1999-11-21 02:26:43 +00:00
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/*
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2005-08-30 18:32:10 +00:00
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* Asterisk -- An open source telephony toolkit.
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1999-11-21 02:26:43 +00:00
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*
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2005-01-21 07:06:25 +00:00
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* Copyright (C) 1999 - 2005, Digium, Inc.
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1999-11-21 02:26:43 +00:00
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*
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2004-12-03 23:46:09 +00:00
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* Mark Spencer <markster@digium.com>
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1999-11-21 02:26:43 +00:00
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*
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2005-08-30 18:32:10 +00:00
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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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1999-11-21 02:26:43 +00:00
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* This program is free software, distributed under the terms of
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2005-08-30 18:32:10 +00:00
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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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2005-10-24 20:12:06 +00:00
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/*! \file
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* \brief Asterisk internal frame definitions.
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2005-11-06 15:09:47 +00:00
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* \arg For an explanation of frames, see \ref Def_Frame
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* \arg Frames are send of Asterisk channels, see \ref Def_Channel
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1999-11-21 02:26:43 +00:00
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*/
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#ifndef _ASTERISK_FRAME_H
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#define _ASTERISK_FRAME_H
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#if defined(__cplusplus) || defined(c_plusplus)
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extern "C" {
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#endif
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2004-03-13 03:52:14 +00:00
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#include <sys/time.h>
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2006-09-03 23:14:54 +00:00
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2009-11-04 14:05:12 +00:00
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#include "asterisk/frame_defs.h"
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2005-04-21 06:02:45 +00:00
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#include "asterisk/endian.h"
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2006-08-29 20:50:36 +00:00
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#include "asterisk/linkedlists.h"
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2003-05-06 22:27:46 +00:00
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2004-12-02 21:57:45 +00:00
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struct ast_codec_pref {
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2009-11-04 14:05:12 +00:00
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char order[sizeof(format_t) * 8];
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char framing[sizeof(format_t) * 8];
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2004-12-02 21:57:45 +00:00
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};
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2009-04-03 22:41:46 +00:00
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/*!
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* \page Def_Frame AST Multimedia and signalling frames
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* \section Def_AstFrame What is an ast_frame ?
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* A frame of data read used to communicate between
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* between channels and applications.
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* Frames are divided into frame types and subclasses.
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*
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* \par Frame types
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* \arg \b VOICE: Voice data, subclass is codec (AST_FORMAT_*)
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* \arg \b VIDEO: Video data, subclass is codec (AST_FORMAT_*)
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* \arg \b DTMF: A DTMF digit, subclass is the digit
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* \arg \b IMAGE: Image transport, mostly used in IAX
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* \arg \b TEXT: Text messages and character by character (real time text)
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* \arg \b HTML: URL's and web pages
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* \arg \b MODEM: Modulated data encodings, such as T.38 and V.150
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* \arg \b IAX: Private frame type for the IAX protocol
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* \arg \b CNG: Comfort noice frames
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* \arg \b CONTROL:A control frame, subclass defined as AST_CONTROL_
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* \arg \b NULL: Empty, useless frame
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*
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* \par Files
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* \arg frame.h Definitions
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* \arg frame.c Function library
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* \arg \ref Def_Channel Asterisk channels
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* \section Def_ControlFrame Control Frames
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* Control frames send signalling information between channels
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* and devices. They are prefixed with AST_CONTROL_, like AST_CONTROL_FRAME_HANGUP
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* \arg \b HANGUP The other end has hungup
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* \arg \b RING Local ring
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* \arg \b RINGING The other end is ringing
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* \arg \b ANSWER The other end has answered
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* \arg \b BUSY Remote end is busy
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* \arg \b TAKEOFFHOOK Make it go off hook (what's "it" ? )
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* \arg \b OFFHOOK Line is off hook
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* \arg \b CONGESTION Congestion (circuit is busy, not available)
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* \arg \b FLASH Other end sends flash hook
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* \arg \b WINK Other end sends wink
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* \arg \b OPTION Send low-level option
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* \arg \b RADIO_KEY Key radio (see app_rpt.c)
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* \arg \b RADIO_UNKEY Un-key radio (see app_rpt.c)
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* \arg \b PROGRESS Other end indicates call progress
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* \arg \b PROCEEDING Indicates proceeding
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* \arg \b HOLD Call is placed on hold
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* \arg \b UNHOLD Call is back from hold
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* \arg \b VIDUPDATE Video update requested
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2010-03-12 22:04:51 +00:00
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* \arg \b SRCUPDATE The source of media has changed (RTP marker bit must change)
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* \arg \b SRCCHANGE Media source has changed (RTP marker bit and SSRC must change)
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2009-04-03 22:41:46 +00:00
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* \arg \b CONNECTED_LINE Connected line has changed
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* \arg \b REDIRECTING Call redirecting information has changed.
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*/
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2005-11-06 15:09:47 +00:00
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2006-08-31 01:59:02 +00:00
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/*!
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* \brief Frame types
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*
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* \note It is important that the values of each frame type are never changed,
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* because it will break backwards compatability with older versions.
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2007-01-20 18:27:35 +00:00
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* This is because these constants are transmitted directly over IAX2.
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2006-08-31 01:59:02 +00:00
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*/
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enum ast_frame_type {
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/*! DTMF end event, subclass is the digit */
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AST_FRAME_DTMF_END = 1,
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/*! Voice data, subclass is AST_FORMAT_* */
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AST_FRAME_VOICE,
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/*! Video frame, maybe?? :) */
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AST_FRAME_VIDEO,
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/*! A control frame, subclass is AST_CONTROL_* */
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AST_FRAME_CONTROL,
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/*! An empty, useless frame */
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AST_FRAME_NULL,
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/*! Inter Asterisk Exchange private frame type */
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AST_FRAME_IAX,
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/*! Text messages */
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AST_FRAME_TEXT,
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/*! Image Frames */
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AST_FRAME_IMAGE,
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/*! HTML Frame */
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AST_FRAME_HTML,
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/*! Comfort Noise frame (subclass is level of CNG in -dBov),
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body may include zero or more 8-bit quantization coefficients */
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AST_FRAME_CNG,
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/*! Modem-over-IP data streams */
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AST_FRAME_MODEM,
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/*! DTMF begin event, subclass is the digit */
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AST_FRAME_DTMF_BEGIN,
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};
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#define AST_FRAME_DTMF AST_FRAME_DTMF_END
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Merged revisions 98943 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r98943 | russell | 2008-01-15 17:26:52 -0600 (Tue, 15 Jan 2008) | 25 lines
Commit a fix for some memory access errors pointed out by the valgrind2.txt
output on issue #11698.
The issue here is that it is possible for an instance of a translator to get
destroyed while the frame allocated as a part of the translator is still being
processed. Specifically, this is possible anywhere between a call to ast_read()
and ast_frame_free(), which is _a lot_ of places in the code. The reason this
happens is that the channel might get masqueraded during this time. During a
masquerade, existing translation paths get destroyed.
So, this patch fixes the issue in an API and ABI compatible way. (This one is
for you, paravoid!)
It changes an int in ast_frame to be used as flag bits. The 1 bit is still used
to indicate that the frame contains timing information. Also, a second flag has
been added to indicate that the frame came from a translator. When a frame with
this flag gets released and has this flag, a function is called in translate.c to
let it know that this frame is doing being processed. At this point, the flag gets
cleared. Also, if the translator was requested to be destroyed while its internal
frame still had this flag set, its destruction has been deffered until it finds out
that the frame is no longer being processed.
Admittedly, this feels like a hack. But, it does fix the issue, and I was not able
to think of a better solution ...
