[SCM] liblivemedia/master: Update upstream changelog.

Sun Nov 4 11:20:53 UTC 2012

The following commit has been merged in the master branch:
commit 605b386e1ae1c3fbeb7dc57fbdcce77c96f1e024
Author: Alessio Treglia <alessio at debian.org>
Date:   Sun Nov 4 11:20:47 2012 +0000

    Update upstream changelog.

diff --git a/debian/upstream.changelog b/debian/upstream.changelog
index 6d4561c..f1af0b6 100644
--- a/debian/upstream.changelog
+++ b/debian/upstream.changelog
@@ -1,3 +1,139 @@
+2012.10.24:
+- Made the "addNewInputSource()" function of "MPEG2TransportStreamFromESSource" "protected:", to allow subclasses of
+  "MPEG2TransportStreamFromESSource" to support adding new kinds of input stream, if desired.
+
+2012.10.22:
+- Updated "AVIFileSink" to add an index at the end.  (Thanks to "qiuchangyong qiuchangyong".)
+- Updated the "MediaSession" implementation to accept "audio/OPUS" as a valid RTP payload format (which can be implemented
+  using a "SimpleRTPSource").
+
+2012.10.21:
+- Fixed a bug in "PresentationTimeSessionNormalizer".  (Thanks to Bruno Marchand.)
+
+2012.10.19:
+- Fixed a bug in the way that our RTSP proxy implementation reinitializes itself after reestablishing connectivity with the
+  back-end server.  It now deletes all "ProxyServerMediaSubsession"s, so that they will get reestablished when the response
+  to the new "DESCRIBE" command comes back.  (This also makes it possible for the back-end stream to restart with different
+  parameters.)  (Thanks to Aashish Kaushik for reporting this problem.)
+- Moved the definition of the "PresentationTimeSessionNormalizer" and "PresentationTimeSubessionNormalizer" classes to
+  "include/ProxyServerMediaSession.hh", so that developers can use them if they wish.  (Note, though, that those classes were
+  intended for use only within our "ProxyServerMediaSession" (etc.) implementation, so don't complain if they don't do what you
+  want.)
+- Added a "#ifndef RTSP_BUFFER_SIZE"/"#endif" around the definition of RTSP_BUFFER_SIZE in "liveMedia/include/RTSPServer.hh",
+  to allow this definition to be changed at compile time if desired.  (Suggestion from Matt Schuckmann.)
+
+2012.10.18:
+- Updated the "RTPInterface" RTP/RTCP-over-TCP handling code to properly handle the case where we know about the
+  'stream channel id' for an embedded RTP or RTCP packet, but haven't yet registered a read handler function for it.
+  (This can occur in rare situations if our server starts sending RTP or RTCP packets before sending the RTSP "PLAY" response,
+  and the client (e.g., VLC) doesn't start reading from its input source until after it receives the "PLAY" response.)
+  (Thanks to Ralf Globisch for noticing this issue, and proposing a solution.)
+
+2012.10.17:
+- Maved "RTSPClient::reset()" from "private:" to "protected:", and made "ProxyRTSPClient::reset()" call "RTSPClient::reset()"
+  (at the end).  This causes the "ProxyRTSPClient" object to properly get reset (and, in particular, the 'session id' to get
+  reset) if we have to restart a connection with a back-end server.
+  (Thanks to Aashish Kaushik for reporting this issue.)
+- Made the implementation of ending RTP/RTCP-over-TCP a little more efficient (by implementing each packet send using 2 calls
+  to "send()", rather than 4).
+
+2012.10.16:
+- Updated our implementation of sending RTP/RTCP-over-TCP so that if the initial '$' send() succeeds, we force the send()s of
+  the remaining data (the 'stream channel id', packet size, and packet data) to succeed, even if it means that we have to 
+  temporarily block by doing so.  This makes servers whose RTP/RTCP-over-TCP streams exceed the capacity of the network
+  handle this a bit more gracefully, avoiding the possibility that 'incomplete' packet data will appear on receivers'
