2011-02-03 16:22:10 +00:00
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/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2010, Digium, Inc.
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*
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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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* \brief Format API
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*
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* \author David Vossel <dvossel@digium.com>
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*/
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#ifndef _AST_FORMAT_H_
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#define _AST_FORMAT_H_
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Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-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
2011-02-22 23:04:49 +00:00
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#include "asterisk/astobj2.h"
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#include "asterisk/silk.h"
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2011-07-07 19:39:17 +00:00
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#include "asterisk/celt.h"
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2011-02-03 16:22:10 +00:00
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#define AST_FORMAT_ATTR_SIZE 128
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#define AST_FORMAT_INC 100000
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/*! This is the value that ends a var list of format attribute
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* key value pairs. */
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#define AST_FORMAT_ATTR_END -1
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/* \brief Format Categories*/
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enum ast_format_type {
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AST_FORMAT_TYPE_AUDIO = 1 * AST_FORMAT_INC,
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AST_FORMAT_TYPE_VIDEO = 2 * AST_FORMAT_INC,
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AST_FORMAT_TYPE_IMAGE = 3 * AST_FORMAT_INC,
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AST_FORMAT_TYPE_TEXT = 4 * AST_FORMAT_INC,
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};
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enum ast_format_id {
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/*! G.723.1 compression */
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AST_FORMAT_G723_1 = 1 + AST_FORMAT_TYPE_AUDIO,
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/*! GSM compression */
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AST_FORMAT_GSM = 2 + AST_FORMAT_TYPE_AUDIO,
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/*! Raw mu-law data (G.711) */
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AST_FORMAT_ULAW = 3 + AST_FORMAT_TYPE_AUDIO,
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/*! Raw A-law data (G.711) */
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AST_FORMAT_ALAW = 4 + AST_FORMAT_TYPE_AUDIO,
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/*! ADPCM (G.726, 32kbps, AAL2 codeword packing) */
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AST_FORMAT_G726_AAL2 = 5 + AST_FORMAT_TYPE_AUDIO,
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/*! ADPCM (IMA) */
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AST_FORMAT_ADPCM = 6 + AST_FORMAT_TYPE_AUDIO,
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/*! LPC10, 180 samples/frame */
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Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-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
2011-02-22 23:04:49 +00:00
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AST_FORMAT_LPC10 = 7 + AST_FORMAT_TYPE_AUDIO,
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2011-02-03 16:22:10 +00:00
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/*! G.729A audio */
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Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-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
2011-02-22 23:04:49 +00:00
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AST_FORMAT_G729A = 8 + AST_FORMAT_TYPE_AUDIO,
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2011-02-03 16:22:10 +00:00
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/*! SpeeX Free Compression */
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Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-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
2011-02-22 23:04:49 +00:00
