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// original source from http://www.dspdimension.com Stephan M. Bernsee
//
// has been modified to use FFT functions from accelerate framework (vdsp)
// tz 11/2011
/****************************************************************************
*
* NAME: smbPitchShift.cpp
* VERSION: 1.2
* HOME URL: http://www.dspdimension.com
* KNOWN BUGS: none
*
* SYNOPSIS: Routine for doing pitch shifting while maintaining
* duration using the Short Time Fourier Transform.
*
* DESCRIPTION: The routine takes a pitchShift factor value which is between 0.5
* (one octave down) and 2. (one octave up). A value of exactly 1 does not change
* the pitch. numSampsToProcess tells the routine how many samples in indata[0...
* numSampsToProcess-1] should be pitch shifted and moved to outdata[0 ...
* numSampsToProcess-1]. The two buffers can be identical (ie. it can process the
* data in-place). fftFrameSize defines the FFT frame size used for the
* processing. Typical values are 1024, 2048 and 4096. It may be any value <=
* MAX_FRAME_LENGTH but it MUST be a power of 2. osamp is the STFT
* oversampling factor which also determines the overlap between adjacent STFT
* frames. It should at least be 4 for moderate scaling ratios. A value of 32 is
* recommended for best quality. sampleRate takes the sample rate for the signal
* in unit Hz, ie. 44100 for 44.1 kHz audio. The data passed to the routine in
* indata[] should be in the range [-1.0, 1.0), which is also the output range
* for the data, make sure you scale the data accordingly (for 16bit signed integers
* you would have to divide (and multiply) by 32768).
*
* COPYRIGHT 1999-2009 Stephan M. Bernsee <smb [AT] dspdimension [DOT] com>
*
* The Wide Open License (WOL)
*
* Permission to use, copy, modify, distribute and sell this software and its
* documentation for any purpose is hereby granted without fee, provided that
* the above copyright notice and this license appear in all source copies.
* THIS SOFTWARE IS PROVIDED "AS IS" WITHOUT EXPRESS OR IMPLIED WARRANTY OF
* ANY KIND. See http://www.dspguru.com/wol.htm for more information.
*
*****************************************************************************/
#include <string.h>
#include <math.h>
#include <stdio.h>
#import <Accelerate/Accelerate.h>
// #define M_PI 3.14159265358979323846
// #define MAX_FRAME_LENGTH 8192
#define MAX_FRAME_LENGTH 4096 // tz
void smbFft(float *fftBuffer, long fftFrameSize, long sign);
double smbAtan2(double x, double y);
void printFFTInitSnapshot(long fftFrameSize2,long stepSize,double freqPerBin,double expct,
long inFifoLatency, long gRover);
void printFFTSnapshot(long i, long k, long qpd, long index,
double magn, double phase, double tmp,
double window, double real, double imag,
long gRover);
// -----------------------------------------------------------------------------------------------------------------
void smbPitchShift(float pitchShift, long numSampsToProcess, long fftFrameSize, long osamp, float sampleRate, float *indata, float *outdata)
/*
Routine smbPitchShift(). See top of file for explanation
Purpose: doing pitch shifting while maintaining duration using the Short
Time Fourier Transform.
