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527 lines
16 KiB
527 lines
16 KiB
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 1999 - 2005, Digium, Inc.
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*
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* Mark Spencer <markster@digium.com>
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*
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*! \file
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*
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* \brief Translate between signed linear and Speex (Open Codec)
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*
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* http://www.speex.org
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* \note This work was motivated by Jeremy McNamara
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* hacked to be configurable by anthm and bkw 9/28/2004
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* \ingroup codecs
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*/
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/*** MODULEINFO
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<depend>libspeex</depend>
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***/
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#include <fcntl.h>
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#include <stdlib.h>
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#include <unistd.h>
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#include <netinet/in.h>
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#include <string.h>
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#include <stdio.h>
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#include <speex/speex.h>
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/* We require a post 1.1.8 version of Speex to enable preprocessing
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and better type handling */
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#ifdef _SPEEX_TYPES_H
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#include <speex/speex_preprocess.h>
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#endif
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/* codec variables */
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static int quality = 3;
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static int complexity = 2;
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static int enhancement = 0;
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static int vad = 0;
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static int vbr = 0;
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static float vbr_quality = 4;
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static int abr = 0;
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static int dtx = 0; /* set to 1 to enable silence detection */
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static int preproc = 0;
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static int pp_vad = 0;
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static int pp_agc = 0;
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static float pp_agc_level = 8000; /* XXX what is this 8000 ? */
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static int pp_denoise = 0;
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static int pp_dereverb = 0;
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static float pp_dereverb_decay = 0.4;
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static float pp_dereverb_level = 0.3;
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#define TYPE_SILENCE 0x2
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#define TYPE_HIGH 0x0
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#define TYPE_LOW 0x1
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#define TYPE_MASK 0x3
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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include "asterisk/lock.h"
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#include "asterisk/translate.h"
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#include "asterisk/module.h"
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#include "asterisk/config.h"
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#include "asterisk/options.h"
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#include "asterisk/logger.h"
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#include "asterisk/channel.h"
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#include "asterisk/utils.h"
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/* Sample frame data */
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#include "slin_speex_ex.h"
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#include "speex_slin_ex.h"
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#define BUFFER_SAMPLES 8000
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#define SPEEX_SAMPLES 160
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struct speex_coder_pvt {
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void *speex;
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SpeexBits bits;
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int framesize;
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int silent_state;
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#ifdef _SPEEX_TYPES_H
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SpeexPreprocessState *pp;
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spx_int16_t buf[BUFFER_SAMPLES];
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#else
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int16_t buf[BUFFER_SAMPLES]; /* input, waiting to be compressed */
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#endif
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};
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static void *lintospeex_new(struct ast_trans_pvt *pvt)
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{
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struct speex_coder_pvt *tmp = pvt->pvt;
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if (!(tmp->speex = speex_encoder_init(&speex_nb_mode)))
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return NULL;
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speex_bits_init(&tmp->bits);
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speex_bits_reset(&tmp->bits);
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speex_encoder_ctl(tmp->speex, SPEEX_GET_FRAME_SIZE, &tmp->framesize);
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ast_log(LOG_WARNING, "speex framesize is %d\n", tmp->framesize);
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speex_encoder_ctl(tmp->speex, SPEEX_SET_COMPLEXITY, &complexity);
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#ifdef _SPEEX_TYPES_H
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if (preproc) {
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tmp->pp = speex_preprocess_state_init(tmp->framesize, 8000); /* XXX what is this 8000 ? */
