+=== release 1.9.2 ===
+
+2016-09-01 Sebastian Dröge <slomo@coaxion.net>
+
+ * configure.ac:
+ releasing 1.9.2
+
+2016-01-27 01:03:52 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * config.h.meson:
+ * examples/meson.build:
+ * gst/meson.build:
+ * gst/rtsp-server/meson.build:
+ * gst/rtsp-sink/meson.build:
+ * meson.build:
+ * pkgconfig/meson.build:
+ * tests/check/meson.build:
+ * tests/meson.build:
+ Add support for Meson as alternative/parallel build system
+ https://github.com/mesonbuild/meson
+
+2016-08-26 21:56:13 +0200 Josep Torra <n770galaxy@gmail.com>
+
+ * configure.ac:
+ * tests/check/Makefile.am:
+ build: silence error about pthread for 'make check' in osx
+ Fixes "clang: error: argument unused during compilation: '-pthread'"
+
+2015-09-25 15:04:00 +0000 Nikita Bobkov <NikitaDBobkov@gmail.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Fix leaking of media in error cases
+ With additional fixes by Kseniya Vasilchuk <vasilchukkseniia@gmail.com>
+ and myself to make the media refcounting a bit easier to follow.
+ https://bugzilla.gnome.org/show_bug.cgi?id=755632
+
+2016-08-02 15:08:22 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Fix leaking of session in error cases
+ https://bugzilla.gnome.org/show_bug.cgi?id=755632
+
+2016-07-11 21:16:04 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From f363b32 to f49c55e
+
+2016-07-06 13:51:15 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.9.1 ===
+
+2016-07-06 13:28:12 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.9.1
+
+2016-06-24 02:02:20 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * configure.ac:
+ configure: Need to add -DGST_STATIC_COMPILATION when building only statically
+ https://bugzilla.gnome.org/show_bug.cgi?id=767463
+
+2016-06-21 11:49:02 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * common:
+ Automatic update of common submodule
+ From ac2f647 to f363b32
+
+2016-04-14 22:56:11 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-sdp.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ sdp: add rollover counters for all sender SSRC
+ We add different crypto sessions in MIKEY, one for each sender
+ SSRC. Currently, all of them will have the same security policy, 0.
+ The rollover counters are obtained from the srtpenc element using the
+ "stats" property.
+ https://bugzilla.gnome.org/show_bug.cgi?id=730539
+
+2016-06-07 20:44:42 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-server.h:
+ docs: fix some typos
+
+2016-05-25 10:28:43 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/Makefile.am:
+ g-i: pass compiler env to g-ir-scanner
+ It's what introspection.mak does as well. Should
+ fix spurious build failures on gnome-continuous
+ (caused by g-ir-scanner getting compiler details
+ via python which is broken in some environments
+ so passing the compiler details bypasses that).
+
+2016-05-18 16:48:44 +0100 Ian <ian.arkver.dev@gmail.com>
+
+ * gst/rtsp-server/rtsp-session.c:
+ rtsp-session: RFC2326 does not allow a space between ; and timeout in the Session header
+ This works with rtspsrc and live555, but fails with e.g. ffmpeg.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766619
+
+2016-03-07 14:48:38 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: Check return value of sscanf
+ And just make sure we always have 0/0 if we have an error
+ CID #1352031
+
+2016-04-25 08:55:25 -0400 Jake Foytik <jake.foytik@ipconfigure.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * tests/check/gst/rtspserver.c:
+ * tests/check/gst/stream.c:
+ rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrc
+ - Unicast udpsrcs are now managed in a hash table. This allows for proper cleanup in with shared streams and fixes a memory leak.
+ - Unicast udpsrcs are now properly cleaned up when shared connections exit. See the update_transport() function.
+ - Create unit test for shared media.
+ https://bugzilla.gnome.org/show_bug.cgi?id=764744
+
+2016-04-11 10:55:23 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Always bind to ANY when address is a multicast address and not only on Windows
+ For IPv6 addresses, binding to a multicast group does not work on Linux
+ either. Always bind to ANY and then later join the multicast group.