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AST_FORMAT_SPEEX = 9 + AST_FORMAT_TYPE_AUDIO,
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2011-02-03 16:22:10 +00:00
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/*! iLBC Free Compression */
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Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-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
2011-02-22 23:04:49 +00:00
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AST_FORMAT_ILBC = 10 + AST_FORMAT_TYPE_AUDIO,
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2011-02-03 16:22:10 +00:00
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/*! ADPCM (G.726, 32kbps, RFC3551 codeword packing) */
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Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-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
2011-02-22 23:04:49 +00:00
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AST_FORMAT_G726 = 11 + AST_FORMAT_TYPE_AUDIO,
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2011-02-03 16:22:10 +00:00
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/*! G.722 */
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Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-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
2011-02-22 23:04:49 +00:00
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AST_FORMAT_G722 = 12 + AST_FORMAT_TYPE_AUDIO,
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2011-02-03 16:22:10 +00:00
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/*! G.722.1 (also known as Siren7, 32kbps assumed) */
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Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-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
2011-02-22 23:04:49 +00:00
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AST_FORMAT_SIREN7 = 13 + AST_FORMAT_TYPE_AUDIO,
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2011-02-03 16:22:10 +00:00
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/*! G.722.1 Annex C (also known as Siren14, 48kbps assumed) */
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Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-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
2011-02-22 23:04:49 +00:00
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AST_FORMAT_SIREN14 = 14 + AST_FORMAT_TYPE_AUDIO,
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2011-02-03 16:22:10 +00:00
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/*! G.719 (64 kbps assumed) */
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Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-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
2011-02-22 23:04:49 +00:00
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AST_FORMAT_G719 = 15 + AST_FORMAT_TYPE_AUDIO,
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2011-02-03 16:22:10 +00:00
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/*! SpeeX Wideband (16kHz) Free Compression */
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Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-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
2011-02-22 23:04:49 +00:00
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AST_FORMAT_SPEEX16 = 16 + AST_FORMAT_TYPE_AUDIO,
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2011-02-03 16:22:10 +00:00
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/*! Raw mu-law data (G.711) */
|
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-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
2011-02-22 23:04:49 +00:00
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AST_FORMAT_TESTLAW = 17 + AST_FORMAT_TYPE_AUDIO,
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/*! SILK format */
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AST_FORMAT_SILK = 18 + AST_FORMAT_TYPE_AUDIO,
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/*! Raw 16-bit Signed Linear (8000 Hz) PCM */
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AST_FORMAT_SLINEAR = 19 + AST_FORMAT_TYPE_AUDIO,
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/*! Raw 16-bit Signed Linear (12000 Hz) PCM */
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AST_FORMAT_SLINEAR12 = 20 + AST_FORMAT_TYPE_AUDIO,
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/*! Raw 16-bit Signed Linear (16000 Hz) PCM */
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AST_FORMAT_SLINEAR16 = 21 + AST_FORMAT_TYPE_AUDIO,
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/*! Raw 16-bit Signed Linear (24000 Hz) PCM */
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AST_FORMAT_SLINEAR24 = 22 + AST_FORMAT_TYPE_AUDIO,
