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freeswitch/libs/spandsp/tests/sig_tone_tests.c

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/*
* SpanDSP - a series of DSP components for telephony
*
* sig_tone_tests.c
*
* Written by Steve Underwood <steveu@coppice.org>
*
* Copyright (C) 2004 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: sig_tone_tests.c,v 1.27 2009/09/23 16:02:59 steveu Exp $
*/
/*! \file */
/*! \page sig_tone_tests_page The signaling tone processor tests
\section sig_tone_tests_sec_1 What does it do?
???.
\section sig_tone_tests_sec_2 How does it work?
???.
*/
#if defined(HAVE_CONFIG_H)
#include "config.h"
#endif
#include <stdlib.h>
#include <stdio.h>
#include <memory.h>
#include <sndfile.h>
//#if defined(WITH_SPANDSP_INTERNALS)
#define SPANDSP_EXPOSE_INTERNAL_STRUCTURES
//#endif
#include "spandsp.h"
#include "spandsp-sim.h"
#define OUT_FILE_NAME "sig_tone.wav"
#define SAMPLES_PER_CHUNK 160
static int sampleno = 0;
static int tone_1_present = 0;
static int tone_2_present = 0;
static int tx_section = 0;
static int dial_pulses = 0;
static void tx_handler(void *user_data, int what, int level, int duration)
{
sig_tone_tx_state_t *s;
s = (sig_tone_tx_state_t *) user_data;
//printf("What - %d, duration - %d\n", what, duration);
if ((what & SIG_TONE_TX_UPDATE_REQUEST))
{
printf("Tx: update request\n");
/* The sig tone transmit side wants to know what to do next */
switch (tx_section)
{
case 0:
printf("33ms break - %d samples\n", ms_to_samples(33));
tx_section++;
sig_tone_tx_set_mode(s, SIG_TONE_1_PRESENT, ms_to_samples(33));
break;
case 1:
printf("67ms make - %d samples\n", ms_to_samples(67));
if (++dial_pulses == 9)
tx_section++;
else
tx_section--;
sig_tone_tx_set_mode(s, 0, ms_to_samples(67));
break;
case 2:
tx_section++;
sig_tone_tx_set_mode(s, SIG_TONE_1_PRESENT, ms_to_samples(600));
break;
case 3:
sig_tone_tx_set_mode(s, SIG_TONE_1_PRESENT | SIG_TONE_TX_PASSTHROUGH, 0);
break;
}
/*endswitch*/
}
/*endif*/
}
/*- End of function --------------------------------------------------------*/
static void rx_handler(void *user_data, int what, int level, int duration)
{
float ms;
ms = 1000.0f*(float) duration/(float) SAMPLE_RATE;
printf("What - %d, duration - %d\n", what, duration);
if ((what & SIG_TONE_1_CHANGE))
{
tone_1_present = what & SIG_TONE_1_PRESENT;
printf("Rx: tone 1 is %s after %d samples (%fms)\n", (tone_1_present) ? "on" : "off", duration, ms);
}
/*endif*/
if ((what & SIG_TONE_2_CHANGE))
{
tone_2_present = what & SIG_TONE_2_PRESENT;
printf("Rx: tone 2 is %s after %d samples (%fms)\n", (tone_2_present) ? "on" : "off", duration, ms);
}
/*endif*/
}
/*- End of function --------------------------------------------------------*/
static void map_frequency_response(sig_tone_rx_state_t *s)
{
int16_t buf[8192];
int i;
int len;
double sumin;
double sumout;
swept_tone_state_t *swept;
/* Things like noise don't highlight the frequency response of the high Q notch
very well. We use a slowly swept frequency to check it. */
swept = swept_tone_init(NULL, 200.0f, 3900.0f, -10.0f, 120*SAMPLE_RATE, 0);
for (;;)
{
if ((len = swept_tone(swept, buf, SAMPLES_PER_CHUNK)) <= 0)
break;
sumin = 0.0;
for (i = 0; i < len; i++)
sumin += (double) buf[i]*(double) buf[i];
sig_tone_rx(s, buf, len);
sumout = 0.0;
for (i = 0; i < len; i++)
