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Next version, in progress:
- Modified the 'multicast loopback' mechanism for getting our own IP address to check the source address of the received
  multicast packet, to make sure that it's valid (e.g., not 127.0.0.1).  (Thanks to Stefan Spurling for this suggestion.)

2012.02.04:
- Updated "WAVAudioFileSource" to read from its input file asynchronously,
  if possible, rather than doing a synchronous (blocking) read.

2012.02.03:
- Updated "RTSPClient" to - after receiving a "SETUP" response for a UDP stream - send a couple of short 'dummy' UDP packets
  to the server.  This will make it more likely that the incoming RTP/UDP packets will successfully traverse a NAT box
  (if the client is behind a NAT).  (Note that we don't do this for RTCP, because the client's regular RTCP "RR" packets will
  have the same effect.)
- Changed the way that the "sessionId" member field in "MediaSubsession" is managed.  Its memory is now managed by "MediaSubsession"
  itself, rather than by "RTSPClient" (as it was previously).  With the previous behavior, "valgrind" (incorrectly) reported
  a possible memory leak.  The new behavior should make 'valgrinerds' happy.

2012.01.26:
- We now make the "MediaLookupTable" class visible in the header file "include/Media.hh".  This allows developers to, if they wish,
  iterate over the whole set of "Medium" objects that they've created.
  (Thanks to Aviad Rozenhek for this suggestion.)
- "HashTable::Iterator::create()" now takes a "HashTable const&" as parameter, rather than a "HashTable&".
  (This makes it possible for iterators to work on (references to) hash tables that we've declared as const - for more type safety.)

2012.01.25:
- Added a new class "MPEG2TransportUDPServerMediaSubsession" (a subclass of "OnDemandServerMediaSubsession") that can be used
  to build a RTSP server that can takes a UDP (raw UDP or RTP/UDP) Transport Stream as input (via IP multicast, or unicast).
  We also updated the "testOnDemandRTSPServer" demo application to show how a RTSP server can take input from the
  (IP multicast) Transport Stream sent by the "testMPEG2TransportStreamer" demo application.
  (Thanks to Achraf Gazdar for this suggestion.)
- Added a new demo application (in "testProgs") "testMPEG2TransportReceiver"
  (which can receive the MPEG-2 Transport/RTP stream sent by "testMPEG2TransportStreamer")

2012.01.13:
- Fixed a bug in "StreamReplicator" that could cause 'replica deactivation' code to be executed more than once on the same replica,
  with bad results.  (Thanks to Mike Stewart for reporting this.)
- Updated the 'connect()'-result test in "RTSPClient.cpp" to check for EWOULDBLOCK as well as EINPROGRESS.
  (Windoze systems can apparently return EWOULDBLOCK.  Thanks to Jeff Shanab for the suggestion.)

2012.01.07:
- Added a new "liveMedia" class "StreamReplicator".  This can be used to create an arbitrary number of 'replicas' of an input
  stream.  We also added a new demo application "testReplicator" (in "testProgs") that demonstrates how to use this class.
- Made a small change to "testRTSPClient" to account for the possibility of a RTCP "BYE" being received after having sent a
  RTSP "TEARDOWN".

2011.12.23:
- Added a new demo application "testRTSPClient" to the "testProgs" directory.  If you're developing your own RTSP client
  application (or want to embed RTSP client functionality into a larger application), then "testRTSPClient" is a better model
  for this than "openRTSP" (because "openRTSP" was designed to be a standalone application, and is complex, with a large number
  of options).
- Fixed a bug in "MultiFramedRTPSource" that could cause an incorrect count of 'truncated' bytes to get reported if the receiving
  sink's buffer size was too large to fit a complete incoming frame.  (Even worse, this bug might also cause a frame of invalid
  data (that should have been truncated from the previous frame) to get delivered to the sink.)

2011.12.20:
- Fixed a "RTSPServer" bug in handling commands (such as "GET_PARAMETER" or "SET_PARAMETER") that have a "Content-Length:" header,
  specifying extra bytes at the end of the message.  (The bug usually affected only RTP-over-TCP streams, not RTP-over-UDP.)
  (Thanks to Jer Morrill for bringing this bug to our attention.)
- We no longer make the 'groupsock' "reuseFlag" a static variable, in case the library is being accessed by multiple threads.
  Instead, we now make it part of the "groupsockPriv" memory that's part of each "UsageEnvironment".
  (Thanks to Michel Promonet for noting this issue.)

2011.12.19:
- Updated the "UserAuthenticationDatabase" destructor (in "RTSPServer.cpp") to delete each of the "password" strings that we
  had previously allocated, thereby eliminating a memory leak.  (Thanks to Michel Promonet for noting this.)
- Fixed a bug with the "#ifdef"s in "liveMedia/include/Locale.hh" (introduced in version 2011.11.27) - to ensure that the new
  'xlocale' functions get used on Linux by default.
  (Thanks to Remi Denis-Courmont for alerting us to this issue.)
- Added a public member function "url()" to "RTSPClient", to get the RTSP URL.  (This might be useful for applications that
  create multiple "RTSPClient" objects, and want to identify each one (e.g., when debugging).)

2011.12.02:
- Updated the "H264VideoStreamFramer" implementation to avoid triggering a "StreamParser" internal error, even for extremely
  large NAL units.

2011.11.29:
- Updated "MPEG2TransportStreamMultiplexor" to change the Program Map PID from 0x10 to 0x30.  (Thanks to Chris Richardson,
  who noted that PIDs below 0x30 are not valid for ATSC or DVB (although they may be valid for other Transport Streams).)
- Updated the "NetAddressList()" constructor (for resolving names->IP addresses) to use "getaddrinfo()" - if available - rather
  than the older, deprecated "gethostbyname()" function.  Also, the "ourIPAddress()" code (in "GroupsockHelper.cpp") now uses
  the "NetAddressList()" code (if the initial multicast loopback method fails) to look up our host name.

2011.11.28:
- Use "uintptr_t" and "intptr_t" in a few places, where appropriate, to eliminate compiler warnings when compiing for 64-bits.
  (Thanks to the VLC developers for this suggestion.)
- When we create each (datagram or stream) socket, set the 'close on exec' property (if we can).
  (Thanks to the VLC developers for this suggestion.)

2011.11.27:
- Updated the definition and implementation of the "Locale" class to - if possible - use "newlocale()" rather than "setlocale()".
  The reason for this is that - on some systems - "setlocale()" may not be thread safe.
  (Thanks to the VLC developers for this suggestion.)

IMPORTANT NOTE: Some systems - e.g., FreeBSD - do not have the "xlocale.h" header file that defines "newlocale()".
  This means that we had to #ifdef the code, and continue to use the old implementation (using "setlocale()"),
  if XLOCALE_NOT_USED is defined.  We have updated the "config.freebsd" configuration file to add "-DXLOCALE_NOT_USED=1"
  to the compile flags.  However, if there are other systems that have "locale.h" but not "xlocale.h", then their configuration
  files will need to be updated also.  If you find such a system, then please let us know (using the "live-devel" mailing list).
- Updated the 'IP address finding' code in "groupsock/GroupsockHelper.cpp" to use "getaddrinfo()" - if available - rather than
  the older, deprecated "gethostbyname()" function.
  (Thanks to the VLC developers for this suggestion.)
- Made the "RTSPClient" member function "setBaseURL()" protected rather than private, to allow subclass constructors to,
  if they wish, construct and set the "rtsp://" URL in the body of their constructor, rather than in the call to the parent
  "RTSPClient" constructor.  (If they want to do this, then they should first pass "NULL" as the "rtspURL" parameter in the
  parent constructor.)
  (Thanks to Jeff Shanab for this suggestion.)

2011.11.26:
- We no longer use "inet_ntoa()" (or "our_inet_ntoa()"), because it's not safe if called from multiple threads.
  Instead, we now use our own "AddressString" class (defined in "groupsock/include/NetAddress.hh").  
  E.g., instead of calling
      our_inet_addr(addr)
  we now call
      AddressString(addr).val()
  (Thanks to the VLC developers for noting the problem with "inet_ntoa()".)
- Updated the "WindowsAudioInputDevice" code to make it work better with Unicode.  (Thanks to Jeff Shanab.)

2011.11.20:
- Fixed "RTCP.cpp" to not call a "BYE" handler (if one's set) until the very end of the incoming RTCP report handler member
  function.  This is to allow for the (common) possibility of the "BYE" handler causing "this" to get deleted.
  (Thanks to Jeff Shanab for noticing this.)
- Updated some counters used by "MPEG2TransportStreamFramer" from "unsigned long" to "u_int64_t", to better support streams
  that last for several days.  (Thanks to Tim Shackleton for this fix.)
- Modified the "RTSPServer" code to properly handle weird RTSP clients that ask for RAW UDP streaming, but also specify two
  port numbers in the "client_port=" part of the "Transport:" header.
  (Thanks to "homepuh(at)yandex.ru" for this suggestion.)
- Made a small change to the "VP8VideoRTPSource" implementation to reflect a minor change (the addition of the "K" header flag)
  in the most recent IETF Internet-Draft (draft-ietf-payload-vp8-02.txt)
- Added an #ifndef to "Boolean.hh" to deal with a Windows development environment that already defines (not "#define"s)
  "True" and "False".  (Thanks to Jeff Shanab for this suggestion.)

2011.11.08:
- Added "VorbisAudioRTPSink" and "VorbisAudioRTPSource" for sending/receiving Vorbos audio RTP streams (based on RFC 5215).
- Added "VP8VideoRTPSink" and "VP8VideoRTPSource" for sending/receiving VP8 video RTP streams.
- Added support for extracting and streaming Vorbis audio tracks from Matroska (including WEBM) files.
- Added support for extracting and streaming VP8 video tracks from Matroska (including WEBM) files.
- Updated the "testOnDemandRTSPServer" and "LIVE555MediaServer" (source-code version only) applications to support streaming
  from ".webm' files.
- Fixed frame durations for data extracted from Matroska tracks that don't have a 'default duration'.
- Fixed a memory leak in "RTSPClient::sendOptionsCmd()".  (Thanks to Gregg Mattinson for reporting this.)
  (Note, however, that "sendOptionsCmd()" - and other functions in the synchronous interface - are currently deprecated.
  Developers should now be using the asynchronous "RTSPClient" interface instead.)

2011.11.02:
- Updated the Matroska file parsing code to be more tolerant of extremely large frames (larger than the "StreamParser"
  "BANK_SIZE"), and bogus data - to avoid tripping into parser 'internal errors'.
- Added an #ifndef/#endif around the "NoReuse dummy;" statement in "RTSPServer.cpp".  This allows a developer - at compile time -
  to allow a "RTSPServer" to run even if there is already a server using the same port.  (The only time you should do this is
  if you anticipate restarting a server immediately after it exits, when the old server is still in a TIME_WAIT state, and
  you're sure that no other server will ever be running with the same port number.)
  (Suggestion by Chris Richardson.)

2011.10.27:
- Added support for seeking within streams that are demultiplexed from a Matroska file.
- Modified the signatures of the virtual functions "ServerMediaSubsession::seekStream()"
  and "OnDemandServerMediaSubsession::seekStreamSource()" to make the "seekNPT" parameter a "double&" rather than a "double".
  This makes it possible for subclasses to make slight changes to the specified seek position when they implement seeking.
  (This is necessary for seeking within Matroska-demuxed streams, because 'cue points' in Matroska files can be coarse-grain.)
- Made a slight modification to "openRTSP" so that it sets the play duration correctly if the RTSP seek position changes as
  a result of the RTSP "PLAY" command.
- Changed the implementation of "ReorderingPacketBuffer" in "MultiFramedRTPSource" to use a doubly-linked list rather than
  a singly-linked list.  This makes it more efficient to add new packets to the list after a packet loss has occurred
  (while we wait for the 'reordering threshold' to expire).  (As always, most of the "ReorderingPacketBuffer" code gets called
  only if a packet is lost (or reordered) in the input stream; not in the common case where all packets arrive in order.)
  (Thanks to Ralf Schroeder for this suggestion.)
- Made the implementation of the "our_random()" function more robust in case it's called concurrently by multiple threads.
  (Of course, as explained in the FAQ, you should calling LIVE555 code from multiple threads only if each thread uses its own
  "UsageEnvironment" and "TaskScheduler" object.)
  (Thanks to Michael Margold for raising this issue.)
- Fixed the way that you can specify - in the C/C++ preprocessor - that you want to use the system-supplied "random()" and
  "srandom()" functions, rather than our own versions.  You can do this by defining USE_SYSTEM_RANDOM (e.g., in a "config." file).

2011.10.18:
- Improved "RTSPServer" support for subdirectories in "rtsp://" URLs (handling this better for non-compliant clients
  that try to do a "SETUP" on agrregate URLs - when there is only a single subsession in the stream).

2011.10.09:
- Updated "RTSPServer" to better support multiple "/" characters (i.e., subdirectories) in URL - i.e., improving the modification
  that we made back in version 2011.08.13.
- Added an #include to "RTSPCommon.hh" to overcome a compilation problem that some
  people had when using Borland development tools on Windows.

2011.10.05:
- Added support for RTSP/RTP streaming, on demand, from a Matroska ('.mkv') file.  The video, audio, and subtitle tracks (if any)
  within the file are demultiplexed and streamed separately, via RTP.  We currently support H.264 video, AAC, AC3 and MP3 audio,
  and text subtitles.  Sometime in the future, we will also support WEBM ('.webm') Matroska files, with VP8 video and Vorbis audio.
  (We don't currently support seeking within the underlying  Matroska file (to implement RTSP seeking), but hope to do so
  in the future.)
  We also updated the "testOnDemandRTSPServer" demo application and the "LIVE555 Media Server" (currently, just the source-code
  version - not the prebuilt application binary versions) to support streaming from '.mkv' files.
- Added a new "RTPSink" subclass "T140TextRTPSink" to implement the 'text conversation' RTP payload format defined
  by RFC 4103.  (Note, however, that the optional support for 'redundancy' - as described in that RFC - is not yet supported.)
  We had already implemented the *reception* of this RTP payload format (using "SimpleRTPSource").
  We use this payload format to stream 'subtitle' tracks from Matroska files.
- Added a new configuration file "config.macosx-32bit" for generating 32-bit-only binaries for Mac OS X.
  (Thanks to Robert Wolff for contributing this.)
- Added a mechanism for (optionally) setting a callback function on a "RTPSink" - to be called if a send of data over
  the sink's socket fails.  (Contributed by Ralf Globisch.)
- Made a minor change to "MultiFramedRTPSink" to ensure that the "doSpecialFrameHandling()" virtual function gets called
  even when sending RTP packets that contain a 0-length payload.
- Fixed a minor memory leak in "H264VideoRTPSink" (the intermediate 'fragmenter' object wasn't getting deleted properly)

2011.09.19:
- We now set RTCP "RR" handlers for "PassiveServerMediaSubsession"s.  As a result, client sessions for multicast streams
  will no longer time out - as long as the clients send RTCP "RR" packets which the server can receive.
- Added a new public member function "setSPSandPPS()" to "H264VideoStreamFramer".  This lets you initialize a 
  "H264VideoStreamFramer" (or a "H264VideoStreamDiscreteFramer") with SPS and PPS NAL units, in case these do not appear
  early (or at all) in the input stream.
- Made some performance improvements to "H264VideoFileServerMediaSubsession" and "MPEG4VideoFileServerMediaSubsession".
- Fixed a bug in "H264VideoRTPSink" that could cause input source processing to not get stopped properly if the sink gets deleted.

2011.09.02:
- Updated "MultiFramedRTPSource" to more cleanly handle the situation where the sender's SSRC changes.  We don't yet support
  streams that contain multiple SSRCs, but we can handle streams where the SSRC changes occasionally - e.g., because a multicast
  source has been restarted.  (When the SSRC changes, we don't check the RTP sequence number of the first packet.)
  (Thanks to Dmitriy Vasil'ev for bringing this issue to our attention.)

2011.08.22:
- Updated the "RTSPServer" code to handle a rare condition whereby "handleRequestBytes()" gets called recursively.
  (This can happen only when we handle a "DESCRIBE" by re-calling the event loop (e.g., to wait for SDP parameters to
  become available).)  If this happens, we need to make sure that we don't delete the "RTSPClientSession" object until we leave the
  outermost call.  (Thanks to Vadim Kosarev for identifying the problem, and proposing the solution.)

2011.08.20:
- Modified the "MediaSession" and "MediaSubsession" classes to make it possible for developers to add support for receiving
  new RTP payload formats, without having to modify the "MediaSession" or "MediaSubsession" code itself.  To do this, developers
  define their own subclasses of "MediaSession" and "MediaSubsession".  For more details, see the comment near the top of
  "liveMedia/include/MediaSession.hh"

2011.08.13:
- Updated the "RTSPServer" implementation to allow multiple "/" characters in the "rtsp://" URL.  This allows the 
  "LIVE555 Media Server" to stream from subdirectories.  (Thanks to Mike Williams for this suggestion.)
- Updated the "RTSPClient" code to more cleanly handle our server's (empty) responses to RTSP "GET_PARAMETER" commands.
  (Thanks to Sebastien Escudier for noting this issue.)
- Updated the "StreamParser" code to allow parsers to test whether we've seen EOF on the input stream.  We also modified
  the "H264VideoStreamParser" code to use this, and fix a bug which was causing it to omit the last NAL unit in the input file.
- Updated "JPEGVideoRTPSource" to generate correct (or at least more correct) JPEG headers when the input stream contains just
  one quantization header.
- Added a 'cast' to the "RTSPServerSupportingHTTPStreaming" code to overcome a compilation problem that some people may
  have been seeing.

2011.07.21:
- Changed some member variables from "private:" to "protected:", to make subclassing easier.
  (Requested by Mojtaba Nouri and James Stafford.)
- Added a sanity check to the "RTP-over-TCP" handling code in "RTPInterface.cpp" to alleviate a strange condition (perhaps a bug)
  that Jeremy Noring saw.

2011.07.18:
- Added a new, optional extension to "RTSPServer" - called "RTSPServerSupportingHTTPStreaming" - that supports streaming
  to iPhones and iPads, using Apple's "HTTP Live Streaming" mechanism.  (For this to work, the files being streamed *must* be
  MPEG Transport Stream files with H.264 video, and must also be 'indexed' (using the same 'index file' mechanism that we use
  for 'trick play').  (The "live555MediaServer" source code has been updated to support this, as an option.)
- Fixed a bug in the various "*Streamer" demo applications that stream from the same input file, in a loop.  We weren't calling
  "stopPlaying()" on the "RTPSink" object at the end of each loop.  This bug directly affected only "testH264VideoStreamFramer",
  but was theoretically an issue for other demo applications as well.  (Thanks to Sergey Kosov for reporting this.)

2011.07.15a:
- Removed some experimental new code that had accidentally been left in the "mediaServer" directory.

2011.07.15:
- Added a new class "ByteStreamMemoryBufferSource" to encapsulate a (static) memory buffer that's used as a byte stream data source.
- Added a new class "TCPStreamSink" to encapsulate a writable TCP socket.  Being a "MediaSink", you can call "startPlaying()"
  on it, to stream from a "FramedSource" to a TCP socket.  This class could be used to implement a HTTP server, for example.
- Changed the signature of "ServerMediaSubsession::seekStream()" to take an additional "numBytes" return parameter.
  This parameter is set to the number of bytes that will be streamed (or 0 if unlimited, or unknown).
  (This can be used to implement a HTTP server, with a "Content-Length:" response header.)
- Added a new virtual function "getStreamSource(streamToken)" to "ServerMediaSubsession".  This provides an easy way for
  "RTSPServer" subclasses to access the media source for each stream, if desired.
- Removed the classes "HTTPSink" and "MPEG1or2VideoHTTPSink".  They weren't being used anymore.

2011.07.08:
- Updated the RTSP server implementation to check the "Content-Length:" header - if present - in incoming requests.  Most commands
  don't include extra data after the <CR><LF><CR><LF>, and thus don't need a "Content-Length:" header.  However, two commands
  - GET_PARAMETER and SET_PARAMETER - include extra data, and thus have a "Content-Length:" header.  We now check this value,
  to make sure that we've received all of the required data for each request.
  (Thanks to Matt Schuckmannn for reporting this issue.)

2011.07.06:
- Modified the change to "ByteStreamFileSource" that we made in version 2011.06.12.  We now call "read()" only if we know that
  the underlying file is non-seekable.  Otherwise - if the file is seekable - we call "fread()", as we did prior to 2011.06.12.
  (Reading from the file using "read()" doesn't work properly if we're also seeking within the file.
  This is why Transport Stream 'trick play' got broken by 2011.06.12, but should be working again now.)

2011.07.05:
- Updated "MPEG2IndexFromTransportStream" to make "MPEG2TransportStreamIndexer" more robust when it sees PCR timestamps decreasing
  in time (something that shouldn't happen).  (Thanks to Mojtaba Nouri for providing an example Transport Stream file that
  illustrates this.)
- Fixed a bug in "MPEG2IndexFromTransportStream" that could cause the "MPEG2TransportStreamIndexer" utility to hang when it
  saw bogus data appear before a MPEG system code.  (Thanks to Julian Scheel for reporting this issue.)
- Modified the "BasicTaskScheduler" implementation to add "dumySocket" only to "fReadSet", but not to "fWriteSet" or "fExceptionSet".
  This should continue to work around the Windows bug (where "select()" returns an error if no sockets are present in "fReadSet"),
  but without causing "select()" to keep returning unnecessarily (chewing up CPU).  (Thanks to Matt Schuckmann for reporting this.)
- We now make all files in the source code distribution read-only, by default.  Developers can, of course, change the permissions
  of any of these files to make them writeable, but having them be read-only by default helps emphasize that modifying the
  supplied source code is discouraged.  Instead, developers should - if possible - extend the supplied source code using
  C++ subclassing.  For more information, see the FAQ: http://www.live555.com/liveMedia/faq.html#modifying-and-extending

2011.06.16:
- Updated "AC3AudioRTPSink" and "AC3AudioRTPSource" to conform to RFC 4184, which defines the RTP payload format for AC-3 audio.
  Previously, we were using a different, incompatible payload format (that had been defined by an initial draft document, but
  which differed from that defined in the final RFC.  Unfortunately, this means that older clients that use our code
  - including existing binary versions of VLC and MPlayer - will no longer be able to play AC-3/RTP streams from our server
  (until they are rebuilt using this newest version of our library).
- Updated "AC3AudioStreamFramer" - and added a new class "AC3AudioFileServerMediaSubsession" - to support streaming from an AC-3
  audio file.  We also added this support to the "testOnDemandRTSPServer" demo application, and to the "LIVE555 Media Server"
  (currently just the source code; not the prebuilt binary versions).
  (Thanks to Ashwani Kathuria for suggesting this.)
- Moved the definition of "class StreamState" from "OnDemandServerMediaSubsession.cpp" to "OnDemandServerMediaSubsession.hh".
  Even though "class StreamState" is used only internally, in the implementation of "OnDemandServerMediaSubsession", moving its
  definition to the header file makes it possible for subclasses of "OnDemandServerMediaSubsession" to access it.

