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/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();
}
}