Index index by Group index by Distribution index by Vendor index by creation date index by Name Mirrors Help Search

live555-devel-2021.05.22-1.1 RPM for ppc64

From OpenSuSE Ports Tumbleweed for ppc64

Name: live555-devel Distribution: openSUSE Tumbleweed
Version: 2021.05.22 Vendor: openSUSE
Release: 1.1 Build date: Thu Jun 24 18:32:51 2021
Group: Development/Languages/C and C++ Build host: obs-power8-03
Size: 631128 Source RPM: live555-2021.05.22-1.1.src.rpm
Summary: LIVE555 Streaming Media
This code forms a set of C++ libraries for multimedia streaming,
using open standard protocols (RTP/RTCP, RTSP, SIP). These libraries
can be used to build streaming applications






* Thu Jun 17 2021
  - Update to 2021.05.22:
    lots of fixes and updates, including the security fix for
    CVE-2021-28899 boo#1185874
    See the list in
  - Change the so version of libliveMedia to 94, libgroupsock to 30
* Sun Oct 18 2020 Dirk Mueller <>
  - update to 2020.10.16:
    - Changed "TLSState::read()" to treat any "SSL_read()" result of <=0 as if the
      TLS connection has closed (unless the error was SSL_ERROR_WANT_READ).
      This fixes a problem that could cause 100% CPU usage in RTSP client applications.
      (Thanks to Larry Wu for reporting this.)
    - Updated "TLSState::setup()" to use "TLS_client_method()" instead of the (deprecated)
* Sat Oct 03 2020 Dirk Mueller <>
  - update to 2020.08.19:
    - Fixed a bug in "QuickTimeFileSink" that could cause malformed "esds" atoms to be
      generated.  (Thanks to Chris Paucar for reporting this issue.)
    - In "MPEG2TransportStreamFromESSource.cpp", changed the name of the constant LOW_WATER_MARK
      to TS_FROM_ES_LOW_WATER_MARK, and "#ifndef"d it, so that, if you wish, you can redefine it
      at compile time.
    - Fixed a bug in the handling of pausing, when streaming from (multi-track) Matroska files.
    - Fixed another bug in the handling of seeking within Matroska files.
    - Fixed a bug in the handling of seeking within Matroska files.
      (Thanks to Jim Ham for reporting this problem.)
    - Changed the parameter signature of the "RawVideoRTPSink" constructor and "createNew()"
      functions so that the "width" parameter comes before the "height" parameter.  This order
    - "width", "height" - is more common, and is the order used when these parameters are
      defined in RFC 4175.
      IMPORTANT NOTE: Because the types of these two parameters are the same, existing application
      code that uses "RawVideoRTPSink" will compile without error; however, it will not work
      properly unless the order of the parameters in the call to "RawVideoRTPSink::createNew()"
      is changed.
    - More cleanup of the implementation of "RawVideoRTPSink".
    - Cleaned up the implementation of "RawVideoRTPSink".
    - Updated the "RawVideoRTPSource" implementation to not set "fCurrentPacketCompletesFrame"
      until we are processing the last line in the packet.
      (Thanks to Andrey Lisovoy for reporting this issue.)
    - Fixed a potential buffer overflow bug in the server handling of a RTSP "PLAY" command,
      when the command specifies seeking by absolute time.
      (Thank to Xiaobo Xiang for reporting this.)
    - Fixed a memory leak in the "sha1()" function (a "EVP_MD_CTX" object was not being deleted).
      (Thanks to Amir Perlman for reporting this.)
    - Moved all definitions of PREFIX from "Makefile.tail" files to "Makefile.head" (so that
      it can be redefined by a "config.*" file, if desired.
      Also changed the definition of EXE in "config.mingw" to be ".exe".
      (Thanks to Eric Beuque for this suggestion.)
    - Fixed a typo in the previous release that could cause a compilation problem for some
      developers.  (Thanks to Eric Beuque for reporting this.)
* Thu Jun 04 2020 Dominique Leuenberger <>
  - Update to version 2020.05.15:
    + Added a new filter class "ADTSAudioStreamDiscreteFramer" that
      prepends ADTS headers to incoming AAC audio frames. This makes
      the AAC audio playable (by media players).
    + Updated "openRTSP" to use a "ADTSAudioStreamDiscreteFramer"
      when outputting a AAC audio stream.
    + Updated the "LIVE555 HLS Proxy" to support AAC audio tracks
      (as well as H.264/5 video).
  - Changes from version 2020.05.14:
    + Updated "H264or5VideoStreamDiscreteFramer" to add VPS,SPS,PPS
      NAL units (if known) to the output stream, each time an
      "access_unit_delimiter" NAL unit is added. This makes it more
      likely that the Transport Stream segments produced by the
      "LIVE555 HLS Proxy" will be understandable by a client browser.
    + Added support for H.265 video streams to the "LIVE555 HLS Proxy".
  - Changes from version 2020.05.13:
    + Made the "MPEG2TransportStreamMultiplexor" segmentation mechanism
      (used by "HLSSegmenter") more robust in case the Transport
      Stream PTS is not monotonic non-decreasing.
* Tue Apr 28 2020 Dominique Leuenberger <>
  - Update to version 2020.04.24:
    + Fixed an apparent bug in "RTSPClient" that was causing it to
      not always send an "Authorization:" header when sending a RTSP
      "OPTIONS" command.
* Thu Apr 23 2020
  - Added pkgconfig(openssl) as a requirement for the devel package:
    iveMedia/TLSState.hh #includes openssl/ssl.h.
* Tue Apr 14 2020
  - Update to version 2020.04.12:
    + Updated "config.linux-with-shared-libraries"
      (and "liveMedia/Makefile.tail") to ensure that "libssl" and
      "libcrypto" are linked when "libliveMedia" is built.
      (Thanks to Felix Kaechele for reporting this.)
  - Changes from version 2020.04.06:
    + Removed support for the classes "RTSPServerSupportingHTTPStreaming"
      and "TCPStreamSink".
      These were being used (in the "LIVE555 Media Server") for
      streaming using "HTTP Live Streaming" (HLS).  This was always
      a hack; it is better to use a dedicated HTTP server to serve
      HLS segments, rather than trying to implement a HTTP server
      (serving 'virtual HLS segments) within our own (primarily RTSP)
      If you are looking for HLS support, note that we provide a
      source-code demo application "testH264VideoToHLSSegments" that
      converts a (static) H.264 Elementary Stream file to HLS
      segments, and the "LIVE555 HLS Proxy", which proxies a live
      RTSP/RTP stream to HLS segments.
* Thu Apr 02 2020 Dominique Leuenberger <>
  - Update to version 2020.03.06 (boo#1146283, CVE-2019-15232):
    + Fixed a problem in "H264or5VideoStreamFramer.cpp" that was
      allegedly causing problems compiling for Windows.
