diff options
| -rw-r--r-- | ChangeLog | 81 | ||||
| -rw-r--r-- | NEWS | 18 | ||||
| -rw-r--r-- | RELEASE | 60 | ||||
| m--------- | common | 0 | ||||
| -rw-r--r-- | configure.ac | 12 | ||||
| -rw-r--r-- | gst-rtsp-server.doap | 12 |
6 files changed, 121 insertions, 62 deletions
@@ -1,9 +1,86 @@ +=== release 1.3.2 === + +2014-05-21 Sebastian Dröge <slomo@coaxion.net> + + * configure.ac: + releasing 1.3.2 + +2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com> + + * common: + Automatic update of common submodule + From 211fa5f to 1f5d3c3 + +2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com> + + * gst/rtsp-server/rtsp-client.c: + client: store TCP ports in transport + Store the TCP ports in the transport when we are doing RTSP over TCP. + This way, we can easily get to the ports from the transport. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776 + +2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com> + + * gst/rtsp-server/rtsp-stream.c: + stream: add signals for new RTP/RTCP encoders + New signals to allow the user to configure the dynamically created + encoders. + https://bugzilla.gnome.org/show_bug.cgi?id=730228 + +2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com> + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + media: Make suspend()/unsuspend() virtual + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109 + +2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com> + + * gst/rtsp-server/rtsp-client.c: + client: fix send-message signal marshaller + Use generic marshalling for the send-message signal. It has + two POINTER arguments, not just one. + https://bugzilla.gnome.org/show_bug.cgi?id=729900 + +2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com> + + * tests/check/gst/media.c: + tests: add and remove pads only once + In this test we simulate a dynamic pad by watching the caps event. + Because of renegotiation in the base payloader now, this caps is sent + multiple times but we can only deal with 1 invocation, use a variable to + only 'add and remove' the pad once. + +2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com> + + * tests/check/gst/rtspserver.c: + tests: add unit test for correct handling of Require headers + https://bugzilla.gnome.org/show_bug.cgi?id=729426 + +2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com> + + * gst/rtsp-server/rtsp-client.c: + rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED + Servers must handle Require headers and must report a failure + if they don't handle any of the Required options, see RFC 2326, + section 12.32: https://tools.ietf.org/html/rfc2326#page-54 + https://bugzilla.gnome.org/show_bug.cgi?id=729426 + +2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com> + + * configure.ac: + Back to development + === release 1.3.1 === -2014-05-03 Sebastian Dröge <slomo@coaxion.net> +2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com> + * ChangeLog: + * NEWS: + * RELEASE: * configure.ac: - releasing 1.3.1 + * gst-rtsp-server.doap: + Release 1.3.1 2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com> @@ -1,4 +1,4 @@ -This is GStreamer RTSP Server 1.3.1 +This is GStreamer RTSP Server 1.3.2 Changes since 1.2: @@ -45,6 +45,8 @@ New API: events and merge custom tags into them consistently. • playbin/playsink has support for application provided audio and video filters. + • GstDiscoverer has new and simplified API to get details about missing + plugins and information to pass to the plugin installer. • The GL library was merged from gst-plugins-gl to gst-plugins-bad, providing a generic infrastructure for handling GL inside GStreamer pipelines and a plugin with some elements using these, especially @@ -62,6 +64,14 @@ Major changes: of the existing V4L2 elements and the corresponding infrastructure. The v4l2videodec element replaces the mfcdec element. + ∘ New downloadbuffer element that replaces the download + buffering feature of queue2. Compared to queue2's code + it is much simpler and only for this single use case. + A noteworthy new feature is that it's downloading gaps + in the already downloaded stream parts when nothing else + is to be downloaded. + This is now used by playbin when download buffering is + enabled. ∘ rtpstreampay and rtpstreamdepay elements for transmitting RTP packets over a stream API (e.g. TCP) according to RFC 4571. @@ -78,7 +88,7 @@ Major changes: are available on OS X and iOS now. • Other changes: - ∘ gst-libav now