xa9, 'g', 'e', 'n'):
+ case FOURCC("gnre"):
+ case FOURCC("\251gen"):
{
metadataKey = "genre";
break;
}
- case FOURCC('c', 'p', 'i', 'l'):
+ case FOURCC("cpil"):
{
if (size == 9 && flags == 21) {
char tmp[16];
@@ -3512,7 +3528,7 @@
}
break;
}
- case FOURCC('t', 'r', 'k', 'n'):
+ case FOURCC("trkn"):
{
if (size == 16 && flags == 0) {
char tmp[16];
@@ -3524,7 +3540,7 @@
}
break;
}
- case FOURCC('d', 'i', 's', 'k'):
+ case FOURCC("disk"):
{
if ((size == 14 || size == 16) && flags == 0) {
char tmp[16];
@@ -3536,17 +3552,17 @@
}
break;
}
- case FOURCC('-', '-', '-', '-'):
+ case FOURCC("----"):
{
buffer[size] = '\0';
switch (mPath[5]) {
- case FOURCC('m', 'e', 'a', 'n'):
+ case FOURCC("mean"):
mLastCommentMean.setTo((const char *)buffer + 4);
break;
- case FOURCC('n', 'a', 'm', 'e'):
+ case FOURCC("name"):
mLastCommentName.setTo((const char *)buffer + 4);
break;
- case FOURCC('d', 'a', 't', 'a'):
+ case FOURCC("data"):
if (size < 8) {
delete[] buffer;
buffer = NULL;
@@ -3654,8 +3670,8 @@
}
int32_t type = U32_AT(&buffer[0]);
- if ((type == FOURCC('n', 'c', 'l', 'x') && size >= 11)
- || (type == FOURCC('n', 'c', 'l', 'c') && size >= 10)) {
+ if ((type == FOURCC("nclx") && size >= 11)
+ || (type == FOURCC("nclc") && size >= 10)) {
// only store the first color specification
int32_t existingColor;
if (!AMediaFormat_getInt32(mLastTrack->meta,
@@ -3663,7 +3679,7 @@
int32_t primaries = U16_AT(&buffer[4]);
int32_t isotransfer = U16_AT(&buffer[6]);
int32_t coeffs = U16_AT(&buffer[8]);
- bool fullRange = (type == FOURCC('n', 'c', 'l', 'x')) && (buffer[10] & 128);
+ bool fullRange = (type == FOURCC("nclx")) && (buffer[10] & 128);
int32_t range = 0;
int32_t standard = 0;
@@ -3709,27 +3725,27 @@
const char *metadataKey = nullptr;
switch (mPath[depth]) {
- case FOURCC('t', 'i', 't', 'l'):
+ case FOURCC("titl"):
{
metadataKey = "title";
break;
}
- case FOURCC('p', 'e', 'r', 'f'):
+ case FOURCC("perf"):
{
metadataKey = "artist";
break;
}
- case FOURCC('a', 'u', 't', 'h'):
+ case FOURCC("auth"):
{
metadataKey = "writer";
break;
}
- case FOURCC('g', 'n', 'r', 'e'):
+ case FOURCC("gnre"):
{
metadataKey = "genre";
break;
}
- case FOURCC('a', 'l', 'b', 'm'):
+ case FOURCC("albm"):
{
if (buffer[size - 1] != '\0') {
char tmp[4];
@@ -3741,7 +3757,7 @@
metadataKey = "album";
break;
}
- case FOURCC('y', 'r', 'r', 'c'):
+ case FOURCC("yrrc"):
{
if (size < 6) {
delete[] buffer;
@@ -4487,7 +4503,7 @@
}
if (!strncasecmp("video/", mime, 6)) {
- uint32_t firstSampleCTS = 0;
+ uint64_t firstSampleCTS = 0;
err = mSampleTable->getMetaDataForSample(0, NULL, NULL, &firstSampleCTS);
// Start offset should be less or equal to composition time of first sample.
// Composition time stamp of first sample cannot be negative.
@@ -4594,8 +4610,8 @@
switch(chunk_type) {
- case FOURCC('t', 'r', 'a', 'f'):
- case FOURCC('m', 'o', 'o', 'f'): {
+ case FOURCC("traf"):
+ case FOURCC("moof"): {
off64_t stop_offset = *offset + chunk_size;
*offset = data_offset;
while (*offset < stop_offset) {
@@ -4604,7 +4620,7 @@
return err;
}
}
- if (chunk_type == FOURCC('m', 'o', 'o', 'f')) {
+ if (chunk_type == FOURCC("moof")) {
// *offset points to the box following this moof. Find the next moof from there.
while (true) {
@@ -4633,7 +4649,7 @@
return ERROR_MALFORMED;
}
- if (chunk_type == FOURCC('m', 'o', 'o', 'f')) {
+ if (chunk_type == FOURCC("moof")) {
mNextMoofOffset = *offset;
break;
} else if (chunk_size == 0) {
@@ -4645,7 +4661,7 @@
break;
}
- case FOURCC('t', 'f', 'h', 'd'): {
+ case FOURCC("tfhd"): {
status_t err;
if ((err = parseTrackFragmentHeader(data_offset, chunk_data_size)) != OK) {
return err;
@@ -4654,7 +4670,7 @@
break;
}
- case FOURCC('t', 'r', 'u', 'n'): {
+ case FOURCC("trun"): {
status_t err;
if (mLastParsedTrackId == mTrackId) {
if ((err = parseTrackFragmentRun(data_offset, chunk_data_size)) != OK) {
@@ -4666,7 +4682,7 @@
break;
}
- case FOURCC('s', 'a', 'i', 'z'): {
+ case FOURCC("saiz"): {
status_t err;
if ((err = parseSampleAuxiliaryInformationSizes(data_offset, chunk_data_size)) != OK) {
return err;
@@ -4674,7 +4690,7 @@
*offset += chunk_size;
break;
}
- case FOURCC('s', 'a', 'i', 'o'): {
+ case FOURCC("saio"): {
status_t err;
if ((err = parseSampleAuxiliaryInformationOffsets(data_offset, chunk_data_size))
!= OK) {
@@ -4684,7 +4700,7 @@
break;
}
- case FOURCC('s', 'e', 'n', 'c'): {
+ case FOURCC("senc"): {
status_t err;
if ((err = parseSampleEncryption(data_offset)) != OK) {
return err;
@@ -4693,7 +4709,7 @@
break;
}
- case FOURCC('m', 'd', 'a', 't'): {
+ case FOURCC("mdat"): {
// parse DRM info if present
ALOGV("MPEG4Source::parseChunk mdat");
// if saiz/saoi was previously observed, do something with the sampleinfos
@@ -4852,7 +4868,9 @@
off64_t offset, bool isSubsampleEncryption, uint32_t flags) {
int32_t ivlength;
- CHECK(AMediaFormat_getInt32(mFormat, AMEDIAFORMAT_KEY_CRYPTO_DEFAULT_IV_SIZE, &ivlength));
+ if (!AMediaFormat_getInt32(mFormat, AMEDIAFORMAT_KEY_CRYPTO_DEFAULT_IV_SIZE, &ivlength)) {
+ return ERROR_MALFORMED;
+ }
// only 0, 8 and 16 byte initialization vectors are supported
if (ivlength != 0 && ivlength != 8 && ivlength != 16) {
@@ -5142,7 +5160,7 @@
sampleCtsOffset = 0;
}
- if (size < (off64_t)(sampleCount * bytesPerSample)) {
+ if (size < (off64_t)sampleCount * bytesPerSample) {
return -EINVAL;
}
@@ -5333,7 +5351,7 @@
sampleIndex, &syncSampleIndex, findFlags);
}
- uint32_t sampleTime;
+ uint64_t sampleTime;
if (err == OK) {
err = mSampleTable->getMetaDataForSample(
sampleIndex, NULL, NULL, &sampleTime);
@@ -5383,7 +5401,7 @@
off64_t offset = 0;
size_t size = 0;
- uint32_t cts, stts;
+ uint64_t cts, stts;
bool isSyncSample;
bool newBuffer = false;
if (mBuffer == NULL) {
@@ -6015,28 +6033,29 @@
static bool isCompatibleBrand(uint32_t fourcc) {
static const uint32_t kCompatibleBrands[] = {
- FOURCC('i', 's', 'o', 'm'),
- FOURCC('i', 's', 'o', '2'),
- FOURCC('a', 'v', 'c', '1'),
- FOURCC('h', 'v', 'c', '1'),
- FOURCC('h', 'e', 'v', '1'),
- FOURCC('a', 'v', '0', '1'),
- FOURCC('3', 'g', 'p', '4'),
- FOURCC('m', 'p', '4', '1'),
- FOURCC('m', 'p', '4', '2'),
- FOURCC('d', 'a', 's', 'h'),
+ FOURCC("isom"),
+ FOURCC("iso2"),
+ FOURCC("avc1"),
+ FOURCC("hvc1"),
+ FOURCC("hev1"),
+ FOURCC("av01"),
+ FOURCC("3gp4"),
+ FOURCC("mp41"),
+ FOURCC("mp42"),
+ FOURCC("dash"),
// Won't promise that the following file types can be played.
// Just give these file types a chance.
- FOURCC('q', 't', ' ', ' '), // Apple's QuickTime
- FOURCC('M', 'S', 'N', 'V'), // Sony's PSP
+ FOURCC("qt "), // Apple's QuickTime
+ FOURCC("MSNV"), // Sony's PSP
+ FOURCC("wmf "),
- FOURCC('3', 'g', '2', 'a'), // 3GPP2
- FOURCC('3', 'g', '2', 'b'),
- FOURCC('m', 'i', 'f', '1'), // HEIF image
- FOURCC('h', 'e', 'i', 'c'), // HEIF image
- FOURCC('m', 's', 'f', '1'), // HEIF image sequence
- FOURCC('h', 'e', 'v', 'c'), // HEIF image sequence
+ FOURCC("3g2a"), // 3GPP2
+ FOURCC("3g2b"),
+ FOURCC("mif1"), // HEIF image
+ FOURCC("heic"), // HEIF image
+ FOURCC("msf1"), // HEIF image sequence
+ FOURCC("hevc"), // HEIF image sequence
};
for (size_t i = 0;
@@ -6104,7 +6123,7 @@
ALOGV("saw chunk type %s, size %" PRIu64 " @ %lld",
chunkstring, chunkSize, (long long)offset);
switch (chunkType) {
- case FOURCC('f', 't', 'y', 'p'):
+ case FOURCC("ftyp"):
{
if (chunkDataSize < 8) {
return false;
@@ -6139,7 +6158,7 @@
break;
}
- case FOURCC('m', 'o', 'o', 'v'):
+ case FOURCC("moov"):
{
moovAtomEndOffset = offset + chunkSize;
diff --git a/media/extractors/mp4/SampleIterator.cpp b/media/extractors/mp4/SampleIterator.cpp
index 1a6d306..ec12130 100644
--- a/media/extractors/mp4/SampleIterator.cpp
+++ b/media/extractors/mp4/SampleIterator.cpp
@@ -301,7 +301,7 @@
}
status_t SampleIterator::findSampleTimeAndDuration(
- uint32_t sampleIndex, uint32_t *time, uint32_t *duration) {
+ uint32_t sampleIndex, uint64_t *time, uint64_t *duration) {
if (sampleIndex >= mTable->mNumSampleSizes) {
return ERROR_OUT_OF_RANGE;
}
@@ -314,8 +314,8 @@
break;
}
if (mTimeToSampleIndex == mTable->mTimeToSampleCount ||
- (mTTSDuration != 0 && mTTSCount > UINT32_MAX / mTTSDuration) ||
- mTTSSampleTime > UINT32_MAX - (mTTSCount * mTTSDuration)) {
+ (mTTSDuration != 0 && mTTSCount > UINT64_MAX / mTTSDuration) ||
+ mTTSSampleTime > UINT64_MAX - (mTTSCount * mTTSDuration)) {
return ERROR_OUT_OF_RANGE;
}
@@ -330,7 +330,7 @@
// below is equivalent to:
// *time = mTTSSampleTime + mTTSDuration * (sampleIndex - mTTSSampleIndex);
- uint32_t tmp;
+ uint64_t tmp;
if (__builtin_sub_overflow(sampleIndex, mTTSSampleIndex, &tmp) ||
__builtin_mul_overflow(mTTSDuration, tmp, &tmp) ||
__builtin_add_overflow(mTTSSampleTime, tmp, &tmp)) {
@@ -340,15 +340,15 @@
int32_t offset = mTable->getCompositionTimeOffset(sampleIndex);
if ((offset < 0 && *time < (offset == INT32_MIN ?
- INT32_MAX : uint32_t(-offset))) ||
- (offset > 0 && *time > UINT32_MAX - offset)) {
- ALOGE("%u + %d would overflow", *time, offset);
+ INT64_MAX : uint64_t(-offset))) ||
+ (offset > 0 && *time > UINT64_MAX - offset)) {
+ ALOGE("%llu + %d would overflow", (unsigned long long) *time, offset);
return ERROR_OUT_OF_RANGE;
}
if (offset > 0) {
*time += offset;
} else {
- *time -= (offset == INT32_MIN ? INT32_MAX : (-offset));
+ *time -= (offset == INT64_MIN ? INT64_MAX : (-offset));
}
*duration = mTTSDuration;
diff --git a/media/extractors/mp4/SampleIterator.h b/media/extractors/mp4/SampleIterator.h
index 6e4f60e..5a0ea76 100644
--- a/media/extractors/mp4/SampleIterator.h
+++ b/media/extractors/mp4/SampleIterator.h
@@ -33,8 +33,8 @@
uint32_t getDescIndex() const { return mChunkDesc; }
off64_t getSampleOffset() const { return mCurrentSampleOffset; }
size_t getSampleSize() const { return mCurrentSampleSize; }
- uint32_t getSampleTime() const { return mCurrentSampleTime; }
- uint32_t getSampleDuration() const { return mCurrentSampleDuration; }
+ uint64_t getSampleTime() const { return mCurrentSampleTime; }
+ uint64_t getSampleDuration() const { return mCurrentSampleDuration; }
uint32_t getLastSampleIndexInChunk() const {
return mCurrentSampleIndex + mSamplesPerChunk -
@@ -63,20 +63,20 @@
uint32_t mTimeToSampleIndex;
uint32_t mTTSSampleIndex;
- uint32_t mTTSSampleTime;
+ uint64_t mTTSSampleTime;
uint32_t mTTSCount;
- uint32_t mTTSDuration;
+ uint64_t mTTSDuration;
uint32_t mCurrentSampleIndex;
off64_t mCurrentSampleOffset;
size_t mCurrentSampleSize;
- uint32_t mCurrentSampleTime;
- uint32_t mCurrentSampleDuration;
+ uint64_t mCurrentSampleTime;
+ uint64_t mCurrentSampleDuration;
void reset();
status_t findChunkRange(uint32_t sampleIndex);
status_t getChunkOffset(uint32_t chunk, off64_t *offset);
- status_t findSampleTimeAndDuration(uint32_t sampleIndex, uint32_t *time, uint32_t *duration);
+ status_t findSampleTimeAndDuration(uint32_t sampleIndex, uint64_t *time, uint64_t *duration);
SampleIterator(const SampleIterator &);
SampleIterator &operator=(const SampleIterator &);
diff --git a/media/extractors/mp4/SampleTable.cpp b/media/extractors/mp4/SampleTable.cpp
index d242798..bf29bf1 100644
--- a/media/extractors/mp4/SampleTable.cpp
+++ b/media/extractors/mp4/SampleTable.cpp
@@ -37,13 +37,13 @@
namespace android {
// static
-const uint32_t SampleTable::kChunkOffsetType32 = FOURCC('s', 't', 'c', 'o');
+const uint32_t SampleTable::kChunkOffsetType32 = FOURCC("stco");
// static
-const uint32_t SampleTable::kChunkOffsetType64 = FOURCC('c', 'o', '6', '4');
+const uint32_t SampleTable::kChunkOffsetType64 = FOURCC("co64");
// static
-const uint32_t SampleTable::kSampleSizeType32 = FOURCC('s', 't', 's', 'z');
+const uint32_t SampleTable::kSampleSizeType32 = FOURCC("stsz");
// static
-const uint32_t SampleTable::kSampleSizeTypeCompact = FOURCC('s', 't', 'z', '2');
+const uint32_t SampleTable::kSampleSizeTypeCompact = FOURCC("stz2");
////////////////////////////////////////////////////////////////////////////////
@@ -614,7 +614,7 @@
return OK;
}
-uint32_t abs_difference(uint32_t time1, uint32_t time2) {
+uint32_t abs_difference(uint64_t time1, uint64_t time2) {
return time1 > time2 ? time1 - time2 : time2 - time1;
}
@@ -662,7 +662,7 @@
}
uint32_t sampleIndex = 0;
- uint32_t sampleTime = 0;
+ uint64_t sampleTime = 0;
for (uint32_t i = 0; i < mTimeToSampleCount; ++i) {
uint32_t n = mTimeToSample[2 * i];
@@ -684,13 +684,13 @@
(compTimeDelta == INT32_MIN ?
INT32_MAX : uint32_t(-compTimeDelta)))
|| (compTimeDelta > 0 &&
- sampleTime > UINT32_MAX - compTimeDelta)) {
- ALOGE("%u + %d would overflow, clamping",
- sampleTime, compTimeDelta);
+ sampleTime > UINT64_MAX - compTimeDelta)) {
+ ALOGE("%llu + %d would overflow, clamping",
+ (unsigned long long) sampleTime, compTimeDelta);
if (compTimeDelta < 0) {
sampleTime = 0;
} else {
- sampleTime = UINT32_MAX;
+ sampleTime = UINT64_MAX;
}
compTimeDelta = 0;
}
@@ -701,10 +701,10 @@
}
++sampleIndex;
- if (sampleTime > UINT32_MAX - delta) {
- ALOGE("%u + %u would overflow, clamping",
- sampleTime, delta);
- sampleTime = UINT32_MAX;
+ if (sampleTime > UINT64_MAX - delta) {
+ ALOGE("%llu + %u would overflow, clamping",
+ (unsigned long long) sampleTime, delta);
+ sampleTime = UINT64_MAX;
} else {
sampleTime += delta;
}
@@ -870,19 +870,19 @@
if (err != OK) {
return err;
}
- uint32_t sample_time = mSampleIterator->getSampleTime();
+ uint64_t sample_time = mSampleIterator->getSampleTime();
err = mSampleIterator->seekTo(mSyncSamples[left]);
if (err != OK) {
return err;
}
- uint32_t upper_time = mSampleIterator->getSampleTime();
+ uint64_t upper_time = mSampleIterator->getSampleTime();
err = mSampleIterator->seekTo(mSyncSamples[left - 1]);
if (err != OK) {
return err;
}
- uint32_t lower_time = mSampleIterator->getSampleTime();
+ uint64_t lower_time = mSampleIterator->getSampleTime();
// use abs_difference for safety
if (abs_difference(upper_time, sample_time) >
@@ -955,9 +955,9 @@
uint32_t sampleIndex,
off64_t *offset,
size_t *size,
- uint32_t *compositionTime,
+ uint64_t *compositionTime,
bool *isSyncSample,
- uint32_t *sampleDuration) {
+ uint64_t *sampleDuration) {
Mutex::Autolock autoLock(mLock);
status_t err;
diff --git a/media/extractors/mp4/SampleTable.h b/media/extractors/mp4/SampleTable.h
index d4b5dc8..57f6e62 100644
--- a/media/extractors/mp4/SampleTable.h
+++ b/media/extractors/mp4/SampleTable.h
@@ -66,9 +66,9 @@
uint32_t sampleIndex,
off64_t *offset,
size_t *size,
- uint32_t *compositionTime,
+ uint64_t *compositionTime,
bool *isSyncSample = NULL,
- uint32_t *sampleDuration = NULL);
+ uint64_t *sampleDuration = NULL);
// call only after getMetaDataForSample has been called successfully.
uint32_t getLastSampleIndexInChunk();
@@ -124,7 +124,7 @@
struct SampleTimeEntry {
uint32_t mSampleIndex;
- uint32_t mCompositionTime;
+ uint64_t mCompositionTime;
};
SampleTimeEntry *mSampleTimeEntries;
diff --git a/media/extractors/mpeg2/MPEG2TSExtractor.cpp b/media/extractors/mpeg2/MPEG2TSExtractor.cpp
index e1509ee..49dd0b4 100644
--- a/media/extractors/mpeg2/MPEG2TSExtractor.cpp
+++ b/media/extractors/mpeg2/MPEG2TSExtractor.cpp
@@ -302,16 +302,21 @@
return AMEDIA_ERROR_UNKNOWN;
}
-void MPEG2TSExtractor::addSource(const sp<AnotherPacketSource> &impl) {
- bool found = false;
+status_t MPEG2TSExtractor::findIndexOfSource(const sp<AnotherPacketSource> &impl, size_t *index) {
for (size_t i = 0; i < mSourceImpls.size(); i++) {
if (mSourceImpls[i] == impl) {
- found = true;
- break;
+ *index = i;
+ return OK;
}
}
- if (!found) {
+ return NAME_NOT_FOUND;
+}
+
+void MPEG2TSExtractor::addSource(const sp<AnotherPacketSource> &impl) {
+ size_t index;
+ if (findIndexOfSource(impl, &index) != OK) {
mSourceImpls.push(impl);
+ mSyncPoints.push();
}
}
@@ -319,6 +324,7 @@
bool haveAudio = false;
bool haveVideo = false;
int64_t startTime = ALooper::GetNowUs();
+ size_t index;
status_t err;
while ((err = feedMore(true /* isInit */)) == OK
@@ -337,8 +343,9 @@
haveVideo = true;
addSource(impl);
if (!isScrambledFormat(*(format.get()))) {
- mSyncPoints.push();
- mSeekSyncPoints = &mSyncPoints.editTop();
+ if (findIndexOfSource(impl, &index) == OK) {
+ mSeekSyncPoints = &mSyncPoints.editItemAt(index);
+ }
}
}
}
@@ -352,10 +359,9 @@
if (format != NULL) {
haveAudio = true;
addSource(impl);
- if (!isScrambledFormat(*(format.get()))) {
- mSyncPoints.push();
- if (!haveVideo) {
- mSeekSyncPoints = &mSyncPoints.editTop();
+ if (!isScrambledFormat(*(format.get())) && !haveVideo) {
+ if (findIndexOfSource(impl, &index) == OK) {
+ mSeekSyncPoints = &mSyncPoints.editItemAt(index);
}
}
}
diff --git a/media/extractors/mpeg2/MPEG2TSExtractor.h b/media/extractors/mpeg2/MPEG2TSExtractor.h
index e425d23..2537d3b 100644
--- a/media/extractors/mpeg2/MPEG2TSExtractor.h
+++ b/media/extractors/mpeg2/MPEG2TSExtractor.h
@@ -95,6 +95,7 @@
status_t seekBeyond(int64_t seekTimeUs);
status_t feedUntilBufferAvailable(const sp<AnotherPacketSource> &impl);
+ status_t findIndexOfSource(const sp<AnotherPacketSource> &impl, size_t *index);
// Add a SynPoint derived from |event|.
void addSyncPoint_l(const ATSParser::SyncEvent &event);
diff --git a/media/libaaudio/examples/loopback/src/loopback.cpp b/media/libaaudio/examples/loopback/src/loopback.cpp
index 2a02b20..3de1514 100644
--- a/media/libaaudio/examples/loopback/src/loopback.cpp
+++ b/media/libaaudio/examples/loopback/src/loopback.cpp
@@ -105,9 +105,14 @@
assert(false);
}
if (framesRead < 0) {
- myData->inputError = framesRead;
- printf("ERROR in read = %d = %s\n", framesRead,
- AAudio_convertResultToText(framesRead));
+ // Expect INVALID_STATE if STATE_STARTING
+ if (myData->framesReadTotal > 0) {
+ myData->inputError = framesRead;
+ printf("ERROR in read = %d = %s\n", framesRead,
+ AAudio_convertResultToText(framesRead));
+ } else {
+ framesRead = 0;
+ }
} else {
myData->framesReadTotal += framesRead;
}
@@ -149,8 +154,10 @@
int32_t totalFramesRead = 0;
do {
actualFramesRead = readFormattedData(myData, numFrames);
- if (actualFramesRead) {
+ if (actualFramesRead > 0) {
totalFramesRead += actualFramesRead;
+ } else if (actualFramesRead < 0) {
+ result = AAUDIO_CALLBACK_RESULT_STOP;
}
// Ignore errors because input stream may not be started yet.
} while (actualFramesRead > 0);
diff --git a/media/libaaudio/src/client/AudioStreamInternal.cpp b/media/libaaudio/src/client/AudioStreamInternal.cpp
index fffcda0..3b03601 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternal.cpp
@@ -62,7 +62,7 @@
, mServiceStreamHandle(AAUDIO_HANDLE_INVALID)
, mInService(inService)
, mServiceInterface(serviceInterface)
- , mAtomicTimestamp()
+ , mAtomicInternalTimestamp()
, mWakeupDelayNanos(AAudioProperty_getWakeupDelayMicros() * AAUDIO_NANOS_PER_MICROSECOND)
, mMinimumSleepNanos(AAudioProperty_getMinimumSleepMicros() * AAUDIO_NANOS_PER_MICROSECOND)
{
@@ -349,8 +349,7 @@
}
}
-aaudio_result_t AudioStreamInternal::requestStop()
-{
+aaudio_result_t AudioStreamInternal::requestStop() {
aaudio_result_t result = stopCallback();
if (result != AAUDIO_OK) {
return result;
@@ -364,7 +363,7 @@
mClockModel.stop(AudioClock::getNanoseconds());
setState(AAUDIO_STREAM_STATE_STOPPING);
- mAtomicTimestamp.clear();
+ mAtomicInternalTimestamp.clear();
return mServiceInterface.stopStream(mServiceStreamHandle);
}
@@ -413,8 +412,8 @@
int64_t *framePosition,
int64_t *timeNanoseconds) {
// Generated in server and passed to client. Return latest.
- if (mAtomicTimestamp.isValid()) {
- Timestamp timestamp = mAtomicTimestamp.read();
+ if (mAtomicInternalTimestamp.isValid()) {
+ Timestamp timestamp = mAtomicInternalTimestamp.read();
int64_t position = timestamp.getPosition() + mFramesOffsetFromService;
if (position >= 0) {
*framePosition = position;
@@ -461,7 +460,7 @@
aaudio_result_t AudioStreamInternal::onTimestampHardware(AAudioServiceMessage *message) {
Timestamp timestamp(message->timestamp.position, message->timestamp.timestamp);
- mAtomicTimestamp.write(timestamp);
+ mAtomicInternalTimestamp.write(timestamp);
return AAUDIO_OK;
}
diff --git a/media/libaaudio/src/client/AudioStreamInternal.h b/media/libaaudio/src/client/AudioStreamInternal.h
index 3bb9e1e..1c88f52 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.h
+++ b/media/libaaudio/src/client/AudioStreamInternal.h
@@ -163,7 +163,7 @@
AAudioServiceInterface &mServiceInterface; // abstract interface to the service
- SimpleDoubleBuffer<Timestamp> mAtomicTimestamp;
+ SimpleDoubleBuffer<Timestamp> mAtomicInternalTimestamp;
AtomicRequestor mNeedCatchUp; // Ask read() or write() to sync on first timestamp.
diff --git a/media/libaaudio/src/client/AudioStreamInternalCapture.cpp b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
index 58ef7b1..7dcb620 100644
--- a/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
@@ -259,6 +259,7 @@
if (callbackResult == AAUDIO_CALLBACK_RESULT_STOP) {
ALOGD("%s(): callback returned AAUDIO_CALLBACK_RESULT_STOP", __func__);
+ result = systemStopFromCallback();
break;
}
}
diff --git a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
index 9af47b2..6af8e7d 100644
--- a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
@@ -71,7 +71,7 @@
mClockModel.stop(AudioClock::getNanoseconds());
setState(AAUDIO_STREAM_STATE_PAUSING);
- mAtomicTimestamp.clear();
+ mAtomicInternalTimestamp.clear();
return mServiceInterface.pauseStream(mServiceStreamHandle);
}
@@ -294,6 +294,7 @@
}
} else if (callbackResult == AAUDIO_CALLBACK_RESULT_STOP) {
ALOGD("%s(): callback returned AAUDIO_CALLBACK_RESULT_STOP", __func__);
+ result = systemStopFromCallback();
break;
}
}
diff --git a/media/libaaudio/src/core/AAudioAudio.cpp b/media/libaaudio/src/core/AAudioAudio.cpp
index 2fb3986..0d71efc 100644
--- a/media/libaaudio/src/core/AAudioAudio.cpp
+++ b/media/libaaudio/src/core/AAudioAudio.cpp
@@ -316,7 +316,7 @@
{
AudioStream *audioStream = convertAAudioStreamToAudioStream(stream);
ALOGD("%s(%p) called", __func__, stream);
- return audioStream->systemStop();
+ return audioStream->systemStopFromApp();
}
AAUDIO_API aaudio_result_t AAudioStream_waitForStateChange(AAudioStream* stream,
diff --git a/media/libaaudio/src/core/AudioStream.cpp b/media/libaaudio/src/core/AudioStream.cpp
index 391af29..e39a075 100644
--- a/media/libaaudio/src/core/AudioStream.cpp
+++ b/media/libaaudio/src/core/AudioStream.cpp
@@ -119,21 +119,29 @@
return AAUDIO_OK;
}
-aaudio_result_t AudioStream::safeStart() {
+aaudio_result_t AudioStream::systemStart() {
std::lock_guard<std::mutex> lock(mStreamLock);
+
if (collidesWithCallback()) {
ALOGE("%s cannot be called from a callback!", __func__);
return AAUDIO_ERROR_INVALID_STATE;
}
- return requestStart();
+
+ aaudio_result_t result = requestStart();
+ if (result == AAUDIO_OK) {
+ // We only call this for logging in "dumpsys audio". So ignore return code.
+ (void) mPlayerBase->start();
+ }
+ return result;
}
-aaudio_result_t AudioStream::safePause() {
+aaudio_result_t AudioStream::systemPause() {
+ std::lock_guard<std::mutex> lock(mStreamLock);
+
if (!isPauseSupported()) {
return AAUDIO_ERROR_UNIMPLEMENTED;
}
- std::lock_guard<std::mutex> lock(mStreamLock);
if (collidesWithCallback()) {
ALOGE("%s cannot be called from a callback!", __func__);
return AAUDIO_ERROR_INVALID_STATE;
@@ -169,7 +177,12 @@
return AAUDIO_ERROR_INVALID_STATE;
}
- return requestPause();
+ aaudio_result_t result = requestPause();
+ if (result == AAUDIO_OK) {
+ // We only call this for logging in "dumpsys audio". So ignore return code.
+ (void) mPlayerBase->pause();
+ }
+ return result;
}
aaudio_result_t AudioStream::safeFlush() {
@@ -192,12 +205,31 @@
return requestFlush();
}
-aaudio_result_t AudioStream::safeStop() {
+aaudio_result_t AudioStream::systemStopFromCallback() {
+ std::lock_guard<std::mutex> lock(mStreamLock);
+ aaudio_result_t result = safeStop();
+ if (result == AAUDIO_OK) {
+ // We only call this for logging in "dumpsys audio". So ignore return code.
+ (void) mPlayerBase->stop();
+ }
+ return result;
+}
+
+aaudio_result_t AudioStream::systemStopFromApp() {
std::lock_guard<std::mutex> lock(mStreamLock);
if (collidesWithCallback()) {
- ALOGE("stream cannot be stopped from a callback!");
+ ALOGE("stream cannot be stopped by calling from a callback!");
return AAUDIO_ERROR_INVALID_STATE;
}
+ aaudio_result_t result = safeStop();
+ if (result == AAUDIO_OK) {
+ // We only call this for logging in "dumpsys audio". So ignore return code.
+ (void) mPlayerBase->stop();
+ }
+ return result;
+}
+
+aaudio_result_t AudioStream::safeStop() {
switch (getState()) {
// Proceed with stopping.
@@ -224,7 +256,7 @@
case AAUDIO_STREAM_STATE_CLOSING:
case AAUDIO_STREAM_STATE_CLOSED:
default:
- ALOGW("requestStop() stream not running, state = %s",
+ ALOGW("%s() stream not running, state = %s", __func__,
AAudio_convertStreamStateToText(getState()));
return AAUDIO_ERROR_INVALID_STATE;
}
@@ -349,21 +381,33 @@
}
}
-aaudio_result_t AudioStream::joinThread(void** returnArg, int64_t timeoutNanoseconds)
+aaudio_result_t AudioStream::joinThread(void** returnArg, int64_t timeoutNanoseconds __unused)
{
if (!mHasThread) {
ALOGE("joinThread() - but has no thread");
return AAUDIO_ERROR_INVALID_STATE;
}
+ aaudio_result_t result = AAUDIO_OK;
+ // If the callback is stopping the stream because the app passed back STOP
+ // then we don't need to join(). The thread is already about to exit.
+ if (pthread_self() != mThread) {
+ // Called from an app thread. Not the callback.
#if 0
- // TODO implement equivalent of pthread_timedjoin_np()
- struct timespec abstime;
- int err = pthread_timedjoin_np(mThread, returnArg, &abstime);
+ // TODO implement equivalent of pthread_timedjoin_np()
+ struct timespec abstime;
+ int err = pthread_timedjoin_np(mThread, returnArg, &abstime);
#else
- int err = pthread_join(mThread, returnArg);
+ int err = pthread_join(mThread, returnArg);
#endif
+ if (err) {
+ ALOGE("%s() pthread_join() returns err = %d", __func__, err);
+ result = AAudioConvert_androidToAAudioResult(-err);
+ }
+ }
+ // This must be set false so that the callback thread can be created
+ // when the stream is restarted.
mHasThread = false;
- return err ? AAudioConvert_androidToAAudioResult(-errno) : mThreadRegistrationResult;
+ return (result != AAUDIO_OK) ? result : mThreadRegistrationResult;
}
aaudio_data_callback_result_t AudioStream::maybeCallDataCallback(void *audioData,
diff --git a/media/libaaudio/src/core/AudioStream.h b/media/libaaudio/src/core/AudioStream.h
index 60200b2..46951f5 100644
--- a/media/libaaudio/src/core/AudioStream.h
+++ b/media/libaaudio/src/core/AudioStream.h
@@ -51,21 +51,6 @@
virtual ~AudioStream();
- /**
- * Lock a mutex and make sure we are not calling from a callback function.
- * @return result of requestStart();
- */
- aaudio_result_t safeStart();
-
- aaudio_result_t safePause();
-
- aaudio_result_t safeFlush();
-
- aaudio_result_t safeStop();
-
- aaudio_result_t safeClose();
-
- // =========== Begin ABSTRACT methods ===========================
protected:
/* Asynchronous requests.
@@ -74,7 +59,7 @@
virtual aaudio_result_t requestStart() = 0;
/**
- * Check the state to see if Pause if currently legal.
+ * Check the state to see if Pause is currently legal.
*
* @param result pointer to return code
* @return true if OK to continue, if false then return result
@@ -356,33 +341,28 @@
mPlayerBase->unregisterWithAudioManager();
}
- // Pass start request through PlayerBase for tracking.
- aaudio_result_t systemStart() {
- mPlayerBase->start();
- // Pass aaudio_result_t around the PlayerBase interface, which uses status__t.
- return mPlayerBase->getResult();
- }
+ aaudio_result_t systemStart();
- // Pass pause request through PlayerBase for tracking.
- aaudio_result_t systemPause() {
- mPlayerBase->pause();
- return mPlayerBase->getResult();
- }
+ aaudio_result_t systemPause();
- // Pass stop request through PlayerBase for tracking.
- aaudio_result_t systemStop() {
- mPlayerBase->stop();
- return mPlayerBase->getResult();
- }
+ aaudio_result_t safeFlush();
+
+ /**
+ * This is called when an app calls AAudioStream_requestStop();
+ * It prevents calls from a callback.
+ */
+ aaudio_result_t systemStopFromApp();
+
+ /**
+ * This is called internally when an app callback returns AAUDIO_CALLBACK_RESULT_STOP.
+ */
+ aaudio_result_t systemStopFromCallback();
+
+ aaudio_result_t safeClose();
protected:
- // PlayerBase allows the system to control the stream.
- // Calling through PlayerBase->start() notifies the AudioManager of the player state.
- // The AudioManager also can start/stop a stream by calling mPlayerBase->playerStart().
- // systemStart() ==> mPlayerBase->start() mPlayerBase->playerStart() ==> requestStart()
- // \ /
- // ------ AudioManager -------
+ // PlayerBase allows the system to control the stream volume.
class MyPlayerBase : public android::PlayerBase {
public:
explicit MyPlayerBase(AudioStream *parent);
@@ -406,20 +386,19 @@
void clearParentReference() { mParent = nullptr; }
+ // Just a stub. The ability to start audio through PlayerBase is being deprecated.
android::status_t playerStart() override {
- // mParent should NOT be null. So go ahead and crash if it is.
- mResult = mParent->safeStart();
- return AAudioConvert_aaudioToAndroidStatus(mResult);
+ return android::NO_ERROR;
}
+ // Just a stub. The ability to pause audio through PlayerBase is being deprecated.
android::status_t playerPause() override {
- mResult = mParent->safePause();
- return AAudioConvert_aaudioToAndroidStatus(mResult);
+ return android::NO_ERROR;
}
+ // Just a stub. The ability to stop audio through PlayerBase is being deprecated.
android::status_t playerStop() override {
- mResult = mParent->safeStop();
- return AAudioConvert_aaudioToAndroidStatus(mResult);
+ return android::NO_ERROR;
}
android::status_t playerSetVolume() override {
@@ -548,6 +527,8 @@
private:
+ aaudio_result_t safeStop();
+
std::mutex mStreamLock;
const android::sp<MyPlayerBase> mPlayerBase;
diff --git a/media/libaaudio/src/legacy/AudioStreamLegacy.cpp b/media/libaaudio/src/legacy/AudioStreamLegacy.cpp
index a6b9f5d..2edab58 100644
--- a/media/libaaudio/src/legacy/AudioStreamLegacy.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamLegacy.cpp
@@ -78,8 +78,9 @@
void AudioStreamLegacy::processCallbackCommon(aaudio_callback_operation_t opcode, void *info) {
aaudio_data_callback_result_t callbackResult;
- // This illegal size can be used to tell AudioFlinger to stop calling us.
- // This takes advantage of AudioFlinger killing the stream.
+ // This illegal size can be used to tell AudioRecord or AudioTrack to stop calling us.
+ // This takes advantage of them killing the stream when they see a size out of range.
+ // That is an undocumented behavior.
// TODO add to API in AudioRecord and AudioTrack
const size_t SIZE_STOP_CALLBACKS = SIZE_MAX;
@@ -95,7 +96,7 @@
ALOGW("processCallbackCommon() data, stream disconnected");
audioBuffer->size = SIZE_STOP_CALLBACKS;
} else if (!mCallbackEnabled.load()) {
- ALOGW("processCallbackCommon() stopping because callback disabled");
+ ALOGW("processCallbackCommon() no data because callback disabled");
audioBuffer->size = SIZE_STOP_CALLBACKS;
} else {
if (audioBuffer->frameCount == 0) {
@@ -115,10 +116,16 @@
}
if (callbackResult == AAUDIO_CALLBACK_RESULT_CONTINUE) {
audioBuffer->size = audioBuffer->frameCount * getBytesPerDeviceFrame();
- } else { // STOP or invalid result
- ALOGW("%s() callback requested stop, fake an error", __func__);
- audioBuffer->size = SIZE_STOP_CALLBACKS;
- // Disable the callback just in case AudioFlinger keeps trying to call us.
+ } else {
+ if (callbackResult == AAUDIO_CALLBACK_RESULT_STOP) {
+ ALOGD("%s() callback returned AAUDIO_CALLBACK_RESULT_STOP", __func__);
+ } else {
+ ALOGW("%s() callback returned invalid result = %d",
+ __func__, callbackResult);
+ }
+ audioBuffer->size = 0;
+ systemStopFromCallback();
+ // Disable the callback just in case the system keeps trying to call us.
mCallbackEnabled.store(false);
}
diff --git a/media/libaaudio/src/legacy/AudioStreamRecord.cpp b/media/libaaudio/src/legacy/AudioStreamRecord.cpp
index 40e22ac..f550089 100644
--- a/media/libaaudio/src/legacy/AudioStreamRecord.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamRecord.cpp
@@ -486,6 +486,9 @@
int64_t *framePosition,
int64_t *timeNanoseconds) {
ExtendedTimestamp extendedTimestamp;
+ if (getState() != AAUDIO_STREAM_STATE_STARTED) {
+ return AAUDIO_ERROR_INVALID_STATE;
+ }
status_t status = mAudioRecord->getTimestamp(&extendedTimestamp);
if (status == WOULD_BLOCK) {
return AAUDIO_ERROR_INVALID_STATE;
diff --git a/media/libaaudio/src/legacy/AudioStreamTrack.cpp b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
index 1ac2558..c995e99 100644
--- a/media/libaaudio/src/legacy/AudioStreamTrack.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
@@ -288,7 +288,7 @@
aaudio_result_t AudioStreamTrack::requestPause() {
if (mAudioTrack.get() == nullptr) {
- ALOGE("requestPause() no AudioTrack");
+ ALOGE("%s() no AudioTrack", __func__);
return AAUDIO_ERROR_INVALID_STATE;
}
@@ -304,7 +304,7 @@
aaudio_result_t AudioStreamTrack::requestFlush() {
if (mAudioTrack.get() == nullptr) {
- ALOGE("requestFlush() no AudioTrack");
+ ALOGE("%s() no AudioTrack", __func__);
return AAUDIO_ERROR_INVALID_STATE;
}
@@ -318,7 +318,7 @@
aaudio_result_t AudioStreamTrack::requestStop() {
if (mAudioTrack.get() == nullptr) {
- ALOGE("requestStop() no AudioTrack");
+ ALOGE("%s() no AudioTrack", __func__);
return AAUDIO_ERROR_INVALID_STATE;
}
diff --git a/media/libaaudio/tests/test_timestamps.cpp b/media/libaaudio/tests/test_timestamps.cpp
index dfa7815..7b1dfd3 100644
--- a/media/libaaudio/tests/test_timestamps.cpp
+++ b/media/libaaudio/tests/test_timestamps.cpp
@@ -35,6 +35,7 @@
#define NUM_SECONDS 1
#define NUM_LOOPS 4
+#define MAX_TESTS 20
typedef struct TimestampInfo {
int64_t framesTotal;
@@ -53,6 +54,49 @@
bool forceUnderruns = false;
} TimestampCallbackData_t;
+struct TimeStampTestLog {
+ aaudio_policy_t isMmap;
+ aaudio_sharing_mode_t sharingMode;
+ aaudio_performance_mode_t performanceMode;
+ aaudio_direction_t direction;
+ aaudio_result_t result;
+};
+
+static int s_numTests = 0;
+// Use a plain old array because we reference this from the callback and do not want any
+// automatic memory allocation.
+static TimeStampTestLog s_testLogs[MAX_TESTS]{};
+
+static void logTestResult(bool isMmap,
+ aaudio_sharing_mode_t sharingMode,
+ aaudio_performance_mode_t performanceMode,
+ aaudio_direction_t direction,
+ aaudio_result_t result) {
+ if(s_numTests >= MAX_TESTS) {
+ printf("ERROR - MAX_TESTS too small = %d\n", MAX_TESTS);
+ return;
+ }
+ s_testLogs[s_numTests].isMmap = isMmap;
+ s_testLogs[s_numTests].sharingMode = sharingMode;
+ s_testLogs[s_numTests].performanceMode = performanceMode;
+ s_testLogs[s_numTests].direction = direction;
+ s_testLogs[s_numTests].result = result;
+ s_numTests++;
+}
+
+static void printTestResults() {
+ for (int i = 0; i < s_numTests; i++) {
+ TimeStampTestLog *log = &s_testLogs[i];
+ printf("%2d: mmap = %3s, sharing = %9s, perf = %11s, dir = %6s ---- %4s\n",
+ i,
+ log->isMmap ? "yes" : "no",
+ getSharingModeText(log->sharingMode),
+ getPerformanceModeText(log->performanceMode),
+ getDirectionText(log->direction),
+ log->result ? "FAIL" : "pass");
+ }
+}
+
// Callback function that fills the audio output buffer.
aaudio_data_callback_result_t timestampDataCallbackProc(
AAudioStream *stream,
@@ -115,6 +159,7 @@
int32_t originalBufferSize = 0;
int32_t requestedBufferSize = 0;
int32_t finalBufferSize = 0;
+ bool isMmap = false;
aaudio_format_t actualDataFormat = AAUDIO_FORMAT_PCM_FLOAT;
aaudio_sharing_mode_t actualSharingMode = AAUDIO_SHARING_MODE_SHARED;
aaudio_sharing_mode_t actualPerformanceMode = AAUDIO_PERFORMANCE_MODE_NONE;
@@ -124,7 +169,8 @@
memset(&sTimestampData, 0, sizeof(sTimestampData));
- printf("------------ testTimeStamps(policy = %d, sharing = %s, perf = %s, dir = %s) -----------\n",
+ printf("\n=================================================================================\n");
+ printf("--------- testTimeStamps(policy = %d, sharing = %s, perf = %s, dir = %s) --------\n",
mmapPolicy,
getSharingModeText(sharingMode),
getPerformanceModeText(performanceMode),
@@ -177,8 +223,8 @@
printf(" chans = %3d, rate = %6d format = %d\n",
actualChannelCount, actualSampleRate, actualDataFormat);
- printf(" Is MMAP used? %s\n", AAudioStream_isMMapUsed(aaudioStream)
- ? "yes" : "no");
+ isMmap = AAudioStream_isMMapUsed(aaudioStream);
+ printf(" Is MMAP used? %s\n", isMmap ? "yes" : "no");
// This is the number of frames that are read in one chunk by a DMA controller
// or a DSP or a mixer.
@@ -218,7 +264,7 @@
for (int second = 0; second < NUM_SECONDS; second++) {
// Give AAudio callback time to run in the background.
- sleep(1);
+ usleep(200 * 1000);
// Periodically print the progress so we know it hasn't died.
printf("framesWritten = %d, XRuns = %d\n",
@@ -234,18 +280,25 @@
}
printf("timestampCount = %d\n", sTimestampData.timestampCount);
- int printed = 0;
- for (int i = 0; i < sTimestampData.timestampCount; i++) {
+ int printedGood = 0;
+ int printedBad = 0;
+ for (int i = 1; i < sTimestampData.timestampCount; i++) {
TimestampInfo *timestamp = &sTimestampData.timestamps[i];
- bool posChanged = (timestamp->timestampPosition != (timestamp - 1)->timestampPosition);
- bool timeChanged = (timestamp->timestampNanos != (timestamp - 1)->timestampNanos);
- if ((printed < 20) && ((i < 10) || posChanged || timeChanged)) {
- printf(" %3d : frames %8lld, xferd %8lld", i,
- (long long) timestamp->framesTotal,
- (long long) timestamp->appPosition);
- if (timestamp->result != AAUDIO_OK) {
- printf(", result = %s\n", AAudio_convertResultToText(timestamp->result));
- } else {
+ if (timestamp->result != AAUDIO_OK) {
+ if (printedBad < 5) {
+ printf(" %3d : frames %8lld, xferd %8lld, result = %s\n",
+ i,
+ (long long) timestamp->framesTotal,
+ (long long) timestamp->appPosition,
+ AAudio_convertResultToText(timestamp->result));
+ printedBad++;
+ }
+ } else {
+ const bool posChanged = (timestamp->timestampPosition !=
+ (timestamp - 1)->timestampPosition);
+ const bool timeChanged = (timestamp->timestampNanos
+ != (timestamp - 1)->timestampNanos);
+ if ((printedGood < 20) && (posChanged || timeChanged)) {
bool negative = timestamp->timestampPosition < 0;
bool retro = (i > 0 && (timestamp->timestampPosition <
(timestamp - 1)->timestampPosition));
@@ -253,17 +306,39 @@
: (retro ? " <= RETROGRADE!" : "");
double latency = calculateLatencyMillis(timestamp->timestampPosition,
- timestamp->timestampNanos,
- timestamp->appPosition,
- timestamp->appNanoseconds,
- actualSampleRate);
- printf(", STAMP: pos = %8lld, nanos = %8lld, lat = %7.1f msec %s\n",
+ timestamp->timestampNanos,
+ timestamp->appPosition,
+ timestamp->appNanoseconds,
+ actualSampleRate);
+ printf(" %3d : frames %8lld, xferd %8lld",
+ i,
+ (long long) timestamp->framesTotal,
+ (long long) timestamp->appPosition);
+ printf(" STAMP: pos = %8lld, nanos = %8lld, lat = %7.1f msec %s\n",
(long long) timestamp->timestampPosition,
(long long) timestamp->timestampNanos,
latency,
message);
+ printedGood++;
}
- printed++;
+ }
+ }
+
+ if (printedGood == 0) {
+ printf("ERROR - AAudioStream_getTimestamp() never gave us a valid timestamp\n");
+ result = AAUDIO_ERROR_INTERNAL;
+ } else {
+ // Make sure we do not get timestamps when stopped.
+ int64_t position;
+ int64_t time;
+ aaudio_result_t tempResult = AAudioStream_getTimestamp(aaudioStream,
+ CLOCK_MONOTONIC,
+ &position, &time);
+ if (tempResult != AAUDIO_ERROR_INVALID_STATE) {
+ printf("ERROR - AAudioStream_getTimestamp() should return"
+ " INVALID_STATE when stopped! %s\n",
+ AAudio_convertResultToText(tempResult));
+ result = AAUDIO_ERROR_INTERNAL;
}
}
@@ -273,12 +348,14 @@
}
finish:
+
+ logTestResult(isMmap, sharingMode, performanceMode, direction, result);
+
if (aaudioStream != nullptr) {
AAudioStream_close(aaudioStream);
}
AAudioStreamBuilder_delete(aaudioBuilder);
printf("result = %d = %s\n", result, AAudio_convertResultToText(result));
-
return result;
}
@@ -292,7 +369,7 @@
// in a buffer if we hang or crash.
setvbuf(stdout, nullptr, _IONBF, (size_t) 0);
- printf("Test Timestamps V0.1.3\n");
+ printf("Test Timestamps V0.1.4\n");
// Legacy
aaudio_policy_t policy = AAUDIO_POLICY_NEVER;
@@ -332,5 +409,7 @@
AAUDIO_PERFORMANCE_MODE_LOW_LATENCY,
AAUDIO_DIRECTION_OUTPUT);
+ printTestResults();
+
return (result == AAUDIO_OK) ? EXIT_SUCCESS : EXIT_FAILURE;
}
diff --git a/media/libaudioclient/Android.bp b/media/libaudioclient/Android.bp
index 827df6a..1417aaf 100644
--- a/media/libaudioclient/Android.bp
+++ b/media/libaudioclient/Android.bp
@@ -50,6 +50,7 @@
"libmediametrics",
"libmediautils",
"libnblog",
+ "libprocessgroup",
"libutils",
],
export_shared_lib_headers: ["libbinder"],
diff --git a/media/libaudioclient/AudioRecord.cpp b/media/libaudioclient/AudioRecord.cpp
index 3223647..72a23e3 100644
--- a/media/libaudioclient/AudioRecord.cpp
+++ b/media/libaudioclient/AudioRecord.cpp
@@ -26,6 +26,7 @@
#include <media/AudioRecord.h>
#include <utils/Log.h>
#include <private/media/AudioTrackShared.h>
+#include <processgroup/sched_policy.h>
#include <media/IAudioFlinger.h>
#include <media/MediaAnalyticsItem.h>
#include <media/TypeConverter.h>
@@ -1398,6 +1399,17 @@
return mAudioRecord->getActiveMicrophones(activeMicrophones).transactionError();
}
+status_t AudioRecord::setMicrophoneDirection(audio_microphone_direction_t direction)
+{
+ AutoMutex lock(mLock);
+ return mAudioRecord->setMicrophoneDirection(direction).transactionError();
+}
+
+status_t AudioRecord::setMicrophoneFieldDimension(float zoom) {
+ AutoMutex lock(mLock);
+ return mAudioRecord->setMicrophoneFieldDimension(zoom).transactionError();
+}
+
// =========================================================================
void AudioRecord::DeathNotifier::binderDied(const wp<IBinder>& who __unused)
diff --git a/media/libaudioclient/AudioSystem.cpp b/media/libaudioclient/AudioSystem.cpp
index baeae8b..4c762ed 100644
--- a/media/libaudioclient/AudioSystem.cpp
+++ b/media/libaudioclient/AudioSystem.cpp
@@ -784,7 +784,8 @@
status_t AudioSystem::setDeviceConnectionState(audio_devices_t device,
audio_policy_dev_state_t state,
const char *device_address,
- const char *device_name)
+ const char *device_name,
+ audio_format_t encodedFormat)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
const char *address = "";
@@ -798,7 +799,7 @@
if (device_name != NULL) {
name = device_name;
}
- return aps->setDeviceConnectionState(device, state, address, name);
+ return aps->setDeviceConnectionState(device, state, address, name, encodedFormat);
}
audio_policy_dev_state_t AudioSystem::getDeviceConnectionState(audio_devices_t device,
@@ -812,7 +813,8 @@
status_t AudioSystem::handleDeviceConfigChange(audio_devices_t device,
const char *device_address,
- const char *device_name)
+ const char *device_name,
+ audio_format_t encodedFormat)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
const char *address = "";
@@ -826,7 +828,7 @@
if (device_name != NULL) {
name = device_name;
}
- return aps->handleDeviceConfigChange(device, address, name);
+ return aps->handleDeviceConfigChange(device, address, name, encodedFormat);
}
status_t AudioSystem::setPhoneState(audio_mode_t state)
@@ -1335,6 +1337,13 @@
return aps->isHapticPlaybackSupported();
}
+status_t AudioSystem::getHwOffloadEncodingFormatsSupportedForA2DP(
+ std::vector<audio_format_t> *formats)
+{
+ const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
+ if (aps == 0) return PERMISSION_DENIED;
+ return aps->getHwOffloadEncodingFormatsSupportedForA2DP(formats);
+}
// ---------------------------------------------------------------------------
diff --git a/media/libaudioclient/AudioTrack.cpp b/media/libaudioclient/AudioTrack.cpp
index b444d2d..e9a0e22 100644
--- a/media/libaudioclient/AudioTrack.cpp
+++ b/media/libaudioclient/AudioTrack.cpp
@@ -29,6 +29,7 @@
#include <media/AudioTrack.h>
#include <utils/Log.h>
#include <private/media/AudioTrackShared.h>
+#include <processgroup/sched_policy.h>
#include <media/IAudioFlinger.h>
#include <media/IAudioPolicyService.h>
#include <media/AudioParameter.h>
diff --git a/media/libaudioclient/IAudioPolicyService.cpp b/media/libaudioclient/IAudioPolicyService.cpp
index 272415c..8c7fac5 100644
--- a/media/libaudioclient/IAudioPolicyService.cpp
+++ b/media/libaudioclient/IAudioPolicyService.cpp
@@ -92,6 +92,7 @@
IS_HAPTIC_PLAYBACK_SUPPORTED,
SET_UID_DEVICE_AFFINITY,
REMOVE_UID_DEVICE_AFFINITY,
+ GET_OFFLOAD_FORMATS_A2DP
};
#define MAX_ITEMS_PER_LIST 1024
@@ -108,7 +109,8 @@
audio_devices_t device,
audio_policy_dev_state_t state,
const char *device_address,
- const char *device_name)
+ const char *device_name,
+ audio_format_t encodedFormat)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
@@ -116,6 +118,7 @@
data.writeInt32(static_cast <uint32_t>(state));
data.writeCString(device_address);
data.writeCString(device_name);
+ data.writeInt32(static_cast <uint32_t>(encodedFormat));
remote()->transact(SET_DEVICE_CONNECTION_STATE, data, &reply);
return static_cast <status_t> (reply.readInt32());
}
@@ -134,13 +137,15 @@
virtual status_t handleDeviceConfigChange(audio_devices_t device,
const char *device_address,
- const char *device_name)
+ const char *device_name,
+ audio_format_t encodedFormat)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
data.writeInt32(static_cast <uint32_t>(device));
data.writeCString(device_address);
data.writeCString(device_name);
+ data.writeInt32(static_cast <uint32_t>(encodedFormat));
remote()->transact(HANDLE_DEVICE_CONFIG_CHANGE, data, &reply);
return static_cast <status_t> (reply.readInt32());
}
@@ -884,7 +889,30 @@
return reply.readInt32();
}
- virtual status_t addStreamDefaultEffect(const effect_uuid_t *type,
+ virtual status_t getHwOffloadEncodingFormatsSupportedForA2DP(
+ std::vector<audio_format_t> *formats)
+ {
+ if (formats == NULL) {
+ return BAD_VALUE;
+ }
+
+ Parcel data, reply;
+ data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_OFFLOAD_FORMATS_A2DP, data, &reply);
+ if (status != NO_ERROR || (status = (status_t)reply.readInt32()) != NO_ERROR) {
+ return status;
+ }
+
+ size_t list_size = reply.readUint32();
+
+ for (size_t i = 0; i < list_size; i++) {
+ formats->push_back(static_cast<audio_format_t>(reply.readInt32()));
+ }
+ return NO_ERROR;
+ }
+
+
+ virtual status_t addStreamDefaultEffect(const effect_uuid_t *type,
const String16& opPackageName,
const effect_uuid_t *uuid,
int32_t priority,
@@ -1096,7 +1124,8 @@
case SET_ASSISTANT_UID:
case SET_A11Y_SERVICES_UIDS:
case SET_UID_DEVICE_AFFINITY:
- case REMOVE_UID_DEVICE_AFFINITY: {
+ case REMOVE_UID_DEVICE_AFFINITY:
+ case GET_OFFLOAD_FORMATS_A2DP: {
if (!isServiceUid(IPCThreadState::self()->getCallingUid())) {
ALOGW("%s: transaction %d received from PID %d unauthorized UID %d",
__func__, code, IPCThreadState::self()->getCallingPid(),
@@ -1121,6 +1150,7 @@
static_cast <audio_policy_dev_state_t>(data.readInt32());
const char *device_address = data.readCString();
const char *device_name = data.readCString();
+ audio_format_t codecFormat = static_cast <audio_format_t>(data.readInt32());
if (device_address == nullptr || device_name == nullptr) {
ALOGE("Bad Binder transaction: SET_DEVICE_CONNECTION_STATE for device %u", device);
reply->writeInt32(static_cast<int32_t> (BAD_VALUE));
@@ -1128,7 +1158,8 @@
reply->writeInt32(static_cast<uint32_t> (setDeviceConnectionState(device,
state,
device_address,
- device_name)));
+ device_name,
+ codecFormat)));
}
return NO_ERROR;
} break;
@@ -1154,13 +1185,16 @@
static_cast <audio_devices_t>(data.readInt32());
const char *device_address = data.readCString();
const char *device_name = data.readCString();
+ audio_format_t codecFormat =
+ static_cast <audio_format_t>(data.readInt32());
if (device_address == nullptr || device_name == nullptr) {
ALOGE("Bad Binder transaction: HANDLE_DEVICE_CONFIG_CHANGE for device %u", device);
reply->writeInt32(static_cast<int32_t> (BAD_VALUE));
} else {
reply->writeInt32(static_cast<uint32_t> (handleDeviceConfigChange(device,
device_address,
- device_name)));
+ device_name,
+ codecFormat)));
}
return NO_ERROR;
} break;
@@ -1745,6 +1779,21 @@
return NO_ERROR;
}
+ case GET_OFFLOAD_FORMATS_A2DP: {
+ CHECK_INTERFACE(IAudioPolicyService, data, reply);
+ std::vector<audio_format_t> encodingFormats;
+ status_t status = getHwOffloadEncodingFormatsSupportedForA2DP(&encodingFormats);
+ reply->writeInt32(status);
+ if (status != NO_ERROR) {
+ return NO_ERROR;
+ }
+ reply->writeUint32(static_cast<uint32_t>(encodingFormats.size()));
+ for (size_t i = 0; i < encodingFormats.size(); i++)
+ reply->writeInt32(static_cast<int32_t>(encodingFormats[i]));
+ return NO_ERROR;
+ }
+
+
case ADD_STREAM_DEFAULT_EFFECT: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
effect_uuid_t type;
diff --git a/media/libaudioclient/aidl/android/media/IAudioRecord.aidl b/media/libaudioclient/aidl/android/media/IAudioRecord.aidl
index 01e0a71..cf9c7f4 100644
--- a/media/libaudioclient/aidl/android/media/IAudioRecord.aidl
+++ b/media/libaudioclient/aidl/android/media/IAudioRecord.aidl
@@ -36,4 +36,12 @@
/* Get a list of current active microphones.
