Release 1.7.21.7.2

5 files changed, 259 insertions, 30 deletions

diff --git a/ChangeLog b/ChangeLog
index a931568..dfd82d0 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,9 +1,237 @@
+=== release 1.7.2 ===
+
+2016-02-19  Sebastian Dröge <slomo@coaxion.net>
+
+	* configure.ac:
+	  releasing 1.7.2
+
+2016-02-18 15:20:05 +0000  Julien Isorce <j.isorce@samsung.com>
+
+	* pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
+	  uninstalled.pc: add support for non libtool build systems
+	  Currently the .la path is provided which requires to use libtool as
+	  mentioned in the GStreamer manual section-helloworld-compilerun.html.
+	  It is fine as long as the application is built using libtool.
+	  So currently it is not possible to compile a GStreamer application
+	  within gst-uninstalled with CMake or other build system different
+	  than autotools.
+	  This patch allows to do the following in gst-uninstalled env:
+	  gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \
+	  gstreamer-rtsp-server-1.0)
+	  Previously it required to prepend libtool --mode=link
+	  https://bugzilla.gnome.org/show_bug.cgi?id=720778
+
+2016-02-09 10:34:22 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* gst/rtsp-sink/gstrtspclientsink.c:
+	  rtspclientsink: remove check for impossible condition
+	  Goto error label checks stream to see if it needs to be unreferenced before
+	  returning, but this goto jumps happens before the stream is ever set, so it
+	  will always be NULL in this error label.
+	  CID #1352034
+
+2016-02-08 23:33:03 +0000  Luis de Bethencourt <luisbg@osg.samsung.com>
+
+	* gst/rtsp-sink/gstrtspclientsink.c:
+	  rtspclientsink: clean switch statements
+	  Coverity demands for fallthrough statements to be clearly commented,
+	  to distinguish from accidental fall throughs. And it also needs all
+	  cases to finish with a break, even if the break is never going to be
+	  executed like in the case of a continue jump.
+	  CID #1352039
+	  CID #1352040
+
+2016-02-05 20:03:01 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* tests/check/Makefile.am:
+	  tests: extend the AM_TESTS_ENVIRONMENT from check.mak
+	  To get the CK_DEFAULT_TIMEOUT defined for all tests
+	  Also removes a 120 seconds timeout that was set as default
+	  explicitly in this module
+	  https://bugzilla.gnome.org/show_bug.cgi?id=761472
+
+2016-02-05 18:11:41 -0300  Thiago Santos <thiagoss@osg.samsung.com>
+
+	* autogen.sh:
+	* common:
+	  Automatic update of common submodule
+	  From 86e4663 to b64f03f
+
+2016-02-02 09:01:51 +0100  Steven Hoving <sh@bigbrother.nl>
+
+	* gst/rtsp-server/rtsp-media.c:
+	  rtsp-media: fix state_lock not locked again when preroll fails
+	  https://bugzilla.gnome.org/show_bug.cgi?id=761399
+
+2016-01-28 22:05:56 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* configure.ac:
+	  configure: Move plugin specific flags below all the others
+	  They use some of the other flags, like $GST_ALL_LDFLAGS which is adding
+	  -no-undefined. And -no-undefined is required on Windows to build DLLs.
+
+2016-01-28 04:58:00 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/rtsp-sink/gstrtspclientsink.c:
+	  rtspclientsink: Simplify slightly using new -base API
+	  Use the new Mikey and SDP API in the base plugins libs
+	  to simplify some code.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=758180
+
+2015-11-17 01:12:28 +1100  Jan Schmidt <jan@centricular.com>
+
+	* .gitignore:
+	* configure.ac:
+	* gst/Makefile.am:
+	* gst/rtsp-sink/Makefile.am:
+	* gst/rtsp-sink/gstrtspclientsink.c:
+	* gst/rtsp-sink/gstrtspclientsink.h:
+	* gst/rtsp-sink/plugin.c:
+	* tests/check/Makefile.am:
+	* tests/check/gst/rtspclientsink.c:
+	  rtspsink: Add rtspclientsink element
+	  Add an rtspclientsink element that accepts streams for which
+	  there is a registered payloader and sends them to
+	  an RTSP server using RECORD.
+	  Sending is synchronised to the pipeline clock. Payload-types
+	  are automatically selected. The 'new-payloader' signal is fired
+	  for custom configuration of payloaders when they are created.
+	  Can now stream a movie like this:
+	  receiver:
+	  ./test-record "( decodebin name=depay0 ! videoconvert ! autovideosink \
+	  decodebin name=depay1 ! audioconvert ! autoaudiosink )"
+	  sender:
+	  gst-launch-1.0 filesrc location=file-with-aac-and-h264.mp4 ! qtdemux name=d ! \
+	  queue ! aacparse ! rtspclientsink location=rtsp://127.0.0.1:8554/test name=s \
+	  https://bugzilla.gnome.org/show_bug.cgi?id=758180
+
+2015-11-17 01:12:28 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/rtsp-server/rtsp-stream.c:
