diff options
| -rw-r--r-- | include/media/AudioResamplerPublic.h | 13 | ||||
| -rw-r--r-- | media/libmedia/AudioTrack.cpp | 82 | ||||
| -rw-r--r-- | services/audioflinger/AudioMixer.cpp | 50 | ||||
| -rw-r--r-- | services/audioflinger/AudioResamplerFirProcessNeon.h | 80 | ||||
| -rw-r--r-- | services/audioflinger/Threads.cpp | 31 |
5 files changed, 132 insertions, 124 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/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp index 0d4b358..836f550 100644 --- a/services/audioflinger/AudioMixer.cpp +++ b/services/audioflinger/AudioMixer.cpp @@ -341,11 +341,46 @@ AudioMixer::RemixBufferProvider::RemixBufferProvider(audio_channel_mask_t inputC ALOGV("RemixBufferProvider(%p)(%#x, %#x, %#x) %zu %zu", this, format, inputChannelMask, outputChannelMask, mInputChannels, mOutputChannels); - // TODO: consider channel representation in index array formulation - // We ignore channel representation, and just use the bits. - memcpy_by_index_array_initialization(mIdxAry, ARRAY_SIZE(mIdxAry), - audio_channel_mask_get_bits(outputChannelMask), - audio_channel_mask_get_bits(inputChannelMask)); + + const audio_channel_representation_t inputRepresentation = + audio_channel_mask_get_representation(inputChannelMask); + const audio_channel_representation_t outputRepresentation = + audio_channel_mask_get_representation(outputChannelMask); + const uint32_t inputBits = audio_channel_mask_get_bits(inputChannelMask); + const uint32_t outputBits = audio_channel_mask_get_bits(outputChannelMask); + + switch (inputRepresentation) { + case AUDIO_CHANNEL_REPRESENTATION_POSITION: + switch (outputRepresentation) { + case AUDIO_CHANNEL_REPRESENTATION_POSITION: + memcpy_by_index_array_initialization(mIdxAry, ARRAY_SIZE(mIdxAry), + outputBits, inputBits); + return; + case AUDIO_CHANNEL_REPRESENTATION_INDEX: + // TODO: output channel index mask not currently allowed + // fall through + default: + break; + } + break; + case AUDIO_CHANNEL_REPRESENTATION_INDEX: + switch (outputRepresentation) { + case AUDIO_CHANNEL_REPRESENTATION_POSITION: + memcpy_by_index_array_initialization_src_index(mIdxAry, ARRAY_SIZE(mIdxAry), + outputBits, inputBits); + return; + case AUDIO_CHANNEL_REPRESENTATION_INDEX: + // TODO: output channel index mask not currently allowed + // fall through + default: + break; + } + break; + default: + break; + } + LOG_ALWAYS_FATAL("invalid channel mask conversion from %#x to %#x", + inputChannelMask, outputChannelMask); } void AudioMixer::RemixBufferProvider::copyFrames(void *dst, const void *src, size_t frames) @@ -605,7 +640,10 @@ status_t AudioMixer::track_t::prepareForDownmix() && mMixerChannelMask == AUDIO_CHANNEL_OUT_STEREO)) { return NO_ERROR; } - if (DownmixerBufferProvider::isMultichannelCapable()) { + // DownmixerBufferProvider is only used for position masks. + if (audio_channel_mask_get_representation(channelMask) + == AUDIO_CHANNEL_REPRESENTATION_POSITION + && DownmixerBufferProvider::isMultichannelCapable()) { DownmixerBufferProvider* pDbp = new DownmixerBufferProvider(channelMask, mMixerChannelMask, AUDIO_FORMAT_PCM_16_BIT /* TODO: use mMixerInFormat, now only PCM 16 */, diff --git