MODIFICATIONS FILE
------------------

$Id: MODS,v 1.308 2002/07/23 20:36:28 ucaccsp Exp $

Copyright (C) 1995-2002 University College London
All rights reserved. 

This software is distributed under license, see the file COPYRIGHT for full
terms and conditions.

The modifications here include the development versions, not all of which 
are released.  This list is not intended to be comprehensive. The current 
release is available at http://www-mice.cs.ucl.ac.uk/multimedia/software/rat/

v0      Initial audio tool
	Old VAT and RTPv2(5) compatibility
	PCM DVI GSM LPC codecs

v1      Vicky's restructuring
	Dropped old VAT compat mode
	Used in ReLaTe trials
	New UI including small mode
	Video Sync code added

v2.0    Transmitter cleaned up
	Lots of bugs fixed
	Replaced DVI and LPC codecs with VAT4 versions
	Released in Boston WWW4

v2.1    Ported to Tcl7.3/Tk4.0
	Added UI response slider

v2.2    Added play and rec options

v2.3    Port to Tk4.1/Tcl7.5

v2.4a0  Port to Windows95
v2.4a1  Fixed active speaker indication
v2.4a2  Fixed GSM
v2.4a3  Release to MERCI partners for testing
v2.4a4  Fixed broken assertions...
v2.4a5  SGI fix
v2.4a6  Security holes fixed
	Assertion handling improved
	Start at encryption and multi-rate audio
v2.4a7  FreeBSD modifications
	Fixed playout calculation (probably)
v2.4a8  Added RTCP SDES TOOL field to stats panel
	Fixed version number
	Made packet repetition an option and set default to OFF
	Fixed encoding selection bugs
	Misc bug fixes....
v2.4a9  Fixed participant display when audio device is busy
	When device is busy audio controls get disabled
	Added code to set microphone gain under Win95
	Modified loop timing for win95 version so that it does not hog CPU
	Made session title appear as window title with -C flag
v2.4a10 Added OS type to RTCP tool string

v2.5a0  Fixed long DVI packet format. Now there is only one state (VAT comp).
	Reduced UI window border width. New look!
	Added scrollbar to participants window
	New participant stats code. Allows multiple windows...
	Attempt to fix some of the stats
	Popup window to request RTP name if resource not set
v2.5a1  Changed redundancy packet format to conform to UCL-INRIA spec
	Added error code to report use of old packet format
	New code to do encoding name stats
	Fixed RTCP RR packets to conform with latest spec
	Made RAT look in .RTPdefaults for user name
	Added IP address display in stats
v2.5a2  Highlight other RAT users in list of participants
	Ignore audio until RTCP packet received
v2.5a3  Added encryption (Darren said it's compatible with vat)

v2.6a0  Added command line option to set redundancy type
	Started to change payload type to match draft-perkins-rtp-redundancy-01.txt
	(Grouped headers, but haven't changed bit order yet...)
v2.6a1  Finished changing the payload type.
	Default redundancy pt=121
	Send initial RTCP message as soon as we start up...
v2.6a2  Allow window to resize horizontally (requested by vilhuber@cs.colorado.edu)
	Updated copyright notice.
	Updated about panel.
	Added -pt pt/codec/clock/channels command line option. 
	Example: "-pt 121/redundancy" to set redundancy payload type to 121
	Remove -rpt option (use -pt instead).
	Remove -f2 option.
	Change -f option to accept things like "-f pcm/dvi" to set redundancy type.
v2.6a3  Fixed packing order so it resizes horizontally correctly.
v2.6a4  New WIN32 version
	Changed priority of thread
	Fixed cushion code to work with I/O switching
	Linked tcl and tk with WIN32 binary so no DLLs are needed
v2.6a5  Partial rewrite of win32 audio driver to use static buffers
	Solved most win32 audio problems
v2.6a6  Made the prompt for .RTPdefaults information less user hostile
	Fixed LPC decoding on SGIs (-fsigned-char does it!?!)
v2.6a7  Removed old bitmap from control panel
	Rewrite of small user interface (command line option -smallui)
v2.6a8  Further rewrite of the small user interface
	Fixed SO_REUSEADDR
	HP-UX fixes
	Added "-crypt <key>" command line option to enable encryption
v2.6a9  Fix m/cast power meters when encryption is turned on.
v2.6a10 HP-UX fixes
v2.6a11 Remove debugging malloc wh  en NDEBUG is defined. 
v2.6a12 Code cleanup in preparation for mixer
	Corrected URL on the about panel
v2.6a13 Code cleanup in preparation for mixer
	Added -li <local_interface> option for multihomed hosts
	Removed annoying transient tag from pop-up windows.
v2.6a14 Added an icon
	Win32 fixes
	Removed -li stuff, since it broke the win32 port.
	More HP-UX changes

v3.0.0  Restructure code for audio translator/mixer
	Extended ascii hostnames (>15 chars) - useful for unicast
	Additional RTP header validation
v3.0.1  Major restucturing of main_rat.c, and split into several separate files.
	Fixes for Linux.
	Include full/half duplex stuff in all versions.
v3.0.2  Bug fixes to auddev_linux.c
	Fixes to RTCP header checks on little-endian machines.
	Now obeys xresources settings.
v3.0.3  Experimental release for Linux
	Some fixes to encryption code: it still doesn't work though...
v3.0.4  Major mods to init_session.c
	Added mixer code to linux port.
v3.0.5  Fake TS start if we TOG packets
	Fix RTCP timestamp epoch to 1900 not 1970 to conform to RFC1889
	Fix display of group address in options panel.
	Fix CNAME.
	Extensive changes for the audio transcoder/mixer.
	New HP-UX audio device driver from UiO added.
v3.0.6  More changes for the audio transcoder/mixer.
	Fixes to Linux audio driver when in half-duplex mode.
	Added the -version option.
v3.0.7  Major changes for the audio transcoder/mixer: First version to work at all!
	Now sources $HOME/.rat as tcl on startup
	Linux fixes
v3.0.8  CNAME now uses dotted-quad representation of the IP address, not FQDN.
	16bit linear audio throughout main code: not all device drivers fixed yet.
	Make DVI the default primary encoding.
	Added L16 (16bit linear, uncompressed) codec.
	Both sides of transcoder/mixer use same SSRC/CNAME.
	Calculate RTCP reporting interval, rather than using constant 6 seconds.
	Major rewrite of transmitter and receiver code (supports multiple red)
	Added CSRC processing to receiver
	Fix indication of active speakers
	Update device buffer cushion code
	Removed small (ReLaTe) UI
	Minor redisign of option UI
	Added lecture mode
	Linux port now works again
v3.0.9  Added minimal LBL conference bus support 
	Enabled voice-switching of video using LBL conference bus/vic
	Fixed SGI 16 bit
	Fixed WIN32 16 bit
	Slight highlighting of participants after they stop speaking
	Now uses tk4.2/tcl7.6
v3.0.10 Prehang and posthang added to new code
	IP_MULTICAST_LOOP is now on for the session port
	We can now parse our BYE message although it is wrong
	Packet repetition added to new code
	Audio DC bias removal added for all platforms
v3.0.11 We can no longer parse our BYE message, since it is wrong!
	More UI changes... (packet format uses pull down menus)
	Disable powermeters option added
	Selective muting of participants possible with middle mouse button.
	Automatic bias removal (turns off if not needed)
	Fix handling of RTP packets with CSRC data
	Rewrite of RTCP handling code...
	Fix RTCP BYE packet
	Transcoder/mixer now passes RTCP packets
	Exponential fade on packets   (no good)
	LPC redundancy alignment code (also no good)
v3.0.12 Linear fade on repeated packets
	Fade on repeated LPC decoded packets
	Prevention of mixing dummies with no decompressed data
	Packet length shown in UI
	Can select N-level redundancy from command line (-f pcm/dvi/lpc for example)
	Fix parsing of command line options
	Lecture mode is on by default
	Turn off lecture mode as soon as we send data
	Added -loopback_rtp option for loopback of the RTP packets (RTCP loopback is always on)
	Added audio device trading via the LBL conference bus
	Added flakeway operation
v3.0.13 Integrated transcoder/mixer into the main Rat code.
v3.0.14 Move to tcl-7.6p2/tk-4.2p2
	Fix linux/HP-UX ports
	Fix for flakeaway
v3.0.15 Assorted bug fixes
	Hang onto multiple encoded data in receiver
	Logic into UI for redundant coding types  
	Fix LPC redundancy (Ho Ho)
	Fix sending L16 with redundancy
	Fix auto-select of "Mike mutes net" when running on a half-duplex machine
	TCL_ALL_EVENTS in event processing loop (sockets from tcl scripts->fileevents)
v3.0.16 Bug fixes to the transmitter
	Fix CSRC processing
v3.0.17 Fixes for Win32
v3.0.18 Fix RTCP BYE packet on little-endian machines
	Muting of output now frees data
	Fix reception of 160ms packets 
	Fix uninitialised variable in rtcp_db 
v3.0.19 Remove spurious debugging messages when we can't open the audio device
	Cleanup HP-UX audio driver
v3.0.20 Can now select line-in/microne input on SGI machines
	Added -K option to enable encryption (synonym for -crypt)
	Cleanup Linux audio driver
	Cleanup Sparc audio driver
	Cleanup for draft-ietf-avt-profile-new-00
	Added -double_speed (16kHz) option
	Transcoder no longer hits stdin/stdout with audio ioctls
	Rewrite and optimize decryption of RTCP packets
	RTCP timer reconsideration done for every RTCP packet received (following presentation at Memphis IETF...)
	RTCP SR and SDES fixes
	Can now send/display PHONE, LOC and EMAIL SDES items.
	Fixed mike mutes net operation for half duplex cards.
	FreeBSD port now works.
	Win32 version uses registry to store defaults.
v3.0.21 Added signal handler, to catch SIGINT and exit cleanly (with RTCP BYE)
	Linux device driver now accepts 16kHz operation.
	Assorted bug fixes to Linux device driver.
	Initialise device_time variable: prevents crash if we can't access the audio device.
	The use of strdup() interacts badly with xmalloc(). Fixed.
	Fixed transcoder RTCP BYE handling
	* Branch here: v3.1 development starts, later v3.0 fixes will have to 
	* be forward ported
v3.0.22 Remove debugging messages in statistics.c
	Remove WBS codec
	Remove -double_speed option
v3.0.23 Linux port uses both audio channels, rather than just the left channel
	Linux port no longer repeatedly resets the volume to 50% 
	Bug fix for HP-UX audio driver (Terje Vernly <terjeve@newlearning.no>)
	Fix CPU usage on Win32
v3.0.24 Fix encryption
	Fix loopback of packts
	Fix crash in LBL conference bus code, sending focus when the CNAME was NULL
	Fix RTCP header validation
	Fix RTP header validation when padding bit is set
	Fix voice switching of video using LBL conference bus with vic
	Modifications to WIN32 silence detection as PC hardware often breaks previous algorithm.
	Assorted Win32 bug fixes
	Win32 HOMEDIR bug fix. 
3.0.25  Assorted Win32 fixes.
3.0.26  Fixed bug in net.c that caused packets to cease processing on some Win32
	boxes.
3.0.27  Fix transcoder problems on half-duplex machines. Thanks to Sachin Sheth 
	for providing the fix.
	Fix fonts for solaris 2.6.
	* Released 15 August 1998
3.0.28  Update copyright notice (the dates were wrong!)
	Endian fix to the RTCP packet forwarding code (Thanks to Sachin Sheth).
	Fixed HOMEDIR problem introduced in 3.0.27 for Win32.
	Fixed catch when sourcing ~/.rat
	* Released 22 September 1998