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98944 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-15 23:31:53 +00:00
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enum {
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/*! This frame contains valid timing information */
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AST_FRFLAG_HAS_TIMING_INFO = (1 << 0),
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};
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2010-03-02 23:11:06 +00:00
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union ast_frame_subclass {
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int integer;
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format_t codec;
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};
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2005-11-06 15:09:47 +00:00
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/*! \brief Data structure associated with a single frame of data
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*/
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1999-11-21 02:26:43 +00:00
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struct ast_frame {
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2001-11-10 20:31:39 +00:00
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/*! Kind of frame */
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2006-08-31 01:59:02 +00:00
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enum ast_frame_type frametype;
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2001-11-10 20:31:39 +00:00
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/*! Subclass, frame dependent */
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2010-03-02 23:11:06 +00:00
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union ast_frame_subclass subclass;
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2001-11-10 20:31:39 +00:00
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/*! Length of data */
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int datalen;
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2009-07-08 23:19:09 +00:00
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/*! Number of samples in this frame */
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2003-02-05 19:26:49 +00:00
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int samples;
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2001-11-10 20:31:39 +00:00
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/*! Was the data malloc'd? i.e. should we free it when we discard the frame? */
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int mallocd;
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2006-08-29 20:50:36 +00:00
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/*! The number of bytes allocated for a malloc'd frame header */
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size_t mallocd_hdr_len;
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2005-10-28 16:20:44 +00:00
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/*! How many bytes exist _before_ "data" that can be used if needed */
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2001-11-10 20:31:39 +00:00
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int offset;
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/*! Optional source of frame for debugging */
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2005-03-21 04:30:57 +00:00
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const char *src;
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2001-11-10 20:31:39 +00:00
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/*! Pointer to actual data */
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2008-05-22 16:29:54 +00:00
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union { void *ptr; uint32_t uint32; char pad[8]; } data;
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2004-03-13 03:52:14 +00:00
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/*! Global delivery time */
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struct timeval delivery;
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2006-08-29 20:50:36 +00:00
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/*! For placing in a linked list */
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AST_LIST_ENTRY(ast_frame) frame_list;
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Merged revisions 98943 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r98943 | russell | 2008-01-15 17:26:52 -0600 (Tue, 15 Jan 2008) | 25 lines
Commit a fix for some memory access errors pointed out by the valgrind2.txt
output on issue #11698.
The issue here is that it is possible for an instance of a translator to get
destroyed while the frame allocated as a part of the translator is still being
processed. Specifically, this is possible anywhere between a call to ast_read()
and ast_frame_free(), which is _a lot_ of places in the code. The reason this
happens is that the channel might get masqueraded during this time. During a
masquerade, existing translation paths get destroyed.
So, this patch fixes the issue in an API and ABI compatible way. (This one is
for you, paravoid!)
It changes an int in ast_frame to be used as flag bits. The 1 bit is still used
to indicate that the frame contains timing information. Also, a second flag has
been added to indicate that the frame came from a translator. When a frame with
this flag gets released and has this flag, a function is called in translate.c to
let it know that this frame is doing being processed. At this point, the flag gets
cleared. Also, if the translator was requested to be destroyed while its internal
frame still had this flag set, its destruction has been deffered until it finds out
that the frame is no longer being processed.
Admittedly, this feels like a hack. But, it does fix the issue, and I was not able
to think of a better solution ...
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98944 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-15 23:31:53 +00:00
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/*! Misc. frame flags */
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unsigned int flags;
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2006-05-31 16:56:50 +00:00
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/*! Timestamp in milliseconds */
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long ts;
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/*! Length in milliseconds */
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long len;
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/*! Sequence number */
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int seqno;
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1999-11-21 02:26:43 +00:00
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};
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2006-04-09 22:31:38 +00:00
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/*!
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* Set the various field of a frame to point to a buffer.
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* Typically you set the base address of the buffer, the offset as
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* AST_FRIENDLY_OFFSET, and the datalen as the amount of bytes queued.
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* The remaining things (to be done manually) is set the number of
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* samples, which cannot be derived from the datalen unless you know
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* the number of bits per sample.
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*/
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#define AST_FRAME_SET_BUFFER(fr, _base, _ofs, _datalen) \
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2006-03-29 02:14:38 +00:00
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{ \
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2008-05-22 16:29:54 +00:00
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(fr)->data.ptr = (char *)_base + (_ofs); \
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2006-03-29 02:14:38 +00:00
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(fr)->offset = (_ofs); \
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(fr)->datalen = (_datalen); \
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}
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2006-01-31 17:18:58 +00:00
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/*! Queueing a null frame is fairly common, so we declare a global null frame object
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for this purpose instead of having to declare one on the stack */
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extern struct ast_frame ast_null_frame;
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2009-02-17 21:40:09 +00:00
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/*! \brief Offset into a frame's data buffer.
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*
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* By providing some "empty" space prior to the actual data of an ast_frame,
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* this gives any consumer of the frame ample space to prepend other necessary
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* information without having to create a new buffer.
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*
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* As an example, RTP can use the data from an ast_frame and simply prepend the
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* RTP header information into the space provided by AST_FRIENDLY_OFFSET instead
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* of having to create a new buffer with the necessary space allocated.
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*/
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#define AST_FRIENDLY_OFFSET 64
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2005-11-06 15:09:47 +00:00
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#define AST_MIN_OFFSET 32 /*! Make sure we keep at least this much handy */
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2004-02-16 15:05:21 +00:00
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2001-11-10 20:31:39 +00:00
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/*! Need the header be free'd? */
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#define AST_MALLOCD_HDR (1 << 0)
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/*! Need the data be free'd? */
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#define AST_MALLOCD_DATA (1 << 1)
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/*! Need the source be free'd? (haha!) */
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#define AST_MALLOCD_SRC (1 << 2)
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1999-11-21 02:26:43 +00:00
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2006-01-12 17:14:58 +00:00
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/* MODEM subclasses */
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/*! T.38 Fax-over-IP */
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#define AST_MODEM_T38 1
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/*! V.150 Modem-over-IP */
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#define AST_MODEM_V150 2
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2001-10-12 17:15:10 +00:00
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/* HTML subclasses */
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2001-11-10 20:31:39 +00:00
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/*! Sending a URL */
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#define AST_HTML_URL 1
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/*! Data frame */
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#define AST_HTML_DATA 2
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/*! Beginning frame */
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#define AST_HTML_BEGIN 4
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/*! End frame */
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#define AST_HTML_END 8
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/*! Load is complete */
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#define AST_HTML_LDCOMPLETE 16
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/*! Peer is unable to support HTML */
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#define AST_HTML_NOSUPPORT 17
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/*! Send URL, and track */
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#define AST_HTML_LINKURL 18
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/*! No more HTML linkage */
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#define AST_HTML_UNLINK 19
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/*! Reject link request */
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#define AST_HTML_LINKREJECT 20
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1999-11-21 02:26:43 +00:00
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/* Data formats for capabilities and frames alike */
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2001-11-10 20:31:39 +00:00
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/*! G.723.1 compression */
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2009-11-04 14:05:12 +00:00
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#define AST_FORMAT_G723_1 (1ULL << 0)
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2001-11-10 20:31:39 +00:00
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/*! GSM compression */
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2009-11-04 14:05:12 +00:00
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#define AST_FORMAT_GSM (1ULL << 1)
|
2001-11-10 20:31:39 +00:00
|
|
|
/*! Raw mu-law data (G.711) */
|
2009-11-04 14:05:12 +00:00
|
|
|
#define AST_FORMAT_ULAW (1ULL << 2)
|
2001-11-10 20:31:39 +00:00
|
|
|
/*! Raw A-law data (G.711) */
|
2009-11-04 14:05:12 +00:00
|
|
|
#define AST_FORMAT_ALAW (1ULL << 3)
|
2006-07-13 20:39:34 +00:00
|
|
|
/*! ADPCM (G.726, 32kbps, AAL2 codeword packing) */
|
2009-11-04 14:05:12 +00:00
|
|
|
#define AST_FORMAT_G726_AAL2 (1ULL << 4)
|
2003-11-04 02:40:09 +00:00
|
|
|
/*! ADPCM (IMA) */
|
2009-11-04 14:05:12 +00:00
|
|
|
#define AST_FORMAT_ADPCM (1ULL << 5)
|
2001-11-10 20:31:39 +00:00
|
|
|
/*! Raw 16-bit Signed Linear (8000 Hz) PCM */
|
2009-11-04 14:05:12 +00:00
|
|
|
#define AST_FORMAT_SLINEAR (1ULL << 6)
|
2001-11-10 20:31:39 +00:00
|
|
|
/*! LPC10, 180 samples/frame */
|
2009-11-04 14:05:12 +00:00
|
|
|
#define AST_FORMAT_LPC10 (1ULL << 7)
|
2002-03-07 16:49:49 +00:00
|
|
|
/*! G.729A audio */
|
2009-11-04 14:05:12 +00:00
|
|
|
#define AST_FORMAT_G729A (1ULL << 8)
|
2002-07-26 15:44:05 +00:00
|
|
|
/*! SpeeX Free Compression */
|
2009-11-04 14:05:12 +00:00
|
|
|
#define AST_FORMAT_SPEEX (1ULL << 9)
|
2003-04-15 04:36:52 +00:00
|
|
|
/*! iLBC Free Compression */
|
2009-11-04 14:05:12 +00:00
|
|
|
#define AST_FORMAT_ILBC (1ULL << 10)
|
2006-07-13 20:39:34 +00:00
|
|
|
/*! ADPCM (G.726, 32kbps, RFC3551 codeword packing) */
|
2009-11-04 14:05:12 +00:00
|
|
|
#define AST_FORMAT_G726 (1ULL << 11)
|
2006-10-25 00:32:23 +00:00
|
|
|
/*! G.722 */
|
2009-11-04 14:05:12 +00:00
|
|
|
#define AST_FORMAT_G722 (1ULL << 12)
|
Add basic (passthrough, playback, record) support for ITU G.722.1 and G.722.1C (also known as Siren7 and Siren14)
This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported.
