+=== release 1.16.0 ===
+
+2019-04-19 00:34:54 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ * meson.build:
+ Release 1.16.0
+
+2019-04-15 20:33:01 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: Notify the stream transport about each written message
+ Otherwise it will never try to send us the next one: it tries to keep
+ exactly one message in-flight all the time.
+ In gst-rtsp-server this is done asynchronously via the GstRTSPWatch but
+ in the client sink we always write data out synchronously.
+
+2019-04-02 08:05:03 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp_server: Free thread pool before clean transport cache
+ If not waiting for free thread pool before clean transport caches, there
+ can be a crash if a thread is executing in transport list loop in
+ function send_tcp_message.
+ Also add a check if priv->send_pool in on_message_sent to avoid that a
+ new thread is pushed during wait of free thread pool. This is possible
+ since when waiting for free thread pool mutex have to be unlocked.
+
+=== release 1.15.90 ===
+
+2019-04-11 00:35:55 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ * meson.build:
+ Release 1.15.90
+
+2019-04-10 10:32:53 +0200 Ulf Olsson <ulfo@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Add support for GCM (RFC 7714)
+ Follow-up to !198
+
+2019-03-28 00:27:37 +0100 Erlend Eriksen <erlend_ne@hotmail.com>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ session pool: fix missing klass-> in klass->create_session
+
+2019-03-23 19:16:17 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ g-i: pass --quiet to g-ir-scanner
+ This suppresses the annoying 'g-ir-scanner: link: cc ..' output
+ that we get even if everything works just fine.
+ We still get g-ir-scanner warnings and compiler warnings if
+ we pass this option.
+
+2019-03-23 19:15:48 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ g-i: silence 'nested extern' compiler warnings when building scanner binary
+ We need a nested extern in our init section for the scanner binary
+ so we can call gst_init to make sure GStreamer types are initialised
+ (they are not all lazy init via get_type functions, but some are in
+ exported variables). There doesn't seem to be any other mechanism to
+ achieve this, so just remove that warning, it's not important at all.
+
+2019-03-21 11:49:10 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ meson: pass -Wno-unused to compiler if gstreamer debug system is disabled
+
+2019-03-14 07:37:26 +0100 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * tests/check/gst/media.c:
+ rtsp-media: Handle set state when preparing.
+ Handle the situation when a call to gst_rtsp_media_set_state is done
+ when media status is preparing.
+ Also add unit test for this scenario.
+ The unit test simulate on a media level when two clients share a (live)
+ media.
+ Both clients have done SETUP and got responses. Now client 1 is doing
+ play and client 2 is just closing the connection.
+ Then without patch there are a problem when
+ client1 is calling gst_rtsp_media_unsuspend in handle_play_request.
+ And client2 is doing closing connection we can end up in a call
+ to gst_rtsp_media_set_state when
+ priv->status == GST_RTSP_MEDIA_STATUS_PREPARING and all the logic for
+ shut down media is jumped over .
+ With this patch and this scenario we wait until
+ priv->status == GST_RTSP_MEDIA_STATUS_PREPARED and then continue to
+ execute after that and now we will execute the logic for
+ shut down media.
+
+2019-03-04 09:13:30 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * meson.build:
+ Back to development
+
+=== release 1.15.2 ===
+
+2019-02-26 11:58:53 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ * meson.build:
+ Release 1.15.2
+
+2019-02-19 09:45:08 +0100 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * tests/check/gst/client.c:
+ rtsp-media: Fix multicast use case with common media
+ Use case
+ client 1: SETUP
+ client 1: PLAY
+ client 2: SETUP
+ client 1: TEARDOWN
+ client 2: PLAY
+ client 2: TEARDOWN
+
+2019-01-16 12:59:11 +0100 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-server: remove recursive behavior
+ Introduce a threadpool to send rtp and rtcp to avoid recursive behavior.
+
+2019-01-25 14:22:42 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Only allow to set either a send_func or send_messages_func but not both
+ And route all messages through the send_func if no send_messages_func
+ was provided.
+ We otherwise break backwards compatibility.
+
+2018-09-17 22:18:46 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-client: Add support for sending buffer lists directly
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
+
+2018-06-27 12:17:07 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtsp-server: Add support for buffer lists
+ This adds new functions for passing buffer lists through the different
+ layers without breaking API/ABI, and enables the appsink to actually
+ provide buffer lists.
+ This should already reduce CPU usage and potentially context switches a
+ bit by passing a whole buffer list from the appsink instead of
+ individual buffers. As a next step it would be necessary to
+ a) Add support for a vector of data for the GstRTSPMessage body
+ b) Add support for sending multiple messages at once to the
+ GstRTSPWatch and let it be handled internally
+ c) Adding API to GOutputStream that works like writev()
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
+
+2018-12-04 14:12:04 +0100 Benjamin Berg <bberg@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: Fix crash in close handler
+ The close handler could trigger a crash because it invalidated the
+ watch_context while still leaving a source attached to it which would be
+ cleaned up at a later point.
+
+2019-01-29 14:42:35 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Use cached address when allocating sockets
+ If an address/port was previously decided upon (ex: multicast in the
+ SDP), then use that instead of re-creating another one
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/57
+
+2018-12-27 11:28:17 +0100 Lars Wiréen <larswi@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Fix race codition in finish_unprepare
+ The previous fix for race condition around finish_unprepare where the
+ function could be called twice assumed that the status wouldn't change
+ during execution of the function. This assumption is incorrect as the
+ state may change, for example if an error message arrives from the
+ pipeline bus.
+ Instead a flag keeping track on whether the finish_unprepare function
+ is currently executing is introduced and checked.
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/59
+
+=== release 1.15.1 ===
+
+2019-01-17 02:26:48 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ * meson.build:
+ Release 1.15.1
+
+2018-12-05 15:07:25 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ Add source elements to the pipeline before activation
+ In plug_src we changed the element state before adding it to
+ the owner container. This prevented the pipeline from intercepting
+ a GST_STREAM_STATUS_TYPE_CREATE message from the pad in order
+ to assign a custom task pool.
+ Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/53
+
+2018-12-05 17:24:59 -0300 Thibault Saunier <tsaunier@igalia.com>
+
+ * common:
+ Automatic update of common submodule
+ From ed78bee to 59cb678
+
+2018-11-20 19:12:09 +0100 Ingo Randolf <ingo.randolf@servus.at>
+
+ * examples/test-appsrc.c:
+ examples: test-appsrc: fix coding style error
+
+2018-11-20 11:07:48 +0100 Ingo Randolf <ingo.randolf@servus.at>
+
+ * examples/test-appsrc.c:
+ examples: test-appsrc: fix buffer leak
+
+2018-11-17 19:19:54 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Update priv->blocked when linked streams are unblocked.
+ Media is considered to be blocked when all streams that belong to
+ that media are blocked.
+ This patch solves the problem of inconsistent updates of
+ priv->blocked that are not synchronized with the media state.
+
+2018-11-17 18:18:27 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Don't block streams before seeking
+ Before the seek operation is performed on media, it's required that
+ its pipeline is prepared <=> the pipeline is in the PAUSED state.
+ At this stage, all transport parts (transport sinks) have been successfully
+ added to the pipeline and there is no need for blocking the streams.
+
+2018-11-17 16:11:53 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: rtspserver: Add shared media test case for TCP
+
+2018-11-06 18:21:54 +0100 Linus Svensson <linussn@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Use seqnum-offset for rtpinfo
+ The sequence number in the rtpinfo is supposed to be the first RTP
+ sequence number. The "seqnum" property on a payloader is supposed to be
+ the number from the last processed RTP packet. The sequence number for
+ payloaders that inherit gstrtpbasepayload will not be correct in case of
+ buffer lists. In order to fix the seqnum property on the payloaders
+ gst-rtsp-server must get the sequence number for rtpinfo elsewhere and
+ "seqnum-offset" from the "stats" property contains the value of the
+ very first RTP packet in a stream. The server will, however, try to look
+ at the last simple in the sink element and only use properties on the
+ payloader in case there no sink elements yet, and by looking at the last
+ sample of the sink gives the server full control of which RTP packet it
+ looks at. If the payloader does not have the "stats" property, "seqnum"
+ is still used since "seqnum-offset" is only present in as part of
+ "stats" and this is still an issue not solved with this patch.
+ Needed for gst-plugins-base!17
+
+2018-11-06 18:10:56 +0100 Linus Svensson <linussn@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Plug memory leak
+ Attaching a GSource to a context will increase the refcount. The idle
+ source will never be free'd since the initial reference is never
+ dropped.
+
+2018-11-12 16:06:39 +0200 Jordan Petridis <jordan@centricular.com>
+
+ * .gitlab-ci.yml:
+ Add Gitlab CI configuration
+ This commit adds a .gitlab-ci.yml file, which uses a feature
+ to fetch the config from a centralized repository. The intent is
+ to have all the gstreamer modules use the same configuration.
+ The configuration is currently hosted at the gst-ci repository
+ under the gitlab/ci_template.yml path.
