Merge "codec2: Queue ROI info buffer for surface mode" into main
diff --git a/Android.bp b/Android.bp
index 72b8721..afb1341 100644
--- a/Android.bp
+++ b/Android.bp
@@ -133,3 +133,19 @@
frozen: true,
}
+
+latest_av_audio_types_aidl = "av-audio-types-aidl-V1"
+
+cc_defaults {
+ name: "latest_av_audio_types_aidl_ndk_shared",
+ shared_libs: [
+ latest_av_audio_types_aidl + "-ndk",
+ ],
+}
+
+cc_defaults {
+ name: "latest_av_audio_types_aidl_ndk_static",
+ static_libs: [
+ latest_av_audio_types_aidl + "-ndk",
+ ],
+}
diff --git a/media/TEST_MAPPING b/media/TEST_MAPPING
index fd46b5b..1a637ac 100644
--- a/media/TEST_MAPPING
+++ b/media/TEST_MAPPING
@@ -47,8 +47,13 @@
],
// Postsubmit tests for TV devices
"tv-postsubmit": [
- {
- "name": "android.media.decoder.cts.DecoderRenderTest"
- }
+ {
+ "name": "CtsMediaDecoderTestCases",
+ "options": [
+ {
+ "include-filter": "android.media.decoder.cts.DecoderRenderTest"
+ }
+ ]
+ }
]
}
diff --git a/media/libaudioclient/AudioTrack.cpp b/media/libaudioclient/AudioTrack.cpp
index 2afe80c..d6b1163 100644
--- a/media/libaudioclient/AudioTrack.cpp
+++ b/media/libaudioclient/AudioTrack.cpp
@@ -1707,14 +1707,14 @@
mSelectedDeviceId = deviceId;
if (mStatus == NO_ERROR) {
if (isOffloadedOrDirect_l()) {
- if (mState == STATE_STOPPED || mState == STATE_FLUSHED) {
- ALOGD("%s(%d): creating a new AudioTrack", __func__, mPortId);
- result = restoreTrack_l("setOutputDevice", true /* forceRestore */);
- } else {
+ if (isPlaying_l()) {
ALOGW("%s(%d). Offloaded or Direct track is not STOPPED or FLUSHED. "
"State: %s.",
__func__, mPortId, stateToString(mState));
result = INVALID_OPERATION;
+ } else {
+ ALOGD("%s(%d): creating a new AudioTrack", __func__, mPortId);
+ result = restoreTrack_l("setOutputDevice", true /* forceRestore */);
}
} else {
// allow track invalidation when track is not playing to propagate
diff --git a/media/libaudiohal/impl/Android.bp b/media/libaudiohal/impl/Android.bp
index dd8f021..1a6b949 100644
--- a/media/libaudiohal/impl/Android.bp
+++ b/media/libaudiohal/impl/Android.bp
@@ -227,11 +227,11 @@
"latest_android_hardware_audio_core_sounddose_ndk_shared",
"latest_android_hardware_audio_effect_ndk_shared",
"latest_android_media_audio_common_types_ndk_shared",
+ "latest_av_audio_types_aidl_ndk_shared",
],
shared_libs: [
"android.hardware.common-V2-ndk",
"android.hardware.common.fmq-V1-ndk",
- "av-audio-types-aidl-V1-ndk",
"libaudio_aidl_conversion_common_cpp",
"libaudio_aidl_conversion_common_ndk",
"libaudio_aidl_conversion_common_ndk_cpp",
diff --git a/media/libaudiohal/impl/StreamHalAidl.cpp b/media/libaudiohal/impl/StreamHalAidl.cpp
index 5d2f9e8..4f01ec7 100644
--- a/media/libaudiohal/impl/StreamHalAidl.cpp
+++ b/media/libaudiohal/impl/StreamHalAidl.cpp
@@ -281,11 +281,12 @@
return OK;
}
-status_t StreamHalAidl::getObservablePosition(int64_t *frames, int64_t *timestamp) {
+status_t StreamHalAidl::getObservablePosition(int64_t* frames, int64_t* timestamp,
+ StatePositions* statePositions) {
ALOGV("%p %s::%s", this, getClassName().c_str(), __func__);
if (!mStream) return NO_INIT;
StreamDescriptor::Reply reply;
- RETURN_STATUS_IF_ERROR(updateCountersIfNeeded(&reply));
+ RETURN_STATUS_IF_ERROR(updateCountersIfNeeded(&reply, statePositions));
*frames = std::max<int64_t>(0, reply.observable.frames);
*timestamp = std::max<int64_t>(0, reply.observable.timeNs);
return OK;
@@ -442,8 +443,12 @@
if (auto state = getState(); state == StreamDescriptor::State::DRAINING) {
// Retrieve the current state together with position counters unconditionally
// to ensure that the state on our side gets updated.
