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Pretty high quality pitch Shifting effect using smbPitchShift
This commit is contained in:
Juan Linietsky
parent
4d9bc8b00c
commit
cff80bb1cc
297
servers/audio/effects/audio_effect_pitch_shift.cpp
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297
servers/audio/effects/audio_effect_pitch_shift.cpp
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#include "audio_effect_pitch_shift.h"
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#include "servers/audio_server.h"
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/****************************************************************************
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*
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* NAME: smbPitchShift.cpp
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* VERSION: 1.2
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* HOME URL: http://blogs.zynaptiq.com/bernsee
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* KNOWN BUGS: none
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*
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* SYNOPSIS: Routine for doing pitch shifting while maintaining
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* duration using the Short Time Fourier Transform.
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*
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* DESCRIPTION: The routine takes a pitchShift factor value which is between 0.5
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* (one octave down) and 2. (one octave up). A value of exactly 1 does not change
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* the pitch. numSampsToProcess tells the routine how many samples in indata[0...
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* numSampsToProcess-1] should be pitch shifted and moved to outdata[0 ...
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* numSampsToProcess-1]. The two buffers can be identical (ie. it can process the
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* data in-place). fftFrameSize defines the FFT frame size used for the
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* processing. Typical values are 1024, 2048 and 4096. It may be any value <=
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* MAX_FRAME_LENGTH but it MUST be a power of 2. osamp is the STFT
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* oversampling factor which also determines the overlap between adjacent STFT
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* frames. It should at least be 4 for moderate scaling ratios. A value of 32 is
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* recommended for best quality. sampleRate takes the sample rate for the signal
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* in unit Hz, ie. 44100 for 44.1 kHz audio. The data passed to the routine in
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* indata[] should be in the range [-1.0, 1.0), which is also the output range
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* for the data, make sure you scale the data accordingly (for 16bit signed integers
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* you would have to divide (and multiply) by 32768).
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*
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* COPYRIGHT 1999-2015 Stephan M. Bernsee <s.bernsee [AT] zynaptiq [DOT] com>
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*
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* The Wide Open License (WOL)
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*
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* Permission to use, copy, modify, distribute and sell this software and its
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* documentation for any purpose is hereby granted without fee, provided that
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* the above copyright notice and this license appear in all source copies.
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* THIS SOFTWARE IS PROVIDED "AS IS" WITHOUT EXPRESS OR IMPLIED WARRANTY OF
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* ANY KIND. See http://www.dspguru.com/wol.htm for more information.
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*
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*****************************************************************************/
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void SMBPitchShift::PitchShift(float pitchShift, long numSampsToProcess, long fftFrameSize, long osamp, float sampleRate, float *indata, float *outdata,int stride) {
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/*
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Routine smbPitchShift(). See top of file for explanation
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Purpose: doing pitch shifting while maintaining duration using the Short
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Time Fourier Transform.
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Author: (c)1999-2015 Stephan M. Bernsee <s.bernsee [AT] zynaptiq [DOT] com>
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*/
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double magn, phase, tmp, window, real, imag;
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double freqPerBin, expct;
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long i,k, qpd, index, inFifoLatency, stepSize, fftFrameSize2;
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/* set up some handy variables */
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fftFrameSize2 = fftFrameSize/2;
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stepSize = fftFrameSize/osamp;
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freqPerBin = sampleRate/(double)fftFrameSize;
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expct = 2.*M_PI*(double)stepSize/(double)fftFrameSize;
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inFifoLatency = fftFrameSize-stepSize;
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if (gRover == 0) gRover = inFifoLatency;
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/* initialize our static arrays */
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/* main processing loop */
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for (i = 0; i < numSampsToProcess; i++){
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/* As long as we have not yet collected enough data just read in */
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gInFIFO[gRover] = indata[i*stride];
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outdata[i*stride] = gOutFIFO[gRover-inFifoLatency];
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gRover++;
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/* now we have enough data for processing */
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if (gRover >= fftFrameSize) {
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gRover = inFifoLatency;
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/* do windowing and re,im interleave */
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for (k = 0; k < fftFrameSize;k++) {
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window = -.5*cos(2.*M_PI*(double)k/(double)fftFrameSize)+.5;
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gFFTworksp[2*k] = gInFIFO[k] * window;