3.0.29  Added -smallui option to use relate interface and to enable power
	messages via the lbl confbus
	Fixed -drop option
	* Released 21 December 1998

3.0.30  Fix statistics in RTCP SR/RR packets
	Fix copyright notice in small UI
	Fix URL in the about panel
	* Released 6 January 1999

3.0.31  Remove obsolete tcl/tk library code
	Fix tcl/tk configuration on win32
	Move to using winsock2.h on win32
	* Released 29 January 1999

3.0.32  Fix Tk resource files on Win32
	* Released 4 February 1999

3.0.33  Fix font in the statistics panel, which I broke in 3.0.31 (oops!)
	Added A-law audio codec.  Back port from 3.2x.
	Added MMX mixing support.  
	Patch from Michael Wallbaum for transcoder on win32.
	Made file processing use binary mode opens.  Win32 cares.
	* Released 1 March 1999

3.0.34  Removed -D$(USER) from Makefile (Reported by Ola Borrebak).
	Added alaw to command line codecs (Reported by Dimitris Terzis).
	Winsock 1/2 detection and selection.  
	Bug fixes for Winsock 1 built with Winsock 2 headers.


v3.1.0  Added the wide-band speech codec (use with the -double_speed hack)
	Cleanup statistics.c
	Fixed Win32 cname
	Made left mouse button on participants toggle stats on and off
	Keep input/output mode selection and gain settings when device trading
	Reorganize the codec_* files
	Add indication of loss rat to UI
	Add automatic gain control
	Fix CPU usage on Win32
	GSM now implements 06.11 repair (so there)
	Fixed self induced LPC bug :-)
	Improved cushion code to give lower delay
	Completely fixed highlighting (honest)
v3.1.1  Default RTP port is 5004 (draft-ietf-avt-profile-new-00)
	Turn off automatic gain control in the transcoder
	Linux port uses both audio channels, rather than just the left channel
	Linux port no longer repeatedly resets the volume to 50% 
	UI user participant code now maintains sorted list
	No longer source $HOME/.rat on startup
	Separate the UI from the media engine (use conference bus for communication)
v3.1.2  Many changes to the conference bus protocol and support code
	Allow multiple UI scripts (use -ui <name> to select)
	Unify Linux/FreeBSD drivers into auddev_oss.c
	Fix HP-UX audio driver (Terje Vernly <terjeve@newlearning.no>)
	Fix encryption code on little-endian machines
	Fix RTCP header validation code
	Fix security hole in confbus (zap [] in strings send to tcl)
	Add Anna Bouch's UI code (use "-ui anna" to enable)
	Add ReLaTe UI code (use "-ui ReLaTe" to enable)

v3.2.0  - Codec interface rewrite. Based on structure holding all codec
	  parameters and pointers to init/encode/decode functions.
	- Multi rate support. Multiple clock support (one per participant plus
	  a few internal ones to keep RTCP etc working).
	- Rewrite of transmit code. Circular buffer is gone and now a list of
	  minibufs is used instead. Within each minibuf the unit_size (which
	  is calculated from number of channels, freq etc) remains constant.
	- Rewrite of receive code to support multiple unit_sizes. Now each
	  receiver has a separate receive buffer. Within each buffer units of
	  different length can be mixed without any problems. The only
	  restriction is that for any particular time segment all coded_units
	  must be compatible (i.e same length /frequency/channels etc).
	- All the code understands number of channels, variable unit sizes and
	  multiple frequencies. This information is deduced from codec
	  information. This includes mix and statistics...
	- Added hooks for up and down sampling of remote participant decoded
	  audio. 
	- Cleaned up include file mess. Now each source file includes only
	  required headers. In the process of doing so I cleaned up a few
	  interfaces so parts of the code that do not need to know about
	  structures don't see the definitions and work with pointers.
	- Various old and new bug fixes :-)
	- Probably some more changes that I can not remember now as I didn't
	  bother to write them down as I was doing them ;-)
	- Reimplemented repair code from previous release and left hooks for
	  pitch repetition and pattern matching. Now repair is multi-channel
	  and multi-freq.
	- Centralized allocation of rx_queue_element's.
	- Added CD input support
	- Assorted bug fixes to the original UI
	- Add reception quality chart to the original UI
	- Fix encryption code on little-endian machines (again)
	- Cleanup RTCP code and headers
	- Many changes to the conference bus (Mbus) code
	- Assorted bug fixes to the transcoder
	- SGI version now sets initial volume to something sensible
	- Incorporated Luigi's mods to FreeBSD driver
	- Added Pattern Matching repair
	- Fix loopback of RTP packets
	- Modified rules_based_silence to work at rates other than 8K.
	- Add balloon help
	- Fix RTP header validation for packets with P bit set [Bill Fenner]
	- Fix transcoder timing (RTCP packets are sent much too often:
	  loopback problem...)
	- Fix freeing of units held in the receive buffer, when an RTCP BYE
	  packet is received.
	- Seems that FreeBSD (unlike the rest of the known universe...) has
	  a SO_REUSEPORT, which we need to set if we're doing TTL zero
	  multicast. Sigh.
	- Fixed timer code - works with all freqs that divide into 96000.
	- Added channel coding functionality - it should now be possible
	  to slot in any scheme.
	- Reimplemented redundancy to comply with new channel coding regime.
	- Added interleaving.
	- Fix decoding of encrypted packet (no longer blocks...)
	- Fix encryption code on little-endian machines (again, again!)
	- Fix fragment size on Linux
	- Added header extension support.
	- Remove LBL conference bus support (exercise for the reader: write
	  an Mbus to LBL conference bus bridge, to allow device trading...)
	- Add binding for Control-Button-1 to mute individual participants
	- Add lip-sync support
	- Major UI rewrite
	- New silence detection algorithm.
	- Fix parsing of RTCP SR/RR packets on little-endian machines
	- Fix of decoding path - repair works again now.
	- Separated out cushion code (cushion.[ch]) and cleaned a little.
	- Separated out the UI update code from the Tcl/Tk maintainence code.
	- Modularise Mbus code
	- Stereo (?)
	- Time code simplified.
	- Multi-channel silence suppression.
	- Misordered packets counted.
	- Remove NET_MUTES_MIKE and MIKE_MUTES_NET options. We now require
	  a full duplex audio device to operate. This should not be a
	  problem, since such hardware is readily available for all
	  platforms we care about (and some we don't...).
	- Minor memory allocation optimization for native units and mbus 
	  block_alloc replacing xmalloc for fixed size allocations.
	- Modularisation of silence detection and voice activity code.
	- Transmit buffer simplification.
	- Changes labels on preferences panel to match sdr, since it makes
	  sense that way, and the old labels were broken dammit! (and I
	  don't care if they were the ones Windows uses....)
	- Remove flakeaway options (Cadbury's must hate us...)
	- Create RTCP database entries for new sources when we find them 
	  in SR/RR packets, rather than ignoring them.
	- Fix operation with -no_ui. This is not very useful though...
	- Hack the RTCP decryption code to accept encrypted packets from
	  vat, even though vat sets the padding bit incorrectly, and gets
	  the length of encrypted SDES packets off by one. 
	- Fix source address checking in the mbus code
	- Added sampling options to transmission menu.
	- Added destruction code to coders and channel coders.
	- Make sure the random number generator is initialised before use.
	- Rejigged code to allow hot swapping of sampling format mid-session.
	- Fix RTCP reporting interval, when running at other than 8kHz
	- Change of RTCP reporting interval when freq changes.
	- Replaced old AGC code.
	- UI tweaks, bitmaps, titling, about panel.
	- Playout point limits now optional.
	- Allow for remote ui, enable with "-ui <mbus_address>"
	- Watch for transmitter device clock being slower than ours and 
	  corrective action.
	- Cushion decrease only when not playing audio - stops interruptions.
	- Allow `wait for mbus init message' on startup with "-wait"
	- Redundancy new format - mods to rfc to match implementation.
	- Removed auto_path initialisation from init.tcl and aatk.tcl.
	- Interleaver (mk II) no longer halts the system (or works).
	- Powermeters frequency independent.
	- Own activity only sent when changing state.
	- Activity changes no longer refresh participant info.
	- Playout buffer size monitoring and device drift bias correction.
	- Fixed problem with codec recognition at startup when rate not 8k.
	- UI displays correct encoding with interleaver and redundancy.
	- Wrote a configure script
	- Graceful failure/stasis when device not available.
	- Bitmaps match audio i/o ports at start up.
	* Released to CAIRN/MECCANO for testing, 8 June 1998