- sendCommand(makeHalCommand<HalCommand::Tag::getStatus>(),
- nullptr, true /*safeFromNonWorkerThread */);
+ sendCommand(makeHalCommand<HalCommand::Tag::getStatus>(), nullptr,
+ true /*safeFromNonWorkerThread */);
+ // For compatibility with HIDL behavior, apply a "soft" position reset
+ // after receiving the "drain ready" callback.
+ std::lock_guard l(mLock);
+ mStatePositions.framesAtFlushOrDrain = mLastReply.observable.frames;
} else {
ALOGW("%s: unexpected onDrainReady in the state %s", __func__, toString(state).c_str());
}
@@ -510,9 +515,9 @@
}
status_t StreamHalAidl::sendCommand(
- const ::aidl::android::hardware::audio::core::StreamDescriptor::Command &command,
+ const ::aidl::android::hardware::audio::core::StreamDescriptor::Command& command,
::aidl::android::hardware::audio::core::StreamDescriptor::Reply* reply,
- bool safeFromNonWorkerThread) {
+ bool safeFromNonWorkerThread, StatePositions* statePositions) {
// TIME_CHECK(); // TODO(b/243839867) reenable only when optimized.
if (!safeFromNonWorkerThread) {
const pid_t workerTid = mWorkerTid.load(std::memory_order_acquire);
@@ -544,6 +549,23 @@
}
mLastReply = *reply;
mLastReplyExpirationNs = uptimeNanos() + mLastReplyLifeTimeNs;
+ if (!mIsInput && reply->status == STATUS_OK) {
+ if (command.getTag() == StreamDescriptor::Command::standby &&
+ reply->state == StreamDescriptor::State::STANDBY) {
+ mStatePositions.framesAtStandby = reply->observable.frames;
+ } else if (command.getTag() == StreamDescriptor::Command::flush &&
+ reply->state == StreamDescriptor::State::IDLE) {
+ mStatePositions.framesAtFlushOrDrain = reply->observable.frames;
+ } else if (!mContext.isAsynchronous() &&
+ command.getTag() == StreamDescriptor::Command::drain &&
+ (reply->state == StreamDescriptor::State::IDLE ||
+ reply->state == StreamDescriptor::State::DRAINING)) {
+ mStatePositions.framesAtFlushOrDrain = reply->observable.frames;
+ } // for asynchronous drain, the frame count is saved in 'onAsyncDrainReady'
+ }
+ if (statePositions != nullptr) {
+ *statePositions = mStatePositions;
+ }
}
}
switch (reply->status) {
@@ -559,7 +581,8 @@
}
status_t StreamHalAidl::updateCountersIfNeeded(
- ::aidl::android::hardware::audio::core::StreamDescriptor::Reply* reply) {
+ ::aidl::android::hardware::audio::core::StreamDescriptor::Reply* reply,
+ StatePositions* statePositions) {
bool doUpdate = false;
{
std::lock_guard l(mLock);
@@ -569,10 +592,13 @@
// Since updates are paced, it is OK to perform them from any thread, they should
// not interfere with I/O operations of the worker.