Along the way, some related work was done:
1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way.
2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec.
3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result).
Review: http://reviewboard.digium.com/r/158/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-13 13:35:24 +00:00
|
|
|
/*! G.722.1 (also known as Siren7, 32kbps assumed) */
|
2009-11-04 14:05:12 +00:00
|
|
|
#define AST_FORMAT_SIREN7 (1ULL << 13)
|
Add basic (passthrough, playback, record) support for ITU G.722.1 and G.722.1C (also known as Siren7 and Siren14)
This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported.
Along the way, some related work was done:
1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way.
2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec.
3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result).
Review: http://reviewboard.digium.com/r/158/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-13 13:35:24 +00:00
|
|
|
/*! G.722.1 Annex C (also known as Siren14, 48kbps assumed) */
|
2009-11-04 14:05:12 +00:00
|
|
|
#define AST_FORMAT_SIREN14 (1ULL << 14)
|
2007-11-06 22:51:48 +00:00
|
|
|
/*! Raw 16-bit Signed Linear (16000 Hz) PCM */
|
2009-11-04 14:05:12 +00:00
|
|
|
#define AST_FORMAT_SLINEAR16 (1ULL << 15)
|
2006-01-07 17:54:22 +00:00
|
|
|
/*! Maximum audio mask */
|
2009-11-04 14:05:12 +00:00
|
|
|
#define AST_FORMAT_AUDIO_MASK 0xFFFF0000FFFFULL
|
2001-11-10 20:31:39 +00:00
|
|
|
/*! JPEG Images */
|
2009-11-04 14:05:12 +00:00
|
|
|
#define AST_FORMAT_JPEG (1ULL << 16)
|
2001-11-10 20:31:39 +00:00
|
|
|
/*! PNG Images */
|
2009-11-04 14:05:12 +00:00
|
|
|
#define AST_FORMAT_PNG (1ULL << 17)
|
2001-11-10 20:31:39 +00:00
|
|
|
/*! H.261 Video */
|
2009-11-04 14:05:12 +00:00
|
|
|
#define AST_FORMAT_H261 (1ULL << 18)
|
2001-11-10 20:31:39 +00:00
|
|
|
/*! H.263 Video */
|
2009-11-04 14:05:12 +00:00
|
|
|
#define AST_FORMAT_H263 (1ULL << 19)
|
2005-03-05 20:57:49 +00:00
|
|
|
/*! H.263+ Video */
|
2009-11-04 14:05:12 +00:00
|
|
|
#define AST_FORMAT_H263_PLUS (1ULL << 20)
|
2006-01-07 17:54:22 +00:00
|
|
|
/*! H.264 Video */
|
2009-11-04 14:05:12 +00:00
|
|
|
#define AST_FORMAT_H264 (1ULL << 21)
|
2007-01-08 11:49:23 +00:00
|
|
|
/*! MPEG4 Video */
|
2009-11-04 14:05:12 +00:00
|
|
|
#define AST_FORMAT_MP4_VIDEO (1ULL << 22)
|
|
|
|
|
#define AST_FORMAT_VIDEO_MASK ((((1ULL << 25)-1) & ~(AST_FORMAT_AUDIO_MASK)) | 0x7FFF000000000000ULL)
|
2008-05-14 13:37:07 +00:00
|
|
|
/*! T.140 RED Text format RFC 4103 */
|
2009-11-04 14:05:12 +00:00
|
|
|
#define AST_FORMAT_T140RED (1ULL << 26)
|
2008-05-30 11:26:05 +00:00
|
|
|
/*! T.140 Text format - ITU T.140, RFC 4103 */
|
2009-11-04 14:05:12 +00:00
|
|
|
#define AST_FORMAT_T140 (1ULL << 27)
|
2008-05-14 13:37:07 +00:00
|
|
|
/*! Maximum text mask */
|
2009-11-04 14:05:12 +00:00
|
|
|
#define AST_FORMAT_MAX_TEXT (1ULL << 28)
|
|
|
|
|
#define AST_FORMAT_TEXT_MASK (((1ULL << 30)-1) & ~(AST_FORMAT_AUDIO_MASK) & ~(AST_FORMAT_VIDEO_MASK))
|
2010-06-16 19:03:24 +00:00
|
|
|
/*! G.719 (64 kbps assumed) */
|
|
|
|
|
#define AST_FORMAT_G719 (1ULL << 32)
|
2010-06-17 17:23:43 +00:00
|
|
|
/*! SpeeX Wideband (16kHz) Free Compression */
|
|
|
|
|
#define AST_FORMAT_SPEEX16 (1ULL << 33)
|
2009-11-04 14:05:12 +00:00
|
|
|
/*! Raw mu-law data (G.711) */
|
2010-06-17 17:23:43 +00:00
|
|
|
#define AST_FORMAT_TESTLAW (1ULL << 47)
|
2009-11-04 14:05:12 +00:00
|
|
|
/*! Reserved bit - do not use */
|
|
|
|
|
#define AST_FORMAT_RESERVED (1ULL << 63)
|
1999-11-21 02:26:43 +00:00
|
|
|
|
2006-05-09 14:25:57 +00:00
|
|
|
enum ast_control_frame_type {
|
2011-04-20 16:54:02 +00:00
|
|
|
AST_CONTROL_HANGUP = 1, /*!< Other end has hungup */
|
|
|
|
|
AST_CONTROL_RING = 2, /*!< Local ring */
|
|
|
|
|
AST_CONTROL_RINGING = 3, /*!< Remote end is ringing */
|
|
|
|
|
AST_CONTROL_ANSWER = 4, /*!< Remote end has answered */
|
|
|
|
|
AST_CONTROL_BUSY = 5, /*!< Remote end is busy */
|
2006-05-09 16:24:07 +00:00
|
|
|
AST_CONTROL_TAKEOFFHOOK = 6, /*!< Make it go off hook */
|
2011-04-20 16:54:02 +00:00
|
|
|
AST_CONTROL_OFFHOOK = 7, /*!< Line is off hook */
|
|
|
|
|
AST_CONTROL_CONGESTION = 8, /*!< Congestion (circuits busy) */
|
|
|
|
|
AST_CONTROL_FLASH = 9, /*!< Flash hook */
|
|
|
|
|
AST_CONTROL_WINK = 10, /*!< Wink */
|
|
|
|
|
AST_CONTROL_OPTION = 11, /*!< Set a low-level option */
|
|
|
|
|
AST_CONTROL_RADIO_KEY = 12, /*!< Key Radio */
|
2006-05-09 16:24:07 +00:00
|
|
|
AST_CONTROL_RADIO_UNKEY = 13, /*!< Un-Key Radio */
|
2011-04-20 16:54:02 +00:00
|
|
|
AST_CONTROL_PROGRESS = 14, /*!< Indicate PROGRESS */
|
2006-05-09 16:24:07 +00:00
|
|
|
AST_CONTROL_PROCEEDING = 15, /*!< Indicate CALL PROCEEDING */
|
2011-04-20 16:54:02 +00:00
|
|
|
AST_CONTROL_HOLD = 16, /*!< Indicate call is placed on hold */
|
|
|
|
|
AST_CONTROL_UNHOLD = 17, /*!< Indicate call is left from hold */
|
|
|
|
|
AST_CONTROL_VIDUPDATE = 18, /*!< Indicate video frame update */
|
|
|
|
|
_XXX_AST_CONTROL_T38 = 19, /*!< T38 state change request/notification \deprecated This is no longer supported. Use AST_CONTROL_T38_PARAMETERS instead. */
|
|
|
|
|
AST_CONTROL_SRCUPDATE = 20, /*!< Indicate source of media has changed */
|
|
|
|
|
AST_CONTROL_TRANSFER = 21, /*!< Indicate status of a transfer request */
|
2009-06-01 20:17:50 +00:00
|
|
|
AST_CONTROL_CONNECTED_LINE = 22,/*!< Indicate connected line has changed */
|
2011-04-20 16:54:02 +00:00
|
|
|
AST_CONTROL_REDIRECTING = 23, /*!< Indicate redirecting id has changed */
|
|
|
|
|
AST_CONTROL_T38_PARAMETERS = 24,/*!< T38 state change request/notification with parameters */
|
|
|
|
|
AST_CONTROL_CC = 25, /*!< Indication that Call completion service is possible */
|
|
|
|
|
AST_CONTROL_SRCCHANGE = 26, /*!< Media source has changed and requires a new RTP SSRC */
|
|
|
|
|
AST_CONTROL_READ_ACTION = 27, /*!< Tell ast_read to take a specific action */
|
|
|
|
|
AST_CONTROL_AOC = 28, /*!< Advice of Charge with encoded generic AOC payload */
|
2010-11-22 19:36:10 +00:00
|
|
|
AST_CONTROL_END_OF_Q = 29, /*!< Indicate that this position was the end of the channel queue for a softhangup. */
|
Enhancements to connected line and redirecting work.
