Allow duplicating thread to use native mixing audio format
Update OutputTrack to take variable formats.
Previously conversion to AUDIO_FORMAT_PCM_16_BIT was required.
Change-Id: I4a96a60ddd8d8dfe651405a0bcd4f98c89bc1ade
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index ee48276..902d5e4 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -255,7 +255,7 @@
class Buffer : public AudioBufferProvider::Buffer {
public:
- int16_t *mBuffer;
+ void *mBuffer;
};
OutputTrack(PlaybackThread *thread,
@@ -271,7 +271,7 @@
AudioSystem::SYNC_EVENT_NONE,
int triggerSession = 0);
virtual void stop();
- bool write(int16_t* data, uint32_t frames);
+ bool write(void* data, uint32_t frames);
bool bufferQueueEmpty() const { return mBufferQueue.size() == 0; }
bool isActive() const { return mActive; }
const wp<ThreadBase>& thread() const { return mThread; }
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index a7ee38b..4225161 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -172,6 +172,18 @@
// and that all "fast" AudioRecord clients read from. In either case, the size can be small.
static const size_t kRecordThreadReadOnlyHeapSize = 0x2000;
+// Returns the source frames needed to resample to destination frames. This is not a precise
+// value and depends on the resampler (and possibly how it handles rounding internally).
+// If srcSampleRate and dstSampleRate are equal, then it returns destination frames, which
+// may not be a true if the resampler is asynchronous.
+static inline size_t sourceFramesNeeded(
+ uint32_t srcSampleRate, size_t dstFramesRequired, uint32_t dstSampleRate) {
+ // +1 for rounding - always do this even if matched ratio
+ // +1 for additional sample needed for interpolation
+ return srcSampleRate == dstSampleRate ? dstFramesRequired :
+ size_t((uint64_t)dstFramesRequired * srcSampleRate / dstSampleRate + 1 + 1);
+}
+
// ----------------------------------------------------------------------------
static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT;
@@ -3453,8 +3465,7 @@
if (sr == mSampleRate) {
desiredFrames = mNormalFrameCount;
} else {
- // +1 for rounding and +1 for additional sample needed for interpolation
- desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1;
+ desiredFrames = sourceFramesNeeded(sr, mNormalFrameCount, mSampleRate);
// add frames already consumed but not yet released by the resampler
// because mAudioTrackServerProxy->framesReady() will include these frames
desiredFrames += mAudioMixer->getUnreleasedFrames(track->name());
@@ -4876,16 +4887,8 @@
ssize_t AudioFlinger::DuplicatingThread::threadLoop_write()
{
- // We convert the duplicating thread format to AUDIO_FORMAT_PCM_16_BIT
- // for delivery downstream as needed. This in-place conversion is safe as
- // AUDIO_FORMAT_PCM_16_BIT is smaller than any other supported format
- // (AUDIO_FORMAT_PCM_8_BIT is not allowed here).
- if (mFormat != AUDIO_FORMAT_PCM_16_BIT) {
- memcpy_by_audio_format(mSinkBuffer, AUDIO_FORMAT_PCM_16_BIT,
- mSinkBuffer, mFormat, writeFrames * mChannelCount);
- }
for (size_t i = 0; i < outputTracks.size(); i++) {
- outputTracks[i]->write(reinterpret_cast<int16_t*>(mSinkBuffer), writeFrames);
+ outputTracks[i]->write(mSinkBuffer, writeFrames);
}
mStandby = false;
return (ssize_t)mSinkBufferSize;
@@ -4912,25 +4915,26 @@
void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
{
Mutex::Autolock _l(mLock);
- // FIXME explain this formula
- size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate();
- // OutputTrack is forced to AUDIO_FORMAT_PCM_16_BIT regardless of mFormat
- // due to current usage case and restrictions on the AudioBufferProvider.
- // Actual buffer conversion is done in threadLoop_write().
- //
- // TODO: This may change in the future, depending on multichannel
- // (and non int16_t*) support on AF::PlaybackThread::OutputTrack
- OutputTrack *outputTrack = new OutputTrack(thread,
+ // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass.
+ // Adjust for thread->sampleRate() to determine minimum buffer frame count.
+ // Then triple buffer because Threads do not run synchronously and may not be clock locked.
+ const size_t frameCount =
+ 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate());
+ // TODO: Consider asynchronous sample rate conversion to handle clock disparity
+ // from different OutputTracks and their associated MixerThreads (e.g. one may
+ // nearly empty and the other may be dropping data).
