If codec_speex fails to register a translator it would cause Asterisk to exit instead of continue as a DECLINED module. * Make unload_module() always return 0. It is silly to fail unloading if any translators we try to unregister were not even registered. Change-Id: Ia262591f68333dad17673ba7104d11c88096f51a
731 lines
21 KiB
C
731 lines
21 KiB
C
/*
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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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* \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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*
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* \ingroup codecs
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*
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* The Speex library - http://www.speex.org
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*
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*/
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/*** MODULEINFO
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<depend>speex</depend>
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<depend>speex_preprocess</depend>
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<use type="external">speexdsp</use>
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<support_level>core</support_level>
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***/
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#include "asterisk.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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*/
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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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#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/utils.h"
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#include "asterisk/frame.h"
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#include "asterisk/linkedlists.h"
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/* For struct ast_rtp_rtcp_report and struct ast_rtp_rtcp_report_block */
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#include "asterisk/rtp_engine.h"
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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 exp_rtcp_fb = 0; /* set to 1 to use experimental RTCP feedback for changing bitrate */
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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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#define BUFFER_SAMPLES 8000
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#define SPEEX_SAMPLES 160
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/* Sample frame data */
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#include "asterisk/slin.h"
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#include "ex_speex.h"
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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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int fraction_lost;
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int quality;
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int default_quality;
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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 int speex_encoder_construct(struct ast_trans_pvt *pvt, const SpeexMode *profile, int sampling_rate)
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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(profile)))
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return -1;
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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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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, sampling_rate);
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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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tmp->fraction_lost = 0;
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tmp->default_quality = vbr ? vbr_quality : quality;
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tmp->quality = tmp->default_quality;
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ast_debug(3, "Default quality (%s): %d\n", vbr ? "vbr" : "cbr", tmp->default_quality);
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return 0;
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}
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static int lintospeex_new(struct ast_trans_pvt *pvt)
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{
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return speex_encoder_construct(pvt, &speex_nb_mode, 8000);
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}
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static int lin16tospeexwb_new(struct ast_trans_pvt *pvt)
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{
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return speex_encoder_construct(pvt, &speex_wb_mode, 16000);
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}
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static int lin32tospeexuwb_new(struct ast_trans_pvt *pvt)
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{
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return speex_encoder_construct(pvt, &speex_uwb_mode, 32000);
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}
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static int speex_decoder_construct(struct ast_trans_pvt *pvt, const SpeexMode *profile)
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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(profile)))
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return -1;
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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 0;
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}
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static int speextolin_new(struct ast_trans_pvt *pvt)
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{
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return speex_decoder_construct(pvt, &speex_nb_mode);
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}
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static int speexwbtolin16_new(struct ast_trans_pvt *pvt)
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{
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return speex_decoder_construct(pvt, &speex_wb_mode);
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}
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static int speexuwbtolin32_new(struct ast_trans_pvt *pvt)
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{
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return speex_decoder_construct(pvt, &speex_uwb_mode);
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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 = pvt->outbuf.i16;
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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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pvt->datalen += 2 * tmp->framesize; /* 2 bytes/sample */
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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.ptr, 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.ptr, 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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struct ast_frame *result = NULL;
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struct ast_frame *last = NULL;
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int samples = 0; /* output samples */
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while (pvt->samples >= tmp->framesize) {
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struct ast_frame *current;
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int is_speech = 1;
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speex_bits_reset(&tmp->bits);
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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 + samples, 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 + samples, &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[samples + 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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/* Use AST_FRAME_CNG to signify the start of any silence period */
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if (is_speech) {
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int datalen = 0; /* output bytes */
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tmp->silent_state = 0;
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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.c, pvt->t->buf_size);
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current = ast_trans_frameout(pvt, datalen, tmp->framesize);
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} else if (tmp->silent_state) {
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current = NULL;
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} else {
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struct ast_frame frm = {
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.frametype = AST_FRAME_CNG,
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.src = pvt->t->name,
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};
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/*
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* XXX I don't think the AST_FRAME_CNG code has ever
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* really worked for speex. There doesn't seem to be
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* any consumers of the frame type. Everyone that
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* references the type seems to pass the frame on.