+ Part of https://gitlab.freedesktop.org/gstreamer/gstreamer-project/issues/29
+
+2018-11-05 05:56:35 +0000 Matthew Waters <matthew@centricular.com>
+
+ * .gitmodules:
+ * gst-rtsp-server.doap:
+ Update git locations to gitlab
+
+2018-11-01 14:20:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/meson.build:
+ meson: add new onvif types
+
+2018-11-01 12:49:51 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/meson.build:
+ Add ONVIF subclass headers to the installed headers in meson.build too
+
+2018-11-01 11:29:01 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-server-object.h:
+ * gst/rtsp-server/rtsp-server.h:
+ rtsp-server: Declare GstRTSPServer struct before anything else
+ It's needed by all kinds of other headers, including the ones that are
+ required for defining the GstRTSPServer struct itself and its API.
+
+2018-11-01 10:23:02 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-onvif-client.h:
+ * gst/rtsp-server/rtsp-onvif-media-factory.h:
+ * gst/rtsp-server/rtsp-onvif-media.h:
+ * gst/rtsp-server/rtsp-onvif-server.h:
+ Mark all ONVIF-specific subclasses as Since 1.14
+
+2018-11-01 10:18:22 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/meson.build:
+ * gst/rtsp-server/rtsp-context.h:
+ * gst/rtsp-server/rtsp-onvif-server.c:
+ * gst/rtsp-server/rtsp-onvif-server.h:
+ * gst/rtsp-server/rtsp-server-object.h:
+ * gst/rtsp-server/rtsp-server-prelude.h:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session.h:
+ Include ONVIF types from single-include rtsp-server.h
+ ... by actually making it a single-include header and moving everything
+ related to the GstRTSPServer type to rtsp-server-object.h instead.
+ Otherwise there are too many circular includes.
+ https://bugzilla.gnome.org/show_bug.cgi?id=797361
+
+2018-10-18 07:25:05 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-latency-bin.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-stream: use idle source in on_message_sent
+ When the underlying layers are running on_message_sent, this sometimes
+ causes the underlying layer to send more data, which will cause the
+ underlying layer to run callback on_message_sent again. This can go on
+ and on.
+ To break this chain, we introduce an idle source that takes care of
+ sending data if there are more to send when running callback
+ https://bugzilla.gnome.org/show_bug.cgi?id=797289
+
+2018-10-20 16:14:53 +0200 Edward Hervey <edward@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Remove timeout GSource on cleanup
+ Avoids ending up with races where a timeout would still be around
+ *after* a client was gone. This could happen rather easily in
+ RTSP-over-HTTP mode on a local connection, where each RTSP message
+ would be sent as a different HTTP connection with the same tunnelid.
+ If not properly removed, that timeout would then try to free again
+ a client (and its contents).
+
+2018-10-04 14:31:59 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/Makefile.am:
+ autotools: fix distcheck
+
+2018-09-12 11:55:15 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/meson.build:
+ * gst/rtsp-server/rtsp-latency-bin.c:
+ * gst/rtsp-server/rtsp-latency-bin.h:
+ * gst/rtsp-server/rtsp-onvif-media.c:
+ onvif: encapsulate onvif part into a bin
+ ...and thus do not let onvif affect pipelines latency
+ https://bugzilla.gnome.org/show_bug.cgi?id=797174
+
+2018-09-27 19:57:13 +0200 Patricia Muscalu <patricia@dovakhiin.com>
+
+ * tests/check/gst/client.c:
+ tests: client: Avoid bind() failures in tests
+ https://bugzilla.gnome.org/show_bug.cgi?id=797059
+
+2018-09-06 16:17:33 +0200 Patricia Muscalu <patricia@axis.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:
+ * tests/check/gst/client.c:
+ * tests/check/gst/mediafactory.c:
+ New property for socket binding to mcast addresses
+ By default the multicast sockets are bound to INADDR_ANY,
+ as it's not allowed to bind sockets to multicast addresses
+ in Windows. This default behaviour can be changed by setting
+ bind-mcast-address property on the media-factory object.
+ https://bugzilla.gnome.org/show_bug.cgi?id=797059
+
+2018-09-24 09:36:21 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/meson.build:
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-context.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-params.c:
+ * gst/rtsp-server/rtsp-permissions.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-server-prelude.h:
+ * 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:
+ * 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:
+ * meson.build:
+ libs: fix API export/import and 'inconsistent linkage' on MSVC
+ Export rtsp-server library API in headers when we're building the
+ library itself, otherwise import the API from the headers.
+ This fixes linker warnings on Windows when building with MSVC.
+ Fix up some missing config.h includes when building the lib which
+ is needed to get the export api define from config.h
+ https://bugzilla.gnome.org/show_bug.cgi?id=797185
+
+2018-09-19 14:31:56 +0200 Edward Hervey <edward@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ rtsp-media-factory: Add missing break statements
+ This resulted in warnings/assertions whenever one accessed the
+ max-mcast-ttl property.
+ CID #1439515
+ CID #1439523
+
+2018-09-19 12:21:30 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ * meson_options.txt:
+ meson: add gobject-cast-checks, glib-asserts, glib-checks options
+
+2018-09-19 12:17:57 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/meson.build:
+ * meson_options.txt:
+ * tests/check/meson.build:
+ meson: add option to disable build of rtspclientsink plugin
+
+2018-09-19 12:10:14 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson_options.txt:
+ meson: re-arrange options
+
+2018-09-01 11:21:15 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * meson.build:
+ * meson_options.txt:
+ * tests/check/meson.build:
+ * tests/meson.build:
+ meson: Use feature option for tests option
+ This was somehow missed the last time around.
+
+2018-08-31 14:42:15 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * gst/rtsp-server/meson.build:
+ * meson.build:
+ meson: Maintain macOS ABI through dylib versioning
+ Requires Meson 0.48, but the feature will be ignored on older versions
+ so it's safe to add it without bumping the requirement.
+ Documentation:
+ https://github.com/mesonbuild/meson/blob/master/docs/markdown/Reference-manual.md#shared_library
+
+2018-08-31 17:20:47 +1000 Matthew Waters <matthew@centricular.com>
+
+ * gst/rtsp-sink/meson.build:
+ * meson.build:
+ meson: add pkg-config file for the rtspclientsink plugin
+
+2018-08-17 09:54:27 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * tests/check/gst/client.c:
+ rtsp-client: Avoid reuse of channel numbers for interleaved
+ If a (strange) client would reuse interleaved channel numbers in
+ multiple SETUP requests, we should not accept them. The channel
+ numbers are used for looking up stream transports in the
+ priv->transports hash table, and transports disappear from the table
+ if channel numbers are reused.
+ RFC 7826 (RTSP 2.0), Section 18.54, clarifies that it is OK for the
+ server to change the channel numbers suggested by the client.
+ https://bugzilla.gnome.org/show_bug.cgi?id=796988
+
+2018-08-17 09:54:27 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * tests/check/gst/client.c:
+ rtsp-client: Add unit test of SETUP for RTSP/RTP/TCP
+ Allow regex for matching transport header against expected pattern.
+ https://bugzilla.gnome.org/show_bug.cgi?id=796988
+
+2018-08-15 18:57:27 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * tests/check/meson.build:
+ meson: There is no gstreamer-plugins-good-1.0.pc
+ There is no installed version of that, only an uninstalled version.
+
+2018-08-14 14:31:55 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * tests/check/gst/stream.c:
+ Fix indentation again
+
+2018-07-26 12:01:16 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ * tests/check/gst/client.c:
+ * tests/check/gst/stream.c:
+ stream: Added a list of multicast client addresses
+ When media is shared, the same media stream can be sent
+ to multiple multicast groups. Currently, there is no API
+ to retrieve multicast addresses from the stream.
+ When calling gst_rtsp_stream_get_multicast_address() function,
+ only the first multicast address is returned.
+ With this patch, each multicast destination requested in SETUP
+ will be stored in an internal list (call to
+ gst_rtsp_stream_add_multicast_client_address()).
+ The list of multicast groups requested by the clients can be
+ retrieved by calling gst_rtsp_stream_get_multicast_client_addresses().
+ There still exist some problems with the current implementation
+ in the multicast case:
+ 1) The receiving part is currently only configured with
+ regard to the first multicast client (see
+ https://bugzilla.gnome.org/show_bug.cgi?id=796917).
+ 2) Secondly, of security reasons, some constraints should be
+ put on the requested multicast destinations (see
+ https://bugzilla.gnome.org/show_bug.cgi?id=796916).
+ Change-Id: I6b060746e472a0734cc2fd828ffe4ea2956733ea
+ https://bugzilla.gnome.org/show_bug.cgi?id=793441
+
+2018-07-25 15:33:18 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ * tests/check/gst/client.c:
+ stream: Choose the maximum ttl value provided by multicast clients
+ The maximum ttl value provided so far by the multicast clients
+ will be chosen and reported in the response to the current
+ client request.
+ Change-Id: I5408646e3b5a0a224d907ae215bdea60c4f1905f
+ https://bugzilla.gnome.org/show_bug.cgi?id=793441
+
+2018-02-23 14:34:32 +0100 Patricia Muscalu <patricia@dovakhiin.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * tests/check/gst/client.c:
+ rtsp-stream: Don't require address pool in the transport specific case
+ If "transport.client-settings" parameter is set to true, the client is
+ allowed to specify destination, ports and ttl.
