codec_codec2.c

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/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 2016, Alexander Traud
 *
 * Alexander Traud <[email protected]>
 *
 * 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 Translate between signed linear and Codec 2
 *
 * \author Alexander Traud <[email protected]>
 *
 * \note http://www.rowetel.com/codec2.html
 *
 * \ingroup codecs
 */

/*** MODULEINFO
 <depend>codec2</depend>
 <support_level>core</support_level>
 ***/

#include "asterisk.h"

#include "asterisk/codec.h" /* for AST_MEDIA_TYPE_AUDIO */
#include "asterisk/frame.h" /* for ast_frame */
#include "asterisk/linkedlists.h" /* for AST_LIST_NEXT, etc */
#include "asterisk/logger.h" /* for ast_log, etc */
#include "asterisk/module.h"
#include "asterisk/rtp_engine.h" /* ast_rtp_engine_(un)load_format */
#include "asterisk/translate.h" /* for ast_trans_pvt, etc */

#include <codec2/codec2.h>

#define BUFFER_SAMPLES 8000
#define CODEC2_SAMPLES 160 /* consider codec2_samples_per_frame(.) */
#define CODEC2_FRAME_LEN 6 /* consider codec2_bits_per_frame(.) */

/* Sample frame data */
#include "asterisk/slin.h"
#include "ex_codec2.h"

struct codec2_translator_pvt {
 struct CODEC2 *state; /* May be encoder or decoder */
 int16_t buf[BUFFER_SAMPLES];
};

static int codec2_new(struct ast_trans_pvt *pvt)
{
 struct codec2_translator_pvt *tmp = pvt->pvt;

 tmp->state = codec2_create(CODEC2_MODE_2400);

 if (!tmp->state) {
 ast_log(LOG_ERROR, "Error creating Codec 2 conversion\n");
 return -1;
 }

 return 0;
}

/*! \brief decode and store in outbuf. */
static int codec2tolin_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
{
 struct codec2_translator_pvt *tmp = pvt->pvt;
 int x;

 for (x = 0; x < f->datalen; x += CODEC2_FRAME_LEN) {
 unsigned char *src = f->data.ptr + x;
 int16_t *dst = pvt->outbuf.i16 + pvt->samples;

 codec2_decode(tmp->state, dst, src);

 pvt->samples += CODEC2_SAMPLES;
 pvt->datalen += CODEC2_SAMPLES * 2;
 }

 return 0;
}

/*! \brief store samples into working buffer for later decode */
static int lintocodec2_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
{
 struct codec2_translator_pvt *tmp = pvt->pvt;

 memcpy(tmp->buf + pvt->samples, f->data.ptr, f->datalen);
 pvt->samples += f->samples;

 return 0;
}

/*! \brief encode and produce a frame */
static struct ast_frame *lintocodec2_frameout(struct ast_trans_pvt *pvt)
{
 struct codec2_translator_pvt *tmp = pvt->pvt;
 struct ast_frame *result = NULL;
 struct ast_frame *last = NULL;
 int samples = 0; /* output samples */

 while (pvt->samples >= CODEC2_SAMPLES) {
 struct ast_frame *current;

 /* Encode a frame of data */
 codec2_encode(tmp->state, pvt->outbuf.uc, tmp->buf + samples);

 samples += CODEC2_SAMPLES;
 pvt->samples -= CODEC2_SAMPLES;

 current = ast_trans_frameout(pvt, CODEC2_FRAME_LEN, CODEC2_SAMPLES);

 if (!current) {
 continue;
 } else if (last) {
 AST_LIST_NEXT(last, frame_list) = current;
 } else {
 result = current;
 }
 last = current;
 }

 /* Move the data at the end of the buffer to the front */
 if (samples) {
 memmove(tmp->buf, tmp->buf + samples, pvt->samples * 2);
 }

 return result;
}

static void codec2_destroy_stuff(struct ast_trans_pvt *pvt)
{
 struct codec2_translator_pvt *tmp = pvt->pvt;

 if (tmp->state) {
 codec2_destroy(tmp->state);
 }
}

static struct ast_translator codec2tolin = {
 .name = "codec2tolin",
 .src_codec = {
 .name = "codec2",
 .type = AST_MEDIA_TYPE_AUDIO,
 .sample_rate = 8000,
 },
 .dst_codec = {
 .name = "slin",
 .type = AST_MEDIA_TYPE_AUDIO,
 .sample_rate = 8000,
 },
 .format = "slin",
 .newpvt = codec2_new,
 .framein = codec2tolin_framein,
 .destroy = codec2_destroy_stuff,
 .sample = codec2_sample,
 .desc_size = sizeof(struct codec2_translator_pvt),
 .buffer_samples = BUFFER_SAMPLES,
 .buf_size = BUFFER_SAMPLES * 2,
};

static struct ast_translator lintocodec2 = {
 .name = "lintocodec2",
 .src_codec = {
 .name = "slin",
 .type = AST_MEDIA_TYPE_AUDIO,
 .sample_rate = 8000,
 },
 .dst_codec = {
 .name = "codec2",
 .type = AST_MEDIA_TYPE_AUDIO,
 .sample_rate = 8000,
 },
 .format = "codec2",
 .newpvt = codec2_new,
 .framein = lintocodec2_framein,
 .frameout = lintocodec2_frameout,
 .destroy = codec2_destroy_stuff,
 .sample = slin8_sample,
 .desc_size = sizeof(struct codec2_translator_pvt),
 .buffer_samples = BUFFER_SAMPLES,
 .buf_size = (BUFFER_SAMPLES * CODEC2_FRAME_LEN + CODEC2_SAMPLES - 1) / CODEC2_SAMPLES,
};

static int unload_module(void)
{
 int res = 0;

 res |= ast_rtp_engine_unload_format(ast_format_codec2);
 res |= ast_unregister_translator(&lintocodec2);
 res |= ast_unregister_translator(&codec2tolin);

 return res;
}

static int load_module(void)
{
 int res = 0;

 res |= ast_register_translator(&codec2tolin);
 res |= ast_register_translator(&lintocodec2);
 res |= ast_rtp_engine_load_format(ast_format_codec2);

 if (res) {
 unload_module();
 return AST_MODULE_LOAD_DECLINE;
 }

 return AST_MODULE_LOAD_SUCCESS;
}

AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Codec 2 Coder/Decoder");