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freeswitch/libs/spandsp/tests/make_g168_css.c
Steve Underwood 32f70174f5 Tweaks to spandsp tests
2011-12-06 01:00:22 +08:00

434 lines
13 KiB
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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.
*/
/*! \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 <sndfile.h>
#if defined(HAVE_FFTW3_H)
#include <fftw3.h>
#else
#include <fftw.h>
#endif
//#if defined(WITH_SPANDSP_INTERNALS)
#define SPANDSP_EXPOSE_INTERNAL_STRUCTURES
//#endif
#include "spandsp.h"
#include "spandsp/g168models.h"
#if !defined(NULL)
#define NULL (void *) 0
#endif
#define FAST_SAMPLE_RATE 44100.0
#define C1_VOICED_SAMPLES 2144 /* 48.62ms at 44100 samples/second => 2144.142 */
#define C1_NOISE_SAMPLES 8820 /* 200ms at 44100 samples/second => 8820.0 */
#define C1_SILENCE_SAMPLES 4471 /* 101.38ms at 44100 samples/second => 4470.858 */
#define C3_VOICED_SAMPLES 3206 /* 72.69ms at 44100 samples/second => 3205.629 */
#define C3_NOISE_SAMPLES 8820 /* 200ms at 44100 samples/second => 8820.0 */
#define C3_SILENCE_SAMPLES 5614 /* 127.31ms at 44100 samples/second => 5614.371 */
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;
}
/*- End of function --------------------------------------------------------*/
static double peak(const int16_t amp[], int len)
{
int16_t peak;
int i;
peak = 0;
for (i = 0; i < len; i++)
{
if (abs(amp[i]) > peak)
peak = abs(amp[i]);
}
return peak/32767.0;
}
/*- End of function --------------------------------------------------------*/
static double rms(const int16_t amp[], int len)
{
double ms;
int i;
ms = 0.0;
for (i = 0; i < len; i++)
ms += amp[i]*amp[i];
return sqrt(ms/len)/32767.0;
}
/*- End of function --------------------------------------------------------*/
static double rms_to_dbm0(double rms)
{
return 20.0*log10(rms) + DBM0_MAX_POWER;
}
/*- End of function --------------------------------------------------------*/
static double rms_to_db(double rms)
{
return 20.0*log10(rms);
}
/*- End of function --------------------------------------------------------*/
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[8830];
int16_t silence_sound[8192];
int i;
int voiced_length;
int randy;
double f;
double pk;
double ms;
double scale;
SNDFILE *filehandle;
SF_INFO info;
awgn_state_t noise_source;
memset(&info, 0, sizeof(info));
info.frames = 0;
info.samplerate = FAST_SAMPLE_RATE;
info.channels = 1;
info.format = SF_FORMAT_WAV | SF_FORMAT_PCM_16;
info.sections = 1;
info.seekable = 1;
if ((filehandle = sf_open("sound_c1.wav", SFM_WRITE, &info)) == NULL)
{
fprintf(stderr, " Failed to open result file\n");
exit(2);
}
printf("Generate C1\n");
/* The sequence is
48.62ms of voiced sound from table C.1 of G.168
200.0ms of pseudo-noise
101.38ms of silence
The above is then repeated phase inverted.
The voice comes straight from C.1, repeated enough times to
fill out the 48.62ms period - i.e. 16 copies of the sequence.
The pseudo noise section is random numbers filtered by the spectral
pattern in Figure C.2 */
/* The set of C1 voice samples is ready for use in the output file. */
voiced_length = sizeof(css_c1)/sizeof(css_c1[0]);
for (i = 0; i < voiced_length; i++)
voiced_sound[i] = css_c1[i];
pk = peak(voiced_sound, voiced_length);
ms = rms(voiced_sound, voiced_length);
printf("Voiced level = %.2fdB, crest factor = %.2fdB\n", rms_to_dbm0(ms), rms_to_db(pk/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
randy = ((rand() >> 10) & 0x1) ? 1.0 : -1.0;
#if defined(HAVE_FFTW3_H)
in[i][0] = randy*pow(10.0, scale/20.0)*35.0;
in[8192 - i][0] = -in[i][0];
#else
in[i].re = randy*pow(10.0, scale/20.0)*35.0;
in[8192 - i].re = -in[i].re;
#endif
}
#if defined(HAVE_FFTW3_H)
fftw_execute(p);
#else
fftw_one(p, in, out);
#endif
for (i = 0; i < 8192; i++)
{
#if defined(HAVE_FFTW3_H)
noise_sound[i] = out[i][1];
#else
noise_sound[i] = out[i].im;
#endif
}
pk = peak(noise_sound, 8192);
ms = rms(noise_sound, 8192);
printf("Filtered noise level = %.2fdB, crest factor = %.2fdB\n", rms_to_dbm0(ms), rms_to_db(pk/ms));
for (i = 0; i < 8192; i++)
silence_sound[i] = 0.0;
for (i = 0; i < 16; i++)
sf_writef_short(filehandle, voiced_sound, voiced_length);
printf("%d samples of voice\n", 16*voiced_length);
sf_writef_short(filehandle, noise_sound, 8192);
sf_writef_short(filehandle, noise_sound, C1_NOISE_SAMPLES - 8192);
printf("%d samples of noise\n", C1_NOISE_SAMPLES);
sf_writef_short(filehandle, silence_sound, C1_SILENCE_SAMPLES);
printf("%d samples of silence\n", C1_SILENCE_SAMPLES);
/* Now phase invert the C1 set of samples. */
voiced_length = sizeof(css_c1)/sizeof(css_c1[0]);
for (i = 0; i < voiced_length; i++)
voiced_sound[i] = -css_c1[i];
for (i = 0; i < 8192; i++)
noise_sound[i] = -noise_sound[i];
for (i = 0; i < 16; i++)
sf_writef_short(filehandle, voiced_sound, voiced_length);
printf("%d samples of voice\n", 16*voiced_length);
sf_writef_short(filehandle, noise_sound, 8192);
sf_writef_short(filehandle, noise_sound, C1_NOISE_SAMPLES - 8192);
printf("%d samples of noise\n", C1_NOISE_SAMPLES);
sf_writef_short(filehandle, silence_sound, C1_SILENCE_SAMPLES);
printf("%d samples of silence\n", C1_SILENCE_SAMPLES);
if (sf_close(filehandle))
{
fprintf(stderr, " Cannot close speech file '%s'\n", "sound_c1.wav");
exit(2);
}
memset(&info, 0, sizeof(info));
info.frames = 0;
info.samplerate = FAST_SAMPLE_RATE;
info.channels = 1;
info.format = SF_FORMAT_WAV | SF_FORMAT_PCM_16;
info.sections = 1;
info.seekable = 1;
if ((filehandle = sf_open("sound_c3.wav", SFM_WRITE, &info)) == NULL)
{
fprintf(stderr, " Failed to open result file\n");
exit(2);
}
printf("Generate C3\n");
/* The sequence is
72.69ms of voiced sound from table C.3 of G.168
200.0ms of pseudo-noise
127.31ms of silence
The above is then repeated phase inverted.
