Asterisk - The Open Source Telephony Project
21.4.1
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Resample slinear audio. More...
#include "asterisk.h"
#include "speex/speex_resampler.h"
#include "asterisk/module.h"
#include "asterisk/translate.h"
#include "asterisk/slin.h"
Go to the source code of this file.
Macros | |
#define | OUTBUF_SAMPLES 11520 |
Functions | |
static void | __reg_module (void) |
static void | __unreg_module (void) |
struct ast_module * | AST_MODULE_SELF_SYM (void) |
static int | load_module (void) |
static void | resamp_destroy (struct ast_trans_pvt *pvt) |
static int | resamp_framein (struct ast_trans_pvt *pvt, struct ast_frame *f) |
static int | resamp_new (struct ast_trans_pvt *pvt) |
static int | unload_module (void) |
Variables | |
static struct ast_module_info | __mod_info = { .name = AST_MODULE, .flags = AST_MODFLAG_LOAD_ORDER , .description = "SLIN Resampling Codec" , .key = "This paragraph is copyright (c) 2006 by Digium, Inc. \In order for your module to load, it must return this \key via a function called \"key\". Any code which \includes this paragraph must be licensed under the GNU \General Public License version 2 or later (at your \option). In addition to Digium's general reservations \of rights, Digium expressly reserves the right to \allow other parties to license this paragraph under \different terms. Any use of Digium, Inc. trademarks or \logos (including \"Asterisk\" or \"Digium\") without \express written permission of Digium, Inc. is prohibited.\n" , .buildopt_sum = "da6642af068ee5e6490c5b1d2cc1d238" , .load = load_module, .unload = unload_module, .load_pri = AST_MODPRI_DEFAULT, .support_level = AST_MODULE_SUPPORT_CORE, } |
static const struct ast_module_info * | ast_module_info = &__mod_info |
static struct ast_codec | codec_list [] |
static int | trans_size |
static struct ast_translator * | translators |
Resample slinear audio.
Definition in file codec_resample.c.