  - Changes from version 2020.02.25:
    + Added full support for the "LIVE555 HLS Proxy"
  - Changes from version 2020.02.24:
    + Commented out a line of code that was preventing
      "RTSPClientConnection" objects from being closed when a RTSP
      server handles a "TEARDOWN" command (on a RTP-over-TCP stream).
  - Changes from version 2020.02.23:
    + Updated our (unicast) RTSP server implementation to handle
      "PAUSE" commands by calling "stopGettingFrames()" on the input
    + Fixed a bug in "H264or5VideoStreamFramer" that was causing it
      to not update its presentation times properly following a pause.
    + Updated "openRTSP" to improve the handling of the "-R <port-num>"
  - Changes from version 2020.02.11:
    + Added support for receiving SRTP (encrypted) RTSP streams.
  - For changes between 2019.06.28 and 2020.02.10, please see the
  - Rebase live555-fpic.patch.
* Mon Jul 15 2019 Dave Plater <>
  - Update to version 2019.06.28, fixes boo#1127341 VUL-1:
    CVE-2019-9215: live555: malformed headers lead to invalid memory
    access in the parseAuthorizationHeader function.
* Mon Jun 24 2019 Dominique Leuenberger <>
  - Convert to dynamic libraries (boo#1121995):
    + Use make ilinux-with-shared-libraries: build the dynamic libs
      instead of the static one.
    + Use make install instead of a manual file copy script: this
      also reveals that we missed quite a bit of code to be installed
    + Split out shared library packages according the SLPP.
* Thu May 16 2019 Martin Liška <>
  - Use FAT LTO objects in order to provide proper static library.
* Mon Feb 04 2019
  - update to 2019.02.03:
    * CVE-2019-7314:A use-after-free error causes the RTSP server to
      crash (boo#1124159)
* Tue Jan 15 2019
  - update to 2018.12.14:
    * Add support for sending (and handling) RTCP "BYE" packets that
      contain a 'reason' string (breaking api change)
* Tue Jan 15 2019
  - update to 2018.11.26:
    * CVE-2019-6256: DoS vulnerability in the server implementation
      of RTSP-over-HTTP (boo#1121892)
* Tue Nov 06 2018 Mathias Homann <>
  - Update to 2018.10.17
    * CVE-2018-4013: remote code execution vulnerability (boo#1114779)
    * changes: see
    * removed xlocale.patch - xlocale.h is now conditionally included, only when
      required. Upstream fix.
* Wed Nov 01 2017
  - Update to 2017.10.28
  - Fixed the handling of the LIVE555 Proxy Server's "-u <username> <password>" command-line option if
    the "REGISTER" command is also implemented (i.e., with "-R").  Now, when we handle "REGISTER",
    the <username> and <password> are used to access the REGISTER'ed back-end stream, if necessary.
  - Changed the server handling of the RTSP "REGISTER" command to (if "reuseConnection" is True) delay
    a short period of time (100ms) between replying to the "REGISTER" command, and actually
    handling it.  This is intended to reduce/avoid the possibility of a subsequent "DESCRIBE" command
    ending up in the client ("REGISTER" sender)'s buffer, before the socket gets reused for handling
    incoming RTSP commands.  (Thanks to Durgesh Tanuku for noting this issue.)
  - Made a change to "SIPClient" to better handle Asterisk SIP servers with authentication.
    (Thanks to Rus.)
  - Made some improvements/bug fixes to AVI indexes in "AVIFileSink".
    (Thanks to Victor V. Vinokurov.)
  - Updated the handling of the "writeTimeoutInMilliseconds" parameter in the "makeSocketBlocking()"
    function to work correctly on Windows. (Thanks to Jeff Shanab for noting this issue.)
  - Added support for adding Opus audio to MPEG Transport Streams.  This is done by setting the
    "mpegVersion" parameter to 3 in "MPEG2TransportStreamFromESSource::addNewAudioSource()" or
    (Thanks to Praveen Mathad for suggesting this.)
  - Updated "BitVector" to support a signed version of "get_expGolomb()", and fixed our H.264/265
    parsing code to use the signed version where appropriate.
    (Thanks to Toson Huang and Long Zhang for reporting this.)
* Mon Aug 07 2017
  - xlocale.patch: don't use obsolete <xlocale.h>
* Sat Jul 08 2017
  - Replace silly -exec rm ; by -delete.
* Mon Jul 03 2017
  - Update to version 2017.06.04
    * Fixed a bug in "RTPInterface::removeStreamSocket()" that could
      cause not all 'TCP stream' records for a given socket number
      to be removed if a TCP socket I/O error occurred (during
      RTP/RTCP-over-TCP streaming).  (Thanks to Gerald Hansink et al
      for reporting this.)
    * In "RTSPClient.cpp", moved the call to "clearServerRequestAlternativeByteHandler()"
      from the "RTSPClient" destructor to the "resetTCPSockets()"
      function (which is called more often). This should eliminate
      a 'pointer to a deleted object' error. (Thanks to Gerald Hansink
      et al for reporting this.)
    * Added a new public member function "numClientSessions()"
      to "GenericMediaServer" (and therefore to "RTSPServer", which
      inherits from this).  This allows a server to - at any time -
      check how many clients are currently accessing the server.
    * Updated the diagnostic output in "RTSPClient" to distinguish
      between opening a new TCP socket and connect()ing on a TCP
      socket. (The distinction is important for "REGISTER", which can
      reuse an existing TCP socket.)
    * Fixed a bug in "base64Decode()" that could be triggered
      if (1) your RTSP server is streaming RTP/RTCP-over-HTTP, and (2)
      the remote client sends bad Base64 data (containing an embedded '\0'
      character). (Thanks to Arkady Bernov for reporting this.)
    * Updated "ProxyServerMediaSession.cpp" to change all 'reset()'
      operations so that they are now run as a 'scheduled task' from
      the event loop - avoiding the possibility of bugs caused by
      'reset()' being called while another operation is in progress.
      (Thanks to Erik Montnemery for reporting this issue, and proposing a fix.)
    * Our "RTSPClient" code now ignores "Connection: close" lines in
      the responses to HTTP "GET" requests (that are used to set up
      RTSP-over-HTTP tunneling). Because this tunneling requires that
      the (separate) input and output TCP connections remain intact,
      we assume that the server - if it includes such a line in the
      response to a HTTP "GET" - doesn't really mean it.
      (Thanks to Nguyen Viet Hung for reporting a server that does this.)
    * Fixed a bug in the handling of 'APP' RTCP subpackets. (Thanks to
      Frederik de Ruyck for reporting this.)