return sendCommand(makeHalCommand<HalCommand::Tag::getStatus>(),
- reply, true /*safeFromNonWorkerThread */);
+ reply, true /*safeFromNonWorkerThread */, statePositions);
} else if (reply != nullptr) { // provide cached reply
std::lock_guard l(mLock);
*reply = mLastReply;
+ if (statePositions != nullptr) {
+ *statePositions = mStatePositions;
+ }
}
return OK;
}
@@ -664,8 +690,19 @@
return BAD_VALUE;
}
int64_t aidlFrames = 0, aidlTimestamp = 0;
- RETURN_STATUS_IF_ERROR(getObservablePosition(&aidlFrames, &aidlTimestamp));
- *dspFrames = aidlFrames;
+ StatePositions statePositions{};
+ RETURN_STATUS_IF_ERROR(
+ getObservablePosition(&aidlFrames, &aidlTimestamp, &statePositions));
+ // Number of audio frames since the stream has exited standby.
+ // See the table at the start of 'StreamHalInterface' on when it needs to reset.
+ int64_t mostRecentResetPoint;
+ if (!mContext.isAsynchronous() && audio_has_proportional_frames(mConfig.format)) {
+ mostRecentResetPoint = statePositions.framesAtStandby;
+ } else {
+ mostRecentResetPoint =
+ std::max(statePositions.framesAtStandby, statePositions.framesAtFlushOrDrain);
+ }
+ *dspFrames = aidlFrames <= mostRecentResetPoint ? 0 : aidlFrames - mostRecentResetPoint;
return OK;
}
@@ -722,8 +759,16 @@
return BAD_VALUE;
}
int64_t aidlFrames = 0, aidlTimestamp = 0;
- RETURN_STATUS_IF_ERROR(getObservablePosition(&aidlFrames, &aidlTimestamp));
- *frames = aidlFrames;
+ StatePositions statePositions{};
+ RETURN_STATUS_IF_ERROR(getObservablePosition(&aidlFrames, &aidlTimestamp, &statePositions));
+ // See the table at the start of 'StreamHalInterface'.
+ if (!mContext.isAsynchronous() && audio_has_proportional_frames(mConfig.format)) {
+ *frames = aidlFrames;
+ } else {
+ const int64_t mostRecentResetPoint =
+ std::max(statePositions.framesAtStandby, statePositions.framesAtFlushOrDrain);
+ *frames = aidlFrames <= mostRecentResetPoint ? 0 : aidlFrames - mostRecentResetPoint;
+ }
timestamp->tv_sec = aidlTimestamp / NANOS_PER_SECOND;
timestamp->tv_nsec = aidlTimestamp - timestamp->tv_sec * NANOS_PER_SECOND;
return OK;
diff --git a/media/libaudiohal/impl/StreamHalAidl.h b/media/libaudiohal/impl/StreamHalAidl.h
index 8a398d8..fff7a92 100644
--- a/media/libaudiohal/impl/StreamHalAidl.h
+++ b/media/libaudiohal/impl/StreamHalAidl.h
@@ -194,6 +194,11 @@
// For tests.
friend class sp<StreamHalAidl>;
+ struct StatePositions {
+ int64_t framesAtFlushOrDrain;
+ int64_t framesAtStandby;
+ };
+
template<class T>
static std::shared_ptr<::aidl::android::hardware::audio::core::IStreamCommon> getStreamCommon(
const std::shared_ptr<T>& stream);
@@ -212,7 +217,8 @@
status_t getLatency(uint32_t *latency);
// Always returns non-negative values.