Author: (c)1999-2009 Stephan M. Bernsee <smb [AT] dspdimension [DOT] com>
*/
{
static float gInFIFO[MAX_FRAME_LENGTH];
static float gOutFIFO[MAX_FRAME_LENGTH];
static float gFFTworksp[2*MAX_FRAME_LENGTH];
static float gLastPhase[MAX_FRAME_LENGTH/2+1];
static float gSumPhase[MAX_FRAME_LENGTH/2+1];
static float gOutputAccum[2*MAX_FRAME_LENGTH];
static float gAnaFreq[MAX_FRAME_LENGTH];
static float gAnaMagn[MAX_FRAME_LENGTH];
static float gSynFreq[MAX_FRAME_LENGTH];
static float gSynMagn[MAX_FRAME_LENGTH];
static long gRover = FALSE, gInit = FALSE;
double magn, phase, tmp, window, real, imag;
double freqPerBin;
double expct; // expected phase difference tz
long i,k, qpd, index, inFifoLatency, stepSize, fftFrameSize2;
// float maxMag; // tz maximum magnitude for pitch detection display
// float displayFreq; // true pitch from last window analysis
/* set up some handy variables */
fftFrameSize2 = fftFrameSize/2;
stepSize = fftFrameSize/osamp;
freqPerBin = sampleRate/(double)fftFrameSize;
expct = 2.*M_PI*(double)stepSize/(double)fftFrameSize;
inFifoLatency = fftFrameSize-stepSize;
if (gRover == FALSE) gRover = inFifoLatency;
/* initialize our static arrays */
if (gInit == FALSE) {
NSLog(@"init static arrays");
printFFTInitSnapshot(fftFrameSize2,stepSize, freqPerBin, expct, inFifoLatency, gRover);
memset(gInFIFO, 0, MAX_FRAME_LENGTH*sizeof(float));
memset(gOutFIFO, 0, MAX_FRAME_LENGTH*sizeof(float));
memset(gFFTworksp, 0, 2*MAX_FRAME_LENGTH*sizeof(float));
memset(gLastPhase, 0, (MAX_FRAME_LENGTH/2+1)*sizeof(float));
memset(gSumPhase, 0, (MAX_FRAME_LENGTH/2+1)*sizeof(float));
memset(gOutputAccum, 0, 2*MAX_FRAME_LENGTH*sizeof(float));
memset(gAnaFreq, 0, MAX_FRAME_LENGTH*sizeof(float));
memset(gAnaMagn, 0, MAX_FRAME_LENGTH*sizeof(float));
gInit = true;
}
/* main processing loop */
for (i = 0; i < numSampsToProcess; i++){
// loading
// load the next section of data, one stepsize chunk at a time, starting at beginning of indata. the chunk gets loaded
// to a slot at the end of the gInFIFO, while at the same time, the chunk at the beginning of gOutFIFO gets loaded to into
// the outdata buffer one chunk at a time starting at the beginning.
//
// the very first time this pitchshifter is called, the gOutFIFO will be initialized with zero's so it looks like
// there will be some latency before the actual 'processed' samples begin to fill outdata.
//
/* As long as we have not yet collected enough data, just read in */
gInFIFO[gRover] = indata[i];
outdata[i] = gOutFIFO[gRover-inFifoLatency];
gRover++;
/* now we have enough data for processing */
if (gRover >= fftFrameSize) {
gRover = inFifoLatency; // gRover cycles up between (fftFrameSize - stepsize) and fftFrameSize
// eg., 896 - 1024 for an osamp of 8 and framesize of 1024
/* do windowing and re,im interleave */
// note that the first time this runs, the inFIFO will be mostly zeroes, but essentially, the fft runs on
// data that keeps getting slid to the left?
// the window is like a triangular hat that gets imposed over the sample buffer before its input to the fft
// the size of the hat is the fftsize and it scales off the data at beginning and end of the buffer
// i think that the vDSP_ctoz function will accomplish the interleaving and complex formatting stuff below
// we would still need to do the windowing, but maybe there's an apple function for that too
for (k = 0; k < fftFrameSize;k++) {
window = -.5*cos(2.*M_PI*(double)k/(double)fftFrameSize)+.5;
gFFTworksp[2*k] = gInFIFO[k] * window; // real part is winowed amplitude of samples
gFFTworksp[2*k+1] = 0.; // imag part is set to 0
// NSLog(@"i: %d, k: %d, window: %f", i, k, window );
}
/* ***************** ANALYSIS ******************* */
/* do transform */
// lets try replacing this with accelerate functions
smbFft(gFFTworksp, fftFrameSize, -1);
/* this is the analysis step */
// this is looping through the fft output bins in the frequency domain