+ https://bugzilla.gnome.org/show_bug.cgi?id=764679
+
+2016-04-14 10:05:02 +0100 Julien Isorce <j.isorce@samsung.com>
+
+ * common:
+ Automatic update of common submodule
+ From 6f2d209 to ac2f647
+
+2016-04-06 10:09:46 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ rtsp-thread-pool: explained why GSource is a part of ThreadImpl
+ Clarified why it is necessary to add source information to
+ GstRTSPThreadImpl. See the reported bug in GLib:
+ https://bugzilla.gnome.org/show_bug.cgi?id=720186
+ for more information.
+ https://bugzilla.gnome.org/show_bug.cgi?id=761702
+
+2016-04-04 12:58:38 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/Makefile.am:
+ examples: Clean up CFLAGS/LDADD even more
+ The internal .la should come first and is part of LDADD, as is
+ GST_CFLAGS/LIBS.
+
+2016-04-04 12:39:39 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/Makefile.am:
+ examples: Clean up CFLAGS/LDADD to link with the correct versions of all libraries
+
+2016-04-03 12:06:29 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/Makefile.am:
+ rtsp-server: Use $(GST_NET_LIBS) / $(GST_NET_CFLAGS)
+
+2015-12-30 18:39:05 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-server: Implement clock signalling according to RFC7273
+ For NTP and PTP clocks we signal the actual clock that is used and signal
+ the direct media clock offset.
+ For all other clocks we at least signal that it's the local sender clock.
+ This allows receivers to know which clock was used to generate the media and
+ its RTP timestamps. Receivers can then implement network synchronization,
+ either absolute or at least relative by getting the sender clock rate directly
+ via NTP/PTP instead of estimating it from RTP timestamps and packet receive
+ times.
+ https://bugzilla.gnome.org/show_bug.cgi?id=760005
+
+2016-03-02 19:42:58 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: Add support for setting the multicast interface
+ https://bugzilla.gnome.org/show_bug.cgi?id=763000
+
+2016-03-02 19:42:13 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-media: Add support for setting the multicast interface
+ https://bugzilla.gnome.org/show_bug.cgi?id=763000
+
+2016-03-07 08:50:01 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: use new gst_element_class_add_static_pad_template()
+ https://bugzilla.gnome.org/show_bug.cgi?id=763196
+
+2016-03-24 13:33:43 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.8.0 ===
+
+2016-03-24 13:00:35 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.8.0
+
+2016-03-16 23:35:09 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Don't set the state of the appsrc from PLAYING to PAUSED again during setup
+ This would get us NO_PREROLL in the bin again and break seeking.
+ Thanks to Carlos Rafael Giani for helping to debug this!
+ https://bugzilla.gnome.org/show_bug.cgi?id=740509
+
+=== release 1.7.91 ===
+
+2016-03-15 12:26:13 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.7.91
+
+2016-03-10 13:54:38 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Ensure that the pipeline is live and later-added udpsrcs are syncing the state with the parent bin
+ Without this, RECORD pipelines are broken because
+ a) we wait for ASYNC_DONE which never happens anymore because udpsrc would be
+ added later. Previously it was there earlier and due to NO_PREROLL caused the
+ pipeline to preroll immediately
+ b) the udpsrc for the pipeline is added later and never set to PLAYING state,
+ as the corresponding code previously was only for PLAY pipelines.
+ https://bugzilla.gnome.org/show_bug.cgi?id=763281
+
+2016-03-11 01:22:54 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Fix typo in the docstring
+ gst_rtsp_stream_set_client_side -> gst_rtsp_stream_is_client_side
+
+2016-03-05 10:52:11 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Disable multicast loopback for all our sockets
+ On Windows this is a receiver-side setting, on Linux a sender-side setting. As
+ we provide a socket ourselves to udpsrc, udpsrc is never setting the multicast
+ loopback setting on the socket... while udpsink does which unfortunately has
+ no effect here on Windows but on Linux.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757488
+
+2016-03-03 15:07:06 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * tests/check/gst/stream.c:
+ stream tests: added new tests
+ Test a case when the address pool only contains multicast addresses
+ and the client is requesting unicast udp.
+ Added tests for multicast ports allocation.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757488
+
+2016-03-04 13:51:12 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Only bind multicast sockets to ANY on Windows
+ On Linux it is still needed to bind to the multicast address
+ to filter out random other packets, while on Windows binding
+ to multicast addresses just fails.