v3.2.1  - Assorted tweaks to the UI
	- Cleanup mbus_encode_str interface
	- a small change to function rx_source_playout in mbus_engine.c
	- fixed bug in ui_update_video_playout
	- lip-sync support
	- play audio packets in no lip-synced mode when no RTCP SR is received 
	  even if lip-sync is enabled. When SR is received, switch to lip-sync 
	  mode
	- bug in mbus resend code fixed.
	- Cleanup parsing of command line options
	- Add patches for HP-UX 10 from Dirk Meersman
	- Remove HP-UX 9 compatibility: it makes the code cleaner, and
	  HP-UX 9 is completely obsolete anyway. Now if only we could do
	  the same with SunOS 4...
	- Only allow lip-sync to be enabled if we are not running on the
	  Mbus base c  hannel
	- Table for format conversions.  Windows ACM PCM Converter support.
	* Released for testing, 10 July 1998

v3.2.2  - Fixes for command line option parsing
	- Improvements to playout calculation (less togged)
	- Increased audio in receive buffer to be 1 cushion's worth. This 
	  makes handling "fast" senders easier and also reduces togged units
	- tcl2c now strips comments, 10-40% size improvement on tcl strings
	  (UNIX only).
	- Removed fixed tables for g711 conversion.  Now calculated at start
	  time.
	- Increased minimum pitch length in pattern match repair (less pings).
	- Add support for TclX, and profiling of the tcl code
	- Attempt to optimize some of the more intensive Tcl routines
	- Reduce rate of power-meter updates, since profiling shows that is
	  an expensive operation. 
	- Added playout buffer size to participant info
	- Converted jitter into ms for ui
	- Increased initial estimate of jitter, and codec independent
	- Reduced average playout delay, now just add/subtract cushion surplus
	  instead of whole cushion length to playout, i.e. we add exactly 
	  the right amount.
	- Rearranged files for more portable Windows configuration.
	* Released for testing, 17 July 1998

v3.2.3  - Added button in the UI to enable sound externalisation (the
	  actual 3d audio code is not present yet).
	- Fix display of loss rates in the participant list ("loss diamonds")
	- Updated sdr plugin (in sdr v2.5a4)
	- Change initial silence detection threshold, such that we transmit
	  until the detector has trained rather than suppressing.
	- Win32 release configuration added.
	- Added reset of saved settings when version change detected.
	- Minor change to low jitter playout handling.
	- Fix parsing of -K/-crypt options
	- Endian fix to RTCP packet forwarding code
	- Remove TclX support (as part of the planned move to using the
	  Tcl plus-patch to cleanup the initialisation code).
	- Fixed playout point calculation problems.  Playout buffer should
	  now have same order as source timestamps, no overlaps allowed.
	- Powermeters tweak.
	- Move to winsock2.h, in preparation for IPv6 port. NOTE: This now
	  requires that winsock2.dll is installed on Windows.
	- Win32 ACM G.723.1 code.
	* Release for testing, 24 August 1998

v3.2.4  - Cleanup networking code
	- Cleanup RTCP code
	- Only send an RTCP BYE packet if the number of members is less
	  than 50. This means we just-about comply with the latest RTP
	  draft, although we should really implement BYE reconsideration.
	- Fixed bug introduced into repair code.
	- Increased powermeter rate and added table to improve vu performance.
	- RTCP timeout frequency independent.
	- Got ride of speaker table code. Highlighting is now from when playout 
	  buffer created destroyed.
	- Fix GSM codec
	* Release for testing, 10 September 1998

v3.2.5  - Fix transmission mute/unmute
	* Release for testing, 10 September 1998

v3.2.6  - Move all system specific configuration options and includes into
	  config_unix.h and config_win32.h to ease porting. Should also
	  simplify things when we move to a common code library among the
	  UCL tools.
	- Audio loopback in hardware for Solaris/SunOS and Windows.
	- Moved select on audio_fd out of networking code.
	- Centralized code that destroys and empties playout buffers.
	- Added "net mutes mike" and called it "Silence Suppression".  Works
	  off of creation and destruction of playout buffers.
	- Assorted UI tweaks
	- Move to using our own version of tcl/tk on unix machines, to 
	  avoid the problems with initialisation scripts being sourced
	  from other installed versions.  We no longer compile the tcl
	  or tk initialisation scripts into rat.
	- Merge GSM codec into codec_gsm.[ch]
	- Modularize networking code
	- Mbus code now batches messages before transmission, to reduce
	  the number of packets sent
	- Update Mbus command set to better match latest specification
	- Add 3d audio code
	- Add IPv6 support under Windows NT 4.0 using the stack from 
	  Microsoft Research (http://www.research.microsoft.com/msripv6/)
	* Release for testing, 9 October 1998

v3.3.0  - Audio interface revamp.  Support for multiple cards.
	- Osprey SunVideo Plus audio support added (buggy).
	- PCA audio support added on FreeBSD (builtin speaker).
	- Documented audio interface.
	- Replaced audio_fd with audio_device to emphasize this is no longer
	  a file descriptor.
	- Added null audio interface for devices not attainable.
	- Safety net when opening an interfac   - Fix LSR and DLSR in our reception report packets (maybe)
	- Fix muting of individual participants
	- Update mbus command set
	- Added virtual externalisation (sound source emanating from outside
	  the head even when listened to with headphones)
	- File loading and saving added.
	- Playout calc refinements.
	- Repair schemes given queryable interface (like other modules).  
	  mbus and ui updated to reflect this.
	- Fix for fatal bug in channel decoder arising from duplicates and 
	  interaction with playout buffer.
	- Merge in IPv6 fixes from Maryann Maher
	- 3D rendering code added, debugged, and integrated into UI.
	- Audio device interface changes - input format may differ 
					   from output format.
					 - format conversion in audio 
					   interface if dev does not
					   support required format.
	- Win32 RTCP tool includes OS version of end host.
	- Win32 Multiple audio device support bug fixed.  Uses matching
	  card and mixer.
	- Fix stereo powermeters (work).
	- Convert to use the common code library
	- Disposed of old packet queues, now use simpler buffer and no
	  nasty macros.  (rx queue still there, when there's time it goes..)
	- Re-wrote codec interface.  Codec wrappers appear in files 
	  codec_*.[ch]
	  and codec implementation files appear in cx_*.[ch].
	- codec_t axed and replaced with codec_id_t.  Look at codec.h for
	  manipulation functions.
	- Codec repair functions appear in codec interfaces rather than
	  reair function.
	- Removed all RAT dependencies from codec code, repair code, 3d code, 
	  appart from a few basic types i.e sample, audio format, and 
	  coded_unit.
	- Disabled all channel coding except vanilla.  All to be 
	  re-implemented.
	- Added ability to read number of samples written to and read from 
	  audio device.
	- Made audio device id's (handles) mirror codec id's.  We now always 
	  have a valid handle for a device, and can read/write to it.
	- Removed interfaces.[ch].
	- Replaced receive.[ch] with source.[ch] and replaced playout
	  buffer code with playout.c.
	- Rewrote channel coder interface to be cleaner.
	- Removed unnecessary dependencies in rate and channel conversion code.
	  Conversion routine now uses coded_units for input and output.
	- Moved codec_state_list onto source structure and channel_coder_state.
	- Fixed dependencies in repair and 3D rendering code. Their only 
	  dependencies are now the codec files.
	- Made converter interfaces same as audio device and codec interfaces.
	- Changed 3d rendered to always output stereo.
	- Fixed potential problem in 3D code where the audio frame size being
	  passed is smaller than the delay the render applies causing samples
	  to be shifted over the stack.
	- Added tests directory where people can through tests for modules if
	  they care.  So far have added test_3d.c which found the last problem
	  and native.c used to fix bug in native coding types.  
	- Fixed nasty bug in 3d rendered where at high sampling rates the delay
	  size is larger than the storage buffers between intervals causing
	  memcpy to overlap src and dst areas.
	- Fixed bug in vdvi frame peek code and sizes returned by
	  vdvi_{en,de}code.  All codecs pass tests in test_codecs.
	- Wrote timestamp code that allows easy comparison between multi-rate
	  clocks.
	- Added first order differencing for sequences of timestamps so
	  wraps in 32 bit clocks do not cause discontinuities in modulo m
	  clocks and vice versa.
	- Added iterators to playout buffer.
	- Updated transmitter to use playout buffer with multiple iterators
	  instead of former custom list code.
	- Modified channel code and source code for playout buffer iterators.
	- Fixed widgetabbing (due to missing tk library scripts).
	- CNAME now uses the IPv6 host address, if running on IPv6.
	- Move mbus heartbeat into the common code library.
	- Repair and duplicate handling put back into new decode path.
	- Separated sample ops into audio_util.[ch].
	- Added MMX mixing code for Win32.
	- Moved bias control out of session pointer and into transmitter.
	- Removed support for multiple mbus channels.
	- Removed unnecessary include files from assorted source files.
	- Cleaned up device reconfiguration code.
	- Fixed l16 48k stereo name - was incorrect.
	- Renamed certain functions to make names congruent to group of calls.
	- Removed input_mode and output_mode from session_struct.
	- Restored playout buffer length reporting functions.
	- Added source flush if mix fails (source sample rate change).
	- Added repair failure code, should only fail if sample rate changes.
	- Added codec list option (-codecs) for launcher apps to know what is
	  available.
	- Added payload mapping from UI.
	- Added Sun's G726 public reference version.
	- Redundancy encoder and decoder re-written but not linked in yet, 
	  needs debugging.
	- Rewrote win32 mixer interface code.  No longer use string matching
	  with pre-defined english names.  Still have to use some string 
	  matching but only between names various controls from the driver
	  spit out.  Took circuitous route since no two driver vendors 
	  implement mixer functions in same way.  Seems to be stable now.
	- Modularized audio ports so that device interfaces now return
	  audio_port_details_t that contains string name and unique 
	  input/output identifier to be used for port selection.
	- Modified UI to have transmit and receive checkbuttons and replaced
	  bitmaps with string names since we cannot guarantee win32 port 
	  names and changing between bitmaps and strings in the ui is ugly.
	- Fixed transmitter bug where temporary audio_buffer iterator
	  was not being freed.