2011.06.14a:
- Removed some junk (debugging) files that had accidentally been left in the "testProgs/" directory.

2011.06.14:
- Fixed a race condition in "H264VideoFileServerMediaSubsession" and "MPEG4VideoFileServerMediaSubsession" that could have been
  triggered when two separate clients tried to stream the same file concurrently.  (Thanks to Bruno Abreu for reporting this.)
- Fixed a minor bug in "Authenticator::operator=()": we were not freeing the LHS object's memory before doing the
  assignment.  (Thanks to Sawan Das for noting this.)
- Updated "config.iphoneos" to change LIBRARY_LINK from "ar cr " to "libtool -s -o ", just as we did for "config.macosx" in the
  previous release.

2011.06.12:
- Updated the RTCP implementation to (if we're a SSM source) reflect incoming RTCP packets back to the multicast group
  *only* if the packet originated from another host.  This means that we won't reflect packets that come from other
  processes on our own host.  The reason for this is that we can't reliably distinguish such packets from packets that were
  actually looped back from us.  (We must not 'reflect' such packets, otherwise we risk a packet storm.)
- Updated the "ByteStreamFileSource" code to make the underlying socket non-blocking.  We also read data using "read()"
  rather than "fread()", to eliminate blocking (if READ_FROM_FILES_SYNCHRONOUSLY is not defined).  (This change should
  affect only reading from pipes, not regular files.)  (Thanks to Tyson Wiser for noting this issue.) 
- Updated the "DelayQueue" destructor to make sure that any remaining delay queue entries are destroyed.
  (Thanks to "Mr. Xiao" and Luke Kucalaba for (independently) reporting this issue.)
- Updated "config.macosx" to change LIBRARY_LINK from "ar cr " to "libtool -s -o ", because someone reported that "ar"
  didn't work for them.
- Added support for receiving the "audio/EAC3" RTP payload format (which is handled the same way as "audio/AC3").
  (Thanks to Gregory McGarry for the patch.)

2011.05.25:
- Fixed a bug in "RTSPClient" which was causing clients to crash if servers returned a 401 error code, but without a
  "WWW-Authenticate:" header.  (Thanks to Sebastien Escudier for reporting this.)

2011.03.14:
- Updated the "MPEG2TransportFileServerMediaSubsession" to use the "streamDuration" parameter (if >0.0) to limit
  the number of Transport Packets that are streamed from the source file.  (This happens only if the file is indexed.)
  This allows the server to implement finite RTSP ranges for such streams.
- Fixed a minor bug in "OnDemandServerMediaSubsession" if "reuseFirstSource" is set.  (Thanks to Andreas Gaer for noting this.)

2011.03.06:
- Changed the implementation of "RTSPClient::teardownMediaSession()" and "RTSPClient::teardownMediaSubsession()"
  (i.e., in the old, now-deprecated synchronous "RTSPClient" interface) to not wait for, or handle, a response to the
  RTSP "TEARDOWN" command.  This avoids the client blocking indefinitely if the server (or its connection) happens to die before
  the RTSP response gets sent.
  (Note, however: All RTSP client applications should be updated to use the new, asynchronous interface.  The old synchronous
  interface will not be supported indefinitely.)

2011.03.05:
- Changed the signature of the "OnDemandServerMediaSubsession::seekStreamSource()"
  virtual function to add a "streamDuration" field.  This field (if >0.0) tells
  the implementation how much data to stream before ending with EOF.
  Our RTSPServer implementation now uses this to implement finite RTSP ranges
  (as specified in the "Range:" header).  As with seeking, only some codecs currently
  implement this: MP3 audio, WAV audio, and DV video.
- We now support streaming from IMA ADPCM ("DVI4") WAV files.
- When streaming MP3 audio files or indexed MPEG Transport Stream Files, we now compute a more accurate bitrate estimate
  (for use in computing RTCP packet frequencies).

2011.01.24:
- Fixed a bug that was accidentally introduced in version 2011.01.10, and which crashes VLC (when VLC is used to play a
  "rtsp://" URL).  To avoid the bug, VLC should therefore use this version or later.

2011.01.21:
- Fixed a bug in "H264VideoStreamFramer" that was introduced in the last release.
  (This affected the "testH264VideoToTransportStream" demo application.)  (Thanks to Dunling Li for reporting this.)
- Fixed a minor syntax bug in "H264VideoStreamFramer.cpp" which (fortunately) had not been an actual problem.
  (Thanks to Guillaume Le Neindre for the report.)

2011.01.20:
- Added a new demo application "testH264VideoToTransportStream", which takes a H.264 Video input file (named "in.264"),
  and converts it to a Transport Stream file (named "out.ts").
  (Note that for this conversion to work properly, the input H.264 file must contain sufficient timing information for us to be
  able to deduce the frame rate.)
  (Thanks to Dunling Li for this suggestion.)
- Changed our RTSP server implementation so that the "SET_PARAMETER" command succeeds (but does nothing) by default.
  (Subclasses can redefine its behavior, if necessary.)  We also now support the special "*" URL, which designates an
  operation on the entire server.  (The only commands for which this is allowed are "OPTIONS", "GET_PARAMETER" and "SET_PARAMETER".)
  (Thanks to Jeremy Noring for noting that some clients need this.)

2011.01.19:
- Fixed a bug in "OnDemandServerMediaSubsession" that was causing unicast RTSP/RTP
  servers to sometimes omit sending the very first packet of a stream.  (This seemed
  to occur only for servers running on Windows.)

2011.01.10:
- Updated "RTSPClient" so that the new asynchronous interface can handle RTSP URLs that contain a "<username>:<password>@"
  before the server host name/address.  This had been supported in the old synchronous interface, but when we implemented the
  new asynchronous interface, we had forgotten to implement it there as well.
- Made a change to the implementation (but not the default behavior) of "RTSPServer" to allow for the possibility of a subclass
  implementing HTTP streaming (using the same HTTP port that we use to support RTSP-over-HTTP tunneling).

2011.01.06:
- We added support for receiving the RTP payload formats "audio/L20", "audio/L24" and "audio/DAT12", as defined in RFC 3190.
  We also recognize the new "fmtp:"-line parameters "emphasis" and "channel-order" that this RFC also defines.
  (Thanks to Maciej Szlapka for noting that these payload formats had not previously been supported.)

2011.01.05:
- Made a small change to the sanity check introduced to "MultiFramedRTPSink" and "BasicUDPSink" in version 2010.12.05 to
  eliminate the possibility of small extra delays accumulating over time.  (Thanks to Warren Young for raising this issue.)
- Made a small change to the 'magic values' that are used to implement the "esds" atom for MPEG-4 video in "QuickTimeFileSink".
  (I have no idea what these values are supposed to do, but Stu Tomlinson reports that the new values work better for him.)

2010.12.31:
- We now support 'trick play' operations (seeking, fast-forward, reverse play) on MPEG Transport Stream files that contain
  H.264 video (rather than just MPEG-2 video, as previously).  To support this, the index file format has been extended in a
  backwards-compatible way, so that existing index files (for MPEG-2 video Transport Streams) will continue to work as before.
  New versions of the "MPEG2TransportStreamIndexer" and "testMPEG2TransportStreamTrickPlay" utilities - and the
  "live555MediaServer" - have also been released.
- Fixed a bug in the definition of the "profile_level_id" field in "H264VideoRTPSink".  (Thanks to Geoff Cleary for noting this.)
- Change the parsing of RTSP "Range:" headers to allow parameters of the form "clock=" or"smtpe=".  (However, we currently don't
  interpret parameters of this form; instead, we just ignore them.)
  (Thanks to Sebastien Escudier for this suggestion.)
- Fixed "openRTSP" to properly reset internal state before repeating the playing of a stream (if the '-c' (play continuously))
  option is used.  (Thanks to Anon Sricharoenchai for noting this.)

2010.12.14:
- Oops - there was a serious bug in the 'event trigger' implementation in the previous release.
  IMPORTANT: You should upgrade to this new version if you plan to use the 'event trigger' mechanism!
  (Thanks to "P.J." for noticing the bug in the previous version.)

2010.12.11:
- Added a new 'event trigger' mechanism to "TaskScheduler".  This makes it possible
  to define new events that can be handled within the event loop.  Unlike other
  library functions, events can be 'triggered' (i.e., fired) from a separate thread.
  This makes it easier to implement input devices than the old 'watchVariable'
  mechanism (which remains).
  Also, the "DeviceSource" class - which is a model for how to implement an input
  device class - has been significantly improved.  It now uses the new 'event trigger'
  mechanism.  Also, more of it has been implemented, making it clearer where new code
  needs to be written.

2010.12.05:
- Significantly improved our support for streaming H.264 video.  In particular, "H264VideoStreamFramer"
  and "H264VideoStreamDiscreteFramer" (a new class) act like their corresponding MPEG4 versions: "H264VideoStreamFramer" reads
  a H.264 Video Elementary Stream byte stream (e.g., from a file), and "H264VideoStreamDiscreteFramer" reads discrete H.264 video
  NAL units (i.e., one-at-a-time), e.g., from a H.264 video encoder.  (Note that developers no longer need to subclass
  "H264VideoStreamFramer".)  We also added a new demo application - "testH264VideoStreamer" - for streaming from
  a H.264 Elementary Stream Video file via multicast.  "testOnDemandRTSPServer" and "live555MediaServer" were also updated to
  stream H.264 Video Elementary Stream files.
- Added a sanity check to "MultiFramedRTPSink" and "BasicUDPSink" to allow for the possibility of the system clock jumping ahead
  in time, and thereby messing up the calculation of how long to wait before sending the next packet.
  (Thanks to Anders Chen for noting this issue.)
- Fixed bugs in "AMRAudioRTPSource" and "QCELPAudioRTPSource" that might sometimes cause an event handler to try to reference
  objects that had already been deleted.  (Thanks to David Cailliere for detecting the problem with "AMRAudioRTPSource";
  it turns out that "QCELPAudioRTPSource" had the same problem.)

2010.11.17:
- Added new a member function "setAuthenticationDatabase()" to "RTSPServer".  This allows a server's manager to change
  (or disable) authentication at runtime.  (Thanks to Jeremy Norling for suggesting this functionality.)

2010.11.10:
- Fixed "openRTSP" to eliminate a recursive call to "shutdown()" if we receive a RTCP "BYE" while in the middle of doing
  a RTSP "TEARDOWN" command.  (Thanks to David Cailliere for noting this issue.)

2010.11.09:
- The previous release had accidentally included some experimental changes to "H264VideoStreamFramer" that were not (yet)
  intended to see the light of day.

2010.11.08:
- Fixed a minor problem with RTSP-over-HTTP support in "RTSPClient" that was causing some servers to complain due to the
  "CSeq:" header value not being incremented properly.  (Thanks to Kamil Dobkowski for the report.)

2010.11.04:
- Backed out and corrected the change to "RTSPClient" that we made in version 2010.10.22.  It turns out that existing RTSP-over-HTTP
  servers that return "401 Unauthorized" in response to the HTTP "GET" *do*, in fact, want clients to resend the "GET" command
  (with a filled-in "Authorization:" header).  However, at least one such server out there closes the TCP connection after
  sending back the "401 Unauthorized" response, so we need to send the second "GET" command using a new TCP connection.
  (Thanks to Kamil Dobkowski for providing access to a server that illustrated this problem.)

2010.10.28:
- Updated "JPEGVideoRTPSource" amd "JPEGVideoRTPSink" to support optional "Restart Marker Headers" in the outgoing RTP packet.
  (We already supported such headers in *incoming* JPEG/RTP packets ("JPEGVideoRTPSource").)
  If "type()" (defined by the "JPEGVideoSource" subclass) returns a value in [64,127], then the "JPEGVideoSource" subclass must
  also redefine "restartInterval()" to return a non-zero value.
  (Thanks to Cristiano Belloni for suggesting this addition.)

2010.10.23a:
- Fixed a bug in the way that "RTSPClient" generates ephemeral RTP/RTCP port number pairs for use when receiving unicast
  RTSP/RTP streams.  The RTP (even) port was guaranteed not to be in use elsewhere on the same host, but that guarantee was not
  necessarily true for the RTCP (next, i.e. odd) port.  This version fixes this.

2010.10.23:
- Our server implementation of "RTSP-over-HTTP tunneling" now allows for the possibility of receiving data from the 
  (Base-64-encoded) initial RTSP command at the same time that we receive the HTTP "POST" command.  (This is a possibility because
  the client does not wait for a response to the "POST" - because there isn't one.)

2010.10.22:
- Made a minor modification to the way that "RTSPClient" does "RTSP-over-HTTP tunneling".  If the initial "GET" request
  returns "401 Unauthorized", then we don't resend it (with an "Authorization:" header).
  Instead, we continue, as normal, sending the subsequent HTTP "POST" (with an "Authorization:" header).
  It's not clear (from the limited documentation on "RTSP-over-HTTP tunneling") which behavior is 'correct', but at least some
  servers seem to handle the new approach better.

2010.10.20:
- Made the implementation of 'RTSP-over-HTTP tunneling' in the RTSP server more tolerant of certain buggy clients.
  (Thanks to Cristiano Belloni for this suggestion.)

2010.10.15:
- Added server support for RTSP/RTP-over-HTTP tunneling.  We added a new "RTSPServer" member function
      Boolean setUpTunnelingOverHTTP(Port httpPort);
  to (attempt to) set up tunneling on the specified HTTP port number.
  We also updated the "testOnDemandRTSPServer" and "live555MediaServer" applications to try to set up HTTP tunneling
  on port 80, then port 8000, and then finally port 8080.

2010.10.06:
- Made a small change to "RTSPClient" to make it better handle RTSP servers that
  erroneously do not include "CSeq:" lines in their responses.
  (Thanks to Sebastien Escudier for this suggestion.)
- Made a minor update to "RTSPClient" to better handle any error in the HTTP "GET" command (if RTSP-over-HTTP tunneling
  is being used).
- Updated the "timestampString()" function in "groupsock/GroupsockHelper.cpp" to allow for the possibility of "ctime()" returning NULL.
- Removed the (never fully implemented) "RTSPOverHTTPServer" class.
  Instead, RTSP-over-HTTP tunneling will be implemented within the existing "RTSPServer"
  code, as that seems to be easier than using a separate class.

2010.09.25:
- Disabled the code for "Groupsock::multicastSendOnly()", because it was apparently causing SSM multicast packets to not be
  received by other applications on the same host (at least, for some systems).

2010.09.23a:
- Backed out the "RTSPServer" change that was in the previous release, because of a report that it doesn't work.

2010.09.23:
- Updated "H2564VideoFileSink" to take an optional "sprop parameter string" parameter as input.  If present, this string is
  decoded, and the resulting data (SPS/PPS NAL units) is prepended to the file.
  Also, updated "openRTSP" to call "MediaSubsession::fmtp_spropparametersets()", and pass this string when creating a
  "H2564VideoFileSink".
- Updated "RTSPServer" to ensure that we transmit RTP and RTCP packets over the same interface that is used by the requesting client
  (in case the server is multi-homed).  (Thanks to David Stegbauer for this suggestion.)

2010.09.22:
- Added a new virtual function "internalError()" to "UsageEnvironment" and "TaskScheduler".  This function is called
  a small number of times within the library code when an unexepected, 'should not occur'-type error condition occurs.
  The default implementation of this function calls "abort()", but subclasses can, of course, redefine this, if desired.
- Updated "RTSPClient::parseTransportParams()" to accept either "port=<num1>-<num2>" or "port=<num>".
  (The RTSP specification is ambiguous about whether or not both are allowed.)
- Made sure that calls to "ntohl()" and "htonl()" take (and return) "u_int32_t" rather than unsigned", for improved portability.
  (Thanks to Anon Sricharoenchai for this suggestion.)

2010.09.10:
- Added a sanity check to "MediaSubsession::getNormalPlayTime()" so that we ignore any old packet that might
  mess up the initial computation of the presentationTime-to-NPT offset following the arrival of new "RTP-Info" data.
  This should never happen with our servers (though John Shao Fa - who proposed this check - claims to be seeing it).
  We add the check, however, to protect against other servers that might be messing up.

2010.09.03:
- Made sure that various calls to "TaskScheduler::scheduleDelayedTask()" are called with a "int64_t" 'delay in microseconds'
  parameter, to avoid possible integer underflow if the system clock gets set back in time.
  (Thanks to Sebastien Escudier for reporting this problem.)

2010.08.31:
- Fixed some problems in the way that we implement RTP/RTCP-over-TCP streams that were showing up when we have more than one
  such stream sharing the same input source ("reuseFirstSource" == True).  (Thanks to John Tam for reporting this.)

2010.08.22:
- Updated the "DarwinInjector" class to use the new, asynchronous "RTSPClient" interface.

2010.07.29:
- Fixed a bug in RTP-over-TCP that was sometimes causing a socket to (attempt to be) read after it had already been closed.
  (Thanks to Wilton @ Invent Vision for reporting this bug.)
- Fixed some bugs in the way that RTP reception statistics were being generated and reported.  (Thanks to David Cailliere.)
- Added a new config file "config.linux-64bit" that should work better than "config.linux" for building for 64-bit Linux systems.

2010.07.13:
- Updated the "openRTSP" and "playSIP" applications to use the new asynchronous "RTSPClient" interface.
- Made the "readSocket()" routine more bullet-proof against Windows brain damage that was apparently causing problems with
  Windows RTSP clients that do RTP-over-TCP.
  (Thanks to Kamil Dobkowski for this suggestion.)

2010.07.07:
- Fixed a problem with RTP-over-TCP reception that was causing some RTP-over-TCP streams to not be received properly.
  All RTSP client applications that can request/receive RTP-over-TCP should upgrade to this new version.
  (Thanks to Kamil Dobkowski for providing a server that we could use to debug the problem.)
- Reinstated the functionality of the "forceMulticastOnUnspecified" flag in the "RTSPClient" "SETUP" command.
  (This functionality had temporarily been disabled, because I wasn't sure that it was still used by anyone.
  But apparently it is.)

2010.06.22:
- Fixed another bug in the new "RTSPClient" implementation (thanks to Sebastien Escudier); we were not recognizing
  "npt=now-" in "Range:" headers.
- Fixed a bug in "openRTSP" (reported by Kenneth Ljungh).  We were closing output files before issuing a RTSP "TEARDOWN" command.
  This problem was exposed only with the new "RTSPClient" implementation, because it reenters the event loop (thereby possibly
  handling incoming RTP packets) after the sending of the "TEARDOWN", before it gets handled and responded to.
  We have now fixed the "openRTSP" code to not close output files until after the RTSP "TEARDOWN" has been completed.

2010.06.18:
- Fixed a bug in the new "RTSPClient" implementation (reported by Sebastien Escudier) - that was causing problems
  when streaming from some RTSP-enabled video cameras.

2010.06.16a:
- Made "RTSPClient"s parsing of "Transport:" headers more bullet-proof against those sent by some servers
  (including, apparently, some recent versions of VLC's built-in RTSP server) that include a "client_port=" field,
  but no "server_port=" field.  (In this case, we assume that the server port is the same as the client port.)

2010.06.16:
- Fixed another bug in the new "RTSPClient" implementation of "GET_PARAMETER".
  (Thanks to Denis Charmet for reporting this.)
- Made "RTSPServer::RTSPClientSession::handleAlternativeRequestByte1()" more bullet-proof against an alleged bug noted by 
  Jeremy Noring.  (The actual bug hasn't been located, but this change apparently stops it from causing problems.)

2010.06.15:
- Removed the "blockUntilReadable()" function, because it's no longer needed, now that all socket reads are asynchronous.
  This means that "readSocket()" no longer calls "select()", which should improve the performance of applications.
- Improved the processing of RTP/RTCP/RTSP-over-TCP (possibly fixing some bugs in the process).
- Fixed bugs in the new "RTSPClient" implementation of "GET_PARAMETER" and "SET_PARAMETER".
  (Thanks to Denis Charmet for reporting this.)

2010.06.11:
- Fixed some potential memory access errors in the previous version of "RTSPClient".
- Removed code from "RTSPClient" that was supposed to deal with buggy (Microsoft?) servers, but which in reality was causing
  clients to not know the duration of time-bounded streams properly.
- Modified the new RTSP client implementation of "RTSP-over-HTTP" to make it more consistent with the old implementation.
  (Doug Porter reported a problem with the implementation in release 2010.05.29.)
- "AMRAudioSource" (and therefore its subclasses also) were not redefining the "MIMEtype()" virtual function.  Fixed this.
  (Thanks to Adam Mich for reporting this.)
- Fixed a minor problem that was causing some "UsageEnvironment"-allocated tables not to get deleted under some circumstances.

