Release 1.3.31.3.3

5 files changed, 110 insertions, 31 deletions

diff --git a/ChangeLog b/ChangeLog
index f4495d0..1777857 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,9 +1,68 @@
+=== release 1.3.3 ===
+
+2014-06-22  Sebastian Dröge <slomo@coaxion.net>
+
+	* configure.ac:
+	  releasing 1.3.3
+
+2014-05-20 14:48:37 -0700  Aleix Conchillo Flaqué <aleix@oblong.com>
+
+	* gst/rtsp-server/rtsp-client.c:
+	* gst/rtsp-server/rtsp-sdp.c:
+	* gst/rtsp-server/rtsp-sdp.h:
+	  mikey: add different key length parameters
+	  Add encryption and authentication key length parameters to MIKEY. For
+	  the encoders, the key lengths are obtained from the cipher and auth
+	  algorithms set in the caps. For the decoders, they are obtained while
+	  parsing the key management from the client.
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472
+
+2014-03-16 17:29:48 +0100  Ognyan Tonchev <otonchev@gmail.com>
+
+	* tests/check/gst/stream.c:
+	  stream tests: Make sure we get right multicast address from stream
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577
+
+2014-06-12 13:49:17 +0200  Ognyan Tonchev <ognyan@axis.com>
+
+	* gst/rtsp-server/rtsp-client.c:
+	  client: ref the context until rtsp watch is alive
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569
+
+2014-06-12 13:48:44 +0200  Ognyan Tonchev <ognyan@axis.com>
+
+	* gst/rtsp-server/rtsp-client.c:
+	  client: Destroy the rtsp watch after connection close
+
+2014-06-13 16:46:06 +0200  Wim Taymans <wtaymans@redhat.com>
+
+	* gst/rtsp-server/rtsp-media.c:
+	  media: fix confusing comment
+
+2014-05-27 12:36:52 +0200  Göran Jönsson <goranjn@axis.com>
+
+	* gst/rtsp-server/rtsp-session.c:
+	  rtsp-session: Timeout in header.
+	  Adding the possbilty to always have timout in header.
+	  This is configurabe with setting "timeout-always-visible".
+	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264
+
+2014-05-21 13:23:40 +0200  Sebastian Dröge <sebastian@centricular.com>
+
+	* configure.ac:
+	  Back to development
+
 === release 1.3.2 ===
 
-2014-05-21  Sebastian Dröge <slomo@coaxion.net>
+2014-05-21 13:06:36 +0200  Sebastian Dröge <sebastian@centricular.com>
 
+	* ChangeLog:
+	* NEWS:
+	* RELEASE:
+	* common:
 	* configure.ac:
-	  releasing 1.3.2
+	* gst-rtsp-server.doap:
+	  Release 1.3.2
 
 2014-05-21 10:54:05 +0200  Sebastian Dröge <sebastian@centricular.com>
 
diff --git a/NEWS b/NEWS
index 1011887..be406ee 100644
--- a/NEWS
+++ b/NEWS
@@ -1,4 +1,4 @@
-This is GStreamer RTSP Server 1.3.2
+This is GStreamer RTSP Server 1.3.3
 
 Changes since 1.2:
 
@@ -30,6 +30,10 @@ New API:
    caps.
  • GstCollectPads has support for flushing and a default handler for
    SEEK events now.
+ • New GstFlowAggregator helper object that simplifies handling of
+   flow returns in elements with multiple source pads. Additionally
+   GstPad now always stores the last flow return and provides an
+   API to retrieve it.
  • GstSegment has new API to offset the running time by a specific
    value and this is used in GstPad to allow positive and negative
    offsets in gst_pad_set_offset() in all situations.
@@ -43,6 +47,7 @@ New API:
  • Support for tiled, raw video formats has been added.
  • GstVideoDecoder and GstAudioDecoder have API to help aggregating tag
    events and merge custom tags into them consistently.
+ • GstBufferPool has support for flushing now.
  • playbin/playsink has support for application provided audio and video
    filters.
  • GstDiscoverer has new and simplified API to get details about missing
@@ -54,6 +59,10 @@ New API:
    DispManX (Raspberry Pi), EAGL (iOS), WGL (Windows) and generic X11,
    Wayland and EGL platforms.
    This replaces eglglessink and also is supposed to replace osxvideosink.
+ • New GstAggregator base class in gst-plugins-bad. This is supposed to
+   replace GstCollectPads in the future and fix long-known shortcomings
+   in its API. Together with the base class some elements are provided
+   already, like a videomixer (compositor).
 