From reviewboard:
Digium has a commercial customer who has made extensive use of the connected party and
redirecting information present in later versions of Asterisk Business Edition and which
is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions
have come about. This patch adds several enhancements to maximize usage of the connected party
and redirecting information functionality.
First, Asterisk trunk already had connected line interception macros. These macros allow you to
manipulate connected line information before it was sent out to its target. This patch adds the
same feature except for redirecting information instead.
Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This
tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI,
mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is
that it can be set to whatever value the administrator likes. Later, when running connected line
and redirecting macros, the admin can read the tag off the appropriate structure to determine what
action to take. You can think of this sort of like a channel variable, except that instead of having
the variable associated with a channel, the variable is associated with a specific identity within
Asterisk.
Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific
caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force
a specific calling presentation value on the outgoing channel.
Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added
to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party
being transferred would not have the opportunity to run a connected line interception macro to
possibly alter the transfer target's connected line information. The issue here was that during a
blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line
update. The way this was corrected was to add this new control frame subclass. Now, we queue an
AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should
be run. When ast_read is called to read the frame, ast_read responds by calling a callback function
associated with the specific read action the control frame describes. In this case, the action taken
is to run the connected line interception macro on the transferee's channel.
Review: https://reviewboard.asterisk.org/r/652/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-17 15:36:31 +00:00
|
|
|
};
|
|
|
|
|
|
|
|
|
|
enum ast_frame_read_action {
|
|
|
|
|
AST_FRAME_READ_ACTION_CONNECTED_LINE_MACRO,
|
|
|
|
|
};
|
|
|
|
|
|
|
|
|
|
struct ast_control_read_action_payload {
|
|
|
|
|
/* An indicator to ast_read of what action to
|
|
|
|
|
* take with the frame;
|
|
|
|
|
*/
|
|
|
|
|
enum ast_frame_read_action action;
|
|
|
|
|
/* The size of the frame's payload
|
|
|
|
|
*/
|
|
|
|
|
size_t payload_size;
|
|
|
|
|
/* A payload for the frame.
|
|
|
|
|
*/
|
|
|
|
|
unsigned char payload[0];
|
2008-02-18 23:47:01 +00:00
|
|
|
};
|
|
|
|
|
|
|
|
|
|
enum ast_control_t38 {
|
|
|
|
|
AST_T38_REQUEST_NEGOTIATE = 1, /*!< Request T38 on a channel (voice to fax) */
|
|
|
|
|
AST_T38_REQUEST_TERMINATE, /*!< Terminate T38 on a channel (fax to voice) */
|
|
|
|
|
AST_T38_NEGOTIATED, /*!< T38 negotiated (fax mode) */
|
|
|
|
|
AST_T38_TERMINATED, /*!< T38 terminated (back to voice) */
|
Improve handling of T.38 re-INVITEs that arrive before a T.38-capable
application is executing on a channel.
This patch addresses an issue found during working with end-users
using res_fax. If an incoming call is answered in the dialplan, or
jumps to the 'fax' extension due to reception of a CNG tone (with
faxdetect enabled), and then the remote endpoint sends a T.38
re-INVITE, it is possible for the channel's T.38 state to be
'T38_STATE_NEGOTIATING' when the application starts up. Unfortunately,
even if the application wants to use T.38, it can't respond to the
peer's negotiation request, because the AST_CONTROL_T38_PARAMETERS
control frame that chan_sip sent originally has been lost, and the
application needs the content of that frame to be able to formulate a
reply.
This patch adds a new 'request' type to AST_CONTROL_T38_PARAMETERS,
AST_T38_REQUEST_PARMS. If the application sends this request, chan_sip
will re-send the original control frame (with
AST_T38_REQUEST_NEGOTIATE as the request type), and the application
can respond as normal. If this occurs within the five second timeout
in chan_sip, the automatic cancellation of the peer reinvite will be
stopped, and the application will 'own' the negotiation process from
that point onwards.
This also improves the code path in chan_sip to allow sip_indicate(),
when called for AST_CONTROL_T38_PARAMETERS, to be able to return a
non-zero response, which should have been in place before since the
control frame *can* fail to be processed properly. It also modifies
ast_indicate() to return whatever result the channel driver returned
for this control frame, rather than converting all non-zero results
into '-1'. Finally, the new request type intentionally returns a
positive value, so that an application that sends
AST_T38_REQUEST_PARMS can know for certain whether the channel driver
accepted it and will be replying with a control frame of its own, or
whether it was ignored (if the sip_indicate()/ast_indicate() path had
properly supported failure responses before, this would not be
necessary).
This patch also modifies res_fax to take advantage of the new request.
In addition, this patch makes sip_t38_abort() actually lock the
private structure before doing its work... bad programmer, no donut.
This patch also enhances chan_sip's 'faxdetect' support to allow
triggering on T.38 re-INVITEs received as well as CNG tone detection.
Review: https://reviewboard.asterisk.org/r/556/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254450 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-25 15:27:31 +00:00
|
|
|
AST_T38_REFUSED, /*!< T38 refused for some reason (usually rejected by remote end) */
|
|
|
|
|
AST_T38_REQUEST_PARMS, /*!< request far end T.38 parameters for a channel in 'negotiating' state */
|
2006-05-09 14:25:57 +00:00
|
|
|
};
|
2001-05-07 02:01:01 +00:00
|
|
|
|
2009-06-26 19:27:24 +00:00
|
|
|
enum ast_control_t38_rate {
|
|
|
|
|
AST_T38_RATE_2400 = 0,
|
|
|
|
|
AST_T38_RATE_4800,
|
|
|
|
|
AST_T38_RATE_7200,
|
|
|
|
|
AST_T38_RATE_9600,
|
|
|
|
|
AST_T38_RATE_12000,
|
|
|
|
|
AST_T38_RATE_14400,
|
|
|
|
|
};
|
|
|
|
|
|
|
|
|
|
enum ast_control_t38_rate_management {
|
2009-07-09 21:20:23 +00:00
|
|
|
AST_T38_RATE_MANAGEMENT_TRANSFERRED_TCF = 0,
|
2009-06-26 19:27:24 +00:00
|
|
|
AST_T38_RATE_MANAGEMENT_LOCAL_TCF,
|
|
|
|
|
};
|
|
|
|
|
|
|
|
|
|
struct ast_control_t38_parameters {
|
Rework of T.38 negotiation and UDPTL API to address interoperability problems
Over the past couple of months, a number of issues with Asterisk
negotiating (and successfully completing) T.38 sessions with various
endpoints have been found. This patch attempts to address many of
them, primarily focused around ensuring that the endpoints'
MaxDatagram size is honored, and in addition by ensuring that T.38
session parameter negotiation is performed correctly according to the
ITU T.38 Recommendation.
The major changes here are:
1) T.38 applications in Asterisk (app_fax) only generate/receive IFP
packets, they do not ever work with UDPTL packets. As a result of
this, they cannot be allowed to generate packets that would overflow
the other endpoints' MaxDatagram size after the UDPTL stack adds any
error correction information. With this patch, the application is told
the maximum *IFP* size it can generate, based on a calculation using
the far end MaxDatagram size and the active error correction mode on
the T.38 session. The same is true for sending *our* MaxDatagram size
to the remote endpoint; it is computed from the value that the
application says it can accept (for a single IFP packet) combined with
the active error correction mode.
2) All treatment of T.38 session parameters as 'capabilities' in
chan_sip has been removed; these parameters are not at all like
audio/video stream capabilities. There are strict rules to follow for
computing an answer to a T.38 offer, and chan_sip now follows those
rules, using the desired parameters from the application (or channel)
that wants to accept the T.38 negotiation.
3) chan_sip now stores and forwards ast_control_t38_parameters
structures for tracking 'our' and 'their' T.38 session parameters;
this greatly simplifies negotiation, especially for pass-through
calls.
4) Since T.38 negotiation without specifying parameters or receiving
the final negotiated parameters is not very worthwhile, the
AST_CONTROL_T38 control frame has been removed. A note has been added
to UPGRADE.txt about this removal, since any out-of-tree applications
that use it will no longer function properly until they are upgraded
to use AST_CONTROL_T38_PARAMETERS.