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*/
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tmp->silent_state = 1;
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/* XXX what now ? format etc... */
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current = ast_frisolate(&frm);
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}
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if (!current) {
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continue;
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} else if (last) {
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AST_LIST_NEXT(last, frame_list) = current;
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} else {
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result = current;
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}
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last = current;
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}
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/* Move the data at the end of the buffer to the front */
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if (samples) {
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memmove(tmp->buf, tmp->buf + samples, pvt->samples * 2);
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}
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return result;
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}
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/*! \brief handle incoming RTCP feedback and possibly edit encoder settings */
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static void lintospeex_feedback(struct ast_trans_pvt *pvt, struct ast_frame *feedback)
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{
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struct speex_coder_pvt *tmp = pvt->pvt;
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struct ast_rtp_rtcp_report *rtcp_report;
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struct ast_rtp_rtcp_report_block *report_block;
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int fraction_lost;
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int percent;
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int bitrate;
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int q;
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if(!exp_rtcp_fb)
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return;
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|
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/* We only accept feedback information in the form of SR and RR reports */
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if (feedback->subclass.integer != AST_RTP_RTCP_SR && feedback->subclass.integer != AST_RTP_RTCP_RR) {
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return;
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}
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rtcp_report = (struct ast_rtp_rtcp_report *)feedback->data.ptr;
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if (rtcp_report->reception_report_count == 0)
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return;
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report_block = rtcp_report->report_block[0];
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fraction_lost = report_block->lost_count.fraction;
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if (fraction_lost == tmp->fraction_lost)
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return;
|
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/* Per RFC3550, fraction lost is defined to be the number of packets lost
|
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* divided by the number of packets expected. Since it's a 8-bit value,
|
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* and we want a percentage value, we multiply by 100 and divide by 256. */
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percent = (fraction_lost*100)/256;
