codec_codec2.c

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/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 2016, Alexander Traud
 *
 * Alexander Traud <pabstraud@compuserve.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 Translate between signed linear and Codec 2
 *
 * \author Alexander Traud <pabstraud@compuserve.com>
 *
 * \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");