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d760e81f37
-Functional changes 1. Dynamic global format list build by codecs defined in codecs.conf 2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf 3. Negotiation of SILK attributes in chan_sip. 4. SPEEX 32khz with translation 5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation using codec_resample.c 6. Various changes to RTP code required to properly handle the dynamic format list and formats with attributes. 7. ConfBridge now dynamically jumps to the best possible sample rate. This allows for conferences to take advantage of HD audio (Which sounds awesome) 8. Audiohooks are no longer limited to 8khz audio, and most effects have been updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT. 9. codec_resample now uses its own code rather than depending on libresample. -Organizational changes Global format list is moved from frame.c to format.c Various format specific functions moved from frame.c to format.c Review: https://reviewboard.asterisk.org/r/1104/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@308582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
141 lines
3.3 KiB
C
141 lines
3.3 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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***/
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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 = f->samples;
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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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