368 lines
13 KiB
C++
368 lines
13 KiB
C++
/*************************************************************************/
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/* audio_effect_pitch_shift.cpp */
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/*************************************************************************/
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/* This file is part of: */
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/* GODOT ENGINE */
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/* https://godotengine.org */
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/*************************************************************************/
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/* Copyright (c) 2007-2022 Juan Linietsky, Ariel Manzur. */
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/* Copyright (c) 2014-2022 Godot Engine contributors (cf. AUTHORS.md). */
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/* */
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/* Permission is hereby granted, free of charge, to any person obtaining */
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/* a copy of this software and associated documentation files (the */
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/* "Software"), to deal in the Software without restriction, including */
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/* without limitation the rights to use, copy, modify, merge, publish, */
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/* distribute, sublicense, and/or sell copies of the Software, and to */
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/* permit persons to whom the Software is furnished to do so, subject to */
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/* the following conditions: */
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/* */
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/* The above copyright notice and this permission notice shall be */
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/* included in all copies or substantial portions of the Software. */
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/* */
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/* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, */
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/* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF */
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/* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.*/
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/* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY */
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/* CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, */
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/* TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE */
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/* SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. */
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/*************************************************************************/
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#include "audio_effect_pitch_shift.h"
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#include "core/math/math_funcs.h"
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#include "servers/audio_server.h"
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/* Thirdparty code, so disable clang-format with Godot style */
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/* clang-format off */
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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 https://dspguru.com/wide-open-license/ 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.*Math_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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}
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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.*Math_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/Math_PI;
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if (qpd >= 0) { qpd += qpd&1;
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} else { qpd -= qpd&1;
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}
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tmp -= Math_PI*(double)qpd;
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/* get deviation from bin frequency from the +/- Pi interval */
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tmp = osamp*tmp/(2.*Math_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.*Math_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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}
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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.*Math_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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}
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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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}
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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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}
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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((double)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 = Math_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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/* Godot code again */
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/* clang-format on */
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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, fft_size, base->oversampling, sample_rate, in_l, out_l, 2);
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shift_r.PitchShift(base->pitch_scale, p_frame_count, fft_size, base->oversampling, 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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static const int fft_sizes[FFT_SIZE_MAX] = { 256, 512, 1024, 2048, 4096 };
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ins->fft_size = fft_sizes[fft_size];
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return ins;
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}
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void AudioEffectPitchShift::set_pitch_scale(float p_pitch_scale) {
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ERR_FAIL_COND(p_pitch_scale <= 0.0);
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pitch_scale = p_pitch_scale;
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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::set_oversampling(int p_oversampling) {
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ERR_FAIL_COND(p_oversampling < 4);
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oversampling = p_oversampling;
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}
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int AudioEffectPitchShift::get_oversampling() const {
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return oversampling;
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}
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void AudioEffectPitchShift::set_fft_size(FFT_Size p_fft_size) {
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ERR_FAIL_INDEX(p_fft_size, FFT_SIZE_MAX);
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fft_size = p_fft_size;
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}
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AudioEffectPitchShift::FFT_Size AudioEffectPitchShift::get_fft_size() const {
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return fft_size;
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}
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void AudioEffectPitchShift::_bind_methods() {
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ClassDB::bind_method(D_METHOD("set_pitch_scale", "rate"), &AudioEffectPitchShift::set_pitch_scale);
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ClassDB::bind_method(D_METHOD("get_pitch_scale"), &AudioEffectPitchShift::get_pitch_scale);
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ClassDB::bind_method(D_METHOD("set_oversampling", "amount"), &AudioEffectPitchShift::set_oversampling);
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ClassDB::bind_method(D_METHOD("get_oversampling"), &AudioEffectPitchShift::get_oversampling);
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ClassDB::bind_method(D_METHOD("set_fft_size", "size"), &AudioEffectPitchShift::set_fft_size);
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ClassDB::bind_method(D_METHOD("get_fft_size"), &AudioEffectPitchShift::get_fft_size);
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ADD_PROPERTY(PropertyInfo(Variant::REAL, "pitch_scale", PROPERTY_HINT_RANGE, "0.01,16,0.01"), "set_pitch_scale", "get_pitch_scale");
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ADD_PROPERTY(PropertyInfo(Variant::REAL, "oversampling", PROPERTY_HINT_RANGE, "4,32,1"), "set_oversampling", "get_oversampling");
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ADD_PROPERTY(PropertyInfo(Variant::INT, "fft_size", PROPERTY_HINT_ENUM, "256,512,1024,2048,4096"), "set_fft_size", "get_fft_size");
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BIND_ENUM_CONSTANT(FFT_SIZE_256);
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BIND_ENUM_CONSTANT(FFT_SIZE_512);
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BIND_ENUM_CONSTANT(FFT_SIZE_1024);
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BIND_ENUM_CONSTANT(FFT_SIZE_2048);
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BIND_ENUM_CONSTANT(FFT_SIZE_4096);
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BIND_ENUM_CONSTANT(FFT_SIZE_MAX);
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}
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AudioEffectPitchShift::AudioEffectPitchShift() {
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pitch_scale = 1.0;
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oversampling = 4;
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fft_size = FFT_SIZE_2048;
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}
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