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-rw-r--r--include/media/AudioTrack.h6
-rw-r--r--include/private/media/AudioTrackShared.h35
-rw-r--r--libs/audioflinger/AudioFlinger.cpp109
-rw-r--r--libs/audioflinger/AudioFlinger.h12
-rw-r--r--libs/audioflinger/AudioPolicyManagerBase.cpp21
-rw-r--r--media/libmedia/AudioRecord.cpp8
-rw-r--r--media/libmedia/AudioTrack.cpp176
7 files changed, 181 insertions, 186 deletions
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index 14b30ae..cc4ab74 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -398,7 +398,8 @@ private:
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
- audio_io_handle_t output);
+ audio_io_handle_t output,
+ bool enforceFrameCount);
sp<IAudioTrack> mAudioTrack;
sp<IMemory> mCblkMemory;
@@ -420,7 +421,8 @@ private:
callback_t mCbf;
void* mUserData;
- uint32_t mNotificationFrames;
+ uint32_t mNotificationFramesReq; // requested number of frames between each notification callback
+ uint32_t mNotificationFramesAct; // actual number of frames between each notification callback
sp<IMemory> mSharedBuffer;
int mLoopCount;
uint32_t mRemainingFrames;
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index ab5ac64..cd47fdf 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -32,6 +32,18 @@ namespace android {
#define MAX_RUN_TIMEOUT_MS 1000
#define WAIT_PERIOD_MS 10
+#define CBLK_UNDERRUN_MSK 0x0001
+#define CBLK_UNDERRUN_ON 0x0001 // underrun (out) or overrrun (in) indication
+#define CBLK_UNDERRUN_OFF 0x0000 // no underrun
+#define CBLK_DIRECTION_MSK 0x0002
+#define CBLK_DIRECTION_OUT 0x0002 // this cblk is for an AudioTrack
+#define CBLK_DIRECTION_IN 0x0000 // this cblk is for an AudioRecord
+#define CBLK_FORCEREADY_MSK 0x0004
+#define CBLK_FORCEREADY_ON 0x0004 // track is considered ready immediately by AudioFlinger
+#define CBLK_FORCEREADY_OFF 0x0000 // track is ready when buffer full
+#define CBLK_INVALID_MSK 0x0008
+#define CBLK_INVALID_ON 0x0008 // track buffer is invalidated by AudioFlinger: must be re-created
+#define CBLK_INVALID_OFF 0x0000
struct audio_track_cblk_t
{
@@ -44,12 +56,12 @@ struct audio_track_cblk_t
volatile uint32_t server;
uint32_t userBase;
uint32_t serverBase;
- void* buffers;
- uint32_t frameCount;
- // Cache line boundary
- uint32_t loopStart;
- uint32_t loopEnd;
- int loopCount;
+ void* buffers;
+ uint32_t frameCount;
+ // Cache line boundary
+ uint32_t loopStart;
+ uint32_t loopEnd;
+ int loopCount;
volatile union {
uint16_t volume[2];
uint32_t volumeLR;
@@ -58,15 +70,16 @@ struct audio_track_cblk_t
// NOTE: audio_track_cblk_t::frameSize is not equal to AudioTrack::frameSize() for
// 8 bit PCM data: in this case, mCblk->frameSize is based on a sample size of
// 16 bit because data is converted to 16 bit before being stored in buffer
- uint32_t frameSize;
+
+ uint8_t frameSize;
uint8_t channelCount;
- uint8_t flowControlFlag; // underrun (out) or overrrun (in) indication
- uint8_t out; // out equals 1 for AudioTrack and 0 for AudioRecord
- uint8_t forceReady;
+ uint16_t flags;
+
uint16_t bufferTimeoutMs; // Maximum cumulated timeout before restarting audioflinger
uint16_t waitTimeMs; // Cumulated wait time
- // Cache line boundary (32 bytes)
+ uint32_t reserved;
+ // Cache line boundary (32 bytes)
audio_track_cblk_t();
uint32_t stepUser(uint32_t frameCount);
bool stepServer(uint32_t frameCount);
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 @@ void AudioFlinger::ThreadBase::processConfigEvents()
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 @@ AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinge
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 @@ String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys)
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 @@ void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) {
default:
break;
}
- Mutex::Autolock _l(mAudioFlinger->mLock);
mAudioFlinger->audioConfigChanged_l(event, mId, param2);
}
@@ -1614,66 +1613,22 @@ uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track
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 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase(
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 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase(