2010.05.29:
- Made a major change to the interface and implementation of "RTSPClient", to make it perform operations asynchronously
  - i.e., without blocking.  Each operation that performs a RTSP command now takes a 'response handler' function as parameter.
  The operation returns immediately, but the 'response handler' function will get called later, when a response arrives
  (or if an error occurs).
  For backwards compatibility, the old 'synchronous' interface still remains, and is implemented using the new 'asynchronous'
  interface.  (However, the old 'synchronous' interface should not be used for new applications, and it will likely be removed
  in a few months.)
  For now, the demo application "openRTSP" still uses the old interface, but will be rewritten over the next few weeks to use the
  new "RTSPClient" interface.
- Modified the "TaskScheduler" definition yet again, because it was too complicated.  There's now just one virtual function
  for setting/clearing socket handlers:
  "virtual void setBackgroundHandling(int socketNum, int conditionSet, BackgroundHandlerProc* handlerProc, void* clientData) = 0;"
  where "conditionSet" is a combination of SOCKET_READABLE, SOCKET_WRITABLE, and SOCKET_EXCEPTION.
  For backwards compatibility, the existing "turnOnBackgroundReadHandling()" and "turnOffBackgroundReadHandling()" functions
  are retained, but are now implemented using "setBackgroundHandling()".
  See "UsageEnvironment/include/UsageEnvironment.hh" for details.

2010.05.28:
- Updated the previous revision of "TaskScheduler" to add two more new virtual functions
  "turnOnBackgroundExceptionHandling()" and "turnOffBackgroundExceptionHandling()",
  because it turns out that Windows needs them for checking on a failed remote "connect()"
  (behaving differently from Posix, which requires only select()ing the socket for
  writing, not an error).

2010.05.27:
- Updated "TaskScheduler" (and our supplied example subclass "BasicTaskScheduler") to include new virtual functions
  "turnOnBackgroundWriteHandling()" and "turnOffBackgoundWriteHandling()".  These are the same as the "Read" versions,
  except that they are for checking when a socket becomes writable.  (They will be used to implement the new, upcoming
  'asynchronous' implementation of "RTSPClient".)
  We also modified the signature of "setResultErrMsg()" to take an optional "err" parameter.  (If  non-zero, this will be used
  instead of "getErrno()".) 
- Added a "config.iphoneos" (for cross-compiling on a Mac for the iPhone).  (Thanks to Mark Vasilkov.)

2010.04.09:
- Fixed a bug in the way that "AMRAudioRTPSource" reimplements "hasBeenSynchronizedUsingRTCP()".
  (Thanks to Ralf Globisch for reporting this bug.)
- Increased "SegmentQueueSize" in "MP3ADU.cpp" from 10 to 20, to allow for some unusually-VBR MP3 files that neeed this.
  (Thanks to Dirk Raffel for reporting this problem.)
- Changed the implementation of "parseRangeHeader()" in "RTSPCommon.cpp", in preparation for upcoming changes to "RTSPClient".

2010.04.01:
- Fixed the "openRTSP" code so that it no longer restricts the play time duration (using the "-d" option), because there is no
  no longer any practical limit on the duration that gets passed to "TaskScheduler::scheduleDelayedTask()"
  (because it is microseconds in 64 bits).
  (Thanks to Brad Thomas for this suggestion.)
- Removed some obsolete (and unused) code from the "RTSPClient" implementation, in preparation for a major reworking of this code.

2010.03.16:
- Made more improvements to "QuickTimeFileSink".  (Thanks to Wolfgang Breyha.)

2010.03.15:
- Fixed a bug in the Transport Stream file indexing mechanism that was preventing index files from properly being generated
  for some Transport Stream files.  (We sometimes weren't analyzing Program Association Tables correctly.)
- Removed the hack that we introduced in version 2009.11.27, because we have now fixed the bug that made that necessary.
- Removed the "RTPOverTCP_OK" hack from "RTPInterface.cpp", because it's no longer needed.

2010.03.14:
- Fixed the RTSP server implementation so that - when streaming via RTP-over-TCP - it can now handle RTSP commands after the
  first "PLAY".  (This fixes a long-standing bug.)
- Updated "QuickTimeFileSink" to support writing files larger than 2GB.  Also fixed a bug involving "esds" atoms for MPEG-4.
  (Thanks to Wolfgang Breyha for this patch.)
- Fixed a bug in the way that RTCP "SDES" items are constructed.  (Thanks to Jeremy Norling for bringing this to our attention.)
- Added a sanity check to the string version of "BasicUsageEnvironment::operator<<()", to check for a NULL string input.
  (Thanks to Paulo Rogerio Panhoto for this suggestion.)

2010.03.08:
- Fixed a rare bug that could sometimes cause RTSP clients to reuse the same (ephemeral) port number.
  (Thanks to Andro Debevere for reporting this.)
- Made changes to "config.armlinux", suggested by Rafael Carre.
- Removed a "|| defined(__CYGWIN32__)" test from "groupsock/GroupsockHelper.cpp", because it's no longer needed.
  (Cygwin has apparently now fixed the bug that made that necessary.)
  (Thanks to Tatterdemalian Anzulovic for reporting this.)

2010.02.10:
- Fixed a possible 1-byte overflow in "RTSPClient::setupMediaSubsession()".  (Thanks to Martin Bonnin for reporting this.)
- Make "parseCLine()" a bit more robust against certain technically illegal, but still parseable SDP "c=" lines.
  (Thanks to Aurelien Nephtali for this suggestion.)

2010.02.09:
- Fixed the way that "RTSPClient" operations handle the "timeout" parameter.
  We now make the socket blocking afterwards (having made it non-blocking beforehand).
  (Thanks to Jeremy Noring for this patch.)

2010.01.22:
- Changed the implementation of "Socket::changePort()" once again, beause it was reported that the change made in version
  2010.01.15 didn't work.  Note, however, that this change also introduces a new function
	     virtual void moveSocketHandling(int oldSocketNum, int newSocketNum) = 0;
  to the "TaskScheduler" API.  We have implemented this for "BasicTaskScheduler".  If you implement your own subclass of
  "TaskScheduler", then you will need to implement this new function as well.

2010.01.16:
- Added ""#include <stdio.h>" to "groupsock/NetInterface.cpp".  This was tripping up some people.
  (Thanks to Ben Hutchings for noticing this.)

2010.01.15:
- Changed the implementation of "Socket::changePort()" to call "bind()" directly, rather than closing the socket and reopening a
  new one.  That old implementation sometimes caused the socket number to change, which messed up our event loop.
  (Thanks to Rob Krakora for noticing this.)
- Improved the implementation of DV video streaming (introduced in the previous release) - in particular,
  to implement "DVVideoRTPSource".

2010.01.13:
- Added support for streaming/receiving DV video via RTP (using the RTP payload format defined in RFC 3189.
  (Thanks to Ben Hutchings for his help.)

2010.01.11:
- Increased the default "FileSink" buffer size in "openRTSP" from 20000 to 100000, because the previous limit was being bumped into
  too many times.
- Added #include "GroupsockHelper.hh" to "DVVideoStreamFramer.cpp", because it defines "gettimeofday()".

2010.01.09:
- In the various demo applications (in "testProgs/"), we changed the calls to "abort()" back to "exit(1)", because we want
  "abort()" to be used only for serious (unexpected) errors; not things like bad command-line arguments.
- Updated the RTSP server implementation to make session ids random (and 8 (hex) digits long), to conform to the RTSP specification.
  (Thanks to Guy Bonneau for this suggestion.)

2010.01.07:
- Fixed a bug in "parseGeneralConfigStr()" (it was mistakenly returning an extra byte).
- A few places in the code we called "exit()" if we encountered an unexpected error.  By request, these have now been changed to
  "abort()". 
- Made the "fSDPLines" member variable (in both "OnDemandServerMediaSubsession" and "PassiveServerMediaSubsession") "protected"
  rather than "private".  This allows subclasses to - if they wish - delete and NULL "fSDPLines" each time a client connects, in
  case the stream's SDP parameters have changed.

2009.11.27:
- Temporarily hacked the RTSP server code to not include "GET_PARAMETER" as an available option in the response to the "OPTIONS"
  command.  This will stop VLC from using "GET_PARAMETER" as a client 'liveness' indicator.  This is something that we don't
  need, because we already use clients' RTCP "RR" packets as a 'liveness' indicator, and because sending "GET_PARAMETER" over a
  RTP-over-TCP session causes problems with our current implementation.  (This hack will be removed when we fix the problem with
  RTP-over-TCP.)

2009.11.12:
- Made "BasicTaskScheduler::DelayQueue.cpp" more robust against an apparent compiler error on some platforms (e.g., Mac OS X)
  that was leading to "select() fails: Invalid argument" errors.
- Added a new compilation configuration file: "config.avr32-linux".  (Thanks to Jorge Munoz Castiner.)
- Made some minor changes to "groupsock/GroupsockHelper.cpp", "liveMedia/InputFile.cpp" and "liveMedia/OutputFile.cpp"
  to make these files compile (or compile better) for WinCE.  (Thanks to Pierre Ynard for these fixes.)

2009.09.28:
- Added an optional "timeout" parameter to "DarwinInjector::setDestination()".  (Suggested by Jeremy Noring.)
- Added support for receiving the "text/T140" RTP payload format.  (Suggested by Pierre Ynard.)

2009.09.21:
- Added patches proposed by Ben Hutchings to (i) improve the top-level Makefile; (ii) build better static libraries (on
  Linux, FreeBSD, and Mac OS X); (iii) have RTSP servers set "b=AS:" lines in their SDP (and set an appropriate socket send buffer
  size), and (iv) have RTSP (and other SDP) clients interpret the "b=AS:" line (if present) to set the socket receive buffer.
- Changed two source files in the "groupsock" project to use the (current standard) <sstream> #include file.
- Increased RTSP_PARAM_STRING_MAX to 200 (request by Stuart Rawling)

2009.09.04:
- Fixed "BasicTaskScheduler"s workaround to a Windows bug (thanks to Vityusha Vinokurov).
- Fixed "DelayQueue::synchronize()" to allow for the possibility of the system clock being reset back in time.
  (Thanks to Sebastien Escudier for pointing out this issue.)
- Made "H264VideoRTPSink::auxSDPLine()" "protected:" rather than "private:" (following a request from Stuart Rawling)

2009.07.28:
- Updated "QuickTimeFileSink" to add a "stss" atom for video streams, following a suggestion by Gerardo Ares.
  (At present we just 'guess' which video 'samples' (frames) are 'key frames', so this might not work properly on some
  video streams.)
- Modified the "config.uClinux" configuration file, following a suggestion by Chetan Raj.
- Changed "RTSPClient"s implementation of the RTSP "TEARDOWN" command to always act as if the command succeeded, regardless of
  the actual response from the server (because, from the client's point of view, the session has ended).
  (This overcomes a potential memory leak, pointer out by Stuart Rawling.)

2009.07.09:
- Modified the RTSP server implementation to - for streams where there is a known duration - always include a range end time
  in the RTSP "PLAY" response, even if the client did not specify one in the "PLAY" request.  This allows VLC's client
  'trick play' to (mostly) work.
- Updated "MediaSession::initiate()" to eliminate a possible memory leak if we get an error in socket creation.
  (Thanks to Denis Charmet.)
- Made a minor change to "MultiFramedRTPSink" to make monitoring/debugging easier.  (Thanks to Guy Bonneau.)
- Begun adding support for DV video.  However, this implementation is still incomplete.  DO NOT USE IT!

2009.06.02:
- Updated the MPEG Transport Stream multiplexor implementation to allow for H.264 video.  (Thanks to Massimo Zito.)
- Updated "MultiFramedRTPSink" to allow for subclasses for RTP payload formats (such as DV, coming soon) that impose
  a granularity on RTP fragment sizes.

2009.04.20:
- Fixed "BasicUsageEnvironment::getErrno()" to always (under Windows) return 
  "WSAGetLastError()" (and to just ignore the "errno" variable>.
  Also fixed a few places in the code where we were still using "errno"
  instead of calling "getErrno()"

2009.04.07:
- Changed many "char*" variables to "char const*" to eliminate possible compiler warnings.
  (Thanks to Sebastien Escudier for pointing out this issue.)

2009.04.06:
- Modified our Windows-only version of "gettimeofday()" so that it now returns times based on the proper epoch.
  (Thanks to Patrick White for this suggestion.)
- Created a new config file for 64-bit Solaris, and renamed the old "config.solaris" file to make it
  clear that it's for 32-bit Solaris only.  (Thanks to Michael Skaastrup.)
- Modified "config.mingw" to add "-DLOCALE_NOT_USED" to the "COMPILE_OPTS =" line.  (The VLC folks seem to want
  this.)
- Made a minor change to some win32-specific code in "RTSPClient.cpp" that the VLC folks seem to like.
  (However, "RTSPClient" is about to undergo a major overhaul (for asynchronous I/O) anyway...)
- Made a small change to "mediaServer/DynamicRTSPServer.cpp" to eliminate compiler warnings on some platforms.

2009.03.22:
- Updated "MediaSession::initiate()" to make the selection of ephemeral client
  port numbers (for RTP and RTCP) more robust, and to eliminate the possibility of
  the same port numbers accidentally getting chosen more than once by different
  clients running on the same host.  (Thanks to Guido Marelli and others for helping
  track down this issue.)

2009.02.23:
- Changed the way in which 'trick mode' transport streams are generated, so that each
  video I-frame is included in the output Transport Stream no more than once.
  This means that now - for low scale values - the output frame rate, and bit rate,
  will be les than that of the original file.  (Beforehand, we maintained the same
  frame rate, which meant that - because each output frame is an I-frame - the output
  stream's bit rate was always significantly larger than the original.  This caused
  problems for some clients.)
- Changed the 'client liveness' checking code in the "RTSPServer" implementation so that it deletes the
  "RTSPClientSession" object even for multicast streams.  
- Made a minor fix to an error report in "RTSPClient".  (Thanks to Igor Milavec.)

2009.02.13:
- Changed calls to the "Locale()" constructor to use an explicit variable name (rather than an anonymous
  declaration), because some compilers weren't handling the anonymous declaration properly.
  (Thanks to Steven Kramer for noticing this.)
- Improved the "RTSPServer" definition and implementation to make it easier to subclass the
  "RTSPServer::RTSPClientSession" class (e.g., to add a custom implementation of the "SET_PARAMETER" command).
  Also added an iterator class for iterating over the "RTSPClientSession" objects within a "RTSPServer".
- Make "MediaSubsession::parseSDPAttribute_framerate()" a little more robust.
  (Thanks to Shawn Van Every for the suggestion.)
- Improved "BasicUsageEnvironment0::reportBackgroundError()" (thanks to Matt Schuckmann)
- Fixed a bug in the "AVIFileSink" code (thanks to Pramod Bhadwat for reporting this).

2009.01.26:
- Updated the header file for "rtcp_from_spec.c" to make the call to "exit()" a noop.
  (Thanks to Chris Silvers for raising this issue.)
- Changed the definition of "RTSPClientSession" (in "RTSPServer") to make it protected (and also to make its
  command handler functions protected and virtual), to allow developers to subclass it (to change the
  implementation of some commands).
  (Thanks to Yedidia Amit for suggesting this.)
- Fixed a minor bug in "scheduleNextQOSMeasurement()" (in the "openRTSP" implementation).
  (Thanks to Gabriele De Luca for noticing this.)

2008.12.20:
- Removed - from "MultiFramedRTPSource.cpp" - the historical sanity check for bizarrely delayed packets.
  Accepting such bizarro packets causes problems in the code; it's best not to accept them at all.
- Added to "FramedFilter" a new public function "detachInputSource()" that sets the filter's input source
  to NULL.  This is useful if you want to close the filter independently of closing its input source.
  (Thanks to Alex Cichowski for this suggestion.)
- Modified the header file for "MultiFramedRTPSink" to make its implementation of the "stopPlaying()" virtual
  function public.  (Thanks to Guy Bonneau for this suggestion.)
- Changed "MediaSession::initiate()" to make the port number selection mechanism (for unicast streams) more robust.
  (Thanks to Pierre Ynard for this suggestion.0

2008.11.13:
- Fixed an obscure bug in "BasicTaskScheduler::SingleStep()" in which the call to "handleAlarm()" would
  (on rare occasions) modify the set of read handlers in such a way as to be inconsistent with the result of
  the earlier call to "select()".  We fixed the bug by moving the call to "handleAlarm()" after the socket
  handler call.  (Thanks to Sigismondo Boschi for tracking down this bug.)
- Fixed a bug in "parseRTSPRequestString()" that could cause a crash when parsing certain weird, non-compliant
  requests.  (Thanks to Sebastien Escudier for reporting this.)

2008.11.04:
- Fixed a bug in "RTPInterface" that was sometimes causing sockets not to be reclaimed properly by a
  RTSP server when streaming RTP/RTCP-over-TCP.  (Thanks to Ralf Globisch for tracking this down.)
- Changed the RTSP 'start time' parameter from "float" to "double", to allow for
  accurate 'start times' that are very large - e.g., Unix 'epoch' times.
  (Thanks to Morgan Toervolt for the suggestion and patch.)
- Made some simplifications to the 'reception stats' code in "RTPSource".
  (Thanks to Guy Bonneau for this suggestion.)
- Fixed a minor bug in "QuickTimeFileSink". (Thanks to Pramod Bhagwat.)

2008.10.07:
- Added an optional "timeout" parameter (in seconds) to the 'connecting' RTSP
  client commands.  (Thanks to Sebastien Escudier for this patch.)
- Increased the size of a "DummySink"s buffer in "MPEG1or2FileServerDemux", to make
  it large enough for any PES packet.
- Fixed a minor memory leak in the "createNew()" functions for
  "AVIFileSink" and "QuickTimeFileSink" if the output fid did not get created.
  (Thanks to Mehmet Ozgul for noting this.)
- Fixed a couple of spelling erros, removed some unnecessary whitespace,
  and updated the address of the FSF in the LGPL preamble (thanks to Erik Hovland).

2008.09.02:
- Fixed a minor bug in "RTSPServer".  (Thanks to Renato Mauro.)
- Minor improvements fo "FileSink" and "MP3FileSource".  (Thanks to Pramod Bhagwat.)
- Changed the default "reclamationTestSeconds" parameter (in "RTSPServer" and "DynamicRTSPServer") from 45 seconds
  to 65 seconds, because 60 seconds is the default timeout value expected by (broken) clients that send
  RTSP commands rather than RTCP "RR" reports to indicate liveness.

2008.07.25:
- Minor bug fix to "MPEG2TransportStreamFramer" (reset "fTSPCRCount" in "doStopGettingFrames()").
  (Thanks to Qiujian Shu for reporting this bug.)
- Modified the previous changes to "QuickTimeFileSink" and "AVIFileSink" so that the
  "createNew()" functions return NULL if the output file cannot be opened.

2008.07.24:
- Changed "MultiFramedRTPSource" to handle the very first packet that we receive as
  if there were packet loss beforehand.  This ensures that we correctly discard any
  initial packets that we receive that occur after the start of a (multi-packet) frame.
- Fixed "QuickTimeFileSink" and "AVIFileSink" to close the output file (if it's not stdout or stderr) in the destructor.

2008.07.22:
- Fixed "MPEG2TransportStreamMultiplexor" to use an appropriate stream for our SCR, even if the input contained
  its own Program Stream Map.
- More minor changes to the code to make Erik Hovland's code-checking tools happy.

2008.07.06:
- Corrected one of Erik Hovland's memory leak fixes made to the previous revision.

2008.07.05:
- Fixed some memory leaks - and several other mostly cosmetic fixes.
  (Thanks to Erik Hovland for noticing these.)

2008.06.26:
- Fixed a bug in "openRTSP" that was sometimes causing an incorrect "Range:" header
  to be sent in "PLAY" requests.

2008.06.25:
- Fixed a bug in the headers sent for the RTSP "GET_PARAMETER" and "SET_PARAMETER" commands.
  (Thanks to "beilyzhang (at) hotmail" for this report.)

2008.06.05:
- Renamed the "ourSourceAddressForMulticast()" function to the less confusing and more explanatory
  "ourIPAddress()".
- The "RTSPClient" change that was made in 2008.04.09 to accommodate multicast
  streams apparently broke (some?) unicast streams.  This has now been fixed.
- Updated "RTSPClient" to handle weird servers that include a "Content-Length:" header
  in the response to the RTSP "SETUP" command.  (If we see such a header, we skip over
  the data referred to in the header.)  (Thanks to Kristian Soerensen for this suggestion.)
- Changed the parsing of lines in "RTSPClient" so that it now accepts \r or \n alone
  at the end of each line.  (\r\n is preferred, but the standard also allows \r or \n)
  (Thanks to David Schueler for this suggestion.)

2008.05.12:
- Made a change to "RTCPInterface" to ensure that "turnOffBackgroundReadHandling()" is called on a socket
  as many times as "turnOnBackgroundReadHandling()".  Apparently some versions of Windows get upset if this
  doesn't happen.  (Thanks to Ken Seo for the suggested fix.)
- Fixed a bug in "H263plusVideoStreamParser".  (Thanks to Andrey Latin.)
- Added more sanity checking to "JPEGVideoRTPSink".

2008.04.09:
- Corrected a type-related bug in "RTPInterface::handleRead()".  (Thanks to Brain Lai for noting this.)
- Fixed a bug in "RTSPClient" that would prevent RTCP "RR" reports from being sent to
  the correct multicast address when we are receiving a multicast stream. (Thanks to Changjin Liu for noticing this.)

2008.04.03:
- Fixed a bug that was introduced in the "2007.12.27" release.  The "timeout" parameter to the call to
  "readSocket" in "SocketDescriptor::tcpReadHandler()" in "RTPInterface.cpp" needed to be initialized
  (to zero) first.  (Thanks to Lodewijk Loos for alerting us to this problem.)
- Added a new config file "config.bfin-linux-uclibc", and renamed "config.bfin_uclinux" as "config.bfin-uclinux".
  (Thanks to Mike Frysinger.)

2008.02.08:
- Added a hack (suggested by "Romain") to "MPEG2TransportStreamFramer" to (perhaps)
  produce more accurate per-transport-packet duration estimates for wildly VBR streams.
- Updated "MPEG2TransportStreamMultiplexor" to support the addition of MPEG-4 Audio or Video Elementary Streams.
- Updated "RTSPOverHTTPServer.cpp" (a work in progress) so that it complies properly for some versions of
  Visual Studio on Windows.  (Thanks to Eric Flickner for noting this.)

