/**************************************************************************** * * NAME: smbPitchShift.c * VERSION: 1.2 * HOME URL: http://blogs.zynaptiq.com/bernsee * 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 1999-2015 Stephan M. Bernsee * * 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 #include #include #define MAX_FRAME_LENGTH 8192 #define false 0 #define true 1 void smbFft(float *fftBuffer, long fftFrameSize, long sign); double smbAtan2(double x, double y); // ----------------------------------------------------------------------------------------------------------------- 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)1999-2015 Stephan M. Bernsee */ { 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, expct; long i,k, qpd, index, inFifoLatency, stepSize, fftFrameSize2; /* 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) { 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++){ /* 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; /* do windowing and re,im interleave */ for (k = 0; k < fftFrameSize;k++) { window = -.5*cos(2.*M_PI*(double)k/(double)fftFrameSize)+.5; gFFTworksp[2*k] = gInFIFO[k] * window; gFFTworksp[2*k+1] = 0.; } /* ***************** ANALYSIS ******************* */ /* do transform */ smbFft(gFFTworksp, fftFrameSize, -1); /* this is the analysis step */ 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 */ 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; } /* ***************** PROCESSING ******************* */ /* this does the actual pitch shifting */ memset(gSynMagn, 0, fftFrameSize*sizeof(float)); memset(gSynFreq, 0, fftFrameSize*sizeof(float)); for (k = 0; k <= fftFrameSize2; k++) { 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]; /* shift accumulator */ memmove(gOutputAccum, gOutputAccum+stepSize, fftFrameSize*sizeof(float)); /* move input FIFO */ for (k = 0; k < inFifoLatency; k++) gInFIFO[k] = gInFIFO[k+stepSize]; } } } // ----------------------------------------------------------------------------------------------------------------- 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); } // ----------------------------------------------------------------------------------------------------------------- // ----------------------------------------------------------------------------------------------------------------- // -----------------------------------------------------------------------------------------------------------------