+	* gst/rtsp-server/rtsp-stream.h:
+	  rtsp-stream: Add functions for using rtsp-stream from the client
+	  Add a boolean to indicate that the rtsp-stream is running on the
+	  'client' side of an RTSP connection, for sending streams via
+	  RECORD. In that case, the roles of the client/server ports
+	  in transport setup are swapped.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=758180
+
+2015-11-17 01:12:28 +1100  Jan Schmidt <jan@centricular.com>
+
+	* gst/rtsp-server/rtsp-sdp.c:
+	* gst/rtsp-server/rtsp-sdp.h:
+	  rtsp-sdp: Add gst_rtsp_sdp_from_stream()
+	  A new function that adds info from a GstRTSPStream into an SDP message.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=758180
+
+2016-01-28 09:22:18 +0100  Steven Hoving <sh@bigbrother.nl>
+
+	* gst/rtsp-server/rtsp-media.c:
+	  rtsp-media: Fix mutex beeing unlocked while they should be locked
+	  https://bugzilla.gnome.org/show_bug.cgi?id=761226
+
+2016-01-15 07:01:37 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/rtsp-server/rtsp-media-factory.c:
+	  rtsp-media-factory: add missing break in "clock" property setter
+	  CID 1348453
+
+2016-01-05 13:10:36 +0100  Srimanta Panda <srimanta@axis.com>
+
+	* gst/rtsp-server/rtsp-stream.c:
+	  rtsp-stream: fixed assert during update transport
+	  When RTSP server trying update transport during multicast, it throws an
+	  assert. The assert is thrown because it is trying to get the parent of
+	  an non-existing funnel element.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=760150
+
+2016-01-03 17:26:31 +0000  Tim-Philipp Müller <tim@centricular.com>
+
+	* gst/rtsp-server/rtsp-permissions.h:
+	* gst/rtsp-server/rtsp-thread-pool.h:
+	* gst/rtsp-server/rtsp-token.h:
+	  docs: remove dummy function declarations with G_INLINE_FUNC for gtk-doc
+	  gtk-doc can handle static inline functions just fine these days,
+	  there's no need for this stuff any more.
+
+2015-10-07 18:53:01 +0900  Hyunjun Ko <zzoon.ko@samsung.com>
+
+	* gst/rtsp-server/rtsp-media.c:
+	* gst/rtsp-server/rtsp-sdp.c:
+	  sdp: replace duplicated codes to call new base sdp apis
+	  https://bugzilla.gnome.org/show_bug.cgi?id=745880
+
+2015-12-30 16:34:30 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* examples/test-netclock.c:
+	  test-netclock: Use the new API to configure a clock directly
+
+2015-12-30 16:31:13 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtsp-server/rtsp-media-factory.c:
+	* gst/rtsp-server/rtsp-media-factory.h:
+	* gst/rtsp-server/rtsp-media.c:
+	* gst/rtsp-server/rtsp-media.h:
+	  rtsp-media: Add API to directly configure a clock on the media pipelines
+
+2015-12-30 16:43:17 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtsp-server/rtsp-media.c:
+	  rtsp-media: Fix typo in docs gst_rtsp_media_set_latncy() -> latency()
+
+2015-12-30 16:30:38 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtsp-server/rtsp-media-factory.c:
+	  rtsp-media-factory: Add FIXME for 2.0
+
+2015-12-30 16:29:45 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtsp-server/rtsp-stream.c:
+	  rtsp-stream: Fix indentation
+
+2015-12-22 12:08:02 +0100  Sebastian Rasmussen <sebras@hotmail.com>
+
+	* gst/rtsp-server/rtsp-media.c:
+	  rtsp-media: Do not prepare media after media times out
+	  Deferred calls to start_prepare() can be deferred past the point until
+	  which wait_preroll() and by proxy gst_rtsp_media_get_status() is
+	  prepared to wait. Previously there was no lock and no check for this
+	  situation. This meant that a media could be prepared and unprepared
+	  simultaneously by two different threads. Now a lock is in place and a
+	  suitable check is done.
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759773
+
+2015-12-09 18:24:24 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* gst/rtsp-server/rtsp-client.c:
+	* gst/rtsp-server/rtsp-media-factory.c:
+	* gst/rtsp-server/rtsp-media-factory.h:
+	* gst/rtsp-server/rtsp-media.c:
+	* gst/rtsp-server/rtsp-media.h:
+	  rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN
+	  Without TEARDOWN it might be desireable to keep the media running and continue
+	  sending data to the client, even if the RTSP connection itself is
+	  disconnected.
+	  Only do this for session medias that have only UDP transports. If there's at
+	  least on TCP transport, it will stop working and cause problems when the
+	  connection is disconnected.
+	  https://bugzilla.gnome.org/show_bug.cgi?id=758999
+
+2015-12-24 15:29:33 +0100  Sebastian Dröge <sebastian@centricular.com>
+
+	* configure.ac:
+	  Back to development
+
 === release 1.7.1 ===
 
