git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@14952 65c4cc65-6c06-0410-ace0-fbb531ad65f3
358 lines
8.3 KiB
C
358 lines
8.3 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* The GSM code is from TOAST. Copyright information for that package is available
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* in the GSM directory.
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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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* 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 Global System for Mobile Communications (GSM)
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*
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* \ingroup codecs
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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 "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/config.h"
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#include "asterisk/options.h"
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#include "asterisk/module.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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#ifdef USE_EXTERNAL_GSM_LIB
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#include <gsm/gsm.h>
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#else
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#include "gsm/inc/gsm.h"
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#endif
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#include "../formats/msgsm.h"
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/* Sample frame data */
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#include "slin_gsm_ex.h"
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#include "gsm_slin_ex.h"
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AST_MUTEX_DEFINE_STATIC(localuser_lock);
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static int localusecnt=0;
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static char *tdesc = "GSM/PCM16 (signed linear) Codec Translator";
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static int useplc = 0;
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struct ast_translator_pvt {
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gsm gsm;
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struct ast_frame f;
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/* Space to build offset */
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char offset[AST_FRIENDLY_OFFSET];
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/* Buffer for our outgoing frame */
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short outbuf[8000];
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/* Enough to store a full second */
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short buf[8000];
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int tail;
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plc_state_t plc;
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};
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#define gsm_coder_pvt ast_translator_pvt
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static struct ast_translator_pvt *gsm_new(void)
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{
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struct gsm_coder_pvt *tmp;
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if ((tmp = ast_malloc(sizeof(*tmp)))) {
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if (!(tmp->gsm = gsm_create())) {
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free(tmp);
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tmp = NULL;
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}
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tmp->tail = 0;
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plc_init(&tmp->plc);
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localusecnt++;
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}
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return tmp;
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}
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static struct ast_frame *lintogsm_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_gsm_ex);
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/* Assume 8000 Hz */
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f.samples = sizeof(slin_gsm_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_gsm_ex;
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return &f;
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}
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static struct ast_frame *gsmtolin_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_GSM;
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f.datalen = sizeof(gsm_slin_ex);
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/* All frames are 20 ms long */
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f.samples = 160;
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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 = gsm_slin_ex;
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return &f;
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}
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static struct ast_frame *gsmtolin_frameout(struct ast_translator_pvt *tmp)
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{
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if (!tmp->tail)
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return NULL;
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/* Signed linear is no particular frame size, so just send whatever
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we have in the buffer in one lump sum */
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tmp->f.frametype = AST_FRAME_VOICE;
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tmp->f.subclass = AST_FORMAT_SLINEAR;
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tmp->f.datalen = tmp->tail * 2;
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/* Assume 8000 Hz */
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tmp->f.samples = tmp->tail;
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tmp->f.mallocd = 0;
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tmp->f.offset = AST_FRIENDLY_OFFSET;
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tmp->f.src = __PRETTY_FUNCTION__;
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tmp->f.data = tmp->buf;
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/* Reset tail pointer */
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tmp->tail = 0;
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return &tmp->f;
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}
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static int gsmtolin_framein(struct ast_translator_pvt *tmp, struct ast_frame *f)
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{
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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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unsigned char data[66];
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int msgsm=0;
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if(f->datalen == 0) { /* perform PLC with nominal framesize of 20ms/160 samples */
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if((tmp->tail + 160) > sizeof(tmp->buf) / 2) {
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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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if(useplc) {
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plc_fillin(&tmp->plc, tmp->buf+tmp->tail, 160);
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tmp->tail += 160;
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}
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return 0;
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}
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if ((f->datalen % 33) && (f->datalen % 65)) {
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ast_log(LOG_WARNING, "Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from %s (%d)?\n", f->src, f->datalen);
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return -1;
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}
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if (f->datalen % 65 == 0)
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msgsm = 1;
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for (x=0;x<f->datalen;x+=(msgsm ? 65 : 33)) {
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if (msgsm) {
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/* Translate MSGSM format to Real GSM format before feeding in */
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conv65(f->data + x, data);
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if (tmp->tail + 320 < sizeof(tmp->buf)/2) {
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if (gsm_decode(tmp->gsm, data, tmp->buf + tmp->tail)) {
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ast_log(LOG_WARNING, "Invalid GSM data (1)\n");
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return -1;
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}
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tmp->tail+=160;
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if (gsm_decode(tmp->gsm, data + 33, tmp->buf + tmp->tail)) {
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ast_log(LOG_WARNING, "Invalid GSM data (2)\n");
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return -1;
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}
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tmp->tail+=160;
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} else {
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ast_log(LOG_WARNING, "Out of (MS) buffer space\n");
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return -1;
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}
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} else {
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if (tmp->tail + 160 < sizeof(tmp->buf)/2) {
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if (gsm_decode(tmp->gsm, f->data + x, tmp->buf + tmp->tail)) {