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/*! Raw 16-bit Signed Linear (32000 Hz) PCM */
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AST_FORMAT_SLINEAR32 = 23 + AST_FORMAT_TYPE_AUDIO,
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/*! Raw 16-bit Signed Linear (44100 Hz) PCM just because we can. */
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AST_FORMAT_SLINEAR44 = 24 + AST_FORMAT_TYPE_AUDIO,
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/*! Raw 16-bit Signed Linear (48000 Hz) PCM */
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AST_FORMAT_SLINEAR48 = 25 + AST_FORMAT_TYPE_AUDIO,
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/*! Raw 16-bit Signed Linear (96000 Hz) PCM */
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AST_FORMAT_SLINEAR96 = 26 + AST_FORMAT_TYPE_AUDIO,
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|
|
/*! Raw 16-bit Signed Linear (192000 Hz) PCM. maybe we're taking this too far. */
|
|
|
|
|
AST_FORMAT_SLINEAR192 = 27 + AST_FORMAT_TYPE_AUDIO,
|
|
|
|
|
AST_FORMAT_SPEEX32 = 28 + AST_FORMAT_TYPE_AUDIO,
|
2011-07-07 19:39:17 +00:00
|
|
|
AST_FORMAT_CELT = 29 + AST_FORMAT_TYPE_AUDIO,
|
2011-02-03 16:22:10 +00:00
|
|
|
|
|
|
|
|
/*! H.261 Video */
|
|
|
|
|
AST_FORMAT_H261 = 1 + AST_FORMAT_TYPE_VIDEO,
|
|
|
|
|
/*! H.263 Video */
|
|
|
|
|
AST_FORMAT_H263 = 2 + AST_FORMAT_TYPE_VIDEO,
|
|
|
|
|
/*! H.263+ Video */
|
|
|
|
|
AST_FORMAT_H263_PLUS = 3 + AST_FORMAT_TYPE_VIDEO,
|
|
|
|
|
/*! H.264 Video */
|
|
|
|
|
AST_FORMAT_H264 = 4 + AST_FORMAT_TYPE_VIDEO,
|
|
|
|
|
/*! MPEG4 Video */
|
|
|
|
|
AST_FORMAT_MP4_VIDEO = 5 + AST_FORMAT_TYPE_VIDEO,
|
|
|
|
|
|
|
|
|
|
/*! JPEG Images */
|
|
|
|
|
AST_FORMAT_JPEG = 1 + AST_FORMAT_TYPE_IMAGE,
|
|
|
|
|
/*! PNG Images */
|
|
|
|
|
AST_FORMAT_PNG = 2 + AST_FORMAT_TYPE_IMAGE,
|
|
|
|
|
|
|
|
|
|
/*! T.140 RED Text format RFC 4103 */
|
|
|
|
|
AST_FORMAT_T140RED = 1 + AST_FORMAT_TYPE_TEXT,
|
|
|
|
|
/*! T.140 Text format - ITU T.140, RFC 4103 */
|
|
|
|
|
AST_FORMAT_T140 = 2 + AST_FORMAT_TYPE_TEXT,
|
|
|
|
|
};
|
|
|
|
|
|
|
|
|
|
/*! Determine what type of media a ast_format_id is. */
|
|
|
|
|
#define AST_FORMAT_GET_TYPE(id) (((int) (id / AST_FORMAT_INC)) * AST_FORMAT_INC)
|
|
|
|
|
|
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-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
2011-02-22 23:04:49 +00:00
|
|
|
|
2011-02-03 16:22:10 +00:00
|
|
|
/*! \brief This structure contains the buffer used for format attributes */
|
|
|
|
|
struct ast_format_attr {
|
|
|
|
|
/*! The buffer formats can use to represent attributes */
|
|
|
|
|
uint8_t format_attr[AST_FORMAT_ATTR_SIZE];
|
|
|
|
|
/*! If a format's payload needs to pass through that a new marker is required
|
|
|
|
|
* for RTP, this variable will be set. */
|
|
|
|
|
uint8_t rtp_marker_bit;
|
|
|
|
|
};
|
|
|
|
|
|
|
|
|
|
/*! \brief Represents a media format within Asterisk. */
|
|
|
|
|
struct ast_format {
|
|
|
|
|
/*! The unique id representing this format from all the other formats. */
|
|
|
|
|
enum ast_format_id id;
|
|
|
|
|
/*! Attribute structure used to associate attributes with a format. */
|
|
|
|
|
struct ast_format_attr fattr;
|
|
|
|
|
};
|
|
|
|
|
|
|
|
|
|
enum ast_format_cmp_res {
|
|
|
|
|
/*! structure 1 is identical to structure 2. */
|
|
|
|
|
AST_FORMAT_CMP_EQUAL = 0,
|
|
|
|
|
/*! structure 1 contains elements not in structure 2. */
|
|
|
|
|
AST_FORMAT_CMP_NOT_EQUAL,
|
|
|
|
|
/*! structure 1 is a proper subset of the elements in structure 2.*/
|
|
|
|
|
AST_FORMAT_CMP_SUBSET,
|
|
|
|
|
};
|
|
|
|
|
|
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-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
2011-02-22 23:04:49 +00:00
|
|
|
/*! \brief Definition of supported media formats (codecs) */
|
|
|
|
|
struct ast_format_list {
|
|
|
|
|
struct ast_format format; /*!< The unique format. */
|
|
|
|
|
char name[64]; /*!< short name */
|
|
|
|
|
unsigned int samplespersecond; /*!< Number of samples per second (8000/16000) */
|
|
|
|
|
char desc[128]; /*!< Description */
|
|
|
|
|
int fr_len; /*!< Single frame length in bytes */
|
|
|
|
|
int min_ms; /*!< Min value */
|
|
|
|
|
int max_ms; /*!< Max value */
|
|
|
|
|
int inc_ms; /*!< Increment */
|
|
|
|
|
int def_ms; /*!< Default value */
|
|
|
|
|
unsigned int flags; /*!< Smoother flags */
|
|
|
|
|
int cur_ms; /*!< Current value */
|
|
|
|
|
int custom_entry;
|
|
|
|
|
};
|
|
|
|
|
|
2011-02-03 16:22:10 +00:00
|
|
|
/*! \brief A format must register an attribute interface if it requires the use of the format attributes void pointer */
|
|
|
|
|
struct ast_format_attr_interface {
|
|
|
|
|
/*! format type */
|
|
|
|
|
enum ast_format_id id;
|
|
|
|
|
|
|
|
|
|
/*! \brief Determine if format_attr 1 is a subset of format_attr 2.
