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inital checkin of spandsp from http://www.soft-switch.org/downloads/snapshots/spandsp/spandsp-20080829.tar.gz
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@9425 d0543943-73ff-0310-b7d9-9358b9ac24b2
This commit is contained in:
Michael Jerris
189
libs/spandsp/src/tone_generate.c
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189
libs/spandsp/src/tone_generate.c
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/*
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* SpanDSP - a series of DSP components for telephony
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*
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* tone_generate.c - General telephony tone generation.
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*
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* Written by Steve Underwood <steveu@coppice.org>
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*
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* Copyright (C) 2001 Steve Underwood
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*
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* All rights reserved.
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU Lesser General Public License version 2.1,
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* as published by the Free Software Foundation.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with this program; if not, write to the Free Software
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* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
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*
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* $Id: tone_generate.c,v 1.43 2008/07/02 14:48:26 steveu Exp $
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*/
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/*! \file */
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#if defined(HAVE_CONFIG_H)
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#include "config.h"
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#endif
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#include <inttypes.h>
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#include <string.h>
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#include <stdlib.h>
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#include <stdio.h>
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#include <time.h>
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#include <fcntl.h>
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#include "floating_fudge.h"
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#if defined(HAVE_TGMATH_H)
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#include <tgmath.h>
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#endif
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#if defined(HAVE_MATH_H)
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#include <math.h>
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#endif
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#include "spandsp/telephony.h"
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#include "spandsp/dc_restore.h"
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#include "spandsp/complex.h"
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#include "spandsp/dds.h"
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#include "spandsp/tone_generate.h"
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#if !defined(M_PI)
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/* C99 systems may not define M_PI */
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#define M_PI 3.14159265358979323846264338327
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#endif
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#define ms_to_samples(t) (((t)*SAMPLE_RATE)/1000)
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void make_tone_gen_descriptor(tone_gen_descriptor_t *s,
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int f1,
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int l1,
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int f2,
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int l2,
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int d1,
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int d2,
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int d3,
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int d4,
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int repeat)
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{
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memset(s, 0, sizeof(*s));
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if (f1)
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{
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s->tone[0].phase_rate = dds_phase_ratef((float) f1);
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if (f2 < 0)
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s->tone[0].phase_rate = -s->tone[0].phase_rate;
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s->tone[0].gain = dds_scaling_dbm0f((float) l1);
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}
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if (f2)
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{
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s->tone[1].phase_rate = dds_phase_ratef((float) abs(f2));
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s->tone[1].gain = (f2 < 0) ? (float) l2/100.0f : dds_scaling_dbm0f((float) l2);
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}
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s->duration[0] = d1*SAMPLE_RATE/1000;
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s->duration[1] = d2*SAMPLE_RATE/1000;
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s->duration[2] = d3*SAMPLE_RATE/1000;
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s->duration[3] = d4*SAMPLE_RATE/1000;
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s->repeat = repeat;
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}
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/*- End of function --------------------------------------------------------*/
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tone_gen_state_t *tone_gen_init(tone_gen_state_t *s, tone_gen_descriptor_t *t)
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{
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int i;
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if (s == NULL)
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return NULL;
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for (i = 0; i < 4; i++)
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{
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s->tone[i] = t->tone[i];
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s->phase[i] = 0;
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}
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for (i = 0; i < 4; i++)
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s->duration[i] = t->duration[i];
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s->repeat = t->repeat;
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s->current_section = 0;
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s->current_position = 0;
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return s;
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}
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/*- End of function --------------------------------------------------------*/
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int tone_gen(tone_gen_state_t *s, int16_t amp[], int max_samples)
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{
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int samples;
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int limit;
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float xamp;
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int i;
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if (s->current_section < 0)
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return 0;
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for (samples = 0; samples < max_samples; )
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{
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limit = samples + s->duration[s->current_section] - s->current_position;
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if (limit > max_samples)
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limit = max_samples;
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s->current_position += (limit - samples);
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if (s->current_section & 1)
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{
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/* A silent section */
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for ( ; samples < limit; samples++)
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amp[samples] = 0;
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}
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else
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{
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if (s->tone[0].phase_rate < 0)
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{
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for ( ; samples < limit; samples++)
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{
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/* There must be two, and only two tones */
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xamp = dds_modf(&s->phase[0], -s->tone[0].phase_rate, s->tone[0].gain, 0)
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*(1.0f + dds_modf(&s->phase[1], s->tone[1].phase_rate, s->tone[1].gain, 0));
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amp[samples] = (int16_t) lrintf(xamp);
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}
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}
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else
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{
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for ( ; samples < limit; samples++)
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{
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xamp = 0.0f;
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for (i = 0; i < 4; i++)
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{
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if (s->tone[i].phase_rate == 0)
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break;
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xamp += dds_modf(&s->phase[i], s->tone[i].phase_rate, s->tone[i].gain, 0);
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}
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/* Saturation of the answer is the right thing at this point.
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However, we are normally generating well controlled tones,
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that cannot clip. So, the overhead of doing saturation is
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a waste of valuable time. */
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amp[samples] = (int16_t) lrintf(xamp);
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}
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}
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}
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if (s->current_position >= s->duration[s->current_section])
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{
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s->current_position = 0;
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if (++s->current_section > 3 || s->duration[s->current_section] == 0)
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{
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if (!s->repeat)
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{
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/* Force a quick exit */
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s->current_section = -1;
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break;
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}
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s->current_section = 0;
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}
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}
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}
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return samples;
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}
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/*- End of function --------------------------------------------------------*/
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/*- End of file ------------------------------------------------------------*/
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