+
+2016-03-03 10:41:51 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Only use the address pool for unicast UDP if it contains unicast addresses
+ Otherwise we fail to allocate UDP ports if the pool only contains multicast
+ addresses, which is something that used to work before. For unicast addresses
+ if the pool contains none, we just allocate them as if there is no pool at
+ all.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757488
+
+2016-03-02 11:48:49 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-server: Fix indentation
+
+2016-03-02 11:47:47 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Don't bind the sockets to multicast addresses
+ This works on Linux but fails completely on Windows. You're supposed
+ to bind to ANY and then join the multicast group.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757488
+
+=== release 1.7.90 ===
+
+2016-03-01 19:00:45 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.7.90
+
+2016-02-26 12:42:51 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From b64f03f to 6f2d209
+
+2016-02-24 00:10:52 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ * tests/check/gst/rtspclientsink.c:
+ rtspsink: Fix some leaks in rtspclientsink and the unit test.
+ https://bugzilla.gnome.org/show_bug.cgi?id=762525
+
+2016-02-23 15:01:22 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * tests/check/gst/media.c:
+ * tests/check/gst/rtspclientsink.c:
+ * tests/check/gst/rtspserver.c:
+ * tests/check/gst/stream.c:
+ tests: unit test fixes
+ Removed port allocation test from the media suite.
+ The port allocation failure is now in the stream suite.
+ rtspserver:
+ Make sure that the media is suspended after the DESCRIBE request
+ before reconfiguring the UDP sinks.
+ rtspclientsink:
+ In the RECORD case we have to set async property to false
+ for the appsink element in the test in order to make sure
+ that the media pipeline doesn't hang in start_preroll().
+ https://bugzilla.gnome.org/show_bug.cgi?id=757488
+
+2016-02-23 14:59:32 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-stream: postpone UDP socket allocation until SETUP
+ Postpone the allocation of the UDP sockets until we know
+ what transport has been chosen by the client.
+ Both unicast and multicast UDP sources are created in one
+ function.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757488
+
+2016-01-13 11:29:35 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: postpone the creation of the UDP sources
+ Code refactoring: allocate the UDP ports after the sender and
+ the reciver parts have been created.
+ We postpone the creation of the UDP sources until the UDP
+ ports have been allocated.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757488
+
+2016-01-13 10:55:40 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: added function for setting UDP sources to PLAYING state
+ Code refactoring: Introduced a function for setting UDP sources
+ to PLAYING state.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757488
+
+2015-11-20 15:34:43 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: added function for creating and configuring UDP sources
+ Code refactoring: create and configure UDP sources in a separate function.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757488
+
+2015-11-20 14:43:38 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: added function for RTP/RTCP socket configuration
+ Code refactoring: configure RTP and RTCP sockets for UDP sinks
+ in a separate function.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757488
+
+2015-11-20 08:38:42 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: added function for creating and configuring UDP sinks
+ Code refactoring: create and configure UDP sinks in a separate function.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757488
+
+2015-11-19 14:09:25 +0100 Patricia Muscalu <patricia@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: added helper function for creating the sender/receiver parts
+ Code refactoring: introduced helper function for creating
+ the receiver and the sender parts of the streaming pipeline.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757488
+
+2016-02-19 12:38:42 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.7.2 ===
+
+2016-02-19 12:03:18 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.7.2
+
+2016-02-18 15:20:05 +0000 Julien Isorce <j.isorce@samsung.com>
+
+ * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
+ uninstalled.pc: add support for non libtool build systems
+ Currently the .la path is provided which requires to use libtool as
+ mentioned in the GStreamer manual section-helloworld-compilerun.html.
+ It is fine as long as the application is built using libtool.
+ So currently it is not possible to compile a GStreamer application
+ within gst-uninstalled with CMake or other build system different
+ than autotools.
+ This patch allows to do the following in gst-uninstalled env:
+ gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \
+ gstreamer-rtsp-server-1.0)
+ Previously it required to prepend libtool --mode=link
+ https://bugzilla.gnome.org/show_bug.cgi?id=720778
+
+2016-02-09 10:34:22 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: remove check for impossible condition
+ Goto error label checks stream to see if it needs to be unreferenced before
+ returning, but this goto jumps happens before the stream is ever set, so it
+ will always be NULL in this error label.
+ CID #1352034
+
+2016-02-08 23:33:03 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: clean switch statements
+ Coverity demands for fallthrough statements to be clearly commented,
+ to distinguish from accidental fall throughs. And it also needs all
+ cases to finish with a break, even if the break is never going to be
+ executed like in the case of a continue jump.