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 @@ bool AudioFlinger::PlaybackThread::Track::isReady() const {
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 @@ void AudioFlinger::PlaybackThread::Track::reset()
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 @@ void AudioFlinger::RecordThread::RecordTrack::stop()
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 @@ AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
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 @@ AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, A
mReqChannelCount = AudioSystem::popCount(channels);
mReqSampleRate = sampleRate;
readInputParameters();
- sendConfigEvent(AudioSystem::INPUT_OPENED);
}
@@ -3689,7 +3643,7 @@ String8 AudioFlinger::RecordThread::getParameters(const String8& keys)
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 @@ void AudioFlinger::RecordThread::audioConfigChanged(int event, int param) {
default:
break;
}
- Mutex::Autolock _l(mAudioFlinger->mLock);
mAudioFlinger->audioConfigChanged_l(event, mId, param2);
}
@@ -3828,6 +3781,8 @@ int AudioFlinger::openOutput(uint32_t *pDevices,
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 @@ int AudioFlinger::openDuplicateOutput(int output1, int output2)
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 @@ int AudioFlinger::openInput(uint32_t *pDevices,
input->standby();
+ // notify client processes of the new input creation
+ thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED);
return mNextThreadId;
}
@@ -4018,22 +3977,16 @@ status_t AudioFlinger::setStreamOutput(uint32_t stream, int output)
}
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;
}
diff --git a/libs/audioflinger/AudioFlinger.h b/libs/audioflinger/AudioFlinger.h
index c4a5305..f35f38b 100644
--- a/libs/audioflinger/AudioFlinger.h
+++ b/libs/audioflinger/AudioFlinger.h
@@ -342,7 +342,7 @@ private:
virtual bool checkForNewParameters_l() = 0;
virtual status_t setParameters(const String8& keyValuePairs);
virtual String8 getParameters(const String8& keys) = 0;
- virtual void audioConfigChanged(int event, int param = 0) = 0;
+ virtual void audioConfigChanged_l(int event, int param = 0) = 0;
void sendConfigEvent(int event, int param = 0);
void sendConfigEvent_l(int event, int param = 0);
void processConfigEvents();
@@ -547,7 +547,7 @@ private:
void restore() { if (mSuspended) mSuspended--; }
bool isSuspended() { return (mSuspended != 0); }
virtual String8 getParameters(const String8& keys);
- virtual void audioConfigChanged(int event, int param = 0);
+ virtual void audioConfigChanged_l(int event, int param = 0);
virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames);
struct stream_type_t {
@@ -613,11 +613,7 @@ private:
// Thread virtuals
virtual bool threadLoop();
- void getTracks(SortedVector < sp<Track> >& tracks,
- SortedVector < wp<Track> >& activeTracks,
- int streamType);
- void putTracks(SortedVector < sp<Track> >& tracks,
- SortedVector < wp<Track> >& activeTracks);
+ void invalidateTracks(int streamType);
virtual bool checkForNewParameters_l();
virtual status_t dumpInternals(int fd, const Vector<String16>& args);
@@ -764,7 +760,7 @@ private:
virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
virtual bool checkForNewParameters_l();
virtual String8 getParameters(const String8& keys);
- virtual void audioConfigChanged(int event, int param = 0);
+ virtual void audioConfigChanged_l(int event, int param = 0);
void readInputParameters();
virtual unsigned int getInputFramesLost();
diff --git a/libs/audioflinger/AudioPolicyManagerBase.cpp b/libs/audioflinger/AudioPolicyManagerBase.cpp
index c8b3f48..381a958 100644
--- a/libs/audioflinger/AudioPolicyManagerBase.cpp
+++ b/libs/audioflinger/AudioPolicyManagerBase.cpp
@@ -1249,6 +1249,17 @@ void AudioPolicyManagerBase::closeA2dpOutputs()
LOGV("setDeviceConnectionState() closing A2DP and duplicated output!");
if (mDuplicatedOutput != 0) {
+ AudioOutputDescriptor *dupOutputDesc = mOutputs.valueFor(mDuplicatedOutput);
+ AudioOutputDescriptor *hwOutputDesc = mOutputs.valueFor(mHardwareOutput);
+ // As all active tracks on duplicated output will be deleted,
+ // and as they were also referenced on hardware output, the reference
+ // count for their stream type must be adjusted accordingly on
+ // hardware output.