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gFFTworksp[2*k+1] = 0.;
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}
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/* ***************** ANALYSIS ******************* */
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/* do transform */
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smbFft(gFFTworksp, fftFrameSize, -1);
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/* this is the analysis step */
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for (k = 0; k <= fftFrameSize2; k++) {
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/* de-interlace FFT buffer */
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real = gFFTworksp[2*k];
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imag = gFFTworksp[2*k+1];
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/* compute magnitude and phase */
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magn = 2.*sqrt(real*real + imag*imag);
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phase = atan2(imag,real);
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/* compute phase difference */
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tmp = phase - gLastPhase[k];
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gLastPhase[k] = phase;
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/* subtract expected phase difference */
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tmp -= (double)k*expct;
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/* map delta phase into +/- Pi interval */
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qpd = tmp/M_PI;
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if (qpd >= 0) qpd += qpd&1;
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else qpd -= qpd&1;
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tmp -= M_PI*(double)qpd;
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/* get deviation from bin frequency from the +/- Pi interval */
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tmp = osamp*tmp/(2.*M_PI);
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/* compute the k-th partials' true frequency */
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tmp = (double)k*freqPerBin + tmp*freqPerBin;
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/* store magnitude and true frequency in analysis arrays */
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gAnaMagn[k] = magn;
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gAnaFreq[k] = tmp;
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}
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/* ***************** PROCESSING ******************* */
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/* this does the actual pitch shifting */
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memset(gSynMagn, 0, fftFrameSize*sizeof(float));
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memset(gSynFreq, 0, fftFrameSize*sizeof(float));
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for (k = 0; k <= fftFrameSize2; k++) {
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index = k*pitchShift;
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if (index <= fftFrameSize2) {
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gSynMagn[index] += gAnaMagn[k];
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gSynFreq[index] = gAnaFreq[k] * pitchShift;
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}
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}
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/* ***************** SYNTHESIS ******************* */
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/* this is the synthesis step */
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for (k = 0; k <= fftFrameSize2; k++) {
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/* get magnitude and true frequency from synthesis arrays */
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magn = gSynMagn[k];
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tmp = gSynFreq[k];
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/* subtract bin mid frequency */
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tmp -= (double)k*freqPerBin;
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/* get bin deviation from freq deviation */
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tmp /= freqPerBin;
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/* take osamp into account */
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tmp = 2.*M_PI*tmp/osamp;
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/* add the overlap phase advance back in */
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tmp += (double)k*expct;
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/* accumulate delta phase to get bin phase */
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gSumPhase[k] += tmp;
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phase = gSumPhase[k];
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/* get real and imag part and re-interleave */
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gFFTworksp[2*k] = magn*cos(phase);
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gFFTworksp[2*k+1] = magn*sin(phase);
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}
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/* zero negative frequencies */
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for (k = fftFrameSize+2; k < 2*fftFrameSize; k++) gFFTworksp[k] = 0.;
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/* do inverse transform */
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smbFft(gFFTworksp, fftFrameSize, 1);
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/* do windowing and add to output accumulator */
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for(k=0; k < fftFrameSize; k++) {
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window = -.5*cos(2.*M_PI*(double)k/(double)fftFrameSize)+.5;
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gOutputAccum[k] += 2.*window*gFFTworksp[2*k]/(fftFrameSize2*osamp);
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}
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for (k = 0; k < stepSize; k++) gOutFIFO[k] = gOutputAccum[k];
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/* shift accumulator */
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memmove(gOutputAccum, gOutputAccum+stepSize, fftFrameSize*sizeof(float));
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/* move input FIFO */
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for (k = 0; k < inFifoLatency; k++) gInFIFO[k] = gInFIFO[k+stepSize];
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}
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}
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}
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void SMBPitchShift::smbFft(float *fftBuffer, long fftFrameSize, long sign)
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/*
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FFT routine, (C)1996 S.M.Bernsee. Sign = -1 is FFT, 1 is iFFT (inverse)
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Fills fftBuffer[0...2*fftFrameSize-1] with the Fourier transform of the
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time domain data in fftBuffer[0...2*fftFrameSize-1]. The FFT array takes
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and returns the cosine and sine parts in an interleaved manner, ie.