+ CID #1352039
+ CID #1352040
+
+2016-02-05 20:03:01 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/Makefile.am:
+ tests: extend the AM_TESTS_ENVIRONMENT from check.mak
+ To get the CK_DEFAULT_TIMEOUT defined for all tests
+ Also removes a 120 seconds timeout that was set as default
+ explicitly in this module
+ https://bugzilla.gnome.org/show_bug.cgi?id=761472
+
+2016-02-05 18:11:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * autogen.sh:
+ * common:
+ Automatic update of common submodule
+ From 86e4663 to b64f03f
+
+2016-02-02 09:01:51 +0100 Steven Hoving <sh@bigbrother.nl>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: fix state_lock not locked again when preroll fails
+ https://bugzilla.gnome.org/show_bug.cgi?id=761399
+
+2016-01-28 22:05:56 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ configure: Move plugin specific flags below all the others
+ They use some of the other flags, like $GST_ALL_LDFLAGS which is adding
+ -no-undefined. And -no-undefined is required on Windows to build DLLs.
+
+2016-01-28 04:58:00 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ rtspclientsink: Simplify slightly using new -base API
+ Use the new Mikey and SDP API in the base plugins libs
+ to simplify some code.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758180
+
+2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
+
+ * .gitignore:
+ * configure.ac:
+ * gst/Makefile.am:
+ * gst/rtsp-sink/Makefile.am:
+ * gst/rtsp-sink/gstrtspclientsink.c:
+ * gst/rtsp-sink/gstrtspclientsink.h:
+ * gst/rtsp-sink/plugin.c:
+ * tests/check/Makefile.am:
+ * tests/check/gst/rtspclientsink.c:
+ rtspsink: Add rtspclientsink element
+ Add an rtspclientsink element that accepts streams for which
+ there is a registered payloader and sends them to
+ an RTSP server using RECORD.
+ Sending is synchronised to the pipeline clock. Payload-types
+ are automatically selected. The 'new-payloader' signal is fired
+ for custom configuration of payloaders when they are created.
+ Can now stream a movie like this:
+ receiver:
+ ./test-record "( decodebin name=depay0 ! videoconvert ! autovideosink \
+ decodebin name=depay1 ! audioconvert ! autoaudiosink )"
+ sender:
+ gst-launch-1.0 filesrc location=file-with-aac-and-h264.mp4 ! qtdemux name=d ! \
+ queue ! aacparse ! rtspclientsink location=rtsp://127.0.0.1:8554/test name=s \
+ https://bugzilla.gnome.org/show_bug.cgi?id=758180
+
+2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-stream: Add functions for using rtsp-stream from the client
+ Add a boolean to indicate that the rtsp-stream is running on the
+ 'client' side of an RTSP connection, for sending streams via
+ RECORD. In that case, the roles of the client/server ports
+ in transport setup are swapped.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758180
+
+2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-sdp.h:
+ rtsp-sdp: Add gst_rtsp_sdp_from_stream()
+ A new function that adds info from a GstRTSPStream into an SDP message.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758180
+
+2016-01-28 09:22:18 +0100 Steven Hoving <sh@bigbrother.nl>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Fix mutex beeing unlocked while they should be locked
+ https://bugzilla.gnome.org/show_bug.cgi?id=761226
+
+2016-01-15 07:01:37 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ rtsp-media-factory: add missing break in "clock" property setter
+ CID 1348453
+
+2016-01-05 13:10:36 +0100 Srimanta Panda <srimanta@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: fixed assert during update transport
+ When RTSP server trying update transport during multicast, it throws an
+ assert. The assert is thrown because it is trying to get the parent of
+ an non-existing funnel element.
+ https://bugzilla.gnome.org/show_bug.cgi?id=760150
+
+2016-01-03 17:26:31 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-permissions.h:
+ * gst/rtsp-server/rtsp-thread-pool.h:
+ * gst/rtsp-server/rtsp-token.h:
+ docs: remove dummy function declarations with G_INLINE_FUNC for gtk-doc
+ gtk-doc can handle static inline functions just fine these days,
+ there's no need for this stuff any more.