*/
void getActiveMicrophones(out MicrophoneInfo[] activeMicrophones);
+
+ /* Set the microphone direction (for processing).
+ */
+ void setMicrophoneDirection(int /*audio_microphone_direction_t*/ direction);
+
+ /* Set the microphone zoom (for processing).
+ */
+ void setMicrophoneFieldDimension(float zoom);
}
diff --git a/media/libaudioclient/include/media/AudioRecord.h b/media/libaudioclient/include/media/AudioRecord.h
index 35a7e05..ebee124 100644
--- a/media/libaudioclient/include/media/AudioRecord.h
+++ b/media/libaudioclient/include/media/AudioRecord.h
@@ -534,6 +534,14 @@
*/
status_t getActiveMicrophones(std::vector<media::MicrophoneInfo>* activeMicrophones);
+ /* Set the Microphone direction (for processing purposes).
+ */
+ status_t setMicrophoneDirection(audio_microphone_direction_t direction);
+
+ /* Set the Microphone zoom factor (for processing purposes).
+ */
+ status_t setMicrophoneFieldDimension(float zoom);
+
/* Get the unique port ID assigned to this AudioRecord instance by audio policy manager.
* The ID is unique across all audioserver clients and can change during the life cycle
* of a given AudioRecord instance if the connection to audioserver is restored.
diff --git a/media/libaudioclient/include/media/AudioSystem.h b/media/libaudioclient/include/media/AudioSystem.h
index 781e9df..a208602 100644
--- a/media/libaudioclient/include/media/AudioSystem.h
+++ b/media/libaudioclient/include/media/AudioSystem.h
@@ -209,12 +209,14 @@
// IAudioPolicyService interface (see AudioPolicyInterface for method descriptions)
//
static status_t setDeviceConnectionState(audio_devices_t device, audio_policy_dev_state_t state,
- const char *device_address, const char *device_name);
+ const char *device_address, const char *device_name,
+ audio_format_t encodedFormat);
static audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
const char *device_address);
static status_t handleDeviceConfigChange(audio_devices_t device,
const char *device_address,
- const char *device_name);
+ const char *device_name,
+ audio_format_t encodedFormat);
static status_t setPhoneState(audio_mode_t state);
static status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config);
static audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage);
@@ -342,6 +344,9 @@
static status_t getMicrophones(std::vector<media::MicrophoneInfo> *microphones);
+ static status_t getHwOffloadEncodingFormatsSupportedForA2DP(
+ std::vector<audio_format_t> *formats);
+
// numSurroundFormats holds the maximum number of formats and bool value allowed in the array.
// When numSurroundFormats is 0, surroundFormats and surroundFormatsEnabled will not be
// populated. The actual number of surround formats should be returned at numSurroundFormats.
diff --git a/media/libaudioclient/include/media/IAudioPolicyService.h b/media/libaudioclient/include/media/IAudioPolicyService.h
index fb4fe93..177adc2 100644
--- a/media/libaudioclient/include/media/IAudioPolicyService.h
+++ b/media/libaudioclient/include/media/IAudioPolicyService.h
@@ -44,12 +44,14 @@
virtual status_t setDeviceConnectionState(audio_devices_t device,
audio_policy_dev_state_t state,
const char *device_address,
- const char *device_name) = 0;
+ const char *device_name,
+ audio_format_t encodedFormat) = 0;
virtual audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
const char *device_address) = 0;
virtual status_t handleDeviceConfigChange(audio_devices_t device,
const char *device_address,
- const char *device_name) = 0;
+ const char *device_name,
+ audio_format_t encodedFormat) = 0;
virtual status_t setPhoneState(audio_mode_t state) = 0;
virtual status_t setForceUse(audio_policy_force_use_t usage,
audio_policy_forced_cfg_t config) = 0;
@@ -186,6 +188,8 @@
audio_format_t *surroundFormats,
bool *surroundFormatsEnabled,
bool reported) = 0;
+ virtual status_t getHwOffloadEncodingFormatsSupportedForA2DP(
+ std::vector<audio_format_t> *formats) = 0;
virtual status_t setSurroundFormatEnabled(audio_format_t audioFormat, bool enabled) = 0;
virtual status_t setAssistantUid(uid_t uid) = 0;
diff --git a/media/libaudiohal/impl/DeviceHalHidl.cpp b/media/libaudiohal/impl/DeviceHalHidl.cpp
index 7a9e843..a1e869f 100644
--- a/media/libaudiohal/impl/DeviceHalHidl.cpp
+++ b/media/libaudiohal/impl/DeviceHalHidl.cpp
@@ -268,6 +268,8 @@
audio_input_flags_t flags,
const char *address,
audio_source_t source,
+ audio_devices_t outputDevice,
+ const char *outputDeviceAddress,
sp<StreamInHalInterface> *inStream) {
if (mDevice == 0) return NO_INIT;
DeviceAddress hidlDevice;
@@ -283,6 +285,17 @@
// for now, only send the main source at 1dbfs
SinkMetadata sinkMetadata = {{{ .source = AudioSource(source), .gain = 1 }}};
#endif
+#if MAJOR_VERSION < 5
+ (void)outputDevice;
+ (void)outputDeviceAddress;
+#else
+ if (outputDevice != AUDIO_DEVICE_NONE) {
+ DeviceAddress hidlOutputDevice;
+ status = deviceAddressFromHal(outputDevice, outputDeviceAddress, &hidlOutputDevice);
+ if (status != OK) return status;
+ sinkMetadata.tracks[0].destination.device(std::move(hidlOutputDevice));
+ }
+#endif
Return<void> ret = mDevice->openInputStream(
handle,
hidlDevice,
diff --git a/media/libaudiohal/impl/DeviceHalHidl.h b/media/libaudiohal/impl/DeviceHalHidl.h
index 291c88f..f7d465f 100644
--- a/media/libaudiohal/impl/DeviceHalHidl.h
+++ b/media/libaudiohal/impl/DeviceHalHidl.h
@@ -86,6 +86,8 @@
audio_input_flags_t flags,
const char *address,
audio_source_t source,
+ audio_devices_t outputDevice,
+ const char *outputDeviceAddress,
sp<StreamInHalInterface> *inStream);
// Returns whether createAudioPatch and releaseAudioPatch operations are supported.
diff --git a/media/libaudiohal/impl/DeviceHalLocal.cpp b/media/libaudiohal/impl/DeviceHalLocal.cpp
index dffe9da..ee68252 100644
--- a/media/libaudiohal/impl/DeviceHalLocal.cpp
+++ b/media/libaudiohal/impl/DeviceHalLocal.cpp
@@ -131,6 +131,8 @@
audio_input_flags_t flags,
const char *address,
audio_source_t source,
+ audio_devices_t /*outputDevice*/,
+ const char */*outputDeviceAddress*/,
sp<StreamInHalInterface> *inStream) {
audio_stream_in_t *halStream;
ALOGV("open_input_stream handle: %d devices: %x flags: %#x "
diff --git a/media/libaudiohal/impl/DeviceHalLocal.h b/media/libaudiohal/impl/DeviceHalLocal.h
index 18bd879..36db72e 100644
--- a/media/libaudiohal/impl/DeviceHalLocal.h
+++ b/media/libaudiohal/impl/DeviceHalLocal.h
@@ -79,6 +79,8 @@
audio_input_flags_t flags,
const char *address,
audio_source_t source,
+ audio_devices_t outputDevice,
+ const char *outputDeviceAddress,
sp<StreamInHalInterface> *inStream);
// Returns whether createAudioPatch and releaseAudioPatch operations are supported.
diff --git a/media/libaudiohal/impl/StreamHalHidl.cpp b/media/libaudiohal/impl/StreamHalHidl.cpp
index c12b362..2e35be6 100644
--- a/media/libaudiohal/impl/StreamHalHidl.cpp
+++ b/media/libaudiohal/impl/StreamHalHidl.cpp
@@ -854,5 +854,29 @@
}
#endif
+#if MAJOR_VERSION < 5
+status_t StreamInHalHidl::setMicrophoneDirection(audio_microphone_direction_t direction __unused) {
+ if (mStream == 0) return NO_INIT;
+ return INVALID_OPERATION;
+}
+
+status_t StreamInHalHidl::setMicrophoneFieldDimension(float zoom __unused) {
+ if (mStream == 0) return NO_INIT;
+ return INVALID_OPERATION;
+}
+#else
+status_t StreamInHalHidl::setMicrophoneDirection(audio_microphone_direction_t direction) {
+ if (!mStream) return NO_INIT;
+ return processReturn("setMicrophoneDirection",
+ mStream->setMicrophoneDirection(static_cast<MicrophoneDirection>(direction)));
+}
+
+status_t StreamInHalHidl::setMicrophoneFieldDimension(float zoom) {
+ if (!mStream) return NO_INIT;
+ return processReturn("setMicrophoneFieldDimension",
+ mStream->setMicrophoneFieldDimension(zoom));
+}
+#endif
+
} // namespace CPP_VERSION
} // namespace android
diff --git a/media/libaudiohal/impl/StreamHalHidl.h b/media/libaudiohal/impl/StreamHalHidl.h
index f7b507e..9ac1067 100644
--- a/media/libaudiohal/impl/StreamHalHidl.h
+++ b/media/libaudiohal/impl/StreamHalHidl.h
@@ -220,6 +220,12 @@
// Get active microphones
virtual status_t getActiveMicrophones(std::vector<media::MicrophoneInfo> *microphones);
+ // Set microphone direction (for processing)
+ virtual status_t setMicrophoneDirection(audio_microphone_direction_t direction) override;
+
+ // Set microphone zoom (for processing)
+ virtual status_t setMicrophoneFieldDimension(float zoom) override;
+
// Called when the metadata of the stream's sink has been changed.
status_t updateSinkMetadata(const SinkMetadata& sinkMetadata) override;
diff --git a/media/libaudiohal/impl/StreamHalLocal.cpp b/media/libaudiohal/impl/StreamHalLocal.cpp
index 26d30d4..fcb809b 100644
--- a/media/libaudiohal/impl/StreamHalLocal.cpp
+++ b/media/libaudiohal/impl/StreamHalLocal.cpp
@@ -368,5 +368,26 @@
}
#endif
+#if MAJOR_VERSION < 5
+status_t StreamInHalLocal::setMicrophoneDirection(audio_microphone_direction_t direction __unused) {
+ return INVALID_OPERATION;
+}
+
+status_t StreamInHalLocal::setMicrophoneFieldDimension(float zoom __unused) {
+ return INVALID_OPERATION;
+}
+#else
+status_t StreamInHalLocal::setMicrophoneDirection(audio_microphone_direction_t direction) {
+ if (mStream->set_microphone_direction == NULL) return INVALID_OPERATION;
+ return mStream->set_microphone_direction(mStream, direction);
+}
+
+status_t StreamInHalLocal::setMicrophoneFieldDimension(float zoom) {
+ if (mStream->set_microphone_field_dimension == NULL) return INVALID_OPERATION;
+ return mStream->set_microphone_field_dimension(mStream, zoom);
+
+}
+#endif
+
} // namespace CPP_VERSION
} // namespace android
diff --git a/media/libaudiohal/impl/StreamHalLocal.h b/media/libaudiohal/impl/StreamHalLocal.h
index 4fd1960..3d6c50e 100644
--- a/media/libaudiohal/impl/StreamHalLocal.h
+++ b/media/libaudiohal/impl/StreamHalLocal.h
@@ -204,6 +204,12 @@
// Get active microphones
virtual status_t getActiveMicrophones(std::vector<media::MicrophoneInfo> *microphones);
+ // Sets microphone direction (for processing)
+ virtual status_t setMicrophoneDirection(audio_microphone_direction_t direction);
+
+ // Sets microphone zoom (for processing)
+ virtual status_t setMicrophoneFieldDimension(float zoom);
+
// Called when the metadata of the stream's sink has been changed.
status_t updateSinkMetadata(const SinkMetadata& sinkMetadata) override;
diff --git a/media/libaudiohal/include/media/audiohal/DeviceHalInterface.h b/media/libaudiohal/include/media/audiohal/DeviceHalInterface.h
index 7de8eb3..e565237 100644
--- a/media/libaudiohal/include/media/audiohal/DeviceHalInterface.h
+++ b/media/libaudiohal/include/media/audiohal/DeviceHalInterface.h
@@ -84,6 +84,8 @@
audio_input_flags_t flags,
const char *address,
audio_source_t source,
+ audio_devices_t outputDevice,
+ const char *outputDeviceAddress,
sp<StreamInHalInterface> *inStream) = 0;
// Returns whether createAudioPatch and releaseAudioPatch operations are supported.
diff --git a/media/libaudiohal/include/media/audiohal/StreamHalInterface.h b/media/libaudiohal/include/media/audiohal/StreamHalInterface.h
index bd71dc0..ed8282f 100644
--- a/media/libaudiohal/include/media/audiohal/StreamHalInterface.h
+++ b/media/libaudiohal/include/media/audiohal/StreamHalInterface.h
@@ -179,6 +179,12 @@
// Get active microphones
virtual status_t getActiveMicrophones(std::vector<media::MicrophoneInfo> *microphones) = 0;
+ // Set direction for capture processing
+ virtual status_t setMicrophoneDirection(audio_microphone_direction_t) = 0;
+
+ // Set zoom factor for capture stream
+ virtual status_t setMicrophoneFieldDimension(float zoom) = 0;
+
struct SinkMetadata {
std::vector<record_track_metadata_t> tracks;
};
diff --git a/media/libeffects/lvm/lib/Bundle/src/LVM_Init.c b/media/libeffects/lvm/lib/Bundle/src/LVM_Init.c
index 0669a81..c57498e 100644
--- a/media/libeffects/lvm/lib/Bundle/src/LVM_Init.c
+++ b/media/libeffects/lvm/lib/Bundle/src/LVM_Init.c
@@ -61,6 +61,72 @@
/* */
/****************************************************************************************/
+/*
+ * 4 Types of Memory Regions of LVM
+ * TODO: Allocate on the fly.
+ * i) LVM_MEMREGION_PERSISTENT_SLOW_DATA - For Instance Handles
+ * ii) LVM_MEMREGION_PERSISTENT_FAST_DATA - Persistent Buffers
+ * iii) LVM_MEMREGION_PERSISTENT_FAST_COEF - For Holding Structure values
+ * iv) LVM_MEMREGION_TEMPORARY_FAST - For Holding Structure values
+ *
+ * LVM_MEMREGION_PERSISTENT_SLOW_DATA:
+ * Total Memory size:
+ * sizeof(LVM_Instance_t) + \
+ * sizeof(LVM_Buffer_t) + \
+ * sizeof(LVPSA_InstancePr_t) + \
+ * sizeof(LVM_Buffer_t) - needed if buffer mode is LVM_MANAGED_BUFFER
+ *
+ * LVM_MEMREGION_PERSISTENT_FAST_DATA:
+ * Total Memory size:
+ * sizeof(LVM_TE_Data_t) + \
+ * 2 * pInstParams->EQNB_NumBands * sizeof(LVM_EQNB_BandDef_t) + \
+ * sizeof(LVCS_Data_t) + \
+ * sizeof(LVDBE_Data_FLOAT_t) + \
+ * sizeof(Biquad_2I_Order2_FLOAT_Taps_t) + \
+ * sizeof(Biquad_2I_Order2_FLOAT_Taps_t) + \
+ * pInstParams->EQNB_NumBands * sizeof(Biquad_2I_Order2_FLOAT_Taps_t) + \
+ * pInstParams->EQNB_NumBands * sizeof(LVEQNB_BandDef_t) + \
+ * pInstParams->EQNB_NumBands * sizeof(LVEQNB_BiquadType_en) + \
+ * 2 * LVM_HEADROOM_MAX_NBANDS * sizeof(LVM_HeadroomBandDef_t) + \
+ * PSA_InitParams.nBands * sizeof(Biquad_1I_Order2_Taps_t) + \
+ * PSA_InitParams.nBands * sizeof(QPD_Taps_t)
+ *
+ * LVM_MEMREGION_PERSISTENT_FAST_COEF:
+ * Total Memory size:
+ * sizeof(LVM_TE_Coefs_t) + \
+ * sizeof(LVCS_Coefficient_t) + \
+ * sizeof(LVDBE_Coef_FLOAT_t) + \
+ * sizeof(Biquad_FLOAT_Instance_t) + \
+ * sizeof(Biquad_FLOAT_Instance_t) + \
+ * pInstParams->EQNB_NumBands * sizeof(Biquad_FLOAT_Instance_t) + \
+ * PSA_InitParams.nBands * sizeof(Biquad_Instance_t) + \
+ * PSA_InitParams.nBands * sizeof(QPD_State_t)
+ *
+ * LVM_MEMREGION_TEMPORARY_FAST (Scratch):
+ * Total Memory Size:
+ * BundleScratchSize + \
+ * MAX_INTERNAL_BLOCKSIZE * sizeof(LVM_FLOAT) + \
+ * MaxScratchOf (CS, EQNB, DBE, PSA)
+ *
+ * a)BundleScratchSize:
+ * 3 * LVM_MAX_CHANNELS \
+ * * (MIN_INTERNAL_BLOCKSIZE + InternalBlockSize) * sizeof(LVM_FLOAT)
+ * This Memory is allocated only when Buffer mode is LVM_MANAGED_BUFFER.
+ * b)MaxScratchOf (CS, EQNB, DBE, PSA)
+ * This Memory is needed for scratch usage for CS, EQNB, DBE, PSA.
+ * CS = (LVCS_SCRATCHBUFFERS * sizeof(LVM_FLOAT)
+ * * pCapabilities->MaxBlockSize)
+ * EQNB = (LVEQNB_SCRATCHBUFFERS * sizeof(LVM_FLOAT)
+ * * pCapabilities->MaxBlockSize)
+ * DBE = (LVDBE_SCRATCHBUFFERS_INPLACE*sizeof(LVM_FLOAT)
+ * * pCapabilities->MaxBlockSize)
+ * PSA = (2 * pInitParams->MaxInputBlockSize * sizeof(LVM_FLOAT))
+ * one MaxInputBlockSize for input and another for filter output
+ * c)MAX_INTERNAL_BLOCKSIZE
+ * This Memory is needed for PSAInput - Temp memory to store output
+ * from McToMono block and given as input to PSA block
+ */
+
LVM_ReturnStatus_en LVM_GetMemoryTable(LVM_Handle_t hInstance,
LVM_MemTab_t *pMemoryTable,
LVM_InstParams_t *pInstParams)
@@ -168,7 +234,13 @@
AlgScratchSize = 0;
if (pInstParams->BufferMode == LVM_MANAGED_BUFFERS)
{
+#ifdef BUILD_FLOAT
+ BundleScratchSize = 3 * LVM_MAX_CHANNELS \
+ * (MIN_INTERNAL_BLOCKSIZE + InternalBlockSize) \
+ * sizeof(LVM_FLOAT);
+#else
BundleScratchSize = 6 * (MIN_INTERNAL_BLOCKSIZE + InternalBlockSize) * sizeof(LVM_INT16);
+#endif
InstAlloc_AddMember(&AllocMem[LVM_MEMREGION_TEMPORARY_FAST], /* Scratch buffer */
BundleScratchSize);
InstAlloc_AddMember(&AllocMem[LVM_MEMREGION_PERSISTENT_SLOW_DATA],
@@ -369,8 +441,13 @@
PSA_MemTab.Region[LVM_PERSISTENT_FAST_COEF].Size);
/* Fast Temporary */
+#ifdef BUILD_FLOAT
+ InstAlloc_AddMember(&AllocMem[LVM_TEMPORARY_FAST],
+ MAX_INTERNAL_BLOCKSIZE * sizeof(LVM_FLOAT));
+#else
InstAlloc_AddMember(&AllocMem[LVM_TEMPORARY_FAST],
MAX_INTERNAL_BLOCKSIZE * sizeof(LVM_INT16));
+#endif
if (PSA_MemTab.Region[LVM_TEMPORARY_FAST].Size > AlgScratchSize)
{
@@ -559,13 +636,20 @@
*/
pInstance->pBufferManagement = InstAlloc_AddMember(&AllocMem[LVM_MEMREGION_PERSISTENT_SLOW_DATA],
sizeof(LVM_Buffer_t));
+#ifdef BUILD_FLOAT
+ BundleScratchSize = (LVM_INT32)
+ (3 * LVM_MAX_CHANNELS \
+ * (MIN_INTERNAL_BLOCKSIZE + InternalBlockSize) \
+ * sizeof(LVM_FLOAT));
+#else
BundleScratchSize = (LVM_INT32)(6 * (MIN_INTERNAL_BLOCKSIZE + InternalBlockSize) * sizeof(LVM_INT16));
+#endif
pInstance->pBufferManagement->pScratch = InstAlloc_AddMember(&AllocMem[LVM_MEMREGION_TEMPORARY_FAST], /* Scratch 1 buffer */
(LVM_UINT32)BundleScratchSize);
#ifdef BUILD_FLOAT
LoadConst_Float(0, /* Clear the input delay buffer */
(LVM_FLOAT *)&pInstance->pBufferManagement->InDelayBuffer,
- (LVM_INT16)(2 * MIN_INTERNAL_BLOCKSIZE));
+ (LVM_INT16)(LVM_MAX_CHANNELS * MIN_INTERNAL_BLOCKSIZE));
#else
LoadConst_16(0, /* Clear the input delay buffer */
(LVM_INT16 *)&pInstance->pBufferManagement->InDelayBuffer,
diff --git a/media/libeffects/lvm/lib/Common/src/LVC_Mixer_SetTimeConstant.c b/media/libeffects/lvm/lib/Common/src/LVC_Mixer_SetTimeConstant.c
index 48f5d54..9d3ee88 100644
--- a/media/libeffects/lvm/lib/Common/src/LVC_Mixer_SetTimeConstant.c
+++ b/media/libeffects/lvm/lib/Common/src/LVC_Mixer_SetTimeConstant.c
@@ -51,7 +51,7 @@
LVM_INT16 NumChannels)
{
#ifdef HIGHER_FS
- LVM_FLOAT DeltaTable[11] = {0.500000f,/*8000*/
+ LVM_FLOAT DeltaTable[13] = {0.500000f,/*8000*/
0.362812f,/*11025*/
0.333333f,/*12000*/
0.250000f,/*16000*/
@@ -60,7 +60,9 @@
0.125000f,/*32000*/
0.090703f,/*44100*/
0.083333f,/*48000*/
+ 0.045352f,/*88200*/
0.041667f,/*96000*/
+ 0.022676f,/*176400*/
0.020833f};/*192000*/
#else
LVM_FLOAT DeltaTable[9] = {0.500000f,/*8000*/
diff --git a/media/libeffects/lvm/lib/Common/src/LVC_Mixer_VarSlope_SetTimeConstant.c b/media/libeffects/lvm/lib/Common/src/LVC_Mixer_VarSlope_SetTimeConstant.c
index 9dc7d21..0e0acf1 100644
--- a/media/libeffects/lvm/lib/Common/src/LVC_Mixer_VarSlope_SetTimeConstant.c
+++ b/media/libeffects/lvm/lib/Common/src/LVC_Mixer_VarSlope_SetTimeConstant.c
@@ -52,7 +52,7 @@
LVM_INT16 NumChannels)
{
#ifdef HIGHER_FS
- LVM_FLOAT DeltaTable[11] = {0.500000f,/*8000*/
+ LVM_FLOAT DeltaTable[13] = {0.500000f,/*8000*/
0.362812f,/*11025*/
0.333333f,/*12000*/
0.250000f,/*16000*/
@@ -61,7 +61,9 @@
0.125000f,/*32000*/
0.090703f,/*44100*/
0.083333f,/*48000*/
+ 0.045352f,/*88200*/
0.041666f,/*96000*/
+ 0.022676f,/*176400*/
0.020833f};/*192000*/
#else
LVM_FLOAT DeltaTable[9] = {0.500000f,/*8000*/
diff --git a/media/libeffects/lvm/lib/Common/src/LVM_GetOmega.c b/media/libeffects/lvm/lib/Common/src/LVM_GetOmega.c
index 7846ca0..6307e68 100644
--- a/media/libeffects/lvm/lib/Common/src/LVM_GetOmega.c
+++ b/media/libeffects/lvm/lib/Common/src/LVM_GetOmega.c
@@ -53,7 +53,9 @@
#define LVVDL_2PiBy_48000_f 0.000130900f
#ifdef HIGHER_FS
+#define LVVDL_2PiBy_88200_f 0.000071238f
#define LVVDL_2PiBy_96000_f 0.000065450f
+#define LVVDL_2PiBy_176400_f 0.000035619f
#define LVVDL_2PiBy_192000_f 0.000032725f
#endif
const LVM_FLOAT LVVDL_2PiOnFsTable[] = {LVVDL_2PiBy_8000_f,
@@ -66,7 +68,9 @@
LVVDL_2PiBy_44100_f,
LVVDL_2PiBy_48000_f
#ifdef HIGHER_FS
+ ,LVVDL_2PiBy_88200_f
,LVVDL_2PiBy_96000_f
+ ,LVVDL_2PiBy_176400_f
,LVVDL_2PiBy_192000_f
#endif
};
diff --git a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Headphone_Coeffs.h b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Headphone_Coeffs.h
index e45d81f..ba05577 100644
--- a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Headphone_Coeffs.h
+++ b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Headphone_Coeffs.h
@@ -239,13 +239,12 @@
#define LVCS_STEREODELAY_CS_24KHZ 279 /* Sample rate 24kS/s */
#define LVCS_STEREODELAY_CS_32KHZ 372 /* Sample rate 32kS/s */
#define LVCS_STEREODELAY_CS_44KHZ 512 /* Sample rate 44kS/s */
-// TODO: this should linearly scale by frequency but is limited to 512 frames until
-// we ensure enough buffer size has been allocated.
-#define LVCS_STEREODELAY_CS_48KHZ 512 /* Sample rate 48kS/s */
-#define LVCS_STEREODELAY_CS_88KHZ 512 /* Sample rate 88.2kS/s */
-#define LVCS_STEREODELAY_CS_96KHZ 512 /* Sample rate 96kS/s */
-#define LVCS_STEREODELAY_CS_176KHZ 512 /* Sample rate 176.4kS/s */
-#define LVCS_STEREODELAY_CS_192KHZ 512 /* Sample rate 196kS/s */
+#define LVCS_STEREODELAY_CS_48KHZ 557 /* Sample rate 48kS/s */
+#define LVCS_STEREODELAY_CS_88KHZ 1024 /* Sample rate 88.2kS/s */
+#define LVCS_STEREODELAY_CS_96KHZ 1115 /* Sample rate 96kS/s */
+#define LVCS_STEREODELAY_CS_176KHZ 2048 /* Sample rate 176.4kS/s */
+#define LVCS_STEREODELAY_CS_192KHZ 2229 /* Sample rate 196kS/s */
+#define LVCS_STEREODELAY_CS_MAX_VAL LVCS_STEREODELAY_CS_192KHZ
/* Reverb coefficients for 8000 Hz sample rate, scaled with 1.038030 */
#define CS_REVERB_8000_A0 0.667271
diff --git a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_ReverbGenerator.h b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_ReverbGenerator.h
index 69892b6..f94d4e4 100644
--- a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_ReverbGenerator.h
+++ b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_ReverbGenerator.h
@@ -65,7 +65,7 @@
/* Filter */
void (*pBiquadCallBack) (Biquad_Instance_t*, LVM_INT16*, LVM_INT16*, LVM_INT16);
#else
- LVM_FLOAT StereoSamples[2 * LVCS_STEREODELAY_CS_48KHZ];
+ LVM_FLOAT StereoSamples[2 * LVCS_STEREODELAY_CS_MAX_VAL];
/* Reverb Level */
LVM_FLOAT ReverbLevel;
/* Filter */
diff --git a/media/libeffects/lvm/tests/Android.bp b/media/libeffects/lvm/tests/Android.bp
index 8ee807c..003ce9e 100644
--- a/media/libeffects/lvm/tests/Android.bp
+++ b/media/libeffects/lvm/tests/Android.bp
@@ -44,3 +44,16 @@
"-Wextra",
],
}
+
+cc_test {
+ name: "snr",
+ host_supported: false,
+
+ srcs: ["snr.cpp"],
+
+ cflags: [
+ "-Wall",
+ "-Werror",
+ "-Wextra",
+ ],
+}
diff --git a/media/libeffects/lvm/tests/build_and_run_all_unit_tests.sh b/media/libeffects/lvm/tests/build_and_run_all_unit_tests.sh
index 861ee64..41a4f04 100755
--- a/media/libeffects/lvm/tests/build_and_run_all_unit_tests.sh
+++ b/media/libeffects/lvm/tests/build_and_run_all_unit_tests.sh
@@ -25,16 +25,17 @@
adb shell mkdir -p $testdir
adb push $ANDROID_BUILD_TOP/cts/tests/tests/media/res/raw/sinesweepraw.raw $testdir
adb push $OUT/testcases/lvmtest/arm64/lvmtest $testdir
+adb push $OUT/testcases/snr/arm64/snr $testdir
flags_arr=(
"-csE"
"-eqE"
"-tE"
"-csE -tE -eqE"
- "-bE"
+ "-bE -M"
"-csE -tE"
"-csE -eqE" "-tE -eqE"
- "-csE -tE -bE -eqE"
+ "-csE -tE -bE -M -eqE"
)
fs_arr=(
@@ -79,6 +80,10 @@
then
adb shell cmp $testdir/sinesweep_2_$((fs)).raw \
$testdir/sinesweep_$((ch))_$((fs)).raw
+ elif [[ $flags == *"-bE"* ]] && [ "$ch" -gt 2 ]
+ then
+ adb shell $testdir/snr $testdir/sinesweep_2_$((fs)).raw \
+ $testdir/sinesweep_$((ch))_$((fs)).raw -thr:90.308998
fi
done
diff --git a/media/libeffects/lvm/tests/snr.cpp b/media/libeffects/lvm/tests/snr.cpp
new file mode 100644
index 0000000..88110c0
--- /dev/null
+++ b/media/libeffects/lvm/tests/snr.cpp
@@ -0,0 +1,103 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+#include <assert.h>
+#include <inttypes.h>
+#include <math.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <vector>
+
+template <typename T, typename A = float>
+std::pair<A, A> getSignalNoise(FILE *finp, FILE *fref) {
+ constexpr size_t framesize = 256;
+ std::vector<T> in(framesize);
+ std::vector<T> ref(framesize);
+ A signal{};
+ A noise{};
+
+ for (;;) {
+ size_t read_samples_in = fread(&in[0], sizeof(T), framesize, finp);
+ const size_t read_samples_ref = fread(&ref[0], sizeof(T), framesize, fref);
+ if (read_samples_in != read_samples_ref) {
+ printf("file sizes do not match (last %zu %zu)", read_samples_in, read_samples_ref);
+ read_samples_in = std::min(read_samples_in, read_samples_ref);
+ }
+ if (read_samples_in == 0) {
+ return { signal, noise };
+ }
+ for (size_t i = 0; i < read_samples_in; ++i) {
+ const A value(ref[i]);
+ const A diff(A(in[i]) - value);
+ signal += value * value;
+ noise += diff * diff;
+ }
+ }
+}
+
+void printUsage() {
+ printf("\nUsage: ");
+ printf("\n snr <ref_file> <test_file> [options]\n");
+ printf("\nwhere, \n <ref_file> is the reference file name");
+ printf("\n on which will be taken as pure signal");
+ printf("\n <test_file> is test file for snr calculation");
+ printf("\n and options are mentioned below");
+ printf("\n");
+ printf("\n -pcm_format:<pcm format of input files>");
+ printf("\n 0 - 16 bit pcm");
+ printf("\n 1 - 32 bit float");
+ printf("\n default 0");
+ printf("\n -thr:<threshold value>");
+ printf("\n default - negative infinity\n\n");
+}
+
+int main(int argc, const char *argv[]) {
+ if (argc < 3) {
+ printUsage();
+ return -1;
+ }
+ int pcm_format = 0;
+ float thr = - std::numeric_limits<float>::infinity();
+ FILE *fref = fopen(argv[1], "rb");
+ FILE *finp = fopen(argv[2], "rb");
+ for (int i = 3; i < argc; i++) {
+ if (!strncmp(argv[i], "-pcm_format:", 12)) {
+ pcm_format = atoi(argv[i] + 12);
+ } else if (!strncmp(argv[i], "-thr:", 5)) {
+ thr = atof(argv[i] + 5);
+ }
+ }
+ if (finp == nullptr || fref == nullptr) {
+ printf("\nError: missing input/reference files\n");
+ return -1;
+ }
+ auto sn = pcm_format == 0
+ ? getSignalNoise<short>(finp, fref)
+ : getSignalNoise<float>(finp, fref);
+ if (sn.first > 0.f && sn.second > 0.f) {
+ float snr = 10.f * log(sn.first / sn.second);
+ // compare the measured snr value with threshold
+ if (snr < thr) {
+ printf("%.6f less than threshold %.6f\n", snr, thr);
+ } else {
+ printf("%.6f\n", snr);
+ }
+ }
+ fclose(finp);
+ fclose(fref);
+
+ return 0;
+}
diff --git a/media/libmedia/Android.bp b/media/libmedia/Android.bp
index 3efb5de..68dae56 100644
--- a/media/libmedia/Android.bp
+++ b/media/libmedia/Android.bp
@@ -213,6 +213,7 @@
"android.hidl.token@1.0-utils",
"liblog",
"libcutils",
+ "libprocessgroup",
"libutils",
"libbinder",
"libsonivox",
diff --git a/media/libmedia/IMediaExtractor.cpp b/media/libmedia/IMediaExtractor.cpp
index e9a6230..fb6d3a2 100644
--- a/media/libmedia/IMediaExtractor.cpp
+++ b/media/libmedia/IMediaExtractor.cpp
@@ -19,6 +19,7 @@
#include <utils/Log.h>
#include <stdint.h>
+#include <time.h>
#include <sys/types.h>
#include <binder/IPCThreadState.h>
@@ -219,10 +220,16 @@
Vector<wp<IMediaSource>> tracks;
Vector<String8> trackDescriptions;
String8 toString() const;
+ time_t when;
} ExtractorInstance;
String8 ExtractorInstance::toString() const {
- String8 str = name;
+ String8 str;
+ char timeString[32];
+ strftime(timeString, sizeof(timeString), "%m-%d %T", localtime(&when));
+ str.append(timeString);
+ str.append(": ");
+ str.append(name);
str.append(" for mime ");
str.append(mime);
str.append(", source ");
@@ -287,6 +294,7 @@
ex.sourceDescription = source->toString();
ex.owner = IPCThreadState::self()->getCallingPid();
ex.extractor = extractor;
+ ex.when = time(NULL);
{
Mutex::Autolock lock(sExtractorsLock);
diff --git a/media/libmedia/IMediaMetadataRetriever.cpp b/media/libmedia/IMediaMetadataRetriever.cpp
index 590ba1a..f9fa86e 100644
--- a/media/libmedia/IMediaMetadataRetriever.cpp
+++ b/media/libmedia/IMediaMetadataRetriever.cpp
@@ -23,6 +23,7 @@
#include <media/IDataSource.h>
#include <media/IMediaHTTPService.h>
#include <media/IMediaMetadataRetriever.h>
+#include <processgroup/sched_policy.h>
#include <utils/String8.h>
#include <utils/KeyedVector.h>
diff --git a/media/libmedia/MediaCodecInfo.cpp b/media/libmedia/MediaCodecInfo.cpp
index 5308e1c..86ad997 100644
--- a/media/libmedia/MediaCodecInfo.cpp
+++ b/media/libmedia/MediaCodecInfo.cpp
@@ -28,6 +28,15 @@
namespace android {
+/** This redundant redeclaration is needed for C++ pre 14 */
+constexpr char MediaCodecInfo::Capabilities::FEATURE_ADAPTIVE_PLAYBACK[];
+constexpr char MediaCodecInfo::Capabilities::FEATURE_DYNAMIC_TIMESTAMP[];
+constexpr char MediaCodecInfo::Capabilities::FEATURE_FRAME_PARSING[];
+constexpr char MediaCodecInfo::Capabilities::FEATURE_INTRA_REFRESH[];
+constexpr char MediaCodecInfo::Capabilities::FEATURE_MULTIPLE_FRAMES[];
+constexpr char MediaCodecInfo::Capabilities::FEATURE_SECURE_PLAYBACK[];
+constexpr char MediaCodecInfo::Capabilities::FEATURE_TUNNELED_PLAYBACK[];
+
void MediaCodecInfo::Capabilities::getSupportedProfileLevels(
Vector<ProfileLevel> *profileLevels) const {
profileLevels->clear();
@@ -40,16 +49,11 @@
colorFormats->appendVector(mColorFormats);
}
-uint32_t MediaCodecInfo::Capabilities::getFlags() const {
- return mFlags;
-}
-
const sp<AMessage> MediaCodecInfo::Capabilities::getDetails() const {
return mDetails;
}
-MediaCodecInfo::Capabilities::Capabilities()
- : mFlags(0) {
+MediaCodecInfo::Capabilities::Capabilities() {
mDetails = new AMessage;
}
@@ -73,12 +77,10 @@
caps->mColorFormats.push_back(color);
}
}
- uint32_t flags = static_cast<uint32_t>(parcel.readInt32());
sp<AMessage> details = AMessage::FromParcel(parcel);
if (details == NULL)
return NULL;
if (caps != NULL) {
- caps->mFlags = flags;
caps->mDetails = details;
}
return caps;
@@ -96,7 +98,6 @@
for (size_t i = 0; i < mColorFormats.size(); i++) {
parcel->writeInt32(mColorFormats.itemAt(i));
}
- parcel->writeInt32(mFlags);
mDetails->writeToParcel(parcel);
return OK;
}
@@ -111,6 +112,14 @@
mCap->mDetails->setInt32(key, value);
}
+void MediaCodecInfo::CapabilitiesWriter::removeDetail(const char* key) {
+ if (mCap->mDetails->removeEntryAt(mCap->mDetails->findEntryByName(key)) == OK) {
+ ALOGD("successfully removed detail %s", key);
+ } else {
+ ALOGD("detail %s wasn't present to remove", key);
+ }
+}
+
void MediaCodecInfo::CapabilitiesWriter::addProfileLevel(
uint32_t profile, uint32_t level) {
ProfileLevel profileLevel;
@@ -129,32 +138,32 @@
}
}
-void MediaCodecInfo::CapabilitiesWriter::addFlags(uint32_t flags) {
- mCap->mFlags |= flags;
-}
-
MediaCodecInfo::CapabilitiesWriter::CapabilitiesWriter(
MediaCodecInfo::Capabilities* cap) : mCap(cap) {
}
-bool MediaCodecInfo::isEncoder() const {
- return mIsEncoder;
+MediaCodecInfo::Attributes MediaCodecInfo::getAttributes() const {
+ return mAttributes;
}
-uint32_t MediaCodecInfo::rank() const {
+uint32_t MediaCodecInfo::getRank() const {
return mRank;
}
-void MediaCodecInfo::getSupportedMimes(Vector<AString> *mimes) const {
- mimes->clear();
+void MediaCodecInfo::getAliases(Vector<AString> *aliases) const {
+ *aliases = mAliases;
+}
+
+void MediaCodecInfo::getSupportedMediaTypes(Vector<AString> *mediaTypes) const {
+ mediaTypes->clear();
for (size_t ix = 0; ix < mCaps.size(); ix++) {
- mimes->push_back(mCaps.keyAt(ix));
+ mediaTypes->push_back(mCaps.keyAt(ix));
}
}
const sp<MediaCodecInfo::Capabilities>
-MediaCodecInfo::getCapabilitiesFor(const char *mime) const {
- ssize_t ix = getCapabilityIndex(mime);
+MediaCodecInfo::getCapabilitiesFor(const char *mediaType) const {
+ ssize_t ix = getCapabilityIndex(mediaType);
if (ix >= 0) {
return mCaps.valueAt(ix);
}
@@ -173,21 +182,26 @@
sp<MediaCodecInfo> MediaCodecInfo::FromParcel(const Parcel &parcel) {
AString name = AString::FromParcel(parcel);
AString owner = AString::FromParcel(parcel);
- bool isEncoder = static_cast<bool>(parcel.readInt32());
+ Attributes attributes = static_cast<Attributes>(parcel.readInt32());
uint32_t rank = parcel.readUint32();
sp<MediaCodecInfo> info = new MediaCodecInfo;
info->mName = name;
info->mOwner = owner;
- info->mIsEncoder = isEncoder;
+ info->mAttributes = attributes;
info->mRank = rank;
+ size_t numAliases = static_cast<size_t>(parcel.readInt32());
+ for (size_t i = 0; i < numAliases; i++) {
+ AString alias = AString::FromParcel(parcel);
+ info->mAliases.add(alias);
+ }
size_t size = static_cast<size_t>(parcel.readInt32());
for (size_t i = 0; i < size; i++) {
- AString mime = AString::FromParcel(parcel);
+ AString mediaType = AString::FromParcel(parcel);
sp<Capabilities> caps = Capabilities::FromParcel(parcel);
if (caps == NULL)
return NULL;
if (info != NULL) {
- info->mCaps.add(mime, caps);
+ info->mCaps.add(mediaType, caps);
}
}
return info;
@@ -196,8 +210,12 @@
status_t MediaCodecInfo::writeToParcel(Parcel *parcel) const {
mName.writeToParcel(parcel);
mOwner.writeToParcel(parcel);
- parcel->writeInt32(mIsEncoder);
+ parcel->writeInt32(mAttributes);
parcel->writeUint32(mRank);
+ parcel->writeInt32(mAliases.size());
+ for (const AString &alias : mAliases) {
+ alias.writeToParcel(parcel);
+ }
parcel->writeInt32(mCaps.size());
for (size_t i = 0; i < mCaps.size(); i++) {
mCaps.keyAt(i).writeToParcel(parcel);
@@ -206,10 +224,10 @@
return OK;
}
-ssize_t MediaCodecInfo::getCapabilityIndex(const char *mime) const {
- if (mime) {
+ssize_t MediaCodecInfo::getCapabilityIndex(const char *mediaType) const {
+ if (mediaType) {
for (size_t ix = 0; ix < mCaps.size(); ix++) {
- if (mCaps.keyAt(ix).equalsIgnoreCase(mime)) {
+ if (mCaps.keyAt(ix).equalsIgnoreCase(mediaType)) {
return ix;
}
}
@@ -217,19 +235,26 @@
return -1;
}
-MediaCodecInfo::MediaCodecInfo() : mRank(0x100) {
+MediaCodecInfo::MediaCodecInfo()
+ : mAttributes((MediaCodecInfo::Attributes)0),
+ mRank(0x100) {
}
void MediaCodecInfoWriter::setName(const char* name) {
mInfo->mName = name;
}
+void MediaCodecInfoWriter::addAlias(const char* name) {
+ mInfo->mAliases.add(name);
+}
+
void MediaCodecInfoWriter::setOwner(const char* owner) {
mInfo->mOwner = owner;
}
-void MediaCodecInfoWriter::setEncoder(bool isEncoder) {
- mInfo->mIsEncoder = isEncoder;
+void MediaCodecInfoWriter::setAttributes(
+ typename std::underlying_type<MediaCodecInfo::Attributes>::type attributes) {
+ mInfo->mAttributes = (MediaCodecInfo::Attributes)attributes;
}
void MediaCodecInfoWriter::setRank(uint32_t rank) {
@@ -237,21 +262,21 @@
}
std::unique_ptr<MediaCodecInfo::CapabilitiesWriter>
- MediaCodecInfoWriter::addMime(const char *mime) {
- ssize_t ix = mInfo->getCapabilityIndex(mime);
+ MediaCodecInfoWriter::addMediaType(const char *mediaType) {
+ ssize_t ix = mInfo->getCapabilityIndex(mediaType);
if (ix >= 0) {
return std::unique_ptr<MediaCodecInfo::CapabilitiesWriter>(
new MediaCodecInfo::CapabilitiesWriter(
mInfo->mCaps.valueAt(ix).get()));
}
sp<MediaCodecInfo::Capabilities> caps = new MediaCodecInfo::Capabilities();
- mInfo->mCaps.add(AString(mime), caps);
+ mInfo->mCaps.add(AString(mediaType), caps);
return std::unique_ptr<MediaCodecInfo::CapabilitiesWriter>(
new MediaCodecInfo::CapabilitiesWriter(caps.get()));
}
-bool MediaCodecInfoWriter::removeMime(const char *mime) {
- ssize_t ix = mInfo->getCapabilityIndex(mime);
+bool MediaCodecInfoWriter::removeMediaType(const char *mediaType) {
+ ssize_t ix = mInfo->getCapabilityIndex(mediaType);
if (ix >= 0) {
mInfo->mCaps.removeItemsAt(ix);
return true;
diff --git a/media/libmedia/TypeConverter.cpp b/media/libmedia/TypeConverter.cpp
index 0ab0e9b..aa77cd3 100644
--- a/media/libmedia/TypeConverter.cpp
+++ b/media/libmedia/TypeConverter.cpp
@@ -74,6 +74,7 @@
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_IN_WIRED_HEADSET),
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_IN_AUX_DIGITAL),
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_IN_HDMI),
+ MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_IN_HDMI_ARC),
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_IN_TELEPHONY_RX),
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_IN_VOICE_CALL),
MAKE_STRING_FROM_ENUM(AUDIO_DEVICE_IN_BACK_MIC),
@@ -208,6 +209,14 @@
MAKE_STRING_FROM_ENUM(AUDIO_FORMAT_MAT_1_0),
MAKE_STRING_FROM_ENUM(AUDIO_FORMAT_MAT_2_0),
MAKE_STRING_FROM_ENUM(AUDIO_FORMAT_MAT_2_1),
+ MAKE_STRING_FROM_ENUM(AUDIO_FORMAT_AAC_LATM),
+ MAKE_STRING_FROM_ENUM(AUDIO_FORMAT_AAC_LATM_LC),
+ MAKE_STRING_FROM_ENUM(AUDIO_FORMAT_AAC_LATM_HE_V1),
+ MAKE_STRING_FROM_ENUM(AUDIO_FORMAT_AAC_LATM_HE_V2),
+ MAKE_STRING_FROM_ENUM(AUDIO_FORMAT_CELT),
+ MAKE_STRING_FROM_ENUM(AUDIO_FORMAT_APTX_ADAPTIVE),
+ MAKE_STRING_FROM_ENUM(AUDIO_FORMAT_LHDC),
+ MAKE_STRING_FROM_ENUM(AUDIO_FORMAT_LHDC_LL),
TERMINATOR
};
diff --git a/media/libmedia/include/media/DrmHal.h b/media/libmedia/include/media/DrmHal.h
index de0f3c7..7be5cf2 100644
--- a/media/libmedia/include/media/DrmHal.h
+++ b/media/libmedia/include/media/DrmHal.h
@@ -38,6 +38,7 @@
using drm::V1_0::IDrmPlugin;
using drm::V1_0::IDrmPluginListener;
using drm::V1_0::KeyStatus;
+using drm::V1_1::SecurityLevel;
using drm::V1_2::OfflineLicenseState;
using ::android::hardware::hidl_vec;
using ::android::hardware::Return;
@@ -62,7 +63,9 @@
virtual status_t initCheck() const;
- virtual bool isCryptoSchemeSupported(const uint8_t uuid[16], const String8 &mimeType);
+ virtual bool isCryptoSchemeSupported(const uint8_t uuid[16],
+ const String8& mimeType,
+ DrmPlugin::SecurityLevel level);
virtual status_t createPlugin(const uint8_t uuid[16],
const String8 &appPackageName);
@@ -223,6 +226,10 @@
status_t getPropertyStringInternal(String8 const &name, String8 &value) const;
status_t getPropertyByteArrayInternal(String8 const &name,
Vector<uint8_t> &value) const;
+ bool matchMimeTypeAndSecurityLevel(sp<IDrmFactory> &factory,
+ const uint8_t uuid[16],
+ const String8 &mimeType,
+ DrmPlugin::SecurityLevel level);
DISALLOW_EVIL_CONSTRUCTORS(DrmHal);
};
diff --git a/media/libmedia/include/media/IDrm.h b/media/libmedia/include/media/IDrm.h
index 49166c6..a32756f 100644
--- a/media/libmedia/include/media/IDrm.h
+++ b/media/libmedia/include/media/IDrm.h
@@ -34,7 +34,9 @@
virtual status_t initCheck() const = 0;
- virtual bool isCryptoSchemeSupported(const uint8_t uuid[16], const String8 &mimeType) = 0;
+ virtual bool isCryptoSchemeSupported(const uint8_t uuid[16],
+ const String8 &mimeType,
+ DrmPlugin::SecurityLevel securityLevel) = 0;
virtual status_t createPlugin(const uint8_t uuid[16],
const String8 &appPackageName) = 0;
diff --git a/media/libmedia/include/media/MediaCodecInfo.h b/media/libmedia/include/media/MediaCodecInfo.h
index b3777d3..54f565a 100644
--- a/media/libmedia/include/media/MediaCodecInfo.h
+++ b/media/libmedia/include/media/MediaCodecInfo.h
@@ -30,6 +30,8 @@
#include <utils/Vector.h>
#include <utils/StrongPointer.h>
+#include <type_traits>
+
namespace android {
struct AMessage;
@@ -51,21 +53,47 @@
struct CapabilitiesWriter;
+ enum Attributes : int32_t {
+ // attribute flags
+ kFlagIsEncoder = 1 << 0,
+ kFlagIsVendor = 1 << 1,
+ kFlagIsSoftwareOnly = 1 << 2,
+ kFlagIsHardwareAccelerated = 1 << 3,
+ };
+
struct Capabilities : public RefBase {
- enum {
- // decoder flags
- kFlagSupportsAdaptivePlayback = 1 << 0,
- kFlagSupportsSecurePlayback = 1 << 1,
- kFlagSupportsTunneledPlayback = 1 << 2,
+ constexpr static char FEATURE_ADAPTIVE_PLAYBACK[] = "feature-adaptive-playback";
+ constexpr static char FEATURE_DYNAMIC_TIMESTAMP[] = "feature-dynamic-timestamp";
+ constexpr static char FEATURE_FRAME_PARSING[] = "feature-frame-parsing";
+ constexpr static char FEATURE_INTRA_REFRESH[] = "feature-frame-parsing";
+ constexpr static char FEATURE_MULTIPLE_FRAMES[] = "feature-multiple-frames";
+ constexpr static char FEATURE_SECURE_PLAYBACK[] = "feature-secure-playback";
+ constexpr static char FEATURE_TUNNELED_PLAYBACK[] = "feature-tunneled-playback";
- // encoder flags
- kFlagSupportsIntraRefresh = 1 << 0,
-
- };
-
+ /**
+ * Returns the supported levels for each supported profile in a target array.
+ *
+ * @param profileLevels target array for the profile levels.
+ */
void getSupportedProfileLevels(Vector<ProfileLevel> *profileLevels) const;
+
+ /**
+ * Returns the supported color formats in a target array. Only used for video/image
+ * components.
+ *
+ * @param colorFormats target array for the color formats.
+ */
void getSupportedColorFormats(Vector<uint32_t> *colorFormats) const;
- uint32_t getFlags() const;
+
+ /**
+ * Returns metadata associated with this codec capability.
+ *
+ * This contains:
+ * - features,
+ * - performance data.
+ *
+ * TODO: expose this as separate API-s and wrap here.
+ */
const sp<AMessage> getDetails() const;
protected:
@@ -73,7 +101,6 @@
SortedVector<ProfileLevel> mProfileLevelsSorted;
Vector<uint32_t> mColorFormats;
SortedVector<uint32_t> mColorFormatsSorted;
- uint32_t mFlags;
sp<AMessage> mDetails;
Capabilities();
@@ -93,8 +120,7 @@
/**
* This class is used for modifying information inside a `Capabilities`
* object. An object of type `CapabilitiesWriter` can be obtained by calling
- * `MediaCodecInfoWriter::addMime()` or
- * `MediaCodecInfoWriter::updateMime()`.
+ * `MediaCodecInfoWriter::addMediaType()`.
*/
struct CapabilitiesWriter {
/**
@@ -122,6 +148,13 @@
*/
void addDetail(const char* key, int32_t value);
/**
+ * Removes a key-value pair from the list of details. If the key is not
+ * present, this call does nothing.
+ *
+ * @param key The key.
+ */
+ void removeDetail(const char* key);
+ /**
* Add a profile-level pair. If this profile-level pair already exists,
* it will be ignored.
*
@@ -136,13 +169,7 @@
* @param format The color format.
*/
void addColorFormat(uint32_t format);
- /**
- * Add flags. The underlying operation is bitwise-or. In other words,
- * bits that have already been set will be ignored.
- *
- * @param flags The additional flags.
- */
- void addFlags(uint32_t flags);
+
private:
/**
* The associated `Capabilities` object.
@@ -158,19 +185,42 @@
friend MediaCodecInfoWriter;
};
- bool isEncoder() const;
- void getSupportedMimes(Vector<AString> *mimes) const;
- const sp<Capabilities> getCapabilitiesFor(const char *mime) const;
+ inline bool isEncoder() const {
+ return getAttributes() & kFlagIsEncoder;
+ }
+
+ Attributes getAttributes() const;
+ void getSupportedMediaTypes(Vector<AString> *mediaTypes) const;
+ const sp<Capabilities> getCapabilitiesFor(const char *mediaType) const;
const char *getCodecName() const;
/**
+ * Returns a vector containing alternate names for the codec.
+ *
+ * \param aliases the destination array for the aliases. This is cleared.
+ *
+ * Multiple codecs may share alternate names as long as their supported media types are
+ * distinct; however, these will result in different aliases for the MediaCodec user as
+ * the canonical codec has to be resolved without knowing the media type in
+ * MediaCodec::CreateByComponentName.
+ */
+ void getAliases(Vector<AString> *aliases) const;
+
+ /**
* Return the name of the service that hosts the codec. This value is not
* visible at the Java level.
*
* Currently, this is the "instance name" of the IOmx service.
*/
const char *getOwnerName() const;
- uint32_t rank() const;
+
+ /**
+ * Returns the rank of the component.
+ *
+ * Technically this is defined to be per media type, but that makes ordering the MediaCodecList
+ * impossible as MediaCodecList is ordered by codec name.
+ */
+ uint32_t getRank() const;
/**
* Serialization over Binder
@@ -181,11 +231,12 @@
private:
AString mName;
AString mOwner;
- bool mIsEncoder;
+ Attributes mAttributes;
KeyedVector<AString, sp<Capabilities> > mCaps;
+ Vector<AString> mAliases;
uint32_t mRank;
- ssize_t getCapabilityIndex(const char *mime) const;
+ ssize_t getCapabilityIndex(const char *mediaType) const;
/**
* Construct an `MediaCodecInfo` object. After the construction, its
@@ -219,6 +270,13 @@
*/
void setName(const char* name);
/**
+ * Adds an alias (alternate name) for the codec. Multiple codecs can share an alternate name
+ * as long as their supported media types are distinct.
+ *
+ * @param name an alternate name.
+ */
+ void addAlias(const char* name);
+ /**
* Set the owner name of the codec.
*
* This "owner name" is the name of the `IOmx` instance that supports this
@@ -228,32 +286,32 @@
*/
void setOwner(const char* owner);
/**
- * Set whether this codec is an encoder or a decoder.
+ * Sets codec attributes.
*
- * @param isEncoder Whether this codec is an encoder or a decoder.
+ * @param attributes Codec attributes.
*/
- void setEncoder(bool isEncoder = true);
+ void setAttributes(typename std::underlying_type<MediaCodecInfo::Attributes>::type attributes);
/**
- * Add a mime to an indexed list and return a `CapabilitiesWriter` object
+ * Add a media type to an indexed list and return a `CapabilitiesWriter` object
* that can be used for modifying the associated `Capabilities`.
*
- * If the mime already exists, this function will return the
- * `CapabilitiesWriter` associated with the mime.
+ * If the media type already exists, this function will return the
+ * `CapabilitiesWriter` associated with the media type.
*
- * @param[in] mime The name of a new mime to add.
+ * @param[in] mediaType The name of a new media type to add.
* @return writer The `CapabilitiesWriter` object for modifying the
- * `Capabilities` associated with the mime. `writer` will be valid
- * regardless of whether `mime` already exists or not.
+ * `Capabilities` associated with the media type. `writer` will be valid
+ * regardless of whether `mediaType` already exists or not.
*/
- std::unique_ptr<MediaCodecInfo::CapabilitiesWriter> addMime(
- const char* mime);
+ std::unique_ptr<MediaCodecInfo::CapabilitiesWriter> addMediaType(
+ const char* mediaType);
/**
- * Remove a mime.
+ * Remove a media type.
*
- * @param mime The name of the mime to remove.
- * @return `true` if `mime` is removed; `false` if `mime` is not found.
+ * @param mediaType The name of the media type to remove.
+ * @return `true` if `mediaType` is removed; `false` if `mediaType` is not found.
*/
- bool removeMime(const char* mime);
+ bool removeMediaType(const char* mediaType);
/**
* Set rank of the codec. MediaCodecList will stable-sort the list according
* to rank in non-descending order.
diff --git a/media/libmediametrics/Android.bp b/media/libmediametrics/Android.bp
index e188e54..15ea578 100644
--- a/media/libmediametrics/Android.bp
+++ b/media/libmediametrics/Android.bp
@@ -1,6 +1,4 @@
-// TODO: change it back to cc_library_shared when there is a way to
-// expose media metrics as stable API.
-cc_library {
+cc_library_shared {
name: "libmediametrics",
srcs: [
@@ -32,12 +30,13 @@
cfi: true,
},
- // enumerate the stable interface
-// this would mean nobody can use the C++ interface. have to rework some things.
