codec_resample.c

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/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 2011, Digium, Inc.
 *
 * Russell Bryant <[email protected]>
 * David Vossel <[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 Resample slinear audio
 *
 * \ingroup codecs
 */

/*** MODULEINFO
 <support_level>core</support_level>
 ***/

#include "asterisk.h"
#include "speex/speex_resampler.h"

#include "asterisk/module.h"
#include "asterisk/translate.h"
#include "asterisk/slin.h"

#define OUTBUF_SAMPLES 11520

static struct ast_translator *translators;
static int trans_size;
static struct ast_codec codec_list[] = {
 {
 .name = "slin",
 .type = AST_MEDIA_TYPE_AUDIO,
 .sample_rate = 8000,
 },
 {
 .name = "slin",
 .type = AST_MEDIA_TYPE_AUDIO,
 .sample_rate = 12000,
 },
 {
 .name = "slin",
 .type = AST_MEDIA_TYPE_AUDIO,
 .sample_rate = 16000,
 },
 {
 .name = "slin",
 .type = AST_MEDIA_TYPE_AUDIO,
 .sample_rate = 24000,
 },
 {
 .name = "slin",
 .type = AST_MEDIA_TYPE_AUDIO,
 .sample_rate = 32000,
 },
 {
 .name = "slin",
 .type = AST_MEDIA_TYPE_AUDIO,
 .sample_rate = 44100,
 },
 {
 .name = "slin",
 .type = AST_MEDIA_TYPE_AUDIO,
 .sample_rate = 48000,
 },
 {
 .name = "slin",
 .type = AST_MEDIA_TYPE_AUDIO,
 .sample_rate = 96000,
 },
 {
 .name = "slin",
 .type = AST_MEDIA_TYPE_AUDIO,
 .sample_rate = 192000,
 },
};

static int resamp_new(struct ast_trans_pvt *pvt)
{
 int err;

 if (!(pvt->pvt = speex_resampler_init(1, pvt->t->src_codec.sample_rate, pvt->t->dst_codec.sample_rate, 5, &err))) {
 return -1;
 }

 ast_assert(pvt->f.subclass.format == NULL);
 pvt->f.subclass.format = ao2_bump(ast_format_cache_get_slin_by_rate(pvt->t->dst_codec.sample_rate));

 return 0;
}

static void resamp_destroy(struct ast_trans_pvt *pvt)
{
 SpeexResamplerState *resamp_pvt = pvt->pvt;

 speex_resampler_destroy(resamp_pvt);
}

static int resamp_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
{
 SpeexResamplerState *resamp_pvt = pvt->pvt;
 unsigned int out_samples = OUTBUF_SAMPLES - pvt->samples;
 unsigned int in_samples;

 if (!f->datalen) {
 return -1;
 }
 in_samples = f->datalen / 2;

 speex_resampler_process_int(resamp_pvt,
 0,
 f->data.ptr,
 &in_samples,
 pvt->outbuf.i16 + pvt->samples,
 &out_samples);

 pvt->samples += out_samples;
 pvt->datalen += out_samples * 2;

 return 0;
}

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

 for (idx = 0; idx < trans_size; idx++) {
 res |= ast_unregister_translator(&translators[idx]);
 }
 ast_free(translators);

 return res;
}

static int load_module(void)
{
 int res = 0;
 int x, y, idx = 0;

 trans_size = ARRAY_LEN(codec_list) * (ARRAY_LEN(codec_list) - 1);
 if (!(translators = ast_calloc(1, sizeof(struct ast_translator) * trans_size))) {
 return AST_MODULE_LOAD_DECLINE;
 }

 for (x = 0; x < ARRAY_LEN(codec_list); x++) {
 for (y = 0; y < ARRAY_LEN(codec_list); y++) {
 if (x == y) {
 continue;
 }
 translators[idx].newpvt = resamp_new;
 translators[idx].destroy = resamp_destroy;
 translators[idx].framein = resamp_framein;
 translators[idx].desc_size = 0;
 translators[idx].buffer_samples = OUTBUF_SAMPLES;
 translators[idx].buf_size = (OUTBUF_SAMPLES * sizeof(int16_t));
 memcpy(&translators[idx].src_codec, &codec_list[x], sizeof(struct ast_codec));
 memcpy(&translators[idx].dst_codec, &codec_list[y], sizeof(struct ast_codec));
 snprintf(translators[idx].name, sizeof(translators[idx].name), "slin %ukhz -> %ukhz",
 translators[idx].src_codec.sample_rate, translators[idx].dst_codec.sample_rate);
 res |= ast_register_translator(&translators[idx]);
 idx++;
 }

 }
 /* in case ast_register_translator() failed, we call unload_module() and
 ast_unregister_translator won't fail.*/
 if (res) {
 unload_module();
 return AST_MODULE_LOAD_DECLINE;
 }

 return AST_MODULE_LOAD_SUCCESS;
}

AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "SLIN Resampling Codec");