v4.0.0  - Update license agreement -- we now use a standard Berkeley style 
	  license for all files. See the file LICENSE for details.
	- Removed output_gain and input_gain from session_struct.  Should
	  fix output gain going to zero on some updates.
	* Released 31 March 1999

v4.0.1  - Clarify license -- the LICENSE and COPYRIGHT files contradicted
	  each other, we now only have one file (COPYRIGHT) which is a
	  standard Berkeley style license.
	- Fixed problem with pseudo-devices (auddev_null and
	  auddev_pca) that use gettimeofday to calculate how many samples are
	  available.  They were failing to include number of samples included
	  in "device" buffer left over from previous reads.  This meant device
	  time was always slow.
	- Go back to ssrc rather than cname in UI for indexing entries.  
	  Issues running multiple instances on single machine otherwise.
	- Playout buffer skew adjustment added so when playout buffer 
	  deviates from expected size because source clock is different and
	  signal is temporarily silent an adjustment is made.  Only intervals 
	  when adjustment made is when packet has arrived in current 
	  processing interval so we do not adapt at the end of talkspurts.
	- Made repair and converters communication like that of input and
	  output ports in ui.  i.e. tool.rat.converters.flush, 
	  tool.rat.converters.add, tool.rat.converter.  Defaults for repair
	  and conversion are first type of module.
	- Issues with mixing multi-rate fixed.  Existing code was not 
	  accounting properly for different sample rates when positioning new 
	  mixer head time.  This probably got introduced when the time
	  stamp method got changed since it definitely worked before (honest).
	- Added status bar's to file window.  Hopefully clearer what is going 
	  on.  Attempted fix button of state problems.
	- Caching of previous talkspurt jitter.  Previously a fresh guess was 
	  made at the start of each spurt.
	- source_get_playout_delay now includes audio we believe to be in
	  device buffer.
	  So buffer shifts now work with cushion...hopefully.
	- Added counter for skew adjustments to make adjustments less
	  aggressive.
	- Win32 mixer corrections - mixer control code now works with all
	  the cards we have. Let this be the end of it!
	- Added read_and_discard for reading all available packets and
	  discarding them. This is needed because there is a long delay 
	  between opening the socket and the first read (due to ui startup, 
	  audio device startup, etc...).  As a result lots of audio was 
	  getting buffered at receiver with continuous sources at start up.
	- Fixes to linux audio driver
	- Fixed initial playout delay estimation.
	- Stopped interaction between skew ajustment and repair.
	- Assorted bug fixes to the UI
	- Attempt to get participant highlighting correct
	- Blending on samples dropped because of skew not quite there.
	- Added unsigned 8 bit to device formats and added to conversion
	  routines.
	- Partial SoundBlaster support on FreeBSD (out s16, in u8).
	- Initial code for the move to using the RTP library - not used yet
	- WAV file MAKEFOURCC bug fixed for non-Win32 little endian machines.
	  All platforms should be able to read WAV and AU files.
	- Changed skew adjust back so it invokes repair to cover would-be
	  gap.
	- Enhanced sample drop during skew detection.  Period that can be
	  be dropped determined by pattern matching samples in frame to
	  to have samples dropped in.  Then blend in start of what would
	  have been played to match point.  Quality improvement
	  more than compensates for extra cpu cycles.
	- Redundancy encoder working again (95% confidence :-).
	- Added dismiss button to file dialog.
	- Add support for DES encryption to the mbus (on by default).
	- Fix encryption. 
	- Fix UI codec failure when changing sampling config and no codec
	  with same friendly name exists.  Also fix for redundant codec
	  displayed when primary updated.
	- Add timestamp to mbus packet headers
	- Reorder initialisation sequence, to handle the new requirement
	  that the mbus can only send reliably to those entities we've
	  received a message from previously. This probably means that 
	  the wait-on-startup hack for ReLaTe is broken, but we don't care
	  since the entire initialisation sequence needs redoing anyway 
	  (the Bremen PSTN-to-Mbone gateway needs more fine grained control
	  than is possible with the current startup sequence).
	- Fixed size of participant window so user can see 2 layers of 
	  encoding when redundancy in use.
	- Fixed channel coder reconfigure issue when changing sampling mode.
	- The name of the RAT binary now includes the version number.
	- Fix muting of individual participants.
	- Added G726-16 using code from Marc Randolph's patch on comp.speech.
	- Add sdr plugin (not tested)
	- Fixed decoding preference problem when redundancy used.
	- Fixed source muting (we discard the packets in statistics, rather
	  than not mixing source).
	- Removed -drop command line option.
	- Moved codecs that had CODEC_PAYLOAD_DYNAMIC into unassigned and 
	  dynamic range so they are available in transmission panel.
	- Removed half-duplex cards appear in devices list on Win32.
	- Fixed ui update when switching channel coder.
	* Released 14 May 1999

v4.0.2  - Added message box for win32 users who launch with no command line
	  options, giving usage instructions.
	- Added WSACleanup() to shutdown win32 sockets nicely.
	- Fix inet_aton(), which prevents a crash on startup on Windows NT
	  service pack 3 and some versions of Windows 95.
	* Released 15 May 1999

v4.0.3  - Fix compilation with v1.2 of the Microsoft Research IPv6 stack.
	- Catch WM_DELETE for preferences window. Reported by Atanu Ghosh.
	- Check address and port is valid in command line parsing so
	  we don't dump core later.  Reported by Lorenzo Vicisano.
	- Unrecognized options are ignored, but no longer stop processing other 
	  options. This means that "rat-4.0.3 -f pcm -t 64 224.1.2.3/1234" now 
	  sets the TTL correctly, even though the -f is unrecognized, meaning 
	  that we can start rat via sdr. Reported by Terry Gibbons.
	- Fix occasional assertion failure in mix.c:133 `len < ms->buf_len' 
	  (it looks like the condition in the assert is off by one, should
	  be 'len <= ms->buf_len'. If this doesn't fix it, there's a check
	  around line 242 of mix.c which sets things up so we don't overflow
	  the buffer - we just decrease the amount we write by one sample
	  there instead. Reported by Roy Bennett and others.
	- Fix payload type mapping for PCMA-16K-Stereo.
	- Mapping for existing rat/vat codec names.
	- Primary payload type mapping using -f.
	- Secondary payload type mapping using -f also.
	- Fix restoration of previous audio port settings from saved values.
	- Separated sound file handling into sndfile_wav.[ch] and 
	  sndfile_au.[ch]
	- Added support for default sndfile extension if none specified.
	  Win32 -> wav, others -> au.
	- Added support for writing alaw and ulaw.
	- Updated man page.
	- Added sndfile_raw.[ch] for handling raw files.
	- Added sndfile_valid_format.
	- Fixed spurious debugging code in the windows audio driver which
	  would cause an infinite loop if activated (rare, but could happen).
	- Split converter code up into convert_*.[ch] and added convert_types.h 
	- Fixed repair when next unit on playout buffer is less than expected 
	  length away.
	- Compilation fixes for Irix 5.3 contributed by David Balazic.
	- Some linux audio drivers, in particular the ES1370 used in the 
	  SoundBlaster PCI 128, don't support /dev/sndstat. Rewrite the
	  query functions to work with /dev/mixer instead. 
	- Handle known skew in the audio sampling rate for those linux audio
	  drivers which report it.
	- Added 'high quality' converter, sinc filter.
	* Released 2 June 1999

v4.0.4  - Add ability to receive on one port yet send to another, requested
	  by Henning Schulzrinne. Usage: "rat addr/rx_port/tx_port"
	- Compilation fixes for Irix 6.5 contributed by Bob Olson.
	- Fix core dump caused by calling sndfile_write_audio with buffer
	  length of 0.
	- Fix core dump when called with invalid command line arguments,
	  reported by Marcus Meissner.
	- Fix passing of encryption key across mbus, reported by Ed Whelan.
	- Remove support for the -wait option. This means we can't work with the 
	  ReLaTe interface anymore but, once the split into multiple processes
	  is complete, ReLaTe will be able to drive the media engine directly.
	- Remove most of the transcoder support - it didn't work anyway, and 
	  needs to be reimplemented from scratch.
	- Start the split into multiple processes: media engine, user interface
	  and controller. 
	- Added basic layered channel coder. Usage: rat -l n <address>/<port>
	  <address>/<port>, where n = no of layers and there is an address and
	  port for each layer. Only works with the WBS codec at present, and
	  probably not very well at that.
	- Move the code to load and save settings into the media engine, and
	  out of the user interface.
	- Fixed WBS codec so that it works with different endian machines.
	- Added hash table to settings_load.
	- Fixes for 64 bit IRIX from Peter Orbaek
	- Some codecs' default payload values clashed with RTP spec:
	  PCM-16-K Stereo was 76, now 124; GSM-16K was 72, now 118;
	  GSM-32K was 73, now 119; GSM-48K was 76, now 123.
	* Released 7 July 1999

