[SCM] liblivemedia/master: Update upstream ChangeLog.

[alessio at users.alioth.debian.org]

The following commit has been merged in the master branch:
commit bdd7af0c5af9c2e3d69fd4957e203b8aed704a53
Author: Alessio Treglia <alessio at debian.org>
Date:   Sat Apr 16 11:00:16 2011 +0200

    Update upstream ChangeLog.

diff --git a/debian/upstream.changelog b/debian/upstream.changelog
index 37a007e..8aa1b52 100644
--- a/debian/upstream.changelog
+++ b/debian/upstream.changelog
@@ -1,3 +1,116 @@
+2011.03.14:
+- Updated the "MPEG2TransportFileServerMediaSubsession" to use the "streamDuration" parameter (if >0.0) to limit
+  the number of Transport Packets that are streamed from the source file.  (This happens only if the file is indexed.)
+  This allows the server to implement finite RTSP ranges for such streams.
+- Fixed a minor bug in "OnDemandServerMediaSubsession" if "reuseFirstSource" is set.  (Thanks to Andreas Gaer for noting this.)
+
+2011.03.06:
+- Changed the implementation of "RTSPClient::teardownMediaSession()" and "RTSPClient::teardownMediaSubsession()"
+  (i.e., in the old, now-deprecated synchronous "RTSPClient" interface) to not wait for, or handle, a response to the
+  RTSP "TEARDOWN" command.  This avoids the client blocking indefinitely if the server (or its connection) happens to die before
+  the RTSP response gets sent.
+  (Note, however: All RTSP client applications should be updated to use the new, asynchronous interface.  The old synchronous
+  interface will not be supported indefinitely.)
+
+2011.03.05:
+- Changed the signature of the "OnDemandServerMediaSubsession::seekStreamSource()"
+  virtual function to add a "streamDuration" field.  This field (if >0.0) tells
+  the implementation how much data to stream before ending with EOF.
+  Our RTSPServer implementation now uses this to implement finite RTSP ranges
+  (as specified in the "Range:" header).  As with seeking, only some codecs currently
+  implement this: MP3 audio, WAV audio, and DV video.
+- We now support streaming from IMA ADPCM ("DVI4") WAV files.
+- When streaming MP3 audio files or indexed MPEG Transport Stream Files, we now compute a more accurate bitrate estimate
+  (for use in computing RTCP packet frequencies).
+
+2011.01.24:
+- Fixed a bug that was accidentally introduced in version 2011.01.10, and which crashes VLC (when VLC is used to play a
+  "rtsp://" URL).  To avoid the bug, VLC should therefore use this version or later.
+
+2011.01.21:
+- Fixed a bug in "H264VideoStreamFramer" that was introduced in the last release.
+  (This affected the "testH264VideoToTransportStream" demo application.)  (Thanks to Dunling Li for reporting this.)
+- Fixed a minor syntax bug in "H264VideoStreamFramer.cpp" which (fortunately) had not been an actual problem.
+  (Thanks to Guillaume Le Neindre for the report.)
+
+2011.01.20:
+- Added a new demo application "testH264VideoToTransportStream", which takes a H.264 Video input file (named "in.264"),
+  and converts it to a Transport Stream file (named "out.ts").
+  (Note that for this conversion to work properly, the input H.264 file must contain sufficient timing information for us to be
+  able to deduce the frame rate.)
+  (Thanks to Dunling Li for this suggestion.)
+- Changed our RTSP server implementation so that the "SET_PARAMETER" command succeeds (but does nothing) by default.
+  (Subclasses can redefine its behavior, if necessary.)  We also now support the special "*" URL, which designates an
+  operation on the entire server.  (The only commands for which this is allowed are "OPTIONS", "GET_PARAMETER" and "SET_PARAMETER".)
+  (Thanks to Jeremy Noring for noting that some clients need this.)
+
+2011.01.19:
+- Fixed a bug in "OnDemandServerMediaSubsession" that was causing unicast RTSP/RTP
+  servers to sometimes omit sending the very first packet of a stream.  (This seemed
+  to occur only for servers running on Windows.)
+
+2011.01.10:
+- Updated "RTSPClient" so that the new asynchronous interface can handle RTSP URLs that contain a "<username>:<password>@"
+  before the server host name/address.  This had been supported in the old synchronous interface, but when we implemented the
+  new asynchronous interface, we had forgotten to implement it there as well.
+- Made a change to the implementation (but not the default behavior) of "RTSPServer" to allow for the possibility of a subclass
+  implementing HTTP streaming (using the same HTTP port that we use to support RTSP-over-HTTP tunneling).
+
+2011.01.06:
+- We added support for receiving the RTP payload formats "audio/L20", "audio/L24" and "audio/DAT12", as defined in RFC 3190.
+  We also recognize the new "fmtp:"-line parameters "emphasis" and "channel-order" that this RFC also defines.
