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/*
* SpanDSP - a series of DSP components for telephony
*
* makecss.c - Create the composite source signal (CSS) for G.168 testing.
*
* Written by Steve Underwood <steveu@coppice.org>
*
* Copyright (C) 2003 Steve Underwood
*
* All rights reserved.
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2, as
* published by the Free Software Foundation.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*
* $Id: make_g168_css.c,v 1.14 2008/05/13 13:17:26 steveu Exp $
*/
/*! \page makecss_page CSS construction for G.168 testing
\section makecss_page_sec_1 What does it do?
???.
\section makecss_page_sec_2 How does it work?
???.
*/
#if defined(HAVE_CONFIG_H)
#include "config.h"
#endif
#include <stdlib.h>
#include <unistd.h>
#include <string.h>
#include <time.h>
#include <stdio.h>
#include <fcntl.h>
#include <audiofile.h>
#if defined(HAVE_FFTW3_H)
#include <fftw3.h>
#else
#include <fftw.h>
#endif
#if defined(HAVE_MATH_H)
#define GEN_CONST
#endif
#include "spandsp.h"
#include "spandsp/g168models.h"
#if !defined(NULL)
#define NULL (void *) 0
#endif
#define FAST_SAMPLE_RATE 44100.0
static double scaling(double f, double start, double end, double start_gain, double end_gain)
{
double scale;
scale = start_gain + (f - start)*(end_gain - start_gain)/(end - start);
return scale;
}
int main(int argc, char *argv[])
{
#if defined(HAVE_FFTW3_H)
double in[8192][2];
double out[8192][2];
#else
fftw_complex in[8192];
fftw_complex out[8192];
#endif
fftw_plan p;
int16_t voiced_sound[8192];
int16_t noise_sound[8192];
int16_t silence_sound[8192];
int i;
int j;
int outframes;
int voiced_length;
float f;
double peak;
double ms;
double scale;
AFfilehandle filehandle;
AFfilesetup filesetup;
awgn_state_t noise_source;
filesetup = afNewFileSetup();
if (filesetup == AF_NULL_FILESETUP)
{
fprintf(stderr, " Failed to create file setup\n");
exit(2);
}
afInitSampleFormat(filesetup, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 16);
afInitRate(filesetup, AF_DEFAULT_TRACK, FAST_SAMPLE_RATE);
afInitFileFormat(filesetup, AF_FILE_WAVE);
afInitChannels(filesetup, AF_DEFAULT_TRACK, 1);
filehandle = afOpenFile("sound_c1.wav", "w", filesetup);
if (filehandle == AF_NULL_FILEHANDLE)
{
fprintf(stderr, " Failed to open result file\n");
exit(2);
}
for (i = 0; i < (int)(sizeof(css_c1)/sizeof(css_c1[0])); i++)
voiced_sound[i] = css_c1[i];
voiced_length = i;
ms = 0;
peak = 0;
for (i = 0; i < voiced_length; i++)
{
if (abs(voiced_sound[i]) > peak)
peak = abs(voiced_sound[i]);
ms += voiced_sound[i]*voiced_sound[i];
}
ms = 20.0*log10(sqrt(ms/voiced_length)/32767.0) + DBM0_MAX_POWER;
printf("Voiced level = %.2fdB\n", ms);
#if defined(HAVE_FFTW3_H)
p = fftw_plan_dft_1d(8192, in, out, FFTW_BACKWARD, FFTW_ESTIMATE);
#else
p = fftw_create_plan(8192, FFTW_BACKWARD, FFTW_ESTIMATE);
#endif
for (i = 0; i < 8192; i++)
{
#if defined(HAVE_FFTW3_H)