-// stubs: {
-// symbol_file: "libmediametrics.map.txt",
-// versions: [
-// "1" ,
-// ]
-// },
+ // enumerate stable entry points, for apex use
+ stubs: {
+ symbol_file: "libmediametrics.map.txt",
+ versions: [
+ "1" ,
+ ]
+ },
}
+
+
diff --git a/media/libmediametrics/IMediaAnalyticsService.cpp b/media/libmediametrics/IMediaAnalyticsService.cpp
index 28a7746..9114927 100644
--- a/media/libmediametrics/IMediaAnalyticsService.cpp
+++ b/media/libmediametrics/IMediaAnalyticsService.cpp
@@ -142,7 +142,7 @@
CHECK_INTERFACE(IMediaAnalyticsService, data, reply);
bool forcenew;
- MediaAnalyticsItem *item = new MediaAnalyticsItem;
+ MediaAnalyticsItem *item = MediaAnalyticsItem::create();
data.readBool(&forcenew);
item->readFromParcel(data);
diff --git a/media/libmediametrics/MediaAnalyticsItem.cpp b/media/libmediametrics/MediaAnalyticsItem.cpp
index 448e2d9..02c23b1 100644
--- a/media/libmediametrics/MediaAnalyticsItem.cpp
+++ b/media/libmediametrics/MediaAnalyticsItem.cpp
@@ -52,6 +52,17 @@
const char * const MediaAnalyticsItem::EnabledPropertyPersist = "persist.media.metrics.enabled";
const int MediaAnalyticsItem::EnabledProperty_default = 1;
+// So caller doesn't need to know size of allocated space
+MediaAnalyticsItem *MediaAnalyticsItem::create()
+{
+ return MediaAnalyticsItem::create(kKeyNone);
+}
+
+MediaAnalyticsItem *MediaAnalyticsItem::create(MediaAnalyticsItem::Key key)
+{
+ MediaAnalyticsItem *item = new MediaAnalyticsItem(key);
+ return item;
+}
// access functions for the class
MediaAnalyticsItem::MediaAnalyticsItem()
@@ -642,6 +653,19 @@
//
int32_t MediaAnalyticsItem::readFromParcel(const Parcel& data) {
+ int32_t version = data.readInt32();
+
+ switch(version) {
+ case 0:
+ return readFromParcel0(data);
+ break;
+ default:
+ ALOGE("Unsupported MediaAnalyticsItem Parcel version: %d", version);
+ return -1;
+ }
+}
+
+int32_t MediaAnalyticsItem::readFromParcel0(const Parcel& data) {
// into 'this' object
// .. we make a copy of the string to put away.
mKey = data.readCString();
@@ -691,8 +715,23 @@
}
int32_t MediaAnalyticsItem::writeToParcel(Parcel *data) {
+
if (data == NULL) return -1;
+ int32_t version = 0;
+ data->writeInt32(version);
+
+ switch(version) {
+ case 0:
+ return writeToParcel0(data);
+ break;
+ default:
+ ALOGE("Unsupported MediaAnalyticsItem Parcel version: %d", version);
+ return -1;
+ }
+}
+
+int32_t MediaAnalyticsItem::writeToParcel0(Parcel *data) {
data->writeCString(mKey.c_str());
data->writeInt32(mPid);
@@ -737,7 +776,6 @@
return 0;
}
-
const char *MediaAnalyticsItem::toCString() {
return toCString(PROTO_LAST);
}
@@ -876,8 +914,6 @@
}
return true;
} else {
- std::string p = this->toString();
- ALOGW("Unable to record: %s [forcenew=%d]", p.c_str(), forcenew);
return false;
}
}
@@ -1035,5 +1071,170 @@
return true;
}
+// a byte array; contents are
+// overall length (uint32) including the length field itself
+// encoding version (uint32)
+// count of properties (uint32)
+// N copies of:
+// property name as length(int16), bytes
+// the bytes WILL include the null terminator of the name
+// type (uint8 -- 1 byte)
+// size of value field (int16 -- 2 bytes)
+// value (size based on type)
+// int32, int64, double -- little endian 4/8/8 bytes respectively
+// cstring -- N bytes of value [WITH terminator]
+
+enum { kInt32 = 0, kInt64, kDouble, kRate, kCString};
+
+bool MediaAnalyticsItem::dumpAttributes(char **pbuffer, size_t *plength) {
+
+ char *build = NULL;
+
+ if (pbuffer == NULL || plength == NULL)
+ return false;
+
+ // consistency for the caller, who owns whatever comes back in this pointer.
+ *pbuffer = NULL;
+
+ // first, let's calculate sizes
+ int32_t goal = 0;
+ int32_t version = 0;
+
+ goal += sizeof(uint32_t); // overall length, including the length field
+ goal += sizeof(uint32_t); // encoding version
+ goal += sizeof(uint32_t); // # properties
+
+ int32_t count = mPropCount;
+ for (int i = 0 ; i < count; i++ ) {
+ Prop *prop = &mProps[i];
+ goal += sizeof(uint16_t); // name length
+ goal += strlen(prop->mName) + 1; // string + null
+ goal += sizeof(uint8_t); // type
+ goal += sizeof(uint16_t); // size of value
+ switch (prop->mType) {
+ case MediaAnalyticsItem::kTypeInt32:
+ goal += sizeof(uint32_t);
+ break;
+ case MediaAnalyticsItem::kTypeInt64:
+ goal += sizeof(uint64_t);
+ break;
+ case MediaAnalyticsItem::kTypeDouble:
+ goal += sizeof(double);
+ break;
+ case MediaAnalyticsItem::kTypeRate:
+ goal += 2 * sizeof(uint64_t);
+ break;
+ case MediaAnalyticsItem::kTypeCString:
+ // length + actual string + null
+ goal += strlen(prop->u.CStringValue) + 1;
+ break;
+ default:
+ ALOGE("found bad Prop type: %d, idx %d, name %s",
+ prop->mType, i, prop->mName);
+ return false;
+ }
+ }
+
+ // now that we have a size... let's allocate and fill
+ build = (char *)malloc(goal);
+ if (build == NULL)
+ return false;
+
+ memset(build, 0, goal);
+
+ char *filling = build;
+
+#define _INSERT(val, size) \
+ { memcpy(filling, &(val), (size)); filling += (size);}
+#define _INSERTSTRING(val, size) \
+ { memcpy(filling, (val), (size)); filling += (size);}
+
+ _INSERT(goal, sizeof(int32_t));
+ _INSERT(version, sizeof(int32_t));
+ _INSERT(count, sizeof(int32_t));
+
+ for (int i = 0 ; i < count; i++ ) {
+ Prop *prop = &mProps[i];
+ int16_t attrNameLen = strlen(prop->mName) + 1;
+ _INSERT(attrNameLen, sizeof(int16_t));
+ _INSERTSTRING(prop->mName, attrNameLen); // termination included
+ int8_t elemtype;
+ int16_t elemsize;
+ switch (prop->mType) {
+ case MediaAnalyticsItem::kTypeInt32:
+ {
+ elemtype = kInt32;
+ _INSERT(elemtype, sizeof(int8_t));
+ elemsize = sizeof(int32_t);
+ _INSERT(elemsize, sizeof(int16_t));
+
+ _INSERT(prop->u.int32Value, sizeof(int32_t));
+ break;
+ }
+ case MediaAnalyticsItem::kTypeInt64:
+ {
+ elemtype = kInt64;
+ _INSERT(elemtype, sizeof(int8_t));
+ elemsize = sizeof(int64_t);
+ _INSERT(elemsize, sizeof(int16_t));
+
+ _INSERT(prop->u.int64Value, sizeof(int64_t));
+ break;
+ }
+ case MediaAnalyticsItem::kTypeDouble:
+ {
+ elemtype = kDouble;
+ _INSERT(elemtype, sizeof(int8_t));
+ elemsize = sizeof(double);
+ _INSERT(elemsize, sizeof(int16_t));
+
+ _INSERT(prop->u.doubleValue, sizeof(double));
+ break;
+ }
+ case MediaAnalyticsItem::kTypeRate:
+ {
+ elemtype = kRate;
+ _INSERT(elemtype, sizeof(int8_t));
+ elemsize = 2 * sizeof(uint64_t);
+ _INSERT(elemsize, sizeof(int16_t));
+
+ _INSERT(prop->u.rate.count, sizeof(uint64_t));
+ _INSERT(prop->u.rate.duration, sizeof(uint64_t));
+ break;
+ }
+ case MediaAnalyticsItem::kTypeCString:
+ {
+ elemtype = kCString;
+ _INSERT(elemtype, sizeof(int8_t));
+ elemsize = strlen(prop->u.CStringValue) + 1;
+ _INSERT(elemsize, sizeof(int16_t));
+
+ _INSERTSTRING(prop->u.CStringValue, elemsize);
+ break;
+ }
+ default:
+ // error if can't encode; warning if can't decode
+ ALOGE("found bad Prop type: %d, idx %d, name %s",
+ prop->mType, i, prop->mName);
+ goto badness;
+ }
+ }
+
+ if (build + goal != filling) {
+ ALOGE("problems populating; wrote=%d planned=%d",
+ (int)(filling-build), goal);
+ goto badness;
+ }
+
+ *pbuffer = build;
+ *plength = goal;
+
+ return true;
+
+ badness:
+ free(build);
+ return false;
+}
+
} // namespace android
diff --git a/media/libmediametrics/MediaMetrics.cpp b/media/libmediametrics/MediaMetrics.cpp
index 9b08aa7..6109190 100644
--- a/media/libmediametrics/MediaMetrics.cpp
+++ b/media/libmediametrics/MediaMetrics.cpp
@@ -34,7 +34,7 @@
// manage the overall record
mediametrics_handle_t mediametrics_create(mediametricskey_t key) {
- android::MediaAnalyticsItem *item = new android::MediaAnalyticsItem(key);
+ android::MediaAnalyticsItem *item = android::MediaAnalyticsItem::create(key);
return (mediametrics_handle_t) item;
}
@@ -187,18 +187,9 @@
return android::MediaAnalyticsItem::isEnabled();
}
-#if 0
-// do not expose this as is.
-// need to revisit (or redefine) how the android::Parcel parameter is handled
-// so that it meets the stable-API criteria for updateable components.
-//
-int32_t mediametrics_writeToParcel(mediametrics_handle_t handle, android::Parcel *parcel) {
+bool mediametrics_getAttributes(mediametrics_handle_t handle, char **buffer, size_t *length) {
android::MediaAnalyticsItem *item = (android::MediaAnalyticsItem *) handle;
- if (item == NULL) {
- return -1;
- }
- return item->writeToParcel(parcel);
+ if (item == NULL) return false;
+ return item->dumpAttributes(buffer, length);
+
}
-#endif
-
-
diff --git a/media/libmediametrics/include/MediaAnalyticsItem.h b/media/libmediametrics/include/MediaAnalyticsItem.h
index b99cd91..2f9e7c2 100644
--- a/media/libmediametrics/include/MediaAnalyticsItem.h
+++ b/media/libmediametrics/include/MediaAnalyticsItem.h
@@ -17,9 +17,10 @@
#ifndef ANDROID_MEDIA_MEDIAANALYTICSITEM_H
#define ANDROID_MEDIA_MEDIAANALYTICSITEM_H
-#include <cutils/properties.h>
#include <string>
#include <sys/types.h>
+
+#include <cutils/properties.h>
#include <utils/Errors.h>
#include <utils/KeyedVector.h>
#include <utils/RefBase.h>
@@ -84,6 +85,10 @@
public:
+ // so clients do not need to know size details
+ static MediaAnalyticsItem* create(Key key);
+ static MediaAnalyticsItem* create();
+
// access functions for the class
MediaAnalyticsItem();
MediaAnalyticsItem(Key);
@@ -175,6 +180,9 @@
int32_t writeToParcel(Parcel *);
int32_t readFromParcel(const Parcel&);
+ // supports the stable interface
+ bool dumpAttributes(char **pbuffer, size_t *plength);
+
std::string toString();
std::string toString(int version);
const char *toCString();
@@ -183,6 +191,11 @@
// are we collecting analytics data
static bool isEnabled();
+ private:
+ // handle Parcel version 0
+ int32_t writeToParcel0(Parcel *);
+ int32_t readFromParcel0(const Parcel&);
+
protected:
// merge fields from arg into this
diff --git a/media/libmediametrics/include/MediaMetrics.h b/media/libmediametrics/include/MediaMetrics.h
index 4d2f352..a4e1ed2 100644
--- a/media/libmediametrics/include/MediaMetrics.h
+++ b/media/libmediametrics/include/MediaMetrics.h
@@ -85,13 +85,9 @@
void mediametrics_setUid(mediametrics_handle_t handle, uid_t uid);
bool mediametrics_isEnabled();
-#if 0
-// do not expose this as is.
-// need to revisit (or redefine) how the android::Parcel parameter is handled
-// so that it meets the stable-API criteria for updateable components.
-//
-int32_t mediametrics_writeToParcel(mediametrics_handle_t handle, android::Parcel *parcel);
-#endif
+// serialized copy of the attributes/values, mostly for upstream getMetrics() calls
+// caller owns the buffer allocated as part of this call.
+bool mediametrics_getAttributes(mediametrics_handle_t handle, char **buffer, size_t *length);
__END_DECLS
diff --git a/media/libmediametrics/libmediametrics.map.txt b/media/libmediametrics/libmediametrics.map.txt
new file mode 100644
index 0000000..c46281a
--- /dev/null
+++ b/media/libmediametrics/libmediametrics.map.txt
@@ -0,0 +1,29 @@
+LIBMEDIAMETRICS_1 {
+ global:
+ mediametrics_addDouble; # apex
+ mediametrics_addInt32; # apex
+ mediametrics_addInt64; # apex
+ mediametrics_addRate; # apex
+ mediametrics_count; # apex
+ mediametrics_create; # apex
+ mediametrics_delete; # apex
+ mediametrics_freeCString; # apex
+ mediametrics_getAttributes; # apex
+ mediametrics_getCString; # apex
+ mediametrics_getDouble; # apex
+ mediametrics_getInt32; # apex
+ mediametrics_getInt64; # apex
+ mediametrics_getKey; # apex
+ mediametrics_getRate; # apex
+ mediametrics_isEnabled; # apex
+ mediametrics_readable; # apex
+ mediametrics_selfRecord; # apex
+ mediametrics_setCString; # apex
+ mediametrics_setDouble; # apex
+ mediametrics_setInt32; # apex
+ mediametrics_setInt64; # apex
+ mediametrics_setRate; # apex
+ mediametrics_setUid; # apex
+ local:
+ *;
+};
diff --git a/media/libmediaplayer2/include/mediaplayer2/MediaPlayer2Interface.h b/media/libmediaplayer2/include/mediaplayer2/MediaPlayer2Interface.h
index 0c8d016..7804a62 100644
--- a/media/libmediaplayer2/include/mediaplayer2/MediaPlayer2Interface.h
+++ b/media/libmediaplayer2/include/mediaplayer2/MediaPlayer2Interface.h
@@ -214,6 +214,8 @@
virtual status_t setParameter(int key, const Parcel &request) = 0;
virtual status_t getParameter(int key, Parcel *reply) = 0;
+ virtual status_t getMetrics(char **buffer, size_t *length) = 0;
+
// Invoke a generic method on the player by using opaque parcels
// for the request and reply.
//
diff --git a/media/libmediaplayer2/include/mediaplayer2/mediaplayer2.h b/media/libmediaplayer2/include/mediaplayer2/mediaplayer2.h
index 78865c4..2993ab1 100644
--- a/media/libmediaplayer2/include/mediaplayer2/mediaplayer2.h
+++ b/media/libmediaplayer2/include/mediaplayer2/mediaplayer2.h
@@ -102,6 +102,7 @@
status_t setAudioAttributes(const jobject attributes);
jobject getAudioAttributes();
status_t getParameter(int key, Parcel* reply);
+ status_t getMetrics(char **buffer, size_t *length);
// Modular DRM
status_t prepareDrm(int64_t srcId,
diff --git a/media/libmediaplayer2/mediaplayer2.cpp b/media/libmediaplayer2/mediaplayer2.cpp
index f75380c..53f2fb1 100644
--- a/media/libmediaplayer2/mediaplayer2.cpp
+++ b/media/libmediaplayer2/mediaplayer2.cpp
@@ -21,7 +21,6 @@
#include <android/binder_ibinder.h>
#include <media/AudioSystem.h>
#include <media/DataSourceDesc.h>
-#include <media/MediaAnalyticsItem.h>
#include <media/MemoryLeakTrackUtil.h>
#include <media/NdkWrapper.h>
#include <media/stagefright/foundation/ADebug.h>
@@ -979,6 +978,22 @@
return status;
}
+// for mediametrics
+status_t MediaPlayer2::getMetrics(char **buffer, size_t *length) {
+ ALOGD("MediaPlayer2::getMetrics()");
+ Mutex::Autolock _l(mLock);
+ if (mPlayer == NULL) {
+ ALOGV("getMetrics: no active player");
+ return INVALID_OPERATION;
+ }
+
+ status_t status = mPlayer->getMetrics(buffer, length);
+ if (status != OK) {
+ ALOGD("getMetrics returns %d", status);
+ }
+ return status;
+}
+
void MediaPlayer2::notify(int64_t srcId, int msg, int ext1, int ext2, const PlayerMessage *obj) {
ALOGV("message received srcId=%lld, msg=%d, ext1=%d, ext2=%d",
(long long)srcId, msg, ext1, ext2);
diff --git a/media/libmediaplayer2/nuplayer2/Android.bp b/media/libmediaplayer2/nuplayer2/Android.bp
index 71cd50f..0f69b2e 100644
--- a/media/libmediaplayer2/nuplayer2/Android.bp
+++ b/media/libmediaplayer2/nuplayer2/Android.bp
@@ -51,6 +51,7 @@
"libui",
"libgui",
"libmedia",
+ "libmediametrics",
"libmediandk",
"libmediandk_utils",
"libpowermanager",
diff --git a/media/libmediaplayer2/nuplayer2/NuPlayer2Decoder.cpp b/media/libmediaplayer2/nuplayer2/NuPlayer2Decoder.cpp
index a5bd62d..9729d86 100644
--- a/media/libmediaplayer2/nuplayer2/NuPlayer2Decoder.cpp
+++ b/media/libmediaplayer2/nuplayer2/NuPlayer2Decoder.cpp
@@ -107,6 +107,8 @@
mStats->setInt64("frames-total", mNumFramesTotal);
mStats->setInt64("frames-dropped-input", mNumInputFramesDropped);
mStats->setInt64("frames-dropped-output", mNumOutputFramesDropped);
+ mStats->setFloat("frame-rate-total", mFrameRateTotal);
+
return mStats;
}
diff --git a/media/libmediaplayer2/nuplayer2/NuPlayer2Driver.cpp b/media/libmediaplayer2/nuplayer2/NuPlayer2Driver.cpp
index 56e9471..1b661f2 100644
--- a/media/libmediaplayer2/nuplayer2/NuPlayer2Driver.cpp
+++ b/media/libmediaplayer2/nuplayer2/NuPlayer2Driver.cpp
@@ -92,6 +92,7 @@
static const char *kPlayerHeight = "android.media.mediaplayer.height";
static const char *kPlayerFrames = "android.media.mediaplayer.frames";
static const char *kPlayerFramesDropped = "android.media.mediaplayer.dropped";
+static const char *kPlayerFrameRate = "android.media.mediaplayer.fps";
static const char *kPlayerAMime = "android.media.mediaplayer.audio.mime";
static const char *kPlayerACodec = "android.media.mediaplayer.audio.codec";
static const char *kPlayerDuration = "android.media.mediaplayer.durationMs";
@@ -125,7 +126,7 @@
mMediaClock(new MediaClock),
mPlayer(new NuPlayer2(pid, uid, mMediaClock, context)),
mPlayerFlags(0),
- mAnalyticsItem(NULL),
+ mMetricsHandle(0),
mClientUid(uid),
mAtEOS(false),
mLooping(false),
@@ -136,9 +137,9 @@
mMediaClock->init();
- // set up an analytics record
- mAnalyticsItem = new MediaAnalyticsItem(kKeyPlayer);
- mAnalyticsItem->setUid(mClientUid);
+ // set up media metrics record
+ mMetricsHandle = mediametrics_create(kKeyPlayer);
+ mediametrics_setUid(mMetricsHandle, mClientUid);
mNuPlayer2Looper->start(
false, /* runOnCallingThread */
@@ -159,10 +160,7 @@
updateMetrics("destructor");
logMetrics("destructor");
- if (mAnalyticsItem != NULL) {
- delete mAnalyticsItem;
- mAnalyticsItem = NULL;
- }
+ mediametrics_delete(mMetricsHandle);
}
status_t NuPlayer2Driver::initCheck() {
@@ -453,15 +451,15 @@
if (mime.startsWith("video/")) {
int32_t width, height;
- mAnalyticsItem->setCString(kPlayerVMime, mime.c_str());
+ mediametrics_setCString(mMetricsHandle, kPlayerVMime, mime.c_str());
if (!name.empty()) {
- mAnalyticsItem->setCString(kPlayerVCodec, name.c_str());
+ mediametrics_setCString(mMetricsHandle, kPlayerVCodec, name.c_str());
}
if (stats->findInt32("width", &width)
&& stats->findInt32("height", &height)) {
- mAnalyticsItem->setInt32(kPlayerWidth, width);
- mAnalyticsItem->setInt32(kPlayerHeight, height);
+ mediametrics_setInt32(mMetricsHandle, kPlayerWidth, width);
+ mediametrics_setInt32(mMetricsHandle, kPlayerHeight, height);
}
int64_t numFramesTotal = 0;
@@ -469,14 +467,18 @@
stats->findInt64("frames-total", &numFramesTotal);
stats->findInt64("frames-dropped-output", &numFramesDropped);
- mAnalyticsItem->setInt64(kPlayerFrames, numFramesTotal);
- mAnalyticsItem->setInt64(kPlayerFramesDropped, numFramesDropped);
+ mediametrics_setInt64(mMetricsHandle, kPlayerFrames, numFramesTotal);
+ mediametrics_setInt64(mMetricsHandle, kPlayerFramesDropped, numFramesDropped);
+ float frameRate = 0;
+ if (stats->findFloat("frame-rate-output", &frameRate)) {
+ mediametrics_setInt64(mMetricsHandle, kPlayerFrameRate, frameRate);
+ }
} else if (mime.startsWith("audio/")) {
- mAnalyticsItem->setCString(kPlayerAMime, mime.c_str());
+ mediametrics_setCString(mMetricsHandle, kPlayerAMime, mime.c_str());
if (!name.empty()) {
- mAnalyticsItem->setCString(kPlayerACodec, name.c_str());
+ mediametrics_setCString(mMetricsHandle, kPlayerACodec, name.c_str());
}
}
}
@@ -487,17 +489,17 @@
// getDuration() uses mLock for mutex -- careful where we use it.
int64_t duration_ms = -1;
getDuration(&duration_ms);
- mAnalyticsItem->setInt64(kPlayerDuration, duration_ms);
+ mediametrics_setInt64(mMetricsHandle, kPlayerDuration, duration_ms);
- mAnalyticsItem->setInt64(kPlayerPlaying, (mPlayingTimeUs+500)/1000 );
+ mediametrics_setInt64(mMetricsHandle, kPlayerPlaying, (mPlayingTimeUs+500)/1000 );
if (mRebufferingEvents != 0) {
- mAnalyticsItem->setInt64(kPlayerRebuffering, (mRebufferingTimeUs+500)/1000 );
- mAnalyticsItem->setInt32(kPlayerRebufferingCount, mRebufferingEvents);
- mAnalyticsItem->setInt32(kPlayerRebufferingAtExit, mRebufferingAtExit);
+ mediametrics_setInt64(mMetricsHandle, kPlayerRebuffering, (mRebufferingTimeUs+500)/1000 );
+ mediametrics_setInt32(mMetricsHandle, kPlayerRebufferingCount, mRebufferingEvents);
+ mediametrics_setInt32(mMetricsHandle, kPlayerRebufferingAtExit, mRebufferingAtExit);
}
- mAnalyticsItem->setCString(kPlayerDataSourceType, mPlayer->getDataSourceType());
+ mediametrics_setCString(mMetricsHandle, kPlayerDataSourceType, mPlayer->getDataSourceType());
}
@@ -507,7 +509,7 @@
}
ALOGV("logMetrics(%p) from %s at state %d", this, where, mState);
- if (mAnalyticsItem == NULL || mAnalyticsItem->isEnabled() == false) {
+ if (mMetricsHandle == 0 || mediametrics_isEnabled() == false) {
return;
}
@@ -516,16 +518,12 @@
// and that always injects 3 fields (duration, playing time, and
// datasource) into the record.
// So the canonical "empty" record has 3 elements in it.
- if (mAnalyticsItem->count() > 3) {
-
- mAnalyticsItem->selfrecord();
-
+ if (mediametrics_count(mMetricsHandle) > 3) {
+ mediametrics_selfRecord(mMetricsHandle);
// re-init in case we prepare() and start() again.
- delete mAnalyticsItem ;
- mAnalyticsItem = new MediaAnalyticsItem(kKeyPlayer);
- if (mAnalyticsItem) {
- mAnalyticsItem->setUid(mClientUid);
- }
+ mediametrics_delete(mMetricsHandle);
+ mMetricsHandle = mediametrics_create(kKeyPlayer);
+ mediametrics_setUid(mMetricsHandle, mClientUid);
} else {
ALOGV("did not have anything to record");
}
@@ -649,19 +647,18 @@
return INVALID_OPERATION;
}
-status_t NuPlayer2Driver::getParameter(int key, Parcel *reply) {
-
- if (key == FOURCC('m','t','r','X')) {
- // mtrX -- a play on 'metrics' (not matrix)
- // gather current info all together, parcel it, and send it back
- updateMetrics("api");
- mAnalyticsItem->writeToParcel(reply);
- return OK;
- }
-
+status_t NuPlayer2Driver::getParameter(int key __unused, Parcel *reply __unused) {
return INVALID_OPERATION;
}
+status_t NuPlayer2Driver::getMetrics(char **buffer, size_t *length) {
+ updateMetrics("api");
+ if (mediametrics_getAttributes(mMetricsHandle, buffer, length))
+ return OK;
+ else
+ return FAILED_TRANSACTION;
+}
+
void NuPlayer2Driver::notifyResetComplete(int64_t /* srcId */) {
ALOGD("notifyResetComplete(%p)", this);
Mutex::Autolock autoLock(mLock);
@@ -867,11 +864,11 @@
// ext1 is our primary 'error type' value. Only add ext2 when non-zero.
// [test against msg is due to fall through from previous switch value]
if (msg == MEDIA2_ERROR) {
- mAnalyticsItem->setInt32(kPlayerError, ext1);
+ mediametrics_setInt32(mMetricsHandle, kPlayerError, ext1);
if (ext2 != 0) {
- mAnalyticsItem->setInt32(kPlayerErrorCode, ext2);
+ mediametrics_setInt32(mMetricsHandle, kPlayerErrorCode, ext2);
}
- mAnalyticsItem->setCString(kPlayerErrorState, stateString(mState).c_str());
+ mediametrics_setCString(mMetricsHandle, kPlayerErrorState, stateString(mState).c_str());
}
mAtEOS = true;
break;
diff --git a/media/libmediaplayer2/nuplayer2/NuPlayer2Driver.h b/media/libmediaplayer2/nuplayer2/NuPlayer2Driver.h
index 0ec3a4b..3d299f3 100644
--- a/media/libmediaplayer2/nuplayer2/NuPlayer2Driver.h
+++ b/media/libmediaplayer2/nuplayer2/NuPlayer2Driver.h
@@ -16,7 +16,7 @@
#include <mediaplayer2/MediaPlayer2Interface.h>
-#include <media/MediaAnalyticsItem.h>
+#include <media/MediaMetrics.h>
#include <media/stagefright/foundation/ABase.h>
#include <mediaplayer2/JObjectHolder.h>
@@ -61,6 +61,7 @@
virtual void setAudioSink(const sp<AudioSink> &audioSink) override;
virtual status_t setParameter(int key, const Parcel &request) override;
virtual status_t getParameter(int key, Parcel *reply) override;
+ virtual status_t getMetrics(char **buf, size_t *length) override;
virtual status_t dump(int fd, const Vector<String16> &args) const override;
@@ -132,7 +133,7 @@
sp<AudioSink> mAudioSink;
uint32_t mPlayerFlags;
- MediaAnalyticsItem *mAnalyticsItem;
+ mediametrics_handle_t mMetricsHandle;
uid_t mClientUid;
bool mAtEOS;
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDrm.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDrm.cpp
index 8d876da..67a0f1e 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDrm.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDrm.cpp
@@ -159,7 +159,8 @@
if (drm != NULL) {
for (size_t i = 0; i < psshDRMs.size(); i++) {
DrmUUID uuid = psshDRMs[i];
- if (drm->isCryptoSchemeSupported(uuid.ptr(), String8()))
+ if (drm->isCryptoSchemeSupported(uuid.ptr(), String8(),
+ DrmPlugin::kSecurityLevelUnknown))
supportedDRMs.add(uuid);
}
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index dadfe28..a1a2660 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -8682,14 +8682,17 @@
if (omxNode->configureVideoTunnelMode(
kPortIndexOutput, OMX_TRUE, 0, &sidebandHandle) == OK) {
// tunneled playback includes adaptive playback
- caps->addFlags(MediaCodecInfo::Capabilities::kFlagSupportsAdaptivePlayback
- | MediaCodecInfo::Capabilities::kFlagSupportsTunneledPlayback);
- } else if (omxNode->setPortMode(
- kPortIndexOutput, IOMX::kPortModeDynamicANWBuffer) == OK ||
- omxNode->prepareForAdaptivePlayback(
- kPortIndexOutput, OMX_TRUE,
- 1280 /* width */, 720 /* height */) == OK) {
- caps->addFlags(MediaCodecInfo::Capabilities::kFlagSupportsAdaptivePlayback);
+ } else {
+ // tunneled playback is not supported
+ caps->removeDetail(MediaCodecInfo::Capabilities::FEATURE_TUNNELED_PLAYBACK);
+ if (omxNode->setPortMode(
+ kPortIndexOutput, IOMX::kPortModeDynamicANWBuffer) != OK &&
+ omxNode->prepareForAdaptivePlayback(
+ kPortIndexOutput, OMX_TRUE,
+ 1280 /* width */, 720 /* height */) != OK) {
+ // adaptive playback is not supported
+ caps->removeDetail(MediaCodecInfo::Capabilities::FEATURE_ADAPTIVE_PLAYBACK);
+ }
}
}
@@ -8697,11 +8700,20 @@
OMX_VIDEO_CONFIG_ANDROID_INTRAREFRESHTYPE params;
InitOMXParams(¶ms);
params.nPortIndex = kPortIndexOutput;
- // TODO: should we verify if fallback is supported?
+
+ OMX_VIDEO_PARAM_INTRAREFRESHTYPE fallbackParams;
+ InitOMXParams(&fallbackParams);
+ fallbackParams.nPortIndex = kPortIndexOutput;
+ fallbackParams.eRefreshMode = OMX_VIDEO_IntraRefreshCyclic;
+
if (omxNode->getConfig(
(OMX_INDEXTYPE)OMX_IndexConfigAndroidIntraRefresh,
- ¶ms, sizeof(params)) == OK) {
- caps->addFlags(MediaCodecInfo::Capabilities::kFlagSupportsIntraRefresh);
+ ¶ms, sizeof(params)) != OK &&
+ omxNode->getParameter(
+ OMX_IndexParamVideoIntraRefresh, &fallbackParams,
+ sizeof(fallbackParams)) != OK) {
+ // intra refresh is not supported
+ caps->removeDetail(MediaCodecInfo::Capabilities::FEATURE_INTRA_REFRESH);
}
}
diff --git a/media/libstagefright/Android.bp b/media/libstagefright/Android.bp
index f45cc58..03eef48 100644
--- a/media/libstagefright/Android.bp
+++ b/media/libstagefright/Android.bp
@@ -160,7 +160,6 @@
"libstagefright_codecbase",
"libstagefright_foundation",
"libstagefright_omx_utils",
- "libstagefright_opus_common",
"libRScpp",
"libhidlallocatorutils",
"libhidlbase",
diff --git a/media/libstagefright/MPEG4Writer.cpp b/media/libstagefright/MPEG4Writer.cpp
index 7df1a2d..c4015fb 100644
--- a/media/libstagefright/MPEG4Writer.cpp
+++ b/media/libstagefright/MPEG4Writer.cpp
@@ -85,7 +85,7 @@
static const int kTimestampDebugCount = 10;
static const int kItemIdBase = 10000;
static const char kExifHeader[] = {'E', 'x', 'i', 'f', '\0', '\0'};
-static const int32_t kTiffHeaderOffset = htonl(sizeof(kExifHeader));
+static const uint8_t kExifApp1Marker[] = {'E', 'x', 'i', 'f', 0xff, 0xe1};
static const uint8_t kMandatoryHevcNalUnitTypes[3] = {
kHevcNalUnitTypeVps,
@@ -125,7 +125,7 @@
bool isAudio() const { return mIsAudio; }
bool isMPEG4() const { return mIsMPEG4; }
bool usePrefix() const { return mIsAvc || mIsHevc || mIsHeic; }
- bool isExifData(const MediaBufferBase *buffer) const;
+ bool isExifData(MediaBufferBase *buffer, uint32_t *tiffHdrOffset) const;
void addChunkOffset(off64_t offset);
void addItemOffsetAndSize(off64_t offset, size_t size, bool isExif);
void flushItemRefs();
@@ -364,7 +364,7 @@
Vector<uint16_t> mProperties;
ItemRefs mDimgRefs;
- ItemRefs mCdscRefs;
+ Vector<uint16_t> mExifList;
uint16_t mImageItemId;
int32_t mIsPrimary;
int32_t mWidth, mHeight;
@@ -1368,14 +1368,16 @@
}
off64_t MPEG4Writer::addSample_l(
- MediaBuffer *buffer, bool usePrefix, bool isExif, size_t *bytesWritten) {
+ MediaBuffer *buffer, bool usePrefix,
+ uint32_t tiffHdrOffset, size_t *bytesWritten) {
off64_t old_offset = mOffset;
if (usePrefix) {
addMultipleLengthPrefixedSamples_l(buffer);
} else {
- if (isExif) {
- ::write(mFd, &kTiffHeaderOffset, 4); // exif_tiff_header_offset field
+ if (tiffHdrOffset > 0) {
+ tiffHdrOffset = htonl(tiffHdrOffset);
+ ::write(mFd, &tiffHdrOffset, 4); // exif_tiff_header_offset field
mOffset += 4;
}
@@ -1803,7 +1805,6 @@
mStartTimestampUs(-1),
mRotation(0),
mDimgRefs("dimg"),
- mCdscRefs("cdsc"),
mImageItemId(0),
mIsPrimary(0),
mWidth(0),
@@ -1984,11 +1985,34 @@
return OK;
}
-bool MPEG4Writer::Track::isExifData(const MediaBufferBase *buffer) const {
- return mIsHeic
- && (buffer->range_length() > sizeof(kExifHeader))
- && !memcmp((uint8_t *)buffer->data() + buffer->range_offset(),
- kExifHeader, sizeof(kExifHeader));
+bool MPEG4Writer::Track::isExifData(
+ MediaBufferBase *buffer, uint32_t *tiffHdrOffset) const {
+ if (!mIsHeic) {
+ return false;
+ }
+
+ // Exif block starting with 'Exif\0\0'
+ size_t length = buffer->range_length();
+ uint8_t *data = (uint8_t *)buffer->data() + buffer->range_offset();
+ if ((length > sizeof(kExifHeader))
+ && !memcmp(data, kExifHeader, sizeof(kExifHeader))) {
+ *tiffHdrOffset = sizeof(kExifHeader);
+ return true;
+ }
+
+ // Exif block starting with fourcc 'Exif' followed by APP1 marker
+ if ((length > sizeof(kExifApp1Marker) + 2 + sizeof(kExifHeader))
+ && !memcmp(data, kExifApp1Marker, sizeof(kExifApp1Marker))
+ && !memcmp(data + sizeof(kExifApp1Marker) + 2, kExifHeader, sizeof(kExifHeader))) {
+ // skip 'Exif' fourcc
+ buffer->set_range(4, buffer->range_length() - 4);
+
+ // 2-byte APP1 + 2-byte size followed by kExifHeader
+ *tiffHdrOffset = 2 + 2 + sizeof(kExifHeader);
+ return true;
+ }
+
+ return false;
}
void MPEG4Writer::Track::addChunkOffset(off64_t offset) {
@@ -2014,7 +2038,7 @@
}
if (isExif) {
- mCdscRefs.value.push_back(mOwner->addItem_l({
+ mExifList.push_back(mOwner->addItem_l({
.itemType = "Exif",
.isPrimary = false,
.isHidden = false,
@@ -2117,7 +2141,16 @@
if (mImageItemId > 0) {
mOwner->addRefs_l(mImageItemId, mDimgRefs);
- mOwner->addRefs_l(mImageItemId, mCdscRefs);
+
+ if (!mExifList.empty()) {
+ // The "cdsc" ref is from the metadata/exif item to the image item.
+ // So the refs all contain the image item.
+ ItemRefs cdscRefs("cdsc");
+ cdscRefs.value.push_back(mImageItemId);
+ for (uint16_t exifItem : mExifList) {
+ mOwner->addRefs_l(exifItem, cdscRefs);
+ }
+ }
}
}
@@ -2269,14 +2302,16 @@
while (!chunk->mSamples.empty()) {
List<MediaBuffer *>::iterator it = chunk->mSamples.begin();
- int32_t isExif;
- if (!(*it)->meta_data().findInt32(kKeyIsExif, &isExif)) {
- isExif = 0;
+ uint32_t tiffHdrOffset;
+ if (!(*it)->meta_data().findInt32(
+ kKeyExifTiffOffset, (int32_t*)&tiffHdrOffset)) {
+ tiffHdrOffset = 0;
}
+ bool isExif = (tiffHdrOffset > 0);
bool usePrefix = chunk->mTrack->usePrefix() && !isExif;
size_t bytesWritten;
- off64_t offset = addSample_l(*it, usePrefix, isExif, &bytesWritten);
+ off64_t offset = addSample_l(*it, usePrefix, tiffHdrOffset, &bytesWritten);
if (chunk->mTrack->isHeic()) {
chunk->mTrack->addItemOffsetAndSize(offset, bytesWritten, isExif);
@@ -3002,10 +3037,11 @@
}
bool isExif = false;
+ uint32_t tiffHdrOffset = 0;
int32_t isMuxerData;
if (buffer->meta_data().findInt32(kKeyIsMuxerData, &isMuxerData) && isMuxerData) {
// We only support one type of muxer data, which is Exif data block.
- isExif = isExifData(buffer);
+ isExif = isExifData(buffer, &tiffHdrOffset);
if (!isExif) {
ALOGW("Ignoring bad Exif data block");
buffer->release();
@@ -3027,7 +3063,7 @@
buffer = NULL;
if (isExif) {
- copy->meta_data().setInt32(kKeyIsExif, 1);
+ copy->meta_data().setInt32(kKeyExifTiffOffset, tiffHdrOffset);
}
bool usePrefix = this->usePrefix() && !isExif;
@@ -3300,7 +3336,8 @@
}
if (!hasMultipleTracks) {
size_t bytesWritten;
- off64_t offset = mOwner->addSample_l(copy, usePrefix, isExif, &bytesWritten);
+ off64_t offset = mOwner->addSample_l(
+ copy, usePrefix, tiffHdrOffset, &bytesWritten);
if (mIsHeic) {
addItemOffsetAndSize(offset, bytesWritten, isExif);
diff --git a/media/libstagefright/MediaCodec.cpp b/media/libstagefright/MediaCodec.cpp
index 7816fae..c7da7c7 100644
--- a/media/libstagefright/MediaCodec.cpp
+++ b/media/libstagefright/MediaCodec.cpp
@@ -862,9 +862,9 @@
//static
sp<CodecBase> MediaCodec::GetCodecBase(const AString &name, const char *owner) {
if (owner) {
- if (strncmp(owner, "default", 8) == 0) {
+ if (strcmp(owner, "default") == 0) {
return new ACodec;
- } else if (strncmp(owner, "codec2", 7) == 0) {
+ } else if (strncmp(owner, "codec2", 6) == 0) {
return CreateCCodec();
}
}
@@ -911,10 +911,10 @@
continue;
}
mCodecInfo = mcl->getCodecInfo(codecIdx);
- Vector<AString> mimes;
- mCodecInfo->getSupportedMimes(&mimes);
- for (size_t i = 0; i < mimes.size(); i++) {
- if (mimes[i].startsWith("video/")) {
+ Vector<AString> mediaTypes;
+ mCodecInfo->getSupportedMediaTypes(&mediaTypes);
+ for (size_t i = 0; i < mediaTypes.size(); i++) {
+ if (mediaTypes[i].startsWith("video/")) {
mIsVideo = true;
break;
}
diff --git a/media/libstagefright/MediaCodecList.cpp b/media/libstagefright/MediaCodecList.cpp
index eaff283..93478e9 100644
--- a/media/libstagefright/MediaCodecList.cpp
+++ b/media/libstagefright/MediaCodecList.cpp
@@ -215,13 +215,9 @@
mCodecInfos.begin(),
mCodecInfos.end(),
[](const sp<MediaCodecInfo> &info1, const sp<MediaCodecInfo> &info2) {
- if (info2 == nullptr) {
- return false;
- } else if (info1 == nullptr) {
- return true;
- } else {
- return info1->rank() < info2->rank();
- }
+ // null is lowest
+ return info1 == nullptr
+ || (info2 != nullptr && info1->getRank() < info2->getRank());
});
}
diff --git a/media/libstagefright/MediaCodecListOverrides.cpp b/media/libstagefright/MediaCodecListOverrides.cpp
index cac53f4..dd7c3e6 100644
--- a/media/libstagefright/MediaCodecListOverrides.cpp
+++ b/media/libstagefright/MediaCodecListOverrides.cpp
@@ -228,18 +228,18 @@
continue;
}
- Vector<AString> mimes;
- info->getSupportedMimes(&mimes);
- for (size_t i = 0; i < mimes.size(); ++i) {
+ Vector<AString> mediaTypes;
+ info->getSupportedMediaTypes(&mediaTypes);
+ for (size_t i = 0; i < mediaTypes.size(); ++i) {
const sp<MediaCodecInfo::Capabilities> &caps =
- info->getCapabilitiesFor(mimes[i].c_str());
+ info->getCapabilitiesFor(mediaTypes[i].c_str());
if (!forceToMeasure &&
(caps->getDetails()->contains("max-supported-instances") ||
caps->getDetails()->contains("max-concurrent-instances"))) {
continue;
}
- size_t max = doProfileCodecs(info->isEncoder(), name, mimes[i], caps);
+ size_t max = doProfileCodecs(info->isEncoder(), name, mediaTypes[i], caps);
if (max > 0) {
CodecSettings settings;
char maxStr[32];
@@ -248,7 +248,7 @@
AString key = name;
key.append(" ");
- key.append(mimes[i]);
+ key.append(mediaTypes[i]);
if (info->isEncoder()) {
encoder_results->add(key, settings);
diff --git a/media/libstagefright/MediaExtractor.cpp b/media/libstagefright/MediaExtractor.cpp
index 9511931..4ed3382 100644
--- a/media/libstagefright/MediaExtractor.cpp
+++ b/media/libstagefright/MediaExtractor.cpp
@@ -57,7 +57,7 @@
}
MediaTrack *MediaExtractorCUnwrapper::getTrack(size_t index) {
- return new MediaTrackCUnwrapper(plugin->getTrack(plugin->data, index));
+ return MediaTrackCUnwrapper::create(plugin->getTrack(plugin->data, index));
}
status_t MediaExtractorCUnwrapper::getTrackMetaData(
diff --git a/media/libstagefright/MediaTrack.cpp b/media/libstagefright/MediaTrack.cpp
index 036e79d..89c9b25 100644
--- a/media/libstagefright/MediaTrack.cpp
+++ b/media/libstagefright/MediaTrack.cpp
@@ -65,6 +65,13 @@
bufferGroup = nullptr;
}
+MediaTrackCUnwrapper *MediaTrackCUnwrapper::create(CMediaTrack *cmediatrack) {
+ if (cmediatrack == nullptr) {
+ return nullptr;
+ }
+ return new MediaTrackCUnwrapper(cmediatrack);
+}
+
MediaTrackCUnwrapper::~MediaTrackCUnwrapper() {
wrapper->free(wrapper->data);
free(wrapper);
diff --git a/media/libstagefright/OggWriter.cpp b/media/libstagefright/OggWriter.cpp
index ad55c56..5c13983 100644
--- a/media/libstagefright/OggWriter.cpp
+++ b/media/libstagefright/OggWriter.cpp
@@ -30,7 +30,7 @@
#include <media/stagefright/MetaData.h>
#include <media/stagefright/OggWriter.h>
#include <media/stagefright/foundation/ADebug.h>
-#include "OpusHeader.h"
+#include <media/stagefright/foundation/OpusHeader.h>
extern "C" {
#include <ogg/ogg.h>
@@ -114,30 +114,17 @@
}
mSampleRate = sampleRate;
-
- OpusHeader header;
- header.channels = nChannels;
- header.num_streams = nChannels;
- header.num_coupled = 0;
- header.channel_mapping = ((nChannels > 8) ? 255 : (nChannels > 2));
- header.gain_db = 0;
- header.skip_samples = 0;
-
- // headers are 21-bytes + something driven by channel count
- // expect numbers in the low 30's here. WriteOpusHeader() will tell us
- // if things are bad.
- unsigned char header_data[100];
- ogg_packet op;
- ogg_page og;
-
- const int packet_size = WriteOpusHeader(header, mSampleRate, (uint8_t*)header_data,
- sizeof(header_data));
-
- if (packet_size < 0) {
- ALOGE("opus header writing failed");
+ uint32_t type;
+ const void *header_data;
+ size_t packet_size;
+ if (!source->getFormat()->findData(kKeyOpusHeader, &type, &header_data, &packet_size)) {
+ ALOGE("opus header not found");
return UNKNOWN_ERROR;
}
- op.packet = header_data;
+
+ ogg_packet op;
+ ogg_page og;
+ op.packet = (unsigned char *)header_data;
op.bytes = packet_size;
op.b_o_s = 1;
op.e_o_s = 0;
diff --git a/media/libstagefright/OmxInfoBuilder.cpp b/media/libstagefright/OmxInfoBuilder.cpp
index 96b896b..382c947 100644
--- a/media/libstagefright/OmxInfoBuilder.cpp
+++ b/media/libstagefright/OmxInfoBuilder.cpp
@@ -57,14 +57,9 @@
}
status_t queryCapabilities(
- const IOmxStore::NodeInfo& node, const char* mime, bool isEncoder,
+ const IOmxStore::NodeInfo& node, const char* mediaType, bool isEncoder,
MediaCodecInfo::CapabilitiesWriter* caps) {
sp<ACodec> codec = new ACodec();
- status_t err = codec->queryCapabilities(
- node.owner.c_str(), node.name.c_str(), mime, isEncoder, caps);
- if (err != OK) {
- return err;
- }
for (const auto& attribute : node.attributes) {
// All features have an int32 value except
// "feature-bitrate-modes", which has a string value.
@@ -81,6 +76,12 @@
attribute.key.c_str(), attribute.value.c_str());
}
}
+ // query capabilities may remove capabilities that are not actually supported by the codec
+ status_t err = codec->queryCapabilities(
+ node.owner.c_str(), node.name.c_str(), mediaType, isEncoder, caps);
+ if (err != OK) {
+ return err;
+ }
return OK;
}
@@ -163,7 +164,10 @@
info = c2i->second.get();
info->setName(nodeName.c_str());
info->setOwner(node.owner.c_str());
- info->setEncoder(isEncoder);
+ info->setAttributes(
+ // all OMX codecs are vendor codecs (in the vendor partition), but
+ // treat OMX.google codecs as non-hardware-accelerated and non-vendor
+ (isEncoder ? MediaCodecInfo::kFlagIsEncoder : 0));
info->setRank(defaultRank);
} else {
// The node has been seen before. Simply retrieve the
@@ -180,7 +184,19 @@
info = c2i->second.get();
info->setName(nodeName.c_str());
info->setOwner(node.owner.c_str());
- info->setEncoder(isEncoder);
+ typename std::underlying_type<MediaCodecInfo::Attributes>::type attrs =
+ MediaCodecInfo::kFlagIsVendor;
+ if (isEncoder) {
+ attrs |= MediaCodecInfo::kFlagIsEncoder;
+ }
+ if (std::count_if(
+ node.attributes.begin(), node.attributes.end(),
+ [](const IOmxStore::Attribute &i) -> bool {
+ return i.key == "attribute::software-codec";
+ })) {
+ attrs |= MediaCodecInfo::kFlagIsHardwareAccelerated;
+ }
+ info->setAttributes(attrs);
info->setRank(defaultRank);
} else {
// If preferPlatformNodes is true, this node must be
@@ -195,12 +211,12 @@
}
}
std::unique_ptr<MediaCodecInfo::CapabilitiesWriter> caps =
- info->addMime(typeName.c_str());
+ info->addMediaType(typeName.c_str());
if (queryCapabilities(
node, typeName.c_str(), isEncoder, caps.get()) != OK) {
- ALOGW("Fail to add mime %s to codec %s",
+ ALOGW("Fail to add media type %s to codec %s",
typeName.c_str(), nodeName.c_str());
- info->removeMime(typeName.c_str());
+ info->removeMediaType(typeName.c_str());
}
}
@@ -219,7 +235,18 @@
info = c2i->second.get();
info->setName(nodeName.c_str());
info->setOwner(node->owner.c_str());
- info->setEncoder(isEncoder);
+ typename std::underlying_type<MediaCodecInfo::Attributes>::type attrs =
+ MediaCodecInfo::kFlagIsVendor;
+ if (isEncoder) {
+ attrs |= MediaCodecInfo::kFlagIsEncoder;
+ }
+ if (std::count_if(
+ node->attributes.begin(), node->attributes.end(),
+ [](const IOmxStore::Attribute &i) -> bool {
+ return i.key == "attribute::software-codec";
+ })) {
+ attrs |= MediaCodecInfo::kFlagIsHardwareAccelerated;
+ }
info->setRank(defaultRank);
} else {
// The node has been seen before. Simply retrieve the
@@ -227,13 +254,13 @@
info = c2i->second.get();
}
std::unique_ptr<MediaCodecInfo::CapabilitiesWriter> caps =
- info->addMime(typeName.c_str());
+ info->addMediaType(typeName.c_str());
if (queryCapabilities(
*node, typeName.c_str(), isEncoder, caps.get()) != OK) {
- ALOGW("Fail to add mime %s to codec %s "
+ ALOGW("Fail to add media type %s to codec %s "
"after software codecs",
typeName.c_str(), nodeName.c_str());
- info->removeMime(typeName.c_str());
+ info->removeMediaType(typeName.c_str());
}
}
}
diff --git a/media/libstagefright/Utils.cpp b/media/libstagefright/Utils.cpp
index 49e485a..2e7da01 100644
--- a/media/libstagefright/Utils.cpp
+++ b/media/libstagefright/Utils.cpp
@@ -37,6 +37,7 @@
#include <media/stagefright/foundation/ALookup.h>
#include <media/stagefright/foundation/AMessage.h>
#include <media/stagefright/foundation/ByteUtils.h>
+#include <media/stagefright/foundation/OpusHeader.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/MediaDefs.h>
#include <media/AudioSystem.h>
@@ -1745,12 +1746,34 @@
} else if (mime == MEDIA_MIMETYPE_VIDEO_VP9) {
meta->setData(kKeyVp9CodecPrivate, 0, csd0->data(), csd0->size());
} else if (mime == MEDIA_MIMETYPE_AUDIO_OPUS) {
- meta->setData(kKeyOpusHeader, 0, csd0->data(), csd0->size());
+ size_t opusHeadSize = csd0->size();
+ size_t codecDelayBufSize = 0;
+ size_t seekPreRollBufSize = 0;
+ void *opusHeadBuf = csd0->data();
+ void *codecDelayBuf = NULL;
+ void *seekPreRollBuf = NULL;
if (msg->findBuffer("csd-1", &csd1)) {
- meta->setData(kKeyOpusCodecDelay, 0, csd1->data(), csd1->size());
+ codecDelayBufSize = csd1->size();
+ codecDelayBuf = csd1->data();
}
if (msg->findBuffer("csd-2", &csd2)) {
- meta->setData(kKeyOpusSeekPreRoll, 0, csd2->data(), csd2->size());
+ seekPreRollBufSize = csd2->size();
+ seekPreRollBuf = csd2->data();
+ }
+ /* Extract codec delay and seek pre roll from csd-0,
+ * if csd-1 and csd-2 are not present */
+ if (!codecDelayBuf && !seekPreRollBuf) {
+ GetOpusHeaderBuffers(csd0->data(), csd0->size(), &opusHeadBuf,
+ &opusHeadSize, &codecDelayBuf,
+ &codecDelayBufSize, &seekPreRollBuf,
+ &seekPreRollBufSize);
+ }
+ meta->setData(kKeyOpusHeader, 0, opusHeadBuf, opusHeadSize);
+ if (codecDelayBuf) {
+ meta->setData(kKeyOpusCodecDelay, 0, codecDelayBuf, codecDelayBufSize);
+ }
+ if (seekPreRollBuf) {
+ meta->setData(kKeyOpusSeekPreRoll, 0, seekPreRollBuf, seekPreRollBufSize);
}
} else if (mime == MEDIA_MIMETYPE_AUDIO_VORBIS) {
meta->setData(kKeyVorbisInfo, 0, csd0->data(), csd0->size());
diff --git a/media/libstagefright/data/media_codecs_google_c2_audio.xml b/media/libstagefright/data/media_codecs_google_c2_audio.xml
index 0b554a2..88cd08d 100644
--- a/media/libstagefright/data/media_codecs_google_c2_audio.xml
+++ b/media/libstagefright/data/media_codecs_google_c2_audio.xml
@@ -93,5 +93,12 @@
<Limit name="complexity" range="0-8" default="5" />
<Feature name="bitrate-modes" value="CQ" />
</MediaCodec>
+ <MediaCodec name="c2.android.opus.encoder" type="audio/opus">
+ <Limit name="channel-count" max="2" />
+ <Limit name="sample-rate" ranges="8000,12000,16000,24000,48000" />
+ <Limit name="bitrate" range="500-512000" />
+ <Limit name="complexity" range="0-10" default="5" />
+ <Feature name="bitrate-modes" value="CQ" />
+ </MediaCodec>
</Encoders>
</Included>
diff --git a/media/libstagefright/data/media_codecs_google_c2_telephony.xml b/media/libstagefright/data/media_codecs_google_c2_telephony.xml
new file mode 100644
index 0000000..d1055b3
--- /dev/null
+++ b/media/libstagefright/data/media_codecs_google_c2_telephony.xml
@@ -0,0 +1,25 @@
+<?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.
+-->
+
+<Included>
+ <Decoders>
+ <MediaCodec name="c2.android.gsm.decoder" type="audio/gsm">
+ <Limit name="channel-count" max="1" />
+ <Limit name="sample-rate" ranges="8000" />
+ <Limit name="bitrate" range="13000" />
+ </MediaCodec>
+ </Decoders>
+</Included>
diff --git a/media/libstagefright/data/media_codecs_google_c2_tv.xml b/media/libstagefright/data/media_codecs_google_c2_tv.xml
new file mode 100644
index 0000000..fa082c7
--- /dev/null
+++ b/media/libstagefright/data/media_codecs_google_c2_tv.xml
@@ -0,0 +1,29 @@
+<?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.
+-->
+
+<Included>
+ <Decoders>
+ <MediaCodec name="c2.android.mpeg2.decoder" type="video/mpeg2">
+ <!-- profiles and levels: ProfileMain : LevelHL -->
+ <Limit name="size" min="16x16" max="1920x1088" />
+ <Limit name="alignment" value="2x2" />
+ <Limit name="block-size" value="16x16" />
+ <Limit name="blocks-per-second" range="1-244800" />
+ <Limit name="bitrate" range="1-20000000" />
+ <Feature name="adaptive-playback" />
+ </MediaCodec>
+ </Decoders>
+</Included>
diff --git a/media/libstagefright/data/media_codecs_google_c2_video.xml b/media/libstagefright/data/media_codecs_google_c2_video.xml
index adb45b3..c49789e 100644
--- a/media/libstagefright/data/media_codecs_google_c2_video.xml
+++ b/media/libstagefright/data/media_codecs_google_c2_video.xml
@@ -71,6 +71,15 @@
<Limit name="bitrate" range="1-40000000" />
<Feature name="adaptive-playback" />
</MediaCodec>
+ <MediaCodec name="c2.android.av1.decoder" type="video/av01">
+ <Limit name="size" min="96x96" max="1920x1080" />
+ <Limit name="alignment" value="2x2" />
+ <Limit name="block-size" value="16x16" />
+ <Limit name="blocks-per-second" min="24" max="2073600" />
+ <Limit name="bitrate" range="1-120000000" />
+ <Limit name="frame-rate" range="1-60" />
+ <Feature name="adaptive-playback" />
+ </MediaCodec>
</Decoders>
<Encoders>
diff --git a/media/libstagefright/foundation/Android.bp b/media/libstagefright/foundation/Android.bp
index dd1d904..533cd72 100644
--- a/media/libstagefright/foundation/Android.bp
+++ b/media/libstagefright/foundation/Android.bp
@@ -72,6 +72,7 @@
"MediaKeys.cpp",
"MetaData.cpp",
"MetaDataBase.cpp",
+ "OpusHeader.cpp",
"avc_utils.cpp",
"base64.cpp",
"hexdump.cpp",
diff --git a/media/libstagefright/opus/OpusHeader.cpp b/media/libstagefright/foundation/OpusHeader.cpp
similarity index 69%
rename from media/libstagefright/opus/OpusHeader.cpp
rename to media/libstagefright/foundation/OpusHeader.cpp
index e4a460c..9faede1 100644
--- a/media/libstagefright/opus/OpusHeader.cpp
+++ b/media/libstagefright/foundation/OpusHeader.cpp
@@ -16,7 +16,7 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "SoftOpus"
-
+#include <algorithm>
#include <cstring>
#include <stdint.h>
@@ -43,9 +43,6 @@
{0, 6, 1, 2, 3, 4, 5, 7},
};
-// Opus always has a 48kHz output rate. This is true for all Opus, not just this
-// implementation.
-constexpr int kRate = 48000;
// Size of the Opus header excluding optional mapping information.
constexpr size_t kOpusHeaderSize = 19;
// Offset to magic string that starts Opus header.
@@ -76,15 +73,12 @@
constexpr size_t kOpusHeaderNumCoupledStreamsOffset = 20;
// Offset to the stream to channel mapping in the Opus header.
constexpr size_t kOpusHeaderStreamMapOffset = 21;
-// Maximum packet size used in Xiph's opusdec.
-constexpr int kMaxOpusOutputPacketSizeSamples = 960 * 6;
// Default audio output channel layout. Used to initialize |stream_map| in
// OpusHeader, and passed to opus_multistream_decoder_create() when the header
// does not contain mapping information. The values are valid only for mono and
// stereo output: Opus streams with more than 2 channels require a stream map.
constexpr int kMaxChannelsWithDefaultLayout = 2;
-constexpr uint8_t kDefaultOpusChannelLayout[kMaxChannelsWithDefaultLayout] = {0, 1};
static uint16_t ReadLE16(const uint8_t* data, size_t data_size, uint32_t read_offset) {
// check whether the 2nd byte is within the buffer
@@ -182,4 +176,88 @@
}
}
+int WriteOpusHeaders(const OpusHeader &header, int inputSampleRate,
+ uint8_t* output, size_t outputSize, uint64_t codecDelay,
+ uint64_t seekPreRoll) {
+ if (outputSize < AOPUS_UNIFIED_CSD_MINSIZE) {
+ ALOGD("Buffer not large enough to hold unified OPUS CSD");
+ return -1;
+ }
+
+ int headerLen = WriteOpusHeader(header, inputSampleRate, output,
+ outputSize);
+ if (headerLen < 0) {
+ ALOGD("WriteOpusHeader failed");
+ return -1;
+ }
+ if (headerLen >= (outputSize - 2 * AOPUS_TOTAL_CSD_SIZE)) {
+ ALOGD("Buffer not large enough to hold codec delay and seek pre roll");
+ return -1;
+ }
+
+ uint64_t length = AOPUS_LENGTH;
+
+ /*
+ Following is the CSD syntax for signalling codec delay and
+ seek pre-roll which is to be appended after OpusHeader
+
+ Marker (8 bytes) | Length (8 bytes) | Samples (8 bytes)
+
+ Markers supported:
+ AOPUSDLY - Signals Codec Delay
+ AOPUSPRL - Signals seek pre roll
+
+ Length should be 8.