for (k = 0; k <= fftFrameSize2; k++) {
/* de-interlace FFT buffer */
real = gFFTworksp[2*k];
imag = gFFTworksp[2*k+1];
/* compute magnitude and phase */
magn = 2.*sqrt(real*real + imag*imag);
phase = atan2(imag,real);
/* compute phase difference */
// the gLastPhase[k] would be the phase from the kth frequency bin from the previous transform over this endlessly
// shifting data
tmp = phase - gLastPhase[k];
gLastPhase[k] = phase;
/* subtract expected phase difference */
tmp -= (double)k*expct;
/* map delta phase into +/- Pi interval */
qpd = tmp/M_PI;
if (qpd >= 0) qpd += qpd&1;
else qpd -= qpd&1;
tmp -= M_PI*(double)qpd;
/* get deviation from bin frequency from the +/- Pi interval */
tmp = osamp*tmp/(2.*M_PI);
/* compute the k-th partials' true frequency */
tmp = (double)k*freqPerBin + tmp*freqPerBin;
/* store magnitude and true frequency in analysis arrays */
gAnaMagn[k] = magn;
gAnaFreq[k] = tmp;
}
/*
// pitch detection ------------------
// find max magnitude for this pass
maxMag = 0.0;
displayFreq = 0.0;
for (k = 0; k <= fftFrameSize2; k++) {
if (gAnaMagn[k] > maxMag) {
maxMag = gAnaMagn[k];
displayFreq = gAnaFreq[k];
}
}
NSLog(@"pitch is: %f", displayFreq);
*/
/* ***************** PROCESSING ******************* */
/* this does the actual pitch shifting */
memset(gSynMagn, 0, fftFrameSize*sizeof(float)); // why do we zero out the buffer to frame size but
memset(gSynFreq, 0, fftFrameSize*sizeof(float)); // only actually seem to use half of frame size?
// so this code assigns the results of the analysis.
// it sets up pitch shifted bins using analyzed magnitude and analyzed freq * pitchShift
for (k = 0; k <= fftFrameSize2; k++) {
index = (long) (k * pitchShift);
// NSLog(@"i: %d, index: %d, k: %d, pitchShift: %f", i, index, k, pitchShift );
if (index <= fftFrameSize2) {
gSynMagn[index] += gAnaMagn[k];
gSynFreq[index] = gAnaFreq[k] * pitchShift;
}
}
/* ***************** SYNTHESIS ******************* */
/* this is the synthesis step */
for (k = 0; k <= fftFrameSize2; k++) {
/* get magnitude and true frequency from synthesis arrays */
magn = gSynMagn[k];
tmp = gSynFreq[k];
/* subtract bin mid frequency */
tmp -= (double)k*freqPerBin;
/* get bin deviation from freq deviation */
tmp /= freqPerBin;
/* take osamp into account */
tmp = 2.*M_PI*tmp/osamp;
/* add the overlap phase advance back in */
tmp += (double)k*expct;
/* accumulate delta phase to get bin phase */
gSumPhase[k] += tmp;
phase = gSumPhase[k];
/* get real and imag part and re-interleave */
gFFTworksp[2*k] = magn*cos(phase);
gFFTworksp[2*k+1] = magn*sin(phase);
}
/* zero negative frequencies */
for (k = fftFrameSize+2; k < 2*fftFrameSize; k++) gFFTworksp[k] = 0.;
/* do inverse transform */
smbFft(gFFTworksp, fftFrameSize, 1);
/* do windowing and add to output accumulator */
for(k=0; k < fftFrameSize; k++) {
window = -.5*cos(2.*M_PI*(double)k/(double)fftFrameSize)+.5;
gOutputAccum[k] += 2.*window*gFFTworksp[2*k]/(fftFrameSize2*osamp);
}
for (k = 0; k < stepSize; k++) gOutFIFO[k] = gOutputAccum[k];
// why use two different methods to copy memory?
/* shift accumulator */
// this shifts in zeroes from beyond the bounds of framesize to fill the upper step size chunk
memmove(gOutputAccum, gOutputAccum+stepSize, fftFrameSize*sizeof(float));
/* move input FIFO */
for (k = 0; k < inFifoLatency; k++) gInFIFO[k] = gInFIFO[k+stepSize];
}
}
}
// -----------------------------------------------------------------------------------------------------------------
// -----------------------------------------------------------------------------------------------------------------
void smb2PitchShift(float pitchShift, long numSampsToProcess, long fftFrameSize, long osamp,
float sampleRate, float *indata, float *outdata,
FFTSetup fftSetup, float *frequency)
/*
Routine smbPitchShift(). See top of file for explanation
Purpose: doing pitch shifting while maintaining duration using the Short
Time Fourier Transform.
