jixin_new/shiku_im/audioRecorder/smbPitchShift.m

// 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);
//	
}





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