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speex_preprocess_ctl(tmp->pp, SPEEX_PREPROCESS_SET_VAD, &pp_vad);
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speex_preprocess_ctl(tmp->pp, SPEEX_PREPROCESS_SET_AGC, &pp_agc);
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speex_preprocess_ctl(tmp->pp, SPEEX_PREPROCESS_SET_AGC_LEVEL, &pp_agc_level);
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speex_preprocess_ctl(tmp->pp, SPEEX_PREPROCESS_SET_DENOISE, &pp_denoise);
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speex_preprocess_ctl(tmp->pp, SPEEX_PREPROCESS_SET_DEREVERB, &pp_dereverb);
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speex_preprocess_ctl(tmp->pp, SPEEX_PREPROCESS_SET_DEREVERB_DECAY, &pp_dereverb_decay);
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speex_preprocess_ctl(tmp->pp, SPEEX_PREPROCESS_SET_DEREVERB_LEVEL, &pp_dereverb_level);
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}
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#endif
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if (!abr && !vbr) {
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speex_encoder_ctl(tmp->speex, SPEEX_SET_QUALITY, &quality);
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if (vad)
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speex_encoder_ctl(tmp->speex, SPEEX_SET_VAD, &vad);
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}
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if (vbr) {
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speex_encoder_ctl(tmp->speex, SPEEX_SET_VBR, &vbr);
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speex_encoder_ctl(tmp->speex, SPEEX_SET_VBR_QUALITY, &vbr_quality);
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}
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if (abr)
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speex_encoder_ctl(tmp->speex, SPEEX_SET_ABR, &abr);
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if (dtx)
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speex_encoder_ctl(tmp->speex, SPEEX_SET_DTX, &dtx);
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tmp->silent_state = 0;
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return tmp;
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}
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static void *speextolin_new(struct ast_trans_pvt *pvt)
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{
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struct speex_coder_pvt *tmp = pvt->pvt;
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if (!(tmp->speex = speex_decoder_init(&speex_nb_mode)))
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return NULL;
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speex_bits_init(&tmp->bits);
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speex_decoder_ctl(tmp->speex, SPEEX_GET_FRAME_SIZE, &tmp->framesize);
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if (enhancement)
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speex_decoder_ctl(tmp->speex, SPEEX_SET_ENH, &enhancement);
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return tmp;
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}
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static struct ast_frame *lintospeex_sample(void)
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{
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static struct ast_frame f;
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f.frametype = AST_FRAME_VOICE;
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f.subclass = AST_FORMAT_SLINEAR;
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f.datalen = sizeof(slin_speex_ex);
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/* Assume 8000 Hz */
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f.samples = sizeof(slin_speex_ex)/2;
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f.mallocd = 0;
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f.offset = 0;
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f.src = __PRETTY_FUNCTION__;
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f.data = slin_speex_ex;
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return &f;
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}
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static struct ast_frame *speextolin_sample(void)
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{
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static struct ast_frame f;
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f.frametype = AST_FRAME_VOICE;
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f.subclass = AST_FORMAT_SPEEX;
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f.datalen = sizeof(speex_slin_ex);
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/* All frames are 20 ms long */
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f.samples = SPEEX_SAMPLES;
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f.mallocd = 0;
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f.offset = 0;
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f.src = __PRETTY_FUNCTION__;
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f.data = speex_slin_ex;
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return &f;
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}
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/*! \brief convert and store into outbuf */
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static int speextolin_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
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{
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struct speex_coder_pvt *tmp = pvt->pvt;
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/* Assuming there's space left, decode into the current buffer at
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the tail location. Read in as many frames as there are */
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int x;
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int res;
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int16_t *dst = (int16_t *)pvt->outbuf;
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/* XXX fout is a temporary buffer, may have different types */
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#ifdef _SPEEX_TYPES_H
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spx_int16_t fout[1024];
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#else
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float fout[1024];
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#endif
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if (f->datalen == 0) { /* Native PLC interpolation */
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if (pvt->samples + tmp->framesize > BUFFER_SAMPLES) {
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ast_log(LOG_WARNING, "Out of buffer space\n");
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return -1;
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}
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#ifdef _SPEEX_TYPES_H