+ for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
+ int refCount = dupOutputDesc->mRefCount[i];
+ hwOutputDesc->changeRefCount((AudioSystem::stream_type)i,-refCount);
+ }
+
mpClientInterface->closeOutput(mDuplicatedOutput);
delete mOutputs.valueFor(mDuplicatedOutput);
mOutputs.removeItem(mDuplicatedOutput);
@@ -1288,11 +1299,6 @@ void AudioPolicyManagerBase::checkOutputForStrategy(routing_strategy strategy, u
for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
if (getStrategy((AudioSystem::stream_type)i) == strategy) {
mpClientInterface->setStreamOutput((AudioSystem::stream_type)i, mHardwareOutput);
- int refCount = a2dpOutputDesc->mRefCount[i];
- // in the case of duplicated output, the ref count is first incremented
- // and then decremented on hardware output tus keeping its value
- hwOutputDesc->changeRefCount((AudioSystem::stream_type)i, refCount);
- a2dpOutputDesc->changeRefCount((AudioSystem::stream_type)i,-refCount);
}
}
// do not change newDevice if it was already set before this call by a previous call to
@@ -1318,11 +1324,6 @@ void AudioPolicyManagerBase::checkOutputForStrategy(routing_strategy strategy, u
for (int i = 0; i < (int)AudioSystem::NUM_STREAM_TYPES; i++) {
if (getStrategy((AudioSystem::stream_type)i) == strategy) {
mpClientInterface->setStreamOutput((AudioSystem::stream_type)i, a2dpOutput);
- int refCount = hwOutputDesc->mRefCount[i];
- // in the case of duplicated output, the ref count is first incremented
- // and then decremented on hardware output tus keeping its value
- a2dpOutputDesc->changeRefCount((AudioSystem::stream_type)i, refCount);
- hwOutputDesc->changeRefCount((AudioSystem::stream_type)i,-refCount);
}
}
}
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index ad037d6..fd2b1ce 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -430,7 +430,7 @@ status_t AudioRecord::openRecord(
mCblkMemory = cblk;
mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());
mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
- mCblk->out = 0;
+ mCblk->flags &= ~CBLK_DIRECTION_MSK;
mCblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
mCblk->waitTimeMs = 0;
return NO_ERROR;
@@ -644,10 +644,10 @@ bool AudioRecord::processAudioBuffer(const sp<ClientRecordThread>& thread)
// Manage overrun callback
if (mActive && (mCblk->framesAvailable_l() == 0)) {
- LOGV("Overrun user: %x, server: %x, flowControlFlag %d", mCblk->user, mCblk->server, mCblk->flowControlFlag);
- if (mCblk->flowControlFlag == 0) {
+ LOGV("Overrun user: %x, server: %x, flags %04x", mCblk->user, mCblk->server, mCblk->flags);
+ if ((mCblk->flags & CBLK_UNDERRUN_MSK) == CBLK_UNDERRUN_OFF) {
mCbf(EVENT_OVERRUN, mUserData, 0);
- mCblk->flowControlFlag = 1;
+ mCblk->flags |= CBLK_UNDERRUN_ON;
}
}
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index cd7bcd5..c350532 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -124,10 +124,6 @@ status_t AudioTrack::set(
if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
return NO_INIT;
}
- int afFrameCount;
- if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
- return NO_INIT;
- }
uint32_t afLatency;
if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
return NO_INIT;
@@ -173,48 +169,13 @@ status_t AudioTrack::set(
return BAD_VALUE;
}
- if (!AudioSystem::isLinearPCM(format)) {
- if (sharedBuffer != 0) {
- frameCount = sharedBuffer->size();
- }
- } else {
- // Ensure that buffer depth covers at least audio hardware latency
- uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);
- if (minBufCount < 2) minBufCount = 2;
-
- int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
-
- if (sharedBuffer == 0) {
- if (frameCount == 0) {
- frameCount = minFrameCount;
- }
- if (notificationFrames == 0) {
