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author | Andy Hung <hunga@google.com> | 2015-02-28 00:38:24 +0000 |
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committer | Android (Google) Code Review <android-gerrit@google.com> | 2015-02-28 00:38:25 +0000 |
commit | d4bd93b22ab9f7dfa32dfc6d48e008db3ea9c634 (patch) | |
tree | 6e1effbb661cd90a1c09aed0db9cc44cedaf1544 /media | |
parent | 9060199701c97b290aa4a4184d147e0311ab8897 (diff) | |
parent | 0e48d25606c82def035ad10a5b3923767a765cdd (diff) | |
download | frameworks_av-d4bd93b22ab9f7dfa32dfc6d48e008db3ea9c634.zip frameworks_av-d4bd93b22ab9f7dfa32dfc6d48e008db3ea9c634.tar.gz frameworks_av-d4bd93b22ab9f7dfa32dfc6d48e008db3ea9c634.tar.bz2 |
Merge "Change AudioTrack resampling buffers from 3 to 2"
Diffstat (limited to 'media')
-rw-r--r-- | media/libmedia/AudioTrack.cpp | 82 |
1 files changed, 23 insertions, 59 deletions
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; |