369 lines
13 KiB
C++
369 lines
13 KiB
C++
/**************************************************************************/
|
|
/* audio_effect_pitch_shift.cpp */
|
|
/**************************************************************************/
|
|
/* This file is part of: */
|
|
/* GODOT ENGINE */
|
|
/* https://godotengine.org */
|
|
/**************************************************************************/
|
|
/* Copyright (c) 2014-present Godot Engine contributors (see AUTHORS.md). */
|
|
/* Copyright (c) 2007-2014 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 */
|
|
/* CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, */
|
|
/* 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 "core/math/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: https://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 https://dspguru.com/wide-open-license/ 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;
|
|
unsigned long fftFrameBufferSize;
|
|
|
|
/* set up some handy variables */
|
|
fftFrameBufferSize = (unsigned long)fftFrameSize*sizeof(float);
|
|
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, fftFrameBufferSize);
|
|
memset(gSynFreq, 0, fftFrameBufferSize);
|
|
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, fftFrameBufferSize);
|
|
|
|
/* 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, fft_size, base->oversampling, sample_rate, in_l, out_l, 2);
|
|
shift_r.PitchShift(base->pitch_scale, p_frame_count, fft_size, base->oversampling, sample_rate, in_r, out_r, 2);
|
|
}
|
|
|
|
Ref<AudioEffectInstance> AudioEffectPitchShift::instantiate() {
|
|
Ref<AudioEffectPitchShiftInstance> ins;
|
|
ins.instantiate();
|
|
ins->base = Ref<AudioEffectPitchShift>(this);
|
|
static const int fft_sizes[FFT_SIZE_MAX] = { 256, 512, 1024, 2048, 4096 };
|
|
ins->fft_size = fft_sizes[fft_size];
|
|
|
|
return ins;
|
|
}
|
|
|
|
void AudioEffectPitchShift::set_pitch_scale(float p_pitch_scale) {
|
|
ERR_FAIL_COND(!(p_pitch_scale > 0.0));
|
|
pitch_scale = p_pitch_scale;
|
|
}
|
|
|
|
float AudioEffectPitchShift::get_pitch_scale() const {
|
|
return pitch_scale;
|
|
}
|
|
|
|
void AudioEffectPitchShift::set_oversampling(int p_oversampling) {
|
|
ERR_FAIL_COND(p_oversampling < 4);
|
|
oversampling = p_oversampling;
|
|
}
|
|
|
|
int AudioEffectPitchShift::get_oversampling() const {
|
|
return oversampling;
|
|
}
|
|
|
|
void AudioEffectPitchShift::set_fft_size(FFTSize p_fft_size) {
|
|
ERR_FAIL_INDEX(p_fft_size, FFT_SIZE_MAX);
|
|
fft_size = p_fft_size;
|
|
}
|
|
|
|
AudioEffectPitchShift::FFTSize AudioEffectPitchShift::get_fft_size() const {
|
|
return fft_size;
|
|
}
|
|
|
|
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);
|
|
|
|
ClassDB::bind_method(D_METHOD("set_oversampling", "amount"), &AudioEffectPitchShift::set_oversampling);
|
|
ClassDB::bind_method(D_METHOD("get_oversampling"), &AudioEffectPitchShift::get_oversampling);
|
|
|
|
ClassDB::bind_method(D_METHOD("set_fft_size", "size"), &AudioEffectPitchShift::set_fft_size);
|
|
ClassDB::bind_method(D_METHOD("get_fft_size"), &AudioEffectPitchShift::get_fft_size);
|
|
|
|
ADD_PROPERTY(PropertyInfo(Variant::FLOAT, "pitch_scale", PROPERTY_HINT_RANGE, "0.01,16,0.01"), "set_pitch_scale", "get_pitch_scale");
|
|
ADD_PROPERTY(PropertyInfo(Variant::FLOAT, "oversampling", PROPERTY_HINT_RANGE, "4,32,1"), "set_oversampling", "get_oversampling");
|
|
ADD_PROPERTY(PropertyInfo(Variant::INT, "fft_size", PROPERTY_HINT_ENUM, "256,512,1024,2048,4096"), "set_fft_size", "get_fft_size");
|
|
|
|
BIND_ENUM_CONSTANT(FFT_SIZE_256);
|
|
BIND_ENUM_CONSTANT(FFT_SIZE_512);
|
|
BIND_ENUM_CONSTANT(FFT_SIZE_1024);
|
|
BIND_ENUM_CONSTANT(FFT_SIZE_2048);
|
|
BIND_ENUM_CONSTANT(FFT_SIZE_4096);
|
|
BIND_ENUM_CONSTANT(FFT_SIZE_MAX);
|
|
}
|
|
|
|
AudioEffectPitchShift::AudioEffectPitchShift() {
|
|
pitch_scale = 1.0;
|
|
oversampling = 4;
|
|
fft_size = FFT_SIZE_2048;
|
|
wet = 0.0;
|
|
dry = 0.0;
|
|
filter = false;
|
|
}
|