+=== release 1.6.0 ===
+
+2015-09-25 Sebastian Dröge <slomo@coaxion.net>
+
+ * configure.ac:
+ releasing 1.6.0
+
+=== release 1.5.91 ===
+
+2015-09-18 20:12:06 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.5.91
+
+2015-09-17 20:07:34 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: fix docs for recently-added get/set_buffer_size API
+ https://bugzilla.gnome.org/show_bug.cgi?id=749095
+
+2015-09-04 11:23:43 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Don't crash on encrypted RTX SDP
+ In parse_keymgmt(), don't mutate the input string that's been passed
+ as const, especially since we might need the original value again if
+ the same key info applies to multiple streams (RTX, for example).
+ https://bugzilla.gnome.org/show_bug.cgi?id=754753
+
+2015-08-22 20:59:40 +1000 Jan Schmidt <jan@centricular.com>
+
+ * examples/test-mp4.c:
+ test-mp4: Support filenames with spaces in them. Error out on too few arguments
+
+2015-08-17 02:36:31 +1000 Jan Schmidt <jan@centricular.com>
+
+ * examples/test-record.c:
+ test-record: Check parameter count and print out help
+ If no launch pipeline was supplied, print out some help
+
+2015-08-31 22:48:34 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-stream: Implement UDP buffer size setting.
+ Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
+ UDP TX buffer size.
+ Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
+
+2015-08-31 22:47:45 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.h:
+ rtsp-media: Fix small typo causing gtk-doc to complain
+
+=== release 1.5.90 ===
+
+2015-08-19 14:15:23 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.5.90
+
+2015-08-12 14:33:44 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ media-factory: get port number through gst_rtsp_url_get_port
+ https://bugzilla.gnome.org/show_bug.cgi?id=753473
+
+2015-08-13 11:24:10 +0200 Francisco Velazquez <francisv@ifi.uio.no>
+
+ * tests/check/gst/media.c:
+ media-test: Removing unnecessary assertion
+ https://bugzilla.gnome.org/show_bug.cgi?id=753385
+
+2015-07-23 14:50:30 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ Document that source keeps a ref on server until it's destroyed
+ https://bugzilla.gnome.org/show_bug.cgi?id=749227
+
+2015-08-08 11:09:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * tests/check/gst/media.c:
+ media-test: Test for multiple dynamic payload
+ https://bugzilla.gnome.org/show_bug.cgi?id=753385
+
+2015-08-08 09:40:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: Only add fakesink once per pipeline
+ The intention is to prevent going PLAYING state before pads are created.
+ If there was mutilple dynamic payload, it would leak few fakesink and
+ actually prevent from ever reaching playing state.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753385
+
+2015-08-08 09:08:37 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ Revert "rtsp-media: Only add 1 fakesink per pipeline"
+ This reverts commit 22bf61f16c1210bb458fc3f53642179a0211104f.
+
+2015-08-07 09:21:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Only add 1 fakesink per pipeline
+ There should be only one fakesink per pipeline, not per dynpay. This
+ would lead to element naming clash.
+
+2015-07-30 15:32:43 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: assertion error due to wrong condition check
+ In media to caps function, reserved_keys array is being used for variable i,
+ leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
+ changed it to variable j
+ https://bugzilla.gnome.org/show_bug.cgi?id=753009
+
+2015-07-29 11:27:05 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Strip keys from the fmtp that we use internally in our caps
+ Skip keys from the fmtp, which we already use ourselves for the
+ caps. Some software is adding random things like clock-rate into
+ the fmtp, and we would otherwise here set a string-typed clock-rate
+ in the caps... and thus fail to create valid RTP caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=753009
+
+2015-07-20 16:37:44 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * gst/rtsp-server/rtsp-thread-pool.c:
+ threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
+ https://bugzilla.gnome.org/show_bug.cgi?id=752640
+
+2015-07-03 22:00:00 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From f74b2df to 9aed1d7
+
+2015-06-25 00:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.5.2 ===
+
+2015-06-24 23:44:37 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.5.2
+
+2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * tests/check/gst/client.c:
+ rtsp-client: allow application to decide what requirements are supported
+ Add "check-requirements" signal and vfunc to allow application
+ (and subclasses) to check the requirements.
+ Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
+ https://bugzilla.gnome.org/show_bug.cgi?id=749417
+
+2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 6015d26 to f74b2df
+
+2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Always use real payloader when creating streams
+ A bin that contains the real payloader might be used as payloader. In this
+ case we have to get the real payloader for the various properties it provides.
+ Example use cases for this are bins that payload some media and then have
+ additional elements that add metadata or RTP extension headers to the stream.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750800
+
+2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-netclock-client.c:
+ test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
+
+2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-netclock-client.c:
+ * examples/test-netclock.c:
+ test-netclock: Use new ntp-time-source property on rtpbin
+ Select the clock time to be used as NTP time source. This allows proper
+ synchronization between receivers, independent of sharing base times, and just
+ requires them to use the same clock.
+
+2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-netclock-client.c:
+ * examples/test-netclock.c:
+ test-netclock: Setting the same base time on sender and receiver is not necessary
+ It's going to be fixed up by rtpbin when using ntp-sync=TRUE
+
+2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
+ https://bugzilla.gnome.org/show_bug.cgi?id=750764
+
+2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * docs/libs/gst-rtsp-server.types:
+ docs: add missing types
+ https://bugzilla.gnome.org/show_bug.cgi?id=750764
+
+2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ docs: add missing apis
+ https://bugzilla.gnome.org/show_bug.cgi?id=750764
+
+2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-netclock-client.c:
+ test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
+
+2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-auth.c:
+ * gst/rtsp-server/rtsp-auth.h:
+ GstRTSPAuth: Add client certificate authentication support
+ https://bugzilla.gnome.org/show_bug.cgi?id=750471
+
+2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-netclock-client.c:
+ test-netclock-client: Use new GstClock API to wait for clock synchronization
+
+2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-netclock-client.c:
+ test-netclock-client: Use a GMainLoop and playbin's source-setup signal
+ A mainloop is needed to get glimagesink to display something on OSX, and
+ the source-setup signal just makes things a little bit easier.
+
+2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * common:
+ Automatic update of common submodule
+ From d9a3353 to 6015d26
+
+2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From d37af32 to d9a3353
+
+2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 21ba2e5 to d37af32
+
+2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From c408583 to 21ba2e5
+
+2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * docs/libs/Makefile.am:
+ docs: remove variables that we define in the snippet from common
+ This is syncing our Makefile.am with upstream gtkdoc.
+
+2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 44a3517 to c408583
+
+2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.5.1 ===
+
+2015-06-07 11:20:01 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.5.1
+
+2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: No flush during Teardown.
+ When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
+ backlog is empty it can happen that just a part of a message will be
+ sent and rest is in backlog queue. If then flush during teardown
+ just a part of message will be sent.This can lead to client miss
+ teardown response since it expect to get the last part of message.
+ The flushing during teardown was introduced to fix a deadlock that now
+ is fixed more generally in handle_request by temporary setting backlog
+ size to unlimited.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
+
+2015-05-27 17:04:41 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/Makefile.am:
+ tests: Use AM_TESTS_ENVIRONMENT
+ Needed by the new automake test runner and the
+ current version of the common submodule.
+
+2015-05-20 17:05:47 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-server: Use single-include rtsp header to make sure we get all definitions
+
+2015-05-05 16:46:57 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Mark some more functions static
+
+2015-05-05 16:46:19 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Only unblock the media in suspend() when actually changing the state
+ Otherwise we're going to lose a few packets for live streams during DESCRIBE.
+
+2015-05-04 16:33:08 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-video-rtx.c:
+ examples: Use AVPF profile for the RTX example
+
+2015-05-04 16:31:20 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ rtsp-sdp: Only add RTX to the SDP when using a feedback profile
+
+2015-04-27 19:35:53 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: get valid clock-rate from last-sample
+ clock-rate in last-sample's caps is integer, not unsigned.