in[i][0] = 0.0;
in[i][1] = 0.0;
#else
in[i].re = 0.0;
in[i].im = 0.0;
#endif
}
for (i = 1; i <= 3715; i++)
{
f = FAST_SAMPLE_RATE*i/8192.0;
#if 1
if (f < 50.0)
scale = -60.0;
else if (f < 100.0)
scale = scaling(f, 50.0, 100.0, -25.8, -12.8);
else if (f < 200.0)
scale = scaling(f, 100.0, 200.0, -12.8, 17.4);
else if (f < 215.0)
scale = scaling(f, 200.0, 215.0, 17.4, 17.8);
else if (f < 500.0)
scale = scaling(f, 215.0, 500.0, 17.8, 12.2);
else if (f < 1000.0)
scale = scaling(f, 500.0, 1000.0, 12.2, 7.2);
else if (f < 2850.0)
scale = scaling(f, 1000.0, 2850.0, 7.2, 0.0);
else if (f < 3600.0)
scale = scaling(f, 2850.0, 3600.0, 0.0, -2.0);
else if (f < 3660.0)
scale = scaling(f, 3600.0, 3660.0, -2.0, -20.0);
else if (f < 3680.0)
scale = scaling(f, 3600.0, 3680.0, -20.0, -30.0);
else
scale = -60.0;
#else
scale = 0.0;
#endif
#if defined(HAVE_FFTW3_H)
in[i][0] = 0.0;
in[i][1] = (rand() & 0x1) ? 1.0 : -1.0;
in[i][1] *= pow(10.0, scale/20.0);
in[i][1] *= 35.0; //305360
//printf("%10d %15.5f %15.5f\n", i, in[i].re, in[i].im);
in[8192 - i][0] = 0.0;
in[8192 - i][1] = -in[i][1];
//printf("%10d %15.5f %15.5f\n", i, in[i][0], in[i][1]);
#else
in[i].re = 0.0;
in[i].im = (rand() & 0x1) ? 1.0 : -1.0;
in[i].im *= pow(10.0, scale/20.0);
in[i].im *= 35.0; //305360
in[8192 - i].re = 0.0;
in[8192 - i].im = -in[i].im;
//printf("%10d %15.5f %15.5f\n", i, in[i].re, in[i].im);
#endif
}
#if defined(HAVE_FFTW3_H)
fftw_execute(p);
#else
fftw_one(p, in, out);
#endif
for (i = 0; i < 8192; i++)
{
//printf("%10d %15.5f %15.5f\n", i, out[i].re, out[i].im);
#if defined(HAVE_FFTW3_H)
noise_sound[i] = out[i][0];
#else
noise_sound[i] = out[i].re;
#endif
}
peak = 0;
ms = 0;
for (i = 0; i < 8192; i++)
{
if (abs(noise_sound[i]) > peak)
peak = abs(noise_sound[i]);
ms += noise_sound[i]*noise_sound[i];
}
printf("Noise level = %.2fdB\n", 20.0*log10(sqrt(ms/8192.0)/32767.0) + DBM0_MAX_POWER);
printf("Crest factor = %.2fdB\n", 20.0*log10(peak/sqrt(ms/8192.0)));
for (i = 0; i < 8192; i++)
silence_sound[i] = 0.0;
for (j = 0; j < 16; j++)
{
outframes = afWriteFrames(filehandle,
AF_DEFAULT_TRACK,
voiced_sound,
voiced_length);
}
printf("%d samples of voice\n", 16*voiced_length);
outframes = afWriteFrames(filehandle,
AF_DEFAULT_TRACK,
noise_sound,
8192);
outframes = afWriteFrames(filehandle,
AF_DEFAULT_TRACK,
noise_sound,
8820 - 8192);
printf("%d samples of noise\n", 8820);
outframes = afWriteFrames(filehandle,
AF_DEFAULT_TRACK,
silence_sound,
4471);
printf("%d samples of silence\n", 4471);
for (j = 0; j < voiced_length; j++)
voiced_sound[j] = -voiced_sound[j];
for (j = 0; j < 8192; j++)
noise_sound[j] = -noise_sound[j];
for (j = 0; j < 16; j++)
{
outframes = afWriteFrames(filehandle,
AF_DEFAULT_TRACK,
voiced_sound,
voiced_length);
}
printf("%d samples of voice\n", 16*voiced_length);
outframes = afWriteFrames(filehandle,
AF_DEFAULT_TRACK,
noise_sound,
8192);
outframes = afWriteFrames(filehandle,