2008.01.19:
- Corrected the "getNormalPlayTime()" function - introduced in the previous release - to allow for 'trick play'
  scale factors other than 1.

2008.01.18:
- Added a new member function
    float MediaSubsession::getNormalPlayTime()
  which - given a stream's current presentation time - returns the "Normal Play Time".
  This function is useful for RTSP streams.
- Added support for a "a=control:" URL specified in the SDP description at the session level.

2008.01.04:
- Changed the "RTSPClient" timeout - introduced in the previous release - from
  5s to 30s.
- Added support for setting and returning the RTSP session start time (as well as the end time).
  (This eliminates the need for the "live-starttime.patch" file that VLC was using.)
- Updated some of the system-specific configuration files, to eliminate the need for many of the patches that
  the VLC developers added to their code.

2007.12.27:
- Added a 5s timeout to the "RTSPClient" code that checks for RTSP responses.
  This mirrors a change that was already being done to VLC's copy of the code.
  (This is still a short-term fix, until the "RTSPClient" code is rewritten to
  properly use asynchronous I/O, using the event loop.)
- Added a timeout to the "readSocket()" call in "SocketDescriptor::tcpReadHandler()" in "RTPInterface.cpp",
  to handle reading RTP-over-TCP data.  This allows for the possibility of non-cooperative RTSP clients.
  (Thanks to Peter Leese for this suggestion.)  This is probably not a complete solution;
  more thought is needed...

2007.12.07:
- Fixed "H2633plusVideoFileServerMediaSubsession" to properly use a dynamic RTP payload type, rather than the
  static type 34 (which is reserved for the now-obsolete old "video/H263" RTP payload format).

2007.12.06:
- Updated "JPEGVideoRTPSource" to take optional 'default width' and 'default height' parameters.
  These parameters can be set by fields in the SDP description, and can be used to specifiy unusually
  large frame widths and/or heights.  (Thanks to Andrey Filippov.)

2007.11.18:
- Fixed a couple of memory leaks in "DarwinInjector".  (Thanks to Eyal Beit-Halachmi for noting these.)
- Removed old 'backwards compatibility' stuff from "FramedSource".  (Noone should be relying upon this any more.)
- Fixed a bounds-checking error in "parseRTSPRequestString()" caused by an int vs. unsigned problem.
  (Thanks to Luigi Auriemma for noting this.)  
- In "RTSPClient.cpp", fixed a couple of "unsigned" vs. "int" nits.  (Thanks for Brain Lai for noting this.)

2007.11.01:
- Several of the options to "openRTSP" have now been changed, with two new options added:
	  -s <initial-seek-time>
	  -z <scale>
  See the "openRTSP" documentation <http://www.live555.com/openRTSP/> for details.
- Fixed a bug in the way that "MPEG4VideoStreamDiscreteFramer" parses 'config' information (for inclusion in
  the stream's SDP description).  (Thanks to Nicola Bova for helping to identify this bug.)
- Eliminated a potential memory (and socket) leak when allocating server RTP,RTCP socket pairs.
  (Thanks to David Pan for reporting this.)
- Updated "ByteStreamFileSource" to treat a 0-byte file read the same as EOF.
- Ensure that we have reasonable OS buffering for writes on non-blocking sockets.

2007.08.03a:
- Removed a debugging printf() that had been left in by mistake.  (Thanks to Massimo Zito for noticing this.)

2007.08.03:
- Updated the "RTSPServer" implementation to work better on systems with more than one IP address.
  The server will now include - in its RTSP responses - the IP address on which the corresponding incoming request
  was received.

2007.07.25:
- Added some sanity checks to various "MediaSink" subclasses, in case "fSource" is NULL.
  (Thanks to Andrey Kaminsky for noting one of these.)

2007.07.10:
- Made sure that "MPEG2TransportStreamFramer"s estimate of the average duration of each Transport Packet
  gets updated correctly after each 'seek' operation.  (Thanks to Massimo Zito for suggesting this.)
- Fixed a bug in "MPEG2TransportStreamFromESSource" that was causing presentation
  timestamps to not be generated correctly.  (Thanks to Massimo Zito for noticing this.)

2007.07.01:
- Fixed a bug in "RTSPServer" and "RTSPClient" that would cause problems when streaming
  files whose names have spaces in them.
- Fixed a bug in "OnDemandServerMediaSubsession::deleteStream".  (Thanks to Igor Bukanov.)
- Make the sockets for "RTPSource", "RTCPInstance" and "BasicUDPSource" non-blocking, even though they will be read
  from only asynchronously, when packets arrive.  The reason for this is that, in some OSs, reads on a blocking
  socket can (allegedly) sometimes block, even if the socket was previously reported (e.g., by "select()") as
  having data available.  (This can supposedly happen if the UDP checksum fails, for example.)
  (Thanks to Marc Neuberger for pointing this out.)
- Fixed a bug in "MPEG2TransportFileServerMediaSubsession.cpp" that was causing 2x
  fast-forward to fail.  (Some debugging code had been left in by mistake.)

2007.05.24:
- Modified "setupDatagramSocket()" to better handle setting "ReceivingInterfaceAddr" when creating a socket
  to be used to send/receive multicast.
- Made a couple of minor changes to overcome some compilation errors that some people were apparently seeing.

2007.05.23:
- The RTSP server now includes the "source=" parameter in the response to a RTSP "SETUP" command.
  This works around a bug in QuickTime Player that would cause it to not send RTCP "RR" packets
  (when playing a unicast stream, and running on Mac OS X).  (Thanks to Dave Singer et al at Apple for tracking
  down this problem.)
- The RTSP server implementation no longer terminates the session if it returns 400 or 405 errors.

2007.04.24a:
- Fixed a problem in the new "RTSPOverHTTPServer" code that was causing some people compilation problems.

2007.04.24:
- Fixed a typo in "testOnDemandRTSPServer.cpp".  (Thanks to Nils Grundback for noticing this.)
- Modified the signature of "RTSPServer::specialClientAccessCheck()" (defined in the previous revision) to add
  a new "clientAddr" parameter (a "struct sockaddr_in"), to allow special access checking based on clients' IP
  address.
- Make sure that the locale is set to "POSIX" when calling "toupper()" or "tolower()" on human-supplied strings.
  (Thanks to Ismail Doenmez for noting this.)

2007.04.20:
- Modified "MPEG2TransportStreamMultiplexor" to set the "discontinuity_indicator" flag for the first
  "adaptation_field" in the output Transport Stream.
- Changed "AMRAudioRTPSource" to more accurately report whether/when an interleaved frame's timestamp has
  been synchronized using RTCP.
  (Thanks to David Bertrand for this patch.)
- Made to a small fix to the previous revision's support for multicast streaming of raw UDP (nonstandard)
- Added a virtual function to "OnDemandServerMediaSubsession" for closing the stream source.
  Subclasses can redefine this, if they wish, to do something smarter.  (Thanks to Igor Bukanov for this patch.)
- Added support to "RTSPServer" for optionally performing special per-client access control, beyond the
  standard Digest Authentication method.  (Thanks to Igor Bukanov for this patch.)
- Made a minor optimization to "Groupsock.cpp".  (Thanks to Maxim Petrov for this suggestion.)
- Added an initial implementation of RTSP-over-HTTP at the server level (it was already supported at the client level).
  (Note: This code has not yet been completed, and doesn't yet work, so don't try to use it.)

2007.02.20:
- Updated "RTSPServer" to support multicast streaming of raw UDP streams (nonstandard).
  (Thanks to Aesmund Grammeltvedt for this modification.)
- Made "RTSPClient" a little more robust in case the TCP connection fails.
- Made "RTPSink::rtpmapLine()" virtual (as requested by Andrew Voznytsa).
- Removed an archaic (no longer used) declaration in "groupsock/include/Groupsock.hh".
- Added an optional "allowKasennaProtocol" parameter to "RTSPClient::describeWithPassword()"
  (similar to "RTSPClient::describeURL()").  (Thanks to Igor Bukanov for this suggestion.)s

2007.01.17:
- Fixed a bug that would cause the "MPEG2TransportStreamIndexer" application to fail when run on a big-endian
  architecture.
- Added a new class "H264VideoFileSink", that prepends each incoming H.264 NAL unit with the prefix 0x000001,
  before writing it to the file.  Also updated the "openRTSP" code to use this, when receiving H.264/RTP streams.
  (Thanks to Chris Kuiper for contributing this.)
- Added a temporary #ifdef to "GroupsockHelper.cpp" to work around a compliation problem when building
  for Cygwin.

2007.01.11:
- Improved the "MPEG2IndexFromTransportStream" class to recogize Transport Stream
  PAT and PMT (tables).  This in turn makes the "MPEG2TransportStreamIndexer"
  utility more robust.
- Fixed a minor bug in "MPEG2TransportStreamFromESSource".

2007.01.09:
- Added RTSP server support for 'trick play' operations on MPEG-2 Transport Stream
  files.  This requires the presence of a ".tsx" index file for each ".ts" file.
  (This functionality will shortly be documented on the LIVE555 web site,
  and announced on the "live-devel" mailing list.) 
- Improved the performance of asynchronous file reading in "ByteStreamFileSource".
  (Thanks to Aesmund Grammeltvedt for this suggestion.)
- Added "-D_FILE_OFFSET_BITS=64" to the "COMPILE_OPTS" line for each of the Linux config files, in order to
  alleviate possible problems with I/O on large files.

2006.12.31:
- We now implement "ByteStreamFileSource" on Windows using synchronous file reads,
  by default.  I had thought that Windows XP had fixed the problem - present in
  earlier versions of Windows - whereby open files are not treated as select()able
  sockets.  But apparently the problem is still there.
- Added code that will - probably in the next release, very soon - support server
  'trick mode' operations on MPEG-2 Transport Stream files.  At present, we have two
  new applications - in the "testProgs" directory - that help support this:
  "MPEG2TransportStreamIndexer" (to create a special index file for a Transport Stream
  file), and "testMPEG2TransportStreamTrickPlay" (to generate a new Transport Stream
  file that simulates a 'trick play' operation performed on the original Transport
  Stream file).  See my forthcoming post to the "live-devel" mailing list for more
  details.
- Fixed a bug in the Base64 encoding routine.  (Thanks to Sebastian Gracias for reporting this.)

2006.12.08:
- Made sure that each TCP socket used by a "RTSPserver" is non-blocking, so that a slow or hanging client
  cannot hang a server.  (Thanks to "jers (at) inwind.it" for this suggestion.)

2006.11.16a:
- Oops - fixed a problem with the packaging of the new "mediaServer" code.

2006.11.16:
- Included the code for the new "LIVE555 Media Server" application
  (in the "mediaServer" subdirectory).  This is a complete RTSP/RTP server, that
  will serve any appropriately-formatted file that resides in the current directory.
  (More documentation will follow.)
- Updated "QuickTimeFileSink" to support H.264 video in ".mov" or ".mp4" output files.
  (Thanks to David Arnold for this.)
- Fixed a potential reference-count related bug in "OnDemandServerMediaSubsession".  (Thanks to Jers.)
- Changed the "RTSPClient" constructor from "private:" to "protected:".

2006.10.27:
- "RTSPClient" now recognizes the "Content-Base:" header in RTSP "DESCRIBE" responses.
  (Thanks to Derk-Jan Hartman for this suggestion.)
- Updated "MultiFramedRTPSource" to avoid excessive recursion (and thus potential
  stack overflow) in some cases when receiving streams that have packet loss.
  (Thanks to Erwin Beckers for raising this as an issue.)
- Updated "RTPSink" and "MultiFramedRTPSink" to ensure that a proper RTP timestamp
  is set in the RTSP "RTP-Info" header that follows a "PAUSE", and that this timestamp
  gets used in the next outgoing RTP packet.  (Thanks to David Bertrand for this suggestion.)
- Changed the special implementation of "gettimeofday()" for Windows to return
  more accurate results.  (Thanks to David Arnold for this change.)

2006.10.18a:
- A minor update to the "MultiFramedRTPSink" change made in the previous release.

2006.10.18:
- Added support for adding optional RTP padding bytes to "MultiFramedRTPSink".
  (Thanks to Victor Kozhuhov for this suggestion.)

2006.10.12a:
- Backed out the change that we made in the 2006.09.18 release: We now don't attempt
  to setsockopt() SO_REUSEPORT and IP_MULTICAST_LOOP on Windows. (Dave Arnold reports
  that doing so leads to UDP packets with TTL=1.)

2006.10.12:
- Fixed a rare memory leak in "MultiFramedRTPSource".  (Thanks to David Bertrand for this fix.)
- Made "RTSPServer::lookupServerMediaSession()" virtual, to allow for future
  subclasses of "RTSPServer" that - for example - create and add "ServerMediaSession"s
  dynamically (depending on whether or not the named file exists).
- Added a "rtspURLPrefix()" member function to "RTSPServer".  This returns the
  "rtsp://" URL prefix that's common to all of the server's "rtsp://" URLs.

2006.10.07:
- The demo applications (in the "testProgs" directory) that stream MPEG-4 
  Elementary Stream video files now use a file name "test.m4e" rather than "test.m4v".
  This is to avoid confusion, because Apple has coopted the ".m4v" suffix for
  MPEG-4 (audio+video) files, which are different.
- Made "MPEG4VideoStreamFramer" more bullet proof against some non-standard MPEG-4 video data.
  (Thanks to Maxim Petrov for this suggestion.)

2006.10.05:
- Fixed a minor bug in "MPEG2TransportStreamFromESSource" (thanks to "jeremy at electrosilk.net")
- Changed some class member function and variables from "private" to "protected", in response to a request
  from David Bertrand.
- Made "DarwinInjector::createNew()" more bullet proof, by giving the "applicationName" parameter
  a default value.

2006.09.20:
- Corrected the bug fix to "H264VideoRTPSink" that was made two revisions ago.

2006.09.18:
- Fixed a bug in the way that "RTPSource" records the maximum sequence number seen so far.
  (The bug could be triggered by packets arriving out of order.)  Thanks to Mallikharjuna Reddy
  for pointing this out. 
- When creating new datagram sockets, we now attempt to setsockopt() SO_REUSEPORT and IP_MULTICAST_LOOP,
  even on Windows.  (A version of Windows prior to XP had problems with this, but it's apparently no longer
  a problem.)

2006.09.12:
- Fixed a minor bug in "H264VideoRTPSink".  (Thanks to Emiliano Parasassi for noting this.)
- If the "totSessionBW" parameter to "RTCPInstance::RTCPInstance()" is zero, print out an error message,
  and use a value of 1 (kbps) instead.

2006.09.09:
- Reenabled asynchronous file reading in "ByteStreamFileSource", after fixing a bug
  that could lead to a 'race condition' in some circumstances.  (The fix was to call
  "FramedSource::afterGetting()" directly - rather than via the event loop - after
  delivering data to the downstream object.)
- Updated "MPEG2TransportStreamFromESSource" again.  Now, we set the
  "PES_packet_length" field to zero if the input data was larger than 2^16 bytes
  long.  (This is in accordance with the MPEG specifications; thanks to "jeremy at electrosilk.net"
  for noticing this..)

2006.09.08:
- Temporarily disabled asynchronous file reading in "ByteStreamFileSource" until we
  fix an apparent bug that is causing MPEG streaming to fail.
- Updated "MPEG2TransportStreamFromESSource" to print a warning message if the input source delivers
  data in chunks that are too big to make up a PES packet.

2006.09.07:
- Fixed a bug that was preventing the RTCP "RR" packet handler from being called
  when we were streaming RTP/RTCP-over-TCP.  (Thanks to "zhangzx at rcs-9000.com"
  for noticing this.)

2006.08.26:
- Changed the implementation of "ByteStreamFileSource" so that reads from the
  open file are now done asynchronously, when data becomes available.  This matches
  the code's event-driven execution model better than the previous implementation,
  which read from files synchronously.  It should also perform better in systems
  that read from more than one file concurrently.
- Added Victor Kozhuhov's modifications to the WAV file support code, to allow for WAV files that contain
  u-law or a-law content (instead of the usual raw PCM).

2006.08.24:
- Fixed the way in which "RTSPServer" handles requests that span more than one
  TCP socket read.  These reads are now all handled asynchronously, in accordance
  with the library's event-driven execution model.  (Thanks to Marc Neuberger for noting this.)
- Fixed a memory leak in "MultiFramedRTPSource" that would occur if we received
  packets with duplicate sequence numbers, or if we received excessively delayed
  packets.  (Thanks to David Bertrand for noticing this.)
- Changed the order of statements in "~RTSPClient()" to fix a bug (noted by Brain Lai).
- Changed "base64Encode()" to take a 'length' parameter, so that it will work with input data that
  contains '\0' characters.  (Thanks to Derk-Jan Hartman for this suggestion.)
- Added support - in "RTSPServer" - for the (non-standard) "x-playNow:" header in RTSP requests.
  (Thanks to Rob Casey for this patch.)
- Updated "MediaSession" to parse the (standard) SDP "framerate" attribute (in addition to "x-framerate", which
  is non-standard).
- Changed the parent class of "H264VideoStreamFramer" to be "FramedFilter" (to be consistent with other
  'Framer' classes) instead of "FramedSource".  (Thanks to Jinfeng Zhang for noticing this.)
- Made the "FramedFilter" implementation more robust, in case the supplied input source parameter is NULL.

2006.08.07:
- Changed the way that "OnDemandServerMediaSubsession" creates server ports.
  It now chooses server port numbers starting with a specific port number
  (which is now an optional parameter to the "OnDemandServerMediaSubsession"
  constructor).  The default value of this initial port number parameter is 6970.
  This matches the port number range used by other common RTSP server implementations,
  including Darwin Streaming Server and Helix.
- Changed the criterion for whether to send out a RTCP "SR".  (Thanks to David Bertrand.)
- Fixed an obscure bug in "OnDemandServerMediaSubsession::deleteStream()"
  (reported by Scott Hayes)
- Removed some archaic code that was developed for a customer several years ago, but
  which is no longer used.

2006.07.04:
- Another update to "H264VideoRTPSink" (requested by David Bertrand)

2006.06.28:
- Another update to "H264VideoRTPSink".  (Thanks to David Bertrand.)

2006.06.27:
- Updated "H364VideoRTPSink"
- Corrected the definition of MAX_PES_PACKET_SIZE in MPEG2TransportStreamFromPESSource.cpp.
  (Thanks to Jiri Pinkava for this fix.)

2006.06.23:
- Fixed some minor issues with the previous addition of "H264VideoRTPSink"

2006.06.22:
- Added "H264VideoRTPSink", for transmitting H.264/RTP streams.  (Thanks to
  Benoit Quirynen and David Bertrand for funding this work.)
- Changed the default definitions of IP_ADD_SOURCE_MEMBERSHIP and
  IP_DROP_SOURCE_MEMBERSHIP (for use if they're not already defined in header files),
  to make it more likely that SSM joins will properly fail if SSM is not actually
  implemented on the host OS.
- Changed some "delete"s into "delete[]".  (Thanks to Wojciech Matyjewicz
  and Mathur for reporting these bugs.)
- Made "sapWatch a bit more robust.  (Thanks to Jiri Pinkava for this fix.)

2006.05.17:
- Fixed "OnDemandRTSPServer" so that it sets a RTCP "RR" handler,
  even if "reuseFirstStream" is True.
- Changed "RTSPServer" (and the "testOnDemandRTSPServer" demo application)
  to use a 45-second client liveness timeout, by default.  (Previously,
  no client liveness checking was being done by default.)
  (Note that client liveness timeouts are done only for unicast streams, not multicast.)
- Fixed a bug in the implementation of "BasicHashTable" that arose when
  we were using hash tables with multi-word-sized keys.

2006.05.15:
- No change from the previous release, but the installed ".tar.gz" file had
  accidentally been deleted.

2006.05.11:
- Cleaned up some code in "BasicTaskScheduler" that was allegedly causing
  problems on a Zaurus.
- Changed "TaskScheduler::rescheduleDelayedTask()" so that its "microseconds"
  parameter is a "int64_t" (to match "scheduleDelayedTask").
  (Thanks to David Bertrand for suggesting this.)
- Updated "ServerMediaSession:::generateSDPDescription()" to change
  "a=rtcp: unicast-reflection" to "a=rtcp-unicast: reflection", to
  reflect the latest version of the IETF's "draft-ietf-avt-rtcpssm"
  Internet-Draft.  (This is for Source-Specific Multicast sessions only.)
- Cleaned up "H263plusVideoRTPSink" slightly, to remove an unnecessary
  member variable.
- Removed support for the (experimental and non-standard) "X-MCT-TEXT"
  payload type.

2006.03.17:
- Fixed a bug in "RTSPClient" (that had been introduced in the 2006.02.15
  release) that was stopping RTP-over-TCP reception from working.

2006.03.15:
- Updated "MediaSession" to parse the "s=" and "i=" lines of input SDP
  descriptions, and added member functions for returning these string values.
  (Thanks to Scott Hays or this.)
- Updated "MPEG2TransportStreamMultiplexor" to use the audio track for PCR
  iff there is no video track present.
- Made a change to "liveMedia/include/H263plusVideoStreamFramer.hh" to overcome
  an error reported by some nit-picking compilers.
- Fixed a minor bug in "MPEG2TransportStreamFromESSource" (in the way that it
  handles EOF on input).

2006.03.03:
- Updated "MPEG4LATMAudioRTPSource" to allow the LATM data length field to
  be omitted from the start of each returned audio frame.  (The "faad" decoder
  (used by VLC, for example) needed this.)
- Renamed "SECOND" in the "BasicUsageEnvironment" library to "DELAY_SECOND"
  (and similarly for "ZERO", "MINUTE", "HOUR", "DAY"), to avoid an alleged
  name conflict with some other code used by VLC.  (Thanks to
  Derk-Jan Hartman for this suggestion.)
- Removed some unnecessary code from "testProgs/playCommon.cpp".

2006.02.25:
- Updated the previous change to "MultiFramedRTPSource::doStopGettingFrames()",
  to also reset the object's fields.  (Thanks to David Bertrand.)

