Merge "Update dynamic resampler buffer fetching" (7dac3313) · Commits · e / os / android_frameworks_av

services/audioflinger/AudioResamplerDyn.cpp

+16 −6

Original line number	Original line	Diff line number	Diff line
	@@ -455,12 +455,13 @@ void AudioResamplerDyn<TC, TI, TO>::resample(TO* out, size_t outFrameCount,
	const Constants& c(mConstants);		const Constants& c(mConstants);
	const TC* const coefs = mConstants.mFirCoefs;		const TC* const coefs = mConstants.mFirCoefs;
	TI* impulse = mInBuffer.getImpulse();		TI* impulse = mInBuffer.getImpulse();
	size_t inputIndex = mInputIndex;		size_t inputIndex = 0;
	uint32_t phaseFraction = mPhaseFraction;		uint32_t phaseFraction = mPhaseFraction;
	const uint32_t phaseIncrement = mPhaseIncrement;		const uint32_t phaseIncrement = mPhaseIncrement;
	size_t outputIndex = 0;		size_t outputIndex = 0;
	size_t outputSampleCount = outFrameCount * 2; // stereo output		size_t outputSampleCount = outFrameCount * 2; // stereo output
	size_t inFrameCount = getInFrameCountRequired(outFrameCount);		size_t inFrameCount = getInFrameCountRequired(outFrameCount) + (phaseFraction != 0);
			ALOG_ASSERT(0 < inFrameCount && inFrameCount < (1U << 31));
	const uint32_t phaseWrapLimit = c.mL << c.mShift;		const uint32_t phaseWrapLimit = c.mL << c.mShift;

	// NOTE: be very careful when modifying the code here. register		// NOTE: be very careful when modifying the code here. register
	@@ -474,11 +475,13 @@ void AudioResamplerDyn<TC, TI, TO>::resample(TO* out, size_t outFrameCount,
	// buffer is empty, fetch a new one		// buffer is empty, fetch a new one
	while (mBuffer.frameCount == 0) {		while (mBuffer.frameCount == 0) {
	mBuffer.frameCount = inFrameCount;		mBuffer.frameCount = inFrameCount;
			ALOG_ASSERT(inFrameCount > 0);
	provider->getNextBuffer(&mBuffer,		provider->getNextBuffer(&mBuffer,
	calculateOutputPTS(outputIndex / 2));		calculateOutputPTS(outputIndex / 2));
	if (mBuffer.raw == NULL) {		if (mBuffer.raw == NULL) {
	goto resample_exit;		goto resample_exit;
	}		}
			inFrameCount -= mBuffer.frameCount;
	if (phaseFraction >= phaseWrapLimit) { // read in data		if (phaseFraction >= phaseWrapLimit) { // read in data
	mInBuffer.template readAdvance<CHANNELS>(		mInBuffer.template readAdvance<CHANNELS>(
	impulse, c.mHalfNumCoefs,		impulse, c.mHalfNumCoefs,
	@@ -487,7 +490,7 @@ void AudioResamplerDyn<TC, TI, TO>::resample(TO* out, size_t outFrameCount,
	while (phaseFraction >= phaseWrapLimit) {		while (phaseFraction >= phaseWrapLimit) {
	inputIndex++;		inputIndex++;
	if (inputIndex >= mBuffer.frameCount) {		if (inputIndex >= mBuffer.frameCount) {
	inputIndex -= mBuffer.frameCount;		inputIndex = 0;
	provider->releaseBuffer(&mBuffer);		provider->releaseBuffer(&mBuffer);
	break;		break;
	}		}
	@@ -535,15 +538,22 @@ void AudioResamplerDyn<TC, TI, TO>::resample(TO* out, size_t outFrameCount,
	done:		done:
	// often arrives here when input buffer runs out		// often arrives here when input buffer runs out
	if (inputIndex >= frameCount) {		if (inputIndex >= frameCount) {
	inputIndex -= frameCount;		inputIndex = 0;
	provider->releaseBuffer(&mBuffer);		provider->releaseBuffer(&mBuffer);
	// mBuffer.frameCount MUST be zero here.		ALOG_ASSERT(mBuffer.frameCount == 0);
	}		}
	}		}

	resample_exit:		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);
			}
	mInBuffer.setImpulse(impulse);		mInBuffer.setImpulse(impulse);
	mInputIndex = inputIndex;
	mPhaseFraction = phaseFraction;		mPhaseFraction = phaseFraction;
	}		}