    * Fixed a bug in the "StreamReplicator" code. (Thanks to Bruno Abreu
      for reporting this.)
    * Increase the RTSP client's socket receive buffer when we'll be
      receiving RTP/RTCP-over-TCP, and increase the RTSP server's
      client connection socket send buffer when it's used to
      "REGISTER" a stream.
    * Fixed a bug (in the sending/ handling of the "REGISTER"/"DEREGISTER"
      commands) that had been accidentally introduced in version
      2016.09.19. (Thanks to Ralf Globisch for noting this.)
    * Performed the annual update of the copyright years and license
      near the start of each file
    * Changed the "RTCPInstance error" message in "RTCP.cpp" to make
      it clear that the problem is caused by the remote endpoint using
      a buggy version of RTP/RTCP-over-TCP streaming.
    * Updated "QuickTimeFileSink" to make the various creation/modification
      times relative to January 1st 1904 in UTC (as Apple recommends),
      rather than in US Pacific Time.
* Sun Oct 16 2016
  - Update to version 2016.10.11
    * After building the source code, we now display a message
      reminding the developer about our FAQ.
    * Added a new "liveMedia" class
      "MPEG2TransportStreamAccumulator" - a filter that can
      be used to combine several (by default, 7) MPEG Transport
      Stream 188-byte 'packets' into a larger chunk of data,
      more appropriate for streaming via RTP (or raw UDP).
    * Added support for an experimental RTSP "DEREGISTER"
      command, which undoes the effect of a "REGISTER" command.
    * Moved the REGISTER/DEREGISTER-specific functionality of
      "RTSPServer.cpp" into a new file "RTSPServerRegister.cpp",
      to make the base RTSP server code (in "RTSPServer.cpp")
      easier to comprehend.
    * Fixed "GenericMediaServer::createNewClientSessionWithId()"
      to make sure that the new 'client session' object (returned
      by a call to "createNewClientSession()") is not NULL
      before it tries to add it to the 'fClientSessions' table.
      (Thanks to Helmut Grohne for discovering this issue.)
    * Updated "RTSPClient::reset()" to reset each of the 'request
      queues' as well.
      (Thanks to Erik Montnemery for noting a problem (with the
      "LIVE555 Proxy Server") that this caused.)
    * Updated "GenericMediaServer::ClientConnection::closeSockets()"
      so that it doesn't try to call "closeSocket()" (=="close()")
      on socket numbers <0.
    * Fixed a problem whereby a 'delayed task' for a
      "MPEG2TransportStreamMultiplexor" object might have gotten run
      after such an object was deleted.
      (Thanks to Bruno Basilio for providing debugging output to
      help track this down.)
    * Updated "Socket::reset()" (in "groupsock/NetInterface.cpp") so
      that it doesn't try to call "closeSocket()" (=="close()") on
      socket numbers <0.
    * Added a comment to "UsageEnvironment/include/UsageEnvironment.hh"
      to note that "triggerEvent()" should not be called with the
      same 'event trigger id' from different threads.
      (This was already noted in a comment in
      "liveMedia/DeviceSource.cpp", but not in
      "UsageEnvironment/include/UsageEnvironment.hh", which is
      where "triggerEvent()" is defined.)
    * Fixed a problem whereby a "Medium" object's "nextTask()"
      (i.e., "fNextTask") could hold an invalid value after a
      'scheduled task' has occurred (but before the next similar
      task is scheduled) - which causes problems should the
      "Medium" object be deleted during that window of time.
      (Thanks to Helmut Grohne for noting this problem.)
    * Added comments to "UsageEnvironment/include/UsageEnvironment.hh"
      to make it clear that "unscheduleDelayedTask()" (or
      "rescheduleDelayedTask()") must not be called on a
      'scheduled task' after it has already occurred.
      (Thanks to Helmut Grohne for motivating this.)
    * Fixed a bug in the handling of the non-standard
      "" header (used by 'SAT>IP' servers)
      that we had introduced in version 2016.01.12.
      (Thanks to Yaobing Deng for noting this.)
    * Fixed a bug in "RTSPServer" that could cause a crash if a
      "RTSPServer" object is deleted after having been used
      for RTSP-over-HTTP streaming.  (Thanks to Pavel Aronov.)
    * Updated "RTSPClient" to recognize a "Connection: Close"
      header in a server's response. It handles this header by
      closing the RTSP TCP connection (because the server is
      assumed to not be using it again), so that we open a new
      TCP  connection for any subsequent commands.
      (Thanks to Nathan (at MediaPortal) for this suggestion.)
    * Made a small optimization to "RTSPServer"s handling of the
      first "SETUP" command from each client.  (Thanks to Maxim
      Dementiev for the suggestion.)
    * Added a new (public) function "canDeliverNewFrameImmediately()"
      to "MPEG2TransportStreamMultiplexor".
      This function may be used by a downstream reader to test
      whether the next call to "doGetNextFrame()" will deliver
      data immediately.  It can be useful if you want to decide
      whether or not to keep accumulating multiple Transport Stream
      'packets' into an outgoing RTP packet.
      (Thanks to Gilles Chanteperdrix for suggesting this.)
    * Made a minor syntactic change to "MediaTranscodingTable.hh"
      to eliminate compiler warnings.
    * Changed the constant "MAX_INPUT_ES_FRAME_SIZE" to a static
      member variable
      that can, if desired, be increased at run time (before a
      "MPEG2TransportStreamFromESSource" object is created).
      (Thanks to Gilles Chanteperdrix for motivating this.)
    * Changed "~ProxyServerMediaSession()" so that it no longer
      deletes the "MediaTranscodingTable" object that it had
      been passed in its constructor.  (The reason for this is
      that the same "MediaTranscodingTable" can be used by more
      than one "ProxyServerMediaSession".)
    * Made the "parseTransportHeaderForREGISTER()" function
      (that's used in the "RTSPServer" implementation) non-static,
      so that it can be used in other, non-RTSP server
      implementations that want to handle the "REGISTER" command.
    * Made the "RTPSink::SSRC()" function "public:" rather than
      (Thanks to Jean-Luc Bonnet for this suggestion.)
    * Added a new virtual function "noteLiveness()" to the
      "ServerMediaSession" class. This function is called
      (by a "GenericMediaServer") whenever there's 'liveness'
      on a "ClientSession".  The default implementation of this
      function is a 'noop', but subclasses can redefine it - e.g.,
      if you want to remove long-unused "ServerMediaSession"s
      from the server.
    * Fixed a bug in the options handling for the command
      "live555ProxyServer" that could erroneously produce a
      "usage" error if the '-R' option is used, but no
      back-end "rtsp://" URL is given.
    * Backed out the change to "MultiFramedRTPSink" that was
      made in 2016.05.17; the 2016.05.16 version turned out
      to be correct.
