Fix issue 2553359: Pandora does not work well with Passion deskdock / Cardock.
The problem is due to a too big difference between the buffer size used at the hardware interface and at the A2DP interface.
When no resampling occurs we don't notice problems but the timing is very tight. As soon as resampling is activated, the AudioTrack underruns.
This is because the AudioTrack buffers are not resized when moving the AudioTrack from hardware to A2DP output.
The AudioTrack buffers are calculated based on a hardware output buffer size of 3072 bytes. Which is much less than the A2DP output buffer size (10240).
The solution consists in creating new tracks with new buffers in AudioFlinger when the A2DP output is opened
instead of just transfering active tracks from hardware output mixer thread to the new A2DP output mixer thread.
To avoid synchronization issues between mixer threads and client processes, this is done by invalidating tracks
by setting a flag in their control block and having AudioTrack release the handle on this track (IAudioTrack)
and create a new IAudioTrack when this flag is detected next time obtainBuffer() or start() is executed.
AudioFlinger modifications:
- invalidate the tracks when setStreamOutput() is called
- make sure that notifications of output opening/closing and change of stream type to output mapping are sent synchronously to client process.
This is necessary so that AudioSystem has the new stream to output mapping when the AudioTrack detects the invalidate flag in the client process.
Previously their were sent when the corresponding thread loop was executed.
AudioTrack modifications:
- move frame count calculation and verification from set() to createTrack() so that is is updated every time a new IAudioTrack is created.
- detect track invalidate flag in obtainBuffer() and start() and create a new IAudioTrack.
AudioTrackShared modifications
- group all flags (out, flowControlFlag, forceReady...) into a single bit filed to save space.
Change-Id: I9ac26b6192230627d35084e1449640caaf7d56ee
diff --git a/libs/audioflinger/AudioFlinger.cpp b/libs/audioflinger/AudioFlinger.cpp
index 58eb590..3b38d83 100644
--- a/libs/audioflinger/AudioFlinger.cpp
+++ b/libs/audioflinger/AudioFlinger.cpp
@@ -873,11 +873,12 @@
LOGV("processConfigEvents() remaining events %d", mConfigEvents.size());
ConfigEvent *configEvent = mConfigEvents[0];
mConfigEvents.removeAt(0);
- // release mLock because audioConfigChanged() will lock AudioFlinger mLock
- // before calling Audioflinger::audioConfigChanged_l() thus creating
- // potential cross deadlock between AudioFlinger::mLock and mLock
+ // release mLock before locking AudioFlinger mLock: lock order is always
+ // AudioFlinger then ThreadBase to avoid cross deadlock
mLock.unlock();
- audioConfigChanged(configEvent->mEvent, configEvent->mParam);
+ mAudioFlinger->mLock.lock();
+ audioConfigChanged_l(configEvent->mEvent, configEvent->mParam);
+ mAudioFlinger->mLock.unlock();
delete configEvent;
mLock.lock();
}
@@ -953,8 +954,6 @@
mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream);
mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream);
}
- // notify client processes that a new input has been opened
- sendConfigEvent(AudioSystem::OUTPUT_OPENED);
}
AudioFlinger::PlaybackThread::~PlaybackThread()
@@ -1234,11 +1233,12 @@
return mOutput->getParameters(keys);
}
-void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) {
+// destroyTrack_l() must be called with AudioFlinger::mLock held
+void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) {
AudioSystem::OutputDescriptor desc;
void *param2 = 0;
- LOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event, param);
+ LOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param);
switch (event) {
case AudioSystem::OUTPUT_OPENED:
@@ -1257,7 +1257,6 @@
default:
break;
}
- Mutex::Autolock _l(mAudioFlinger->mLock);
mAudioFlinger->audioConfigChanged_l(event, mId, param2);
}
@@ -1614,66 +1613,22 @@
return mixerStatus;
}
-void AudioFlinger::MixerThread::getTracks(
- SortedVector < sp<Track> >& tracks,
- SortedVector < wp<Track> >& activeTracks,
- int streamType)
+void AudioFlinger::MixerThread::invalidateTracks(int streamType)
{
- LOGV ("MixerThread::getTracks() mixer %p, mTracks.size %d, mActiveTracks.size %d", this, mTracks.size(), mActiveTracks.size());
+ LOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", this, streamType, mTracks.size());
Mutex::Autolock _l(mLock);
size_t size = mTracks.size();
for (size_t i = 0; i < size; i++) {
sp<Track> t = mTracks[i];
if (t->type() == streamType) {
- tracks.add(t);
- int j = mActiveTracks.indexOf(t);
- if (j >= 0) {
- t = mActiveTracks[j].promote();
- if (t != NULL) {
- activeTracks.add(t);
+ t->mCblk->lock.lock();
+ t->mCblk->flags |= CBLK_INVALID_ON;
+ t->mCblk->cv.signal();
+ t->mCblk->lock.unlock();
}
}