+ */
+
+ // Add codec delay
+ memcpy(output + headerLen, AOPUS_CSD_CODEC_DELAY_MARKER, AOPUS_MARKER_SIZE);
+ headerLen += AOPUS_MARKER_SIZE;
+ memcpy(output + headerLen, &length, AOPUS_LENGTH_SIZE);
+ headerLen += AOPUS_LENGTH_SIZE;
+ memcpy(output + headerLen, &codecDelay, AOPUS_CSD_SIZE);
+ headerLen += AOPUS_CSD_SIZE;
+
+ // Add skip pre roll
+ memcpy(output + headerLen, AOPUS_CSD_SEEK_PREROLL_MARKER, AOPUS_MARKER_SIZE);
+ headerLen += AOPUS_MARKER_SIZE;
+ memcpy(output + headerLen, &length, AOPUS_LENGTH_SIZE);
+ headerLen += AOPUS_LENGTH_SIZE;
+ memcpy(output + headerLen, &seekPreRoll, AOPUS_CSD_SIZE);
+ headerLen += AOPUS_CSD_SIZE;
+
+ return headerLen;
+}
+
+void GetOpusHeaderBuffers(const uint8_t *data, size_t data_size,
+ void **opusHeadBuf, size_t *opusHeadSize,
+ void **codecDelayBuf, size_t *codecDelaySize,
+ void **seekPreRollBuf, size_t *seekPreRollSize) {
+ *codecDelayBuf = NULL;
+ *codecDelaySize = 0;
+ *seekPreRollBuf = NULL;
+ *seekPreRollSize = 0;
+ *opusHeadBuf = (void *)data;
+ *opusHeadSize = data_size;
+ if (data_size >= AOPUS_UNIFIED_CSD_MINSIZE) {
+ size_t i = 0;
+ while (i < data_size - AOPUS_TOTAL_CSD_SIZE) {
+ uint8_t *csdBuf = (uint8_t *)data + i;
+ if (!memcmp(csdBuf, AOPUS_CSD_CODEC_DELAY_MARKER, AOPUS_MARKER_SIZE)) {
+ *opusHeadSize = std::min(*opusHeadSize, i);
+ *codecDelayBuf = csdBuf + AOPUS_MARKER_SIZE + AOPUS_LENGTH_SIZE;
+ *codecDelaySize = AOPUS_CSD_SIZE;
+ i += AOPUS_TOTAL_CSD_SIZE;
+ } else if (!memcmp(csdBuf, AOPUS_CSD_SEEK_PREROLL_MARKER, AOPUS_MARKER_SIZE)) {
+ *opusHeadSize = std::min(*opusHeadSize, i);
+ *seekPreRollBuf = csdBuf + AOPUS_MARKER_SIZE + AOPUS_LENGTH_SIZE;
+ *seekPreRollSize = AOPUS_CSD_SIZE;
+ i += AOPUS_TOTAL_CSD_SIZE;
+ } else {
+ i++;
+ }
+ }
+ }
+}
+
} // namespace android
diff --git a/media/libstagefright/foundation/include/media/stagefright/foundation/OpusHeader.h b/media/libstagefright/foundation/include/media/stagefright/foundation/OpusHeader.h
new file mode 100644
index 0000000..9bffccb
--- /dev/null
+++ b/media/libstagefright/foundation/include/media/stagefright/foundation/OpusHeader.h
@@ -0,0 +1,66 @@
+/*
+ * 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.
+ */
+
+/*
+ * The Opus specification is part of IETF RFC 6716:
+ * http://tools.ietf.org/html/rfc6716
+ */
+
+#ifndef OPUS_HEADER_H_
+#define OPUS_HEADER_H_
+
+namespace android {
+
+/* Constants used for delimiting Opus CSD */
+#define AOPUS_CSD_CODEC_DELAY_MARKER "AOPUSDLY"
+#define AOPUS_CSD_SEEK_PREROLL_MARKER "AOPUSPRL"
+#define AOPUS_CSD_SIZE 8
+#define AOPUS_LENGTH 8
+#define AOPUS_MARKER_SIZE 8
+#define AOPUS_LENGTH_SIZE 8
+#define AOPUS_TOTAL_CSD_SIZE \
+ ((AOPUS_MARKER_SIZE) + (AOPUS_LENGTH_SIZE) + (AOPUS_CSD_SIZE))
+#define AOPUS_CSD0_MINSIZE 19
+#define AOPUS_UNIFIED_CSD_MINSIZE \
+ ((AOPUS_CSD0_MINSIZE) + 2 * (AOPUS_TOTAL_CSD_SIZE))
+
+/* CSD0 at max can be 22 bytes + max number of channels (255) */
+#define AOPUS_CSD0_MAXSIZE 277
+#define AOPUS_UNIFIED_CSD_MAXSIZE \
+ ((AOPUS_CSD0_MAXSIZE) + 2 * (AOPUS_TOTAL_CSD_SIZE))
+
+struct OpusHeader {
+ int channels;
+ int channel_mapping;
+ int num_streams;
+ int num_coupled;
+ int16_t gain_db;
+ int skip_samples;
+ uint8_t stream_map[8];
+};
+
+bool ParseOpusHeader(const uint8_t* data, size_t data_size, OpusHeader* header);
+int WriteOpusHeader(const OpusHeader &header, int input_sample_rate, uint8_t* output, size_t output_size);
+void GetOpusHeaderBuffers(const uint8_t *data, size_t data_size,
+ void **opusHeadBuf, size_t *opusHeadSize,
+ void **codecDelayBuf, size_t *codecDelaySize,
+ void **seekPreRollBuf, size_t *seekPreRollSize);
+int WriteOpusHeaders(const OpusHeader &header, int inputSampleRate,
+ uint8_t* output, size_t outputSize, uint64_t codecDelay,
+ uint64_t seekPreRoll);
+} // namespace android
+
+#endif // OPUS_HEADER_H_
diff --git a/media/libstagefright/include/media/stagefright/MPEG4Writer.h b/media/libstagefright/include/media/stagefright/MPEG4Writer.h
index 1abef8c..803155d 100644
--- a/media/libstagefright/include/media/stagefright/MPEG4Writer.h
+++ b/media/libstagefright/include/media/stagefright/MPEG4Writer.h
@@ -257,7 +257,9 @@
void initInternal(int fd, bool isFirstSession);
// Acquire lock before calling these methods
- off64_t addSample_l(MediaBuffer *buffer, bool usePrefix, bool isExif, size_t *bytesWritten);
+ off64_t addSample_l(
+ MediaBuffer *buffer, bool usePrefix,
+ uint32_t tiffHdrOffset, size_t *bytesWritten);
void addLengthPrefixedSample_l(MediaBuffer *buffer);
void addMultipleLengthPrefixedSamples_l(MediaBuffer *buffer);
uint16_t addProperty_l(const ItemProperty &);
diff --git a/media/libstagefright/include/media/stagefright/MetaDataBase.h b/media/libstagefright/include/media/stagefright/MetaDataBase.h
index b99c14c..2910bd3 100644
--- a/media/libstagefright/include/media/stagefright/MetaDataBase.h
+++ b/media/libstagefright/include/media/stagefright/MetaDataBase.h
@@ -221,7 +221,8 @@
kKeyFrameCount = 'nfrm', // int32_t, total number of frame in video track
kKeyExifOffset = 'exof', // int64_t, Exif data offset
kKeyExifSize = 'exsz', // int64_t, Exif data size
- kKeyIsExif = 'exif', // bool (int32_t) buffer contains exif data block
+ kKeyExifTiffOffset = 'thdr', // int32_t, if > 0, buffer contains exif data block with
+ // tiff hdr at specified offset
kKeyPcmBigEndian = 'pcmb', // bool (int32_t)
// Key for ALAC Magic Cookie
diff --git a/media/libstagefright/omx/1.0/Omx.cpp b/media/libstagefright/omx/1.0/Omx.cpp
index 4e2d398..121bb1a 100644
--- a/media/libstagefright/omx/1.0/Omx.cpp
+++ b/media/libstagefright/omx/1.0/Omx.cpp
@@ -124,11 +124,11 @@
} else {
uint32_t quirks = 0;
for (const auto& quirk : codec->second.quirkSet) {
- if (quirk == "requires-allocate-on-input-ports") {
+ if (quirk == "quirk::requires-allocate-on-input-ports") {
quirks |= OMXNodeInstance::
kRequiresAllocateBufferOnInputPorts;
}
- if (quirk == "requires-allocate-on-output-ports") {
+ if (quirk == "quirk::requires-allocate-on-output-ports") {
quirks |= OMXNodeInstance::
kRequiresAllocateBufferOnOutputPorts;
}
diff --git a/media/libstagefright/opus/Android.bp b/media/libstagefright/opus/Android.bp
deleted file mode 100644
index c5086ec..0000000
--- a/media/libstagefright/opus/Android.bp
+++ /dev/null
@@ -1,21 +0,0 @@
-cc_library_shared {
- name: "libstagefright_opus_common",
- vendor_available: true,
-
- export_include_dirs: ["include"],
-
- srcs: ["OpusHeader.cpp"],
-
- shared_libs: ["liblog"],
-
- cflags: ["-Werror"],
-
- sanitize: {
- integer_overflow: true,
- cfi: true,
- diag: {
- integer_overflow: true,
- cfi: true,
- },
- },
-}
\ No newline at end of file
diff --git a/media/libstagefright/opus/include/OpusHeader.h b/media/libstagefright/opus/include/OpusHeader.h
deleted file mode 100644
index f9f79cd..0000000
--- a/media/libstagefright/opus/include/OpusHeader.h
+++ /dev/null
@@ -1,41 +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.
- */
-
-/*
- * The Opus specification is part of IETF RFC 6716:
- * http://tools.ietf.org/html/rfc6716
- */
-
-#ifndef OPUS_HEADER_H_
-#define OPUS_HEADER_H_
-
-namespace android {
-
-struct OpusHeader {
- int channels;
- int channel_mapping;
- int num_streams;
- int num_coupled;
- int16_t gain_db;
- int skip_samples;
- uint8_t stream_map[8];
-};
-
-bool ParseOpusHeader(const uint8_t* data, size_t data_size, OpusHeader* header);
-int WriteOpusHeader(const OpusHeader &header, int input_sample_rate, uint8_t* output, size_t output_size);
-} // namespace android
-
-#endif // OPUS_HEADER_H_
diff --git a/media/libstagefright/webm/Android.bp b/media/libstagefright/webm/Android.bp
index 1f840b7..64ecc2d 100644
--- a/media/libstagefright/webm/Android.bp
+++ b/media/libstagefright/webm/Android.bp
@@ -28,7 +28,6 @@
shared_libs: [
"libstagefright_foundation",
- "libstagefright_opus_common",
"libutils",
"liblog",
],
diff --git a/media/libstagefright/webm/WebmWriter.cpp b/media/libstagefright/webm/WebmWriter.cpp
index 7b4b23a..b0a303e 100644
--- a/media/libstagefright/webm/WebmWriter.cpp
+++ b/media/libstagefright/webm/WebmWriter.cpp
@@ -24,7 +24,7 @@
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/hexdump.h>
-#include <OpusHeader.h>
+#include <media/stagefright/foundation/OpusHeader.h>
#include <utils/Errors.h>
diff --git a/media/libstagefright/xmlparser/MediaCodecsXmlParser.cpp b/media/libstagefright/xmlparser/MediaCodecsXmlParser.cpp
index 2dec9fa..6e541ba 100644
--- a/media/libstagefright/xmlparser/MediaCodecsXmlParser.cpp
+++ b/media/libstagefright/xmlparser/MediaCodecsXmlParser.cpp
@@ -26,8 +26,9 @@
#include <sys/stat.h>
#include <expat.h>
-#include <cctype>
#include <algorithm>
+#include <cctype>
+#include <string>
namespace android {
@@ -326,8 +327,8 @@
case SECTION_DECODER:
case SECTION_ENCODER:
{
- if (strEq(name, "Quirk")) {
- (void)addQuirk(attrs);
+ if (strEq(name, "Quirk") || strEq(name, "Attribute")) {
+ (void)addQuirk(attrs, name);
} else if (strEq(name, "Type")) {
(void)addTypeFromAttributes(attrs,
(mCurrentSection == SECTION_ENCODER));
@@ -348,6 +349,8 @@
if (outside &&
(strEq(name, "Limit") || strEq(name, "Feature"))) {
ALOGW("ignoring %s specified outside of a Type", name);
+ } else if (strEq(name, "Alias")) {
+ (void)addAlias(attrs);
} else if (strEq(name, "Limit")) {
(void)addLimit(attrs);
} else if (strEq(name, "Feature")) {
@@ -579,7 +582,7 @@
return OK;
}
-status_t MediaCodecsXmlParser::addQuirk(const char **attrs) {
+status_t MediaCodecsXmlParser::addQuirk(const char **attrs, const char *tag) {
if (mCurrentCodec == mCodecMap.end()) {
return BAD_VALUE;
}
@@ -606,7 +609,12 @@
return BAD_VALUE;
}
- mCurrentCodec->second.quirkSet.emplace(name);
+ std::string tagString = tag;
+ std::transform(tagString.begin(), tagString.end(), tagString.begin(), ::tolower);
+ tagString.append("::");
+ tagString.append(name);
+ mCurrentCodec->second.quirkSet.emplace(tagString.c_str());
+ ALOGI("adding %s to %s", tagString.c_str(), mCurrentCodec->first.c_str());
return OK;
}
@@ -760,6 +768,7 @@
strEq(a_name, "quality") ||
strEq(a_name, "size") ||
strEq(a_name, "measured-blocks-per-second") ||
+ strHasPrefix(a_name, "performance-point-") ||
strHasPrefix(a_name, "measured-frame-rate-")) {
// "range" is specified in exactly one of the following forms:
// 1) min-max
@@ -964,6 +973,34 @@
return OK;
}
+status_t MediaCodecsXmlParser::addAlias(const char **attrs) {
+ size_t i = 0;
+ const char *name = nullptr;
+
+ while (attrs[i] != nullptr) {
+ if (strEq(attrs[i], "name")) {
+ if (attrs[++i] == nullptr) {
+ ALOGE("addAlias: name is null");
+ return BAD_VALUE;
+ }
+ name = attrs[i];
+ } else {
+ ALOGE("addAlias: unrecognized attribute: %s", attrs[i]);
+ return BAD_VALUE;
+ }
+ ++i;
+ }
+
+ // Every feature must have a name.
+ if (name == nullptr) {
+ ALOGE("alias with no 'name' attribute");
+ return BAD_VALUE;
+ }
+
+ mCurrentCodec->second.aliases.emplace_back(name);
+ return OK;
+}
+
const MediaCodecsXmlParser::AttributeMap&
MediaCodecsXmlParser::getServiceAttributeMap() const {
return mServiceAttributeMap;
@@ -1041,11 +1078,18 @@
NodeInfo nodeInfo;
nodeInfo.name = codecName;
+ // NOTE: no aliases are exposed in role info
+ // attribute quirks are exposed as node attributes
nodeInfo.attributeList.reserve(typeAttributeMap.size());
for (const auto& attribute : typeAttributeMap) {
nodeInfo.attributeList.push_back(
Attribute{attribute.first, attribute.second});
}
+ for (const std::string &quirk : codec.second.quirkSet) {
+ if (strHasPrefix(quirk.c_str(), "attribute::")) {
+ nodeInfo.attributeList.push_back(Attribute{quirk, "present"});
+ }
+ }
nodeList->insert(std::make_pair(
std::move(order), std::move(nodeInfo)));
}
diff --git a/media/libstagefright/xmlparser/include/media/stagefright/xmlparser/MediaCodecsXmlParser.h b/media/libstagefright/xmlparser/include/media/stagefright/xmlparser/MediaCodecsXmlParser.h
index cc69e52..fd949da 100644
--- a/media/libstagefright/xmlparser/include/media/stagefright/xmlparser/MediaCodecsXmlParser.h
+++ b/media/libstagefright/xmlparser/include/media/stagefright/xmlparser/MediaCodecsXmlParser.h
@@ -65,6 +65,7 @@
size_t order; ///< Order of appearance in the file (starting from 0)
QuirkSet quirkSet; ///< Set of quirks requested by this codec
TypeMap typeMap; ///< Map of types supported by this codec
+ std::vector<std::string> aliases; ///< Name aliases for this codec
};
typedef std::pair<std::string, CodecProperties> Codec;
@@ -76,6 +77,7 @@
struct NodeInfo {
std::string name;
std::vector<Attribute> attributeList;
+ // note: aliases are not exposed here as they are not part of the role map
};
/**
@@ -171,8 +173,9 @@
void addMediaCodec(bool encoder, const char *name,
const char *type = nullptr);
- status_t addQuirk(const char **attrs);
+ status_t addQuirk(const char **attrs, const char *tag);
status_t addTypeFromAttributes(const char **attrs, bool encoder);
+ status_t addAlias(const char **attrs);
status_t addLimit(const char **attrs);
status_t addFeature(const char **attrs);
void addType(const char *name);
diff --git a/media/mediaserver/mediaserver.rc b/media/mediaserver/mediaserver.rc
index f6c325c..8cfcd79 100644
--- a/media/mediaserver/mediaserver.rc
+++ b/media/mediaserver/mediaserver.rc
@@ -2,5 +2,7 @@
class main
user media
group audio camera inet net_bt net_bt_admin net_bw_acct drmrpc mediadrm
+ # TODO(b/123275379): Remove updatable when http://aosp/878198 has landed
+ updatable
ioprio rt 4
writepid /dev/cpuset/foreground/tasks /dev/stune/foreground/tasks
diff --git a/media/ndk/NdkMediaDrm.cpp b/media/ndk/NdkMediaDrm.cpp
index 55afb33..9082f62 100644
--- a/media/ndk/NdkMediaDrm.cpp
+++ b/media/ndk/NdkMediaDrm.cpp
@@ -274,7 +274,7 @@
}
String8 mimeStr = mimeType ? String8(mimeType) : String8("");
- return drm->isCryptoSchemeSupported(uuid, mimeStr);
+ return drm->isCryptoSchemeSupported(uuid, mimeStr, DrmPlugin::kSecurityLevelUnknown);
}
EXPORT
diff --git a/media/ndk/NdkMediaExtractor.cpp b/media/ndk/NdkMediaExtractor.cpp
index 8296598..28e4f12 100644
--- a/media/ndk/NdkMediaExtractor.cpp
+++ b/media/ndk/NdkMediaExtractor.cpp
@@ -46,6 +46,18 @@
sp<ABuffer> mPsshBuf;
};
+sp<ABuffer> U32ArrayToSizeBuf(size_t numSubSamples, uint32_t *data) {
+ if (numSubSamples > SIZE_MAX / sizeof(size_t)) {
+ return NULL;
+ }
+ sp<ABuffer> sizebuf = new ABuffer(numSubSamples * sizeof(size_t));
+ size_t *sizes = (size_t *)sizebuf->data();
+ for (size_t i = 0; sizes != NULL && i < numSubSamples; i++) {
+ sizes[i] = data[i];
+ }
+ return sizebuf;
+}
+
extern "C" {
EXPORT
@@ -339,7 +351,7 @@
if (!meta->findData(kKeyEncryptedSizes, &type, &crypteddata, &cryptedsize)) {
return NULL;
}
- size_t numSubSamples = cryptedsize / sizeof(size_t);
+ size_t numSubSamples = cryptedsize / sizeof(uint32_t);
const void *cleardata;
size_t clearsize;
@@ -373,6 +385,16 @@
mode = CryptoPlugin::kMode_AES_CTR;
}
+ if (sizeof(uint32_t) != sizeof(size_t)) {
+ sp<ABuffer> clearbuf = U32ArrayToSizeBuf(numSubSamples, (uint32_t *)cleardata);
+ sp<ABuffer> cryptedbuf = U32ArrayToSizeBuf(numSubSamples, (uint32_t *)crypteddata);
+ cleardata = clearbuf == NULL ? NULL : clearbuf->data();
+ crypteddata = crypteddata == NULL ? NULL : cryptedbuf->data();
+ if(crypteddata == NULL || cleardata == NULL) {
+ return NULL;
+ }
+ }
+
return AMediaCodecCryptoInfo_new(
numSubSamples,
(uint8_t*) key,
diff --git a/packages/MediaComponents/apex/java/android/media/session/MediaSession.java b/packages/MediaComponents/apex/java/android/media/session/MediaSession.java
index 73e16a6..3cbeff9 100644
--- a/packages/MediaComponents/apex/java/android/media/session/MediaSession.java
+++ b/packages/MediaComponents/apex/java/android/media/session/MediaSession.java
@@ -1077,8 +1077,7 @@
private static RemoteUserInfo createRemoteUserInfo(String packageName, int pid, int uid,
ISessionControllerCallback caller) {
- return new RemoteUserInfo(packageName, pid, uid,
- caller != null ? caller.asBinder() : null);
+ return new RemoteUserInfo(packageName, pid, uid);
}
@Override
diff --git a/packages/MediaComponents/apex/java/android/service/media/MediaBrowserService.java b/packages/MediaComponents/apex/java/android/service/media/MediaBrowserService.java
index a66ec35..76c99b9 100644
--- a/packages/MediaComponents/apex/java/android/service/media/MediaBrowserService.java
+++ b/packages/MediaComponents/apex/java/android/service/media/MediaBrowserService.java
@@ -544,8 +544,7 @@
throw new IllegalStateException("This should be called inside of onGetRoot or"
+ " onLoadChildren or onLoadItem methods");
}
- return new RemoteUserInfo(mCurConnection.pkg, mCurConnection.pid, mCurConnection.uid,
- mCurConnection.callbacks.asBinder());
+ return new RemoteUserInfo(mCurConnection.pkg, mCurConnection.pid, mCurConnection.uid);
}
/**
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 26f76c0..0d6ef46 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -2379,7 +2379,8 @@
return BAD_VALUE;
}
- sp<ThreadBase> thread = openInput_l(module, input, config, *devices, address, source, flags);
+ sp<ThreadBase> thread = openInput_l(
+ module, input, config, *devices, address, source, flags, AUDIO_DEVICE_NONE, String8{});
if (thread != 0) {
// notify client processes of the new input creation
@@ -2395,7 +2396,9 @@
audio_devices_t devices,
const String8& address,
audio_source_t source,
- audio_input_flags_t flags)
+ audio_input_flags_t flags,
+ audio_devices_t outputDevice,
+ const String8& outputDeviceAddress)
{
AudioHwDevice *inHwDev = findSuitableHwDev_l(module, devices);
if (inHwDev == NULL) {
@@ -2424,7 +2427,8 @@
sp<DeviceHalInterface> inHwHal = inHwDev->hwDevice();
sp<StreamInHalInterface> inStream;
status_t status = inHwHal->openInputStream(
- *input, devices, &halconfig, flags, address.string(), source, &inStream);
+ *input, devices, &halconfig, flags, address.string(), source,
+ outputDevice, outputDeviceAddress, &inStream);
ALOGV("openInput_l() openInputStream returned input %p, devices %#x, SamplingRate %d"
", Format %#x, Channels %#x, flags %#x, status %d addr %s",
inStream.get(),
@@ -2447,7 +2451,8 @@
ALOGV("openInput_l() reopening with proposed sampling rate and channel mask");
inStream.clear();
status = inHwHal->openInputStream(
- *input, devices, &halconfig, flags, address.string(), source, &inStream);
+ *input, devices, &halconfig, flags, address.string(), source,
+ outputDevice, outputDeviceAddress, &inStream);
// FIXME log this new status; HAL should not propose any further changes
}
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 6c698f6..c1169d2 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -579,6 +579,10 @@
virtual binder::Status stop();
virtual binder::Status getActiveMicrophones(
std::vector<media::MicrophoneInfo>* activeMicrophones);
+ virtual binder::Status setMicrophoneDirection(
+ int /*audio_microphone_direction_t*/ direction);
+ virtual binder::Status setMicrophoneFieldDimension(float zoom);
+
private:
const sp<RecordThread::RecordTrack> mRecordTrack;
@@ -620,7 +624,9 @@
audio_devices_t device,
const String8& address,
audio_source_t source,
- audio_input_flags_t flags);
+ audio_input_flags_t flags,
+ audio_devices_t outputDevice,
+ const String8& outputDeviceAddress);
sp<ThreadBase> openOutput_l(audio_module_handle_t module,
audio_io_handle_t *output,
audio_config_t *config,
diff --git a/services/audioflinger/PatchPanel.cpp b/services/audioflinger/PatchPanel.cpp
index 63a9ec4..3381e4d 100644
--- a/services/audioflinger/PatchPanel.cpp
+++ b/services/audioflinger/PatchPanel.cpp
@@ -211,6 +211,8 @@
((patch->sinks[0].type == AUDIO_PORT_TYPE_DEVICE) &&
((patch->sinks[0].ext.device.hw_module != srcModule) ||
!audioHwDevice->supportsAudioPatches()))) {
+ audio_devices_t outputDevice = AUDIO_DEVICE_NONE;
+ String8 outputDeviceAddress;
if (patch->num_sources == 2) {
if (patch->sources[1].type != AUDIO_PORT_TYPE_MIX ||
(patch->num_sinks != 0 && patch->sinks[0].ext.device.hw_module !=
@@ -261,6 +263,8 @@
goto exit;
}
newPatch.mPlayback.setThread(reinterpret_cast<PlaybackThread*>(thread.get()));
+ outputDevice = device;
+ outputDeviceAddress = address;
}
audio_devices_t device = patch->sources[0].ext.device.type;
String8 address = String8(patch->sources[0].ext.device.address);
@@ -293,7 +297,9 @@
device,
address,
AUDIO_SOURCE_MIC,
- flags);
+ flags,
+ outputDevice,
+ outputDeviceAddress);
ALOGV("mAudioFlinger.openInput_l() returned %p inChannelMask %08x",
thread.get(), config.channel_mask);
if (thread == 0) {
diff --git a/services/audioflinger/RecordTracks.h b/services/audioflinger/RecordTracks.h
index 85f5456..32af7d5 100644
--- a/services/audioflinger/RecordTracks.h
+++ b/services/audioflinger/RecordTracks.h
@@ -71,6 +71,9 @@
status_t getActiveMicrophones(std::vector<media::MicrophoneInfo>* activeMicrophones);
+ status_t setMicrophoneDirection(audio_microphone_direction_t direction);
+ status_t setMicrophoneFieldDimension(float zoom);
+
static bool checkServerLatencySupported(
audio_format_t format, audio_input_flags_t flags) {
return audio_is_linear_pcm(format)
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 607d2d1..31a8c7d 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -7582,6 +7582,20 @@
return status;
}
+status_t AudioFlinger::RecordThread::setMicrophoneDirection(audio_microphone_direction_t direction)
+{
+ ALOGV("RecordThread::setMicrophoneDirection");
+ AutoMutex _l(mLock);
+ return mInput->stream->setMicrophoneDirection(direction);
+}
+
+status_t AudioFlinger::RecordThread::setMicrophoneFieldDimension(float zoom)
+{
+ ALOGV("RecordThread::setMicrophoneFieldDimension");
+ AutoMutex _l(mLock);
+ return mInput->stream->setMicrophoneFieldDimension(zoom);
+}
+
void AudioFlinger::RecordThread::updateMetadata_l()
{
if (mInput == nullptr || mInput->stream == nullptr ||
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 5d06773..aab7601 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -1545,6 +1545,9 @@
status_t getActiveMicrophones(std::vector<media::MicrophoneInfo>* activeMicrophones);
+ status_t setMicrophoneDirection(audio_microphone_direction_t direction);
+ status_t setMicrophoneFieldDimension(float zoom);
+
void updateMetadata_l() override;
bool fastTrackAvailable() const { return mFastTrackAvail; }
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 9a7f1f1..d23d19d 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -1710,6 +1710,18 @@
mRecordTrack->getActiveMicrophones(activeMicrophones));
}
+binder::Status AudioFlinger::RecordHandle::setMicrophoneDirection(
+ int /*audio_microphone_direction_t*/ direction) {
+ ALOGV("%s()", __func__);
+ return binder::Status::fromStatusT(mRecordTrack->setMicrophoneDirection(
+ static_cast<audio_microphone_direction_t>(direction)));
+}
+
+binder::Status AudioFlinger::RecordHandle::setMicrophoneFieldDimension(float zoom) {
+ ALOGV("%s()", __func__);
+ return binder::Status::fromStatusT(mRecordTrack->setMicrophoneFieldDimension(zoom));
+}
+
// ----------------------------------------------------------------------------
#undef LOG_TAG
#define LOG_TAG "AF::RecordTrack"
@@ -2004,6 +2016,27 @@
}
}
+status_t AudioFlinger::RecordThread::RecordTrack::setMicrophoneDirection(
+ audio_microphone_direction_t direction) {
+ sp<ThreadBase> thread = mThread.promote();
+ if (thread != 0) {
+ RecordThread *recordThread = (RecordThread *)thread.get();
+ return recordThread->setMicrophoneDirection(direction);
+ } else {
+ return BAD_VALUE;
+ }
+}
+
+status_t AudioFlinger::RecordThread::RecordTrack::setMicrophoneFieldDimension(float zoom) {
+ sp<ThreadBase> thread = mThread.promote();
+ if (thread != 0) {
+ RecordThread *recordThread = (RecordThread *)thread.get();
+ return recordThread->setMicrophoneFieldDimension(zoom);
+ } else {
+ return BAD_VALUE;
+ }
+}
+
// ----------------------------------------------------------------------------
#undef LOG_TAG
#define LOG_TAG "AF::PatchRecord"
diff --git a/services/audiopolicy/AudioPolicyInterface.h b/services/audiopolicy/AudioPolicyInterface.h
index 1c2b9d7..cf2ce99 100644
--- a/services/audiopolicy/AudioPolicyInterface.h
+++ b/services/audiopolicy/AudioPolicyInterface.h
@@ -75,14 +75,16 @@
virtual status_t setDeviceConnectionState(audio_devices_t device,
audio_policy_dev_state_t state,
const char *device_address,
- const char *device_name) = 0;
+ const char *device_name,
+ audio_format_t encodedFormat) = 0;
// retrieve a device connection status
virtual audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
const char *device_address) = 0;
// indicate a change in device configuration
virtual status_t handleDeviceConfigChange(audio_devices_t device,
const char *device_address,
- const char *device_name) = 0;
+ const char *device_name,
+ audio_format_t encodedFormat) = 0;
// indicate a change in phone state. Valid phones states are defined by audio_mode_t
virtual void setPhoneState(audio_mode_t state) = 0;
// force using a specific device category for the specified usage
@@ -234,6 +236,9 @@
virtual bool isHapticPlaybackSupported() = 0;
+ virtual status_t getHwOffloadEncodingFormatsSupportedForA2DP(
+ std::vector<audio_format_t> *formats) = 0;
+
virtual void setAppState(uid_t uid, app_state_t state);
};
diff --git a/services/audiopolicy/common/include/policy.h b/services/audiopolicy/common/include/policy.h
index 46a2a40..837ca47 100644
--- a/services/audiopolicy/common/include/policy.h
+++ b/services/audiopolicy/common/include/policy.h
@@ -76,6 +76,21 @@
}
/**
+ * Check whether audio device has encoding capability.
+ *
+ * @param[in] device to consider
+ *
+ * @return true if device has encoding capability, false otherwise..
+ */
+static inline bool device_has_encoding_capability(audio_devices_t device)
+{
+ if (device & AUDIO_DEVICE_OUT_ALL_A2DP) {
+ return true;
+ }
+ return false;
+}
+
+/**
* Returns the priority of a given audio source for capture. The priority is used when more than one
* capture session is active on a given input stream to determine which session drives routing and
* effect configuration.
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h
index fa9ba0b..d4cfd1e 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h
@@ -23,11 +23,12 @@
#include "AudioIODescriptorInterface.h"
#include "AudioPort.h"
#include "ClientDescriptor.h"
+#include "DeviceDescriptor.h"
#include "EffectDescriptor.h"
+#include "IOProfile.h"
namespace android {
-class IOProfile;
class AudioMix;
class AudioPolicyClientInterface;
@@ -42,10 +43,16 @@
audio_port_handle_t getId() const;
audio_module_handle_t getModuleHandle() const;
+ audio_devices_t getDeviceType() const { return (mDevice != nullptr) ?
+ mDevice->type() : AUDIO_DEVICE_NONE; }
+ sp<DeviceDescriptor> getDevice() const { return mDevice; }
+ void setDevice(const sp<DeviceDescriptor> &device) { mDevice = device; }
+ DeviceVector supportedDevices() const {
+ return mProfile != nullptr ? mProfile->getSupportedDevices() : DeviceVector(); }
+
void dump(String8 *dst) const override;
audio_io_handle_t mIoHandle = AUDIO_IO_HANDLE_NONE; // input handle
- audio_devices_t mDevice = AUDIO_DEVICE_NONE; // current device this input is routed to
AudioMix *mPolicyMix = nullptr; // non NULL when used by a dynamic policy
const sp<IOProfile> mProfile; // I/O profile this output derives from
@@ -61,6 +68,7 @@
bool isSourceActive(audio_source_t source) const;
audio_source_t source() const;
bool isSoundTrigger() const;
+ audio_attributes_t getHighestPriorityAttributes() const;
void setClientActive(const sp<RecordClientDescriptor>& client, bool active);
int32_t activeCount() { return mGlobalActiveCount; }
void trackEffectEnabled(const sp<EffectDescriptor> &effect, bool enabled);
@@ -71,8 +79,7 @@
void setPatchHandle(audio_patch_handle_t handle) override;
status_t open(const audio_config_t *config,
- audio_devices_t device,
- const String8& address,
+ const sp<DeviceDescriptor> &device,
audio_source_t source,
audio_input_flags_t flags,
audio_io_handle_t *input);
@@ -99,6 +106,8 @@
audio_patch_handle_t mPatchHandle = AUDIO_PATCH_HANDLE_NONE;
audio_port_handle_t mId = AUDIO_PORT_HANDLE_NONE;
+ sp<DeviceDescriptor> mDevice = nullptr; /**< current device this input is routed to */
+
// Because a preemptible capture session can preempt another one, we end up in an endless loop
// situation were each session is allowed to restart after being preempted,
// thus preempting the other one which restarts and so on.
@@ -120,8 +129,8 @@
sp<AudioInputDescriptor> getInputFromId(audio_port_handle_t id) const;
// count active capture sessions using one of the specified devices.
- // ignore devices if AUDIO_DEVICE_IN_DEFAULT is passed
- uint32_t activeInputsCountOnDevices(audio_devices_t devices = AUDIO_DEVICE_IN_DEFAULT) const;
+ // ignore devices if empty vector is passed
+ uint32_t activeInputsCountOnDevices(const DeviceVector &devices) const;
/**
* return io handle of active input or 0 if no input is active
@@ -130,8 +139,6 @@
*/
Vector<sp <AudioInputDescriptor> > getActiveInputs();
- audio_devices_t getSupportedDevices(audio_io_handle_t handle) const;
-
sp<AudioInputDescriptor> getInputForClient(audio_port_handle_t portId);
void trackEffectEnabled(const sp<EffectDescriptor> &effect, bool enabled);
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
index ed995e0..e1ecc61 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
@@ -26,13 +26,14 @@
#include "AudioIODescriptorInterface.h"
#include "AudioPort.h"
#include "ClientDescriptor.h"
+#include "DeviceDescriptor.h"
+#include <map>
namespace android {
class IOProfile;
class AudioMix;
class AudioPolicyClientInterface;
-class DeviceDescriptor;
// descriptor for audio outputs. Used to maintain current configuration of each opened audio output
// and keep track of the usage of this output by each audio stream type.
@@ -48,14 +49,12 @@
void log(const char* indent);
audio_port_handle_t getId() const;
- virtual audio_devices_t device() const;
- virtual bool sharesHwModuleWith(const sp<AudioOutputDescriptor>& outputDesc);
- virtual audio_devices_t supportedDevices();
+ virtual DeviceVector devices() const { return mDevices; }
+ bool sharesHwModuleWith(const sp<AudioOutputDescriptor>& outputDesc);
+ virtual DeviceVector supportedDevices() const { return mDevices; }
virtual bool isDuplicated() const { return false; }
virtual uint32_t latency() { return 0; }
virtual bool isFixedVolume(audio_devices_t device);
- virtual sp<AudioOutputDescriptor> subOutput1() { return 0; }
- virtual sp<AudioOutputDescriptor> subOutput2() { return 0; }
virtual bool setVolume(float volume,
audio_stream_type_t stream,
audio_devices_t device,
@@ -119,7 +118,7 @@
return mActiveClients;
}
- audio_devices_t mDevice = AUDIO_DEVICE_NONE; // current device this output is routed to
+ DeviceVector mDevices; /**< current devices this output is routed to */
nsecs_t mStopTime[AUDIO_STREAM_CNT];
int mMuteCount[AUDIO_STREAM_CNT]; // mute request counter
bool mStrategyMutedByDevice[NUM_STRATEGIES]; // strategies muted because of incompatible
@@ -151,14 +150,16 @@
virtual ~SwAudioOutputDescriptor() {}
void dump(String8 *dst) const override;
- virtual audio_devices_t device() const;
- virtual bool sharesHwModuleWith(const sp<AudioOutputDescriptor>& outputDesc);
- virtual audio_devices_t supportedDevices();
+ virtual DeviceVector devices() const;
+ void setDevices(const DeviceVector &devices) { mDevices = devices; }
+ bool sharesHwModuleWith(const sp<SwAudioOutputDescriptor>& outputDesc);
+ virtual DeviceVector supportedDevices() const;
+ virtual bool deviceSupportsEncodedFormats(audio_devices_t device);
virtual uint32_t latency();
virtual bool isDuplicated() const { return (mOutput1 != NULL && mOutput2 != NULL); }
virtual bool isFixedVolume(audio_devices_t device);
- virtual sp<AudioOutputDescriptor> subOutput1() { return mOutput1; }
- virtual sp<AudioOutputDescriptor> subOutput2() { return mOutput2; }
+ sp<SwAudioOutputDescriptor> subOutput1() { return mOutput1; }
+ sp<SwAudioOutputDescriptor> subOutput2() { return mOutput2; }
void changeStreamActiveCount(
const sp<TrackClientDescriptor>& client, int delta) override;
virtual bool setVolume(float volume,
@@ -171,22 +172,49 @@
const struct audio_port_config *srcConfig = NULL) const;
virtual void toAudioPort(struct audio_port *port) const;
- status_t open(const audio_config_t *config,
- audio_devices_t device,
- const String8& address,
- audio_stream_type_t stream,
- audio_output_flags_t flags,
- audio_io_handle_t *output);
- // Called when a stream is about to be started
- // Note: called before setClientActive(true);
- status_t start();
- // Called after a stream is stopped.
- // Note: called after setClientActive(false);
- void stop();
- void close();
- status_t openDuplicating(const sp<SwAudioOutputDescriptor>& output1,
- const sp<SwAudioOutputDescriptor>& output2,
- audio_io_handle_t *ioHandle);
+ status_t open(const audio_config_t *config,
+ const DeviceVector &devices,
+ audio_stream_type_t stream,
+ audio_output_flags_t flags,
+ audio_io_handle_t *output);
+
+ // Called when a stream is about to be started
+ // Note: called before setClientActive(true);
+ status_t start();
+ // Called after a stream is stopped.
+ // Note: called after setClientActive(false);
+ void stop();
+ void close();
+ status_t openDuplicating(const sp<SwAudioOutputDescriptor>& output1,
+ const sp<SwAudioOutputDescriptor>& output2,
+ audio_io_handle_t *ioHandle);
+
+ /**
+ * @brief supportsDevice
+ * @param device to be checked against
+ * @return true if the device is supported by type (for non bus / remote submix devices),
+ * true if the device is supported (both type and address) for bus / remote submix
+ * false otherwise
+ */
+ bool supportsDevice(const sp<DeviceDescriptor> &device) const;
+
+ /**
+ * @brief supportsAllDevices
+ * @param devices to be checked against
+ * @return true if the device is weakly supported by type (e.g. for non bus / rsubmix devices),
+ * true if the device is supported (both type and address) for bus / remote submix
+ * false otherwise
+ */
+ bool supportsAllDevices(const DeviceVector &devices) const;
+
+ /**
+ * @brief filterSupportedDevices takes a vector of devices and filters them according to the
+ * device supported by this output (the profile from which this output derives from)
+ * @param devices reference device vector to be filtered
+ * @return vector of devices filtered from the supported devices of this output (weakly or not
+ * depending on the device type)
+ */
+ DeviceVector filterSupportedDevices(const DeviceVector &devices) const;
const sp<IOProfile> mProfile; // I/O profile this output derives from
audio_io_handle_t mIoHandle; // output handle
@@ -208,7 +236,6 @@
void dump(String8 *dst) const override;
- virtual audio_devices_t supportedDevices();
virtual bool setVolume(float volume,
audio_stream_type_t stream,
audio_devices_t device,
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h
index 955e87b..2932296 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h
@@ -16,15 +16,17 @@
#pragma once
+#include "DeviceDescriptor.h"
#include <utils/RefBase.h>
#include <media/AudioPolicy.h>
#include <utils/KeyedVector.h>
#include <system/audio.h>
#include <utils/String8.h>
-namespace android {
+#include <DeviceDescriptor.h>
+#include <AudioOutputDescriptor.h>
-class SwAudioOutputDescriptor;
+namespace android {
/**
* custom mix entry in mPolicyMixes
@@ -74,9 +76,21 @@
status_t getOutputForAttr(audio_attributes_t attributes, uid_t uid,
sp<SwAudioOutputDescriptor> &desc);
- audio_devices_t getDeviceAndMixForInputSource(audio_source_t inputSource,
- audio_devices_t availableDeviceTypes,
- AudioMix **policyMix);
+ sp<DeviceDescriptor> getDeviceAndMixForInputSource(audio_source_t inputSource,
+ const DeviceVector &availableDeviceTypes,
+ AudioMix **policyMix);
+
+ /**
+ * @brief try to find a matching mix for a given output descriptor and returns the associated
+ * output device.
+ * @param output to be considered
+ * @param availableOutputDevices list of output devices currently reachable
+ * @param policyMix to be returned if any mix matching ouput descriptor
+ * @return device selected from the mix attached to the output, null pointer otherwise
+ */
+ sp<DeviceDescriptor> getDeviceAndMixForOutput(const sp<SwAudioOutputDescriptor> &output,
+ const DeviceVector &availableOutputDevices,
+ AudioMix **policyMix = nullptr);
status_t getInputMixForAttr(audio_attributes_t attr, AudioMix **policyMix);
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPort.h b/services/audiopolicy/common/managerdefinitions/include/AudioPort.h
index bb9cad8..1b5a2d6 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioPort.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioPort.h
@@ -65,6 +65,7 @@
uint32_t getFlags() const { return mFlags; }
virtual void attach(const sp<HwModule>& module);
+ virtual void detach();
bool isAttached() { return mModule != 0; }
// Audio port IDs are in a different namespace than AudioFlinger unique IDs
@@ -161,7 +162,7 @@
const struct audio_port_config *srcConfig = NULL) const = 0;
virtual sp<AudioPort> getAudioPort() const = 0;
virtual bool hasSameHwModuleAs(const sp<AudioPortConfig>& other) const {
- return (other != 0) &&
+ return (other != 0) && (other->getAudioPort() != 0) && (getAudioPort() != 0) &&
(other->getAudioPort()->getModuleHandle() == getAudioPort()->getModuleHandle());
}
unsigned int mSamplingRate = 0u;
diff --git a/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
index d02123c..cc43fe6 100644
--- a/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
@@ -44,8 +44,18 @@
const FormatVector& encodedFormats() const { return mEncodedFormats; }
+ audio_format_t getEncodedFormat() { return mCurrentEncodedFormat; }
+
+ void setEncodedFormat(audio_format_t format) {
+ mCurrentEncodedFormat = format;
+ }
+
bool equals(const sp<DeviceDescriptor>& other) const;
+ bool hasCurrentEncodedFormat() const;
+
+ bool supportsFormat(audio_format_t format);
+
// AudioPortConfig
virtual sp<AudioPort> getAudioPort() const { return (AudioPort*) this; }
virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
@@ -53,6 +63,8 @@
// AudioPort
virtual void attach(const sp<HwModule>& module);
+ virtual void detach();
+
virtual void toAudioPort(struct audio_port *port) const;
virtual void importAudioPort(const sp<AudioPort>& port, bool force = false);
@@ -67,6 +79,7 @@
audio_devices_t mDeviceType;
FormatVector mEncodedFormats;
audio_port_handle_t mId = AUDIO_PORT_HANDLE_NONE;
+ audio_format_t mCurrentEncodedFormat;
};
class DeviceVector : public SortedVector<sp<DeviceDescriptor> >
@@ -86,9 +99,10 @@
audio_devices_t types() const { return mDeviceTypes; }
- // If 'address' is empty, a device with a non-empty address may be returned
- // if there is no device with the specified 'type' and empty address.
- sp<DeviceDescriptor> getDevice(audio_devices_t type, const String8 &address = {}) const;
+ // If 'address' is empty and 'codec' is AUDIO_FORMAT_DEFAULT, a device with a non-empty
+ // address may be returned if there is no device with the specified 'type' and empty address.
+ sp<DeviceDescriptor> getDevice(audio_devices_t type, const String8 &address,
+ audio_format_t codec) const;
DeviceVector getDevicesFromTypeMask(audio_devices_t types) const;
/**
@@ -164,6 +178,23 @@
return !operator==(right);
}
+ /**
+ * @brief getFirstValidAddress
+ * @return the first valid address of a list of device, "" if no device with valid address
+ * found.
+ * This helper function helps maintaining compatibility with legacy where we used to have a
+ * devices mask and an address.
+ */
+ String8 getFirstValidAddress() const
+ {
+ for (const auto &device : *this) {
+ if (device->address() != "") {
+ return device->address();
+ }
+ }
+ return String8("");
+ }
+
std::string toString() const;
void dump(String8 *dst, const String8 &tag, int spaces = 0, bool verbose = true) const;
diff --git a/services/audiopolicy/common/managerdefinitions/include/HwModule.h b/services/audiopolicy/common/managerdefinitions/include/HwModule.h
index 2b57fa9..eb34da4 100644
--- a/services/audiopolicy/common/managerdefinitions/include/HwModule.h
+++ b/services/audiopolicy/common/managerdefinitions/include/HwModule.h
@@ -46,6 +46,22 @@
const DeviceVector &getDeclaredDevices() const { return mDeclaredDevices; }
void setDeclaredDevices(const DeviceVector &devices);
+ DeviceVector getAllDevices() const
+ {
+ DeviceVector devices = mDeclaredDevices;
+ devices.merge(mDynamicDevices);
+ return devices;
+ }
+ void addDynamicDevice(const sp<DeviceDescriptor> &device)
+ {
+ mDynamicDevices.add(device);
+ }
+
+ bool removeDynamicDevice(const sp<DeviceDescriptor> &device)
+ {
+ return mDynamicDevices.remove(device) >= 0;
+ }
+ DeviceVector getDynamicDevices() const { return mDynamicDevices; }
const InputProfileCollection &getInputProfiles() const { return mInputProfiles; }
const OutputProfileCollection &getOutputProfiles() const { return mOutputProfiles; }
@@ -104,6 +120,7 @@
InputProfileCollection mInputProfiles; // input profiles exposed by this module
uint32_t mHalVersion; // audio HAL API version
DeviceVector mDeclaredDevices; // devices declared in audio_policy configuration file.
+ DeviceVector mDynamicDevices; /**< devices that can be added/removed at runtime (e.g. rsbumix)*/
AudioRouteVector mRoutes;
AudioPortVector mPorts;
};
@@ -113,13 +130,63 @@
public:
sp<HwModule> getModuleFromName(const char *name) const;
- sp<HwModule> getModuleForDevice(audio_devices_t device) const;
+ sp<HwModule> getModuleForDeviceTypes(audio_devices_t device,
+ audio_format_t encodedFormat) const;
- sp<DeviceDescriptor> getDeviceDescriptor(const audio_devices_t device,
- const char *device_address,
- const char *device_name,
+ sp<HwModule> getModuleForDevice(const sp<DeviceDescriptor> &device,
+ audio_format_t encodedFormat) const;
+
+ DeviceVector getAvailableDevicesFromModuleName(const char *name,
+ const DeviceVector &availableDevices) const;
+
+ /**
+ * @brief getDeviceDescriptor returns a device descriptor associated to the device type and
+ * device address (if matchAddress is true).
+ * It may loop twice on all modules to check if allowToCreate is true
+ * -first loop will check if the device is found on a module since declared in the list
+ * of device port in configuration file
+ * -(allowToCreate is true)second loop will check if the device is weakly supported by one
+ * or more profiles on a given module and will add as a supported device for this module.
+ * The device will also be added to the dynamic list of device of this module
+ * @param type of the device requested
+ * @param address of the device requested
+ * @param name of the device that requested
+ * @param encodedFormat if not AUDIO_FORMAT_DEFAULT, must match one supported format
+ * @param matchAddress true if a strong match is required
+ * @param allowToCreate true if allowed to create dynamic device (e.g. hdmi, usb...)
+ * @return device descriptor associated to the type (and address if matchAddress is true)
+ */
+ sp<DeviceDescriptor> getDeviceDescriptor(const audio_devices_t type,
+ const char *address,
+ const char *name,
+ audio_format_t encodedFormat,
+ bool allowToCreate = false,
bool matchAddress = true) const;
+ /**
+ * @brief createDevice creates a new device from the type and address given. It checks that
+ * according to the device type, a module is supporting this device (weak check).
+ * This concerns only dynamic device, aka device with a specific address and not
+ * already supported by module/underlying profiles.
+ * @param type of the device to be created
+ * @param address of the device to be created
+ * @param name of the device to be created
+ * @return device descriptor if a module is supporting this type, nullptr otherwise.
+ */
+ sp<DeviceDescriptor> createDevice(const audio_devices_t type,
+ const char *address,
+ const char *name,
+ const audio_format_t encodedFormat) const;
+
+ /**
+ * @brief cleanUpForDevice: loop on all profiles of all modules to remove device from
+ * the list of supported device. If this device is a dynamic device (aka a device not in the
+ * xml file with a runtime address), it is also removed from the module collection of dynamic
+ * devices.
+ * @param device that has been disconnected
+ */
+ void cleanUpForDevice(const sp<DeviceDescriptor> &device);
+
void dump(String8 *dst) const;
};
diff --git a/services/audiopolicy/common/managerdefinitions/include/IOProfile.h b/services/audiopolicy/common/managerdefinitions/include/IOProfile.h
index ca6ca56..e0b56d4 100644
--- a/services/audiopolicy/common/managerdefinitions/include/IOProfile.h
+++ b/services/audiopolicy/common/managerdefinitions/include/IOProfile.h
@@ -57,12 +57,25 @@
}
}
- // This method is used for input and direct output, and is not used for other output.
- // If parameter updatedSamplingRate is non-NULL, it is assigned the actual sample rate.
- // For input, flags is interpreted as audio_input_flags_t.
- // TODO: merge audio_output_flags_t and audio_input_flags_t.
- bool isCompatibleProfile(audio_devices_t device,
- const String8& address,
+ /**
+ * @brief isCompatibleProfile: This method is used for input and direct output,
+ * and is not used for other output.
+ * Checks if the IO profile is compatible with specified parameters.
+ * For input, flags is interpreted as audio_input_flags_t.
+ * TODO: merge audio_output_flags_t and audio_input_flags_t.
+ *
+ * @param devices vector of devices to be checked for compatibility
+ * @param samplingRate to be checked for compatibility. Must be specified
+ * @param updatedSamplingRate if non-NULL, it is assigned the actual sample rate.
+ * @param format to be checked for compatibility. Must be specified
+ * @param updatedFormat if non-NULL, it is assigned the actual format
+ * @param channelMask to be checked for compatibility. Must be specified
+ * @param updatedChannelMask if non-NULL, it is assigned the actual channel mask
+ * @param flags to be checked for compatibility
+ * @param exactMatchRequiredForInputFlags true if exact match is required on flags
+ * @return true if the profile is compatible, false otherwise.
