Donate to e Foundation | Murena handsets with /e/OS | Own a part of Murena! Learn more

Commit d20db4c4 authored by Andy Hung's avatar Andy Hung Committed by Android (Google) Code Review
Browse files

Merge "Fix resampler to allow output of single frame"

parents 4cc224ac 24781fff

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