+=== release 1.11.1 ===
+
+2017-01-12 Sebastian Dröge <slomo@coaxion.net>
+
+ * configure.ac:
+ releasing 1.11.1
+
+2017-01-10 08:34:50 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: corrected if-statement in _get_server_port()
+ This bug was accidentally introduced while fixing a segfault
+ in _get_server_port() function.
+ https://bugzilla.gnome.org/show_bug.cgi?id=776345
+
+2017-01-09 14:12:05 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * tests/check/gst/stream.c:
+ rtsp-stream: fixed segmenation fault in _get_server_port()
+ Calling function gst_rtsp_stream_get_server_port() results in
+ segmenation fault in the RTP/RTSP/TCP case.
+ Port that the server will use to receive RTCP makes only
+ sense in the UDP case, however the function should handle
+ the TCP case in a nicer way.
+ https://bugzilla.gnome.org/show_bug.cgi?id=776345
+
+2017-01-09 12:22:40 +0300 Aleksandr Slobodeniuk <alenuke@yandex.ru>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ dosc: Fix a little typo
+ https://bugzilla.gnome.org/show_bug.cgi?id=777037
+
+2017-01-04 16:20:54 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * pkgconfig/Makefile.am:
+ * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
+ * pkgconfig/meson.build:
+ meson: generate pkg-config -uninstalled pc files
+ Generating those files is useful for users building the GStreamer stack
+ using meson and having to link it to another project which is still
+ using the autotools.
+ https://bugzilla.gnome.org/show_bug.cgi?id=776810
+
+2017-01-04 16:11:08 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
+ pkgconfig: fix -uninstalled pc file
+ pcfiledir was never defined so the paths were wrong.
+ https://bugzilla.gnome.org/show_bug.cgi?id=776867
+
+2016-12-21 13:41:50 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * tests/check/gst/rtspserver.c:
+ rtsp-stream: Fixed TCP transport case
+ Make sure that the appsink element is actually added to
+ the bin before trying to link it with the elements in it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=776343
+
+2016-12-16 17:26:04 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * .gitignore:
+ * Makefile.am:
+ * configure.ac:
+ * gst-rtsp.spec.in:
+ Remove generated .spec file
+ Likely extremely bitrotten, and we should not ship this anyway.
+
+2016-12-03 08:21:02 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * common:
+ Automatic update of common submodule
+ From f980fd9 to 39ac2f5
+
+2016-12-02 15:40:09 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: Fix pt map caps
+ Since decryption is handled within rtpbin, all outcoming stream
+ caps will be application/x-rtp (i.e. regular rtp)
+ Fixes RECORD with SRTP streams
+
+2016-12-02 15:38:04 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ media-factory: Create media objects with the proper transport mode
+ The function called immediately afterwards (collect_streams()) will
+ need it to work properly
+
+2016-12-02 14:36:50 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ rtsp-auth: Don't remove digest-auth nonces that already/still have a client connected
+
+2016-12-01 18:04:34 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ rtsp-media-factory: Don't create a pipeline for the media pipeline string
+ We're going to put a pipeline into a pipeline otherwise, which is not
+ exactly ideal.
+
+2016-10-25 15:41:28 +0300 Kseniia Vasilchuk <vasilchukkseniia@gmail.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: Fix race condition around finish_unprepare() if called multiple time
+ https://bugzilla.gnome.org/show_bug.cgi?id=755329
+
+2016-11-30 14:06:36 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: Don't leave stale pointer after unref
+ Fix a warning on shutdown - don't keep a pointer to an
+ alread-unreffed object.
+
+2016-11-26 11:24:50 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * .gitmodules:
+ common: use https protocol for common submodule
+ https://bugzilla.gnome.org/show_bug.cgi?id=775110
+
+2016-11-21 23:29:56 +1100 Matthew Waters <matthew@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: block the output of rtpbin instead of the source pipeline
+ 85c52e194bcb81928b96614be0ae47d59eccb1ce introduced a more correct
+ detection of the srtp rollover counter to add to the SDP.
+ Unfortunately, it was incomplete for live pipelines where the logic
+ blocks the source bin before creating the SDP and thus would never have
+ the necessary informaiton to create a correct SDP with srtp encryption.
+ Move the pad blocks to rtpbin's output pads instead so that the
+ necessary information can be created before we need the information for
+ the SDP.
+ https://bugzilla.gnome.org/show_bug.cgi?id=770239
+
+2016-11-21 16:02:39 +0100 Dag Gullberg <dagg@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: add IDLE timeout, before session exists
+ The RTSP server will not timeout an idle RTSP connection
+ (note this is different from doing timeout on a RTSP
+ session).
+ At least for Apache this is a problem when running RTSP over
+ HTTPS since it uses one of the threads (there is a rather
+ limited number) that are available for handling requests.