+ */
+ bool isCompatibleProfile(const DeviceVector &devices,
uint32_t samplingRate,
uint32_t *updatedSamplingRate,
audio_format_t format,
@@ -78,49 +91,61 @@
bool hasSupportedDevices() const { return !mSupportedDevices.isEmpty(); }
- bool supportDevice(audio_devices_t device) const
+ bool supportsDeviceTypes(audio_devices_t device) const
{
if (audio_is_output_devices(device)) {
- return mSupportedDevices.types() & device;
+ if (deviceSupportsEncodedFormats(device)) {
+ return mSupportedDevices.types() & device;
+ }
+ return false;
}
return mSupportedDevices.types() & (device & ~AUDIO_DEVICE_BIT_IN);
}
- bool supportDeviceAddress(const String8 &address) const
+ /**
+ * @brief supportsDevice
+ * @param device to be checked against
+ * forceCheckOnAddress if true, check on type and address whatever the type, otherwise
+ * the address enforcement is limited to "offical devices" that distinguishe on address
+ * @return true if the device is supported by type (for non bus / remote submix devices),
+ * true if the device is supported (both type and address) for bus / remote submix
+ * false otherwise
+ */
+ bool supportsDevice(const sp<DeviceDescriptor> &device, bool forceCheckOnAddress = false) const
{
- return mSupportedDevices[0]->address() == address;
- }
-
- // chose first device present in mSupportedDevices also part of deviceType
- audio_devices_t getSupportedDeviceForType(audio_devices_t deviceType) const
- {
- for (size_t k = 0; k < mSupportedDevices.size(); k++) {
- audio_devices_t profileType = mSupportedDevices[k]->type();
- if (profileType & deviceType) {
- return profileType;
- }
+ if (!device_distinguishes_on_address(device->type()) && !forceCheckOnAddress) {
+ return supportsDeviceTypes(device->type());
}
- return AUDIO_DEVICE_NONE;
+ return mSupportedDevices.contains(device);
}
- audio_devices_t getSupportedDevicesType() const { return mSupportedDevices.types(); }
+ bool deviceSupportsEncodedFormats(audio_devices_t device) const
+ {
+ if (device == AUDIO_DEVICE_NONE) {
+ return true; // required for isOffloadSupported() check
+ }
+ DeviceVector deviceList =
+ mSupportedDevices.getDevicesFromTypeMask(device);
+ if (!deviceList.empty()) {
+ return deviceList.itemAt(0)->hasCurrentEncodedFormat();
+ }
+ return false;
+ }
void clearSupportedDevices() { mSupportedDevices.clear(); }
void addSupportedDevice(const sp<DeviceDescriptor> &device)
{
mSupportedDevices.add(device);
}
-
+ void removeSupportedDevice(const sp<DeviceDescriptor> &device)
+ {
+ mSupportedDevices.remove(device);
+ }
void setSupportedDevices(const DeviceVector &devices)
{
mSupportedDevices = devices;
}
- sp<DeviceDescriptor> getSupportedDeviceByAddress(audio_devices_t type, String8 address) const
- {
- return mSupportedDevices.getDevice(type, address);
- }
-
const DeviceVector &getSupportedDevices() const { return mSupportedDevices; }
bool canOpenNewIo() {
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp
index 0bc88a5..c880e67 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp
@@ -21,7 +21,6 @@
#include <policy.h>
#include <AudioPolicyInterface.h>
#include "AudioInputDescriptor.h"
-#include "IOProfile.h"
#include "AudioGain.h"
#include "HwModule.h"
@@ -55,30 +54,7 @@
audio_source_t AudioInputDescriptor::source() const
{
- audio_source_t source = AUDIO_SOURCE_DEFAULT;
-
- for (bool activeOnly : { true, false }) {
- int32_t topPriority = -1;
- app_state_t topState = APP_STATE_IDLE;
- for (const auto &client : getClientIterable()) {
- if (activeOnly && !client->active()) {
- continue;
- }
- app_state_t curState = client->appState();
- if (curState >= topState) {
- int32_t curPriority = source_priority(client->source());
- if (curPriority > topPriority) {
- source = client->source();
- topPriority = curPriority;
- }
- topState = curState;
- }
- }
- if (source != AUDIO_SOURCE_DEFAULT) {
- break;
- }
- }
- return source;
+ return getHighestPriorityAttributes().source;
}
void AudioInputDescriptor::toAudioPortConfig(struct audio_port_config *dstConfig,
@@ -148,6 +124,34 @@
return false;
}
+audio_attributes_t AudioInputDescriptor::getHighestPriorityAttributes() const
+{
+ audio_attributes_t attributes = { .source = AUDIO_SOURCE_DEFAULT };
+
+ for (bool activeOnly : { true, false }) {
+ int32_t topPriority = -1;
+ app_state_t topState = APP_STATE_IDLE;
+ for (const auto &client : getClientIterable()) {
+ if (activeOnly && !client->active()) {
+ continue;
+ }
+ app_state_t curState = client->appState();
+ if (curState >= topState) {
+ int32_t curPriority = source_priority(client->source());
+ if (curPriority > topPriority) {
+ attributes = client->attributes();
+ topPriority = curPriority;
+ }
+ topState = curState;
+ }
+ }
+ if (attributes.source != AUDIO_SOURCE_DEFAULT) {
+ break;
+ }
+ }
+ return attributes;
+}
+
bool AudioInputDescriptor::isSoundTrigger() const {
// sound trigger and non sound trigger clients are not mixed on a given input
// so check only first client
@@ -180,8 +184,7 @@
}
status_t AudioInputDescriptor::open(const audio_config_t *config,
- audio_devices_t device,
- const String8& address,
+ const sp<DeviceDescriptor> &device,
audio_source_t source,
audio_input_flags_t flags,
audio_io_handle_t *input)
@@ -198,24 +201,26 @@
mDevice = device;
- ALOGV("opening input for device %08x address %s profile %p name %s",
- mDevice, address.string(), mProfile.get(), mProfile->getName().string());
+ ALOGV("opening input for device %s profile %p name %s",
+ mDevice->toString().c_str(), mProfile.get(), mProfile->getName().string());
+
+ audio_devices_t deviceType = mDevice->type();
status_t status = mClientInterface->openInput(mProfile->getModuleHandle(),
input,
&lConfig,
- &mDevice,
- address,
+ &deviceType,
+ mDevice->address(),
source,
flags);
- LOG_ALWAYS_FATAL_IF(mDevice != device,
+ LOG_ALWAYS_FATAL_IF(mDevice->type() != deviceType,
"%s openInput returned device %08x when given device %08x",
- __FUNCTION__, mDevice, device);
+ __FUNCTION__, mDevice->type(), deviceType);
if (status == NO_ERROR) {
LOG_ALWAYS_FATAL_IF(*input == AUDIO_IO_HANDLE_NONE,
- "%s openInput returned input handle %d for device %08x",
- __FUNCTION__, *input, device);
+ "%s openInput returned input handle %d for device %s",
+ __FUNCTION__, *input, mDevice->toString().c_str());
mSamplingRate = lConfig.sample_rate;
mChannelMask = lConfig.channel_mask;
mFormat = lConfig.format;
@@ -252,15 +257,21 @@
void AudioInputDescriptor::close()
{
if (mIoHandle != AUDIO_IO_HANDLE_NONE) {
+ // clean up active clients if any (can happen if close() is called to force
+ // clients to reconnect
+ for (const auto &client : getClientIterable()) {
+ if (client->active()) {
+ ALOGW("%s client with port ID %d still active on input %d",
+ __func__, client->portId(), mId);
+ setClientActive(client, false);
+ stop();
+ }
+ }
+
mClientInterface->closeInput(mIoHandle);
LOG_ALWAYS_FATAL_IF(mProfile->curOpenCount < 1, "%s profile open count %u",
__FUNCTION__, mProfile->curOpenCount);
- // do not call stop() here as stop() is supposed to be called after
- // setClientActive(client, false) and we don't know how many clients
- // are still active at this time
- if (isActive()) {
- mProfile->curActiveCount--;
- }
+
mProfile->curOpenCount--;
LOG_ALWAYS_FATAL_IF(mProfile->curOpenCount < mProfile->curActiveCount,
"%s(%d): mProfile->curOpenCount %d < mProfile->curActiveCount %d.",
@@ -423,7 +434,7 @@
dst->appendFormat(" Sampling rate: %d\n", mSamplingRate);
dst->appendFormat(" Format: %d\n", mFormat);
dst->appendFormat(" Channels: %08x\n", mChannelMask);
- dst->appendFormat(" Devices %08x\n", mDevice);
+ dst->appendFormat(" Devices %s\n", mDevice->toString().c_str());
getEnabledEffects().dump(dst, 1 /*spaces*/, false /*verbose*/);
dst->append(" AudioRecord Clients:\n");
ClientMapHandler<RecordClientDescriptor>::dump(dst);
@@ -452,14 +463,13 @@
return NULL;
}
-uint32_t AudioInputCollection::activeInputsCountOnDevices(audio_devices_t devices) const
+uint32_t AudioInputCollection::activeInputsCountOnDevices(const DeviceVector &devices) const
{
uint32_t count = 0;
for (size_t i = 0; i < size(); i++) {
const sp<AudioInputDescriptor> inputDescriptor = valueAt(i);
if (inputDescriptor->isActive() &&
- ((devices == AUDIO_DEVICE_IN_DEFAULT) ||
- ((inputDescriptor->mDevice & devices & ~AUDIO_DEVICE_BIT_IN) != 0))) {
+ (devices.isEmpty() || devices.contains(inputDescriptor->getDevice()))) {
count++;
}
}
@@ -479,13 +489,6 @@
return activeInputs;
}
-audio_devices_t AudioInputCollection::getSupportedDevices(audio_io_handle_t handle) const
-{
- sp<AudioInputDescriptor> inputDesc = valueFor(handle);
- audio_devices_t devices = inputDesc->mProfile->getSupportedDevicesType();
- return devices;
-}
-
sp<AudioInputDescriptor> AudioInputCollection::getInputForClient(audio_port_handle_t portId)
{
for (size_t i = 0; i < size(); i++) {
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
index 97504ab..78b3f45 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
@@ -82,25 +82,10 @@
return mId;
}
-audio_devices_t AudioOutputDescriptor::device() const
-{
- return mDevice;
-}
-
-audio_devices_t AudioOutputDescriptor::supportedDevices()
-{
- return mDevice;
-}
-
bool AudioOutputDescriptor::sharesHwModuleWith(
const sp<AudioOutputDescriptor>& outputDesc)
{
- if (outputDesc->isDuplicated()) {
- return sharesHwModuleWith(outputDesc->subOutput1()) ||
- sharesHwModuleWith(outputDesc->subOutput2());
- } else {
- return hasSameHwModuleAs(outputDesc);
- }
+ return hasSameHwModuleAs(outputDesc);
}
void AudioOutputDescriptor::changeStreamActiveCount(const sp<TrackClientDescriptor>& client,
@@ -282,7 +267,7 @@
dst->appendFormat(" Sampling rate: %d\n", mSamplingRate);
dst->appendFormat(" Format: %08x\n", mFormat);
dst->appendFormat(" Channels: %08x\n", mChannelMask);
- dst->appendFormat(" Devices: %08x\n", device());
+ dst->appendFormat(" Devices: %s\n", devices().toString().c_str());
dst->appendFormat(" Global active count: %u\n", mGlobalActiveCount);
dst->append(" Stream volume activeCount muteCount\n");
for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) {
@@ -330,17 +315,18 @@
AudioOutputDescriptor::dump(dst);
}
-audio_devices_t SwAudioOutputDescriptor::device() const
+DeviceVector SwAudioOutputDescriptor::devices() const
{
if (isDuplicated()) {
- return (audio_devices_t)(mOutput1->mDevice | mOutput2->mDevice);
- } else {
- return mDevice;
+ DeviceVector devices = mOutput1->devices();
+ devices.merge(mOutput2->devices());
+ return devices;
}
+ return mDevices;
}
bool SwAudioOutputDescriptor::sharesHwModuleWith(
- const sp<AudioOutputDescriptor>& outputDesc)
+ const sp<SwAudioOutputDescriptor>& outputDesc)
{
if (isDuplicated()) {
return mOutput1->sharesHwModuleWith(outputDesc) || mOutput2->sharesHwModuleWith(outputDesc);
@@ -352,12 +338,39 @@
}
}
-audio_devices_t SwAudioOutputDescriptor::supportedDevices()
+DeviceVector SwAudioOutputDescriptor::supportedDevices() const
{
if (isDuplicated()) {
- return (audio_devices_t)(mOutput1->supportedDevices() | mOutput2->supportedDevices());
+ DeviceVector supportedDevices = mOutput1->supportedDevices();
+ supportedDevices.merge(mOutput2->supportedDevices());
+ return supportedDevices;
+ }
+ return mProfile->getSupportedDevices();
+}
+
+bool SwAudioOutputDescriptor::supportsDevice(const sp<DeviceDescriptor> &device) const
+{
+ return supportedDevices().contains(device);
+}
+
+bool SwAudioOutputDescriptor::supportsAllDevices(const DeviceVector &devices) const
+{
+ return supportedDevices().containsAllDevices(devices);
+}
+
+DeviceVector SwAudioOutputDescriptor::filterSupportedDevices(const DeviceVector &devices) const
+{
+ DeviceVector filteredDevices = supportedDevices();
+ return filteredDevices.filter(devices);
+}
+
+bool SwAudioOutputDescriptor::deviceSupportsEncodedFormats(audio_devices_t device)
+{
+ if (isDuplicated()) {
+ return (mOutput1->deviceSupportsEncodedFormats(device)
+ || mOutput2->deviceSupportsEncodedFormats(device));
} else {
- return mProfile->getSupportedDevicesType();
+ return mProfile->deviceSupportsEncodedFormats(device);
}
}
@@ -443,12 +456,15 @@
}
status_t SwAudioOutputDescriptor::open(const audio_config_t *config,
- audio_devices_t device,
- const String8& address,
+ const DeviceVector &devices,
audio_stream_type_t stream,
audio_output_flags_t flags,
audio_io_handle_t *output)
{
+ mDevices = devices;
+ const String8& address = devices.getFirstValidAddress();
+ audio_devices_t device = devices.types();
+
audio_config_t lConfig;
if (config == nullptr) {
lConfig = AUDIO_CONFIG_INITIALIZER;
@@ -459,7 +475,6 @@
lConfig = *config;
}
- mDevice = device;
// if the selected profile is offloaded and no offload info was specified,
// create a default one
if ((mProfile->getFlags() & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
@@ -477,19 +492,19 @@
mFlags = (audio_output_flags_t)(mFlags | flags);
- ALOGV("opening output for device %08x address %s profile %p name %s",
- mDevice, address.string(), mProfile.get(), mProfile->getName().string());
+ ALOGV("opening output for device %s profile %p name %s",
+ mDevices.toString().c_str(), mProfile.get(), mProfile->getName().string());
status_t status = mClientInterface->openOutput(mProfile->getModuleHandle(),
output,
&lConfig,
- &mDevice,
+ &device,
address,
&mLatency,
mFlags);
- LOG_ALWAYS_FATAL_IF(mDevice != device,
+ LOG_ALWAYS_FATAL_IF(mDevices.types() != device,
"%s openOutput returned device %08x when given device %08x",
- __FUNCTION__, mDevice, device);
+ __FUNCTION__, mDevices.types(), device);
if (status == NO_ERROR) {
LOG_ALWAYS_FATAL_IF(*output == AUDIO_IO_HANDLE_NONE,
@@ -548,6 +563,17 @@
void SwAudioOutputDescriptor::close()
{
if (mIoHandle != AUDIO_IO_HANDLE_NONE) {
+ // clean up active clients if any (can happen if close() is called to force
+ // clients to reconnect
+ for (const auto &client : getClientIterable()) {
+ if (client->active()) {
+ ALOGW("%s client with port ID %d still active on output %d",
+ __func__, client->portId(), mId);
+ setClientActive(client, false);
+ stop();
+ }
+ }
+
AudioParameter param;
param.add(String8("closing"), String8("true"));
mClientInterface->setParameters(mIoHandle, param.toString());
@@ -556,11 +582,6 @@
LOG_ALWAYS_FATAL_IF(mProfile->curOpenCount < 1, "%s profile open count %u",
__FUNCTION__, mProfile->curOpenCount);
- // do not call stop() here as stop() is supposed to be called after setClientActive(false)
- // and we don't know how many streams are still active at this time
- if (isActive()) {
- mProfile->curActiveCount--;
- }
mProfile->curOpenCount--;
mIoHandle = AUDIO_IO_HANDLE_NONE;
}
@@ -605,11 +626,6 @@
mSource->dump(dst, 0, 0);
}
-audio_devices_t HwAudioOutputDescriptor::supportedDevices()
-{
- return mDevice;
-}
-
void HwAudioOutputDescriptor::toAudioPortConfig(
struct audio_port_config *dstConfig,
const struct audio_port_config *srcConfig) const
@@ -657,7 +673,7 @@
for (size_t i = 0; i < this->size(); i++) {
const sp<SwAudioOutputDescriptor> outputDesc = this->valueAt(i);
if (outputDesc->isStreamActive(stream, inPastMs, sysTime)
- && ((outputDesc->device() & APM_AUDIO_OUT_DEVICE_REMOTE_ALL) == 0)) {
+ && ((outputDesc->devices().types() & APM_AUDIO_OUT_DEVICE_REMOTE_ALL) == 0)) {
return true;
}
}
@@ -670,7 +686,7 @@
nsecs_t sysTime = systemTime();
for (size_t i = 0; i < size(); i++) {
const sp<SwAudioOutputDescriptor> outputDesc = valueAt(i);
- if (((outputDesc->device() & APM_AUDIO_OUT_DEVICE_REMOTE_ALL) != 0) &&
+ if (((outputDesc->devices().types() & APM_AUDIO_OUT_DEVICE_REMOTE_ALL) != 0) &&
outputDesc->isStreamActive(stream, inPastMs, sysTime)) {
// do not consider re routing (when the output is going to a dynamic policy)
// as "remote playback"
@@ -686,7 +702,10 @@
{
for (size_t i = 0; i < size(); i++) {
sp<SwAudioOutputDescriptor> outputDesc = valueAt(i);
- if (!outputDesc->isDuplicated() && outputDesc->device() & AUDIO_DEVICE_OUT_ALL_A2DP) {
+ if (!outputDesc->isDuplicated() &&
+ outputDesc->devices().types() & AUDIO_DEVICE_OUT_ALL_A2DP &&
+ outputDesc->deviceSupportsEncodedFormats(
+ AUDIO_DEVICE_OUT_BLUETOOTH_A2DP)) {
return this->keyAt(i);
}
}
@@ -700,10 +719,9 @@
if ((primaryOutput != NULL) && (primaryOutput->mProfile != NULL)
&& (primaryOutput->mProfile->getModule() != NULL)) {
sp<HwModule> primaryHwModule = primaryOutput->mProfile->getModule();
- Vector <sp<IOProfile>> primaryHwModuleOutputProfiles =
- primaryHwModule->getOutputProfiles();
- for (size_t i = 0; i < primaryHwModuleOutputProfiles.size(); i++) {
- if (primaryHwModuleOutputProfiles[i]->supportDevice(AUDIO_DEVICE_OUT_ALL_A2DP)) {
+
+ for (const auto &outputProfile : primaryHwModule->getOutputProfiles()) {
+ if (outputProfile->supportsDeviceTypes(AUDIO_DEVICE_OUT_ALL_A2DP)) {
return true;
}
}
@@ -754,13 +772,6 @@
return false;
}
-audio_devices_t SwAudioOutputCollection::getSupportedDevices(audio_io_handle_t handle) const
-{
- sp<SwAudioOutputDescriptor> outputDesc = valueFor(handle);
- audio_devices_t devices = outputDesc->mProfile->getSupportedDevicesType();
- return devices;
-}
-
sp<SwAudioOutputDescriptor> SwAudioOutputCollection::getOutputForClient(audio_port_handle_t portId)
{
for (size_t i = 0; i < size(); i++) {
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp
index 4d0916e..3b9411a 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp
@@ -280,13 +280,32 @@
return BAD_VALUE;
}
-audio_devices_t AudioPolicyMixCollection::getDeviceAndMixForInputSource(audio_source_t inputSource,
- audio_devices_t availDevices,
- AudioMix **policyMix)
+sp<DeviceDescriptor> AudioPolicyMixCollection::getDeviceAndMixForOutput(
+ const sp<SwAudioOutputDescriptor> &output,
+ const DeviceVector &availableOutputDevices,
+ AudioMix **policyMix)
+{
+ for (size_t i = 0; i < size(); i++) {
+ if (valueAt(i)->getOutput() == output) {
+ AudioMix *mix = valueAt(i)->getMix();
+ if (policyMix != nullptr)
+ *policyMix = mix;
+ // This Desc is involved in a Mix, which has the highest prio
+ audio_devices_t deviceType = mix->mDeviceType;
+ String8 address = mix->mDeviceAddress;
+ ALOGV("%s: device (0x%x, addr=%s) forced by mix",
+ __FUNCTION__, deviceType, address.c_str());
+ return availableOutputDevices.getDevice(deviceType, address, AUDIO_FORMAT_DEFAULT);
+ }
+ }
+ return nullptr;
+}
+
+sp<DeviceDescriptor> AudioPolicyMixCollection::getDeviceAndMixForInputSource(
+ audio_source_t inputSource, const DeviceVector &availDevices, AudioMix **policyMix)
{
for (size_t i = 0; i < size(); i++) {
AudioMix *mix = valueAt(i)->getMix();
-
if (mix->mMixType != MIX_TYPE_RECORDERS) {
continue;
}
@@ -295,17 +314,22 @@
mix->mCriteria[j].mValue.mSource == inputSource) ||
(RULE_EXCLUDE_ATTRIBUTE_CAPTURE_PRESET == mix->mCriteria[j].mRule &&
mix->mCriteria[j].mValue.mSource != inputSource)) {
- if (availDevices & AUDIO_DEVICE_IN_REMOTE_SUBMIX) {
+ // assuming PolicyMix only for remote submix for input
+ // so mix->mDeviceType can only be AUDIO_DEVICE_OUT_REMOTE_SUBMIX
+ audio_devices_t device = AUDIO_DEVICE_IN_REMOTE_SUBMIX;
+ auto mixDevice =
+ availDevices.getDevice(device, mix->mDeviceAddress, AUDIO_FORMAT_DEFAULT);
+ if (mixDevice != nullptr) {
if (policyMix != NULL) {
*policyMix = mix;
}
- return AUDIO_DEVICE_IN_REMOTE_SUBMIX;
+ return mixDevice;
}
break;
}
}
}
- return AUDIO_DEVICE_NONE;
+ return nullptr;
}
status_t AudioPolicyMixCollection::getInputMixForAttr(audio_attributes_t attr, AudioMix **policyMix)
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp
index 19dde6a..9fcf5e7 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp
@@ -31,9 +31,15 @@
// --- AudioPort class implementation
void AudioPort::attach(const sp<HwModule>& module)
{
+ ALOGV("%s: attaching module %s to port %s", __FUNCTION__, getModuleName(), mName.string());
mModule = module;
}
+void AudioPort::detach()
+{
+ mModule = nullptr;
+}
+
// Note that is a different namespace than AudioFlinger unique IDs
audio_port_handle_t AudioPort::getNextUniqueId()
{
diff --git a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
index 04cbcd1..dc5b238 100644
--- a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
@@ -18,6 +18,7 @@
//#define LOG_NDEBUG 0
#include <audio_utils/string.h>
+#include <set>
#include "DeviceDescriptor.h"
#include "TypeConverter.h"
#include "AudioGain.h"
@@ -37,11 +38,16 @@
AUDIO_PORT_ROLE_SOURCE),
mTagName(tagName), mDeviceType(type), mEncodedFormats(encodedFormats)
{
+ mCurrentEncodedFormat = AUDIO_FORMAT_DEFAULT;
if (type == AUDIO_DEVICE_IN_REMOTE_SUBMIX || type == AUDIO_DEVICE_OUT_REMOTE_SUBMIX ) {
mAddress = String8("0");
}
- /* FIXME: read from APM config file */
- if (type == AUDIO_DEVICE_OUT_HDMI) {
+ /* If framework runs against a pre 5.0 Audio HAL, encoded formats are absent from the config.
+ * FIXME: APM should know the version of the HAL and don't add the formats for V5.0.
+ * For now, the workaround to remove AC3 and IEC61937 support on HDMI is to declare
+ * something like 'encodedFormats="AUDIO_FORMAT_PCM_16_BIT"' on the HDMI devicePort.
+ */
+ if (type == AUDIO_DEVICE_OUT_HDMI && mEncodedFormats.isEmpty()) {
mEncodedFormats.add(AUDIO_FORMAT_AC3);
mEncodedFormats.add(AUDIO_FORMAT_IEC61937);
}
@@ -58,15 +64,57 @@
mId = getNextUniqueId();
}
+void DeviceDescriptor::detach() {
+ mId = AUDIO_PORT_HANDLE_NONE;
+ AudioPort::detach();
+}
+
+template<typename T>
+bool checkEqual(const T& f1, const T& f2)
+{
+ std::set<typename T::value_type> s1(f1.begin(), f1.end());
+ std::set<typename T::value_type> s2(f2.begin(), f2.end());
+ return s1 == s2;
+}
+
bool DeviceDescriptor::equals(const sp<DeviceDescriptor>& other) const
{
// Devices are considered equal if they:
// - are of the same type (a device type cannot be AUDIO_DEVICE_NONE)
// - have the same address
+ // - have the same encodingFormats (if device supports encoding)
if (other == 0) {
return false;
}
- return (mDeviceType == other->mDeviceType) && (mAddress == other->mAddress);
+
+ return (mDeviceType == other->mDeviceType) && (mAddress == other->mAddress) &&
+ checkEqual(mEncodedFormats, other->mEncodedFormats);
+}
+
+bool DeviceDescriptor::hasCurrentEncodedFormat() const
+{
+ if (!device_has_encoding_capability(type())) {
+ return true;
+ }
+ if (mEncodedFormats.isEmpty()) {
+ return true;
+ }
+
+ return (mCurrentEncodedFormat != AUDIO_FORMAT_DEFAULT);
+}
+
+bool DeviceDescriptor::supportsFormat(audio_format_t format)
+{
+ if (mEncodedFormats.isEmpty()) {
+ return true;
+ }
+
+ for (const auto& devFormat : mEncodedFormats) {
+ if (devFormat == format) {
+ return true;
+ }
+ }
+ return false;
}
void DeviceVector::refreshTypes()
@@ -161,12 +209,17 @@
return deviceTypes;
}
-sp<DeviceDescriptor> DeviceVector::getDevice(audio_devices_t type, const String8& address) const
+sp<DeviceDescriptor> DeviceVector::getDevice(audio_devices_t type, const String8& address,
+ audio_format_t format) const
{
sp<DeviceDescriptor> device;
for (size_t i = 0; i < size(); i++) {
if (itemAt(i)->type() == type) {
- if (address == "" || itemAt(i)->address() == address) {
+ // Assign device if address is empty or matches and
+ // format is default or matches
+ if (((address == "" || itemAt(i)->address() == address) &&
+ format == AUDIO_FORMAT_DEFAULT) ||
+ itemAt(i)->supportsFormat(format)) {
device = itemAt(i);
if (itemAt(i)->address() == address) {
break;
@@ -174,8 +227,8 @@
}
}
}
- ALOGV("DeviceVector::%s() for type %08x address \"%s\" found %p",
- __func__, type, address.string(), device.get());
+ ALOGV("DeviceVector::%s() for type %08x address \"%s\" found %p format %08x",
+ __func__, type, address.string(), device.get(), format);
return device;
}
diff --git a/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp b/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
index 80af88d..85d9bce 100644
--- a/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
@@ -52,6 +52,9 @@
sp<DeviceDescriptor> devDesc = new DeviceDescriptor(device);
devDesc->setAddress(address);
+ addDynamicDevice(devDesc);
+ // Reciprocally attach the device to the module
+ devDesc->attach(this);
profile->addSupportedDevice(devDesc);
return addOutputProfile(profile);
@@ -97,6 +100,9 @@
{
for (size_t i = 0; i < mOutputProfiles.size(); i++) {
if (mOutputProfiles[i]->getName() == name) {
+ for (const auto &device : mOutputProfiles[i]->getSupportedDevices()) {
+ removeDynamicDevice(device);
+ }
mOutputProfiles.removeAt(i);
break;
}
@@ -114,6 +120,9 @@
sp<DeviceDescriptor> devDesc = new DeviceDescriptor(device);
devDesc->setAddress(address);
+ addDynamicDevice(devDesc);
+ // Reciprocally attach the device to the module
+ devDesc->attach(this);
profile->addSupportedDevice(devDesc);
ALOGV("addInputProfile() name %s rate %d mask 0x%08x",
@@ -126,6 +135,9 @@
{
for (size_t i = 0; i < mInputProfiles.size(); i++) {
if (mInputProfiles[i]->getName() == name) {
+ for (const auto &device : mInputProfiles[i]->getSupportedDevices()) {
+ removeDynamicDevice(device);
+ }
mInputProfiles.removeAt(i);
break;
}
@@ -247,6 +259,7 @@
}
}
mDeclaredDevices.dump(dst, String8("Declared"), 2, true);
+ mDynamicDevices.dump(dst, String8("Dynamic"), 2, true);
mRoutes.dump(dst, 2);
}
@@ -260,44 +273,153 @@
return nullptr;
}
-sp <HwModule> HwModuleCollection::getModuleForDevice(audio_devices_t device) const
+sp <HwModule> HwModuleCollection::getModuleForDeviceTypes(audio_devices_t type,
+ audio_format_t encodedFormat) const
{
for (const auto& module : *this) {
- const auto& profiles = audio_is_output_device(device) ?
+ const auto& profiles = audio_is_output_device(type) ?
module->getOutputProfiles() : module->getInputProfiles();
for (const auto& profile : profiles) {
- if (profile->supportDevice(device)) {
- return module;
+ if (profile->supportsDeviceTypes(type)) {
+ if (encodedFormat != AUDIO_FORMAT_DEFAULT) {
+ DeviceVector declaredDevices = module->getDeclaredDevices();
+ sp <DeviceDescriptor> deviceDesc =
+ declaredDevices.getDevice(type, String8(), encodedFormat);
+ if (deviceDesc) {
+ return module;
+ }
+ } else {
+ return module;
+ }
}
}
}
return nullptr;
}
-sp<DeviceDescriptor> HwModuleCollection::getDeviceDescriptor(const audio_devices_t device,
- const char *device_address,
- const char *device_name,
+sp<HwModule> HwModuleCollection::getModuleForDevice(const sp<DeviceDescriptor> &device,
+ audio_format_t encodedFormat) const
+{
+ return getModuleForDeviceTypes(device->type(), encodedFormat);
+}
+
+DeviceVector HwModuleCollection::getAvailableDevicesFromModuleName(
+ const char *name, const DeviceVector &availableDevices) const
+{
+ sp<HwModule> module = getModuleFromName(name);
+ if (module == nullptr) {
+ return DeviceVector();
+ }
+ return availableDevices.getDevicesFromHwModule(module->getHandle());
+}
+
+sp<DeviceDescriptor> HwModuleCollection::getDeviceDescriptor(const audio_devices_t deviceType,
+ const char *address,
+ const char *name,
+ const audio_format_t encodedFormat,
+ bool allowToCreate,
bool matchAddress) const
{
- String8 address = (device_address == nullptr || !matchAddress) ?
- String8("") : String8(device_address);
+ String8 devAddress = (address == nullptr || !matchAddress) ? String8("") : String8(address);
// handle legacy remote submix case where the address was not always specified
- if (device_distinguishes_on_address(device) && (address.length() == 0)) {
- address = String8("0");
+ if (device_distinguishes_on_address(deviceType) && (devAddress.length() == 0)) {
+ devAddress = String8("0");
}
for (const auto& hwModule : *this) {
- DeviceVector declaredDevices = hwModule->getDeclaredDevices();
- sp<DeviceDescriptor> deviceDesc = declaredDevices.getDevice(device, address);
- if (deviceDesc) {
- return deviceDesc;
+ DeviceVector moduleDevices = hwModule->getAllDevices();
+ auto moduleDevice = moduleDevices.getDevice(deviceType, devAddress, encodedFormat);
+ if (moduleDevice) {
+ if (encodedFormat != AUDIO_FORMAT_DEFAULT) {
+ moduleDevice->setEncodedFormat(encodedFormat);
+ if (moduleDevice->address() != devAddress) {
+ moduleDevice->setAddress(devAddress);
+ }
+ }
+ if (allowToCreate) {
+ moduleDevice->attach(hwModule);
+ }
+ return moduleDevice;
}
}
+ if (!allowToCreate) {
+ ALOGE("%s: could not find HW module for device %s %04x address %s", __FUNCTION__,
+ name, deviceType, address);
+ return nullptr;
+ }
+ return createDevice(deviceType, address, name, encodedFormat);
+}
- sp<DeviceDescriptor> devDesc = new DeviceDescriptor(device);
- devDesc->setName(String8(device_name));
- devDesc->setAddress(address);
- return devDesc;
+sp<DeviceDescriptor> HwModuleCollection::createDevice(const audio_devices_t type,
+ const char *address,
+ const char *name,
+ const audio_format_t encodedFormat) const
+{
+ sp<HwModule> hwModule = getModuleForDeviceTypes(type, encodedFormat);
+ if (hwModule == 0) {
+ ALOGE("%s: could not find HW module for device %04x address %s", __FUNCTION__, type,
+ address);
+ return nullptr;
+ }
+ sp<DeviceDescriptor> device = new DeviceDescriptor(type, String8(name));
+ device->setName(String8(name));
+ device->setAddress(String8(address));
+ device->setEncodedFormat(encodedFormat);
+
+ // Add the device to the list of dynamic devices
+ hwModule->addDynamicDevice(device);
+ // Reciprocally attach the device to the module
+ device->attach(hwModule);
+ ALOGD("%s: adding dynamic device %s to module %s", __FUNCTION__,
+ device->toString().c_str(), hwModule->getName());
+
+ const auto &profiles = (audio_is_output_device(type) ? hwModule->getOutputProfiles() :
+ hwModule->getInputProfiles());
+ for (const auto &profile : profiles) {
+ // Add the device as supported to all profile supporting "weakly" or not the device
+ // according to its type
+ if (profile->supportsDevice(device, false /*matchAdress*/)) {
+
+ // @todo quid of audio profile? import the profile from device of the same type?
+ const auto &isoTypeDeviceForProfile =
+ profile->getSupportedDevices().getDevice(type, String8(), AUDIO_FORMAT_DEFAULT);
+ device->importAudioPort(isoTypeDeviceForProfile, true /* force */);
+
+ ALOGV("%s: adding device %s to profile %s", __FUNCTION__,
+ device->toString().c_str(), profile->getTagName().c_str());
+ profile->addSupportedDevice(device);
+ }
+ }
+ return device;
+}
+
+void HwModuleCollection::cleanUpForDevice(const sp<DeviceDescriptor> &device)
+{
+ for (const auto& hwModule : *this) {
+ DeviceVector moduleDevices = hwModule->getAllDevices();
+ if (!moduleDevices.contains(device)) {
+ continue;
+ }
+ device->detach();
+ // Only remove from dynamic list, not from declared list!!!
+ if (!hwModule->getDynamicDevices().contains(device)) {
+ return;
+ }
+ hwModule->removeDynamicDevice(device);
+ ALOGV("%s: removed dynamic device %s from module %s", __FUNCTION__,
+ device->toString().c_str(), hwModule->getName());
+
+ const IOProfileCollection &profiles = audio_is_output_device(device->type()) ?
+ hwModule->getOutputProfiles() : hwModule->getInputProfiles();
+ for (const auto &profile : profiles) {
+ // For cleanup, strong match is required
+ if (profile->supportsDevice(device, true /*matchAdress*/)) {
+ ALOGV("%s: removing device %s from profile %s", __FUNCTION__,
+ device->toString().c_str(), profile->getTagName().c_str());
+ profile->removeSupportedDevice(device);
+ }
+ }
+ }
}
void HwModuleCollection::dump(String8 *dst) const
diff --git a/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp b/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp
index 3788244..fe2eaee 100644
--- a/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp
@@ -25,11 +25,7 @@
namespace android {
-// checks if the IO profile is compatible with specified parameters.
-// Sampling rate, format and channel mask must be specified in order to
-// get a valid a match
-bool IOProfile::isCompatibleProfile(audio_devices_t device,
- const String8& address,
+bool IOProfile::isCompatibleProfile(const DeviceVector &devices,
uint32_t samplingRate,
uint32_t *updatedSamplingRate,
audio_format_t format,
@@ -46,14 +42,8 @@
getType() == AUDIO_PORT_TYPE_MIX && getRole() == AUDIO_PORT_ROLE_SINK;
ALOG_ASSERT(isPlaybackThread != isRecordThread);
-
- if (device != AUDIO_DEVICE_NONE) {
- // just check types if multiple devices are selected
- if (popcount(device & ~AUDIO_DEVICE_BIT_IN) > 1) {
- if ((mSupportedDevices.types() & device) != device) {
- return false;
- }
- } else if (mSupportedDevices.getDevice(device, address) == 0) {
+ if (!devices.isEmpty()) {
+ if (!mSupportedDevices.containsAllDevices(devices)) {
return false;
}
}
diff --git a/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp b/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp
index 1154654..98d375c 100644
--- a/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp
@@ -140,6 +140,8 @@
static constexpr const char *roleSource = "source"; /**< <attribute role source value>. */
/** optional: device address, char string less than 64. */
static constexpr const char *address = "address";
+ /** optional: the list of encoded audio formats that are known to be supported. */
+ static constexpr const char *encodedFormats = "encodedFormats";
};
static Return<Element> deserialize(const xmlNode *cur, PtrSerializingCtx serializingContext);
@@ -511,7 +513,13 @@
ALOGW("%s: bad type %08x", __func__, type);
return Status::fromStatusT(BAD_VALUE);
}
- Element deviceDesc = new DeviceDescriptor(type, String8(name.c_str()));
+ std::string encodedFormatsLiteral = getXmlAttribute(cur, Attributes::encodedFormats);
+ ALOGV("%s: %s %s=%s", __func__, tag, Attributes::encodedFormats, encodedFormatsLiteral.c_str());
+ FormatVector encodedFormats;
+ if (!encodedFormatsLiteral.empty()) {
+ encodedFormats = formatsFromString(encodedFormatsLiteral, " ");
+ }
+ Element deviceDesc = new DeviceDescriptor(type, encodedFormats, String8(name.c_str()));
std::string address = getXmlAttribute(cur, Attributes::address);
if (!address.empty()) {
diff --git a/services/audiopolicy/common/managerdefinitions/src/VolumeCurve.cpp b/services/audiopolicy/common/managerdefinitions/src/VolumeCurve.cpp
index 620f361..2625733 100644
--- a/services/audiopolicy/common/managerdefinitions/src/VolumeCurve.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/VolumeCurve.cpp
@@ -26,16 +26,22 @@
{
ALOG_ASSERT(!mCurvePoints.isEmpty(), "Invalid volume curve");
- size_t nbCurvePoints = mCurvePoints.size();
- // the volume index in the UI is relative to the min and max volume indices for this stream
- int nbSteps = 1 + mCurvePoints[nbCurvePoints - 1].mIndex - mCurvePoints[0].mIndex;
if (indexInUi < volIndexMin) {
+ // an index of 0 means mute request when volIndexMin > 0
+ if (indexInUi == 0) {
+ ALOGV("VOLUME forcing mute for index 0 with min index %d", volIndexMin);
+ return VOLUME_MIN_DB;
+ }
ALOGV("VOLUME remapping index from %d to min index %d", indexInUi, volIndexMin);
indexInUi = volIndexMin;
} else if (indexInUi > volIndexMax) {
ALOGV("VOLUME remapping index from %d to max index %d", indexInUi, volIndexMax);
indexInUi = volIndexMax;
}
+
+ size_t nbCurvePoints = mCurvePoints.size();
+ // the volume index in the UI is relative to the min and max volume indices for this stream
+ int nbSteps = 1 + mCurvePoints[nbCurvePoints - 1].mIndex - mCurvePoints[0].mIndex;
int volIdx = (nbSteps * (indexInUi - volIndexMin)) / (volIndexMax - volIndexMin);
// Where would this volume index been inserted in the curve point
diff --git a/services/audiopolicy/config/audio_policy_configuration_generic.xml b/services/audiopolicy/config/audio_policy_configuration_generic.xml
index 58768c3..40dcc22 100644
--- a/services/audiopolicy/config/audio_policy_configuration_generic.xml
+++ b/services/audiopolicy/config/audio_policy_configuration_generic.xml
@@ -37,4 +37,10 @@
<!-- End of Volume section -->
+ <!-- Surround Sound configuration -->
+
+ <xi:include href="surround_sound_configuration_5_0.xml"/>
+
+ <!-- End of Surround Sound configuration -->
+
</audioPolicyConfiguration>
diff --git a/services/audiopolicy/config/audio_policy_configuration_generic_tv.xml b/services/audiopolicy/config/audio_policy_configuration_generic_tv.xml
new file mode 100644
index 0000000..5f1ca31
--- /dev/null
+++ b/services/audiopolicy/config/audio_policy_configuration_generic_tv.xml
@@ -0,0 +1,49 @@
+<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
+<!-- Copyright (C) 2019 The Android Open Source Project
+
+ Licensed under the Apache License, Version 2.0 (the "License");
+ you may not use this file except in compliance with the License.
+ You may obtain a copy of the License at
+
+ http://www.apache.org/licenses/LICENSE-2.0
+
+ Unless required by applicable law or agreed to in writing, software
+ distributed under the License is distributed on an "AS IS" BASIS,
+ WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ See the License for the specific language governing permissions and
+ limitations under the License.
+-->
+
+<audioPolicyConfiguration version="1.0" xmlns:xi="http://www.w3.org/2001/XInclude">
+ <!-- version section contains a “version” tag in the form “major.minor” e.g version=”1.0” -->
+
+ <!-- Global configuration Decalaration -->
+ <globalConfiguration speaker_drc_enabled="false"/>
+
+ <modules>
+ <!-- Primary Audio HAL -->
+ <xi:include href="primary_audio_policy_configuration_tv.xml"/>
+
+ <!-- Usb Audio HAL -->
+ <xi:include href="usb_audio_policy_configuration.xml"/>
+
+ <!-- Remote Submix Audio HAL -->
+ <xi:include href="r_submix_audio_policy_configuration.xml"/>
+
+ </modules>
+ <!-- End of Modules section -->
+
+ <!-- Volume section -->
+
+ <xi:include href="audio_policy_volumes.xml"/>
+ <xi:include href="default_volume_tables.xml"/>
+
+ <!-- End of Volume section -->
+
+ <!-- Surround Sound configuration -->
+
+ <xi:include href="surround_sound_configuration_5_0.xml"/>
+
+ <!-- End of Surround Sound configuration -->
+
+</audioPolicyConfiguration>
diff --git a/services/audiopolicy/config/audio_policy_configuration_stub.xml b/services/audiopolicy/config/audio_policy_configuration_stub.xml
index 26c381f..8350eb8 100644
--- a/services/audiopolicy/config/audio_policy_configuration_stub.xml
+++ b/services/audiopolicy/config/audio_policy_configuration_stub.xml
@@ -15,6 +15,9 @@
-->
<audioPolicyConfiguration version="1.0" xmlns:xi="http://www.w3.org/2001/XInclude">
+ <!-- Global configuration Decalaration -->
+ <globalConfiguration speaker_drc_enabled="false"/>
+
<modules>
<!-- Stub Audio HAL -->
<xi:include href="stub_audio_policy_configuration.xml"/>
@@ -26,5 +29,6 @@
<xi:include href="audio_policy_volumes.xml"/>
<xi:include href="default_volume_tables.xml"/>
+ <xi:include href="surround_sound_configuration_5_0.xml"/>
</audioPolicyConfiguration>
diff --git a/services/audiopolicy/config/primary_audio_policy_configuration.xml b/services/audiopolicy/config/primary_audio_policy_configuration.xml
index 5b7ae7f..eedc96b 100644
--- a/services/audiopolicy/config/primary_audio_policy_configuration.xml
+++ b/services/audiopolicy/config/primary_audio_policy_configuration.xml
@@ -1,5 +1,5 @@
<?xml version="1.0" encoding="UTF-8"?>
-<!-- Default Primary Audio HAL Module Audio Policy Configuration include flie -->
+<!-- Default Primary Audio HAL Module Audio Policy Configuration include file -->
<module name="primary" halVersion="2.0">
<attachedDevices>
<item>Speaker</item>
diff --git a/services/audiopolicy/config/primary_audio_policy_configuration_tv.xml b/services/audiopolicy/config/primary_audio_policy_configuration_tv.xml
new file mode 100644
index 0000000..826015a
--- /dev/null
+++ b/services/audiopolicy/config/primary_audio_policy_configuration_tv.xml
@@ -0,0 +1,26 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!-- Default Primary Audio HAL Module Audio Policy Configuration include file for TV -->
+<module name="primary" halVersion="2.0">
+ <attachedDevices>
+ <item>Speaker</item>
+ </attachedDevices>
+ <defaultOutputDevice>Speaker</defaultOutputDevice>
+ <mixPorts>
+ <mixPort name="primary output" role="source" flags="AUDIO_OUTPUT_FLAG_PRIMARY">
+ <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+ samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+ </mixPort>
+ <mixPort name="direct" role="source" flags="AUDIO_OUTPUT_FLAG_DIRECT" />
+ <mixPort name="tunnel" role="source"
+ flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_HW_AV_SYNC" />
+ </mixPorts>
+ <devicePorts>
+ <devicePort tagName="Speaker" type="AUDIO_DEVICE_OUT_SPEAKER" role="sink" />
+ <devicePort tagName="Out Aux Digital" type="AUDIO_DEVICE_OUT_AUX_DIGITAL" role="sink"
+ encodedFormats="AUDIO_FORMAT_AC3 AUDIO_FORMAT_IEC61937" />
+ </devicePorts>
+ <routes>
+ <route type="mix" sink="Speaker" sources="primary output"/>
+ <route type="mix" sink="Out Aux Digital" sources="primary output,direct,tunnel"/>
+ </routes>
+</module>
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/device_for_strategy_accessibility.pfw b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/device_for_strategy_accessibility.pfw
index eb11980..7c87c80 100644
--- a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/device_for_strategy_accessibility.pfw
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/device_for_strategy_accessibility.pfw
@@ -28,6 +28,7 @@
TelephonyMode IsNot InCall
TelephonyMode IsNot InCommunication
AvailableOutputDevices Includes RemoteSubmix
+ AvailableOutputDevicesAddresses Includes 0
component: /Policy/policy/strategies/accessibility/selected_output_devices/mask
remote_submix = 1
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/device_for_strategy_dtmf.pfw b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/device_for_strategy_dtmf.pfw
index 883c741..c830c42 100644
--- a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/device_for_strategy_dtmf.pfw
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/device_for_strategy_dtmf.pfw
@@ -20,6 +20,7 @@
TelephonyMode IsNot InCall
TelephonyMode IsNot InCommunication
AvailableOutputDevices Includes RemoteSubmix
+ AvailableOutputDevicesAddresses Includes 0
component: /Policy/policy/strategies/dtmf/selected_output_devices/mask
remote_submix = 1
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/device_for_strategy_enforced_audible.pfw b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/device_for_strategy_enforced_audible.pfw
index f504631..c641138 100644
--- a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/device_for_strategy_enforced_audible.pfw
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/device_for_strategy_enforced_audible.pfw
@@ -61,6 +61,7 @@
domain: Device2
conf: RemoteSubmix
AvailableOutputDevices Includes RemoteSubmix
+ AvailableOutputDevicesAddresses Includes 0
component: /Policy/policy/strategies/enforced_audible/selected_output_devices/mask
remote_submix = 1
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/device_for_strategy_media.pfw b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/device_for_strategy_media.pfw
index bdb6ae0..f8bab3d 100644
--- a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/device_for_strategy_media.pfw
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/device_for_strategy_media.pfw
@@ -19,6 +19,7 @@
domain: Device2
conf: RemoteSubmix
AvailableOutputDevices Includes RemoteSubmix
+ AvailableOutputDevicesAddresses Includes 0
component: /Policy/policy/strategies/media/selected_output_devices/mask
speaker = 0
diff --git a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/device_for_strategy_rerouting.pfw b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/device_for_strategy_rerouting.pfw
index 04e62f7..28a3629 100644
--- a/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/device_for_strategy_rerouting.pfw
+++ b/services/audiopolicy/engineconfigurable/parameter-framework/examples/Settings/device_for_strategy_rerouting.pfw
@@ -24,6 +24,7 @@
domain: Device2
conf: RemoteSubmix
AvailableOutputDevices Includes RemoteSubmix
+ AvailableOutputDevicesAddresses Includes 0
component: /Policy/policy/strategies/rerouting/selected_output_devices/mask
remote_submix = 1
diff --git a/services/audiopolicy/engineconfigurable/wrapper/ParameterManagerWrapper.cpp b/services/audiopolicy/engineconfigurable/wrapper/ParameterManagerWrapper.cpp
index fc6c1e4..1934fa4 100644
--- a/services/audiopolicy/engineconfigurable/wrapper/ParameterManagerWrapper.cpp
+++ b/services/audiopolicy/engineconfigurable/wrapper/ParameterManagerWrapper.cpp
@@ -295,8 +295,8 @@
auto criterionType = criterion->getCriterionType();
int deviceAddressId;
- if (not criterionType->getNumericalValue(devDesc->mAddress.string(), deviceAddressId)) {
- ALOGE("%s: unknown device address reported (%s)", __FUNCTION__, devDesc->mAddress.c_str());
+ if (not criterionType->getNumericalValue(devDesc->address().string(), deviceAddressId)) {
+ ALOGW("%s: unknown device address reported (%s)", __FUNCTION__, devDesc->address().c_str());
return BAD_TYPE;
}
int currentValueMask = criterion->getCriterionState();
diff --git a/services/audiopolicy/engineconfigurable/wrapper/config/policy_criterion_types.xml.in b/services/audiopolicy/engineconfigurable/wrapper/config/policy_criterion_types.xml.in
index 6cb799f..fe17369 100644
--- a/services/audiopolicy/engineconfigurable/wrapper/config/policy_criterion_types.xml.in
+++ b/services/audiopolicy/engineconfigurable/wrapper/config/policy_criterion_types.xml.in
@@ -16,7 +16,12 @@
<criterion_types>
<criterion_type name="OutputDevicesMaskType" type="inclusive"/>
<criterion_type name="InputDevicesMaskType" type="inclusive"/>
- <criterion_type name="OutputDevicesAddressesType" type="inclusive"/>
+ <criterion_type name="OutputDevicesAddressesType" type="inclusive">
+ <values>
+ <!-- legacy remote submix -->
+ <value literal="0" numerical="1"/>
+ </values>
+ </criterion_type>
<criterion_type name="InputDevicesAddressesType" type="inclusive"/>
<criterion_type name="AndroidModeType" type="exclusive"/>
<criterion_type name="BooleanType" type="exclusive">
diff --git a/services/audiopolicy/enginedefault/src/Engine.cpp b/services/audiopolicy/enginedefault/src/Engine.cpp
index 0ef6f52..cc5a025 100644
--- a/services/audiopolicy/enginedefault/src/Engine.cpp
+++ b/services/audiopolicy/enginedefault/src/Engine.cpp
@@ -322,7 +322,7 @@
// a primary device
// FIXME: this is not the right way of solving this problem
audio_devices_t availPrimaryOutputDevices =
- (primaryOutput->supportedDevices() | AUDIO_DEVICE_OUT_HEARING_AID) &
+ (primaryOutput->supportedDevices().types() | AUDIO_DEVICE_OUT_HEARING_AID) &
availableOutputDevices.types();
if (((availableInputDevices.types() &
@@ -475,7 +475,7 @@
// compressed format as they would likely not be mixed and dropped.
for (size_t i = 0; i < outputs.size(); i++) {
sp<AudioOutputDescriptor> desc = outputs.valueAt(i);
- audio_devices_t devices = desc->device() &
+ audio_devices_t devices = desc->devices().types() &
(AUDIO_DEVICE_OUT_HDMI | AUDIO_DEVICE_OUT_SPDIF | AUDIO_DEVICE_OUT_HDMI_ARC);
if (desc->isActive() && !audio_is_linear_pcm(desc->mFormat) &&
devices != AUDIO_DEVICE_NONE) {
@@ -506,7 +506,7 @@
if (strategy != STRATEGY_SONIFICATION) {
// no sonification on remote submix (e.g. WFD)
if (availableOutputDevices.getDevice(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
- String8("0")) != 0) {
+ String8("0"), AUDIO_FORMAT_DEFAULT) != 0) {
device2 = availableOutputDevices.types() & AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
}
}
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index 5544821..cf9c298 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -81,43 +81,49 @@
status_t AudioPolicyManager::setDeviceConnectionState(audio_devices_t device,
audio_policy_dev_state_t state,
const char *device_address,
- const char *device_name)
+ const char *device_name,
+ audio_format_t encodedFormat)
{
- status_t status = setDeviceConnectionStateInt(device, state, device_address, device_name);
+ status_t status = setDeviceConnectionStateInt(device, state, device_address,
+ device_name, encodedFormat);
nextAudioPortGeneration();
return status;
}
-void AudioPolicyManager::broadcastDeviceConnectionState(audio_devices_t device,
- audio_policy_dev_state_t state,
- const String8 &device_address)
+void AudioPolicyManager::broadcastDeviceConnectionState(const sp<DeviceDescriptor> &device,
+ audio_policy_dev_state_t state)
{
- AudioParameter param(device_address);
+ AudioParameter param(device->address());
const String8 key(state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE ?
AudioParameter::keyStreamConnect : AudioParameter::keyStreamDisconnect);
- param.addInt(key, device);
+ param.addInt(key, device->type());
mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
}
-status_t AudioPolicyManager::setDeviceConnectionStateInt(audio_devices_t device,
+status_t AudioPolicyManager::setDeviceConnectionStateInt(audio_devices_t deviceType,
audio_policy_dev_state_t state,
const char *device_address,
- const char *device_name)
+ const char *device_name,
+ audio_format_t encodedFormat)
{
- ALOGV("setDeviceConnectionStateInt() device: 0x%X, state %d, address %s name %s",
- device, state, device_address, device_name);
+ ALOGV("setDeviceConnectionStateInt() device: 0x%X, state %d, address %s name %s format 0x%X",
+ deviceType, state, device_address, device_name, encodedFormat);
// connect/disconnect only 1 device at a time
- if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE;
+ if (!audio_is_output_device(deviceType) && !audio_is_input_device(deviceType)) return BAD_VALUE;
- sp<DeviceDescriptor> devDesc =
- mHwModules.getDeviceDescriptor(device, device_address, device_name);
+ sp<DeviceDescriptor> device =
+ mHwModules.getDeviceDescriptor(deviceType, device_address, device_name, encodedFormat,
+ state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE);
+ if (device == 0) {
+ return INVALID_OPERATION;
+ }
// handle output devices
- if (audio_is_output_device(device)) {
+ if (audio_is_output_device(deviceType)) {
SortedVector <audio_io_handle_t> outputs;
- ssize_t index = mAvailableOutputDevices.indexOf(devDesc);
+ ssize_t index = mAvailableOutputDevices.indexOf(device);
// save a copy of the opened output descriptors before any output is opened or closed
// by checkOutputsForDevice(). This will be needed by checkOutputForAllStrategies()
@@ -127,21 +133,23 @@
// handle output device connection
case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: {
if (index >= 0) {
- ALOGW("setDeviceConnectionState() device already connected: %x", device);
+ ALOGW("%s() device already connected: %s", __func__, device->toString().c_str());
return INVALID_OPERATION;
}
- ALOGV("setDeviceConnectionState() connecting device %x", device);
+ ALOGV("%s() connecting device %s format %x",
+ __func__, device->toString().c_str(), encodedFormat);
// register new device as available
- index = mAvailableOutputDevices.add(devDesc);
+ index = mAvailableOutputDevices.add(device);
if (index >= 0) {
- sp<HwModule> module = mHwModules.getModuleForDevice(device);
+ sp<HwModule> module = mHwModules.getModuleForDevice(device, encodedFormat);
if (module == 0) {
- ALOGD("setDeviceConnectionState() could not find HW module for device %08x",
- device);
- mAvailableOutputDevices.remove(devDesc);
+ ALOGD("setDeviceConnectionState() could not find HW module for device %s",
+ device->toString().c_str());
+ mAvailableOutputDevices.remove(device);
return INVALID_OPERATION;
}
+ ALOGV("setDeviceConnectionState() module name=%s", module->getName());
mAvailableOutputDevices[index]->attach(module);
} else {
return NO_MEMORY;
@@ -149,48 +157,51 @@
// Before checking outputs, broadcast connect event to allow HAL to retrieve dynamic
// parameters on newly connected devices (instead of opening the outputs...)
- broadcastDeviceConnectionState(device, state, devDesc->address());
+ broadcastDeviceConnectionState(device, state);
- if (checkOutputsForDevice(devDesc, state, outputs, devDesc->address()) != NO_ERROR) {
- mAvailableOutputDevices.remove(devDesc);
+ if (checkOutputsForDevice(device, state, outputs) != NO_ERROR) {
+ mAvailableOutputDevices.remove(device);
- broadcastDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
- devDesc->address());
+ mHwModules.cleanUpForDevice(device);
+
+ broadcastDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE);
return INVALID_OPERATION;
}
// Propagate device availability to Engine
- mEngine->setDeviceConnectionState(devDesc, state);
+ mEngine->setDeviceConnectionState(device, state);
// outputs should never be empty here
ALOG_ASSERT(outputs.size() != 0, "setDeviceConnectionState():"
"checkOutputsForDevice() returned no outputs but status OK");
- ALOGV("setDeviceConnectionState() checkOutputsForDevice() returned %zu outputs",
- outputs.size());
+ ALOGV("%s() checkOutputsForDevice() returned %zu outputs", __func__, outputs.size());
} break;
// handle output device disconnection
case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: {
if (index < 0) {
- ALOGW("setDeviceConnectionState() device not connected: %x", device);
+ ALOGW("%s() device not connected: %s", __func__, device->toString().c_str());
return INVALID_OPERATION;
}
- ALOGV("setDeviceConnectionState() disconnecting output device %x", device);
+ ALOGV("%s() disconnecting output device %s", __func__, device->toString().c_str());
// Send Disconnect to HALs
- broadcastDeviceConnectionState(device, state, devDesc->address());
+ broadcastDeviceConnectionState(device, state);
// remove device from available output devices
- mAvailableOutputDevices.remove(devDesc);
+ mAvailableOutputDevices.remove(device);
- checkOutputsForDevice(devDesc, state, outputs, devDesc->address());
+ checkOutputsForDevice(device, state, outputs);
+
+ // Reset active device codec
+ device->setEncodedFormat(AUDIO_FORMAT_DEFAULT);
// Propagate device availability to Engine
- mEngine->setDeviceConnectionState(devDesc, state);
+ mEngine->setDeviceConnectionState(device, state);
} break;
default:
- ALOGE("setDeviceConnectionState() invalid state: %x", state);
+ ALOGE("%s() invalid state: %x", __func__, state);
return BAD_VALUE;
}
@@ -199,8 +210,8 @@
if (!outputs.isEmpty()) {
for (audio_io_handle_t output : outputs) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(output);
- // close unused outputs after device disconnection or direct outputs that have been
- // opened by checkOutputsForDevice() to query dynamic parameters
+ // close unused outputs after device disconnection or direct outputs that have
+ // been opened by checkOutputsForDevice() to query dynamic parameters
if ((state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) ||
(((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) &&
(desc->mDirectOpenCount == 0))) {
@@ -214,29 +225,28 @@
});
if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
- audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
- updateCallRouting(newDevice);
+ DeviceVector newDevices = getNewOutputDevices(mPrimaryOutput, false /*fromCache*/);
+ updateCallRouting(newDevices);
}
- const audio_devices_t msdOutDevice = getModuleDeviceTypes(
- mAvailableOutputDevices, AUDIO_HARDWARE_MODULE_ID_MSD);
+ const DeviceVector msdOutDevices = getMsdAudioOutDevices();
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (desc != mPrimaryOutput)) {
- audio_devices_t newDevice = getNewOutputDevice(desc, true /*fromCache*/);
+ DeviceVector newDevices = getNewOutputDevices(desc, true /*fromCache*/);
// do not force device change on duplicated output because if device is 0, it will
// also force a device 0 for the two outputs it is duplicated to which may override
// a valid device selection on those outputs.
- bool force = (msdOutDevice == AUDIO_DEVICE_NONE || msdOutDevice != desc->device())
+ bool force = (msdOutDevices.isEmpty() || msdOutDevices != desc->devices())
&& !desc->isDuplicated()
- && (!device_distinguishes_on_address(device)
+ && (!device_distinguishes_on_address(deviceType)
// always force when disconnecting (a non-duplicated device)
|| (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE));
- setOutputDevice(desc, newDevice, force, 0);
+ setOutputDevices(desc, newDevices, force, 0);
}
}
if (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) {
- cleanUpForDevice(devDesc);
+ cleanUpForDevice(device);
}
mpClientInterface->onAudioPortListUpdate();
@@ -244,67 +254,66 @@
} // end if is output device
// handle input devices
- if (audio_is_input_device(device)) {
+ if (audio_is_input_device(deviceType)) {
SortedVector <audio_io_handle_t> inputs;
- ssize_t index = mAvailableInputDevices.indexOf(devDesc);
+ ssize_t index = mAvailableInputDevices.indexOf(device);
switch (state)
{
// handle input device connection
case AUDIO_POLICY_DEVICE_STATE_AVAILABLE: {
if (index >= 0) {
- ALOGW("setDeviceConnectionState() device already connected: %d", device);
+ ALOGW("%s() device already connected: %s", __func__, device->toString().c_str());
return INVALID_OPERATION;
}
- sp<HwModule> module = mHwModules.getModuleForDevice(device);
+ sp<HwModule> module = mHwModules.getModuleForDevice(device, AUDIO_FORMAT_DEFAULT);
if (module == NULL) {
- ALOGW("setDeviceConnectionState(): could not find HW module for device %08x",
- device);
+ ALOGW("setDeviceConnectionState(): could not find HW module for device %s",
+ device->toString().c_str());
return INVALID_OPERATION;
}
// Before checking intputs, broadcast connect event to allow HAL to retrieve dynamic
// parameters on newly connected devices (instead of opening the inputs...)
- broadcastDeviceConnectionState(device, state, devDesc->address());
+ broadcastDeviceConnectionState(device, state);
- if (checkInputsForDevice(devDesc, state, inputs, devDesc->address()) != NO_ERROR) {
- broadcastDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
- devDesc->address());
+ if (checkInputsForDevice(device, state, inputs) != NO_ERROR) {
+ broadcastDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE);
+
+ mHwModules.cleanUpForDevice(device);
+
return INVALID_OPERATION;
}
- index = mAvailableInputDevices.add(devDesc);
- if (index >= 0) {
- mAvailableInputDevices[index]->attach(module);
- } else {
+ if (mAvailableInputDevices.add(device) < 0) {
return NO_MEMORY;
}
// Propagate device availability to Engine
- mEngine->setDeviceConnectionState(devDesc, state);
+ mEngine->setDeviceConnectionState(device, state);
} break;
// handle input device disconnection
case AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE: {
if (index < 0) {
- ALOGW("setDeviceConnectionState() device not connected: %d", device);
+ ALOGW("%s() device not connected: %s", __func__, device->toString().c_str());
return INVALID_OPERATION;
}
- ALOGV("setDeviceConnectionState() disconnecting input device %x", device);
+ ALOGV("%s() disconnecting input device %s", __func__, device->toString().c_str());
// Set Disconnect to HALs
- broadcastDeviceConnectionState(device, state, devDesc->address());
+ broadcastDeviceConnectionState(device, state);
- checkInputsForDevice(devDesc, state, inputs, devDesc->address());
- mAvailableInputDevices.remove(devDesc);
+ checkInputsForDevice(device, state, inputs);
+ mAvailableInputDevices.remove(device);
// Propagate device availability to Engine
- mEngine->setDeviceConnectionState(devDesc, state);
+ mEngine->setDeviceConnectionState(device, state);
} break;
default:
- ALOGE("setDeviceConnectionState() invalid state: %x", state);
+ ALOGE("%s() invalid state: %x", __func__, state);
return BAD_VALUE;
}
@@ -314,19 +323,19 @@
updateDevicesAndOutputs();
if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
- audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
- updateCallRouting(newDevice);
+ DeviceVector newDevices = getNewOutputDevices(mPrimaryOutput, false /*fromCache*/);
+ updateCallRouting(newDevices);
}
if (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) {
- cleanUpForDevice(devDesc);
+ cleanUpForDevice(device);
}
mpClientInterface->onAudioPortListUpdate();
return NO_ERROR;
} // end if is input device
- ALOGW("setDeviceConnectionState() invalid device: %x", device);
+ ALOGW("%s() invalid device: %s", __func__, device->toString().c_str());
return BAD_VALUE;
}
@@ -334,7 +343,8 @@
const char *device_address)
{
sp<DeviceDescriptor> devDesc =
- mHwModules.getDeviceDescriptor(device, device_address, "",
+ mHwModules.getDeviceDescriptor(device, device_address, "", AUDIO_FORMAT_DEFAULT,
+ false /* allowToCreate */,
(strlen(device_address) != 0)/*matchAddress*/);
if (devDesc == 0) {
@@ -350,55 +360,65 @@
} else if (audio_is_input_device(device)) {
deviceVector = &mAvailableInputDevices;
} else {
- ALOGW("getDeviceConnectionState() invalid device type %08x", device);
+ ALOGW("%s() invalid device type %08x", __func__, device);
return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE;
}
- return (deviceVector->getDevice(device, String8(device_address)) != 0) ?
+ return (deviceVector->getDevice(
+ device, String8(device_address), AUDIO_FORMAT_DEFAULT) != 0) ?
AUDIO_POLICY_DEVICE_STATE_AVAILABLE : AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE;
}
status_t AudioPolicyManager::handleDeviceConfigChange(audio_devices_t device,
const char *device_address,
- const char *device_name)
+ const char *device_name,
+ audio_format_t encodedFormat)
{
status_t status;
String8 reply;
AudioParameter param;
int isReconfigA2dpSupported = 0;
- ALOGV("handleDeviceConfigChange(() device: 0x%X, address %s name %s",
- device, device_address, device_name);
+ ALOGV("handleDeviceConfigChange(() device: 0x%X, address %s name %s encodedFormat: 0x%X",
+ device, device_address, device_name, encodedFormat);
// connect/disconnect only 1 device at a time
if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE;
// Check if the device is currently connected
- sp<DeviceDescriptor> devDesc =
- mHwModules.getDeviceDescriptor(device, device_address, device_name);
- ssize_t index = mAvailableOutputDevices.indexOf(devDesc);
- if (index < 0) {
+ DeviceVector availableDevices = getAvailableOutputDevices();
+ DeviceVector deviceList = availableDevices.getDevicesFromTypeMask(device);
+ if (deviceList.empty()) {
// Nothing to do: device is not connected
return NO_ERROR;
}
+ sp<DeviceDescriptor> devDesc = deviceList.itemAt(0);
// For offloaded A2DP, Hw modules may have the capability to
- // configure codecs. Check if any of the loaded hw modules
- // supports this.
- // If supported, send a set parameter to configure A2DP codecs
- // and return. No need to toggle device state.
+ // configure codecs.
+ // Handle two specific cases by sending a set parameter to
+ // configure A2DP codecs. No need to toggle device state.