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ast_log(LOG_WARNING, "Invalid GSM data\n");
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return -1;
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}
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tmp->tail+=160;
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} else {
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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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}
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}
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/* just add the last 20ms frame; there must have been at least one */
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if(useplc) plc_rx(&tmp->plc, tmp->buf+tmp->tail-160, 160);
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return 0;
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}
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static int lintogsm_framein(struct ast_translator_pvt *tmp, struct ast_frame *f)
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{
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/* Just add the frames to our stream */
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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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if (tmp->tail + f->datalen/2 < sizeof(tmp->buf) / 2) {
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memcpy((tmp->buf + tmp->tail), f->data, f->datalen);
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tmp->tail += f->datalen/2;
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} else {
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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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return 0;
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}
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static struct ast_frame *lintogsm_frameout(struct ast_translator_pvt *tmp)
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{
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int x=0;
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/* We can't work on anything less than a frame in size */
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if (tmp->tail < 160)
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return NULL;
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tmp->f.frametype = AST_FRAME_VOICE;
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tmp->f.subclass = AST_FORMAT_GSM;
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tmp->f.mallocd = 0;
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tmp->f.offset = AST_FRIENDLY_OFFSET;
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tmp->f.src = __PRETTY_FUNCTION__;
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tmp->f.data = tmp->outbuf;
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while(tmp->tail >= 160) {
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if ((x+1) * 33 >= sizeof(tmp->outbuf)) {
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ast_log(LOG_WARNING, "Out of buffer space\n");
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break;
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}
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/* Encode a frame of data */
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gsm_encode(tmp->gsm, tmp->buf, ((gsm_byte *) tmp->outbuf) + (x * 33));
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/* Assume 8000 Hz -- 20 ms */
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tmp->tail -= 160;
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/* Move the data at the end of the buffer to the front */
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if (tmp->tail)
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memmove(tmp->buf, tmp->buf + 160, tmp->tail * 2);
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x++;
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}
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tmp->f.datalen = x * 33;
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tmp->f.samples = x * 160;
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return &tmp->f;
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}
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static void gsm_destroy_stuff(struct ast_translator_pvt *pvt)
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{
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if (pvt->gsm)
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gsm_destroy(pvt->gsm);
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free(pvt);
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localusecnt--;
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}
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static struct ast_translator gsmtolin =
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{ "gsmtolin",
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AST_FORMAT_GSM, AST_FORMAT_SLINEAR,
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gsm_new,
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gsmtolin_framein,
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gsmtolin_frameout,
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gsm_destroy_stuff,
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gsmtolin_sample
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};
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static struct ast_translator lintogsm =
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{ "lintogsm",
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AST_FORMAT_SLINEAR, AST_FORMAT_GSM,
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gsm_new,
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lintogsm_framein,
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lintogsm_frameout,
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gsm_destroy_stuff,
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lintogsm_sample
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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;
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struct ast_variable *var;
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if ((cfg = ast_config_load("codecs.conf"))) {
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if ((var = ast_variable_browse(cfg, "plc"))) {
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while (var) {
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if (!strcasecmp(var->name, "genericplc")) {
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useplc = ast_true(var->value) ? 1 : 0;
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if (option_verbose > 2)
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ast_verbose(VERBOSE_PREFIX_3 "codec_gsm: %susing generic PLC\n", useplc ? "" : "not ");
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}
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var = var->next;
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}
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}
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ast_config_destroy(cfg);
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}
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}
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int reload(void)
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{
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parse_config();
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return 0;
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}
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int unload_module(void)
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{
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int res;
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ast_mutex_lock(&localuser_lock);
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res = ast_unregister_translator(&lintogsm);
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if (!res)
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res = ast_unregister_translator(&gsmtolin);
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if (localusecnt)
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res = -1;
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ast_mutex_unlock(&localuser_lock);
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return res;
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}
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int load_module(void)
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{
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int res;
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parse_config();
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res=ast_register_translator(&gsmtolin);
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if (!res)
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res=ast_register_translator(&lintogsm);
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else
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ast_unregister_translator(&gsmtolin);
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return res;
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}
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char *description(void)
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{
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return tdesc;
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}
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int usecount(void)
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{
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int res;
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STANDARD_USECOUNT(res);
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return res;
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}
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char *key()
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{
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return ASTERISK_GPL_KEY;
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}
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