+ https://bugzilla.gnome.org/show_bug.cgi?id=771830
+
+2016-11-23 09:45:08 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * .gitignore:
+ .gitignore more
+
+2016-11-21 13:05:50 +0100 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Set close-socket FALSE on UDP src:es
+ With this RTSP server can use the sockets independent on the udpsrc
+ state.
+ When the udp src is finalized it will unref socket and when g_socket
+ is finalized the socket will be closed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=765673
+
+2016-11-18 17:47:13 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: Move to new helper function to parse authentication responses
+ https://bugzilla.gnome.org/show_bug.cgi?id=774416
+
+2016-11-16 08:42:24 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/Makefile.am:
+ * examples/test-auth-digest.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * win32/common/libgstrtspserver.def:
+ rtsp-auth: Add support for Digest authentication
+ https://bugzilla.gnome.org/show_bug.cgi?id=774416
+
+2016-11-17 09:41:53 -0800 Scott D Phillips <scott.d.phillips@intel.com>
+
+ * Makefile.am:
+ * gst/rtsp-server/meson.build:
+ * meson.build:
+ * tests/check/meson.build:
+ * win32/MANIFEST:
+ * win32/common/libgstrtspserver.def:
+ Enable building with MSVC
+ https://bugzilla.gnome.org/show_bug.cgi?id=774640
+
+2016-11-18 20:23:14 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * meson.build:
+ meson: gstreamer gst_check_dep does not exist on windows
+
+2016-11-17 09:43:37 -0800 Scott D Phillips <scott.d.phillips@intel.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: update do_send_message to match type GstRTSPClientSendFunc
+ This type mismatch fails building with MSVC
+ https://bugzilla.gnome.org/show_bug.cgi?id=774640
+
+2016-11-11 14:42:08 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ rtsp-sdp: Fix indentation
+
+2016-11-10 05:16:00 +0000 Neha Arora <arora.neha@samsung.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Only signal "new-state" if the state has actually changed
+ https://bugzilla.gnome.org/show_bug.cgi?id=774173
+
+2016-08-24 11:39:13 +0200 Branko Subasic <branko@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: emit signal in the beginning of each rtsp request
+ These signals let the application validate the requests, configure the
+ media/stream in a certain way and also generate error status code in
+ case of error or bad request.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758062
+
+2016-11-01 18:10:35 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ meson: update version
+
+=== release 1.11.0 ===
+
+2016-11-01 18:53:15 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.10.0 ===
+
+2016-11-01 18:06:46 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.10.0
+
+2016-10-28 18:38:01 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/gst/rtspserver.c:
+ * tests/check/gst/stream.c:
+ tests: try to avoid using the same ports in different tests
+ Causes problems with client multicast tests otherwise if
+ tests are run in parallel.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773640
+
+2016-10-28 17:50:59 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/gst/client.c:
+ tests: client: use fail_unless_equals_foo() for better failure reporting
+
+2016-09-26 11:16:04 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Session filter in unwatch session
+ Call session filter with filter_session_media as paramer in
+ client_unwatch_session if using drop_backlog = FALSE.
+ In client_unwatch_session its allowed to grow the watchs backlog.
+ If using drop_backlog = FALSE and the backlog is full it will cause
+ a deadlock when setting session media state to NULL
+ if the backlog is not allowed to grow.
+ https://bugzilla.gnome.org/show_bug.cgi?id=771983
+
+2016-10-20 21:40:18 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ meson: add fallbacks for gst modules
+ For gst-all.
+
+2016-09-14 17:48:39 +0300 Nikita Bobkov <NikitaDBobkov@gmail.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Fix factory leaking in find_media() in error cases
+ https://bugzilla.gnome.org/show_bug.cgi?id=771488
+
+2016-10-06 11:47:50 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: Fix randomly missing streams from SDP with dynamic elements
+ When using dynamic elements, gst_rtsp_stream_join_bin() is called from
+ "pad-added" signal. In that case priv->srcpad could already have its caps,
+ and they'll be sent to priv->send_src[0] pad. That means that when it
+ connects "notify::caps" signal, that pad could already have received its
+ caps and the signal won't be emitted anymore.
+ In that case priv->caps stay to NULL and when building the SDP that stream
+ gets ignored. Leading to missing video or audio when playing in client side.
+ https://bugzilla.gnome.org/show_bug.cgi?id=772478
+
+2016-09-30 11:42:08 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ meson: update version
+
=== release 1.9.90 ===
-2016-09-30 Sebastian Dröge <slomo@coaxion.net>
+2016-09-30 13:04:12 +0300 Sebastian Dröge <sebastian@centricular.com>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
* configure.ac:
- releasing 1.9.90
+ * gst-rtsp-server.doap:
+ Release 1.9.90
2016-09-17 13:17:19 +0100 Ian Jamison <ian.dev@arkver.com>
projects / platform / upstream / gstreamer.git / blobdiff