-2015-12-24  Sebastian Dröge <slomo@coaxion.net>
+2015-12-24 14:54:06 +0100  Sebastian Dröge <sebastian@centricular.com>
 
+	* ChangeLog:
+	* NEWS:
+	* RELEASE:
 	* configure.ac:
-	  releasing 1.7.1
+	* gst-rtsp-server.doap:
+	  Release 1.7.1
 
 2015-12-21 00:43:49 +0100  Koop Mast <kwm@rainbow-runner.nl>
 
diff --git a/NEWS b/NEWS
index a4bffc6..a4a5e76 100644
--- a/NEWS
+++ b/NEWS
@@ -1,2 +1,2 @@
-This is GStreamer 1.7.1
+This is GStreamer 1.7.2
 
diff --git a/RELEASE b/RELEASE
index f2322a4..37b4315 100644
--- a/RELEASE
+++ b/RELEASE
@@ -1,8 +1,7 @@
 
-Release notes for GStreamer RTSP Server Library 1.7.1
+Release notes for GStreamer RTSP Server Library 1.7.2
 
-
-The GStreamer team is pleased to announce the first release of the unstable
+The GStreamer team is pleased to announce the second release of the unstable
 1.7 release series. The 1.7 release series is adding new features on top of
 the 1.0, 1.2, 1.4 and 1.6 series and is part of the API and ABI-stable 1.x release
 series of the GStreamer multimedia framework. The unstable 1.7 release series
@@ -13,31 +12,26 @@ API can still change until that point.
 Binaries for Android, iOS, Mac OS X and Windows will be provided separately
 during the unstable 1.7 release series.
 
