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https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328247 | lmadsen | 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) | 14 lines Merged revisions 328209 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines Introduce <support_level> tags in MODULEINFO. This change introduces MODULEINFO into many modules in Asterisk in order to show the community support level for those modules. This is used by changes committed to menuselect by Russell Bryant recently (r917 in menuselect). More information about the support level types and what they mean is available on the wiki at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328259 65c4cc65-6c06-0410-ace0-fbb531ad65f3
147 lines
3.4 KiB
C
147 lines
3.4 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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<depend>resample</depend>
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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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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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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_SIZE 8096
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static struct ast_translator *translators;
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static int trans_size;
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static int id_list[] = {
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AST_FORMAT_SLINEAR,
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AST_FORMAT_SLINEAR12,
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AST_FORMAT_SLINEAR16,
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AST_FORMAT_SLINEAR24,
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AST_FORMAT_SLINEAR32,
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AST_FORMAT_SLINEAR44,
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AST_FORMAT_SLINEAR48,
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AST_FORMAT_SLINEAR96,
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AST_FORMAT_SLINEAR192,
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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, ast_format_rate(&pvt->t->src_format), ast_format_rate(&pvt->t->dst_format), 5, &err))) {
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return -1;
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}
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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_SIZE / sizeof(int16_t)) - 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(id_list) * ARRAY_LEN(id_list);
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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(id_list); x++) {
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for (y = 0; y < ARRAY_LEN(id_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_SIZE / sizeof(int16_t));
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translators[idx].buf_size = OUTBUF_SIZE;
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ast_format_set(&translators[idx].src_format, id_list[x], 0);
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ast_format_set(&translators[idx].dst_format, id_list[y], 0);
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snprintf(translators[idx].name, sizeof(translators[idx].name), "slin %dkhz -> %dkhz",
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ast_format_rate(&translators[idx].src_format), ast_format_rate(&translators[idx].dst_format));
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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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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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