Mercurial > hg > audiostuff
view spandsp-0.0.6pre17/src/tone_generate.c @ 4:26cd8f1ef0b1
import spandsp-0.0.6pre17
author | Peter Meerwald <pmeerw@cosy.sbg.ac.at> |
---|---|
date | Fri, 25 Jun 2010 15:50:58 +0200 |
parents | |
children |
line wrap: on
line source
/* * 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 1.53.4.1 2009/12/23 14:23:49 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 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_NONSTD(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 ------------------------------------------------------------*/