    * Rearranged "#include"s to avoid an 'excessive #include
      nesting' error with some old compilers.
    * Made a (mostly inconsequential) fix to the previous bugfix
      for "MultiFramedRTPSink".
    * Fixed a bug in "MultiFramedRTPSink" that affected
      subclasses that redefine "frameSpecificHeaderSize()"
      (for frame-specific headers that precede multiple frames
      in a RTP packet).  (Currently, the only subclass that
      this affected was "VorbisAudioRTPSink".)
      (Thanks to Gilles Chanteperdrix for reporting this bug.)
    * Made a minor update to the "ProxyServerMediaSession" code
      to better support optional media transcoding.
    * Made it easier to set the MTU for all outgoing RTP
      packets, instead of having to call "setPacketSizes()"
      after each "MultiFramedRTPSink" is created.  If you wish,
      you can define the compile-time constants (macros)
      when compiling "MultiFramedRTPSink.cpp".  (These constants
      have default values of 1456 and 1000 respectively, just as
    * Updated "GroupsockHelper.{hh,cpp}" to (supposedly) support
      'MinGW' better
    * Fixed a bug the "ProxyServerMediaSubsession" code that
      could cause an infinite loop if the 'back-end' server was
      slow to respond to "SETUP" requests.
      (Thanks to Erik Montnemery for helping to debug this.)
    * Added support for parsing/streaming Matroska files that
      contain PCM audio tracks.
      (Thanks to Michel Promonet.)
    * Added some more debugging fprintf()s to the
      "ProxyServerMediaSubsession" code to try to track down a
    * Simplified the "genMakefiles" script (moving duplicate
      code into a 'for' loop).
    * Updated the proxy server implementation to better handle
      'front-end' clients that have asked to stream only some
      of the substreams of a multi-stream session.  Now, if a
      substream is closed (because all 'front-end' clients have
      stopped requesting it), but other front-end clients are
      still streaming other substreams, then we will send - to
      the 'back-end' server - only a substream-specific "PAUSE"
      command; not a "PAUSE" command for the entire stream.
      (Thanks to Lakshmi Narayanan for noting this issue.)
    * Added an optional "-p <RTSP-port-number>" option to the
      "LIVE555 Proxy Server", to allow the user to specify a
      RTSP server port number other than the standard port
      numbers: 554 and 8554. (These standard port numbers are
      still tried if the specified port number can't be used.)
      (Thanks to Denis Genestier for this suggestion.)
    * Updated the "ProxyServerMediaSession" to add a Boolean
      virtual function "allowProxyingForSubsession()".
      By default, this always returns True.  However,
      subclasses can redefine this if they wish to restrict
      which subsessions of a stream get proxied - e.g., if
      you want to proxy only video tracks.
    * Improved the "WAVAudioFileSource" code (for parsing
      WAV-format audio files) to make it more tolerant of
      unusual formats.
    * Made it possible to build a version of the "liveMedia"
      library that doesn't contain any RTSP server code;
      e.g., if you are developing only a RTSP client, and
      want to save space. To do this, omit any files that
      contain "Server" or "RTPSink" in their name, and define
      OMIT_REGISTER_HANDLING when compiling "RTSPClient.cpp".
      (Thanks to Jeff Shanab for this suggestion.)
    * Added an option "-E <absolute-seek-end-time>" to
      (Thanks to Hans Maes for suggesting this.)
    * Fixed a bug that was causing "playSIP" to crash.
      (Thanks to Vilaysak Thipavong for reporting this.)
    * Updated "QuickTimeFileSink" to make it usable with non-RTP
      input sources. It still needs to have a "MediaSession"
      that describes the input source; however, this input source
      no longer needs to be RTP; it can, instead, be a UDP or
      other type of source. (Of course, audio/video synchronization
      and hint tracks can't be done in this case.)
    * Changed the name of a variable in the "Makefile.tail" file
      for the "BasicUsageEnvironment" project, in response to
      a complaint that the old name clashed with something in
      some Windows development environment
    * Updated "ProxyServerMediaSession.cpp" to add some
      'internal error' debugging fprintf()s to try to catch a
      possible bug that was reported recently.
    * When a server calls "startStream()" to start a RTSP stream
      for a client, we now no longer make a slight adjustment to
      the RTP timestamp sequence (using the "presetNextTimestamp()"
      call) if there is already another ongoing stream using the
      same "RTPSink".  The effect of this is only minor, but it
      ensures that the addition of an addition 'destination' to
      an ongoing RTSP/RTP stream does not cause any change to the
      contents of the RTP/RTCP packets.
      (Thanks to Erik Montnemery for noting this issue.)
    * This release has no source-code changes from the previous
      release.  However, a test file was mistakenly left in the
      previous version; this produced an excessively-large tar file.
      This has now been removed.
    * Added a hack to "RTSPClient" to handle the non-standard
      "" header - used by 'SAT>IP' servers -
      by using its value in the 'base URL' for subsequent requests.
      (Thanks to Julian Scheel for proposing this.)
    * Updated "QuickTimeFileSink" to add a sanity check to try
      to prevent an occasional problem with H.264 video tracks
      that contain 'sync frames'.
    * Updated the "config.linux-with-shared-libraries"
      configuration file to use the $(CC) and $(CXX)
      macros, to allow for cross-compiling.  (Thanks to Michel
    * Updated the years in the copyright notice on each file.
    * Changed the "ProxyServerMediaSession" code once again.  We
      backed out the changes in the previous two releases,
      and now respond to failures of the back-end "SETUP"
      or "PLAY" commands by doing a full reset - which involves
      deleting the "ProxyServerMediaSubsession" object, and
      doing another "DESCRIBE" to create a new one.  However, we
      can't do this immediately - because the "SETUP" and "PLAY"
      commands can be sent from within
      Instead, we wait until the next 'liveness' command, which
      will get sent immediately when we return to the event loop.
    * Our proxy server code no longer converts the "mode" string
      to lower case before passing it to
      "MPEG4GenericRTPSink::createNew()".  (This turned out to
      be unnecessary, and was breaking some clients that weren't
      treating this string as case-insensitive when they saw it
      in the stream's SDP descriptor.)  (Thanks to Craig Matsuura
      for noting this issue.)
    * Updated the fix in the previous revision to apply to the
      back-end "PLAY" command as well as the back-end "SETUP"
      command, because both of these back-end commands can get
      sent from within
      "ProxyServerMediaSubsession::createNewStreamSource()", so
      we can't allow the "ProxyServerMediaSubsession" object
      to get deleted in either case.
