+=== 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 13:04:12 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.9.90
+
+2016-09-17 13:17:19 +0100 Ian Jamison <ian.dev@arkver.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-server: Hint that set_multicast_iface expects the name of the interface
+ To prevent any possibly confusion with IPs or anything else.
+ https://bugzilla.gnome.org/show_bug.cgi?id=771530
+
+2016-09-18 09:58:55 -0400 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Call g_free() instead of g_object_unref() on multicast-iface strings
+ https://bugzilla.gnome.org/show_bug.cgi?id=763000#c5
+
+2016-09-14 11:31:15 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ configure: Depend on gstreamer 1.9.2.1
+
+2016-09-10 20:52:31 +1000 Jan Schmidt <jan@centricular.com>
+
+ * autogen.sh:
+ * common:
+ Automatic update of common submodule
+ From b18d820 to f980fd9
+
+2016-09-10 09:58:31 +1000 Jan Schmidt <jan@centricular.com>
+
+ * autogen.sh:
+ * common:
+ Automatic update of common submodule
+ From 6f2d209 to b18d820
+
+2016-09-07 18:44:34 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Remove unused _locked() variant of a function
+ It was added during refactoring.
+
+2016-09-07 10:21:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: cosmetic cleanup
+ https://bugzilla.gnome.org/show_bug.cgi?id=766612
+
+2016-09-07 10:16:19 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: Compare IP addresses case insensitive in more places
+ https://bugzilla.gnome.org/show_bug.cgi?id=766612
+
+2016-09-07 10:12:18 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * common:
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: Fix leaked joined_bin
+ There is no need to keep a strong ref on it, and _leave_bin() was
+ setting it to NULL before calling g_clear_object() so it was leaked.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766612
+
+2016-09-06 19:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Compare IP address strings case insensitive
+ Otherwise IPv6 addresses might fail this comparision.
+
+2016-09-06 19:10:21 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Bind multicast sockets to ANY as before
+ https://bugzilla.gnome.org/show_bug.cgi?id=766612#c48
+
+2016-09-05 18:31:36 +0300 Kseniia <vasilchukkseniia@gmail.com>
+
+ * gst/rtsp-server/rtsp-session.c:
+ rtsp-session: Fix segfault when doing keep-alive after removing the session
+ If keep-alive happens after removing the session but before finalizing the
+ stream transport, we would segfault.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750544
+
+2016-09-05 18:04:50 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Always create multicast UDP elements if the protocol flag is set
+ Adding them later will cause deadlocks due to
+ 1) pre-rolling and staying in PAUSED with the unicast/TCP sinks
+ 2) adding the multicast sink
+ 3) waiting for it to get data to preroll again
+ 3) never happens because the queues after the tee are full.
+
+2016-09-05 16:32:57 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Fix up various multicast related issues
+
+2016-09-05 13:40:59 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/gst/stream.c:
+ tests: Fix compilation
+
+2016-07-28 15:33:05 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * tests/check/gst/stream.c:
+ stream: revert back to create udpsrc/udpsink on DESCRIBE for unicast
+ This is basically reverting changes introduced in commit f62a9a7,
+ because it was introducing various regressions:
+ - It introduces a leak of udpsrc elements that got wrongly fixed by adding
+ an hash table in commit cba045e. We should have at most 4 udpsrc for unicast:
+ ipv4/ipv6, rtp/rtcp. They can be reused for all unicast clients.
+ - If a mcast client connects, it creates a new socket in SETUP to try to respect
+ the destination/port given by the client in the transport, and overrides the
+ socket already set on the udpsink element. That means that if we already had a
+ client connected, the source address on the udp packets it receives suddenly
+ changes.
+ - If a 2nd mcast client connects, the destination/port in its transport is
+ ignored but its transport wasn't updated.
+ What this patch does:
+ - Revert back to create udpsrc/udpsink for unicast clients on DESCRIBE.
+ - Always have a tee+queue when udp is enabled. This could be optimized
+ again in a later patch, but is more complicated. If no unicast clients
+ connects then those elements are useless, this could be also optimized
+ in a later patch.
+ - When mcast transport is added, it creates a new set of udpsrc/udpsink,
+ seperated from those for unicast clients. Since we already support only
+ one mcast address, we also create only one set of elements.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766612
+
+2016-07-28 15:20:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: factor our plug_src function
+ https://bugzilla.gnome.org/show_bug.cgi?id=766612
+
+2016-07-21 21:46:16 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: factor out plug_sink function
+ https://bugzilla.gnome.org/show_bug.cgi?id=766612
+
+2016-07-20 23:05:09 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: small documentation clarification
+ https://bugzilla.gnome.org/show_bug.cgi?id=766612
+
+2016-07-20 15:35:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: rename addr_v4/6 to mcast_addr_v4/6 for clarity
+ https://bugzilla.gnome.org/show_bug.cgi?id=766612
+
+2016-07-14 11:10:31 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: Keep a ref on joined bin
+ https://bugzilla.gnome.org/show_bug.cgi?id=766612
+
+2016-07-20 15:11:32 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: code cleanup
+ https://bugzilla.gnome.org/show_bug.cgi?id=766612
+
+2016-07-20 23:18:23 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: small fix in error code path
+ https://bugzilla.gnome.org/show_bug.cgi?id=766612
+
+2016-07-20 20:09:57 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ Revert "rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc"
+ This partly reverts commit cba045e1b19fad6e689e10206f57903e15f1229a,
+ but keeps unit tests.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766612
+
+2016-09-01 12:33:00 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.9.2 ===
+
+2016-09-01 12:32:51 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.9.2
+
+2016-01-27 01:03:52 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * config.h.meson:
+ * examples/meson.build:
+ * gst/meson.build:
+ * gst/rtsp-server/meson.build:
+ * gst/rtsp-sink/meson.build:
+ * meson.build:
+ * pkgconfig/meson.build:
+ * tests/check/meson.build:
+ * tests/meson.build:
+ Add support for Meson as alternative/parallel build system
+ https://github.com/mesonbuild/meson
+
+2016-08-26 21:56:13 +0200 Josep Torra <n770galaxy@gmail.com>
+
+ * configure.ac:
+ * tests/check/Makefile.am:
+ build: silence error about pthread for 'make check' in osx
+ Fixes "clang: error: argument unused during compilation: '-pthread'"
+
+2015-09-25 15:04:00 +0000 Nikita Bobkov <NikitaDBobkov@gmail.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Fix leaking of media in error cases
+ With additional fixes by Kseniya Vasilchuk <vasilchukkseniia@gmail.com>
+ and myself to make the media refcounting a bit easier to follow.
+ https://bugzilla.gnome.org/show_bug.cgi?id=755632
+
+2016-08-02 15:08:22 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Fix leaking of session in error cases
+ https://bugzilla.gnome.org/show_bug.cgi?id=755632
+
+2016-07-11 21:16:04 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From f363b32 to f49c55e
+
+2016-07-06 13:51:15 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.9.1 ===
+
+2016-07-06 13:28:12 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.9.1
+
+2016-06-24 02:02:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * configure.ac:
+ configure: Need to add -DGST_STATIC_COMPILATION when building only statically
+ https://bugzilla.gnome.org/show_bug.cgi?id=767463
+
+2016-06-21 11:49:02 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * common:
+ Automatic update of common submodule
+ From ac2f647 to f363b32
+
+2016-04-14 22:56:11 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-sdp.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ sdp: add rollover counters for all sender SSRC
+ We add different crypto sessions in MIKEY, one for each sender
+ SSRC. Currently, all of them will have the same security policy, 0.
+ The rollover counters are obtained from the srtpenc element using the
+ "stats" property.
+ https://bugzilla.gnome.org/show_bug.cgi?id=730539
+
+2016-06-07 20:44:42 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-server.h:
+ docs: fix some typos
+
+2016-05-25 10:28:43 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/Makefile.am:
+ g-i: pass compiler env to g-ir-scanner
+ It's what introspection.mak does as well. Should
+ fix spurious build failures on gnome-continuous
+ (caused by g-ir-scanner getting compiler details
+ via python which is broken in some environments
+ so passing the compiler details bypasses that).
+
+2016-05-18 16:48:44 +0100 Ian <ian.arkver.dev@gmail.com>
+
+ * gst/rtsp-server/rtsp-session.c:
+ rtsp-session: RFC2326 does not allow a space between ; and timeout in the Session header
+ This works with rtspsrc and live555, but fails with e.g. ffmpeg.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766619
+
+2016-03-07 14:48:38 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: Check return value of sscanf
+ And just make sure we always have 0/0 if we have an error
+ CID #1352031
+
+2016-04-25 08:55:25 -0400 Jake Foytik <jake.foytik@ipconfigure.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * tests/check/gst/rtspserver.c:
+ * tests/check/gst/stream.c:
+ rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc
+ - Unicast udpsrcs are now managed in a hash table. This allows for proper cleanup in with shared streams and fixes a memory leak.
+ - Unicast udpsrcs are now properly cleaned up when shared connections exit. See the update_transport() function.
+ - Create unit test for shared media.
+ https://bugzilla.gnome.org/show_bug.cgi?id=764744
+
+2016-04-11 10:55:23 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Always bind to ANY when address is a multicast address and not only on Windows
+ For IPv6 addresses, binding to a multicast group does not work on Linux
+ either. Always bind to ANY and then later join the multicast group.
+ https://bugzilla.gnome.org/show_bug.cgi?id=764679
+
+2016-04-14 10:05:02 +0100 Julien Isorce <j.isorce@samsung.com>
+
+ * common:
+ Automatic update of common submodule
+ From 6f2d209 to ac2f647
+
+2016-04-06 10:09:46 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ rtsp-thread-pool: explained why GSource is a part of ThreadImpl
+ Clarified why it is necessary to add source information to
+ GstRTSPThreadImpl. See the reported bug in GLib:
+ https://bugzilla.gnome.org/show_bug.cgi?id=720186
+ for more information.
+ https://bugzilla.gnome.org/show_bug.cgi?id=761702
+
+2016-04-04 12:58:38 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/Makefile.am:
+ examples: Clean up CFLAGS/LDADD even more
+ The internal .la should come first and is part of LDADD, as is
+ GST_CFLAGS/LIBS.
+
+2016-04-04 12:39:39 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/Makefile.am:
+ examples: Clean up CFLAGS/LDADD to link with the correct versions of all libraries
+
+2016-04-03 12:06:29 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/Makefile.am:
+ rtsp-server: Use $(GST_NET_LIBS) / $(GST_NET_CFLAGS)
+
+2015-12-30 18:39:05 +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:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-server: Implement clock signalling according to RFC7273
+ For NTP and PTP clocks we signal the actual clock that is used and signal
+ the direct media clock offset.
+ For all other clocks we at least signal that it's the local sender clock.
+ This allows receivers to know which clock was used to generate the media and
+ its RTP timestamps. Receivers can then implement network synchronization,
+ either absolute or at least relative by getting the sender clock rate directly
+ via NTP/PTP instead of estimating it from RTP timestamps and packet receive
+ times.
+ https://bugzilla.gnome.org/show_bug.cgi?id=760005
+
+2016-03-02 19:42:58 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: Add support for setting the multicast interface
+ https://bugzilla.gnome.org/show_bug.cgi?id=763000
+
+2016-03-02 19:42:13 +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:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-media: Add support for setting the multicast interface
+ https://bugzilla.gnome.org/show_bug.cgi?id=763000
+
+2016-03-07 08:50:01 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: use new gst_element_class_add_static_pad_template()
+ https://bugzilla.gnome.org/show_bug.cgi?id=763196
+
+2016-03-24 13:33:43 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.8.0 ===
+
+2016-03-24 13:00:35 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.8.0
+
+2016-03-16 23:35:09 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Don't set the state of the appsrc from PLAYING to PAUSED again during setup
+ This would get us NO_PREROLL in the bin again and break seeking.
+ Thanks to Carlos Rafael Giani for helping to debug this!
+ https://bugzilla.gnome.org/show_bug.cgi?id=740509
+
+=== release 1.7.91 ===
+
+2016-03-15 12:26:13 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.7.91
+
+2016-03-10 13:54:38 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Ensure that the pipeline is live and later-added udpsrcs are syncing the state with the parent bin
+ Without this, RECORD pipelines are broken because
+ a) we wait for ASYNC_DONE which never happens anymore because udpsrc would be
+ added later. Previously it was there earlier and due to NO_PREROLL caused the
+ pipeline to preroll immediately
+ b) the udpsrc for the pipeline is added later and never set to PLAYING state,
+ as the corresponding code previously was only for PLAY pipelines.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763281
+
+2016-03-11 01:22:54 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Fix typo in the docstring
+ gst_rtsp_stream_set_client_side -> gst_rtsp_stream_is_client_side
+
+2016-03-05 10:52:11 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Disable multicast loopback for all our sockets
+ On Windows this is a receiver-side setting, on Linux a sender-side setting. As
+ we provide a socket ourselves to udpsrc, udpsrc is never setting the multicast
+ loopback setting on the socket... while udpsink does which unfortunately has
+ no effect here on Windows but on Linux.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757488
+
+2016-03-03 15:07:06 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * tests/check/gst/stream.c:
+ stream tests: added new tests
+ Test a case when the address pool only contains multicast addresses
+ and the client is requesting unicast udp.
+ Added tests for multicast ports allocation.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757488
+
+2016-03-04 13:51:12 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Only bind multicast sockets to ANY on Windows
+ On Linux it is still needed to bind to the multicast address
+ to filter out random other packets, while on Windows binding
+ to multicast addresses just fails.
+
+2016-03-03 10:41:51 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Only use the address pool for unicast UDP if it contains unicast addresses
+ Otherwise we fail to allocate UDP ports if the pool only contains multicast
+ addresses, which is something that used to work before. For unicast addresses
+ if the pool contains none, we just allocate them as if there is no pool at
+ all.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757488
+
+2016-03-02 11:48:49 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-server: Fix indentation
+
+2016-03-02 11:47:47 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Don't bind the sockets to multicast addresses
+ This works on Linux but fails completely on Windows. You're supposed
+ to bind to ANY and then join the multicast group.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757488
+
+=== release 1.7.90 ===
+
+2016-03-01 19:00:45 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 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 12:03:18 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 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 14:54:06 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.7.1
+
+2015-12-21 00:43:49 +0100 Koop Mast <kwm@rainbow-runner.nl>
+
+ * configure.ac:
+ configure: Make -Bsymbolic check work with clang.
+ Update the -Bsymbolic check with the version glib has. This version
+ works with clang.
+ https://bugzilla.gnome.org/show_bug.cgi?id=759713
+
+2015-11-17 22:30:54 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ rtsp-session-pool: Avoid dollar sign ($) in session ids
+ Live555 in VLC strips off dollar signs and then gets very confused,
+ we don't loose too much entropy by just skipping it.
+
+2015-11-10 14:17:18 -0500 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/rtsp-server/rtsp-address-pool.h:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media-factory-uri.h:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-mount-points.h:
+ * gst/rtsp-server/rtsp-permissions.h:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-media.h:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.h:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.h:
+ * gst/rtsp-server/rtsp-thread-pool.h:
+ * gst/rtsp-server/rtsp-token.h:
+ rtsp-server: Add g_autoptr() support to all types
+ https://bugzilla.gnome.org/show_bug.cgi?id=754464
+
+2015-12-08 08:27:20 +0100 Srimanta Panda <srimanta@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: fixed valgrind error
+ Fixed the valgrind error in unit test. The UDP source created during
+ gst_rtsp_stream_join_bin() was not released while destroying the rtp
+ bin.
+ https://bugzilla.gnome.org/show_bug.cgi?id=759010
+
+2015-12-07 09:11:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * autogen.sh:
+ * common:
+ Automatic update of common submodule
+ From b319909 to 86e4663
+
+2015-11-18 11:14:39 +0100 Srimanta Panda <srimanta@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: suspend media during setup request
+ SETUP request from clients needs to suspend the media to clear the
+ prerolled buffers. Otherwise it will not affect the prerolled buffer
+ and the prerolled buffers will be incorrect (for example block-size
+ from setup request will not affect the prerolled buffer unless the
+ media is suspended).
+ https://bugzilla.gnome.org/show_bug.cgi?id=758268
+
+2015-12-04 08:01:37 +0100 Srimanta Panda <srimanta@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: create stream pipeline based on transport
+ Based on the protocol, create the rtsp stream pipeline. If only TCP or
+ only UDP is set as the transport protocol, it will not add the extra tee
+ or queue element to the pipeline. Both these elements will be added, if
+ it supports both TCP and UDP protocols. This improves the pipeline
+ performance when one protocol is present.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758179
+
+2015-11-19 15:01:16 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Only create RTP sending/receiving rtpbin pads if needed
+ Adding them when not needed will start some logic inside rtpbin that might be
+ problematic. Also if e.g. for a sender media we suddenly receive RTP data, we
+ would start up a rtpjitterbuffer and behave in weird ways.
+ We still set up the UDP sources for RTP receiving for a sender media to be
+ able to receive any packets sent by the client for NAT traversal. They will
+ all go to a fakesink though.
+ Having an rtpjitterbuffer in the media pipeline will cause the pipeline to be
+ NO_PREROLL, which will cause deadlocks when seeking the media as it will never
+ receive ASYNC_DONE after a seek.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758319
+
+2015-11-17 12:44:38 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Disable multicast loopback for the multicast udp sources too
+ On POSIX this setting is for sender sockets, on Windows for receiver sockets.
+ Previously we were only setting this for sender sockets, which caused looped
+ back packets to be received on Windows if a multicast transport was used.
+
+2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
+
+ * examples/test-record-auth.c:
+ * examples/test-record.c:
+ examples: Actually use the provided port in the record examples
+
+2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
+
+ * examples/test-record-auth.c:
+ test-record-auth: Add the option to build in TLS support
+
+2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
+
+ * examples/test-auth.c:
+ test-auth: Use an 'anonymous' user for unauthenticated default
+ There's a comment on one of the resources that 'user' and 'admin'
+ shouldn't even be able to see it, but they can if the default
+ token is 'admin2', since that gives them access anyway.
+
+2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
+
+ * examples/.gitignore:
+ * examples/Makefile.am:
+ * examples/test-record-auth.c:
+ Add test-record-auth example
+
+2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * tests/check/gst/client.c:
+ rtsp-client: Report RECORD and ANNOUNCE as supported in the OPTIONS
+
+2015-11-11 14:58:33 +0100 Marcus Prebble <prebble@axis.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ rtsp-server: Change the logic so we don't pop a NULL context
+ When doing a port scan (e.g. with nmap) the call to GST_RTSP_CHECK()
+ will sometimes fail. This call is made before any context is pushed
+ resulting in an attempt to pop a NULL context.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757949
+
+2015-10-22 14:32:30 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * tests/check/gst/rtspserver.c:
+ rtspserver: Add udp-mcast transport SETUP test
+ Refactor utility functions in the test file so they can handle
+ more than UDP and TCP as lower transport.
+ https://bugzilla.gnome.org/show_bug.cgi?id=756969
+
+2015-10-22 09:15:21 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Always unref return value of gst_object_get_parent()
+ Fixes a leak of a GstBin in the udp-mcast case.
+ https://bugzilla.gnome.org/show_bug.cgi?id=756968
+
+2015-10-21 14:37:19 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From b99800a to b319909
+
+2015-10-20 17:29:42 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Use new GST_ENABLE_EXTRA_CHECKS #define
+ https://bugzilla.gnome.org/show_bug.cgi?id=756870
+
+2015-10-21 14:28:47 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From 6babecd to b99800a
+
+2015-10-02 22:25:47 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Update GLib dependency to 2.40.0
+
+2015-10-02 16:11:05 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * examples/test-mp4.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: listen to sender ssrc signals
+ https://bugzilla.gnome.org/show_bug.cgi?id=746747
+
+2015-09-29 13:00:51 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * common:
+ common: update for new suppression
+ Makes check-valgrind pass with glib 2.46
+
+2015-09-28 17:40:59 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Take reference to media that will be prepared
+ default_prepare() takes a transfer-none reference GstRTSPMedia object.
+ Later on a g_idle_source_new() is created and a pointer to the media
+ object is passed as user data. If the media is freed before the idle
+ source is dispatched the media object pointer is invalid, but the idle
+ source callback expects it to still be valid. To fix this a reference to
+ the media object is taken when registering the source callback function
+ and a corresponding release of the reference is done when the souce is
+ destroyed.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755748
+
+2015-08-20 17:01:24 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * examples/test-launch.c:
+ * examples/test-mp4.c:
+ * examples/test-ogg.c:
+ * examples/test-record.c:
+ * examples/test-uri.c:
+ rtsp-server: Fix memory leaks when context parse fails
+ When g_option_context_parse fails, context and error variables are not getting free'd
+ which results in memory leaks. Free'ing the same.
+ And replacing g_error_free with g_clear_error, which checks if the error being passed
+ is not NULL and sets the variable to NULL on free'ing.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753863
+
+2015-09-25 23:51:17 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.6.0 ===
+
+2015-09-25 23:32:52 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.6.0
+
+=== release 1.5.91 ===
+
+2015-09-18 20:12:06 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.5.91
+
+2015-09-17 20:07:34 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: fix docs for recently-added get/set_buffer_size API
+ https://bugzilla.gnome.org/show_bug.cgi?id=749095
+
+2015-09-04 11:23:43 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Don't crash on encrypted RTX SDP
+ In parse_keymgmt(), don't mutate the input string that's been passed
+ as const, especially since we might need the original value again if
+ the same key info applies to multiple streams (RTX, for example).
+ https://bugzilla.gnome.org/show_bug.cgi?id=754753
+
+2015-08-22 20:59:40 +1000 Jan Schmidt <jan@centricular.com>
+
+ * examples/test-mp4.c:
+ test-mp4: Support filenames with spaces in them. Error out on too few arguments
+
+2015-08-17 02:36:31 +1000 Jan Schmidt <jan@centricular.com>
+
+ * examples/test-record.c:
+ test-record: Check parameter count and print out help
+ If no launch pipeline was supplied, print out some help
+
+2015-08-31 22:48:34 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-stream: Implement UDP buffer size setting.
+ Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
+ UDP TX buffer size.
+ Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
+
+2015-08-31 22:47:45 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.h:
+ rtsp-media: Fix small typo causing gtk-doc to complain
+
+=== release 1.5.90 ===
+
+2015-08-19 14:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.5.90
+
+2015-08-12 14:33:44 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ media-factory: get port number through gst_rtsp_url_get_port
+ https://bugzilla.gnome.org/show_bug.cgi?id=753473
+
+2015-08-13 11:24:10 +0200 Francisco Velazquez <francisv@ifi.uio.no>
+
+ * tests/check/gst/media.c:
+ media-test: Removing unnecessary assertion
+ https://bugzilla.gnome.org/show_bug.cgi?id=753385
+
+2015-07-23 14:50:30 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ Document that source keeps a ref on server until it's destroyed
+ https://bugzilla.gnome.org/show_bug.cgi?id=749227
+
+2015-08-08 11:09:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * tests/check/gst/media.c:
+ media-test: Test for multiple dynamic payload
+ https://bugzilla.gnome.org/show_bug.cgi?id=753385
+
+2015-08-08 09:40:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: Only add fakesink once per pipeline
+ The intention is to prevent going PLAYING state before pads are created.
+ If there was mutilple dynamic payload, it would leak few fakesink and
+ actually prevent from ever reaching playing state.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753385
+
+2015-08-08 09:08:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ Revert "rtsp-media: Only add 1 fakesink per pipeline"
+ This reverts commit 22bf61f16c1210bb458fc3f53642179a0211104f.
+
+2015-08-07 09:21:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Only add 1 fakesink per pipeline
+ There should be only one fakesink per pipeline, not per dynpay. This
+ would lead to element naming clash.
+
+2015-07-30 15:32:43 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: assertion error due to wrong condition check
+ In media to caps function, reserved_keys array is being used for variable i,
+ leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
+ changed it to variable j
+ https://bugzilla.gnome.org/show_bug.cgi?id=753009
+
+2015-07-29 11:27:05 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Strip keys from the fmtp that we use internally in our caps
+ Skip keys from the fmtp, which we already use ourselves for the
+ caps. Some software is adding random things like clock-rate into
+ the fmtp, and we would otherwise here set a string-typed clock-rate
+ in the caps... and thus fail to create valid RTP caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=753009
+
+2015-07-20 16:37:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
+ https://bugzilla.gnome.org/show_bug.cgi?id=752640
+
+2015-07-03 22:00:00 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From f74b2df to 9aed1d7
+
+2015-06-25 00:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.5.2 ===
+
+2015-06-24 23:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.5.2
+
+2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * tests/check/gst/client.c:
+ rtsp-client: allow application to decide what requirements are supported
+ Add "check-requirements" signal and vfunc to allow application
+ (and subclasses) to check the requirements.
+ Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
+ https://bugzilla.gnome.org/show_bug.cgi?id=749417
+
+2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 6015d26 to f74b2df
+
+2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Always use real payloader when creating streams
+ A bin that contains the real payloader might be used as payloader. In this
+ case we have to get the real payloader for the various properties it provides.
+ Example use cases for this are bins that payload some media and then have
+ additional elements that add metadata or RTP extension headers to the stream.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750800
+
+2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-netclock-client.c:
+ test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
+
+2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-netclock-client.c:
+ * examples/test-netclock.c:
+ test-netclock: Use new ntp-time-source property on rtpbin
+ Select the clock time to be used as NTP time source. This allows proper
+ synchronization between receivers, independent of sharing base times, and just
+ requires them to use the same clock.
+
+2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-netclock-client.c:
+ * examples/test-netclock.c:
+ test-netclock: Setting the same base time on sender and receiver is not necessary
+ It's going to be fixed up by rtpbin when using ntp-sync=TRUE
+
+2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
+ https://bugzilla.gnome.org/show_bug.cgi?id=750764
+
+2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * docs/libs/gst-rtsp-server.types:
+ docs: add missing types
+ https://bugzilla.gnome.org/show_bug.cgi?id=750764
+
+2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ docs: add missing apis
+ https://bugzilla.gnome.org/show_bug.cgi?id=750764
+
+2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-netclock-client.c:
+ test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
+
+2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ GstRTSPAuth: Add client certificate authentication support
+ https://bugzilla.gnome.org/show_bug.cgi?id=750471
+
+2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-netclock-client.c:
+ test-netclock-client: Use new GstClock API to wait for clock synchronization
+
+2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-netclock-client.c:
+ test-netclock-client: Use a GMainLoop and playbin's source-setup signal
+ A mainloop is needed to get glimagesink to display something on OSX, and
+ the source-setup signal just makes things a little bit easier.
+
+2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * common:
+ Automatic update of common submodule
+ From d9a3353 to 6015d26
+
+2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From d37af32 to d9a3353
+
+2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 21ba2e5 to d37af32
+
+2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From c408583 to 21ba2e5
+
+2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * docs/libs/Makefile.am:
+ docs: remove variables that we define in the snippet from common
+ This is syncing our Makefile.am with upstream gtkdoc.
+
+2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 44a3517 to c408583
+
+2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.5.1 ===
+
+2015-06-07 11:20:01 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.5.1
+
+2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: No flush during Teardown.
+ When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
+ backlog is empty it can happen that just a part of a message will be
+ sent and rest is in backlog queue. If then flush during teardown
+ just a part of message will be sent.This can lead to client miss
+ teardown response since it expect to get the last part of message.
+ The flushing during teardown was introduced to fix a deadlock that now
+ is fixed more generally in handle_request by temporary setting backlog
+ size to unlimited.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
+
+2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/Makefile.am:
+ tests: Use AM_TESTS_ENVIRONMENT
+ Needed by the new automake test runner and the
+ current version of the common submodule.
+
+2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-server: Use single-include rtsp header to make sure we get all definitions
+
+2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Mark some more functions static
+
+2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Only unblock the media in suspend() when actually changing the state
+ Otherwise we're going to lose a few packets for live streams during DESCRIBE.
+
+2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-video-rtx.c:
+ examples: Use AVPF profile for the RTX example
+
+2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ rtsp-sdp: Only add RTX to the SDP when using a feedback profile
+
+2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: get valid clock-rate from last-sample
+ clock-rate in last-sample's caps is integer, not unsigned.
+ To get this value properly, variable needs to be type-casted to int.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747614
+
+2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * autogen.sh:
+ * common:
+ autogen.sh: only run autopoint if gettext requested in configure.ac
+ Not just because there happens to be a po directory.
+ https://bugzilla.gnome.org/show_bug.cgi?id=748058
+
+2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ Revert "configure.ac: uncomment gettext version setup"
+ This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
+ We don't need a gettext setup here and there's no po
+ directory either, so no reason why autopoint would be
+ run in the first place.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=748058
+
+2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
+
+ * examples/test-multicast.c:
+ * examples/test-multicast2.c:
+ * examples/test-sdp.c:
+ * examples/test-video-rtx.c:
+ * examples/test-video.c:
+ * tests/test-cleanup.c:
+ * tests/test-reuse.c:
+ Fix timeout function signatures across tests and examples
+
+2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/Makefile.am:
+ tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
+ Make sure the test environment is set up.
+ https://bugzilla.gnome.org//show_bug.cgi?id=747624
+
+2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ configure: bump automake requirement to 1.14 and autoconf to 2.69
+ This is only required for builds from git, people can still
+ build tarballs if they only have older autotools.
+ https://bugzilla.gnome.org//show_bug.cgi?id=747624
+
+2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * configure.ac:
+ configure.ac: uncomment gettext version setup
+ Fixes autogen.sh. It would run autopoint, which would complain
+ that it could not find the gettext version in configure.ac.
+ https://bugzilla.gnome.org/show_bug.cgi?id=748058
+
+2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * examples/test-video-rtx.c:
+ test-video-rtx: set exact payload type to PCMA payloader
+ Setting wrong payload type causes failure to do retransmission through audio stream
+ https://bugzilla.gnome.org/show_bug.cgi?id=747839
+
+2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-stream: fix to get valid each stream data for request-aux-sender signal
+ Because of duplicated g_signal_connect for request-aux-sender signal,
+ wrong stream pointer is passed to the signal handler.
+ Instead of passing each stream, pass stream array and get the relevant stream.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747839
+
+2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * acinclude.m4:
+ * autogen.sh:
+ Update autogen.sh to latest version from common
+ Fixes build after aclocal_check etc. helpers have been removed.
+
+2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From bc76a8b to c8fb372
+
+2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Limit the queues to 1 buffer
+ We only need them to be able to pre-roll, queueing up more data here
+ is only going to harm latency and memory usage.
+
+2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Update comment and ASCII art to the latest code
+ We have a queue in front of the udpsink too to prevent the pipeline from
+ locking up.
+
+2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-media: Properly return first rtptime
+ Instead we where returning first GstBuffer timestamp. This would result
+ in clock skew and unwanted behaviour in RTSP playback.
+ https://bugzilla.gnome.org/show_bug.cgi?id=746479
+
+2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Don't leave buffer mapped
+ If the seq is NULL, the RTP buffer was left mapped. We should always
+ unmap the buffer.
+
+2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
+
+ * README:
+ Fix typo in README
+
+2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * tests/check/gst/client.c:
+ Fix double semicolons
+
+2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
+ This gives more accurate values than asking the payloader. There might be
+ queueing happening between the payloader and the sink.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745704
+
+2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Don't seek for PLAY if the position will not change
+ https://bugzilla.gnome.org/show_bug.cgi?id=745704
+
+2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Don't include payload type in the caps for framesize
+ When the sdp media attribute framesize are converted to caps
+ the <payload> should not be included.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
+ Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
+
+2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ rtsp-sdp: add payload type to the sdp framesize attribute
+ The sdp framesize attribute is desribed in RFC6064. It is specified
+ for payloading of H263 and has the following form
+ a=framesize:<payload type> <width>-<height>. The <width>-<height> part
+ should be added to the caps in a payloader and the <payload type> should
+ be added by the rtsp-server.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
+
+2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * examples/test-uri.c:
+ examples: test-uri: fix tainted variable
+ Insignificant but this keeps Coverity happy.
+ CID #1268404
+
+2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
+
+ * examples/.gitignore:
+ * examples/Makefile.am:
+ * examples/test-netclock-client.c:
+ * examples/test-netclock.c:
+ examples: Add a simple example of network synch for live streams.
+ An example server and client that works for synchronising live streams
+ only - as it can't support pause/play.
+
+2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ rtsp-media-factory: Add functions to set/get the media gtype
+ Allow specifying the GType of a GstRtspMedia subclass to create
+ as a simpler way to get the factory to create a custom
+ GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
+
+2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: fix double unlock in _get_buffer_size()
+ Fixes an abort when calling gst_rtsp_media_get_buffer_size()
+ because of double g_mutex_unlock () usage.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745434
+
+2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ rtsp-session: Use monotonic time for RTSP session timeout
+ Changed RTSP session timeout handling to monotonic time
+ and deprecating the API for current system time.
+ This fixes timeouts when the system time changes.
+ https://bugzilla.gnome.org/show_bug.cgi?id=743346
+
+2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-client: Only error out in PLAY if seeking actually failed
+ If the media was just not seekable, we continue from whatever position we are
+ and let the client decide if that is what is wanted or not.
+ Only if the actual seek failed, we can't really recover and should error out.
+
+2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Add necessary queues between tee and multiudpsink
+ https://bugzilla.gnome.org/show_bug.cgi?id=744379
+
+2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: If seeking fails, don't wait forever for the media to preroll again
+ Instead error out properly the same way as if the SEEKING query already
+ failed.
+
+2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-stream: minor code formatting fix
+
+2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: fix logic for collect_streams
+ Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
+ all streams it knows if it got any, and can check if the transport mode is OK.
+ CID #1268400
+
+2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Don't set the transport mode based on what elements we find
+ Just print a warning if the one that was set before disagrees with what
+ elements we found. It must already be set to something before as this
+ function is called after we received the SDP from ANNOUNCE in RECORD mode,
+ and we would reject ANNOUNCE if the RECORD flag was not set.
+
+2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: rtspserver: rename shadowed variable
+ We have two different 'sink' variables here,
+ rename one of them for clarity.
+
+2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: fix awkward if clause
+
+2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/test-uri.c:
+ examples: test-uri: improve uri argument handling and accept file names
+ Print an error if the argument passed is not a URI and can't
+ be converted into one, or no arguments have been provided.
+
+2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/test-uri.c:
+ examples: test-uri: don't remove mount point after 10 seconds
+ It's very irritating when trying to test stuff repeatedly
+ and serves no real purpose other than showing that it can
+ be done.
+
+2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/.gitignore:
+ examples: add new test-record to .gitignore
+
+2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-record.c:
+ * 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:
+ * tests/check/gst/rtspserver.c:
+ rtsp-media: Use flags to distinguish between PLAY and RECORD media
+
+2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-record.c:
+ test-record: Set latency for playback-style example to 2s instead of 200ms
+
+2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: add some unit tests for ANNOUNCE and RECORD
+ https://bugzilla.gnome.org/show_bug.cgi?id=743175
+
+2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: fix a couple of leaks in handle_announce
+
+2015-01-19 13:20:39 +0100 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: Expose latency setting for setting the rtpbin latency
+
+2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-record.c:
+ test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
+
+2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
+
+2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/Makefile.am:
+ * examples/test-record.c:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * 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:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ Add initial support for RECORD
+ We currently only support media that is RECORD or PLAY only, not both at once.
+ https://bugzilla.gnome.org/show_bug.cgi?id=743175
+
+2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: RTCP and RTP transport cache cookies seperated
+ RTCP packets were not sent because the same tr_cache_cookie was used for
+ both RTP and RTCP. So only one of the tr_cache lists were populated
+ depending on which one was sent first. If the tr_cache list is not
+ populated then no packets can be sent. Most often this happened to be
+ RTCP. Now seperate RTCP and RTP transport cache cookies are added which
+ resulted in both the tr_cache_lists to be populated regardless of which
+ one was sent first.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
+
+2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: fix false compiler warning
+ rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
+
+2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: log interleaved data received
+
+2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
+
+2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
+
+2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Use a random session ID in the SDP
+ RFC4566 Section 5.2 says that it should make the username, session id,
+ nettype, addrtype and unicast address tuple globally unique. Always using
+ 1188340656180883 is not going to guarantee that: https://xkcd.com/221/
+ Instead let's create a 64 bit random number, which at least brings us
+ closer to the goal of global uniqueness.
+ https://tools.ietf.org/html/rfc4566#section-5.2
+
+2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-launch.c:
+ * examples/test-mp4.c:
+ * examples/test-ogg.c:
+ * examples/test-uri.c:
+ examples: Don't call gst_init() and gst_get_option_group()
+ The latter calls the former at the appropriate time.
+
+2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Drop trailing \0 of RTSP DATA messages
+ We add a trailing \0 in GstRTSPConnection to make parsing of
+ string message bodies easier (e.g. the SDP from DESCRIBE) but
+ for actual data this means we have to drop it or otherwise
+ create invalid data.
+
+2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
+ Fixes crash when two threads access handle_new_sample() at the same
+ time, one for RTP, one for RTCP.
+ Otherwise, when iterating over the transports cache, it might be modified by
+ another thread at the same time if the transports cookie has changed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=742954
+
+2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Set format=TIME on our app sources for TCP
+
+2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
+ This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
+ RFC 2326 states that session IDs may consist of alphanumeric as well as
+ the safe characters $-_.+ -- N.B. the percent character is not allowed.
+ Previously the session ID was URI-escaped, this meant that any character
+ which was not alphanumeric or any of the characters +-._~ would be
+ percent encoded. While the RFC (surprisingly) mentions that linear white
+ space in session IDs should be URI-escaped, it does not say anything
+ about other characters. Moreover no white space is allowed in the
+ session ID. Finally the percent character which is the result of
+ URI-escaping is not allowed in a session ID.
+ So there is no reason to do any URI-escaping, and now it is removed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=742869
+
+2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From f2c6b95 to bc76a8b
+
+2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * Makefile.am:
+ Fix 'make check' from top-level directory
+
+2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * examples/test-launch.c:
+ * examples/test-mp4.c:
+ * examples/test-ogg.c:
+ * examples/test-uri.c:
+ examples: Add command-line parsing and take a 'port' argument
+ This allows users to run multiple servers on different ports for testing.
+ Only done for examples that actually take arguments and hence are capable of
+ outputting different streams for each instance on each port.
+ https://bugzilla.gnome.org/show_bug.cgi?id=742115
+
+2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ rtsp-client: Add a send_message default signal handler
+ This allows subclasses to easily hook into the response sending
+ mechanism without doing everything from a signal, which seems
+ awkward from subclasses.
+
+2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From ef1ffdc to f2c6b95
+
+2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * Makefile.am:
+ * configure.ac:
+ configure: add --disable-examples switch
+ https://bugzilla.gnome.org/show_bug.cgi?id=741678
+
+2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
+
+ * examples/.gitignore:
+ * examples/Makefile.am:
+ * examples/test-video-rtx.c:
+ examples: add a retransmisison example implementing RFC4588
+ Currently only SSRC-multiplexed rtx streams are supported
+
+2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Fix some minor memory leaks
+
+2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Some minor cleanup
+
+2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Fix compiler warnings
+ rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+ ^
+ rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+ ^
+
+2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * 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:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ media: implement ssrc-multiplexed retransmission support
+ based off RFC 4588 and the server-rtpaux example in -good
+
+2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp: Ref transports in hash table.
+ Also ref streams for transports.
+ This solves a crash when reciving a rtcp after teardown but before
+ client finalize.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
+
+2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * common:
+ Automatic update of common submodule
+ From 7bb2bce to ef1ffdc
+
+2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: refactor cleanup of cached media
+
+2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
+
+ * tests/check/gst/client.c:
+ tests: Remove FIXME
+ The session leak is now fixed, lets remove those FIXME comments.
+
+2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: Test to setup two sessions on one connection
+ https://bugzilla.gnome.org/show_bug.cgi?id=739112
+
+2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: Test setup with tcp transport
+ https://bugzilla.gnome.org/show_bug.cgi?id=739112
+
+2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: Configure transport after creating session media
+ The default implementation of configure_client_transport() in
+ rtsp-client uses the session media when it chooses channels for
+ interleaved traffic.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739112
+
+2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ client: Stop caching media in client when doing setup
+ If the media has been managed by a session media, it should not be
+ cached in the client any longer. The GstRTSPSessionMedia object is now
+ responsible for unpreparing the GstRTSPMedia object using
+ gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
+ session media.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739112
+
+2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: unref srtp decoder when leaving bin
+ https://bugzilla.gnome.org/show_bug.cgi?id=739481
+
+2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: mikey memory leaks
+ https://bugzilla.gnome.org/show_bug.cgi?id=739383
+
+2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From 84d06cd to 7bb2bce
+
+2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * Makefile.am:
+ Parallelise 'make check-valgrind'
+
+2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From a8c8939 to 84d06cd
+
+2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 36388a1 to a8c8939
+
+2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: deactivate media when shutting down from paused
+ This was only done when going directly from playing.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
+
+2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-context.h:
+ rtsp-client: add stream transport to context
+ We add the stream transport to the context so we can get the configured
+ client stream transport in the setup request signal.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
+
+2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: release lock even not all transports have been removed
+ We don't want to keep the lock even we return FALSE because not all the
+ transports have been removed. This could lead into a deadlock.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737797
+
+2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
+ These were renamed in GstRTPBasePayload in 1.0
+
+2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: set session media to NULL without the lock
+ We need to set session medias to NULL without the client lock otherwise
+ we can end up in a deadlock if another thread is waiting for the lock
+ and media unprepare is also waiting for that thread to end.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737690
+
+2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Set state to UNPREPARING in all cases
+
+2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: set state to unpreparing when unprepare is initiated
+ https://bugzilla.gnome.org/show_bug.cgi?id=737675
+
+2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Remove backlog limit while processings requests
+ If the backlog limit is kept two cases of deadlocks may be
+ encountered when streaming over TCP. Without the backlog
+ limit this deadlocks can not happen, at the expence of
+ memory usage.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
+
+2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: do not free main context before rtsp watch
+ https://bugzilla.gnome.org/show_bug.cgi?id=737110
+
+2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: Extend unit test timeout to accomodate for valgrind
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
+
+2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ rtsp-*: Treat sending packets to clients as keepalive
+ As long as gst-rtsp-server can successfully send RTP/RTCP data to
+ clients then the client must be reading. This change makes the server
+ timeout the connection if the client stops reading.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
+
+2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Allow backlog to grow while expiring session
+ Allow the send backlog in the RTSP watch to grow to unlimited size while
+ attempting to bring the media pipeline to NULL due to a session
+ expiring. Without this change the appsink element cannot change state
+ because it is blocked while rendering data in the new_sample callback.
+ This callback will block until it has successfully put the data into the
+ send backlog. There is a chance that the send backlog is full at this
+ point which means that the callback may block for a long time, possibly
+ forever. Therefore the media pipeline may also be prevented from
+ changing state for a long time.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
+
+2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Make old compilers happy
+ rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
+ Just in case that guint8 doesn't fit in a pointer. Just in case ...
+
+2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: raise the backlog limits before pausing
+ We need to raise the backlog limits before pausing the pipeline or else
+ the appsink might be blocking in the render method in wait_backlog() and
+ we would deadlock waiting for paused.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
+
+2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: make define for the WATCH_BACKLOG
+ See https://bugzilla.gnome.org/show_bug.cgi?id=736322
+
+2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: simplify session transport handling
+ link/unlink of the transport in a session was done to keep track of all
+ TCP transports and to send RTP/RTCP data to the streams. We can simplify
+ that by putting all the TCP transports in a hashtable indexed with the
+ channel number.
+ We also don't need to link/unlink the transports when we pause/resume
+ the streams. The same effect is already achieved when we pause/play the
+ media. Indeed, when we pause the media, the transport is removed from
+ the media and the callbacks will not be called anymore.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=736041
+
+2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ stream-transport: make method to handle received data
+ Make a method to handle the data received on a channel. It sends the
+ data to the stream of the transport on the RTP or RTCP pads based on
+ the channel number.