v4.1.0  - Fixed slow memory leak reported by Emmanuel Duros.  Vanilla 
	  decoder was not checking all pb_add functions returned success 
	  and so pointer was getting trashed.
	- Fix RTCP RR DLSR field when reporting on a source from which
	  we've received no SR packets.
	- Changed drop preference for incoming packet buffer overflows.  Now
	  drop incoming packet rather than earliest packet in buffer - makes 
	  source packet processing better ordered.
	- Remove address sanity checking for the layered coder from the
	  command line parser: it should be done when the socket is opened
	  since that makes the code address format agnostic.
	- Added network_exit.
	- Added per source gain control.  Now appears in participant panel 
	  on audio pane.  Suggested by Jerry Isdale.
	- Fixes for the Detexis Musica IPv6 stack on Windows NT.
	- Send rtp.source.cname messages to the video tool when lip-sync is
	  enabled, so it can match sources up to those in the video session.
	- Cleanup documentation of the audev_*.c files
	- Fix loading of preferences: powermeter, lip-sync
	- Fix the fileboxes on the Windows release version.
	- Fix spurious transmission of tool.rat.lecture.mode() mbus messages 
	- Fix spurious transmission of rtp.source.active() mbus messages when
	  we are sending.
	- Fix (hopefully) assertion failure in the mixer when trying to zero 
	  blocks longer than the buffer.
	- Improved layered channel coder, thereby making it incompatible with
	  4.0.4.
	- Added persistent database pdb.[ch].
	- Moved render_3D_data from the rtcp database into persistent database
	  as first step towards using common rtp code.  Lots of other fields to
	  do, but looks to work in principle.
	- Add RPM spec file for RedHat Linux. Based on that contributed by
	  Stephen Adler
	- Convert to using the RTP code in the common library.
	- Fixed getenv not checking for null string after getenv in 
	  auddev_osprey.c.  Reported by Dirk Kutscher 
	  <dku@informatik.uni-bremen.de>.
	- Add playout_calc.[ch].
	- Add ts_mul and ts_div to keep playout calc clean.
	- Change default units per packet to 1, since PIPVIC2 says that the
	  end to end delay is too large otherwise.
	- Rewrite the reception quality matrix
	- Consistency changes: audio device and repair query now consistent 
	  with the majority of interfaces.
	- More consistency changes: channel_get_coder now consistent with
	  other interfaces.
	- Add ability to mtrace between participants by clicking on the squares
	  in the reception quality matrix
	- Split setting load between .RTPdefaults and .RATdefaults.
	- Add cutesy popup boxes to the reception quality matrix
	- Add device buffer occupancy check to auddev_luigi to try to fix
	  sb16 problems.
	- Add toged and cont_toged back.
	- Move source_check_buffering into rtp_callback.
	- Fix mixer handling overlapping blocks.
	- Changed win32 audio read and write methods, added audio event to
	  reduce CPU usage.  Fixed handling of non-blocksize writes.
	- Modified auddev_null on Win32 to use WaitForSingleObject rather
	  than Sleep on Win32.  Reduces CPU considerably.
	- Added colour to powermeters.
	- Add fixed size packet buffer to make for easy discards when media 
	  engine has blocked for an inordinate period.
	- Added source_get_bps (bits per second :-).
	- Removed gain setting stuff from luigi_audio_open.  Problems reported
	  by Bill Fenner and code purpose is defunct.
	- Complete split into media engine, user interface and controller
	  processes. Only works on Unix at present, still debugging the
	  Windows port.
	* Released 5 October 1999

v4.1.1  - Added transcoder audio device.  Contributed by Michael Wallbaum 
	  <wallbaum@informatik.rwth-aachen.de>
	- Now send RTP BYE packets on exit
	- Fixed converter table initialization.  Was not initializing 
	  interpolation and extrapolation correctly after last consistency 
	  update.
	- Unrolled loops in extrapolation converter.  
	- Replaced divisions in interpolation converter that were not powers
	  of two with a multiplication scaling factor and a power of 2 
	  denominator.  Improves performance 3-4 for upsampling by 3,5,6.
	- Fixed 2 step sample rate conversion support.
	- Fixed Windows port.
	- Initialize sp->cur_ts in main_engine, before tx_start.  Should fix 
	  occasional assert failure during transmitter startup.
	- Put cont_toged check into source_process packets.  Added check for
	  cont_toged in rtp_callback.
	- Added time constants and replaced lots of ts_map32 functions with
	  them in source.c.
	- Rewrite initialization and rendezvous code between the controller
	  and the user interface and media engine. This should remove many
	  race conditions and generally clean things up and make them more
	  reliable.
	- use mbus_rendezvous_waiting() and mbus_rendezvous_go()
	- do a real handshake between controller and the subprocesses
	- Fix parsing of many command line options. NOTE: many of the options
	  are parsed in a backwards incompatible manner now, to make the code
	  cleaner. 
	- Tuples for mbus command switches in mbus_{control,engine,ui}.
	* Released 2 November 1999

v4.1.2  - Added echo suppression back, it fell out of code base at some point.
	- Fix ui sending input and output gain values before it's got values
	  from audio engine.  Was happening because scale widget invokes 
	  -command option when created.  Now we don't send gain info until
	  we've heard some from engine.
	- Add signal handler to main_control.
	- Add back default port 5004.
	- Fix luigi_audio_drain to reset, sync, and trigger reading.
	- Moved packet reception handling from rtp_callback to 
	  source_add_packet.
	- Changed playout buffer length estimation to use transit times and 
	  average transit.  Previously used observations of playout buffer 
	  lengths, but this maybe unreliable due to packet losses.
	- Revised skew adaptation.  Now compares transit time in use with 
	  transit average, instead of monitoring playout buffer length which 
	  is tainted by packet losses.
	- Playout variable now refers to playout offset only.
	- VAT style reduction in playout estimate when marker bit set.
	- Playout recalc trigger when jitter > playout.
	- Device reconfig no longer resets ports if device unchanged.
	- Shuffled front panel.  Added bps field.
	- Changed source processing to use next_played rather than last_played.
	  Cleans some processing issues up and allows for codecs with 
	  non-constant frame durations.
	- Added source_get_skew_rate to return relative skew.
	- Added skew to stats panel.
	- Made stats panel consistent with UI.
	- Split reception panel into Decoder and Playout.
	- Added note field for local user.
	- Playout calc revisions:  only update transit if:
		(a) source is new.
		(b) source quiet for longer than transit_reset period.
		(c) playout re-calc triggered by toged_cont.
	- toged_cont is now no. thrown on ground in last 8 packets.
	- Global change of integer types, and removal of compatibility defines:
		u_int8  -> u_int8_t
		u_int16 -> u_int16_t
		u_int32 -> u_int32_t
	  and the same for the signed varients.
	- And change them all to be uint32, etc
	- Packet spikes now accounted for in packet processing.  Transit 
	  average and playout not updated during spike period.
	- Playout panel now displays spike info also.
	- Mbus updates from Markus Germeier, which fix compatibility with
	  the Uni Bremen implementation:
		- mbus.bye() signals an entity when it leaves. (mbus.quit 
		  commands an entity to leave)
		- Generate and work with 64 bit DES keys. Generate parity bits
		  for key.
		- mbus_heartbeat(m, 1) should send an mbus.hello() every 
		  second and not every two seconds
		- HMAC-MD5 should read HMAC-MD5-96
		- Honor the MBUS environment variable for the MBUS 
		  configuration file
		- The TZI Mbus implementations generate spaces in 
		  configuration file, which mislead the UCL Mbus to dump core.
		- Allow Spaces in .mbus
		- Address must be unique
		- Delete failed reliable messages if err_handler is present
	- Fix uninitialized use of session->last_update in RTP code
	- Fix assertion failures in RTP library when sending RTCP packets
	- Recognize, but ignore, profile specific extensions to RTCP SR/RR 
	  packets
	- Update RTCP reporting interval calculation
	- Implement reverse reconsideration of RTCP reporting interval
	- Implement encryption in the RTP code
	- Fix redundancy decoder
	- Fix path searching when starting subprocesses
	- Add patch matching segment insertion when correcting for slow sources
	- Migrated Makefiles and Headers to use autoconf
	- Fix crash on exit
	- Fix system name in tool version
	- Add version number to mbus config file. Should prevent crashes
	  when we update the format of that file.
	- Set thread priority of media engine to THREAD_PRIORITY_ABOVE_NORMAL 
	  on Windows.
	- Fix CPU usage of UI
	- Update to common library 1.0.10
	* Released 13 December 1999

v4.1.3  - Rename common library to libuclmmbase.a on unix
	- Update RPM spec file
	- Update installation instructions
	- Fix "-t" option, to set the TTL. The v4.1.2 release would always
	  send with TTL=15, making it useless for wide area.
	- Fix assertion failures in source.c
	- Cleanup code for updating SDES items in the UI
	- Remove core files & directories on "make distclean"
	- Add check audio in adjusted playout point packets can be played out
	  and correct if would be dropped.
	- Add failure return option for pattern matches used in skew detection.
	- Add session_validate(), source_validate() and pdb_item_validate() 
	  debugging routines. These mostly just check a magic number stored
	  in the structure, to check that the structure hasn't been totally
	  corrupted. There's a lot of scope for extending them in future...
	- Waveform repair schemes now use last available native coding.
	- Minor decode path change.  Decode, sample rate conversion, 3d 
	  rendering, gain scaling, are all performed if no native coding
	  available rather than individually test.  Fixes problems introduced
	  with media_data duplication in playout buffer shift code.
	- Various fixes for potential buffer overflows
	- Update copyright notices to welcome in the new millennium
	- Fix mbus address of the media engine (had a spurious space in 
	  instance:pid)
	- Remove lip-sync command line option and toggle button in the user 
	  interface.
	- Automatically enable/disable lip-sync when we detect a video tool 
	  via our mbus interface.
	- Rewrite the lip-sync code to send our playout delay to the video 
	  tool. We send "rtp.source.playout(ssrc delay-in-ms)", which might be
	  sufficient.
	- Added back backwards compatible name mappings for command line 
	  parsing.
	- Robustified command line parsing of codecs.
	- Fix ordering of command line argument parsing.
	- Accept -crypt as a synonym for -K.
	- Add encrkey field to the session struct.
	- Fix encryption key handling at startup.
	- Fix saving of audioChannelParameters setting.
	- Fix use of mbus structure after freeing it.
	- Fix freeing of memory on shutdown.
	- Add assorted debugging code.
	- Fix reception of RTP packets with padding (this is what caused a
	  crash when using the GSM codec with encryption).
	- Update for changes in the mbus library.
	- Add format conversions for a-law devices.
	- Incorporated auddev_atm.c and wrote auddev_atm.h.
	* Released 31 January 2000 [X3D SDK CDROM]

v4.1.4  - Fix dependencies in Makefile for rpm target
	- Fix handling of odd numbered RTP ports (pick next lower port).
	- Fix loopback setting load and carry thru when changing device.
	- Fix -version command line option.
	* Released 1 February 2000

