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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"
ASTERISK_FILE_VERSION(__FILE__, "$Revision: 385582 $")
#include "asterisk/module.h"
#include "asterisk/translate.h"
#include "asterisk/slin.h"
#define OUTBUF_SIZE 8096
static struct ast_translator *translators;
static int trans_size;
static int id_list[] = {
AST_FORMAT_SLINEAR,
AST_FORMAT_SLINEAR12,
AST_FORMAT_SLINEAR16,
AST_FORMAT_SLINEAR24,
AST_FORMAT_SLINEAR32,
AST_FORMAT_SLINEAR44,
AST_FORMAT_SLINEAR48,
AST_FORMAT_SLINEAR96,
AST_FORMAT_SLINEAR192,
};
static int resamp_new(struct ast_trans_pvt *pvt)
{
int err;
if (!(pvt->pvt = speex_resampler_init(1, ast_format_rate(&pvt->t->src_format), ast_format_rate(&pvt->t->dst_format), 5, &err))) {
return -1;
}
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_SIZE / sizeof(int16_t)) - 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(id_list) * (ARRAY_LEN(id_list) - 1);
if (!(translators = ast_calloc(1, sizeof(struct ast_translator) * trans_size))) {
return AST_MODULE_LOAD_FAILURE;
}
for (x = 0; x < ARRAY_LEN(id_list); x++) {
for (y = 0; y < ARRAY_LEN(id_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_SIZE / sizeof(int16_t));
translators[idx].buf_size = OUTBUF_SIZE;
ast_format_set(&translators[idx].src_format, id_list[x], 0);
ast_format_set(&translators[idx].dst_format, id_list[y], 0);
snprintf(translators[idx].name, sizeof(translators[idx].name), "slin %dkhz -> %dkhz",
ast_format_rate(&translators[idx].src_format), ast_format_rate(&translators[idx].dst_format));
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_FAILURE;
}
return AST_MODULE_LOAD_SUCCESS;
}
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "SLIN Resampling Codec");
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