Author: (c)1999-2009 Stephan M. Bernsee <smb [AT] dspdimension [DOT] com>
*/
{
static float gInFIFO[MAX_FRAME_LENGTH];
static float gOutFIFO[MAX_FRAME_LENGTH];
static float gFFTworksp[2*MAX_FRAME_LENGTH];
static float gLastPhase[MAX_FRAME_LENGTH/2+1];
static float gSumPhase[MAX_FRAME_LENGTH/2+1];
static float gOutputAccum[2*MAX_FRAME_LENGTH];
static float gAnaFreq[MAX_FRAME_LENGTH];
static float gAnaMagn[MAX_FRAME_LENGTH];
static float gSynFreq[MAX_FRAME_LENGTH];
static float gSynMagn[MAX_FRAME_LENGTH];
static COMPLEX_SPLIT A;
static long gRover = FALSE, gInit = FALSE;
double magn, phase, tmp, window, real, imag;
double freqPerBin;
double expct; // expected phase difference tz
long i,k, qpd, index, inFifoLatency, stepSize, fftFrameSize2;
int stride;
size_t bufferCapacity; // In samples
int log2n, n, nOver2; // params for fft setup
float maxMag; // tz maximum magnitude for pitch detection display
float displayFreq; // true pitch from last window analysis
int pitchCount = 0; // number of times pitch gets measured
float freqTotal = 0; // sum of all pitch measurements
/* set up some handy variables */
fftFrameSize2 = fftFrameSize/2;
stepSize = fftFrameSize/osamp;
freqPerBin = sampleRate/(double)fftFrameSize;
expct = 2.*M_PI*(double)stepSize/(double)fftFrameSize;
inFifoLatency = fftFrameSize-stepSize;
if (gRover == FALSE) gRover = inFifoLatency;
stride = 1;
log2n = log2f(fftFrameSize); // log base2 of max number of frames, eg., 10 for 1024
n = 1 << log2n; // actual max number of frames, eg., 1024 - what a silly way to compute it
nOver2 = fftFrameSize/2;
bufferCapacity = fftFrameSize;
// index = 0;
/* initialize our static arrays */
if (gInit == FALSE) {
NSLog(@"init static arrays");
printFFTInitSnapshot(fftFrameSize2,stepSize, freqPerBin, expct, inFifoLatency, gRover);
memset(gInFIFO, 0, MAX_FRAME_LENGTH*sizeof(float));
memset(gOutFIFO, 0, MAX_FRAME_LENGTH*sizeof(float));
memset(gFFTworksp, 0, 2*MAX_FRAME_LENGTH*sizeof(float));
memset(gLastPhase, 0, (MAX_FRAME_LENGTH/2+1)*sizeof(float));
memset(gSumPhase, 0, (MAX_FRAME_LENGTH/2+1)*sizeof(float));
memset(gOutputAccum, 0, 2*MAX_FRAME_LENGTH*sizeof(float));
memset(gAnaFreq, 0, MAX_FRAME_LENGTH*sizeof(float));
memset(gAnaMagn, 0, MAX_FRAME_LENGTH*sizeof(float));
// split complex number buffer
A.realp = (float *)malloc(nOver2 * sizeof(float)); //
A.imagp = (float *)malloc(nOver2 * sizeof(float)); // why is it set to half the frame size
gInit = true;
}
// NSLog(@"before load");
/* main processing loop */
for (i = 0; i < numSampsToProcess; i++){
// loading
// load the next section of data, one stepsize chunk at a time, starting at beginning of indata. the chunk gets loaded
// to a slot at the end of the gInFIFO, while at the same time, the chunk at the beginning of gOutFIFO gets loaded to into
// the outdata buffer one chunk at a time starting at the beginning.
//
// the very first time this pitchshifter is called, the gOutFIFO will be initialized with zero's so it looks like
// there will be some latency before the actual 'processed' samples begin to fill outdata.