v4.1.5  - Add support for ALSA audio drivers on Linux, contributed by
	  Robert Olson of Argonne National Laboratory.
	- Add rtt calculation to rtp_callback.c.
	- Fix don't start mbus and processes if no command line args given.
	- Add controller validation of command-line addresses before 
	  launching sub-processes.
	- Fix controller only quits upon hearing byes from one of it's 
	  children.
	- Add playout buffer debugging/validation code.
	- Add engine parsing of sdes commands accept my_ssrc as an acceptable
	  ssrc identifier.  Then control does not need to know to set name,
	  etc, from command line.
	- Add check in settings that sdes values read in are not already set.
	- Add filter on rtt values to avoid displaying garbage values from rtp
	  misimplementations.
	- Fix participant muting (reported by Bob Olson).
	- Correct size of pb->buf allocation in pktbuf.c.
	- Fix slow memory leak in pktbuf (didn't free pb->buf).
	- Removed unused variables from the session struct
	- Add checks for sp->ui_on to the user interface dispatcher code.
	  This has currently broken the user interface, but is needed to
	  make the initialization sequence reliable.
	- Major update to the initialization sequence
		- media engine solicits for the user interface
		- user interface requests state information from the media
		  engine
		- split ui.c into ui_send_*.c
	- Load/save the note setting.
	- Remove all traces of operating specific defines.  Configure no longer
	  subsitutes operating system.
	- Fix @OSTYPE@ in SDES Tool field (broke with latter).
	- Add SIGCHLD to main_control to shutdown process tree when child 
	  processes die in a timely manner.
	- Fix shutdown sequence.
	- Patch G711 A-law, Sun's reference implementation had a few glitches 
	  and you only notice when speaking with real h/w implementations.
	- Update README.mbus, giving documentation of how to remote control
	  RAT using the message bus interface, and how to replace the user
	  interface by one of your own choosing.
	- Fix two step sample format conversion assertion failure.
	- Fix device sample conversion layer freeing already freed pointer
	  when swapping between devices with and without layer.
	- Add RTCP DELETE messages trigger deletion corresponding pdb entry
	  and source (if relevent).
	- Tweak source ageing, to fix problems with users being hilighted
	  after they finish speaking. Problem occured if less than 50
	  packets arrived in a talkspurt.
	- Save settings now passes through settings file, replacing entries
	  RAT has update values for, and then appends entries not in settings
	  file. Requested by Frode Kileng.
	- Add use of username when no RTPdefault exists for participant name.
	- Fix handling of audioInputMute in RATdefaults. Reported by 
	  Frode Kileng.
	- Fix rtp.source.active for local user in response to rtp.query(), 
	  didn't highlight local users transmit state.
	- Update SDR plugin.
	- Transfer string associative array code out of settings.c into new 
	  common library files asarray.[ch].
	- Fix -allowloopback option.
	- Add controller checks for child process termination on Win32. 
	- Add message loop and watch for WM_DESTROY to controller.  Will now
	  shutdown engine and ui when Win32 task manager sends these.  NB,
	  Visual Studio debugger terminates processes and still leaves
	  children running.  AFAIK there is no way for children to spot this,
	  i.e. WaitForSingleObject watching process does not appear to work
	  for children watching parents.
	- Use _vsnprintf for snprintf functionality on Win32.
	- Fix "make tgz" target to include the sdr plugin 
	* Released 14 February 2000

v4.1.6  - Fix Windows port
	* Released 15 February 2000

v4.1.7  - Update ALSA detection in configure script
	- Fix display of names when SDES NAME is not known (fallback to
	  CNAME or SSRC in that order).
	- Add support for the rtp.addr.query() mbus message to media engine.
	- Remove support for the tool.rat.codec.details() mbus message.
	- Add support for the
		tool.rat.codecs.request()
		tool.rat.codecs.flush()
		tool.rat.codecs.add()
	  mbus messages.
	- Add audio clip support to balloon help.
	- Tidied mixer interface and cleaned session_struct out.
	- Fix mixer wrap handling for multi-channel.
	- Fix default device if not present in saved settings.
	- Add support for rtp.source.mute ("ALL" 0|1), which is much more efficient
	  than iterating all sources over the mbus.
	- Fix handling of WM_DELETE_WINDOW for the about box
	- Fix handling of WM_DELETE_WINDOW for participant popup boxes
	- Remove all "Dismiss" buttons - just close the windows instead
	- Add balloon help button to the main window, moving the application name 
	  to the title bar
	- Partially fix processing of Tcl events in the UI during the rendezvous
	  with the media engine
	- Partially fix audio processing on windows 2000
	* Released 24 February 2000

v4.1.8  - Partially fix layered codec (recognises multiple addresses on
	  command line but doesn't send to all of them)
	- Slay zombies when children die
	- Add repeat+ repair type.  It does standard repeat and then reverse
	  blends samples from the end of the previous frame with the start
	  of this frame.  Perhaps reduces some glitching.
	- Add README.reporting-bugs describing "How to Report Bugs Effectively"
	- Update to the ALSA support, contributed by Bob Lindell of ISI
	- Add local tone generator.
	- Add pointer to the RAT users mailing list <rat-users@cs.ucl.ac.uk>
	  an open list for discussion of issues relating to RAT.
	- RTT calculation fix (ntp32_sub was broken).
	- Add checks to main processing loop for liveness of controller 
	  process.
	- Add checks for sibling and parent process in ui and engine so proc's
	  don't wait for messages that are never going to come.
	- Change skew correction code to use audio_blend rather than repeat 
	  procedure in situ.  Fixes a memory corruption bug.
	- Add null handler for RX_APP to rtp_callback, circumvents abort call.
	- Add linear8 to sndfile import and export functions.
	- Add handling to sinc_converter for conversion chunks that are 
	  smaller than hold buffer size.  We can now convert a sample at a 
	  time if desired rather than blocks of at least the duration of the 
	  hold buffer.
	- Add manual silence detection algorithm.
	- Add UI and settings for silence detection algorithm alternatives.
	- Add sample rate and channel converter to sound file importer.
	- Add support for m*11025 native encoding types.  Although RAT does
	  not support these sampling rates, one of the sample rate converters
	  supports conversion from m*11025 to n*8000.
	- Fix noise during sample insertion from skew adaptation.
	- Fix fatal mix error when downsampling.  Trim length did not
	  compensate for the sample rate, causing trim length to wrap and
	  attempt to remove zillions of samples.
	- Fix mixer rounding error in timestamps vs samples.  Would 
	  periodically fail mix_verify by 1 otherwise.
	- Add installer for the Unix ports. This displays the copyright notice, 
	  and gets confirmation from the user that they accept the conditions,
	  before extracting the .tar.gz file. Use "make installer" to build and
	  then distribute the rat-4.1.8-installer binary only.
	- Add transit estimate reset when device format changes at either end.
	- Add settings to default to first device when device in settings file
	  does not exist on host.
	* Released 20 March 2000.

4.1.9   - fix configure script handling of X_CFLAGS during broken header tests
	  reported by Atanu.
	- remove variable component (e->playout) from spike threshold in 
	  source processing, believe this is part of the bad spike messages 
	  (160>nnnnn).
	- Changes to Win32 driver read code to remove race condition (audio
	  callback occuring as audio being taken from list of ready buffers).
	- Move participant database clean up into session_exit.
	- Destroy local file player and tone generator when reconfiguring
	  device.
	- Changes to Win32 driver interface write code.  Appears to resolve
	  problems with a few troublesome cards.
	- Replace puts with bgerror where appropriate in ui script.
	* Tagged 30 March 2000 (not released).

v4.2.0  - Fix mixer writing to new head position so it mixes multiple streams
	  again!
	- Fix phase skips of tone generator.
	- Add RTP flush after device reconfiguration.
	- Clean up and fix controller command line parsing.
	- Remove timers.[ch] to give a single internal representation of time.
	- Add creation time to mix_create and remove mixer re-sync function.
	- Fix MMX mix bounds calculation.
	- Add sample rate factor into cushion.
	- Change initial cushion value to be more conservative.
	- Fix entries in cushion histogram being entered when less than
	  minimum allowed cushion.
	- Remove second audio_rw_process in engine main loop.  Process only
	  blocks for audio nowadays.
	* Released 5 April 2000.

v4.2.1	- Fix handling of 8bit PCM .WAV files (unsigned 8bit, not signed).
        - Fix shuffle participants not to print newline if last entry.
	- Fix parsing of -t flag, was mis-coded as -ttl in layering command
	  line parsing changes in 4.2.0.
	* Released 6 April 2000.

v4.2.2  - Catch mbus initialization failure and warn user in ui, control, 
	  and engine.
	- Round ttl to maximum of 127.
	- Change parse_args_early in main_control.c to pass over all args.
	  (problem reported by Didier Lambllais).
	- Add -N,-E,-P,-L for sdes item setting from command-line (name, email,
	  phone, location).
	- Fix -f codec communication from controller to engine.
	- Break controller address and option parsing so options are not
	  communicated to engine before rendezvous has occurred.  Otherwise
	  options can be lost due engine's settings_load occuring
	  after rendezvous.
	- OSS (Linux) audio driver updates from Stuart Levy <slevy@ncsa.uiuc.edu>
		- Should make mixer controls work with the Sound Blaster 
		  Live! cards
		- Update behaviour of the line-in controls, to hopefully 
		  work better with AC97 compatible cards, if you see a 
		  difference in the behaviour of line-in please let us know.
		- Update behaviour of output gain controls, to set all outputs. 
		  This probably needs more work, let us know if it breaks anything
		  (the entire OSS audio output control is broken - we need to deal
		  better with the output mixer...)
	- SGI audio driver updates from Stuart Levy <slevy@ncsa.uiuc.edu>
		- Add "APanel" input source. When selected, rat carefully
		  avoids ever specifying the input source to the audio
		  library; thus whatever source is selected by the SGI
		  "apanel" program applies. Use this if you have problems
		  with rat switching to the built-in microphone on O2
		  machines.
		- If anyone could contribute support for the new (post
		  Irix 6.3) audio library, we'd be grateful... 
	- Fix library ordering in makefile (reported by Stuart Levy)
	- Document configure options which may be needed to build on Irix 
	  in INSTALL.TXT
	- Fix RTP SR timestamps and RTP jitter calculation. Reported by
	  Robert Olson.
	* Released 17 April 2000.

