diff options
| -rw-r--r-- | include/media/AudioResamplerPublic.h | 13 | ||||
| -rw-r--r-- | media/libmedia/AudioTrack.cpp | 82 | ||||
| -rw-r--r-- | services/audioflinger/Threads.cpp | 31 |
3 files changed, 48 insertions, 78 deletions
diff --git a/include/media/AudioResamplerPublic.h b/include/media/AudioResamplerPublic.h index 97847a0..b705efa 100644 --- a/include/media/AudioResamplerPublic.h +++ b/include/media/AudioResamplerPublic.h @@ -26,4 +26,17 @@ // TODO: replace with an API #define AUDIO_RESAMPLER_DOWN_RATIO_MAX 256 +// 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). +// Nevertheless, this should be an upper bound on the requirements of the resampler. +// If srcSampleRate and dstSampleRate are equal, then it returns destination frames, which +// may not be 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 (resampler may use phases not ratio) + // +1 for additional sample needed for interpolation + return srcSampleRate == dstSampleRate ? dstFramesRequired : + size_t((uint64_t)dstFramesRequired * srcSampleRate / dstSampleRate + 1 + 1); +} + #endif // ANDROID_AUDIO_RESAMPLER_PUBLIC_H diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp index d4bacc0..1d5fc95 100644 --- a/media/libmedia/AudioTrack.cpp +++ b/media/libmedia/AudioTrack.cpp @@ -66,12 +66,11 @@ status_t AudioTrack::getMinFrameCount( return BAD_VALUE; } - // FIXME merge with similar code in createTrack_l(), except we're missing - // some information here that is available in createTrack_l(): + // FIXME handle in server, like createTrack_l(), possible missing info: // audio_io_handle_t output // audio_format_t format // audio_channel_mask_t channelMask - // audio_output_flags_t flags + // audio_output_flags_t flags (FAST) uint32_t afSampleRate; status_t status; status = AudioSystem::getOutputSamplingRate(&afSampleRate, streamType); @@ -101,16 +100,16 @@ status_t AudioTrack::getMinFrameCount( minBufCount = 2; } - *frameCount = (sampleRate == 0) ? afFrameCount * minBufCount : - afFrameCount * minBufCount * uint64_t(sampleRate) / afSampleRate; - // The formula above should always produce a non-zero value, but return an error - // in the unlikely event that it does not, as that's part of the API contract. + *frameCount = minBufCount * sourceFramesNeeded(sampleRate, afFrameCount, afSampleRate); + // The formula above should always produce a non-zero value under normal circumstances: + // AudioTrack.SAMPLE_RATE_HZ_MIN <= sampleRate <= AudioTrack.SAMPLE_RATE_HZ_MAX. + // Return error in the unlikely event that it does not, as that's part of the API contract. if (*frameCount == 0) { - ALOGE("AudioTrack::getMinFrameCount failed for streamType %d, sampleRate %d", + ALOGE("AudioTrack::getMinFrameCount failed for streamType %d, sampleRate %u", streamType, sampleRate); return BAD_VALUE; } - ALOGV("getMinFrameCount=%zu: afFrameCount=%zu, minBufCount=%d, afSampleRate=%d, afLatency=%d", + ALOGV("getMinFrameCount=%zu: afFrameCount=%zu, minBufCount=%u, afSampleRate=%u, afLatency=%u", *frameCount, afFrameCount, minBufCount, afSampleRate, afLatency); return NO_ERROR; } @@ -1015,11 +1014,9 @@ status_t AudioTrack::createTrack_l() // The client's AudioTrack buffer is divided into n parts for purpose of wakeup by server, where // n = 1 fast track with single buffering; nBuffering is ignored // n = 2 fast track with double buffering - // n = 2 normal track, no sample rate conversion - // n = 3 normal track, with sample rate conversion - // (pessimistic; some non-1:1 conversion ratios don't actually need triple-buffering) - // n > 3 very high latency or very small notification interval; nBuffering is ignored - const uint32_t nBuffering = (mSampleRate == afSampleRate) ? 2 : 3; + // n = 2 normal track, (including those with sample rate conversion) + // n >= 3 very high latency or very small notification interval (unused). + const uint32_t nBuffering = 2; mNotificationFramesAct = mNotificationFramesReq; @@ -1060,39 +1057,9 @@ status_t AudioTrack::createTrack_l() // But when initializing a shared buffer AudioTrack via set(), // there _is_ a frameCount parameter. We silently ignore it. frameCount = mSharedBuffer->size() / mFrameSize; - - } else if (!(mFlags & AUDIO_OUTPUT_FLAG_FAST)) { - - // FIXME move these calculations and associated checks to server - - // Ensure that buffer depth covers at least audio hardware latency - uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate); - ALOGV("afFrameCount=%zu, minBufCount=%d, afSampleRate=%u, afLatency=%d", - afFrameCount, minBufCount, afSampleRate, afLatency); - if (minBufCount <= nBuffering) { - minBufCount = nBuffering; - } - - size_t minFrameCount = afFrameCount * minBufCount * uint64_t(mSampleRate) / afSampleRate; - ALOGV("minFrameCount: %zu, afFrameCount=%zu, minBufCount=%d, sampleRate=%u, afSampleRate=%u" - ", afLatency=%d", - minFrameCount, afFrameCount, minBufCount, mSampleRate, afSampleRate, afLatency); - - if (frameCount == 0) { - frameCount = minFrameCount; - } else if (frameCount < minFrameCount) { - // not ALOGW because it happens all the time when playing key clicks over A2DP - ALOGV("Minimum buffer size corrected from %zu to %zu", - frameCount, minFrameCount); - frameCount = minFrameCount; - } - // Make sure that application is notified with sufficient margin before underrun - if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { - mNotificationFramesAct = frameCount/nBuffering; - } - } else { - // For fast tracks, the frame count calculations and checks are done by server + // For fast and normal streaming tracks, + // the frame count calculations and checks are done by server } IAudioFlinger::track_flags_t trackFlags = IAudioFlinger::TRACK_DEFAULT; @@ -1175,23 +1142,10 @@ status_t AudioTrack::createTrack_l() if (trackFlags & IAudioFlinger::TRACK_FAST) { ALOGV("AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu", frameCount); mAwaitBoost = true; - if (mSharedBuffer == 0) { - // Theoretically double-buffering is not required for fast tracks, - // due to tighter scheduling. But in practice, to accommodate kernels with - // scheduling jitter, and apps with computation jitter, we use double-buffering. - if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { - mNotificationFramesAct = frameCount/nBuffering; - } - } } else { ALOGV("AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu", frameCount); // once denied, do not request again if IAudioTrack is re-created mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST); - if (mSharedBuffer == 0) { - if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount/nBuffering) { - mNotificationFramesAct = frameCount/nBuffering; - } - } } } if (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { @@ -1214,6 +1168,16 @@ status_t AudioTrack::createTrack_l() //return NO_INIT; } } + // Make sure that application is notified with sufficient margin before underrun + if (mSharedBuffer == 0 && audio_is_linear_pcm(mFormat)) { + // Theoretically double-buffering is not required for fast tracks, + // due to tighter scheduling. But in practice, to accommodate kernels with + // scheduling jitter, and apps with computation jitter, we use double-buffering + // for fast tracks just like normal streaming tracks. + if (mNotificationFramesAct == 0 || mNotificationFramesAct > frameCount / nBuffering) { + mNotificationFramesAct = frameCount / nBuffering; + } + } // We retain a copy of the I/O handle, but don't own the reference mOutput = output; 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; |