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fc63e07135
Patch by: junky Review: https://reviewboard.asterisk.org/r/1743/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362307 65c4cc65-6c06-0410-ace0-fbb531ad65f3
152 lines
3.6 KiB
C
152 lines
3.6 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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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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/* 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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