I wrote C++ phase vocoder to change pitch based on MATLAB code from DAFX book. MATLAB code and sample audio file are here. I tested both on a pure sine wave and MATLAB code outputs good result, but C++ adds some buzzing. I spent days trying to understand why and I would appreciate your thoughts.

Here are relevant C++ methods. They use other classes, but the code excerpts below should be clear enough. The first method pitchShift is called for every FFTWindowLength = 1024 frames read from a file. The second method synthesizeHop is called from pitchShift for every hop.

vector<double> PitchShiftResampling::pitchShift(long numSampsToProcess, AudioBuffer audioBuffer){

static double fftProcessData[MAX_FRAME_LENGTH];
long i, s, k;
long gRover = PitchShiftResampling::inFifoLatency;

vector<double> output(numSampsToProcess);

// TODO: Make it work for stereo. For now, work with only one channel
vector<double> indata = audioBuffer.getSampleData(0);

FFT* fft = new FFT(fftFrameSize, audioBuffer.getNumChannels()); 

/* main processing loop */
for (i = 0; i < numSampsToProcess; i++){

    output[i] = gOutFIFO[gRover-inFifoLatency];

    /* As long as we have not yet collected enough data just read in */
    gInFIFO[gRover] = indata[i];

    /* now we have enough data for processing */
    if (gRover >= fftFrameSize) {

        gRover = inFifoLatency;

        /* do windowing */
        for (s = 0; s < fftFrameSize; s++) {
            window = 0.5 * (1.0 - cos (2.0*M_PI*(double)s/(double)(fftFrameSize)));
            fftProcessData[s] = window * gInFIFO[s];

        /* do FFT centering */
        Util::fftshift(fftProcessData, MAX_FRAME_LENGTH);

        /* do FFT */


        /* move input FIFO */
        for (k = 0; k < inFifoLatency; k++) {
            gInFIFO[k] = gInFIFO[k+stepSize];

delete fft;
return output;


void PitchShiftResampling::synthesizeHop(FFT *fft) {

int k;

vector<double> omega;
for (k = 0; k < fftFrameSize; k++) {
    omega.push_back(2 * M_PI * stepSize * k / fftFrameSize);

for (k = 0; k <= fftFrameSize2; k++) {

    double currentAnaPhase = fft->phaseMagn->phas[0][k];

    double deltaPhi = omega[k] + Util::princarg(currentAnaPhase - prevAnaPhase[k] - omega[k]);

    double currentSynPhase = Util::princarg(prevSynPhase[k] + deltaPhi * pitchShift); 

    double magn = fft->phaseMagn->magn[0][k];
    double real = magn * cos(currentSynPhase);
    double imag = magn * sin(currentSynPhase);

    // fftw library uses following format for spectrum data
    // r0, r1, r2, ...., r_(n/2), i_(n+1)/2-1, ...., i2, i1
    // where (r0,0) is f_0, (r1,i1) is f_1, and so on (f_0 does not have imaginary part because the input array is real).
    if (k == 0) {
        fft->fftFrameSpectrum->currentChannelData->spectrum[k] = real;
    if (k > 0) {
        fft->fftFrameSpectrum->currentChannelData->spectrum[k] = real;
        fft->fftFrameSpectrum->currentChannelData->spectrum[fftFrameSize - k] = imag;

    prevAnaPhase[k] = currentAnaPhase;
    prevSynPhase[k] = currentSynPhase;


/* IFFT */
Util::fftshift(fft->fftFrameSpectrum->currentChannelData->output, fftFrameSize); 

vector<double> grain;
/* do windowing and add to output accumulator */
for(k=0; k < fftFrameSize; k++) {
    window = 0.5 * (1.0 - cos (2.0*M_PI*(double)k/(double)(fftFrameSize)));
    grain.push_back(window * fft->fftFrameSpectrum->currentChannelData->output[k] / fftFrameSize);

vector<double> grain2;
vector<double> grain3;

for(k=0; k < fftFrameSize; k++) {
for(k=0; k < lx; k++) {
    double a = grain2[ix[k]];
    double val = a * dx1[k] + grain2[ix1[k]] * dx[k];

for(k=0; k < lx; k++) {
    gOutputAccum[k] += grain3[k];

for (k = 0; k < stepSize; k++) {
    gOutFIFO[k] = gOutputAccum[k];

/* shift accumulator */
memmove(gOutputAccum, gOutputAccum+stepSize, fftFrameSize*2*sizeof(float));


I tested princarg, fftshift functions, made sure that I correctly assign values to fftw library spectrum data structure. I also made sure that overlap add is implemented correctly and that interpolation part is equivalent to MATLAB code.

Here is a screenshot of a resulting sine wave, processed in C++ using 0.8 pitch shift ratio (input was 440Hz sine wave):

440Hz sine wave processed in C++ using 0.8 pitch shift ratio

There are glitches appearing in sine wave every $2.5$ cycles, even for different pitch shift ratios. Glitches become worse in C++ as the pitch ratio decreases below 1, but sine waves processed in MATLAB look almost perfect.

Any hints or ideas about where to look for error are greatly appreciated!

EDIT: Problem solved

Thanks guys! I suspected there is an issue related to the overlap and it turned out my fftshift function had a bug. Apparently I didn't test it carefully enough. Shifting was off by one which resulted in that glitch in every hop.

I replaced

std::rotate(&in[0], &in[n - n2], &in[n - 1]);


for (int i = 0; i < n2; i++)
    double tmp = in[i];
    in[i] = in[i + n2];
    in[i + n2] = tmp;

where n is the length of the array and n2 = n / 2 and the buzzing is gone!

  • $\begingroup$ Hope is FFTsize/4 ? if the glitches appear always every 2.5 cycles, your problem seems to be in the overlap point position. $\endgroup$
    – ederwander
    Commented Dec 14, 2016 at 23:37

1 Answer 1


you might have a windowing offset error. that it's so skinny, might be an offset of one. this is sometimes called the "fence-post error" or "off-by-one" error because MATLAB and C++ might look at indexing slightly differently.


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