+
+2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * examples/test-mp4.c:
+ test: add example of dumping RTCP reports
+
+2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-media: Make sure that sequence numbers are monotonic after pause
+ The sequence number is not monotonic for RTP packets after pause. The
+ reason is basepayloader generates a randon sequence number when the
+ pipeline goes from ready to pause. With this fix generation of sequence
+ number will be monotonic when going from pause to play request.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736017
+
+2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Protect saved clients watch with a mutex
+ Fixes a crash when close() is called while merging clients
+ in handle_tunnel(). In that case close() would destroy the
+ watch while it is still being used in handle_tunnel().
+ https://bugzilla.gnome.org/show_bug.cgi?id=735570
+
+2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Remove the multicast group udp sources when removing from the bin
+
+2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-media: Query position and stop time only on the RTP parts of the pipeline
+ The RTCP parts, in specific the RTCP udpsinks, are not flushed when
+ seeking and will always continue counting the time. This leads to
+ the NPT after a backwards seek to be something completely different
+ to the actual seek position.
+ https://bugzilla.gnome.org/show_bug.cgi?id=732644
+
+2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/test-appsrc.c:
+ examples: fix another reference leak
+ gst_rtsp_media_get_element() returns a new ref.
+
+2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * examples/test-appsrc.c:
+ examples: unref element after usage
+ gst_bin_get_by_name_recurse_up() returns an element
+ reference that must be unreffed after usage.
+ https://bugzilla.gnome.org/show_bug.cgi?id=734546
+
+2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
+
+ * gst/rtsp-server/rtsp-media.c:
+ signals: Fix copy-pasto in target-state signal offset
+
+2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
+
+ * Makefile.am:
+ * common:
+ Makefile: Add usage of build-checks step
+ Allows building checks without running them
+
+2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Listen on the multicast group for RTP/RTCP packets
+ When a UDP multicast transport is used it is expected that the server listens
+ for RTP and RTCP packets on the multicast group with the corresponding port.
+ Without this we will never get RTCP packets from clients in multicast mode.
+ https://bugzilla.gnome.org/show_bug.cgi?id=732238
+
+2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.4.0 ===
+
+2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.4.0
+
+2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
+
+ * gst/rtsp-server/rtsp-media.h:
+ media: correct misspelled words in description
+ https://bugzilla.gnome.org/show_bug.cgi?id=733244
+
+=== release 1.3.91 ===
+
+2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.3.91
+
+2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ docs: update docs
+
+2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: implement client REMOVE filter
+
+2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: expose _close() method
+ Expose a previously internal close method to close the client
+ connection.
+
+2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ session-pool: signal session-removed outside of the lock
+ Release the lock before emiting the session-removed signal.
+
+2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ filter: Release lock in filter functions
+ Release the object lock before calling the filter functions. We need to
+ keep a cookie to detect when the list changed during the filter
+ callback. We also keep a hashtable to make sure we only call the filter
+ function once for each object in case of concurrent modification.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
+
+2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: check if watch is set in handle_teardown()
+ The unit tests run without a watch
+
+2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * tests/check/gst/client.c:
+ client tests: send teardown to cleanup session
+
+2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * tests/check/gst/rtspserver.c:
+ server tests: send teardown to cleanup session
+
+2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: keep ref to client for the session removed handler
+ This extra ref will be dropped when all client sessions have been
+ removed. A session is removed when a client sends teardown, closes its
+ endpoint of the TCP connection or the sessions expires.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
+
+2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * tests/check/gst/client.c:
+ client: manage media in session as a last step
+ Once we manage a media in a session, we can't unmanage it anymore
+ without destroying it. Therefore, first check everything before we
+ manage the media, otherwise if something is wrong we have no way to
+ unmanage the media.
+ If we created a new session and something went wrong, remove the session
+ again. Fixes a leak in the unit test.
+
+2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/test-mp4.c:
+ * examples/test-ogg.c:
+ examples: print 'stream ready at url' for mp4 and ogg example
+
+2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ rtsp: fix for MIKEY api change
+
+2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: free watch context only once
+ The watch context is freed when the source is destroyed. Avoids
+ a CRITICAL when we try to unref the context twice.
+
+2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix build
+
+2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: protect sessions with lock
+ Protect the list of sessions with the lock.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
+
+2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ Client: keep a ref to the session
+ Don't just keep a weak ref to the session objects but use a hard ref. We
+ will be notified when a session is removed from the pool (expired) with
+ the new session-removed signal.
+ Don't automatically close the RTSP connection when all the sessions of
+ a client are removed, a client can continue to operate and it can create
+ a new session if it wants. If you want to remove the client from the
+ server, you have to use gst_rtsp_server_client_filter() now.
+ Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
+ See https://bugzilla.gnome.org/show_bug.cgi?id=732226
+
+2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ session-pool: add session-removed signal
+ Add a signal to be notified when a session is removed from the pool.
+
+2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-server.h:
+ Make rtsp-server.h a single-include header, use it for G-I
+ https://bugzilla.gnome.org/show_bug.cgi?id=732411
+
+=== release 1.3.90 ===
+
+2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.3.90
+
+2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: crypto can be NULL
+
+2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-mount-points.c:
+ introspection: add missing allow-none annotations
+ https://bugzilla.gnome.org/show_bug.cgi?id=730952
+
+2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
+
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-token.c:
+ introspection: add (nullable) annotations to return values
+ https://bugzilla.gnome.org/show_bug.cgi?id=730952
+
+2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ gi: improve annotations
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
+
+2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-server.c:
+ signals: use generic marshal function
+ Use the generic C marshal function.
+ Use more explicit type instead of G_TYPE_POINTER
+
+2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-context.h:
+ context: add type macro
+
+2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-sdp.h:
+ sdp: hide key length defines
+ They don't have a namespace.
+
+2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.3.3 ===
+
+2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 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 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * common:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 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 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.3.1
+
+2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From bcb1518 to 211fa5f
+
+2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * .gitignore:
+ Update .gitignore
+
+2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/gst/sessionmedia.c:
+ tests: fix memory leak in sessionmedia unit test
+
+2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: emit a signal before sending a message
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970
+
+2014-05-01 06:07:08 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: pass context to send_message
+ Pass the current context to send_message, we will need it later.
+
+2014-05-01 05:29:54 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix typo in comment
+
+2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: Do not stop thread twice if default_prepare() fails
+
+2014-04-15 16:51:17 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: set the watch to flushing before going to NULL
+ First set the watch to flushing so that we unblock any current and
+ future attempt to send data on the watch, Then set the pipeline to
+ NULL.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153
+
+2014-04-11 23:52:49 +0200 Linus Svensson <linusp.svensson@gmail.com>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * tests/check/gst/sessionpool.c:
+ rtsp-session-pool: Fixes annotation
+ Fixes annotation for gst_rtsp_session_pool_create() and memory leaks
+ in the sessionpool test.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060
+
+2014-04-09 16:44:21 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: make media_prepare virtual
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
+
+2014-04-12 05:57:00 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * tests/check/gst/media.c:
+ media: stop the thread in more error cases
+
+2014-04-12 05:53:15 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * tests/check/gst/media.c:
+ media: allow NULL as the thread
+ Use the default context whan passing a NULL thread.
+
+2014-04-10 16:39:11 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: indent cleanup
+ Coverity was moaning about unreachable code, and I think it was just
+ confused by { being before the label. We'll see if it pops up again.
+ Coverity 1197705
+
+2014-04-01 13:04:21 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ client: Add drop-backlog property
+ When we have too many messages queued for a client (currently hardcoded
+ to 100) we overflow and drop the messages. Add a drop-backlog property
+ to control this behaviour. Setting this property to FALSE will retry
+ to send the messages to the client by waiting for more room in the
+ backlog.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
+
+2014-04-03 12:19:51 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: support for POST before GET when setting up a tunnel
+
+2014-04-02 12:03:32 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: remove watch of the second client after http tunnel setup
+ The second client will be freed after the HTTP tunnel has been set up.
+ Make sure it's RTSP watch is never dispatched again.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488
+
+2014-03-31 11:00:11 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * tests/check/gst/media.c:
+ media: Make media_prepare() fail if port allocation fails
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376
+
+2014-04-01 16:55:13 +0200 Linus Svensson <linussn@axis.com>
+
+ * tests/check/gst/media.c:
+ media test: cleanup the thread pool in tests
+
+2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * tests/check/gst/media.c:
+ rtsp-media: Unblock blocked streams in unprepare
+ The streams will be blocked when a live media is prepared.
+ The streams should be unblocked in gst_rtsp_media_unprepare.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231
+
+2014-04-08 14:49:41 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: release the state lock when going to NULL
+ Set our state to UNPREPARING and release the state-lock before
+ setting the pipeline to the NULL state. This way, any pad-added
+ callback will be able to take the state-lock and check that we are now
+ unpreparing instead of deadlocking.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102
+
+2014-04-08 12:08:17 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: protect status with lock
+ Make sure we only update the status with the lock.
+
+2014-04-04 17:39:36 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ rtsp: update for MIKEY API changes
+
+2014-04-03 12:52:51 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: parse the mikey response from the client
+ Parse the mikey response from the client and update the policy for
+ each SSRC.
+
+2014-04-02 12:36:16 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: add method to set crypto info
+ Make a method to configure the crypto information of a stream.
+ Set udpsrc in READY instead of PAUSED so that we can configure caps
+ later.
+
+2014-04-03 12:57:13 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: cleanup error paths
+
+2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: fix docs
+
+2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * examples/test-video.c:
+ test: enable SRTP only on RTSPS
+ We only want to enable SRTP when doing rtsp over TLS so that we can
+ exchange the keys in a secure way.
+
+2014-03-25 12:41:33 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * examples/test-video.c:
+ test: print an error on failure
+
+2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * configure.ac:
+ * examples/test-video.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * tests/check/Makefile.am:
+ stream: add SRTP support
+ Install srtp encoder and decoder elements in rtpbin
+ Add MIKEY in SDP
+
+2014-03-16 19:45:26 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/gst/sessionpool.c:
+ tests: Add unit tests for sessionpool
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470
+
+2014-03-22 13:24:27 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * tests/check/gst/threadpool.c:
+ tests: Improve code coverage of rtsp-threadpool tests
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873
+
+2014-03-23 21:26:00 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * tests/check/gst/sessionmedia.c:
+ tests: Improve code coverage for rtsp-session-media
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940
+
+2014-03-23 21:24:48 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ gobject-introspection: Add annotations to support language bindings
+ In addition a few cosmetic changes:
+ * Adjust the order of arguments
+ * Fix typo: occured -> occurred
+ * Fix indentation after Return:-clauses
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
+
+2014-03-14 19:03:24 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Don't mix IPv4 and IPv6 addresses
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
+
+2014-03-13 14:27:15 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: take caps after the session manager
+ Take the caps for the SDP after they leave the rtpbin so that we can
+ also get the properties added by rtpbin elements.
+
+2014-03-13 14:20:17 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: release lock while pushing out packets
+ Keep a cache of the transports and use this to iterate the transport
+ while pushing packets. This allows us to release the lock early.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=725898
+
+2014-03-06 13:52:02 +0100 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ rtsp-client: vmethod for modifying tunnel GET response
+ Add a vmethod tunnel_http_response where the response to the HTTP GET
+ for tunneled connections can be modified.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879
+
+2014-03-03 16:56:53 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ sdp: make 1 media line per profile
+ If we have multiple profiles (AVP or AVPF) for a stream, make one m=
+ line in the SDP for each profile. The client is then supposed to pick
+ one of the profiles in the SETUP request. Because the m= lines have the
+ same pt, the client also knows that only 1 option is possible.
+
+2014-03-03 16:55:48 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ factory: add profile property and pass to media and streams
+
+2014-03-03 15:12:55 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * examples/test-multicast.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ sdp: pass multicast connection for multicast-only stream
+ Pass the multicast address of the stream in the connection info in the
+ SDP so that clients try a multicast connection first.
+ Only allow multicast connections in the test-multicast example. Also
+ increase the TTL a little.
+
+2014-03-02 05:12:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * .gitignore:
+ .gitignore: Ignore gcov intermediate files
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484
+
+2014-03-03 12:17:48 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: release some locks in error cases
+
+2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ docs: Enable and fix gtk-doc warnings
+ * Makefile: Enable gtk-doc warnings, like the rest of GStreamer
+ * addresspool/mediafactory: Add missing annotation colon
+ * stream: Annotate return value
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
+
+2014-02-28 09:36:49 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From fe1672e to bcb1518
+
+2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 1a07da9 to fe1672e
+
+2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/Makefile.am:
+ examples: use LDADD for libs instead of LDFLAGS
+
+2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ configure: make sure releases are in .doap file
+
+2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/test-cgroups.c:
+ examples: test-cgroups: don't put code with side effects into g_assert()
+ The g_assert() might get compiled out with the right
+ compiler/preprocessor flags.
+
+2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/.gitignore:
+ examples: add cgroup test binary to .gitignore
+
+2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/test-cgroups.c:
+ examples: fix cgroup test build
+ Fixes build failure caused by compiler warning:
+ test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
+
+2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * .gitignore:
+ .gitignore: ignore temp files created in the course of 'make check'
+
+2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: don't loose frames handling new PLAY request
+ If client supplied a range check if the range specifies the start point.
+ If not, then do an accurate seek to the current position. If a start
+ point was specified do do a key unit seek to make sure the streaming
+ starts with decodeable frames.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
+
+2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ Revert "media: only flush when setting a new start position"
+ This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a.
+ We need to do the flush in all cases, demuxer block currently for
+ non-flushing seeks.
+
+2014-02-18 16:38:39 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: only flush when setting a new start position
+ Only flush the pipeline when we change the start position with
+ a seek.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=724611
+
+2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: set ttl-mc before adding the socket
+ Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
+ never be set on socket.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
+
+2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: stop thread if media is already prepared
+ in gst_rtsp_media_prepare() the thread is not used if media is already
+ prepared (e.g. media shared) so we want to stop the thread. otherwise, a
+ leak occurs.
+ https://bugzilla.gnome.org/show_bug.cgi?id=724182
+
+2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * Makefile.am:
+ build: Ship gst-rtsp-server.doap file
+
+2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: Fix another compiler warning with gcc
+
+2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-mount-points.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * tests/check/gst/client.c:
+ rtsp-server: Fix lots of compiler warnings with clang
+
+2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ * tests/Makefile.am:
+ configure: Synchronise with the configure scripts of the other modules
+
+2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
+
+2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ Revert "rtsp-server: support build against last stable release"
+ This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
+ Let us require 1.2.3 now, which is going to be released in a few
+ minutes.
+
+2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ session: improve RTP-Info
+ Ignore streams that can't generate RTP-Info instead of failing.
+ Don't return the empty string when all streams are unconfigured but
+ return NULL so that we don't generate and empty RTP-Info header.
+ Improve docs a little.
+
+2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
+
+ * gst/rtsp-server/rtsp-session-media.c:
+ Don't free rtpinfo GString when it is NULL
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
+
+2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: only set keyframe flag when modifying start
+ Only set the keyframe flag when we modify the start position. The
+ keyframe flag should probably be ignored when no change is requested but
+ until we can claim this is all documented properly and all demuxer
+ implement this, avoid setting the flag.
+ See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
+
+2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ thread-pool: Unref source after mainloop has quit to avoid races in GLib
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
+
+2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: handle NULL seqnum and rtptime arguments
+
+2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ * tests/check/gst/threadpool.c:
+ thread-pool: Unref reused threads in gst_rtsp_thread_stop()
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
+
+2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: add fallback for missing stats property
+ Use a fallback when the payloader does not have a stats property
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
+
+2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * common:
+ Automatic update of common submodule
+ From f7bc1c3 to 1a07da9
+
+2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: don't leak stats structure
+ Don't leak the stats structure and deal with NULL stats.
+
+2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: Get rtpinfo properties atomically from payloader
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
+
+2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: refactor state change functions and signals
+ Make functions to set the target state and the pipeline state and emit
+ the signals from those functions.
+
+2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: add signal to notify of pending state changes
+
+2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-server: support build against last stable release
+ Until 1.2.3 is out with the new get_type function and we
+ can require that.
+
+2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: fix compilation
+
+2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: add property to configure profiles
+
+2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: let stream check supported transport
+ Delegate the check if a transport is allowed to the stream.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=720696
+
+2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: add method to check supported transport
+ Add a method to check if a transport is supported
+
+2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ configure.ac: Only check for gstreamer-check, not check
+ We include check in gstreamer-check since quite some time now.
+
+2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: return clock-rate from get_rtpinfo
+ And use it to correct the rtptime to the requested start-time.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=712198
+
+2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ session-media: calculate start-time
+
+2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: also return the running-time
+ Return the running-time in the rtpinfo as well.
+
+2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-media.h:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ session-media: let the session-media make the RTPInfo
+ Add method to create the RTPInfo for a stream-transport.
+ Add method to create the RTPInfo for all stream-transports in a
+ session-media.
+ Use the session-media RTPInfo code in client. This allows us to refactor
+ another method to link the TCP callbacks.
+
+2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ mount-points: sort sequence before g_sequence_lookup
+ * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
+ sort sequence if dirty, otherwise lookup will fail.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
+
+2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ configure: rename package from gst-rtsp to gst-rtsp-server
+ To match git module name and avoid confusion with the
+ rtsp lib in gst-plugins-base and rtsp plugin in -good.
+
+2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ configure: bump core/base/good requirement to 1.2.0
+ Bump to released stable version and make implicit
+ requirements explicit.
+
+2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * autogen.sh:
+ * common:
+ * configure.ac:
+ Fix broken gettext setup which is not used anyway
+
+2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From dbedaa0 to d48bed3
+
+2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: add setup_sdp vmethod
+ gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
+ gst_rtsp_media_setup_sdp.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
+
+2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Check return value of sscanf
+ streamid is only valid if sscanf matched something.
+
+2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Fix iteration
+ Wouldn't even enter the code block otherwise (i++ was used as the check
+ and not the postfix).
+
+2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: add vmethod to configure media and streams
+ Implement a vmethod that can be used to configure the media and the
+ streams based on the current context. Handle the blocksize handling in
+ the default handler.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=720667
+
+2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * .gitignore:
+ Make git ignore more unit test binaries
+
+2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-address-pool.h:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-context.h:
+ * gst/rtsp-server/rtsp-media-factory-uri.h:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-mount-points.h:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-media.h:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.h:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.h:
+ * gst/rtsp-server/rtsp-thread-pool.h:
+ * gst/rtsp-server/rtsp-token.h:
+ rtsp-server: add padding to many public structures
+ Not mini objects though, since they are not subclassable
+ anyway, nor kept on the stack or inlined in a structure.
+
+2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ media: add new create_rtpbin vmethod
+ * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
+ https://bugzilla.gnome.org/show_bug.cgi?id=719734
+
+2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
+
+ * tests/check/gst/media.c:
+ tests: fix memory leak, free test's thread pool
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
+
+2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ stream-transport: free url in finalize
+
+2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: also do state change in suspended state
+
+2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ media: also handle prepare and range in suspended state
+ When we are suspended, we are already prepared.
+ We can get the range in the suspended state.
+
+2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/gst/sessionmedia.c:
+ check: add test for uri in setup
+ Added unit tests for the new functionality in GstRTSPStreamTransport.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
+
+2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: store setup uri and use in PLAY response
+ Store the uri used when doing the setup and use that in the PLAY
+ response.
+ fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
+
+2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ stream-transport: add method to get/set url
+
+2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: suspend after SDP and unsuspend before PLAYING
+ Based on patches by Ognyan Tonchev <ognyan@axis.com>
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
+
+2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.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:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * tests/check/gst/media.c:
+ * tests/check/gst/mediafactory.c:
+ media: add suspend modes
+ Add support for different suspend modes. The stream is suspended right after
+ producing the SDP and after PAUSE. Different suspend modes are available that
+ affect the state of the pipeline. NONE leaves the pipeline state unchanged and
+ is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
+ state and RESET will bring the pipeline to the NULL state.
+ A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
+ this means that the pipeline needs to be prerolled again.
+ Base on patches by Ognyan Tonchev <ognyan@axis.com>
+ See https://bugzilla.gnome.org/show_bug.cgi?id=711257
+
+2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: start live streams in blocked state
+ Start live streams in the blocked state and make them preroll using the
+ messages. This ensure that no data is played by the sink until we explicitly
+ unblock the stream right before going to PLAYING.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=711257
+
+2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: refactor starting and waiting for preroll
+ Based on patches from Ognyan Tonchev <ognyan@axis.com>
+ See https://bugzilla.gnome.org/show_bug.cgi?id=711257
+
+2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: add API to block streams
+ Add an API to block on the streams and make it post a message.
+ Based on patch by Ognyan Tonchev <ognyan@axis.com>
+ See https://bugzilla.gnome.org/show_bug.cgi?id=711257
+
+2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
+
+ * docs/libs/Makefile.am:
+ docs: Specify the override file
+ Even if it's empty (for now) it avoids make distcheck complaining
+
+2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: move default implementations to where they are used
+
+2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: take the right lock in gst_rtsp_media_set_pipeline_state()
+ We need to take the state_lock when calling this method.
+
+2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: handle add-added on non-bins too
+ Handle dynamic payloaders that are not bins, as used in the unit-test.
+
+2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media/-factory: Fix request pad name comments
+ These must be escaped for gtk-doc to parse the comments without warnings.
+
+2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
+
+ rtsp-media: remove transports if media is in error status
+ * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
+ trying to change to GST_STATE_NULL and media is in error status, we
+ remove all transports.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
+
+2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: use element metadata to find payloader
+ Use the element metadata to find the payloader instead of checking
+ for the base class.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
+
+2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
+
+ rtsp-stream: add getter for payload type
+ * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
+ * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
+ element and create the stream with this one instead of the dynpay%d
+ element.
+ https://bugzilla.gnome.org/show_bug.cgi?id=712396
+
+2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-context.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-mount-points.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-token.c:
+ rtsp-*: Refer to NULL as a constant in comments
+ Plus one typo fix.
