Merge "NuPlayerDrm: explicitly use 32-bit subsample sizes" into tm-qpr-dev
diff --git a/media/codec2/sfplugin/CCodec.cpp b/media/codec2/sfplugin/CCodec.cpp
index 296d7ed..43bdb21 100644
--- a/media/codec2/sfplugin/CCodec.cpp
+++ b/media/codec2/sfplugin/CCodec.cpp
@@ -1899,6 +1899,7 @@
comp = state->comp;
}
status_t err = comp->stop();
+ mChannel->stopUseOutputSurface();
if (err != C2_OK) {
// TODO: convert err into status_t
mCallback->onError(UNKNOWN_ERROR, ACTION_CODE_FATAL);
@@ -1972,6 +1973,7 @@
}
mChannel->reset();
+ mChannel->stopUseOutputSurface();
// thiz holds strong ref to this while the thread is running.
sp<CCodec> thiz(this);
std::thread([thiz, sendCallback] { thiz->release(sendCallback); }).detach();
diff --git a/media/codec2/sfplugin/CCodecBufferChannel.cpp b/media/codec2/sfplugin/CCodecBufferChannel.cpp
index 5ecb130..9d2e1bd 100644
--- a/media/codec2/sfplugin/CCodecBufferChannel.cpp
+++ b/media/codec2/sfplugin/CCodecBufferChannel.cpp
@@ -1578,6 +1578,17 @@
mFirstValidFrameIndex = mFrameIndex.load(std::memory_order_relaxed);
}
+void CCodecBufferChannel::stopUseOutputSurface() {
+ if (mOutputSurface.lock()->surface) {
+ C2BlockPool::local_id_t outputPoolId;
+ {
+ Mutexed<BlockPools>::Locked pools(mBlockPools);
+ outputPoolId = pools->outputPoolId;
+ }
+ if (mComponent) mComponent->stopUsingOutputSurface(outputPoolId);
+ }
+}
+
void CCodecBufferChannel::reset() {
stop();
if (mInputSurface != nullptr) {
@@ -1593,14 +1604,6 @@
Mutexed<Output>::Locked output(mOutput);
output->buffers.reset();
}
- if (mOutputSurface.lock()->surface) {
- C2BlockPool::local_id_t outputPoolId;
- {
- Mutexed<BlockPools>::Locked pools(mBlockPools);
- outputPoolId = pools->outputPoolId;
- }
- mComponent->stopUsingOutputSurface(outputPoolId);
- }
}
void CCodecBufferChannel::release() {
diff --git a/media/codec2/sfplugin/CCodecBufferChannel.h b/media/codec2/sfplugin/CCodecBufferChannel.h
index f29a225..61fb06f 100644
--- a/media/codec2/sfplugin/CCodecBufferChannel.h
+++ b/media/codec2/sfplugin/CCodecBufferChannel.h
@@ -149,6 +149,12 @@
std::map<size_t, sp<MediaCodecBuffer>> &&clientInputBuffers);
/**
+ * Stop using buffers of the current output surface for other Codec
+ * instances to use the surface safely.
+ */
+ void stopUseOutputSurface();
+
+ /**
* Stop queueing buffers to the component. This object should never queue
* buffers after this call, until start() is called.
*/
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 56ee014..3d44aec 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -9694,6 +9694,12 @@
if (isOutput()) {
ret = AudioSystem::startOutput(portId);
} else {
+ {
+ // Add the track record before starting input so that the silent status for the
+ // client can be cached.
