Mercurial > hg > audiostuff
view spandsp-0.0.3/spandsp-0.0.3/src/tone_generate.c @ 5:f762bf195c4b
import spandsp-0.0.3
author | Peter Meerwald <pmeerw@cosy.sbg.ac.at> |
---|---|
date | Fri, 25 Jun 2010 16:00:21 +0200 |
parents | |
children |
line wrap: on
line source
/* * SpanDSP - a series of DSP components for telephony * * tone_generate.c - General telephony tone generation, and specific * generation of Bell MF, MFC/R2, and network supervisory tones. * * 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 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: tone_generate.c,v 1.31 2006/11/28 16:59:56 steveu Exp $ */ /*! \file */ #ifdef 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 "spandsp/telephony.h" #include "spandsp/dc_restore.h" #include "spandsp/dds.h" #include "spandsp/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) 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 >= 1) { s->phase_rate[0] = dds_phase_ratef((float) f1); s->gain[0] = dds_scaling_dbm0f((float) l1); } s->modulate = (f2 < 0); if (f2) { s->phase_rate[1] = dds_phase_ratef((float) abs(f2)); s->gain[1] = (s->modulate) ? (float) l2/100.0f : dds_scaling_dbm0f((float) l2); } s->duration[0] = d1*8; s->duration[1] = d2*8; s->duration[2] = d3*8; s->duration[3] = d4*8; s->repeat = repeat; } /*- End of function --------------------------------------------------------*/ void make_tone_descriptor(tone_gen_descriptor_t *desc, cadenced_tone_t *tone) { make_tone_gen_descriptor(desc, tone->f1, tone->level1, tone->f2, tone->level2, tone->on_time1, tone->off_time1, tone->on_time2, tone->off_time2, tone->repeat); } /*- End of function --------------------------------------------------------*/ void tone_gen_init(tone_gen_state_t *s, tone_gen_descriptor_t *t) { int i; s->phase_rate[0] = t->phase_rate[0]; s->gain[0] = t->gain[0]; s->phase_rate[1] = t->phase_rate[1]; s->gain[1] = t->gain[1]; s->modulate = t->modulate; for (i = 0; i < 4; i++) s->duration[i] = t->duration[i]; s->repeat = t->repeat; s->phase[0] = 0; s->phase[1] = 0; s->current_section = 0; s->current_position = 0; } /*- End of function --------------------------------------------------------*/ int tone_gen(tone_gen_state_t *s, int16_t amp[], int max_samples) { int samples; int limit; float xamp; float yamp; 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 { for ( ; samples < limit; samples++) { xamp = 0.0; if (s->phase_rate[0]) xamp = dds_modf(&(s->phase[0]), s->phase_rate[0], s->gain[0], 0); if (s->phase_rate[1]) { yamp = dds_modf(&(s->phase[1]), s->phase_rate[1], s->gain[1], 0); if (s->modulate) xamp *= (1.0f + yamp); else xamp += yamp; } /* 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. */ amp[samples] = (int16_t) lrintf(xamp); } } 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 ------------------------------------------------------------*/