[SCM] liblivemedia/master: Update upstream changelog.

[alessio at users.alioth.debian.org]

The following commit has been merged in the master branch:
commit 6cf66fd16aede2d131c85db4b869479eaabfea1a
Author: Alessio Treglia <alessio at debian.org>
Date:   Thu Sep 6 17:11:59 2012 +0200

    Update upstream changelog.

diff --git a/debian/upstream.changelog b/debian/upstream.changelog
index d68b107..6d4561c 100644
--- a/debian/upstream.changelog
+++ b/debian/upstream.changelog
@@ -1,8 +1,145 @@
-Next version, in progress:
+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.)
+- Fixed a bug in "VorbisAudioRTPSink".  (Thanks to Owen Wallace for reporting this.)
+
+2012.08.30:
+- Fixed an obscure bug in the "MatroskaFile" constructor if the specified input file does not exist.
+  Also, fixed an obscure bug in "RTSPServer" if it receives a malformed request.
+  (Thanks to Kevin Kuo for reporting this.)
+
+2012.08.29:
+- The private->protected change to "fPreferredFrameSize" in the previous release was done in the wrong header file.  It should
+  have been done in "WAVAudioFileSource.hh", not "ByteStreamFileSource.hh".
+
+2012.08.28:
+- Added a new 'filter' class "EndianSwap24", for byte-swapping 24-bit values between little and big-endian.
+- Updated "WAVAudioFileServerMediaSubsession" to support the streaming of 24-bit (or 20-bit) WAV audio files.
+  (Thanks to 'Raph' for this suggestion, and for providing an example file.)
+- Brought the "testWAVAudioStreamer" demo application up-to-date, by increasing the variety of WAV file formats that it can support.
+- Added a "-U <absolute-start-time-string>" option to "openRTSP", to support seeking by 'absolute' time (if, of course, the
+  server supports this).  (Thanks to Michel Promonet for this suggestion.)
+- Updated the RTSP server implementation so that, by default, if a client tries to "PLAY" a stream using an 'absolute' time,
+  then the server will refuse by sending back a regular (NPT) "Range:" header instead.  (If you want the server to handle
+  seeking by 'absolute' time, then you need to reimplement "ServerMediaSession::seekStream()" or
+  "OnDemandServerMediaSubsession::seekStreamSource()".)
+- Made the "fPreferredFrameSize" variable in "ByteStreamFileSource.hh" "protected:" rather than "private:", to allow subclasses
+  of "WAVAudioFileSource" to change its value, if desired.
+
+2012.08.20:
+- Added optional RTSP server and RTSP client support for streams that are indexed by 'absolute' time - i.e., using strings
+  of the form "YYYYMMDDTHHMMSSZ" or "YYYYMMDDTHHMMSS.<frac>Z".
+  - For RTSP server developers (i.e., developers who have their own subclasses of "OnDemandServerMediaSubsession"):
+    - To automatically have your streams advertised (in their SDP description) as supporting absolute time indexing,
+      reimplement (in your subclass) the virtual function:
+        "virtual void getAbsoluteTimeRange(char*& absStartTime, char*& absEndTime) const;"
+	  (see "liveMedia/include/ServerMediaSession.hh").
+      This function should set "absStartTime" to a string value (of the form noted above), and should set "absEndTime" to
+      a corresponding string value, if the stream has an end time, otherwise NULL.
+    - To implement seeking by absolute time, reimplement (in your subclass) the virtual function:
+        "virtual void seekStreamSource(FramedSource* inputSource, char*& absStart, char*& absEnd);"
+	  (see "liveMedia/include/OnDemandServerMediaSubsession.hh").
+      "absStart" (and "absEnd", if non-NULL) are strings (of the form noted above).
+      (The function *may* change them, to make them more accurate.)      
+  - For RTSP client developers:
+    - To check whether a stream supports indexing by absolute time,
+      call "MediaSession::absStartTime()" or "MediaSubsession::absStartTime()", and check whether this (string) value is non-NULL.
+    - To play a stream, indexed by absolute time, call one of the new, alternative forms of "RTSPClient::sendPlayCommand()"
+      that take "absStartTime" (and optional "absEndTime") strings as parameters.
+      (see "liveMedia/include/RTSPClient.hh")
+- Updated the "BasicTaskScheduler" class to make the 'maxSchedulerGranularity' time (in microseconds) an optional parameter to
+  "BasicTaskScheduler::createNew()".  This parameter (default value: 10000 (i.e., 10 ms)) specifies the maximum time that we
+  wait (in "select()") before returning to the event loop to handle non-socket or non-timer-based events,
+  such as 'triggered events'.  You can change this is you wish (but only if you know what you're doing!),
+  or set it to 0, to specify no such maximum time.
+  (You should set it to 0 only if you know that you will not be using 'event triggers'.)
+
+2012.08.17:
+- Fixed the RTSP server support for MPEG Transport Stream 'trick play' to stream the correct amount of data when a stream
+  'duration' is specified for fast forward or reverse play.  (Thanks to Michael Boom for reporting this bug.)
+- Made some changes to "liveMedia/include/RTSPServer.hh" to make old compilers happy.
+
+2012.08.12:
+- Modified the change to the definitions of EINTR, EAGAIN, EINPROGRESS, EDOULDBLOCK in "groupsock/include/NetCommon.h"
+  (made in the previous release) to ensure that they are always mapped to the "WSA*" equivalents.
+
+2012.08.08
+- A major update to the RTSP server implementation.  A single RTSP client session (i.e, the streaming of one particular stream
+  to one particular client) can now use an arbitrary number (>=1) of TCP connections.
