diff options
Diffstat (limited to 'services/audioflinger/Threads.cpp')
-rw-r--r-- | services/audioflinger/Threads.cpp | 31 |
1 files changed, 12 insertions, 19 deletions
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp index 15dd408..8fcc09c 100644 --- a/services/audioflinger/Threads.cpp +++ b/services/audioflinger/Threads.cpp @@ -172,18 +172,6 @@ static int sFastTrackMultiplier = kFastTrackMultiplier; // 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; @@ -1495,20 +1483,25 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrac audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); *flags &= ~IAudioFlinger::TRACK_FAST; - // For compatibility with AudioTrack calculation, buffer depth is forced - // to be at least 2 x the normal mixer frame count and cover audio hardware latency. - // This is probably too conservative, but legacy application code may depend on it. - // If you change this calculation, also review the start threshold which is related. + } + } + // For normal PCM streaming tracks, update minimum frame count. + // For compatibility with AudioTrack calculation, buffer depth is forced + // to be at least 2 x the normal mixer frame count and cover audio hardware latency. + // This is probably too conservative, but legacy application code may depend on it. + // If you change this calculation, also review the start threshold which is related. + if (!(*flags & IAudioFlinger::TRACK_FAST) + && audio_is_linear_pcm(format) && sharedBuffer == 0) { uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); if (minBufCount < 2) { minBufCount = 2; } - size_t minFrameCount = mNormalFrameCount * minBufCount; - if (frameCount < minFrameCount) { + size_t minFrameCount = + minBufCount * sourceFramesNeeded(sampleRate, mNormalFrameCount, mSampleRate); + if (frameCount < minFrameCount) { // including frameCount == 0 frameCount = minFrameCount; } - } } *pFrameCount = frameCount; |