33b5c57199
- Renames PackedIntArray to PackedInt32Array. - Renames PackedFloatArray to PackedFloat32Array. - Adds PackedInt64Array and PackedFloat64Array. - Renames Variant::REAL to Variant::FLOAT for consistency. Packed arrays are for storing large amount of data and creating stuff like meshes, buffers. textures, etc. Forcing them to be 64 is a huge waste of memory. That said, many users requested the ability to have 64 bits packed arrays for their games, so this is just an optional added type. For Variant, the float datatype is always 64 bits, and exposed as `float`. We still have `real_t` which is the datatype that can change from 32 to 64 bits depending on a compile flag (not entirely working right now, but that's the idea). It affects math related datatypes and code only. Neither Variant nor PackedArray make use of real_t, which is only intended for math precision, so the term is removed from there to keep only float.
367 lines
13 KiB
C++
367 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) 2007-2020 Juan Linietsky, Ariel Manzur. */
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/* Copyright (c) 2014-2020 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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/* 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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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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/* 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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/* 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((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::FLOAT, "pitch_scale", PROPERTY_HINT_RANGE, "0.01,16,0.01"), "set_pitch_scale", "get_pitch_scale");
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ADD_PROPERTY(PropertyInfo(Variant::FLOAT, "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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