+  For more details, see:
+      http://lists.live555.com/pipermail/live-devel/2012-July/015571.html
+- Put #ifndef/#endif around the (Windows-only) definitions of EINTR, EAGAIN, EINPROGRESS, EDOULDBLOCK in
+  "groupsock/include/NetCommon.h", in case they are already defined by the system's header files.
+  (Thanks to Hyunho Kim for bringing this to our attention.)
+- Made a small modification to "H264VideoStreamFramer.cpp" to eliminate some 'unused variable' compiler warnings
+  (when compiling without DEBUG defined).
+
+2012.07.26:
+- Made the "RTPInterface" code for handling RTP/RTCP-over-TCP a little more robust.
+- Updated the "WAVAudioFileSource" WAV-file parser to skip any extraneous 'chunks' between RIFF and FORMAT.
+  (Thanks to Sid Price for suggesting this.)
+
+2012.07.24:
+- Updated the "RTPInterface" code to better handle socket errors when reading from TCP sockets.  (Thanks to Barry Stump for
+  bringing this issue to our attention.)
+- Made sure that some code that calls "fileno()" for doing asynchronous file reading never gets compiled under Windows
+  (where all file reading is synchronous), because some versions of Windows redefine the result type of "fileno()".
+  (Thanks to Sid Price for bringing this to our attention.)
+- A member variable of "ProxyServerMediaSession" was not being initialized correctly (causing a "valgrind" error).
+  (Thanks to Jesus Leganes for reporting this.)
+- Removed some unnecessary "abort()" calls from the code.
+
+2012.07.18:
+- Fixed a minor bug in the "H264VideoStreamParser" code that could sometimes cause a very short trailing NAL unit to not
+  get parsed properly.  (Thanks to John McNamara for reporting this issue.)
+- Made some changes to RTSP header parsing, to make it more robust.  (Thanks to Lionel Orry for the suggestion.)
+
+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.)
+
+2012.07.06:
+- Modified the implementation of "ServerMediaSession::generateSDPDescription()" to ignore (skip over) subsessions that use
+  medium or codec names that we don't understand.  (Previously, subsessions like this would cause a NULL SDP description to be
+  returned.)  This allows "ProxyServerMediaSession" to properly handle sessions like this.  (Thanks to Kiran Thakkar for
+  alerting us to an example that illustrated this problem.)
+
+2012.07.03:
+- Moved "ProxyRTSPClient" - which had previously been hidden inside the implementation of "ProxyServerMediaSession" - to
+  the "ProxyServerMediaSession.hh" header file, in case developers want to subclass it.  We also define a new virtual function
+  "ProxyServerMediaSession::createNewProxyRTSPClient()" whichh can be reimplemented to create such subclasses.
+  (Thanks to Jesus Leganes for this suggestion.)
+- Made "RTSPClient::sendRequest()" a virtual function, and made it "protected:" rather than "private:", in case subclasses
+  want to redefine it (e.g., to do additional functions such as logging).
+  (Thanks to Jesus Leganes for this suggestion.)
+
+2012.06.26:
+- Moved the code for ignoring the SIGPIPE signal to "RTSPCommon.cpp", and made it a function.  We now call this function from
+  the "RTSPClient" code, as well as the "RTSPServer" code.  (Thanks to Barry Stump for the suggestion.)
+- Added a new public member flag "describeCompletedFlag" to "ProxyServerMediaSession".  This can be used as a 'watch variable'
+  in a call to "doEventLoop()", to 'block' in the event loop until the back-end "DESCRIBE" command sent for a newly-created
+  "ProxyServerMediaSession" has completed.
+
+2012.06.23:
+- Fixed a bug in "H264VideoStreamParser" that affected the "testH264VideoToTransportStream" demo application.
+  (Thanks to John McNamara for helping to debug the problem.)
+- We no longer set a non-default "FileSink" buffer size in the "testH264VideoToTransportStream" demo application, because the
+  "FileSink" is fed by a "MPEG2TransportStreamFromESSource", ehich delivers only one 188-byte Transport Stream packet at a time.
+
+2012.06.17:
+- Fixed a bug in "HTTP Live Streaming" server support.  (Thanks to Daniel Wang for reporting the problem.)
+- Updated the implementation of "triggerEvent()" in "BasicTaskScheduler" to avoid a possible race condition (if "triggerEvent()"
+  is called from a non-LIVE555 thread).  (Thanks to Matthais Doering for noting this issue.)
+- Updated the "ProxyServerMediaSession" code to ensure that all front-end "RTCPInstance" objects are created with non-zero
+  bandwidth estimates.
+
+2012.06.12:
 - If a SDP description specifies a port number for a unicast stream, then we still choose an ephemeral client port number,
   just as we would if the SDP description had not specified a port number (the usual case). Renato Mauro reports that some Sony
   network cameras do this; this change lets the client choose a different port number (which is useful if one host is receiving
   from more than one such camera at once).
+- Fixed some minor bugs with Matroska file parsing.  (Thanks to Petr Novak.)
+- A field in "RTPTransmissionStats" was not being initialized in the constructor.  (Thanks to Michel Promonet for noting this.)
 - Updated "UsageEnvironment/include/Boolean.hh" to use the 'bool' type defined for "MSVC++ 8.0, Visual Studio 2005 and higher",
   if that development environment is being used.  Also, fixed a few places in the code where we were using boolean types
   incorrectly.  (Thanks to Nikolai Vorontsov for these suggestions.)

-- 
liblivemedia packaging