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asterisk/codecs/codec_alaw.c
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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2005, Digium, Inc.
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*
* Mark Spencer <markster@digium.com>
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*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
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*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
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/*! \file
*
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* \brief codec_alaw.c - translate between signed linear and alaw
*
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* \ingroup codecs
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*/
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/*** MODULEINFO
<support_level>core</support_level>
***/
#include "asterisk.h"
#include "asterisk/module.h"
#include "asterisk/config.h"
#include "asterisk/translate.h"
#include "asterisk/alaw.h"
#include "asterisk/utils.h"
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#define BUFFER_SAMPLES 8096 /* size for the translation buffers */
/* Sample frame data */
#include "asterisk/slin.h"
#include "ex_alaw.h"
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/*! \brief decode frame into lin and fill output buffer. */
static int alawtolin_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
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{
int i = f->samples;
unsigned char *src = f->data.ptr;
int16_t *dst = pvt->outbuf.i16 + pvt->samples;
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pvt->samples += i;
pvt->datalen += i * 2; /* 2 bytes/sample */
while (i--)
*dst++ = AST_ALAW(*src++);
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return 0;
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}
/*! \brief convert and store input samples in output buffer */
static int lintoalaw_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
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{
int i = f->samples;
char *dst = pvt->outbuf.c + pvt->samples;
int16_t *src = f->data.ptr;
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pvt->samples += i;
pvt->datalen += i; /* 1 byte/sample */
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while (i--)
*dst++ = AST_LIN2A(*src++);
return 0;
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}
static struct ast_translator alawtolin = {
.name = "alawtolin",
.src_codec = {
.name = "alaw",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 8000,
},
.dst_codec = {
.name = "slin",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 8000,
},
.format = "slin",
.framein = alawtolin_framein,
.sample = alaw_sample,
.buffer_samples = BUFFER_SAMPLES,
.buf_size = BUFFER_SAMPLES * 2,
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};
static struct ast_translator lintoalaw = {
.name = "lintoalaw",
.src_codec = {
.name = "slin",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 8000,
},
.dst_codec = {
.name = "alaw",
.type = AST_MEDIA_TYPE_AUDIO,
.sample_rate = 8000,
},
.format = "alaw",
.framein = lintoalaw_framein,
.sample = slin8_sample,
.buffer_samples = BUFFER_SAMPLES,
.buf_size = BUFFER_SAMPLES,
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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(&lintoalaw);
res |= ast_unregister_translator(&alawtolin);
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return res;
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}
static int load_module(void)
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{
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int res;
res = ast_register_translator(&alawtolin);
res |= ast_register_translator(&lintoalaw);
if (res) {
unload_module();
return AST_MODULE_LOAD_DECLINE;
}
return AST_MODULE_LOAD_SUCCESS;
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}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "A-law Coder/Decoder",
.support_level = AST_MODULE_SUPPORT_CORE,
.load = load_module,
.unload = unload_module,
);