Hello, I'm working on Audio Algorithm: Pitch-Shifter, unfortunately I can't find Text material about it design, But I found one of the code, implemented on c#: pastebin link.
I would post code here as well, and if you have any idea how I can reduce time or implement new one, please share it, Thanks in advance.
/****************************************************************************
*
* NAME: PitchShift.cs
* VERSION: 1.2
* HOME URL: http://www.dspdimension.com
* 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 1999-2006 Stephan M. Bernsee <smb [AT] dspdimension [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.
*
*****************************************************************************/
/****************************************************************************
*
* This code was converted to C# by Michael Knight
* madmik3 at gmail dot com.
* http://sites.google.com/site/mikescoderama/
*
*****************************************************************************/
using System.Collections;
using System.Collections.Generic;
using UnityEngine;
public static class PitchShifter
{
#region Private Static Memebers
private static int MAX_FRAME_LENGTH = 16000;
private static float[] gInFIFO = new float[MAX_FRAME_LENGTH];
private static float[] gOutFIFO = new float[MAX_FRAME_LENGTH];
private static float[] gFFTworksp = new float[2 * MAX_FRAME_LENGTH];
private static float[] gLastPhase = new float[MAX_FRAME_LENGTH / 2 + 1];
private static float[] gSumPhase = new float[MAX_FRAME_LENGTH / 2 + 1];
private static float[] gOutputAccum = new float[2 * MAX_FRAME_LENGTH];
private static float[] gAnaFreq = new float[MAX_FRAME_LENGTH];
private static float[] gAnaMagn = new float[MAX_FRAME_LENGTH];
private static float[] gSynFreq = new float[MAX_FRAME_LENGTH];
private static float[] gSynMagn = new float[MAX_FRAME_LENGTH];
private static long gRover, gInit;
public static int Progress = 0;
#endregion
#region Public Static Methods
public static float[] PitchShift(float pitchShift, long numSampsToProcess,
float sampleRate, float[] indata)
{
return PitchShift(pitchShift, numSampsToProcess, (long)2048, (long)10, sampleRate, indata);
}
public static float[] PitchShift(float pitchShift, long numSampsToProcess, long fftFrameSize,
long osamp, float sampleRate, float[] indata)
{
int MAX_FRAME_LENGTH = 16000;
gInFIFO = new float[MAX_FRAME_LENGTH];
gOutFIFO = new float[MAX_FRAME_LENGTH];
gFFTworksp = new float[2 * MAX_FRAME_LENGTH];
gLastPhase = new float[MAX_FRAME_LENGTH / 2 + 1];
gSumPhase = new float[MAX_FRAME_LENGTH / 2 + 1];
gOutputAccum = new float[2 * MAX_FRAME_LENGTH];
gAnaFreq = new float[MAX_FRAME_LENGTH];
gAnaMagn = new float[MAX_FRAME_LENGTH];
gSynFreq = new float[MAX_FRAME_LENGTH];
gSynMagn = new float[MAX_FRAME_LENGTH];
gRover = 0; gInit=0;
float magn, phase, tmp, window, real, imag;
float freqPerBin, expct;
long i, k, qpd, index, inFifoLatency, stepSize, fftFrameSize2;
float[] outdata = indata;
/* set up some handy variables */
fftFrameSize2 = fftFrameSize / 2;
stepSize = fftFrameSize / osamp;
freqPerBin = sampleRate / (float)fftFrameSize;
expct = 2.0f * Mathf.PI * (float)stepSize / (float)fftFrameSize;
inFifoLatency = fftFrameSize - stepSize;
if (gRover == 0) gRover = inFifoLatency;
/* main processing loop */
Debug.Log(numSampsToProcess);
for (i = 0; i < numSampsToProcess; i++)
{
Progress = (int)(((i + 1) * 100) / (1.0f * numSampsToProcess));
/* As long as we have not yet collected enough data just read in */
gInFIFO[gRover] = indata[i];
outdata[i] = 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 = -.5f * Mathf.Cos(2.0f * Mathf.PI * (float)k / (float)fftFrameSize) + .5f;
gFFTworksp[2 * k] = (float)(gInFIFO[k] * window);
gFFTworksp[2 * k + 1] = 0.0F;
}
/* ***************** ANALYSIS ******************* */
/* do transform */
ShortTimeFourierTransform(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 = 2.0f * Mathf.Sqrt(real * real + imag * imag);
phase = Mathf.Atan2(imag, real);
/* compute phase difference */
tmp = phase - gLastPhase[k];
gLastPhase[k] = (float)phase;
/* subtract expected phase difference */
tmp -= (float)k * expct;
/* map delta phase into +/- Pi interval */
qpd = (long)(tmp / Mathf.PI);
