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Commit d8b2e2b9 authored by Glenn Kasten's avatar Glenn Kasten
Browse files

Remove aliasing 

Code was aliasing mBuffer as buffer, but continuing to use both buffer
and mBuffer after that point.  This was at best misleading, and at worst
could confuse the compiler into generating bad code.  There was no
performance advantage to the alias, in fact removing it saves 16 bytes.

Change-Id: I55023ddba465d9be82f66745b088d18af658ac60
parent f4aaf1f5

services/audioflinger/AudioResamplerSinc.cpp

+11 −12
Original line number Diff line number Diff line
@@ -199,33 +199,32 @@ void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount, 
    size_t outputSampleCount = outFrameCount * 2;

    size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;



    AudioBufferProvider::Buffer& buffer(mBuffer);

    while (outputIndex < outputSampleCount) {

        // buffer is empty, fetch a new one

        while (buffer.frameCount == 0) {

            buffer.frameCount = inFrameCount;

            provider->getNextBuffer(&buffer);

            if (buffer.raw == NULL) {

        while (mBuffer.frameCount == 0) {

            mBuffer.frameCount = inFrameCount;

            provider->getNextBuffer(&mBuffer);

            if (mBuffer.raw == NULL) {

                goto resample_exit;

            }

            const uint32_t phaseIndex = phaseFraction >> kNumPhaseBits;

            if (phaseIndex == 1) {

                // read one frame

                read<CHANNELS>(impulse, phaseFraction, buffer.i16, inputIndex);

                read<CHANNELS>(impulse, phaseFraction, mBuffer.i16, inputIndex);

            } else if (phaseIndex == 2) {

                // read 2 frames

                read<CHANNELS>(impulse, phaseFraction, buffer.i16, inputIndex);

                read<CHANNELS>(impulse, phaseFraction, mBuffer.i16, inputIndex);

                inputIndex++;

                if (inputIndex >= mBuffer.frameCount) {

                    inputIndex -= mBuffer.frameCount;

                    provider->releaseBuffer(&buffer);

                    provider->releaseBuffer(&mBuffer);

                } else {

                    read<CHANNELS>(impulse, phaseFraction, buffer.i16, inputIndex);

                    read<CHANNELS>(impulse, phaseFraction, mBuffer.i16, inputIndex);

                }

           }

        }

        int16_t *in = buffer.i16;

        const size_t frameCount = buffer.frameCount;

        int16_t *in = mBuffer.i16;

        const size_t frameCount = mBuffer.frameCount;



        // Always read-in the first samples from the input buffer

        int16_t* head = impulse + halfNumCoefs*CHANNELS;
@@ -264,7 +263,7 @@ void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount, 
        // if done with buffer, save samples

        if (inputIndex >= frameCount) {

            inputIndex -= frameCount;

            provider->releaseBuffer(&buffer);

            provider->releaseBuffer(&mBuffer);

        }

    }