+ There is no need for pre-configured address pool.
+ Change-Id: I6ae578fb5164d78e8ec1e2ee82dc4eaacd0912d1
+ https://bugzilla.gnome.org/show_bug.cgi?id=793441
+
+2018-07-24 14:02:40 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * tests/check/gst/client.c:
+ client: Don't reserve multicast address in the client setting case
+ When two multicast clients request specific transport
+ configurations, and "transport.client-settings" parameter is
+ set to true, it's wrong to actually require that these two
+ clients request the same multicast group.
+ Removed test_client_multicast_invalid_transport_specific test
+ cases as they wrongly require that the requested destination
+ address is supposed to be present in the address pool, also in
+ the case when "transport.client-settings" parameter is set to true.
+ Change-Id: I4580182ef35996caf644686d6139f72ec599c9fa
+ https://bugzilla.gnome.org/show_bug.cgi?id=793441
+
+2018-07-24 09:35:46 +0200 Patricia Muscalu <patricia@axis.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:
+ * tests/check/gst/mediafactory.c:
+ Add new API for setting/getting maximum multicast ttl value
+ Change-Id: I5ef4758188c14785e17fb8fbf42a3dc0cb054233
+ https://bugzilla.gnome.org/show_bug.cgi?id=793441
+
+2018-07-31 21:17:41 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: avoid duplicating the first multicast client
+ In dcb4533fedae3ac62bc25a916eb95927b7d69aec , we made it so
+ clients were dynamically added and removed to the multicast
+ udp sinks, as such we should no longer add a first client in
+ set_multicast_socket_for_udpsink
+ https://bugzilla.gnome.org/show_bug.cgi?id=793441
+
+2018-08-14 14:25:53 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ Revert "rtsp-stream: avoid duplicating the first multicast client"
+ This reverts commit 33570944401747f44d8ebfec535350651413fb92.
+ Commits where accidentially squashed together
+
+2018-08-14 14:25:42 +0300 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-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ * tests/check/gst/client.c:
+ * tests/check/gst/mediafactory.c:
+ Revert "Add new API for setting/getting maximum multicast ttl value"
+ This reverts commit 7f0ae77e400fb8a0462a76a5dd2e63e12c4a2e52.
+ Commits where accidentially squashed together
+
+2018-08-14 14:25:37 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * tests/check/gst/client.c:
+ Revert "rtsp-stream: Don't require address pool in the transport specific case"
+ This reverts commit a9db3e7f092cfeb5475e9aa24b1e91906c141d52.
+ Commits where accidentially squashed together
+
+2018-08-14 14:25:14 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ * tests/check/gst/client.c:
+ * tests/check/gst/stream.c:
+ Revert "stream: Choose the maximum ttl value provided by multicast clients"
+ This reverts commit 499e437e501215849d24cdaa157e0edf4de097d0.
+ Commits where accidentially squashed together
+
+2018-08-14 14:10:56 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-auth-digest.c:
+ examples: Fix indentation
+
+2018-07-25 15:33:18 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ * tests/check/gst/client.c:
+ * tests/check/gst/stream.c:
+ stream: Choose the maximum ttl value provided by multicast clients
+ The maximum ttl value provided so far by the multicast clients
+ will be chosen and reported in the response to the current
+ client request.
+ https://bugzilla.gnome.org/show_bug.cgi?id=793441
+
+2018-02-23 14:34:32 +0100 Patricia Muscalu <patricia@dovakhiin.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * tests/check/gst/client.c:
+ rtsp-stream: Don't require address pool in the transport specific case
+ If "transport.client-settings" parameter is set to true, the client is
+ allowed to specify destination, ports and ttl.
+ There is no need for pre-configured address pool.
+ https://bugzilla.gnome.org/show_bug.cgi?id=793441
+
+2018-07-24 09:35:46 +0200 Patricia Muscalu <patricia@axis.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-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ * tests/check/gst/client.c:
+ * tests/check/gst/mediafactory.c:
+ Add new API for setting/getting maximum multicast ttl value
+ https://bugzilla.gnome.org/show_bug.cgi?id=793441
+
+2018-07-31 21:17:41 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: avoid duplicating the first multicast client
+ In dcb4533fedae3ac62bc25a916eb95927b7d69aec , we made it so
+ clients were dynamically added and removed to the multicast
+ udp sinks, as such we should no longer add a first client in
+ set_multicast_socket_for_udpsink
+ https://bugzilla.gnome.org/show_bug.cgi?id=793441
+
+2018-08-06 15:33:04 -0400 Thibault Saunier <tsaunier@igalia.com>
+
+ * gst/rtsp-server/Makefile.am:
+ rtsp-server: Add gstreamer-base gir dir in autotools
+
+2018-07-25 19:54:55 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-client: always allocate both IPV4 and IPV6 sockets
+ multiudpsink does not support setting the socket* properties
+ after it has started, which meant that rtsp-server could no
+ longer serve on both IPV4 and IPV6 sockets since the patches
+ from https://bugzilla.gnome.org/show_bug.cgi?id=757488 were
+ merged.
+ When first connecting an IPV6 client then an IPV4 client,
+ multiudpsink fell back to using the IPV6 socket.
+ When first connecting an IPV4 client, then an IPV6 client,
+ multiudpsink errored out, released the IPV4 socket, then
+ crashed when trying to send a message on NULL nevertheless,
+ that is however a separate issue.
+ This could probably be fixed by handling the setting of
+ sockets in multiudpsink after it has started, that will
+ however be a much more significant effort.
+ For now, this commit simply partially reverts the behaviour
+ of rtsp-stream: it will continue to only create the udpsinks
+ when needed, as was the case since the patches were merged,
+ it will however when creating them, always allocate both
+ sockets and set them on the sink before it starts, as was
+ the case prior to the patches.
+ Transport configuration will only error out if the allocation
+ of UDP sockets fails for the actual client's family, this
+ also downgrades the GST_ERRORs in alloc_ports_one_family
+ to GST_WARNINGs, as failing to allocate is no longer
+ necessarily fatal.
+ https://bugzilla.gnome.org/show_bug.cgi?id=796875
+
+2018-07-25 17:22:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * meson.build:
+ * meson_options.txt:
+ meson: Convert common options to feature options
+ These are necessary for gst-build to set options correctly. The
+ remaining automagic option is cgroup support in examples.
+ https://bugzilla.gnome.org/show_bug.cgi?id=795107
+
+2018-07-23 18:03:51 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Slightly simplify locking
+
+2018-06-28 11:22:21 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ Limit queued TCP data messages to one per stream
+ Before, the watch backlog size in GstRTSPClient was changed
+ dynamically between unlimited and a fixed size, trying to avoid both
+ unlimited memory usage and deadlocks while waiting for place in the
+ queue. (Some of the deadlocks were described in a long comment in
+ handle_request().)
+ In the previous commit, we changed to a fixed backlog size of 100.
+ This is possible, because we now handle RTP/RTCP data messages differently
+ from RTSP request/response messages.
+ The data messages are messages tunneled over TCP. We allow at most one
+ queued data message per stream in GstRTSPClient at a time, and
+ successfully sent data messages are acked by sending a "message-sent"
+ callback from the GstStreamTransport. Until that ack comes, the
+ GstRTSPStream does not call pull_sample() on its appsink, and
+ therefore the streaming thread in the pipeline will not be blocked
+ inside GstRTSPClient, waiting for a place in the queue.
+ pull_sample() is called when we have both an ack and a "new-sample"
+ signal from the appsink. Then, we know there is a buffer to write.
+ RTSP request/response messages are not acked in the same way as data
+ messages. The rest of the 100 places in the queue are used for
+ them. If the queue becomes full of request/response messages, we
+ return an error and close the connection to the client.
+ Change-Id: I275310bc90a219ceb2473c098261acc78be84c97
+
+2018-06-28 11:22:13 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Use fixed backlog size
+ Change to using a fixed backlog size WATCH_BACKLOG_SIZE.
+ Preparation for the next commit, which changes to a different way of
+ avoiding both deadlocks and unlimited memory usage with the watch
+ backlog.
+
+2018-07-16 21:57:08 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: unref clock (if set) when finalizing
+ https://bugzilla.gnome.org/show_bug.cgi?id=796814
+
+2018-07-16 21:56:44 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ rtsp-media: add gst_rtsp_media_*_set_clock to docs
+ https://bugzilla.gnome.org/show_bug.cgi?id=796814
+
+2018-07-12 19:01:54 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ media-factory: unref old clock when setting new clock
+ https://bugzilla.gnome.org/show_bug.cgi?id=796724
+
+2018-06-29 15:20:57 -0700 Brendan Shanks <brendan.shanks@teradek.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ media-factory: unref clock in finalize
+ https://bugzilla.gnome.org/show_bug.cgi?id=796724
+
+2018-07-12 18:57:21 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-onvif-media.c:
+ rtsp-onvif-media: fix g-ir-scanner warnings
+
+2018-07-10 23:56:23 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * .gitignore:
+ .gitignore: add another example binary
+
+2018-07-10 23:55:20 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/meson.build:
+ meson: add new test-appsrc2 example to meson build
+
+2018-07-10 23:53:41 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/Makefile.am:
+ examples: fix build of new test-appsrc2 example
+ Need to link against libgstapp-1.0.