//
/* As long as we have not yet collected enough data, just read in */
gInFIFO[gRover] = indata[i];
outdata[i] = gOutFIFO[gRover-inFifoLatency];
gRover++;
/* now we have enough data for processing */
if (gRover >= fftFrameSize) {
gRover = inFifoLatency; // gRover cycles up between (fftFrameSize - stepsize) and fftFrameSize
// eg., 896 - 1024 for an osamp of 8 and framesize of 1024
/* do windowing and re,im interleave */
// note that the first time this runs, the inFIFO will be mostly zeroes, but essentially, the fft runs on
// data that keeps getting slid to the left?
// the window is like a triangular hat that gets imposed over the sample buffer before its input to the fft
// the size of the hat is the fftsize and it scales off the data at beginning and end of the buffer
// i think that the vDSP_ctoz function will accomplish the interleaving and complex formatting stuff below
// we would still need to do the windowing, but maybe there's an apple function for that too
// for (k = 0; k < fftFrameSize;k++) {
// window = -.5*cos(2.*M_PI*(double)k/(double)fftFrameSize)+.5;
// gFFTworksp[2*k] = gInFIFO[k] * window; // real part is winowed amplitude of samples
// gFFTworksp[2*k+1] = 0.; // imag part is set to 0
// // NSLog(@"i: %d, k: %d, window: %f", i, k, window );
// }
for (k = 0; k < fftFrameSize;k++) {
window = -.5*cos(2.*M_PI*(double)k/(double)fftFrameSize)+.5;
gFFTworksp[k] = gInFIFO[k] * window; // real part is winowed amplitude of samples
// gFFTworksp[2*k+1] = 0.; // imag part is set to 0
// NSLog(@"i: %d, k: %d, window: %f", i, k, window );
}
// cast to complex interleaved then convert to split complex vector
vDSP_ctoz((COMPLEX*)gFFTworksp, 2, &A, 1, nOver2);
// Carry out a Forward FFT transform.
// NSLog(@"before transform");
vDSP_fft_zrip(fftSetup, &A, stride, log2n, FFT_FORWARD);
// NSLog(@"after transform");
// convert from split complex to complex interleaved for analysis
vDSP_ztoc(&A, 1, (COMPLEX *)gFFTworksp, 2, nOver2);
/* ***************** ANALYSIS ******************* */
/* do transform */
// lets try replacing this with accelerate functions
// smbFft(gFFTworksp, fftFrameSize, -1);
/* this is the analysis step */
// this is looping through the fft output bins in the frequency domain
for (k = 0; k <= fftFrameSize2; k++) {
/* de-interlace FFT buffer */
real = gFFTworksp[2*k];
imag = gFFTworksp[2*k+1];
/* compute magnitude and phase */
magn = 2.*sqrt(real*real + imag*imag);
phase = atan2(imag,real);
/* compute phase difference */
// the gLastPhase[k] would be the phase from the kth frequency bin from the previous transform over this endlessly
// shifting data
tmp = phase - gLastPhase[k];
gLastPhase[k] = phase;
/* subtract expected phase difference */
tmp -= (double)k*expct;
/* map delta phase into +/- Pi interval */
qpd = tmp/M_PI;
if (qpd >= 0) qpd += qpd&1;
else qpd -= qpd&1;
tmp -= M_PI*(double)qpd;
/* get deviation from bin frequency from the +/- Pi interval */
tmp = osamp*tmp/(2.*M_PI);
/* compute the k-th partials' true frequency */
tmp = (double)k*freqPerBin + tmp*freqPerBin;
/* store magnitude and true frequency in analysis arrays */
gAnaMagn[k] = magn;
gAnaFreq[k] = tmp;
}
// pitch detection ------------------
// find max magnitude for this pass
maxMag = 0.0;
displayFreq = 0.0;
for (k = 0; k <= fftFrameSize2; k++) {
if (gAnaMagn[k] > maxMag) {
maxMag = gAnaMagn[k];
displayFreq = gAnaFreq[k];
}
}
freqTotal += displayFreq;
pitchCount++;
/* ***************** PROCESSING ******************* */
/* this does the actual pitch shifting */
memset(gSynMagn, 0, fftFrameSize*sizeof(float)); // why do we zero out the buffer to frame size but
memset(gSynFreq, 0, fftFrameSize*sizeof(float)); // only actually seem to use half of frame size?