- status_t getObservablePosition(int64_t *frames, int64_t *timestamp);
+ status_t getObservablePosition(int64_t* frames, int64_t* timestamp,
+ StatePositions* statePositions = nullptr);
// Always returns non-negative values.
status_t getHardwarePosition(int64_t *frames, int64_t *timestamp);
@@ -268,11 +274,13 @@
// Note: Since `sendCommand` takes mLock while holding mCommandReplyLock, never call
// it with `mLock` being held.
status_t sendCommand(
- const ::aidl::android::hardware::audio::core::StreamDescriptor::Command &command,
+ const ::aidl::android::hardware::audio::core::StreamDescriptor::Command& command,
::aidl::android::hardware::audio::core::StreamDescriptor::Reply* reply = nullptr,
- bool safeFromNonWorkerThread = false);
+ bool safeFromNonWorkerThread = false,
+ StatePositions* statePositions = nullptr);
status_t updateCountersIfNeeded(
- ::aidl::android::hardware::audio::core::StreamDescriptor::Reply* reply = nullptr);
+ ::aidl::android::hardware::audio::core::StreamDescriptor::Reply* reply = nullptr,
+ StatePositions* statePositions = nullptr);
const std::shared_ptr<::aidl::android::hardware::audio::core::IStreamCommon> mStream;
const std::shared_ptr<::aidl::android::media::audio::IHalAdapterVendorExtension> mVendorExt;
@@ -280,6 +288,9 @@
std::mutex mLock;
::aidl::android::hardware::audio::core::StreamDescriptor::Reply mLastReply GUARDED_BY(mLock);
int64_t mLastReplyExpirationNs GUARDED_BY(mLock) = 0;
+ // Cached values of observable positions when the stream last entered certain state.
+ // Updated for output streams only.
+ StatePositions mStatePositions GUARDED_BY(mLock) = {};
// mStreamPowerLog is used for audio signal power logging.
StreamPowerLog mStreamPowerLog;
std::atomic<pid_t> mWorkerTid = -1;
@@ -328,10 +339,14 @@
// Requests notification when data buffered by the driver/hardware has been played.
status_t drain(bool earlyNotify) override;
- // Notifies to the audio driver to flush the queued data.
+ // Notifies to the audio driver to flush (that is, drop) the queued data. Stream must
+ // already be paused before calling 'flush'.
status_t flush() override;
// Return a recent count of the number of audio frames presented to an external observer.
+ // This excludes frames which have been written but are still in the pipeline. See the
+ // table at the start of the 'StreamOutHalInterface' for the specification of the frame
+ // count behavior w.r.t. 'flush', 'drain' and 'standby' operations.
status_t getPresentationPosition(uint64_t *frames, struct timespec *timestamp) override;
// Notifies the HAL layer that the framework considers the current playback as completed.
@@ -413,6 +428,7 @@
// Return a recent count of the number of audio frames received and
// the clock time associated with that frame count.
+ // The count must not reset to zero when a PCM input enters standby.
status_t getCapturePosition(int64_t *frames, int64_t *time) override;
// Get active microphones
diff --git a/media/libaudiohal/impl/StreamHalHidl.h b/media/libaudiohal/impl/StreamHalHidl.h
index 80379d0..433e0a3 100644
--- a/media/libaudiohal/impl/StreamHalHidl.h
+++ b/media/libaudiohal/impl/StreamHalHidl.h
@@ -161,10 +161,14 @@
// Requests notification when data buffered by the driver/hardware has been played.
virtual status_t drain(bool earlyNotify);
- // Notifies to the audio driver to flush the queued data.
+ // Notifies to the audio driver to flush (that is, drop) the queued data. Stream must
+ // already be paused before calling 'flush'.
virtual status_t flush();
// Return a recent count of the number of audio frames presented to an external observer.
+ // This excludes frames which have been written but are still in the pipeline. See the
+ // table at the start of the 'StreamOutHalInterface' for the specification of the frame
+ // count behavior w.r.t. 'flush', 'drain' and 'standby' operations.
virtual status_t getPresentationPosition(uint64_t *frames, struct timespec *timestamp);
// Notifies the HAL layer that the framework considers the current playback as completed.