|
|
|
|
|
*
|
|
|
|
|
* \retval ast_format_cmp_res representing the result of comparing fattr1 and fattr2.
|
|
|
|
|
*/
|
|
|
|
|
enum ast_format_cmp_res (* const format_attr_cmp)(const struct ast_format_attr *fattr1, const struct ast_format_attr *fattr2);
|
|
|
|
|
|
|
|
|
|
/*! \brief Get joint attributes of same format type if they exist.
|
|
|
|
|
*
|
|
|
|
|
* \retval 0 if joint attributes exist
|
|
|
|
|
* \retval -1 if no joint attributes are present
|
|
|
|
|
*/
|
|
|
|
|
int (* const format_attr_get_joint)(const struct ast_format_attr *fattr1, const struct ast_format_attr *fattr2, struct ast_format_attr *result);
|
|
|
|
|
|
|
|
|
|
/*! \brief Set format capabilities from a list of key value pairs ending with AST_FORMAT_ATTR_END.
|
|
|
|
|
* \note This function does not need to call va_end of the va_list. */
|
|
|
|
|
void (* const format_attr_set)(struct ast_format_attr *format_attr, va_list ap);
|
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-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
2011-02-22 23:04:49 +00:00
|
|
|
|
|
|
|
|
/*!
|
|
|
|
|
* \brief Find out if format capabilities in va_list are in format.
|
|
|
|
|
* \note This function does not need to call va_end of the va_list.
|
|
|
|
|
*
|
|
|
|
|
* \note This function is optional. In many cases the format_attr_cmp
|
|
|
|
|
* function can be used to derive these results. If it is possible
|
|
|
|
|
* that some format attributes have no bearing on the equality of two formats, this
|
|
|
|
|
* function must exist.
|
|
|
|
|
*
|
|
|
|
|
* \retval 0 if all attributes exist
|
|
|
|
|
* \retval -1 if any of the attributes not present
|
|
|
|
|
*/
|
|
|
|
|
int (* const format_attr_isset)(const struct ast_format_attr *format_attr, va_list ap);
|
|
|
|
|
|
|
|
|
|
/*
|
|
|
|
|
* \brief Return a value for a specific format key. Return that value in the void pointer.
|
|
|
|
|
*
|
|
|
|
|
* \note It is not expected that all key value pairs can be returned, but those that can should
|
|
|
|
|
* be documented as such.
|
|
|
|
|
*
|
|
|
|
|
* \note This function is optional if key value pairs are not allowed to be accessed. This
|
|
|
|
|
* will result in -1 always being returned.
|
|
|
|
|
*
|
|
|
|
|
* \retval 0 Success, value was found and copied into void pointer.
|
|
|
|
|
* \retval -1 failure, Value was either not found, or not allowed to be accessed.
|
|
|
|
|
*/
|
|
|
|
|
int (* const format_attr_get_val)(const struct ast_format_attr *format_attr, int key, void *val);
|
2011-02-03 16:22:10 +00:00
|
|
|
};
|
|
|
|
|
|
|
|
|
|
/*!
|
|
|
|
|
* \brief This function is used to set an ast_format object to represent a media format
|
|
|
|
|
* with optional format attributes represented by format specific key value pairs.