2006.02.15:
- Changed "GetFileSize()" (defined in "liveMedia/include/InputFile.hh") and
  "ByteStreamFileSource::fileSize()" to return a "u_int64_t", to allow for
  (potentially) large files.
- Changed the "microseconds" parameter to "TaskScheduler::scheduleDelayedTask()"
  from "int" to "int64_t".  This allows for durations greater than the previous
  ~2000 second limit.
- Updated "RTSPClient" to support handling RTSP requests sent by the server.
  (Some servers send periodic requests back to the client as a keep-alive test.)
  At present, we just respond to such requests with a "not allowed" error.
  (Thanks to Khanh Mai for this change.)
- Updated "MultiFramedRTPSource::doStopGettingFrames()" to reset the
  "ReorderingPacketBuffer".  (Thanks to David Bertrand for this fix.)
- Corrected a small bug in the previous "RTSPClient" modification
  (noticed by Glen Gray).
- Fixed a bug in "ADTSAudioFileSource" when parsing the optional "crc_check"
  field.  (Thanks to Paul Li for noticing this.)

2006.01.27:
- Updated "RTSPClient" to:
  - allow the "parameterName" parameter to "getMediaSessionParameter()" to be NULL
  - add a "setUserAgentString()" member function, to allow clients to
    specify an alternative string for "User-Agent:" headers
  - parse the optional "timeout" parameter in response "Session:" headers,
    and added a "sessionTimeoutParameter()" member function to retrieve it.
  (Thanks to Glen Gray for proposing these changes.)
- Fixed "H263plusVideoRTPSource" to properly initialize two member variables
  in its constructor.  (Thanks to David Bertrand for this fix.)
- Added Bernhard Feiten's support for H.263plus parsing/framing

2006.01.24:
- Corrected the maximum PES packet size (in "MPEG1or2DemuxedElementaryStream.cpp")
  to allow for a 6-byte header.  (Thanks to Jiri Pinkava for this fix.)
- Fixed "MPEG1or2AudioStreamFramer" to avoid a potential divide-by-zero when
  it's fed certain malformed MPEG audio data.  (Thanks to Eric Peters for this.)
- Updated the top-level "Makefile.tail" to make it clear that "testProgs" can't
  be built until after the 'library' directories are built. This ensures that
  parallelizing "make"s don't do the wrong thing.

2006.01.05:
- Updated "MPEG4VideoStreamFramer" to make the parsing of MPEG-4 video a bit
  more robust.  (Thanks to Khanh Mai for this suggestion.)

2006.01.04:
- Updated "OnDemandServerMediaSubsession" to tear down the stream (thus sending
  a RTCP "BYE") when the input source ends, iff the stream does not have a known
  duration.  (If the stream has a known duration, then we keep the stream alive,
  in case a client wants to seek backwards in it.)

2005.12.30:
- Extended the previous release's fix to also work with the RTSP
  "OPTIONS" command.

2005.12.29:
- Include (if necessary) authentication headers in the initial HTTP
  "GET" and "POST" commands, if we're setting up RTSP-over-HTTP tunneling.
  This allows "RTSPClient" to work properly with some Axis cameras, which
  require authentication on these HTTP commands.

2005.12.23:
- Fixed the RTCP port number in the RTSP "Transport:" header "port=" part
  (for multicast streams).
- Updated the change that we made in version 2005.11.02 so that it also
  sets the port number correctly.

2005.12.15:
- Made calls to "select()" more robust, by checking for (and ignoring)
  EAGAIN or EINTR errors.
- Fixed a bug in "MPEG4VideoFileServerMediaSubsession" that could occur if a
  bad MPEG-4 video file (not containing any initial 'config' data) is read.
  (Thanks to Trevor Pering for noticing this.)
- Removed an undocumented, experimental and rarely-used option from "openRTSP".

2005.12.09:
- Updated the RTCP implementation so that reception stats for a SSRC are now kept
  around even after a RTCP BYE is received, in case we want to use these stats
  for QOS reports.  (However, if a SSRC is reaped due to inactivity, then the
  reception stats are deleted.)

2005.12.05:
- On some systems (notably FreeBSD), "struct sockaddr_in" includes a "sin_len"
  field, which should be set before the struct is used.  We now replace declarations
  of "struct sockaddr_in" with a macro MAKE_SOCKADDR_IN() (defined in
  "groupsock/include/GroupsockHelper.hh").  This macro sets the "sin_len" field
  iff "HAVE_SOCKADDR_LEN" is defined on the command line.
  (Thanks to Boris Nikolaus for noting this issue.)
- Fixed a bug in the timestamp generation for certain rare MPEG-4 video streams.
  (Thanks to Bernhard Feiten for reporting this.)

2005.11.30a:
- Improved the calculation of the per-Transport Packet duration in
  "MPEG2TransportStreamFramer", so that the overall transmit duration
  tries to match the client playout duration (as determined by PCRs).
- Minor patch to avoid errors when compiling with GCC 4.1.
  (Patch contributed by Hanno Boeck.)
- Patch to the special 'Kasenna'-specific support in "RTSPClient".
  (Patch contributed by Glen Gray.)

2005.11.11:
- Modified "MPEG2TransportStreamFramer" to take the PCR "discontinuity_indicator"
  flag into account when updating the estimated duration of each TS packet.

2005.11.03:
- Minor bug fix to the previous version's update to "RTSPClient".

2005.11.02a:
- Updated the previous release to add a new (optional)
  "forceMulticastOnUnspecified" parameter to "RTSPClient::setupMediaSubsession()".
  This forces the client to request a multicast stream if the original SDP
  response address was unspecified (0.0.0.0).  (Note that not all servers
  will be able to handle this.)  We also handle the "port=" field in RTSP
  "SETUP" response "Transport:" headers.
- Fixed "RTSPServer" to include a proper "port=" field in the "SETUP"
  "Transport:" response when streaming multicast.

2005.11.02:
- Updated "RTSPClient" to use the "destination=" address in the "SETUP"
  response's "Transport:" header - for multicast streams.  We do this because
  some weird servers (e.g., Axis cameras, reportedly) do not specify the
  multicast address earlier, in the "DESCRIBE" response's SDP.

2005.10.28b:
- Added a "rtpTimestampFrequency" parameter (with default value 90000)
  to "H263plusVideoRTPSink", "MPEG4ESVideoRTPSink".

2005.10.28a:
- Removed some unnecessary "delete"s, and changed some "delete"s to
  "Medium::close()"s.  (Minor bug noted by Zhixue Zhang.)

2005.10.28:
- Fixed a potential 'divide by zero' problem in "MP3Internals.cpp".
  (Thanks to Eric Peters for noting this.)

2005.10.27:
- Added a "numEntries()" pure virtual function to "HashTable".
  (Thanks to David Bertrand for this suggestion.)

2005.10.23:
- When creating a Transport Stream from AC-3 audio input, we now use a stream_id
  of 0x06 rather than 0x81.
- Add support for receiving the various G.726 RTP payload formats, described
  in RFC 3551.
- Added an implementation of the "doStopGettingFrames()" virtual function to
  "BasicUDPSource".  (Thanks to Adrian Hornsby for this suggestion.)
- Changed the reimplemented virtual functions in "BasicUsageEnvironment0" and
  "BasicTaskScheduler0" from "protected" to "public".  (Thanks to David Bertrand
  for this suggestion.)
- Added support to "RTSPClient" for the "GET_PARAMETER" RTSP command.
  (Thanks to Glen Gray for this.)
- Added a Windows version of the "genWindowsMakefiles" script.  This should
  make it easier for Windows users to unpack and build the code.
  (Thanks to Erwin Herzog for this script.)

2005.10.05:
- Fixed a couple of bugs noticed by David Bertrand.

2005.09.23:
- This software is now called "LIVE555 Streaming Media", and our preferred domain name
  is now "live555.com".

2005.09.20:
- Updated "MPEG2TransportStreamMultiplexor" to increase the frequency of PMTs
  and PATs in the output Transport Stream. 
- More improvements to "MPEG2TransportStreamFromESSource".
- Updated "RTSPServer" to allow for empty stream names from certain non-standard
  RTSP clients (such as Amino STBs).
- Fixed "RTSPClient::tearDownMediaSession()" to delete each subsession's
  "sessionId" field.
- Fixed a minor memory leak in "RTSPClient" (when implementing HTTP tunneling)

2005.09.09:
- Made the destructors for "BasicUsageEnvironment" and "BasicUsageEnvironment0"
  protected rather than public, to match the base class "UsageEnvironment".
- More improvements to "MPEG2TransportStreamFromPESSource" and
  "MPEG2TransportStreamFromESSource".

2005.09.06:
- Modified "MPEG2TransportStreamMultiplexor", and added
  "MPEG2TransportStreamFromESSource".

2005.08.26:
- Updated "MediaSession" to parse the SDP "a=type:" attribute.  (Patch by
  Derk-Jan Hartman.)
- Added support to "RTSPClient" for handling Microsoft servers' non-standard use
  of the RTSP/RTP protocol.  Also, fixed the parsing and generation of floating
  point numbers in the RTSP "Range:" and "Scale:" headers to work correctly in
  non-POSIX locales.  (Thanks to Derk-Jan Hartman for this suggestion.)
- Fixed a memory leak in "MPEG2TransportStreamFromPESSource".
- Changed "MP3StreamState::readFromStream()" from "private" to "protected", to
  satisfy a request by Pierre l'Hussiez.

2005.08.09:
- Improved the Transport Stream generation software to (in the future) allow
  for generating Transport Streams from Elementary Stream data as well as
  from PES packet data.

2005.07.23:
- Improved "H264VideoRTPSource" to properly handle 'aggregate' packets
  that contain more than one NAL unit.

2005.07.21:
- We no longer set a RTCP "RR" handler when we're streaming, unicast on-demand,
  from a shared input source.  (The reason for this is that, in this case,
  we also share a RTCP instance for all client sessions (streams).
  A better fix will eventually be needed.)

2005.07.19:
- Updated H.264/RTP support.

2005.07.15:
- Oops, DEBUG was not supposed to be defined in "RTSPServer.cpp".
- Added "-D_LARGEFILE_SOURCE=1" to the "COMPILE_OPTS =" line of the various
  "config.*linux*" files, to ensure that code that uses "fseeko()" will
  compile properly for all Linux systems.

2005.07.13:
- When handling a 'passive' session, we no longer set a RTCP "RR" handler,
  because (unlike unicast, on-demand sessions) the RTCP instance usually
  outlives RTSP client sessions.  (Thanks to Jon Sheller for reporting
  this bug.)

2005.07.12:
- Minor updates to the Base64 code for H.264/RTP receiving.

2005.07.10:
- Removed a debugging version of "testOnDemandRTSPServer" that had mistakenly been
  left in the proevious revision.  If you downloaded the previous revision, then
  please download this version instead!
- Fixed "MPEG1or2VideoStreamFramer" to better handle MPEG video streams where the
  first GOP time code has a non-zero 'pictures' count.
  (Thanks to Eric Peters for reporting this problem.)
- Some enhancements to the Base64 code, for eventual use by "H264VideoRTPSource".

2005.07.08:
- Added - to "RTCPInstance" - the ability to assign 'handler' functions
  to be called on the arrival of RTCP "SR" or "RR" packets (as well as "BYE").
  We now use this mechanism (specifically, a RTCP "RR" handler) to improve the
  optional 'liveness' test for "RTSPServer".  (Liveness is now indicated by
  either a RTSP command over the TCP connection, or the arrival of a RTCP
  "RR" packet.)
- Moved routines for Base64 encoding and decoding into the "liveMedia"
  library.  (Base64 encoding was already being used by "RTSPClient";
  Base64 decoding will soon be used by "H264VideoRTPSource".)

2005.07.01:
- Updated "testOnDemandRTSPServer" to demonstrate AAC audio streaming (from an
  ADTS-format file)
- Made some modifications to (supposedly) make the code compile better for
  WinCE.

2005.06.30:
- Added support for reading and streaming from AAC audio files in ADTS format.
  (Thanks to Manthan Systems for funding this work.)
- Removed duplicate code in "testOnDemandRTSPServer.cpp".
- Removed duplication of a common error message from "RTSPClient.cpp".

2005.06.29:
- Added support for Basic authentication (in addition to Digest authentication)
  to "RTSPClient".
- Removed a minor memory leak in "PassiveServerMediaSubsession".
- Improved "MPEG4ESVideoRTPSource" to better note when received RTP packets
  begin or end a MPEG-4 'frame'.  (Thanks to Saumya Chandra for noting the
  limitations of the previous code.)
- Changed some member functions in "PassiveServerMediaSubsession" from
  "private" to "protected", to allow subclassing.  (Thanks to
  Fabrice Aeschbacher for this suggestion.)

2005.06.16:
- Fixed a bug in the generic MPEG (1, 2 or 4) video parsing code that was causing some
  MPEG-4 streams not to be parsed correctly.
- Updated "MPEG4VideoStreamFramer" to better handle MPEG-4 video streams that contain
  'B' frames.  (Now, more accurate presentation times and frame durations are computed
  for such streams.)

2005.06.07:
- Modified the Windows-specific code in "ByteStreamFileSource.cpp" to compile properly
  under MINGW.

2005.06.04:
- Updated "MPEG2TransportStreamFramer::doStopGettingFrames()" to clear the PID status
  table.  This makes it possible to handle PCR discontinuities when seeking (e.g.)
  within a stream.
- Updated the "ByteStreamFileSource::seekToByte*()" operations to take 64-bit
  parameters, to allow for seeking within files that are >4 GB in size.

2005.05.26a:
- Updated the 'seeking' operations in "ByteStreamFileSource" to handle files
 that are > 2^31 bytes in size.

2005.05.26:
- Updated "RTCPInstance::setByeHandler()" to (by default) handle only "BYE"s that
  come from active participants.  This means, for example, that a receiver of a
  multicast stream can now quit without its RTCP "BYE" being handled by other
  multicast receivers.  (By default, multicast receivers will now handle only
  "BYE"s that come from the sender.)
- "RTSPServer" now includes "rtptime" in the returned "RTPInfo:" header, because
  the bugfix in the previous release also happened to fix the problem that this was
  having with QuickTime Player.

2005.05.24:
- Fixed "RTPSink" so that it now resets "fTimestampBase" only on the first time
  that a timestamp is ever computed; not on the first time after "startPlaying()"
  is called.  (Thanks to David Bertrand for noting this.)

2005.05.23:
- Added support to "RTSPServer" for (optionally) reclaiming client connection
  state (and stopping the stream) if no command has been received on the
  RTSP TCP connection within a specified period of time.  This is useful for
  use in closed systems where it is known that the RTSP client always uses
  the "GET_PARAMETER" command as a periodic 'keep-alive'.  It stops the
  stream from continuing indefinitely if the client suddenly dies.

2005.05.12:
- Removed some unnecessary floating point code from "MP3Internals.cpp".
  (This was causing unnecessary slowness on some hardware without floating point.)

2005.05.09:
- Modified "MPEG2TransportStreamFramer" to give greater weight (0.5) to more recent
  data when computing the estimate of per-transport-packet duration.  (This causes
  it to respond more accurately to sudden changes in bitrate (i.e., in VBR streams).)
- Added some (otherwise unnecessary) null destructors to stop gcc 4.0 from printing
  warning messages.  (Thanks to Emiliano Parasassi for pointing this out.)

2005.05.05:
- "RTSPServer" now uses separate buffers for requests and responses.  (This fixes
  an issue that arose in the handling of requests from a set-top box.)

2005.05.04:
- Updated "RTSPServer" to make the parsing of "CSeq:" headers in RTSP requests
  more robust.

2005.04.26:
- Made sure that "MultiFramedRTPSource"s properly stop receiving incoming RTP
  packets, when they're asked to stop.  (This fixes a bug that was being triggered
  by some streams that continued sending RTP packets after a RTCP "BYE".  Thanks to
  Eberhardt Garner for helping track this down.)
- Fixed a minor memory leak in "openRTSP".
- Updated "FramedSource.hh" to no longer (by default) support the old form of
  the 'after getting' function.
  
2005.04.23:
- Added "H264VideoRTPSource", for receiving H.264/RTP streams.
  (Thanks to Erik Hellerud for this.)
- Made some minor modifications to "ServerMediaS(ubs)ession",
  to support a customer's project.

2005.04.22:
- Improved "MPEG1or2VideoRTPSink" to allow more than one complete video 'slice'
  from the same picture to be packed into an outgoing RTP packet.
- Eliminated a bogus "IP_ADD_MEMBERSHIP" error message that Windows (for some
  reason) sometimes triggers for no reason.

2005.04.20:
- Changed the name of the (platform-independent) "_close()" function to
  "closeSocket()", to prevent an apparent function name conflict in Windows.

2005.04.13:
- Modified "ourSourceAddressForMulticast()" to try the
  "gethostname()"/"gethostbyName()" method if the 'multicast loopback' method fails.
- Made some modifications to "ServerMediaSubsession" to support "SIPServer"
  (whose implementation is currently in progress).

2005.04.07:
- Fixed a bug in the "base64Encode()" routine in "RTSPClient.cpp".
  (Thanks to Raphael Rigo for noticing this.)
- Updated "MPEG1or2VideoStreamDiscreteFramer" to handle the "iFramesOnly" and
  "vshPeriod" parameters (just as the original "MPEG1or2VideoStreamFramer"
  already does).

2005.03.31:
- Removed some testing code that had accidentally been left in the previous
  release.  If you use "openRTSP", then please use this release instead of the
  previous one. 
- Fixed a bug in "RTPServer" when handling streams without stream names.
  (The bug had been introduced in version 2005.03.28.)

2005.03.30:
- Improved the support for seeking within RTSP/RTP-streamed MPEG-1 or 2
  Program Stream files.  (A/V sync after a seek still isn't perfect, though.)

2005.03.28:
- The "RTSPServer" implementation now supports raw-UDP streaming, for those
  clients that request it.  (This is usually requested only by some specialized
  clients - such as set-top boxes - and only for data such as MPEG Transport
  Streams where audio/video is muxed together (and no RTP timestamps are needed).)
- Updated the calculation of MP3 sampling frequencies to allow for 'MPEG2.5'
  (Thanks to Massimo Buffo for this fix.)

2005.03.23:
- Changed "BasicTaskScheduler::SingleStep()" to call only one read handler during
  each iteration of the event loop.  This prevents potential problems if
  the event loop is called reentrantly from within a read handler.
- Added a new, alternative version of "ByteStreamFileSource::createNew()" that
  takes an open file (FILE*) as parameter, instead of a file name.
  (Thanks to Mike Yan for this suggestion.)
- Updated the top-level Makefile to build the lower-level directories using
  "$(MAKE)" rather than "make".  This allows a command other than "make"
  (e.g., "gmake") to be used.  (Thanks to Sergio.Gelato for this fix.)
- Minor changes to (apparently) make Sun's C++ compiler happy.
  (Thanks to Sergio.Gelato for these.) 
- Some changes to "RTSPServer" to make it more tolerant of some strange, non-standard
  clients.
- Support seeking within RTSP/RTP-streamed MPEG-1 or 2 Program Stream files.

2005.03.11:
  Updated "MPEG2TransportStreamFromPESSource" once again to better handle
  Program Streams that include a "program_stream_map".

2005.03.10:
- Modified "BasicUDPSource" to make its OS socket receive buffer at least 50 kBytes
  (the same as we do for "MultiFramedRTPSource").

2005.03.07:
- Updated "MPEG2TransportStreamFromPESSource" (and thus also the
  "testMPEG1or2ProgramToTransportStream" demo application) to handle MPEG
  Program Streams that include MPEG-4 Elementary Stream data.  (For this to work,
  the Program Stream must include a "program_stream_map".)

2005.03.05:
- Modifying "ReceivingInterfaceAddr" now works before receiving
  unicast streams.  (Previously, it worked only when receiving multicast
  streams.)  Also, added a new option "-I <interface-ip-address>"
  to "openRTSP", to allow the user to specify which interface is used.
  (Thanks to Luca Abeni for this patch.)
- Fixed a "delete[]" vs "delete" bug in "MPEG4VideoStreamFramer",

2005.02.28:
- Fixed a bug (that had been introduced in the 2005.01.07 version)
  in the MP3 frame<->ADU conversion code.

2005.02.25:
- Fixed some minor memory leaks.
- Made some more changes to prepare for RTSP server seeking within MPEG-1 or 2
  Program Stream files.  (This is still not done yet.)

2005.02.14:
- Changed the generation of "rtptime" in "RTPInfo:" RTSP headers to use 
  %u instead of %d.  (Thanks to Regis Feneon for noting this.)
- Made some changes to prepare for RTSP server seeking within MPEG-1 or 2
  Program Stream files.  (This is still not done yet.)

2005.02.09:
- Updated "MultiFramedRTPSink" to allow for frame-specific special headers to
  appear before each frame in RTP packets.  (This in addition to the (possible)
  special header that may appear at the start of the packet.)  This functionality
  is added using a new virtual function "frameSpecificHeaderSize()" (that,
  by default, returns 0.)  Thanks to Matt Romaine for this.
- Added a new "BufferedPacket" virtual member function
  "getNextEnclosedFrameParameters()", which will (eventually) replace the existing
  "nextEnclosedFrameSize()" virtual member function.  This (plus a corresponding
  change to "MultiFramedRTPSource") now makes it possible to give correct 
  presentation times to multiple frames within incoming RTP packets.

2005.02.07:
- The "MultiFramedRTPSink::setPacketSizes()" function now applies just to one
  specific "MultiFramedRTPSink" object; not to all such objects.
- Added a new version of "RTSPServer::removeServerMediaSession()" that takes
  the (string) stream name - rather than the "serverMediaSession" object
  - as argument.