- notificationFrames = frameCount/2;
- }
- // Make sure that application is notified with sufficient margin
- // before underrun
- if (notificationFrames > frameCount/2) {
- notificationFrames = frameCount/2;
- }
- if (frameCount < minFrameCount) {
- LOGE("Invalid buffer size: minFrameCount %d, frameCount %d", minFrameCount, frameCount);
- return BAD_VALUE;
- }
- } else {
- // Ensure that buffer alignment matches channelcount
- if (((uint32_t)sharedBuffer->pointer() & (channelCount | 1)) != 0) {
- LOGE("Invalid buffer alignement: address %p, channelCount %d", sharedBuffer->pointer(), channelCount);
- return BAD_VALUE;
- }
- frameCount = sharedBuffer->size()/channelCount/sizeof(int16_t);
- }
- }
-
mVolume[LEFT] = 1.0f;
mVolume[RIGHT] = 1.0f;
+ mFrameCount = frameCount;
+ mNotificationFramesReq = notificationFrames;
// create the IAudioTrack
status_t status = createTrack(streamType, sampleRate, format, channelCount,
- frameCount, flags, sharedBuffer, output);
+ frameCount, flags, sharedBuffer, output, true);
if (status != NO_ERROR) {
return status;
@@ -238,10 +199,7 @@ status_t AudioTrack::set(
mMuted = false;
mActive = 0;
mCbf = cbf;
- mNotificationFrames = notificationFrames;
- mRemainingFrames = notificationFrames;
mUserData = user;
- mLatency = afLatency + (1000*mFrameCount) / sampleRate;
mLoopCount = 0;
mMarkerPosition = 0;
mMarkerReached = false;
@@ -281,7 +239,7 @@ int AudioTrack::channelCount() const
uint32_t AudioTrack::frameCount() const
{
- return mFrameCount;
+ return mCblk->frameCount;
}
int AudioTrack::frameSize() const
@@ -303,6 +261,7 @@ sp<IMemory>& AudioTrack::sharedBuffer()
void AudioTrack::start()
{
sp<AudioTrackThread> t = mAudioTrackThread;
+ status_t status;
LOGV("start %p", this);
if (t != 0) {
@@ -325,11 +284,18 @@ void AudioTrack::start()
setpriority(PRIO_PROCESS, 0, THREAD_PRIORITY_AUDIO_CLIENT);
}
- status_t status = mAudioTrack->start();
+ if (mCblk->flags & CBLK_INVALID_MSK) {
+ LOGW("start() track %p invalidated, creating a new one", this);
+ // no need to clear the invalid flag as this cblk will not be used anymore
+ // force new track creation
+ status = DEAD_OBJECT;
+ } else {
+ status = mAudioTrack->start();
+ }
if (status == DEAD_OBJECT) {
LOGV("start() dead IAudioTrack: creating a new one");
status = createTrack(mStreamType, mCblk->sampleRate, mFormat, mChannelCount,
- mFrameCount, mFlags, mSharedBuffer, getOutput());
+ mFrameCount, mFlags, mSharedBuffer, getOutput(), false);
if (status == NO_ERROR) {
status = mAudioTrack->start();
if (status == NO_ERROR) {
@@ -479,14 +445,14 @@ status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount
}
if (loopStart >= loopEnd ||
- loopEnd - loopStart > mFrameCount) {
- LOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, user %d", loopStart, loopEnd, loopCount, mFrameCount, cblk->user);
+ loopEnd - loopStart > cblk->frameCount) {
+ LOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, user %d", loopStart, loopEnd, loopCount, cblk->frameCount, cblk->user);
return BAD_VALUE;
}
- if ((mSharedBuffer != 0) && (loopEnd > mFrameCount)) {
+ if ((mSharedBuffer != 0) && (loopEnd > cblk->frameCount)) {
LOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, framecount %d",
- loopStart, loopEnd, mFrameCount);
+ loopStart, loopEnd, cblk->frameCount);
return BAD_VALUE;
}
@@ -566,7 +532,7 @@ status_t AudioTrack::setPosition(uint32_t position)
if (position > mCblk->user) return BAD_VALUE;
mCblk->server = position;
- mCblk->forceReady = 1;
+ mCblk->flags |= CBLK_FORCEREADY_ON;
return NO_ERROR;
}
@@ -586,7 +552,7 @@ status_t AudioTrack::reload()
flush();
- mCblk->stepUser(mFrameCount);
+ mCblk->stepUser(mCblk->frameCount);
return NO_ERROR;
}
@@ -607,7 +573,8 @@ status_t AudioTrack::createTrack(