v4.2.3	- Change the session struct to be a two element array in main_engine.c
	- Change all references to sp-> into sp[0]-> in main() of the media engine
	- A few trivial changes to the initialization sequence, in support
	  of the transcoder. There are a number of things which will hinder
	  this:
	  	- mbus_rendezvous_waiting() and mbus_rendezvous_go() are
		  incompatible with another mbus needing to run in parallel.
		- some parts of the code are non-reentrant (i.e. use static
		  variables) and will have to be fixed
	  it's probably not too bad, though. All are wrapped in #ifdef
	  IS_TRANSCODER, for now.
	- Update OSS (i.e. Linux) audio device detection code
		- Better card detection and probing
		- On systems where the audio driver supports two half-duplex 
		  devices, rather than a single full-duplex device (e.g. the
		  NM256) we attempt to pair them into a form usable by rat.
		  Reported by Thierry Turletti.
	- Workaround failure to get host IP address when setting the RTP
	  CNAME (we use user@127.0.0.1 as the CNAME in this case). Reported 
	  by Joe Love.
	- Add support for "-logstats" option, to log RTP and codec statistics
	  to a file. Requested by Anna Watson.
	- Update "make installer" target
	* Released 3 May 2000

v4.2.4	- Further updates to OSS (i.e. Linux) audio support
	- Further updates to support the transcoder
		- Add -T option to media engine to enable transcoder operation.
		  This will need a custom user interface and controller.
		- Remove the IS_TRANSCODER defines.
	- Update installation instructions
	- Fix buffer overflow when reading from "No audio device". This was
	  the cause of the
		"playout.c:86: pb_validate:Assertion 'pb->magic==0x83838383' 
	  error reported by Christa Wenger, Marcos Bedinelli and Fang Yuan.
	- Remove spurious calls to tx_process_audio() and tx_send() from
	  the main loop.
	- User interface hides the main window is hidden until it is finished
	  initialising. This should prevent errors due to manipulation of the
	  user interface before the system has finished initialisation, as 
	  reported by Peter Parnes.
	- Fix reception of packets with an RTP header extension. Reported by 
  	  Peter Parnes.
	* Released 16 May 2000

v4.2.5	- Fix assert failure in auddev_trans.c, contributed by Jakub Segen
	- Fix audio device pairing with the OSS driver, in those cases where
	  there is only a single mixer channel. Reported by Thierry Turletti.
	- Rewrite code to rendezvous with the controller in the media engine,
	  to support future transcoder operation.
	- Cleanup main_control.c - code moved into process.c and cmd_parser.c
	* Released 16 June 2000

v4.2.6	- Compilation fixes for Win32 contributed by Marcin Michalak.
	- Merge ALSA fixes from Robert Olson
	- Merge OSS fixes for AC97 based soundcards from Robert Olson:	
		"I went ahead and did the (more) proper fixes for AC97
		cards. I don't have a reliable way to detect which cards
		are AC97-based, so there's currently a hard-coded list of
		them in the OSS code:
		   "OSS: AudioPCI 97 (STAC9708)",
		   "OSS: AudioPCI 97 (CRY13/0x43525913)",
		These two are the one that Jay told me about and my ES1371
		card. If you have a card that you think this will help for
		(probably SBLive falls into this category, and any of the
		ES1371 based cards), you can force rat into AC97 mode by
		setting the OSS_IS_AC97 environment variable to 1, in bash:
			 export OSS_IS_AC97=1
		or in tcsh:
			 setenv OSS_IS_AC97 1
		If this makes things work for you, please mail the output of
			 cat /dev/sndstat
		to rat-users@cs.ucl.ac.uk and we can add your card."
	* Released 30 June 2000

v4.2.7	- Split main() in the controller into two parts: one for normal
	  operation as an audio tool, one for operation as a transcoder.
	- Add extra debugging code to transmit.c
	- Fix buffer overflow in tx_unit_create()
	- Add missing initialization of reversal buffer for repetition plus
	  (problem reported by Lingfen Sun).
	- Only perform audio bias removal for live capture, not when reading
	  from a file. This fixes the problem reported by Paxton Smith, where
	  excess RTP packets are sent when playing from a file.
	  	- Note that there are still some extra packets sent, due to 
		  the pre- and/or post-hang in the voice activity detection
		  code. This makes for a much nicer sound for speech (since
		  we have continuous audio during short (<200ms at present)
		  silence periods, but does cause some spurious transmission
		  in this case. The following #define's in parameters.c can
		  be changed to adjust this behaviour:
			#define VAD_SIG_LECT     40
			#define VAD_SIG_CONF     60
			#define VAD_PRE_LECT     60
			#define VAD_PRE_CONF     20
			#define VAD_POST_LECT   200
			#define VAD_POST_CONF   200
	- Fix reading of .wav files 
		- The last unit of audio data was not being sent, which also 
		  contributed to Paxton Smith's problem.
	- Fix bias removal for stereo signals.
	- Fix bias removal maintenance ( >> != / ).
	- Fix arithmetic overflow problems in the bias removal code.
	- Remove debugging code in auddev_null.c which caused the NULL audio
	  device to (incorrectly) generate non-silent a-law audio.
	- Detect whether FreeBSD driver is newpcm or luigi.  There are
	  differences in detecting full duplex caps and handling mixer 
	  settings.  See comments in auddev_luigi.c for an explanation.
	- Change ts_t to timestamp_t.  ts_t is a system defined type
	  on Solaris8.
	- Cleanup and document the receiver source processing (starting 
	  with the source_process() function and friends).
	* Released 6 August 2000

v4.2.8	- Mbus address of media engine now contains session identifier (this 
	  is always zero when running as an audio tool, but the transcoder 
	  has sessions 0 and 1).
	- Fix bug in mbus transmit code which would cause messages to have the
	  same sequence number as the preceeding ack.
	- Update controller initialization sequence to handle transcoder.
	- All fatal_error() messages are now prefixed with the RAT_VERSION
	  text, rather than the argv[0] application name.
	- Fix RTP header extension code (patch from Jori <jori@lumumba.luc.ac.be>)
	  as part of RTP interoperability tests.
	- FreeBSD 4.1 doesn't support binding to an IPv6 multicast address, must
	  bind to in6addr_any instead (reported by Shirasaki Yasuhiro).
	- Fix validation of RTCP packets with padding (reported by Vladimir Brauner).
	- Fix padding octet count in RTCP packets sent with padding (Vladimir Brauner).
	- Fix RTCP timer reconsideration (Vladimir Brauner).
	- Minimal transcoder support: the media engine can transcode, although it 
	  is very buggy. The controller supports just enough to test the media 
	  engine, the UI does not support transcoding yet.
	* Released 22 August 2000

v4.2.9	- Fix spurious assertions in source.c -- I think (hope?) this will fix the 
	  source.c:1588: source_process: Assertion `md->nrep < 5 && md->nrep > 0' failed.
	  crash reported by Bob Olson and Justin Pietsch.
	- Add other_session field to the session struct, and fix transcoder to make
	  calls to rtp_add_csrc() and rtp_del_csrc(), which will eventually support
	  pass through of RTCP. This changes the size of the session struct, and
	  means that you'll have to "make clean" after updating.
	- Modularize user interface: bargraph and chart
	- Fix labelling of transcoder input/output ports
	* Released 8 September 2000

v4.2.10 - Added auddev_newpcm.[ch] with newpcm style mixer control.  Includes
	  support for loopback and preservation of mixer settings so mixer is
	  returned to original state upon exit. [oh]
		  Note: both driver interfaces (luigi, newpcm) are pulled in
		  by configure and this is possible whilst luigi pcm and
		  newpcm maintain the same function call interface.  This may
		  break in future.  Both are necessary because mixer operation
		  is not consistent between the two interfaces.
	- Catch SIGTERM and SIGHUP on unix (Robert Olson)
	- Update Windows project file ratui.dsp (Paxton Smith)
	- Add extra debugging code to sanity check the playout time: since
	  4.2.9 fixed the md->nrep assertion failures in source.c, we now
	  see problems due to poor playout time calculation.
	- Compilation fixes for installer under Red Hat 7's weird gcc varient.
	* Released 1 November 2000

v4.2.11	- Fix port restoration when changing device (Shirasaki Yasuhiro)
	- Remove assert that made duff port restoration core dump in newpcm, 
	  just check port is correct range, if not print warning.
	- Add check that requested sampling rate is within device
	  capabilities on newpcm.  Driver does not do bounds checking when 
	  setting rate and will mis-report actual rate when a duff rate has 
	  been set (reported by Lars Eggert).
	- Break up newpcm writes to n separate writes of blocksize rather
	  than one big write as current newpcm debug code complains if
	  this is done.
	- Fix erroneous debug messages for gain settings in newpcm.
	- Change newpcm_audio_open() to use SNDCTL_DSP_SETFRAGMENT from 
	  SNDCTL_DSP_SETBLKSIZE.  The latter creates just 2 frags in 
	  secondary buffer and this is not enough for cushion fill levels.
	- Fix breakage due to "gcc 2.96" shipped with Red Hat Linux 7
	- Remove inline math routines from cx_gsm.[ch] since they broke
	  with some versions of gcc (signed vs unsigned comparison). 
	* Released 8 January 2001

v4.2.12	- Better diagnostic message when mbus config file cannot be locked.
	- Remove SNDCTL_DSP_SYNC from newpcm reset function as it can block.
	- Change G726-32 packing to fit AVT profile.  Because of packing order
	  difference this and future versions will not be interoperable with
	  previous RAT releases.
	- Set initial cushion size to zero to force audio_rw_process to write
	  an empty cushion of audio after device is opened.
	- Change mix_put_audio return value when gap detected.  Was returning
	  TRUE which indicates mixer and source formats not compatible, this
	  caused source media and channel buffers to be wrongly flushed.  
	- Remove debug sample that made concealment distinctly audible.
	  Should not have gone into code base.
	- Re-jig source_process so that frames generated through loss
	  concealment get passed onto mixer.  This must have broke during
	  tidying.
	- Remove repeat+ from loss concealment schemes, didn't add anything.
	- Add L8 encoding.
	- Change pktbuf to use playout buffer (better use of code).
	- Reset transit histories when source created.  This should fix:
		- Sometime see messages of the form: 
			"source.c:767 Spike (160 > 2708390)"
		  and hear nothing.
		- If a receiving stream contains a lot of jitter and 
		  out-of-order packets it crashes sometimes. (jitter around 
		  1000 ms and random out-of-order). Reported by Peter Parnes.
		- Occasional crashes due to "too many blocks allocated" with 
		  many allocations in channel_types.c:29, channel_types.c:34, 
		  playout.c:188 and source.c:893.  Seems to be due to packets 
		  being received off the network, but never played out. 
		  Reported by Angel Mateo, and also observed at UCL. Not 
		  something I've found a way of reproducing.
		- Occasional weird behaviour of the playout buffer, where 
		  sources are buffered for several _minutes_ before being 
		  played out? Reported by Steven Dawson on the CAIRN list. 
	- Reduce source packet buffer size.
	- Change all G726 packings to gratuitously painful little endian.
	- Increase initial cushion estimate to 2 * min cushion size.
	- Enable 11025 sample rates and codecs.
	- Fix detection of incompatible conversions, M * 11025 <-> N * 8000
	  is not supported.
	* Released 14 January 2001.

