747beb1ed1
In all non-pbx modules, AST_MODULE_LOAD_FAILURE has been changed to AST_MODULE_LOAD_DECLINE. This prevents asterisk from exiting if a module can't be loaded. If the user wishes to retain the FAILURE behavior for a specific module, they can use the "require" or "preload-require" keyword in modules.conf. A new API was added to logger: ast_is_logger_initialized(). This allows asterisk.c/check_init() to print to the error log once the logger subsystem is ready instead of just to stdout. If something does fail before the logger is initialized, we now print to stderr instead of stdout. Change-Id: I5f4b50623d9b5a6cb7c5624a8c5c1274c13b2b25
190 lines
4.2 KiB
C
190 lines
4.2 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2011, Digium, Inc.
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*
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* Russell Bryant <russell@digium.com>
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* David Vossel <dvossel@digium.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*!
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* \file
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*
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* \brief Resample slinear audio
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*
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* \ingroup codecs
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*/
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/*** MODULEINFO
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<support_level>core</support_level>
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***/
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#include "asterisk.h"
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#include "speex/speex_resampler.h"
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#include "asterisk/module.h"
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#include "asterisk/translate.h"
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#include "asterisk/slin.h"
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#define OUTBUF_SAMPLES 11520
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static struct ast_translator *translators;
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static int trans_size;
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static struct ast_codec codec_list[] = {
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{
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.name = "slin",
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.type = AST_MEDIA_TYPE_AUDIO,
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.sample_rate = 8000,
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},
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{
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.name = "slin",
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.type = AST_MEDIA_TYPE_AUDIO,
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.sample_rate = 12000,
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},
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{
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.name = "slin",
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.type = AST_MEDIA_TYPE_AUDIO,
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.sample_rate = 16000,
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},
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{
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.name = "slin",
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.type = AST_MEDIA_TYPE_AUDIO,
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.sample_rate = 24000,
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},
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{
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.name = "slin",
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.type = AST_MEDIA_TYPE_AUDIO,
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.sample_rate = 32000,
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},
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{
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.name = "slin",
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.type = AST_MEDIA_TYPE_AUDIO,
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.sample_rate = 44100,
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},
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{
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.name = "slin",
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.type = AST_MEDIA_TYPE_AUDIO,
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.sample_rate = 48000,
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},
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{
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.name = "slin",
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.type = AST_MEDIA_TYPE_AUDIO,
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.sample_rate = 96000,
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},
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{
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.name = "slin",
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.type = AST_MEDIA_TYPE_AUDIO,
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.sample_rate = 192000,
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},
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};
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static int resamp_new(struct ast_trans_pvt *pvt)
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{
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int err;
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if (!(pvt->pvt = speex_resampler_init(1, pvt->t->src_codec.sample_rate, pvt->t->dst_codec.sample_rate, 5, &err))) {
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return -1;
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}
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ast_assert(pvt->f.subclass.format == NULL);
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pvt->f.subclass.format = ao2_bump(ast_format_cache_get_slin_by_rate(pvt->t->dst_codec.sample_rate));
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return 0;
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}
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static void resamp_destroy(struct ast_trans_pvt *pvt)
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{
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SpeexResamplerState *resamp_pvt = pvt->pvt;
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speex_resampler_destroy(resamp_pvt);
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}
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static int resamp_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
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{
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SpeexResamplerState *resamp_pvt = pvt->pvt;
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unsigned int out_samples = OUTBUF_SAMPLES - pvt->samples;
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unsigned int in_samples;
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if (!f->datalen) {
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return -1;
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}
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in_samples = f->datalen / 2;
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speex_resampler_process_int(resamp_pvt,
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0,
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f->data.ptr,
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&in_samples,
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pvt->outbuf.i16 + pvt->samples,
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&out_samples);
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pvt->samples += out_samples;
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pvt->datalen += out_samples * 2;
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return 0;
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}
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static int unload_module(void)
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{
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int res = 0;
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int idx;
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for (idx = 0; idx < trans_size; idx++) {
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res |= ast_unregister_translator(&translators[idx]);
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}
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ast_free(translators);
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return res;
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}
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static int load_module(void)
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{
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int res = 0;
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int x, y, idx = 0;
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trans_size = ARRAY_LEN(codec_list) * (ARRAY_LEN(codec_list) - 1);
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if (!(translators = ast_calloc(1, sizeof(struct ast_translator) * trans_size))) {
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return AST_MODULE_LOAD_DECLINE;
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}
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for (x = 0; x < ARRAY_LEN(codec_list); x++) {
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for (y = 0; y < ARRAY_LEN(codec_list); y++) {
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if (x == y) {
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continue;
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}
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translators[idx].newpvt = resamp_new;
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translators[idx].destroy = resamp_destroy;
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translators[idx].framein = resamp_framein;
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translators[idx].desc_size = 0;
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translators[idx].buffer_samples = OUTBUF_SAMPLES;
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translators[idx].buf_size = (OUTBUF_SAMPLES * sizeof(int16_t));
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memcpy(&translators[idx].src_codec, &codec_list[x], sizeof(struct ast_codec));
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memcpy(&translators[idx].dst_codec, &codec_list[y], sizeof(struct ast_codec));
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snprintf(translators[idx].name, sizeof(translators[idx].name), "slin %ukhz -> %ukhz",
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translators[idx].src_codec.sample_rate, translators[idx].dst_codec.sample_rate);
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res |= ast_register_translator(&translators[idx]);
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idx++;
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}
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}
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/* in case ast_register_translator() failed, we call unload_module() and
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ast_unregister_translator won't fail.*/
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if (res) {
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unload_module();
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return AST_MODULE_LOAD_DECLINE;
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}
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return AST_MODULE_LOAD_SUCCESS;
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}
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AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "SLIN Resampling Codec");
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