Review: https://reviewboard.asterisk.org/r/310/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@208464 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-23 21:57:24 +00:00
|
|
|
enum ast_control_t38 request_response; /*!< Request or response of the T38 control frame */
|
|
|
|
|
unsigned int version; /*!< Supported T.38 version */
|
|
|
|
|
unsigned int max_ifp; /*!< Maximum IFP size supported */
|
|
|
|
|
enum ast_control_t38_rate rate; /*!< Maximum fax rate supported */
|
|
|
|
|
enum ast_control_t38_rate_management rate_management; /*!< Rate management setting */
|
|
|
|
|
unsigned int fill_bit_removal:1; /*!< Set if fill bit removal can be used */
|
|
|
|
|
unsigned int transcoding_mmr:1; /*!< Set if MMR transcoding can be used */
|
|
|
|
|
unsigned int transcoding_jbig:1; /*!< Set if JBIG transcoding can be used */
|
2009-06-26 19:27:24 +00:00
|
|
|
};
|
|
|
|
|
|
2009-04-03 16:47:27 +00:00
|
|
|
enum ast_control_transfer {
|
|
|
|
|
AST_TRANSFER_SUCCESS = 0, /*!< Transfer request on the channel worked */
|
|
|
|
|
AST_TRANSFER_FAILED, /*!< Transfer request on the channel failed */
|
|
|
|
|
};
|
|
|
|
|
|
2004-03-14 18:01:29 +00:00
|
|
|
#define AST_SMOOTHER_FLAG_G729 (1 << 0)
|
2006-09-18 23:32:57 +00:00
|
|
|
#define AST_SMOOTHER_FLAG_BE (1 << 1)
|
2004-03-14 18:01:29 +00:00
|
|
|
|
2001-05-07 02:01:01 +00:00
|
|
|
/* Option identifiers and flags */
|
|
|
|
|
#define AST_OPTION_FLAG_REQUEST 0
|
|
|
|
|
#define AST_OPTION_FLAG_ACCEPT 1
|
|
|
|
|
#define AST_OPTION_FLAG_REJECT 2
|
|
|
|
|
#define AST_OPTION_FLAG_QUERY 4
|
|
|
|
|
#define AST_OPTION_FLAG_ANSWER 5
|
2001-11-10 20:31:39 +00:00
|
|
|
#define AST_OPTION_FLAG_WTF 6
|
|
|
|
|
|
2005-11-06 15:09:47 +00:00
|
|
|
/*! Verify touchtones by muting audio transmission
|
2001-11-10 20:31:39 +00:00
|
|
|
(and reception) and verify the tone is still present */
|
|
|
|
|
#define AST_OPTION_TONE_VERIFY 1
|
2001-05-07 02:01:01 +00:00
|
|
|
|
2005-11-06 15:09:47 +00:00
|
|
|
/*! Put a compatible channel into TDD (TTY for the hearing-impared) mode */
|
2001-11-10 20:31:39 +00:00
|
|
|
#define AST_OPTION_TDD 2
|
2001-05-07 02:01:01 +00:00
|
|
|
|
2005-11-06 15:09:47 +00:00
|
|
|
/*! Relax the parameters for DTMF reception (mainly for radio use) */
|
2003-02-05 19:26:49 +00:00
|
|
|
#define AST_OPTION_RELAXDTMF 3
|
|
|
|
|
|
2005-11-06 15:09:47 +00:00
|
|
|
/*! Set (or clear) Audio (Not-Clear) Mode */
|
2003-03-23 17:14:29 +00:00
|
|
|
#define AST_OPTION_AUDIO_MODE 4
|
|
|
|
|
|
2005-11-06 15:09:47 +00:00
|
|
|
/*! Set channel transmit gain
|
|
|
|
|
* Option data is a single signed char
|
2005-09-07 01:30:01 +00:00
|
|
|
representing number of decibels (dB)
|
|
|
|
|
to set gain to (on top of any gain
|
|
|
|
|
specified in channel driver)
|
|
|
|
|
*/
|
|
|
|
|
#define AST_OPTION_TXGAIN 5
|
|
|
|
|
|
2005-11-06 15:09:47 +00:00
|
|
|
/*! Set channel receive gain
|
|
|
|
|
* Option data is a single signed char
|
2005-09-07 01:30:01 +00:00
|
|
|
representing number of decibels (dB)
|
|
|
|
|
to set gain to (on top of any gain
|
|
|
|
|
specified in channel driver)
|
|
|
|
|
*/
|
|
|
|
|
#define AST_OPTION_RXGAIN 6
|
|
|
|
|
|
2006-04-22 11:30:06 +00:00
|
|
|
/* set channel into "Operator Services" mode */
|
|
|
|
|
#define AST_OPTION_OPRMODE 7
|
|
|
|
|
|
2006-05-16 21:43:20 +00:00
|
|
|
/*! Explicitly enable or disable echo cancelation for the given channel */
|
|
|
|
|
#define AST_OPTION_ECHOCAN 8
|
|
|
|
|
|
2010-09-10 22:04:53 +00:00
|
|
|
/*! \brief Handle channel write data
|
|
|
|
|
* If a channel needs to process the data from a func_channel write operation
|
|
|
|
|
* after func_channel_write executes, it can define the setoption callback
|
|
|
|
|
* and process this option. A pointer to an ast_chan_write_info_t will be passed.
|
|
|
|
|
* */
|
|
|
|
|
#define AST_OPTION_CHANNEL_WRITE 9
|
|
|
|
|
|
2008-02-18 23:47:01 +00:00
|
|
|
/* !
|
|
|
|
|
* Read-only. Allows query current status of T38 on the channel.
|
|
|
|
|
* data: ast_t38state
|
|
|
|
|
*/
|
|
|
|
|
#define AST_OPTION_T38_STATE 10
|
|
|
|
|
|
2009-04-09 16:19:35 +00:00
|
|
|
/*! Request that the channel driver deliver frames in a specific format */
|
|
|
|
|
#define AST_OPTION_FORMAT_READ 11
|
|
|
|
|
|
|
|
|
|
/*! Request that the channel driver be prepared to accept frames in a specific format */
|
|
|
|
|
#define AST_OPTION_FORMAT_WRITE 12
|
|
|
|
|
|
|
|
|
|
/*! Request that the channel driver make two channels of the same tech type compatible if possible */
|
|
|
|
|
#define AST_OPTION_MAKE_COMPATIBLE 13
|
|
|
|
|
|
2009-06-16 21:10:15 +00:00
|
|
|
/*! Get or set the digit detection state of the channel */
|
|
|
|
|
#define AST_OPTION_DIGIT_DETECT 14
|
|
|
|
|
|
|
|
|
|
/*! Get or set the fax tone detection state of the channel */
|
|
|
|
|
#define AST_OPTION_FAX_DETECT 15
|
|
|
|
|
|
Merge Call completion support into trunk.
From Reviewboard:
CCSS stands for Call Completion Supplementary Services. An admittedly out-of-date
overview of the architecture can be found in the file doc/CCSS_architecture.pdf
in the CCSS branch. Off the top of my head, the big differences between what is
implemented and what is in the document are as follows:
1. We did not end up modifying the Hangup application at all.
2. The document states that a single call completion monitor may be used across
multiple calls to the same device. This proved to not be such a good idea
when implementing protocol-specific monitors, and so we ended up using one
monitor per-device per-call.
3. There are some configuration options which were conceived after the document
was written. These are documented in the ccss.conf.sample that is on this
review request.
For some basic understanding of terminology used throughout this code, see the
ccss.tex document that is on this review.
This implements CCBS and CCNR in several flavors.
First up is a "generic" implementation, which can work over any channel technology
provided that the channel technology can accurately report device state. Call
completion is requested using the dialplan application CallCompletionRequest and can
be canceled using CallCompletionCancel. Device state subscriptions are used in order
to monitor the state of called parties.
Next, there is a SIP-specific implementation of call completion. This method uses the
methods outlined in draft-ietf-bliss-call-completion-06 to implement call completion
using SIP signaling. There are a few things to note here:
* The agent/monitor terminology used throughout Asterisk sometimes is the reverse of
what is defined in the referenced draft.
* Implementation of the draft required support for SIP PUBLISH. I attempted to write
this in a generic-enough fashion such that if someone were to want to write PUBLISH
support for other event packages, such as dialog-state or presence, most of the effort
would be in writing callbacks specific to the event package.