+ To get this value properly, variable needs to be type-casted to int.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747614
+
+2015-04-26 15:00:05 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * autogen.sh:
+ * common:
+ autogen.sh: only run autopoint if gettext requested in configure.ac
+ Not just because there happens to be a po directory.
+ https://bugzilla.gnome.org/show_bug.cgi?id=748058
+
+2015-04-26 14:58:49 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ Revert "configure.ac: uncomment gettext version setup"
+ This reverts commit 1545d8fef7065081079172ec264a0061039ac075.
+ We don't need a gettext setup here and there's no po
+ directory either, so no reason why autopoint would be
+ run in the first place.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=748058
+
+2015-04-23 18:53:08 +0100 Alistair Buxton <a.j.buxton@gmail.com>
+
+ * examples/test-multicast.c:
+ * examples/test-multicast2.c:
+ * examples/test-sdp.c:
+ * examples/test-video-rtx.c:
+ * examples/test-video.c:
+ * tests/test-cleanup.c:
+ * tests/test-reuse.c:
+ Fix timeout function signatures across tests and examples
+
+2015-04-23 17:27:40 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/Makefile.am:
+ tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
+ Make sure the test environment is set up.
+ https://bugzilla.gnome.org//show_bug.cgi?id=747624
+
+2015-04-23 17:22:59 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ configure: bump automake requirement to 1.14 and autoconf to 2.69
+ This is only required for builds from git, people can still
+ build tarballs if they only have older autotools.
+ https://bugzilla.gnome.org//show_bug.cgi?id=747624
+
+2015-04-20 08:49:57 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * configure.ac:
+ configure.ac: uncomment gettext version setup
+ Fixes autogen.sh. It would run autopoint, which would complain
+ that it could not find the gettext version in configure.ac.
+ https://bugzilla.gnome.org/show_bug.cgi?id=748058
+
+2015-04-15 10:06:30 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * examples/test-video-rtx.c:
+ test-video-rtx: set exact payload type to PCMA payloader
+ Setting wrong payload type causes failure to do retransmission through audio stream
+ https://bugzilla.gnome.org/show_bug.cgi?id=747839
+
+2015-04-15 09:45:23 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-stream: fix to get valid each stream data for request-aux-sender signal
+ Because of duplicated g_signal_connect for request-aux-sender signal,
+ wrong stream pointer is passed to the signal handler.
+ Instead of passing each stream, pass stream array and get the relevant stream.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747839
+
+2015-04-06 10:32:52 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * acinclude.m4:
+ * autogen.sh:
+ Update autogen.sh to latest version from common
+ Fixes build after aclocal_check etc. helpers have been removed.
+
+2015-04-03 18:58:26 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From bc76a8b to c8fb372
+
+2015-03-23 21:03:20 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Limit the queues to 1 buffer
+ We only need them to be able to pre-roll, queueing up more data here
+ is only going to harm latency and memory usage.
+
+2015-03-23 20:59:52 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Update comment and ASCII art to the latest code
+ We have a queue in front of the udpsink too to prevent the pipeline from
+ locking up.
+
+2015-03-21 11:04:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-media: Properly return first rtptime
+ Instead we where returning first GstBuffer timestamp. This would result
+ in clock skew and unwanted behaviour in RTSP playback.
+ https://bugzilla.gnome.org/show_bug.cgi?id=746479
+
+2015-03-18 16:44:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Don't leave buffer mapped
+ If the seq is NULL, the RTP buffer was left mapped. We should always
+ unmap the buffer.
+
+2015-03-15 12:27:39 +0000 Sebastian Dröge <sebastian@centricular.com>
+
+ * README:
+ Fix typo in README
+
+2015-03-10 09:39:22 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * tests/check/gst/client.c:
+ Fix double semicolons
+
+2015-03-09 16:00:07 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink
+ This gives more accurate values than asking the payloader. There might be
+ queueing happening between the payloader and the sink.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745704
+
+2015-03-09 13:00:25 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Don't seek for PLAY if the position will not change
+ https://bugzilla.gnome.org/show_bug.cgi?id=745704
+
+2015-03-09 10:21:49 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Don't include payload type in the caps for framesize
+ When the sdp media attribute framesize are converted to caps
+ the <payload> should not be included.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
+ Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
+
+2014-02-26 22:34:06 +0100 Linus Svensson <linussn@axis.com>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ rtsp-sdp: add payload type to the sdp framesize attribute
+ The sdp framesize attribute is desribed in RFC6064. It is specified
+ for payloading of H263 and has the following form
+ a=framesize:<payload type> <width>-<height>. The <width>-<height> part
+ should be added to the caps in a payloader and the <payload type> should
+ be added by the rtsp-server.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
+
+2015-03-03 13:51:01 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * examples/test-uri.c:
+ examples: test-uri: fix tainted variable
+ Insignificant but this keeps Coverity happy.