-
   
 
 Bugs fixed in this release
      
-      * 753863 : rtsp-server: examples: Fix memory leaks when context parse fails
-      * 756969 : rtsp-server unit tests don't test udp-mcast transport
-      * 757949 : gst_rtsp_server_io_func() pops a context that has not been pushed
-      * 758179 : GstRTSPStream : Create pipeline based on enabled transport type
-      * 758268 : handle_setup_request() expect the media to be suspended
-      * 758319 : rtsp-server: Seeking often hangs forever, waiting for prerolling to happen again
-      * 758364 : rtsp-session-pool: Avoid dollar sign ($) in session ids
-      * 759010 : Valgrind test are faling for rtsp-server for master
+      * 758180 : Add rtspclientsink plugin
+      * 758999 : rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN
+      * 759773 : Prevent simultaneous prepare/unprepare of RTSP media
+      * 760150 : Updating transport for multicast case gives assertion
 
 ==== Download ====
 
 You can find source releases of gst-rtsp-server in the download
-directory: http://gstreamer.freedesktop.org/src/gst-rtsp-server/
+directory: https://gstreamer.freedesktop.org/src/gst-rtsp-server/
 
 The git repository and details how to clone it can be found at
 http://cgit.freedesktop.org/gstreamer/gst-rtsp-server/
 
 ==== Homepage ====
 
-The project's website is http://gstreamer.freedesktop.org/
+The project's website is https://gstreamer.freedesktop.org/
 
 ==== Support and Bugs ====
 
@@ -62,17 +56,14 @@ subscribe to the gstreamer-devel list.
         
 Contributors to this release
     
-      * David Svensson Fors
       * Hyunjun Ko
       * Jan Schmidt
-      * Koop Mast
-      * Marcus Prebble
-      * Nicolas Dufresne
-      * Olivier Crête
+      * Julien Isorce
+      * Luis de Bethencourt
       * Sebastian Dröge
       * Sebastian Rasmussen
       * Srimanta Panda
+      * Steven Hoving
+      * Thiago Santos
       * Tim-Philipp Müller
-      * Vineeth TM
-      * Xavier Claessens
  
\ No newline at end of file
diff --git a/configure.ac b/configure.ac
index 83ac172..35e3987 100644
--- a/configure.ac
+++ b/configure.ac
@@ -2,7 +2,7 @@ AC_PREREQ(2.69)
 dnl initialize autoconf
 dnl when going to/from release please set the nano (fourth number) right !
 dnl releases only do Wall, cvs and prerelease does Werror too
-AC_INIT([GStreamer RTSP Server Library], [1.7.1.1],
+AC_INIT([GStreamer RTSP Server Library], [1.7.2],
     [http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],
     [gst-rtsp-server])
 AG_GST_INIT
@@ -53,13 +53,13 @@ dnl      1.2.5 => 205
 dnl      1.10.9 (who knows) => 1009
 dnl
 dnl sets GST_LT_LDFLAGS
-AS_LIBTOOL(GST, 701, 0, 701)
+AS_LIBTOOL(GST, 702, 0, 702)
 
 dnl *** required versions of GStreamer stuff ***
-GST_REQ=1.7.1.1
-GSTPB_REQ=1.7.1.1
-GSTPG_REQ=1.7.1.1
-GSTPD_REQ=1.7.1.1
+GST_REQ=1.7.2
+GSTPB_REQ=1.7.2
+GSTPG_REQ=1.7.2
+GSTPD_REQ=1.7.2
 
 dnl *** autotools stuff ****
 
diff --git a/gst-rtsp-server.doap b/gst-rtsp-server.doap
index b202efe..eae21a2 100644
--- a/gst-rtsp-server.doap
+++ b/gst-rtsp-server.doap
@@ -32,6 +32,16 @@ RTSP server library based on GStreamer
 
  <release>
   <Version>
+   <revision>1.7.2</revision>
+   <branch>master</branch>
+   <name></name>
+   <created>2016-02-19</created>
+   <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.7.2.tar.xz" />
+  </Version>
+ </release>
+
+ <release>
+  <Version>
    <revision>1.7.1</revision>
    <branch>master</branch>
    <name></name>