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bitrate = 0;
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q = -1;
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ast_debug(3, "Fraction lost changed: %d --> %d percent loss\n", fraction_lost, percent);
|
|
/* Handle change */
|
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speex_encoder_ctl(tmp->speex, SPEEX_GET_BITRATE, &bitrate);
|
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ast_debug(3, "Current bitrate: %d\n", bitrate);
|
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ast_debug(3, "Current quality: %d/%d\n", tmp->quality, tmp->default_quality);
|
|
/* FIXME BADLY Very ugly example of how this could be handled: probably sucks */
|
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if (percent < 10) {
|
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/* Not that bad, default quality is fine */
|
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q = tmp->default_quality;
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} else if (percent < 20) {
|
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/* Quite bad, let's go down a bit */
|
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q = tmp->default_quality-1;
|
|
} else if (percent < 30) {
|
|
/* Very bad, let's go down even more */
|
|
q = tmp->default_quality-2;
|
|
} else {
|
|
/* Really bad, use the lowest quality possible */
|
|
q = 0;
|
|
}
|
|
if (q < 0)
|
|
q = 0;
|
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if (q != tmp->quality) {
|
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ast_debug(3, " -- Setting to %d\n", q);
|
|
if (vbr) {
|
|
float vbr_q = q;
|
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speex_encoder_ctl(tmp->speex, SPEEX_SET_VBR_QUALITY, &vbr_q);
|
|
} else {
|
|
speex_encoder_ctl(tmp->speex, SPEEX_SET_QUALITY, &q);
|
|
}
|
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tmp->quality = q;
|
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}
|
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tmp->fraction_lost = fraction_lost;
|
|
}
|
|
|
|
static void speextolin_destroy(struct ast_trans_pvt *arg)
|
|
{
|
|
struct speex_coder_pvt *pvt = arg->pvt;
|
|
|
|
speex_decoder_destroy(pvt->speex);
|
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speex_bits_destroy(&pvt->bits);
|
|
}
|
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|
|
static void lintospeex_destroy(struct ast_trans_pvt *arg)
|
|
{
|
|
struct speex_coder_pvt *pvt = arg->pvt;
|
|
#ifdef _SPEEX_TYPES_H
|
|
if (preproc)
|
|
speex_preprocess_state_destroy(pvt->pp);
|
|
#endif
|
|
speex_encoder_destroy(pvt->speex);
|
|
speex_bits_destroy(&pvt->bits);
|
|
}
|
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|
|
static struct ast_translator speextolin = {
|
|
.name = "speextolin",
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.src_codec = {
|
|
.name = "speex",
|
|
.type = AST_MEDIA_TYPE_AUDIO,
|
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.sample_rate = 8000,
|
|
},
|
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.dst_codec = {
|
|
.name = "slin",
|
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.type = AST_MEDIA_TYPE_AUDIO,
|
|
.sample_rate = 8000,
|
|
},
|
|
.format = "slin",
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|
.newpvt = speextolin_new,
|
|
.framein = speextolin_framein,
|
|
.destroy = speextolin_destroy,
|
|
.sample = speex_sample,
|
|
.desc_size = sizeof(struct speex_coder_pvt),
|
|
.buffer_samples = BUFFER_SAMPLES,
|
|
.buf_size = BUFFER_SAMPLES * 2,
|
|
.native_plc = 1,
|
|
};
|
|
|
|
static struct ast_translator lintospeex = {
|
|
.name = "lintospeex",
|
|
.src_codec = {
|
|
.name = "slin",
|
|
.type = AST_MEDIA_TYPE_AUDIO,
|
|
.sample_rate = 8000,
|
|
},
|
|
.dst_codec = {
|
|
.name = "speex",
|
|
.type = AST_MEDIA_TYPE_AUDIO,
|
|
.sample_rate = 8000,
|
|
},
|
|
.format = "speex",
|
|
.newpvt = lintospeex_new,
|
|
.framein = lintospeex_framein,
|
|
.frameout = lintospeex_frameout,
|
|
.feedback = lintospeex_feedback,
|
|
.destroy = lintospeex_destroy,
