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 no 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.

​