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Commit 24781fff authored by Andy Hung's avatar Andy Hung
Browse files

Fix resampler to allow output of single frame 



Bug: 13073201
Change-Id: If7818a3389a191a37277bbd8e96a59ef8ce68509
Signed-off-by: default avatarAndy Hung <hunga@google.com>
parent 1ebd04b5

services/audioflinger/AudioResampler.cpp

+2 −2
Original line number Diff line number Diff line
@@ -341,7 +341,7 @@ void AudioResamplerOrder1::resampleStereo16(int32_t* out, size_t outFrameCount, 
    uint32_t phaseIncrement = mPhaseIncrement;

    size_t outputIndex = 0;

    size_t outputSampleCount = outFrameCount * 2;

    size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;

    size_t inFrameCount = getInFrameCountRequired(outFrameCount);



    // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d",

    //      outFrameCount, inputIndex, phaseFraction, phaseIncrement);
@@ -439,7 +439,7 @@ void AudioResamplerOrder1::resampleMono16(int32_t* out, size_t outFrameCount, 
    uint32_t phaseIncrement = mPhaseIncrement;

    size_t outputIndex = 0;

    size_t outputSampleCount = outFrameCount * 2;

    size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;

    size_t inFrameCount = getInFrameCountRequired(outFrameCount);



    // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d",

    //      outFrameCount, inputIndex, phaseFraction, phaseIncrement);

services/audioflinger/AudioResampler.h

+32 −0
Original line number Diff line number Diff line
@@ -110,6 +110,38 @@ protected: 
    uint64_t mLocalTimeFreq;

    int64_t mPTS;



    // returns the inFrameCount required to generate outFrameCount frames.

    //

    // Placed here to be a consistent for all resamplers.

    //

    // Right now, we use the upper bound without regards to the current state of the

    // input buffer using integer arithmetic, as follows:

    //

    // (static_cast<uint64_t>(outFrameCount)*mInSampleRate + (mSampleRate - 1))/mSampleRate;

    //

    // The double precision equivalent (float may not be precise enough):

    // ceil(static_cast<double>(outFrameCount) * mInSampleRate / mSampleRate);

    //

    // this relies on the fact that the mPhaseIncrement is rounded down from

    // #phases * mInSampleRate/mSampleRate and the fact that Sum(Floor(x)) <= Floor(Sum(x)).

    // http://www.proofwiki.org/wiki/Sum_of_Floors_Not_Greater_Than_Floor_of_Sums

    //

    // (so long as double precision is computed accurately enough to be considered

    // greater than or equal to the Floor(x) value in int32_t arithmetic; thus this

    // will not necessarily hold for floats).

    //

    // TODO:

    // Greater accuracy and a tight bound is obtained by:

    // 1) subtract and adjust for the current state of the AudioBufferProvider buffer.

    // 2) using the exact integer formula where (ignoring 64b casting)

    //  inFrameCount = (mPhaseIncrement * (outFrameCount - 1) + mPhaseFraction) / phaseWrapLimit;

    //  phaseWrapLimit is the wraparound (1 << kNumPhaseBits), if not specified explicitly.

    //

    inline size_t getInFrameCountRequired(size_t outFrameCount) {

        return (static_cast<uint64_t>(outFrameCount)*mInSampleRate

                + (mSampleRate - 1))/mSampleRate;

    }



private:

    const src_quality mQuality;

services/audioflinger/AudioResamplerCubic.cpp

+2 −2
Original line number Diff line number Diff line
@@ -60,7 +60,7 @@ void AudioResamplerCubic::resampleStereo16(int32_t* out, size_t outFrameCount, 
    uint32_t phaseIncrement = mPhaseIncrement;

    size_t outputIndex = 0;

    size_t outputSampleCount = outFrameCount * 2;

    size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;

    size_t inFrameCount = getInFrameCountRequired(outFrameCount);



    // fetch first buffer

    if (mBuffer.frameCount == 0) {
@@ -128,7 +128,7 @@ void AudioResamplerCubic::resampleMono16(int32_t* out, size_t outFrameCount, 
    uint32_t phaseIncrement = mPhaseIncrement;

    size_t outputIndex = 0;

    size_t outputSampleCount = outFrameCount * 2;

    size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;

    size_t inFrameCount = getInFrameCountRequired(outFrameCount);



    // fetch first buffer

    if (mBuffer.frameCount == 0) {

services/audioflinger/AudioResamplerDyn.cpp

+1 −1
Original line number Diff line number Diff line
@@ -466,7 +466,7 @@ void AudioResamplerDyn::resample(int32_t* out, size_t outFrameCount, 
    const uint32_t phaseIncrement = mPhaseIncrement;

    size_t outputIndex = 0;

    size_t outputSampleCount = outFrameCount * 2;   // stereo output

    size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;

    size_t inFrameCount = getInFrameCountRequired(outFrameCount);

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



    // NOTE: be very careful when modifying the code here. register

services/audioflinger/AudioResamplerSinc.cpp

+1 −1
Original line number Diff line number Diff line
@@ -540,7 +540,7 @@ void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount, 
    uint32_t phaseIncrement = mPhaseIncrement;

    size_t outputIndex = 0;

    size_t outputSampleCount = outFrameCount * 2;

    size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;

    size_t inFrameCount = getInFrameCountRequired(outFrameCount);



    while (outputIndex < outputSampleCount) {

        // buffer is empty, fetch a new one