a/services/audioflinger/AudioResamplerFirProcessNeon.h b/services/audioflinger/AudioResamplerFirProcessNeon.h index d4fa7ad..29ff179 100644 --- a/services/audioflinger/AudioResamplerFirProcessNeon.h +++ b/services/audioflinger/AudioResamplerFirProcessNeon.h @@ -115,13 +115,13 @@ inline void ProcessL<2, 16>(int32_t* const out, "1: \n" - "vld2.16 {q2, q3}, [%[sP]] \n"// (3+0d) load 8 16-bits stereo samples - "vld2.16 {q5, q6}, [%[sN]]! \n"// (3) load 8 16-bits stereo samples + "vld2.16 {q2, q3}, [%[sP]] \n"// (3+0d) load 8 16-bits stereo frames + "vld2.16 {q5, q6}, [%[sN]]! \n"// (3) load 8 16-bits stereo frames "vld1.16 {q8}, [%[coefsP0]:128]! \n"// (1) load 8 16-bits coefs "vld1.16 {q10}, [%[coefsN0]:128]! \n"// (1) load 8 16-bits coefs - "vrev64.16 q2, q2 \n"// (1) reverse 8 frames of the left positive - "vrev64.16 q3, q3 \n"// (0 combines+) reverse right positive + "vrev64.16 q2, q2 \n"// (1) reverse 8 samples of positive left + "vrev64.16 q3, q3 \n"// (0 combines+) reverse positive right "vmlal.s16 q0, d4, d17 \n"// (1) multiply (reversed) samples left "vmlal.s16 q0, d5, d16 \n"// (1) multiply (reversed) samples left @@ -247,8 +247,8 @@ inline void Process<2, 16>(int32_t* const out, "1: \n" - "vld2.16 {q2, q3}, [%[sP]] \n"// (3+0d) load 8 16-bits stereo samples - "vld2.16 {q5, q6}, [%[sN]]! \n"// (3) load 8 16-bits stereo samples + "vld2.16 {q2, q3}, [%[sP]] \n"// (3+0d) load 8 16-bits stereo frames + "vld2.16 {q5, q6}, [%[sN]]! \n"// (3) load 8 16-bits stereo frames "vld1.16 {q8}, [%[coefsP0]:128]! \n"// (1) load 8 16-bits coefs "vld1.16 {q9}, [%[coefsP1]:128]! \n"// (1) load 8 16-bits coefs for interpolation "vld1.16 {q10}, [%[coefsN1]:128]! \n"// (1) load 8 16-bits coefs @@ -260,8 +260,8 @@ inline void Process<2, 16>(int32_t* const out, "vqrdmulh.s16 q9, q9, d2[0] \n"// (2) interpolate (step2) 1st set of coefs "vqrdmulh.s16 q11, q11, d2[0] \n"// (2) interpolate (step2) 2nd set of coefs - "vrev64.16 q2, q2 \n"// (1) reverse 8 frames of the left positive - "vrev64.16 q3, q3 \n"// (1) reverse 8 frames of the right positive + "vrev64.16 q2, q2 \n"// (1) reverse 8 samples of positive left + "vrev64.16 q3, q3 \n"// (1) reverse 8 samples of positive right "vadd.s16 q8, q8, q9 \n"// (1+1d) interpolate (step3) 1st set "vadd.s16 q10, q10, q11 \n"// (1+1d) interpolate (step3) 2nd set @@ -323,7 +323,7 @@ inline void ProcessL<1, 16>(int32_t* const out, "vld1.32 {q8, q9}, [%[coefsP0]:128]! \n"// load 8 32-bits coefs "vld1.32 {q10, q11}, [%[coefsN0]:128]! \n"// load 8 32-bits coefs - "vrev64.16 q2, q2 \n"// reverse 8 frames of the positive side + "vrev64.16 q2, q2 \n"// reverse 8 samples of the positive side "vshll.s16 q12, d4, #15 \n"// extend samples to 31 bits "vshll.s16 q13, d5, #15 \n"// extend samples to 31 bits @@ -331,10 +331,10 @@ inline void ProcessL<1, 16>(int32_t* const out, "vshll.s16 q14, d6, #15 \n"// extend samples to 31 bits "vshll.s16 q15, d7, #15 \n"// extend samples to 31 bits - "vqrdmulh.s32 q12, q12, q9 \n"// multiply samples by interpolated coef - "vqrdmulh.s32 q13, q13, q8 \n"// multiply samples by interpolated coef - "vqrdmulh.s32 q14, q14, q10 \n"// multiply samples by interpolated coef - "vqrdmulh.s32 