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SourceMod Developer
Join Date: Aug 2009
Location: OnGameFrame()
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07-11-2012
, 05:36
Re: VoiceHook - Developer preview of something.
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#100
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Quote:
Originally Posted by Mitchell
I get this when trying to compile that code snippet you gave us:
Code:
1>------ Build started: Project: autogain_mm, Configuration: Release - Orange Box Valve Win32 ------
1> autogain_mm.cpp
1>m:\scripts\mm_dev\mmsource-1-8-mmsource-1.8.7\autogain\smbPitchShift.cpp(253): error C2668: 'log' : ambiguous call to overloaded function
1> C:\Program Files\Microsoft Visual Studio 10.0\VC\include\math.h(575): could be 'long double log(long double)'
1> C:\Program Files\Microsoft Visual Studio 10.0\VC\include\math.h(527): or 'float log(float)'
1> C:\Program Files\Microsoft Visual Studio 10.0\VC\include\math.h(120): or 'double log(double)'
1> while trying to match the argument list '(long)'
I used Tony's Autogain as a base, i just replaced the code with the snippet you gave us, and included the #include "smbPitchShift.cpp" at the top.
some other forums said to replace '#include <math.h>' with '#include <cmath>'
and saw no change to the error |
Here's the modified smbPitchShift.cpp I use:
Code:
/****************************************************************************
*
* NAME: smbPitchShift.cpp
* 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 1999-2009 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.
*
*****************************************************************************/
#include <string.h>
#include <math.h>
#include <stdio.h>
#define M_PI 3.14159265358979323846
#define MAX_FRAME_LENGTH 8192
void smbFft(float *fftBuffer, long fftFrameSize, long sign);
double smbAtan2(double x, double y);
// -----------------------------------------------------------------------------------------------------------------
void smbPitchShift(float pitchShift, long numSampsToProcess, long fftFrameSize, long osamp, float sampleRate, float *indata, float *outdata)
/*
Routine smbPitchShift(). See top of file for explanation
Purpose: doing pitch shifting while maintaining duration using the Short
Time Fourier Transform.
Author: (c)1999-2009 Stephan M. Bernsee <smb [AT] dspdimension [DOT] com>
*/
{
static float gInFIFO[MAX_FRAME_LENGTH];
static float gOutFIFO[MAX_FRAME_LENGTH];
static float gFFTworksp[2*MAX_FRAME_LENGTH];
static float gLastPhase[MAX_FRAME_LENGTH/2+1];
static float gSumPhase[MAX_FRAME_LENGTH/2+1];
static float gOutputAccum[2*MAX_FRAME_LENGTH];
static float gAnaFreq[MAX_FRAME_LENGTH];
static float gAnaMagn[MAX_FRAME_LENGTH];
static float gSynFreq[MAX_FRAME_LENGTH];
static float gSynMagn[MAX_FRAME_LENGTH];
static long gRover = false, gInit = false;
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 == false) gRover = inFifoLatency;
/* initialize our static arrays */
if (gInit == false) {
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));
gInit = true;
}
/* 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*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 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((float)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;
}
}
}
// -----------------------------------------------------------------------------------------------------------------
/*
12/12/02, smb
PLEASE NOTE:
There have been some reports on domain errors when the atan2() function was used
as in the above code. Usually, a domain error should not interrupt the program flow
(maybe except in Debug mode) but rather be handled "silently" and a global variable
should be set according to this error. However, on some occasions people ran into
this kind of scenario, so a replacement atan2() function is provided here.
If you are experiencing domain errors and your program stops, simply replace all
instances of atan2() with calls to the smbAtan2() function below.
*/
double smbAtan2(double x, double y)
{
double signx;
if (x > 0.) signx = 1.;
else signx = -1.;
if (x == 0.) return 0.;
if (y == 0.) return signx * M_PI / 2.;
return atan2(x, y);
}
// -----------------------------------------------------------------------------------------------------------------
// -----------------------------------------------------------------------------------------------------------------
// -----------------------------------------------------------------------------------------------------------------
Make sure you include it as an actual object file in the project, not just as a header.
Quote:
Originally Posted by GoD-Tony
I was still hearing a bit of static with the bad solution I was using. After some more digging, it should really be:
Code:
short sample = (pVoiceData[i+1] & 0xFF) << 8 | pVoiceData[i] & 0xFF;
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Copying from IRC for everyone else, this should actually be:
Code:
short sample = ((short)pVoiceData[i+1] << 8) | (pVoiceData[i] & 0xFF);
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