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* Fixed the iax.conf bandwidth option. This is the root cause of ASTERISK-24150. * Added checks in iax2_request() to ensure that there are actual formats requested for the new channel to prevent any more fracks from issues like ASTERISK-24150. This is a consequence of the iax.conf bandwidth option not working. * Fixed struct iax2_codec_pref.order member size mismatch issue when converting to and from the codec preference order list passed over the wire. In addition the values sent over the wire are now compatible with previous Asterisk versions. * Fixed several issues dealing with the struct iax2_codec_pref members. Off-by-one, array limit errors, and the order/framing members always need to be updated together. * Made iax2_request() setup the channel's native format preference order according to the user's wishes. The new media format strategy needs the order specified earler. * Fixed usage of ast_format_compatibility_bitfield2format(). The function can return NULL if the bitfield was not associated with a function. * Deleted dead code iax2_codec_pref_getsize() and iax2_codec_pref_setsize(). * Made iax2_parse_allow_disallow() and iax2_codec_pref_string() call iax2_codec_pref_to_cap() instead of inlining it. * Made IAX_CAPABILITY_MEDBANDWIDTH, IAX_CAPABILITY_LOWBANDWIDTH, and IAX_CAPABILITY_LOWFREE constants again as they were in Asterisk v1.8. * Renamed prefs to prefs_global so it won't get confused with the local pref versions. * Fixed too small buffer in handle_cli_iax2_show_peer(). * Fixed ast_cli() calls in handle_cli_iax2_show_peer() to output complete lines. * Changed struct create_addr_info.prefs to be struct iax2_codec_pref as an optimization so iax2_request() and iax2_call() do less work. * Fixed a potential deadlock in ast_iax2_new() on an off-nominal path when the pbx could not get started. * Made set_config() setup a local prefs list along side the local capability format bitfield. Once the config is loaded, then the local copies are put into the global versions. * Fix unininialized codec_buf in function_iaxpeer(). ASTERISK-24150 #close Reported by: Scott Griepentrog Review: https://reviewboard.asterisk.org/r/3890/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420364 65c4cc65-6c06-0410-ace0-fbb531ad65f3
130 lines
3.8 KiB
C
130 lines
3.8 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2014, Digium, Inc.
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*
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* Joshua Colp <jcolp@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 Media Format Bitfield Compatibility API
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*
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* \author Joshua Colp <jcolp@digium.com>
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*/
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#ifndef _AST_FORMAT_COMPATIBILITY_H_
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#define _AST_FORMAT_COMPATIBILITY_H_
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struct ast_format;
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struct ast_codec;
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/*
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* Legacy bitfields for specific formats
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*/
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/*! G.723.1 compression */
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#define AST_FORMAT_G723 (1ULL << 0)
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/*! GSM compression */
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#define AST_FORMAT_GSM (1ULL << 1)
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/*! Raw mu-law data (G.711) */
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#define AST_FORMAT_ULAW (1ULL << 2)
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/*! Raw A-law data (G.711) */
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#define AST_FORMAT_ALAW (1ULL << 3)
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/*! ADPCM (G.726, 32kbps, AAL2 codeword packing) */
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#define AST_FORMAT_G726_AAL2 (1ULL << 4)
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/*! ADPCM (IMA) */
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#define AST_FORMAT_ADPCM (1ULL << 5)
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/*! Raw 16-bit Signed Linear (8000 Hz) PCM */
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#define AST_FORMAT_SLIN (1ULL << 6)
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/*! LPC10, 180 samples/frame */
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#define AST_FORMAT_LPC10 (1ULL << 7)
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/*! G.729A audio */
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#define AST_FORMAT_G729 (1ULL << 8)
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/*! SpeeX Free Compression */
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#define AST_FORMAT_SPEEX (1ULL << 9)
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/*! iLBC Free Compression */
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#define AST_FORMAT_ILBC (1ULL << 10)
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/*! ADPCM (G.726, 32kbps, RFC3551 codeword packing) */
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#define AST_FORMAT_G726 (1ULL << 11)
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/*! G.722 */
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#define AST_FORMAT_G722 (1ULL << 12)
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/*! G.722.1 (also known as Siren7, 32kbps assumed) */
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#define AST_FORMAT_SIREN7 (1ULL << 13)
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/*! G.722.1 Annex C (also known as Siren14, 48kbps assumed) */
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#define AST_FORMAT_SIREN14 (1ULL << 14)
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/*! Raw 16-bit Signed Linear (16000 Hz) PCM */
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#define AST_FORMAT_SLIN16 (1ULL << 15)
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/*! G.719 (64 kbps assumed) */
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#define AST_FORMAT_G719 (1ULL << 32)
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/*! SpeeX Wideband (16kHz) Free Compression */
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#define AST_FORMAT_SPEEX16 (1ULL << 33)
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/*! Opus audio (8kHz, 16kHz, 24kHz, 48Khz) */
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#define AST_FORMAT_OPUS (1ULL << 34)
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/*! Raw testing-law data (G.711) */
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#define AST_FORMAT_TESTLAW (1ULL << 47)
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/*! H.261 Video */
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#define AST_FORMAT_H261 (1ULL << 18)
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/*! H.263 Video */
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#define AST_FORMAT_H263 (1ULL << 19)
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/*! H.263+ Video */
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#define AST_FORMAT_H263P (1ULL << 20)
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/*! H.264 Video */
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#define AST_FORMAT_H264 (1ULL << 21)
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/*! MPEG4 Video */
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#define AST_FORMAT_MP4 (1ULL << 22)
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/*! VP8 Video */
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#define AST_FORMAT_VP8 (1ULL << 23)
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/*! JPEG Images */
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#define AST_FORMAT_JPEG (1ULL << 16)
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/*! PNG Images */
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#define AST_FORMAT_PNG (1ULL << 17)
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/*! T.140 RED Text format RFC 4103 */
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#define AST_FORMAT_T140_RED (1ULL << 26)
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/*! T.140 Text format - ITU T.140, RFC 4103 */
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#define AST_FORMAT_T140 (1ULL << 27)
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/*!
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* \brief Convert a format structure to its respective bitfield
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*
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* \param format The media format
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*
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* \retval non-zero success
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* \retval zero format not supported
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*/
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uint64_t ast_format_compatibility_format2bitfield(const struct ast_format *format);
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/*!
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* \brief Convert a codec structure to its respective bitfield
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*
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* \param codec The media codec
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*
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* \retval non-zero success
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* \retval zero format not supported
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*/
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uint64_t ast_format_compatibility_codec2bitfield(const struct ast_codec *codec);
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/*!
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* \brief Convert a bitfield to its respective format structure
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*
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* \param bitfield The bitfield for the media format
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*
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* \retval non-NULL success
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* \retval NULL failure (The format bitfield value is not supported)
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*
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* \note The reference count of the returned format is NOT incremented
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*/
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struct ast_format *ast_format_compatibility_bitfield2format(uint64_t bitfield);
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#endif /* _AST_FORMAT_COMPATIBILITY_H */
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