2005.01.29:
- We now remove reception and transmission statistics records (from each
  "RTPSource" and "RTPSink", respectively), whenever each RTCP SSRC membership
  gets 'reaped'.  We also added more information to "RTPTransmissionStats".
  (Thanks to WIS Technologies for funding this work.)
- Some minor fixes to "JPEGVideoRTPSource".  (Thanks to Sergey Khlutchin for
  reporting this.)
- Made the "ServerMediaSession" constructor "protected", to allow for
  subclassing.  (This was requested by Eric Peters.)
- Updated "win32config.Borland" and modified some header files, so that the
  code can be built using Borland's "C++ Builder 5.0".
  (Thanks to David Wu for this suggestion.)
- Added a "setPacketSizes()" function to "MultiFramedRTPSink" to make it
  possible to change the 'maximum' and 'preferred' size of outgoing
  RTP packets.  (Thanks to Christian Gerstner for this suggestion.)

2005.01.24:
- Some MIME-type parameters in SDP "a=fmtp:" lines are Boolean, taking only values
  0 or 1.  Sometimes these parameters are set (to 1), without an explicit "=1"
  being in the SDP "a=fmtp:" line.  (This is not legal, but it sometimes occurs.)
  Our SDP parsing code (in "MediaSession.cpp") now checks for this.

2005.01.23:
- More improvements to "AVIFileSink".  MPEG-1, 2 or 4, JPEG and H.263 video is
  now supported, along with raw PCM or u-law audio.  (However, audio is not
  yet working perfectly, and MPEG audio is currently not supported at all.)
  (Thanks to WIS Technologies for funding this work.)
- Changed the (many) calls to "gettimeofday()" to pass NULL as the 'timezone'
  parameter, because this isn't used.

2005.01.13:
- Fixed a bug in "liveMedia/OnDemandServerMediaSubsession.cpp" that was reportedly
  causing crashes on some systems.  (Thanks to Brian Wang for finding this.)
- Updated "MPEG1or2VideoStreamDiscreteFramer" and
  "MPEG4VideoStreamDiscreteFramer" to set appropriate presentation
  timestamps for B-frames.  (Unlike I and P-frames, B-frames' timestamps are
  not monotonically increasing.)  (Thanks to WIS Technologies for funding this work.)

2005.01.07:
- Fixed a bug in "JPEGVideoRTPSink" that was causing incorrect packets to
  be generated when streaming using non-standard JPEG quantization tables
  (which must be included in the RTP packets).
- Updated the handling of incoming MP3 ADUs to allow for the possibility of
  receiving ADUs that contain 'ancillary data' at the end.  (For example,
  ADUs for "mp3Pro" frames are like this.)  Note, however, that our
  MP3 frame-to-ADU generating code currently doesn't include 'ancillary
  data' at the end of the generated MP3 ADUs.  This will need to be fixed
  in order to be able to correctly *transmit* ADUs for "mp3Pro" frames.
- More work on "AVIFileSink".  Recording of MPEG-4 video-only files now works.

2004.12.29:
- Added a new liveMedia 'sink' class - "AVIFileSink" - for writing an AVI-format
  file.  Note: This is not fully working yet, so don't try using it yet!
  (Thanks to WIS Technologies for funding this work.)

2004.12.23:
- Fixed a bug in the previous release (when receiving MPEG-2 Transport RTP streams).

2004.12.22:
- Updated "MPEG4GenericBufferedPacket::nextEnclosedFrameSize() to (i) allow
  for "generic" mode, and (ii) properly check for the absence of an
  "AU Header section".  (Thanks to Erick van Rijk for this suggestion.)
- When receiving a MPEG-2 Transport Stream (either RTP *or* raw-UDP), pass it
  through a "MPEG2TransportStreamFramer", so that "durationInMicroseconds" is
  set appropriately, based on the stream's embedded PCR values.
  (Thanks to Dermot McGahon for this suggestion.)
- Changed the "num_packets_lost" statistics line (printed by "openRTSP -Q")
  to format as an int rather than as an unsigned.  This is in case this value
  is negative (which can happen if duplicate packets are received).
  (Thanks to Norbert Donath for noting this.)

2004.12.15:
- Changed the RTSP servers in the various test programs to use port 8554
  instead of port 7070 for RTSP (as an alternative to the standard port
  554).  It turns out that port 8554 is the official IANA-reserved port
  number for RTSP (alternative).
- Changed "RTSPClient" to include the "Scale:" header if the scale is being changed
  to 1.0 from something else.  (Thanks to Matt Romaine for this suggestion.)

2004.12.09:
- Fixed a bug in the handling of "audio/X-MP3-DRAFT-00" RTP streams.
  (Thanks to Dhananjay Deshpande for noticing this.)

2004.12.07:
- Updated "RTSPClient" to parse the "Scale:" header (if any) in a response to
  a "PLAY" command, and set a corresponding field in the "MediaSession" or
  "MediaSubsession" structure.

2004.12.06:
- Fixed a minor bug in "BasicTaskScheduler::SingleStep()" that would be triggered
  if the (optional) "maxDelayTime" parameter were unusually large.
- The "RTSPServer" implementation of RTP/RTC-over-TCP streaming will now work even
  if the client's RTSP "SETUP" command doesn't include an "interleaved=" field.

2004.11.30:
- Added "RTSPServer" support for the RTSP "Scale:" header, which is used to
  implement 'trick play': fast forward and reverse play.  The actual implementation
  of these depends on the media type; it is currently implemented for MPEG-1 or 2
  (including MP3) audio (forward play only), and for WAV (PCM) audio (forward or
  reverse play).
  (Thanks to Sony Corporation for funding this work.)

2004.11.26:
- Improved "H263plusVideoRTPSource" to better recognize which packets begin
  a frame.  (This should improve its performance on lossy networks.)
- Improved the parsing of MPEG-1 or 2 audio streams to ignore more bogus
  'syncwords'.
- We now look for the 'Xing' VBR Table of Contents, and use this (if present)
  to implement seeking within VBR files.
- Updated "ADUFromMP3Source" to allow the internal data buffers to be flushed
  (e.g., if there is a discontinuity in the MP3 input data).
- When streaming a seekable MPEG audio stream on demand, insert filters that
  convert the input stream to ADU format, and then back to MP3.  This allows
  us to seek within the stream without being tripped up by the MP3
  'bit reservoir' (back-pointer).

2004.11.20:
 - Fixed a bug in "MediaSession::initiateByMediaType()".  (It was not allowing
   for the possibility that a subsession had already been initiated.)

2004.11.19:
- For consistency (and ease of comparison), the "MIMEtype()"
  codec name substring for each "MediaSource" subsession is in upper case.

2004.11.18:
- Added an optional "scale" parameter to the "RTSPClient" "play...()"
  methods, to allow a client to specify fast forward or rewind 'trick play'.
  (Thanks to Dermot McGahon for this suggestion.)

2004.11.16:
- Fixed a bug in the way that "QuickTimeFileSink" records the duration of
  multi-channel audio tracks in ".mov" or ".mp4"-format files.
  (Thanks to Orban/CRL Inc. for funding this work.)

2004.11.11a:
- Another minor update to "RTSPClient".

2004.11.11:
- Updated the "RTSPServer" implementation to handle the case where a "PLAY"
  request does not contain a "Range:" header.  (Suggestion by Brian Wang.)
- Reinstated the "MPEG4LATMAudioRTPSource" fix from version "2004.08.24".
  (It had accidentally gotten deleted.)
- Included Dermot McGahon's patch to "RTSPClient" to support the bogus
  non-standard 'Kasenna' variant of RTSP.

2004.11.09a:
- Fixed the previous revision (RTSP seeking support) to work properly with VLC.
- Fixed a bug that was causing receivers of SSM streams to not receive RTCP packets.  (Thanks to Alessandro Gaiarin for reporting this.)

2004.11.09:
- Added support for seeking within streams.  This currently works only for
  WAV audio file streams, and MPEG audio (include MP3) file sources.
  (However, it doesn't yet work properly for VBR MP3 files; this will get fixed.
  Also, it doesn't yet work properly with VLC.)
  (Thanks to Sony Corporation for funding this work.)

2004.11.06:
- Improved the way that we specify output buffers for "RTPSink" objects.
  We now do this using a "maxSize" static member variable (instead of 
  using a 'number of packets' variable as we did before).  We also limit
  the size of the output buffer used for RTCP packets, to save space.
- When streaming a MPEG Transport Stream, don't set the RTP 'M' bit.
- When recording a ".mov" or ".mp4" file using "QuickTimeFileSink.cpp",
  we no longer fill in the "width" and "height" fields for audio tracks.

2004.11.04:
- When streaming MPEG audio or WAV audio files on demand, we now return the
  correct file duration (in the RTSP "Range:" header and the SDP "a=range:"
  attribute), so that media players will report the correct stream duration.
  (Thanks to Sony Corporation for funding this work.)
- In "MultiFramedRTPSink", we now wait until we receive the first data before 
  initializing "fNextSendTime" with the current wall-clock time.
  (Thanks to Eric Peters for suggesting this.)

2004.11.02:
- Updated the RTSP "Range:" header (in a "PLAY" response) and the SDP "a=range:"
  attribute to support streaming from sources with finite durations.
  (Such sources can support seeking via RTSP.)  The "ServerMediaSubsession"
  class now has a "duration()" virtual function.  (In the future,
  "FileServerMediaSubsession" subclasses will redefine this, so that seeking
  within the file will work.)
  (Thanks to Sony Corporation for funding this work.)

2004.11.01:
- Added support to "RTSPClient" for streaming RTSP/RTP over HTTP, using the technique
  described in Apple's document: <http://developer.apple.com/documentation/QuickTime/
QTSS/Concepts/chapter_2_section_14.html>.
  This can be used for streaming from a Darwin Streaming Server from behind a
  HTTP-only firewall.  (Note that our own RTSP server implementation doesn't yet
  implement RTSP-over-HTTP, but this is on the 'to do' list.) 
  (Thanks to Orban/CRL Inc. and WIS Technologies for funding this work.)

2004.10.28a:
- Fixed "OnDemandServerMediaSubsession" so that "PAUSE"/"PLAY" works properly on
  unicast streaming from files.

2004.10.28:
- Removed (perhaps temporarily) the "rtptime=" parameter from the "RTP-Info:" header
  that's returned in response to the RTSP "PLAY" command.  It appears that this may
  be messing up A/V sync in QuickTime Player.

2004.10.26:
- Updated "JPEGVideoRTPSink" (and "JPEGVideoSource") to allow for the
  possibility of sending custom quantiziation tables in JPEG/RTP packets.
- Fixed some compile warnings about unused named parameters.  (Thanks to
  Thiago Correa for these suggestions.)

2004.10.22:
- Added a new "MPEG1or2VideoStreamDiscreteFramer" class, similar to the existing
  "MPEG4VideoStreamDiscreteFramer".

2004.10.21:
- Modified the order of SDP lines generated by a RTSP server
  (in "ServerMediaSession.cpp") to correspond to the exact order specified by
  the SDP RFC.  JMIF clients are supposedly anal retentive about this.
  (Thanks to Fabrice Aeschbacher for noticing this.)

2004.10.19:
- Updated "MPEG2TransportStreamFramer" to properly handle input data that
  doesn't begin with a 'sync' byte (0x47).

2004.10.18:
- Added a new "removeServerMediaSession()" member function to "RTSPServer".
- Updated the implementation of the RTSPServer "RTPInfo:" header to include a
  "rtptime=" field.
- Improved the "WindowsAudioInputDevice" trick for making the microphone the
  first-listed device.  (Thanks to Syncanph Xie for this suggestion.)

2004.10.14:
- Added a new argument "-4" to "openRTSP" (with corresponding updates to
  "QuickTimeFileSink"), to generate a "MP4"-format file.  (Currently, this
  works for MPEG-4 audio only.)
  (Thanks to Orban/CRL Inc. for funding this work.)

2004.10.13:
- Fixed a typo that was causing the Windows version not to build.
  (Thanks to Thiago Correa for noticing this.)

2004.10.11:
- Updates to "RTSPServer":
  - Include a "Date:" header in each response
  - Include a "Range:" header in each "PLAY" response.
  - Include "RTP-Info:" header in each "PLAY" response.
- Made "SimpleRTPSink" more idiot-proof.

2004.10.07:
- Improved the parser in "MPEG4VideoStreamFramer" to be more tolerant of
  bad/unexpected data.
- Moved the code for opening input files by name (which includes a special-case
  hack for "stdin") to a single common file - "InputFile.cpp" - to get rid of
  duplicated code.

2004.10.04:
- Fixed "UserAuthenticationDatabase::addUserRecord()" (in "RTSPServer.cpp") to
  make a copy of the "password" string, before adding it to the hash table.
  (The hash table already makes a copy of the "username" (key) string.)

2004.10.03:
- Added support for receiving raw UDP streams (in addition to normal RTP/UDP streams)
  (Thanks to Derk-Jan Hartman)

2004.09.30:
- Updated the RTSP server "UserAuthenticationDatabase" to (i) allow for it to
  be subclassed, and (ii) allow it to store passwords that are really
  md5(<username>:<realm>:<password>)

2004.09.24:
  Added support for access control (using digest authentication) to "RTSPServer".
  Also, added example code to the "testOnDemandRTSPServer" test program to illustrate
  how to use this. (Thanks to Orban/CRL Inc. and WIS Technologies for
  funding this work.)

2004.09.22:
- Moved the digest authentication support out of "RTSPClient.cpp"
  (and "SIPClient.cpp") into a new file: "DigestAuthentication.cpp".  There's also
  a corresponding new header file: "DigestAuthentication.hh".  This is in preparation
  for adding digest authentication support to our RTSP server implementation also.

2004.09.20:
- More improvements to the multi-unicast streaming support (from a single input
  source).  RTP-over-TCP streaming is now supported in this case.  Updated
  "testOnDemandRTSPServer" to add a variable "reuseFirstSource" (default: False)
  that can be used to implement this.

2004.09.09:
- Made a minor improvement to the multi-unicast streaming support.

2004.09.08:
- Added optional "author" and "copyright" parameters to "DarwinInjector::setDestination()".
- Added an optional "miscSDPLines" parameter to "ServerMediaSession::createNew()",
  to allow the caller to add extra SDP lines to the session description.
- Fixed a minor bug in the media tables reclamation code.
- Added still more support for multi-unicast RTSP/RTP streaming from a
  single input source.  (Thanks to Orban/CRL, Inc. and WIS Technologies for funding
  this work.)

2004.09.05:
- We now more gracefully handle malformed "m=" SDP lines.  (Thanks to
  Derk-Jan Hartman for noticing this problem.)
- In "RTSPClient", we now make the "CSeq" number a static variable, to avoid
  potential problems if we connect to the same server with the same URL
  more than once.  (Thanks to Keith Gurganus for the suggestion.)
- Modified "RTSPServer" to use the destination interface IP address in
  its "rtsp://" URL.

2004.09.02:
- Modified "GroupsockHelper.cpp" to allegedly support building for WinCE.

2004.08.26:
- Fixed a bug that could cause "AMRAudioRTPSource"s to not get reclaimed.
  (Thanks to Chenglim Ear for reporting this.)
- More support for multi-unicast RTSP/RTP streaming from a single input source.

2004.08.24:
- Modified "LATMAudioRTPSource" to include the initial data length field
  in the data that's delivered to a client.  (A decoder was needing to see this.)
- Added a new "MPEG4VideoStreamDiscreteFramer" class, as a (more efficient)
  alternative to "MPEG4VideoStreamFramer" when the input source is a sequence of
  discrete MPEG-4 frames, rather than a byte stream.
- Fixed a minor bug in "Media.cpp". (Thanks to Thiago Correa.)
- Added more support for multi-unicast RTSP/RTP streaming from a single input
  source.  (This now working, except for TCP connections - to be completed.)

2004.08.17:
- Made the parsing of MPEG-1 or 2 video more robust, in case a Video Sequence Header
  doesn't appear exactly where it should.
- Added initial support for multi-unicast RTSP/RTP streaming from a single input
  source.  (This isn't fully working yet - more support needs to be added.)

2004.08.13:
- Added a "numChannels" parameter to "MPEG4GenericRTPSink" (optional)
  and "MPEG4LATMAudioRTPSink".

2004.08.12:
- Fixed a couple of small memory leaks (noticed by Gabriel Bouvigne).

2004.07.31:
- Added support for MPEG-4 LATM audio RTP streaming.  (MPEG-4 LATM audio RTP
  *receiving* had already been implemented.)  (Thanks to Orban/CRL, Inc. for
  funding this work.)

2004.07.27:
- Updated the MPEG Program-to-Transport Stream conversion mechanism to set the
  correct stream tags depending on whether the input data is MPEG-1 or 2.

2004.07.23:
- Added a "-O" (upper-case letter 'oh') option to "openRTSP".  This tells the
  program to not send an initial "OPTIONS" request prior to "DESCRIBE".
- Modified "BasicTaskScheduler" to better allow subclassing.
- Added an optional "maximum delay time" option to
  "BasicTaskScheduler0::SingleStep()".  This allows subclasses to impose a
  limit on how long "select()" can delay, in case it wants to also do polling.

2004.07.22:
- Fixed a serious bug that was causing some MPEG Program Stream files to be parsed
  incorrectly - especially if they have non-MPEG header data at the front.
- Modified the way in which "Medium" objects are reclaimed, to overcome an
  obscure bug.

2004.07.20:
- Added a comment to "DeviceSource.cpp" to clarify the purpose of "fTo", because
  several people have gotten this wrong.
- Made ~BasicUsageEnvironment protected (just like ~UsageEnvironment)
- Fixed "samplingFrequencyFromAudioSpecificConfig()" to remove a memory leak.

2004.07.16:
- Fixed a bug in "AMRAudioRTPSource", in the way that it handled input RTP packets
  that contained more than one AMR frame.
  (Thanks to Gabriel Bouvigne for reporting this problem.)
- We now automatically reclaim the memory that was allocated for the "UsageEnvironment"
  "liveMediaPriv" and "groupsockPriv" structures, once their tables become empty.

2004.07.14:
- Updated "QuickTimeFileSink's" MPEG-4 audio recording mechanism again, this time
  to support aacPlus as well as regular AAC.

2004.07.13:
- Updated "QuickTimeFileSink" to support recording MPEG-4 audio (including
  hint tracks).
- Modified "MPEG4VideoStreamFramer" to allow simplified subclasses that take
  discrete frames as input (and therefore don't need a full parser).

2004.07.07:
- Added the (rather bogus) "a=x-qt-text-nam:" and "a=x-qt-text-inf:" attributes
  to the SDP descriptions delivered by "RTSPServer" (in response to a "DESCRIBE"
  operation).  These attributes get the same strings as the "s=" and "i=" lines
  (respectively).  However, QuickTime Player doesn't do anything with those lines,
  but does recognize the "a=x-qt-*" lines.

2004.07.06:
- Modified the code so that it will build for WinCE.
  (Thanks to Gabriel Bouvigne for these changes.)

2004.07.02:
- Fixed a bug in HashTable::RemoveNext() (noted by Greg Bothe)
- Added partial support for recording MPEG-4 audio in "QuickTimeFileSink".
  (Hinting is not yet working properly for this media type.)
- Added an option to "MPEG1or2AudioStreamFramer" to allow its presentation times
  to be resynchronized by its input source.

2004.06.18:
- Fixed a bug in "MPEG1or2AudioStreamFramer" (in the way it was checking for
  syncwords at the start of each frame).
- Updated the "MPEGVideoStreamFramer" (base) class to report "fNumTruncatedBytes".
- Increased "OutPacketBuffer::numPacketsLimit" from 20 to 30.

2004.06.17:
- Another improvement to the "MPEG1or2Demux" parsing code.
- Updated "QuickTimeFileSink" to support recording MPEG-4 video tracks into
  ".mov" files.  (There's also basic support for recording MPEG-4 audio, but
  this isn't quite working yet.)

2004.06.14:
- Updated "MPEG1or2Demux" to make the checking for a PACK_START_CODE more
  robust if it's not immediately present where we expect to see it.
- Fixed a bug in "JPEGVideoRTPSource" that could cause a problem on
  64-bit architectures.
  (Thanks to Andrey Filippov for noticing this.)

2004.06.11:
- Removed an error from a error message in StreamParser.

2004.06.09:
- Updated the implementation of the aggregate and non-aggregate RTSP "PLAY"
  operations to not include a "Range:" header if the 'start' parameter is < 0.
  (This is used when resuming from a "PAUSE", for example.)
  Also, removed an unnecessary "Range:" header from the "PAUSE" command.
  (Thanks to Gabriel Bouvigne for this suggestion.)
- Reimplemented "MultiFramedRTPSource::doGetNextFrame1()" to use iteration
  rather than recursion, in the case where a large frame is fragmented over
  multiple incoming RTP packets.  This avoids the possibility of stack overflow
  occurring if there are very large fragmented frames in the input data.

2004.06.07:
- Added optional "start" and "end" parameters to the aggregate "PLAY" operation
  in "RTSPClient".  (We had already done this for the non-aggregate "PLAY" operation.)
- Made the input banks buffers in "StreamParser" heap-allocated, rather than making
  them member variables.  This avoids any potential compiler problems with having
  such large structures as member variables.

2004.06.03:
- Added the new 'liveMedia' class "MPEG2TransportFileServerMediaSubsession", for
  unicast on-demand streaming of MPEG-2 Transport Stream files.
- Added the new test programs "testMPEG2TransportStreamer" and
  "testMPEG1or2ProgramToTransportStream", and added support for Transport File
  streaming to "testOnDemandRTSPServer".
  (Thanks to DVAgroup Inc. for funding this work.)

2004.06.02:
- Fixed "MediaSession" to no longer apply the normal RTP 'M' bit rule
  when processing incoming "video/MP2T" (or "video/MP1S" or "video/MP2P")
  streams.  (For such streams, the 'M' bit does not mean 'end of frame'.)

2004.06.01:
- Updated "RTSPClient" to handle "Session:" headers that have a trailing
  ";timeout=" part.  (Thanks to Gabriel Bouvigne for suggesting this.)

2004.05.31:
- We now handle SDP "m=" lines with an optional <number-or-ports> field
  (although we don't actually do anything with this value).  Thanks to
  Derk-Jan Hartman for this suggestion.

2004.05.29a:
- Made a correction to the previous change.

