[SCM] liblivemedia/master: Update upstream.changelog.

[alessio at users.alioth.debian.org]

The following commit has been merged in the master branch:
commit fc8ab0ac8cc927a9652d933961674a6c0d8084c5
Author: Alessio Treglia <alessio at debian.org>
Date:   Sun May 6 06:09:22 2012 -0700

    Update upstream.changelog.
    
    Git-Dch: Ignore

diff --git a/debian/upstream.changelog b/debian/upstream.changelog
index dfaabc3..9d8f673 100644
--- a/debian/upstream.changelog
+++ b/debian/upstream.changelog
@@ -1,3 +1,109 @@
+2012.04.03:
+- Fixed the "testReplicator" demo application code to actually do what it claims - transmit one replica stream via UDP,
+  while writing the other replica stream to a file.
+  (We had accidentally omitted the line that transmits one replica stream via UDP.)
+- Fixed a bug in the implementation of "StreamReplicator::deleteReplica()".  (Thanks to Bruno Abreu for reporting this.)
+- Fixed a bug in "RTSPServer" that would occur if you tried to add two different "ServerMediaSession" objects using the same
+  stream name.  (Thanks to Vadim Kosarev for noting this.)
+- Added "-DXLOCALE_NOT_USED=1" to "config.cygwin" (on the suggestion of Warren Young).
+
+2012.04.27:
+- Modified the "RTSPClient" implementation to be more careful about not accessing the "RTSPClient" object's state after
+  calling "handleRequestError()", in case the handler function handles the error by deleting the "RTSPClient" object itself.
+  (The "testRTSPClient" demo application does this, for example.)  (Thanks to Gord Umphrey for reporting a problem,
+  and to Guy Bonneau for pointing out this as a possible cause.)
+- Fixed a typo in the "Makefile.tail" file for the "mediaServer" directory.  (Thanks to 'Nix Lo' for the report.)
+- Removed an unneeded "typedef" from "Locale.hh"; it was causing compiler warnings.  (Thanks to Barry Stump and Warren Young
+  for reporting this.)
+
+2012.04.26:
+- Added a debugging error message to various "*Sink" classes, to warn when the "numTruncatedBytes" parameter in the 'after getting'
+  function is >0.  When this happens, you need to increase the "bufferSize" parameter in the appropriate "*::createNew()" call.
+- Changed the buffer size used by the "testH264VideoToTransportStream" demo application from 10000 to 100000, because input H.264
+  NAL units are often larger than 10000 bytes.
+- Changed the default 'buffer size' parameter in "H264VideoFileSink::createNew()" from 10000 to 100000, because input H.264
+  NAL units are often larger than 10000 bytes. 
+- Changed the signature of "AuxHandlerFunc()" (used by the "setAuxilliaryReadHandler()" hack) to pass "packetSize" by reference
+  instead of by value.  (Thanks to Keary Griffin for this suggestion; he was using this mechanism to implement SRTP.)
+- Changed the way that we disable the handling of SIGPIPE signals in "RTSPServer.cpp" (so that the server doesn't get killed
+  when clients, running on the same host, get killed).  (Phillipe Clavel reported that the old method
+  - "signal(SIGPIPE, SIG_IGN);" - wasn't working properly on Mac OS X.)
+
+2012.04.21:
+- Made some cosmetic changes to the "live555ProxyServer" code, prior to its official announcement.
+
+2012.04.18:
+- Added "ProxyServerMediaSession" - a subclass of "ServerMediaSession" that can be used to create a (unicast) RTSP servers
+  that acts as a 'proxy' for another (unicast or multicast) RTSP/RTP stream.
+- Added a new application "live555ProxyServer", and included it in a new subdirectory "proxyServer/" in the 
+  "LIVE555 Streaming Media" distribution.  This application - which uses the new "ProxyServerMediaSession" class - acts as
+  a unicast RTSP server 'proxy' for one or more 'back end' (unicast or multicast) RTSP streams, specified on the command line.
+- Fixed an obscure bug in RTP/RTCP-over-TCP reading.  (Thanks to Shiyong Zhang for reporting this.)
+- Changed the definition of "Boolean" in "UsageEnvironment/include/Boolean.hh" from "unsigned" to "unsigned char",
+  to avoid an apparent conflict when compiling for Mac OS X (and also to save some space).
+  (Thanks to Barry Stump for the suggestion.)
+- Madea a minor change to "RTPInterface" to accommodate RTSP clients that call 'startPlaying()' on a "RTPSource" object prior
+  to the handling of a RTSP "SETUP" response.  (This change affected only RTP-over-TCP streams.)
+- Changed the "RTSPClient" "fVerbosityLevel" field from "private:" to "protected:", to allow subclasses to use it in their own
+  debugging output.
+- Made some changes to "GroupsockHelper.cpp" that were allegedly needed for compilation for 'Android'.
+- Made a minor change to "NetCommon.h" that allegedly improves/fixes compilation for some version of Windoze.
+- Removed the "MP3HTTPSource" class; it was old code that shouldn't be used.
+
+2012.04.04:
+- Made some changes to "H264VideoRTPSink" and "T140TextRTPSink" to correct some minor bugs.
