From 717007429a50c02d2acc704a8c1ebbe6760a2c22 Mon Sep 17 00:00:00 2001 From: Andy Hung Date: Mon, 2 Jun 2014 18:54:08 -0700 Subject: Update resampler to fetch exactly the frames needed This avoids unnecessary overfetching/releasing, which could cause problems with the upstream AudioBufferProvider. A one input sample delay is placed on the stream compared with the previous implementation. This change only affects the Dynamic resampler. Change-Id: Ic7fcff130e0081b4724cfb5a00dc8e8b4a8b1af3 --- services/audioflinger/AudioResamplerDyn.cpp | 56 ++++++++++++++++++----------- 1 file changed, 36 insertions(+), 20 deletions(-) (limited to 'services/audioflinger/AudioResamplerDyn.cpp') diff --git a/services/audioflinger/AudioResamplerDyn.cpp b/services/audioflinger/AudioResamplerDyn.cpp index a4446a4..318eb57 100644 --- a/services/audioflinger/AudioResamplerDyn.cpp +++ b/services/audioflinger/AudioResamplerDyn.cpp @@ -460,9 +460,15 @@ void AudioResamplerDyn::resample(TO* out, size_t outFrameCount, const uint32_t phaseIncrement = mPhaseIncrement; size_t outputIndex = 0; size_t outputSampleCount = outFrameCount * 2; // stereo output - size_t inFrameCount = getInFrameCountRequired(outFrameCount) + (phaseFraction != 0); - ALOG_ASSERT(0 < inFrameCount && inFrameCount < (1U << 31)); const uint32_t phaseWrapLimit = c.mL << c.mShift; + size_t inFrameCount = (phaseIncrement * (uint64_t)outFrameCount + phaseFraction) + / phaseWrapLimit; + // sanity check that inFrameCount is in signed 32 bit integer range. + ALOG_ASSERT(0 <= inFrameCount && inFrameCount < (1U << 31)); + + //ALOGV("inFrameCount:%d outFrameCount:%d" + // " phaseIncrement:%u phaseFraction:%u phaseWrapLimit:%u", + // inFrameCount, outFrameCount, phaseIncrement, phaseFraction, phaseWrapLimit); // NOTE: be very careful when modifying the code here. register // pressure is very high and a small change might cause the compiler @@ -472,10 +478,17 @@ void AudioResamplerDyn::resample(TO* out, size_t outFrameCount, // the following logic is a bit convoluted to keep the main processing loop // as tight as possible with register allocation. while (outputIndex < outputSampleCount) { - // buffer is empty, fetch a new one - while (mBuffer.frameCount == 0) { + //ALOGV("LOOP: inFrameCount:%d outputIndex:%d outFrameCount:%d" + // " phaseFraction:%u phaseWrapLimit:%u", + // inFrameCount, outputIndex, outFrameCount, phaseFraction, phaseWrapLimit); + + // check inputIndex overflow + ALOG_ASSERT(inputIndex <= mBuffer.frameCount, "inputIndex%d > frameCount%d", + inputIndex, mBuffer.frameCount); + // Buffer is empty, fetch a new one if necessary (inFrameCount > 0). + // We may not fetch a new buffer if the existing data is sufficient. + while (mBuffer.frameCount == 0 && inFrameCount > 0) { mBuffer.frameCount = inFrameCount; - ALOG_ASSERT(inFrameCount > 0); provider->getNextBuffer(&mBuffer, calculateOutputPTS(outputIndex / 2)); if (mBuffer.raw == NULL) { @@ -486,9 +499,9 @@ void AudioResamplerDyn::resample(TO* out, size_t outFrameCount, mInBuffer.template readAdvance( impulse, c.mHalfNumCoefs, reinterpret_cast(mBuffer.raw), inputIndex); + inputIndex++; phaseFraction -= phaseWrapLimit; while (phaseFraction >= phaseWrapLimit) { - inputIndex++; if (inputIndex >= mBuffer.frameCount) { inputIndex = 0; provider->releaseBuffer(&mBuffer); @@ -497,6 +510,7 @@ void AudioResamplerDyn::resample(TO* out, size_t outFrameCount, mInBuffer.template readAdvance( impulse, c.mHalfNumCoefs, reinterpret_cast(mBuffer.raw), inputIndex); + inputIndex++; phaseFraction -= phaseWrapLimit; } } @@ -507,9 +521,6 @@ void AudioResamplerDyn::resample(TO* out, size_t outFrameCount, const int halfNumCoefs = c.mHalfNumCoefs; const TO* const volumeSimd = mVolumeSimd; - // reread the last input in. - mInBuffer.template readAgain(impulse, halfNumCoefs, in, inputIndex); - // main processing loop while (CC_LIKELY(outputIndex < outputSampleCount)) { // caution: fir() is inlined and may be large. @@ -518,6 +529,10 @@ void AudioResamplerDyn::resample(TO* out, size_t outFrameCount, // from the input samples in impulse[-halfNumCoefs+1]... impulse[halfNumCoefs] // from the polyphase filter of (phaseFraction / phaseWrapLimit) in coefs. // + //ALOGV("LOOP2: inFrameCount:%d outputIndex:%d outFrameCount:%d" + // " phaseFraction:%u phaseWrapLimit:%u", + // inFrameCount, outputIndex, outFrameCount, phaseFraction, phaseWrapLimit); + ALOG_ASSERT(phaseFraction < phaseWrapLimit); fir( &out[outputIndex], phaseFraction, phaseWrapLimit, @@ -527,17 +542,20 @@ void AudioResamplerDyn::resample(TO* out, size_t outFrameCount, phaseFraction += phaseIncrement; while (phaseFraction >= phaseWrapLimit) { - inputIndex++; if (inputIndex >= frameCount) { goto done; // need a new buffer } mInBuffer.template readAdvance(impulse, halfNumCoefs, in, inputIndex); + inputIndex++; phaseFraction -= phaseWrapLimit; } } done: - // often arrives here when input buffer runs out - if (inputIndex >= frameCount) { + // We arrive here when we're finished or when the input buffer runs out. + // Regardless we need to release the input buffer if we've acquired it. + if (inputIndex > 0) { // we've acquired a buffer (alternatively could check frameCount) + ALOG_ASSERT(inputIndex == frameCount, "inputIndex(%d) != frameCount(%d)", + inputIndex, frameCount); // must have been fully read. inputIndex = 0; provider->releaseBuffer(&mBuffer); ALOG_ASSERT(mBuffer.frameCount == 0); @@ -545,14 +563,12 @@ done: } resample_exit: - // Release frames to avoid the count being inaccurate for pts timing. - // TODO: Avoid this extra check by making fetch count exact. This is tricky - // due to the overfetching mechanism which loads unnecessarily when - // mBuffer.frameCount == 0. - if (inputIndex) { - mBuffer.frameCount = inputIndex; - provider->releaseBuffer(&mBuffer); - } + // inputIndex must be zero in all three cases: + // (1) the buffer never was been acquired; (2) the buffer was + // released at "done:"; or (3) getNextBuffer() failed. + ALOG_ASSERT(inputIndex == 0, "Releasing: inputindex:%d frameCount:%d phaseFraction:%u", + inputIndex, mBuffer.frameCount, phaseFraction); + ALOG_ASSERT(mBuffer.frameCount == 0); // there must be no frames in the buffer mInBuffer.setImpulse(impulse); mPhaseFraction = phaseFraction; } -- cgit v1.1