+ // Case 1: A2DP active device switches from primary to primary
+ // module
+ // Case 2: A2DP device config changes on primary module.
if (device & AUDIO_DEVICE_OUT_ALL_A2DP) {
- reply = mpClientInterface->getParameters(
- AUDIO_IO_HANDLE_NONE,
- String8(AudioParameter::keyReconfigA2dpSupported));
- AudioParameter repliedParameters(reply);
- repliedParameters.getInt(
- String8(AudioParameter::keyReconfigA2dpSupported), isReconfigA2dpSupported);
- if (isReconfigA2dpSupported) {
- const String8 key(AudioParameter::keyReconfigA2dp);
- param.add(key, String8("true"));
- mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
- return NO_ERROR;
+ sp<HwModule> module = mHwModules.getModuleForDeviceTypes(device, encodedFormat);
+ audio_module_handle_t primaryHandle = mPrimaryOutput->getModuleHandle();
+ if (availablePrimaryOutputDevices().contains(devDesc) &&
+ (module != 0 && module->getHandle() == primaryHandle)) {
+ reply = mpClientInterface->getParameters(
+ AUDIO_IO_HANDLE_NONE,
+ String8(AudioParameter::keyReconfigA2dpSupported));
+ AudioParameter repliedParameters(reply);
+ repliedParameters.getInt(
+ String8(AudioParameter::keyReconfigA2dpSupported), isReconfigA2dpSupported);
+ if (isReconfigA2dpSupported) {
+ const String8 key(AudioParameter::keyReconfigA2dp);
+ param.add(key, String8("true"));
+ mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
+ devDesc->setEncodedFormat(encodedFormat);
+ return NO_ERROR;
+ }
}
}
@@ -406,7 +426,8 @@
// This will force reading again the device configuration
status = setDeviceConnectionState(device,
AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
- device_address, device_name);
+ device_address, device_name,
+ devDesc->getEncodedFormat());
if (status != NO_ERROR) {
ALOGW("handleDeviceConfigChange() error disabling connection state: %d",
status);
@@ -415,7 +436,7 @@
status = setDeviceConnectionState(device,
AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
- device_address, device_name);
+ device_address, device_name, encodedFormat);
if (status != NO_ERROR) {
ALOGW("handleDeviceConfigChange() error enabling connection state: %d",
status);
@@ -425,16 +446,59 @@
return NO_ERROR;
}
-uint32_t AudioPolicyManager::updateCallRouting(audio_devices_t rxDevice, uint32_t delayMs)
+status_t AudioPolicyManager::getHwOffloadEncodingFormatsSupportedForA2DP(
+ std::vector<audio_format_t> *formats)
+{
+ ALOGV("getHwOffloadEncodingFormatsSupportedForA2DP()");
+ char *tok = NULL, *saveptr;
+ status_t status = NO_ERROR;
+ char encoding_formats_list[PROPERTY_VALUE_MAX];
+ audio_format_t format = AUDIO_FORMAT_DEFAULT;
+ // FIXME This list should not come from a property but the supported encoded
+ // formats of declared A2DP devices in primary module
+ property_get("persist.bluetooth.a2dp_offload.cap", encoding_formats_list, "");
+ tok = strtok_r(encoding_formats_list, "-", &saveptr);
+ for (;tok != NULL; tok = strtok_r(NULL, "-", &saveptr)) {
+ if (strcmp(tok, "sbc") == 0) {
+ ALOGV("%s: SBC offload supported\n",__func__);
+ format = AUDIO_FORMAT_SBC;
+ } else if (strcmp(tok, "aptx") == 0) {
+ ALOGV("%s: APTX offload supported\n",__func__);
+ format = AUDIO_FORMAT_APTX;
+ } else if (strcmp(tok, "aptxhd") == 0) {
+ ALOGV("%s: APTX HD offload supported\n",__func__);
+ format = AUDIO_FORMAT_APTX_HD;
+ } else if (strcmp(tok, "ldac") == 0) {
+ ALOGV("%s: LDAC offload supported\n",__func__);
+ format = AUDIO_FORMAT_LDAC;
+ } else if (strcmp(tok, "aac") == 0) {
+ ALOGV("%s: AAC offload supported\n",__func__);
+ format = AUDIO_FORMAT_AAC;
+ } else {
+ ALOGE("%s: undefined token - %s\n",__func__, tok);
+ continue;
+ }
+ formats->push_back(format);
+ }
+ return status;
+}
+
+uint32_t AudioPolicyManager::updateCallRouting(const DeviceVector &rxDevices, uint32_t delayMs)
{
bool createTxPatch = false;
+ bool createRxPatch = false;
uint32_t muteWaitMs = 0;
- if(!hasPrimaryOutput() || mPrimaryOutput->device() == AUDIO_DEVICE_OUT_STUB) {
+ if(!hasPrimaryOutput() || mPrimaryOutput->devices().types() == AUDIO_DEVICE_OUT_STUB) {
return muteWaitMs;
}
- audio_devices_t txDevice = getDeviceAndMixForInputSource(AUDIO_SOURCE_VOICE_COMMUNICATION);
- ALOGV("updateCallRouting device rxDevice %08x txDevice %08x", rxDevice, txDevice);
+ ALOG_ASSERT(!rxDevices.isEmpty(), "updateCallRouting() no selected output device");
+
+ audio_attributes_t attr = { .source = AUDIO_SOURCE_VOICE_COMMUNICATION };
+ auto txSourceDevice = getDeviceAndMixForAttributes(attr);
+ ALOG_ASSERT(txSourceDevice != 0, "updateCallRouting() input selected device not available");
+ ALOGV("updateCallRouting device rxDevice %s txDevice %s",
+ rxDevices.itemAt(0)->toString().c_str(), txSourceDevice->toString().c_str());
// release existing RX patch if any
if (mCallRxPatch != 0) {
@@ -447,49 +511,88 @@
mCallTxPatch.clear();
}
- // If the RX device is on the primary HW module, then use legacy routing method for voice calls
- // via setOutputDevice() on primary output.
- // Otherwise, create two audio patches for TX and RX path.
- if (availablePrimaryOutputDevices() & rxDevice) {
- muteWaitMs = setOutputDevice(mPrimaryOutput, rxDevice, true, delayMs);
+ auto telephonyRxModule =
+ mHwModules.getModuleForDeviceTypes(AUDIO_DEVICE_IN_TELEPHONY_RX, AUDIO_FORMAT_DEFAULT);
+ auto telephonyTxModule =
+ mHwModules.getModuleForDeviceTypes(AUDIO_DEVICE_OUT_TELEPHONY_TX, AUDIO_FORMAT_DEFAULT);
+ // retrieve Rx Source and Tx Sink device descriptors
+ sp<DeviceDescriptor> rxSourceDevice =
+ mAvailableInputDevices.getDevice(AUDIO_DEVICE_IN_TELEPHONY_RX,
+ String8(),
+ AUDIO_FORMAT_DEFAULT);
+ sp<DeviceDescriptor> txSinkDevice =
+ mAvailableOutputDevices.getDevice(AUDIO_DEVICE_OUT_TELEPHONY_TX,
+ String8(),
+ AUDIO_FORMAT_DEFAULT);
+
+ // RX and TX Telephony device are declared by Primary Audio HAL
+ if (isPrimaryModule(telephonyRxModule) && isPrimaryModule(telephonyTxModule) &&
+ (telephonyRxModule->getHalVersionMajor() >= 3)) {
+ if (rxSourceDevice == 0 || txSinkDevice == 0) {
+ // RX / TX Telephony device(s) is(are) not currently available
+ ALOGE("updateCallRouting() no telephony Tx and/or RX device");
+ return muteWaitMs;
+ }
+ // do not create a patch (aka Sw Bridging) if Primary HW module has declared supporting a
+ // route between telephony RX to Sink device and Source device to telephony TX
+ const auto &primaryModule = telephonyRxModule;
+ createRxPatch = !primaryModule->supportsPatch(rxSourceDevice, rxDevices.itemAt(0));
+ createTxPatch = !primaryModule->supportsPatch(txSourceDevice, txSinkDevice);
+ } else {
+ // If the RX device is on the primary HW module, then use legacy routing method for
+ // voice calls via setOutputDevice() on primary output.
+ // Otherwise, create two audio patches for TX and RX path.
+ createRxPatch = !(availablePrimaryOutputDevices().contains(rxDevices.itemAt(0))) &&
+ (rxSourceDevice != 0);
// If the TX device is also on the primary HW module, setOutputDevice() will take care
// of it due to legacy implementation. If not, create a patch.
- if ((availablePrimaryInputDevices() & txDevice & ~AUDIO_DEVICE_BIT_IN)
- == AUDIO_DEVICE_NONE) {
- createTxPatch = true;
- }
+ createTxPatch = !(availablePrimaryModuleInputDevices().contains(txSourceDevice)) &&
+ (txSinkDevice != 0);
+ }
+ // Use legacy routing method for voice calls via setOutputDevice() on primary output.
+ // Otherwise, create two audio patches for TX and RX path.
+ if (!createRxPatch) {
+ muteWaitMs = setOutputDevices(mPrimaryOutput, rxDevices, true, delayMs);
} else { // create RX path audio patch
- mCallRxPatch = createTelephonyPatch(true /*isRx*/, rxDevice, delayMs);
- createTxPatch = true;
+ mCallRxPatch = createTelephonyPatch(true /*isRx*/, rxDevices.itemAt(0), delayMs);
+ ALOG_ASSERT(createTxPatch, "No Tx Patch will be created, nor legacy routing done");
}
if (createTxPatch) { // create TX path audio patch
- mCallTxPatch = createTelephonyPatch(false /*isRx*/, txDevice, delayMs);
+ mCallTxPatch = createTelephonyPatch(false /*isRx*/, txSourceDevice, delayMs);
}
return muteWaitMs;
}
sp<AudioPatch> AudioPolicyManager::createTelephonyPatch(
- bool isRx, audio_devices_t device, uint32_t delayMs) {
+ bool isRx, const sp<DeviceDescriptor> &device, uint32_t delayMs) {
PatchBuilder patchBuilder;
- sp<DeviceDescriptor> txSourceDeviceDesc;
+ if (device == nullptr) {
+ return nullptr;
+ }
if (isRx) {
- patchBuilder.addSink(findDevice(mAvailableOutputDevices, device)).
- addSource(findDevice(mAvailableInputDevices, AUDIO_DEVICE_IN_TELEPHONY_RX));
+ patchBuilder.addSink(device).
+ addSource(mAvailableInputDevices.getDevice(
+ AUDIO_DEVICE_IN_TELEPHONY_RX, String8(), AUDIO_FORMAT_DEFAULT));
} else {
- patchBuilder.addSource(txSourceDeviceDesc = findDevice(mAvailableInputDevices, device)).
- addSink(findDevice(mAvailableOutputDevices, AUDIO_DEVICE_OUT_TELEPHONY_TX));
+ patchBuilder.addSource(device).
+ addSink(mAvailableOutputDevices.getDevice(
+ AUDIO_DEVICE_OUT_TELEPHONY_TX, String8(), AUDIO_FORMAT_DEFAULT));
}
- audio_devices_t outputDevice = isRx ? device : AUDIO_DEVICE_OUT_TELEPHONY_TX;
- SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(outputDevice, mOutputs);
- audio_io_handle_t output = selectOutput(outputs);
+ // @TODO: still ignoring the address, or not dealing platform with mutliple telephonydevices
+ const sp<DeviceDescriptor> outputDevice = isRx ?
+ device : mAvailableOutputDevices.getDevice(
+ AUDIO_DEVICE_OUT_TELEPHONY_TX, String8(), AUDIO_FORMAT_DEFAULT);
+ SortedVector<audio_io_handle_t> outputs =
+ getOutputsForDevices(DeviceVector(outputDevice), mOutputs);
+ audio_io_handle_t output = selectOutput(outputs, AUDIO_OUTPUT_FLAG_NONE, AUDIO_FORMAT_INVALID);
// request to reuse existing output stream if one is already opened to reach the target device
if (output != AUDIO_IO_HANDLE_NONE) {
sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
- ALOG_ASSERT(!outputDesc->isDuplicated(),
- "%s() %#x device output %d is duplicated", __func__, outputDevice, output);
+ ALOG_ASSERT(!outputDesc->isDuplicated(), "%s() %s device output %d is duplicated", __func__,
+ outputDevice->toString().c_str(), output);
patchBuilder.addSource(outputDesc, { .stream = AUDIO_STREAM_PATCH });
}
@@ -499,7 +602,7 @@
// call TX device but this information is not in the audio patch and logic here must be
// symmetric to the one in startInput()
for (const auto& activeDesc : mInputs.getActiveInputs()) {
- if (activeDesc->hasSameHwModuleAs(txSourceDeviceDesc)) {
+ if (activeDesc->hasSameHwModuleAs(device)) {
closeActiveClients(activeDesc);
}
}
@@ -599,17 +702,17 @@
}
if (hasPrimaryOutput()) {
- // Note that despite the fact that getNewOutputDevice() is called on the primary output,
+ // Note that despite the fact that getNewOutputDevices() is called on the primary output,
// the device returned is not necessarily reachable via this output
- audio_devices_t rxDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
+ DeviceVector rxDevices = getNewOutputDevices(mPrimaryOutput, false /*fromCache*/);
// force routing command to audio hardware when ending call
// even if no device change is needed
- if (isStateInCall(oldState) && rxDevice == AUDIO_DEVICE_NONE) {
- rxDevice = mPrimaryOutput->device();
+ if (isStateInCall(oldState) && rxDevices.isEmpty()) {
+ rxDevices = mPrimaryOutput->devices();
}
if (state == AUDIO_MODE_IN_CALL) {
- updateCallRouting(rxDevice, delayMs);
+ updateCallRouting(rxDevices, delayMs);
} else if (oldState == AUDIO_MODE_IN_CALL) {
if (mCallRxPatch != 0) {
mpClientInterface->releaseAudioPatch(mCallRxPatch->mAfPatchHandle, 0);
@@ -619,18 +722,18 @@
mpClientInterface->releaseAudioPatch(mCallTxPatch->mAfPatchHandle, 0);
mCallTxPatch.clear();
}
- setOutputDevice(mPrimaryOutput, rxDevice, force, 0);
+ setOutputDevices(mPrimaryOutput, rxDevices, force, 0);
} else {
- setOutputDevice(mPrimaryOutput, rxDevice, force, 0);
+ setOutputDevices(mPrimaryOutput, rxDevices, force, 0);
}
}
// reevaluate routing on all outputs in case tracks have been started during the call
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
- audio_devices_t newDevice = getNewOutputDevice(desc, true /*fromCache*/);
+ DeviceVector newDevices = getNewOutputDevices(desc, true /*fromCache*/);
if (state != AUDIO_MODE_IN_CALL || desc != mPrimaryOutput) {
- setOutputDevice(desc, newDevice, (newDevice != AUDIO_DEVICE_NONE), 0 /*delayMs*/);
+ setOutputDevices(desc, newDevices, !newDevices.isEmpty(), 0 /*delayMs*/);
}
}
@@ -654,7 +757,7 @@
}
void AudioPolicyManager::setForceUse(audio_policy_force_use_t usage,
- audio_policy_forced_cfg_t config)
+ audio_policy_forced_cfg_t config)
{
ALOGV("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mEngine->getPhoneState());
if (config == mEngine->getForceUse(usage)) {
@@ -680,26 +783,27 @@
delayMs = TOUCH_SOUND_FIXED_DELAY_MS;
}
if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
- audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, true /*fromCache*/);
- waitMs = updateCallRouting(newDevice, delayMs);
+ DeviceVector newDevices = getNewOutputDevices(mPrimaryOutput, true /*fromCache*/);
+ waitMs = updateCallRouting(newDevices, delayMs);
}
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i);
- audio_devices_t newDevice = getNewOutputDevice(outputDesc, true /*fromCache*/);
+ DeviceVector newDevices = getNewOutputDevices(outputDesc, true /*fromCache*/);
if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (outputDesc != mPrimaryOutput)) {
- waitMs = setOutputDevice(outputDesc, newDevice, (newDevice != AUDIO_DEVICE_NONE),
- delayMs);
+ // As done in setDeviceConnectionState, we could also fix default device issue by
+ // preventing the force re-routing in case of default dev that distinguishes on address.
+ // Let's give back to engine full device choice decision however.
+ waitMs = setOutputDevices(outputDesc, newDevices, !newDevices.isEmpty(), delayMs);
}
- if (forceVolumeReeval && (newDevice != AUDIO_DEVICE_NONE)) {
- applyStreamVolumes(outputDesc, newDevice, waitMs, true);
+ if (forceVolumeReeval && !newDevices.isEmpty()) {
+ applyStreamVolumes(outputDesc, newDevices.types(), waitMs, true);
}
}
for (const auto& activeDesc : mInputs.getActiveInputs()) {
- audio_devices_t newDevice = getNewInputDevice(activeDesc);
+ auto newDevice = getNewInputDevice(activeDesc);
// Force new input selection if the new device can not be reached via current input
- if (activeDesc->mProfile->getSupportedDevices().types() &
- (newDevice & ~AUDIO_DEVICE_BIT_IN)) {
+ if (activeDesc->mProfile->getSupportedDevices().contains(newDevice)) {
setInputDevice(activeDesc->mIoHandle, newDevice);
} else {
closeInput(activeDesc->mIoHandle);
@@ -715,7 +819,7 @@
// Find an output profile compatible with the parameters passed. When "directOnly" is set, restrict
// search to profiles for direct outputs.
sp<IOProfile> AudioPolicyManager::getProfileForOutput(
- audio_devices_t device,
+ const DeviceVector& devices,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
@@ -736,7 +840,7 @@
for (const auto& hwModule : mHwModules) {
for (const auto& curProfile : hwModule->getOutputProfiles()) {
- if (!curProfile->isCompatibleProfile(device, String8(""),
+ if (!curProfile->isCompatibleProfile(devices,
samplingRate, NULL /*updatedSamplingRate*/,
format, NULL /*updatedFormat*/,
channelMask, NULL /*updatedChannelMask*/,
@@ -744,7 +848,11 @@
continue;
}
// reject profiles not corresponding to a device currently available
- if ((mAvailableOutputDevices.types() & curProfile->getSupportedDevicesType()) == 0) {
+ if (!mAvailableOutputDevices.containsAtLeastOne(curProfile->getSupportedDevices())) {
+ continue;
+ }
+ // reject profiles if connected device does not support codec
+ if (!curProfile->deviceSupportsEncodedFormats(devices.types())) {
continue;
}
if (!directOnly) return curProfile;
@@ -765,7 +873,7 @@
audio_io_handle_t AudioPolicyManager::getOutput(audio_stream_type_t stream)
{
routing_strategy strategy = getStrategy(stream);
- audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
+ DeviceVector devices = getDevicesForStrategy(strategy, false /*fromCache*/);
// Note that related method getOutputForAttr() uses getOutputForDevice() not selectOutput().
// We use selectOutput() here since we don't have the desired AudioTrack sample rate,
@@ -773,10 +881,11 @@
// getOutput() solely on audio_stream_type such as AudioSystem::getOutputFrameCount()
// and AudioSystem::getOutputSamplingRate().
- SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device, mOutputs);
- audio_io_handle_t output = selectOutput(outputs);
+ SortedVector<audio_io_handle_t> outputs = getOutputsForDevices(devices, mOutputs);
+ audio_io_handle_t output = selectOutput(outputs, AUDIO_OUTPUT_FLAG_NONE, AUDIO_FORMAT_INVALID);
- ALOGV("getOutput() stream %d selected device %08x, output %d", stream, device, output);
+ ALOGV("getOutput() stream %d selected devices %s, output %d", stream,
+ devices.toString().c_str(), output);
return output;
}
@@ -813,12 +922,11 @@
audio_output_flags_t *flags,
audio_port_handle_t *selectedDeviceId)
{
- DeviceVector outputDevices;
+ DeviceVector devices;
routing_strategy strategy;
- audio_devices_t device;
- const audio_port_handle_t requestedDeviceId = *selectedDeviceId;
- audio_devices_t msdDevice =
- getModuleDeviceTypes(mAvailableOutputDevices, AUDIO_HARDWARE_MODULE_ID_MSD);
+ audio_devices_t deviceType = AUDIO_DEVICE_NONE;
+ const audio_port_handle_t requestedPortId = *selectedDeviceId;
+ DeviceVector msdDevices = getMsdAudioOutDevices();
status_t status = getAudioAttributes(resultAttr, attr, *stream);
if (status != NO_ERROR) {
@@ -829,17 +937,16 @@
" session %d selectedDeviceId %d",
__func__,
resultAttr->usage, resultAttr->content_type, resultAttr->tags, resultAttr->flags,
- session, requestedDeviceId);
+ session, requestedPortId);
*stream = streamTypefromAttributesInt(resultAttr);
strategy = getStrategyForAttr(resultAttr);
// First check for explicit routing (eg. setPreferredDevice)
- if (requestedDeviceId != AUDIO_PORT_HANDLE_NONE) {
- sp<DeviceDescriptor> deviceDesc =
- mAvailableOutputDevices.getDeviceFromId(requestedDeviceId);
- device = deviceDesc->type();
+ sp<DeviceDescriptor> requestedDevice = mAvailableOutputDevices.getDeviceFromId(requestedPortId);
+ if (requestedDevice != nullptr) {
+ deviceType = requestedDevice->type();
} else {
// If no explict route, is there a matching dynamic policy that applies?
sp<SwAudioOutputDescriptor> desc;
@@ -852,7 +959,8 @@
*output = desc->mIoHandle;
AudioMix *mix = desc->mPolicyMix;
sp<DeviceDescriptor> deviceDesc =
- mAvailableOutputDevices.getDevice(mix->mDeviceType, mix->mDeviceAddress);
+ mAvailableOutputDevices.getDevice(
+ mix->mDeviceType, mix->mDeviceAddress, AUDIO_FORMAT_DEFAULT);
*selectedDeviceId = deviceDesc != 0 ? deviceDesc->getId() : AUDIO_PORT_HANDLE_NONE;
ALOGV("%s returns output %d", __func__, *output);
return NO_ERROR;
@@ -863,7 +971,7 @@
ALOGW("%s no policy mix found for usage AUDIO_USAGE_VIRTUAL_SOURCE", __func__);
return BAD_VALUE;
}
- device = getDeviceForStrategy(strategy, false /*fromCache*/);
+ deviceType = getDeviceForStrategy(strategy, false /*fromCache*/);
}
if ((resultAttr->flags & AUDIO_FLAG_HW_AV_SYNC) != 0) {
@@ -875,42 +983,45 @@
// FIXME: provide a more generic approach which is not device specific and move this back
// to getOutputForDevice.
// TODO: Remove check of AUDIO_STREAM_MUSIC once migration is completed on the app side.
- if (device == AUDIO_DEVICE_OUT_TELEPHONY_TX &&
+ if (deviceType == AUDIO_DEVICE_OUT_TELEPHONY_TX &&
(*stream == AUDIO_STREAM_MUSIC || resultAttr->usage == AUDIO_USAGE_VOICE_COMMUNICATION) &&
audio_is_linear_pcm(config->format) &&
isInCall()) {
- if (requestedDeviceId != AUDIO_PORT_HANDLE_NONE) {
+ if (requestedPortId != AUDIO_PORT_HANDLE_NONE) {
*flags = (audio_output_flags_t)AUDIO_OUTPUT_FLAG_INCALL_MUSIC;
} else {
// Get the devce type directly from the engine to bypass preferred route logic
- device = mEngine->getDeviceForStrategy(strategy);
+ deviceType = mEngine->getDeviceForStrategy(strategy);
}
}
ALOGV("%s device 0x%x, sampling rate %d, format %#x, channel mask %#x, "
"flags %#x",
- __func__, device, config->sample_rate, config->format, config->channel_mask, *flags);
+ __func__,
+ deviceType, config->sample_rate, config->format, config->channel_mask, *flags);
*output = AUDIO_IO_HANDLE_NONE;
- if (msdDevice != AUDIO_DEVICE_NONE) {
- *output = getOutputForDevice(msdDevice, session, *stream, config, flags);
- if (*output != AUDIO_IO_HANDLE_NONE && setMsdPatch(device) == NO_ERROR) {
- ALOGV("%s() Using MSD device 0x%x instead of device 0x%x",
- __func__, msdDevice, device);
- device = msdDevice;
+ if (!msdDevices.isEmpty()) {
+ *output = getOutputForDevices(msdDevices, session, *stream, config, flags);
+ sp<DeviceDescriptor> deviceDesc =
+ mAvailableOutputDevices.getDevice(deviceType, String8(), AUDIO_FORMAT_DEFAULT);
+ if (*output != AUDIO_IO_HANDLE_NONE && setMsdPatch(deviceDesc) == NO_ERROR) {
+ ALOGV("%s() Using MSD devices %s instead of device %s",
+ __func__, msdDevices.toString().c_str(), deviceDesc->toString().c_str());
+ deviceType = msdDevices.types();
} else {
*output = AUDIO_IO_HANDLE_NONE;
}
}
+ devices = mAvailableOutputDevices.getDevicesFromTypeMask(deviceType);
if (*output == AUDIO_IO_HANDLE_NONE) {
- *output = getOutputForDevice(device, session, *stream, config, flags);
+ *output = getOutputForDevices(devices, session, *stream, config, flags);
}
if (*output == AUDIO_IO_HANDLE_NONE) {
return INVALID_OPERATION;
}
- outputDevices = mAvailableOutputDevices.getDevicesFromTypeMask(device);
- *selectedDeviceId = getFirstDeviceId(outputDevices);
+ *selectedDeviceId = getFirstDeviceId(devices);
ALOGV("%s returns output %d selectedDeviceId %d", __func__, *output, *selectedDeviceId);
@@ -931,7 +1042,7 @@
if (*portId != AUDIO_PORT_HANDLE_NONE) {
return INVALID_OPERATION;
}
- const audio_port_handle_t requestedDeviceId = *selectedDeviceId;
+ const audio_port_handle_t requestedPortId = *selectedDeviceId;
audio_attributes_t resultAttr;
status_t status = getOutputForAttrInt(&resultAttr, output, session, attr, stream, uid,
config, flags, selectedDeviceId);
@@ -946,20 +1057,20 @@
sp<TrackClientDescriptor> clientDesc =
new TrackClientDescriptor(*portId, uid, session, resultAttr, clientConfig,
- requestedDeviceId, *stream,
+ requestedPortId, *stream,
getStrategyForAttr(&resultAttr),
*flags);
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueFor(*output);
outputDesc->addClient(clientDesc);
ALOGV("%s returns output %d selectedDeviceId %d for port ID %d",
- __func__, *output, requestedDeviceId, *portId);
+ __func__, *output, requestedPortId, *portId);
return NO_ERROR;
}
-audio_io_handle_t AudioPolicyManager::getOutputForDevice(
- audio_devices_t device,
+audio_io_handle_t AudioPolicyManager::getOutputForDevices(
+ const DeviceVector &devices,
audio_session_t session,
audio_stream_type_t stream,
const audio_config_t *config,
@@ -1017,7 +1128,7 @@
if (((*flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0) ||
!(mEffects.isNonOffloadableEffectEnabled() || mMasterMono)) {
- profile = getProfileForOutput(device,
+ profile = getProfileForOutput(devices,
config->sample_rate,
config->format,
config->channel_mask,
@@ -1037,7 +1148,7 @@
(config->channel_mask == desc->mChannelMask) &&
(session == desc->mDirectClientSession)) {
desc->mDirectOpenCount++;
- ALOGI("getOutputForDevice() reusing direct output %d for session %d",
+ ALOGI("%s reusing direct output %d for session %d", __func__,
mOutputs.keyAt(i), session);
return mOutputs.keyAt(i);
}
@@ -1051,8 +1162,7 @@
sp<SwAudioOutputDescriptor> outputDesc =
new SwAudioOutputDescriptor(profile, mpClientInterface);
- DeviceVector outputDevices = mAvailableOutputDevices.getDevicesFromTypeMask(device);
- String8 address = getFirstDeviceAddress(outputDevices);
+ String8 address = getFirstDeviceAddress(devices);
// MSD patch may be using the only output stream that can service this request. Release
// MSD patch to prioritize this request over any active output on MSD.
@@ -1062,7 +1172,7 @@
for (size_t j = 0; j < patch->mPatch.num_sinks; ++j) {
const struct audio_port_config *sink = &patch->mPatch.sinks[j];
if (sink->type == AUDIO_PORT_TYPE_DEVICE &&
- (sink->ext.device.type & device) != AUDIO_DEVICE_NONE &&
+ (sink->ext.device.type & devices.types()) != AUDIO_DEVICE_NONE &&
(address.isEmpty() || strncmp(sink->ext.device.address, address.string(),
AUDIO_DEVICE_MAX_ADDRESS_LEN) == 0)) {
releaseAudioPatch(patch->mHandle, mUidCached);
@@ -1071,15 +1181,15 @@
}
}
- status = outputDesc->open(config, device, address, stream, *flags, &output);
+ status = outputDesc->open(config, devices, stream, *flags, &output);
// only accept an output with the requested parameters
if (status != NO_ERROR ||
(config->sample_rate != 0 && config->sample_rate != outputDesc->mSamplingRate) ||
(config->format != AUDIO_FORMAT_DEFAULT && config->format != outputDesc->mFormat) ||
(config->channel_mask != 0 && config->channel_mask != outputDesc->mChannelMask)) {
- ALOGV("getOutputForDevice() failed opening direct output: output %d sample rate %d %d,"
- "format %d %d, channel mask %04x %04x", output, config->sample_rate,
+ ALOGV("%s failed opening direct output: output %d sample rate %d %d,"
+ "format %d %d, channel mask %04x %04x", __func__, output, config->sample_rate,
outputDesc->mSamplingRate, config->format, outputDesc->mFormat,
config->channel_mask, outputDesc->mChannelMask);
if (output != AUDIO_IO_HANDLE_NONE) {
@@ -1097,7 +1207,7 @@
addOutput(output, outputDesc);
mPreviousOutputs = mOutputs;
- ALOGV("getOutputForDevice() returns new direct output %d", output);
+ ALOGV("%s returns new direct output %d", __func__, output);
mpClientInterface->onAudioPortListUpdate();
return output;
}
@@ -1118,14 +1228,14 @@
if (audio_is_linear_pcm(config->format)) {
// get which output is suitable for the specified stream. The actual
// routing change will happen when startOutput() will be called
- SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device, mOutputs);
+ SortedVector<audio_io_handle_t> outputs = getOutputsForDevices(devices, mOutputs);
// at this stage we should ignore the DIRECT flag as no direct output could be found earlier
*flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_DIRECT);
output = selectOutput(outputs, *flags, config->format,
config->channel_mask, config->sample_rate);
}
- ALOGW_IF((output == 0), "getOutputForDevice() could not find output for stream %d, "
+ ALOGW_IF((output == 0), "getOutputForDevices() could not find output for stream %d, "
"sampling rate %d, format %#x, channels %#x, flags %#x",
stream, config->sample_rate, config->format, config->channel_mask, *flags);
@@ -1133,13 +1243,14 @@
}
sp<DeviceDescriptor> AudioPolicyManager::getMsdAudioInDevice() const {
- sp<HwModule> msdModule = mHwModules.getModuleFromName(AUDIO_HARDWARE_MODULE_ID_MSD);
- if (msdModule != 0) {
- DeviceVector msdInputDevices = mAvailableInputDevices.getDevicesFromHwModule(
- msdModule->getHandle());
- if (!msdInputDevices.isEmpty()) return msdInputDevices.itemAt(0);
- }
- return 0;
+ auto msdInDevices = mHwModules.getAvailableDevicesFromModuleName(AUDIO_HARDWARE_MODULE_ID_MSD,
+ mAvailableInputDevices);
+ return msdInDevices.isEmpty()? nullptr : msdInDevices.itemAt(0);
+}
+
+DeviceVector AudioPolicyManager::getMsdAudioOutDevices() const {
+ return mHwModules.getAvailableDevicesFromModuleName(AUDIO_HARDWARE_MODULE_ID_MSD,
+ mAvailableOutputDevices);
}
const AudioPatchCollection AudioPolicyManager::getMsdPatches() const {
@@ -1160,7 +1271,7 @@
return msdPatches;
}
-status_t AudioPolicyManager::getBestMsdAudioProfileFor(audio_devices_t outputDevice,
+status_t AudioPolicyManager::getBestMsdAudioProfileFor(const sp<DeviceDescriptor> &outputDevice,
bool hwAvSync, audio_port_config *sourceConfig, audio_port_config *sinkConfig) const
{
sp<HwModule> msdModule = mHwModules.getModuleFromName(AUDIO_HARDWARE_MODULE_ID_MSD);
@@ -1168,9 +1279,9 @@
ALOGE("%s() unable to get MSD module", __func__);
return NO_INIT;
}
- sp<HwModule> deviceModule = mHwModules.getModuleForDevice(outputDevice);
+ sp<HwModule> deviceModule = mHwModules.getModuleForDevice(outputDevice, AUDIO_FORMAT_DEFAULT);
if (deviceModule == nullptr) {
- ALOGE("%s() unable to get module for %#x", __func__, outputDevice);
+ ALOGE("%s() unable to get module for %s", __func__, outputDevice->toString().c_str());
return NO_INIT;
}
const InputProfileCollection &inputProfiles = msdModule->getInputProfiles();
@@ -1180,7 +1291,7 @@
}
const OutputProfileCollection &outputProfiles = deviceModule->getOutputProfiles();
if (outputProfiles.isEmpty()) {
- ALOGE("%s() no output profiles for device %#x", __func__, outputDevice);
+ ALOGE("%s() no output profiles for device %s", __func__, outputDevice->toString().c_str());
return NO_INIT;
}
AudioProfileVector msdProfiles;
@@ -1201,8 +1312,8 @@
compressedFormatsOrder, surroundChannelMasksOrder, true /*preferHigherSamplingRates*/,
&bestSinkConfig);
if (result != NO_ERROR) {
- ALOGD("%s() no matching profiles found for device: %#x, hwAvSync: %d",
- __func__, outputDevice, hwAvSync);
+ ALOGD("%s() no matching profiles found for device: %s, hwAvSync: %d",
+ __func__, outputDevice->toString().c_str(), hwAvSync);
return result;
}
sinkConfig->sample_rate = bestSinkConfig.sample_rate;
@@ -1231,11 +1342,10 @@
return NO_ERROR;
}
-PatchBuilder AudioPolicyManager::buildMsdPatch(audio_devices_t outputDevice) const
+PatchBuilder AudioPolicyManager::buildMsdPatch(const sp<DeviceDescriptor> &outputDevice) const
{
PatchBuilder patchBuilder;
- patchBuilder.addSource(getMsdAudioInDevice()).
- addSink(findDevice(mAvailableOutputDevices, outputDevice));
+ patchBuilder.addSource(getMsdAudioInDevice()).addSink(outputDevice);
audio_port_config sourceConfig = patchBuilder.patch()->sources[0];
audio_port_config sinkConfig = patchBuilder.patch()->sinks[0];
// TODO: Figure out whether MSD module has HW_AV_SYNC flag set in the AP config file.
@@ -1253,15 +1363,18 @@
return patchBuilder;
}
-status_t AudioPolicyManager::setMsdPatch(audio_devices_t outputDevice) {
- ALOGV("%s() for outputDevice %#x", __func__, outputDevice);
- if (outputDevice == AUDIO_DEVICE_NONE) {
+status_t AudioPolicyManager::setMsdPatch(const sp<DeviceDescriptor> &outputDevice) {
+ sp<DeviceDescriptor> device = outputDevice;
+ if (device == nullptr) {
// Use media strategy for unspecified output device. This should only
// occur on checkForDeviceAndOutputChanges(). Device connection events may
// therefore invalidate explicit routing requests.
- outputDevice = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/);
+ DeviceVector devices = getDevicesForStrategy(STRATEGY_MEDIA, false /*fromCache*/);
+ LOG_ALWAYS_FATAL_IF(devices.isEmpty(), "no outpudevice to set Msd Patch");
+ device = devices.itemAt(0);
}
- PatchBuilder patchBuilder = buildMsdPatch(outputDevice);
+ ALOGV("%s() for device %s", __func__, device->toString().c_str());
+ PatchBuilder patchBuilder = buildMsdPatch(device);
const struct audio_patch* patch = patchBuilder.patch();
const AudioPatchCollection msdPatches = getMsdPatches();
if (!msdPatches.isEmpty()) {
@@ -1277,8 +1390,9 @@
patch, 0 /*delayMs*/, mUidCached, nullptr /*patchDescPtr*/);
ALOGE_IF(status != NO_ERROR, "%s() error %d creating MSD audio patch", __func__, status);
ALOGI_IF(status == NO_ERROR, "%s() Patch created from MSD_IN to "
- "device:%#x (format:%#x channels:%#x samplerate:%d)", __func__, outputDevice,
- patch->sources[0].format, patch->sources[0].channel_mask, patch->sources[0].sample_rate);
+ "device:%s (format:%#x channels:%#x samplerate:%d)", __func__,
+ device->toString().c_str(), patch->sources[0].format,
+ patch->sources[0].channel_mask, patch->sources[0].sample_rate);
return status;
}
@@ -1289,7 +1403,7 @@
uint32_t samplingRate)
{
// select one output among several that provide a path to a particular device or set of
- // devices (the list was previously build by getOutputsForDevice()).
+ // devices (the list was previously build by getOutputsForDevices()).
// The priority is as follows:
// 1: the output supporting haptic playback when requesting haptic playback
// 2: the output with the highest number of requested policy flags
@@ -1451,17 +1565,20 @@
bool force = !outputDesc->isActive() &&
(outputDesc->getPatchHandle() == AUDIO_PATCH_HANDLE_NONE);
- audio_devices_t device = AUDIO_DEVICE_NONE;
+ DeviceVector devices;
AudioMix *policyMix = NULL;
const char *address = NULL;
if (outputDesc->mPolicyMix != NULL) {
policyMix = outputDesc->mPolicyMix;
+ audio_devices_t newDeviceType;
address = policyMix->mDeviceAddress.string();
if ((policyMix->mRouteFlags & MIX_ROUTE_FLAG_RENDER) == MIX_ROUTE_FLAG_RENDER) {
- device = policyMix->mDeviceType;
+ newDeviceType = policyMix->mDeviceType;
} else {
- device = AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
+ newDeviceType = AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
}
+ devices.add(mAvailableOutputDevices.getDevice(newDeviceType,
+ String8(address), AUDIO_FORMAT_DEFAULT));
}
// requiresMuteCheck is false when we can bypass mute strategy.
@@ -1476,8 +1593,8 @@
outputDesc->setClientActive(client, true);
if (client->hasPreferredDevice(true)) {
- device = getNewOutputDevice(outputDesc, false /*fromCache*/);
- if (device != outputDesc->device()) {
+ devices = getNewOutputDevices(outputDesc, false /*fromCache*/);
+ if (devices != outputDesc->devices()) {
checkStrategyRoute(getStrategy(stream), outputDesc->mIoHandle);
}
}
@@ -1486,10 +1603,10 @@
selectOutputForMusicEffects();
}
- if (outputDesc->streamActiveCount(stream) == 1 || device != AUDIO_DEVICE_NONE) {
+ if (outputDesc->streamActiveCount(stream) == 1 || !devices.isEmpty()) {
// starting an output being rerouted?
- if (device == AUDIO_DEVICE_NONE) {
- device = getNewOutputDevice(outputDesc, false /*fromCache*/);
+ if (devices.isEmpty()) {
+ devices = getNewOutputDevices(outputDesc, false /*fromCache*/);
}
routing_strategy strategy = getStrategy(stream);
@@ -1498,13 +1615,13 @@
(beaconMuteLatency > 0);
uint32_t waitMs = beaconMuteLatency;
for (size_t i = 0; i < mOutputs.size(); i++) {
- sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
if (desc != outputDesc) {
// An output has a shared device if
// - managed by the same hw module
// - supports the currently selected device
const bool sharedDevice = outputDesc->sharesHwModuleWith(desc)
- && (desc->supportedDevices() & device) != AUDIO_DEVICE_NONE;
+ && (!desc->filterSupportedDevices(devices).isEmpty());
// force a device change if any other output is:
// - managed by the same hw module
@@ -1514,7 +1631,7 @@
// In this case, the audio HAL must receive the new device selection so that it can
// change the device currently selected by the other output.
if (sharedDevice &&
- desc->device() != device &&
+ desc->devices() != devices &&
desc->getPatchHandle() != AUDIO_PATCH_HANDLE_NONE) {
force = true;
}
@@ -1537,13 +1654,13 @@
}
const uint32_t muteWaitMs =
- setOutputDevice(outputDesc, device, force, 0, NULL, address, requiresMuteCheck);
+ setOutputDevices(outputDesc, devices, force, 0, NULL, requiresMuteCheck);
// apply volume rules for current stream and device if necessary
checkAndSetVolume(stream,
- mVolumeCurves->getVolumeIndex(stream, outputDesc->device()),
+ mVolumeCurves->getVolumeIndex(stream, outputDesc->devices().types()),
outputDesc,
- outputDesc->device());
+ outputDesc->devices().types());
// update the outputs if starting an output with a stream that can affect notification
// routing
@@ -1574,12 +1691,13 @@
// Automatically enable the remote submix input when output is started on a re routing mix
// of type MIX_TYPE_RECORDERS
- if (audio_is_remote_submix_device(device) && policyMix != NULL &&
+ if (audio_is_remote_submix_device(devices.types()) && policyMix != NULL &&
policyMix->mMixType == MIX_TYPE_RECORDERS) {
setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
address,
- "remote-submix");
+ "remote-submix",
+ AUDIO_FORMAT_DEFAULT);
}
return NO_ERROR;
@@ -1619,13 +1737,13 @@
if (outputDesc->streamActiveCount(stream) == 1) {
// Automatically disable the remote submix input when output is stopped on a
// re routing mix of type MIX_TYPE_RECORDERS
- if (audio_is_remote_submix_device(outputDesc->mDevice) &&
+ if (audio_is_remote_submix_device(outputDesc->devices().types()) &&
outputDesc->mPolicyMix != NULL &&
outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) {
setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
outputDesc->mPolicyMix->mDeviceAddress,
- "remote-submix");
+ "remote-submix", AUDIO_FORMAT_DEFAULT);
}
}
bool forceDeviceUpdate = false;
@@ -1640,33 +1758,31 @@
// store time at which the stream was stopped - see isStreamActive()
if (outputDesc->streamActiveCount(stream) == 0 || forceDeviceUpdate) {
outputDesc->mStopTime[stream] = systemTime();
- audio_devices_t newDevice = getNewOutputDevice(outputDesc, false /*fromCache*/);
+ DeviceVector newDevices = getNewOutputDevices(outputDesc, false /*fromCache*/);
// delay the device switch by twice the latency because stopOutput() is executed when
// the track stop() command is received and at that time the audio track buffer can
// still contain data that needs to be drained. The latency only covers the audio HAL
// and kernel buffers. Also the latency does not always include additional delay in the
// audio path (audio DSP, CODEC ...)
- setOutputDevice(outputDesc, newDevice, false, outputDesc->latency()*2);
+ setOutputDevices(outputDesc, newDevices, false, outputDesc->latency()*2);
// force restoring the device selection on other active outputs if it differs from the
// one being selected for this output
uint32_t delayMs = outputDesc->latency()*2;
for (size_t i = 0; i < mOutputs.size(); i++) {
- sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i);
+ sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
if (desc != outputDesc &&
desc->isActive() &&
outputDesc->sharesHwModuleWith(desc) &&
- (newDevice != desc->device())) {
- audio_devices_t newDevice2 = getNewOutputDevice(desc, false /*fromCache*/);
- bool force = desc->device() != newDevice2;
+ (newDevices != desc->devices())) {
+ DeviceVector newDevices2 = getNewOutputDevices(desc, false /*fromCache*/);
+ bool force = desc->devices() != newDevices2;
- setOutputDevice(desc,
- newDevice2,
- force,
- delayMs);
+ setOutputDevices(desc, newDevices2, force, delayMs);
+
// re-apply device specific volume if not done by setOutputDevice()
if (!force) {
- applyStreamVolumes(desc, newDevice2, delayMs);
+ applyStreamVolumes(desc, newDevices2.types(), delayMs);
}
}
}
@@ -1739,29 +1855,27 @@
attr->source, config->sample_rate, config->format, config->channel_mask, session, flags);
status_t status = NO_ERROR;
- // handle legacy remote submix case where the address was not always specified
- String8 address = String8("");
audio_source_t halInputSource;
- audio_source_t inputSource = attr->source;
+ audio_attributes_t attributes = *attr;
AudioMix *policyMix = NULL;
- DeviceVector inputDevices;
+ sp<DeviceDescriptor> device;
sp<AudioInputDescriptor> inputDesc;
sp<RecordClientDescriptor> clientDesc;
audio_port_handle_t requestedDeviceId = *selectedDeviceId;
bool isSoundTrigger;
- audio_devices_t device;
// The supplied portId must be AUDIO_PORT_HANDLE_NONE
if (*portId != AUDIO_PORT_HANDLE_NONE) {
return INVALID_OPERATION;
}
- if (inputSource == AUDIO_SOURCE_DEFAULT) {
- inputSource = AUDIO_SOURCE_MIC;
+ if (attr->source == AUDIO_SOURCE_DEFAULT) {
+ attributes.source = AUDIO_SOURCE_MIC;
}
// Explicit routing?
- sp<DeviceDescriptor> deviceDesc = mAvailableInputDevices.getDeviceFromId(*selectedDeviceId);
+ sp<DeviceDescriptor> explicitRoutingDevice =
+ mAvailableInputDevices.getDeviceFromId(*selectedDeviceId);
// special case for mmap capture: if an input IO handle is specified, we reuse this input if
// possible
@@ -1802,7 +1916,7 @@
}
}
*inputType = API_INPUT_LEGACY;
- device = inputDesc->mDevice;
+ device = inputDesc->getDevice();
ALOGI("%s reusing MMAP input %d for session %d", __FUNCTION__, *input, session);
goto exit;
@@ -1811,44 +1925,40 @@
*input = AUDIO_IO_HANDLE_NONE;
*inputType = API_INPUT_INVALID;
- halInputSource = inputSource;
+ halInputSource = attributes.source;
- if (inputSource == AUDIO_SOURCE_REMOTE_SUBMIX &&
- strncmp(attr->tags, "addr=", strlen("addr=")) == 0) {
- status = mPolicyMixes.getInputMixForAttr(*attr, &policyMix);
+ if (attributes.source == AUDIO_SOURCE_REMOTE_SUBMIX &&
+ strncmp(attributes.tags, "addr=", strlen("addr=")) == 0) {
+ status = mPolicyMixes.getInputMixForAttr(attributes, &policyMix);
if (status != NO_ERROR) {
goto error;
}
*inputType = API_INPUT_MIX_EXT_POLICY_REROUTE;
- device = AUDIO_DEVICE_IN_REMOTE_SUBMIX;
- address = String8(attr->tags + strlen("addr="));
+ device = mAvailableInputDevices.getDevice(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
+ String8(attr->tags + strlen("addr=")),
+ AUDIO_FORMAT_DEFAULT);
} else {
- if (deviceDesc != 0) {
- device = deviceDesc->type();
+ if (explicitRoutingDevice != nullptr) {
+ device = explicitRoutingDevice;
} else {
- device = getDeviceAndMixForInputSource(inputSource, &policyMix);
+ device = getDeviceAndMixForAttributes(attributes, &policyMix);
}
- if (device == AUDIO_DEVICE_NONE) {
- ALOGW("getInputForAttr() could not find device for source %d", inputSource);
+ if (device == nullptr) {
+ ALOGW("getInputForAttr() could not find device for source %d", attributes.source);
status = BAD_VALUE;
goto error;
}
- if (policyMix != NULL) {
- address = policyMix->mDeviceAddress;
- if (policyMix->mMixType == MIX_TYPE_RECORDERS) {
- // there is an external policy, but this input is attached to a mix of recorders,
- // meaning it receives audio injected into the framework, so the recorder doesn't
- // know about it and is therefore considered "legacy"
- *inputType = API_INPUT_LEGACY;
- } else {
- // recording a mix of players defined by an external policy, we're rerouting for
- // an external policy
- *inputType = API_INPUT_MIX_EXT_POLICY_REROUTE;
- }
- } else if (audio_is_remote_submix_device(device)) {
- address = String8("0");
+ if (policyMix != nullptr) {
+ ALOG_ASSERT(policyMix->mMixType == MIX_TYPE_RECORDERS, "Invalid Mix Type");
+ // there is an external policy, but this input is attached to a mix of recorders,
+ // meaning it receives audio injected into the framework, so the recorder doesn't
+ // know about it and is therefore considered "legacy"
+ *inputType = API_INPUT_LEGACY;
+ } else if (audio_is_remote_submix_device(device->type())) {
+ device = mAvailableInputDevices.getDevice(AUDIO_DEVICE_IN_REMOTE_SUBMIX, String8("0"),
+ AUDIO_FORMAT_DEFAULT);
*inputType = API_INPUT_MIX_CAPTURE;
- } else if (device == AUDIO_DEVICE_IN_TELEPHONY_RX) {
+ } else if (device->type() == AUDIO_DEVICE_IN_TELEPHONY_RX) {
*inputType = API_INPUT_TELEPHONY_RX;
} else {
*inputType = API_INPUT_LEGACY;
@@ -1856,7 +1966,7 @@
}
- *input = getInputForDevice(device, address, session, inputSource,
+ *input = getInputForDevice(device, session, attributes.source,
config, flags,
policyMix);
if (*input == AUDIO_IO_HANDLE_NONE) {
@@ -1866,16 +1976,16 @@
exit:
- inputDevices = mAvailableInputDevices.getDevicesFromTypeMask(device);
- *selectedDeviceId = getFirstDeviceId(inputDevices);
+ *selectedDeviceId = mAvailableInputDevices.contains(device) ?
+ device->getId() : AUDIO_PORT_HANDLE_NONE;
- isSoundTrigger = inputSource == AUDIO_SOURCE_HOTWORD &&
+ isSoundTrigger = attributes.source == AUDIO_SOURCE_HOTWORD &&
mSoundTriggerSessions.indexOfKey(session) > 0;
*portId = AudioPort::getNextUniqueId();
- clientDesc = new RecordClientDescriptor(*portId, uid, session,
- *attr, *config, requestedDeviceId,
- inputSource,flags, isSoundTrigger);
+ clientDesc = new RecordClientDescriptor(*portId, uid, session, *attr, *config,
+ requestedDeviceId, attributes.source, flags,
+ isSoundTrigger);
inputDesc = mInputs.valueFor(*input);
inputDesc->addClient(clientDesc);
@@ -1889,8 +1999,7 @@
}
-audio_io_handle_t AudioPolicyManager::getInputForDevice(audio_devices_t device,
- String8 address,
+audio_io_handle_t AudioPolicyManager::getInputForDevice(const sp<DeviceDescriptor> &device,
audio_session_t session,
audio_source_t inputSource,
const audio_config_base_t *config,
@@ -1926,8 +2035,7 @@
audio_input_flags_t profileFlags = flags;
for (;;) {
profileFormat = config->format; // reset each time through loop, in case it is updated
- profile = getInputProfile(device, address,
- profileSamplingRate, profileFormat, profileChannelMask,
+ profile = getInputProfile(device, profileSamplingRate, profileFormat, profileChannelMask,
profileFlags);
if (profile != 0) {
break; // success
@@ -1936,9 +2044,9 @@
} else if (profileFlags != AUDIO_INPUT_FLAG_NONE) {
profileFlags = AUDIO_INPUT_FLAG_NONE; // retry
} else { // fail
- ALOGW("getInputForDevice() could not find profile for device 0x%X, "
- "sampling rate %u, format %#x, channel mask 0x%X, flags %#x",
- device, config->sample_rate, config->format, config->channel_mask, flags);
+ ALOGW("%s could not find profile for device %s, sampling rate %u, format %#x, "
+ "channel mask 0x%X, flags %#x", __func__, device->toString().c_str(),
+ config->sample_rate, config->format, config->channel_mask, flags);
return input;
}
}
@@ -1995,14 +2103,7 @@
lConfig.channel_mask = profileChannelMask;
lConfig.format = profileFormat;
- if (address == "") {
- DeviceVector inputDevices = mAvailableInputDevices.getDevicesFromTypeMask(device);
- // the inputs vector must be of size >= 1, but we don't want to crash here
- address = getFirstDeviceAddress(inputDevices);
- }
-
- status_t status = inputDesc->open(&lConfig, device, address,
- halInputSource, profileFlags, &input);
+ status_t status = inputDesc->open(&lConfig, device, halInputSource, profileFlags, &input);
// only accept input with the exact requested set of parameters
if (status != NO_ERROR || input == AUDIO_IO_HANDLE_NONE ||
@@ -2059,7 +2160,7 @@
// indicate active capture to sound trigger service if starting capture from a mic on
// primary HW module
- audio_devices_t device = getNewInputDevice(inputDesc);
+ sp<DeviceDescriptor> device = getNewInputDevice(inputDesc);
setInputDevice(input, device, true /* force */);
if (inputDesc->activeCount() == 1) {
@@ -2070,8 +2171,8 @@
MIX_STATE_MIXING);
}
- audio_devices_t primaryInputDevices = availablePrimaryInputDevices();
- if (((device & primaryInputDevices & ~AUDIO_DEVICE_BIT_IN) != 0) &&
+ DeviceVector primaryInputDevices = availablePrimaryModuleInputDevices();
+ if (primaryInputDevices.contains(device) &&
mInputs.activeInputsCountOnDevices(primaryInputDevices) == 1) {
SoundTrigger::setCaptureState(true);
}
@@ -2079,7 +2180,7 @@
// automatically enable the remote submix output when input is started if not
// used by a policy mix of type MIX_TYPE_RECORDERS
// For remote submix (a virtual device), we open only one input per capture request.
- if (audio_is_remote_submix_device(inputDesc->mDevice)) {
+ if (audio_is_remote_submix_device(inputDesc->getDeviceType())) {
String8 address = String8("");
if (inputDesc->mPolicyMix == NULL) {
address = String8("0");
@@ -2089,7 +2190,7 @@
if (address != "") {
setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
- address, "remote-submix");
+ address, "remote-submix", AUDIO_FORMAT_DEFAULT);
}
}
}
@@ -2130,7 +2231,7 @@
// automatically disable the remote submix output when input is stopped if not
// used by a policy mix of type MIX_TYPE_RECORDERS
- if (audio_is_remote_submix_device(inputDesc->mDevice)) {
+ if (audio_is_remote_submix_device(inputDesc->getDeviceType())) {
String8 address = String8("");
if (inputDesc->mPolicyMix == NULL) {
address = String8("0");
@@ -2140,17 +2241,15 @@
if (address != "") {
setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
- address, "remote-submix");
+ address, "remote-submix", AUDIO_FORMAT_DEFAULT);
}
}
-
- audio_devices_t device = inputDesc->mDevice;
resetInputDevice(input);
// indicate inactive capture to sound trigger service if stopping capture from a mic on
// primary HW module
- audio_devices_t primaryInputDevices = availablePrimaryInputDevices();
- if (((device & primaryInputDevices & ~AUDIO_DEVICE_BIT_IN) != 0) &&
+ DeviceVector primaryInputDevices = availablePrimaryModuleInputDevices();
+ if (primaryInputDevices.contains(inputDesc->getDevice()) &&
mInputs.activeInputsCountOnDevices(primaryInputDevices) == 0) {
SoundTrigger::setCaptureState(false);
}
@@ -2228,6 +2327,10 @@
int indexMax)
{
ALOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax);
+ if (indexMin < 0 || indexMax < 0) {
+ ALOGE("%s for stream %d: invalid min %d or max %d", __func__, stream , indexMin, indexMax);
+ return;
+ }
mVolumeCurves->initStreamVolume(stream, indexMin, indexMax);
// initialize other private stream volumes which follow this one
@@ -2244,10 +2347,11 @@
audio_devices_t device)
{
- // VOICE_CALL stream has minVolumeIndex > 0 but can be muted directly by an
- // app that has MODIFY_PHONE_STATE permission.
+ // VOICE_CALL and BLUETOOTH_SCO stream have minVolumeIndex > 0 but
+ // can be muted directly by an app that has MODIFY_PHONE_STATE permission.
if (((index < mVolumeCurves->getVolumeIndexMin(stream)) &&
- !(stream == AUDIO_STREAM_VOICE_CALL && index == 0)) ||
+ !((stream == AUDIO_STREAM_VOICE_CALL || stream == AUDIO_STREAM_BLUETOOTH_SCO) &&
+ index == 0)) ||
(index > mVolumeCurves->getVolumeIndexMax(stream))) {
return BAD_VALUE;
}
@@ -2280,7 +2384,7 @@
status_t status = NO_ERROR;
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
- audio_devices_t curDevice = desc->device();
+ audio_devices_t curDevice = desc->devices().types();
for (int curStream = 0; curStream < AUDIO_STREAM_FOR_POLICY_CNT; curStream++) {
if (!(streamsMatchForvolume(stream, (audio_stream_type_t)curStream))) {
continue;
@@ -2356,8 +2460,8 @@
// 4: the first output in the list
routing_strategy strategy = getStrategy(AUDIO_STREAM_MUSIC);
- audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
- SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device, mOutputs);
+ DeviceVector devices = getDevicesForStrategy(strategy, false /*fromCache*/);
+ SortedVector<audio_io_handle_t> outputs = getOutputsForDevices(devices, mOutputs);
if (outputs.size() == 0) {
return AUDIO_IO_HANDLE_NONE;
@@ -2562,27 +2666,31 @@
if (mix.mMixType == MIX_TYPE_PLAYERS) {
setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
- address.string(), "remote-submix");
+ address.string(), "remote-submix", AUDIO_FORMAT_DEFAULT);
} else {
setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
- address.string(), "remote-submix");
+ address.string(), "remote-submix", AUDIO_FORMAT_DEFAULT);
}
} else if ((mix.mRouteFlags & MIX_ROUTE_FLAG_RENDER) == MIX_ROUTE_FLAG_RENDER) {
String8 address = mix.mDeviceAddress;
- audio_devices_t device = mix.mDeviceType;
+ audio_devices_t type = mix.mDeviceType;
ALOGV(" registerPolicyMixes() mix %zu of %zu is RENDER, dev=0x%X addr=%s",
- i, mixes.size(), device, address.string());
+ i, mixes.size(), type, address.string());
+
+ sp<DeviceDescriptor> device = mHwModules.getDeviceDescriptor(
+ mix.mDeviceType, mix.mDeviceAddress,
+ String8(), AUDIO_FORMAT_DEFAULT);
+ if (device == nullptr) {
+ res = INVALID_OPERATION;
+ break;
+ }
bool foundOutput = false;
for (size_t j = 0 ; j < mOutputs.size() ; j++) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(j);
- sp<AudioPatch> patch = mAudioPatches.valueFor(desc->getPatchHandle());
- if ((patch != 0) && (patch->mPatch.num_sinks != 0)
- && (patch->mPatch.sinks[0].type == AUDIO_PORT_TYPE_DEVICE)
- && (patch->mPatch.sinks[0].ext.device.type == device)
- && (strncmp(patch->mPatch.sinks[0].ext.device.address, address.string(),
- AUDIO_DEVICE_MAX_ADDRESS_LEN) == 0)) {
+
+ if (desc->supportedDevices().contains(device)) {
if (mPolicyMixes.registerMix(address, mix, desc) != NO_ERROR) {
res = INVALID_OPERATION;
} else {
@@ -2594,12 +2702,12 @@
if (res != NO_ERROR) {
ALOGE(" Error registering mix %zu for device 0x%X addr %s",
- i, device, address.string());
+ i, type, address.string());
res = INVALID_OPERATION;
break;
} else if (!foundOutput) {
ALOGE(" Output not found for mix %zu for device 0x%X addr %s",
- i, device, address.string());
+ i, type, address.string());
res = INVALID_OPERATION;
break;
}
@@ -2640,13 +2748,13 @@
AUDIO_POLICY_DEVICE_STATE_AVAILABLE) {
setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
- address.string(), "remote-submix");
+ address.string(), "remote-submix", AUDIO_FORMAT_DEFAULT);
}
if (getDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, address.string()) ==
AUDIO_POLICY_DEVICE_STATE_AVAILABLE) {
setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
- address.string(), "remote-submix");
+ address.string(), "remote-submix", AUDIO_FORMAT_DEFAULT);
}
rSubmixModule->removeOutputProfile(address);
rSubmixModule->removeInputProfile(address);
@@ -2690,11 +2798,11 @@
// reevaluate outputs for all given devices
for (size_t i = 0; i < devices.size(); i++) {
sp<DeviceDescriptor> devDesc = mHwModules.getDeviceDescriptor(
- devices[i].mType, devices[i].mAddress, String8());
+ devices[i].mType, devices[i].mAddress, String8(),
+ AUDIO_FORMAT_DEFAULT);
SortedVector<audio_io_handle_t> outputs;
if (checkOutputsForDevice(devDesc, AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
- outputs,
- devDesc->address()) != NO_ERROR) {
+ outputs) != NO_ERROR) {
ALOGE("setUidDeviceAffinities() error in checkOutputsForDevice for device=%08x"
" addr=%s", devices[i].mType, devices[i].mAddress.string());
return INVALID_OPERATION;
@@ -2712,11 +2820,11 @@
// reevaluate outputs for all found devices
for (size_t i = 0; i < devices.size(); i++) {
sp<DeviceDescriptor> devDesc = mHwModules.getDeviceDescriptor(
- devices[i].mType, devices[i].mAddress, String8());
+ devices[i].mType, devices[i].mAddress, String8(),
+ AUDIO_FORMAT_DEFAULT);
SortedVector<audio_io_handle_t> outputs;
if (checkOutputsForDevice(devDesc, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
- outputs,
- devDesc->address()) != NO_ERROR) {
+ outputs) != NO_ERROR) {
ALOGE("%s() error in checkOutputsForDevice for device=%08x addr=%s",
__FUNCTION__, devices[i].mType, devices[i].mAddress.string());
return INVALID_OPERATION;
@@ -2836,7 +2944,7 @@
// See if there is a profile to support this.