+ https://bugzilla.gnome.org/show_bug.cgi?id=714988
+
+2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ rtsp-*: Fix type name typos in comments
+ * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
+ * rtsp-auth: Refer to part of constant name as text
+ * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
+ * rtsp-session-media: Fix GstRTSPSessionMedia typo
+ * rtsp-stream: Fix typo when refering to GstBin
+ https://bugzilla.gnome.org/show_bug.cgi?id=714988
+
+2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * docs/README:
+ * docs/libs/gst-rtsp-server-docs.sgml:
+ * docs/libs/gst-rtsp-server-sections.txt:
+ docs: Improve documentation
+ * Include annotation-glossary to quiet gtk-doc
+ * Rename remaining ClientState -> Context
+ * Rename object hierarchy file
+ * Remove stale chapter references
+ * Add missing function and object references
+ * Include missing GstRTSPAddressPoolResult
+ https://bugzilla.gnome.org/show_bug.cgi?id=714988
+
+2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-server: sprinkle some allow-none annotations for g-i
+
+2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: add method to filter transports
+ Add a method to safely iterate and collect the stream transports
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
+
+2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ rtsp: allow NULL func in filters
+ Passing a null function make the filters return a list of
+ refcounted objects.
+
+2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
+
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * tests/check/gst/addresspool.c:
+ address-pool: fix address increment
+ Use a guint instead of guint8 to increment the address. It's still not
+ completely correct because a guint might not be able to hold the complete
+ address range, but that's an enhacement for later.
+ Add unit test to test improved behaviour.
+ https://bugzilla.gnome.org/show_bug.cgi?id=708237
+
+2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * tests/check/gst/client.c:
+ client: allow absolute path in requests
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
+
+2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: make make_path_from_uri a vmethod
+
+2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ * tests/check/Makefile.am:
+ * tests/check/gst/stream.c:
+ stream: Add functions to get rtp and rtcp sockets
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
+
+2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
+
+ * gst/rtsp-server/rtsp-context.c:
+ * gst/rtsp-server/rtsp-context.h:
+ context: defing a GType for the context
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
+
+2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-context.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-mount-points.c:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ Fixed several GIR warnings
+
+2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ auth: small typos
+
+2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/gst/token.c:
+ tests: Add unit tests for token
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
+
+2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/rtsp-server/rtsp-token.c:
+ token: Validate args for gst_rtsp_token_is_allowed
+ See https://bugzilla.gnome.org/show_bug.cgi?id=710520
+
+2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/rtsp-server/rtsp-token.c:
+ token: Fix bug when creating empty token
+ We always want to have a valid GstStructure in the token.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
+
+2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ thread-pool: avoid race in shutdown
+ If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
+ don't actually stop the mainloop ever. Solve this race by adding an idle source
+ to the mainloop that calls the _quit. This way we immediately exit the mainloop
+ if quit was called before we started it.
+
+2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/gst/permissions.c:
+ tests: Add unit tests for permissions
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
+
+2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * tests/check/gst/mediafactory.c:
+ tests: Test mediafactory permissions
+ See https://bugzilla.gnome.org/show_bug.cgi?id=710202
+
+2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/rtsp-server/rtsp-permissions.c:
+ permissions: Fix refcounting when adding/removing roles
+ Previously a role that was removed was unreffed twice, and when
+ replacing an existing role the replaced role was freed while still being
+ referenced. Both bugs are now fixed.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=710202
+
+2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * tests/check/gst/media.c:
+ * tests/check/gst/mediafactory.c:
+ * tests/check/gst/rtspserver.c:
+ tests: Check gst_rtsp_url_parse return value
+ See https://bugzilla.gnome.org/show_bug.cgi?id=710202
+
+2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From 865aa20 to dbedaa0
+
+2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ rtsp-server: Fix socket leak
+ https://bugzilla.gnome.org/show_bug.cgi?id=710088
+
+2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ rtsp-session-pool: Make sure session IDs are properly URI-escaped
+ https://bugzilla.gnome.org/show_bug.cgi?id=643812
+
+2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
+
+ * examples/.gitignore:
+ * examples/test-video.c:
+ examples: fix compilation when WITH_AUTH is defined
+ https://bugzilla.gnome.org/show_bug.cgi?id=710228
+
+2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * .gitignore:
+ gitignore: Add new test binary
+
+2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/gst/threadpool.c:
+ thread-pool: Add unit test for the thread pools
+ https://bugzilla.gnome.org/show_bug.cgi?id=710228
+
+2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ thread-pool: Fix thread leak when reusing threads
+ https://bugzilla.gnome.org/show_bug.cgi?id=709730
+
+2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * tests/check/gst/rtspserver.c:
+ tests: fixed racy behavior in rtspserver tests
+ https://bugzilla.gnome.org/show_bug.cgi?id=710078
+
+2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * tests/check/gst/addresspool.c:
+ tests: Improve address pool unit tests
+ Add a range with mixed IPV4 and IPV6 addresses to pool.
+ Get an IPV4 address from an IPV6-only pool.
+ Get an IPV6 address from an IPV4-only pool.
+ Reserve a IPV6 address from an IPV4-only pool.
+ Check for unicast addresses in multicast-only pool.
+ Check for unicast addresses in uni-/multicast-mixed pool.
+ https://bugzilla.gnome.org/show_bug.cgi?id=710128
+
+2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: append query string in PAUSE/PLAY/TEARDOWN as well
+
+2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: Add query to control path
+ If the SETUP url contains a query it must be appended to the control
+ path so that it matches any already created stream in the media. The
+ query will also be appended to the session media path.
+
+2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: remove old line
+
+2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: Correct control comparison
+ https://bugzilla.gnome.org/show_bug.cgi?id=709176
+
+2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: Check dynamically if the pipeline supports seeking
+ We should not depend on whether or not the pipeline state change
+ returned NO_PREROLL or not. A media could dynamically change its
+ element and switch from seekable to non seekable so it's best to test
+ the seekable nature of the pipeline dynamically when we try to do a seek.
+
+2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: Return FALSE if seeking is not supported
+
+2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: don't seek accurate by default
+ Accurate seeking is perhaps a little overkill in the most common situation and
+ causes some formats (mp3) over slow media to seek extremely slowly.
+
+2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: fix unit test
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
+
+2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: Reply 400 if media cannot be constructed
+ Reply 400 Bad Request instead of 503 Service Unavailable if media
+ cannot be constructed in SETUP.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
+
+2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: Send setup reply once only
+ If find_media() failed in handle_setup_request() two replies was sent.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
+
+2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * common:
+ Automatic update of common submodule
+ From 6b03ba7 to 865aa20
+
+2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: Emit client-connected signal earlier
+ Emit client-connected before the client ref is given to a GSource,
+ otherwise client-connected can be emitted after the client object has
+ been freed.
+
+2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
+
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-address-pool.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * tests/check/gst/addresspool.c:
+ addresspool: return reason of failure
+ Let gst_rtsp_address_pool_reserve_address() return the reason why
+ the address could not be reserved.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
+
+2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
+
+ * autogen.sh:
+ autogen.sh: Sync behaviour with other GStreamer modules
+ Allows building from outside of tree amongst other things
+
+2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
+
+ * common:
+ Automatic update of common submodule
+ From b613661 to 6b03ba7
+
+2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * common:
+ Automatic update of common submodule
+ From 74a6857 to b613661
+
+2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * common:
+ Automatic update of common submodule
+ From 01a7a46 to 74a6857
+
+2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: Do not read beyond end of path string
+ If the setup was done without a control url, make sure we don't try to read the
+ non-existing control string and crash.
+
+2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: Fix RTPInfo header
+ Refactor the method to make the content_base.
+ Use the content-base and the control url to construct the RTPInfo
+ url.
+
+2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: map url to path only in describe
+ Only map the request url to a path in the DESCRIBE method. The SDP then
+ contains the base and control urls that should be used to SETUP/PAUSE/
+ PLAY/TEARDOWN the media.
+
+2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ Revert "client: map URL to path in requests"
+ This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
+ This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
+ contains the base and control urls which are used in the SETUP, PLAY,
+ PAUSE and TEARDOWN requests.
+
+2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: map URL to path in requests
+
+2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-mount-points.c:
+ * gst/rtsp-server/rtsp-mount-points.h:
+ mount-points: make vmethod to make path from uri
+ Make a vmethod to transform an url into a path. The path is then used to lookup
+ the factory. This makes it possible to also use other bits of the url, such as
+ the query parameters, to locate the factory.
+
+2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ * gst/rtsp-server/rtsp-thread-pool.h:
+ thread-pool: Add cleanup to wait for the threadpool to finish
+ Also fix race condition if two threads are asking for the first
+ thread from the thread pool at once. This would case two internal
+ GThreadPools to be created.
+ https://bugzilla.gnome.org/show_bug.cgi?id=707753
+
+2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * tests/check/gst/client.c:
+ client: free threadpool
+ https://bugzilla.gnome.org/show_bug.cgi?id=707638
+
+2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * tests/check/gst/mountpoints.c:
+ mountpoints tests: unref matched factories
+ https://bugzilla.gnome.org/show_bug.cgi?id=707638
+
+2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * tests/check/gst/media.c:
+ media tests: unref thread pool and caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=707638
+
+2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ auth, media, media-factory: unref permissions
+ https://bugzilla.gnome.org/show_bug.cgi?id=707638
+
+2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/Makefile.am:
+ Makefile: add rule for appsrc example
+
+2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-appsrc.c:
+ tests: add appsrc example
+ Add an example on how to use appsrc to feed the server pipeline with data.
+
+2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: remove query part from content-base string
+ Make sure that after the control url has been resolved, it's
+ not a part of the query-string.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
+
+2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: don't check url in response
+ There is no url or method in the response to check
+
+2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ Add handle-response signal for when we receive a GET_PARAMETER response
+
+2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ Fix gst_rtsp_server_client_filter, using wrong variable type
+
+2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
+ For AAC we need to check for framed=true instead of parsed=true.
+ https://bugzilla.gnome.org/show_bug.cgi?id=701384
+
+2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: optimize pipeline for protocols
+ When TCP is not an allowed protocol for the stream, avoid creating the
+ appsrc/appsink/queue and tee elements.
+
+2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: set protocols on streams
+
+2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: use protocols supported by stream
+
+2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ media-factory: allow all protocols
+
+2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: configure protocols in new streams
+
+2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: add protocols property
+
+2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: send state in "new-state" signal
+ https://bugzilla.gnome.org/show_bug.cgi?id=705110
+
+2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
+
+ * configure.ac:
+ build: add subdir-objects to AM_INIT_AUTOMAKE
+ Fixes warnings with automake 1.14
+ https://bugzilla.gnome.org/show_bug.cgi?id=705350
+
+2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ server: add method to iterate clients of server
+
+2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ Add vmethod for rtsp-media subclass to access rtpbin
+
+2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.h:
+ small documentation fix
+
+2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ Do not take range header if range is invalid
+
+2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-media.c:
+ media: add docs for new method
+
+2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ Add API to rtsp-media set the pipeline's state
+
+2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ Update current position/duration when gst_rtsp_media_get_range_string is called
+
+2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-cgroups.c:
+ tests: add some more docs
+
+2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-cgroups.c:
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-context.c:
+ * gst/rtsp-server/rtsp-context.h:
+ * gst/rtsp-server/rtsp-params.c:
+ * gst/rtsp-server/rtsp-params.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ * gst/rtsp-server/rtsp-thread-pool.h:
+ * tests/check/gst/client.c:
+ ClientState -> Context
+ Rename the clientstate to context and put the code in a separate file.
+
+2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-auth.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ auth: add support for default token
+ The default token is used when the user is not authenticated and can be used to
+ give minimal permissions.
+
+2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-auth.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ auth: use defines when possible
+
+2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-address-pool.c:
+ address-pool: improve docs
+
+2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-permissions.c:
+ permissions: add the role to the copy
+
+2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-permissions.c:
+ permissions: Also copy the roles
+
+2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-permissions.c:
+ permissions: Make it build
+
+2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-address-pool.h:
+ docs: small fixes
+
+2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * tests/check/gst/client.c:
+ docs: improve docs
+
+2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-address-pool.h:
+ * tests/check/gst/addresspool.c:
+ * tests/check/gst/rtspserver.c:
+ address-pool: cleanups
+ Remove redundant method, improve docs.
+
+2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-permissions.c:
+ * gst/rtsp-server/rtsp-permissions.h:
+ * gst/rtsp-server/rtsp-token.c:
+ docs: improve docs
+
+2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-permissions.c:
+ permissions: implement _remove_role
+
+2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-permissions.c:
+ permissions: update docs
+
+2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/gst/client.c:
+ tests: simplify tests
+ Client settings are now disabled by default so we don't need an auth
+ module to disable them.
+
+2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ auth: add default authorizations
+ When no auth module is specified, use our table of defaults to look up the
+ default value of the check instead of always allowing everything. This was
+ we can disallow client settings by default.
+
+2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/README:
+ README: update readme
+
+2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ * gst/rtsp-server/rtsp-thread-pool.h:
+ thread-pool: add more docs
+
+2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ * gst/rtsp-server/rtsp-thread-pool.h:
+ thread-pool: fix race in thread reuse
+ If we try to reuse a thread right after we made it stop, we end up using a
+ stopped thread. Catch this case and only reuse threads that are not stopping.
+
+2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: add small debug
+
+2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/gst/client.c:
+ client: fix test
+ Add some permissions to media so we can use the auth and enable
+ client settings.
+
+2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: support pushed context in handle_request
+ If we already have a pushed state, reuse it and add our own things. This makes
+ it easier to write tests.
+
+2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ auth: don't auth on methods
+ Don't authorize on methods anymore but on the resources that we
+ try to access, this is more flexible.
+ Move the authorization checks to where they are needed and let the
+ check return the response on error.
+
+2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-mount-points.c:
+ mount-points: add some debug
+
+2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/gst/client.c:
+ tests: almost fix test
+
+2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ auth: let the auth module check client_settings
+ Let the auth module decide if client settings are allowed for the
+ current client.
+
+2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-token.c:
+ * gst/rtsp-server/rtsp-token.h:
+ token: add method to check boolean permission
+
+2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-auth.c:
+ * examples/test-cgroups.c:
+ * gst/rtsp-server/rtsp-token.c:
+ * gst/rtsp-server/rtsp-token.h:
+ token: simplify token constructor
+ Use variable arguments to make easier API.
+
+2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-auth.c:
+ * examples/test-cgroups.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ media-factory: add convenience API for factory
+
+2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-auth.c:
+ * examples/test-cgroups.c:
+ * gst/rtsp-server/rtsp-permissions.c:
+ * gst/rtsp-server/rtsp-permissions.h:
+ permissions: simplify API a little
+ Avoid passing GstStructure in the add_role method, use varargs instead
+ to construct the structure behind the scenes. We can then also use the
+ structure name as the role and simplify some more logic.
+
+2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ auth: fix typo
+
+2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ auth: handle unauthorized response
+ Move handling of the unauthorized response to the auth module, it can add
+ the appropriate headers to request authorization for the required method
+ much better than the client.
+
+2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: allow for sending any message, not only requests
+ Change the _send_request() method to _send_message() so that we
+ can both send requests and replies.
+
+2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-server.h:
+ docs: fix docs
+
+2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-video.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ auth: move TLS handling to auth module
+ Remove the TLS settings on the server and move it to the auth module because
+ that is where security related bits go.
+
+2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: add state push/pop
+
+2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: add connection to state
+
+2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-mount-points.c:
+ mount-points: fix debug
+
+2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/gst/media.c:
+ tests: fix media test
+
+2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ thread-pool: we don't require a state
+
+2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: let context ref the server
+ So that we don't risk losing the server object early anc crash.
+
+2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/gst/client.c:
+ tests: fix client test
+
+2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/README:
+ * docs/libs/gst-rtsp-server-docs.sgml:
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media-factory-uri.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-mount-points.c:
+ * gst/rtsp-server/rtsp-params.c:
+ * gst/rtsp-server/rtsp-permissions.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ * gst/rtsp-server/rtsp-token.c:
+ docs: improve docs
+
+2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ session-pool: make vmethod to create a session
+ Make a vmethod to create a sessions so that subclasses can create
+ custom session objects
+
+2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-mount-points.h:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-stream.h:
+ docs: more updates
+
+2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-docs.sgml:
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-address-pool.h:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-permissions.c:
+ * gst/rtsp-server/rtsp-permissions.h:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-media.h:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.h:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-thread-pool.h:
+ docs: update docs
+
+2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ * examples/Makefile.am:
+ configure: compile cgroup example conditionally
+ Only compile the cgroup example when we have libcgroup
+
+2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ * examples/Makefile.am:
+ * examples/test-cgroups.c:
+ examples: add cgroups example
+
+2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/gst/rtspserver.c:
+ tests: fix compilation
+
+2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ thread-pool: fix vmethod invocation
+
+2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ * gst/rtsp-server/rtsp-thread-pool.h:
+ thread-pool: store thread type in thread
+
+2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: pass thread from pool to media _prepare
+ Get a thread from the configured threadpool and pass it to the prepare method of
+ the media.
+
+2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: Accept a thread in _prepare
+ Remove out own threadpool handling and use the provided thread and
+ maincontext for the bus messages and the state changes.
+
+2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: configure client thread pool
+
+2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: add method to configure thread pool
+
+2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ server: use thread pool
+ Use the thread pool instead of doing our own thing.
+
+2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ * gst/rtsp-server/rtsp-thread-pool.h:
+ thread-pool: add object to manage threads
+ Add an object to manage the client and media threads.
+
+2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ auth: debug authorization check
+
+2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: start media pipeline in context
+ Start the media pipeline in the provided context (or our default one
+ when NULL). This makes sure that we run the bus thread in this context and that
+ all media threads are children of this context.
+
+2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ factory: pass permissions to media by default
+
+2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-auth.c:
+ test: add permissions to auth test
+ Ass some permissions to the media factory in the test.
+
+2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ auth: simplify auth checks
+ Remove client from methods, it's now in the state
+ Perform the check specified by the string, use the information from the
+ thread local context.
+
+2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: add state to current thread
+ Add the client to the ClientState object.
+ Place the ClientState on the current thread.
+
+2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * 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:
+ media: make it possible to set permissions
+ Make it possible to set permissions on media and media factory objects
+
+2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-permissions.c:
+ * gst/rtsp-server/rtsp-permissions.h:
+ permissions: add permissions object
+ Add a mini object to store permissions based on a role.
+
+2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-auth.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ auth: add auth checks
+ Add an enum with auth checks and implement the checks in the auth object.
+ Perform the checks from the client.
+
+2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-auth.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.h:
+ auth: use the token after authentication
+ After we authenticated a user, keep the Token around in the state.
+
+2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * tests/check/gst/media.c:
+ media: add optional context for bus messages
+ Add an optional mainloop to _prepare that will handle the bus messages instead
+ of always using the shared mainloop.
+
+2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-token.c:
+ * gst/rtsp-server/rtsp-token.h:
+ token: add authorization token
+ Add a simply miniobject that contains the authorizations. The object contains a
+ GstStructure that hold all authorization fields. When a user is authenticated,
+ the auth module will create a Token for the user. The token is then used to
+ check what operations the user is allowed to do and various other configuration
+ values.
+
+2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-auth.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * 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:
+ auth: remove auth from media and factory
+ Remove the auth object from media and factory. We want to have the RTSPClient
+ authenticate and authorize resources, there is no need to place another auth
+ manager on the media/factory.
+
+2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-auth.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.h:
+ auth: add support for multiple basic auth tokens
+ Make it possible to add multiple basic authorisation tokens to one authorization
+ object. Associate with each token an authorization group that will define what
+ capabilities are allowed.
+
+2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: error out on non-aggregate control
+ We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
+
+2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: rework setup request a little
+ Cache the media in DESCRIBE based on the longest matching path with the uri
+ that we can find in the mount points.
+ Rework the setup request a little to get the media from the session or from
+ the longest matching path, this way we can derive the control string as
+ everything after the path instead of hardcoding it.
+ Find the stream based on the control string and only open a session when all
+ this can be done.
+
+2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: add method to find a stream by control url
+
+2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: add method to check control url of stream
+
+2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-media.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ session: use path matching for session media
+ Use a path string instead of a uri to lookup session media in the sessions. Also
+ use path matching to find the largest possible path that matches.
+
+2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-mount-points.c:
+ * gst/rtsp-server/rtsp-mount-points.h:
+ * tests/check/gst/mountpoints.c:
+ mount-points: remove useless vmethod
+ Making lookups in the mount points should not be done with a URL, if there is a
+ mapping to be done from URL to mount points, we'll need to do it somewhere
+ else.
+
+2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-mount-points.c:
+ * gst/rtsp-server/rtsp-mount-points.h:
+ * tests/check/gst/mountpoints.c:
+ mount-points: improve mount point searching
+ Use a GSequence to keep track of the mount points.
+ Match a URL to the longest matching registered mount point. This should be the
+ URL to perform aggreagate control and the remainder is the stream specific
+ control part.
+ Add some unit tests for this.
+
+2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst/rtsp-server/Makefile.am:
+ rtsp-server: Allow building of static library
+
+2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/gst/mediafactory.c:
+ tests: fix compilation
+
+2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ sdp: get control string from stream
+ Use the control string as configured in the stream.
+
+2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: add methods and property to set control string
+
+2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: cleanups
+ Rename variables for clarity
+ Keep media in state when we can
+
+2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: add more support for IPv6
+ Rename _get_address to _get_multicast_address in GstRTSPStream to
+ make it clear that this function only deals with multicast.
+ Make it possible to have both an IPv4 and IPv6 multicast address on
+ a stream. Give the client an IPv4 or IPv6 address depending on the
+ address it used to connect to the server.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
+
+2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix comment
+
+2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: handle failed port allocation
+ Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
+ can't allocate any family at all. Also keep track of what port families we
+ allocated.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
+
+2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: improve docs
+
+2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ stream-transport: remove old if 0 block
+
+2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * tests/check/gst/client.c:
+ tests: fix tests
+ gst_rtsp_client_get_uri() has been removed
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
+
+2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: add method to filter managed sessions
+ Add a method to filter the sessions managed by this client connection.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=703016
+
+2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: remove _get_uri() method
+ Remove the get_uri() method on the client. A client has no uri, the uri
+ property is an internal property to manage the last cached media for
+ the client.
