kenmirrors
/
freeswitch
mirror of https://github.com/signalwire/freeswitch.git
1
0
Fork 0
Code Issues Projects Releases Wiki Activity
freeswitch/libs/spandsp/src/tone_generate.c

242 lines
7.4 KiB
C
Raw Blame History

/*
* SpanDSP - a series of DSP components for telephony
*
* tone_generate.c - General telephony tone generation.
*
* Written by Steve Underwood <steveu@coppice.org>
*
* Copyright (C) 2001 Steve Underwood
*
* All rights reserved.
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU Lesser General Public License version 2.1,
* 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 Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this program; if not, write to the Free Software
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*
* $Id: tone_generate.c,v 1.50 2009/02/10 13:06:47 steveu Exp $
*/
/*! \file */
#if defined(HAVE_CONFIG_H)
#include "config.h"
#endif
#include <inttypes.h>
#include <string.h>
#include <stdlib.h>
#include <stdio.h>
#include <time.h>
#include <fcntl.h>
#if defined(HAVE_TGMATH_H)
#include <tgmath.h>
#endif
#if defined(HAVE_MATH_H)
#include <math.h>
#endif
#include "floating_fudge.h"
#include "spandsp/telephony.h"
#include "spandsp/fast_convert.h"
#include "spandsp/dc_restore.h"
#include "spandsp/complex.h"
#include "spandsp/dds.h"
#include "spandsp/tone_generate.h"
#include "spandsp/private/tone_generate.h"
#if !defined(M_PI)
/* C99 systems may not define M_PI */
#define M_PI 3.14159265358979323846264338327
#endif
#define ms_to_samples(t) (((t)*SAMPLE_RATE)/1000)
SPAN_DECLARE(void) make_tone_gen_descriptor(tone_gen_descriptor_t *s,
int f1,
int l1,
int f2,
int l2,
int d1,
int d2,
int d3,
int d4,
int repeat)
{
memset(s, 0, sizeof(*s));
if (f1)
{
#if defined(SPANDSP_USE_FIXED_POINT)
s->tone[0].phase_rate = dds_phase_rate((float) f1);
if (f2 < 0)
s->tone[0].phase_rate = -s->tone[0].phase_rate;
s->tone[0].gain = dds_scaling_dbm0((float) l1);
#else
s->tone[0].phase_rate = dds_phase_ratef((float) f1);
if (f2 < 0)
s->tone[0].phase_rate = -s->tone[0].phase_rate;
s->tone[0].gain = dds_scaling_dbm0f((float) l1);
#endif
}
if (f2)
{
#if defined(SPANDSP_USE_FIXED_POINT)
s->tone[1].phase_rate = dds_phase_rate((float) abs(f2));
s->tone[1].gain = (f2 < 0) ? (float) 32767.0f*l2/100.0f : dds_scaling_dbm0((float) l2);
#else
s->tone[1].phase_rate = dds_phase_ratef((float) abs(f2));
s->tone[1].gain = (f2 < 0) ? (float) l2/100.0f : dds_scaling_dbm0f((float) l2);
#endif
}
s->duration[0] = d1*SAMPLE_RATE/1000;
s->duration[1] = d2*SAMPLE_RATE/1000;
s->duration[2] = d3*SAMPLE_RATE/1000;
s->duration[3] = d4*SAMPLE_RATE/1000;
s->repeat = repeat;
}
/*- End of function --------------------------------------------------------*/
SPAN_DECLARE(tone_gen_state_t *) tone_gen_init(tone_gen_state_t *s, tone_gen_descriptor_t *t)
{
int i;
if (s == NULL)
return NULL;
for (i = 0; i < 4; i++)
{
s->tone[i] = t->tone[i];
s->phase[i] = 0;
}
for (i = 0; i < 4; i++)
s->duration[i] = t->duration[i];
s->repeat = t->repeat;
s->current_section = 0;
s->current_position = 0;
return s;
}
/*- End of function --------------------------------------------------------*/
SPAN_DECLARE(int) tone_gen_release(tone_gen_state_t *s)
{
return 0;
}
/*- End of function --------------------------------------------------------*/
SPAN_DECLARE(int) tone_gen_free(tone_gen_state_t *s)
{
if (s)
free(s);
return 0;
}
/*- End of function --------------------------------------------------------*/
SPAN_DECLARE(int) tone_gen(tone_gen_state_t *s, int16_t amp[], int max_samples)
{
int samples;
int limit;
#if defined(SPANDSP_USE_FIXED_POINT)
int16_t xamp;
#else
float xamp;
#endif
int i;
if (s->current_section < 0)
return 0;
for (samples = 0; samples < max_samples; )
{
limit = samples + s->duration[s->current_section] - s->current_position;
if (limit > max_samples)
limit = max_samples;
s->current_position += (limit - samples);
if (s->current_section & 1)
{
/* A silent section */
for ( ; samples < limit; samples++)
amp[samples] = 0;
}
else
{
if (s->tone[0].phase_rate < 0)
{
/* Modulated tone */
for ( ; samples < limit; samples++)
{
/* There must be two, and only two, tones */
#if defined(SPANDSP_USE_FIXED_POINT)
xamp = ((int32_t) dds_mod(&s->phase[0], -s->tone[0].phase_rate, s->tone[0].gain, 0)
*(32767 + (int32_t) dds_mod(&s->phase[1], s->tone[1].phase_rate, s->tone[1].gain, 0))) >> 15;
amp[samples] = xamp;
#else
xamp = dds_modf(&s->phase[0], -s->tone[0].phase_rate, s->tone[0].gain, 0)
*(1.0f + dds_modf(&s->phase[1], s->tone[1].phase_rate, s->tone[1].gain, 0));
amp[samples] = (int16_t) lfastrintf(xamp);
#endif
}
}
else
{
for ( ; samples < limit; samples++)
{
#if defined(SPANDSP_USE_FIXED_POINT)
xamp = 0;
#else
xamp = 0.0f;
#endif
for (i = 0; i < 4; i++)
{
if (s->tone[i].phase_rate == 0)
break;
#if defined(SPANDSP_USE_FIXED_POINT)
xamp += dds_mod(&s->phase[i], s->tone[i].phase_rate, s->tone[i].gain, 0);
#else
xamp += dds_modf(&s->phase[i], s->tone[i].phase_rate, s->tone[i].gain, 0);
#endif
}
/* Saturation of the answer is the right thing at this point.
However, we are normally generating well controlled tones,
that cannot clip. So, the overhead of doing saturation is
a waste of valuable time. */
#if defined(SPANDSP_USE_FIXED_POINT)
amp[samples] = xamp;
#else
amp[samples] = (int16_t) lfastrintf(xamp);
#endif
}
}
}
if (s->current_position >= s->duration[s->current_section])
{
s->current_position = 0;
if (++s->current_section > 3 || s->duration[s->current_section] == 0)
{
if (!s->repeat)
{
/* Force a quick exit */
s->current_section = -1;
break;
}
s->current_section = 0;
}
}
}
return samples;
}
/*- End of function --------------------------------------------------------*/
/*- End of file ------------------------------------------------------------*/
Reference in New Issue View Git Blame Copy Permalink