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godot/servers/audio/effects/audio_effect_pitch_shift.cpp

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/*************************************************************************/
/* audio_effect_pitch_shift.cpp */
/*************************************************************************/
/* This file is part of: */
/* GODOT ENGINE */
/* http://www.godotengine.org */
/*************************************************************************/
/* Copyright (c) 2007-2017 Juan Linietsky, Ariel Manzur. */
/* */
/* Permission is hereby granted, free of charge, to any person obtaining */
/* a copy of this software and associated documentation files (the */
/* "Software"), to deal in the Software without restriction, including */
/* without limitation the rights to use, copy, modify, merge, publish, */
/* distribute, sublicense, and/or sell copies of the Software, and to */
/* permit persons to whom the Software is furnished to do so, subject to */
/* the following conditions: */
/* */
/* The above copyright notice and this permission notice shall be */
/* included in all copies or substantial portions of the Software. */
/* */
/* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, */
/* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF */
/* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.*/
/* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY */
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/* TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE */
/* SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. */
/*************************************************************************/
#include "audio_effect_pitch_shift.h"
#include "math_funcs.h"
#include "servers/audio_server.h"
/* Thirdparty code, so disable clang-format with Godot style */
/* clang-format off */
/****************************************************************************
*
* NAME: smbPitchShift.cpp
* 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 <s.bernsee [AT] zynaptiq [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.
*
*****************************************************************************/
void SMBPitchShift::PitchShift(float pitchShift, long numSampsToProcess, long fftFrameSize, long osamp, float sampleRate, float *indata, float *outdata,int stride) {
/*
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 <s.bernsee [AT] zynaptiq [DOT] com>
*/
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.*Math_PI*(double)stepSize/(double)fftFrameSize;
inFifoLatency = fftFrameSize-stepSize;
if (gRover == 0) gRover = inFifoLatency;
/* initialize our static arrays */
/* 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*stride];
outdata[i*stride] = 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.*Math_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/Math_PI;
if (qpd >= 0) qpd += qpd&1;
else qpd -= qpd&1;
tmp -= Math_PI*(double)qpd;
/* get deviation from bin frequency from the +/- Pi interval */
tmp = osamp*tmp/(2.*Math_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.*Math_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.*Math_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 SMBPitchShift::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((double)fftFrameSize)/log(2.)+.5); k++) {
le <<= 1;
le2 = le>>1;
ur = 1.0;
ui = 0.0;
arg = Math_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;
}
}
}
/* Godot code again */
/* clang-format on */
void AudioEffectPitchShiftInstance::process(const AudioFrame *p_src_frames, AudioFrame *p_dst_frames, int p_frame_count) {
float sample_rate = AudioServer::get_singleton()->get_mix_rate();
float *in_l = (float *)p_src_frames;
float *in_r = in_l + 1;
float *out_l = (float *)p_dst_frames;
float *out_r = out_l + 1;
shift_l.PitchShift(base->pitch_scale, p_frame_count, 2048, 4, sample_rate, in_l, out_l, 2);
shift_r.PitchShift(base->pitch_scale, p_frame_count, 2048, 4, sample_rate, in_r, out_r, 2);
}
Ref<AudioEffectInstance> AudioEffectPitchShift::instance() {
Ref<AudioEffectPitchShiftInstance> ins;
ins.instance();
ins->base = Ref<AudioEffectPitchShift>(this);
return ins;
}
void AudioEffectPitchShift::set_pitch_scale(float p_adjust) {
pitch_scale = p_adjust;
}
float AudioEffectPitchShift::get_pitch_scale() const {
return pitch_scale;
}
void AudioEffectPitchShift::_bind_methods() {
ClassDB::bind_method(D_METHOD("set_pitch_scale", "rate"), &AudioEffectPitchShift::set_pitch_scale);
ClassDB::bind_method(D_METHOD("get_pitch_scale"), &AudioEffectPitchShift::get_pitch_scale);
ADD_PROPERTY(PropertyInfo(Variant::REAL, "pitch_scale", PROPERTY_HINT_RANGE, "0.01,16,0.01"), "set_pitch_scale", "get_pitch_scale");
}
AudioEffectPitchShift::AudioEffectPitchShift() {
pitch_scale = 1.0;
}
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