This patch is the foundation of an entire new way of looking at media in Asterisk. The code present in this patch is everything required to complete phase1 of my Media Architecture proposal. For more information about this project visit the link below. https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal The primary function of this patch is to convert all the usages of format bitfields in Asterisk to use the new format and format_cap APIs. Functionally no change in behavior should be present in this patch. Thanks to twilson and russell for all the time they spent reviewing these changes. Review: https://reviewboard.asterisk.org/r/1083/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306010 65c4cc65-6c06-0410-ace0-fbb531ad65f3
233 lines
5.9 KiB
C
233 lines
5.9 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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/*** MODULEINFO
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<depend>gsm</depend>
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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 "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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#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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int datalen = 0;
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int samples = 0;
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/* We can't work on anything less than a frame in size */
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if (pvt->samples < GSM_SAMPLES)
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return NULL;
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while (pvt->samples >= GSM_SAMPLES) {
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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 + datalen);
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datalen += GSM_FRAME_LEN;
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samples += GSM_SAMPLES;
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pvt->samples -= GSM_SAMPLES;
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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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return ast_trans_frameout(pvt, datalen, samples);
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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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.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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.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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/*! \brief standard module glue */
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static int reload(void)
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{
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return AST_MODULE_LOAD_SUCCESS;
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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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if (!res)
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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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ast_format_set(&gsmtolin.src_format, AST_FORMAT_GSM, 0);
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ast_format_set(&gsmtolin.dst_format, AST_FORMAT_SLINEAR, 0);
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ast_format_set(&lintogsm.src_format, AST_FORMAT_SLINEAR, 0);
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ast_format_set(&lintogsm.dst_format, AST_FORMAT_GSM, 0);
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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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if (res)
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return AST_MODULE_LOAD_FAILURE;
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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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.load = load_module,
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.unload = unload_module,
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.reload = reload,
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);
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