codec_alaw.c

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/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 1999 - 2005, Digium, Inc.
 *
 * Mark Spencer <markster@digium.com>
 *
 * 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.
 *
 * 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.
 */
 
/*! \file
 *
 * \brief codec_alaw.c - translate between signed linear and alaw
 *
 * \ingroup codecs
 */
 
/*** 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"
 
#define BUFFER_SAMPLES   8096   /* size for the translation buffers */
 
/* Sample frame data */
#include "asterisk/slin.h"
#include "ex_alaw.h"
 
/*! \brief decode frame into lin and fill output buffer. */
static int alawtolin_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
{
    int i = f->samples;
    unsigned char *src = f->data.ptr;
    int16_t *dst = pvt->outbuf.i16 + pvt->samples;
 
    pvt->samples += i;
    pvt->datalen += i * 2;  /* 2 bytes/sample */
 
    while (i--)
        *dst++ = AST_ALAW(*src++);
 
    return 0;
}
 
/*! \brief convert and store input samples in output buffer */
static int lintoalaw_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
{
    int i = f->samples;
    char *dst = pvt->outbuf.c + pvt->samples;
    int16_t *src = f->data.ptr;
 
    pvt->samples += i;
    pvt->datalen += i;  /* 1 byte/sample */
 
    while (i--)
        *dst++ = AST_LIN2A(*src++);
 
    return 0;
}
 
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,
};
 
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,
};
 
static int unload_module(void)
{
    int res;
 
    res = ast_unregister_translator(&lintoalaw);
    res |= ast_unregister_translator(&alawtolin);
 
    return res;
}
 
static int load_module(void)
{
    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;
}
 
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,
);