+
+2018-07-11 01:25:51 +1000 Jan Schmidt <jan@centricular.com>
+
+ * examples/.gitignore:
+ * examples/Makefile.am:
+ * examples/test-appsrc2.c:
+ examples: Add test-appsrc2
+ Add an example of feeding both audio and video into an RTSP
+ pipeline via appsrc.
+
+2016-01-08 18:12:14 -0500 Louis-Francis Ratté-Boulianne <lfrb@collabora.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: Strip transport parts as whitespaces could be around commas
+ https://bugzilla.gnome.org/show_bug.cgi?id=758428
+
+2018-06-27 08:30:42 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: avoid pushing data on unlinked udpsrc pad during setup
+ Fix race when setting up source elements.
+ Since we set the source element(s) to PLAYING state before hooking
+ them up to the downstream funnel, it's possible for the source element
+ to receive packets before we actually get to linking it to the funnel,
+ in which case buffers would be pushed out on an unlinked pad, causing
+ it to error out and stop receiving more data.
+ We fix this by blocking the source's srcpad until we have linked it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=796160
+
+2018-03-21 10:56:51 +0100 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Fix mismatch between allowed and configured protocols
+ https://bugzilla.gnome.org/show_bug.cgi?id=796679
+
+2017-02-01 09:44:50 +0100 Ulf Olsson <ulfo@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Emit a signal when the SRTP decoder is created
+ https://bugzilla.gnome.org/show_bug.cgi?id=778080
+
+2018-03-13 11:10:35 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Don't require presence of sinks in _get_*_socket()
+ Transport specific sink elements are added to the pipeline
+ in PLAY request and sockets are already created in SETUP so
+ it's actually wrong to require the presence of sinks in
+ _get_*_socket() functions.
+ https://bugzilla.gnome.org/show_bug.cgi?id=793441
+
+2018-02-14 10:41:02 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Update transport for multicast clients as well
+ If a multicast client requests different transport settings
+ than the existing one make sure that this new transport
+ configuruation is propagated to the multicast udp sink.
+ https://bugzilla.gnome.org/show_bug.cgi?id=793441
+
+2018-02-13 11:04:36 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Set the multicast TTL parameter on multicast udp sinks
+ And not on unicast udp sinks
+ https://bugzilla.gnome.org/show_bug.cgi?id=793441
+
+2018-06-24 12:44:26 +0200 Tim-Philipp Müller <tim@centricular.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:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ Update for g_type_class_add_private() deprecation in recent GLib
+
+2018-06-24 12:45:49 +0200 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ Fix indentation
+
+2018-06-22 23:17:08 +1000 Jan Schmidt <jan@centricular.com>
+
+ * examples/Makefile.am:
+ * examples/test-video-disconnect.c:
+ examples: Add test-video-disconnect example
+ Simple example which cuts off all clients 10 seconds
+ after the first one connects.
+
+2018-06-20 04:37:11 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * examples/test-auth-digest.c:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ rtsp-auth: Add support for parsing .htdigest files
+ Passwords are usually not stored in clear text, but instead
+ stored already hashed in a .htdigest file.
+ Add support for parsing such files, add API to allow setting
+ a custom realm in RTSPAuth, and update the digest example.
+ https://bugzilla.gnome.org/show_bug.cgi?id=796637
+
+2018-06-19 14:53:02 +1000 Matthew Waters <matthew@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ * gst/rtsp-sink/gstrtspclientsink.h:
+ rtspclientsink: fix waiting for multiple streams
+ We were previously only ever waiting for a single stream to notify it's
+ blocked status through GstRTSPStreamBlocking. Actually count streams to
+ wait for.
+ Fixes rtspclientsink sending SDP's without out some of the input
+ streams.
+ https://bugzilla.gnome.org/show_bug.cgi?id=796624
+
+2018-06-20 04:30:04 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ docs: add missing auth methods
+
+2018-06-20 00:10:18 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: only create funnel if it didn't exist already.
+ This precented using multiple protocols for the same stream.
+ https://bugzilla.gnome.org/show_bug.cgi?id=796634
+
+2018-06-20 01:35:47 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * examples/meson.build:
+ meson: build auth-digest example
+
+2018-06-05 08:44:44 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ Get payloader stats only for the sending streams
+ Get/set payloader properties only for streams that actually
+ contain a payloader element.
+ https://bugzilla.gnome.org/show_bug.cgi?id=796523
+
+2018-05-18 14:53:49 +0200 Edward Hervey <edward@centricular.com>
+
+ * gst/rtsp-server/Makefile.am:
+ Makefile: Don't hardcode libtool for g-i build
+ Similar to the other commits in core/base/bad
+
+2018-05-08 14:13:31 +0200 Johan Bjäreholt <johanbj@axis.com>
+
+ * gst/rtsp-server/rtsp-onvif-media-factory.h:
+ rtsp-onvif-media-factory: export gst_rtsp_onvif_media_factory_requires_backchannel
+ https://bugzilla.gnome.org/show_bug.cgi?id=796229
+
+2018-05-09 04:09:02 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: Don't deadlock in preroll on early close
+ If the connection is closed very early, the flushing
+ marker might not get set and rtspclientsink can get
+ deadlocked waiting for preroll forever.
+ https://bugzilla.gnome.org/show_bug.cgi?id=786961
+
+2018-05-05 19:51:52 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * meson.build:
+ * meson_options.txt:
+ meson: Update option names to omit disable_ and with- prefixes
+ Also yield common options to the outer project (gst-build in our case)
+ so that they don't have to be set manually.
+
+2018-04-25 11:00:32 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ meson: use -Wl,-Bsymbolic-functions where supported
+ Just like the autotools build.
+
+2018-04-22 20:09:01 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ * tests/check/Makefile.am:
+ configure: check for -good and -bad plugins only in uninstalled setup
+ Avoids confusing configure messages looking or a -good .pc file
+ that doesn't exist.
+ Also use plugindir variables that common macros set while at it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=795466
+
+2018-04-17 11:03:11 +0200 Joakim Johansson <joakimj@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Fix session timeout
+ When streaming data over TCP then is not the keep-alive
+ functionality working.
+ The reason is that the function do_send_data have changed
+ to boolean but the code is still checking the received result
+ from send_func with GST_RTSP_OK.
+ The result is that a successful send_func will always lead to
+ that do_send_data is returning false and the keep-alive will
+ not be updated.
+ https://bugzilla.gnome.org/show_bug.cgi?id=795321
+
+2018-04-02 22:49:35 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ * gst/rtsp-sink/gstrtspclientsink.h:
+ Implement support for ULP Forward Error Correction
+ In this initial commit, interface is only exposed for RECORD,
+ further work will be needed in rtspsrc to support this for
+ PLAY.
+ https://bugzilla.gnome.org/show_bug.cgi?id=794911
+
+2018-04-17 17:47:30 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-onvif-media.c:
+ Revert "rtsp-server: Switch around sendonly/recvonly attributes"
+ This reverts commit 3d275b1345b76151418e3f56ed014d9089ac1a57.
+ While RFC 3264 (SDP) says that sendonly/recvonly are from the point of view of
+ the requester, the actual RTSP RFCs (RFC 2326 / 7826) disagree and say
+ the opposite, just like the ONVIF standard.
+ Let's follow those RFCs as we're doing RTSP here, and add a property at
+ a later time if needed to switch to the SDP RFC behaviour.
+ https://bugzilla.gnome.org/show_bug.cgi?id=793964
+
+2018-04-16 10:53:52 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From 3fa2c9e to ed78bee
+
+2018-04-04 10:06:06 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * tests/check/gst/rtspclientsink.c:
+ gst: Run everything through gst-indent again
+
+2018-04-03 08:57:47 +0200 Branko Subasic <branko@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * tests/check/gst/media.c:
+ rtsp-media: query the position on active streams if media is complete
+ If the media is complete, i.e. one or more streams have been configured
+ with sinks, then we want to query the position on those streams only.
+ A query on an incomplete stream may return a position that originates from
+ an earlier preroll.
+ https://bugzilla.gnome.org/show_bug.cgi?id=794964
+
+2018-04-02 12:35:04 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: make sure not to use freed string
+ Set transport string to NULL after freeing it, so that
+ at worst we get a NULL pointer if constructing a new
+ transport string fails (which shouldn't really fail here).
+ Also check return value of that, just in case.
+ CID 1433768.
+
+2018-03-30 23:34:01 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: do not free string passed to take_header
+
+2018-03-30 23:10:10 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: do not take lock in request_aux_receiver
+ Added it right before pushing the previous commit, it is
+ incorrect and deadlocks because this function gets called
+ from the join_bin thread, which already holds the lock,
+ that's the reason why request_aux_sender didn't take the
+ lock either.