* A subportion of supporting PUBLISH reception was that we had to implement a PIDF
parser. The PIDF support added is a bit minimal. I first wrote a validation
routine to ensure that the PIDF document is formatted properly. The rest of the
PIDF reading is done in-line in the call-completion-specific PUBLISH-handling
code. In other words, while there is PIDF support here, it is not in any state
where it could easily be applied to other event packages as is.
Finally, there are a variety of ISDN-related call completion protocols supported. These
were written by Richard Mudgett, and as such I can't really say much about their
implementation. There are notes in the CHANGES file that indicate the ISDN protocols
over which call completion is supported.
Review: https://reviewboard.asterisk.org/r/523
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256528 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-09 15:31:32 +00:00
|
|
|
/*! Get the device name from the channel */
|
|
|
|
|
#define AST_OPTION_DEVICE_NAME 16
|
|
|
|
|
|
|
|
|
|
/*! Get the CC agent type from the channel */
|
|
|
|
|
#define AST_OPTION_CC_AGENT_TYPE 17
|
|
|
|
|
|
2010-06-08 05:29:08 +00:00
|
|
|
/*! Get or set the security options on a channel */
|
|
|
|
|
#define AST_OPTION_SECURE_SIGNALING 18
|
|
|
|
|
#define AST_OPTION_SECURE_MEDIA 19
|
|
|
|
|
|
2006-04-22 11:30:06 +00:00
|
|
|
struct oprmode {
|
|
|
|
|
struct ast_channel *peer;
|
|
|
|
|
int mode;
|
|
|
|
|
} ;
|
|
|
|
|
|
2001-05-07 02:01:01 +00:00
|
|
|
struct ast_option_header {
|
|
|
|
|
/* Always keep in network byte order */
|
|
|
|
|
#if __BYTE_ORDER == __BIG_ENDIAN
|
2006-09-18 22:02:04 +00:00
|
|
|
uint16_t flag:3;
|
|
|
|
|
uint16_t option:13;
|
2001-05-07 02:01:01 +00:00
|
|
|
#else
|
|
|
|
|
#if __BYTE_ORDER == __LITTLE_ENDIAN
|
2006-09-18 22:02:04 +00:00
|
|
|
uint16_t option:13;
|
|
|
|
|
uint16_t flag:3;
|
2001-05-07 02:01:01 +00:00
|
|
|
#else
|
|
|
|
|
#error Byte order not defined
|
|
|
|
|
#endif
|
|
|
|
|
#endif
|
2006-09-18 22:02:04 +00:00
|
|
|
uint8_t data[0];
|
2001-05-07 02:01:01 +00:00
|
|
|
};
|
1999-11-21 02:26:43 +00:00
|
|
|
|
2006-09-18 23:32:57 +00:00
|
|
|
|
|
|
|
|
/*! \brief Definition of supported media formats (codecs) */
|
|
|
|
|
struct ast_format_list {
|
2009-11-04 14:05:12 +00:00
|
|
|
format_t bits; /*!< bitmask value */
|
2006-09-18 23:32:57 +00:00
|
|
|
char *name; /*!< short name */
|
2007-11-06 22:51:48 +00:00
|
|
|
int samplespersecond; /*!< Number of samples per second (8000/16000) */
|
2006-09-18 23:32:57 +00:00
|
|
|
char *desc; /*!< 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 */
|
|
|
|
|
};
|
|
|
|
|
|
|
|
|
|
|
2005-11-06 15:09:47 +00:00
|
|
|
/*! \brief Requests a frame to be allocated
|
|
|
|
|
*
|
2001-11-10 20:31:39 +00:00
|
|
|
* \param source
|
|
|
|
|
* Request a frame be allocated. source is an optional source of the frame,
|
|
|
|
|
* len is the requested length, or "0" if the caller will supply the buffer
|
|
|
|
|
*/
|
|
|
|
|
#if 0 /* Unimplemented */
|
1999-11-21 02:26:43 +00:00
|
|
|
struct ast_frame *ast_fralloc(char *source, int len);
|
2001-11-10 20:31:39 +00:00
|
|
|
#endif
|
1999-11-21 02:26:43 +00:00
|
|
|
|
2006-09-03 23:14:54 +00:00
|
|
|
/*!
|
2009-06-16 18:54:30 +00:00
|
|
|
* \brief Frees a frame or list of frames
|
2006-09-03 23:14:54 +00:00
|
|
|
*
|
2009-06-16 18:54:30 +00:00
|
|
|
* \param fr Frame to free, or head of list to free
|
2006-09-03 23:14:54 +00:00
|
|
|
* \param cache Whether to consider this frame for frame caching
|
2001-11-10 20:31:39 +00:00
|
|
|
*/
|
2006-09-03 23:14:54 +00:00
|
|
|
void ast_frame_free(struct ast_frame *fr, int cache);
|
|
|
|
|
|
2007-10-10 06:41:51 +00:00
|
|
|
#define ast_frfree(fr) ast_frame_free(fr, 1)
|
1999-11-21 02:26:43 +00:00
|
|
|
|
2006-07-28 18:59:59 +00:00
|
|
|
/*! \brief Makes a frame independent of any static storage
|
2001-11-10 20:31:39 +00:00
|
|
|
* \param fr frame to act upon
|
|
|
|
|
* Take a frame, and if it's not been malloc'd, make a malloc'd copy
|
|
|
|
|
* and if the data hasn't been malloced then make the
|
|
|
|
|
* data malloc'd. If you need to store frames, say for queueing, then
|
|
|
|
|
* you should call this function.
|
2005-11-06 15:09:47 +00:00
|
|
|
* \return Returns a frame on success, NULL on error
|
2009-06-16 18:54:30 +00:00
|
|
|
* \note This function may modify the frame passed to it, so you must
|
|
|
|
|
* not assume the frame will be intact after the isolated frame has
|
|
|
|
|
* been produced. In other words, calling this function on a frame
|
|
|
|
|
* should be the last operation you do with that frame before freeing
|
|
|
|
|
* it (or exiting the block, if the frame is on the stack.)
|
2001-11-10 20:31:39 +00:00
|
|
|
*/
|
1999-11-21 02:26:43 +00:00
|
|
|
struct ast_frame *ast_frisolate(struct ast_frame *fr);
|
|
|
|
|
|
2005-11-06 15:09:47 +00:00
|
|
|
/*! \brief Copies a frame
|
2001-11-10 20:31:39 +00:00
|
|
|
* \param fr frame to copy
|
2006-07-28 18:59:59 +00:00
|
|
|
* Duplicates a frame -- should only rarely be used, typically frisolate is good enough
|
2005-11-06 15:09:47 +00:00
|
|
|
* \return Returns a frame on success, NULL on error
|
2001-11-10 20:31:39 +00:00
|
|
|
*/
|
2006-07-28 18:59:59 +00:00
|
|
|
struct ast_frame *ast_frdup(const struct ast_frame *fr);
|
2000-01-05 00:19:54 +00:00
|
|
|
|
2005-04-04 02:13:40 +00:00
|
|
|
void ast_swapcopy_samples(void *dst, const void *src, int samples);
|
2005-04-03 22:57:18 +00:00
|
|
|
|
|
|
|
|
/* Helpers for byteswapping native samples to/from
|
|
|
|
|
little-endian and big-endian. */
|
|
|
|
|
#if __BYTE_ORDER == __LITTLE_ENDIAN
|
|
|
|
|
#define ast_frame_byteswap_le(fr) do { ; } while(0)
|
2008-05-22 16:29:54 +00:00
|
|
|
#define ast_frame_byteswap_be(fr) do { struct ast_frame *__f = (fr); ast_swapcopy_samples(__f->data.ptr, __f->data.ptr, __f->samples); } while(0)
|
2005-04-03 22:57:18 +00:00
|
|
|
#else
|
2008-05-22 16:29:54 +00:00
|
|
|
#define ast_frame_byteswap_le(fr) do { struct ast_frame *__f = (fr); ast_swapcopy_samples(__f->data.ptr, __f->data.ptr, __f->samples); } while(0)
|
2005-04-03 22:57:18 +00:00
|
|
|
#define ast_frame_byteswap_be(fr) do { ; } while(0)
|
|
|
|
|
#endif
|
|
|
|
|
|
|
|
|
|
|
2005-11-06 15:09:47 +00:00
|
|
|
/*! \brief Get the name of a format
|
2003-08-16 05:10:35 +00:00
|
|
|
* \param format id of format
|
2008-01-10 20:13:24 +00:00
|
|
|
* \return A static string containing the name of the format or "unknown" if unknown.