|
|
.sample = slin8_sample,
|
|
.desc_size = sizeof(struct speex_coder_pvt),
|
|
.buffer_samples = BUFFER_SAMPLES,
|
|
.buf_size = BUFFER_SAMPLES * 2, /* XXX maybe a lot less ? */
|
|
};
|
|
|
|
static struct ast_translator speexwbtolin16 = {
|
|
.name = "speexwbtolin16",
|
|
.src_codec = {
|
|
.name = "speex",
|
|
.type = AST_MEDIA_TYPE_AUDIO,
|
|
.sample_rate = 16000,
|
|
},
|
|
.dst_codec = {
|
|
.name = "slin",
|
|
.type = AST_MEDIA_TYPE_AUDIO,
|
|
.sample_rate = 16000,
|
|
},
|
|
.format = "slin16",
|
|
.newpvt = speexwbtolin16_new,
|
|
.framein = speextolin_framein,
|
|
.destroy = speextolin_destroy,
|
|
.sample = speex16_sample,
|
|
.desc_size = sizeof(struct speex_coder_pvt),
|
|
.buffer_samples = BUFFER_SAMPLES,
|
|
.buf_size = BUFFER_SAMPLES * 2,
|
|
.native_plc = 1,
|
|
};
|
|
|
|
static struct ast_translator lin16tospeexwb = {
|
|
.name = "lin16tospeexwb",
|
|
.src_codec = {
|
|
.name = "slin",
|
|
.type = AST_MEDIA_TYPE_AUDIO,
|
|
.sample_rate = 16000,
|
|
},
|
|
.dst_codec = {
|
|
.name = "speex",
|
|
.type = AST_MEDIA_TYPE_AUDIO,
|
|
.sample_rate = 16000,
|
|
},
|
|
.format = "speex16",
|
|
.newpvt = lin16tospeexwb_new,
|
|
.framein = lintospeex_framein,
|
|
.frameout = lintospeex_frameout,
|
|
.feedback = lintospeex_feedback,
|
|
.destroy = lintospeex_destroy,
|
|
.sample = slin16_sample,
|
|
.desc_size = sizeof(struct speex_coder_pvt),
|
|
.buffer_samples = BUFFER_SAMPLES,
|
|
.buf_size = BUFFER_SAMPLES * 2, /* XXX maybe a lot less ? */
|
|
};
|
|
|
|
static struct ast_translator speexuwbtolin32 = {
|
|
.name = "speexuwbtolin32",
|
|
.src_codec = {
|
|
.name = "speex",
|
|
.type = AST_MEDIA_TYPE_AUDIO,
|
|
.sample_rate = 32000,
|
|
},
|
|
.dst_codec = {
|
|
.name = "slin",
|
|
.type = AST_MEDIA_TYPE_AUDIO,
|
|
.sample_rate = 32000,
|
|
},
|
|
.format = "slin32",
|
|
.newpvt = speexuwbtolin32_new,
|
|
.framein = speextolin_framein,
|
|
.destroy = speextolin_destroy,
|
|
.desc_size = sizeof(struct speex_coder_pvt),
|
|
.buffer_samples = BUFFER_SAMPLES,
|
|
.buf_size = BUFFER_SAMPLES * 2,
|
|
.native_plc = 1,
|
|
};
|
|
|
|
static struct ast_translator lin32tospeexuwb = {
|
|
.name = "lin32tospeexuwb",
|
|
.src_codec = {
|
|
.name = "slin",
|
|
.type = AST_MEDIA_TYPE_AUDIO,
|
|
.sample_rate = 32000,
|
|
},
|
|
.dst_codec = {
|
|
.name = "speex",
|
|
.type = AST_MEDIA_TYPE_AUDIO,
|
|
.sample_rate = 32000,
|
|
},
|
|
.format = "speex32",
|
|
.newpvt = lin32tospeexuwb_new,
|
|
.framein = lintospeex_framein,
|
|
.frameout = lintospeex_frameout,
|
|
.feedback = lintospeex_feedback,
|
|
.destroy = lintospeex_destroy,
|
|
.desc_size = sizeof(struct speex_coder_pvt),
|
|
.buffer_samples = BUFFER_SAMPLES,
|
|
.buf_size = BUFFER_SAMPLES * 2, /* XXX maybe a lot less ? */
|
|
};
|
|
|
|
static int parse_config(int reload)
|
|
{
|
|
struct ast_flags config_flags = { reload ? CONFIG_FLAG_FILEUNCHANGED : 0 };
|
|
struct ast_config *cfg = ast_config_load("codecs.conf", config_flags);
|
|
struct ast_variable *var;
|
|
int res;
|
|
float res_f;
|
|
|
|
if (cfg == CONFIG_STATUS_FILEMISSING || cfg == CONFIG_STATUS_FILEUNCHANGED || cfg == CONFIG_STATUS_FILEINVALID)
|
|
return 0;
|
|
|
|
for (var = ast_variable_browse(cfg, "speex"); var; var = var->next) {
|
|
if (!strcasecmp(var->name, "quality")) {
|
|
res = abs(atoi(var->value));
|
|
if (res > -1 && res < 11) {
|
|
ast_verb(3, "CODEC SPEEX: Setting Quality to %d\n",res);
|
|
quality = res;
|
|
} else
|
|
ast_log(LOG_ERROR,"Error Quality must be 0-10\n");
|
|
} else if (!strcasecmp(var->name, "complexity")) {
|
|
res = abs(atoi(var->value));
|
|
if (res > -1 && res < 11) {
|
|
ast_verb(3, "CODEC SPEEX: Setting Complexity to %d\n",res);
|
|
complexity = res;
|
|
} else
|
|
ast_log(LOG_ERROR,"Error! Complexity must be 0-10\n");
|
|
} else if (!strcasecmp(var->name, "vbr_quality")) {
|
|
if (sscanf(var->value, "%30f", &res_f) == 1 && res_f >= 0 && res_f <= 10) {
|
|
ast_verb(3, "CODEC SPEEX: Setting VBR Quality to %f\n",res_f);
|
|
vbr_quality = res_f;
|
|
} else
|
|
ast_log(LOG_ERROR,"Error! VBR Quality must be 0-10\n");