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fftBuffer[0] = cosPart[0], fftBuffer[1] = sinPart[0], asf. fftFrameSize
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must be a power of 2. It expects a complex input signal (see footnote 2),
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ie. when working with 'common' audio signals our input signal has to be
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passed as {in[0],0.,in[1],0.,in[2],0.,...} asf. In that case, the transform
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of the frequencies of interest is in fftBuffer[0...fftFrameSize].
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*/
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{
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float wr, wi, arg, *p1, *p2, temp;
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float tr, ti, ur, ui, *p1r, *p1i, *p2r, *p2i;
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long i, bitm, j, le, le2, k;
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for (i = 2; i < 2*fftFrameSize-2; i += 2) {
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for (bitm = 2, j = 0; bitm < 2*fftFrameSize; bitm <<= 1) {
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if (i & bitm) j++;
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j <<= 1;
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}
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if (i < j) {
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p1 = fftBuffer+i; p2 = fftBuffer+j;
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temp = *p1; *(p1++) = *p2;
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*(p2++) = temp; temp = *p1;
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*p1 = *p2; *p2 = temp;
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}
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}
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for (k = 0, le = 2; k < (long)(log(fftFrameSize)/log(2.)+.5); k++) {
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le <<= 1;
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le2 = le>>1;
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ur = 1.0;
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ui = 0.0;
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arg = M_PI / (le2>>1);
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wr = cos(arg);
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wi = sign*sin(arg);
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for (j = 0; j < le2; j += 2) {
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p1r = fftBuffer+j; p1i = p1r+1;
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p2r = p1r+le2; p2i = p2r+1;
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for (i = j; i < 2*fftFrameSize; i += le) {
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tr = *p2r * ur - *p2i * ui;
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ti = *p2r * ui + *p2i * ur;
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*p2r = *p1r - tr; *p2i = *p1i - ti;
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*p1r += tr; *p1i += ti;
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p1r += le; p1i += le;
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p2r += le; p2i += le;
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}
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tr = ur*wr - ui*wi;
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ui = ur*wi + ui*wr;
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ur = tr;
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}
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}
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}
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void AudioEffectPitchShiftInstance::process(const AudioFrame *p_src_frames,AudioFrame *p_dst_frames,int p_frame_count) {
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float sample_rate = AudioServer::get_singleton()->get_mix_rate();
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float *in_l = (float*)p_src_frames;
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float *in_r = in_l + 1;
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float *out_l = (float*)p_dst_frames;
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float *out_r = out_l + 1;
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shift_l.PitchShift(base->pitch_scale,p_frame_count,2048,4,sample_rate,in_l,out_l,2);
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shift_r.PitchShift(base->pitch_scale,p_frame_count,2048,4,sample_rate,in_r,out_r,2);
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}
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Ref<AudioEffectInstance> AudioEffectPitchShift::instance() {
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Ref<AudioEffectPitchShiftInstance> ins;
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ins.instance();
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ins->base=Ref<AudioEffectPitchShift>(this);
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return ins;
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}
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void AudioEffectPitchShift::set_pitch_scale(float p_adjust) {
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pitch_scale=p_adjust;
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}
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float AudioEffectPitchShift::get_pitch_scale() const {
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return pitch_scale;
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}
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void AudioEffectPitchShift::_bind_methods() {
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ClassDB::bind_method(_MD("set_pitch_scale","rate"),&AudioEffectPitchShift::set_pitch_scale);
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ClassDB::bind_method(_MD("get_pitch_scale"),&AudioEffectPitchShift::get_pitch_scale);
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ADD_PROPERTY(PropertyInfo(Variant::REAL,"pitch_scale",PROPERTY_HINT_RANGE,"0.01,16,0.01"),_SCS("set_pitch_scale"),_SCS("get_pitch_scale"));
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}
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AudioEffectPitchShift::AudioEffectPitchShift() {
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pitch_scale=1.0;
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}
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89
servers/audio/effects/audio_effect_pitch_shift.h