// so this code assigns the results of the analysis.
// it sets up pitch shifted bins using analyzed magnitude and analyzed freq * pitchShift
for (k = 0; k <= fftFrameSize2; k++) {
index = (long) (k * pitchShift);
// NSLog(@"i: %d, index: %d, k: %d, pitchShift: %f", i, index, k, pitchShift );
if (index <= fftFrameSize2) {
gSynMagn[index] += gAnaMagn[k];
gSynFreq[index] = gAnaFreq[k] * pitchShift;
}
}
/* ***************** SYNTHESIS ******************* */
/* this is the synthesis step */
for (k = 0; k <= fftFrameSize2; k++) {
/* get magnitude and true frequency from synthesis arrays */
magn = gSynMagn[k];
tmp = gSynFreq[k];
/* subtract bin mid frequency */
tmp -= (double)k*freqPerBin;
/* get bin deviation from freq deviation */
tmp /= freqPerBin;
/* take osamp into account */
tmp = 2.*M_PI*tmp/osamp;
/* add the overlap phase advance back in */
tmp += (double)k*expct;
/* accumulate delta phase to get bin phase */
gSumPhase[k] += tmp;
phase = gSumPhase[k];
/* get real and imag part and re-interleave */
gFFTworksp[2*k] = magn*cos(phase);
gFFTworksp[2*k+1] = magn*sin(phase);
}
/* zero negative frequencies */
for (k = fftFrameSize+2; k < 2*fftFrameSize; k++) gFFTworksp[k] = 0.;
// convert from complex interleaved to split complex vector
vDSP_ctoz((COMPLEX*)gFFTworksp, 2, &A, 1, nOver2);
// Carry out an inverse FFT transform.
vDSP_fft_zrip(fftSetup, &A, stride, log2n, FFT_INVERSE );
// scale it
// the suggested scale factor makes the sound barely audible
// so we should probably experiment with various things
// I have a hunch that the stfft needs a different kind of scaling
// float scale = (float) 1.0 / (2 * n);
// float scale = (float) 1.0 / (osamp);
float scale = .25;
vDSP_vsmul(A.realp, 1, &scale, A.realp, 1, nOver2 );
vDSP_vsmul(A.imagp, 1, &scale, A.imagp, 1, nOver2 );
// covert from split complex to complex interleaved
vDSP_ztoc(&A, 1, (COMPLEX *) gFFTworksp, 2, nOver2);
/* do inverse transform */
// smbFft(gFFTworksp, fftFrameSize, 1);
/* do windowing and add to output accumulator */
/*
for(k=0; k < fftFrameSize; k++) {
window = -.5*cos(2.*M_PI*(double)k/(double)fftFrameSize)+.5;
gOutputAccum[k] += 2.*window*gFFTworksp[2*k]/(fftFrameSize2*osamp);
}
*/
/* do windowing and add to output accumulator */
for(k=0; k < fftFrameSize; k++) {
window = -.5*cos(2.*M_PI*(double)k/(double)fftFrameSize)+.5;
gOutputAccum[k] += 2.*window*gFFTworksp[k]/(fftFrameSize2*osamp);
}
for (k = 0; k < stepSize; k++) gOutFIFO[k] = gOutputAccum[k];
// why use two different methods to copy memory?
/* shift accumulator */
// this shifts in zeroes from beyond the bounds of framesize to fill the upper step size chunk
memmove(gOutputAccum, gOutputAccum+stepSize, fftFrameSize*sizeof(float));
/* move input FIFO */
for (k = 0; k < inFifoLatency; k++) gInFIFO[k] = gInFIFO[k+stepSize];
}
}
// NSLog(@"pitchCount: %d", pitchCount);
*frequency = (float) (freqTotal / pitchCount);
// NSLog(@"pitch is: %f", *frequency );
}
// -----------------------------------------------------------------------------------------------------------------
void smbFft(float *fftBuffer, long fftFrameSize, long sign)
/*
FFT routine, (C)1996 S.M.Bernsee. Sign = -1 is FFT, 1 is iFFT (inverse)
Fills fftBuffer[0...2*fftFrameSize-1] with the Fourier transform of the
time domain data in fftBuffer[0...2*fftFrameSize-1]. The FFT array takes
and returns the cosine and sine parts in an interleaved manner, ie.
fftBuffer[0] = cosPart[0], fftBuffer[1] = sinPart[0], asf. fftFrameSize
must be a power of 2. It expects a complex input signal (see footnote 2),
ie. when working with 'common' audio signals our input signal has to be
passed as {in[0],0.,in[1],0.,in[2],0.,...} asf. In that case, the transform
of the frequencies of interest is in fftBuffer[0...fftFrameSize].