+
+2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.h:
+ media-factory: fix typo
+
+2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Do not leak the query in default_query_stop
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
+
+2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: don't unlock when conversion fails
+ Don't unlock the state lock when conversion fails because it was not locked.
+
+2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ Add query_position and query_stop vmethods to rtsp-media
+
+2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ Fix typo in property install for rtsp-media's time-provider
+
+2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: clean some variables
+ Clean some variables and add some guards to _send_request()
+
+2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ Add gst_rtsp_client_send_request API
+ This makes it possible to send arbitrary messages to a client, such as
+ SET_PARAMETER or GET_PARAMETER
+
+2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: add _get_element() method
+ Add method to get the element used when creating the media.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
+
+2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: fix docs
+
+2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: allow access to the rtp session
+ https://bugzilla.gnome.org/show_bug.cgi?id=703004
+
+2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ dscp qos support in gst-rtsp-stream
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
+
+2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/gst/rtspserver.c:
+ tests: fix test
+ Actually do what the comment says. Also keep the old code around, not sure what
+ should happen when you get a 454 from a TEARDOWN, does it close the connection?
+ it currently doesn't.
+
+2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: also watch newly created session
+ When we newly created a session, start watching it immediately instead of
+ on the next request.
+
+2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * tests/check/gst/client.c:
+ tests: add unit test for new-session
+ See https://bugzilla.gnome.org/show_bug.cgi?id=701587
+
+2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: emit new-session when new session is created
+ Only emit new-session when we created a new session for a client, not when a
+ client picked up a previous session.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
+
+2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: handle asterisk as path in requests
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
+
+2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: handle segment query format mismatch
+ It's possible that the segment query returns with a different format than what
+ we asked for, handle this case also.
+
+2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: use segment stop in collect_media_stats
+ Use segment stop instead of duration as range end point.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
+
+2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * tests/check/gst/media.c:
+ rtsp-media: Do not leak the element in take_pipeline
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
+
+2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ rtsp-client: Make configure_client_transport virtual
+ This patch makes configure_client_transport virtual. The functionality is
+ needed to handle some weird clients sending multicast transport settings as url
+ options.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
+
+2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ rtsp-client: Make param_set and param_get virtual
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
+
+2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: convert_range replaces get_range_times
+ get_range_times worked for handling UTC ranges for seeks, but we also
+ need to convert back from NPT to the requested unit in
+ get_range_string. convert_range is now used for both.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
+
+2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-sdp.h:
+ sdp: cleanup sdp info
+ We don't need to pass the proto, we can more easily check a boolean.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
+
+2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ use 0.0.0.0 or :: for c= line instead of server address
+
+2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ use local address, not remote, in SDP
+ See https://bugzilla.gnome.org/show_bug.cgi?id=702063
+
+2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 098c0d7 to 01a7a46
+
+2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: possibility to override range time conversion
+ Make it possible to override the conversion from GstRTSPTimeRange to
+ GstClockTimes, that is done before seeking on the media
+ pipeline. Overriding can be useful for UTC ranges, where the default
+ conversion gives nanoseconds since 1900.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
+
+2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ rtsp-server: Expose the use_client_settings API
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
+
+2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtspstream: handle both ipv4 and ipv6 clients
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
+
+2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
+ This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
+ We already have a way to place extra attributes in the SDP by using a string
+ property with prefix x- or a- in the caps.
+
+2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
+ This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
+ We already have a way to place extra attributes in the SDP, just make a string
+ property in the payloader with a- or x- prefix.
+
+2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ rtsp: place a- and x- properties as attributes
+ application/x-rtp has properties with a- and x- prefixes that should be
+ placed as attributes in the SDP for the media instead of being added to the
+ fmtp.
+
+2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/Makefile.am:
+ * examples/test-video.c:
+ example: add TLS example
+
+2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ server: add support for TLS
+ Add methods to set and get a TLS certificate.
+ Add vmethod to configure a new connection. By default, configure the TLS
+ certificate in a new connection if needed.
+
+2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ server: remove accept_client vmethod
+ This vmethod is not very useful so remove it.
+
+2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: don't crash on NULL GError
+
+2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ rtsp-session-pool: corrected session timeout detection
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
+
+2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: improve debug
+
+2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-server.c:
+ server: refactor connection setup
+ Let the server accept the socket connection and construct a GstRTSPConnection
+ from it. Remove the code from the client and let the client only deal with
+ a fully configure GstRTSPConnection object.
+ We will need this later when the server will configure the connection for
+ TLS.
+
+2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: keep the transport object alive
+ Keep the transport object alive while we have it as qdata on the
+ source.
+
+2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-server.c:
+ rtsp-server: Do not crash on nmapping of server
+ * generate error when gst_rtsp_connection_accept fails
+ * do not stop accepting incoming connections because
+ accepting a client fails
+ https://bugzilla.gnome.org/show_bug.cgi?id=701072
+
+2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
+ https://bugzilla.gnome.org/show_bug.cgi?id=700953
+
+2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ rtsp-sdp: Parse framerate caps field and set SDP attribute
+ The SDP attribute and its format is described in RFC4566.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
+
+2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ rtsp-sdp: Parse width/height from caps and set SDP attribute
+ The SDP attribute and its format is described in RFC6064.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
+
+2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ * tests/check/gst/client.c:
+ rtsp-sdp: add bandwidth line
+ https://bugzilla.gnome.org/show_bug.cgi?id=699220
+
+2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 5edcd85 to 098c0d7
+
+2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * tests/check/gst/media.c:
+ tests: add dynamic payloader prepare/unprepare check
+
+2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: release lock when removing fakesink
+
+2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: set elements to NULL before removing
+ When removing a stream, set the elements to NULL first. This avoids
+ element-is-not-in-NULL-state errors when we dispose the elements.
+
+2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * common:
+ Automatic update of common submodule
+ From 3cb3d3c to 5edcd85
+
+2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: listen to pad-removed signals
+ Listen to the pad-removed signal and remove the stream associated with the
+ removed pad.
+ Add signal to be notified of the removed pad.
+ Remove the fakesink in unprepare()
+ Fix signatures of the signal methods
+
+2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-sdp.c:
+ tests: add example of reusable pipelines
+
+2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: add method to get the srcpad
+
+2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * tests/check/gst/media.c:
+ check: add media prepare/unprepare test
+ See https://bugzilla.gnome.org/show_bug.cgi?id=698376
+
+2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: disconnect from signal handlers in unprepare()
+ We connected to the pad-added and no-more-pads signals in prepare() so
+ we need to disconnect from them in unprepare().
+ See https://bugzilla.gnome.org/show_bug.cgi?id=698376
+
+2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: don't free streams array
+ Don't free the streams array in the unprepare() method, they were not
+ added in prepare().
+ See https://bugzilla.gnome.org/show_bug.cgi?id=698376
+
+2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: don't unref the pipeline in unprepare
+ Unprepare() should undo what prepare() does. Because the pipeline is
+ not created in prepare(), we should not unref it in unprepare()
+
+2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: clear session and caps for reuse
+ Set the session and caps to NULL after unref otherwise we might unref
+ them again later.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=698376
+
+2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: send out teardown signal before tearing down
+ The advantage is that in the signal handler you get direct access to
+ information about what streams are about to get torn down (in the
+ GstRTSPClientState).
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
+
+2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: expose connection
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
+
+2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * common:
+ Automatic update of common submodule
+ From aed87ae to 3cb3d3c
+
+2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-media.h:
+ media: add method to get the base_time of the pipeline
+ Together with a shared clock, this base-time could eventually be sent to
+ the client so that it can reconstruct the exact running-time of the clock
+ on the server.
+
+2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-sdp.c:
+ media: add GstNetTimeProvider support
+ Add a property to let the media provide a GstNetTimeProvider for its clock.
+ Make methods to get the clock and nettimeprovider
+ Add a x-gst-clock property to the SDP with the IP and port number of the nettime
+ provider and also the current time of the clock. This should make it possible
+ for (GStreamer) clients to slave their clock to the server clock.
+
+2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 04c7a1e to aed87ae
+
+2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: wait for buffering to complete
+ Wait for buffering to complete before changing the state to the target state.
+
+2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: small cleanup
+
+2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: remove extra unref in test_setup_non_existing_stream
+ The unref is not needed anymore, teardown runs without it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=696542
+
+2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: GSocketService cleanup in test_bind_already_in_use
+ Use g_socket_service_stop so the rtspserver test stops listening for
+ incoming connections in test_bind_already_in_use.
+ https://bugzilla.gnome.org/show_bug.cgi?id=696541
+
+2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
+ Instead use a GWeakRef which is safe to use
+ This is a known GLib bug, see:
+ https://bugzilla.gnome.org/show_bug.cgi?id=667145
+
+2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * tests/check/gst/media.c:
+ * tests/check/gst/rtspserver.c:
+ rtsp-media/client: Reply to PLAY request with same type of Range
+ Remember the type of Range from the PLAY request and use the same type for
+ the reply.
+
+2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * tests/check/gst/client.c:
+ rtsp-client: expose uri
+
+2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/check/gst/mediafactory.c:
+ tests: Hold ref while creating second media
+ To test if the media aren't shared, make sure we keep the first one while creating a second
+ otherwise the same memory address may be reused.
+
+2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ * configure.ac:
+ configure: remove out-of-date comment
+
+2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ * .gitignore:
+ .gitignore: ignore more build files
+
+2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ * tests/check/Makefile.am:
+ tests: use right _LIBS variable for gst-plugins-base libs
+
+2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/Makefile.am:
+ check: add librtp to libs
+
+2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: Add test to check selecting a port the server will send from
+
+2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: Make sure packets are actually received
+
+2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: Select unicast address from pool if appropriate
+
+2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: Properties are always there in Gst 1.0
+
+2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/check/gst/addresspool.c:
+ tests: Add tests for unicast addresses in pool
+
+2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * tests/check/gst/addresspool.c:
+ address-pool: Verify that multicast addresses are used for multicast and vice-versa
+
+2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-address-pool.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * tests/check/gst/addresspool.c:
+ address-pool: Add unicast addresses
+
+2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * configure.ac:
+ * gst/rtsp-server/rtsp-server.c:
+ * tests/check/gst/rtspserver.c:
+ rtsp-server: Limit the number of threads per server instance
+ If we exceed the maximum, just round robin the clients over the existing
+ threads.
+
+2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ rtsp-server: No need to store the GMainContext in the client context
+
+2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: Add test for client disconnection
+
+2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: Test client and session timeouts with multiple threads
+
+2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-mount-points.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ Document locking and its order
+
+2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: Test that slow DESCRIBE don't block other clients
+
+2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/check/gst/client.c:
+ tests: Add tests for client-requested multicast address
+
+2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ docs: Put the various functions in the right sections
+
+2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * docs/libs/gst-rtsp-server-docs.sgml:
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-address-pool.h:
+ docs: Generate docs for GstRTSPAddressPool
+
+2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ client: Check client provided addresses against the address pool
+
+2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-address-pool.h:
+ * tests/check/gst/addresspool.c:
+ address-pool: Add API to request a specific address from the pool
+ Also add relevant unit tests.
+
+2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/check/gst/mediafactory.c:
+ tests: Check the passing around of a RTSPAddressPool
+ Make sure the RTSPAddressPool is propagated from the MediaFactory all the
+ way down to the stream.
+
+2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * tests/check/gst/addresspool.c:
+ tests: Add more tests for the address pool
+
+2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-address-pool.c:
+ address-pool: Fix off by one error
+ When splitting a port range, the port after a skip is not part of range.
+
+2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ * common:
+ Automatic update of common submodule
+ From 2de221c to 04c7a1e
+
+2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
+
+ * configure.ac:
+ configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
+ AM_CONFIG_HEADER was removed in automake 1.13
+ https://bugzilla.gnome.org/show_bug.cgi?id=693368
+
+2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From a942293 to 2de221c
+
+2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: make sure the watch exists while sending data
+ Protect the send_func with a lock. This allows us to wait for sending
+ to complete before changing the send_func and user_data. We add an
+ extra ref to the watch to make sure that it remains valid during
+ sending.
+ When closing the connection, set the send_func to NULL
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
+
+2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * tests/check/Makefile.am:
+ tests: use GST_*_1_0 environment variables everywhere
+ The _1_0 suffixed environment variables override the
+ non-suffixed ones, so if we're in an environment that
+ sets the _1_0 suffixed ones, such as jhbuild, we need
+ to set those to make sure ours actually always get
+ used.
+
+2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From acb04d9 to a942293
+
+2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: set the client backlog
+ Set the client backlog to a reasonable default
+
+2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Make the element a constructor parameter
+ https://bugzilla.gnome.org/show_bug.cgi?id=689594
+
+2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * docs/libs/Makefile.am:
+ docs: Link with gcov library when gcov is enabled
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
+
+2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: match prepare with unprepare
+ Really unprepare when there were an equal amount of prepare calls.
+
+2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: media has to be unprepared in finalize
+ Because unprepare takes away the last ref on the media.
+
+2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
+ This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
+ We can't use the refcount to trigger unprepare because it is the unprepare call
+ that removes the last refcount after all messages are consumed. What we should
+ probably do is make a prepared refcount and only unprepare when the refcount
+ reaches 0.
+
+2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: let the source unref the last media ref
+ the last ref to the media is held by the source so we don't need to add more ref
+ and unrefs, we simply destroy the media when the source is gone.
+
+2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: improve debug
+
+2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: check state
+ Make sure we are in the right state when collecting the position and duration.
+ Only make ourselves PREPARED when we were previously PREPARING.
+
+2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: use g_object_ref/unref for GObjects
+
+2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
+ Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
+ GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
+ isn't being used anymore.
+
+2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ Fix compiler warning
+
+2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
+
+2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session-media.h:
+ small cleanup
+
+2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * tests/check/gst/media.c:
+ media: avoid element leak
+
+2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: require an element in media constructor
+
+2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ Revert "client: TEARDOWN brings that state to Init again"
+ This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
+ The object is already disposed, there is no point in setting the state.
+
+2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: TEARDOWN brings that state to Init again
+
+2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * examples/test-auth.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media-factory-uri.h:
+ * 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:
+ * gst/rtsp-server/rtsp-mount-points.c:
+ * gst/rtsp-server/rtsp-mount-points.h:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-media.h:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ * tests/check/gst/media.c:
+ rtsp: make object details private
+ Make all object details private
+ Add methods to access private bits
+
+2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/Makefile.am:
+ * tests/check/gst/media.c:
+ tests: add media tests
+
+2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: check if prepared for some methods
+ Check that the media object is prepared before doing seek and getting the
+ current position etc.
+ Add some g_return checks.
+
+2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/Makefile.am:
+ * tests/check/gst/mediafactory.c:
+ tests: add mediafactory test
+
+2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: improve debug
+
+2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: unref pipeline in finalize to avoid leaking it
+
+2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp: use gst_object_unref on GstObjects
+
+2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ media-factory: require an url
+
+2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-uri.c:
+ examples: fix include
+
+2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.h:
+ server: remove unused include
+
+2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/Makefile.am:
+ * tests/check/gst/mountpoints.c:
+ tests: add test for mountpoints
+
+2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix factory leak
+ Keep the factory in the state object only for authorization checks and make
+ sure we unref it on failure. Also don't keep invalid objects in the state
+ object.
+
+2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-mount-points.c:
+ mounts: add g_return_if guards
+
+2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/gst/client.c:
+ tests: add more tests
+
+2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: improve debug
+
+2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: improve debug and fix leaks
+ Cleanup the uri and session when there is a bad request.
+
+2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * common:
+ update common
+
+2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/gst/client.c:
+ test: add test for session in options request
+
+2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: use 454 when session can't be found
+ We should use 454 when a session can't be found because there was no session
+ pool configured in the server. This is not a server configuration problem
+ because the server on which the request is done might not be the same one that
+ will keep the sessions for us and so it does not need to support sessions.
+
+2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: only free connection when there is one
+ It's possible that the client doesn't have a connection when we try to free it.
+
+2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/Makefile.am:
+ * tests/check/gst/client.c:
+ tests: add unit test for the client object
+
+2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: small cleanup
+
+2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.h:
+ client: remove unused include
+
+2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix compilation
+
+2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: call destroy without the lock
+
+2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: make the client usable without a socket
+ Make a method to let the client handle a message and a callback when the client
+ wants us to send a response message back. This makes it possible to also use the
+ client object without the sockets, which should make it easier to test.
+
+2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: small cleanup
+
+2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-server.c:
+ client: remove reference to server
+ We don't need to keep a ref to the server
+
+2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: add locking
+ Also add some g_return_if()
+
+2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: log more errors
+
+2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix compilation
+
+2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: add generic close-after-send support
+ Add a property to send_response() to close the connection after the response has
+ been sent to the client.
+
+2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/README:
+ * docs/libs/gst-rtsp-server-docs.sgml:
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * docs/libs/gst-rtsp-server.types:
+ * examples/test-auth.c:
+ * examples/test-launch.c:
+ * examples/test-mp4.c:
+ * examples/test-multicast.c:
+ * examples/test-multicast2.c:
+ * examples/test-ogg.c:
+ * examples/test-readme.c:
+ * examples/test-sdp.c:
+ * examples/test-uri.c:
+ * examples/test-video.c:
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media-mapping.c:
+ * gst/rtsp-server/rtsp-media-mapping.h:
+ * gst/rtsp-server/rtsp-mount-points.c:
+ * gst/rtsp-server/rtsp-mount-points.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * tests/check/gst/rtspserver.c:
+ MediaMapping -> MountPoints
+ Describes better what the object manages.
+
+2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ configure: bump required version of -base
+
+2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: fix seeking
+
+2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: support more Range formats
+ Use the new -base methods to convert the Range string into a seek start and stop
+ value.
+
+2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-launch.c:
+ examples: fix whitespace
+
+2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-auth.c:
+ test-auth: add example of how to remove sessions
+ Add an example of the session filter api.
+
+2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-uri.c:
+ test-uri: remove mapping example
+
+2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-uri.c:
+ test-uri: fix callback signature
+
+2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ factory: keep ref to factory while media active
+ While the media from a factory is alive, keep a ref to the factory.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
+
+2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ factory-uri: add some debug
+
+2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: set udp sources to PLAYING
+ Set the UDP sources to PLAYING and locked state before we add it to the pipeline
+ so that it doesn't cause our pipeline to produce ASYNC-DONE.
+
+2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ factory-uri: take ref to factory
+ Take a ref to the factory that we place in our list.
+
+2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/Makefile.am:
+ * tests/test-reuse.c:
+ test: add test for server reuse
+ See https://bugzilla.gnome.org/show_bug.cgi?id=688395
+
+2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: start and stop multiple times
+ Stop listening on the RTSP port when the GSource is removed, so clients
+ can't connect and the server can be started again.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
+
+2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: fix small leak
+
+2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: unref source in finish_unprepare
+ The source is created in prepare, unref it in finish_unprepare.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=688707
+
+2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: remove bus watch before finalizing
+ * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
+ * An extra media ref is added for the bus watch. This extra ref is unreffed by
+ the GDestroyNotify function.
+ * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
+ * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
+ gst_rtsp_media_unprepare before unreffing the media.
+ This way, the bus watch will be removed before the media is finalized.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
+
+2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: wait until the TEARDOWN response is sent to close the connection
+ Responses can be sent async so we need to wait until the TEARDOWN response has
+ been written before we close the connection to the client. This avoids the risk
+ of writing/polling closed sockets.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
+
+2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: plug socket leak
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
+
+2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ * common:
+ Automatic update of common submodule
+ From 6bb6951 to a72faea
+
+2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ rtsp-server: don't use deprecated API
+
+2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: fix unused-but-set-variable compiler warning
+ rtsp-client.c:1260:21: error: variable 'protocols' set but not used
+
+2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * TODO:
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp: cleanups
+
+2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/Makefile.am:
+ * examples/test-multicast2.c:
+ examples: add another multicast example
+ Add an example for how to configure separate multicast ranges for each media
+ stream.
+
+2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-multicast.c:
+ test: set shared
+
+2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-media.h:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ stream: use the address managed by the stream
+ Use the address managed by the stream for multicast. This allows us to have 1
+ multicast address for each stream.
+ Because the address is now managed by the stream we don't have to pass it around
+ anymore.
+ Set the address pool on the streams.
+
+2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp: improve debug
+
+2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: add signal for new streams
+ This allows applications to listen for new streams and configure properties on
+ them, like the address pool.
+
+2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: configure address pool in new streams
+
+2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: add methods to deal with address pool
+ Add methods to get and set the address pool for the stream
+ Add method to allocate and get the multicast addresses for this stream.
+
+2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: remove MTU property
+ It is a stream property
+
+2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: set blocksize only on stream
+ Set the blocksize only on the current stream.
+
+2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: share src and sink sockets
+ the allocated socket is in the used-socket property, not socket.
+
+2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-address-pool.h:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-media.h:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * tests/check/gst/addresspool.c:
+ rtsp: make address-pool return an address object
+ Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
+ store more info in the structure and allows us to more easily return the address
+ to the right pool when no longer needed.
+ Pass the address to the StreamTransport so that we can return it to the pool
+ when the stream transport is freed or changed.
+
+2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/Makefile.am:
+ * examples/test-multicast.c:
+ examples: add multicast example
+ Show how to set up the multicast address pool so that media can be
+ server with multicast.
+
+2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * 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: use AddressPool
+ Remove the multicast_group property.
+ Use the configured addresspool to allocate multicast addresses.
+
+2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-address-pool.h:
+ address-pool: add clear method
+
+2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-address-pool.c:
+ address-pool: small cleanups
+
+2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/Makefile.am:
+ * tests/check/gst/addresspool.c:
+ tests: add addresspool unit test
+
+2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-address-pool.h:
+ address-pool: add object to manage multicast addresses
+ Make an object that can manage a rage of multicast addresses and ports.
+
+2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: set default max-threads property
+
+2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: wait for concurrent _prepare
+ If a prepare is busy, wait for the result.
+
+2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: add lock around message handler
+ We don't want to dispatch messages while we are still processing the result of
+ the state change.
+
+2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: add lock to protect state changes
+
+2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: add locking
+
+2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ stream-transport: add keep-alive method
+
+2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ stream-transport: add method to handle RTP/RTCP
+ Call new methods instead of poking into the structures directly.
