Asterisk - The Open Source Telephony Project
GIT-master-f36a736
include
asterisk
transcap.h
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/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 1999 - 2005, Digium, Inc.
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*
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* Matthew Fredrickson <creslin@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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/*! \file
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* \brief General Asterisk channel transcoding definitions.
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*/
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#ifndef _ASTERISK_TRANSCAP_H
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#define _ASTERISK_TRANSCAP_H
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/* These definitions are taken directly out of libpri.h and used here.
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* DO NOT change them as it will cause unexpected behavior in channels
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* that utilize these fields.
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*/
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/*! \name AstTranscode General Asterisk channel transcoding definitions.
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*
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* @{
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*/
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#define AST_TRANS_CAP_SPEECH 0x0
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#define AST_TRANS_CAP_DIGITAL 0x08
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#define AST_TRANS_CAP_RESTRICTED_DIGITAL 0x09
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#define AST_TRANS_CAP_3_1K_AUDIO 0x10
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#define AST_TRANS_CAP_7K_AUDIO 0x11
/* Depriciated ITU Q.931 (05/1998)*/
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#define AST_TRANS_CAP_DIGITAL_W_TONES 0x11
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#define AST_TRANS_CAP_VIDEO 0x18
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/*! @} */
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#define IS_DIGITAL(cap)\
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(cap) & AST_TRANS_CAP_DIGITAL ? 1 : 0
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#endif
/* _ASTERISK_TRANSCAP_H */
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