+
+2018-03-29 22:49:26 +0200 Mathieu Duponchelle <mathieu@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-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-server: add API to enable retransmission requests
+ "do-retransmission" was previously set when rtx-time != 0,
+ which made no sense as do-retransmission is used to enable
+ the sending of retransmission requests, where as rtx-time
+ is used by the peer to enable storing of buffers in order
+ to respond to retransmission requests.
+ rtsp-media now also provides a callback for the
+ request-aux-receiver signal.
+ https://bugzilla.gnome.org/show_bug.cgi?id=794822
+
+2018-03-29 16:18:42 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: add rtx ssrc to mikey's crypto sessions
+ https://bugzilla.gnome.org/show_bug.cgi?id=794813
+
+2018-03-29 16:15:45 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: Handle the KeyMgmt header in ANNOUNCE response
+ This in order to be able to decrypt the RTCP backchannel
+ https://bugzilla.gnome.org/show_bug.cgi?id=794813
+
+2018-03-29 16:12:26 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Send KeyMgmt header in ANNOUNCE response
+ When sending back an encrypted RTCP back channel, it is useful
+ for the client to know the encryption key.
+ https://bugzilla.gnome.org/show_bug.cgi?id=794813
+
+2018-03-29 16:06:31 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-stream: extract handle_keymgmt from rtsp-client
+ rtspclientsink will also need to parse KeyMgmt headers
+ sent by the server to decrypt the RTCP backchannel stream
+ https://bugzilla.gnome.org/show_bug.cgi?id=794813
+
+2018-03-29 02:51:02 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ * tests/check/gst/rtspclientsink.c:
+ rtspclientsink: Fix client ports for the RTCP backchannel
+ This was broken since the work for delayed transport creation
+ was merged: the creation of the transports string depends on
+ calling stream_get_server_port, which only starts returning
+ something meaningful after a call to stream_allocate_udp_sockets
+ has been made, this function expects a transport that we parse
+ from the transport string ...
+ Significant refactoring is in order, but does not look entirely
+ trivial, for now we put a band aid on and create a second transport
+ string after the stream has been completed, to pass it in
+ the request headers instead of the previous, incomplete one.
+ https://bugzilla.gnome.org/show_bug.cgi?id=794789
+
+2018-02-15 13:26:16 +0100 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client:Error handling when equal http session cookie
+ There are some clients that are sending same session cookie on random
+ basis.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753616
+
+2018-03-20 16:21:37 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory-uri.c:
+ rtsp-media-factory-uri: Fix compilation with latest GLib
+ rtsp-media-factory-uri.c: In function ‘rtsp_media_factory_uri_create_element’:
+ rtsp-media-factory-uri.c:621:17: error: assignment from incompatible pointer type [-Werror=incompatible-pointer-types]
+ data->factory = g_object_ref (factory);
+ ^
+
+2018-03-20 10:21:36 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * meson.build:
+ Back to development
+
+=== release 1.14.0 ===
+
+2018-03-19 20:27:04 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ * meson.build:
+ Release 1.14.0
+
+=== release 1.13.91 ===
+
+2018-03-13 19:28:33 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ * meson.build:
+ Release 1.13.91
+
+2018-03-13 13:30:41 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/meson.build:
+ * 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-onvif-client.h:
+ * gst/rtsp-server/rtsp-onvif-media-factory.h:
+ * gst/rtsp-server/rtsp-onvif-media.h:
+ * gst/rtsp-server/rtsp-onvif-server.h:
+ * gst/rtsp-server/rtsp-params.h:
+ * gst/rtsp-server/rtsp-permissions.h:
+ * gst/rtsp-server/rtsp-sdp.h:
+ * gst/rtsp-server/rtsp-server-prelude.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: GST_EXPORT -> GST_RTSP_SERVER_API
+ We need different export decorators for the different libs.
+ For now no actual change though, just rename before the release,
+ and add prelude headers to define the new decorator to GST_EXPORT.
+
+2018-03-07 12:20:05 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-onvif-media-factory.c:
+ rtsp-onvif-media-factory: Document that backchannel pipelines must end with async=false sinks
+ https://bugzilla.gnome.org/show_bug.cgi?id=794143
+
+=== release 1.13.90 ===
+
+2018-03-03 22:49:34 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ * meson.build:
+ Release 1.13.90
+
+2018-03-02 16:24:23 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-permissions.c:
+ permissions: add Since tags and example for new API
+
+2018-03-02 01:36:23 +0100 Mathieu Duponchelle <mathieu@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-permissions.c:
+ * gst/rtsp-server/rtsp-permissions.h:
+ * tests/check/gst/permissions.c:
+ permissions: more bindings-friendly API
+ https://bugzilla.gnome.org/show_bug.cgi?id=793975
+
+2018-03-01 19:28:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * meson.build:
+ meson: enable more warnings
+
+2018-02-28 21:12:43 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Place netaddress meta on packets received via TCP
+ This allows us to later map signals from rtpbin/rtpsource back to the
+ corresponding stream transport, and allows to do keep-alive based on
+ RTCP packets in case of TCP media transport.
+ https://bugzilla.gnome.org/show_bug.cgi?id=789646
+
+2018-02-27 20:34:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: if OPEN failed, unqueue next command
+ As READY_TO_PAUSED can no longer return async, the RECORD
+ command will be queued before the OPEN command fails
+ (for example in case the server could not be connected),
+ and record then waits for ever.
+ https://bugzilla.gnome.org/show_bug.cgi?id=793896
+
+2018-02-26 22:59:17 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: fix retrieval of custom payloader caps
+ If a bin is passed as the custom payloader, the caps of
+ its factory will be empty, the correct way to obtain the caps
+ is to query its sinkpad.
+
+2018-02-26 22:59:00 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: fix extra unref of custom payloader
+
+2018-02-26 22:57:39 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rspclientsink: fix recent code indentation
+
+2018-02-26 20:27:57 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: add missing get_type prototype
+
+2018-02-24 03:52:15 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: allow setting payloader as pad property
+ This was a FIXME item, and can be quite useful, also
+ allowing to specify payloader properties from the command
+ line, which is always nice.
+ https://bugzilla.gnome.org/show_bug.cgi?id=793776
+
+2018-02-26 14:16:54 +0100 Carlos Rafael Giani <dv@pseudoterminal.org>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Replace g_print() log line
+ https://bugzilla.gnome.org/show_bug.cgi?id=793838
+
+2018-02-22 20:17:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * tests/check/gst/rtspclientsink.c:
+ rtsp-media: fix RECORD getting stuck
+ The test_record case was working because async=false had
+ been added in https://bugzilla.gnome.org/show_bug.cgi?id=757488
+ but that was incorrect, as it should not be needed.
+ Removing async=false made the test fail as expected, this is
+ fixed by not trying to preroll when preparing the media for
+ RECORD, as start_prepare is called upon receiving ANNOUNCE,
+ and our peer will not start sending media until it has received
+ a response to that request, and sent and received a response
+ to RECORD as well, thus obviously preventing preroll.
+ https://bugzilla.gnome.org/show_bug.cgi?id=793738
+
+2018-02-23 03:26:21 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ rtsp-auth: fix set_tls_authentication_mode annotation
+
+2018-02-19 11:57:29 +0100 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
+
+ * gst/rtsp-server/rtsp-onvif-media.c:
+ rtp-server: remove redefined variable
+ res is a boolean variable which is defined in the function scope and
+ redefined, with no reason, in the loop scope. This patch removes the
+ redefinition.
+ https://bugzilla.gnome.org/show_bug.cgi?id=793592
+
+2018-02-05 11:49:07 +0100 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ stream: Add functions for checking if stream is receiver or sender
+ ...and replace all checks for RECORD in GstRTSPMedia which are really
+ for "sender-only". This way the code becomes more generic and introducing
+ support for onvif-backchannel later on will require no changes in
+ GstRTSPMedia.
+
+2017-10-21 14:06:30 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-onvif-media-factory.c:
+ * gst/rtsp-server/rtsp-onvif-media-factory.h:
+ onvif: Make requires_backchannel() public
+ ...in order to let subclasses building the onvif part of the pipeline
+ check whether backchannel shall be included or not.
+
+2018-01-22 12:46:34 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-onvif-media.c:
+ rtsp-server: Switch around sendonly/recvonly attributes
+ They are wrong in the ONVIF streaming spec. The backchannel should be
+ recvonly and the normal media should be sendonly: direction is always
+ from the point of view of the SDP offerer (the server) according to
+ RFC 3264.
+
+2017-09-25 19:41:05 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * docs/libs/gst-rtsp-server-docs.sgml:
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * examples/.gitignore:
+ * examples/Makefile.am:
+ * examples/test-onvif-backchannel.c:
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-onvif-client.c:
+ * gst/rtsp-server/rtsp-onvif-client.h:
+ * gst/rtsp-server/rtsp-onvif-media-factory.c:
+ * gst/rtsp-server/rtsp-onvif-media-factory.h:
+ * gst/rtsp-server/rtsp-onvif-media.c:
+ * gst/rtsp-server/rtsp-onvif-media.h:
+ * gst/rtsp-server/rtsp-onvif-server.c:
+ * gst/rtsp-server/rtsp-onvif-server.h:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-sdp.h:
+ rtsp: Add support for ONVIF backchannel
+ This adds a new RTSP server, client, media-factory and media subclass
+ for handling the specifics of the backchannel. Ideally this later can be
+ extended with other ONVIF specific features.