@@ -259,6 +263,7 @@
// Return a recent count of the number of audio frames received and
// the clock time associated with that frame count.
+ // The count must not reset to zero when a PCM input enters standby.
virtual status_t getCapturePosition(int64_t *frames, int64_t *time);
// Get active microphones
diff --git a/media/libaudiohal/include/media/audiohal/StreamHalInterface.h b/media/libaudiohal/include/media/audiohal/StreamHalInterface.h
index eb14f6b..585a895 100644
--- a/media/libaudiohal/include/media/audiohal/StreamHalInterface.h
+++ b/media/libaudiohal/include/media/audiohal/StreamHalInterface.h
@@ -135,6 +135,38 @@
virtual ~StreamOutHalInterfaceLatencyModeCallback() = default;
};
+/**
+ * On position reporting. There are two methods: 'getRenderPosition' and
+ * 'getPresentationPosition'. The first difference is that they may have a
+ * time offset because "render" position relates to what happens between
+ * ADSP and DAC, while "observable" position is relative to the external
+ * observer. The second difference is that 'getRenderPosition' always
+ * resets on standby (for all types of stream data) according to its
+ * definition. Since the original C definition of 'getRenderPosition' used
+ * 32-bit frame counters, and also because in complex playback chains that
+ * include wireless devices the "observable" position has more practical
+ * meaning, 'getRenderPosition' does not exist in the AIDL HAL interface.
+ * The table below summarizes frame count behavior for 'getPresentationPosition':
+ *
+ * | Mixed | Direct | Direct
+ * | | non-offload | offload
+ * ==============|============|==============|==============
+ * PCM and | Continuous | |
+ * encapsulated | | |
+ * bitstream | | |
+ * --------------|------------| Continuous† |
+ * Bitstream | | | Reset on
+ * encapsulated | | | flush, drain
+ * into PCM | | | and standby
+ * | Not | |
+ * --------------| supported |--------------|
+ * Bitstream | | Reset on |
+ * | | flush, drain |
+ * | | and standby |
+ * | | |
+ *
+ * † - on standby, reset of the frame count happens at the framework level.
+ */
class StreamOutHalInterface : public virtual StreamHalInterface {
public:
// Return the audio hardware driver estimated latency in milliseconds.
@@ -173,10 +205,14 @@
// Requests notification when data buffered by the driver/hardware has been played.
virtual status_t drain(bool earlyNotify) = 0;
- // Notifies to the audio driver to flush the queued data.
+ // Notifies to the audio driver to flush (that is, drop) the queued data. Stream must
+ // already be paused before calling 'flush'.
virtual status_t flush() = 0;
// Return a recent count of the number of audio frames presented to an external observer.
+ // This excludes frames which have been written but are still in the pipeline. See the
+ // table at the start of the 'StreamOutHalInterface' for the specification of the frame
+ // count behavior w.r.t. 'flush', 'drain' and 'standby' operations.
virtual status_t getPresentationPosition(uint64_t *frames, struct timespec *timestamp) = 0;
// Notifies the HAL layer that the framework considers the current playback as completed.
@@ -270,6 +306,7 @@
// Return a recent count of the number of audio frames received and
// the clock time associated with that frame count.
+ // The count must not reset to zero when a PCM input enters standby.
virtual status_t getCapturePosition(int64_t *frames, int64_t *time) = 0;
// Get active microphones
diff --git a/media/libstagefright/VideoRenderQualityTracker.cpp b/media/libstagefright/VideoRenderQualityTracker.cpp
index eb9ac0f..bf29b1d 100644
--- a/media/libstagefright/VideoRenderQualityTracker.cpp
+++ b/media/libstagefright/VideoRenderQualityTracker.cpp
@@ -302,13 +302,6 @@
mRenderDurationMs += (actualRenderTimeUs - mLastRenderTimeUs) / 1000;
}
- // Now that a frame has been rendered, the previously skipped frames can be processed as skipped
- // frames since the app is not skipping them to terminate playback.