sumout += (double) buf[i]*(double) buf[i];
/*endfor*/
printf("%7.1f %f\n", swept_tone_current_frequency(swept), 10.0*log10(sumout/sumin));
}
/*endfor*/
swept_tone_free(swept);
}
/*- End of function --------------------------------------------------------*/
int main(int argc, char *argv[])
{
int16_t amp[SAMPLES_PER_CHUNK];
int16_t out_amp[2*SAMPLES_PER_CHUNK];
SNDFILE *outhandle;
int outframes;
int i;
int type;
int rx_samples;
int tx_samples;
sig_tone_tx_state_t tx_state;
sig_tone_rx_state_t rx_state;
awgn_state_t noise_source;
codec_munge_state_t *munge;
if ((outhandle = sf_open_telephony_write(OUT_FILE_NAME, 2)) == NULL)
{
fprintf(stderr, " Cannot create audio file '%s'\n", OUT_FILE_NAME);
exit(2);
}
/*endif*/
awgn_init_dbm0(&noise_source, 1234567, -20.0f);
for (type = 1; type <= 3; type++)
{
sampleno = 0;
tone_1_present = 0;
tone_2_present = 0;
tx_section = 0;
munge = NULL;
switch (type)
{
case 1:
printf("2280Hz tests.\n");
munge = codec_munge_init(MUNGE_CODEC_ALAW, 0);
sig_tone_tx_init(&tx_state, SIG_TONE_2280HZ, tx_handler, &tx_state);
sig_tone_rx_init(&rx_state, SIG_TONE_2280HZ, rx_handler, &rx_state);
rx_state.current_rx_tone |= SIG_TONE_RX_PASSTHROUGH;
break;
case 2:
printf("2600Hz tests.\n");
munge = codec_munge_init(MUNGE_CODEC_ULAW, 0);
sig_tone_tx_init(&tx_state, SIG_TONE_2600HZ, tx_handler, &tx_state);
sig_tone_rx_init(&rx_state, SIG_TONE_2600HZ, rx_handler, &rx_state);
rx_state.current_rx_tone |= SIG_TONE_RX_PASSTHROUGH;
break;
case 3:
printf("2400Hz/2600Hz tests.\n");
munge = codec_munge_init(MUNGE_CODEC_ULAW, 0);
sig_tone_tx_init(&tx_state, SIG_TONE_2400HZ_2600HZ, tx_handler, &tx_state);
sig_tone_rx_init(&rx_state, SIG_TONE_2400HZ_2600HZ, rx_handler, &rx_state);
rx_state.current_rx_tone |= SIG_TONE_RX_PASSTHROUGH;
break;
}
/*endswitch*/
/* Set to the default of hook condition */
sig_tone_rx_set_mode(&rx_state, SIG_TONE_RX_PASSTHROUGH | SIG_TONE_RX_FILTER_TONE, 0);
sig_tone_tx_set_mode(&tx_state, SIG_TONE_1_PRESENT | SIG_TONE_2_PRESENT | SIG_TONE_TX_PASSTHROUGH, 0);
map_frequency_response(&rx_state);
sig_tone_rx_set_mode(&rx_state, SIG_TONE_RX_PASSTHROUGH, 0);
for (sampleno = 0; sampleno < 30000; sampleno += SAMPLES_PER_CHUNK)
{
if (sampleno == 8000)
{
/* 100ms seize */
printf("100ms seize - %d samples\n", ms_to_samples(100));
dial_pulses = 0;
sig_tone_tx_set_mode(&tx_state, 0, ms_to_samples(100));
}
for (i = 0; i < SAMPLES_PER_CHUNK; i++)
amp[i] = awgn(&noise_source);
/*endfor*/
tx_samples = sig_tone_tx(&tx_state, amp, SAMPLES_PER_CHUNK);
for (i = 0; i < tx_samples; i++)
out_amp[2*i] = amp[i];
/*endfor*/
codec_munge(munge, amp, tx_samples);
rx_samples = sig_tone_rx(&rx_state, amp, tx_samples);
for (i = 0; i < rx_samples; i++)
out_amp[2*i + 1] = amp[i];
/*endfor*/
outframes = sf_writef_short(outhandle, out_amp, rx_samples);
if (outframes != rx_samples)
{
fprintf(stderr, " Error writing audio file\n");
exit(2);
}
/*endif*/
}
/*endfor*/
}
/*endfor*/
if (sf_close(outhandle) != 0)
{
fprintf(stderr, " Cannot close audio file '%s'\n", OUT_FILE_NAME);
exit(2);
}
/*endif*/
printf("Tests completed.\n");
return 0;
}
/*- End of function --------------------------------------------------------*/
/*- End of file ------------------------------------------------------------*/
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