+
+2017-10-12 21:00:16 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Add support for sending+receiving medias
+ We need to add an appsrc/appsink in that case because otherwise the
+ media bin will be a sink and a source for rtpbin, causing a pipeline
+ loop.
+ https://bugzilla.gnome.org/show_bug.cgi?id=788950
+
+2018-02-15 19:44:28 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ * meson.build:
+ Back to development
+
+=== release 1.13.1 ===
+
+2018-02-15 17:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * NEWS:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ * meson.build:
+ Release 1.13.1
+
+2018-02-14 17:11:19 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ session-pool: remove nullable return annotation
+ create_watch can only return NULL from the API guards, no
+ need for nullable.
+
+2018-02-13 18:59:16 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ set_clock functions: Add nullable annotations
+
+2018-02-10 00:07:25 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-client.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:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ All around: add annotations and API guards
+
+2018-02-12 19:12:35 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * tests/test-cleanup.c:
+ test-cleanup: bind any port
+ The meson test suite runs tests in parallel, trying to bind
+ a single port made the test fail.
+
+2018-02-08 19:15:10 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ meson: make version numbers ints and fix int/string comparison
+ WARNING: Trying to compare values of different types (str, int).
+ The result of this is undefined and will become a hard error
+ in a future Meson release.
+
+2018-02-06 18:00:33 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-context.c:
+ gst_rtsp_context_get_current: add (skip) annotation
+ The return value type is defined with G_DEFINE_POINTER_TYPE,
+ and gi emits the following warning:
+ Invalid non-constant return of bare structure or union; register as
+ boxed type or (skip)
+
+2018-02-06 17:58:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: add type annotations
+ gi doesn't seem to be able to figure out the type of the
+ signal parameters when defined with G_DEFINE_POINTER_TYPE
+
+2018-02-04 12:24:09 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ autotools: use -fno-strict-aliasing where supported
+ https://bugzilla.gnome.org/show_bug.cgi?id=769183
+
+2018-01-30 20:35:21 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ meson: use -fno-strict-aliasing where supported
+ https://bugzilla.gnome.org/show_bug.cgi?id=769183
+
+2018-01-25 12:09:03 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-mount-points.c:
+ mount-points: bail out of loop again when matching mount points
+ Previous patch led to us iterating the entire sequence. Bail out
+ of the loop again if we have a match but are moving away from it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=771555
+
+2018-01-25 12:06:57 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/gst/mountpoints.c:
+ tests: mountpoints: add more checks for mount point path matching
+ https://bugzilla.gnome.org/show_bug.cgi?id=771555
+
+2016-09-16 20:41:19 +0000 Andrew Bott <andrew.bott@blackmoth.com>
+
+ * gst/rtsp-server/rtsp-mount-points.c:
+ mount-points: fix matching of paths where there's also an entry with a common prefix
+ e.g. with the following mount points
+ /raw
+ /raw/snapshot
+ /raw/video
+ _match() would not match /raw/video and /raw/snapshot correctly.
+ https://bugzilla.gnome.org/show_bug.cgi?id=771555
+
+2018-01-18 23:53:20 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-permissions.c:
+ * gst/rtsp-server/rtsp-permissions.h:
+ * tests/check/gst/permissions.c:
+ permissions: add some new API to make this usable from bindings
+ https://bugzilla.gnome.org/show_bug.cgi?id=787073
+
+2018-01-18 11:32:32 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-token.c:
+ rtsp-token: annotate constructors for bindings
+ This maps _new_empty() to _new(), which also makes RTSPToken()
+ work properly now. Since this API wasn't usable from bindings
+ before, this should hopefully be fine.
+ https://bugzilla.gnome.org/show_bug.cgi?id=787073
+
+2018-01-18 11:07:45 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-token.c:
+ * gst/rtsp-server/rtsp-token.h:
+ * tests/check/gst/token.c:
+ rtsp-token: add some API to set fields from bindings
+ The existing functions are all vararg-based and as such
+ not usable from bindings.
+ https://bugzilla.gnome.org/show_bug.cgi?id=787073
+
+2018-01-13 15:02:28 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/gst/rtspclientsink.c:
+ * tests/check/gst/rtspserver.c:
+ * tests/check/gst/sessionpool.c:
+ * tests/check/gst/stream.c:
+ tests: fix indentation
+ Fix and "fix".
+
+2018-01-13 14:58:55 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: rtspserver: fix another ref leak
+ Even if this didn't show up in valgrind.
+
+2018-01-13 14:58:00 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/gst/rtspclientsink.c:
+ tests: rtspclientsink: fix leak
+
+2018-01-02 14:19:31 +0100 Branko Subasic <branko@axis.com>
+
+ * tests/check/gst/rtspserver.c:
+ test: rtspserver: plug memory leak in test_no_session_timeout
+ In test_no_session_timeout, unref the rtsp session object when the
+ test is done.
+ https://bugzilla.gnome.org/show_bug.cgi?id=792127
+
+2017-12-20 14:17:02 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtpsclientsink: Initialize and clear newly added mutex and cond
+ While it *did* work, glib would automatically create new mutex and cond
+ ... which never got freed
+
+2017-12-19 11:34:37 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Set multicast TTL on the multicast sockets
+ And not if we do unicast UDP.
+ https://bugzilla.gnome.org/show_bug.cgi?id=791743
+
+2017-12-19 11:14:48 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Decide based on the sockets, not the addresses if we already allocated a socket
+ In the multicast case (as in test-multicast, not test-multicast2), the
+ address could be allocated/reserved (and thus set) already without
+ allocating the actual socket. We need to allocate the socket here still
+ instead of just claiming that it was already allocated.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=791743#c2
+
+2017-12-16 21:46:53 +0100 Patricia Muscalu <patricia@dovakhiin.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ * gst/rtsp-sink/gstrtspclientsink.h:
+ rtspclientsink: Use the new rtsp-stream API
+ https://bugzilla.gnome.org/show_bug.cgi?id=790412
+
+2017-12-16 21:01:43 +0100 Patricia Muscalu <patricia@dovakhiin.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ * gst/rtsp-sink/gstrtspclientsink.h:
+ rtspclientsink: Wait until OPEN has been scheduled
+ Make sure that the sink thread has started opening connection
+ to the server before continuing.
+ https://bugzilla.gnome.org/show_bug.cgi?id=790412
+
+2017-12-14 14:53:35 +1100 Matthew Waters <matthew@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From e8c7a71 to 3fa2c9e
+
+2017-12-07 16:08:29 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-server: Minor doc fixes
+ Mostly for g-i
+
+2017-12-06 20:47:22 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * Makefile.am:
+ * tests/Makefile.am:
+ tests: disable all tests when --disable-tests is used
+ Move conditional subdir include into top level.
+ Based on patch by: Joel Holdsworth
+ https://bugzilla.gnome.org/show_bug.cgi?id=757703
+
+2017-12-06 20:42:39 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ * meson_options.txt:
+ * tests/meson.build:
+ meson: build more tests and add options to disable tests and examples
+
+2017-11-26 13:26:39 -0300 Thibault Saunier <tsaunier@gnome.org>
+
+ * gst/rtsp-server/rtsp-session.c:
+ Fix build when -Werror=deprecated-declarations is on
+ As gst_rtsp_session_next_timeout is deprecated.
+ ```
+ ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-session.c:760:3: error: ‘gst_rtsp_session_next_timeout’ is deprecated: Use 'gst_rtsp_session_next_timeout_usec' instead [-Werror=deprecated-declarations]
+ res = (gst_rtsp_session_next_timeout (session, now) == 0);
+ ^~~
+ ../subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-session.c:685:1: note: declared here
+ gst_rtsp_session_next_timeout (GstRTSPSession * session, GTimeVal * now)
+ ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ ```
+
+2017-11-27 20:18:24 +1100 Matthew Waters <matthew@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From 3f4aa96 to e8c7a71
+
+2017-11-25 20:34:16 +0100 Patricia Muscalu <patricia@dovakhiin.com>
+
+ * tests/check/gst/media.c:
+ check/media: Add seekability test case: not all streams are active
+ Media contains two streams but only one is complete and prepared
+ for playing.
+ https://bugzilla.gnome.org/show_bug.cgi?id=790674
+
+2017-11-25 20:32:02 +0100 Patricia Muscalu <patricia@dovakhiin.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Do not reset 'blocking' if stream is already blocked
+ https://bugzilla.gnome.org/show_bug.cgi?id=790674
+
+2017-11-25 20:45:44 +0100 Patricia Muscalu <patricia@dovakhiin.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Fix missing lock in gst_rtsp_media_seekable()
+ https://bugzilla.gnome.org/show_bug.cgi?id=790674
+
+2017-11-26 16:29:49 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ meson: remove vs_module_defs_dir variable which is no longer needed
+
+2017-11-26 14:46:05 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-session.h:
+ rtsp: fix distcheck
+
+2017-11-26 12:53:42 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * Makefile.am:
+ * gst/rtsp-server/meson.build:
+ * win32/MANIFEST:
+ * win32/common/libgstrtspserver.def:
+ win32: remove .def file with exports
+ They're no longer needed, symbol exporting is now explicit
+ via GST_EXPORT in all cases (autotools, meson, incl. MSVC).