+
+2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-media.h:
+ session-media: add locking
+
+2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ session: add locking
+
+2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: free old socket
+
+2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-mapping.c:
+ * gst/rtsp-server/rtsp-media-mapping.h:
+ mapping: add locking
+
+2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ media-factory: add locking
+
+2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ auth: add locking
+
+2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ server: add max-thread property
+
+2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ server: use a threadpool for the mainloops
+
+2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: rename method
+ gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
+ don't really create the client from the socket, we use the socket for the
+ client.
+
+2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-server.c:
+ server: rework maincontext handling in clients
+ Make a separate method to attach a client to a MainContext.
+ Let the server decide in what GMainContext the client will operate and give this
+ context to the client in attach. Then the server can later decide to use a
+ separate thread for each client or just use the mainthread.
+
+2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ session: move session header code in session object
+
+2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ * COPYING:
+ * COPYING.LIB:
+ * examples/test-auth.c:
+ * examples/test-launch.c:
+ * examples/test-mp4.c:
+ * examples/test-ogg.c:
+ * examples/test-readme.c:
+ * examples/test-sdp.c:
+ * examples/test-uri.c:
+ * examples/test-video.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media-factory-uri.h:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media-mapping.c:
+ * gst/rtsp-server/rtsp-media-mapping.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-params.c:
+ * gst/rtsp-server/rtsp-params.h:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-sdp.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-media.h:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ * tests/check/gst/rtspserver.c:
+ * tests/test-cleanup.c:
+ Fix FSF address
+
+2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session.c:
+ rtsp-server: added annotations to indicate type of ownership transfer of return values
+ https://bugzilla.gnome.org/show_bug.cgi?id=680777
+
+2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ * configure.ac:
+ No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
+
+2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
+
+ * Makefile.am:
+ * bindings/Makefile.am:
+ * bindings/vala/Makefile.am:
+ * bindings/vala/gst-rtsp-server-0.10.deps:
+ * bindings/vala/gst-rtsp-server-0.10.vapi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.deps:
+ * bindings/vala/packages/gst-rtsp-server-0.10.files:
+ * bindings/vala/packages/gst-rtsp-server-0.10.gi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
+ * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
+ * configure.ac:
+ bindings: remove vala bindings
+ They'll be reunited with the other GStreamer bindings
+ https://bugzilla.gnome.org/show_bug.cgi?id=680777
+
+2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-media.h:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ rtsp: only create transport when needed
+ Only create the StreamTransport when configured.
+
+2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: small cleanup
+
+2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ rtsp: refactor configuration of transport
+ Move the configuration of the transport to a place where it makes
+ more sense.
+
+2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: refactor transport parsing
+
+2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: refuse to change the MTU on shared media
+ If we change the MTU of chared media, it changes for all clients.
+ We don't want to set the MTU to something large for clients that
+ stream over UDP.
+
+2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-mp4.c:
+ * gst/rtsp-server/rtsp-media.c:
+ small fixes to docs and debug
+
+2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: transports must already have been removed
+
+2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: improve join and leave of the pipeline
+ simplify code
+ Do the cleanup properly
+ Add some docs
+
+2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: move unprepare below default implementation
+ Makes it easier to find the default implementation
+
+2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: signal unprepared when we actually finish
+
+2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: no need to unlock, unprepare does that when needed
+
+2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media-mapping.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-params.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.h:
+ docs: update docs
+
+2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-mapping.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp: fix MTU setting
+ Fix setting of the MTU. There is no need for a vmethod.
+
+2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/README:
+ docs: update docs
+
+2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ configure: bump version number after refactoring
+
+2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media-factory-uri.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:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-media.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp: massive refactoring
+ Make GObjects from the remaining simple structures.
+ Remove GstRTSPSessionStream, it's not needed.
+ Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
+ Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
+ a GstRTSPStream should be transported to a client.
+ Rename GstRTSPMediaFactory::get_element -> create_element because that
+ more accurately describes what it does.
+ Make nice methods instead of poking in the structures.
+ Move some methods inside the relevant object source code.
+ Use GPtrArray to store objects instead of plain arrays, it is more
+ natural and allows us to more easily clean up.
+ Move the allocation of udp ports to the Stream object. The Stream object
+ contains the elements needed to stream the media to a client.
+ Improve the prepare and unprepare methods. Unprepare should now undo
+ everything prepare did. Improve also async unprepare when doing EOS on
+ shutdown. Make sure we always unprepare correctly.
+
+2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Unref server address clients connected to
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
+
+2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ rtsp-server: don't ref server socket if it is NULL
+ Fixes test_bind_already_in_use unit test again after commit 6a497440.
+ https://bugzilla.gnome.org/show_bug.cgi?id=686644
+
+2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
+
+ * tests/check/Makefile.am:
+ tests: Add libgio link dependency
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
+
+2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-media-mapping.c:
+ * gst/rtsp-server/rtsp-media-mapping.h:
+ rtsp-media-mapping: rename find_media vfunc to find_factory
+ The virtual method and class method should have the same name
+ so it is correctly represented in GIR file
+ https://bugzilla.gnome.org/show_bug.cgi?id=680777
+
+2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-mapping.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ rtsp-server: fixed comments and GIR annotations
+ https://bugzilla.gnome.org/show_bug.cgi?id=680777
+
+2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/rtsp-server/rtsp-media-mapping.c:
+ media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
+
+2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ rtsp-server: allow binding on port 0 (binds on a random port)
+
+2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ rtsp-server: add bound-port property
+ bound-port can be used to retrieve the port number when the server is bound on
+ port 0, which binds on a random port.
+
+2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ rtsp-media-factory: make ::get_element overridable by GI bindings
+ The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
+ for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
+ as the invoker for ::get_element(), making it overridable by GI generated
+ bindings.
+
+2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ rtsp-media-factory-uri: don't autoplug parsers in a loop
+ Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
+ h264parse forever.
+
+2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/rtsp-server/Makefile.am:
+ Explicitly link against gio. Fix link error on mac.
+
+2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
+
+ * gst/rtsp-server/rtsp-session.c:
+ session: add ttl to the transport header in SETUP
+ See https://bugzilla.gnome.org/show_bug.cgi?id=685561
+
+2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media.c:
+ client: Use client transport settings for multicast if allowed.
+ This patch makes it possible for the client to send transport settings for
+ multicast (destination && ttl). Client settings must be explicitly allowed or
+ the server will use its own settings.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
+
+2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * common:
+ Automatic update of common submodule
+ From 6c0b52c to 6bb6951
+
+2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: do not destroy the rtsp watch
+ Don't destroy the client watch while dispatching. The rtsp watch is
+ automatically destroyed after the rtsp watch function closed() has
+ been called.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
+
+2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * common:
+ Automatic update of common submodule
+ From 4f962f7 to 6c0b52c
+
+2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: fix check for seekability
+
+2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: use more GIO
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
+
+2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: remove obsolete includes
+
+2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
+
+ rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
+ * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
+ be available in "on_new_ssrc". The transports are added in
+ gst_rtsp_media_set_state when going to PLAYING state. However,
+ "on_new_ssrc" might be called before this happens.
+ https://bugzilla.gnome.org/show_bug.cgi?id=683304
+
+2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ rtsp-client: add signals for rtsp requests (fixes #683287)
+
+2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ add new-session signal to rtsp-client (fixes #683058)
+
+2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 668acee to 4f962f7
+
+2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * tests/check/gst/rtspserver.c:
+ rtsp-server: fixed segfault in gst_rtsp_server_create_socket
+ Do not assume that *error is set in g_socket_address_enumerator_next.
+ Added test_bind_already_in_use unit-test.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
+
+2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * common:
+ Automatic update of common submodule
+ From 94ccf4c to 668acee
+
+2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ rtsp-client: make create_sdp virtual method
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
+
+2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 98e386f to 94ccf4c
+
+2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix docs
+
+2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ rtsp-server: use an existing socket to establish HTTP tunnel
+ Make it possible to transfer a socket from an HTTP server to be used as
+ an RTSP over HTTP tunnel.
+
+2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ rtsp: Handle the blocksize parameter
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
+
+2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/gst/rtspserver.c:
+ Have unit test get header from source dir, not installed dir
+ This makes compilation of unit tests work in a build directory other
+ than the source directory.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
+
+2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: update for gst_element_make_from_uri() changes
+
+2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * configure.ac:
+ * tests/Makefile.am:
+ * tests/check/Makefile.am:
+ * tests/check/gst/rtspserver.c:
+ rtsp: add unit test
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
+
+2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: don't collect media stats when going to NULL
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
+
+2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: don't leak transports
+
+2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: free transport on no_stream in SETUP handler
+
+2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: changed session media iteration
+ In client_unlink_session: now don't iterate in session->medias
+ list where items are removed by gst_rtsp_session_release_media.
+ Instead, repeatedly remove the first item.
+
+2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
+ GstRTSPSessionMedia is not a GObject type. When the
+ GstRTSPSession is freed, it will free the media.
+
+2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ factory: plug pad leak in collect_streams
+ In gst_rtsp_media_factory_collect_streams: unref the srcpad that
+ was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
+ will take one reference, and the other reference will otherwise
+ give a memory leak.
+
+2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
+
+ * configure.ac:
+ configure: suppress some warnings when debug is disabled
+ Warnings about unused variables should be suppressed if core has the
+ debug system disabled.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
+
+2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * docs/libs/Makefile.am:
+ docs: fix build in uninstalled setup
+ Include gst-plugins-base libs properly.
+
+2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
+
+ * docs/libs/gst-rtsp-server.types:
+ docs: include headers defining rtsp-server object types
+ Fixes compiler warnings during docs build.
+ https://bugzilla.gnome.org/show_bug.cgi?id=676824
+
+2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
+
+ * configure.ac:
+ configure: Add warning flags for compiler when configuring
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
+
+2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 03a0e57 to 98e386f
+
+2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 1fab359 to 03a0e57
+
+2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix GSocketAddress leak in gst_rtsp_client_accept
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
+
+2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From f1b5a96 to 1fab359
+
+2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 92b7266 to f1b5a96
+
+2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From ec1c4a8 to 92b7266
+
+2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 3429ba6 to ec1c4a8
+
+2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-server.c:
+ rtsp: fix compiler warnings
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
+
+2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From dc70203 to 3429ba6
+
+2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * 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:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ rtsp-server: port to new thread API
+
+2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 6db25be to dc70203
+
+2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-server: Fix compilation and compiler warnings
+
+2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * autogen.sh:
+ * configure.ac:
+ * gst/rtsp-server/Makefile.am:
+ configure: Modernize autotools setup a bit
+ Also we now only create tar.bz2 and tar.xz tarballs.
+
+2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 464fe15 to 6db25be
+
+2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 7fda524 to 464fe15
+
+2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * configure.ac:
+ * docs/libs/Makefile.am:
+ * docs/version.entities.in:
+ * gst-rtsp.spec.in:
+ * gst/rtsp-server/Makefile.am:
+ * pkgconfig/Makefile.am:
+ * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
+ * pkgconfig/gstreamer-rtsp-server.pc.in:
+ * tests/Makefile.am:
+ rtsp-server: Update versioning
+
+2012-03-29 15:12:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ Merge remote-tracking branch 'origin/0.10'
+ Conflicts:
+ gst/rtsp-server/rtsp-session-pool.c
+
+2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ rtsp-server: Don't use deprecated GLib API
+
+2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Replace master with 0.11
+
+2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * docs/README:
+ A couple minor typo fixes
+
+2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: fix state of the appqueue
+
+2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ factory: use videoconvert
+
+2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ factory: change to new style caps
+
+2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ rtsp-server: port to GIO
+ Port to GIO
+
+2012-03-07 15:03:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ configure: fix build
+
+2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * docs/README:
+ docs: fix for gst_rtsp_server_set_port() -> _set_service()
+ https://bugzilla.gnome.org/show_bug.cgi?id=666548
+
+2012-02-13 11:42:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ * examples/Makefile.am:
+ First rule of gst-rtsp-server club: don't talk about gst-phonon
+
+2012-02-13 11:40:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ * pkgconfig/Makefile.am:
+ * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
+ * pkgconfig/gstreamer-rtsp-server.pc.in:
+ pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc
+ For consistency with all other modules.
+
+2012-02-13 11:06:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: update for new map API
+
+2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * .gitignore:
+ * bindings/Makefile.am:
+ * bindings/python/Makefile.am:
+ * bindings/python/arg-types.py:
+ * bindings/python/codegen/Makefile.am:
+ * bindings/python/codegen/__init__.py:
+ * bindings/python/codegen/argtypes.py:
+ * bindings/python/codegen/code-coverage.py:
+ * bindings/python/codegen/codegen.py:
+ * bindings/python/codegen/definitions.py:
+ * bindings/python/codegen/defsparser.py:
+ * bindings/python/codegen/docextract.py:
+ * bindings/python/codegen/docgen.py:
+ * bindings/python/codegen/fileprefix.override:
+ * bindings/python/codegen/fileprefixmodule.c:
+ * bindings/python/codegen/h2def.py:
+ * bindings/python/codegen/mergedefs.py:
+ * bindings/python/codegen/mkskel.py:
+ * bindings/python/codegen/override.py:
+ * bindings/python/codegen/reversewrapper.py:
+ * bindings/python/codegen/scmexpr.py:
+ * bindings/python/rtspserver-types.defs:
+ * bindings/python/rtspserver.defs:
+ * bindings/python/rtspserver.override:
+ * bindings/python/rtspservermodule.c:
+ * bindings/python/test.py:
+ * configure.ac:
+ python: remove pygst-based python bindings
+ pygi is the future, apparently.
+
+2012-01-25 14:12:41 +0100 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+
+ * common:
+ Automatic update of common submodule
+ From c463bc0 to 7fda524
+
+2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 2a59016 to c463bc0
+
+2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 0807187 to 2a59016
+
+2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 11f0cd5 to 0807187
+
+2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-auth.c:
+ example: update for new caps
+
+2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-video.c:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ rtsp-server: port some more to 0.11
+ Fix caps.
+ Remove bufferlist stuff
+ Update for new API.
+ Add queue before appsink now that preroll-queue-len is gone.
+ Update for request pad changes.
+
+2011-11-03 16:14:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
+
+ * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
+ bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
+ Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
+
+2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
+
+ * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
+ bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership.
+ Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
+
+2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: add a seekable boolean
+ Maintain the seekable state with a new variable instead of reusing the
+ is_live variable.
+
+2011-09-16 11:31:17 -0400 Victor Gottardi <vgottardi@hotmail.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ Disallow seek in live media
+
+2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
+
+ * gst/rtsp-server/rtsp-server.c:
+ #ifdef statements for windows socket creation were missing
+
+2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From a39eb83 to 11f0cd5
+
+2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 605cd9a to a39eb83
+
+2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: use method to access property
+
+2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ media-factory: add protocols property
+ Add a property to configure the allowed protocols in the media created from the
+ factory.
+
+2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ media-factory: add media-configure signal
+ Add signal to allow the application to configure the media after it was created
+ from the factory.
+
+2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: use method to access property
+
+2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ media-factory: add protocols property
+ Add a property to configure the allowed protocols in the media created from the
+ factory.
+
+2011-08-16 15:03:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ media-factory: add media-configure signal
+ Add signal to allow the application to configure the media after it was created
+ from the factory.
+
+2011-08-16 14:50:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: use media multicast group
+
+2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.h:
+ retab some .h
+
+2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-sdp.h:
+ sdp: copy and free the server ip address
+ Copy and free the server ip address to make memory management easier later.
+
+2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ media-factory: configure multicast in media
+
+2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: add property for multicast group
+ Add a property to configure the multicast group in the media.
+ Based on patches from Marc Leeman and Robert Krakora.
+
+2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ media-factory: add property for multicast group
+ Add a property to configure the multicast group in the media factory.
+ Based on patches from Marc Leeman and Robert Krakora.
+
+2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: do configuration of transport in one place
+ Move the configuration of the transport destination address to where we also
+ configure the other bits.
+
+2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: use media multicast group
+
+2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.h:
+ retab some .h
+
+2011-08-16 13:31:52 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-sdp.h:
+ sdp: copy and free the server ip address
+ Copy and free the server ip address to make memory management easier later.
+
+2011-08-16 13:27:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ media-factory: configure multicast in media
+
+2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: add property for multicast group
+ Add a property to configure the multicast group in the media.
+ Based on patches from Marc Leeman and Robert Krakora.
+
+2011-08-16 13:13:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ media-factory: add property for multicast group
+ Add a property to configure the multicast group in the media factory.
+ Based on patches from Marc Leeman and Robert Krakora.
+
+2011-08-16 12:51:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: do configuration of transport in one place
+ Move the configuration of the transport destination address to where we also
+ configure the other bits.
+
+2011-08-16 12:11:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: destroy pipeline on client disconnect with no prior TEARDOWN.
+ The problem occurs when the client abruptly closes the connection without
+ issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
+ server is where the pipeline gets torn down. Since this handler is not called,
+ the pipeline remains and is up and running. Subsequent clients get their own
+ pipelines and if the do not issue TEARDOWNs then those pipelines will also
+ remain up and running. This is a resource leak.
+
+2011-08-16 11:53:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ media-factory: add a "media-constructed" signal to GstRTSPMediaFactory
+ For example, it can be used to retrieve source elements like appsrc, in a more
+ convenient way than subclassing get_element.
+
+2011-08-16 11:12:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
+
+ * gst/rtsp-server/rtsp-server.c:
+ rtsp-server: hold on to reference while using object
+
+2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: use new api
+
+2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ configure: use unstable api
+
+2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix reference counting
+
+2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ fix compiler warnings about unused variables
+
+2011-07-19 16:10:39 +0200 Stefan Sauer <ensonic@google.com>
+
+ * examples/test-launch.c:
+ * examples/test-readme.c:
+ * examples/test-uri.c:
+ * examples/test-video.c:
+ examples: tell rtsp uri when ready
+
+2011-06-23 11:30:14 -0700 David Schleef <ds@schleef.org>
+
+ * common:
+ Automatic update of common submodule
+ From 69b981f to 605cd9a
+
+2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: update for buffer API change
+
+2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ Makefile.am: 0.10 => @GST_MAJORMINOR@
+
+2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer
+
+2011-06-07 10:59:03 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * gst/rtsp-server/.gitignore:
+ .gitignore: 0.10 => 0.11
+
+2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ Makefile.am: 0.10 => @GST_MAJORMINOR@
+
+2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 9e5bbd5 to 69b981f
+
+2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From fd35073 to 9e5bbd5
+
+2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 46dfcea to fd35073
+
+2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media.c:
+ media: port to new caps API
+
+2011-05-17 09:45:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+
+2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
+
+ * bindings/vala/gst-rtsp-server-0.10.vapi:
+ Updated Vala bindings.
+ Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
+
+2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ Add a signal for newly connected clients.
+ Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
+
+2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * bindings/python/rtspserver.override:
+ python: override gst_rtsp_media_mapping_add_factory to fix refcounting
+
+2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-funnel.c:
+ * gst/rtsp-server/rtsp-funnel.h:
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-server: port to 0.11
+
+2011-04-26 19:14:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * common:
+ add common
+
+2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+ Conflicts:
+ common
+ configure.ac
+
+2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From c3cafe1 to 46dfcea
+
+2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * bindings/python/Makefile.am:
+ * bindings/python/rtspserver.defs:
+ python bindings: wrap GstRTSPMediaFactoryClass vfuncs
+
+2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * bindings/python/arg-types.py:
+ python bindings: add GstRTSPUrlParam
+ Needed to implement MediaFactory virtual proxies
+
+2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * bindings/python/arg-types.py:
+ python bindings: fix returning GstRTSPUrl types
+
+2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * bindings/python/arg-types.py:
+ python bindings: add arg type for GstRTSPUrl
+
+2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * bindings/python/rtspserver.defs:
+ python bindings: fix the definition of MediaFactory.collect_stream
+
+2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 1ccbe09 to c3cafe1
+
+2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 193b717 to 1ccbe09
+
+2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From b77e2bf to 193b717
+
+2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * Makefile.am:
+ build: Include lcov.mak to allow test coverage report generation
+
+2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From d8814b6 to b77e2bf
+
+2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 6aaa286 to d8814b6
+
+2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 6aec6b9 to 6aaa286
+
+2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * autogen.sh:
+ autogen: wingo signed comment
+
+2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ session: use full charset for RTSP session ID
+ As specified in RFC 2326 section 3.4 use full valid charset to make guessing
+ session ID more difficult.
+ https://bugzilla.gnome.org/show_bug.cgi?id=643812
+
+2011-03-07 10:23:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ rtsp-server: Don't install the funnel header
+
+2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 1de7f6a to 6aec6b9
+
+2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ configure: require core/base 0.10.31
+ Needed at least for gst_plugin_feature_rank_compare_func().
+
+2011-02-14 12:56:29 +0200 Stefan Kost <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From f94d739 to 1de7f6a
+
+2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: remove more unused code
+
+2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: remove duplicate filtering
+ Remove the duplicate filtering code now that we have a released -good version.
+ Give a warning instead.
+
+2011-01-31 17:38:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ media: fix default buffer size
+
+2011-01-31 17:37:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ media-factory: add property to configure the buffer-size
+ Add a property to configure the kernel UDP buffer size.
+
+2011-01-31 17:28:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: add property to configure kernel buffer sizes
+ Add a property to configure the kernel UDP buffer size.
+
+2011-01-26 15:52:54 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ configure: set PYGOBJECT_REQ before using it
+ https://bugzilla.gnome.org/show_bug.cgi?id=640641
+
+2011-01-24 11:59:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * docs/Makefile.am:
+ docs: recursive into sub-directories on 'make upload'
+
+2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-docs.sgml:
+ * docs/version.entities.in:
+ docs: mention full version these docs are for, not just major-minor
+
+2011-01-24 12:07:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ back to development
+
+=== release 0.10.8 ===
+
+2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ release 0.10.8
+
+2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ rtsp-server: clarify docs a little
+
+2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: init debug category before starting thread
+
+2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ auth: add realm to make it more spec compliant
+
+2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ server: add locking
+
+2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-video.c:
+ example: improve example docs a little
+
+2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: ensure the watch has a ref to the server
+
+2011-01-12 18:24:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: simpify channel function
+
+2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ server: simplify management of channel and source
+ We don't need to keep around the channel and source objects. Let the mainloop
+ and the source manage the source and channel respectively.