* Fri Oct 16 2015
  - Update to version 2015.10.12:
    * The change that we made to the "ProxyServerMediaSession" code
      in version 2015.07.31 (to reset the proxy server's state if
      a back-end "SETUP" command fails) was too aggressive; it was
      deleting the "ProxyServerMediaSubsession" object.  This was
      a problem, because "SETUP" commands can be called from within
      Instead, we now deal with a failed back-end "SETUP" command
      simply by resetting the 'back-end' connection. (Thanks to
      Hardik Sangani for reporting this issue.)
  - 2015.09.24:
    * Fixed a bug in "RTSPClient" that could cause a crash if the
      TCP connection was lost while resending a RTSP command.
      (Thanks to ChaSeop Im for reporting this.)
    * Moved some more generic 'media server' functionality from
      "RTSPServer" to its parent class "GenericMediaServer".
    * Added a new pure virtual function "getRTPSinkandRTCP()"
      to "ServerMediaSubsession" to allow callers to get ('const')
      access to a stream's "RTPSink" and/or "RTCPInstance" (and
      thus their corresponding "Groupsock" objects) after the
      stream has been created (using "getStreamParameters()".
    * Updated "Groupsock" to allow for the possibility of there
      being more than one 'destRecord' for each sessionId.
      (This is something that doesn't happen in the normal case;
      it's only a special case for WebRTC.)
  - 2015.08.07:
    * If a "RTCPInstance" happens to have both a source and a sink
      (an unusual situation), we now include both "SR" and "RR"
      reports in each outgoing RTCP report packet.
    * When a "RTPSink" is being closed, we no longer turn off
      background reading on its 'groupsock' (because, being a
      "RTPSink", we never turned it on), just in case the
      'groupsock' is also being shared with something else
      (e.g., a "RTPSource") that does background read handling).
  - 2015.08.06:
    * Fixed a bug that would cause the destruction of a
      "RTCPInstance" that was sharing a 'groupsock'
      with a "RTPSource" (i.e., for multiplexed RTP and RTCP) to
      stop the "RTPSource" from continuing to receive incoming RTP
      packets.  This normally wasn't a major problem, because the
      destruction of the "RTCPInstance" was usually followed
      immediately by the destruction of the "RTPSource".
      However, it's also possible for the "RTPSource" to stay alive
      long after the "RTCPInstance" is deleted; in this case things
      will now work correctly.
  - 2015.07.31:
    * Fixed a minor memory leak in the "ProxyServerMediaSession"
      code ("PresentationTimeSessionNormalizer"s and
      "PresentationTimeSubsessionNormalizer"s weren't being deleted
      properly). (Thanks to Dnyanesh Gate for reporting this.)
    * Made the "ProxyServerMediaSession" code a bit more
      bullet-proof, by resetting the 'back-end' connection if a
      "SETUP" command fails. (Thanks to Craig Matsuura for providing
      a real-world example of "SETUP" failing.)
    * Fixed the 'estimated bitrate' values in
      "testMPEG1or2VideoReceiver.cpp" and
      "testMPEG2TransportReceiver.cpp" to match those in the
      corresponding "test*Streamer.cpp" files.
      (Thanks to Alex Anderson for reporting this.)
  - 2015.07.23:
    * Fixed a potential buffer overflow bug in "RTSPServer".
      (Thanks to "an anonymous researcher working with Beyond
      Security's SecuriTeam Secure Disclosure" for discovering this.)
  - 2015.07.19:
    * Fixed a bug in "RTPInterface::sendDataOverTCP()"; it was
      disabling transmission on its socket if the "send()" call
      failed.  We now do this only if the error was not "EAGAIN".
      (Thanks to Erik Oomen for bringing this to our attention.)
    * Changed "QuickTimeFileSink" to try to work around an issue with
      QuickTime sometimes complaining about the frame number in the
      last 'sync frame' being 'out of range'.
    * Changed the parameter signature for
      "ProxyServerMediaSession::createNew()" (and the
      "ProxyServerMediaSession" constructor) to take a
      "GenericMediaServer*" rather than a "RTSPServer" as parameter.
      This makes it possible to create proxy servers that use protocols
      other than RTSP at the 'front-end'.  (The 'back-end' protocol will
      still be RTSP, however.)
    * Defined a new class "MediaTranscodingTable" that can be used to
      generate "FramedFilter" (subclass) objects that perform media
      transcoding.  Added a parameter of this type (with default value
      NULL) to the "ProxyServerMediaSession" constructor and
      "createNew()" function.  This makes it possible to - if you wish
    - add transcoding functionality to a proxy server.  (This feature
      is still experimental, and might be changed in the future.)
    * Added optional "initialPortNum" and "multiplexRTCPWithRTP"
      parameters to the "ProxyServerMediaSession" constructor - to be
      passed to the "ProxyServerMediaSubsession" objects that it creates.
      This allows subclasses to change these parameters if they wish.
    * Updated "ProxyServerMediaSession" to make it possible for
      subclasses to create subclasses of "Groupsock" and/or
      "RTCPInstance", if they wish.
  - 2015.06.25:
    * Changed the definition of the "doEventLoop()" "watchVariable" to
      make it 'volatile'. (Ditto for the "fTriggersAwaitingHandling"
      field in the "BasicTaskScheduler" implementation.) This is to
      alleviate a concern about aggressive optimizing compilers
      possibly generating incorrect code.  (Thanks to Remi
      Denis-Courmont for bringing this issue to our attention.)
  - 2015.06.24:
    * Updated the implementation of "GenericMediaServer" to move the
      code that removes and deletes all "ClientConnection",
      "ClientSession", and "ServerMediaS(ubs)ession" objects from the
      "GenericMediaServer" destructor to a member function "cleanup()".
      This member function MUST be called from the destructor of any
      subclass of "GenericMediaServer".  (Putting this code in the
      destructor of "GenericMediaServer" itself was a bug, because the
      "ClientConnection", "ClientSession", and
      "ServerMediaS(ubs)ession" objects may themselves have been
      subclassed, and there may be a problem deleting them after the
      "GenericMediaServer" subclass destructor has already been called.
      (Thanks to Christopher Benne for noting this.)
    * Fixed the way that "RTSPClient" handles responses to
    "GET_PARAMETER" to properly allow for possible additional
      pipelined responses appearing afterwards.
      (Thanks to Paul Clark for identifying this problem.)
    * Moved the "ClientSession" liveness checking/timeout mechanism
      from "RTSPServer" to its new abstract base class
      "GenericMediaServer".  (The API and functionality of the
      "RTSPServer" class remains unchanged.)
    * Updated the "OnDemandServerMediaSubsession" code to make it
      possible for subclasses to create and use subclasses of
    * Undid the change that we made to "RTSPClient.hh" in the
      previous version.  There is no longer a demonstrated need to
      make "RTSPClient::connectToServer()" virtual.