if (qpd >= 0) qpd += qpd & 1;
else qpd -= qpd & 1;
tmp -= Mathf.PI * (float)qpd;
/* get deviation from bin frequency from the +/- Pi interval */
tmp = osamp * tmp / (2.0f * Mathf.PI);
/* compute the k-th partials' true frequency */
tmp = (float)k * freqPerBin + tmp * freqPerBin;
/* store magnitude and true frequency in analysis arrays */
gAnaMagn[k] = (float)magn;
gAnaFreq[k] = (float)tmp;
}
/* ***************** PROCESSING ******************* */
/* this does the actual pitch shifting */
for (int zero = 0; zero < fftFrameSize; zero++)
{
gSynMagn[zero] = 0;
gSynFreq[zero] = 0;
}
for (k = 0; k <= fftFrameSize2; k++)
{
index = (long)(k * pitchShift);
if (index <= fftFrameSize2)
{
gSynMagn[index] += gAnaMagn[k];
gSynFreq[index] = gAnaFreq[k] * pitchShift;
}
else break;
}
/* ***************** 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 -= (float)k * freqPerBin;
/* get bin deviation from freq deviation */
tmp /= freqPerBin;
/* take osamp into account */
tmp = 2.0f * Mathf.PI * tmp / osamp;
/* add the overlap phase advance back in */
tmp += (float)k * expct;
/* accumulate delta phase to get bin phase */
gSumPhase[k] += (float)tmp;
phase = gSumPhase[k];
/* get real and imag part and re-interleave */
gFFTworksp[2 * k] = (float)(magn * Mathf.Cos(phase));
gFFTworksp[2 * k + 1] = (float)(magn * Mathf.Sin(phase));
}
/* zero negative frequencies */
for (k = fftFrameSize + 2; k < 2 * fftFrameSize; k++) gFFTworksp[k] = 0.0F;
/* do inverse transform */
ShortTimeFourierTransform(gFFTworksp, fftFrameSize, 1);
/* do windowing and add to output accumulator */
for (k = 0; k < fftFrameSize; k++)
{
window = -.5f * Mathf.Cos(2.0f * Mathf.PI * (float)k / (float)fftFrameSize) + .5f;
gOutputAccum[k] += (float)(2.0 * 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));
for (k = 0; k < fftFrameSize; k++)
{
gOutputAccum[k] = gOutputAccum[k + stepSize];
}
/* move input FIFO */
for (k = 0; k < inFifoLatency; k++) gInFIFO[k] = gInFIFO[k + stepSize];
}
}
return outdata;
}
#endregion
#region Private Static Methods
public static void ShortTimeFourierTransform(float[] fftBuffer, long fftFrameSize, long sign)
{
float wr, wi, arg, temp;
float tr, ti, ur, ui;
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) != 0) j++;
j <<= 1;
}
if (i < j)
{
temp = fftBuffer[i];
fftBuffer[i] = fftBuffer[j];
fftBuffer[j] = temp;
temp = fftBuffer[i + 1];
fftBuffer[i + 1] = fftBuffer[j + 1];
fftBuffer[j + 1] = temp;
}
}
long max = (long)(Mathf.Log(fftFrameSize) / Mathf.Log(2.0f) + .5);
for (k = 0, le = 2; k < max; k++)
{
le <<= 1;
le2 = le >> 1;
ur = 1.0F;
ui = 0.0F;
arg = (float)Mathf.PI / (le2 >> 1);
wr = (float)Mathf.Cos(arg);
wi = (float)(sign * Mathf.Sin(arg));
for (j = 0; j < le2; j += 2)
{
for (i = j; i < 2 * fftFrameSize; i += le)
{
tr = fftBuffer[i + le2] * ur - fftBuffer[i + le2 + 1] * ui;
ti = fftBuffer[i + le2] * ui + fftBuffer[i + le2 + 1] * ur;
fftBuffer[i + le2] = fftBuffer[i] - tr;
fftBuffer[i + le2 + 1] = fftBuffer[i + 1] - ti;
fftBuffer[i] += tr;
fftBuffer[i + 1] += ti;
}
tr = ur * wr - ui * wi;
ui = ur * wi + ui * wr;
ur = tr;
}
}
}
#endregion
}
The
window
parameter in the interleavingk
loop of the secondPitchShift()
function is independent of the sample numberi
. You can pre-compute and store these values in an arraywindow[ fftFrameSize ]
, before the outeri
loop, in a dynamic programming sense, and usewindow[ k ]
in the interleaving loop.A C++17 update before the outer loop should look like that:
And the updated interleaving loop should look like that:
The same
window
array can be used later in the following loop.where
osamp_factor
can be computed before the outer loop as:Finally, a more recent C++ version of the C# code is available on Stephan Bernsee's Blog. You may also review the source code of the Rubber Band Library for audio time stretching.
The following is a test program for the proposed updates.
Thank you very much for your help, Sir.
With pleasure.
Please re-check the post and the test program, as the +0.5 factor at the end of the statement in the second loop was overlooked. Actually, both loops use the same expression for
window
. Therefore, the second vectorc_window
in redundant, and it is sufficient to use the former vector only.