- for (int64_t contentTimeUs : mPendingSkippedFrameContentTimeUsList) {
- processMetricsForSkippedFrame(contentTimeUs);
- }
- mPendingSkippedFrameContentTimeUsList = {};
-
// We can render a pending queued frame if it's the last frame of the video, so release it
// immediately.
if (contentTimeUs == mTunnelFrameQueuedContentTimeUs && mTunnelFrameQueuedContentTimeUs != -1) {
@@ -332,9 +325,25 @@
(long long) contentTimeUs, (long long) nextExpectedFrame.contentTimeUs);
break;
}
+ // Process all skipped frames before the dropped frame.
+ while (!mPendingSkippedFrameContentTimeUsList.empty()) {
+ if (mPendingSkippedFrameContentTimeUsList.front() >= nextExpectedFrame.contentTimeUs) {
+ break;
+ }
+ processMetricsForSkippedFrame(mPendingSkippedFrameContentTimeUsList.front());
+ mPendingSkippedFrameContentTimeUsList.pop_front();
+ }
processMetricsForDroppedFrame(nextExpectedFrame.contentTimeUs,
nextExpectedFrame.desiredRenderTimeUs);
}
+ // Process all skipped frames before the rendered frame.
+ while (!mPendingSkippedFrameContentTimeUsList.empty()) {
+ if (mPendingSkippedFrameContentTimeUsList.front() >= nextExpectedFrame.contentTimeUs) {
+ break;
+ }
+ processMetricsForSkippedFrame(mPendingSkippedFrameContentTimeUsList.front());
+ mPendingSkippedFrameContentTimeUsList.pop_front();
+ }
processMetricsForRenderedFrame(nextExpectedFrame.contentTimeUs,
nextExpectedFrame.desiredRenderTimeUs, actualRenderTimeUs,
freezeEventOut, judderEventOut);
diff --git a/media/libstagefright/timedtext/test/fuzzer/Android.bp b/media/libstagefright/timedtext/test/fuzzer/Android.bp
index 6590ebb..8724d51 100644
--- a/media/libstagefright/timedtext/test/fuzzer/Android.bp
+++ b/media/libstagefright/timedtext/test/fuzzer/Android.bp
@@ -48,8 +48,16 @@
],
fuzz_config: {
cc: [
- "android-media-fuzzing-reports@google.com",
+ "android-media-playback@google.com",
],
- componentid: 155276,
+ componentid: 42195,
+ hotlists: [
+ "4593311",
+ ],
+ description: "This fuzzer targets the APIs of libstagefright_timedtext",
+ vector: "local_no_privileges_required",
+ service_privilege: "constrained",
+ users: "multi_user",
+ fuzzed_code_usage: "shipped",
},
}
diff --git a/media/module/extractors/mp4/MPEG4Extractor.cpp b/media/module/extractors/mp4/MPEG4Extractor.cpp
index b3707c8..cb2994e 100644
--- a/media/module/extractors/mp4/MPEG4Extractor.cpp
+++ b/media/module/extractors/mp4/MPEG4Extractor.cpp
@@ -1615,39 +1615,6 @@
mLastTrack->timescale = ntohl(timescale);
- // 14496-12 says all ones means indeterminate, but some files seem to use
- // 0 instead. We treat both the same.