q15, q15, q11 \n"// multiply samples by interpolated coef + "vqrdmulh.s32 q12, q12, q9 \n"// multiply samples + "vqrdmulh.s32 q13, q13, q8 \n"// multiply samples + "vqrdmulh.s32 q14, q14, q10 \n"// multiply samples + "vqrdmulh.s32 q15, q15, q11 \n"// multiply samples "vadd.s32 q0, q0, q12 \n"// accumulate result "vadd.s32 q13, q13, q14 \n"// accumulate result @@ -380,13 +380,13 @@ inline void ProcessL<2, 16>(int32_t* const out, "1: \n" - "vld2.16 {q2, q3}, [%[sP]] \n"// load 4 16-bits stereo samples - "vld2.16 {q5, q6}, [%[sN]]! \n"// load 4 16-bits stereo samples - "vld1.32 {q8, q9}, [%[coefsP0]:128]! \n"// load 4 32-bits coefs - "vld1.32 {q10, q11}, [%[coefsN0]:128]! \n"// load 4 32-bits coefs + "vld2.16 {q2, q3}, [%[sP]] \n"// load 8 16-bits stereo frames + "vld2.16 {q5, q6}, [%[sN]]! \n"// load 8 16-bits stereo frames + "vld1.32 {q8, q9}, [%[coefsP0]:128]! \n"// load 8 32-bits coefs + "vld1.32 {q10, q11}, [%[coefsN0]:128]! \n"// load 8 32-bits coefs - "vrev64.16 q2, q2 \n"// reverse 8 frames of the positive side - "vrev64.16 q3, q3 \n"// reverse 8 frames of the positive side + "vrev64.16 q2, q2 \n"// reverse 8 samples of positive left + "vrev64.16 q3, q3 \n"// reverse 8 samples of positive right "vshll.s16 q12, d4, #15 \n"// extend samples to 31 bits "vshll.s16 q13, d5, #15 \n"// extend samples to 31 bits @@ -394,15 +394,15 @@ inline void ProcessL<2, 16>(int32_t* const out, "vshll.s16 q14, d10, #15 \n"// extend samples to 31 bits "vshll.s16 q15, d11, #15 \n"// extend samples to 31 bits - "vqrdmulh.s32 q12, q12, q9 \n"// multiply samples by interpolated coef - "vqrdmulh.s32 q13, q13, q8 \n"// multiply samples by interpolated coef - "vqrdmulh.s32 q14, q14, q10 \n"// multiply samples by interpolated coef - "vqrdmulh.s32 q15, q15, q11 \n"// multiply samples by interpolated coef + "vqrdmulh.s32 q12, q12, q9 \n"// multiply samples by coef + "vqrdmulh.s32 q13, q13, q8 \n"// multiply samples by coef + "vqrdmulh.s32 q14, q14, q10 \n"// multiply samples by coef + "vqrdmulh.s32 q15, q15, q11 \n"// multiply samples by coef "vadd.s32 q0, q0, q12 \n"// accumulate result "vadd.s32 q13, q13, q14 \n"// accumulate result - "vadd.s32 q0, q0, q15 \n"// (+1) accumulate result - "vadd.s32 q0, q0, q13 \n"// (+1) accumulate result + "vadd.s32 q0, q0, q15 \n"// accumulate result + "vadd.s32 q0, q0, q13 \n"// accumulate result "vshll.s16 q12, d6, #15 \n"// extend samples to 31 bits "vshll.s16 q13, d7, #15 \n"// extend samples to 31 bits @@ -410,15 +410,15 @@ inline void ProcessL<2, 16>(int32_t* const out, "vshll.s16 q14, d12, #15 \n"// extend samples to 31 bits "vshll.s16 q15, d13, #15 \n"// extend samples to 31 bits - "vqrdmulh.s32 q12, q12, q9 \n"// multiply samples by interpolated coef - "vqrdmulh.s32 q13, q13, q8 \n"// multiply samples by interpolated coef - "vqrdmulh.s32 q14, q14, q10 \n"// multiply samples by interpolated coef - "vqrdmulh.s32 q15, q15, q11 \n"// multiply samples by interpolated coef + "vqrdmulh.s32 q12, q12, q9 \n"// multiply samples by coef + "vqrdmulh.s32 q13, q13, q8 \n"// multiply samples by coef + "vqrdmulh.s32 q14, q14, q10 \n"// multiply samples by coef + "vqrdmulh.s32 q15, q15, q11 \n"// multiply samples