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89
servers/audio/effects/audio_effect_pitch_shift.h
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@ -0,0 +1,89 @@
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#ifndef AUDIO_EFFECT_PITCH_SHIFT_H
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#define AUDIO_EFFECT_PITCH_SHIFT_H
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#include "servers/audio/audio_effect.h"
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class SMBPitchShift {
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enum {
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MAX_FRAME_LENGTH=8192
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};
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float gInFIFO[MAX_FRAME_LENGTH];
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float gOutFIFO[MAX_FRAME_LENGTH];
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float gFFTworksp[2*MAX_FRAME_LENGTH];
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float gLastPhase[MAX_FRAME_LENGTH/2+1];
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float gSumPhase[MAX_FRAME_LENGTH/2+1];
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float gOutputAccum[2*MAX_FRAME_LENGTH];
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float gAnaFreq[MAX_FRAME_LENGTH];
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float gAnaMagn[MAX_FRAME_LENGTH];
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float gSynFreq[MAX_FRAME_LENGTH];
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float gSynMagn[MAX_FRAME_LENGTH];
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long gRover;
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void smbFft(float *fftBuffer, long fftFrameSize, long sign);
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public:
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void PitchShift(float pitchShift, long numSampsToProcess, long fftFrameSize, long osamp, float sampleRate, float *indata, float *outdata, int stride);
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SMBPitchShift() {
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gRover=0;
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memset(gInFIFO, 0, MAX_FRAME_LENGTH*sizeof(float));
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memset(gOutFIFO, 0, MAX_FRAME_LENGTH*sizeof(float));
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memset(gFFTworksp, 0, 2*MAX_FRAME_LENGTH*sizeof(float));
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memset(gLastPhase, 0, (MAX_FRAME_LENGTH/2+1)*sizeof(float));
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memset(gSumPhase, 0, (MAX_FRAME_LENGTH/2+1)*sizeof(float));
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memset(gOutputAccum, 0, 2*MAX_FRAME_LENGTH*sizeof(float));
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memset(gAnaFreq, 0, MAX_FRAME_LENGTH*sizeof(float));
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memset(gAnaMagn, 0, MAX_FRAME_LENGTH*sizeof(float));
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}
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};
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class AudioEffectPitchShift;
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class AudioEffectPitchShiftInstance : public AudioEffectInstance {
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GDCLASS(AudioEffectPitchShiftInstance,AudioEffectInstance)
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friend class AudioEffectPitchShift;
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Ref<AudioEffectPitchShift> base;
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SMBPitchShift shift_l;
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SMBPitchShift shift_r;
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public:
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virtual void process(const AudioFrame *p_src_frames,AudioFrame *p_dst_frames,int p_frame_count);
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};
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class AudioEffectPitchShift : public AudioEffect {
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GDCLASS(AudioEffectPitchShift,AudioEffect)
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friend class AudioEffectPitchShiftInstance;
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float pitch_scale;
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int window_size;
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float wet;
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float dry;
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bool filter;
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protected:
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static void _bind_methods();
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public:
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Ref<AudioEffectInstance> instance();
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void set_pitch_scale(float p_adjust);
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float get_pitch_scale() const;
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AudioEffectPitchShift();
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};
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#endif // AUDIO_EFFECT_PITCH_SHIFT_H
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@ -48,6 +48,7 @@
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#include "audio/effects/audio_effect_delay.h"
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#include "audio/effects/audio_effect_compressor.h"
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#include "audio/effects/audio_effect_limiter.h"
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#include "audio/effects/audio_effect_pitch_shift.h"
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static void _debugger_get_resource_usage(List<ScriptDebuggerRemote::ResourceUsage>* r_usage) {
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@ -111,6 +112,7 @@ void register_server_types() {
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ClassDB::register_class<AudioEffectDelay>();
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ClassDB::register_class<AudioEffectCompressor>();
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ClassDB::register_class<AudioEffectLimiter>();
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ClassDB::register_class<AudioEffectPitchShift>();
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}
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