+- Improved the way that "H264VideoStreamDIscreteFramer" detects and reports NAL units that erroneously begin with
+  MPEG 'start codes'.
+- Fixed a bug in the way in which the "ServerMediaSession" reference count is updated by "RTSPServer::RTSPClientSession" objects.
+  (Thanks to Daniel Liu for reporting this.)
+- Modified the "MultiFramedRTPSource" code to optimize the case when the 'packet reordering threshold' has been set to 0.
+  (Doing this is not recommended, however, unless you're only going to be on networks where packet reordering is extremely
+  unlikely.)
+- Added a new member function "reassignInputSource()" to "FramedFilter", to allow a filter's input source to be changed.
+- Changed some comments in "testRTSPClient.cpp" to make it clearer that if you (for whatever reason) choose to reclaim the
+  "UsageEnvironment" and "TaskScheduler" objects, then you can do so only *outside* the event loop (e.g., in "main()", after
+  "doEventLoop()" has returned).
+
+2012.03.22:
+- Fixed a bug in the way that "MPEG1or2DemuxedServerMediaSubsession" creates "AC3AudioStreamFramer" objects.
+  This was causing errors in the way that AC3 audio tracks in VOB files were being streamed by "testOnDemandRTSPServer".
+  (Thanks to "Rustam" for reporting this issue.)
+- Corrected a potentially misleading error message in "RTSPClient".  (Thanks to Sebastien Escudier for reporting this.)
+
+2012.03.20:
+- Fixed a bug in the "RTSPServer" implementation that could prevent it from properly handling 'pipelined' requests
+  (such as "SETUP" requests) from a single client.
+- Updated the "RTSPClient" code to (when "verbosityLevel" > 0) output a warning message when the server 'skips over' one of our
+  earlier requests.  If this happens, it indicates a bug in the server (perhaps a bug in the way that the server handles
+  pipelined requests - such as the bug that we just fixed in our own server here).
+- Fixed a bug in the "RTSPClient" code that prevented it from properly handling 'pipelined' responses from a single server.
+- Fixed a bug in the Matroska file parsing code that could cause an infinite loop when streaming from a multi-track file.
+- Updated the "RTSPServer" code to no longer attempt to seek, or set the scale of, a subsession within a multi-subsession
+  stream.  (Instead, these operations can be done only on the 'aggregate' session.)
+- Added a new member function "addFilter()" to "MediaSubsession".  This allows RTP receivers to add a filter (such as a 'framer')
+  in front of the subsession's "readSource()", changing "readSource()" to be this new filter.
+- Fixed a minor bug in "ServerMediaSession" that was causing it to generate incomplete default 'info' and 'description' strings
+  in SDP descriptions.
+- Fixed a minor bug in "T140TextRTPSink": Make sure that the 'idle timer' (for delivering empty frames downstream) gets turned
+  off when the input source closes.
+- Fixed the testing of the "mpeg4Mode" parameter to the "MPEG4GenericRTPSink" constructor, to make it case-insensitive.
+- Added new versions of "H264VideoRTPSink::createNew()" that (optionally) take SPS and PPS NAL units
+  (either in raw binary form, or as a 'sprop-parameter-string') as parameters.  This is useful if you know this information in
+  advance, rather than having to get it from the input 'framer' object.
+- Added a new version of "H264VideoStreamFramer::setSPSandPPS()" that takes a 'sprop-parameter-string'
+  (instead of the raw binary NAL units) as parameter.
+- Added a new version of "MPEG4ESVideoRTPSink::createNew()" that takes stream configuration information as a parameter.  This is
+  useful if you know this information in advance, and don't want to rely upon the sink getting this from the input 'framer' object.
+- Added a member function "MPEG4VideoStreamFramer::setConfigInfo()" that can be used to (optionally) set stream configuration
+  information, without requiring the framer to read it from the input source.
+- Added a new version of "VorbisAudioRTPSink::createNew()" that takes a Base-64-encoded 'configuration' string - rather than
+  raw configuration headers - as parameter.
+- Added a minor hack to "MediaSubsession" to allow ADU-ized MP3 frames to optionally be received 'as is',
+  instead of always converting them back to MP3 frames for delivery.
+- Added a new member function "ServerMediaSession::numSubsessions()" to return the number of "ServerMediaSubsessions" that have
+  been added to the "ServerMediaSession".
+- Changed the signature of "base64Decode()" to make its string argument "char const*" instead of "char*" (for extra type safety).
+
 2012.02.29:
 - We no longer define RTSPCLIENT_SYNCHRONOUS_INTERFACE by default.  Consequently, the old, now-deprecated 'synchronous' "RTSPClient"
   interface will no longer be available, by default.  If you still want this, however, you can get it by "#define"ing

-- 
liblivemedia packaging