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when a file is invalid from when a file is missing. This is most important when we have two configuration files. Consider the following example: Old system: sip.conf users.conf Old result New result ======== ========== ========== ========== Missing Missing SIP doesn't load SIP doesn't load Missing OK SIP doesn't load SIP doesn't load Missing Invalid SIP doesn't load SIP doesn't load OK Missing SIP loads SIP loads OK OK SIP loads SIP loads OK Invalid SIP loads incompletely SIP doesn't load Invalid Missing SIP doesn't load SIP doesn't load Invalid OK SIP doesn't load SIP doesn't load Invalid Invalid SIP doesn't load SIP doesn't load So in the case when users.conf doesn't load because there's a typo that disrupts the syntax, we may only partially load users, instead of failing with an error, which may cause some calls not to get processed. Worse yet, the old system would do this with no indication that anything was even wrong. (closes issue #10690) Reported by: dtyoo Patches: 20080716__bug10690.diff.txt uploaded by Corydon76 (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142992 65c4cc65-6c06-0410-ace0-fbb531ad65f3
503 lines
15 KiB
C
503 lines
15 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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* \extref 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>speexdsp</use>
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***/
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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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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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#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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/* 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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/* 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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#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 int 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 -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, 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 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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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 -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 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.ptr = 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.ptr = 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 = 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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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 + 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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}
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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 + samples, pvt->samples * 2);
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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.c, 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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.native_plc = 1,
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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 int parse_config(int reload)
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{
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struct ast_flags config_flags = { reload ? CONFIG_FLAG_FILEUNCHANGED : 0 };
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struct ast_config *cfg = ast_config_load("codecs.conf", config_flags);
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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 == CONFIG_STATUS_FILEMISSING || cfg == CONFIG_STATUS_FILEUNCHANGED || cfg == CONFIG_STATUS_FILEINVALID)
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return 0;
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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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ast_verb(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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ast_verb(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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ast_verb(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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ast_verb(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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ast_verb(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 (res > 0)
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ast_verb(3, "CODEC SPEEX: Setting ABR target bitrate to %d\n",res);
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else
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ast_verb(3, "CODEC SPEEX: Disabling ABR\n");
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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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ast_verb(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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ast_verb(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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ast_verb(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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ast_verb(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;
|
|
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, "%f", &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, "%f", &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, "%f", &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");
|
|
}
|
|
}
|
|
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)
|
|
{
|
|
int res;
|
|
|
|
res = ast_unregister_translator(&lintospeex);
|
|
res |= ast_unregister_translator(&speextolin);
|
|
|
|
return res;
|
|
}
|
|
|
|
static int load_module(void)
|
|
{
|
|
int res;
|
|
|
|
if (parse_config(0))
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
res=ast_register_translator(&speextolin);
|
|
if (!res)
|
|
res=ast_register_translator(&lintospeex);
|
|
else
|
|
ast_unregister_translator(&speextolin);
|
|
if (res)
|
|
return AST_MODULE_LOAD_FAILURE;
|
|
return AST_MODULE_LOAD_SUCCESS;
|
|
}
|
|
|
|
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "Speex Coder/Decoder",
|
|
.load = load_module,
|
|
.unload = unload_module,
|
|
.reload = reload,
|
|
);
|