[SCM] liblivemedia/master: Refresh upstream changelog.
alessio at users.alioth.debian.org
alessio at users.alioth.debian.org
Mon Mar 5 09:39:39 UTC 2012
The following commit has been merged in the master branch:
Author: Alessio Treglia <alessio at debian.org>
Date: Mon Mar 5 10:39:04 2012 +0100
Refresh upstream changelog.
diff --git a/debian/upstream.changelog b/debian/upstream.changelog
index 1cd3e50..dfaabc3 100644
@@ -1,3 +1,54 @@
+- 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
+ RTSPCLIENT_SYNCHRONOUS_INTERFACE before "RTSPClient.hh" gets included the first time.
+- Modified the 'multicast loopback' mechanism for getting our own IP address to check the source address of the received
+ multicast packet, to make sure that it's valid (e.g., not 127.0.0.1). (Thanks to Stefan Spurling for this suggestion.)
+- Updated "MediaSubsession::initiate()" to better handle the (relatively rare) case of UDP-only (i.e., non-RTP) streams that
+ specify a port number in the SDP description. In this case, because RTP is not being used, we accept the provided port number
+ even if it's odd, and we don't bother creating a RTCP 'groupsock'. (Thanks to John Orr for this suggestion.)
+- Updated "WAVAudioFileSource" to read from its input file asynchronously,
+ if possible, rather than doing a synchronous (blocking) read.
+- 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.
+- We now make the "MediaLookupTable" class visible in the header file "include/Media.hh". This allows developers to, if they wish,
+ iterate over the whole set of "Medium" objects that they've created.
+ (Thanks to Aviad Rozenhek for this suggestion.)
+- "HashTable::Iterator::create()" now takes a "HashTable const&" as parameter, rather than a "HashTable&".
+ (This makes it possible for iterators to work on (references to) hash tables that we've declared as const - for more type safety.)
+- Added a new class "MPEG2TransportUDPServerMediaSubsession" (a subclass of "OnDemandServerMediaSubsession") that can be used
+ to build a RTSP server that can takes a UDP (raw UDP or RTP/UDP) Transport Stream as input (via IP multicast, or unicast).
+ We also updated the "testOnDemandRTSPServer" demo application to show how a RTSP server can take input from the
+ (IP multicast) Transport Stream sent by the "testMPEG2TransportStreamer" demo application.
+ (Thanks to Achraf Gazdar for this suggestion.)
+- Added a new demo application (in "testProgs") "testMPEG2TransportReceiver"
+ (which can receive the MPEG-2 Transport/RTP stream sent by "testMPEG2TransportStreamer")
+- Fixed a bug in "StreamReplicator" that could cause 'replica deactivation' code to be executed more than once on the same replica,
+ with bad results. (Thanks to Mike Stewart for reporting this.)
+- Updated the 'connect()'-result test in "RTSPClient.cpp" to check for EWOULDBLOCK as well as EINPROGRESS.
+ (Windoze systems can apparently return EWOULDBLOCK. Thanks to Jeff Shanab for the suggestion.)
+- Added a new "liveMedia" class "StreamReplicator". This can be used to create an arbitrary number of 'replicas' of an input
+ stream. We also added a new demo application "testReplicator" (in "testProgs") that demonstrates how to use this class.
+- Made a small change to "testRTSPClient" to account for the possibility of a RTCP "BYE" being received after having sent a
+ RTSP "TEARDOWN".
- Added a new demo application "testRTSPClient" to the "testProgs" directory. If you're developing your own RTSP client
application (or want to embed RTSP client functionality into a larger application), then "testRTSPClient" is a better model
@@ -485,10 +536,10 @@
- Backed out the "RTSPServer" change that was in the previous release, because of a report that it doesn't work.
-- Updated "H2564VideoFileSink" to take an optional "sprop parameter string" parameter as input. If present, this string is
+- Updated "H264VideoFileSink" to take an optional "sprop parameter string" parameter as input. If present, this string is
decoded, and the resulting data (SPS/PPS NAL units) is prepended to the file.
Also, updated "openRTSP" to call "MediaSubsession::fmtp_spropparametersets()", and pass this string when creating a
- Updated "RTSPServer" to ensure that we transmit RTP and RTCP packets over the same interface that is used by the requesting client
(in case the server is multi-homed). (Thanks to David Stegbauer for this suggestion.)
@@ -2164,7 +2215,7 @@ QTSS/Concepts/chapter_2_section_14.html>.
- Fixed a bug in "AMRAudioRTPSource", in the way that it handled input RTP packets
that contained more than one AMR frame.
(Thanks to Gabriel Bouvigne for reporting this problem.)
-- We no automatically reclaim the memory that was allocated for the "UsageEnvironment"
+- We now automatically reclaim the memory that was allocated for the "UsageEnvironment"
"liveMediaPriv" and "groupsockPriv" structures, once their tables become empty.
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