+
+2017-11-26 12:28:40 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ autotools: stop controlling symbol visibility with -export-symbols-regex
+ Instead, use -fvisibility=hidden and explicit exports via GST_EXPORT.
+ This should result in consistent behaviour for the autotools and
+ Meson builds.
+
+2017-11-26 12:47:08 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ rtsp-server: add missing GST_EXPORT and export deprecated funcs
+
+2017-11-25 07:53:30 +0100 Edward Hervey <edward@centricular.com>
+
+ * tests/check/gst/media.c:
+ check: Add seekability testing on medias
+ Make sure that once GstRTSPMedia are prepared they returned
+ the expected seekability results
+ https://bugzilla.gnome.org/show_bug.cgi?id=790674
+
+2017-11-24 17:34:31 +0100 Edward Hervey <edward@centricular.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ * win32/common/libgstrtspserver.def:
+ rtsp-media: Enable seeking query before pipeline is complete
+ SDP are now provided *before* the pipeline is fully complete. In order
+ to know whether a media is seekable or not therefore requires asking
+ the invididual streams.
+ API: gst_rtsp_stream_seekable
+ https://bugzilla.gnome.org/show_bug.cgi?id=790674
+
+2017-11-23 20:34:03 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Fix handling in default_unsuspend()
+ Handle the case when streams are not blocked and media
+ is suspended from PAUSED.
+ Change-Id: I2f3d222ea7b9b20a0732ea5dc81a32d17ab75040
+ https://bugzilla.gnome.org/show_bug.cgi?id=790674
+
+2017-11-23 18:51:21 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * tests/check/gst/media.c:
+ check/media: Fix thread pool leak.
+ Change-Id: I0f92b1caca0ee518ae64a7dacfbd28a214c3eea1
+ https://bugzilla.gnome.org/show_bug.cgi?id=790674
+
+2017-11-23 18:39:44 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Removed fakesink elements
+ There is not need of adding fakesink elements to the media
+ pipeline in the dynamic-payloader case.
+ The media pipeline itself is dynamically updated with
+ the receiver and sender parts that are based on the client
+ transport information known after SETUP has been received.
+ Change-Id: I4e88c9b500c04030669822f0d03b1842913f6cb9
+ https://bugzilla.gnome.org/show_bug.cgi?id=790674
+
+2017-11-23 09:10:54 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Corrected ASYNC_DONE handling
+ Media is complete when all the transport based parts are
+ added to the media pipeline. At this point ASYNC_DONE is
+ posted by the media pipeline and media is ready to enter
+ the PREPARED state.
+ Change-Id: I50fb8dfed88ebaf057d9a35fca2d7f0a70e9d1fa
+ https://bugzilla.gnome.org/show_bug.cgi?id=790674
+
+2017-11-22 12:24:38 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * tests/check/gst/media.c:
+ check/media: Check that prepared media can provide a SDP
+ Whenever a RTSPMedia is prepared, it should be able to provide a SDP
+
+2017-11-21 09:53:19 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Don't leak addr
+ CID #1422260
+
+2017-11-21 09:53:08 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ Run gst-indent
+
+2017-11-20 18:30:19 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Don't unblock with remaining dynamic payloaders
+ If we still have some dynamic paylaoders which haven't posted
+ no-more-pads yet, don't go to PREPARED if one of the streams
+ blocked.
+ The risk was that we would end up not exposing/using all specified
+ streams.
+ The downside is that if you have _multiple_ _live_ _dynamic_ payloaders
+ then it will take a bit more time to start. But only if those 3
+ conditions are present.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769521
+
+2017-11-20 16:49:29 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Fix doc
+
+2017-11-20 16:48:55 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Don't set float on a gint64 variable
+ Just use 0. Fixes 'undefined' behaviour from clang
+
+2017-11-20 18:29:02 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Fix previous commit
+ We only want to count dynamic payloaders
+
+2017-11-20 09:32:07 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * tests/check/gst/media.c:
+ rtsp-media: Handle multiple dynamic elements
+ If we have more than one dynamic payloader in the pipeline, we need
+ to wait until the *last* one emits 'no-more-pads' before switching
+ to PREPARED.
+ Failure to do so would result in a race where some of the streams
+ wouldn't properly be prepared
+ https://bugzilla.gnome.org/show_bug.cgi?id=769521
+
+2017-11-16 12:18:20 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * win32/common/libgstrtspserver.def:
+ win32: Fix exported symbols list
+
+2017-11-15 19:52:29 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Only update the RTP udpsink if it actually exists
+ For send-only streams it does not exist, but the RTCP udpsink might.
+
+2017-11-15 18:15:53 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * win32/common/libgstrtspserver.def:
+ win32: Update exports
+
+2017-10-23 09:49:09 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-media: seek on media pipelines that are complete
+ Make sure that a seek is performed on pipelines that
+ contain at least one sink element.
+ Change-Id: Icf398e10add3191d104b1289de612412da326819
+ https://bugzilla.gnome.org/show_bug.cgi?id=788340
+
+2017-10-17 10:44:33 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ * tests/check/gst/client.c:
+ * tests/check/gst/media.c:
+ * tests/check/gst/rtspserver.c:
+ * tests/check/gst/stream.c:
+ Dynamically reconfigure pipeline in PLAY based on transports
+ The initial pipeline does not contain specific transport
+ elements. The receiver and the sender parts are added
+ after PLAY.
+ If the media is shared, the streams are dynamically
+ reconfigured after each PLAY.
+ https://bugzilla.gnome.org/show_bug.cgi?id=788340
+
+2017-10-16 12:40:57 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: obtain stream position from pad
+ If no sinks have been added yet, obtain the current and
+ the stop position of the stream from the send_src pad.
+ Change-Id: Iacd4ab4bdc69f6b49370d06012880ce48a7d595a
+ https://bugzilla.gnome.org/show_bug.cgi?id=788340
+
+2017-10-16 11:35:10 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-media.h:
+ rtsp-session-media: add function to get a list of transports
+ Change-Id: I817e10624da0f3200f24d1b232cff481099278e3
+ https://bugzilla.gnome.org/show_bug.cgi?id=788340
+
+2017-10-16 11:15:55 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-stream: add functions to get rtp and rtcp multicast sockets
+ Change-Id: Iddfe6e0bd250cb0159096d5eba9e4202d22b56db
+ https://bugzilla.gnome.org/show_bug.cgi?id=788340
+
+2017-10-20 12:21:48 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: set async=sync=false only for RTCP appsink
+ Change-Id: I929a218a9adf4759f61322b6f2063aacc5595f90
+ https://bugzilla.gnome.org/show_bug.cgi?id=788340
+
+2017-10-16 10:10:17 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: return minimum value in query position case
+ The minimum position should be returned as we are interested
+ in the whole interval.
+ Change-Id: I30e297fc040c995ae40c25dee8ff56321612fe2b
+ https://bugzilla.gnome.org/show_bug.cgi?id=788340
+
+2017-08-09 11:52:38 +0200 Jonathan Karlsson <jonakn@axis.com>
+
+ * gst/rtsp-server/rtsp-session.c:
+ * tests/check/gst/rtspserver.c:
+ rtsp-session: Handle the case when timeout=0
+ According to the documentation, a timeout of value 0 means
+ that the session never timeouts. This adds handling of that.
+ If timeout=0 we just return with a -1 from
+ gst_rtsp_session_next_timeout_usec ().
+ https://bugzilla.gnome.org/show_bug.cgi?id=785058
+
+2017-07-17 17:15:22 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: Add "accept-certificate" signal for manually checking a TLS certificate for validity
+ https://bugzilla.gnome.org/show_bug.cgi?id=785024
+
+2017-10-26 14:43:19 +0200 Mathieu Duponchelle <mathieu@centricular.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ docs: add media factory transport mode accessors
+ and fix the documentation for the return value of the getter
+
+2017-10-09 12:43:01 +0200 Branko Subasic <branko@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: unref 'pipelined_requests' in finalize
+ The hash table priv->pipelined_requests is not unref:ed in the
+ finalize funktion. Make sure it is.
+ https://bugzilla.gnome.org/show_bug.cgi?id=788704
+
+2017-10-09 14:44:40 +0200 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Initialize scalar variable
+ CID 1418985
+
+2017-10-06 10:27:34 +0200 Edward Hervey <edward@centricular.com>
+
+ * win32/common/libgstrtspserver.def:
+ win32: Update export file
+
+2017-04-22 09:26:07 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ Start support for RTSP 2.0
+ This adds basic support for new 2.0 features, though the protocol is
+ subposdely backward incompatible, most semantics are the sames.
+ This commit adds:
+ - features:
+ * version negotiation
+ * pipelined requests support
+ * Media-Properties support
+ * Accept-Ranges support
+ - APIs:
+ * gst_rtsp_media_seekable
+ The RTSP methods that have been removed when using 2.0 now return
+ BAD_REQUEST.