+  TCP connections.
+
+2012.10.12:
+- We backed out the change that we made (back in version 2012.06.12) to ignore SDP-specified port numbers for unicast streams,
+  because this broke our SIP client (which has no way of telling the server that it wants to use a different port number).
+  Now, if you want to ignore SDP-specified port numbers, you'll need to compile "MediaSession.cpp" with the definition
+  -DIGNORE_UNICAST_SDP_PORTS=1
+
+2010.10.11:
+- Fixed a bug that would cause "playSIP" to crash if run without a username/password parameter.  (Thanks to Sam Machin.)
+- Added "#include <errno.h>" to "groupsock/include/NetCommon.h", to reduce the likelihood of compiler warning messages
+  when compiling for Windows.  (Thanks to Bruno Marchand for the suggestion.)
+- Updated "StreamReplicator" to call "stopPlaying()" on the upstream source if/when all replicas have closed.
+  (Thanks to Bruno Marchand for the suggestion.)
+
+2012.10.04:
+- Updated the fix that we made in version 2012.09.11: When a "RTSPClient" requests RTP/RTCP-over-TCP streaming, it is
+  prepared to handle incoming RTP/RTCP-over-TCP data as soon as it receives the "SETUP" response, but doesn't start sending
+  RTCP "RR" packets until after it receives the "PLAY" response.
+
+2012.10.01:
+- Made the code for receiving RTP/RTSP-over-TCP streams a little more robust.  (Thanks to Rex Wolf.)
+- Made a minor fix to the way that "SIPClient" sends a "BYE".  (Thanks to Frederic Nadeau.)
+- Made the "fDestinationsHashTable" field in "OnDemandServerMediaSubsession" "protected:" rather than "private:", to allow
+  subclasses to access it, if desired.  (They should only do lookups in the table, though.)
+  (Thanks to Matt Schuckmann for this suggestion.)
+
+2012.09.27:
+- Fixed the RTSP proxying implementation so that the presentation times of relayed frames are properly aligned with
+  'wall clock' time, so that receivers will get these presentation times accurately (using RTCP).  We also defer RTCP "SR"
+  reports on the outgoing ('front-end') streams until the incomong streams' presentation times have been RTCP-synchronized
+  (because until then, the incoming presentation times are not accurate).
+  (Thanks to Bruno Marchand for bringing this issue to our attention.)
+- Updated the "ProxyRTSPClient" implementation so that if the connection to the back-end server fails, it tries to restore the
+  connection by sending a new "DESCRIBE" command. This is more reliable (especially for back-end servers that use authentication.)
+  (Thanks to Lei Wu for this suggestion.)
+- Added optional Boolean parameters to the "createNew()" functions for the "MPEG4VideoStreamDiscreteFramer",
+  "MPEG1or2VideoStreamDiscreteFramer", and "DVVideoStreamFramer" filters, to specify that the input frames' presentation
+  times should be passed through unmodified.  (The default behavior for these filters is to modify at least some presentation
+  times.)  This parameter is necessary for our RTSP proxying implementation (because it wants to relay frames' presentation times
+  unmodified.)
+- Updated "RTSPClient.hh" to move the "fCSeq" and "fCurrentAuthenticator" from "private:" to "protected:".
+  This makes it possible for subclasses to implement variants of RTSP commands.
+  (Thanks to Matt Schuckmann for this suggestion.)
+- Updated the "MediaSession" implementation to accept "audio/ILBC" as a valid RTP payload format (which can be implemented
+  using a "SimpleRTPSource").
+- Added a "usesTCPTransport()" Boolean function to "RTSPServer::RTSPClientSession" so that subclasses can easily know whether
+  (at least one subsession of) the session is being streamed via TCP.
+- Removed a small piece of unnecessary code from the "testRTSPClient" code.  (Thanks to Anton Zvyagintsev for noticing this.)