+  (Thanks to Maciej Szlapka for noting that these payload formats had not previously been supported.)
+
+2011.01.05:
+- Made a small change to the sanity check introduced to "MultiFramedRTPSink" and "BasicUDPSink" in version 2010.12.05 to
+  eliminate the possibility of small extra delays accumulating over time.  (Thanks to Warren Young for raising this issue.)
+- Made a small change to the 'magic values' that are used to implement the "esds" atom for MPEG-4 video in "QuickTimeFileSink".
+  (I have no idea what these values are supposed to do, but Stu Tomlinson reports that the new values work better for him.)
+
+2010.12.31:
+- We now support 'trick play' operations (seeking, fast-forward, reverse play) on MPEG Transport Stream files that contain
+  H.264 video (rather than just MPEG-2 video, as previously).  To support this, the index file format has been extended in a
+  backwards-compatible way, so that existing index files (for MPEG-2 video Transport Streams) will continue to work as before.
+  New versions of the "MPEG2TransportStreamIndexer" and "testMPEG2TransportStreamTrickPlay" utilities - and the
+  "live555MediaServer" - have also been released.
+- Fixed a bug in the definition of the "profile_level_id" field in "H264VideoRTPSink".  (Thanks to Geoff Cleary for noting this.)
+- Change the parsing of RTSP "Range:" headers to allow parameters of the form "clock=" or"smtpe=".  (However, we currently don't
+  interpret parameters of this form; instead, we just ignore them.)
+  (Thanks to Sebastien Escudier for this suggestion.)
+- Fixed "openRTSP" to properly reset internal state before repeating the playing of a stream (if the '-c' (play continuously))
+  option is used.  (Thanks to Anon Sricharoenchai for noting this.)
+
+2010.12.14:
+- Oops - there was a serious bug in the 'event trigger' implementation in the previous release.
+  IMPORTANT: You should upgrade to this new version if you plan to use the 'event trigger' mechanism!
+  (Thanks to "P.J." for noticing the bug in the previous version.)
+
+2010.12.11:
+- Added a new 'event trigger' mechanism to "TaskScheduler".  This makes it possible
+  to define new events that can be handled within the event loop.  Unlike other
+  library functions, events can be 'triggered' (i.e., fired) from a separate thread.
+  This makes it easier to implement input devices than the old 'watchVariable'
+  mechanism (which remains).
+  Also, the "DeviceSource" class - which is a model for how to implement an input
+  device class - has been significantly improved.  It now uses the new 'event trigger'
+  mechanism.  Also, more of it has been implemented, making it clearer where new code
+  needs to be written. 
+
+2010.12.05:
+- Significantly improved our support for streaming H.264 video.  In particular, "H264VideoStreamFramer"
+  and "H264VideoStreamDiscreteFramer" (a new class) act like their corresponding MPEG4 versions: "H264VideoStreamFramer" reads
+  a H.264 Video Elementary Stream byte stream (e.g., from a file), and "H264VideoStreamDiscreteFramer" reads discrete H.264 video
+  NAL units (i.e., one-at-a-time), e.g., from a H.264 video encoder.  (Note that developers no longer need to subclass
+  "H264VideoStreamFramer".)  We also added a new demo application - "testH264VideoStreamer" - for streaming from
+  a H.264 Elementary Stream Video file via multicast.  "testOnDemandRTSPServer" and "live555MediaServer" were also updated to
+  stream H.264 Video Elementary Stream files.
+- Added a sanity check to "MultiFramedRTPSink" and "BasicUDPSink" to allow for the possibility of the system clock jumping ahead
+  in time, and thereby messing up the calculation of how long to wait before sending the next packet.
+  (Thanks to Anders Chen for noting this issue.)
+- Fixed bugs in "AMRAudioRTPSource" and "QCELPAudioRTPSource" that might sometimes cause an event handler to try to reference
+  objects that had already been deleted.  (Thanks to David Cailliere for detecting the problem with "AMRAudioRTPSource";
+  it turns out that "QCELPAudioRTPSource" had the same problem.)
+
 2010.11.17:
 - Added new a member function "setAuthenticationDatabase()" to "RTSPServer".  This allows a server's manager to change
   (or disable) authentication at runtime.  (Thanks to Jeremy Norling for suggesting this functionality.)
@@ -104,7 +217,7 @@
 
 2010.08.31:
 - Fixed some problems in the way that we implement RTP/RTCP-over-TCP streams that were showing up when we have more than one
-  suck stream sharing the same input source ("reuseFirstSource" == True).  (Thanks to John Tam for reporting this.)
+  such stream sharing the same input source ("reuseFirstSource" == True).  (Thanks to John Tam for reporting this.)
 
 2010.08.22:
 - Updated the "DarwinInjector" class to use the new, asynchronous "RTSPClient" interface.

-- 
liblivemedia packaging