Release 1.7.90

diff --git a/ChangeLog b/ChangeLog
index dfd82d0..97df09b 100644 (file)
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,9 +1,116 @@
+=== release 1.7.90 ===
+
+2016-03-01  Sebastian Dröge <slomo@coaxion.net>
+
+       * configure.ac:
+         releasing 1.7.90
+
+2016-02-26 12:42:51 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+       * common:
+         Automatic update of common submodule
+         From b64f03f to 6f2d209
+
+2016-02-24 00:10:52 +1100  Jan Schmidt <jan@centricular.com>
+
+       * gst/rtsp-sink/gstrtspclientsink.c:
+       * tests/check/gst/rtspclientsink.c:
+         rtspsink: Fix some leaks in rtspclientsink and the unit test.
+         https://bugzilla.gnome.org/show_bug.cgi?id=762525
+
+2016-02-23 15:01:22 +0100  Patricia Muscalu <patricia@axis.com>
+
+       * tests/check/gst/media.c:
+       * tests/check/gst/rtspclientsink.c:
+       * tests/check/gst/rtspserver.c:
+       * tests/check/gst/stream.c:
+         tests: unit test fixes
+         Removed port allocation test from the media suite.
+         The port allocation failure is now in the stream suite.
+         rtspserver:
+         Make sure that the media is suspended after the DESCRIBE request
+         before reconfiguring the UDP sinks.
+         rtspclientsink:
+         In the RECORD case we have to set async property to false
+         for the appsink element in the test in order to make sure
+         that the media pipeline doesn't hang in start_preroll().
+         https://bugzilla.gnome.org/show_bug.cgi?id=757488
+
+2016-02-23 14:59:32 +0100  Patricia Muscalu <patricia@axis.com>
+
+       * gst/rtsp-server/rtsp-client.c:
+       * gst/rtsp-server/rtsp-stream.c:
+       * gst/rtsp-server/rtsp-stream.h:
+         rtsp-stream: postpone UDP socket allocation until SETUP
+         Postpone the allocation of the UDP sockets until we know
+         what transport has been chosen by the client.
+         Both unicast and multicast UDP sources are created in one
+         function.
+         https://bugzilla.gnome.org/show_bug.cgi?id=757488
+
+2016-01-13 11:29:35 +0100  Patricia Muscalu <patricia@axis.com>
+
+       * gst/rtsp-server/rtsp-stream.c:
+         rtsp-stream: postpone the creation of the UDP sources
+         Code refactoring: allocate the UDP ports after the sender and
+         the reciver parts have been created.
+         We postpone the creation of the UDP sources until the UDP
+         ports have been allocated.
+         https://bugzilla.gnome.org/show_bug.cgi?id=757488
+
+2016-01-13 10:55:40 +0100  Patricia Muscalu <patricia@axis.com>
+
+       * gst/rtsp-server/rtsp-stream.c:
+         rtsp-stream: added function for setting UDP sources to PLAYING state
+         Code refactoring: Introduced a function for setting UDP sources
+         to PLAYING state.
+         https://bugzilla.gnome.org/show_bug.cgi?id=757488
+
+2015-11-20 15:34:43 +0100  Patricia Muscalu <patricia@axis.com>
+
+       * gst/rtsp-server/rtsp-stream.c:
+         rtsp-stream: added function for creating and configuring UDP sources
+         Code refactoring: create and configure UDP sources in a separate function.
+         https://bugzilla.gnome.org/show_bug.cgi?id=757488
+
+2015-11-20 14:43:38 +0100  Patricia Muscalu <patricia@axis.com>
+
+       * gst/rtsp-server/rtsp-stream.c:
+         rtsp-stream: added function for RTP/RTCP socket configuration
+         Code refactoring: configure RTP and RTCP sockets for UDP sinks
+         in a separate function.
+         https://bugzilla.gnome.org/show_bug.cgi?id=757488
+
+2015-11-20 08:38:42 +0100  Patricia Muscalu <patricia@axis.com>
+
+       * gst/rtsp-server/rtsp-stream.c:
+         rtsp-stream: added function for creating and configuring UDP sinks
+         Code refactoring: create and configure UDP sinks in a separate function.
+         https://bugzilla.gnome.org/show_bug.cgi?id=757488
+
+2015-11-19 14:09:25 +0100  Patricia Muscalu <patricia@axis.com>
+
+       * gst/rtsp-server/rtsp-stream.c:
+         rtsp-stream: added helper function for creating the sender/receiver parts
+         Code refactoring: introduced helper function for creating
+         the receiver and the sender parts of the streaming pipeline.
+         https://bugzilla.gnome.org/show_bug.cgi?id=757488
+
+2016-02-19 12:38:42 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+       * configure.ac:
+         Back to development
+
 === release 1.7.2 ===
 