    * Made a syntactic change to "MatroskaFile.cpp" to eliminate
      some compiler warnings.
  - 2015.06.21:
    * Updated "RTSPClient" to put "port=" rather than "client_port="
      in "Transport:" headers when requesting a multicast stream,
      in accordance with RFC 2326.
      (Thanks to Julian Scheel for noting this.)
    * Updated "MultiFramedRTPSource" so that it doesn't deliver
      0-length frames to the downstream object - in case the
      downstream object interprets this as being an error.
      (Thanks to Julian Scheel for the suggestion.)
    * Made the member function "RTSPClient::connectToServer()"
      virtual, in response to a request from a developer who wanted
      to reimplement this in their "RTSPClient" subclass.
    * Changed the "Groupsock::output()" function to no longer
      take a 'TTL' parameter. (Instead, we now use the TTL (usually
      255) that was provided when the "Groupsock" object was
    * Cleaned up the "GroupEId" class that's used by "Groupsock".
      (Previously, that class had some extra, experimental
      functionality that turned out not to be useful.)
    * Cleaned up the "destRecord" structure that's used in
      "Groupsock" to represent the (possibly multiple) destinations
      for each "Groupsock" object.
    * Updated the "groupsock" library and
      "OnDemandServerMediaSubsession" to better support (in some
      future release) sockets whose destination endpoints are set
      via STUN packet exchanges.
  - 2015.06.11:
    * Fixed a bug in "RTSPClient" that had accidentally been
      introduced in version - 2015.06.04 that prevented "Session:"
      headers from being included in some requests.
  - 2015.06.10:
    * Fixed the return type of the "createNewClientConnection()"
      virtual function, redefined in
    * More changes to satisfy anal-retentive compilers.
    * Removed the "DarwinInjector" code; that functionality has
      not been supported for some time.
  - 2015.06.09a:
    * More changes to supposedly satisfy anal-retentive compilers.
  - 2015.06.09:
    * Added some "friend" declarations to "GenericMediaServer.hh"
      and "RTSPServer.hh" in an attempt to placate an anal-retentive
      Windows compiler.  (Issue reported by Deanna Earley.)
  - 2015.06.07:
    * Restructured the "RTSPServer" class into an abstract base
      class "GenericMediaServer" and a subclass "RTSPServer".
      This makes it possible to develop other kinds of media server
      that use the same "ServerMediaSession"/"ServerMediaSubsession"
      objects to represent the stream(s) that they serve, but using
      protocols other than RTSP.
    * Added a new virtual function "createGroupsock" to
      "OnDemandServerMediaSubsession". This makes it possible for
      subclasses of "OnDemandServerMediaSubsession" to automatically
      use subclasses of "Groupsock" (e.g., those that implement
    * Moved the "ignoreSigPipeOnSocket()" function from
      "RTSPCommon.hh" ("liveMedia" library) to "GroupsockHelper.hh"
      ("groupsock" library), because the function is not specific to
  - 2015.06.04:
    * Added optional support for including the RTSP "Speed:" header
      in "PLAY" requests. (Thanks to Sarma Kolavasi.)
    * Updated the implementation of "setResultErrMsg()" in
      "BasicUsageEnvironment" to work properly in Windows.
      (Thanks to Stas Tsymbalov.)
  - 2015.05.31:
    * Updated the "ProxyServerMediaSession" code to recover better
      if a back-end RTSP "PLAY" command fails (for whatever reason).
      Should this happen, we now reset the connection to the
      'back-end' server.  (This will cause the initial 'front-end'
      client connection (that caused the "PLAY" command to be sent)
      to fail, but subsequent 'front-end' client requests will now
      have a better chance of succeeding.)
  - 2015.05.28:
    * Fixed a bug in error reporting in the "groupsock" library.
      In a couple of places, we were using the result of
      "getResultMsg()" directly in a call to "setResultMsg()", but
      unfortunately those functions are implemented (at least in
      "BasicUsageEnvironment") using the same buffer.
      (Thanks to Stas Tsymbalov for reporting this.)
    * Updated the "MPEGVideoStreamFramer" class (and thereby its
      subclasses, including "H264VideoStreamFramer" to implement
      the "doStopGettingFrames()" virtual function by calling
      "flushInput()".  This should fix a potential problem whereby
      these classes might not work correctly if the downstream
      reader calls "stopPlaying()", and then resumes reading.
      (Thanks to Stas Tsymbalov for bringing this issue to our
  - 2015.05.25:
    * Fixed a bug in "StreamReplicator::removeStreamReplica()":
      It should have been calling "deactivateStreamReplica()"
    * before* possibly deleting the "StreamReplicator" object
      (if this was the last replica, and
      "fDeleteWhenLastReplicaDies" was True).
      (Thanks to Stas Tsymbalov for reporting this.)
    * Fixed some potential problems with "StreamReplica"
      deactivation.  (Thanks to Stas Tsymbalov.)
    * Updated the "RTSPServer" implementation to call
      "ignoreSigPipeOnSocket()" on 'client connection' sockets,
      rather than just on the main server socket.  This is to
      ensure that the server doesn't get killed if a client -
      running on the same host - gets killed.  (Note that, because
      of this fix, it should never be necessarily to set the
      "MSG_NOSIGNAL" flag on any of our calls to "send()".)
  - 2015.05.12:
    * Updated the previous revision to change the order in which
      fields are deleted in the "RTSPServer" destructor, to avoid
      a possible crash if "RTSPServer" objects are deleted.
      (Thanks to ChaSeop Im for noting the problem.)
  - 2015.05.03:
    * Updated the "RTSPServer" implementation to fix a bug in
      RTP/RTCP-over-TCP streaming. Before, if the
      "RTSPClientConnection" object closed before the
      "RTSPClientSession" object, and the TCP connection was also
      being used for RTP/RTCP-over-TCP streaming, then the streaming
      state (in the "RTSPClientSession") would stay alive, even
      though the TCP socket had closed (and the socket number
      possibly reused for a subsequent connection). This could cause
      a problem when the "RTSPClientSession" was later reclaimed
      (due to inactivity). Now, whenever a "RTSPClientConnection"
      object is closed (due to the RTSP TCP connection closing), we
      make sure that we also close any stream that had been using
      the same TCP connection for RTP/RTCP-over-TCP streaming.
      (Thanks to Kirill Zhegulev for noting this issue.)
    * Removed extraneous comments near the top of
  - 2015.04.22:
    * Updated "config.iphone" and "config.iphone-simulator" to work
      with the latest Xcode. (Thanks to Braden Ackerman.)
    * Fixed a rare memory leak in "MultiFramedRTPSource" that might
      occur if it's reading an incoming packet over TCP - requiring
      >1 read for the packet - and the "MultiFramedRTPSource" gets
      closed or paused while this is happening.