|
|
|
|
|
*
|
|
|
|
|
* \param format to set
|
|
|
|
|
* \param id, format id to set on format
|
|
|
|
|
* \param set_attributes, are there attributes to set on this format. 0 == false, 1 == True.
|
|
|
|
|
* \param var list of attribute key value pairs, must end with AST_FORMAT_ATTR_END;
|
|
|
|
|
*
|
|
|
|
|
* \details Example usage.
|
|
|
|
|
* ast_format_set(format, AST_FORMAT_ULAW, 0); // no capability attributes are needed for ULAW
|
|
|
|
|
*
|
|
|
|
|
* ast_format_set(format, AST_FORMAT_SILK, 1, // SILK has capability attributes.
|
|
|
|
|
* AST_FORMAT_SILK_ATTR_RATE, 24000,
|
|
|
|
|
* AST_FORMAT_SILK_ATTR_RATE, 16000,
|
|
|
|
|
* AST_FORMAT_SILK_ATTR_RATE, 12000,
|
|
|
|
|
* AST_FORMAT_SILK_ATTR_RATE, 8000,
|
|
|
|
|
* AST_FORMAT_ATTR_END);
|
|
|
|
|
*
|
|
|
|
|
* \note This function will initialize the ast_format structure.
|
|
|
|
|
*
|
|
|
|
|
* \return Pointer to ast_format object, same pointer that is passed in
|
|
|
|
|
* by the first argument.
|
|
|
|
|
*/
|
|
|
|
|
struct ast_format *ast_format_set(struct ast_format *format, enum ast_format_id id, int set_attributes, ... );
|
|
|
|
|
|
|
|
|
|
/*!
|
|
|
|
|
* \brief After ast_format_set has been used on a function, this function can be used to
|
|
|
|
|
* set additional format attributes to the structure.
|
|
|
|
|
*
|
|
|
|
|
* \param format to set
|
|
|
|
|
* \param var list of attribute key value pairs, must end with AST_FORMAT_ATTR_END;
|
|
|
|
|
*
|
|
|
|
|
* \details Example usage.
|
|
|
|
|
* ast_format_set(format, AST_FORMAT_SILK, 0);
|
|
|
|
|
* ast_format_append(format, // SILK has capability attributes.
|
|
|
|
|
* AST_FORMAT_SILK_ATTR_RATE, 24000,
|
|
|
|
|
* AST_FORMAT_SILK_ATTR_RATE, 16000,
|
|
|
|
|
* AST_FORMAT_SILK_ATTR_RATE, 12000,
|
|
|
|
|
* AST_FORMAT_SILK_ATTR_RATE, 8000,
|
|
|
|
|
* AST_FORMAT_ATTR_END);
|
|
|
|
|
*
|
|
|
|
|
* \return Pointer to ast_format object, same pointer that is passed in
|
|
|
|
|
* by the first argument.
|
|
|
|
|
*/
|
|
|
|
|
struct ast_format *ast_format_append(struct ast_format *format, ... );
|
|
|
|
|
|
|
|
|
|
/*!
|
|
|
|
|
* \brief Clears the format stucture.
|
|
|
|
|
*/
|
|
|
|
|
void ast_format_clear(struct ast_format *format);
|
|
|
|
|
|
|
|
|
|
/*!
|
|
|
|
|
* \brief This function is used to set an ast_format object to represent a media format
|
|
|
|
|
* with optional capability attributes represented by format specific key value pairs.
|
|
|
|
|
*
|
|
|
|
|
* \details Example usage. Is this SILK format capable of 8khz
|
|
|
|
|
* is_8khz = ast_format_isset(format, AST_FORMAT_SILK_CAP_RATE, 8000);
|
|
|
|
|
*
|
|
|
|
|
* \return 0, The format key value pairs are within the capabilities defined in this structure.
|
|
|
|
|
* \return -1, The format key value pairs are _NOT_ within the capabilities of this structure.