2004.05.29:
- Added a call to "DarwinInjector" to set the TCP socket's send buffer to 100 kBytes.

2004.05.28:
- Renamed some variables in "MP3Internals.cpp" (and made them static) to
  overcome a name conflict seen by the VLC developers.  (Thanks to
  Derk-Jan Hartman for pointing this out.)

2004.05.26:
- Fixed a bug (apparently introduced with recent header file changes) that
  could cause MP3 files to not be read correctly on Windows.  (In particular,
  this could cause "testMP3Streamer" to not work correctly on Windows.)
  Thanks to David Skiba for the bug report.

2004.05.24:
- Added a new member function "lastReceivedSSRC()" to "RTPSource".  This makes it
  possible for a receiver to demultiplex incoming RTP data based on SSRC.
  (Later this functionality should perhaps be added to "(MultiFramed)RTPSource"
  instead.)
- More updates to "DarwinInjector".

2004.05.19:
- Another update to "openRTSP" for a customer job.

2004.05.17:
- Added a new test program "testMPEG4VideoToDarwin.cpp" that is similar
  to "testMPEG4VideoStreamer", except that instead of transmitting the
  video RTP/RTCP packets via multicast, it transmits them - via a RTSP TCP
  connection - to a remote Darwin Streaming Server.

2004.05.12:
- Added a new "liveMedia" class - "DarwinInjector" - that can be used to send an
  audio and/or video RTP/RTCP stream to a remote Apple 'Darwin' (aka. QuickTime)
  Streaming Server, for playing by (potentially multiple) RTSP clients.
- Added a new test program "testMPEG1or2AudioVideoToDarwin.cpp" that is similar
  to "testMPEG1or2AudioVideoStreamer", except that instead of transmitting the
  audio/video RTP/RTCP packets via multicast, it transmits them - via a RTSP TCP
  connection - to a remote Darwin Streaming Server.

2004.05.09:
- Removed a line of junk that had erroneously been left at the start of Makefile.tail.
- The "testRelay" test program has been reprogrammed in the usual liveMedia
  'source->sink' style.

2004.05.07:
- Another update to RTSPClient for a customer job.

2004.05.04:
- Added an #include to "BasicUDPSink.cpp" to prevent a compilation problem on
  some platforms.

2004.05.02:
- Some updates to RTSPClient and openRTSP for a customer job.

2004.04.27:
- Fixed the round-trip time estimation routine in "RTPSink" to allow for
  the possibility of the computed round-trip time being negative.  This can
  happen if there is clock drift between the sender and receiver, and if the
  actual round-trip time was quite small.  If this happens, the returned
  round-trip time value is zero.
  (Thanks to Dixon Siu for alerting us to this issue.)

2004.04.23:
- Updated some #ifdef's so that the code will compile with "mingw" on Windows.
  (Thanks to Joey Parrish for this.)
- Added a "BasicUDPSink" class to "liveMedia", for UDP streaming without RTP.
- Added new 'filter' classes to "liveMedia" for
  - converting MPEG Program Streams to Transport Streams.
  - 'framing' MPEG Transport Streams to return the 'duration' of each Transport
  packet (i.e., the inter-packet time gap).

2004.04.09:
- Fixed a byte-ordering bug in the code for streaming from WAV audio files
  (used in "testWAVAudioStreamer" and "testOnDemandRTSPServer").  We were converting
  16-bit samples from host to network order before streaming them.  That was wrong,
  because audio samples are always stored in WAV files in little-endian order.
  The correct thing to do was to convert from little-endian to big-endian order.

2004.03.27:
- Fixed a minor bug in the top-level Makefile.  (Thanks to Matteo Nastasi for
  noting this.)

2004.03.27:
- Modified "H261VideoRTPSource" so that it returns individual packet payloads to
  the caller, rather than waiting to form complete frames.  Also, added a member
  function "lastSpecialHeader()" that returns the 4-byte special header for the
  most recently-read payload.

2004.03.23:
- Added support for unicast RTSP/RTP streaming from VOB files, and added an example
  of this to "testOnDemandRTSPServer".

2004.03.17:
- Made a series of minor bug fixes and improvements suggested by Sony.

2004.03.15:
- Fixed a bug in "OnDemandServerMediaSubsession" (the RTCP instance was being
  deleted at the wrong time).  (Thanks to Clark Taylor for noticing this.)
- Minor modifications to the 2004.02.26 "RTPSink" changes.
- Updated the "AMRAudioRTPSource" implementation to support 'bandwidth-efficient'
  mode.  (Warning: This has not yet been tested.)

2004.03.12:
- Updated the "RTSPClient" class to support an aggregate "PAUSE" operation.

2004.03.11:
- Added an implementation of AMR audio RTP receiving ("AMRAudioRTPSource"), to
  match the existing"AMRAudioRTPSink".  Also, added a new class "AMRAudioFileSink"
  (a subclass of "FileSink") that outputs an AMR audio input stream to a file,
  including the file 'magic number' and frame headers.  "openRTSP" was also updated
  to output received AMR audio streams using "AMRAudioFileSink"s.
  (Thanks to SIGOS Systemintegration GmbH for funding this work.)
- Cleaned up the "RTPSource" changes that were made in version 2004.02.26.

2004.03.05:
- Updated the "RTSPServer" implementation to properly handle aggregate
  operations for which the URL ends with a "/".  Some clients, such as
  "gmp4player" do this.

2004.03.03a:
- Fixed a bug in "MultiFramedRTPSource.cpp" where we were not properly
  skipping over "contributing source" fields, if they were present in the
  RTP header.  (Thanks to Phillip Bruce for noticing this.)

2004.03.03:
- Fixed the "testMP3Streamer" test program to keep the RTP sink, RTCP instance,
  Groupsocks, and RTSP server (if used) open across all iterations of the loop.
  (Thanks to Frank Xia for noticing this problem.)

2004.03.02a:
- The previous version accidentally had the built-in RTSP server enabled in
  "testMP3Streamer". It should be disabled by default.

2004.03.02:
- Fixed a few potential errors that were found by running "valgrind" on some of
  the test programs (on Linux).

2004.03.01:
- Added two new options to "openRTSP" (and "playSIP"):
  "-E <max-inter-packet-gap-time>"
      Tells the program to close the stream if no new packets have been
      received in at least <max-inter-packet-gap-time> seconds.
  "-B <input-socket-buffer-size>"
      Tells the program to set the network socket input buffer (for each
      input stream) to <input-socket-buffer-size> bytes.  (This can be useful
      when testing QOS for different input buffer sizes.)
  (Thanks to SIGOS Systemintegration GmbH for funding this work.)

2004.02.26:
- Added - to "RTPSink" - a database of statistics from incoming RTCP
  "Reception Report" (RR) packets.  (This mirrors the similar database
  for RTCP SR packets that was already part of "RTPSource".)  (Thanks to
  Clark Taylor for contributing this code.)

2004.02.23:
- Fixed a bug in "MP3ADUinterleaving.cpp" that was causing frame durations
  to not be set properly when handling interleaved MP3 ADUs.

2004.02.20a:
- Corrected a typo in the previous "#ifdef"s

2004.02.20:
- Added "#ifdef"s to the two files that #include <strstream.h>, so that they
  will compile OK with GCC v3.*.  (Thanks to Goetz Waschk for this.)
- Changed the stream parser debugging statements to make it clear that the
  use of C++ language exceptions by the stream parsing code is normal, and
  is not an error.

2004.02.19:
- Updated the "RTSPServer" implementation to support RTP-over-TCP streaming
  (if requested by the client).

2004.02.13:
- Changed the implementation of the "BasicUsageEnvironment" "operator<<" 
  member functions so that they use "fprintf(stderr, ..." rather than
  "cerr << ...".  People compiling in some environments were getting
  complaints about "cerr" being undefined (presumably because the right
  #include files weren't being found).  However, everyone should have
  "stdio.h".  (Thanks to Clark Taylor for this suggestion.)
- Changed the "genMakefiles" script to complain if the user tries to
  run it without an "<os-target"> parameter.

2004.02.09:
- Updated "RTSPServer" once again - this time to ensure that each incoming request
  is read completely (i.e., up until the trailing <CR><LF><CR><LF>).
  (Thanks to Regis Feneon for helping to fix this.)
- Added a new "WAVAudioFileServerMediaSubsession" class to support on-demand
  streaming from WAV audio files.  Also, updated the "testOnDemandRTSPServer"
  test program to illustrate this.

2004.02.05:
- Updated the "WindowsAudioInputDevice" project to build two different versions
  of the library: One that uses Windows' built-in mixer; another that doesn't.

2004.02.04:
- Fixed an obscure bug in "MPEG4ESVideoRTPSink" that could (very rarely)
  cause incorrect RTP timestamps to be set for the last fragmented packet
  of large MPEG-4 frames.  This could cause QuickTime Player's video playback
  to freeze.

2004.02.03:
- Added new "liveMedia" classes "MPEG1or2FileServerDemux" and
  "MPEG1or2DemuxedServerMediaSubsession" for supporting on-demand RTSP/RTP
  streaming of MPEG-1 or 2 Program Stream files.  Also, updated the
  "testOnDemandRTSPServer" test program to support streaming of such
  a file.
- We no longer set SO_REUSEPORT for stream (TCP) sockets, because it usually
  doesn't make sense to share TCP ports.  (As a side effect, this means that it
  will no longer be possible to run more than one RTSP server (using the same
  TCP port) simultaneously on the same computer - something that should never have
  been possible in the first place.)
- Fixed a minor RTSP server bug that was causing incorrect "destination"s to be set
  (for multicast streams) in the "SETUP" response "Transport:" header.

2004.01.28:
- Yet another bug fix to "MPEG4VideoStreamFramer", to properly handle the
  case where MPEG-4 "GOV" headers don't occur on exact 'second' boundaries.
  (Thanks to Michael Niedermayer (via Michael Hess) for clarifying this.)

2004.01.27:
- Fixed a bug in the previously added (but normally "#ifdef"d out)
  debugging code in "MultiFramedRTPSource.cpp".
- Made a fix to the (normally "#ifdef"d out) debugging code in
  "AC3AudioStreamFramer.cpp".  (Thanks to Yigal for noticing this.)

2004.01.24:
- Changed "openRTSP" to use 20 kByte "FileSink" receive buffers by default,
  instead of 10 kBytes.  (Some frames in MPEG-4 streams were exceeding the
  previous limit.)
- Added code to "MultiFramedRTPSource.cpp" to easily let the developer
  simulate packet loss.

2004.01.23:
- Changed "RTSPServer" to include a "Content-Base:" header in "DESCRIBE"
  responses.  This overcomes a bug in QuickTime Player, which sends incorrect
  "SETUP" URLs otherwise.
- Fixed a bug in "MPEG4VideoStreamFramer" that was causing some frames
  to erroneously be given a duration of 0.

2004.01.22:
- Made another improvement to the "RTSPServer" implementation.  It once again
  properly handles streams without a stream name (i.e., where the "streamName"
  parameter to "ServerMediaSession::createNew()" was NULL.
- Updated the (little-used) "ByteStreamMultiFileSource" class to take
  "preferredFrameSize" and "playTimePerFrame" parameters, just like
  "ByteStreamFileSource".  Also added a member function to ask whether
  a new file has just been read from.

2004.01.21:
- Made several improvements/bugfixes to the "RTSPServer" implementation:
  - Worked around an apparent bug in QuickTime Player: It sometimes
  doesn't include a proper URL in RTSP "SETUP" requests.
  - Corrected the 'frame' (really tick) rate and frame duration that
  are computed for 'fixed_vop_rate' MPEG-4 video streams.
  - Compensated for some buggy MPEG-4 video streams that don't set
  timestamp information properly.
- MPEG-4 video end codes will now be included in the outgoing RTP stream,
  and received properly by RTSP clients.
- Fixed "RTSPClient" to be more tolerant of RTSP "Transport:" headers that
  contain more than one successive ';' character.  (Darwin Streaming Server
  sometimes does this.)

2004.01.19:
- Changed "RTSPServer" to not create per-session state when handling
  "DESCRIBE", but instead to wait until "SETUP".  This fixes a problem
  that was causing the "VLC" media player to fail to work with our
  "RTSPServer" implementation, because VLC uses one RTSP TCP connection
  to do the "DESCRIBE", and then another to do "SETUP" and "PLAY".
  (Thanks to Emmanuel Dufour for pointing this out.)
- Updated "MPEG4VideoStreamFramer" to include the VIDEO_SEQUENCE_END_CODE
  in the output stream if it sees it in the input.

2004.01.10:
- Fixed the "RTSPServer" implementation to properly return a "server_port" field
  in the RTSP "SETUP" response for unicast streams.  Also, the server ports
  (RTP and RTCP) are now chosen separately, and will no longer be the same
  as the client ports if the client and server are running on the same machine.
  (That did not work on some Linux systems.)
- Fixed a bug in "MP3ADU" that was causing frame durations to not be set properly
  when translating between MP3 frames and ADU frames.

2004.01.09:
- Modified "RTSPClient" to handle embedded NULL characters in a SDP
  description.  (These are not legal, but they have been seen in some cases.)

2004.01.06:
- Modified the FramedSource 'after getting' function signature to take two new
  parameters: (1) The number of bytes (if any) that were truncated from the
  delivered data, and (2) The duration (in microseconds) of the delivered frame.
  For backwards compatibility with old code, the old 'after getting' function
  signature is still supported, but will be removed in some future version of
  the code.  (The "getPlayTime()" function was also removed.)

2003.12.26:
- Made "FramedFilter" objects implement the "stopPlaying()" function by
  also calling "stopPlaying()" on their source object.
- Fixed an obscure bug in "RTSPClient".

2003.12.20:
- Made the "video/H263-2000" MIME type use the same RTP payload format as
  "video/H263-1998".  (Thanks to Norbert Doneth for suggesting this.)

2003.12.19:
- Fixed a minor bug in "RTSPServer" (some allocated memory wasn't being
  freed properly).  (Thanks to Konstantin Lunin for finding this.)

2003.12.16:
- Added "-DSOCKLEN_T=socklen_t" to the "COMPILE_OPTS =" line in
  the file "config.solaris", so that the code will compile
  correctly for 64-bit Solaris systems.

2003.11.25:
- Put some error checks back into "WindowsAudioInputDevice".

2003.11.21:
- Some case-insensitive string comparisons weren't working properly in Windows; fixed.

2003.11.19:
- Added a global Boolean variable to flag when RTP I/O over a TCP connection fails.
  This hack allows higher-level code to check for this.

2003.11.16:
- Fixed "RTPInterface" to not infinite loop when trying to read from a TCP
  connection that has closed.
- Modified "BasicTaskScheduler" to overcome a bug in Windows that sometimes
  causes "select()" to fail.

2003.11.13:
- Made "WindowsAudioInputDevice" a bit more robust against broken audio drivers.

2003.11.06:
- Modified "groupsock/inet.c" to allow the system-supplied "random()" and "srandom()"
  functions to be used instead of our own "our_..." implementation.
  This is done if USE_SYSTEM_RANDOM is defined (e.g., in the "config.*" file).

2003.11.06:
- Modified the "MPEGVideoStreamParser" constructor, in response to a complaint
that some compilers would choke on the old code.

2003.10.30:
- Added a warning output message to "MultiFramedRTPSink::afterGettingFrame1()"
  whenever the input frame appears to exceed the maximum buffer size.  This
  maximum buffer size can be increased by changing
  "OutPacketBuffer::numPacketsLimit".

2003.10.27:
- Changed the "MultiFramedRTPSource" error message (added in 2003.10.04) to apply
only to frame sizes > 1000, so that (e.g.) QCELP RTP sources don't trigger it.

2003.10.24:
- Fixed the RTSP server implementation to put SDP lines in the correct order
(as defined by the SDP speecification).  Also, added "Cseq:" lines that were
missing from "404" error responses.  (Thanks to Bill May for noting these bugs.)
- Improved the implementation of MPEG-4 video framing and streaming.

2003.10.08:
- Fixed a bug in the previous "FileSink" update that was causing "openRTSP"
to crash when recording a MPEG-4 video stream (using the "-m" option).
(Thanks to Jiangzhou Scu for noticing this bug.)

2003.10.07:
- Updated the "FileSink" class to take an optional Boolean parameter
"oneFilePerFrame".  If set, a separate file will be output for each
incoming frame.  The frame's presentation time is used as a
file name suffix, to distinguish the files.
- Added a new option "-m" to "openRTSP" (and "playSIP").  This option
causes a separate file to be written for each frame (using the
above-mentioned new feature of "FileSink").

2003.10.05:
- Renamed the new openRTSP/playSIP "-S" option to "-b".

2003.10.04:
- Added an error message to "MultiFramedRTPSource" to report when
the client's receive buffer is too small for an incoming frame.
(This could happen with large JPEG frames, for example.)
- Updated the signature to "FileSink" to take an optional 'buffer size'
parameter as argument.  (The default value is 10000.)
- Added a "-S <file-sink-buffer-size>" option to "openRTSP" and "playSIP".

2003.09.30:
Improved the default implementation of
"RTPSink::hasBeenSynchronizedUsingRTCP()" to more accurately report
whether or not the most recently-delivered packet's presentation time
was synchronized using RTCP SRs.  (Thanks to Luca Abeni for pointing
out the possibility of a problem with the old implementation.)

2003.09.25:
Improved the support for unicast RTSP/RTP streaming.  Also, added a new
test program "testOnDemandRTSPServer" that demonstrates how to use this.

2003.09.19:
- Added support for unicast streaming from a "RTSPServer".
  (A test program that demonstrates this will be available shortly.)
  (Thanks to TNO Fysisch en Elektronisch Laboratorium (Netherlands)
  for funding this work.)
- Modified the "strDup()" function to return NULL (instead of crashing)
  if passed a NULL parameter.

2003.09.11:
- Another improvement/bugfix to the "MultiFramedRTPSink"
  implementation.
- Changed the implementation of "RTSPServer" and "ServerMediaSession"
  in preparation for implementing unicast streaming from RTSP servers.
  (This is something that's not ready yet, but close...)

2003.09.05:
Fixed the implementation of "JPEGVideoRTPSource" so that it prepends
a proper JFIF JPEG header to the start of each incoming JPEG frame.
This means that programs such as "openRTSP" and "MPlayer" will now
receive/play JPEG/RTP streams correctly.

2003.09.03:
Improved the implementation of "MultiFramedRTPSink" to more efficiently
handle the case where input frames get broken up into multiple outgoing
RTP packets.  We now eliminate an uncessary "memmove()" for each
fragment.  In particular, this should make MPEG-1 or 2 and motion-JPEG
video streaming more efficient.

2003.09.01:
- Added support for reading and streaming AMR audio files (as defined in
RFC 3267).  Also added a new test program "testAMRAudioStreamer".
(Thanks to TNO Fysisch en Elektronisch Laboratorium (Netherlands) for
funding this work.)

2003.08.28a:
- When "openRTSP" writes out a recorded MPEG-4 Elementary Stream video file,
it now writes SDP 'config' information to the front of the file beforehand.
This 'config' information contains VOL etc. headers that can be useful
for playback.

2003.08.28:
- Added 'framer' and RTP sink classes for MPEG-4 Elementary Stream video.
(Thanks to TNO Fysisch en Elektronisch Laboratorium (Netherlands) for
funding this work.)
- Added a new test program "testMPEG4VideoStreamer".
- Fixed a bug with RTP-over-TCP reading that could cause a crash if the
same TCP socket number gets reused for a later stream.  Also, made this
code more thread-safe.

2003.08.21:
Renamed all of the MPEG-1 or 2 class and program names from "MPEG*" to
"MPEG1or2*".  This is to distinguish these from the classes and programs
that use MPEG-4.

2003.08.19:
Made the maximum buffer size in "MediaSink.cpp" a parameter that can be 
set (changed) at runtime.  (Reset the default back to 20* packet size)

2003.08.18:
- Fixed "JPEGVideoRTPSink" to use the standard RTP payload format code
(26) for JPEG RTP streams.
- Updated "MediaSink.cpp" to increase the maximum buffer size for incoming
data.  This makes it possible to stream large JPEG frames (which are
fragmented over multiple outgoing RTP packets).

2003.08.16:
- Added new classes "AudioRTPSink" and "VideoRTPSink" as parent classes
of the various audio and video RTP sink classes, respectively.
- Added a new "JPEGVideoRTPSink" class, and a "JPEGVideoSource" class
that can be used as an abstract base class for particular JPEG sources.

2003.08.07:
Minor change to "MediaSink".

2003.07.29:
Updated "RTSPClient" to include a "tearDownMediaSession()" operation,
in addition to "tearDownMediaSubsession().  "openRTSP" now uses just
the "tearDownMediaSession()" operation.  (Darwin Streaming Server was
barfing with "tearDownMediaSubsession()").  Thanks to Edward Estabrook
for suggesting this.

2003.07.27:
Added a new "-F <fileName-prefix>" option to "openRTSP" (and "playSIP").
This can be useful if you are running "openRTSP" several times,
in the same directory, to play several different RTSP streams
(e.g., from a script).  (Thanks to Norbert Donath for this suggestion.)

2003.07.14:
Updated the "WAVAudioFileSource" implementation (and the
"testWAVAudioStreamer" test program) to better report any reason for
a WAV file to be invalid.

2003.07.13:
- Added new filter classes for converting between 16-bit PCM and
8-bit u-law audio, and between host and network order for 16-bit values.
(See "liveMedia/include/uLawAudioFilter.hh" for details.)
- Added a new class "WAVAudioFileSource" for reading/processing
WAV audio files.
- Added a new test program "testWAVAudioStreamer" that reads a WAV
audio file - "test.wav" - and streams it via multicast (using a built-in
RTSP server).
- Updated "MediaSession" to recognize "L8" and "L16" RTP audio streams
(to be handled by "SimpleRTPSource").

2003.07.07:
- Fixed a bug that would cause "vobStreamer" to crash if (incorrectly)
run with no arguments.
- Updated the "test*Streamer" test programs to print out the "rtsp://"
URL of the built-in RTSP server (if it's enabled).