4.2.13	- Fix spurious assert in G726 decoder
	- Fix G726-40 unpack (bits per sample mis-represented as 3!)
	- Fix sinc converter selection bug.  Should fix:
		(a) Crash in convert_sinc.c:
			convert_sinc.c:134: failed assertion
			convert_sinc.c:122: failed assertion
			util.c:87: _block_alloc: Assertion `size > 0' failed.  
		    reported by Greg Herlein.
		(b) Another crash in convert_sinc.c:
			convert_sinc.c:1107: sinc_upsample_stereo: 
			Assertion `out == dst + dst_len' failed.
 		    reported by <thavaqos@yahoo.com>
	- Change linear converter to use same structure arrangement as sinc
	  converter and fix linear converter selection bug (derived from 
	  convert_sinc.c).
	- Fix reverse reconsideration of RTCP interval
	- Mark entries in the RTP database as active senders if we get an
	  RTCP SR packet from them.
	- Added support for G.728 encoding and decoding.  See README.g728
	  for build instructions and comments.
	- Add support for non-AC97 mixers on FreeBSD's newpcm.
	- Add support for new device names under FreeBSD-5 (audio<m>.<n>)
	- OSS/Linux audio driver now uses /dev/dsp rather than /dev/audio,
	  since some drivers don't support /dev/audio.
	- Cleanup OSS/Linux audio driver code
	- Added check for device busy vs device not configured in newpcm
	  probing.
	- Fixed re-use of "thedev" string in newpcm_audio_open that caused 
	  device name to be trashed.  Reported by Lars Eggert.  Thanks!
	- Fix RTCP bandwidth fraction estimation (Timur Friedman)
	- Add support for displaying RTCP SDES PRIV items (we send a test
	  item also).
	* Released 3 March 2001

v4.2.14	- Add "-noui" option to start the media engine without user interface
	- Fix CSRC handling (Patch from ivo.sedlacek@anfdata.cz)
	- Fix uninitialized field in SDES PRIV handling (Robert Olson)
	- Fix G726-24 and G726-40 out by one in codewords unpacked.
	- Rename rtp_callback to rtp_callback_proc to avoid name clash 
	  following recent common library changes.
        - Fixed all MSVC warnings about implicit type conversions.
        - Update MSVC project files to reflect new paths in common library.
	- Update configure to remove need for patching in FreeBSD ports 
	  collection.
	- Add shell script target "rat" as target in Makefile.  Again to
	  make FreeBSD ports less effort, like zero :-D.
	* Released 6 April 2001

v4.2.15	- Fix common include path for use with standard Tcl/Tk (Pekka Savola)
	- Fix handling of undefined PATH variable, which caused a crash on
	  startup in some cases (patch from Pekka Savola).
	* Released 13 April 2001

v4.2.16	- Fix PATH handling
	- Fix so that we no longer send an RTCP BYE if we leave the session
	  before sending any other packets.  
	- Fix initial estimate of average RTCP packet size.  
	- Implement RTCP BYE reconsideration.  
	- Fix RTCP transmission interval.  
	- Fix handling of probationary sources in the RTP code (a source is
	  no longer treated as probationary if we've received RTCP packet
	  from it).
	* Released 24 April 2001

v4.2.17	- Fixes to auddev_oss.c from Robert Olson
	- Fix includes for <time.h> (should fix Red Hat 7.1 compilation)
	- Fix IPv6 support (Kris Hasler)
	* Released 18 May 2001

v4.2.18 - IPv6 fixes in common
        - Config for Windows2k IPv6 stack
        * Released 25 May 2001

v4.2.19 - Fix off-by-one error when removing half duplex audio devices in 
          the OSS driver (Eric Gerlach and Robert Olson)
	- Fix generation of rat shell script (David Martinez)
	- Fix assert failure with Repair=None (Jakub Segen)
	- Add initial IXJ audio driver for QuickNet devices (contributed
	  by James MacLean <macleajb@EDnet.NS.CA>)
	- Fixes to device pairing for SB Live (Robert Olson)
	- Fix complience with draft-ietf-mmusic-mbus-transport-06.txt
	  (patch from Lars Fischer)
	- Fix handling of EAGAIN in auddev_oss.c (James MacLean)
	* Released 21 June 2001

v4.2.20	- Fix forgotton malloc in auddev_oss.c mixer allocation (patch from
	  Eric Gerlach)
	- Fix configure script to detect PhoneJack audio devices
	- Add PHONE type to audio_types.h
	- Fix misquoted mbus message in ui_send_rtp.c (Eric Gerlach)
	* 24 September 2001

v4.2.21	- Fix FreeBSD support to work with 4.5-RELEASE (Orion Hodson)
	- Fix SoundBlaster Live! support on win32 (Chris Greenhalgh)
	- Accept (and ignore) unknown SSRC in SDES (Chris Greenhalgh)
	- Add . to path in rat script. Helpful for folks working on RAT
	  without dot in their path.
	- Change sample rate variable types from uint16_t to uint32_t
	  and add 96kHz timebase (Orion Hodson)
	- Update configure script to better support IPv6 (Suzuki Shinsuke)
	- Fix count of CSRCs in mixer
	- Fix compilation on alpha (David Martinez)
	- Fixes for MacOS X and Win32 (Bill May)
        - Explicitly set send/recv buffer sizes (1Mbyte)
	- Assorted performance tweaks to RTP code (Colin Perkins)
	* 15 April 2002

v4.2.22	- Fix compilation on Debian/SPARC (David Martinez/Petter Reinholdsten)
	- Reduce device buffer to 1/8 second on Win32 (Justin Binns)
	- Fix compilation on Win32
	* 23 July 2002



TODO -- They're features not bugs dammit!
----

- Assorted audio driver problems:
	- Broken auddev_pca with adding of audio interface conversion code.
	  FreeBSD 3.1 pca audio does not seem to work anymore.
	- SunVideoPlus interface code does not work properly.  The driver Sun
	  distributes only works in 8k mono.  MMAC produce the card and wrote
	  a driver for Sun: an exceptionally buggy and unstable driver. 
	  MMAC now has a better version now, but expect you to pay for the 
	  developer kit to access features.  $1200 for a card that does not
	  perform as advertised.  The marketing blurb states it has G728, 
	  G722, and "CD Quality" audio: it may do, but you can't access them.
	- SGI O2's don't have autosensing audio interface selection like 
	  Indy's.
	- OSS/Linux driver is buggy
		- Thierry still has problems with the NM256 card and device 
		  pairing
		- Some cards are not detected correctly
- Update the 3d audio rendering code
	- should decimate when using sampling rates other than that for which 
	  it was originally designed for.
	- could use finite precision maths for better performance (eliminate 
	  type conversions).
- sndfile routines should use conversion functions in audio_fmt.c, not their own.
- Channel coders need intelligent bail out fn's for when units overflow 
  (rare, but real problem).
- UI lets you select packets larger than the MTU, which can't be sent
- Reimplement transcoder.
	- Convertors need to be moved into the session struct, rather than being 
	  statically allocated.
	- settings load/save code needs to be updated for the transcoder. We should 
	  save settings into a different set of files, and save each side of the 
	  transcoder individually.
- Add support for conversion between m*8000 and n*11025
- Sample based codecs should accept frame lengths of 0-200ms, not just 
  multiples of frame lengths in tables.  Should add codec_is_sample_based 
  test and data to codec tables and relax receiver sanity checking so that
  sample based frames do not have to be n * codec__peek_frame_size.
- Should use the empty RRs coming back from receivers who are not seeing
  traffic, and display this in the loss matrix.
- Layering is broken (hasn't been updated for multiple processes, etc)
- Weird interaction with rat-3.0.x -- seems that we can generate packets 
  which some instances of rat-3.0 won't play. You get some people in the
  session who's tools report receiving your data, but they don't play it
  out. Suspect we're hitting the limitations of the old playout algorithm,
  rather than a specific bug, and the correct solution is to get everyone
  to update to rat-4.x
- Handling of SSRC collisions is broken
- Rework media engine initialization, so we can query it before sending
  rtp.addr()
- when rat starts it turns of "playback-mute" on the mic input which
  gives feedback for the mic. Muting it via the audio control turns of the
  feedback. Note that record-mute and playback-mute is not the same thing.
  Reported by Peter Parnes. [Depends on feedback settings and platform, oh].
- Add ability to change RTP sessions whilst rat is running. Requested by
  lots of people (Robert Olson, Joerg Ott, etc).
- When weak RTP validation is enabled, spurious sources can be
  added to the database (from the CSRC area of a packet?). Reported
  by Xiaotao Wu <xiaotaow@cs.columbia.edu>
- Crashes on WinNT if another version of Tcl/Tk is installed on the machine?
  Reported by Didier Lamballais <Didier.Lamballais@irisa.fr>.
- Excessive end-to-end delay on Windows? Gerry Wolfson. [The scale of delay
  is order of magnitude greater than observed on UCL machines, direct sound
  sound driver anyone...].
- Crash when playing .wav files? <thavaqos@yahoo.com>  [Can't Replicate???]
- Build problems on Windows: some of the *.dsp/*.dsw files have unix end of
  line characters which cause problems for some versions of VC++? Reported
  by Fulvio Risso <risso@polito.it>