+ Mutex::Autolock _l(mLock);
+ setClientSilencedState_l(portId, false /*silenced*/);
+ }
ret = AudioSystem::startInput(portId);
}
@@ -9712,6 +9718,7 @@
} else {
mHalStream->stop();
}
+ eraseClientSilencedState_l(portId);
return PERMISSION_DENIED;
}
@@ -9720,6 +9727,9 @@
mChannelMask, mSessionId, isOutput(),
client.attributionSource,
IPCThreadState::self()->getCallingPid(), portId);
+ if (!isOutput()) {
+ track->setSilenced_l(isClientSilenced_l(portId));
+ }
if (isOutput()) {
// force volume update when a new track is added
@@ -9777,6 +9787,7 @@
}
mActiveTracks.remove(track);
+ eraseClientSilencedState_l(track->portId());
mLock.unlock();
if (isOutput()) {
@@ -10567,6 +10578,7 @@
broadcast_l();
}
}
+ setClientSilencedIfExists_l(portId, silenced);
}
void AudioFlinger::MmapCaptureThread::toAudioPortConfig(struct audio_port_config *config)
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index da5adea..ad5617d 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -1172,7 +1172,7 @@
volatile int32_t mSuspended;
int64_t mBytesWritten;
- int64_t mFramesWritten; // not reset on standby
+ std::atomic<int64_t> mFramesWritten; // not reset on standby
int64_t mLastFramesWritten = -1; // track changes in timestamp
// server frames written.
int64_t mSuspendedFrames; // not reset on standby
@@ -1386,6 +1386,7 @@
virtual bool hasFastMixer() const = 0;
virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex __unused) const
{ FastTrackUnderruns dummy; return dummy; }
+ const std::atomic<int64_t>& framesWritten() const { return mFramesWritten; }
protected:
// accessed by both binder threads and within threadLoop(), lock on mutex needed
@@ -2130,6 +2131,26 @@
virtual bool isStreamInitialized() { return false; }
+ void setClientSilencedState_l(audio_port_handle_t portId, bool silenced) {
+ mClientSilencedStates[portId] = silenced;
+ }
+
+ size_t eraseClientSilencedState_l(audio_port_handle_t portId) {
+ return mClientSilencedStates.erase(portId);
+ }
+
+ bool isClientSilenced_l(audio_port_handle_t portId) const {
+ const auto it = mClientSilencedStates.find(portId);
+ return it != mClientSilencedStates.end() ? it->second : false;
+ }
+
+ void setClientSilencedIfExists_l(audio_port_handle_t portId, bool silenced) {
+ const auto it = mClientSilencedStates.find(portId);
+ if (it != mClientSilencedStates.end()) {
+ it->second = silenced;
+ }
+ }
+
protected:
void dumpInternals_l(int fd, const Vector<String16>& args) override;
void dumpTracks_l(int fd, const Vector<String16>& args) override;
@@ -2149,6 +2170,7 @@
AudioHwDevice* const mAudioHwDev;
ActiveTracks<MmapTrack> mActiveTracks;
float mHalVolFloat;
+ std::map<audio_port_handle_t, bool> mClientSilencedStates;
int32_t mNoCallbackWarningCount;
static constexpr int32_t kMaxNoCallbackWarnings = 5;
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index b422bb2..95ca855 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -1094,12 +1094,22 @@
__func__, mId, (int)mThreadIoHandle);
}
- // states to reset position info for non-offloaded/direct tracks
- if (!isOffloaded() && !isDirect()
+ PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+
+ // states to reset position info for pcm tracks
+ if (audio_is_linear_pcm(mFormat)
&& (state == IDLE || state == STOPPED || state == FLUSHED)) {
mFrameMap.reset();
+
+ if (!isFastTrack() && (isDirect() || isOffloaded())) {
+ // Start point of track -> sink frame map. If the HAL returns a
+ // frame position smaller than the first written frame in
+ // updateTrackFrameInfo, the timestamp can be interpolated
+ // instead of using a larger value.
+ mFrameMap.push(mAudioTrackServerProxy->framesReleased(),
+ playbackThread->framesWritten());
+ }
}
- PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
if (isFastTrack()) {
// refresh fast track underruns on start because that field is never cleared
// by the fast mixer; furthermore, the same track can be recycled, i.e. start