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speex_decode_int(tmp->speex, NULL, dst + pvt->samples);
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#else
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speex_decode(tmp->speex, NULL, fout);
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for (x=0;x<tmp->framesize;x++) {
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dst[pvt->samples + x] = (int16_t)fout[x];
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}
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#endif
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pvt->samples += tmp->framesize;
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return 0;
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}
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/* Read in bits */
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speex_bits_read_from(&tmp->bits, f->data, f->datalen);
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for (;;) {
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#ifdef _SPEEX_TYPES_H
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res = speex_decode_int(tmp->speex, &tmp->bits, fout);
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#else
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res = speex_decode(tmp->speex, &tmp->bits, fout);
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#endif
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if (res < 0)
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break;
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if (pvt->samples + tmp->framesize > BUFFER_SAMPLES) {
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ast_log(LOG_WARNING, "Out of buffer space\n");
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return -1;
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}
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for (x = 0 ; x < tmp->framesize; x++)
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dst[pvt->samples + x] = (int16_t)fout[x];
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pvt->samples += tmp->framesize;
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pvt->datalen += 2 * tmp->framesize; /* 2 bytes/sample */
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}
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return 0;
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}
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/*! \brief store input frame in work buffer */
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static int lintospeex_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
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{
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struct speex_coder_pvt *tmp = pvt->pvt;
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/* XXX We should look at how old the rest of our stream is, and if it
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is too old, then we should overwrite it entirely, otherwise we can
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get artifacts of earlier talk that do not belong */
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memcpy(tmp->buf + pvt->samples, f->data, f->datalen);
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pvt->samples += f->samples;
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return 0;
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}
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/*! \brief convert work buffer and produce output frame */
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static struct ast_frame *lintospeex_frameout(struct ast_trans_pvt *pvt)
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{
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struct speex_coder_pvt *tmp = pvt->pvt;
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int is_speech=1;
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int datalen = 0; /* output bytes */
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int samples = 0; /* output samples */
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/* We can't work on anything less than a frame in size */
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if (pvt->samples < tmp->framesize)
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return NULL;
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speex_bits_reset(&tmp->bits);
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while (pvt->samples >= tmp->framesize) {
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#ifdef _SPEEX_TYPES_H
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/* Preprocess audio */
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if (preproc)
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is_speech = speex_preprocess(tmp->pp, tmp->buf, NULL);
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/* Encode a frame of data */
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if (is_speech) {
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/* If DTX enabled speex_encode returns 0 during silence */
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is_speech = speex_encode_int(tmp->speex, tmp->buf, &tmp->bits) || !dtx;
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} else {
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/* 5 zeros interpreted by Speex as silence (submode 0) */
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speex_bits_pack(&tmp->bits, 0, 5);
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}
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#else
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{
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float fbuf[1024];
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int x;
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/* Convert to floating point */
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for (x = 0; x < tmp->framesize; x++)
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fbuf[x] = tmp->buf[x];
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/* Encode a frame of data */
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is_speech = speex_encode(tmp->speex, fbuf, &tmp->bits) || !dtx;
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}
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#endif
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samples += tmp->framesize;
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pvt->samples -= tmp->framesize;
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/* Move the data at the end of the buffer to the front */
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if (pvt->samples)
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memmove(tmp->buf, tmp->buf + tmp->framesize, pvt->samples * 2);
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}
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/* Use AST_FRAME_CNG to signify the start of any silence period */
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if (is_speech) {
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tmp->silent_state = 0;