+
+2011-01-12 18:17:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * Makefile.am:
+ * configure.ac:
+ build tests
+
+2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/.gitignore:
+ * tests/Makefile.am:
+ * tests/test-cleanup.c:
+ tests: add tests directory and cleanup test
+
+2011-01-12 18:14:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-mapping.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ server: improve debugging in various objects
+
+2011-01-12 16:38:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: chain up to the parent finalize
+
+2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
+
+ * bindings/python/rtspserver-types.defs:
+ * bindings/python/rtspserver.defs:
+ * bindings/python/rtspserver.override:
+ * bindings/python/test.py:
+ gst-rtsp-server: update python bindings
+
+2011-01-12 15:37:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: use the response from the clientstate
+ Create the response object only once and store in the client state.
+ Make all methods use the state response,
+
+2011-01-12 15:36:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: use signal to keep track of clients
+ Keep track of all the clients that the server creates and remove them when they
+ fire the 'closed' signal.
+
+2011-01-12 15:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: emit signal when closing
+
+2011-01-12 13:57:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/.gitignore:
+ * examples/Makefile.am:
+ * examples/test-auth.c:
+ * examples/test-video.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.h:
+ media: enable per factory authorisations
+ Allow for adding a GstRTSPAuth on the factory and media level and check
+ permissions when accessing the factory.
+ Add hints to the auth methods for future more fine grained authorisation.
+ Add example application for per factory authentication.
+
+2011-01-12 13:16:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-params.c:
+ * gst/rtsp-server/rtsp-params.h:
+ rtsp-server: Pass ClientState structure arround
+ Pass the collected information for the ongoing request in a GstRTSPClientState
+ structure that we can then pass around to simplify the method arguments. This
+ will also be handy when we implement logging functionality.
+
+2011-01-12 12:07:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ media-factory: add methods to configure authorisation
+
+2011-01-12 12:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: unref auth in finalize
+
+2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: unref auth in finalize
+
+2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-docs.sgml:
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * docs/libs/gst-rtsp-server.types:
+ docs: add more docs
+
+2011-01-12 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ server: separate create and accept
+ Create separate create and accept methods so that subclasses can create custom
+ client object.
+ Configure the server in the client object and prepare for keeping track of
+ connected clients.
+
+2011-01-12 10:42:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: add support for setting the server.
+ Add support for keeping a ref to the server that started this client
+ connection.
+
+2011-01-12 10:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ auth: fix memleak and add some docs
+ Fix a memleak of the basic auth token.
+ Add docs for the helper function
+
+2011-01-12 00:35:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ client: delegate setup of auth to the manager
+ Delegate the configuration of the authentication tokens to the manager object
+ when configured.
+
+2011-01-12 00:17:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-video.c:
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ auth: add authentication object
+ Add an object that can check the authorization of requests.
+ Implement basic authentication.
+ Add example authentication to test-video
+
+2011-01-12 00:20:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ server: move includes back
+ the includes are needed for sockaddr_in.
+
+2011-01-11 22:41:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ rtsp: move network includes where they are needed
+
+2011-01-07 23:45:32 +0200 Sreerenj Balachandran <sreerenj.balachandran@nokia.com>
+
+ * gst/rtsp-server/rtsp-media.h:
+ rtsp-media.h: Minor corrections in comments.
+ Fixes #638944
+
+2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From e572c87 to f94d739
+
+2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * .gitignore:
+ * docs/.gitignore:
+ * docs/libs/.gitignore:
+ * examples/.gitignore:
+ * gst/rtsp-server/.gitignore:
+ gitignore: updates
+
+2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * docs/libs/Makefile.am:
+ docs: We don't build ps/pdf for API reference docs
+
+2011-01-10 16:39:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From ccbaa85 to e572c87
+
+2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 46445ad to ccbaa85
+
+2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-funnel.c:
+ * gst/rtsp-server/rtsp-funnel.h:
+ * gst/rtsp-server/rtsp-media.c:
+ funnel: rename fsfunnel to rtspfunnel
+ Rename the funnel to avoid conflicts with the farsight one.
+
+2011-01-10 13:41:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/fs-funnel.c:
+ * gst/rtsp-server/fs-funnel.h:
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: add and use fsfunnel
+ Add a copy of fsfunnel to the build because input-selector removed the (broken)
+ select-all property that we need.
+
+2011-01-08 01:58:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ gobject-introspection: use PKG_CONFIG_PATH specified at configure time
+ Use PKG_CONFIG_PATH specified at configure time (if any) as well
+ for the g-ir-compiler, rather than just assuming the env var has
+ been set.
+
+2011-01-08 01:55:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * .gitignore:
+ * Makefile.am:
+ * configure.ac:
+ * m4/Makefile.am:
+ * m4/codeset.m4:
+ build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4
+
+2011-01-08 01:15:35 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ * gst/rtsp-server/Makefile.am:
+ gobject-introspection: fix g-i build for uninstalled setup
+ Requires gst-plugins-base git (> 0.10.31.2).
+
+2011-01-07 11:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-uri.c:
+ examples: add some more options and comments
+
+2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ factory-uri: use right property type
+
+2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ factory-uri: attempt to configure buffer-lists
+ Attempt to configure buffer lists in the payloader for improved performance.
+
+2011-01-05 12:06:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: attempt to configure bigger UDP buffers
+ Attempt to configure bigger udp kernel send buffers to avoid overflowing the
+ send buffers with high bitrate streams.
+
+2011-01-05 11:26:30 +0100 Jonas Larsson <jonas at hallerud dot se>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: use the socket length from getsockname
+ Use the length returned by getsockname to perform the getnameinfo call because
+ the size can depend on the socket type and platform.
+ Fixes #638723
+
+2010-12-30 12:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/libs/gst-rtsp-server-docs.sgml:
+ * docs/libs/gst-rtsp-server-sections.txt:
+ docs: add uri factory to the docs
+
+2010-12-30 12:41:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.h:
+ docs: improve docs
+
+2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ rtsp-server: add support for buffer lists
+ Add support for sending bufferlists received from appsink.
+ Fixes #635832
+
+2010-12-28 18:35:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-sdp.c:
+ media: make method to retrieve the play range
+ Make a method to retrieve the playback range so that we can conditionally create
+ a different range for the SDP and the PLAY requests.
+
+2010-12-28 18:34:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: add signal to notify of state changes
+
+2010-12-28 18:31:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.h:
+ client: cleanup headers
+
+2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix typo
+
+2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media-factory-uri.h:
+ factory-uri: add support for gstpay
+ Add an option to prefer gstpay over decoder + raw payloader.
+
+2010-12-23 15:58:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media-factory-uri.h:
+ factory-uri: rework the autoplugger.
+ Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
+ before payloaders.
+
+2010-12-21 17:37:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ factory-uri: use better factory filter
+ Make better payloader filter based on autoplug rank and RTP use case.
+
+2010-12-20 17:48:41 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 169462a to 46445ad
+
+2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: set SO_REUSEADDR before bind
+ Set the SO_REUSEADDR _before_ bind() to make it actually work.
+
+2010-12-13 16:58:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: emit prepared signal when prepared
+ Make a 'prepared' signal and emit it when we successfully prepared the element.
+ This signal can be used to configure the media object after it has been prepared
+ for streaming.
+
+2010-12-15 14:58:00 +0200 Stefan Kost <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 011bcc8 to 169462a
+
+2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
+
+ python an optional dependency
+ * configure.ac: Move up valgrind and g-i checks. Make the python
+ dependency optional, as it was before.
+
+2010-12-13 11:43:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' into 0.11
+ Conflicts:
+ common
+ configure.ac
+
+2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: update range when active clients changed
+ When we changed the number of active clients, update the current range
+ information because we want the second client connecting to a shared resource
+ continue from where the stream currently.
+
+2010-12-12 04:06:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media-factory-uri.h:
+ factory-uri: add colorspace and fix pt
+ Rework the way we pass data to the autoplugger.
+ When we have raw caps, plug a converter element to make pluggin to raw
+ payloaders more successful.
+ Make sure all dynamically plugged payloaders have a unique payload types.
+
+2010-12-11 18:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/Makefile.am:
+ * examples/test-uri.c:
+ example: add example of the uri factory
+
+2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ * gst/rtsp-server/rtsp-media-factory-uri.h:
+ * gst/rtsp-server/rtsp-server.h:
+ factory-uri: add a factory to stream any URI
+ Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
+ when we have one.
+
+2010-12-11 17:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: ignore spurious ASYNC_DONE messages
+ When we are dynamically adding pads, the addition of the udpsrc elements will
+ trigger an ASYNC_DONE. We have to ignore this because we only want to react to
+ the real ASYNC_DONE when everything is prerolled.
+
+2010-12-11 13:41:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ media-factory: make lock macro
+
+2010-12-11 10:53:28 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-server: Remove unused variable and dead assignment
+
+2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * examples/test-launch.c:
+ * examples/test-mp4.c:
+ * examples/test-ogg.c:
+ * examples/test-readme.c:
+ * examples/test-sdp.c:
+ * examples/test-video.c:
+ examples: Run gst-indent
+
+2010-12-11 10:48:42 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-mapping.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-params.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ rtsp-server: Run gst-indent
+ Since it wasn't using the upstream common previously, there was no
+ indentation check before commiting.
+
+2010-12-11 10:48:25 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtsp-server/rtsp-media-mapping.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ rtsp-server: Some more doc fixups
+
+2010-12-07 18:56:03 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * Makefile.am:
+ Makefile: Add cruft-cleaning support
+
+2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * Makefile.am:
+ * configure.ac:
+ * docs/Makefile.am:
+ * docs/libs/Makefile.am:
+ * docs/libs/gst-rtsp-server-docs.sgml:
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * docs/libs/gst-rtsp-server.types:
+ * docs/version.entities.in:
+ docs: Add gtk-doc build system
+
+2010-12-07 18:14:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ Makefile.am: Use standard GIR make behaviour
+
+2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
+
+ * autogen.sh:
+ * configure.ac:
+ autogen/configure: Bring more in sync to standard gst module behaviour
+
+2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: warn and fail when gstrtpbin is not found
+
+2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ configure: open 0.11 branch
+
+2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * .gitmodules:
+ * common:
+ Add common submodule
+
+2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * common/ChangeLog:
+ * common/Makefile.am:
+ * common/c-to-xml.py:
+ * common/check.mak:
+ * common/coverage/coverage-report-entry.pl:
+ * common/coverage/coverage-report.pl:
+ * common/coverage/coverage-report.xsl:
+ * common/coverage/lcov.mak:
+ * common/gettext.patch:
+ * common/glib-gen.mak:
+ * common/gst-autogen.sh:
+ * common/gst-xmlinspect.py:
+ * common/gst.supp:
+ * common/gstdoc-scangobj:
+ * common/gtk-doc-plugins.mak:
+ * common/gtk-doc.mak:
+ * common/m4/.gitignore:
+ * common/m4/Makefile.am:
+ * common/m4/README:
+ * common/m4/as-ac-expand.m4:
+ * common/m4/as-auto-alt.m4:
+ * common/m4/as-compiler-flag.m4:
+ * common/m4/as-compiler.m4:
+ * common/m4/as-docbook.m4:
+ * common/m4/as-libtool-tags.m4:
+ * common/m4/as-libtool.m4:
+ * common/m4/as-python.m4:
+ * common/m4/as-scrub-include.m4:
+ * common/m4/as-version.m4:
+ * common/m4/ax_create_stdint_h.m4:
+ * common/m4/check.m4:
+ * common/m4/glib-gettext.m4:
+ * common/m4/gst-arch.m4:
+ * common/m4/gst-args.m4:
+ * common/m4/gst-check.m4:
+ * common/m4/gst-debuginfo.m4:
+ * common/m4/gst-default.m4:
+ * common/m4/gst-doc.m4:
+ * common/m4/gst-error.m4:
+ * common/m4/gst-feature.m4:
+ * common/m4/gst-function.m4:
+ * common/m4/gst-gettext.m4:
+ * common/m4/gst-glib2.m4:
+ * common/m4/gst-libxml2.m4:
+ * common/m4/gst-plugindir.m4:
+ * common/m4/gst-valgrind.m4:
+ * common/m4/gtk-doc.m4:
+ * common/m4/introspection.m4:
+ * common/m4/pkg.m4:
+ * common/mangle-tmpl.py:
+ * common/plugins.xsl:
+ * common/po.mak:
+ * common/release.mak:
+ * common/scangobj-merge.py:
+ * common/upload.mak:
+ common: Remove static version
+
+2010-11-08 17:04:00 +0000 Bastien Nocera <hadess@hadess.net>
+
+ * common/m4/introspection.m4:
+ Update introspection.m4 to match usage
+
+2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * README:
+ README: update
+ Remove old stuff from the README
+
+2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ back to development
+
+=== release 0.10.7 ===
+
+2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ release 0.10.7
+
+2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-ogg.c:
+ test-ogg: remove parsers
+ Remove the parsers, they are not needed anymore as oggdemux now outputs normal
+ buffers with timestamps. Using the parsers also seems to break things.
+
+2010-09-23 12:44:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * bindings/vala/gst-rtsp-server-0.10.vapi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
+ Updated Vala bindings
+
+2010-09-22 23:13:37 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * common/m4/introspection.m4:
+ * configure.ac:
+ * gst/rtsp-server/Makefile.am:
+ Added initial gobject-introspection support
+
+2010-09-23 11:32:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ media-factory: don't use host for shared hash key
+ When we generate the key to share made between connections, don't include the
+ host used to connect so that we can share media even if between clients that
+ connected with localhost and ones with the ip address.
+
+2010-09-22 21:16:03 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * bindings/vala/Makefile.am:
+ build: fix distcheck
+
+2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * bindings/vala/gst-rtsp-server-0.10.vapi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.gi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
+ Update Vala bindings
+
+2010-09-22 18:12:50 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * bindings/vala/Makefile.am:
+ * configure.ac:
+ Fix configure checks and installation location for Vala bindings
+ Fixes bug #628676.
+
+2010-09-22 16:32:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ back to development
+
+=== release 0.10.6 ===
+
+2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ configure: release 0.10.6
+
+2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: help the compiler a little
+
+2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-session.c:
+ media: cleanup media transport before freeing
+ Cleanup the media transport data before freeing. In particular, remove the qdata
+ from the rtpsource object.
+
+2010-08-20 18:17:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * 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:
+ media-factory: add eos-shutdown property
+ Add an eos-shutdown property that will send an EOS to the pipeline before
+ shutting it down. This allows for nice cleanup in case of a muxer.
+ Fixes #625597
+
+2010-08-20 15:58:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: use multiudpsink send-duplicates when we can
+ If we have a new enough multiudpsink with the send-duplicates property, use this
+ instead of doing our own filtering. Our custom filtering code should eventually
+ be removed when we can depend on a released -good.
+
+2010-08-20 13:19:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: don't leak destinations
+ Refactor and cleanup the destinations array when the stream is destroyed.
+
+2010-08-20 13:09:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: don't add udp addresses multiple times
+ Keep track of the udp addresses we added to udpsink and never add the same udp
+ destination twice. This avoids duplicate packets when using multicast.
+
+2010-08-20 10:18:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: disable use of SO_LINGER
+ SO_LINGER cause the client to fail to receive a TEARDOWN message because the
+ server close()s the connection.
+
+2010-08-19 18:52:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: use 5 second linger period in SO_LINGER
+ Wait 5 seconds before clearing the send buffers and reseting the connection with
+ the client when we do a close. This should be enough time to get the message to
+ the client.
+ See #622757
+
+2010-08-16 12:32:28 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: use SO_LINGER
+ SO_LINGER on the socket will make sure that any pending data on the socket is
+ flushed ASAP and that the socket connection is reset. This makes sure that the
+ socket can be reused immediately.
+ Fixes 622757
+
+2010-08-16 12:24:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/README:
+ README: add blurb about shared media factories
+
+2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
+
+ * gst/rtsp-server/rtsp-media.c:
+ Add stdlib.h for atoi()
+
+2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * bindings/python/Makefile.am:
+ * bindings/vala/Makefile.am:
+ build: distcheck fixes
+ Fix 'make distcheck', somewhat (it still fails because it tries to
+ install files into /usr/share/vala/vapi/ irrespective of the
+ configured prefix).
+
+2010-05-20 14:09:18 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ configure: bump core/base requirements to released version
+ Makes things less confusing for people.
+
+2010-04-25 16:35:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ configure: fail if GStreamer core/base requirements are not met
+
+2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: improve client cleanups
+ Make sure the session does not timeout when using TCP. We need to do this
+ because quicktime player does not send RTCP for some reason in tunneled
+ mode.
+ Refactor some cleanup code.
+ Fixes #612915
+
+2010-04-06 17:07:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ session: add support for prevent session timeouts
+ Add an atomix counter to prevent session timeouts when we are, for example,
+ streaming over TCP.
+
+2010-04-06 15:45:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix unlink on session timeouts
+ When our session times out, make sure we unlink all streams in this
+ session.
+ Remove the tunnelid when closing the connection.
+
+2010-04-06 15:44:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session.c:
+ session: small cleanups
+
+2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: handle lost_tunnel callbacks
+ Handle lost_tunnel callbacks and use it to store the tunnelid back into the
+ hashtable so that we can reuse it for when the client reopens the POST
+ socket.
+ Close the connection after a TEARDOWN.
+ Make sure or watchid is cleared when the watch is removed.
+ Fixes #612915
+
+2010-03-19 18:03:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ rtsp-server: add more support for multicast
+
+2010-03-19 15:15:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: allow configuration of allowed lower transport
+
+2010-03-16 18:37:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-sdp.h:
+ * gst/rtsp-server/rtsp-server.c:
+ rtsp: keep track of server ip and ipv6
+ Keep track of how the client connected to the server and setup the udp ports
+ with the same protocol.
+ Copy the server ip address in the SDP so that clients can send RTCP back to
+ us.
+
+2010-03-16 18:34:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session.c:
+ session: indent
+
+2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: use right size for malloc
+
+2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: comment ipv6 server listening address
+
+2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: allow for ipv6 sockets
+
+2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ server: rework server part
+ Allow setting a bind address, make sure we can deal with ipv6.
+ Remove the port property and change with the service property.
+
+2010-03-09 13:44:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.h:
+ media: update comments a little
+
+2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: make content-base better
+ Use the URI formatting functions to make a content-base. Also make sure that
+ there is a trailing / at the end.
+
+2010-03-09 13:42:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: guard against invalid paths
+
+2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-video.c:
+ test: catch server bind errors
+
+2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtspmedia: emit "unprepared" if _prepare fails.
+ Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
+ media object is removed from its factory's cache.
+
+2010-03-05 19:08:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: collect media position when seek completes
+
+2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: call unlink_streams in client finalize
+ Fixes #599027
+
+2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: limit the time to wait to something huge
+ Avoid waiting forever but limit the timeout to 20 seconds.
+
+2010-03-05 17:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ sdp: reindent and check for prepared status
+
+2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-session.c:
+ media: avoid doing _get_state() for state changes
+ When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
+ until the media is prerolled or in error. This avoids doing a blocking call of
+ gst_element_get_state() that can cause lockups when there is an error.
+ Fixes #611899
+
+2010-03-05 16:20:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: reindent
+
+2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ media-factory: better error handling
+ Improve the error handling a bit.
+
+2010-03-05 13:31:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: rework transport parsing
+ Rework the transport parsing code so that we can ignore transports we don't
+ support instead of just picking the first one we can parse.
+ Configure a (for now hardcoded) destination for multicast transports.
+
+2010-03-05 13:28:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: set multicast sink parameters
+ Disable loop and automatic multicast join on the udpsink elements.
+ Add some more debug info.
+ Reset some state variables in the right place.
+ Use the right port numbers for multicast.
+
+2010-03-05 13:27:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session.c:
+ session: handle transport setup correctly
+ Handle UDP, MCAST and TCP transport negotiation more correctly.
+ Store the server session SSRC in the transport.
+
+2010-01-27 18:38:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: implement error_full
+ Implement error_full to avoid some segfaults when the rtspconnection calls it.
+ See #608245
+
+2009-12-25 18:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/README:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-server.c:
+ docs: update docs and comments
+
+2009-12-25 15:22:23 +0100 Nikolay Ivanov <ivnik@mail.ru>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ sdp: make server work better when behind a proxy
+
+2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
+
+2009-11-21 19:20:23 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-mapping.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ Use GStreamer's debugging subsystem
+
+2009-11-21 01:00:39 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ server: Set ghost pad active in gst_rtsp_media_factory_collect_streams
+
+2009-11-05 11:22:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ back to development
+
+=== release 0.10.5 ===
+
+2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ release 0.10.5
+
+2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ configure: bump required versions
+
+2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: call weak-unref on client->sessions from finalize
+ Fixes bug #596305
+
+2009-10-09 23:08:18 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: Fixed crasher where caps got unref'ed too often
+
+2009-10-09 16:26:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * configure.ac:
+ * pkgconfig/.gitignore:
+ * pkgconfig/Makefile.am:
+ * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
+ Added pkg-config file to use gst-rtsp-server uninstalled
+
+2009-09-11 13:52:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: add some docs
+
+2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp: Use gst_rtsp_watch_send_message().
+ Use gst_rtsp_watch_send_message() since the old API which used
+ gst_rtsp_watch_queue_message() has been deprecated.
+
+2009-08-05 11:53:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ back to development
+
+=== release 0.10.4 ===
+
+2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ Release 0.10.4
+
+2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ rtsp: allocate channels in TCP mode
+ When the client does not provide us with channels in TCP mode, allocate channels
+ ourselves.
+
+2009-07-24 12:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: don't crash when tunnelid is missing
+ When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
+ don't crash but return an error response to the client.
+ Fixes #589489
+
+2009-07-13 11:31:23 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * bindings/vala/gst-rtsp-server-0.10.vapi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.gi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
+ bindings: update vala bindings with new method
+
+2009-06-30 21:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ sessionpool: add function to filter sessions
+ Add generic function to retrieve/remove sessions.