uses libav 10, and gained support for H265/HEVC. + ∘ gst-libav now uses libav 10.1, and gained support for H265/HEVC. ∘ Support for hardware codecs and special memory types has been improved with bugfixes and feature additions in various plugins and base classes. @@ -95,6 +105,9 @@ Major changes: reliable now and supports more HLS features like trick modes. Also fragments are pushed downstream while they're downloaded now instead of waiting for each fragment to finish. + ∘ dashdemux and mssdemux are now also pushing fragments downstream + while they're downloaded instead of waiting for each fragment to + finish. ∘ videoflip can automatically flip based on the orientation tag. ∘ openjpeg supports the OpenJPEG2 API. ∘ gst-rtsp-server supports SRTP and MIKEY now. @@ -107,4 +120,3 @@ 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. - @@ -1,8 +1,8 @@ -Release notes for GStreamer RTSP Server Library 1.3.1 +Release notes for GStreamer RTSP Server Library 1.3.2 -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 1.3 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 @@ -28,24 +28,16 @@ change. +Features of this release + + Bugs fixed in this release - * 725484 : gst-rtsp-server: Ignore gcov intermediate files - * 725528 : rtspserver: Enable and fix gtk-doc warnings - * 725879 : rtsp-client: headers in GET response not configurable for tunnels - * 726362 : rtsp-stream: fix a typo where IPv4 and IPv6 addresses were confused. - * 726470 : tests: Add unit tests for sessionpool - * 726873 : rtsp-threadpool: Improve code coverage of check tests - * 726940 : rtsp-session-media: add more tests to improve code coverage - * 726941 : docs: Add annotations to support language bindings - * 727102 : rtsp-media: deadlock with dynamic pipelines when preroll fails - * 727231 : rtsp-server: The media streams leak - * 727376 : crash if media_prepare() fails to allocate UDP ports - * 727488 : There is a race when disconnecting POST channel in tunneled mode - * 728029 : rtsp-media: Make media_prepare() virtual - * 728060 : rtsp-session-pool: Incorrect annotation and leak in unit test - * 728153 : Problem with send_lock when data in backlog and recive a teardown request. - * 728970 : rtsp-client: add signal before sending response + * 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 ==== Download ==== @@ -84,41 +76,9 @@ Applications Contributors to this release - * Aleix Conchillo Flaque * Aleix Conchillo Flaqué - * Alessandro Decina - * Alexander Schrab - * Andrey Utkin - * Branko Subasic - * David Schleef - * David Svensson Fors - * Edward Hervey - * Emmanuel Pacaud - * Fabian Deutsch - * George McCollister - * Göran Jönsson - * Jonas Holmberg - * Linus Svensson - * Lubosz Sarnecki - * Luis de Bethencourt - * Mark Nauwelaerts - * Miguel Angel Cabrera Moya * Ognyan Tonchev - * Olivier Crête - * Patricia Muscalu - * Patrick Radizi - * Robert Krakora * Sebastian Dröge - * Sebastian Pölsterl - * Sebastian Rasmussen - * Stefan Kost - * Stefan Sauer - * Thijs Vermeir - * Thomas Vander Stichele * Tim-Philipp Müller - * Victor Gottardi - * Vincent Penquerc'h * Wim Taymans - * Youness Alaoui - * mat
\ No newline at end of file diff --git a/common b/common -Subproject 1f5d3c3163cc3399251827235355087c2affa79 +Subproject 211fa5f2d0930dfd6891b386d42edba6d88c2a1 diff --git a/configure.ac b/configure.ac index e56ba81..42aea34 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], [1.3.1.1], +AC_INIT([GStreamer RTSP Server Library], [1.3.2], [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, 301, 0, 301) +AS_LIBTOOL(GST, 302, 0, 302) dnl *** required versions of GStreamer stuff *** -GST_REQ=1.3.1.1 -GSTPB_REQ=1.3.1.1 -GSTPG_REQ=1.3.1.1 -GSTPD_REQ=1.3.1.1 +GST_REQ=1.3.2 +GSTPB_REQ=1.3.2 +GSTPG_REQ=1.3.2 +GSTPD_REQ=1.3.2 dnl *** autotools stuff **** diff --git a/gst-rtsp-server.doap b/gst-rtsp-server.doap index 2167481..2201ab2 100644 --- a/gst-rtsp-server.doap +++ b/gst-rtsp-server.doap @@ -28,7 +28,17 @@ RTSP server library based on GStreamer <location rdf:resource="git://anongit.freedesktop.org/gstreamer/gst-rtsp-server"/> <browse rdf:resource="http://cgit.freedesktop.org/gstreamer/gst-rtsp-server"/> </GitRepository> - </repository> +</repository> + + <release> + <Version> + <revision>1.3.2</revision> + <branch>1.3</branch> + <name></name> + <created>2014-05-21</created> + <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.3.2.tar.xz" /> + </Version> + </release> <release> <Version> |