using System; namespace NAudio.Dsp { /**************************************************************************** * * NAME: PitchShift.cs * VERSION: 1.0 * 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 * * 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/ * *****************************************************************************/ /// /// SMB Pitch Shifter /// public class SmbPitchShifter { private static int MAX_FRAME_LENGTH = 16000; private float[] gInFIFO = new float[MAX_FRAME_LENGTH]; private float[] gOutFIFO = new float[MAX_FRAME_LENGTH]; private float[] gFFTworksp = new float[2*MAX_FRAME_LENGTH]; private float[] gLastPhase = new float[MAX_FRAME_LENGTH/2 + 1]; private float[] gSumPhase = new float[MAX_FRAME_LENGTH/2 + 1]; private float[] gOutputAccum = new float[2*MAX_FRAME_LENGTH]; private float[] gAnaFreq = new float[MAX_FRAME_LENGTH]; private float[] gAnaMagn = new float[MAX_FRAME_LENGTH]; private float[] gSynFreq = new float[MAX_FRAME_LENGTH]; private float[] gSynMagn = new float[MAX_FRAME_LENGTH]; private long gRover; /// /// Pitch Shift /// public void PitchShift(float pitchShift, long numSampsToProcess, float sampleRate, float[] indata) { PitchShift(pitchShift, numSampsToProcess, 2048L, 10L, sampleRate, indata); } /// /// Pitch Shift /// public void PitchShift(float pitchShift, long numSampsToProcess, long fftFrameSize, long osamp, float sampleRate, float[] indata) { double magn, phase, tmp, window, real, imag; double 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/(double) fftFrameSize; expct = 2.0*Math.PI*(double) stepSize/(double) fftFrameSize; inFifoLatency = fftFrameSize - stepSize; if (gRover == 0) gRover = inFifoLatency; /* 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]; 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 = -.5*Math.Cos(2.0*Math.PI*(double) k/(double) fftFrameSize) + .5; 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 = 2.0*Math.Sqrt(real*real + imag*imag); phase = Math.Atan2(imag, real); /* compute phase difference */ tmp = phase - gLastPhase[k]; gLastPhase[k] = (float) phase; /* subtract expected phase difference */ tmp -= (double) k*expct; /* map delta phase into +/- Pi interval */ qpd = (long) (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/(2.0*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] = (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; } } /* ***************** 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 = 2.0*Math.PI*tmp/osamp; /* add the overlap phase advance back in */ tmp += (double) 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*Math.Cos(phase)); gFFTworksp[2*k + 1] = (float) (magn*Math.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 = -.5*Math.Cos(2.0*Math.PI*(double) k/(double) fftFrameSize) + .5; 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]; } } } /// /// Short Time Fourier Transform /// public 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) (Math.Log(fftFrameSize)/Math.Log(2.0) + .5); for (k = 0, le = 2; k < max; k++) { le <<= 1; le2 = le >> 1; ur = 1.0F; ui = 0.0F; arg = (float) Math.PI/(le2 >> 1); wr = (float) Math.Cos(arg); wi = (float) (sign*Math.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; } } } } }