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
- audio_io_handle_t output)
+ audio_io_handle_t output,
+ bool enforceFrameCount)
{
status_t status;
const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
@@ -616,6 +583,61 @@ status_t AudioTrack::createTrack(
return NO_INIT;
}
+ int afSampleRate;
+ if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
+ return NO_INIT;
+ }
+ int afFrameCount;
+ if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
+ return NO_INIT;
+ }
+ uint32_t afLatency;
+ if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
+ return NO_INIT;
+ }
+
+ mNotificationFramesAct = mNotificationFramesReq;
+ if (!AudioSystem::isLinearPCM(format)) {
+ if (sharedBuffer != 0) {
+ frameCount = sharedBuffer->size();
+ }
+ } else {
+ // Ensure that buffer depth covers at least audio hardware latency
+ uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);
+ if (minBufCount < 2) minBufCount = 2;
+
+ int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
+
+ if (sharedBuffer == 0) {
+ if (frameCount == 0) {
+ frameCount = minFrameCount;
+ }
+ if (mNotificationFramesAct == 0) {
+ mNotificationFramesAct = frameCount/2;
+ }
+ // Make sure that application is notified with sufficient margin
+ // before underrun
+ if (mNotificationFramesAct > (uint32_t)frameCount/2) {
+ mNotificationFramesAct = frameCount/2;
+ }
+ if (frameCount < minFrameCount) {
+ if (enforceFrameCount) {
+ LOGE("Invalid buffer size: minFrameCount %d, frameCount %d", minFrameCount, frameCount);
+ return BAD_VALUE;
+ } else {
+ frameCount = minFrameCount;
+ }
+ }
+ } else {
+ // Ensure that buffer alignment matches channelcount
+ if (((uint32_t)sharedBuffer->pointer() & (channelCount | 1)) != 0) {
+ LOGE("Invalid buffer alignement: address %p, channelCount %d", sharedBuffer->pointer(), channelCount);
+ return BAD_VALUE;
+ }
+ frameCount = sharedBuffer->size()/channelCount/sizeof(int16_t);
+ }
+ }
+
sp<IAudioTrack> track = audioFlinger->createTrack(getpid(),
streamType,
sampleRate,
@@ -641,20 +663,20 @@ status_t AudioTrack::createTrack(
mCblkMemory.clear();
mCblkMemory = cblk;
mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());
- mCblk->out = 1;
- // Update buffer size in case it has been limited by AudioFlinger during track creation
- mFrameCount = mCblk->frameCount;
+ mCblk->flags |= CBLK_DIRECTION_OUT;
if (sharedBuffer == 0) {
mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
} else {
mCblk->buffers = sharedBuffer->pointer();
// Force buffer full condition as data is already present in shared memory
- mCblk->stepUser(mFrameCount);
+ mCblk->stepUser(mCblk->frameCount);
}
mCblk->volumeLR = (int32_t(int16_t(mVolume[LEFT] * 0x1000)) << 16) | int16_t(mVolume[RIGHT] * 0x1000);
mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
mCblk->waitTimeMs = 0;
+ mRemainingFrames = mNotificationFramesAct;
+ mLatency = afLatency + (1000*mCblk->frameCount) / sampleRate;
return NO_ERROR;
}
@@ -685,8 +707,15 @@ status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
cblk->lock.unlock();
return WOULD_BLOCK;
}
-
- result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
+ if (!(cblk->flags & CBLK_INVALID_MSK)) {
+ result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
+ }
+ if (cblk->flags & CBLK_INVALID_MSK) {
+ LOGW("obtainBuffer() track %p invalidated, creating a new one", this);
+ // no need to clear the invalid flag as this cblk will not be used anymore
+ cblk->lock.unlock();
+ goto create_new_track;
+ }
if (__builtin_expect(result!=NO_ERROR, false)) {
cblk->waitTimeMs += waitTimeMs;
if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) {
@@ -700,8 +729,9 @@ status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
result = mAudioTrack->start();
if (result == DEAD_OBJECT) {