-2016-02-19  Sebastian Dröge <slomo@coaxion.net>
+2016-02-19 12:03:18 +0200  Sebastian Dröge <sebastian@centricular.com>
 
+       * ChangeLog:
+       * NEWS:
+       * RELEASE:
        * configure.ac:
-         releasing 1.7.2
+       * gst-rtsp-server.doap:
+         Release 1.7.2
 
 2016-02-18 15:20:05 +0000  Julien Isorce <j.isorce@samsung.com>
 

diff --git a/NEWS b/NEWS
index a4a5e76..8f11fc4 100644 (file)
--- a/NEWS
+++ b/NEWS
@@ -1,2 +1,2 @@
-This is GStreamer 1.7.2
+This is GStreamer 1.7.90
 

diff --git a/RELEASE b/RELEASE
index 37b4315..6aa933c 100644 (file)
--- a/RELEASE
+++ b/RELEASE
@@ -1,25 +1,21 @@
 
-Release notes for GStreamer RTSP Server Library 1.7.2
+Release notes for GStreamer RTSP Server Library 1.7.90
 
-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 GStreamer team is pleased to announce the first release candidate of the stable
+1.8 release series. The 1.8 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
-will lead to the stable 1.8 release series in the next weeks. Any newly added
-API can still change until that point.
+series of the GStreamer multimedia framework.
 
 
 Binaries for Android, iOS, Mac OS X and Windows will be provided separately
-during the unstable 1.7 release series.
+during the stable 1.8 release series.
 
   
 
 Bugs fixed in this release
      
-      * 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
+      * 757488 : gst-rtsp-server: multicast functionality is broken if using the same port ranges for multicast and unicast
+      * 762525 : rtspclientsink test: Memory leaks
 
 ==== Download ====
 
@@ -56,14 +52,7 @@ subscribe to the gstreamer-devel list.
         
 Contributors to this release
     
-      * Hyunjun Ko
       * Jan Schmidt
-      * Julien Isorce
-      * Luis de Bethencourt
+      * Patricia Muscalu
       * Sebastian Dröge
-      * Sebastian Rasmussen
-      * Srimanta Panda
-      * Steven Hoving
-      * Thiago Santos
-      * Tim-Philipp Müller
  
\ No newline at end of file

diff --git a/configure.ac b/configure.ac
index 7f57cfa..1261bc3 100644 (file)
--- 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], [1.7.2.1],
+AC_INIT([GStreamer RTSP Server Library], [1.7.90],
     [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, 702, 0, 702)
+AS_LIBTOOL(GST, 790, 0, 790)
 
 dnl *** required versions of GStreamer stuff ***
-GST_REQ=1.7.2.1
-GSTPB_REQ=1.7.2.1
-GSTPG_REQ=1.7.2.1
-GSTPD_REQ=1.7.2.1
+GST_REQ=1.7.90
+GSTPB_REQ=1.7.90
+GSTPG_REQ=1.7.90
+GSTPD_REQ=1.7.90
 
 dnl *** autotools stuff ****
 

diff --git a/gst-rtsp-server.doap b/gst-rtsp-server.doap
index eae21a2..52eb47d 100644 (file)
--- a/gst-rtsp-server.doap
+++ b/gst-rtsp-server.doap
@@ -32,6 +32,16 @@ RTSP server library based on GStreamer
 
  <release>
   <Version>
+   <revision>1.7.90</revision>
+   <branch>master</branch>
+   <name></name>
+   <created>2016-03-01</created>
+   <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.7.90.tar.xz" />
+  </Version>
+ </release>
+
+ <release>
+  <Version>
    <revision>1.7.2</revision>
    <branch>master</branch>
    <name></name>