path: root/servers/audio/effects/audio_effect_pitch_shift.cpp

#include "audio_effect_pitch_shift.h"
#include "servers/audio_server.h"
#include "math_funcs.h"
/****************************************************************************
*
* 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(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;
		}
	}
}


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(_MD("set_pitch_scale","rate"),&AudioEffectPitchShift::set_pitch_scale);
	ClassDB::bind_method(_MD("get_pitch_scale"),&AudioEffectPitchShift::get_pitch_scale);

	ADD_PROPERTY(PropertyInfo(Variant::REAL,"pitch_scale",PROPERTY_HINT_RANGE,"0.01,16,0.01"),_SCS("set_pitch_scale"),_SCS("get_pitch_scale"));

}

AudioEffectPitchShift::AudioEffectPitchShift() {
	pitch_scale=1.0;

}