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} else {
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if (tmp->silent_state) {
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return NULL;
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} else {
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tmp->silent_state = 1;
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speex_bits_reset(&tmp->bits);
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memset(&pvt->f, 0, sizeof(pvt->f));
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pvt->f.frametype = AST_FRAME_CNG;
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pvt->f.samples = samples;
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/* XXX what now ? format etc... */
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}
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}
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/* Terminate bit stream */
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speex_bits_pack(&tmp->bits, 15, 5);
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datalen = speex_bits_write(&tmp->bits, pvt->outbuf, pvt->t->buf_size);
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return ast_trans_frameout(pvt, datalen, samples);
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}
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static void speextolin_destroy(struct ast_trans_pvt *arg)
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{
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struct speex_coder_pvt *pvt = arg->pvt;
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speex_decoder_destroy(pvt->speex);
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speex_bits_destroy(&pvt->bits);
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}
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static void lintospeex_destroy(struct ast_trans_pvt *arg)
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{
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struct speex_coder_pvt *pvt = arg->pvt;
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#ifdef _SPEEX_TYPES_H
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if (preproc)
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speex_preprocess_state_destroy(pvt->pp);
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#endif
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speex_encoder_destroy(pvt->speex);
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speex_bits_destroy(&pvt->bits);
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}
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static struct ast_translator speextolin = {
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.name = "speextolin",
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.srcfmt = AST_FORMAT_SPEEX,
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.dstfmt = AST_FORMAT_SLINEAR,
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.newpvt = speextolin_new,
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.framein = speextolin_framein,
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.destroy = speextolin_destroy,
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.sample = speextolin_sample,
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.desc_size = sizeof(struct speex_coder_pvt),
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.buffer_samples = BUFFER_SAMPLES,
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.buf_size = BUFFER_SAMPLES * 2,
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};
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static struct ast_translator lintospeex = {
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.name = "lintospeex",
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.srcfmt = AST_FORMAT_SLINEAR,
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.dstfmt = AST_FORMAT_SPEEX,
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.newpvt = lintospeex_new,
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.framein = lintospeex_framein,
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.frameout = lintospeex_frameout,
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.destroy = lintospeex_destroy,
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.sample = lintospeex_sample,
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.desc_size = sizeof(struct speex_coder_pvt),
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.buffer_samples = BUFFER_SAMPLES,
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.buf_size = BUFFER_SAMPLES * 2, /* XXX maybe a lot less ? */
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};
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static void parse_config(void)
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{
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struct ast_config *cfg = ast_config_load("codecs.conf");
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struct ast_variable *var;
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int res;
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float res_f;
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if (cfg == NULL)
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return;
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for (var = ast_variable_browse(cfg, "speex"); var; var = var->next) {
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if (!strcasecmp(var->name, "quality")) {
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res = abs(atoi(var->value));
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if (res > -1 && res < 11) {
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if (option_verbose > 2)
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ast_verbose(VERBOSE_PREFIX_3 "CODEC SPEEX: Setting Quality to %d\n",res);
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quality = res;
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} else
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ast_log(LOG_ERROR,"Error Quality must be 0-10\n");
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} else if (!strcasecmp(var->name, "complexity")) {
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res = abs(atoi(var->value));
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if (res > -1 && res < 11) {
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if (option_verbose > 2)
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ast_verbose(VERBOSE_PREFIX_3 "CODEC SPEEX: Setting Complexity to %d\n",res);
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complexity = res;
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} else
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ast_log(LOG_ERROR,"Error! Complexity must be 0-10\n");
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} else if (!strcasecmp(var->name, "vbr_quality")) {
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if (sscanf(var->value, "%f", &res_f) == 1 && res_f >= 0 && res_f <= 10) {
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if (option_verbose > 2)
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ast_verbose(VERBOSE_PREFIX_3 "CODEC SPEEX: Setting VBR Quality to %f\n",res_f);