The voice comes straight from C.3, repeated enough times to
fill out the 72.69ms period - i.e. 14 copies of the sequence.
The pseudo noise section is AWGN filtered by the spectral
pattern in Figure C.2. Since AWGN has the quality of being its
own Fourier transform, we can use an approach like the one above
for the C1 signal, using AWGN samples instead of randomly alternating
ones and zeros. */
/* Take the supplied set of C3 voice samples. */
voiced_length = (sizeof(css_c3)/sizeof(css_c3[0]));
for (i = 0; i < voiced_length; i++)
voiced_sound[i] = css_c3[i];
pk = peak(voiced_sound, voiced_length);
ms = rms(voiced_sound, voiced_length);
printf("Voiced level = %.2fdB, crest factor = %.2fdB\n", rms_to_dbm0(ms), rms_to_db(pk/ms));
awgn_init_dbm0(&noise_source, 7162534, rms_to_dbm0(ms));
for (i = 0; i < 8192; i++)
noise_sound[i] = awgn(&noise_source);
pk = peak(noise_sound, 8192);
ms = rms(noise_sound, 8192);
printf("Unfiltered noise level = %.2fdB, crest factor = %.2fdB\n", rms_to_dbm0(ms), rms_to_db(pk/ms));
/* Now filter them */
#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] = noise_sound[i]*pow(10.0, scale/20.0)*0.0106;
in[8192 - i][0] = -in[i][0];
#else
in[i].re = noise_sound[i]*pow(10.0, scale/20.0)*0.0106;
in[8192 - i].re = -in[i].re;
#endif
}
#if defined(HAVE_FFTW3_H)
fftw_execute(p);
#else
fftw_one(p, in, out);
#endif
for (i = 0; i < 8192; i++)
{
#if defined(HAVE_FFTW3_H)
noise_sound[i] = out[i][1];
#else
noise_sound[i] = out[i].im;
#endif
}
pk = peak(noise_sound, 8192);
ms = rms(noise_sound, 8192);
printf("Filtered noise level = %.2fdB, crest factor = %.2fdB\n", rms_to_dbm0(ms), rms_to_db(pk/ms));
for (i = 0; i < 14; i++)
sf_writef_short(filehandle, voiced_sound, voiced_length);
printf("%d samples of voice\n", 14*voiced_length);
sf_writef_short(filehandle, noise_sound, 8192);
sf_writef_short(filehandle, noise_sound, C3_NOISE_SAMPLES - 8192);
printf("%d samples of noise\n", C3_NOISE_SAMPLES);
sf_writef_short(filehandle, silence_sound, C3_SILENCE_SAMPLES);
printf("%d samples of silence\n", C3_SILENCE_SAMPLES);
/* Now phase invert the C3 set of samples. */
voiced_length = (sizeof(css_c3)/sizeof(css_c3[0]));
for (i = 0; i < voiced_length; i++)
voiced_sound[i] = -css_c3[i];
for (i = 0; i < 8192; i++)
noise_sound[i] = -noise_sound[i];
for (i = 0; i < 14; i++)
sf_writef_short(filehandle, voiced_sound, voiced_length);
printf("%d samples of voice\n", 14*voiced_length);
sf_writef_short(filehandle, noise_sound, 8192);
sf_writef_short(filehandle, noise_sound, C3_NOISE_SAMPLES - 8192);
printf("%d samples of noise\n", C3_NOISE_SAMPLES);
sf_writef_short(filehandle, silence_sound, C3_SILENCE_SAMPLES);
printf("%d samples of silence\n", C3_SILENCE_SAMPLES);
if (sf_close(filehandle))
{
fprintf(stderr, " Cannot close speech file '%s'\n", "sound_c3.wav");
exit(2);
}
fftw_destroy_plan(p);
return 0;
}
/*- End of function --------------------------------------------------------*/
/*- End of file ------------------------------------------------------------*/
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