+
+2015-10-07 18:53:01 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ sdp: replace duplicated codes to call new base sdp apis
+ https://bugzilla.gnome.org/show_bug.cgi?id=745880
+
+2015-12-30 16:34:30 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-netclock.c:
+ test-netclock: Use the new API to configure a clock directly
+
+2015-12-30 16:31:13 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ rtsp-media: Add API to directly configure a clock on the media pipelines
+
+2015-12-30 16:43:17 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Fix typo in docs gst_rtsp_media_set_latncy() -> latency()
+
+2015-12-30 16:30:38 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ rtsp-media-factory: Add FIXME for 2.0
+
+2015-12-30 16:29:45 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Fix indentation
+
+2015-12-22 12:08:02 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Do not prepare media after media times out
+ Deferred calls to start_prepare() can be deferred past the point until
+ which wait_preroll() and by proxy gst_rtsp_media_get_status() is
+ prepared to wait. Previously there was no lock and no check for this
+ situation. This meant that a media could be prepared and unprepared
+ simultaneously by two different threads. Now a lock is in place and a
+ suitable check is done.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759773
+
+2015-12-09 18:24:24 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN
+ Without TEARDOWN it might be desireable to keep the media running and continue
+ sending data to the client, even if the RTSP connection itself is
+ disconnected.
+ Only do this for session medias that have only UDP transports. If there's at
+ least on TCP transport, it will stop working and cause problems when the
+ connection is disconnected.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758999
+
+2015-12-24 15:29:33 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.7.1 ===
+
+2015-12-24 14:54:06 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.7.1
+
+2015-12-21 00:43:49 +0100 Koop Mast <kwm@rainbow-runner.nl>
+
+ * configure.ac:
+ configure: Make -Bsymbolic check work with clang.
+ Update the -Bsymbolic check with the version glib has. This version
+ works with clang.
+ https://bugzilla.gnome.org/show_bug.cgi?id=759713
+
+2015-11-17 22:30:54 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ rtsp-session-pool: Avoid dollar sign ($) in session ids
+ Live555 in VLC strips off dollar signs and then gets very confused,
+ we don't loose too much entropy by just skipping it.
+
+2015-11-10 14:17:18 -0500 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/rtsp-server/rtsp-address-pool.h:
+ * gst/rtsp-server/rtsp-auth.h:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media-factory-uri.h:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-mount-points.h:
+ * gst/rtsp-server/rtsp-permissions.h:
+ * gst/rtsp-server/rtsp-server.h:
+ * gst/rtsp-server/rtsp-session-media.h:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ * gst/rtsp-server/rtsp-session.h:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ * gst/rtsp-server/rtsp-stream.h:
+ * gst/rtsp-server/rtsp-thread-pool.h:
+ * gst/rtsp-server/rtsp-token.h:
+ rtsp-server: Add g_autoptr() support to all types
+ https://bugzilla.gnome.org/show_bug.cgi?id=754464
+
+2015-12-08 08:27:20 +0100 Srimanta Panda <srimanta@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: fixed valgrind error
+ Fixed the valgrind error in unit test. The UDP source created during
+ gst_rtsp_stream_join_bin() was not released while destroying the rtp
+ bin.
+ https://bugzilla.gnome.org/show_bug.cgi?id=759010
+
+2015-12-07 09:11:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * autogen.sh:
+ * common:
+ Automatic update of common submodule
+ From b319909 to 86e4663
+
+2015-11-18 11:14:39 +0100 Srimanta Panda <srimanta@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: suspend media during setup request
+ SETUP request from clients needs to suspend the media to clear the
+ prerolled buffers. Otherwise it will not affect the prerolled buffer
+ and the prerolled buffers will be incorrect (for example block-size
+ from setup request will not affect the prerolled buffer unless the
+ media is suspended).
+ https://bugzilla.gnome.org/show_bug.cgi?id=758268
+
+2015-12-04 08:01:37 +0100 Srimanta Panda <srimanta@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: create stream pipeline based on transport
+ Based on the protocol, create the rtsp stream pipeline. If only TCP or
+ only UDP is set as the transport protocol, it will not add the extra tee
+ or queue element to the pipeline. Both these elements will be added, if
+ it supports both TCP and UDP protocols. This improves the pipeline
+ performance when one protocol is present.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758179
+
+2015-11-19 15:01:16 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Only create RTP sending/receiving rtpbin pads if needed
+ Adding them when not needed will start some logic inside rtpbin that might be
+ problematic. Also if e.g. for a sender media we suddenly receive RTP data, we
+ would start up a rtpjitterbuffer and behave in weird ways.