+
+ sp<OutputTrack> outputTrack = new OutputTrack(thread,
this,
mSampleRate,
- AUDIO_FORMAT_PCM_16_BIT,
+ mFormat,
mChannelMask,
frameCount,
IPCThreadState::self()->getCallingUid());
if (outputTrack->cblk() != NULL) {
thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f);
mOutputTracks.add(outputTrack);
- ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread);
+ ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread);
updateWaitTime_l();
}
}
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 1e750bd..7757ea2 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -1710,14 +1710,13 @@
mActive = false;
}
-bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames)
+bool AudioFlinger::PlaybackThread::OutputTrack::write(void* data, uint32_t frames)
{
Buffer *pInBuffer;
Buffer inBuffer;
- uint32_t channelCount = mChannelCount;
bool outputBufferFull = false;
inBuffer.frameCount = frames;
- inBuffer.i16 = data;
+ inBuffer.raw = data;
uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
@@ -1727,13 +1726,17 @@
if (thread != 0) {
MixerThread *mixerThread = (MixerThread *)thread.get();
if (mFrameCount > frames) {
+ // For the first write after being inactive, ensure that we have
+ // enough frames to fill mFrameCount (which should be multiples of
+ // the minimum buffer requirements of the downstream MixerThread).
+ // This provides enough frames for the downstream mixer to begin
+ // (see AudioFlinger::PlaybackThread::Track::isReady()).
if (mBufferQueue.size() < kMaxOverFlowBuffers) {
uint32_t startFrames = (mFrameCount - frames);
pInBuffer = new Buffer;
- pInBuffer->mBuffer = new int16_t[startFrames * channelCount];
+ pInBuffer->mBuffer = calloc(1, startFrames * mFrameSize);
pInBuffer->frameCount = startFrames;
- pInBuffer->i16 = pInBuffer->mBuffer;
- memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t));
+ pInBuffer->raw = pInBuffer->mBuffer;
mBufferQueue.add(pInBuffer);
} else {
ALOGW("OutputTrack::write() %p no more buffers in queue", this);
@@ -1774,20 +1777,20 @@
uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount :
pInBuffer->frameCount;
- memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
+ memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * mFrameSize);
Proxy::Buffer buf;
buf.mFrameCount = outFrames;
buf.mRaw = NULL;
mClientProxy->releaseBuffer(&buf);
pInBuffer->frameCount -= outFrames;
- pInBuffer->i16 += outFrames * channelCount;
+ pInBuffer->raw = (int8_t *)pInBuffer->raw + outFrames * mFrameSize;
mOutBuffer.frameCount -= outFrames;
- mOutBuffer.i16 += outFrames * channelCount;
+ mOutBuffer.raw = (int8_t *)mOutBuffer.raw + outFrames * mFrameSize;
if (pInBuffer->frameCount == 0) {
if (mBufferQueue.size()) {
mBufferQueue.removeAt(0);
- delete [] pInBuffer->mBuffer;
+ free(pInBuffer->mBuffer);
delete pInBuffer;
ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this,
mThread.unsafe_get(), mBufferQueue.size());
@@ -1803,11 +1806,10 @@
if (thread != 0 && !thread->standby()) {
if (mBufferQueue.size() < kMaxOverFlowBuffers) {
pInBuffer = new Buffer;
- pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount];
+ pInBuffer->mBuffer = malloc(inBuffer.frameCount * mFrameSize);
pInBuffer->frameCount = inBuffer.frameCount;
- pInBuffer->i16 = pInBuffer->mBuffer;
- memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount *
- sizeof(int16_t));
+ pInBuffer->raw = pInBuffer->mBuffer;
+ memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * mFrameSize);
mBufferQueue.add(pInBuffer);
ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this,
mThread.unsafe_get(), mBufferQueue.size());
@@ -1818,23 +1820,10 @@
}
}
- // Calling write() with a 0 length buffer, means that no more data will be written:
- // If no more buffers are pending, fill output track buffer to make sure it is started
- // by output mixer.
- if (frames == 0 && mBufferQueue.size() == 0) {
- // FIXME borken, replace by getting framesReady() from proxy
- size_t user = 0; // was mCblk->user
- if (user < mFrameCount) {
- frames = mFrameCount - user;
- pInBuffer = new Buffer;
- pInBuffer->mBuffer = new int16_t[frames * channelCount];
- pInBuffer->frameCount = frames;
- pInBuffer->i16 = pInBuffer->mBuffer;
- memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t));
- mBufferQueue.add(pInBuffer);
- } else if (mActive) {
- stop();
- }
+ // Calling write() with a 0 length buffer means that no more data will be written:
+ // We rely on stop() to set the appropriate flags to allow the remaining frames to play out.
+ if (frames == 0 && mBufferQueue.size() == 0 && mActive) {
+ stop();
}
return outputBufferFull;
@@ -1860,7 +1849,7 @@
for (size_t i = 0; i < size; i++) {
Buffer *pBuffer = mBufferQueue.itemAt(i);
- delete [] pBuffer->mBuffer;
+ free(pBuffer->mBuffer);
delete pBuffer;
}
mBufferQueue.clear();