      (Thanks to Kirill Zhegulev for noting this.)
  - 2015.04.16:
    * Added the "f" (force symbolic link) flag to the "ln" command
      in the "make install" Makefile rules, in case you're
      reinstalling the same version of a library.
      (Thanks to Luca Ceresoli for noting the need for this.)
  - 2015.04.15:
    * Removed the previous (20 kByte) hard-wired limitation in the
      size of incoming packets for "MultiFramedRTPSource".  (Now,
      any size packet up to the maximum size of 65535 can be
    * Added a (u_int16_t) field "desiredMaxIncomingPacketSize" to
      "RTSPClient".  If set to a value >0, then a "Blocksize:"
      header with this value (minus an allowance for IP, UDP, and
      RTP headers) will be sent with each "SETUP" request.
      (Thanks to Deanna Earley for noting the optional RTSP
      "Blocksize" header.)
  - 2015.04.01:
    * By default, "H264or5VideoStreamDiscreteFramer" sets
      "fPictureEndMarker" (and thus the RTP 'M' bit) if the NAL
      unit is VCL.  Because this isn't always the right thing to do
      (e.g., if we're delivering multiple 'slice' NAL units per
      'access unit' (picture)), we now move this test into a virtual
      If desired, you can implement a subclass that redefines this
      virtual function. (Thanks to Chris Richardson for bringing
      this issue to our attention.)
    * Made a minor syntactic change to
      "ProxyServerMediaSubsession.cpp" to ensure that it compiles
      with some old versions of VC++.
  - 2015.03.19:
    * Updated the "RTSPClient" code for handling a
      "WWW-Authenticate:" header in a "401 Unauthorized" response.
      We now check for the "stale=TRUE" parameter.  If it's set,
      then we resend the command, even if we already handled an
      earlier "WWW-Authenticate:" header. (Thanks to Deanna Earley
      for noting the need to handle "stale=TRUE".)
  - 2015.03.16:
    * Made a small change to the "BasicTaskScheduler"
      implementation to reduce the likelihood of a race condition
      with external thread(s) calling "triggerEvent()".
  - 2015.03.06a:
    * Oops - forgot to add '\0'-termination to the previous fix.
  - 2015.03.06:
    * Updated "RTSPClient" to decode %-encoded characters, should
      they appear in the <username> and/or <password> fields in a
      "rtsp://" URL. (Thanks to Deanna Earley for suggesting this.)
  - 2015.03.01:
    * Updated the "H264or5VideoRTPSink" implementation to make sure
      that any stale fragmented data is flushed (discarded) if a
      server's stream is paused.  This ensures that - after we
      resume from the pause - that we never stream data with old
      presentation times. (Thanks to Gilles Chanteperdrix for
      discovering and reporting this issue.)
  - 2015.02.26:
    * Fixed a bug in "ProxyServerMediaSubsession" that could cause a
      crash if the parent "ProxyServerMediaSession" object is
      removed from the RTSP server and deleted. (Thanks to Sergio ?
      for first reporting this problem.  Thanks to Chiung Ikhwan
      for discovering the source of the bug.)
  - 2015.02.23:
    * Fixed a bug in
      (Thanks to Gilles Chanteperdrix for noting this.)
  - 2015.02.17:
    * Latest version of the "LIVE555 Streaming Media" code
    (reinstalled due to a server crash).
  - 2015.02.13:
    * Oops - removed the "#define DEBUG" that had inadvertently
      been left in "RTCP.cpp" in the previous version.
  - 2015.02.12:
    * Updated the previous release of "RTCP.cpp" to ensure that it
      will compile for Windows.
  - 2015.02.10:
    * Added experimental support for sending RTCP "APP" packets,
      and handling incoming RTCP "APP" packets.  (Thanks to Nick
      Ogden for suggesting this, and providing an example
  - 2015.02.05:
    * Made the "ProxyServerMediaSession" code a bit more
  - 2015.02.04:
    * Fixed a bug in "DigestAuthentication" that could cause the
      proxy server code to crash if it was given a username and
      password for its 'back end' server.
      (Thanks to Sergio Andrade  for reporting this.)
    * Fixed a minor bug in "MatroskaFileParser".
    * Did some syntactic cleanup on a few files to avoid compiler
      warnings with the newest version of "gcc".
* Sat Jan 31 2015
  - fixed paths in live555.pc
  - update to version 2015.01.27:
    * Fixed a bug in "MPEG2TransportStreamFromESSource" that could
      sometimes cause an abort if more than one Elementary Stream
      Source were multiplexed into a single Transport Stream.
      (Thanks to Marc Palau for reporting this issue.)
  - version 2015.01.19:
    * Fixed an obscure bug in "RTSPClient" that might conceivably
      have caused a crash if it received a completely empty RTSP
  - version 2015.01.04:
    * Updated "config.iphone-simulator" to work with the latest Xcode.
      (Thanks to Braden Ackerman.)
    * In the "BasicUsageEnvironment" implementation, renamed
      "EventTime" to "_EventTime" to avoid a reported naming conflict.
  - version 2014.12.17:
    * Updated "RTSPServerSupportingHTTPStreaming" to make sure that
      the data stream source gets closed when it's no longer needed.
  - version 2014.12.16:
    * Changed the FD_SETSIZE check (introduced in version 2014.12.11)
      so that it's not done in Windows (because in Windows,
      FD_SETSIZE has different semantics).
      (Thanks to Deanna Earley for reporting this.)
  - version 2014.12.13:
    * Updated the H.264/H.265 parsing code in "H264or5VideoStreamFramer"
      to be a little smarter about how it computes a file's frame rate
      (when streaming a 'raw' H.264 or H.265 file).
      (Thanks to Michel Promonet for inspiring this.)
    * Updated "config.iphoneos" to work with the latest Xcode.
      (Thanks to Braden Ackerman.)
  - version 2014.12.11:
    * Changed our implementation of "setBackgroundHandling()" and
      "moveBackgroundHandling()" in "BasicTaskScheduler" to check for
      (and disallow) socket numbers >= FD_SETSIZE, because <sys/select.h>
      has a bug (at least, in most systems) that causes buffer overflow
      in this case.  (Thanks to Michel Promonet for pointing this out.)
  - version 2014.12.09:
    * Needed to make the "QuickTimeFileSink" constructor and destructor
      protected: to allow subclassing.
  - version 2014.12.08:
    * Fixed a bug in parsing 'absolute' RTSP "Range:" headers with no end
      time. (Thanks to Ken Chow for reporting this.)