|
|
|
|
|
*/
|
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-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
2011-02-22 23:04:49 +00:00
|
|
|
int ast_format_isset(const struct ast_format *format, ... );
|
|
|
|
|
|
|
|
|
|
/*!
|
|
|
|
|
* \brief Get a value from a format containing attributes.
|
|
|
|
|
* \note The key represents the format attribute to be retrieved, and the void pointer
|
|
|
|
|
* is to the structure that value will be stored in. It must be known what structure a
|
|
|
|
|
* key represents.
|
|
|
|
|
*
|
|
|
|
|
* \retval 0, success
|
|
|
|
|
* \retval -1, failure
|
|
|
|
|
*/
|
|
|
|
|
int ast_format_get_value(const struct ast_format *format, int key, void *value);
|
2011-02-03 16:22:10 +00:00
|
|
|
|
|
|
|
|
/*!
|
|
|
|
|
* \brief Compare ast_formats structures
|
|
|
|
|
*
|
|
|
|
|
* \retval ast_format_cmp_res representing the result of comparing format1 and format2.
|
|
|
|
|
*/
|
|
|
|
|
enum ast_format_cmp_res ast_format_cmp(const struct ast_format *format1, const struct ast_format *format2);
|
|
|
|
|
|
|
|
|
|
/*!
|
|
|
|
|
* \brief Find joint format attributes of two ast_format
|
|
|
|
|
* structures containing the same uid and return the intersection in the
|
|
|
|
|
* result structure.
|
|
|
|
|
*
|
|
|
|
|
* retval 0, joint attribute capabilities exist.
|
|
|
|
|
* retval -1, no joint attribute capabilities exist.
|
|
|
|
|
*/
|
|
|
|
|
int ast_format_joint(const struct ast_format *format1, const struct ast_format *format2, struct ast_format *result);
|
|
|
|
|
|
|
|
|
|
/*!
|
|
|
|
|
* \brief copy format src into format dst.
|
|
|
|
|
*/
|
|
|
|
|
void ast_format_copy(struct ast_format *dst, const struct ast_format *src);
|
|
|
|
|
|
|
|
|
|
/*!
|
|
|
|
|
* \brief Set the rtp mark value on the format to indicate to the interface
|
|
|
|
|
* writing this format's payload that a new RTP marker is necessary.
|
|
|
|
|
*/
|
|
|
|
|
void ast_format_set_video_mark(struct ast_format *format);
|
|
|
|
|
|
|
|
|
|
/*!
|
|
|
|
|
* \brief Determine of the marker bit is set or not on this format.
|
|
|
|
|
*
|
|
|
|
|
* \retval 1, true
|
|
|
|
|
* \retval 0, false
|
|
|
|
|
*/
|
|
|
|
|
int ast_format_get_video_mark(const struct ast_format *format);
|
|
|
|
|
|
|
|
|
|
/*!
|
|
|
|
|
* \brief ast_format to old bitfield format represenatation
|
|
|
|
|
*
|
|
|
|
|
* \note This is only to be used for IAX2 compatibility
|
|
|
|
|
*
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* \retval iax2 representation of ast_format
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* \retval 0, if no representation existis for iax2
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*/
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uint64_t ast_format_to_old_bitfield(const struct ast_format *format);
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/*!
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* \brief ast_format_id to old bitfield format represenatation
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*
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*/
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uint64_t ast_format_id_to_old_bitfield(enum ast_format_id id);
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/*!
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* \brief convert old bitfield format to ast_format represenatation
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* \note This is only to be used for IAX2 compatibility
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*
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* \retval on success, pointer to the dst format in the input parameters
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* \retval on failure, NULL
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*/
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struct ast_format *ast_format_from_old_bitfield(struct ast_format *dst, uint64_t src);
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/*!
