/*

     File: FFTHelper.cpp

 Abstract: This class demonstrates how to use the Accelerate framework to take Fast Fourier Transforms (FFT) of the audio data. FFTs are used to perform analysis on the captured audio data

  Version: 2.0

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#include "FFTHelper.h"

// Utility includes

#include "CABitOperations.h"

const Float32 kAdjust0DB = 1.5849e-13;

FFTHelper::FFTHelper ( UInt32 inMaxFramesPerSlice )

: mSpectrumAnalysis(NULL),

mFFTNormFactor(1.0/(2*inMaxFramesPerSlice)),

mFFTLength(inMaxFramesPerSlice/2),

mLog2N(Log2Ceil(inMaxFramesPerSlice))

{

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

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

    mSpectrumAnalysis = vDSP_create_fftsetup(mLog2N, kFFTRadix2);

}

FFTHelper::~FFTHelper()

{

    vDSP_destroy_fftsetup(mSpectrumAnalysis);

    free (mDspSplitComplex.realp);

    free (mDspSplitComplex.imagp);

}

void FFTHelper::ComputeFFT(Float32* inAudioData, Float32* outFFTData)

{

    if (inAudioData == NULL || outFFTData == NULL) return;

    //Generate a split complex vector from the real data

    vDSP_ctoz((COMPLEX *)inAudioData, 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

    vDSP_zvmags(&mDspSplitComplex, 1, outFFTData, 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(outFFTData, 1, &kAdjust0DB, outFFTData, 1, mFFTLength);

    Float32 one = 1;

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

}