+ We still set up the UDP sources for RTP receiving for a sender media to be
+ able to receive any packets sent by the client for NAT traversal. They will
+ all go to a fakesink though.
+ Having an rtpjitterbuffer in the media pipeline will cause the pipeline to be
+ NO_PREROLL, which will cause deadlocks when seeking the media as it will never
+ receive ASYNC_DONE after a seek.
+ https://bugzilla.gnome.org/show_bug.cgi?id=758319
+
+2015-11-17 12:44:38 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Disable multicast loopback for the multicast udp sources too
+ On POSIX this setting is for sender sockets, on Windows for receiver sockets.
+ Previously we were only setting this for sender sockets, which caused looped
+ back packets to be received on Windows if a multicast transport was used.
+
+2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
+
+ * examples/test-record-auth.c:
+ * examples/test-record.c:
+ examples: Actually use the provided port in the record examples
+
+2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
+
+ * examples/test-record-auth.c:
+ test-record-auth: Add the option to build in TLS support
+
+2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
+
+ * examples/test-auth.c:
+ test-auth: Use an 'anonymous' user for unauthenticated default
+ There's a comment on one of the resources that 'user' and 'admin'
+ shouldn't even be able to see it, but they can if the default
+ token is 'admin2', since that gives them access anyway.
+
+2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
+
+ * examples/.gitignore:
+ * examples/Makefile.am:
+ * examples/test-record-auth.c:
+ Add test-record-auth example
+
+2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * tests/check/gst/client.c:
+ rtsp-client: Report RECORD and ANNOUNCE as supported in the OPTIONS
+
+2015-11-11 14:58:33 +0100 Marcus Prebble <prebble@axis.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ rtsp-server: Change the logic so we don't pop a NULL context
+ When doing a port scan (e.g. with nmap) the call to GST_RTSP_CHECK()
+ will sometimes fail. This call is made before any context is pushed
+ resulting in an attempt to pop a NULL context.
+ https://bugzilla.gnome.org/show_bug.cgi?id=757949
+
+2015-10-22 14:32:30 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * tests/check/gst/rtspserver.c:
+ rtspserver: Add udp-mcast transport SETUP test
+ Refactor utility functions in the test file so they can handle
+ more than UDP and TCP as lower transport.
+ https://bugzilla.gnome.org/show_bug.cgi?id=756969
+
+2015-10-22 09:15:21 +0200 David Svensson Fors <davidsf@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Always unref return value of gst_object_get_parent()
+ Fixes a leak of a GstBin in the udp-mcast case.
+ https://bugzilla.gnome.org/show_bug.cgi?id=756968
+
+2015-10-21 14:37:19 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From b99800a to b319909
+
+2015-10-20 17:29:42 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Use new GST_ENABLE_EXTRA_CHECKS #define
+ https://bugzilla.gnome.org/show_bug.cgi?id=756870
+
+2015-10-21 14:28:47 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From 6babecd to b99800a
+
+2015-10-02 22:25:47 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Update GLib dependency to 2.40.0
+
+2015-10-02 16:11:05 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * examples/test-mp4.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: listen to sender ssrc signals
+ https://bugzilla.gnome.org/show_bug.cgi?id=746747
+
+2015-09-29 13:00:51 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * common:
+ common: update for new suppression
+ Makes check-valgrind pass with glib 2.46
+
+2015-09-28 17:40:59 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Take reference to media that will be prepared
+ default_prepare() takes a transfer-none reference GstRTSPMedia object.
+ Later on a g_idle_source_new() is created and a pointer to the media
+ object is passed as user data. If the media is freed before the idle
+ source is dispatched the media object pointer is invalid, but the idle
+ source callback expects it to still be valid. To fix this a reference to
+ the media object is taken when registering the source callback function
+ and a corresponding release of the reference is done when the souce is
+ destroyed.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755748
+
+2015-08-20 17:01:24 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * examples/test-launch.c:
+ * examples/test-mp4.c:
+ * examples/test-ogg.c:
+ * examples/test-record.c:
+ * examples/test-uri.c:
+ rtsp-server: Fix memory leaks when context parse fails
+ When g_option_context_parse fails, context and error variables are not getting free'd
+ which results in memory leaks. Free'ing the same.