LOGW("obtainBuffer() dead IAudioTrack: creating a new one");
+create_new_track:
result = createTrack(mStreamType, cblk->sampleRate, mFormat, mChannelCount,
- mFrameCount, mFlags, mSharedBuffer, getOutput());
+ mFrameCount, mFlags, mSharedBuffer, getOutput(), false);
if (result == NO_ERROR) {
cblk = mCblk;
cblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
@@ -826,13 +856,13 @@ bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
// Manage underrun callback
if (mActive && (mCblk->framesReady() == 0)) {
- LOGV("Underrun user: %x, server: %x, flowControlFlag %d", mCblk->user, mCblk->server, mCblk->flowControlFlag);
- if (mCblk->flowControlFlag == 0) {
+ LOGV("Underrun user: %x, server: %x, flags %04x", mCblk->user, mCblk->server, mCblk->flags);
+ if ((mCblk->flags & CBLK_UNDERRUN_MSK) == CBLK_UNDERRUN_OFF) {
mCbf(EVENT_UNDERRUN, mUserData, 0);
if (mCblk->server == mCblk->frameCount) {
mCbf(EVENT_BUFFER_END, mUserData, 0);
}
- mCblk->flowControlFlag = 1;
+ mCblk->flags |= CBLK_UNDERRUN_ON;
if (mSharedBuffer != 0) return false;
}
}
@@ -932,7 +962,7 @@ bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
while (frames);
if (frames == 0) {
- mRemainingFrames = mNotificationFrames;
+ mRemainingFrames = mNotificationFramesAct;
} else {
mRemainingFrames = frames;
}
@@ -949,7 +979,7 @@ status_t AudioTrack::dump(int fd, const Vector<String16>& args) const
result.append(" AudioTrack::dump\n");
snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, mVolume[0], mVolume[1]);
result.append(buffer);
- snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, mChannelCount, mFrameCount);
+ snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, mChannelCount, mCblk->frameCount);
result.append(buffer);
snprintf(buffer, 255, " sample rate(%d), status(%d), muted(%d)\n", (mCblk == 0) ? 0 : mCblk->sampleRate, mStatus, mMuted);
result.append(buffer);
@@ -986,7 +1016,7 @@ audio_track_cblk_t::audio_track_cblk_t()
: lock(Mutex::SHARED), cv(Condition::SHARED), user(0), server(0),
userBase(0), serverBase(0), buffers(0), frameCount(0),
loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), volumeLR(0),
- flowControlFlag(1), forceReady(0)
+ flags(0)
{
}
@@ -996,7 +1026,7 @@ uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount)
u += frameCount;
// Ensure that user is never ahead of server for AudioRecord
- if (out) {
+ if (flags & CBLK_DIRECTION_MSK) {
// If stepServer() has been called once, switch to normal obtainBuffer() timeout period
if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) {
bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
@@ -1013,7 +1043,7 @@ uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount)
this->user = u;
// Clear flow control error condition as new data has been written/read to/from buffer.
- flowControlFlag = 0;
+ flags &= ~CBLK_UNDERRUN_MSK;
return u;
}
@@ -1038,7 +1068,7 @@ bool audio_track_cblk_t::stepServer(uint32_t frameCount)
uint32_t s = this->server;
s += frameCount;
- if (out) {
+ if (flags & CBLK_DIRECTION_MSK) {
// Mark that we have read the first buffer so that next time stepUser() is called
// we switch to normal obtainBuffer() timeout period
if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) {
@@ -1089,7 +1119,7 @@ uint32_t audio_track_cblk_t::framesAvailable_l()
uint32_t u = this->user;
uint32_t s = this->server;
- if (out) {
+ if (flags & CBLK_DIRECTION_MSK) {
uint32_t limit = (s < loopStart) ? s : loopStart;
return limit + frameCount - u;
} else {
@@ -1102,7 +1132,7 @@ uint32_t audio_track_cblk_t::framesReady()
uint32_t u = this->user;
uint32_t s = this->server;
- if (out) {
+ if (flags & CBLK_DIRECTION_MSK) {
if (u < loopEnd) {
return u - s;
} else {
generated by cgit v1.1 at 2025-08-15 07:21:28 +0000