|
|
} else if (!strcasecmp(var->name, "abr_quality")) {
|
|
ast_log(LOG_ERROR,"Error! ABR Quality setting obsolete, set ABR to desired bitrate\n");
|
|
} else if (!strcasecmp(var->name, "enhancement")) {
|
|
enhancement = ast_true(var->value) ? 1 : 0;
|
|
ast_verb(3, "CODEC SPEEX: Perceptual Enhancement Mode. [%s]\n",enhancement ? "on" : "off");
|
|
} else if (!strcasecmp(var->name, "vbr")) {
|
|
vbr = ast_true(var->value) ? 1 : 0;
|
|
ast_verb(3, "CODEC SPEEX: VBR Mode. [%s]\n",vbr ? "on" : "off");
|
|
} else if (!strcasecmp(var->name, "abr")) {
|
|
res = abs(atoi(var->value));
|
|
if (res >= 0) {
|
|
if (res > 0)
|
|
ast_verb(3, "CODEC SPEEX: Setting ABR target bitrate to %d\n",res);
|
|
else
|
|
ast_verb(3, "CODEC SPEEX: Disabling ABR\n");
|
|
abr = res;
|
|
} else
|
|
ast_log(LOG_ERROR,"Error! ABR target bitrate must be >= 0\n");
|
|
} else if (!strcasecmp(var->name, "vad")) {
|
|
vad = ast_true(var->value) ? 1 : 0;
|
|
ast_verb(3, "CODEC SPEEX: VAD Mode. [%s]\n",vad ? "on" : "off");
|
|
} else if (!strcasecmp(var->name, "dtx")) {
|
|
dtx = ast_true(var->value) ? 1 : 0;
|
|
ast_verb(3, "CODEC SPEEX: DTX Mode. [%s]\n",dtx ? "on" : "off");
|
|
} else if (!strcasecmp(var->name, "preprocess")) {
|
|
preproc = ast_true(var->value) ? 1 : 0;
|
|
ast_verb(3, "CODEC SPEEX: Preprocessing. [%s]\n",preproc ? "on" : "off");
|
|
} else if (!strcasecmp(var->name, "pp_vad")) {
|
|
pp_vad = ast_true(var->value) ? 1 : 0;
|
|
ast_verb(3, "CODEC SPEEX: Preprocessor VAD. [%s]\n",pp_vad ? "on" : "off");
|
|
} else if (!strcasecmp(var->name, "pp_agc")) {
|
|
pp_agc = ast_true(var->value) ? 1 : 0;
|
|
ast_verb(3, "CODEC SPEEX: Preprocessor AGC. [%s]\n",pp_agc ? "on" : "off");
|
|
} else if (!strcasecmp(var->name, "pp_agc_level")) {
|
|
if (sscanf(var->value, "%30f", &res_f) == 1 && res_f >= 0) {
|
|
ast_verb(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;
|
|
ast_verb(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;
|
|
ast_verb(3, "CODEC SPEEX: Preprocessor Dereverb. [%s]\n",pp_dereverb ? "on" : "off");
|
|
} else if (!strcasecmp(var->name, "pp_dereverb_decay")) {
|
|
if (sscanf(var->value, "%30f", &res_f) == 1 && res_f >= 0) {
|
|
ast_verb(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, "%30f", &res_f) == 1 && res_f >= 0) {
|
|
ast_verb(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");
|
|
} else if (!strcasecmp(var->name, "experimental_rtcp_feedback")) {
|
|
exp_rtcp_fb = ast_true(var->value) ? 1 : 0;
|
|
ast_verb(3, "CODEC SPEEX: Experimental RTCP Feedback. [%s]\n",exp_rtcp_fb ? "on" : "off");
|
|
}
|
|
}
|
|
ast_config_destroy(cfg);
|
|
return 0;
|
|
}
|
|
|
|
static int reload(void)
|
|
{
|
|
if (parse_config(1))
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
return AST_MODULE_LOAD_SUCCESS;
|
|
}
|
|
|
|
static int unload_module(void)
|
|
{
|
|
ast_unregister_translator(&speextolin);
|
|
ast_unregister_translator(&lintospeex);
|
|
ast_unregister_translator(&speexwbtolin16);
|
|
ast_unregister_translator(&lin16tospeexwb);
|
|
ast_unregister_translator(&speexuwbtolin32);
|
|
ast_unregister_translator(&lin32tospeexuwb);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int load_module(void)
|
|
{
|
|
int res = 0;
|
|
|
|
if (parse_config(0)) {
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
|
|
/* XXX It is most likely a bug in this module if we fail to register a translator */
|
|
res |= ast_register_translator(&speextolin);
|
|
res |= ast_register_translator(&lintospeex);
|
|
res |= ast_register_translator(&speexwbtolin16);
|
|
res |= ast_register_translator(&lin16tospeexwb);
|
|
res |= ast_register_translator(&speexuwbtolin32);
|
|
res |= ast_register_translator(&lin32tospeexuwb);
|
|
if (res) {
|
|
unload_module();
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
|
|
return AST_MODULE_LOAD_SUCCESS;
|
|
}
|
|
|
|
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "Speex Coder/Decoder",
|
|
.support_level = AST_MODULE_SUPPORT_CORE,
|
|
.load = load_module,
|
|
.unload = unload_module,
|
|
.reload = reload,
|
|
);
|