2003.07.01a:
Updated the "WindowsAudioInputDevice" subdirectory to:
- Change the name of the library from "WindowsAudioInputDevice.lib" to
"libWindowsAudioInputDevice.lib".
- Added a console test program - "showAudioInputPorts" - that lists
the audio input ports that are currently available.

2003.07.01:
Added a new subdirectory "WindowsAudioInputDevice", which is a
Windows-specific implementation of the "AudioInputDevice" class.
This can be used by Windows applications to read PCM audio from
a sound card.

2003.06.28:
Added an optional parameter (default, True) to "SimpleRTPSource"
that says whether to use the RTP "M" (marker) bit to indicate the
last (or only) fragment of a video frame.  (Suggested by
Cezar Plesca.)

2003.06.27:
Added a "AudioDeviceSource" class to the "liveMedia" library.
This class is a generic audio input device (such as a microphone
or a sound card) - to be subclassed for specific OS platforms.

2003.06.26:
- Updated the "RTSPServer" implementation to respond to "PAUSE"
requests.  (It doesnt actually pause the stream; just returns "OK".
This makes QuickTime Player happier.)
- Added a "-o" option to "openRTSP" (and modified "RTSPClient"
accordingly).  This option causes openRTSP to send an "OPTIONS" command
to the server, and prints out the response.

2003.06.17:
Added support for MPEG-4 generic RTP sinks.  (Currently, only the
"AAC-hbr" audio mode is supported, and only one AAC frame is packed
into each outgoing RTP packet.)

2003.06.13:
- Fixed a bug in "ServerMediaSubsession" that could cause a memory leak.
(Thanks to Frederik Bonte for finding this.)

2003.06.07:
- Fixed a minor bug in "MPEGVideoRTPSink": The RTP "M" bit is now set
only on the last fragment, if a frame is fragmented.  (Thanks to
Tym Altman for pointing this out.)
- Improved support for AudioRTPSinks that have more than one audio channel.

2003.05.31:
Fixed a bug in "SimpleRTPSink" that would cause it to not generate correct
RTP timestamps on outgoing packets.

2003.05.29:
Fixed a bug in "RTSPServer" that would cause it to get into an infinite
loop if a client's connection was terminated unexpectedly.

2003.05.28:
Fixed some typos in comments.

2003.05.23:
Some minor changes to reduce the number of bogus warning messages
displayed when compiling using Visual C++.

2003.05.22b:
Fixed a bug in "SIPClient" (thanks to Reini Urban for helping track this
down).

2003.05.22a:
Fixed a couple more minor bugs that were causing "valgrind" to complain.

2003.05.22:
Fixed a bug in "SIPClient" (a field was not getting initialized properly)

2003.05.21:
- Because not all platforms define 'errno' the same way, 'errno' is now
implemented as a pure virtual function "getErrno()" in
"UsageEnvironment".  "BasicUsageEnvironment" defines this function to
just return "errno", but other subclasses could implement it in other
ways.
- Added support for setting and reading information in "RTP-Info:"
headers in RTSP "PLAY" response. (Thanks to Romulus Grigoras for
contributing this.)

2003.05.19:
Fixed a bug in "SIPClient" that could cause a crash.  (Thanks to
Reini Urban for finding this.)

2003.05.16:
- Removed all calls to "fprintf(stderr, ...)" and "cerr << ..." from
the library code.  Instead, we now use "operator<<" virtual functions
that are defined on the "UsageEnvironment".  "BasicUsageEnvironment"
defines these by outputting to "cerr", but other subclasses of
"UsageEnvironment" can define them to do console output whichever
way they wish.

2003.05.15:
- Moved the "select()" call in "readSocket()" (GroupsockHelper.cpp) into
  a separate function, to allow it to be easily reimplemented
  if desired.
- More minor changes to "RTPSource" stats handling, and the "-Q" option
  to "openRTSP"/"playSIP".

2003.05.12:
- Restructured "BasicTaskScheduler" as two hierarchical classes:
  "BasicTaskScheduler0" (an abstract base class), and
  "BasicTaskScheduler".
  This makes it easier to subclass (e.g., to reimplement "SingleStep").
  A subclass reimplementation would use "BasicTaskScheduler0".
- Removed "our_bcopy()", and use "memmove()" instead, because everyone
  seems to implement that.
- Fixed a bug in "RTPSource" that could cause packet loss stats to be
  reported incorrectly if the first RTP sequence number received was 0.

2003.05.06:
Minor changes to the "openRTSP" QOS stats reports.

2003.05.05:
Changed the "start" and "end" parameters to
"RTSPClient::playMediaSubsession()" from int to float, for increased
granularity.  (Suggestion by Cezar Plesca.)

2003.05.03:
- Added support for pausing/resuming to "RTSPClient".
(Thanks to Romulus Grigoras for contributing this.)
- Fixed a couple of bugs in "SIPClient".

2003.04.28:
- Cleaned up and improved the "DelayQueue" implementation.
(This time it works properly, I hope.)

2003.04.27:
- Temporarily backed out the previous change to "DelayQueue", because it
broke MPEG video streaming (causing it to play too slowly).

2003.04.26:
- Made "DelayQueue" a little more accurate, by adjusting for the time
that has elapsed since the last alarm, when adding a new entry to the queue.
- Fixed a bug in "RTPSource" that could cause packet loss statistics to not
be reported accurately (in RTCP) if incoming packets were misordered.
- Updated "openRTSP" and "playSIP" to take a new "-Q" option, which
prints out QOS data at the end of the session.

2003.04.24:
- Added a mechanism for registering an optional, auxilliary read handler
with a "RTPSource" or "RTCPInstance".  Such a handler would get called
after each new packet is read.
- Changed "SimpleRTPSink" to (by default) set the RTP "M" bit on
video streams iff the packet contains the last (or only) fragment of
a frame.

2003.04.23:
- More improvements/fixes to the handling of SSM sessions.
- Fixed a bug in RTSPServer that could cause a crash if the client
quit suddenly.

2003.04.20:
- Updated "MediaSession" to recognize
  a=source-filter: incl ...
lines in SDP descriptions (for SSM sessions), and to do the appropriate
SSM-style multicast joins in this case (and also to send back RTCP via
unicast).
- Updated the RTCP implementation to reflect incoming unicast RTCP packets
back to the multicast group in the case where we're a SSM source.
- Update "vobStreamer" and the various "test*Streamer" test programs to
properly behave as SSM sources, when we have specified this.

2003.04.18:
- Updated the "*ServerMediaSession" classes to (optionally) output
SSM-specific information in SDP descriptions.
- Updated "vobStreamer" to choose a random multicast address in the SSM
range.

2003.04.12:
- Fixed "MPEGVideoStreamFramer" to allow for the possibility of
the GOP "time_code" field remaining unchanged in the source stream.
- Elimination of more unnecessary global variables.

2003.04.11:
- Added support for an optional user name and password inside a RTSP
or SIP URL.
- The media lookup table (used in "Media.cpp") is now allocated
dynamically, and stored in the "UsageEnvironment", rather than
being a global variable.
This makes it possible to have different threads, within the same
address spaces, accessing the library code.
(Each thread would need to have its own "UsageEnvironment" and
"TaskScheduler", though.)

2003.04.09:
Changed "BasicTaskScheduler::SingleStep()" to execute only one delay
queue task (along with any data input events) during each iteration.
This prevents a possible livelock if the delay queue always has events
outstanding.  (Thanks to Ruth Sadler for pointing this out.)

2003.04.04:
- More improvements to "vobStreamer" (which has also been renamed
from "VOBStreamer".

2003.04.02:
- Made another change to the order that things are closed when exiting
"openRTSP", to help avoid a potential race condition that can occur
when using the "-t" option.  (Thanks to Romulus Grigoras for reporting
this.)
- Updated "MPEGDemux" to properly handle the case where someone tries
to read from it after its input source has already closed.
- Several improvements to "VOBStreamer".
- Added "ByteStreamMultiFileSource" to the "liveMedia" library.
This generalizes "ByteStreamFileSource" to allow more than one input
file to be read, in sequence.

2003.04.01:
Fixed a bug in "openRTSP" that could cause it to crash when exiting
(if the "-t" option was used).

2003.03.31:
Added a new test program "VOBStreamer" to "testProgs".
This program reads a "VOB" file (e.g., from a non-encrypted DVD),
and streams the component video (MPEG) and audio (AC3) component streams
using RTP multicast.
(Online documentation to follow...)

2003.03.30:
Added "AC3AudioRTPSource" and "AC3AudioRTPSink".

2003.03.28:
- Fixed a bug in "MPEGVideoStreamFramer" that would cause it to generate
incorrect presentation times if the MPEG video stream started out with
a non-zero 'time_code'.
- Updated "AC3AudioStreamFramer" to allow clients to get the stream's
sampling rate before it reads the first frame.

2003.03.25:
Began adding support for AC3 audio/RTP streaming.

2003.03.22:
Added support for receiving H.261/RTP streams

2003.03.14:
- Removed calls to "strdup()", because that's a C-library function whose
result should not be deleted using "delete" (or "delete[]").  Instead,
we now provide our own C++ equvalent, called "strDup()".
- Also, changed several "delete"s to "delete[]".
(Thanks to Bill Kain for noting the need for this fix.)

2003:03;11:
Updated SIPClient to allow it to use an arbitrary source port number.
This makes it possible to use it on the same host on which a SIP server
is already running.

2003.03.09:
- Made "MediaSession"s parsing of SDP descriptions more robust, to allow for
blank lines in the SDP description.  ("Be liberal in what you accept...")

2003.03.07:
- Added an optional "MIME subtype" parameter to "SIPClient", to allow
SIP "INVITE"s to use a dynamic RTP payload format.  Updated "playSIP"
accordingly, to add an optional "-D <mime-subtype-name>" argument.
- Modified "FileSink" to close the source (and stop playing) if it gets
an EOF when writing to the output file.  (Thanks to Ruud Schramp.)

2003.03.03:
Improved the implementation of "MPEG4GenericRTPSource".

2003.02.28:
- Fixed a bug that would cause RTSP clients to not set the server port
number (for RTCP) correctly in some situations.  (Thanks to Alex Pollard.)
- Fixed a bug in "RTSPServer" that would cause "Transport:" headers to not
get generated correctly.

2003.02.27:
- Improved the "MultiFramedRTPSource" implementation to more cleanly handle
packet loss in the case where frames are split into multiple RTP packets.
- Parsed the optional "/<num-channels>" parameter that can appear at the
end of "a=rtpmap:" lines for audio sessions.

2003.02.17:
More work on the experimental '-R <rtsp-url>' option to "playSIP"
(and "openRTSP") that allows it to inject the incoming stream into
a separate RTSP server.  This code is now working, although 
currently only for a single PCMU or GSM audio stream.  Also, RTCP
packets are not yet relayed between the source and destination.

2003.02.16:
Improved RTSPClient and SIPClient to check for (and discard)
any '\r' or '\n' that appears at the start of a response message.
(Some weirdo servers can do this.)

2003.02.10:
Began adding a new option to "playSIP" (and "openRTSP") to allow the
incoming media stream to be injected into a separate, destination
RTSP server.  This still needs lots of work...

2003.02.08:
Fixed the "JPEGVideoRTPSource" implementation to properly prepend a
synthesized JPEG header to each received frame of RTP data.  This now
makes it possible to receive and play motion-JPEG RTP streams.

2003.02.06a:
- Fixed a bug in the SDP "config" attribute parsing.  (This affected
the use of "MPEG4LATMAudioRTPSource".
- Cleaned up the "createNew()" routines in each of the *RTPSource classes.
- Added two new RTPSource classes - "MPEG4ESVideoRTPSource" and
"MPEG4GenericRTPSource", for MPEG-4.  However, these have not been
fully implemented yet.

2003.02.06:
Fixed some code that was 64-bit-unsafe.  (Thanks to Philipp Thomas for
noticing this.)

2003.02.05:
- Cleaned up "RTSPClient", and added a method for doing an aggregate
"PLAY" operation (in addition to the existing method that does a
"PLAY" on an individual subsession).
- Modified the "openRTSP" test program so that it does a single
aggregate "PLAY", rather than a series of non-aggregate "PLAY"s, one for
each subsession.  (RealNetworks' server doesn't support the latter.)
- Added a new routine for parsing "AudioSpecificConfig" strings that
can appear in SDP descriptions.
- Added Morgan Multimedia's implementation of "JPEGVideoRTPSource"
to the "liveMedia" library.  (I haven't yet tested this.)

2003.02.04:
Removed "-DUSE_OUR_BZERO=1" from "config.linux", because it no longer
seems to be necessary, and it was breaking compilation for some people.

2003.02.03b:
Added an #include to "BasicUsageEnvironment/Lock.cpp" to fix a problem
that someone encountered with NULL not being defined.

2003.02.03a:
Created a new header file "groupsock/include/NetCommon.h" that contains
all networking-related #includes.  This also does the correct #includes
for Windows, allowing Windows code to use Winsock-2 rather than Winsock-1.
Also, modified "win32config" to use a "TARGETOS" of WINNT rather than
WIN95, to make this all work.

Thanks to Doug Kosovic for figuring out the magic incantations
needed to get this all working.

2003.02.03:
- Improved "SIPClient" (in the "liveMedia") library, so that "INVITE"
requests are retransmitted, as necessary, for reliability.
- Added a "-A <rtp-payload-format>" option to "playSIP".  This allows the
user to specify which audio RTP payload format should be received.
(At present only static payload formats can be specified, and only
audio codecs - not video.)

2003.01.28:
- Added support (in the "liveMedia" library) for a basic SIP client
- Added a new test program "playSIP", similar to "openRTSP".  These two
applications are now built from the same code base.

2003.01.17:
Further improvements to the MPEG-4 audio support.  We can now read
(from the "MediaSubsession") the StreamMuxConfig "config" stream that
was present in the SDP description.  Also added a routine that parses
such a string, producing binary "AudioSpecificConfig" data.
(See "MPEG4LATMAudioRTPSource.cpp" for details.)

2003.01.16:
Improved support for receiving MPEG-4 LATM audio.  Note that currently,
this works only for streams that do not have a StreamMuxConfig present
in the stream.

2003.01.10:
Added basic support for MPEG-4 LATM audio.

2002.12.21:
Fixed a bug in the previous update to "QuickTimeFileSink"

2002.12.20:
- Another change to "QuickTimeFileSink", again to change the way that track
durations are computed for ".mov" files.  Now, each track's duration is
set to be the maximum of (i) the sum of the sample durations listed
in the 'stts' atom, and (ii) the the sum of the durations listed in the
track's 'edit list' (if any).
- Updated the MP3 file parsing code to check for a 'Xing' VBR header in
the first frame.  ("liveCaster" can use this information to print more
sensible information about each file that it streams.)

2002.12.10:
Yet another change to "QuickTimeFileSink", this time to make sure that
the duration of video tracks corresponds exactly to the sum of video
sample durations listed in the 'stts' atom.

2002.11.30:
Renamed "TaskScheduler::blockMyself()" to "doEventLoop()", to better
describe what this member function actually does.

2002.11.25:
Another change to "QuickTimeFileSink".  Now, when synchronzed tracks
are requested, the durations of video frame samples are adjusted so
that they correspond to actual RTP presentation times.  (This is not
done for audio samples, however, because having audio samples vary
in duration might break audio codecs.)

2002.11.22:
Updated "QuickTimeFileSink" once again - this time to add more statistics
to the 'udta'/'hinf' atom for each generated hint track.

2002.11.18:
Modified the support for creating hint tracks in "QuickTimeFileSink" so
that it it now works properly for H.263+ video sessions.

2002.11.15:
- Made a small modification to "groupsock/GroupsockHelper.hh" to ensure that
it compiles on all platforms.
- Modified "MP3FileSource" so that it can handle layer I or layer II files,
as well as layer III (MP3).

2002.11.14:
Updated "QuickTimeFileSink" to support (optional) hint tracks in output
QuickTime files.  Also added a "-H" option to "openRTSP" that
(when used with "-q") will add hint tracks to the output ".mov" file.
(Note that the hint tracks currently don't work well for H.263+ video
tracks, and currently don't work at all for QCELP.)

2002.11.04:
Updated the expiration date in "openRTSP.cpp"

2002.10.22:
- Undid the previous change.  The top-level Unix Makefile now does
      cd <dir> ; make
once again.  Unfortunately "--directory" doesn't seem to work properly
with the version of "make" that's installed with FreeBSD.
- Changed the implementation of the "-n" option in "openRTSP".  If "-y"
is also specified, then the user will be notified only if (i) data has
arrived for all subsessions, and (ii) all subsessions have been
synchronized.

2002.10.21:
Changed the top-level Unix Makefile to do
	make --directory=<dir>
rather than
       cd <dir>; make
as the latter apparently didn't work with cygwin.
(Thanks to "Sycotic" Smith for the tip.)

2002.10.20:
- Updated "QuickTimeFileSink" once again to improve the implementation of
synchronized audio/video tracks in output QuickTime files.
- Added a configuration file for "cygwin".  (Thanks to "Sycotic" Smith
for working on this.)

2002.10.19:
Updated "QuickTimeFileSink" to use QuickTime Edit Lists to synchronize
the media tracks.  (It appears that this does not always produce playable
QuickTime files, so this implementation may have to be changed later.)

2002.10.11:
Fixed a bug in "MultiFramedRTPSource" that could cause a RTP receiver to
hang if it received a malformed RTP packet with no data after a special,
media-specific header.

2002.10.10:
Fixed the "RTSPClient" implementation so that it now understands
"Content-Length" as well as "Content-length".  (This now allows it
to work with our own RTSP server implementation once again!)

2002.10.07:
Improved the implementation of RTP (and RTCP) reception to eliminate an
unnecessary "memmove()" each time a packet is received.

2002.10.04:
Updated the "liveMedia" library to support optionally sending and receiving
RTP and RTCP packets over a TCP connection (e.g., the TCP connection used
for a RTSP session).  Also, updated the "openRTSP" test program to take
an optional "-t" argument, meaning: stream over TCP.

2002.10.01:
Fixed the modification to "MultiFramedRTPSource" that was made in the
2002.08.29 release, and also made a corresponding change to
"QuickTimeFileSink", so that it now correctly records H.263+/RTP streams
once again.

2002.09.30:
- Removed the "Media::addNew()" member function, because it was always
  called each time a new Media object was created.  Instead, its
  function was just moved into the Media::Media() constructor.
- Made another fix to "testMPEGAudioVideoStreamer".

2002.09.28:
Fixed a bug in "testMPEGAudioVideoStreamer" that could cause a crash when
its input file was being read for the 2nd or more time.  (Also updated
"liveMedia/FramedSource.cpp".)

2002.09.27:
Updated "QCELPAudioRTPSource" so that "hasBeenSynchronizedUsingRTCP()"
returns True only after a full interleave cycle of RTP packets has been
received.  This ensures that when it returns true, the receiver will
be reading a frame that came from a synchronized RTP packet.

2002.09.26:
- Fixed QCELPAudioRTPSource so that it now returns correct presentation
timestamps on each frame that is read from it (even if the input data
was interleaved).
- Updated QuickTimeFileSink to correctly write ".mov" files for half-rate
QCELP audio tracks.

2002.09.25:
Added a new test program - "sapWatch" - that reads and prints SDP/SAP
announcements (sent to the default SDP/SAP directory)

2002.09.24:
- Fixed a bug in "QCELPAudioRTPSource"
- The RTCP implementation now properly recognizes incoming RTCP
reports from other processes on the same computer.  (Previously, these
would be rejected as being loop-back packets.)

2002.09.19:
- Updated "RTPSource"s so that an accurate "presentationTime" variable
is now returned whenever a client reads from such a source. These times
are kept accurate by RTCP "Sender Report" packets sent by the sender.

2002.09.13:
- Improved the "BasicHashTable" implementation so that it's
no longer a quick-and-dirty hack.
- Changed the parameter signature to "ByteStreamFileSource" so that the
"playTimePerFrame" parameter is now an unsigned (microseconds) instead
of a float.  This eliminates a round-off problem.

2002.09.11:
Modified "ByteStreamFileSource::doGetNextFrame()" so that - if the
"playTimePerFrame" parameter was set - the presentation time gets set
based on this, rather than the current 'wall clock' time.

2002.09.06:
Updated "win32config.Borland" based on feedback by Vesselin Kostadinov.
(Also updated an #ifdef in "liveMedia/include/Media.hh")

2002.09.05:
Added code to ignore an "EAGAIN" error on a 'groupsock' read.
(This can happen in Linux.)

2002.09.03:
Fixed a bug in the RTSP server implementation that could
have caused a memory smash.

2002.08.30:
Added header files containing version strings (and corresponding integers)
for each library.  Client code can use these to print out version
information, or check for version compatibility.

2002.08.29:
Modified "MultiFramedRTPSource" and its subclasses to properly handle
the case where a frame is fragmented over several successive incoming
RTP packets.  Each subclass's implementation of "processSpecialHeader()"
can choose between having incomplete fragments returned immediately to
the client (the default behavior), or waiting until all of the fragments
have arrived.  The implementation of "H263plusVideoRTPSource" was changed
to do the latter (because the current "ffmpeg" H.263+ decoder relies upon
getting complete frames for input).

Added an optional "applicationName" parameter to "RTSPClient::createNew()",
and updated the RTSP client implementation to add a "User-Agent:" field,
which will contain this "applicationName" string (if present).

2002.08.27:
Added an optional Boolean parameter "iFramesOnly" to
"MPEGVideoStreamFramer::createNew()".  If this parameter is True,
then the framer object will return only data from "I" frames.
(By default, this parameter is False, meaning that all frames will
be returned.)  This parameter can be used to reduce the bandwidth
of streamed MPEG (1 or 2) video.

2002.08.06:
Changed "groupsock/GroupsockHelper.cpp" to do a "#include <errno.h>"
rather than "extern int errno".  The latter was apparently breaking in
some versions of Linux.

2002.08.05:
Added support for the video/MP2P MIME type (RFC 2250 MPEG Program Streams)

2002.06.25:
Source file distribution copied from live.sourceforge.net