// AUDIO_DEVICE_NONE
- sp<IOProfile> profile = getProfileForOutput(AUDIO_DEVICE_NONE /*ignore device */,
+ sp<IOProfile> profile = getProfileForOutput(DeviceVector() /*ignore device */,
offloadInfo.sample_rate,
offloadInfo.format,
offloadInfo.channel_mask,
@@ -2850,7 +2958,7 @@
const audio_attributes_t& attributes) {
audio_output_flags_t output_flags = AUDIO_OUTPUT_FLAG_NONE;
audio_attributes_flags_to_audio_output_flags(attributes.flags, output_flags);
- sp<IOProfile> profile = getProfileForOutput(AUDIO_DEVICE_NONE /*ignore device */,
+ sp<IOProfile> profile = getProfileForOutput(DeviceVector() /*ignore device */,
config.sample_rate,
config.format,
config.channel_mask,
@@ -3044,8 +3152,7 @@
return BAD_VALUE;
}
- if (!outputDesc->mProfile->isCompatibleProfile(devDesc->type(),
- devDesc->address(),
+ if (!outputDesc->mProfile->isCompatibleProfile(DeviceVector(devDesc),
patch->sources[0].sample_rate,
NULL, // updatedSamplingRate
patch->sources[0].format,
@@ -3066,7 +3173,7 @@
// TODO: reconfigure output format and channels here
ALOGV("createAudioPatch() setting device %08x on output %d",
devices.types(), outputDesc->mIoHandle);
- setOutputDevice(outputDesc, devices.types(), true, 0, handle);
+ setOutputDevices(outputDesc, devices, true, 0, handle);
index = mAudioPatches.indexOfKey(*handle);
if (index >= 0) {
if (patchDesc != 0 && patchDesc != mAudioPatches.valueAt(index)) {
@@ -3095,14 +3202,13 @@
return BAD_VALUE;
}
}
- sp<DeviceDescriptor> devDesc =
+ sp<DeviceDescriptor> device =
mAvailableInputDevices.getDeviceFromId(patch->sources[0].id);
- if (devDesc == 0) {
+ if (device == 0) {
return BAD_VALUE;
}
- if (!inputDesc->mProfile->isCompatibleProfile(devDesc->type(),
- devDesc->address(),
+ if (!inputDesc->mProfile->isCompatibleProfile(DeviceVector(device),
patch->sinks[0].sample_rate,
NULL, /*updatedSampleRate*/
patch->sinks[0].format,
@@ -3116,9 +3222,9 @@
return INVALID_OPERATION;
}
// TODO: reconfigure output format and channels here
- ALOGV("createAudioPatch() setting device %08x on output %d",
- devDesc->type(), inputDesc->mIoHandle);
- setInputDevice(inputDesc->mIoHandle, devDesc->type(), true, handle);
+ ALOGV("%s() setting device %s on output %d", __func__,
+ device->toString().c_str(), inputDesc->mIoHandle);
+ setInputDevice(inputDesc->mIoHandle, device, true, handle);
index = mAudioPatches.indexOfKey(*handle);
if (index >= 0) {
if (patchDesc != 0 && patchDesc != mAudioPatches.valueAt(index)) {
@@ -3138,16 +3244,16 @@
return BAD_VALUE;
}
}
- sp<DeviceDescriptor> srcDeviceDesc =
+ sp<DeviceDescriptor> srcDevice =
mAvailableInputDevices.getDeviceFromId(patch->sources[0].id);
- if (srcDeviceDesc == 0) {
+ if (srcDevice == 0) {
return BAD_VALUE;
}
//update source and sink with our own data as the data passed in the patch may
// be incomplete.
struct audio_patch newPatch = *patch;
- srcDeviceDesc->toAudioPortConfig(&newPatch.sources[0], &patch->sources[0]);
+ srcDevice->toAudioPortConfig(&newPatch.sources[0], &patch->sources[0]);
for (size_t i = 0; i < patch->num_sinks; i++) {
if (patch->sinks[i].type != AUDIO_PORT_TYPE_DEVICE) {
@@ -3155,26 +3261,26 @@
return INVALID_OPERATION;
}
- sp<DeviceDescriptor> sinkDeviceDesc =
+ sp<DeviceDescriptor> sinkDevice =
mAvailableOutputDevices.getDeviceFromId(patch->sinks[i].id);
- if (sinkDeviceDesc == 0) {
+ if (sinkDevice == 0) {
return BAD_VALUE;
}
- sinkDeviceDesc->toAudioPortConfig(&newPatch.sinks[i], &patch->sinks[i]);
+ sinkDevice->toAudioPortConfig(&newPatch.sinks[i], &patch->sinks[i]);
// create a software bridge in PatchPanel if:
// - source and sink devices are on different HW modules OR
// - audio HAL version is < 3.0
// - audio HAL version is >= 3.0 but no route has been declared between devices
- if (!srcDeviceDesc->hasSameHwModuleAs(sinkDeviceDesc) ||
- (srcDeviceDesc->getModuleVersionMajor() < 3) ||
- !srcDeviceDesc->getModule()->supportsPatch(srcDeviceDesc, sinkDeviceDesc)) {
+ if (!srcDevice->hasSameHwModuleAs(sinkDevice) ||
+ (srcDevice->getModuleVersionMajor() < 3) ||
+ !srcDevice->getModule()->supportsPatch(srcDevice, sinkDevice)) {
// support only one sink device for now to simplify output selection logic
if (patch->num_sinks > 1) {
return INVALID_OPERATION;
}
SortedVector<audio_io_handle_t> outputs =
- getOutputsForDevice(sinkDeviceDesc->type(), mOutputs);
+ getOutputsForDevices(DeviceVector(sinkDevice), mOutputs);
// if the sink device is reachable via an opened output stream, request to go via
// this output stream by adding a second source to the patch description
audio_io_handle_t output = selectOutput(outputs);
@@ -3232,11 +3338,11 @@
return BAD_VALUE;
}
- setOutputDevice(outputDesc,
- getNewOutputDevice(outputDesc, true /*fromCache*/),
- true,
- 0,
- NULL);
+ setOutputDevices(outputDesc,
+ getNewOutputDevices(outputDesc, true /*fromCache*/),
+ true,
+ 0,
+ NULL);
} else if (patch->sources[0].type == AUDIO_PORT_TYPE_DEVICE) {
if (patch->sinks[0].type == AUDIO_PORT_TYPE_MIX) {
sp<AudioInputDescriptor> inputDesc = mInputs.getInputFromId(patch->sinks[0].id);
@@ -3359,8 +3465,8 @@
void AudioPolicyManager::checkStrategyRoute(routing_strategy strategy,
audio_io_handle_t ouptutToSkip)
{
- audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
- SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(device, mOutputs);
+ DeviceVector devices = getDevicesForStrategy(strategy, false /*fromCache*/);
+ SortedVector<audio_io_handle_t> outputs = getOutputsForDevices(devices, mOutputs);
for (size_t j = 0; j < mOutputs.size(); j++) {
if (mOutputs.keyAt(j) == ouptutToSkip) {
continue;
@@ -3379,8 +3485,8 @@
}
}
} else {
- audio_devices_t newDevice = getNewOutputDevice(outputDesc, false /*fromCache*/);
- setOutputDevice(outputDesc, newDevice, false);
+ setOutputDevices(
+ outputDesc, getNewOutputDevices(outputDesc, false /*fromCache*/), false);
}
}
}
@@ -3443,7 +3549,8 @@
{
*session = (audio_session_t)mpClientInterface->newAudioUniqueId(AUDIO_UNIQUE_ID_USE_SESSION);
*ioHandle = (audio_io_handle_t)mpClientInterface->newAudioUniqueId(AUDIO_UNIQUE_ID_USE_INPUT);
- *device = getDeviceAndMixForInputSource(AUDIO_SOURCE_HOTWORD);
+ audio_attributes_t attr = { .source = AUDIO_SOURCE_HOTWORD };
+ *device = getDeviceAndMixForAttributes(attr)->type();
return mSoundTriggerSessions.acquireSession(*session, *ioHandle);
}
@@ -3469,10 +3576,11 @@
return INVALID_OPERATION;
}
- sp<DeviceDescriptor> srcDeviceDesc =
+ sp<DeviceDescriptor> srcDevice =
mAvailableInputDevices.getDevice(source->ext.device.type,
- String8(source->ext.device.address));
- if (srcDeviceDesc == 0) {
+ String8(source->ext.device.address),
+ AUDIO_FORMAT_DEFAULT);
+ if (srcDevice == 0) {
ALOGW("%s source->ext.device.type %08x not found", __FUNCTION__, source->ext.device.type);
return BAD_VALUE;
}
@@ -3483,7 +3591,7 @@
sp<AudioPatch> patchDesc = new AudioPatch(&dummyPatch, uid);
sp<SourceClientDescriptor> sourceDesc =
- new SourceClientDescriptor(*portId, uid, *attributes, patchDesc, srcDeviceDesc,
+ new SourceClientDescriptor(*portId, uid, *attributes, patchDesc, srcDevice,
streamTypefromAttributesInt(attributes),
getStrategyForAttr(attributes));
@@ -3504,18 +3612,20 @@
audio_attributes_t attributes = sourceDesc->attributes();
routing_strategy strategy = getStrategyForAttr(&attributes);
audio_stream_type_t stream = sourceDesc->stream();
- sp<DeviceDescriptor> srcDeviceDesc = sourceDesc->srcDevice();
+ sp<DeviceDescriptor> srcDevice = sourceDesc->srcDevice();
- audio_devices_t sinkDevice = getDeviceForStrategy(strategy, true);
- sp<DeviceDescriptor> sinkDeviceDesc =
- mAvailableOutputDevices.getDevice(sinkDevice, String8(""));
+ DeviceVector sinkDevices = getDevicesForStrategy(strategy, true);
+ ALOG_ASSERT(!sinkDevices.isEmpty(), "connectAudioSource(): no device found for strategy");
+ sp<DeviceDescriptor> sinkDevice = sinkDevices.itemAt(0);
+ ALOG_ASSERT(mAvailableOutputDevices.contains(sinkDevice), "%s: Device %s not available",
+ __FUNCTION__, sinkDevice->toString().c_str());
audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
- if (srcDeviceDesc->hasSameHwModuleAs(sinkDeviceDesc) &&
- srcDeviceDesc->getModuleVersionMajor() >= 3 &&
- sinkDeviceDesc->getModule()->supportsPatch(srcDeviceDesc, sinkDeviceDesc) &&
- srcDeviceDesc->getAudioPort()->mGains.size() > 0) {
+ if (srcDevice->hasSameHwModuleAs(sinkDevice) &&
+ srcDevice->getModuleVersionMajor() >= 3 &&
+ sinkDevice->getModule()->supportsPatch(srcDevice, sinkDevice) &&
+ srcDevice->getAudioPort()->mGains.size() > 0) {
ALOGV("%s Device to Device route supported by >=3.0 HAL", __FUNCTION__);
// TODO: may explicitly specify whether we should use HW or SW patch
// create patch between src device and output device
@@ -3532,12 +3642,12 @@
getOutputForAttrInt(&resultAttr, &output, AUDIO_SESSION_NONE,
&attributes, &stream, sourceDesc->uid(), &config, &flags, &selectedDeviceId);
if (output == AUDIO_IO_HANDLE_NONE) {
- ALOGV("%s no output for device %08x", __FUNCTION__, sinkDevice);
+ ALOGV("%s no output for device %08x", __FUNCTION__, sinkDevices.types());
return INVALID_OPERATION;
}
sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
if (outputDesc->isDuplicated()) {
- ALOGV("%s output for device %08x is duplicated", __FUNCTION__, sinkDevice);
+ ALOGV("%s output for device %08x is duplicated", __FUNCTION__, sinkDevices.types());
return INVALID_OPERATION;
}
status_t status = outputDesc->start();
@@ -3551,7 +3661,7 @@
// - the sink is defined by whatever output device is currently selected for the output
// though which this patch is routed.
PatchBuilder patchBuilder;
- patchBuilder.addSource(srcDeviceDesc).addSource(outputDesc, { .stream = stream });
+ patchBuilder.addSource(srcDevice).addSource(outputDesc, { .stream = stream });
status = mpClientInterface->createAudioPatch(patchBuilder.patch(),
&afPatchHandle,
0);
@@ -3753,14 +3863,16 @@
status_t status = setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_HDMI,
AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
address.c_str(),
- name.c_str());
+ name.c_str(),
+ AUDIO_FORMAT_DEFAULT);
if (status != NO_ERROR) {
continue;
}
status = setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_HDMI,
AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
address.c_str(),
- name.c_str());
+ name.c_str(),
+ AUDIO_FORMAT_DEFAULT);
profileUpdated |= (status == NO_ERROR);
}
// FIXME: Why doing this for input HDMI devices if we don't augment their reported formats?
@@ -3773,14 +3885,16 @@
status_t status = setDeviceConnectionStateInt(AUDIO_DEVICE_IN_HDMI,
AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
address.c_str(),
- name.c_str());
+ name.c_str(),
+ AUDIO_FORMAT_DEFAULT);
if (status != NO_ERROR) {
continue;
}
status = setDeviceConnectionStateInt(AUDIO_DEVICE_IN_HDMI,
AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
address.c_str(),
- name.c_str());
+ name.c_str(),
+ AUDIO_FORMAT_DEFAULT);
profileUpdated |= (status == NO_ERROR);
}
@@ -3978,8 +4092,6 @@
// mAvailableOutputDevices and mAvailableInputDevices now contain all attached devices
// open all output streams needed to access attached devices
- audio_devices_t outputDeviceTypes = mAvailableOutputDevices.types();
- audio_devices_t inputDeviceTypes = mAvailableInputDevices.types() & ~AUDIO_DEVICE_BIT_IN;
for (const auto& hwModule : mHwModulesAll) {
hwModule->setHandle(mpClientInterface->loadHwModule(hwModule->getName()));
if (hwModule->getHandle() == AUDIO_MODULE_HANDLE_NONE) {
@@ -4008,51 +4120,49 @@
if ((outProfile->getFlags() & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
continue;
}
- audio_devices_t profileType = outProfile->getSupportedDevicesType();
- if ((profileType & mDefaultOutputDevice->type()) != AUDIO_DEVICE_NONE) {
- profileType = mDefaultOutputDevice->type();
+ const DeviceVector &supportedDevices = outProfile->getSupportedDevices();
+ DeviceVector availProfileDevices = supportedDevices.filter(mAvailableOutputDevices);
+ sp<DeviceDescriptor> supportedDevice = 0;
+ if (supportedDevices.contains(mDefaultOutputDevice)) {
+ supportedDevice = mDefaultOutputDevice;
} else {
- // chose first device present in profile's SupportedDevices also part of
- // outputDeviceTypes
- profileType = outProfile->getSupportedDeviceForType(outputDeviceTypes);
+ // choose first device present in profile's SupportedDevices also part of
+ // mAvailableOutputDevices.
+ if (availProfileDevices.isEmpty()) {
+ continue;
+ }
+ supportedDevice = availProfileDevices.itemAt(0);
}
- if ((profileType & outputDeviceTypes) == 0) {
+ if (!mAvailableOutputDevices.contains(supportedDevice)) {
continue;
}
sp<SwAudioOutputDescriptor> outputDesc = new SwAudioOutputDescriptor(outProfile,
mpClientInterface);
- const DeviceVector &supportedDevices = outProfile->getSupportedDevices();
- const DeviceVector &devicesForType = supportedDevices.getDevicesFromTypeMask(
- profileType);
- String8 address = getFirstDeviceAddress(devicesForType);
audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
- status_t status = outputDesc->open(nullptr, profileType, address,
- AUDIO_STREAM_DEFAULT, AUDIO_OUTPUT_FLAG_NONE, &output);
-
+ status_t status = outputDesc->open(nullptr, DeviceVector(supportedDevice),
+ AUDIO_STREAM_DEFAULT,
+ AUDIO_OUTPUT_FLAG_NONE, &output);
if (status != NO_ERROR) {
- ALOGW("Cannot open output stream for device %08x on hw module %s",
- outputDesc->mDevice,
- hwModule->getName());
- } else {
- for (const auto& dev : supportedDevices) {
- ssize_t index = mAvailableOutputDevices.indexOf(dev);
- // give a valid ID to an attached device once confirmed it is reachable
- if (index >= 0 && !mAvailableOutputDevices[index]->isAttached()) {
- mAvailableOutputDevices[index]->attach(hwModule);
- }
- }
- if (mPrimaryOutput == 0 &&
- outProfile->getFlags() & AUDIO_OUTPUT_FLAG_PRIMARY) {
- mPrimaryOutput = outputDesc;
- }
- addOutput(output, outputDesc);
- setOutputDevice(outputDesc,
- profileType,
- true,
- 0,
- NULL,
- address);
+ ALOGW("Cannot open output stream for devices %s on hw module %s",
+ supportedDevice->toString().c_str(), hwModule->getName());
+ continue;
}
+ for (const auto &device : availProfileDevices) {
+ // give a valid ID to an attached device once confirmed it is reachable
+ if (!device->isAttached()) {
+ device->attach(hwModule);
+ }
+ }
+ if (mPrimaryOutput == 0 &&
+ outProfile->getFlags() & AUDIO_OUTPUT_FLAG_PRIMARY) {
+ mPrimaryOutput = outputDesc;
+ }
+ addOutput(output, outputDesc);
+ setOutputDevices(outputDesc,
+ DeviceVector(supportedDevice),
+ true,
+ 0,
+ NULL);
}
// open input streams needed to access attached devices to validate
// mAvailableInputDevices list
@@ -4067,75 +4177,59 @@
continue;
}
// chose first device present in profile's SupportedDevices also part of
- // inputDeviceTypes
- audio_devices_t profileType = inProfile->getSupportedDeviceForType(inputDeviceTypes);
-
- if ((profileType & inputDeviceTypes) == 0) {
+ // available input devices
+ const DeviceVector &supportedDevices = inProfile->getSupportedDevices();
+ DeviceVector availProfileDevices = supportedDevices.filter(mAvailableInputDevices);
+ if (availProfileDevices.isEmpty()) {
+ ALOGE("%s: Input device list is empty!", __FUNCTION__);
continue;
}
sp<AudioInputDescriptor> inputDesc =
new AudioInputDescriptor(inProfile, mpClientInterface);
- DeviceVector inputDevices = mAvailableInputDevices.getDevicesFromTypeMask(profileType);
- // the inputs vector must be of size >= 1, but we don't want to crash here
- String8 address = getFirstDeviceAddress(inputDevices);
- ALOGV(" for input device 0x%x using address %s", profileType, address.string());
- ALOGE_IF(inputDevices.size() == 0, "Input device list is empty!");
-
audio_io_handle_t input = AUDIO_IO_HANDLE_NONE;
status_t status = inputDesc->open(nullptr,
- profileType,
- address,
+ availProfileDevices.itemAt(0),
AUDIO_SOURCE_MIC,
AUDIO_INPUT_FLAG_NONE,
&input);
-
- if (status == NO_ERROR) {
- for (const auto& dev : inProfile->getSupportedDevices()) {
- ssize_t index = mAvailableInputDevices.indexOf(dev);
- // give a valid ID to an attached device once confirmed it is reachable
- if (index >= 0) {
- sp<DeviceDescriptor> devDesc = mAvailableInputDevices[index];
- if (!devDesc->isAttached()) {
- devDesc->attach(hwModule);
- devDesc->importAudioPort(inProfile, true);
- }
- }
- }
- inputDesc->close();
- } else {
- ALOGW("Cannot open input stream for device %08x on hw module %s",
- profileType,
+ if (status != NO_ERROR) {
+ ALOGW("Cannot open input stream for device %s on hw module %s",
+ availProfileDevices.toString().c_str(),
hwModule->getName());
+ continue;
}
+ for (const auto &device : availProfileDevices) {
+ // give a valid ID to an attached device once confirmed it is reachable
+ if (!device->isAttached()) {
+ device->attach(hwModule);
+ device->importAudioPort(inProfile, true);
+ }
+ }
+ inputDesc->close();
}
}
// make sure all attached devices have been allocated a unique ID
- for (size_t i = 0; i < mAvailableOutputDevices.size();) {
- if (!mAvailableOutputDevices[i]->isAttached()) {
- ALOGW("Output device %08x unreachable", mAvailableOutputDevices[i]->type());
- mAvailableOutputDevices.remove(mAvailableOutputDevices[i]);
- continue;
+ auto checkAndSetAvailable = [this](auto& devices) {
+ for (size_t i = 0; i < devices.size();) {
+ const auto &device = devices[i];
+ if (!device->isAttached()) {
+ ALOGW("device %s is unreachable", device->toString().c_str());
+ devices.remove(device);
+ continue;
+ }
+ // Device is now validated and can be appended to the available devices of the engine
+ mEngine->setDeviceConnectionState(device, AUDIO_POLICY_DEVICE_STATE_AVAILABLE);
+ i++;
}
- // The device is now validated and can be appended to the available devices of the engine
- mEngine->setDeviceConnectionState(mAvailableOutputDevices[i],
- AUDIO_POLICY_DEVICE_STATE_AVAILABLE);
- i++;
- }
- for (size_t i = 0; i < mAvailableInputDevices.size();) {
- if (!mAvailableInputDevices[i]->isAttached()) {
- ALOGW("Input device %08x unreachable", mAvailableInputDevices[i]->type());
- mAvailableInputDevices.remove(mAvailableInputDevices[i]);
- continue;
- }
- // The device is now validated and can be appended to the available devices of the engine
- mEngine->setDeviceConnectionState(mAvailableInputDevices[i],
- AUDIO_POLICY_DEVICE_STATE_AVAILABLE);
- i++;
- }
+ };
+ checkAndSetAvailable(mAvailableOutputDevices);
+ checkAndSetAvailable(mAvailableInputDevices);
+
// make sure default device is reachable
- if (mDefaultOutputDevice == 0 || mAvailableOutputDevices.indexOf(mDefaultOutputDevice) < 0) {
- ALOGE("Default device %08x is unreachable", mDefaultOutputDevice->type());
+ if (mDefaultOutputDevice == 0 || !mAvailableOutputDevices.contains(mDefaultOutputDevice)) {
+ ALOGE_IF(mDefaultOutputDevice != 0, "Default device %s is unreachable",
+ mDefaultOutputDevice->toString().c_str());
status = NO_INIT;
}
// If microphones address is empty, set it according to device type
@@ -4208,44 +4302,28 @@
nextAudioPortGeneration();
}
-void AudioPolicyManager::findIoHandlesByAddress(const sp<SwAudioOutputDescriptor>& desc /*in*/,
- const audio_devices_t device /*in*/,
- const String8& address /*in*/,
- SortedVector<audio_io_handle_t>& outputs /*out*/) {
- sp<DeviceDescriptor> devDesc =
- desc->mProfile->getSupportedDeviceByAddress(device, address);
- if (devDesc != 0) {
- ALOGV("findIoHandlesByAddress(): adding opened output %d on same address %s",
- desc->mIoHandle, address.string());
- outputs.add(desc->mIoHandle);
- }
-}
-
-status_t AudioPolicyManager::checkOutputsForDevice(const sp<DeviceDescriptor>& devDesc,
+status_t AudioPolicyManager::checkOutputsForDevice(const sp<DeviceDescriptor>& device,
audio_policy_dev_state_t state,
- SortedVector<audio_io_handle_t>& outputs,
- const String8& address)
+ SortedVector<audio_io_handle_t>& outputs)
{
- audio_devices_t device = devDesc->type();
+ audio_devices_t deviceType = device->type();
+ const String8 &address = device->address();
sp<SwAudioOutputDescriptor> desc;
- if (audio_device_is_digital(device)) {
+ if (audio_device_is_digital(deviceType)) {
// erase all current sample rates, formats and channel masks
- devDesc->clearAudioProfiles();
+ device->clearAudioProfiles();
}
if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) {
// first list already open outputs that can be routed to this device
for (size_t i = 0; i < mOutputs.size(); i++) {
desc = mOutputs.valueAt(i);
- if (!desc->isDuplicated() && (desc->supportedDevices() & device)) {
- if (!device_distinguishes_on_address(device)) {
- ALOGV("checkOutputsForDevice(): adding opened output %d", mOutputs.keyAt(i));
- outputs.add(mOutputs.keyAt(i));
- } else {
- ALOGV(" checking address match due to device 0x%x", device);
- findIoHandlesByAddress(desc, device, address, outputs);
- }
+ if (!desc->isDuplicated() && desc->supportsDevice(device)
+ && desc->deviceSupportsEncodedFormats(deviceType)) {
+ ALOGV("checkOutputsForDevice(): adding opened output %d on device %s",
+ mOutputs.keyAt(i), device->toString().c_str());
+ outputs.add(mOutputs.keyAt(i));
}
}
// then look for output profiles that can be routed to this device
@@ -4253,13 +4331,10 @@
for (const auto& hwModule : mHwModules) {
for (size_t j = 0; j < hwModule->getOutputProfiles().size(); j++) {
sp<IOProfile> profile = hwModule->getOutputProfiles()[j];
- if (profile->supportDevice(device)) {
- if (!device_distinguishes_on_address(device) ||
- profile->supportDeviceAddress(address)) {
- profiles.add(profile);
- ALOGV("checkOutputsForDevice(): adding profile %zu from module %s",
- j, hwModule->getName());
- }
+ if (profile->supportsDevice(device)) {
+ profiles.add(profile);
+ ALOGV("checkOutputsForDevice(): adding profile %zu from module %s",
+ j, hwModule->getName());
}
}
}
@@ -4267,7 +4342,7 @@
ALOGV(" found %zu profiles, %zu outputs", profiles.size(), outputs.size());
if (profiles.isEmpty() && outputs.isEmpty()) {
- ALOGW("checkOutputsForDevice(): No output available for device %04x", device);
+ ALOGW("checkOutputsForDevice(): No output available for device %04x", deviceType);
return BAD_VALUE;
}
@@ -4283,8 +4358,8 @@
if (!desc->isDuplicated() && desc->mProfile == profile) {
// matching profile: save the sample rates, format and channel masks supported
// by the profile in our device descriptor
- if (audio_device_is_digital(device)) {
- devDesc->importAudioPort(profile);
+ if (audio_device_is_digital(deviceType)) {
+ device->importAudioPort(profile);
}
break;
}
@@ -4300,20 +4375,20 @@
}
ALOGV("opening output for device %08x with params %s profile %p name %s",
- device, address.string(), profile.get(), profile->getName().string());
+ deviceType, address.string(), profile.get(), profile->getName().string());
desc = new SwAudioOutputDescriptor(profile, mpClientInterface);
audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
- status_t status = desc->open(nullptr, device, address,
+ status_t status = desc->open(nullptr, DeviceVector(device),
AUDIO_STREAM_DEFAULT, AUDIO_OUTPUT_FLAG_NONE, &output);
if (status == NO_ERROR) {
// Here is where the out_set_parameters() for card & device gets called
if (!address.isEmpty()) {
- char *param = audio_device_address_to_parameter(device, address);
+ char *param = audio_device_address_to_parameter(deviceType, address);
mpClientInterface->setParameters(output, String8(param));
free(param);
}
- updateAudioProfiles(devDesc, output, profile->getAudioProfiles());
+ updateAudioProfiles(device, output, profile->getAudioProfiles());
if (!profile->hasValidAudioProfile()) {
ALOGW("checkOutputsForDevice() missing param");
desc->close();
@@ -4328,7 +4403,8 @@
config.offload_info.channel_mask = config.channel_mask;
config.offload_info.format = config.format;
- status_t status = desc->open(&config, device, address, AUDIO_STREAM_DEFAULT,
+ status_t status = desc->open(&config, DeviceVector(device),
+ AUDIO_STREAM_DEFAULT,
AUDIO_OUTPUT_FLAG_NONE, &output);
if (status != NO_ERROR) {
output = AUDIO_IO_HANDLE_NONE;
@@ -4337,14 +4413,15 @@
if (output != AUDIO_IO_HANDLE_NONE) {
addOutput(output, desc);
- if (device_distinguishes_on_address(device) && address != "0") {
+ if (device_distinguishes_on_address(deviceType) && address != "0") {
sp<AudioPolicyMix> policyMix;
- if (mPolicyMixes.getAudioPolicyMix(address, policyMix) != NO_ERROR) {
- ALOGE("checkOutputsForDevice() cannot find policy for address %s",
+ if (mPolicyMixes.getAudioPolicyMix(address, policyMix) == NO_ERROR) {
+ policyMix->setOutput(desc);
+ desc->mPolicyMix = policyMix->getMix();
+ } else {
+ ALOGW("checkOutputsForDevice() cannot find policy for address %s",
address.string());
}
- policyMix->setOutput(desc);
- desc->mPolicyMix = policyMix->getMix();
} else if (((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) &&
hasPrimaryOutput()) {
@@ -4376,28 +4453,28 @@
output = AUDIO_IO_HANDLE_NONE;
}
if (output == AUDIO_IO_HANDLE_NONE) {
- ALOGW("checkOutputsForDevice() could not open output for device %x", device);
+ ALOGW("checkOutputsForDevice() could not open output for device %x", deviceType);
profiles.removeAt(profile_index);
profile_index--;
} else {
outputs.add(output);
// Load digital format info only for digital devices
- if (audio_device_is_digital(device)) {
- devDesc->importAudioPort(profile);
+ if (audio_device_is_digital(deviceType)) {
+ device->importAudioPort(profile);
}
- if (device_distinguishes_on_address(device)) {
- ALOGV("checkOutputsForDevice(): setOutputDevice(dev=0x%x, addr=%s)",
- device, address.string());
- setOutputDevice(desc, device, true/*force*/, 0/*delay*/,
- NULL/*patch handle*/, address.string());
+ if (device_distinguishes_on_address(deviceType)) {
+ ALOGV("checkOutputsForDevice(): setOutputDevices %s",
+ device->toString().c_str());
+ setOutputDevices(desc, DeviceVector(device), true/*force*/, 0/*delay*/,
+ NULL/*patch handle*/);
}
ALOGV("checkOutputsForDevice(): adding output %d", output);
}
}
if (profiles.isEmpty()) {
- ALOGW("checkOutputsForDevice(): No output available for device %04x", device);
+ ALOGW("checkOutputsForDevice(): No output available for device %04x", deviceType);
return BAD_VALUE;
}
} else { // Disconnect
@@ -4406,10 +4483,10 @@
desc = mOutputs.valueAt(i);
if (!desc->isDuplicated()) {
// exact match on device
- if (device_distinguishes_on_address(device) &&
- (desc->supportedDevices() == device)) {
- findIoHandlesByAddress(desc, device, address, outputs);
- } else if (!(desc->supportedDevices() & mAvailableOutputDevices.types())) {
+ if (device_distinguishes_on_address(deviceType) && desc->supportsDevice(device)
+ && desc->deviceSupportsEncodedFormats(deviceType)) {
+ outputs.add(mOutputs.keyAt(i));
+ } else if (!mAvailableOutputDevices.containsAtLeastOne(desc->supportedDevices())) {
ALOGV("checkOutputsForDevice(): disconnecting adding output %d",
mOutputs.keyAt(i));
outputs.add(mOutputs.keyAt(i));
@@ -4420,7 +4497,7 @@
for (const auto& hwModule : mHwModules) {
for (size_t j = 0; j < hwModule->getOutputProfiles().size(); j++) {
sp<IOProfile> profile = hwModule->getOutputProfiles()[j];
- if (profile->supportDevice(device)) {
+ if (profile->supportsDevice(device)) {
ALOGV("checkOutputsForDevice(): "
"clearing direct output profile %zu on module %s",
j, hwModule->getName());
@@ -4432,24 +4509,22 @@
return NO_ERROR;
}
-status_t AudioPolicyManager::checkInputsForDevice(const sp<DeviceDescriptor>& devDesc,
+status_t AudioPolicyManager::checkInputsForDevice(const sp<DeviceDescriptor>& device,
audio_policy_dev_state_t state,
- SortedVector<audio_io_handle_t>& inputs,
- const String8& address)
+ SortedVector<audio_io_handle_t>& inputs)
{
- audio_devices_t device = devDesc->type();
sp<AudioInputDescriptor> desc;
- if (audio_device_is_digital(device)) {
+ if (audio_device_is_digital(device->type())) {
// erase all current sample rates, formats and channel masks
- devDesc->clearAudioProfiles();
+ device->clearAudioProfiles();
}
if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) {
// first list already open inputs that can be routed to this device
for (size_t input_index = 0; input_index < mInputs.size(); input_index++) {
desc = mInputs.valueAt(input_index);
- if (desc->mProfile->supportDevice(device)) {
+ if (desc->mProfile->supportsDeviceTypes(device->type())) {
ALOGV("checkInputsForDevice(): adding opened input %d", mInputs.keyAt(input_index));
inputs.add(mInputs.keyAt(input_index));
}
@@ -4463,19 +4538,16 @@
profile_index++) {
sp<IOProfile> profile = hwModule->getInputProfiles()[profile_index];
- if (profile->supportDevice(device)) {
- if (!device_distinguishes_on_address(device) ||
- profile->supportDeviceAddress(address)) {
- profiles.add(profile);
- ALOGV("checkInputsForDevice(): adding profile %zu from module %s",
- profile_index, hwModule->getName());
- }
+ if (profile->supportsDevice(device)) {
+ profiles.add(profile);
+ ALOGV("checkInputsForDevice(): adding profile %zu from module %s",
+ profile_index, hwModule->getName());
}
}
}
if (profiles.isEmpty() && inputs.isEmpty()) {
- ALOGW("checkInputsForDevice(): No input available for device 0x%X", device);
+ ALOGW("%s: No input available for device %s", __func__, device->toString().c_str());
return BAD_VALUE;
}
@@ -4490,8 +4562,8 @@
for (input_index = 0; input_index < mInputs.size(); input_index++) {
desc = mInputs.valueAt(input_index);
if (desc->mProfile == profile) {
- if (audio_device_is_digital(device)) {
- devDesc->importAudioPort(profile);
+ if (audio_device_is_digital(device->type())) {
+ device->importAudioPort(profile);
}
break;
}
@@ -4510,18 +4582,18 @@
audio_io_handle_t input = AUDIO_IO_HANDLE_NONE;
status_t status = desc->open(nullptr,
device,
- address,
AUDIO_SOURCE_MIC,
AUDIO_INPUT_FLAG_NONE,
&input);
if (status == NO_ERROR) {
+ const String8& address = device->address();
if (!address.isEmpty()) {
- char *param = audio_device_address_to_parameter(device, address);
+ char *param = audio_device_address_to_parameter(device->type(), address);
mpClientInterface->setParameters(input, String8(param));
free(param);
}
- updateAudioProfiles(devDesc, input, profile->getAudioProfiles());
+ updateAudioProfiles(device, input, profile->getAudioProfiles());
if (!profile->hasValidAudioProfile()) {
ALOGW("checkInputsForDevice() direct input missing param");
desc->close();
@@ -4534,20 +4606,21 @@
} // endif input != 0
if (input == AUDIO_IO_HANDLE_NONE) {
- ALOGW("checkInputsForDevice() could not open input for device 0x%X", device);
+ ALOGW("%s could not open input for device %s", __func__,
+ device->toString().c_str());
profiles.removeAt(profile_index);
profile_index--;
} else {
inputs.add(input);
- if (audio_device_is_digital(device)) {
- devDesc->importAudioPort(profile);
+ if (audio_device_is_digital(device->type())) {
+ device->importAudioPort(profile);
}
ALOGV("checkInputsForDevice(): adding input %d", input);
}
} // end scan profiles
if (profiles.isEmpty()) {
- ALOGW("checkInputsForDevice(): No input available for device 0x%X", device);
+ ALOGW("%s: No input available for device %s", __func__, device->toString().c_str());
return BAD_VALUE;
}
} else {
@@ -4555,7 +4628,7 @@
// check if one opened input is not needed any more after disconnecting one device
for (size_t input_index = 0; input_index < mInputs.size(); input_index++) {
desc = mInputs.valueAt(input_index);
- if (!(desc->mProfile->supportDevice(mAvailableInputDevices.types()))) {
+ if (!mAvailableInputDevices.containsAtLeastOne(desc->supportedDevices())) {
ALOGV("checkInputsForDevice(): disconnecting adding input %d",
mInputs.keyAt(input_index));
inputs.add(mInputs.keyAt(input_index));
@@ -4567,7 +4640,7 @@
profile_index < hwModule->getInputProfiles().size();
profile_index++) {
sp<IOProfile> profile = hwModule->getInputProfiles()[profile_index];
- if (profile->supportDevice(device)) {
+ if (profile->supportsDevice(device)) {
ALOGV("checkInputsForDevice(): clearing direct input profile %zu on module %s",
profile_index, hwModule->getName());
profile->clearAudioProfiles();
@@ -4641,7 +4714,7 @@
// MSD patches may have been released to support a non-MSD direct output. Reset MSD patch if
// no direct outputs are open.
- if (mHwModules.getModuleFromName(AUDIO_HARDWARE_MODULE_ID_MSD) != 0) {
+ if (!getMsdAudioOutDevices().isEmpty()) {
bool directOutputOpen = false;
for (size_t i = 0; i < mOutputs.size(); i++) {
if (mOutputs[i]->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
@@ -4668,7 +4741,7 @@
nextAudioPortGeneration();
- audio_devices_t device = inputDesc->mDevice;
+ sp<DeviceDescriptor> device = inputDesc->getDevice();
ssize_t index = mAudioPatches.indexOfKey(inputDesc->getPatchHandle());
if (index >= 0) {
sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
@@ -4680,26 +4753,27 @@
inputDesc->close();
mInputs.removeItem(input);
- audio_devices_t primaryInputDevices = availablePrimaryInputDevices();
- if (((device & primaryInputDevices & ~AUDIO_DEVICE_BIT_IN) != 0) &&
+ DeviceVector primaryInputDevices = availablePrimaryModuleInputDevices();
+ if (primaryInputDevices.contains(device) &&
mInputs.activeInputsCountOnDevices(primaryInputDevices) == 0) {
SoundTrigger::setCaptureState(false);
}
}
-SortedVector<audio_io_handle_t> AudioPolicyManager::getOutputsForDevice(
- audio_devices_t device,
- const SwAudioOutputCollection& openOutputs)
+SortedVector<audio_io_handle_t> AudioPolicyManager::getOutputsForDevices(
+ const DeviceVector &devices,
+ const SwAudioOutputCollection& openOutputs)
{
SortedVector<audio_io_handle_t> outputs;
- ALOGVV("getOutputsForDevice() device %04x", device);
+ ALOGVV("%s() devices %s", __func__, devices.toString().c_str());
for (size_t i = 0; i < openOutputs.size(); i++) {
- ALOGVV("output %zu isDuplicated=%d device=%04x",
+ ALOGVV("output %zu isDuplicated=%d device=%s",
i, openOutputs.valueAt(i)->isDuplicated(),
- openOutputs.valueAt(i)->supportedDevices());
- if ((device & openOutputs.valueAt(i)->supportedDevices()) == device) {
- ALOGVV("getOutputsForDevice() found output %d", openOutputs.keyAt(i));
+ openOutputs.valueAt(i)->supportedDevices().toString().c_str());
+ if (openOutputs.valueAt(i)->supportsAllDevices(devices)
+ && openOutputs.valueAt(i)->deviceSupportsEncodedFormats(devices.types())) {
+ ALOGVV("%s() found output %d", __func__, openOutputs.keyAt(i));
outputs.add(openOutputs.keyAt(i));
}
}
@@ -4721,10 +4795,10 @@
void AudioPolicyManager::checkOutputForStrategy(routing_strategy strategy)
{
- audio_devices_t oldDevice = getDeviceForStrategy(strategy, true /*fromCache*/);
- audio_devices_t newDevice = getDeviceForStrategy(strategy, false /*fromCache*/);
- SortedVector<audio_io_handle_t> srcOutputs = getOutputsForDevice(oldDevice, mPreviousOutputs);
- SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevice(newDevice, mOutputs);
+ DeviceVector oldDevices = getDevicesForStrategy(strategy, true /*fromCache*/);
+ DeviceVector newDevices = getDevicesForStrategy(strategy, false /*fromCache*/);
+ SortedVector<audio_io_handle_t> srcOutputs = getOutputsForDevices(oldDevices, mPreviousOutputs);
+ SortedVector<audio_io_handle_t> dstOutputs = getOutputsForDevices(newDevices, mOutputs);
// also take into account external policy-related changes: add all outputs which are
// associated with policies in the "before" and "after" output vectors
@@ -4744,7 +4818,7 @@
}
}
- if (srcOutputs != dstOutputs) {
+ if (!dstOutputs.isEmpty() && srcOutputs != dstOutputs) {
// get maximum latency of all source outputs to determine the minimum mute time guaranteeing
// audio from invalidated tracks will be rendered when unmuting
uint32_t maxLatency = 0;
@@ -4754,14 +4828,17 @@
maxLatency = desc->latency();
}
}
- ALOGV("checkOutputForStrategy() strategy %d, moving from output %d to output %d",
- strategy, srcOutputs[0], dstOutputs[0]);
+ ALOGV_IF(!(srcOutputs.isEmpty() || dstOutputs.isEmpty()),
+ "%s: strategy %d, moving from output %s to output %s", __func__, strategy,
+ std::to_string(srcOutputs[0]).c_str(),
+ std::to_string(dstOutputs[0]).c_str());
// mute strategy while moving tracks from one output to another
for (audio_io_handle_t srcOut : srcOutputs) {
sp<SwAudioOutputDescriptor> desc = mPreviousOutputs.valueFor(srcOut);
if (desc != 0 && isStrategyActive(desc, strategy)) {
setStrategyMute(strategy, true, desc);
- setStrategyMute(strategy, false, desc, maxLatency * LATENCY_MUTE_FACTOR, newDevice);
+ setStrategyMute(strategy, false, desc, maxLatency * LATENCY_MUTE_FACTOR,
+ newDevices.types());
}
sp<SourceClientDescriptor> source =
getSourceForStrategyOnOutput(srcOut, strategy);
@@ -4880,26 +4957,35 @@
return device;
}
-audio_devices_t AudioPolicyManager::getNewOutputDevice(const sp<AudioOutputDescriptor>& outputDesc,
- bool fromCache)
+DeviceVector AudioPolicyManager::getNewOutputDevices(const sp<SwAudioOutputDescriptor>& outputDesc,
+ bool fromCache)
{
+ DeviceVector devices;
+
ssize_t index = mAudioPatches.indexOfKey(outputDesc->getPatchHandle());
if (index >= 0) {
sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
if (patchDesc->mUid != mUidCached) {
- ALOGV("getNewOutputDevice() device %08x forced by patch %d",
- outputDesc->device(), outputDesc->getPatchHandle());
- return outputDesc->device();
+ ALOGV("%s device %s forced by patch %d", __func__,
+ outputDesc->devices().toString().c_str(), outputDesc->getPatchHandle());
+ return outputDesc->devices();
}
}
// Honor explicit routing requests only if no client using default routing is active on this
// input: a specific app can not force routing for other apps by setting a preferred device.
bool active; // unused
- sp<DeviceDescriptor> deviceDesc =
+ sp<DeviceDescriptor> device =
findPreferredDevice(outputDesc, STRATEGY_NONE, active, mAvailableOutputDevices);
- if (deviceDesc != nullptr) {
- return deviceDesc->type();
+ if (device != nullptr) {
+ return DeviceVector(device);
+ }
+
+ // Legacy Engine cannot take care of bus devices and mix, so we need to handle the conflict
+ // of setForceUse / Default Bus device here
+ device = mPolicyMixes.getDeviceAndMixForOutput(outputDesc, mAvailableOutputDevices);
+ if (device != nullptr) {
+ return DeviceVector(device);
}
// check the following by order of priority to request a routing change if necessary:
@@ -4925,66 +5011,65 @@
// FIXME: extend use of isStrategyActiveOnSameModule() to all strategies
// with a refined rule considering mutually exclusive devices (using same backend)
// as opposed to all streams on the same audio HAL module.
- audio_devices_t device = AUDIO_DEVICE_NONE;
if (isStrategyActive(outputDesc, STRATEGY_ENFORCED_AUDIBLE) &&
mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) {
- device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache);
+ devices = getDevicesForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache);
} else if (isInCall() ||
isStrategyActiveOnSameModule(outputDesc, STRATEGY_PHONE)) {
- device = getDeviceForStrategy(STRATEGY_PHONE, fromCache);
+ devices = getDevicesForStrategy(STRATEGY_PHONE, fromCache);
} else if (isStrategyActiveOnSameModule(outputDesc, STRATEGY_SONIFICATION)) {
- device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache);
+ devices = getDevicesForStrategy(STRATEGY_SONIFICATION, fromCache);
} else if (isStrategyActive(outputDesc, STRATEGY_ENFORCED_AUDIBLE)) {
- device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache);
+ devices = getDevicesForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache);
} else if (isStrategyActive(outputDesc, STRATEGY_ACCESSIBILITY)) {
- device = getDeviceForStrategy(STRATEGY_ACCESSIBILITY, fromCache);
+ devices = getDevicesForStrategy(STRATEGY_ACCESSIBILITY, fromCache);
} else if (isStrategyActive(outputDesc, STRATEGY_SONIFICATION_RESPECTFUL)) {
- device = getDeviceForStrategy(STRATEGY_SONIFICATION_RESPECTFUL, fromCache);
+ devices = getDevicesForStrategy(STRATEGY_SONIFICATION_RESPECTFUL, fromCache);
} else if (isStrategyActive(outputDesc, STRATEGY_MEDIA)) {
- device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache);
+ devices = getDevicesForStrategy(STRATEGY_MEDIA, fromCache);
} else if (isStrategyActive(outputDesc, STRATEGY_DTMF)) {
- device = getDeviceForStrategy(STRATEGY_DTMF, fromCache);
+ devices = getDevicesForStrategy(STRATEGY_DTMF, fromCache);
} else if (isStrategyActive(outputDesc, STRATEGY_TRANSMITTED_THROUGH_SPEAKER)) {
- device = getDeviceForStrategy(STRATEGY_TRANSMITTED_THROUGH_SPEAKER, fromCache);
+ devices = getDevicesForStrategy(STRATEGY_TRANSMITTED_THROUGH_SPEAKER, fromCache);
} else if (isStrategyActive(outputDesc, STRATEGY_REROUTING)) {
- device = getDeviceForStrategy(STRATEGY_REROUTING, fromCache);
+ devices = getDevicesForStrategy(STRATEGY_REROUTING, fromCache);
}
- ALOGV("getNewOutputDevice() selected device %x", device);
- return device;
+ ALOGV("getNewOutputDevice() selected devices %s", devices.toString().c_str());
+ return devices;
}
-audio_devices_t AudioPolicyManager::getNewInputDevice(const sp<AudioInputDescriptor>& inputDesc)
+sp<DeviceDescriptor> AudioPolicyManager::getNewInputDevice(
+ const sp<AudioInputDescriptor>& inputDesc)
{
- audio_devices_t device = AUDIO_DEVICE_NONE;
+ sp<DeviceDescriptor> device;
ssize_t index = mAudioPatches.indexOfKey(inputDesc->getPatchHandle());
if (index >= 0) {
sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
if (patchDesc->mUid != mUidCached) {
- ALOGV("getNewInputDevice() device %08x forced by patch %d",
- inputDesc->mDevice, inputDesc->getPatchHandle());
- return inputDesc->mDevice;
+ ALOGV("getNewInputDevice() device %s forced by patch %d",
+ inputDesc->getDevice()->toString().c_str(), inputDesc->getPatchHandle());
+ return inputDesc->getDevice();
}
}
// Honor explicit routing requests only if no client using default routing is active on this
// input: a specific app can not force routing for other apps by setting a preferred device.
bool active;
- sp<DeviceDescriptor> deviceDesc =
- findPreferredDevice(inputDesc, AUDIO_SOURCE_DEFAULT, active, mAvailableInputDevices);
- if (deviceDesc != nullptr) {
- return deviceDesc->type();
+ device = findPreferredDevice(inputDesc, AUDIO_SOURCE_DEFAULT, active, mAvailableInputDevices);
+ if (device != nullptr) {
+ return device;
}
// If we are not in call and no client is active on this input, this methods returns
// AUDIO_DEVICE_NONE, causing the patch on the input stream to be released.
- audio_source_t source = inputDesc->source();
- if (source == AUDIO_SOURCE_DEFAULT && isInCall()) {
- source = AUDIO_SOURCE_VOICE_COMMUNICATION;
+ audio_attributes_t attributes = inputDesc->getHighestPriorityAttributes();
+ if (attributes.source == AUDIO_SOURCE_DEFAULT && isInCall()) {
+ attributes.source = AUDIO_SOURCE_VOICE_COMMUNICATION;
}
- if (source != AUDIO_SOURCE_DEFAULT) {
- device = getDeviceAndMixForInputSource(source);
+ if (attributes.source != AUDIO_SOURCE_DEFAULT) {
+ device = getDeviceAndMixForAttributes(attributes);
}
return device;
@@ -5006,36 +5091,37 @@
if (stream < (audio_stream_type_t) 0 || stream >= AUDIO_STREAM_PUBLIC_CNT) {
return AUDIO_DEVICE_NONE;
}
- audio_devices_t activeDevices = AUDIO_DEVICE_NONE;
- audio_devices_t devices = AUDIO_DEVICE_NONE;
+ DeviceVector activeDevices;
+ DeviceVector devices;
for (int curStream = 0; curStream < AUDIO_STREAM_FOR_POLICY_CNT; curStream++) {
if (!streamsMatchForvolume(stream, (audio_stream_type_t)curStream)) {
continue;
}
routing_strategy curStrategy = getStrategy((audio_stream_type_t)curStream);
- audio_devices_t curDevices =
- getDeviceForStrategy((routing_strategy)curStrategy, false /*fromCache*/);
- devices |= curDevices;
- for (audio_io_handle_t output : getOutputsForDevice(curDevices, mOutputs)) {
+ DeviceVector curDevices =
+ getDevicesForStrategy((routing_strategy)curStrategy, false /*fromCache*/);
+ devices.merge(curDevices);
+ for (audio_io_handle_t output : getOutputsForDevices(curDevices, mOutputs)) {
sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
if (outputDesc->isStreamActive((audio_stream_type_t)curStream)) {
- activeDevices |= outputDesc->device();
+ activeDevices.merge(outputDesc->devices());
}
}
}
// Favor devices selected on active streams if any to report correct device in case of
// explicit device selection
- if (activeDevices != AUDIO_DEVICE_NONE) {
+ if (!activeDevices.isEmpty()) {
devices = activeDevices;
}
/*Filter SPEAKER_SAFE out of results, as AudioService doesn't know about it
and doesn't really need to.*/
- if (devices & AUDIO_DEVICE_OUT_SPEAKER_SAFE) {
- devices |= AUDIO_DEVICE_OUT_SPEAKER;
- devices &= ~AUDIO_DEVICE_OUT_SPEAKER_SAFE;
+ DeviceVector speakerSafeDevices = devices.getDevicesFromTypeMask(AUDIO_DEVICE_OUT_SPEAKER_SAFE);
+ if (!speakerSafeDevices.isEmpty()) {
+ devices.merge(mAvailableOutputDevices.getDevicesFromTypeMask(AUDIO_DEVICE_OUT_SPEAKER));
+ devices.remove(speakerSafeDevices);
}
- return devices;
+ return devices.types();
}
routing_strategy AudioPolicyManager::getStrategy(audio_stream_type_t stream) const
@@ -5126,34 +5212,33 @@
return 0;
}
-audio_devices_t AudioPolicyManager::getDeviceForStrategy(routing_strategy strategy,
- bool fromCache)
+DeviceVector AudioPolicyManager::getDevicesForStrategy(routing_strategy strategy, bool fromCache)
{
// Honor explicit routing requests only if all active clients have a preferred route in which
// case the last active client route is used
- sp<DeviceDescriptor> deviceDesc = findPreferredDevice(mOutputs, strategy, mAvailableOutputDevices);
- if (deviceDesc != nullptr) {
- return deviceDesc->type();
+ sp<DeviceDescriptor> device = findPreferredDevice(mOutputs, strategy, mAvailableOutputDevices);
+ if (device != nullptr) {
+ return DeviceVector(device);
}
if (fromCache) {
- ALOGVV("getDeviceForStrategy() from cache strategy %d, device %x",
- strategy, mDeviceForStrategy[strategy]);
- return mDeviceForStrategy[strategy];
+ ALOGVV("%s from cache strategy %d, device %s", __func__, strategy,
+ mDevicesForStrategy[strategy].toString().c_str());
+ return mDevicesForStrategy[strategy];
}
- return mEngine->getDeviceForStrategy(strategy);
+ return mAvailableOutputDevices.getDevicesFromTypeMask(mEngine->getDeviceForStrategy(strategy));
}
void AudioPolicyManager::updateDevicesAndOutputs()
{
for (int i = 0; i < NUM_STRATEGIES; i++) {
- mDeviceForStrategy[i] = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/);
+ mDevicesForStrategy[i] = getDevicesForStrategy((routing_strategy)i, false /*fromCache*/);
}
mPreviousOutputs = mOutputs;
}
uint32_t AudioPolicyManager::checkDeviceMuteStrategies(const sp<AudioOutputDescriptor>& outputDesc,
- audio_devices_t prevDevice,
+ audio_devices_t prevDeviceType,
uint32_t delayMs)
{
// mute/unmute strategies using an incompatible device combination
@@ -5164,13 +5249,14 @@
}
uint32_t muteWaitMs = 0;
- audio_devices_t device = outputDesc->device();
- bool shouldMute = outputDesc->isActive() && (popcount(device) >= 2);
+ audio_devices_t deviceType = outputDesc->devices().types();
+ bool shouldMute = outputDesc->isActive() && (popcount(deviceType) >= 2);
for (size_t i = 0; i < NUM_STRATEGIES; i++) {
- audio_devices_t curDevice = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/);
- curDevice = curDevice & outputDesc->supportedDevices();
- bool mute = shouldMute && (curDevice & device) && (curDevice != device);
+ audio_devices_t curDeviceType =
+ getDeviceForStrategy((routing_strategy)i, false /*fromCache*/);
+ curDeviceType = curDeviceType & outputDesc->supportedDevices().types();
+ bool mute = shouldMute && (curDeviceType & deviceType) && (curDeviceType != deviceType);
bool doMute = false;
if (mute && !outputDesc->mStrategyMutedByDevice[i]) {
@@ -5184,12 +5270,11 @@
for (size_t j = 0; j < mOutputs.size(); j++) {
sp<AudioOutputDescriptor> desc = mOutputs.valueAt(j);
// skip output if it does not share any device with current output
- if ((desc->supportedDevices() & outputDesc->supportedDevices())
- == AUDIO_DEVICE_NONE) {
+ if (!desc->supportedDevices().containsAtLeastOne(outputDesc->supportedDevices())) {
continue;
}
ALOGVV("checkDeviceMuteStrategies() %s strategy %zu (curDevice %04x)",
- mute ? "muting" : "unmuting", i, curDevice);
+ mute ? "muting" : "unmuting", i, curDeviceType);
setStrategyMute((routing_strategy)i, mute, desc, mute ? 0 : delayMs);
if (isStrategyActive(desc, (routing_strategy)i)) {
if (mute) {
@@ -5209,7 +5294,7 @@
// temporary mute output if device selection changes to avoid volume bursts due to
// different per device volumes
- if (outputDesc->isActive() && (device != prevDevice)) {
+ if (outputDesc->isActive() && (deviceType != prevDeviceType)) {
uint32_t tempMuteWaitMs = outputDesc->latency() * 2;
// temporary mute duration is conservatively set to 4 times the reported latency
uint32_t tempMuteDurationMs = outputDesc->latency() * 4;
@@ -5223,7 +5308,7 @@
// delayed device change
setStrategyMute((routing_strategy)i, true, outputDesc, delayMs);
setStrategyMute((routing_strategy)i, false, outputDesc,
- delayMs + tempMuteDurationMs, device);
+ delayMs + tempMuteDurationMs, deviceType);
}
}
}
@@ -5237,46 +5322,45 @@
return 0;
}
-uint32_t AudioPolicyManager::setOutputDevice(const sp<AudioOutputDescriptor>& outputDesc,
- audio_devices_t device,
- bool force,
- int delayMs,
- audio_patch_handle_t *patchHandle,
- const char *address,
- bool requiresMuteCheck)
+uint32_t AudioPolicyManager::setOutputDevices(const sp<SwAudioOutputDescriptor>& outputDesc,
+ const DeviceVector &devices,
+ bool force,
+ int delayMs,
+ audio_patch_handle_t *patchHandle,
+ bool requiresMuteCheck)
{
- ALOGV("setOutputDevice() device %04x delayMs %d", device, delayMs);
- AudioParameter param;
+ ALOGV("%s device %s delayMs %d", __func__, devices.toString().c_str(), delayMs);
uint32_t muteWaitMs;
if (outputDesc->isDuplicated()) {
- muteWaitMs = setOutputDevice(outputDesc->subOutput1(), device, force, delayMs,
- nullptr /* patchHandle */, nullptr /* address */, requiresMuteCheck);
- muteWaitMs += setOutputDevice(outputDesc->subOutput2(), device, force, delayMs,
- nullptr /* patchHandle */, nullptr /* address */, requiresMuteCheck);
+ muteWaitMs = setOutputDevices(outputDesc->subOutput1(), devices, force, delayMs,
+ nullptr /* patchHandle */, requiresMuteCheck);
+ muteWaitMs += setOutputDevices(outputDesc->subOutput2(), devices, force, delayMs,
+ nullptr /* patchHandle */, requiresMuteCheck);
return muteWaitMs;
}
- // no need to proceed if new device is not AUDIO_DEVICE_NONE and not supported by current
- // output profile
- if ((device != AUDIO_DEVICE_NONE) &&
- ((device & outputDesc->supportedDevices()) == AUDIO_DEVICE_NONE)) {
- return 0;
- }
// filter devices according to output selected
- device = (audio_devices_t)(device & outputDesc->supportedDevices());
+ DeviceVector filteredDevices = outputDesc->filterSupportedDevices(devices);
- audio_devices_t prevDevice = outputDesc->mDevice;
+ // no need to proceed if new device is not AUDIO_DEVICE_NONE and not supported by current
+ // output profile
+ if (!devices.isEmpty() && filteredDevices.isEmpty()) {
+ ALOGV("%s: unsupported device %s for output", __func__, devices.toString().c_str());
+ return 0;
+ }
- ALOGV("setOutputDevice() prevDevice 0x%04x", prevDevice);
+ DeviceVector prevDevices = outputDesc->devices();
- if (device != AUDIO_DEVICE_NONE) {
- outputDesc->mDevice = device;
+ ALOGV("setOutputDevices() prevDevice %s", prevDevices.toString().c_str());
+
+ if (!filteredDevices.isEmpty()) {
+ outputDesc->setDevices(filteredDevices);
}
// if the outputs are not materially active, there is no need to mute.
if (requiresMuteCheck) {
- muteWaitMs = checkDeviceMuteStrategies(outputDesc, prevDevice, delayMs);
+ muteWaitMs = checkDeviceMuteStrategies(outputDesc, prevDevices.types(), delayMs);
} else {
ALOGV("%s: suppressing checkDeviceMuteStrategies", __func__);
muteWaitMs = 0;
@@ -5287,42 +5371,32 @@
// OR the requested device is the same as current device
// AND force is not specified
// AND the output is connected by a valid audio patch.
