summaryrefslogtreecommitdiff
path: root/servers/audio
diff options
context:
space:
mode:
authorJuan Linietsky <reduzio@gmail.com>2017-01-23 01:45:43 -0300
committerJuan Linietsky <reduzio@gmail.com>2017-01-23 01:45:43 -0300
commitcff80bb1cc0aa5a548254b9c89d0fab28982b504 (patch)
treecea35b87f03ec7a37d82b77a2c9a4ee196c69272 /servers/audio
parent4d9bc8b00cd9a561005ad145b3b95c58d4fbfa00 (diff)
Pretty high quality pitch Shifting effect using smbPitchShift
Diffstat (limited to 'servers/audio')
-rw-r--r--servers/audio/effects/audio_effect_pitch_shift.cpp297
-rw-r--r--servers/audio/effects/audio_effect_pitch_shift.h89
2 files changed, 386 insertions, 0 deletions
diff --git a/servers/audio/effects/audio_effect_pitch_shift.cpp b/servers/audio/effects/audio_effect_pitch_shift.cpp
new file mode 100644
index 0000000000..a11ae37675
--- /dev/null
+++ b/servers/audio/effects/audio_effect_pitch_shift.cpp
@@ -0,0 +1,297 @@
+#include "audio_effect_pitch_shift.h"
+#include "servers/audio_server.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.*M_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.*M_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/M_PI;
+ if (qpd >= 0) qpd += qpd&1;
+ else qpd -= qpd&1;
+ tmp -= M_PI*(double)qpd;
+
+ /* get deviation from bin frequency from the +/- Pi interval */
+ tmp = osamp*tmp/(2.*M_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.*M_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.*M_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 = M_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;
+
+}
diff --git a/servers/audio/effects/audio_effect_pitch_shift.h b/servers/audio/effects/audio_effect_pitch_shift.h
new file mode 100644
index 0000000000..d1343a0745
--- /dev/null
+++ b/servers/audio/effects/audio_effect_pitch_shift.h
@@ -0,0 +1,89 @@
+#ifndef AUDIO_EFFECT_PITCH_SHIFT_H
+#define AUDIO_EFFECT_PITCH_SHIFT_H
+
+
+#include "servers/audio/audio_effect.h"
+
+class SMBPitchShift {
+
+ enum {
+ MAX_FRAME_LENGTH=8192
+ };
+
+ float gInFIFO[MAX_FRAME_LENGTH];
+ float gOutFIFO[MAX_FRAME_LENGTH];
+ float gFFTworksp[2*MAX_FRAME_LENGTH];
+ float gLastPhase[MAX_FRAME_LENGTH/2+1];
+ float gSumPhase[MAX_FRAME_LENGTH/2+1];
+ float gOutputAccum[2*MAX_FRAME_LENGTH];
+ float gAnaFreq[MAX_FRAME_LENGTH];
+ float gAnaMagn[MAX_FRAME_LENGTH];
+ float gSynFreq[MAX_FRAME_LENGTH];
+ float gSynMagn[MAX_FRAME_LENGTH];
+ long gRover;
+
+ void smbFft(float *fftBuffer, long fftFrameSize, long sign);
+public:
+ void PitchShift(float pitchShift, long numSampsToProcess, long fftFrameSize, long osamp, float sampleRate, float *indata, float *outdata, int stride);
+
+ SMBPitchShift() {
+ gRover=0;
+ memset(gInFIFO, 0, MAX_FRAME_LENGTH*sizeof(float));
+ memset(gOutFIFO, 0, MAX_FRAME_LENGTH*sizeof(float));
+ memset(gFFTworksp, 0, 2*MAX_FRAME_LENGTH*sizeof(float));
+ memset(gLastPhase, 0, (MAX_FRAME_LENGTH/2+1)*sizeof(float));
+ memset(gSumPhase, 0, (MAX_FRAME_LENGTH/2+1)*sizeof(float));
+ memset(gOutputAccum, 0, 2*MAX_FRAME_LENGTH*sizeof(float));
+ memset(gAnaFreq, 0, MAX_FRAME_LENGTH*sizeof(float));
+ memset(gAnaMagn, 0, MAX_FRAME_LENGTH*sizeof(float));
+ }
+
+
+};
+
+
+class AudioEffectPitchShift;
+
+class AudioEffectPitchShiftInstance : public AudioEffectInstance {
+ GDCLASS(AudioEffectPitchShiftInstance,AudioEffectInstance)
+friend class AudioEffectPitchShift;
+ Ref<AudioEffectPitchShift> base;
+
+ SMBPitchShift shift_l;
+ SMBPitchShift shift_r;
+
+
+public:
+
+ virtual void process(const AudioFrame *p_src_frames,AudioFrame *p_dst_frames,int p_frame_count);
+
+};
+
+
+class AudioEffectPitchShift : public AudioEffect {
+ GDCLASS(AudioEffectPitchShift,AudioEffect)
+
+friend class AudioEffectPitchShiftInstance;
+
+ float pitch_scale;
+ int window_size;
+ float wet;
+ float dry;
+ bool filter;
+
+protected:
+
+ static void _bind_methods();
+public:
+
+
+ Ref<AudioEffectInstance> instance();
+
+ void set_pitch_scale(float p_adjust);
+ float get_pitch_scale() const;
+
+ AudioEffectPitchShift();
+};
+
+
+#endif // AUDIO_EFFECT_PITCH_SHIFT_H