}
- }
- size = activeTracks.size();
- for (size_t i = 0; i < size; i++) {
- mActiveTracks.remove(activeTracks[i]);
- }
-
- size = tracks.size();
- for (size_t i = 0; i < size; i++) {
- sp<Track> t = tracks[i];
- mTracks.remove(t);
- deleteTrackName_l(t->name());
- }
-}
-
-void AudioFlinger::MixerThread::putTracks(
- SortedVector < sp<Track> >& tracks,
- SortedVector < wp<Track> >& activeTracks)
-{
- LOGV ("MixerThread::putTracks() mixer %p, tracks.size %d, activeTracks.size %d", this, tracks.size(), activeTracks.size());
- Mutex::Autolock _l(mLock);
- size_t size = tracks.size();
- for (size_t i = 0; i < size ; i++) {
- sp<Track> t = tracks[i];
- int name = getTrackName_l();
-
- if (name < 0) return;
-
- t->mName = name;
- t->mThread = this;
- mTracks.add(t);
-
- int j = activeTracks.indexOf(t);
- if (j >= 0) {
- mActiveTracks.add(t);
- // force buffer refilling and no ramp volume when the track is mixed for the first time
- t->mFillingUpStatus = Track::FS_FILLING;
- }
- }
-}
// getTrackName_l() must be called with ThreadBase::mLock held
int AudioFlinger::MixerThread::getTrackName_l()
@@ -2348,7 +2303,7 @@
memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
// Force underrun condition to avoid false underrun callback until first data is
// written to buffer
- mCblk->flowControlFlag = 1;
+ mCblk->flags = CBLK_UNDERRUN_ON;
} else {
mBuffer = sharedBuffer->pointer();
}
@@ -2371,7 +2326,7 @@
memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
// Force underrun condition to avoid false underrun callback until first data is
// written to buffer
- mCblk->flowControlFlag = 1;
+ mCblk->flags = CBLK_UNDERRUN_ON;
mBufferEnd = (uint8_t *)mBuffer + bufferSize;
}
}
@@ -2589,9 +2544,9 @@
if (mFillingUpStatus != FS_FILLING) return true;
if (mCblk->framesReady() >= mCblk->frameCount ||
- mCblk->forceReady) {
+ (mCblk->flags & CBLK_FORCEREADY_MSK)) {
mFillingUpStatus = FS_FILLED;
- mCblk->forceReady = 0;
+ mCblk->flags &= ~CBLK_FORCEREADY_MSK;
return true;
}
return false;
@@ -2706,8 +2661,8 @@
TrackBase::reset();
// Force underrun condition to avoid false underrun callback until first data is
// written to buffer
- mCblk->flowControlFlag = 1;
- mCblk->forceReady = 0;
+ mCblk->flags |= CBLK_UNDERRUN_ON;
+ mCblk->flags &= ~CBLK_FORCEREADY_MSK;
mFillingUpStatus = FS_FILLING;
mResetDone = true;
}
@@ -2818,7 +2773,7 @@
TrackBase::reset();
// Force overerrun condition to avoid false overrun callback until first data is
// read from buffer
- mCblk->flowControlFlag = 1;
+ mCblk->flags |= CBLK_UNDERRUN_ON;
}
}
@@ -2851,7 +2806,7 @@
PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get();
if (mCblk != NULL) {
- mCblk->out = 1;
+ mCblk->flags |= CBLK_DIRECTION_OUT;
mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
mCblk->volume[0] = mCblk->volume[1] = 0x1000;
mOutBuffer.frameCount = 0;
@@ -3274,7 +3229,6 @@
mReqChannelCount = AudioSystem::popCount(channels);
mReqSampleRate = sampleRate;
readInputParameters();
- sendConfigEvent(AudioSystem::INPUT_OPENED);
}
@@ -3689,7 +3643,7 @@
return mInput->getParameters(keys);
}
-void AudioFlinger::RecordThread::audioConfigChanged(int event, int param) {
+void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) {
AudioSystem::OutputDescriptor desc;
void *param2 = 0;
@@ -3708,7 +3662,6 @@
default:
break;
}
- Mutex::Autolock _l(mAudioFlinger->mLock);
mAudioFlinger->audioConfigChanged_l(event, mId, param2);
}
@@ -3828,6 +3781,8 @@
if (pChannels) *pChannels = channels;
if (pLatencyMs) *pLatencyMs = thread->latency();
+ // notify client processes of the new output creation
+ thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
return mNextThreadId;
}
@@ -3849,6 +3804,8 @@
DuplicatingThread *thread = new DuplicatingThread(this, thread1, ++mNextThreadId);
thread->addOutputTrack(thread2);
mPlaybackThreads.add(mNextThreadId, thread);
+ // notify client processes of the new output creation
+ thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
return mNextThreadId;
}
@@ -3978,6 +3935,8 @@
input->standby();
+ // notify client processes of the new input creation
+ thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
return mNextThreadId;
}
@@ -4018,22 +3977,16 @@
}
LOGV("setStreamOutput() stream %d to output %d", stream, output);
+ audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream);
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
if (thread != dstThread &&
thread->type() != PlaybackThread::DIRECT) {
MixerThread *srcThread = (MixerThread *)thread;
- SortedVector < sp<MixerThread::Track> > tracks;
- SortedVector < wp<MixerThread::Track> > activeTracks;
- srcThread->getTracks(tracks, activeTracks, stream);
- if (tracks.size()) {
- dstThread->putTracks(tracks, activeTracks);
+ srcThread->invalidateTracks(stream);
}
}
- }
-
- dstThread->sendConfigEvent(AudioSystem::STREAM_CONFIG_CHANGED, stream);
return NO_ERROR;
}