ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes all traces of it. Previously exported symbols removed: * __ast_register_file * __ast_unregister_file * ast_complete_source_filename This also removes the mtx_prof static variable that was declared when MTX_PROFILE was enabled. This variable was only used in lock.c so it is now initialized in that file only. ASTERISK-26480 #close Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966
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_FAILURE;
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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_FAILURE;
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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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