- int64_t duration = 0;
- if (version == 1) {
- if (mDataSource->readAt(
- timescale_offset + 4, &duration, sizeof(duration))
- < (ssize_t)sizeof(duration)) {
- return ERROR_IO;
- }
- if (duration != -1) {
- duration = ntoh64(duration);
- }
- } else {
- uint32_t duration32;
- if (mDataSource->readAt(
- timescale_offset + 4, &duration32, sizeof(duration32))
- < (ssize_t)sizeof(duration32)) {
- return ERROR_IO;
- }
- if (duration32 != 0xffffffff) {
- duration = ntohl(duration32);
- }
- }
- if (duration != 0 && mLastTrack->timescale != 0) {
- long double durationUs = ((long double)duration * 1000000) / mLastTrack->timescale;
- if (durationUs < 0 || durationUs > INT64_MAX) {
- ALOGE("cannot represent %lld * 1000000 / %lld in 64 bits",
- (long long) duration, (long long) mLastTrack->timescale);
- return ERROR_MALFORMED;
- }
- AMediaFormat_setInt64(mLastTrack->meta, AMEDIAFORMAT_KEY_DURATION, durationUs);
- }
-
uint8_t lang[2];
off64_t lang_offset;
if (version == 1) {
@@ -3907,17 +3874,18 @@
}
int32_t id;
+ int64_t duration;
if (version == 1) {
// we can get ctime value from U64_AT(&buffer[4])
// we can get mtime value from U64_AT(&buffer[12])
id = U32_AT(&buffer[20]);
- // we can get duration value from U64_AT(&buffer[28])
+ duration = U64_AT(&buffer[28]);
} else if (version == 0) {
// we can get ctime value from U32_AT(&buffer[4])
// we can get mtime value from U32_AT(&buffer[8])
id = U32_AT(&buffer[12]);
- // we can get duration value from U32_AT(&buffer[20])
+ duration = U32_AT(&buffer[20]);
} else {
return ERROR_UNSUPPORTED;
}
@@ -3926,6 +3894,15 @@
return ERROR_MALFORMED;
AMediaFormat_setInt32(mLastTrack->meta, AMEDIAFORMAT_KEY_TRACK_ID, id);
+ if (duration != 0 && mHeaderTimescale != 0) {
+ long double durationUs = ((long double)duration * 1000000) / mHeaderTimescale;
+ if (durationUs < 0 || durationUs > INT64_MAX) {
+ ALOGE("cannot represent %lld * 1000000 / %lld in 64 bits",
+ (long long) duration, (long long) mHeaderTimescale);
+ return ERROR_MALFORMED;
+ }
+ AMediaFormat_setInt64(mLastTrack->meta, AMEDIAFORMAT_KEY_DURATION, durationUs);
+ }
size_t matrixOffset = dynSize + 16;
int32_t a00 = U32_AT(&buffer[matrixOffset]);
diff --git a/media/ndk/Android.bp b/media/ndk/Android.bp
index 9ec7700..3d873df 100644
--- a/media/ndk/Android.bp
+++ b/media/ndk/Android.bp
@@ -192,7 +192,6 @@
header_libs: [
"libstagefright_headers",
"libmedia_headers",
- "libstagefright_headers",
],
shared_libs: [
diff --git a/services/audioparameterparser/Android.bp b/services/audioparameterparser/Android.bp
index f5feece..1c1c1e1 100644
--- a/services/audioparameterparser/Android.bp
+++ b/services/audioparameterparser/Android.bp
@@ -35,10 +35,10 @@
name: "android.hardware.audio.parameter_parser.example_defaults",
defaults: [
"latest_android_hardware_audio_core_ndk_shared",
+ "latest_av_audio_types_aidl_ndk_shared",
],
shared_libs: [
- "av-audio-types-aidl-V1-ndk",
"libbase",
"libbinder_ndk",
],
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
index d027564..747af4a 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
@@ -561,6 +561,7 @@
audio_port_config config = {};
devicePort->toAudioPortConfig(&config);
config.config_mask = AUDIO_PORT_CONFIG_GAIN;
+ config.gain.mode = gains[0]->getMode();
config.gain.values[0] = gainValueMb;
return mClientInterface->setAudioPortConfig(&config, 0) == NO_ERROR;
}