*/
{
float wr, wi, arg, *p1, *p2, temp;
float tr, ti, ur, ui, *p1r, *p1i, *p2r, *p2i;
long i, bitm, j, le, le2, k;
for (i = 2; i < 2*fftFrameSize-2; i += 2) {
for (bitm = 2, j = 0; bitm < 2*fftFrameSize; bitm <<= 1) {
if (i & bitm) j++;
j <<= 1;
}
if (i < j) {
p1 = fftBuffer+i; p2 = fftBuffer+j;
temp = *p1; *(p1++) = *p2;
*(p2++) = temp; temp = *p1;
*p1 = *p2; *p2 = temp;
}
}
for (k = 0, le = 2; k < (long)(log(fftFrameSize)/log(2.)+.5); k++) {
le <<= 1;
le2 = le>>1;
ur = 1.0;
ui = 0.0;
arg = M_PI / (le2>>1);
wr = cos(arg);
wi = sign*sin(arg);
for (j = 0; j < le2; j += 2) {
p1r = fftBuffer+j; p1i = p1r+1;
p2r = p1r+le2; p2i = p2r+1;
for (i = j; i < 2*fftFrameSize; i += le) {
tr = *p2r * ur - *p2i * ui;
ti = *p2r * ui + *p2i * ur;
*p2r = *p1r - tr; *p2i = *p1i - ti;
*p1r += tr; *p1i += ti;
p1r += le; p1i += le;
p2r += le; p2i += le;
}
tr = ur*wr - ui*wi;
ui = ur*wi + ui*wr;
ur = tr;
}
}
}
// -----------------------------------------------------------------------------------------------------------------
/*
12/12/02, smb
PLEASE NOTE:
There have been some reports on domain errors when the atan2() function was used
as in the above code. Usually, a domain error should not interrupt the program flow
(maybe except in Debug mode) but rather be handled "silently" and a global variable
should be set according to this error. However, on some occasions people ran into
this kind of scenario, so a replacement atan2() function is provided here.
If you are experiencing domain errors and your program stops, simply replace all
instances of atan2() with calls to the smbAtan2() function below.
*/
double smbAtan2(double x, double y)
{
double signx;
if (x > 0.) signx = 1.;
else signx = -1.;
if (x == 0.) return 0.;
if (y == 0.) return signx * M_PI / 2.;
return atan2(x, y);
}
// tz utility functions
void printFFTInitSnapshot(long fftFrameSize2,long stepSize,double freqPerBin,double expct,
long inFifoLatency, long gRover) {
NSLog(@"fft init snapshot");
NSLog(@"fftFrameSize2: %ld", fftFrameSize2);
NSLog(@"stepSize: %ld", stepSize);
NSLog(@"freqPerBin: %f", freqPerBin);
NSLog(@"expct: %f", expct);
NSLog(@"inFifoLatency: %ld", inFifoLatency);
NSLog(@"gRover: %ld", gRover);
}
void printFFTSnapshot(long i, long k, long qpd, long index,
double magn, double phase, double tmp,
double window, double real, double imag,
long gRover){
NSLog(@"fft snapshot");
NSLog(@"i: %ld, k: %ld, qpd: %ld, index: %ld", i,k,qpd,index);
NSLog(@"magn: %f, phase: %f, tmp: %f", magn, phase, tmp );
NSLog(@"window: %f, real: %f, imag: %f ", window, real, imag);
NSLog(@"gRover %ld", gRover);
}
// -----------------------------------------------------------------------------------------------------------------
// -----------------------------------------------------------------------------------------------------------------
// -----------------------------------------------------------------------------------------------------------------