root/rat/trunk/MODS @ 2879

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

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.
        * 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. 



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

- Assorted audio driver problems:
        - FreeBSD driver bug on SoundBlaster 16 has small write buffers 
          and stops working full duplex mode after a time.  Hard to fix because
          of cushion.
        - 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 (have tried nearly all
          options but current driver is v. unstable). Has only worked in 8k 
          mono.  New driver is coming soon...
        - SGI O2's don't have autosensing audio interface selection like 
          Indy's.
        - Need to allow manual input and output switching.
- 3d renderer should decimate when using sampling rates other than
  that for which it was originally designed for.
- 3d renderer could use finite precision maths for better performance 
  (eliminate type conversions).
- sndfile routines should use conversion functions in audio_fmt.c, not their 
  own.
- Bias estimate should be sample continuous, not block stepped.
- 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
- Release audio option
- Add/restore audio device trading 
- Disabling participant list doesn't work if the window has been resized:
  not sure if this is a feature of Tk or the window manager, but there's 
  little we can do about it.
- Playout delays may be out of date in both tools (rat & vic) when lip-sync 
  is enabled and when no talkspurt has been sent/received for a while (e.g 
  when the speaker stops talking for a while)
- When silence suppression is off, then no mbus lip-sync messages are exchanged
  between rat & vic since this is done on a per-talkspurt basis by rat.  We 
  may use a timer to send such messages every so often. 
- Streaming option -S from command line to stream files, raw and sample based
  files would be good.
- All supported codecs should have payload types by default.
- Reimplement transcoder.
- Add support for conversion between m*8000 and n*11025
        - n*11025 sampling isn't well supported in general
- Add mp3 decoder.  xaudio and ms acm decoders look v. easy to add.
- Playout bug with overlapping audio reception.  Pushes incoming audio too
  far back.  Mute source briefly to get around this.
- Better code documentation.
- 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.
- The new loading of settings isn't quite complete yet - some variables 
  need to be activated in the user interface (they are loaded fine; it's
  just not made evident to the user)