by coef "vadd.s32 q4, q4, q12 \n"// accumulate result "vadd.s32 q13, q13, q14 \n"// accumulate result - "vadd.s32 q4, q4, q15 \n"// (+1) accumulate result - "vadd.s32 q4, q4, q13 \n"// (+1) accumulate result + "vadd.s32 q4, q4, q15 \n"// accumulate result + "vadd.s32 q4, q4, q13 \n"// accumulate result "subs %[count], %[count], #8 \n"// update loop counter "sub %[sP], %[sP], #32 \n"// move pointer to next set of samples @@ -485,7 +485,7 @@ inline void Process<1, 16>(int32_t* const out, "vadd.s32 q10, q10, q14 \n"// interpolate (step3) "vadd.s32 q11, q11, q15 \n"// interpolate (step3) - "vrev64.16 q2, q2 \n"// reverse 8 frames of the positive side + "vrev64.16 q2, q2 \n"// reverse 8 samples of the positive side "vshll.s16 q12, d4, #15 \n"// extend samples to 31 bits "vshll.s16 q13, d5, #15 \n"// extend samples to 31 bits @@ -549,8 +549,8 @@ inline void Process<2, 16>(int32_t* const out, "1: \n" - "vld2.16 {q2, q3}, [%[sP]] \n"// load 4 16-bits stereo samples - "vld2.16 {q5, q6}, [%[sN]]! \n"// load 4 16-bits stereo samples + "vld2.16 {q2, q3}, [%[sP]] \n"// load 8 16-bits stereo frames + "vld2.16 {q5, q6}, [%[sN]]! \n"// load 8 16-bits stereo frames "vld1.32 {q8, q9}, [%[coefsP0]:128]! \n"// load 8 32-bits coefs "vld1.32 {q12, q13}, [%[coefsP1]:128]! \n"// load 8 32-bits coefs "vld1.32 {q10, q11}, [%[coefsN1]:128]! \n"// load 8 32-bits coefs @@ -571,8 +571,8 @@ inline void Process<2, 16>(int32_t* const out, "vadd.s32 q10, q10, q14 \n"// interpolate (step3) "vadd.s32 q11, q11, q15 \n"// interpolate (step3) - "vrev64.16 q2, q2 \n"// reverse 8 frames of the positive side - "vrev64.16 q3, q3 \n"// reverse 8 frames of the positive side + "vrev64.16 q2, q2 \n"// reverse 8 samples of positive left + "vrev64.16 q3, q3 \n"// reverse 8 samples of positive right "vshll.s16 q12, d4, #15 \n"// extend samples to 31 bits "vshll.s16 q13, d5, #15 \n"// extend samples to 31 bits @@ -587,8 +587,8 @@ inline void Process<2, 16>(int32_t* const out, "vadd.s32 q0, q0, q12 \n"// accumulate result "vadd.s32 q13, q13, q14 \n"// accumulate result - "vadd.s32 q0, q0, q15 \n"// (+1) accumulate result - "vadd.s32 q0, q0, q13 \n"// (+1) accumulate result + "vadd.s32 q0, q0, q15 \n"// accumulate result + "vadd.s32 q0, q0, q13 \n"// accumulate result "vshll.s16 q12, d6, #15 \n"// extend samples to 31 bits "vshll.s16 q13, d7, #15 \n"// extend samples to 31 bits @@ -603,8 +603,8 @@ inline void Process<2, 16>(int32_t* const out, "vadd.s32 q4, q4, q12 \n"// accumulate result "vadd.s32 q13, q13, q14 \n"// accumulate result - "vadd.s32 q4, q4, q15 \n"// (+1) accumulate result - "vadd.s32 q4, q4, q13 \n"// (+1) accumulate result + "vadd.s32 q4, q4, q15 \n"// accumulate result + "vadd.s32 q4, q4, q13 \n"// accumulate result "subs %[count], %[count], #8 \n"// update loop counter "sub %[sP], %[sP], #32 \n"// move pointer to next set of samples diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp index 384bd25..40ab0af 100644 --- a/services/audioflinger/Threads.cpp +++ b/services/audioflinger/Threads.cpp @@ -174,18 +174,6 @@ 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; @@ -1497,20 +1485,25 @@ 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; |