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* \brief convert old bitfield format to ast_format_id value
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*/
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enum ast_format_id ast_format_id_from_old_bitfield(uint64_t src);
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|
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-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
2011-02-22 23:04:49 +00:00
|
|
|
/*!
|
|
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* \brief Retrieve the global format list in a read only array.
|
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* \note ast_format_list_destroy must be called on every format
|
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* list retrieved from this function.
|
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*/
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const struct ast_format_list *ast_format_list_get(size_t *size);
|
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/*!
|
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* \brief Destroy an ast_format_list gotten from ast_format_list_get()
|
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|
*/
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const struct ast_format_list *ast_format_list_destroy(const struct ast_format_list *list);
|
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/*! \brief Get the name of a format
|
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* \param format id of format
|
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* \return A static string containing the name of the format or "unknown" if unknown.
|
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*/
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const char* ast_getformatname(const struct ast_format *format);
|
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/*! \brief Returns a string containing all formats pertaining to an format id.
|
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* \param buf a buffer for the output string
|
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|
* \param size size of buf (bytes)
|
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|
* \param format id.
|
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|
|
* \return The return value is buf.
|
|
|
|
|
*/
|
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char* ast_getformatname_multiple_byid(char *buf, size_t size, enum ast_format_id id);
|
|
|
|
|
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|
|
|
|
/*!
|
|
|
|
|
* \brief Gets a format from a name.
|
|
|
|
|
* \param name string of format
|
|
|
|
|
* \param format structure to return the format in.
|
|
|
|
|
* \return This returns the format pointer given to it on success and NULL on failure
|
|
|
|
|
*/
|
|
|
|
|
struct ast_format *ast_getformatbyname(const char *name, struct ast_format *format);
|
|
|
|
|
|
|
|
|
|
/*!
|
|
|
|
|
* \brief Get a name from a format
|
|
|
|
|
* \param format to get name of
|
|
|
|
|
* \return This returns a static string identifying the format on success, 0 on error.
|
|
|
|
|
*/
|
|
|
|
|
const char *ast_codec2str(struct ast_format *format);
|
|
|
|
|
|
|
|
|
|
/*!
|
|
|
|
|
* \brief Get the sample rate for a given format.
|
|
|
|
|
*/
|
|
|
|
|
int ast_format_rate(const struct ast_format *format);
|
|
|
|
|
|
2011-02-03 16:22:10 +00:00
|
|
|
/*!
|
|
|
|
|
* \brief register ast_format_attr_interface with core.
|
|
|
|
|
*
|
|
|
|
|
* \retval 0 success
|
|
|
|
|
* \retval -1 failure
|
|
|
|
|
*/
|
|
|
|
|
int ast_format_attr_reg_interface(const struct ast_format_attr_interface *interface);
|
|
|
|
|
|
|
|
|
|
/*!
|
|
|
|
|
* \brief unregister format_attr interface with core.
|
|
|
|
|
*
|
|
|
|
|
* \retval 0 success
|
|
|
|
|
* \retval -1 failure
|
|
|
|
|
*/
|
|
|
|
|
int ast_format_attr_unreg_interface(const struct ast_format_attr_interface *interface);
|
|
|
|
|
|
|
|
|
|
/*!
|
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-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
2011-02-22 23:04:49 +00:00
|
|
|
* \brief Determine if a format is 16bit signed linear of any sample rate.
|
2011-02-03 16:22:10 +00:00
|
|
|
*/
|
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-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
2011-02-22 23:04:49 +00:00
|
|
|
int ast_format_is_slinear(const struct ast_format *format);
|
2011-02-03 16:22:10 +00:00
|
|
|
|
Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff
-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
2011-02-22 23:04:49 +00:00
|
|
|
/*!
|
|
|
|
|
* \brief Get the best slinear format id for a given sample rate
|
|
|
|
|
*/
|
|
|
|
|
enum ast_format_id ast_format_slin_by_rate(unsigned int rate);
|
2011-02-03 16:22:10 +00:00
|
|
|
#endif /* _AST_FORMAT_H */
|