+ https://bugzilla.gnome.org/show_bug.cgi?id=781446
+
+2017-06-02 15:37:54 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: Use stream duration as stream-stop if segment was not configured with a stop
+ Allowing client to know stream duration when no seeking happened.
+ https://bugzilla.gnome.org/show_bug.cgi?id=783435
+
+2017-09-25 19:40:17 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ rtsp-media-factory: Don't cache any media if NULL was returned as key
+ The docs already mentioned this, but we actually stored it in the hash
+ table with key==NULL and leaked its reference forever.
+
+2017-09-18 19:31:31 +0200 Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ * gst/rtsp-sink/gstrtspclientsink.h:
+ rtspclientsink: Use a mutex for protecting against concurrent send/receives
+ This is a simple port of:
+ * a722f6e8329032c6eda4865d6a07f4ba5981d7ea
+ * c438545dc9e2f14f657bc0ef261fff726449867b
+ * cd17c71dcea5c9310d21f1347c7520983e5869ac
+ in gst-plugins-good.
+
+2017-08-31 13:24:15 +0530 Satya Prakash Gupta <sp.gupta@samsung.com>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ sdp: fix Memory leak in error case
+ https://bugzilla.gnome.org/show_bug.cgi?id=787059
+
+2017-08-18 17:37:01 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * pkgconfig/meson.build:
+ meson: don't install -uninstalled.pc file
+ https://bugzilla.gnome.org/show_bug.cgi?id=786457
+
+2017-08-17 12:26:17 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From 48a5d85 to 3f4aa96
+
+2017-08-14 21:04:23 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Fix typo in debug message
+
+2017-08-11 14:14:32 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ meson: hide symbols by default unless explicitly exported
+
+2017-08-10 14:20:12 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
+ pkgconfig: remove -I@srcdir@/.. which duplicates abs_top_srcdir
+ Fixes meson warning about undefined @srcdir@.
+
+2017-07-21 13:36:00 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/meson.build:
+ meson: skip tests on windows for now
+ As we do in the other modules. As libgstcheck is currently not
+ built on windows. Fixes "Fallback variable 'gst_check_dep' in
+ the subproject 'gstreamer' does not exist"" Meson error.
+
+2017-06-22 07:25:07 -0700 Julien Isorce <julien.isorce@gmail.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: fix connection delay due to wrong assumption on last-sample
+ Commit 852cc09f542af5cadd79ffd7fe79d6475cf57e14 assumed that
+ multiudpsink's last-sample always comes from the payloader. Which
+ is wrong if auxiliary streams are multiplexed in the same stream.
+ So check the buffer's ssrc against the caps'ssrc before to use its
+ seqnum. If not the same ssrc just use the payloader as done prior
+ the commit above or when there is no last-sample yet.
+ https://bugzilla.gnome.org/show_bug.cgi?id=784094
+
+2017-06-23 16:19:04 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * meson.build:
+ meson: Allow using glib as a subproject
+
+2017-06-26 09:55:49 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ meson: fix with-package-name option
+ https://bugzilla.gnome.org/show_bug.cgi?id=784082
+
+2017-06-09 20:16:28 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * Makefile.am:
+ Distribute meson_options.txt
+
+2017-06-09 20:11:47 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * Makefile.am:
+ And config.h.meson is no longer dist either
+
+2017-06-09 21:27:09 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * config.h.meson:
+ * meson.build:
+ meson: config.h.meson is no longer needed
+
+2017-06-07 13:04:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * tests/check/meson.build:
+ * tests/meson.build:
+ meson: Fix building tests and activate them again
+
+2017-06-07 12:55:41 -0400 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * tests/check/meson.build:
+ meson: Do not use path separator in test names
+ Avoiding warnings like:
+ WARNING: Target "elements/audioamplify" has a path separator in its name.
+
+2017-05-20 15:07:31 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ * meson_options.txt:
+ meson: add options to set package name and origin
+ https://bugzilla.gnome.org/show_bug.cgi?id=782172
+
+2017-05-18 10:35:18 +0100 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-params.h:
+ * gst/rtsp-server/rtsp-permissions.h:
+ * gst/rtsp-server/rtsp-sdp.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:
+ Mark symbols explicitly for export with GST_EXPORT
+
+2017-05-16 14:44:43 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * configure.ac:
+ * gst/rtsp-sink/Makefile.am:
+ Remove plugin specific static build option
+ Static and dynamic plugins now have the same interface. The standard
+ --enable-static/--enable-shared toggle are sufficient.
+
+2017-05-04 18:59:14 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ * meson.build:
+ Back to development
+
+=== release 1.12.0 ===
+
+2017-05-04 15:40:46 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ * meson.build:
+ Release 1.12.0
+
+=== release 1.11.91 ===
+
+2017-04-27 17:42:02 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ * meson.build:
+ Release 1.11.91
+
+2017-04-24 20:30:37 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From 60aeef6 to 48a5d85
+
+2017-04-13 14:20:10 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ gi: Fix some annotations and docstrings
+
+2017-04-13 13:52:26 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * gst/rtsp-server/meson.build:
+ * meson.build:
+ * meson_options.txt:
+ meson: Build gir
+
+2017-04-10 23:51:12 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * autogen.sh:
+ * common:
+ Automatic update of common submodule
+ From 39ac2f5 to 60aeef6
+
+=== release 1.11.90 ===
+
+2017-04-07 16:35:03 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ * meson.build:
+ Release 1.11.90
+
+2017-03-27 18:19:33 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/test-launch.c:
+ examples: make test-launch pipeline shared by default as well
+
+2017-02-27 19:10:44 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
+ gstreamer-rtsp-server: Add both srcdir and builddir to the include path
+ Just the build dir is not going to work for srcdir!=builddir.
+
+2017-02-24 15:59:54 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * meson.build:
+ meson: Update version
+
+2017-02-24 15:37:49 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.11.2 ===
+
+2017-02-24 15:10:07 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.11.2
+
+2017-02-14 20:40:26 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * Makefile.am:
+ meson: dist meson build files
+ Ship meson build files in tarballs, so people who use tarballs
+ in their builds can start playing with meson already.
+
+2017-02-07 23:39:37 +1100 Jan Schmidt <jan@centricular.com>
+
+ * examples/test-record.c:
+ examples/test-record: Add extra line to initial printout
+ Add an example line of how to deliver a stream to the
+ RTSP RECORD example
+
+2017-01-19 14:57:19 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Also handle the (S|G)ET_PARAMETER case of size==0 || !data as keep-alive
+ If there is no Content-Length header, no body would be allocated and the
+ '\0' would also not be appended to the body.
+
+2017-01-19 14:24:07 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Fix handling of keep-alive GET_PARAMETER/SET_PARAMETER
+ While they logically have 0 bytes length, GstRTSPConnection is appending
+ a '\0' to everything making the size be 1 instead.
+
+2017-01-13 12:39:36 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ meson: bump version
+
+2017-01-12 19:04:23 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-session.c:
+ rtsp-session: Only remove deprecated API if requested to do so, not just when disabling
+ gst_rtsp_session_is_expired() and gst_rtsp_session_next_timeout() were
+ affected.
+
+2017-01-12 16:32:59 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.11.1 ===
+
+2017-01-12 16:14:46 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ * win32/common/libgstrtspserver.def:
+ Release 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 Sebastian Dröge <slomo@coaxion.net>
+2016-03-01 19:00:45 +0200 Sebastian Dröge <sebastian@centricular.com>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
* configure.ac:
- releasing 1.7.90
+ * gst-rtsp-server.doap:
+ Release 1.7.90
2016-02-26 12:42:51 +0200 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
* pkgconfig/Makefile.am:
- * pkgconfig/gst-rtsp-server-uninstalled.pc.in:
- * pkgconfig/gst-rtsp-server.pc.in:
* 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
2011-01-10 15:10:53 +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-funnel.c:
* gst/rtsp-server/rtsp-funnel.h:
* gst/rtsp-server/rtsp-media.c:
2009-01-29 13:31:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
* examples/Makefile.am:
- * examples/main.c:
* examples/test-mp4.c:
* examples/test-ogg.c:
* examples/test-video.c:
* bindings/vala/gst-rtsp-server-0.10.deps:
* bindings/vala/gst-rtsp-server-0.10.vapi:
- * bindings/vala/gst-rtsp-server.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:
- * bindings/vala/packages/gst-rtsp-server.deps:
- * bindings/vala/packages/gst-rtsp-server.excludes:
- * 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:
Regenerated Vala bindings
2008-12-08 13:19:40 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
* gst/rtsp-server/rtsp-session.c:
* gst/rtsp-server/rtsp-session.h:
* src/Makefile.am:
- * src/main.c:
- * src/rtsp-client.c:
- * src/rtsp-client.h:
- * src/rtsp-media.c:
- * src/rtsp-media.h:
- * src/rtsp-server.c:
- * src/rtsp-server.h:
- * src/rtsp-session-pool.c:
- * src/rtsp-session-pool.h:
- * src/rtsp-session.c:
- * src/rtsp-session.h:
Split in library and example program
2008-11-10 20:59:35 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
projects / platform / upstream / gstreamer.git / blobdiff