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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2011, Digium, Inc.
*
* Russell Bryant <russell@digium.com>
* David Vossel <dvossel@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 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");
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