- // Doing this check here allows the caller to call setOutputDevice() without conditions
- if ((device == AUDIO_DEVICE_NONE || device == prevDevice) &&
- !force &&
- outputDesc->getPatchHandle() != 0) {
- ALOGV("setOutputDevice() setting same device 0x%04x or null device", device);
+ // Doing this check here allows the caller to call setOutputDevices() without conditions
+ if ((filteredDevices.isEmpty() || filteredDevices == prevDevices) &&
+ !force && outputDesc->getPatchHandle() != 0) {
+ ALOGV("%s setting same device %s or null device, force=%d, patch handle=%d", __func__,
+ filteredDevices.toString().c_str(), force, outputDesc->getPatchHandle());
return muteWaitMs;
}
- ALOGV("setOutputDevice() changing device");
+ ALOGV("%s changing device to %s", __func__, filteredDevices.toString().c_str());
// do the routing
- if (device == AUDIO_DEVICE_NONE) {
+ if (filteredDevices.isEmpty()) {
resetOutputDevice(outputDesc, delayMs, NULL);
} else {
- DeviceVector deviceList;
- if ((address == NULL) || (strlen(address) == 0)) {
- deviceList = mAvailableOutputDevices.getDevicesFromTypeMask(device);
- } else {
- sp<DeviceDescriptor> deviceDesc = mAvailableOutputDevices.getDevice(
- device, String8(address));
- if (deviceDesc) deviceList.add(deviceDesc);
+ PatchBuilder patchBuilder;
+ patchBuilder.addSource(outputDesc);
+ ALOG_ASSERT(filteredDevices.size() <= AUDIO_PATCH_PORTS_MAX, "Too many sink ports");
+ for (const auto &filteredDevice : filteredDevices) {
+ patchBuilder.addSink(filteredDevice);
}
- if (!deviceList.isEmpty()) {
- PatchBuilder patchBuilder;
- patchBuilder.addSource(outputDesc);
- ALOG_ASSERT(deviceList.size() <= AUDIO_PATCH_PORTS_MAX, "Too many sink ports");
- for (const auto &device : deviceList) {
- patchBuilder.addSink(device);
- }
- installPatch(__func__, patchHandle, outputDesc.get(), patchBuilder.patch(), delayMs);
- }
+ installPatch(__func__, patchHandle, outputDesc.get(), patchBuilder.patch(), delayMs);
}
// update stream volumes according to new device
- applyStreamVolumes(outputDesc, device, delayMs);
+ applyStreamVolumes(outputDesc, filteredDevices.types(), delayMs);
return muteWaitMs;
}
@@ -5351,18 +5425,17 @@
}
status_t AudioPolicyManager::setInputDevice(audio_io_handle_t input,
- audio_devices_t device,
+ const sp<DeviceDescriptor> &device,
bool force,
audio_patch_handle_t *patchHandle)
{
status_t status = NO_ERROR;
sp<AudioInputDescriptor> inputDesc = mInputs.valueFor(input);
- if ((device != AUDIO_DEVICE_NONE) && ((device != inputDesc->mDevice) || force)) {
- inputDesc->mDevice = device;
+ if ((device != nullptr) && ((device != inputDesc->getDevice()) || force)) {
+ inputDesc->setDevice(device);
- DeviceVector deviceList = mAvailableInputDevices.getDevicesFromTypeMask(device);
- if (!deviceList.isEmpty()) {
+ if (mAvailableInputDevices.contains(device)) {
PatchBuilder patchBuilder;
patchBuilder.addSink(inputDesc,
// AUDIO_SOURCE_HOTWORD is for internal use only:
@@ -5374,7 +5447,7 @@
}
return result; }).
//only one input device for now
- addSource(deviceList.itemAt(0));
+ addSource(device);
status = installPatch(__func__, patchHandle, inputDesc.get(), patchBuilder.patch(), 0);
}
}
@@ -5404,8 +5477,7 @@
return status;
}
-sp<IOProfile> AudioPolicyManager::getInputProfile(audio_devices_t device,
- const String8& address,
+sp<IOProfile> AudioPolicyManager::getInputProfile(const sp<DeviceDescriptor> &device,
uint32_t& samplingRate,
audio_format_t& format,
audio_channel_mask_t& channelMask,
@@ -5425,7 +5497,7 @@
for (const auto& profile : hwModule->getInputProfiles()) {
// profile->log();
//updatedFormat = format;
- if (profile->isCompatibleProfile(device, address, samplingRate,
+ if (profile->isCompatibleProfile(DeviceVector(device), samplingRate,
&samplingRate /*updatedSamplingRate*/,
format,
&format, /*updatedFormat*/
@@ -5436,7 +5508,7 @@
true /*exactMatchRequiredForInputFlags*/)) {
return profile;
}
- if (firstInexact == nullptr && profile->isCompatibleProfile(device, address,
+ if (firstInexact == nullptr && profile->isCompatibleProfile(DeviceVector(device),
samplingRate,
&updatedSamplingRate,
format,
@@ -5460,32 +5532,34 @@
return NULL;
}
-
-audio_devices_t AudioPolicyManager::getDeviceAndMixForInputSource(audio_source_t inputSource,
- AudioMix **policyMix)
+sp<DeviceDescriptor> AudioPolicyManager::getDeviceAndMixForAttributes(
+ const audio_attributes_t &attributes, AudioMix **policyMix)
{
// Honor explicit routing requests only if all active clients have a preferred route in which
// case the last active client route is used
- sp<DeviceDescriptor> deviceDesc =
- findPreferredDevice(mInputs, inputSource, mAvailableInputDevices);
- if (deviceDesc != nullptr) {
- return deviceDesc->type();
+ sp<DeviceDescriptor> device =
+ findPreferredDevice(mInputs, attributes.source, mAvailableInputDevices);
+ if (device != nullptr) {
+ return device;
}
-
- audio_devices_t availableDeviceTypes = mAvailableInputDevices.types() & ~AUDIO_DEVICE_BIT_IN;
- audio_devices_t selectedDeviceFromMix =
- mPolicyMixes.getDeviceAndMixForInputSource(inputSource, availableDeviceTypes, policyMix);
-
- if (selectedDeviceFromMix != AUDIO_DEVICE_NONE) {
- return selectedDeviceFromMix;
- }
- return getDeviceForInputSource(inputSource);
+ sp<DeviceDescriptor> selectedDeviceFromMix =
+ mPolicyMixes.getDeviceAndMixForInputSource(attributes.source, mAvailableInputDevices,
+ policyMix);
+ return (selectedDeviceFromMix != nullptr) ?
+ selectedDeviceFromMix : getDeviceForAttributes(attributes);
}
-audio_devices_t AudioPolicyManager::getDeviceForInputSource(audio_source_t inputSource)
+sp<DeviceDescriptor> AudioPolicyManager::getDeviceForAttributes(const audio_attributes_t &attributes)
{
- return mEngine->getDeviceForInputSource(inputSource);
+ audio_devices_t device = mEngine->getDeviceForInputSource(attributes.source);
+ if (attributes.source == AUDIO_SOURCE_REMOTE_SUBMIX &&
+ strncmp(attributes.tags, "addr=", strlen("addr=")) == 0) {
+ return mAvailableInputDevices.getDevice(AUDIO_DEVICE_IN_REMOTE_SUBMIX,
+ String8(attributes.tags + strlen("addr=")),
+ AUDIO_FORMAT_DEFAULT);
+ }
+ return mAvailableInputDevices.getDevice(device, String8(), AUDIO_FORMAT_DEFAULT);
}
float AudioPolicyManager::computeVolume(audio_stream_type_t stream,
@@ -5603,6 +5677,15 @@
float minDst = (float)mVolumeCurves->getVolumeIndexMin(dstStream);
float maxDst = (float)mVolumeCurves->getVolumeIndexMax(dstStream);
+ // preserve mute request or correct range
+ if (srcIndex < minSrc) {
+ if (srcIndex == 0) {
+ return 0;
+ }
+ srcIndex = minSrc;
+ } else if (srcIndex > maxSrc) {
+ srcIndex = maxSrc;
+ }
return (int)(minDst + ((srcIndex - minSrc) * (maxDst - minDst)) / (maxSrc - minSrc));
}
@@ -5630,7 +5713,7 @@
}
if (device == AUDIO_DEVICE_NONE) {
- device = outputDesc->device();
+ device = outputDesc->devices().types();
}
float volumeDb = computeVolume(stream, index, device);
@@ -5701,7 +5784,7 @@
audio_devices_t device)
{
if (device == AUDIO_DEVICE_NONE) {
- device = outputDesc->device();
+ device = outputDesc->devices().types();
}
ALOGVV("setStreamMute() stream %d, mute %d, mMuteCount %d device %04x",
@@ -5798,9 +5881,9 @@
return false;
}
-bool AudioPolicyManager::isStrategyActiveOnSameModule(const sp<AudioOutputDescriptor>& outputDesc,
- routing_strategy strategy, uint32_t inPastMs,
- nsecs_t sysTime) const
+bool AudioPolicyManager::isStrategyActiveOnSameModule(const sp<SwAudioOutputDescriptor>& outputDesc,
+ routing_strategy strategy, uint32_t inPastMs,
+ nsecs_t sysTime) const
{
for (size_t i = 0; i < mOutputs.size(); i++) {
sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
@@ -5859,6 +5942,8 @@
releaseAudioPatch(patchDesc->mHandle, patchDesc->mUid);
}
}
+
+ mHwModules.cleanUpForDevice(deviceDesc);
}
void AudioPolicyManager::modifySurroundFormats(
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
index 9eb1dcf..de6d489 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -97,12 +97,14 @@
virtual status_t setDeviceConnectionState(audio_devices_t device,
audio_policy_dev_state_t state,
const char *device_address,
- const char *device_name);
+ const char *device_name,
+ audio_format_t encodedFormat);
virtual audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
const char *device_address);
virtual status_t handleDeviceConfigChange(audio_devices_t device,
const char *device_address,
- const char *device_name);
+ const char *device_name,
+ audio_format_t encodedFormat);
virtual void setPhoneState(audio_mode_t state);
virtual void setForceUse(audio_policy_force_use_t usage,
audio_policy_forced_cfg_t config);
@@ -239,6 +241,9 @@
bool reported);
virtual status_t setSurroundFormatEnabled(audio_format_t audioFormat, bool enabled);
+ virtual status_t getHwOffloadEncodingFormatsSupportedForA2DP(
+ std::vector<audio_format_t> *formats);
+
// return the strategy corresponding to a given stream type
routing_strategy getStrategy(audio_stream_type_t stream) const;
@@ -313,36 +318,40 @@
// where conditions are changing (setDeviceConnectionState(), setPhoneState()...) AND
// before updateDevicesAndOutputs() is called.
virtual audio_devices_t getDeviceForStrategy(routing_strategy strategy,
- bool fromCache);
+ bool fromCache)
+ {
+ return getDevicesForStrategy(strategy, fromCache).types();
+ }
+
+ DeviceVector getDevicesForStrategy(routing_strategy strategy, bool fromCache);
bool isStrategyActive(const sp<AudioOutputDescriptor>& outputDesc, routing_strategy strategy,
uint32_t inPastMs = 0, nsecs_t sysTime = 0) const;
- bool isStrategyActiveOnSameModule(const sp<AudioOutputDescriptor>& outputDesc,
- routing_strategy strategy, uint32_t inPastMs = 0,
- nsecs_t sysTime = 0) const;
+ bool isStrategyActiveOnSameModule(const sp<SwAudioOutputDescriptor>& outputDesc,
+ routing_strategy strategy, uint32_t inPastMs = 0,
+ nsecs_t sysTime = 0) const;
// change the route of the specified output. Returns the number of ms we have slept to
// allow new routing to take effect in certain cases.
- virtual uint32_t setOutputDevice(const sp<AudioOutputDescriptor>& outputDesc,
- audio_devices_t device,
- bool force = false,
- int delayMs = 0,
- audio_patch_handle_t *patchHandle = NULL,
- const char *address = nullptr,
- bool requiresMuteCheck = true);
+ uint32_t setOutputDevices(const sp<SwAudioOutputDescriptor>& outputDesc,
+ const DeviceVector &device,
+ bool force = false,
+ int delayMs = 0,
+ audio_patch_handle_t *patchHandle = NULL,
+ bool requiresMuteCheck = true);
status_t resetOutputDevice(const sp<AudioOutputDescriptor>& outputDesc,
int delayMs = 0,
audio_patch_handle_t *patchHandle = NULL);
status_t setInputDevice(audio_io_handle_t input,
- audio_devices_t device,
+ const sp<DeviceDescriptor> &device,
bool force = false,
audio_patch_handle_t *patchHandle = NULL);
status_t resetInputDevice(audio_io_handle_t input,
audio_patch_handle_t *patchHandle = NULL);
// select input device corresponding to requested audio source
- virtual audio_devices_t getDeviceForInputSource(audio_source_t inputSource);
+ sp<DeviceDescriptor> getDeviceForAttributes(const audio_attributes_t &attributes);
// compute the actual volume for a given stream according to the requested index and a particular
// device
@@ -391,15 +400,13 @@
// when a device is disconnected, checks if an output is not used any more and
// returns its handle if any.
// transfers the audio tracks and effects from one output thread to another accordingly.
- status_t checkOutputsForDevice(const sp<DeviceDescriptor>& devDesc,
+ status_t checkOutputsForDevice(const sp<DeviceDescriptor>& device,
audio_policy_dev_state_t state,
- SortedVector<audio_io_handle_t>& outputs,
- const String8& address);
+ SortedVector<audio_io_handle_t>& outputs);
- status_t checkInputsForDevice(const sp<DeviceDescriptor>& devDesc,
+ status_t checkInputsForDevice(const sp<DeviceDescriptor>& device,
audio_policy_dev_state_t state,
- SortedVector<audio_io_handle_t>& inputs,
- const String8& address);
+ SortedVector<audio_io_handle_t>& inputs);
// close an output and its companion duplicating output.
void closeOutput(audio_io_handle_t output);
@@ -437,8 +444,8 @@
// must be called every time a condition that affects the device choice for a given output is
// changed: connected device, phone state, force use, output start, output stop..
// see getDeviceForStrategy() for the use of fromCache parameter
- audio_devices_t getNewOutputDevice(const sp<AudioOutputDescriptor>& outputDesc,
- bool fromCache);
+ DeviceVector getNewOutputDevices(const sp<SwAudioOutputDescriptor>& outputDesc,
+ bool fromCache);
// updates cache of device used by all strategies (mDeviceForStrategy[])
// must be called every time a condition that affects the device choice for a given strategy is
@@ -448,7 +455,7 @@
void updateDevicesAndOutputs();
// selects the most appropriate device on input for current state
- audio_devices_t getNewInputDevice(const sp<AudioInputDescriptor>& inputDesc);
+ sp<DeviceDescriptor> getNewInputDevice(const sp<AudioInputDescriptor>& inputDesc);
virtual uint32_t getMaxEffectsCpuLoad()
{
@@ -460,16 +467,16 @@
return mEffects.getMaxEffectsMemory();
}
- SortedVector<audio_io_handle_t> getOutputsForDevice(audio_devices_t device,
- const SwAudioOutputCollection& openOutputs);
+ SortedVector<audio_io_handle_t> getOutputsForDevices(
+ const DeviceVector &devices, const SwAudioOutputCollection& openOutputs);
// mute/unmute strategies using an incompatible device combination
// if muting, wait for the audio in pcm buffer to be drained before proceeding
// if unmuting, unmute only after the specified delay
// Returns the number of ms waited
virtual uint32_t checkDeviceMuteStrategies(const sp<AudioOutputDescriptor>& outputDesc,
- audio_devices_t prevDevice,
- uint32_t delayMs);
+ audio_devices_t prevDeviceType,
+ uint32_t delayMs);
audio_io_handle_t selectOutput(const SortedVector<audio_io_handle_t>& outputs,
audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
@@ -477,13 +484,22 @@
audio_channel_mask_t channelMask = AUDIO_CHANNEL_NONE,
uint32_t samplingRate = 0);
// samplingRate, format, channelMask are in/out and so may be modified
- sp<IOProfile> getInputProfile(audio_devices_t device,
- const String8& address,
+ sp<IOProfile> getInputProfile(const sp<DeviceDescriptor> & device,
uint32_t& samplingRate,
audio_format_t& format,
audio_channel_mask_t& channelMask,
audio_input_flags_t flags);
- sp<IOProfile> getProfileForOutput(audio_devices_t device,
+ /**
+ * @brief getProfileForOutput
+ * @param devices vector of descriptors, may be empty if ignoring the device is required
+ * @param samplingRate
+ * @param format
+ * @param channelMask
+ * @param flags
+ * @param directOnly
+ * @return IOProfile to be used if found, nullptr otherwise
+ */
+ sp<IOProfile> getProfileForOutput(const DeviceVector &devices,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
@@ -501,19 +517,26 @@
return mAudioPatches.removeAudioPatch(handle);
}
- audio_devices_t availablePrimaryOutputDevices() const
+ bool isPrimaryModule(const sp<HwModule> &module) const
{
- if (!hasPrimaryOutput()) {
- return AUDIO_DEVICE_NONE;
+ if (module == 0 || !hasPrimaryOutput()) {
+ return false;
}
- return mPrimaryOutput->supportedDevices() & mAvailableOutputDevices.types();
+ return module->getHandle() == mPrimaryOutput->getModuleHandle();
}
- audio_devices_t availablePrimaryInputDevices() const
+ DeviceVector availablePrimaryOutputDevices() const
{
if (!hasPrimaryOutput()) {
- return AUDIO_DEVICE_NONE;
+ return DeviceVector();
}
- return mAvailableInputDevices.getDeviceTypesFromHwModule(
+ return mAvailableOutputDevices.filter(mPrimaryOutput->supportedDevices());
+ }
+ DeviceVector availablePrimaryModuleInputDevices() const
+ {
+ if (!hasPrimaryOutput()) {
+ return DeviceVector();
+ }
+ return mAvailableInputDevices.getDevicesFromHwModule(
mPrimaryOutput->getModuleHandle());
}
/**
@@ -530,8 +553,9 @@
return (devices.size() > 0) ? devices.itemAt(0)->address() : String8("");
}
- uint32_t updateCallRouting(audio_devices_t rxDevice, uint32_t delayMs = 0);
- sp<AudioPatch> createTelephonyPatch(bool isRx, audio_devices_t device, uint32_t delayMs);
+ uint32_t updateCallRouting(const DeviceVector &rxDevices, uint32_t delayMs = 0);
+ sp<AudioPatch> createTelephonyPatch(bool isRx, const sp<DeviceDescriptor> &device,
+ uint32_t delayMs);
sp<DeviceDescriptor> findDevice(
const DeviceVector& devices, audio_devices_t device) const;
audio_devices_t getModuleDeviceTypes(
@@ -581,7 +605,16 @@
DeviceVector mAvailableInputDevices; // all available input devices
bool mLimitRingtoneVolume; // limit ringtone volume to music volume if headset connected
- audio_devices_t mDeviceForStrategy[NUM_STRATEGIES];
+
+ /**
+ * @brief mDevicesForStrategy vector of devices that are assigned for a given strategy.
+ * Note: in case of removal of device (@see setDeviceConnectionState), the device descriptor
+ * will be removed from the @see mAvailableOutputDevices or @see mAvailableInputDevices
+ * but the devices for strategies will be reevaluated within the
+ * @see setDeviceConnectionState function.
+ */
+ DeviceVector mDevicesForStrategy[NUM_STRATEGIES];
+
float mLastVoiceVolume; // last voice volume value sent to audio HAL
bool mA2dpSuspended; // true if A2DP output is suspended
@@ -637,13 +670,14 @@
// Support for Multi-Stream Decoder (MSD) module
sp<DeviceDescriptor> getMsdAudioInDevice() const;
+ DeviceVector getMsdAudioOutDevices() const;
const AudioPatchCollection getMsdPatches() const;
- status_t getBestMsdAudioProfileFor(audio_devices_t outputDevice,
+ status_t getBestMsdAudioProfileFor(const sp<DeviceDescriptor> &outputDevice,
bool hwAvSync,
audio_port_config *sourceConfig,
audio_port_config *sinkConfig) const;
- PatchBuilder buildMsdPatch(audio_devices_t outputDevice) const;
- status_t setMsdPatch(audio_devices_t outputDevice = AUDIO_DEVICE_NONE);
+ PatchBuilder buildMsdPatch(const sp<DeviceDescriptor> &outputDevice) const;
+ status_t setMsdPatch(const sp<DeviceDescriptor> &outputDevice = nullptr);
// If any, resolve any "dynamic" fields of an Audio Profiles collection
void updateAudioProfiles(const sp<DeviceDescriptor>& devDesc, audio_io_handle_t ioHandle,
@@ -654,22 +688,12 @@
// It can give a chance to HAL implementer to retrieve dynamic capabilities associated
// to this device for example.
// TODO avoid opening stream to retrieve capabilities of a profile.
- void broadcastDeviceConnectionState(audio_devices_t device,
- audio_policy_dev_state_t state,
- const String8 &device_address);
+ void broadcastDeviceConnectionState(const sp<DeviceDescriptor> &device,
+ audio_policy_dev_state_t state);
// updates device caching and output for streams that can influence the
// routing of notifications
void handleNotificationRoutingForStream(audio_stream_type_t stream);
- // find the outputs on a given output descriptor that have the given address.
- // to be called on an AudioOutputDescriptor whose supported devices (as defined
- // in mProfile->mSupportedDevices) matches the device whose address is to be matched.
- // see deviceDistinguishesOnAddress(audio_devices_t) for whether the device type is one
- // where addresses are used to distinguish between one connected device and another.
- void findIoHandlesByAddress(const sp<SwAudioOutputDescriptor>& desc /*in*/,
- const audio_devices_t device /*in*/,
- const String8& address /*in*/,
- SortedVector<audio_io_handle_t>& outputs /*out*/);
uint32_t curAudioPortGeneration() const { return mAudioPortGeneration; }
// internal method, get audio_attributes_t from either a source audio_attributes_t
// or audio_stream_type_t, respectively.
@@ -687,15 +711,14 @@
audio_output_flags_t *flags,
audio_port_handle_t *selectedDeviceId);
// internal method to return the output handle for the given device and format
- audio_io_handle_t getOutputForDevice(
- audio_devices_t device,
+ audio_io_handle_t getOutputForDevices(
+ const DeviceVector &devices,
audio_session_t session,
audio_stream_type_t stream,
const audio_config_t *config,
audio_output_flags_t *flags);
// internal method to return the input handle for the given device and format
- audio_io_handle_t getInputForDevice(audio_devices_t device,
- String8 address,
+ audio_io_handle_t getInputForDevice(const sp<DeviceDescriptor> &device,
audio_session_t session,
audio_source_t inputSource,
const audio_config_base_t *config,
@@ -713,14 +736,15 @@
// select input device corresponding to requested audio source and return associated policy
// mix if any. Calls getDeviceForInputSource().
- audio_devices_t getDeviceAndMixForInputSource(audio_source_t inputSource,
- AudioMix **policyMix = NULL);
+ sp<DeviceDescriptor> getDeviceAndMixForAttributes(const audio_attributes_t &attributes,
+ AudioMix **policyMix = NULL);
// Called by setDeviceConnectionState().
- status_t setDeviceConnectionStateInt(audio_devices_t device,
- audio_policy_dev_state_t state,
- const char *device_address,
- const char *device_name);
+ status_t setDeviceConnectionStateInt(audio_devices_t deviceType,
+ audio_policy_dev_state_t state,
+ const char *device_address,
+ const char *device_name,
+ audio_format_t encodedFormat);
void updateMono(audio_io_handle_t output) {
AudioParameter param;
param.addInt(String8(AudioParameter::keyMonoOutput), (int)mMasterMono);
diff --git a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
index 2c904d9..49c541c 100644
--- a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
@@ -32,7 +32,8 @@
status_t AudioPolicyService::setDeviceConnectionState(audio_devices_t device,
audio_policy_dev_state_t state,
const char *device_address,
- const char *device_name)
+ const char *device_name,
+ audio_format_t encodedFormat)
{
if (mAudioPolicyManager == NULL) {
return NO_INIT;
@@ -49,7 +50,7 @@
Mutex::Autolock _l(mLock);
AutoCallerClear acc;
return mAudioPolicyManager->setDeviceConnectionState(device, state,
- device_address, device_name);
+ device_address, device_name, encodedFormat);
}
audio_policy_dev_state_t AudioPolicyService::getDeviceConnectionState(
@@ -66,7 +67,8 @@
status_t AudioPolicyService::handleDeviceConfigChange(audio_devices_t device,
const char *device_address,
- const char *device_name)
+ const char *device_name,
+ audio_format_t encodedFormat)
{
if (mAudioPolicyManager == NULL) {
return NO_INIT;
@@ -79,7 +81,7 @@
Mutex::Autolock _l(mLock);
AutoCallerClear acc;
return mAudioPolicyManager->handleDeviceConfigChange(device, device_address,
- device_name);
+ device_name, encodedFormat);
}
status_t AudioPolicyService::setPhoneState(audio_mode_t state)
@@ -1138,6 +1140,17 @@
surroundFormatsEnabled, reported);
}
+status_t AudioPolicyService::getHwOffloadEncodingFormatsSupportedForA2DP(
+ std::vector<audio_format_t> *formats)
+{
+ if (mAudioPolicyManager == NULL) {
+ return NO_INIT;
+ }
+ Mutex::Autolock _l(mLock);
+ AutoCallerClear acc;
+ return mAudioPolicyManager->getHwOffloadEncodingFormatsSupportedForA2DP(formats);
+}
+
status_t AudioPolicyService::setSurroundFormatEnabled(audio_format_t audioFormat, bool enabled)
{
if (mAudioPolicyManager == NULL) {
diff --git a/services/audiopolicy/service/AudioPolicyService.h b/services/audiopolicy/service/AudioPolicyService.h
index 959e757..c073b7c 100644
--- a/services/audiopolicy/service/AudioPolicyService.h
+++ b/services/audiopolicy/service/AudioPolicyService.h
@@ -61,13 +61,15 @@
virtual status_t setDeviceConnectionState(audio_devices_t device,
audio_policy_dev_state_t state,
const char *device_address,
- const char *device_name);
+ const char *device_name,
+ audio_format_t encodedFormat);
virtual audio_policy_dev_state_t getDeviceConnectionState(
audio_devices_t device,
const char *device_address);
virtual status_t handleDeviceConfigChange(audio_devices_t device,
const char *device_address,
- const char *device_name);
+ const char *device_name,
+ audio_format_t encodedFormat);
virtual status_t setPhoneState(audio_mode_t state);
virtual status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config);
virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage);
@@ -218,6 +220,8 @@
audio_format_t *surroundFormats,
bool *surroundFormatsEnabled,
bool reported);
+ virtual status_t getHwOffloadEncodingFormatsSupportedForA2DP(
+ std::vector<audio_format_t> *formats);
virtual status_t setSurroundFormatEnabled(audio_format_t audioFormat, bool enabled);
virtual status_t setAssistantUid(uid_t uid);
diff --git a/services/audiopolicy/tests/Android.mk b/services/audiopolicy/tests/Android.mk
index 2ccb542..e4fba0f 100644
--- a/services/audiopolicy/tests/Android.mk
+++ b/services/audiopolicy/tests/Android.mk
@@ -29,6 +29,8 @@
LOCAL_MULTILIB := $(AUDIOSERVER_MULTILIB)
+LOCAL_COMPATIBILITY_SUITE := device-tests
+
include $(BUILD_NATIVE_TEST)
# system/audio.h utilities test
@@ -55,4 +57,6 @@
LOCAL_MULTILIB := $(AUDIOSERVER_MULTILIB)
+LOCAL_COMPATIBILITY_SUITE := device-tests
+
include $(BUILD_NATIVE_TEST)
diff --git a/services/audiopolicy/tests/audiopolicymanager_tests.cpp b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
index 24326bb..e9f4657 100644
--- a/services/audiopolicy/tests/audiopolicymanager_tests.cpp
+++ b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
@@ -117,9 +117,14 @@
explicit PatchCountCheck(AudioPolicyManagerTestClient *client)
: mClient{client},
mInitialCount{mClient->getActivePatchesCount()} {}
- void assertDelta(int delta) const {
- ASSERT_EQ(mInitialCount + delta, mClient->getActivePatchesCount()); }
- void assertNoChange() const { assertDelta(0); }
+ int deltaFromSnapshot() const {
+ size_t currentCount = mClient->getActivePatchesCount();
+ if (mInitialCount <= currentCount) {
+ return currentCount - mInitialCount;
+ } else {
+ return -(static_cast<int>(mInitialCount - currentCount));
+ }
+ }
private:
const AudioPolicyManagerTestClient *mClient;
const size_t mInitialCount;
@@ -139,7 +144,7 @@
int sampleRate,
audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
audio_port_handle_t *portId = nullptr);
- PatchCountCheck snapPatchCount() { return PatchCountCheck(mClient.get()); }
+ PatchCountCheck snapshotPatchCount() { return PatchCountCheck(mClient.get()); }
std::unique_ptr<AudioPolicyManagerTestClient> mClient;
std::unique_ptr<AudioPolicyTestManager> mManager;
@@ -225,7 +230,7 @@
TEST_F(AudioPolicyManagerTest, CreateAudioPatchFailure) {
audio_patch patch{};
audio_patch_handle_t handle = AUDIO_PATCH_HANDLE_NONE;
- const PatchCountCheck patchCount = snapPatchCount();
+ const PatchCountCheck patchCount = snapshotPatchCount();
ASSERT_EQ(BAD_VALUE, mManager->createAudioPatch(nullptr, &handle, 0));
ASSERT_EQ(BAD_VALUE, mManager->createAudioPatch(&patch, nullptr, 0));
ASSERT_EQ(BAD_VALUE, mManager->createAudioPatch(&patch, &handle, 0));
@@ -252,20 +257,20 @@
ASSERT_EQ(INVALID_OPERATION, mManager->createAudioPatch(&patch, &handle, 0));
// Verify that the handle is left unchanged.
ASSERT_EQ(AUDIO_PATCH_HANDLE_NONE, handle);
- patchCount.assertNoChange();
+ ASSERT_EQ(0, patchCount.deltaFromSnapshot());
}
TEST_F(AudioPolicyManagerTest, CreateAudioPatchFromMix) {
audio_patch_handle_t handle = AUDIO_PATCH_HANDLE_NONE;
uid_t uid = 42;
- const PatchCountCheck patchCount = snapPatchCount();
+ const PatchCountCheck patchCount = snapshotPatchCount();
ASSERT_FALSE(mManager->getConfig().getAvailableInputDevices().isEmpty());
PatchBuilder patchBuilder;
patchBuilder.addSource(mManager->getConfig().getAvailableInputDevices()[0]).
addSink(mManager->getConfig().getDefaultOutputDevice());
ASSERT_EQ(NO_ERROR, mManager->createAudioPatch(patchBuilder.patch(), &handle, uid));
ASSERT_NE(AUDIO_PATCH_HANDLE_NONE, handle);
- patchCount.assertDelta(1);
+ ASSERT_EQ(1, patchCount.deltaFromSnapshot());
}
// TODO: Add patch creation tests that involve already existing patch
@@ -350,84 +355,82 @@
}
TEST_F(AudioPolicyManagerTestMsd, PatchCreationOnSetForceUse) {
- const PatchCountCheck patchCount = snapPatchCount();
+ const PatchCountCheck patchCount = snapshotPatchCount();
mManager->setForceUse(AUDIO_POLICY_FORCE_FOR_ENCODED_SURROUND,
AUDIO_POLICY_FORCE_ENCODED_SURROUND_ALWAYS);
- patchCount.assertDelta(1);
+ ASSERT_EQ(1, patchCount.deltaFromSnapshot());
}
TEST_F(AudioPolicyManagerTestMsd, GetOutputForAttrEncodedRoutesToMsd) {
- const PatchCountCheck patchCount = snapPatchCount();
+ const PatchCountCheck patchCount = snapshotPatchCount();
audio_port_handle_t selectedDeviceId;
getOutputForAttr(&selectedDeviceId,
AUDIO_FORMAT_AC3, AUDIO_CHANNEL_OUT_5POINT1, 48000, AUDIO_OUTPUT_FLAG_DIRECT);
ASSERT_EQ(selectedDeviceId, mMsdOutputDevice->getId());
- patchCount.assertDelta(1);
+ ASSERT_EQ(1, patchCount.deltaFromSnapshot());
}
TEST_F(AudioPolicyManagerTestMsd, GetOutputForAttrPcmRoutesToMsd) {
- const PatchCountCheck patchCount = snapPatchCount();
+ const PatchCountCheck patchCount = snapshotPatchCount();
audio_port_handle_t selectedDeviceId;
getOutputForAttr(&selectedDeviceId,
AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO, 48000);
ASSERT_EQ(selectedDeviceId, mMsdOutputDevice->getId());
- patchCount.assertDelta(1);
+ ASSERT_EQ(1, patchCount.deltaFromSnapshot());
}
TEST_F(AudioPolicyManagerTestMsd, GetOutputForAttrEncodedPlusPcmRoutesToMsd) {
- const PatchCountCheck patchCount = snapPatchCount();
+ const PatchCountCheck patchCount = snapshotPatchCount();
audio_port_handle_t selectedDeviceId;
getOutputForAttr(&selectedDeviceId,
AUDIO_FORMAT_AC3, AUDIO_CHANNEL_OUT_5POINT1, 48000, AUDIO_OUTPUT_FLAG_DIRECT);
ASSERT_EQ(selectedDeviceId, mMsdOutputDevice->getId());
- patchCount.assertDelta(1);
+ ASSERT_EQ(1, patchCount.deltaFromSnapshot());
getOutputForAttr(&selectedDeviceId,
AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO, 48000);
ASSERT_EQ(selectedDeviceId, mMsdOutputDevice->getId());
- patchCount.assertDelta(1);
+ ASSERT_EQ(1, patchCount.deltaFromSnapshot());
}
TEST_F(AudioPolicyManagerTestMsd, GetOutputForAttrUnsupportedFormatBypassesMsd) {
- const PatchCountCheck patchCount = snapPatchCount();
+ const PatchCountCheck patchCount = snapshotPatchCount();
audio_port_handle_t selectedDeviceId;
getOutputForAttr(&selectedDeviceId,
AUDIO_FORMAT_DTS, AUDIO_CHANNEL_OUT_5POINT1, 48000, AUDIO_OUTPUT_FLAG_DIRECT);
ASSERT_NE(selectedDeviceId, mMsdOutputDevice->getId());
- patchCount.assertNoChange();
+ ASSERT_EQ(0, patchCount.deltaFromSnapshot());
}
TEST_F(AudioPolicyManagerTestMsd, GetOutputForAttrFormatSwitching) {
// Switch between formats that are supported and not supported by MSD.
{
- const PatchCountCheck patchCount = snapPatchCount();
+ const PatchCountCheck patchCount = snapshotPatchCount();
audio_port_handle_t selectedDeviceId, portId;
getOutputForAttr(&selectedDeviceId,
AUDIO_FORMAT_AC3, AUDIO_CHANNEL_OUT_5POINT1, 48000, AUDIO_OUTPUT_FLAG_DIRECT,
&portId);
ASSERT_EQ(selectedDeviceId, mMsdOutputDevice->getId());
- patchCount.assertDelta(1);
+ ASSERT_EQ(1, patchCount.deltaFromSnapshot());
mManager->releaseOutput(portId);
- patchCount.assertDelta(1); // compared to the state at the block entry
- // TODO: make PatchCountCheck asserts more obvious. It's easy to
- // miss the fact that it is immutable.
+ ASSERT_EQ(1, patchCount.deltaFromSnapshot());
}
{
- const PatchCountCheck patchCount = snapPatchCount();
+ const PatchCountCheck patchCount = snapshotPatchCount();
audio_port_handle_t selectedDeviceId, portId;
getOutputForAttr(&selectedDeviceId,
AUDIO_FORMAT_DTS, AUDIO_CHANNEL_OUT_5POINT1, 48000, AUDIO_OUTPUT_FLAG_DIRECT,
&portId);
ASSERT_NE(selectedDeviceId, mMsdOutputDevice->getId());
- patchCount.assertDelta(-1);
+ ASSERT_EQ(-1, patchCount.deltaFromSnapshot());
mManager->releaseOutput(portId);
- patchCount.assertNoChange();
+ ASSERT_EQ(0, patchCount.deltaFromSnapshot());
}
{
- const PatchCountCheck patchCount = snapPatchCount();
+ const PatchCountCheck patchCount = snapshotPatchCount();
audio_port_handle_t selectedDeviceId;
getOutputForAttr(&selectedDeviceId,
AUDIO_FORMAT_AC3, AUDIO_CHANNEL_OUT_5POINT1, 48000, AUDIO_OUTPUT_FLAG_DIRECT);
ASSERT_EQ(selectedDeviceId, mMsdOutputDevice->getId());
- patchCount.assertNoChange();
+ ASSERT_EQ(0, patchCount.deltaFromSnapshot());
}
}
diff --git a/services/mediacodec/Android.mk b/services/mediacodec/Android.mk
index 3b6dc80..6a71d7d 100644
--- a/services/mediacodec/Android.mk
+++ b/services/mediacodec/Android.mk
@@ -69,9 +69,12 @@
include $(CLEAR_VARS)
# seccomp is not required for coverage build.
ifneq ($(NATIVE_COVERAGE),true)
-LOCAL_REQUIRED_MODULES_arm := crash_dump.policy mediacodec.policy
-LOCAL_REQUIRED_MODULES_x86 := crash_dump.policy mediacodec.policy
+LOCAL_REQUIRED_MODULES_arm := crash_dump.policy mediaswcodec.policy
+LOCAL_REQUIRED_MODULES_arm64 := crash_dump.policy mediaswcodec.policy
+LOCAL_REQUIRED_MODULES_x86 := crash_dump.policy mediaswcodec.policy
+LOCAL_REQUIRED_MODULES_x86_64 := crash_dump.policy mediaswcodec.policy
endif
+
LOCAL_SRC_FILES := \
main_swcodecservice.cpp \
MediaCodecUpdateService.cpp \
@@ -107,8 +110,12 @@
LOCAL_MODULE := mediaswcodec
LOCAL_INIT_RC := mediaswcodec.rc
-LOCAL_32_BIT_ONLY := true
LOCAL_SANITIZE := scudo
+ifeq ($(TARGET_ARCH), $(filter $(TARGET_ARCH), x86_64 arm64))
+ LOCAL_MULTILIB := both
+ LOCAL_MODULE_STEM_32 := $(LOCAL_MODULE)32
+ LOCAL_MODULE_STEM_64 := $(LOCAL_MODULE)
+endif
sanitizer_runtime_libraries :=
llndk_libraries :=
@@ -137,4 +144,16 @@
include $(BUILD_PREBUILT)
endif
+####################################################################
+
+# sw service seccomp policy
+ifeq ($(TARGET_ARCH), $(filter $(TARGET_ARCH), x86 x86_64 arm arm64))
+include $(CLEAR_VARS)
+LOCAL_MODULE := mediaswcodec.policy
+LOCAL_MODULE_CLASS := ETC
+LOCAL_MODULE_PATH := $(TARGET_OUT)/etc/seccomp_policy
+LOCAL_SRC_FILES := seccomp_policy/mediaswcodec-$(TARGET_ARCH).policy
+include $(BUILD_PREBUILT)
+endif
+
include $(call all-makefiles-under, $(LOCAL_PATH))
diff --git a/services/mediacodec/main_swcodecservice.cpp b/services/mediacodec/main_swcodecservice.cpp
index 1168825..05b5695 100644
--- a/services/mediacodec/main_swcodecservice.cpp
+++ b/services/mediacodec/main_swcodecservice.cpp
@@ -26,12 +26,10 @@
using namespace android;
-// TODO: replace policy with software codec-only policies
-// Must match location in Android.mk.
static const char kSystemSeccompPolicyPath[] =
- "/system/etc/seccomp_policy/mediacodec.policy";
+ "/system/etc/seccomp_policy/mediaswcodec.policy";
static const char kVendorSeccompPolicyPath[] =
- "/vendor/etc/seccomp_policy/mediacodec.policy";
+ "/vendor/etc/seccomp_policy/mediaswcodec.policy";
// Disable Scudo's mismatch allocation check, as it is being triggered
// by some third party code.
@@ -47,8 +45,11 @@
::android::hardware::configureRpcThreadpool(64, false);
- // codec libs are currently 32-bit only
+#ifdef __LP64__
+ loadFromApex("/apex/com.android.media.swcodec/lib64");
+#else
loadFromApex("/apex/com.android.media.swcodec/lib");
+#endif
::android::hardware::joinRpcThreadpool();
}
diff --git a/services/mediacodec/mediaswcodec.rc b/services/mediacodec/mediaswcodec.rc
index dfe3381..3549666 100644
--- a/services/mediacodec/mediaswcodec.rc
+++ b/services/mediacodec/mediaswcodec.rc
@@ -2,5 +2,6 @@
class main
user mediacodec
group camera drmrpc mediadrm
+ updatable
ioprio rt 4
writepid /dev/cpuset/foreground/tasks
diff --git a/services/mediacodec/registrant/Android.bp b/services/mediacodec/registrant/Android.bp
index 8c40ad1..80d3630 100644
--- a/services/mediacodec/registrant/Android.bp
+++ b/services/mediacodec/registrant/Android.bp
@@ -38,6 +38,7 @@
"libcodec2_soft_mp3dec",
"libcodec2_soft_vorbisdec",
"libcodec2_soft_opusdec",
+ "libcodec2_soft_opusenc",
"libcodec2_soft_vp8dec",
"libcodec2_soft_vp9dec",
"libcodec2_soft_av1dec",
@@ -49,7 +50,5 @@
"libcodec2_soft_gsmdec",
"libcodec2_soft_xaacdec",
],
-
- compile_multilib: "32",
}
diff --git a/services/mediacodec/seccomp_policy/mediacodec-x86.policy b/services/mediacodec/seccomp_policy/mediacodec-x86.policy
index 966e214..6d88c84 100644
--- a/services/mediacodec/seccomp_policy/mediacodec-x86.policy
+++ b/services/mediacodec/seccomp_policy/mediacodec-x86.policy
@@ -18,15 +18,19 @@
openat: 1
open: 1
getuid32: 1
+getuid: 1
+getrlimit: 1
writev: 1
ioctl: 1
close: 1
mmap2: 1
+mmap: 1
fstat64: 1
stat64: 1
statfs64: 1
madvise: 1
fstatat64: 1
+newfstatat: 1
futex: 1
munmap: 1
faccessat: 1
diff --git a/services/mediacodec/seccomp_policy/mediaswcodec-arm.policy b/services/mediacodec/seccomp_policy/mediaswcodec-arm.policy
new file mode 100644
index 0000000..588141a
--- /dev/null
+++ b/services/mediacodec/seccomp_policy/mediaswcodec-arm.policy
@@ -0,0 +1,60 @@
+# 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.
+
+futex: 1
+# ioctl calls are filtered via the selinux policy.
+ioctl: 1
+sched_yield: 1
+close: 1
+dup: 1
+ppoll: 1
+mprotect: arg2 in ~PROT_EXEC || arg2 in ~PROT_WRITE
+mmap2: arg2 in ~PROT_EXEC || arg2 in ~PROT_WRITE
+
+# mremap: Ensure |flags| are (MREMAP_MAYMOVE | MREMAP_FIXED) TODO: Once minijail
+# parser support for '<' is in this needs to be modified to also prevent
+# |old_address| and |new_address| from touching the exception vector page, which
+# on ARM is statically loaded at 0xffff 0000. See
+# http://infocenter.arm.com/help/index.jsp?topic=/com.arm.doc.ddi0211h/Babfeega.html
+# for more details.
+mremap: arg3 == 3
+munmap: 1
+prctl: 1
+getuid32: 1
+writev: 1
+sigaltstack: 1
+clone: 1
+exit: 1
+lseek: 1
+rt_sigprocmask: 1
+openat: 1
+fstat64: 1
+write: 1
+nanosleep: 1
+setpriority: 1
+set_tid_address: 1
+getdents64: 1
+readlinkat: 1
+read: 1
+pread64: 1
+fstatfs64: 1
+gettimeofday: 1
+faccessat: 1
+_llseek: 1
+fstatat64: 1
+ugetrlimit: 1
+exit_group: 1
+restart_syscall: 1
+rt_sigreturn: 1
+getrandom: 1
diff --git a/services/mediacodec/seccomp_policy/mediaswcodec-arm64.policy b/services/mediacodec/seccomp_policy/mediaswcodec-arm64.policy
new file mode 100644
index 0000000..1bee1b5
--- /dev/null
+++ b/services/mediacodec/seccomp_policy/mediaswcodec-arm64.policy
@@ -0,0 +1,61 @@
+# Copyright (C) 2019 The Android Open Source Project
+#
+# Licensed under the Apache License, Version 2.0 (the "License");
+# you may not use this file except in compliance with the License.
+# You may obtain a copy of the License at
+#
+# http://www.apache.org/licenses/LICENSE-2.0
+#
+# Unless required by applicable law or agreed to in writing, software
+# distributed under the License is distributed on an "AS IS" BASIS,
+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+# See the License for the specific language governing permissions and
+# limitations under the License.
+
+futex: 1
+# ioctl calls are filtered via the selinux policy.
+ioctl: 1
+sched_yield: 1
+close: 1
+dup: 1
+ppoll: 1
+mprotect: arg2 in ~PROT_EXEC || arg2 in ~PROT_WRITE
+mmap: arg2 in ~PROT_EXEC || arg2 in ~PROT_WRITE
+getuid: 1
+getrlimit: 1
+fstat: 1
+newfstatat: 1
+fstatfs: 1
+
+# mremap: Ensure |flags| are (MREMAP_MAYMOVE | MREMAP_FIXED) TODO: Once minijail
+# parser support for '<' is in this needs to be modified to also prevent
+# |old_address| and |new_address| from touching the exception vector page, which
+# on ARM is statically loaded at 0xffff 0000. See
+# http://infocenter.arm.com/help/index.jsp?topic=/com.arm.doc.ddi0211h/Babfeega.html
+# for more details.
+mremap: arg3 == 3
+munmap: 1
+prctl: 1
+writev: 1
+sigaltstack: 1
+clone: 1
+exit: 1
+lseek: 1
+rt_sigprocmask: 1
+openat: 1
+write: 1
+nanosleep: 1
+setpriority: 1
+set_tid_address: 1
+getdents64: 1
+readlinkat: 1
+read: 1
+pread64: 1
+gettimeofday: 1
+faccessat: 1
+exit_group: 1
+restart_syscall: 1
+rt_sigreturn: 1
+getrandom: 1
+madvise: 1
+
diff --git a/services/mediacodec/seccomp_policy/mediaswcodec-x86.policy b/services/mediacodec/seccomp_policy/mediaswcodec-x86.policy
new file mode 120000
index 0000000..ab2592a
--- /dev/null
+++ b/services/mediacodec/seccomp_policy/mediaswcodec-x86.policy
@@ -0,0 +1 @@
+mediacodec-x86.policy
\ No newline at end of file
diff --git a/services/mediacodec/seccomp_policy/mediaswcodec-x86_64.policy b/services/mediacodec/seccomp_policy/mediaswcodec-x86_64.policy
new file mode 120000
index 0000000..ab2592a
--- /dev/null
+++ b/services/mediacodec/seccomp_policy/mediaswcodec-x86_64.policy
@@ -0,0 +1 @@
+mediacodec-x86.policy
\ No newline at end of file
diff --git a/services/mediaextractor/mediaextractor.rc b/services/mediaextractor/mediaextractor.rc
index 5fc2941..6b2d0a5 100644
--- a/services/mediaextractor/mediaextractor.rc
+++ b/services/mediaextractor/mediaextractor.rc
@@ -2,5 +2,7 @@
class main
user mediaex
group drmrpc mediadrm
+ # TODO(b/123275379): Remove updatable when http://aosp/878198 has landed
+ updatable
ioprio rt 4
writepid /dev/cpuset/foreground/tasks
diff --git a/services/oboeservice/AAudioServiceEndpoint.h b/services/oboeservice/AAudioServiceEndpoint.h
index 43b0a37..3616fa2 100644
--- a/services/oboeservice/AAudioServiceEndpoint.h
+++ b/services/oboeservice/AAudioServiceEndpoint.h
@@ -121,7 +121,7 @@
mutable std::mutex mLockStreams;
std::vector<android::sp<AAudioServiceStreamBase>> mRegisteredStreams;
- SimpleDoubleBuffer<Timestamp> mAtomicTimestamp;
+ SimpleDoubleBuffer<Timestamp> mAtomicEndpointTimestamp;
android::AudioClient mMmapClient; // set in open, used in open and startStream
diff --git a/services/oboeservice/AAudioServiceEndpointShared.cpp b/services/oboeservice/AAudioServiceEndpointShared.cpp
index 2f1ec7e..0a415fd 100644
--- a/services/oboeservice/AAudioServiceEndpointShared.cpp
+++ b/services/oboeservice/AAudioServiceEndpointShared.cpp
@@ -181,8 +181,8 @@
// Get timestamp that was written by the real-time service thread, eg. mixer.
aaudio_result_t AAudioServiceEndpointShared::getFreeRunningPosition(int64_t *positionFrames,
int64_t *timeNanos) {
- if (mAtomicTimestamp.isValid()) {
- Timestamp timestamp = mAtomicTimestamp.read();
+ if (mAtomicEndpointTimestamp.isValid()) {
+ Timestamp timestamp = mAtomicEndpointTimestamp.read();
*positionFrames = timestamp.getPosition();
*timeNanos = timestamp.getNanoseconds();
return AAUDIO_OK;
diff --git a/services/oboeservice/AAudioServiceStreamBase.cpp b/services/oboeservice/AAudioServiceStreamBase.cpp
index defbb7b..b16b5dc 100644
--- a/services/oboeservice/AAudioServiceStreamBase.cpp
+++ b/services/oboeservice/AAudioServiceStreamBase.cpp
@@ -43,7 +43,7 @@
AAudioServiceStreamBase::AAudioServiceStreamBase(AAudioService &audioService)
: mUpMessageQueue(nullptr)
, mTimestampThread("AATime")
- , mAtomicTimestamp()
+ , mAtomicStreamTimestamp()
, mAudioService(audioService) {
mMmapClient.clientUid = -1;
mMmapClient.clientPid = -1;
@@ -182,7 +182,7 @@
setSuspended(false);
// Start with fresh presentation timestamps.
- mAtomicTimestamp.clear();
+ mAtomicStreamTimestamp.clear();
mClientHandle = AUDIO_PORT_HANDLE_NONE;
result = startDevice();
@@ -291,16 +291,20 @@
}
// implement Runnable, periodically send timestamps to client
+__attribute__((no_sanitize("integer")))
void AAudioServiceStreamBase::run() {
ALOGD("%s() %s entering >>>>>>>>>>>>>> TIMESTAMPS", __func__, getTypeText());
TimestampScheduler timestampScheduler;
timestampScheduler.setBurstPeriod(mFramesPerBurst, getSampleRate());
timestampScheduler.start(AudioClock::getNanoseconds());
int64_t nextTime = timestampScheduler.nextAbsoluteTime();
+ int32_t loopCount = 0;
while(mThreadEnabled.load()) {
+ loopCount++;
if (AudioClock::getNanoseconds() >= nextTime) {
aaudio_result_t result = sendCurrentTimestamp();
if (result != AAUDIO_OK) {
+ ALOGE("%s() timestamp thread got result = %d", __func__, result);
break;
}
nextTime = timestampScheduler.nextAbsoluteTime();
@@ -310,7 +314,8 @@
AudioClock::sleepUntilNanoTime(nextTime);
}
}
- ALOGD("%s() %s exiting <<<<<<<<<<<<<< TIMESTAMPS", __func__, getTypeText());
+ ALOGD("%s() %s exiting after %d loops <<<<<<<<<<<<<< TIMESTAMPS",
+ __func__, getTypeText(), loopCount);
}
void AAudioServiceStreamBase::disconnect() {
diff --git a/services/oboeservice/AAudioServiceStreamBase.h b/services/oboeservice/AAudioServiceStreamBase.h
index 7904b25..ffc768b 100644
--- a/services/oboeservice/AAudioServiceStreamBase.h
+++ b/services/oboeservice/AAudioServiceStreamBase.h
@@ -301,7 +301,7 @@
// TODO rename mClientHandle to mPortHandle to be more consistent with AudioFlinger.
audio_port_handle_t mClientHandle = AUDIO_PORT_HANDLE_NONE;
- SimpleDoubleBuffer<Timestamp> mAtomicTimestamp;
+ SimpleDoubleBuffer<Timestamp> mAtomicStreamTimestamp;
android::AAudioService &mAudioService;
diff --git a/services/oboeservice/AAudioServiceStreamMMAP.cpp b/services/oboeservice/AAudioServiceStreamMMAP.cpp
index 9377945..837b080 100644
--- a/services/oboeservice/AAudioServiceStreamMMAP.cpp
+++ b/services/oboeservice/AAudioServiceStreamMMAP.cpp
@@ -162,7 +162,7 @@
aaudio_result_t result = serviceEndpointMMAP->getFreeRunningPosition(positionFrames, timeNanos);
if (result == AAUDIO_OK) {
Timestamp timestamp(*positionFrames, *timeNanos);
- mAtomicTimestamp.write(timestamp);
+ mAtomicStreamTimestamp.write(timestamp);
*positionFrames = timestamp.getPosition();
*timeNanos = timestamp.getNanoseconds();
} else if (result != AAUDIO_ERROR_UNAVAILABLE) {
@@ -184,8 +184,8 @@
static_cast<AAudioServiceEndpointMMAP *>(endpoint.get());
// TODO Get presentation timestamp from the HAL
- if (mAtomicTimestamp.isValid()) {
- Timestamp timestamp = mAtomicTimestamp.read();
+ if (mAtomicStreamTimestamp.isValid()) {
+ Timestamp timestamp = mAtomicStreamTimestamp.read();
*positionFrames = timestamp.getPosition();
*timeNanos = timestamp.getNanoseconds() + serviceEndpointMMAP->getHardwareTimeOffsetNanos();
return AAUDIO_OK;
diff --git a/services/oboeservice/AAudioServiceStreamShared.cpp b/services/oboeservice/AAudioServiceStreamShared.cpp
index d5450fe..14742dd 100644
--- a/services/oboeservice/AAudioServiceStreamShared.cpp
+++ b/services/oboeservice/AAudioServiceStreamShared.cpp
@@ -238,15 +238,15 @@
}
void AAudioServiceStreamShared::markTransferTime(Timestamp ×tamp) {
- mAtomicTimestamp.write(timestamp);
+ mAtomicStreamTimestamp.write(timestamp);
}
// Get timestamp that was written by mixer or distributor.
aaudio_result_t AAudioServiceStreamShared::getFreeRunningPosition(int64_t *positionFrames,
int64_t *timeNanos) {
// TODO Get presentation timestamp from the HAL
- if (mAtomicTimestamp.isValid()) {
- Timestamp timestamp = mAtomicTimestamp.read();
+ if (mAtomicStreamTimestamp.isValid()) {
+ Timestamp timestamp = mAtomicStreamTimestamp.read();
*positionFrames = timestamp.getPosition();
*timeNanos = timestamp.getNanoseconds();
return AAUDIO_OK;
diff --git a/services/soundtrigger/SoundTriggerHalHidl.cpp b/services/soundtrigger/SoundTriggerHalHidl.cpp
index 1d37a8e..68d54c7 100644
--- a/services/soundtrigger/SoundTriggerHalHidl.cpp
+++ b/services/soundtrigger/SoundTriggerHalHidl.cpp
@@ -168,18 +168,23 @@
int ret;
SoundModelHandle halHandle;
sp<V2_1_ISoundTriggerHw> soundtrigger_2_1 = toService2_1(soundtrigger);
+ sp<V2_2_ISoundTriggerHw> soundtrigger_2_2 = toService2_2(soundtrigger);
if (sound_model->type == SOUND_MODEL_TYPE_KEYPHRASE) {
- if (!soundtrigger_2_1) {
- ISoundTriggerHw::PhraseSoundModel halSoundModel;
- convertPhraseSoundModelToHal(&halSoundModel, sound_model);
- AutoMutex lock(mHalLock);
- hidlReturn = soundtrigger->loadPhraseSoundModel(
- halSoundModel,
- this, modelId, [&](int32_t retval, auto res) {
- ret = retval;
- halHandle = res;
- });
- } else {
+ if (soundtrigger_2_2) {
+ V2_2_ISoundTriggerHw::PhraseSoundModel halSoundModel;
+ auto result = convertPhraseSoundModelToHal(&halSoundModel, sound_model);
+ if (result.first) {
+ AutoMutex lock(mHalLock);
+ hidlReturn = soundtrigger_2_2->loadPhraseSoundModel_2_1(
+ halSoundModel,
+ this, modelId, [&](int32_t retval, auto res) {
+ ret = retval;
+ halHandle = res;
+ });
+ } else {
+ return NO_MEMORY;
+ }
+ } else if (soundtrigger_2_1) {
V2_1_ISoundTriggerHw::PhraseSoundModel halSoundModel;
auto result = convertPhraseSoundModelToHal(&halSoundModel, sound_model);
if (result.first) {
@@ -193,18 +198,32 @@
} else {
return NO_MEMORY;
}
- }
- } else {
- if (!soundtrigger_2_1) {
- ISoundTriggerHw::SoundModel halSoundModel;
- convertSoundModelToHal(&halSoundModel, sound_model);
+ } else {
+ ISoundTriggerHw::PhraseSoundModel halSoundModel;
+ convertPhraseSoundModelToHal(&halSoundModel, sound_model);
AutoMutex lock(mHalLock);
- hidlReturn = soundtrigger->loadSoundModel(halSoundModel,
+ hidlReturn = soundtrigger->loadPhraseSoundModel(
+ halSoundModel,
this, modelId, [&](int32_t retval, auto res) {
ret = retval;
halHandle = res;
});
- } else {
+ }
+ } else {
+ if (soundtrigger_2_2) {
+ V2_2_ISoundTriggerHw::SoundModel halSoundModel;
+ auto result = convertSoundModelToHal(&halSoundModel, sound_model);
+ if (result.first) {
+ AutoMutex lock(mHalLock);
+ hidlReturn = soundtrigger_2_2->loadSoundModel_2_1(halSoundModel,
+ this, modelId, [&](int32_t retval, auto res) {
+ ret = retval;
+ halHandle = res;
+ });
+ } else {
+ return NO_MEMORY;
+ }
+ } else if (soundtrigger_2_1) {
V2_1_ISoundTriggerHw::SoundModel halSoundModel;
auto result = convertSoundModelToHal(&halSoundModel, sound_model);
if (result.first) {
@@ -217,6 +236,15 @@
} else {
return NO_MEMORY;
}
+ } else {
+ ISoundTriggerHw::SoundModel halSoundModel;
+ convertSoundModelToHal(&halSoundModel, sound_model);
+ AutoMutex lock(mHalLock);
+ hidlReturn = soundtrigger->loadSoundModel(halSoundModel,
+ this, modelId, [&](int32_t retval, auto res) {
+ ret = retval;
+ halHandle = res;
+ });
}
}
@@ -282,16 +310,20 @@
model->mRecognitionCookie = cookie;
sp<V2_1_ISoundTriggerHw> soundtrigger_2_1 = toService2_1(soundtrigger);
+ sp<V2_2_ISoundTriggerHw> soundtrigger_2_2 = toService2_2(soundtrigger);
Return<int32_t> hidlReturn(0);
- if (!soundtrigger_2_1) {
- ISoundTriggerHw::RecognitionConfig halConfig;
- convertRecognitionConfigToHal(&halConfig, config);
- {
+ if (soundtrigger_2_2) {
+ V2_2_ISoundTriggerHw::RecognitionConfig halConfig;
+ auto result = convertRecognitionConfigToHal(&halConfig, config);
+ if (result.first) {
AutoMutex lock(mHalLock);
- hidlReturn = soundtrigger->startRecognition(model->mHalHandle, halConfig, this, handle);
+ hidlReturn = soundtrigger_2_2->startRecognition_2_1(
+ model->mHalHandle, halConfig, this, handle);
+ } else {
+ return NO_MEMORY;
}
- } else {
+ } else if (soundtrigger_2_1) {
V2_1_ISoundTriggerHw::RecognitionConfig halConfig;
auto result = convertRecognitionConfigToHal(&halConfig, config);
if (result.first) {
@@ -301,6 +333,13 @@
} else {
return NO_MEMORY;
}
+ } else {
+ ISoundTriggerHw::RecognitionConfig halConfig;
+ convertRecognitionConfigToHal(&halConfig, config);
+ {
+ AutoMutex lock(mHalLock);
+ hidlReturn = soundtrigger->startRecognition(model->mHalHandle, halConfig, this, handle);
+ }
}
if (!hidlReturn.isOk()) {