+
+2012.09.13:
+- Minor bugfix in "RTSPClient" when imnplementing RTSP-over-HTTP: Don't enqueue "POST" request records (after sending the command),
+  because we don't expect a response to this request.  (Thanks to Daniel Peng.)
+- Print an error message (in "RTCP.cpp" and "MultiFramedRTPSource.cpp") when we hit a buffer limit when reading
+  a (RTCP or RTP) packet over TCP.
+- Increased MAX_PACKET_SIZE in "MultiFramedRTPSource.cpp" (for reading incoming packets) to 20000, because of a report
+  of a server sending packets larger than 10000 bytes.
+- Added support for receiving the "application/VND.ONVIF.METADATA" RTP payload format.  This is just a XML document packed into
+  simple RTP packets (but with the "M" bit used to indicate the end of the document).
+  (Thanks to Michel Promonet for this suggestion.)
+- Updated the meaning of the "doNormalMBitRule" parameter to "SimpleRTPSource::createNew()" and "SimpleRTPSink::createNew()".
+  It now applies for all media types other than "audio".  (Previously, it applied only to "video".)
+  This change makes it possible to use "doNormalMBitRule" = True for the "application/VND.ONVIF.METADATA" RTP payload format.
+
+2012.09.12:
+- Fixed a bug in the URL parsing code in "RTSPCommon.cpp" that would cause parsing to fail for RTSP urls that don't have any
+  slashes after the "host" or "host:port" part.  (Such URLs would usually be used only for operations - such as "OPTIONS" - on the
+  entire server.)  (Thanks to Tamas Vincze for reporting this.)
+
+2012.09.11:
+- Fixed a problem whereby a "RTSPClient" streaming RTP/RTCP-over-TCP would sometimes send its first RTCP "RR" packet before
+  the server had had a chance to handle the "PLAY" command.  This caused a problem for our server, because it would receive
+  the RTCP "RR" packet over the RTSP command connection, before it knew how to deal with this.
+  (Thanks to Ralf Globisch for reporting this issue.)
+- Fixed a bug in "MPEG2TransportStreamFromESSource": Its destructor wasn't stopping the delivery from upstream objects.
+  (Thanks to Jing Li for reporting this.)
+- Updated "AVIFileSink" to add a 'start code' before each H.264 NAL unit that it writes to the file.
+  (Thanks to 'giuchangyong' for the suggestion.)
+- Added "-DTIME_BASE=int" to the COMPILE_OPTS for each of the "config.macosx*" configuration files (because someone reported
+  that it seems to be necessary for them).
+
+2012.09.07:
+- The socket error handling improvement that we made back in version 2012.07.24 inadvertently broke a workaround that we had
+  made (much earlier) for a bug in Windows.  This caused problems for some Windows clients that receive via RTP-over-TCP.
+  This version should fix those problems.
+- Modified the implementation of "MatroskaFileParser::parseEBMLVal_float()" to remove some aliasing that might cause
+  some compilers to generate incorrect code when optimizing.  (Thanks to 'Owen' for noting this issue.)
+
+2012.09.06:
+- Fixed a bug in "RTSPServer" that would cause incorrect RTSP responses to be sent back in response to "PAUSE"
+  (and some "GET_PARAMETER" and "SET_PARAMETER") commands.  (Thanks to Matthias Meding for reporting this.)
+
 2012.08.31:
 - Fixed a bug in "WAVAudioFileServerMediaSubsession" that could cause problems when seeking within very large WAV files.
   (Thanks to 'Raph' for noting this problem.)
@@ -96,7 +232,7 @@
 
 2012.07.14:
 - Updated the RTSP server code to properly handle "Range:" headers of the form "npt=-<endtime>".
-  (Thanks to Michael Boom for brining this to our attention.)
+  (Thanks to Michael Boom for bringing this to our attention.)
 
 2012.07.06:
 - Modified the implementation of "ServerMediaSession::generateSDPDescription()" to ignore (skip over) subsessions that use

-- 
liblivemedia packaging

