Asterisk - The Open Source Telephony Project GIT-master-f36a736
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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.
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#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 = AST_BUILDOPT_SUM, .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.
#define OUTBUF_SAMPLES 11520 |
Definition at line 39 of file codec_resample.c.
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Definition at line 189 of file codec_resample.c.
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Definition at line 189 of file codec_resample.c.
struct ast_module * AST_MODULE_SELF_SYM | ( | void | ) |
Definition at line 189 of file codec_resample.c.
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Definition at line 149 of file codec_resample.c.
References ARRAY_LEN, ast_calloc, AST_MODULE_LOAD_DECLINE, AST_MODULE_LOAD_SUCCESS, ast_register_translator, codec_list, name, OUTBUF_SAMPLES, resamp_destroy(), resamp_framein(), resamp_new(), trans_size, and unload_module().
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Definition at line 105 of file codec_resample.c.
References ast_trans_pvt::pvt, and speex_resampler_destroy().
Referenced by load_module().
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Definition at line 112 of file codec_resample.c.
References ast_frame::data, ast_frame::datalen, ast_trans_pvt::datalen, ast_trans_pvt::i16, ast_trans_pvt::outbuf, OUTBUF_SAMPLES, ast_frame::ptr, ast_trans_pvt::pvt, ast_trans_pvt::samples, and speex_resampler_process_int().
Referenced by load_module().
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Definition at line 91 of file codec_resample.c.
References ao2_bump, ast_assert, ast_format_cache_get_slin_by_rate(), ast_translator::dst_codec, ast_trans_pvt::f, ast_frame_subclass::format, NULL, ast_trans_pvt::pvt, ast_codec::sample_rate, speex_resampler_init(), ast_translator::src_codec, ast_frame::subclass, and ast_trans_pvt::t.
Referenced by load_module().
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Definition at line 136 of file codec_resample.c.
References ast_free, ast_unregister_translator(), and trans_size.
Referenced by load_module().
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Definition at line 189 of file codec_resample.c.
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Definition at line 189 of file codec_resample.c.
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Definition at line 43 of file codec_resample.c.
Referenced by load_module().
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Definition at line 42 of file codec_resample.c.
Referenced by load_module(), and unload_module().
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Definition at line 41 of file codec_resample.c.