diff options
Diffstat (limited to 'services/audioflinger/AudioResamplerDyn.cpp')
| -rw-r--r-- | services/audioflinger/AudioResamplerDyn.cpp | 56 |
1 files changed, 41 insertions, 15 deletions
diff --git a/services/audioflinger/AudioResamplerDyn.cpp b/services/audioflinger/AudioResamplerDyn.cpp index 3abe8fd..318eb57 100644 --- a/services/audioflinger/AudioResamplerDyn.cpp +++ b/services/audioflinger/AudioResamplerDyn.cpp @@ -455,13 +455,20 @@ void AudioResamplerDyn<TC, TI, TO>::resample(TO* out, size_t outFrameCount, const Constants& c(mConstants); const TC* const coefs = mConstants.mFirCoefs; TI* impulse = mInBuffer.getImpulse(); - size_t inputIndex = mInputIndex; + size_t inputIndex = 0; uint32_t phaseFraction = mPhaseFraction; const uint32_t phaseIncrement = mPhaseIncrement; size_t outputIndex = 0; size_t outputSampleCount = outFrameCount * 2; // stereo output - size_t inFrameCount = getInFrameCountRequired(outFrameCount); 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 @@ -471,29 +478,39 @@ void AudioResamplerDyn<TC, TI, TO>::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; provider->getNextBuffer(&mBuffer, calculateOutputPTS(outputIndex / 2)); if (mBuffer.raw == NULL) { goto resample_exit; } + inFrameCount -= mBuffer.frameCount; if (phaseFraction >= phaseWrapLimit) { // read in data mInBuffer.template readAdvance<CHANNELS>( impulse, c.mHalfNumCoefs, reinterpret_cast<TI*>(mBuffer.raw), inputIndex); + inputIndex++; phaseFraction -= phaseWrapLimit; while (phaseFraction >= phaseWrapLimit) { - inputIndex++; if (inputIndex >= mBuffer.frameCount) { - inputIndex -= mBuffer.frameCount; + inputIndex = 0; provider->releaseBuffer(&mBuffer); break; } mInBuffer.template readAdvance<CHANNELS>( impulse, c.mHalfNumCoefs, reinterpret_cast<TI*>(mBuffer.raw), inputIndex); + inputIndex++; phaseFraction -= phaseWrapLimit; } } @@ -504,9 +521,6 @@ void AudioResamplerDyn<TC, TI, TO>::resample(TO* out, size_t outFrameCount, const int halfNumCoefs = c.mHalfNumCoefs; const TO* const volumeSimd = mVolumeSimd; - // reread the last input in. - mInBuffer.template readAgain<CHANNELS>(impulse, halfNumCoefs, in, inputIndex); - // main processing loop while (CC_LIKELY(outputIndex < outputSampleCount)) { // caution: fir() is inlined and may be large. @@ -515,6 +529,10 @@ void AudioResamplerDyn<TC, TI, TO>::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<CHANNELS, LOCKED, STRIDE>( &out[outputIndex], phaseFraction, phaseWrapLimit, @@ -524,26 +542,34 @@ void AudioResamplerDyn<TC, TI, TO>::resample(TO* out, size_t outFrameCount, phaseFraction += phaseIncrement; while (phaseFraction >= phaseWrapLimit) { - inputIndex++; if (inputIndex >= frameCount) { goto done; // need a new buffer } mInBuffer.template readAdvance<CHANNELS>(impulse, halfNumCoefs, in, inputIndex); + inputIndex++; phaseFraction -= phaseWrapLimit; } } done: - // often arrives here when input buffer runs out - if (inputIndex >= frameCount) { - 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); - // mBuffer.frameCount MUST be zero here. + ALOG_ASSERT(mBuffer.frameCount == 0); } } resample_exit: + // 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); - mInputIndex = inputIndex; mPhaseFraction = phaseFraction; } |