/*************************************************************************/ /* 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 */ /* 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 "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 * * 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 */ 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 AudioEffectPitchShift::instance() { Ref ins; ins.instance(); ins->base = Ref(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; }