    * Added a new option "-K" to "openRTSP, to tell the client to
      periodically send "OPTIONS" requests as 'keep-alives' for buggy
      servers that don't use incoming RTCP "RR" packets to indicate client
      liveness.  (Thanks to Peter Schlaile for this suggestion.)
    * Added a new 'protected' virtual member function "noteRecordedFrame()"
      to "QuickTimeFileSink". This function is called whenever a frame is
      recorded to the output file. The default implementation of this
      virtual function does nothing, but subclasses can redefine it if
      they wish.
  - version 2014.11.28:
    * When "RTSPClient" parses a RTSP response, we first skip over any
      blank lines that may be at the start of the response.  This can
      happen if the previous response (e.g., to a "DESCRIBE") contained
      extra whitespace. (Thanks to ilwoo Nam for giving an example of
      a server that exhibited this behavior.)
  - version 2014.11.12:
    * We had forgotten to initialize the "RTSPClient" member variable
    "fAllowBasicAuthentication" that we introduced in the previous
  - version 2014.11.07:
    * Added a new "RTSPClient" member function "disallowBasicAuthentication()"
      that you can call if you don't want a RTSP client to perform 'basic'
      authentication (whcih involves sending the username and password over
      the network), even if the server asks for this.
      (Thanks to Tomasz Pala for this suggestion.)
    * Updated the debugging printout code in "RTCP.cpp" to identify all
      known RTCP payload types, even if we don't currently handle them.
      We also - when doing debugging printout - parse and print out
      the contents of SDES RTCP packets.
  - version 2014.11.01:
    * Updated "RTSPClient" so that it reuses "fCurrentAuthenticator"
      if we previously updated it with data from a "WWW-Authenticate:"
      response, even if a non_NULL "authenticator" parameter was
      passed as a parameter to the command.  This reduces the number
      of authetication exchanges that take place if the server asks
      for authentication on more than one command in a RTSP session.
      (Thanks to Tomasz Pala for this suggestion.)
    * Updated "DigestAuthenticator" to allow for the possibility of
      "username" or "password" being NULL.
    * Updated the "RTSPServer" implementation to add an access check
      before the first "SETUP" (the one that doesn't include a
      session id), because it's possible, in principle, for a client
      to send such a "SETUP" without first sending a "DESCRIBE".
      Therefore, we need to perform access checks on both commands.
  - version 2014.10.28:
    * Added support for the VP9 video RTP payload format (sending and
      receiving), including the demultiplexing and streaming of a VP9
      video track from a Matroska-format file.
    * Made "VP8VideoRTPSource" more robust against a bad first-byte
      header field in the payload.
  - version 2014.10.21:
    * Increased the max output packet size for "MultiFramedRTPSink"
      and "RTCPInstance" from 1448 to 1456, because we had a report
      of problems when proxying incoming JPEG/RTP packets of this
      size (and because 1456 bytes still gives a packet size of no
      more than 1500 bytes when we add
      in IP, UDP, and UMTP headers).
  - version 2014.10.20:
    * Increased the RTSP request and response buffer sizes from 10000
      to 20000 bytes, because we saw a RTSP stream (VP8 video) that
      had an extremely large "configuration=" string that was hiting
      the previous limit.
  - version 2014.10.16:
    * Fixed the "RTSPServer" implementation to handle a rare race
      condition that could cause a "ServerMediaSession" object to
      be deleted while it was being used to implement "DESCRIBE".
      (Thanks to Michel Promonet for reporting this.)
  - version 2014.10.07:
    * Fixed a bug in the "MultiFramedRTPSource" implementation where
      we weren't properly checking the size of incoming RTP packets
      that have the "CC" field (i.e., number of "CSRC" fields) non-zero.
    * Updated "Groupsock::output()" to be a virtual function.
      (This makes it possible to implement "Groupsock" subclasses that
      implement 'bump-in-the-stack' protocols (such as SRT(C)P) below
  - version 2014.10.03:
    * Fixed a problem in the "timestampString()" routine that occurs
      if "time_t" is 64 bits, but we're on a 32-bit machine.
      (Thanks to Deanna Earley for reporting this.)
    * Updated the debugging output code in "RTCP.cpp" to make it
      clearer that SDES and APP packets are not invalid; just not
      (yet) handled by us.
* Thu Oct 30 2014
  - BuildRequire pkg-config to get rpm Provides/Requires pkgconfig(live555)
* Mon Oct 06 2014
  - Added support for pkg-config by creating the relevant .pc file
* Thu Oct 02 2014
  - Update to 2014.09.22:
    + Changed the way in which the "RTSPServer" code handles incoming
      "OPTIONS" commands that contain a "Session:" header. If the
      "Session:" header contains a session id that does not exist,
      then we now return a "Session Not Found" error (even though the
      handling of the "OPTIONS" command is not session-specific).
      This new behavior will help proxy servers (that use our
      "RTSPServer" implementation as a 'back-end' server) better
      detect when the back-end server has restarted while streaming.
    + For all other changes since 2013.04.30, please see
* Mon Mar 04 2013
  - Update to version 2013.04.30:
    + One year worth of updates... see changelog.
* Sun Feb 05 2012
  - Update to version 2012.02.04:
    + Updated "WAVAudioFileSource" to read from its input file
      asynchronously, if possible, rather than doing a synchronous
      (blocking) read.
  - Changes from version 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.
  - Drop patches that were required by VideoLAN: fixed upstream:
    + live-getaddrinfo.patch
    + live-inet_ntop.patch
    + live-uselocale.patch
* Tue Jan 31 2012
  - Update to version 2012.01.26.
* Wed Nov 16 2011
  - Rewrite part of the .spec file.., Cleaner installation.
* Wed Nov 16 2011
  - Add VideoLAN required patches for proper funtioning of live555:
    + live-getaddrinfo.patch
    + live-inet_ntop.patch
    + live-uselocale.patch
* Wed Nov 16 2011
  - Update to version 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()".
* Sat Oct 22 2011
  - Update to version 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).
  - Add a -devel subpackage, obsolete the now empty subpackage by it.
  - Drop rpmlintrc file, as the devel files are now in a devel
* Thu Jun 30 2011
  - Update to version 2011.06.16
* Sat Oct 02 2010
  - Update to 2010.09.25
* Tue Sep 01 2009
  - 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.)
  - 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!
  - 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.
* Sun Apr 26 2009
  - 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()"
  - Changed many "char*" variables to "char const*" to eliminate possible compiler warnings.
    (Thanks to Sebastien Escudier for pointing out this issue.)
  - 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 ichael Skaastrup.)
  - Modified "config.mingw" to add "-DLOCALE_NOT_USED" to the "COMPILE_OPTS =" line.  (The VLC folks seem to want
  - 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.



Generated by rpm2html 1.8.1

Fabrice Bellet, Thu Dec 9 11:18:58 2021