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Commit 411cb8e4 authored by Andy Hung's avatar Andy Hung
Browse files

Update dynamic resampler buffer fetching 

Make the criteria tight for fetching to avoid storing
excessive frame data internal to the resampler.
This should reduce jitter in frame delivery computation.

Bug: 14962343
Change-Id: I7adaf714d11c272696ccdbf218bda994c7217477
parent 49869253

services/audioflinger/AudioResamplerDyn.cpp

+16 −6
Original line number 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 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);

    size_t inFrameCount = getInFrameCountRequired(outFrameCount) + (phaseFraction != 0);

    ALOG_ASSERT(0 < inFrameCount && inFrameCount < (1U << 31));

    const uint32_t phaseWrapLimit = c.mL << c.mShift;



    // 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

        while (mBuffer.frameCount == 0) {

            mBuffer.frameCount = inFrameCount;

            ALOG_ASSERT(inFrameCount > 0);

            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,
@@ -487,7 +490,7 @@ void AudioResamplerDyn<TC, TI, TO>::resample(TO* out, size_t outFrameCount, 
                while (phaseFraction >= phaseWrapLimit) {

                    inputIndex++;

                    if (inputIndex >= mBuffer.frameCount) {

                        inputIndex -= mBuffer.frameCount;

                        inputIndex = 0;

                        provider->releaseBuffer(&mBuffer);

                        break;

                    }
@@ -535,15 +538,22 @@ void AudioResamplerDyn<TC, TI, TO>::resample(TO* out, size_t outFrameCount, 
done:

        // often arrives here when input buffer runs out

        if (inputIndex >= frameCount) {

            inputIndex -= frameCount;

            inputIndex = 0;

            provider->releaseBuffer(&mBuffer);

            // mBuffer.frameCount MUST be zero here.

            ALOG_ASSERT(mBuffer.frameCount == 0);

        }

    }



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);

    mInputIndex = inputIndex;

    mPhaseFraction = phaseFraction;

}