253 lines
6.1 KiB
C
253 lines
6.1 KiB
C
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/*
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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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/*** MODULEINFO
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<depend>gsm</depend>
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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 "asterisk/translate.h"
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#include "asterisk/config.h"
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#include "asterisk/module.h"
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#include "asterisk/utils.h"
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#include "asterisk/linkedlists.h"
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#ifdef HAVE_GSM_HEADER
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#include "gsm.h"
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#elif defined(HAVE_GSM_GSM_HEADER)
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#include <gsm/gsm.h>
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#endif
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#include "../formats/msgsm.h"
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#define BUFFER_SAMPLES 8000
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#define GSM_SAMPLES 160
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#define GSM_FRAME_LEN 33
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#define MSGSM_FRAME_LEN 65
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/* Sample frame data */
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#include "asterisk/slin.h"
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#include "ex_gsm.h"
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struct gsm_translator_pvt { /* both gsm2lin and lin2gsm */
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gsm gsm;
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int16_t buf[BUFFER_SAMPLES]; /* lin2gsm, temporary storage */
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};
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static int gsm_new(struct ast_trans_pvt *pvt)
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{
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struct gsm_translator_pvt *tmp = pvt->pvt;
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return (tmp->gsm = gsm_create()) ? 0 : -1;
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}
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/*! \brief decode and store in outbuf. */
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static int gsmtolin_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
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{
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struct gsm_translator_pvt *tmp = pvt->pvt;
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int x;
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int16_t *dst = pvt->outbuf.i16;
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/* guess format from frame len. 65 for MSGSM, 33 for regular GSM */
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int flen = (f->datalen % MSGSM_FRAME_LEN == 0) ?
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MSGSM_FRAME_LEN : GSM_FRAME_LEN;
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for (x=0; x < f->datalen; x += flen) {
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unsigned char data[2 * GSM_FRAME_LEN];
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unsigned char *src;
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int len;
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if (flen == MSGSM_FRAME_LEN) {
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len = 2*GSM_SAMPLES;
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src = data;
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/* Translate MSGSM format to Real GSM format before feeding in */
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/* XXX what's the point here! we should just work
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* on the full format.
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*/
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conv65(f->data.ptr + x, data);
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} else {
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len = GSM_SAMPLES;
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src = f->data.ptr + x;
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}
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/* XXX maybe we don't need to check */
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if (pvt->samples + len > 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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if (gsm_decode(tmp->gsm, src, dst + pvt->samples)) {
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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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pvt->samples += GSM_SAMPLES;
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pvt->datalen += 2 * GSM_SAMPLES;
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if (flen == MSGSM_FRAME_LEN) {
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if (gsm_decode(tmp->gsm, data + GSM_FRAME_LEN, dst + pvt->samples)) {
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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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pvt->samples += GSM_SAMPLES;
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pvt->datalen += 2 * GSM_SAMPLES;
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}
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}
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return 0;
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}
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/*! \brief store samples into working buffer for later decode */
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static int lintogsm_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
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{
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struct gsm_translator_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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if (pvt->samples + f->samples > 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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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 encode and produce a frame */
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static struct ast_frame *lintogsm_frameout(struct ast_trans_pvt *pvt)
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{
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struct gsm_translator_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 >= GSM_SAMPLES) {
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struct ast_frame *current;
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/* Encode a frame of data */
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gsm_encode(tmp->gsm, tmp->buf + samples, (gsm_byte *) pvt->outbuf.c);
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samples += GSM_SAMPLES;
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pvt->samples -= GSM_SAMPLES;
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current = ast_trans_frameout(pvt, GSM_FRAME_LEN, GSM_SAMPLES);
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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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static void gsm_destroy_stuff(struct ast_trans_pvt *pvt)
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{
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struct gsm_translator_pvt *tmp = pvt->pvt;
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if (tmp->gsm)
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gsm_destroy(tmp->gsm);
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}
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static struct ast_translator gsmtolin = {
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.name = "gsmtolin",
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.src_codec = {
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.name = "gsm",
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.type = AST_MEDIA_TYPE_AUDIO,
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.sample_rate = 8000,
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},
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.dst_codec = {
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.name = "slin",
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.type = AST_MEDIA_TYPE_AUDIO,
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.sample_rate = 8000,
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},
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.format = "slin",
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.newpvt = gsm_new,
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.framein = gsmtolin_framein,
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.destroy = gsm_destroy_stuff,
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.sample = gsm_sample,
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.buffer_samples = BUFFER_SAMPLES,
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.buf_size = BUFFER_SAMPLES * 2,
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.desc_size = sizeof (struct gsm_translator_pvt ),
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};
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static struct ast_translator lintogsm = {
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.name = "lintogsm",
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.src_codec = {
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.name = "slin",
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.type = AST_MEDIA_TYPE_AUDIO,
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.sample_rate = 8000,
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},
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.dst_codec = {
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.name = "gsm",
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.type = AST_MEDIA_TYPE_AUDIO,
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.sample_rate = 8000,
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},
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.format = "gsm",
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.newpvt = gsm_new,
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.framein = lintogsm_framein,
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.frameout = lintogsm_frameout,
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.destroy = gsm_destroy_stuff,
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.sample = slin8_sample,
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.desc_size = sizeof (struct gsm_translator_pvt ),
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.buf_size = (BUFFER_SAMPLES * GSM_FRAME_LEN + GSM_SAMPLES - 1)/GSM_SAMPLES,
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};
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static int unload_module(void)
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{
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int res;
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res = ast_unregister_translator(&lintogsm);
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res |= ast_unregister_translator(&gsmtolin);
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return res;
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}
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static int load_module(void)
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{
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int res;
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res = ast_register_translator(&gsmtolin);
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res |= ast_register_translator(&lintogsm);
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if (res) {
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unload_module();
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return AST_MODULE_LOAD_DECLINE;
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}
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return AST_MODULE_LOAD_SUCCESS;
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}
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AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "GSM Coder/Decoder",
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.support_level = AST_MODULE_SUPPORT_CORE,
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.load = load_module,
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.unload = unload_module,
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);
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