 
 Major changes:
@@ -97,7 +106,8 @@ Major changes:
    ∘ dvbsrc supports more delivery mechanisms and other features
      now, including DVB S2 and T2 support.
    ∘ The MPEGTS library has support for many more descriptors.
-   ∘ Major improvements to tsdemux, especially time related.
+   ∘ Major improvements to tsdemux and tsparse, especially time and
+     seeking related.
    ∘ souphttpsrc now has support for keep-alive connections,
      compression, configurable number of retries and configuration
      for SSL certificate validation.
@@ -110,9 +120,16 @@ Major changes:
      finish.
    ∘ videoflip can automatically flip based on the orientation tag.
    ∘ openjpeg supports the OpenJPEG2 API.
+   ∘ waylandsink was refactored and should be more useful now. It also
+     includes a small library which most likely is going to be removed
+     in the future and will result in extensions to the GstVideoOverlay
+     interface.
    ∘ gst-rtsp-server supports SRTP and MIKEY now.
+   ∘ gst-libav encoders are now negotiating any profile/level settings
+     with downstream via caps.
    ∘ Lots of fixes for coverity warnings all over the place.
-   ∘ 400+ fixed bug reports, and many other bug fixes and other
+   ∘ Negotiation related performance improvements.
+   ∘ 500+ fixed bug reports, and many other bug fixes and other
      improvements everywhere that had no bug report.
 
 Things to look out for:
@@ -120,3 +137,5 @@ Things to look out for:
    element.
  • The mfcdec element was removed and replaced by v4l2videodec.
  • osxvideosink is only available in OS X 10.6 or newer.
+ • The GstDeviceMonitor API will likely change slightly before the
+   1.4.0 release.
diff --git a/RELEASE b/RELEASE
index c7d8057..341e1e8 100644
--- a/RELEASE
+++ b/RELEASE
@@ -1,8 +1,7 @@
 
-Release notes for GStreamer RTSP Server Library 1.3.2
+Release notes for GStreamer RTSP Server Library 1.3.3
 
-
-The GStreamer team is pleased to announce the second release of the unstable
+The GStreamer team is pleased to announce the third release of the unstable
 1.3 release series. The 1.3 release series is adding new features on top of
 the 1.0 and 1.2 series and is part of the API and ABI-stable 1.x release
 series of the GStreamer multimedia framework. The unstable 1.3 release series
@@ -10,22 +9,15 @@ will lead to the stable 1.4 release series in the next weeks, and newly added
 API can still change until that point.
 
 
+This is hopefully the last 1.3 development release and will be followed by
+the first 1.4.0 release candidate (1.3.90) in 1-2 weeks. Which then hopefully
+is followed by 1.4.0 soonish in early July.
+
 
 Binaries for Android, iOS, Mac OS X and Windows will be provided separately
 during the unstable 1.3 release series.
 
 
-
-The versioning scheme that is used in general is that 1.x.y is API and
-ABI backwards compatible with previous 1.x.y releases. If x is an even
-number it is a stable release series and all releases in this series
-will only contain important bugfixes, e.g. the 1.0 series with 1.0.7. If
-x is odd it is a development release series that will lead to the next
-stable release series 1.x+1 and contains new features and bigger
-changes. During the development release series, new API can still
-change.
-
-
   
 
 Features of this release
@@ -33,11 +25,10 @@ Features of this release
 
 Bugs fixed in this release
      
-      * 729426 : Should respond " 551 Option not supported " in case a Require header is received
-      * 729776 : Set client port from URL
-      * 729900 : rtsp-client: wrong marshalling in send-message signal
-      * 730109 : media: Make suspend()/unsuspend() virtual
-      * 730228 : stream: add signals for new RTP/RTCP encoders
+      * 728264 : No timeout in header when timeout is 60s
+      * 730472 : mikey: add different key length parameters
+      * 731569 : Server does not free all resources if session timeout
+      * 731577 : new unit test for gst/stream
 
 ==== Download ====
 
@@ -77,8 +68,8 @@ Applications
 Contributors to this release
     
       * Aleix Conchillo Flaqué
+      * Göran Jönsson
       * Ognyan Tonchev
       * Sebastian Dröge
-      * Tim-Philipp Müller
       * Wim Taymans
  
\ No newline at end of file
diff --git a/configure.ac b/configure.ac
index 1b053e7..49687d2 100644
--- a/configure.ac
+++ b/configure.ac
@@ -2,7 +2,7 @@ AC_PREREQ(2.62)
 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], [1.3.2.1],
+AC_INIT([GStreamer RTSP Server Library], [1.3.3],
     [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, 302, 0, 302)
+AS_LIBTOOL(GST, 303, 0, 303)
 
 dnl *** required versions of GStreamer stuff ***
-GST_REQ=1.3.2.1
-GSTPB_REQ=1.3.2.1
-GSTPG_REQ=1.3.2.1
-GSTPD_REQ=1.3.2.1
+GST_REQ=1.3.3
+GSTPB_REQ=1.3.3
+GSTPG_REQ=1.3.3
+GSTPD_REQ=1.3.3
 
 dnl *** autotools stuff ****
 
diff --git a/gst-rtsp-server.doap b/gst-rtsp-server.doap
index 2201ab2..28fb725 100644
--- a/gst-rtsp-server.doap
+++ b/gst-rtsp-server.doap
@@ -32,6 +32,16 @@ RTSP server library based on GStreamer
 
  <release>
   <Version>
+   <revision>1.3.3</revision>
+   <branch>1.3</branch>
+   <name></name>
+   <created>2014-06-22</created>
+   <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.3.3.tar.xz" />
+  </Version>
+ </release>
+
+ <release>
+  <Version>
    <revision>1.3.2</revision>
    <branch>1.3</branch>
    <name></name>