|
2003-08-16 05:10:35 +00:00
|
|
|
*/
|
2009-11-04 14:05:12 +00:00
|
|
|
char* ast_getformatname(format_t format);
|
2003-08-16 05:10:35 +00:00
|
|
|
|
2005-11-06 15:09:47 +00:00
|
|
|
/*! \brief Get the names of a set of formats
|
2004-05-19 03:39:44 +00:00
|
|
|
* \param buf a buffer for the output string
|
2005-10-26 23:11:36 +00:00
|
|
|
* \param size size of buf (bytes)
|
2004-05-19 03:39:44 +00:00
|
|
|
* \param format the format (combined IDs of codecs)
|
|
|
|
|
* Prints a list of readable codec names corresponding to "format".
|
2004-12-02 21:57:45 +00:00
|
|
|
* ex: for format=AST_FORMAT_GSM|AST_FORMAT_SPEEX|AST_FORMAT_ILBC it will return "0x602 (GSM|SPEEX|ILBC)"
|
2004-05-19 03:39:44 +00:00
|
|
|
* \return The return value is buf.
|
|
|
|
|
*/
|
2009-11-04 14:05:12 +00:00
|
|
|
char* ast_getformatname_multiple(char *buf, size_t size, format_t format);
|
2004-05-19 03:39:44 +00:00
|
|
|
|
2001-11-10 20:31:39 +00:00
|
|
|
/*!
|
2005-11-06 15:09:47 +00:00
|
|
|
* \brief Gets a format from a name.
|
2001-11-10 20:31:39 +00:00
|
|
|
* \param name string of format
|
2005-11-06 15:09:47 +00:00
|
|
|
* \return This returns the form of the format in binary on success, 0 on error.
|
2001-11-10 20:31:39 +00:00
|
|
|
*/
|
2009-11-04 14:05:12 +00:00
|
|
|
format_t ast_getformatbyname(const char *name);
|
2000-01-05 00:19:54 +00:00
|
|
|
|
2005-11-06 15:09:47 +00:00
|
|
|
/*! \brief Get a name from a format
|
2003-08-16 14:48:05 +00:00
|
|
|
* Gets a name from a format
|
2005-11-06 15:09:47 +00:00
|
|
|
* \param codec codec number (1,2,4,8,16,etc.)
|
|
|
|
|
* \return This returns a static string identifying the format on success, 0 on error.
|
2003-08-16 14:48:05 +00:00
|
|
|
*/
|
2009-11-04 14:05:12 +00:00
|
|
|
char *ast_codec2str(format_t codec);
|
2003-08-16 14:48:05 +00:00
|
|
|
|
2007-12-11 09:26:25 +00:00
|
|
|
/*! \name AST_Smoother
|
|
|
|
|
*/
|
|
|
|
|
/*@{ */
|
|
|
|
|
/*! \page ast_smooth The AST Frame Smoother
|
2006-11-07 13:16:05 +00:00
|
|
|
The ast_smoother interface was designed specifically
|
|
|
|
|
to take frames of variant sizes and produce frames of a single expected
|
|
|
|
|
size, precisely what you want to do.
|
|
|
|
|
|
|
|
|
|
The basic interface is:
|
|
|
|
|
|
|
|
|
|
- Initialize with ast_smoother_new()
|
|
|
|
|
- Queue input frames with ast_smoother_feed()
|
|
|
|
|
- Get output frames with ast_smoother_read()
|
|
|
|
|
- when you're done, free the structure with ast_smoother_free()
|
|
|
|
|
- Also see ast_smoother_test_flag(), ast_smoother_set_flags(), ast_smoother_get_flags(), ast_smoother_reset()
|
|
|
|
|
*/
|
2002-06-14 15:35:24 +00:00
|
|
|
struct ast_smoother;
|
|
|
|
|
|
2006-06-05 18:05:53 +00:00
|
|
|
struct ast_smoother *ast_smoother_new(int bytes);
|
|
|
|
|
void ast_smoother_set_flags(struct ast_smoother *smoother, int flags);
|
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int ast_smoother_get_flags(struct ast_smoother *smoother);
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2006-09-18 23:32:57 +00:00
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int ast_smoother_test_flag(struct ast_smoother *s, int flag);
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2006-06-05 18:05:53 +00:00
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void ast_smoother_free(struct ast_smoother *s);
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void ast_smoother_reset(struct ast_smoother *s, int bytes);
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Merged revisions 180372 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r180372 | kpfleming | 2009-03-05 12:22:16 -0600 (Thu, 05 Mar 2009) | 9 lines
Fix problems when RTP packet frame size is changed
During some code analysis, I found that calling ast_rtp_codec_setpref() on an ast_rtp session does not work as expected; it does not adjust the smoother that may on the RTP session, in fact it summarily drops it, even if it has data in it, even if the current format's framing size has not changed. This is not good.
This patch changes this behavior, so that if the packetization size for the current format changes, any existing smoother is safely updated to use the new size, and if no smoother was present, one is created. A new API call for smoothers, ast_smoother_reconfigure(), was required to implement these changes.
Review: http://reviewboard.digium.com/r/184/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180373 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-05 18:29:38 +00:00
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/*!
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* \brief Reconfigure an existing smoother to output a different number of bytes per frame
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* \param s the smoother to reconfigure
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* \param bytes the desired number of bytes per output frame
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* \return nothing
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*
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*/
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void ast_smoother_reconfigure(struct ast_smoother *s, int bytes);
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2006-06-05 18:05:53 +00:00
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int __ast_smoother_feed(struct ast_smoother *s, struct ast_frame *f, int swap);
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struct ast_frame *ast_smoother_read(struct ast_smoother *s);
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2005-04-05 17:04:51 +00:00
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#define ast_smoother_feed(s,f) __ast_smoother_feed(s, f, 0)
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2005-04-03 22:57:18 +00:00
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#if __BYTE_ORDER == __LITTLE_ENDIAN
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2005-04-05 17:04:51 +00:00
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#define ast_smoother_feed_be(s,f) __ast_smoother_feed(s, f, 1)
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#define ast_smoother_feed_le(s,f) __ast_smoother_feed(s, f, 0)
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2005-04-03 22:57:18 +00:00
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#else
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2005-04-05 17:04:51 +00:00
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#define ast_smoother_feed_be(s,f) __ast_smoother_feed(s, f, 0)
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#define ast_smoother_feed_le(s,f) __ast_smoother_feed(s, f, 1)
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2005-04-03 22:57:18 +00:00
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#endif
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2007-12-11 09:26:25 +00:00
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/*@} Doxygen marker */
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2002-06-14 15:35:24 +00:00
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Add basic (passthrough, playback, record) support for ITU G.722.1 and G.722.1C (also known as Siren7 and Siren14)
This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported.
Along the way, some related work was done:
1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way.
2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec.
3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result).
Review: http://reviewboard.digium.com/r/158/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-13 13:35:24 +00:00
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const struct ast_format_list *ast_get_format_list_index(int index);
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const struct ast_format_list *ast_get_format_list(size_t *size);
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2006-06-05 18:05:53 +00:00
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void ast_frame_dump(const char *name, struct ast_frame *f, char *prefix);
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2003-03-12 06:00:18 +00:00
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2006-10-30 16:33:02 +00:00
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/*! \page AudioCodecPref Audio Codec Preferences
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2007-02-24 19:49:11 +00:00
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2006-06-05 15:20:09 +00:00
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In order to negotiate audio codecs in the order they are configured
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2007-02-24 19:49:11 +00:00
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in \<channel\>.conf for a device, we set up codec preference lists
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2006-06-05 15:20:09 +00:00
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in addition to the codec capabilities setting. The capabilities
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setting is a bitmask of audio and video codecs with no internal
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order. This will reflect the offer given to the other side, where
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the prefered codecs will be added to the top of the list in the
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order indicated by the "allow" lines in the device configuration.
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Video codecs are not included in the preference lists since they
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can't be transcoded and we just have to pick whatever is supported
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*/
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2007-07-16 02:51:56 +00:00
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/*!
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*\brief Initialize an audio codec preference to "no preference".
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* \arg \ref AudioCodecPref
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*/
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2006-06-05 18:05:53 +00:00
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void ast_codec_pref_init(struct ast_codec_pref *pref);
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2004-12-02 21:57:45 +00:00
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2007-07-16 02:51:56 +00:00
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/*!
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* \brief Codec located at a particular place in the preference index.