+ And replacing g_error_free with g_clear_error, which checks if the error being passed
+ is not NULL and sets the variable to NULL on free'ing.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753863
+
+2015-09-25 23:51:17 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.6.0 ===
+
+2015-09-25 23:32:52 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.6.0
+
+=== release 1.5.91 ===
+
+2015-09-18 20:12:06 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.5.91
+
+2015-09-17 20:07:34 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: fix docs for recently-added get/set_buffer_size API
+ https://bugzilla.gnome.org/show_bug.cgi?id=749095
+
+2015-09-04 11:23:43 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Don't crash on encrypted RTX SDP
+ In parse_keymgmt(), don't mutate the input string that's been passed
+ as const, especially since we might need the original value again if
+ the same key info applies to multiple streams (RTX, for example).
+ https://bugzilla.gnome.org/show_bug.cgi?id=754753
+
+2015-08-22 20:59:40 +1000 Jan Schmidt <jan@centricular.com>
+
+ * examples/test-mp4.c:
+ test-mp4: Support filenames with spaces in them. Error out on too few arguments
+
+2015-08-17 02:36:31 +1000 Jan Schmidt <jan@centricular.com>
+
+ * examples/test-record.c:
+ test-record: Check parameter count and print out help
+ If no launch pipeline was supplied, print out some help
+
+2015-08-31 22:48:34 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-stream: Implement UDP buffer size setting.
+ Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
+ UDP TX buffer size.
+ Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
+
+2015-08-31 22:47:45 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.h:
+ rtsp-media: Fix small typo causing gtk-doc to complain
+
+=== release 1.5.90 ===
+
+2015-08-19 14:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.5.90
+
+2015-08-12 14:33:44 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ media-factory: get port number through gst_rtsp_url_get_port
+ https://bugzilla.gnome.org/show_bug.cgi?id=753473
+
+2015-08-13 11:24:10 +0200 Francisco Velazquez <francisv@ifi.uio.no>
+
+ * tests/check/gst/media.c:
+ media-test: Removing unnecessary assertion
+ https://bugzilla.gnome.org/show_bug.cgi?id=753385
+
+2015-07-23 14:50:30 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ Document that source keeps a ref on server until it's destroyed
+ https://bugzilla.gnome.org/show_bug.cgi?id=749227
+
+2015-08-08 11:09:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * tests/check/gst/media.c:
+ media-test: Test for multiple dynamic payload
+ https://bugzilla.gnome.org/show_bug.cgi?id=753385
+
+2015-08-08 09:40:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: Only add fakesink once per pipeline
+ The intention is to prevent going PLAYING state before pads are created.
+ If there was mutilple dynamic payload, it would leak few fakesink and
+ actually prevent from ever reaching playing state.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753385
+
+2015-08-08 09:08:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ Revert "rtsp-media: Only add 1 fakesink per pipeline"
+ This reverts commit 22bf61f16c1210bb458fc3f53642179a0211104f.
+
+2015-08-07 09:21:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Only add 1 fakesink per pipeline
+ There should be only one fakesink per pipeline, not per dynpay. This
+ would lead to element naming clash.
+
+2015-07-30 15:32:43 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: assertion error due to wrong condition check
+ In media to caps function, reserved_keys array is being used for variable i,
+ leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
+ changed it to variable j
+ https://bugzilla.gnome.org/show_bug.cgi?id=753009
+
+2015-07-29 11:27:05 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Strip keys from the fmtp that we use internally in our caps
+ Skip keys from the fmtp, which we already use ourselves for the
+ caps. Some software is adding random things like clock-rate into
+ the fmtp, and we would otherwise here set a string-typed clock-rate
+ in the caps... and thus fail to create valid RTP caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=753009
+
+2015-07-20 16:37:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
+ https://bugzilla.gnome.org/show_bug.cgi?id=752640
+
+2015-07-03 22:00:00 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From f74b2df to 9aed1d7
+
+2015-06-25 00:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
=== release 1.5.2 ===
-2015-06-24 Sebastian Dröge <slomo@coaxion.net>
+2015-06-24 23:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
* configure.ac:
- releasing 1.5.2
+ * gst-rtsp-server.doap:
+ Release 1.5.2
2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.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