AF_DEFAULT_TRACK,
noise_sound,
8820 - 8192);
printf("%d samples of noise\n", 8820);
outframes = afWriteFrames(filehandle,
AF_DEFAULT_TRACK,
silence_sound,
4471);
printf("%d samples of silence\n", 4471);
if (afCloseFile(filehandle) != 0)
{
fprintf(stderr, " Cannot close speech file '%s'\n", "sound_c1.wav");
exit(2);
}
afInitSampleFormat(filesetup, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 16);
afInitRate(filesetup, AF_DEFAULT_TRACK, FAST_SAMPLE_RATE);
afInitFileFormat(filesetup, AF_FILE_WAVE);
afInitChannels(filesetup, AF_DEFAULT_TRACK, 1);
filehandle = afOpenFile("sound_c3.wav", "w", filesetup);
if (filehandle == AF_NULL_FILEHANDLE)
{
fprintf(stderr, " Failed to open result file\n");
exit(2);
}
for (i = 0; i < (int) (sizeof(css_c3)/sizeof(css_c3[0])); i++)
voiced_sound[i] = css_c3[i];
voiced_length = i;
ms = 0;
for (i = 0; i < voiced_length; i++)
{
if (abs(voiced_sound[i]) > peak)
peak = abs(voiced_sound[i]);
ms += voiced_sound[i]*voiced_sound[i];
}
ms = 20.0*log10(sqrt(ms/voiced_length)/32767.0) + DBM0_MAX_POWER;
printf("Voiced level = %.2fdB\n", ms);
awgn_init_dbm0(&noise_source, 7162534, ms);
for (i = 0; i < 8192; i++)
noise_sound[i] = awgn(&noise_source);
peak = 0;
ms = 0;
for (i = 0; i < 8192; i++)
{
if (abs(noise_sound[i]) > peak)
peak = abs(noise_sound[i]);
ms += noise_sound[i]*noise_sound[i];
}
printf("Noise level = %.2fdB\n", 20.0*log10(sqrt(ms/8192.0)/32767.0) + DBM0_MAX_POWER);
printf("Crest factor = %.2fdB\n", 20.0*log10(peak/sqrt(ms/8192.0)));
for (j = 0; j < 14; j++)
{
outframes = afWriteFrames(filehandle,
AF_DEFAULT_TRACK,
voiced_sound,
voiced_length);
}
printf("%d samples of voice\n", 14*i);
outframes = afWriteFrames(filehandle,
AF_DEFAULT_TRACK,
noise_sound,
8192);
outframes = afWriteFrames(filehandle,
AF_DEFAULT_TRACK,
noise_sound,
8820 - 8192);
printf("%d samples of noise\n", 8820);
outframes = afWriteFrames(filehandle,
AF_DEFAULT_TRACK,
silence_sound,
5614);
printf("%d samples of silence\n", 5614);
for (j = 0; j < voiced_length; j++)
voiced_sound[j] = -voiced_sound[j];
for (j = 0; j < 8192; j++)
noise_sound[j] = -noise_sound[j];
for (j = 0; j < 14; j++)
{
outframes = afWriteFrames(filehandle,
AF_DEFAULT_TRACK,
voiced_sound,
voiced_length);
}
printf("%d samples of voice\n", 14*i);
outframes = afWriteFrames(filehandle,
AF_DEFAULT_TRACK,
noise_sound,
8192);
outframes = afWriteFrames(filehandle,
AF_DEFAULT_TRACK,
noise_sound,
8820 - 8192);
printf("%d samples of noise\n", 8820);
outframes = afWriteFrames(filehandle,
AF_DEFAULT_TRACK,
silence_sound,
5614);
printf("%d samples of silence\n", 5614);
if (afCloseFile(filehandle) != 0)
{
fprintf(stderr, " Cannot close speech file '%s'\n", "sound_c3.wav");
exit(2);
}
afFreeFileSetup(filesetup);
fftw_destroy_plan(p);
return 0;
}
/*- End of function --------------------------------------------------------*/
/*- End of file ------------------------------------------------------------*/
|