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* \arg \ref AudioCodecPref
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*/
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2009-11-04 14:05:12 +00:00
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format_t ast_codec_pref_index(struct ast_codec_pref *pref, int index);
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2004-12-02 21:57:45 +00:00
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2006-06-05 15:20:09 +00:00
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/*! \brief Remove audio a codec from a preference list */
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2009-11-04 14:05:12 +00:00
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void ast_codec_pref_remove(struct ast_codec_pref *pref, format_t format);
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2004-12-02 21:57:45 +00:00
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2006-06-05 15:20:09 +00:00
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/*! \brief Append a audio codec to a preference list, removing it first if it was already there
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*/
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2009-11-04 14:05:12 +00:00
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int ast_codec_pref_append(struct ast_codec_pref *pref, format_t format);
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2004-12-02 21:57:45 +00:00
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2008-01-17 22:50:13 +00:00
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/*! \brief Prepend an audio codec to a preference list, removing it first if it was already there
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*/
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2009-11-04 14:05:12 +00:00
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void ast_codec_pref_prepend(struct ast_codec_pref *pref, format_t format, int only_if_existing);
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2008-01-17 22:50:13 +00:00
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2006-06-05 15:20:09 +00:00
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/*! \brief Select the best audio format according to preference list from supplied options.
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2004-12-02 21:57:45 +00:00
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If "find_best" is non-zero then if nothing is found, the "Best" format of
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the format list is selected, otherwise 0 is returned. */
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2009-11-04 14:05:12 +00:00
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format_t ast_codec_choose(struct ast_codec_pref *pref, format_t formats, int find_best);
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2004-12-02 21:57:45 +00:00
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2006-09-18 23:32:57 +00:00
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/*! \brief Set packet size for codec
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*/
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2009-11-04 14:05:12 +00:00
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int ast_codec_pref_setsize(struct ast_codec_pref *pref, format_t format, int framems);
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2006-09-18 23:32:57 +00:00
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/*! \brief Get packet size for codec
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*/
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2009-11-04 14:05:12 +00:00
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struct ast_format_list ast_codec_pref_getsize(struct ast_codec_pref *pref, format_t format);
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2006-09-18 23:32:57 +00:00
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2006-06-05 15:20:09 +00:00
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/*! \brief Parse an "allow" or "deny" line in a channel or device configuration
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and update the capabilities mask and pref if provided.
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Video codecs are not added to codec preference lists, since we can not transcode
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2007-01-01 19:48:31 +00:00
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\return Returns number of errors encountered during parsing
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2006-06-05 15:20:09 +00:00
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*/
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2009-11-04 14:05:12 +00:00
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int ast_parse_allow_disallow(struct ast_codec_pref *pref, format_t *mask, const char *list, int allowing);
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2004-12-02 21:57:45 +00:00
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2006-06-05 15:20:09 +00:00
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/*! \brief Dump audio codec preference list into a string */
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2006-06-05 18:05:53 +00:00
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int ast_codec_pref_string(struct ast_codec_pref *pref, char *buf, size_t size);
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2004-12-02 21:57:45 +00:00
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2010-12-01 17:01:56 +00:00
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/*! \brief Shift an audio codec preference list up or down 65 bytes so that it becomes an ASCII string
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* \note Due to a misunderstanding in how codec preferences are stored, this
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* list starts at 'B', not 'A'. For backwards compatibility reasons, this
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* cannot change.
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* \param pref A codec preference list structure
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* \param buf A string denoting codec preference, appropriate for use in line transmission
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* \param size Size of \a buf
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* \param right Boolean: if 0, convert from \a buf to \a pref; if 1, convert from \a pref to \a buf.
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*/
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2006-06-05 18:05:53 +00:00
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void ast_codec_pref_convert(struct ast_codec_pref *pref, char *buf, size_t size, int right);
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2004-12-02 21:57:45 +00:00
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2005-11-06 15:09:47 +00:00
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/*! \brief Returns the number of samples contained in the frame */
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2006-06-05 18:05:53 +00:00
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int ast_codec_get_samples(struct ast_frame *f);
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2005-05-14 23:57:44 +00:00
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2005-11-06 15:09:47 +00:00
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/*! \brief Returns the number of bytes for the number of samples of the given format */
|
2009-11-04 14:05:12 +00:00
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int ast_codec_get_len(format_t format, int samples);
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2005-08-22 19:29:29 +00:00
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2005-12-20 17:52:31 +00:00
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/*! \brief Appends a frame to the end of a list of frames, truncating the maximum length of the list */
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2006-06-05 18:05:53 +00:00
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struct ast_frame *ast_frame_enqueue(struct ast_frame *head, struct ast_frame *f, int maxlen, int dupe);
|
2005-12-20 17:52:31 +00:00
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2005-11-06 15:09:47 +00:00
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/*! \brief Gets duration in ms of interpolation frame for a format */
|
2009-11-04 14:05:12 +00:00
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static inline int ast_codec_interp_len(format_t format)
|
2005-05-12 19:01:03 +00:00
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{
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|
return (format == AST_FORMAT_ILBC) ? 30 : 20;
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}
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2005-10-28 16:20:44 +00:00
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/*!
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\brief Adjusts the volume of the audio samples contained in a frame.
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\param f The frame containing the samples (must be AST_FRAME_VOICE and AST_FORMAT_SLINEAR)
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\param adjustment The number of dB to adjust up or down.
|
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|
\return 0 for success, non-zero for an error
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*/
|
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int ast_frame_adjust_volume(struct ast_frame *f, int adjustment);
|
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|
2005-10-28 21:49:27 +00:00
|
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/*!
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|
\brief Sums two frames of audio samples.
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|
\param f1 The first frame (which will contain the result)
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|
\param f2 The second frame
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\return 0 for success, non-zero for an error
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The frames must be AST_FRAME_VOICE and must contain AST_FORMAT_SLINEAR samples,
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and must contain the same number of samples.
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*/
|
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int ast_frame_slinear_sum(struct ast_frame *f1, struct ast_frame *f2);
|
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2008-03-05 01:54:16 +00:00
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/*!
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* \brief Get the sample rate for a given format.
|
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|
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*/
|
2009-11-04 14:05:12 +00:00
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static force_inline int ast_format_rate(format_t format)
|
2008-03-05 01:54:16 +00:00
|
|
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{
|
Add basic (passthrough, playback, record) support for ITU G.722.1 and G.722.1C (also known as Siren7 and Siren14)
This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported.
Along the way, some related work was done:
1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way.
2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec.
3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result).
Review: http://reviewboard.digium.com/r/158/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-13 13:35:24 +00:00
|
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switch (format) {
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case AST_FORMAT_G722:
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case AST_FORMAT_SLINEAR16:
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|
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case AST_FORMAT_SIREN7:
|
2010-06-17 17:23:43 +00:00
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case AST_FORMAT_SPEEX16:
|
2008-03-05 01:54:16 +00:00
|
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return 16000;
|
Add basic (passthrough, playback, record) support for ITU G.722.1 and G.722.1C (also known as Siren7 and Siren14)
This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported.
Along the way, some related work was done:
1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way.
2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec.
3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result).
Review: http://reviewboard.digium.com/r/158/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-13 13:35:24 +00:00
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case AST_FORMAT_SIREN14:
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return 32000;
|
2010-06-16 19:03:24 +00:00
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case AST_FORMAT_G719:
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|
return 48000;
|
Add basic (passthrough, playback, record) support for ITU G.722.1 and G.722.1C (also known as Siren7 and Siren14)
This patch adds passthrough, file recording and file playback support for the codecs listed above, with negotiation over SIP/SDP supported. Due to Asterisk's current limitation of treating a codec/bitrate combination as a unique codec, only G.722.1 at 32 kbps and G.722.1C at 48 kbps are supported.
Along the way, some related work was done:
1) The rtpPayloadType structure definition, used as a return result for an API call in rtp.h, was moved from rtp.c to rtp.h so that the API call was actually usable. The only previous used of the API all was chan_h323.c, which had a duplicate of the structure definition instead of doing it the right way.
2) The hardcoded SDP sample rates for various codecs in chan_sip.c were removed, in favor of storing these sample rates in rtp.c along with the codec definitions there. A new API call was added to allow retrieval of the sample rate for a given codec.
3) Some basic 'a=fmtp' parsing for SDP was added to chan_sip, because chan_sip *must* decline any media streams offered for these codecs that are not at the bitrates that we support (otherwise Bad Things (TM) would result).
Review: http://reviewboard.digium.com/r/158/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-13 13:35:24 +00:00
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default:
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return 8000;
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}
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2008-03-05 01:54:16 +00:00
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}
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2010-04-21 11:27:27 +00:00
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/*!
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* \brief Clear all audio samples from an ast_frame. The frame must be AST_FRAME_VOICE and AST_FORMAT_SLINEAR
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*/
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int ast_frame_clear(struct ast_frame *frame);
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1999-11-21 02:26:43 +00:00
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#if defined(__cplusplus) || defined(c_plusplus)
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}
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#endif
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2005-08-30 18:32:10 +00:00
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#endif /* _ASTERISK_FRAME_H */
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