+
+2009-06-22 18:57:25 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ configure: bump core/base requirements to release
+
+2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: fix indentation
+
+2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-media.c:
+ Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.
+
+2009-06-13 16:05:02 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-media.c:
+ set state and remove elements of media in for loop
+
+2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
+
+ * bindings/vala/gst-rtsp-server-0.10.vapi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.gi:
+ Added gst_rtsp_media_remove_elements function to Vala bindings
+
+2009-06-13 14:38:20 +0200 Sebastian <sebastian@ubuntu.(none)>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ Added gst_rtsp_media_remove_elements function
+
+2009-06-12 22:22:40 +0200 Sebastian <sebastian@ubuntu.(none)>
+
+ * gst/rtsp-server/rtsp-media.c:
+ Don't use name for gstrtpbin so we can add multiple instances to the pipeline
+
+2009-06-12 19:28:04 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * bindings/vala/gst-rtsp-server-0.10.vapi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.gi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
+ Updated Vala bindings
+
+2009-06-12 18:05:30 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ Added vmethod unprepare to GstRTSPMedia
+ The default implementation sets the state of the pipeline to GST_STATE_NULL
+
+2009-06-12 17:51:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ Made collect_streams function public
+
+2009-06-12 17:45:29 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ Added vmethod create_pipeline to GstRTSPMediaFactory
+ The pipeline is created in this method and the GstRTSPMedia's element is added to it
+
+2009-06-11 11:27:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: use g_source_destroy()
+ We need to use g_source_destroy() because we might have added the source to a
+ different main context than the default one.
+
+2009-06-10 00:01:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-params.c:
+ * gst/rtsp-server/rtsp-params.h:
+ rtsp: prepare for handling GET/SET_PARAMETER
+ Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
+ is a body now.
+ Fix return codes of handlers.
+
+2009-06-04 19:20:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: don't leak session pads
+
+2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: clean up the messages a bit
+
+2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ sdp: warn and skip streams without media
+
+2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * bindings/vala/gst-rtsp-server-0.10.vapi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
+ vala: Fixed typo in header file of RTSPMediaStream
+
+2009-05-27 11:15:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: fix message
+ Fix a debug message
+ Make dumping RTCP stats configurable
+
+2009-05-26 19:20:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: be less verbose and leak less
+
+2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: don't leak the destination address
+
+2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ rtsp: use RTCP to keep the session alive
+ Use the RTCP rtcp-from stats field to find the associated session and use this
+ to keep the session alive.
+
+2009-05-26 17:27:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session.c:
+ session: add 5sec to the real session timeout
+ Allow the session to live 5sec longer before really timing out. This should give
+ clients some extra time to keep the session active.
+
+2009-05-26 17:25:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: replay OK to GET/SET_PARAMETER
+ Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
+ so that we return OK for those requests.
+
+2009-05-26 11:42:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: keep track of active transports
+ Keep track of which transport is active to avoid closing the connection too
+ soon.
+ Remove the destination transport also when going to NULL.
+ Print some stats about the SDES and other RTCP messages we receive from the
+ clients.
+
+2009-05-24 20:00:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/.gitignore:
+ * examples/Makefile.am:
+ * examples/test-sdp.c:
+ example: add SDP relay example
+
+2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: also count active TCP connections
+
+2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ rtsp: add support for dynamic elements
+ Add support for dynamic elements.
+ Don't set live pipelines back to paused.
+
+2009-05-24 19:33:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ sdp: don't add encoding name when absent in caps
+
+2009-05-23 16:30:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: warn when we can't do RTP-Info
+
+2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ factory: factor out the stream construction
+
+2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: only add RTP-Info when we have the info
+ Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
+ depayloader.
+
+2009-05-17 14:04:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ back to development
+
+=== release 0.10.3 ===
+
+2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ release: 0.10.3
+ - Fixes a bug where it put the wrong verion in pkgconfig
+ - Link RTP and RTCP sources
+
+2009-05-15 17:58:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: link the RTP udpsrc to the session manager
+ Link the RTP udpsrc and the appsrc to the session manager so that they don't
+ shut down when the client sends a packet to open firewalls.
+
+2009-05-15 17:10:44 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * pkgconfig/gst-rtsp-server.pc.in:
+ Don't use hard-coded version number in pkg-config file
+
+2009-05-11 10:51:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ back to development
+
+=== release 0.10.2 ===
+
+2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ release 0.10.2
+
+2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * .gitignore:
+ * common/m4/.gitignore:
+ * examples/.gitignore:
+ * pkgconfig/.gitignore:
+ add some .gitignore files
+
+2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: seek to key frames
+
+2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: emit the unprepared signal by id
+ Emit the unprepared signal by id instead of name and set the media as
+ reused.
+
+2009-04-21 22:23:54 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-media.c:
+ Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare
+
+2009-04-18 16:10:59 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * gst/rtsp-server/rtsp-server.c:
+ Added finalize function to GstRTPSPServer to unref session pool and media mapping
+
+2009-04-17 21:13:07 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * bindings/vala/gst-rtsp-server-0.10.vapi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.gi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
+ Updated vala bindings
+
+2009-04-14 23:38:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ server: use appsink and appsrc with the API
+ Use the appsink/appsrc API instead of the signals for higher
+ performance.
+
+2009-04-14 23:38:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-ogg.c:
+ tests: set the payload type correctly
+
+2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ factory: connect to the unprepare signal
+ Connect to the unprepare signal for non-reusable media so that we can remove
+ them from the cache.
+
+2009-04-03 22:45:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: add signal to notify of unprepare
+
+2009-04-03 22:22:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ media: more work on making the media shared
+ Add a reusable flag to medias, indicating that they can be reused after a state
+ change to NULL.
+ Small cleanups.
+
+2009-04-03 19:47:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-readme.c:
+ examples: mark the example as shared for testing
+
+2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ client: support shared media
+ Always perform the state actions even if the target state of the pipeline is
+ already correct, we still want to add/remove the transports when we are dealing
+ with shared media.
+ Keep a counter of the number of active transports for a media so that we can use
+ this to perform a state change when needed.
+ Perform a state change of the pipeline only when the first transport was added
+ or when there are no active transports.
+
+2009-04-03 09:03:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix refcounting crasher
+ Don't need to remove the weak refs in the finalize methods, they are already
+ removed in the dispose.
+ Don't register the callback with a DestroyNofity.
+
+2009-04-01 01:01:46 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ Fix rtsp client refcount management in TCP mode.
+ Don't unref a client ref we never had. Fixes an unref
+ of an already-free client object after a client
+ teardown request for me.
+
+2009-04-01 00:45:17 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session.c:
+ docs: fix typo in API docs
+
+2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ More seeking fixes.
+ Keep the udp sources in playing even if we go to paused. unlock the sources when
+ we shut down.
+ Add some more debug info.
+ Only seek when we need to.
+ Keep track of the position when we go to paused.
+
+2009-03-12 20:32:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ Add beginnings of seeking.
+ Parse the Range header and perform a seek on the pipeline for the requested
+ position. It's disabled currently until I figure out what's going wrong.
+
+2009-03-12 20:31:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ allow pause requests for now.
+ --
+
+2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ Remove weak ref on the session in teardown
+ We need to remove our weakref from the session when we do a teardown because
+ else we close the TCP connection prematurely.
+
+2009-03-11 19:38:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ Do some more session cleanup
+ Make session timeout kill the TCP connection that currently watches the
+ session.
+ Remove the client timeout property.
+
+2009-03-11 16:45:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ Add TCP transports
+ Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
+ connection.
+
+2009-03-11 16:39:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/Makefile.am:
+ * examples/test-launch.c:
+ Add example server that takes launch lines
+ Add an example server that streams any -launch line.
+
+2009-03-06 19:34:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-readme.c:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ Add support for live streams
+ Add support for live streams and ranges
+ Start on handling TCP data transfer.
+
+2009-03-04 16:33:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ Free the pipeline before other things
+ ---
+
+2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ Only free the pending tunnel if there is one
+ --
+
+2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-server: Add support for tunneling
+ Add support for tunneling over HTTP.
+ Use new connection methods to retrieve the url.
+ Dispatch messages based on the message type instead of blindly
+ assuming it's always a request.
+ Keep track of the watch id so that we can remove it later.
+ Set the media pipeline to NULL before unreffing the pipeline.
+
+2009-02-19 15:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ Fix for channel -> watch rename in gstreamer
+ Rename the RTSPChannel to RTSPWatch and remove an unused variable.
+
+2009-02-18 18:57:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ Use ASYNC RTSP io
+ Use the async RTSP channels instead of spawning a new thread for each client.
+ If a sessionid is specified in a request, fail if we don't have the session.
+
+2009-02-18 17:49:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ Add better debug info
+ Add some better debug info.
+
+2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/test-video.c:
+ Time out sessions
+ Add support for session timeouts in the example.
+
+2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ Pass GTimeVal around for performance reasons
+ Get the current time only once and pass it around so that sessions don't have to
+ get the current time anymore.
+ Add experimental support for a GSource that dispatches when the session needs to
+ be cleaned up.
+
+2009-02-13 19:56:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ Add better support for session timeouts
+ Add a method to request the number of milliseconds when a session will timeout.
+
+2009-02-13 19:54:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ Add suport for RTP manager monitoring
+ Add the first stage in monitoring the rtp manager.
+ Make sure we don't update the state to something we don't want.
+
+2009-02-13 19:52:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ Add support for session keepalive
+ Get and update the session timeout for all requests. get the session as early as
+ possible.
+
+2009-02-13 16:39:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ Handle media bus messages
+ Handle media bus messages in a custom mainloop and dispatch them to the
+ RTSPMedia objects. Let the default implementation handle some common messages.
+
+2009-02-13 12:57:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ Some more session timeout handling
+ Move the session header setting code to a central place so that we always add
+ the timeout parameter too.
+ Handle timeouts by running the session cleanup code.
+ Stop media before cleaning up.
+
+2009-02-10 16:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ Add timeout property
+ Add a timeout property ot the client and make the other properties into GObject
+ properties.
+
+2009-02-10 16:21:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ Use getters and setters in property code
+ Use the getters and setters for the timeout property instead of locking
+ ourselves.
+
+2009-02-04 20:13:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
+
+2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ Add more timeout stuff
+ Add method to check if a session is expired.
+ Add method to perform cleanup on a session pool.
+
+2009-02-04 19:52:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ Add beginnings of session timeouts and limits
+ Add the timeout value to the Session header for unusual timeout values.
+ Allow us to configure a limit to the amount of active sessions in a pool. Set a
+ limit on the amount of retry we do after a sessionid collision.
+ Add properties to the sessionid and the timeout of a session. Keep track of
+ creation time and last access time for sessions.
+
+2009-02-04 17:00:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ Cleanup of sessions and more
+ Fix the refcounting of media and sessions in the client. Properly clean up the
+ session data when the client performs a teardown.
+ Add Server header to responses.
+ Allow for multiple uri setups in one session.
+ Add Range header to the PLAY response and add the range attribute to the SDP
+ message.
+ Fix the session pool remove method, it used the wrong key in the hashtable. Also
+ give the ownership of the sessionid to the session object.
+
+2009-02-04 09:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ Rename a variable
+ Rename the 'server_port' variable to simply 'port'.
+
+2009-02-03 19:32:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ Rework the way we handle transports for streams
+ Make the media accept an array of transports for the streams that we have
+ configured for the play/pause requests.
+ Implement server states for a client and its media.
+ Require 0.10.22.1 (git HEAD) of gstreamer.
+
+2009-01-31 19:50:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ Drop const from functions dealing with urls
+ Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
+ have the right const in them.
+
+2009-01-30 17:06:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ Fix various leaks
+ Fix some leaks.
+
+2009-01-30 16:24:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ More cleanups
+ Don't keep a reference to the GstRTSPMedia in the stream.
+ Free more things when freeing the GstRTSPMedia.
+
+2009-01-30 14:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/README:
+ * 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:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ More docs and small cleanups
+ Add some more docs and update the README
+ Cleanup some method names.
+ Remove an unneeded idx field in the GstRTSPMediaStream
+
+2009-01-30 13:24:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * docs/README:
+ * examples/Makefile.am:
+ * examples/test-readme.c:
+ Add a README and more example code
+ Add a README file that contains a small introduction on how to use the server
+ along with the example code explained in the readme.
+
+2009-01-30 11:06:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-server.c:
+ Fix some leaks and change default port
+ Fix some memory leaks by setting the udpsrc elements to the unlocked state after
+ we finished the initial preroll. If we keep them locked, setting the pipeline to
+ NULL will not stop and clean up the sources correctly.
+ Change the default RTSP port to 8554 aka the official alternative RTSP port.
+
+2009-01-29 18:55:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ Cleanups to the session object
+ Remove some unneeded variables in the session state of a stream such as the
+ owner media and the server transport.
+ Get the configuration of a media stream in a session based on the media_stream
+ in the original object instead of our cached index.
+ Free more data in the finalize method.
+
+2009-01-29 18:51:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ Cleanups and reuse media from DESCRIBE
+ Handle thread create errors.
+ Rename some internal methods to better match what they actually do.
+ Handle misconfiguration of session_pool and media_mapping gracefully.
+ Cache the DESCRIBE media and uri in the client connection and reuse them when
+ we receive a SETUP request in the same connection for the same uri.
+ Cleanup the client connection object.
+
+2009-01-29 17:20:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * 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:
+ Add shared properties to media and factory
+ Add the shared property to media.
+ Implement some simple caching in the factory depending on if the media is shared
+ or not.
+
+2009-01-29 17:19:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ Add a little comment
+ Add some comment about the content-base header.
+
+2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/Makefile.am:
+ * examples/test-mp4.c:
+ * examples/test-ogg.c:
+ * examples/test-video.c:
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * 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:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-sdp.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ Reorganize things, prepare for media sharing
+ Added various other test server examples
+ Move the SDP message generation to a separate helper.
+ Refactor common code for finding the session.
+ Add content-base for realplayer compatibility
+ Clean up request uris before processing for better vlc compatibility.
+ Move prerolling and pipeline construction to the RTSPMedia object.
+ Use multiudpsink for future pipeline reuse.
+
+2009-01-30 11:23:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ Back to development
+ Back to 0.10.1.1
+
+=== release 0.10.1 ===
+
+2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * configure.ac:
+ Make 0.10.1 release
+ Release 0.10.1
+
+2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * bindings/vala/Makefile.am:
+ Fix make dist
+ Add more directories and files to the dist.
+
+2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * bindings/python/Makefile.am:
+ * bindings/python/rtspserver.override:
+ Fixed compile error of python bindings
+
+2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * bindings/vala/gst-rtsp-server-0.10.vapi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
+ Marked values as nullable accordingly
+
+2009-01-23 20:31:11 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * bindings/vala/gst-rtsp-server-0.10.vapi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
+ * bindings/vala/packages/gst-rtsp-server-0.10.gi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
+ Updated Vala bindings
+
+2009-01-22 18:35:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-mapping.c:
+ * gst/rtsp-server/rtsp-media-mapping.h:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ Cleanups and doc updates
+ Add some more documentation and do some minor cleanups here and there.
+
+2009-01-22 17:58:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * 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:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ More improvements
+ Rename GstRTSPMediaBin to GstRTSPMedia
+ Parse the request url into a GstRTSPUri object and pass this object to the
+ various handlers and methods that require the uri.
+
+2009-01-22 16:54:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/main.c:
+ Update example
+ Add some more docs and remove some old code from the example.
+
+2009-01-22 16:53:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ Handle state change failures better
+ Handle state change failures better when changing the state of the pipeline to
+ determine the SDP.
+
+2009-01-22 16:51:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ Make element creation more extendible
+ Add get_element vmethod to the default MediaFactory so that subclasses can just
+ override that method and still use the default logic for making a MediaBin from
+ that.
+
+2009-01-22 15:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/main.c:
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media-mapping.c:
+ * gst/rtsp-server/rtsp-media-mapping.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ Make the server handle arbitrary pipelines
+ Make GstMediaFactory an object that can instantiate GstMediaBin objects.
+ The GstMediaBin object has a handle to a bin with elements and to a list of
+ GstMediaStream objects that this bin produces.
+ Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
+ with methods to register and remove those mappings.
+ Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
+ used by the server instance.
+ Modify the example application so that it shows how to create custom pipelines
+ attached to a specific mount point.
+ Various misc cleanps.
+
+2009-01-20 19:47:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ Allow setting a custom media factory for a server
+
+2009-01-20 19:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ Allow setting a custom media factory for a client.
+
+2009-01-20 19:45:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ Add Makefile entry for the media factory
+
+2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ Add media factory to map urls to media pipeline objects.
+
+2009-01-20 19:43:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ Add comments. Remove unused field
+
+2009-01-20 19:41:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ Allow custom session pools to override the session id allocation algorithms Add some comments.
+
+2009-01-20 19:40:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session.h:
+ Add some comments.
+
+2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ Move the connection code in one place Add some comments
+
+2009-01-20 13:19:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ Make vmethod to create and accept new clients. Add some docs.
+
+2009-01-19 19:36:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations.
+
+2009-01-19 19:34:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ Name the parameters more appropriately.
+
+2009-01-19 19:32:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ Do some more cleanup of the session pool.
+
+2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-client.c:
+ Check if return value of gst_rtsp_session_get_media is not NULL
+
+2009-01-08 15:02:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/Makefile.am:
+ Install rtsp-session and rtsp-session-pool headers
+
+2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * .gitignore:
+ * Makefile.am:
+ * acinclude.m4:
+ * bindings/python/Makefile.am:
+ * bindings/python/arg-types.py:
+ * bindings/python/codegen/Makefile.am:
+ * bindings/python/codegen/__init__.py:
+ * bindings/python/codegen/argtypes.py:
+ * bindings/python/codegen/code-coverage.py:
+ * bindings/python/codegen/codegen.py:
+ * bindings/python/codegen/definitions.py:
+ * bindings/python/codegen/defsparser.py:
+ * bindings/python/codegen/docextract.py:
+ * bindings/python/codegen/docgen.py:
+ * bindings/python/codegen/fileprefix.override:
+ * bindings/python/codegen/fileprefixmodule.c:
+ * bindings/python/codegen/h2def.py:
+ * bindings/python/codegen/mergedefs.py:
+ * bindings/python/codegen/mkskel.py:
+ * bindings/python/codegen/override.py:
+ * bindings/python/codegen/reversewrapper.py:
+ * bindings/python/codegen/scmexpr.py:
+ * bindings/python/rtspserver-types.defs:
+ * bindings/python/rtspserver.defs:
+ * bindings/python/rtspserver.override:
+ * bindings/python/rtspservermodule.c:
+ * configure.ac:
+ Add python bindings.
+
+2009-01-08 14:53:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * bindings/Makefile.am:
+ * configure.ac:
+ Don't go into python dir when requirements for python bindings are missing
+
+2009-01-08 14:49:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * bindings/Makefile.am:
+ * bindings/vala/Makefile.am:
+ * configure.ac:
+ Install Vala bindings if vala is available
+
+2008-12-12 16:22:02 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * bindings/vala/gst-rtsp-server-0.10.deps:
+ * bindings/vala/gst-rtsp-server-0.10.vapi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.deps:
+ * bindings/vala/packages/gst-rtsp-server-0.10.excludes:
+ * bindings/vala/packages/gst-rtsp-server-0.10.files:
+ * bindings/vala/packages/gst-rtsp-server-0.10.gi:
+ * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
+ * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
+ Regenerated Vala bindings
+
+2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
+
+ * bindings/vala/gst-rtsp-server.vapi:
+ * bindings/vala/packages/gst-rtsp-server.metadata:
+ Fixed typo in included headers for vala bindings
+
+2009-01-08 14:42:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * Makefile.am:
+ * configure.ac:
+ * pkgconfig/Makefile.am:
+ * pkgconfig/gst-rtsp-server.pc.in:
+ Added pkgconfig file
+
+2008-11-30 23:57:26 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
+
+ * bindings/vala/gst-rtsp-server.vapi:
+ * bindings/vala/packages/gst-rtsp-server.excludes:
+ * bindings/vala/packages/gst-rtsp-server.gi:
+ * bindings/vala/packages/gst-rtsp-server.metadata:
+ Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h
+
+2008-11-30 23:41:20 +0100 Sebastian Pölsterl <marduk@k-d-w.org>
+
+ * bindings/vala/gst-rtsp-server.vapi:
+ * bindings/vala/packages/gst-rtsp-server.deps:
+ * bindings/vala/packages/gst-rtsp-server.files:
+ * bindings/vala/packages/gst-rtsp-server.gi:
+ * bindings/vala/packages/gst-rtsp-server.metadata:
+ * bindings/vala/packages/gst-rtsp-server.namespace:
+ Added Vala bindings
+
+2008-10-25 23:36:16 +0200 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/rtsp-server/rtsp-session.c:
+ Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b)
+
+2008-11-13 19:43:10 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
+
+ * examples/Makefile.am:
+ * gst/rtsp-server/Makefile.am:
+ Put GStreamer version in library name
+
+2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * examples/Makefile.am:
+ * gst/rtsp-server/Makefile.am:
+ Fix some issues to pass distcheck
+
+2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-server.c:
+ Added port property to GstRTSPServer class.
+
+2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * Makefile.am:
+ * autogen.sh:
+ * configure.ac:
+ * examples/Makefile.am:
+ * examples/main.c:
+ * gst/Makefile.am:
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ * src/Makefile.am:
+ Split in library and example program
+
+2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
+
+ * src/rtsp-client.h:
+ Removed obsolete variable
+
+2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
+
+ * src/rtsp-client.c:
+ * src/rtsp-client.h:
+ Removed pipeline variable GstRTSPClient, because it's only used in one function
+
+2009-01-08 11:22:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * src/rtsp-media.c:
+ Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead.
+
+2008-10-23 12:23:27 +0200 Wim Taymans <wim@metal.(none)>
+
+ * src/rtsp-session.c:
+ Initialize some more vars.
+
+2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
+
+ * src/rtsp-session.c:
+ Initialize variable to avoid compiler warning.
+
+2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
+
+ * .gitignore:
+ Add a reasonable generic .gitignore
+
projects / platform / upstream / gstreamer.git / blobdiff