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vbr_quality = res_f;
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} else
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ast_log(LOG_ERROR,"Error! VBR Quality must be 0-10\n");
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} else if (!strcasecmp(var->name, "abr_quality")) {
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ast_log(LOG_ERROR,"Error! ABR Quality setting obsolete, set ABR to desired bitrate\n");
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} else if (!strcasecmp(var->name, "enhancement")) {
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enhancement = ast_true(var->value) ? 1 : 0;
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if (option_verbose > 2)
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ast_verbose(VERBOSE_PREFIX_3 "CODEC SPEEX: Perceptual Enhancement Mode. [%s]\n",enhancement ? "on" : "off");
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} else if (!strcasecmp(var->name, "vbr")) {
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vbr = ast_true(var->value) ? 1 : 0;
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if (option_verbose > 2)
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ast_verbose(VERBOSE_PREFIX_3 "CODEC SPEEX: VBR Mode. [%s]\n",vbr ? "on" : "off");
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} else if (!strcasecmp(var->name, "abr")) {
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res = abs(atoi(var->value));
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if (res >= 0) {
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if (option_verbose > 2) {
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if (res > 0)
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ast_verbose(VERBOSE_PREFIX_3 "CODEC SPEEX: Setting ABR target bitrate to %d\n",res);
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else
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ast_verbose(VERBOSE_PREFIX_3 "CODEC SPEEX: Disabling ABR\n");
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}
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abr = res;
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} else
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ast_log(LOG_ERROR,"Error! ABR target bitrate must be >= 0\n");
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} else if (!strcasecmp(var->name, "vad")) {
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vad = ast_true(var->value) ? 1 : 0;
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if (option_verbose > 2)
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ast_verbose(VERBOSE_PREFIX_3 "CODEC SPEEX: VAD Mode. [%s]\n",vad ? "on" : "off");
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} else if (!strcasecmp(var->name, "dtx")) {
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dtx = ast_true(var->value) ? 1 : 0;
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if (option_verbose > 2)
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ast_verbose(VERBOSE_PREFIX_3 "CODEC SPEEX: DTX Mode. [%s]\n",dtx ? "on" : "off");
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} else if (!strcasecmp(var->name, "preprocess")) {
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preproc = ast_true(var->value) ? 1 : 0;
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if (option_verbose > 2)
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ast_verbose(VERBOSE_PREFIX_3 "CODEC SPEEX: Preprocessing. [%s]\n",preproc ? "on" : "off");
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} else if (!strcasecmp(var->name, "pp_vad")) {
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pp_vad = ast_true(var->value) ? 1 : 0;
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if (option_verbose > 2)
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ast_verbose(VERBOSE_PREFIX_3 "CODEC SPEEX: Preprocessor VAD. [%s]\n",pp_vad ? "on" : "off");
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} else if (!strcasecmp(var->name, "pp_agc")) {
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pp_agc = ast_true(var->value) ? 1 : 0;
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if (option_verbose > 2)
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ast_verbose(VERBOSE_PREFIX_3 "CODEC SPEEX: Preprocessor AGC. [%s]\n",pp_agc ? "on" : "off");
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|
} else if (!strcasecmp(var->name, "pp_agc_level")) {
|
|
if (sscanf(var->value, "%f", &res_f) == 1 && res_f >= 0) {
|
|
if (option_verbose > 2)
|
|
ast_verbose(VERBOSE_PREFIX_3 "CODEC SPEEX: Setting preprocessor AGC Level to %f\n",res_f);
|
|
pp_agc_level = res_f;
|
|
} else
|
|
ast_log(LOG_ERROR,"Error! Preprocessor AGC Level must be >= 0\n");
|
|
} else if (!strcasecmp(var->name, "pp_denoise")) {
|
|
pp_denoise = ast_true(var->value) ? 1 : 0;
|
|
if (option_verbose > 2)
|
|
ast_verbose(VERBOSE_PREFIX_3 "CODEC SPEEX: Preprocessor Denoise. [%s]\n",pp_denoise ? "on" : "off");
|
|
} else if (!strcasecmp(var->name, "pp_dereverb")) {
|
|
pp_dereverb = ast_true(var->value) ? 1 : 0;
|
|
if (option_verbose > 2)
|
|
ast_verbose(VERBOSE_PREFIX_3 "CODEC SPEEX: Preprocessor Dereverb. [%s]\n",pp_dereverb ? "on" : "off");
|
|
} else if (!strcasecmp(var->name, "pp_dereverb_decay")) {
|
|
if (sscanf(var->value, "%f", &res_f) == 1 && res_f >= 0) {
|
|
if (option_verbose > 2)
|
|
ast_verbose(VERBOSE_PREFIX_3 "CODEC SPEEX: Setting preprocessor Dereverb Decay to %f\n",res_f);
|
|
pp_dereverb_decay = res_f;
|
|
} else
|
|
ast_log(LOG_ERROR,"Error! Preprocessor Dereverb Decay must be >= 0\n");
|
|
} else if (!strcasecmp(var->name, "pp_dereverb_level")) {
|
|
if (sscanf(var->value, "%f", &res_f) == 1 && res_f >= 0) {
|
|
if (option_verbose > 2)
|
|
ast_verbose(VERBOSE_PREFIX_3 "CODEC SPEEX: Setting preprocessor Dereverb Level to %f\n",res_f);
|
|
pp_dereverb_level = res_f;
|
|
} else
|
|
ast_log(LOG_ERROR,"Error! Preprocessor Dereverb Level must be >= 0\n");
|
|
}
|
|
}
|
|
ast_config_destroy(cfg);
|
|
}
|
|
|
|
static int reload(void *mod)
|
|
{
|
|
/*
|
|
* XXX reloading while there are active sessions is
|
|
* somewhat silly because the old state presumably
|
|
* wouldn't work anymore...
|
|
* maybe we shuld do a standard hangup localusers ?
|
|
*/
|
|
ast_mutex_lock(&__mod_desc->lock);
|
|
parse_config();
|
|
ast_mutex_lock(&__mod_desc->lock);
|
|
return 0;
|
|
}
|
|
|
|
static int unload_module(void *mod)
|
|
{
|
|
int res;
|
|
res = ast_unregister_translator(&lintospeex);
|
|
res |= ast_unregister_translator(&speextolin);
|
|
return res;
|
|
}
|
|
|
|
static int load_module(void *mod)
|
|
{
|
|
int res;
|
|
parse_config();
|
|
res=ast_register_translator(&speextolin, mod);
|
|
if (!res)
|
|
res=ast_register_translator(&lintospeex, mod);
|
|
else
|
|
ast_unregister_translator(&speextolin);
|
|
return res;
|
|
}
|
|
|
|
static const char *description(void)
|
|
{
|
|
return "Speex/PCM16 (signed linear) Codec Translator";
|
|
}
|
|
|
|
static const char *key(void)
|
|
{
|
|
return ASTERISK_GPL_KEY;
|
|
}
|
|
|
|
STD_MOD(MOD_1, reload, NULL, NULL);
|
|
|