/*

     File: FFTBufferManager.cpp

 Abstract: This class manages buffering and computation for FFT analysis on input audio data. The methods provided are used to grab the audio, buffer it, and perform the FFT when sufficient data is available

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#include "FFTBufferManager.h"

#include "CABitOperations.h"

#include "CAStreamBasicDescription.h"

#define min(x,y) (x < y) ? x : y

FFTBufferManager::FFTBufferManager(UInt32 inNumberFrames) :

mNeedsAudioData(0),

mHasAudioData(0),

mFFTNormFactor(1.0/(2*inNumberFrames)),

mAdjust0DB(1.5849e-13),

m24BitFracScale(16777216.0f),

mFFTLength(inNumberFrames/2),

mLog2N(Log2Ceil(inNumberFrames)),

mNumberFrames(inNumberFrames),

mAudioBufferSize(inNumberFrames * sizeof(Float32)),

mAudioBufferCurrentIndex(0)

{

    mAudioBuffer = (Float32*) calloc(mNumberFrames,sizeof(Float32));

    mDspSplitComplex.realp = (Float32*) calloc(mFFTLength,sizeof(Float32));

    mDspSplitComplex.imagp = (Float32*) calloc(mFFTLength, sizeof(Float32));

    mSpectrumAnalysis = vDSP_create_fftsetup(mLog2N, kFFTRadix2);

    OSAtomicIncrement32Barrier(&mNeedsAudioData);

}

FFTBufferManager::~FFTBufferManager()

{

    vDSP_destroy_fftsetup(mSpectrumAnalysis);

    free(mAudioBuffer);

    free (mDspSplitComplex.realp);

    free (mDspSplitComplex.imagp);

}

void FFTBufferManager::GrabAudioData(AudioBufferList *inBL)

{

    if (mAudioBufferSize < inBL->mBuffers[0].mDataByteSize) return;

    UInt32 bytesToCopy = min(inBL->mBuffers[0].mDataByteSize, mAudioBufferSize - mAudioBufferCurrentIndex);

    memcpy(mAudioBuffer+mAudioBufferCurrentIndex, inBL->mBuffers[0].mData, bytesToCopy);

    mAudioBufferCurrentIndex += bytesToCopy / sizeof(Float32);

    if (mAudioBufferCurrentIndex >= mAudioBufferSize / sizeof(Float32))

    {

        OSAtomicIncrement32Barrier(&mHasAudioData);

        OSAtomicDecrement32Barrier(&mNeedsAudioData);

    }

}

Boolean FFTBufferManager::ComputeFFT(int32_t *outFFTData)

{

    if (HasNewAudioData())

    {

        //Generate a split complex vector from the real data

        vDSP_ctoz((COMPLEX *)mAudioBuffer, 2, &mDspSplitComplex, 1, mFFTLength);

        //Take the fft and scale appropriately

        vDSP_fft_zrip(mSpectrumAnalysis, &mDspSplitComplex, 1, mLog2N, kFFTDirection_Forward);

        vDSP_vsmul(mDspSplitComplex.realp, 1, &mFFTNormFactor, mDspSplitComplex.realp, 1, mFFTLength);

        vDSP_vsmul(mDspSplitComplex.imagp, 1, &mFFTNormFactor, mDspSplitComplex.imagp, 1, mFFTLength);

        //Zero out the nyquist value

        mDspSplitComplex.imagp[0] = 0.0;

        //Convert the fft data to dB

        Float32 tmpData[mFFTLength];

        vDSP_zvmags(&mDspSplitComplex, 1, tmpData, 1, mFFTLength);

        //In order to avoid taking log10 of zero, an adjusting factor is added in to make the minimum value equal -128dB

        vDSP_vsadd(tmpData, 1, &mAdjust0DB, tmpData, 1, mFFTLength);

        Float32 one = 1;

        vDSP_vdbcon(tmpData, 1, &one, tmpData, 1, mFFTLength, 0);

        //Convert floating point data to integer (Q7.24)

        vDSP_vsmul(tmpData, 1, &m24BitFracScale, tmpData, 1, mFFTLength);

        for(UInt32 i=0; i<mFFTLength; ++i)

            outFFTData[i] = (SInt32) tmpData[i];

        OSAtomicDecrement32Barrier(&mHasAudioData);

        OSAtomicIncrement32Barrier(&mNeedsAudioData);

        mAudioBufferCurrentIndex = 0;

        return true;

    }

    else if (mNeedsAudioData == 0)

        OSAtomicIncrement32Barrier(&mNeedsAudioData);

    return false;

}