+ CID #1268404
+
+2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
+
+ * examples/.gitignore:
+ * examples/Makefile.am:
+ * examples/test-netclock-client.c:
+ * examples/test-netclock.c:
+ examples: Add a simple example of network synch for live streams.
+ An example server and client that works for synchronising live streams
+ only - as it can't support pause/play.
+
+2015-03-03 01:49:42 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ rtsp-media-factory: Add functions to set/get the media gtype
+ Allow specifying the GType of a GstRtspMedia subclass to create
+ as a simpler way to get the factory to create a custom
+ GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
+
+2015-02-27 17:45:42 +0100 Gregor Boirie <gregor.boirie@parrot.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: fix double unlock in _get_buffer_size()
+ Fixes an abort when calling gst_rtsp_media_get_buffer_size()
+ because of double g_mutex_unlock () usage.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745434
+
+2015-02-19 10:43:16 +0200 Kent-Inge Ingesson <kenti@axis.com>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-session.h:
+ rtsp-session: Use monotonic time for RTSP session timeout
+ Changed RTSP session timeout handling to monotonic time
+ and deprecating the API for current system time.
+ This fixes timeouts when the system time changes.
+ https://bugzilla.gnome.org/show_bug.cgi?id=743346
+
+2015-02-13 12:21:16 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-client: Only error out in PLAY if seeking actually failed
+ If the media was just not seekable, we continue from whatever position we are
+ and let the client decide if that is what is wanted or not.
+ Only if the actual seek failed, we can't really recover and should error out.
+
+2015-02-12 10:46:28 +0100 Andreas Frisch <fraxinas@opendreambox.org>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Add necessary queues between tee and multiudpsink
+ https://bugzilla.gnome.org/show_bug.cgi?id=744379
+
+2015-02-12 16:48:46 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: If seeking fails, don't wait forever for the media to preroll again
+ Instead error out properly the same way as if the SEEKING query already
+ failed.
+
+2015-02-11 17:24:38 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-stream: minor code formatting fix
+
+2015-02-10 16:39:58 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: fix logic for collect_streams
+ Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
+ all streams it knows if it got any, and can check if the transport mode is OK.
+ CID #1268400
+
+2015-02-09 10:21:50 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Don't set the transport mode based on what elements we find
+ Just print a warning if the one that was set before disagrees with what
+ elements we found. It must already be set to something before as this
+ function is called after we received the SDP from ANNOUNCE in RECORD mode,
+ and we would reject ANNOUNCE if the RECORD flag was not set.
+
+2015-02-08 18:05:50 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: rtspserver: rename shadowed variable
+ We have two different 'sink' variables here,
+ rename one of them for clarity.
+
+2015-02-08 12:08:36 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: fix awkward if clause
+
+2015-02-06 19:34:17 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/test-uri.c:
+ examples: test-uri: improve uri argument handling and accept file names
+ Print an error if the argument passed is not a URI and can't
+ be converted into one, or no arguments have been provided.
+
+2015-02-06 19:15:40 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/test-uri.c:
+ examples: test-uri: don't remove mount point after 10 seconds
+ It's very irritating when trying to test stuff repeatedly
+ and serves no real purpose other than showing that it can
+ be done.
+
+2015-01-21 17:32:21 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/.gitignore:
+ examples: add new test-record to .gitignore
+
+2015-01-28 18:54:01 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-record.c:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * tests/check/gst/rtspserver.c:
+ rtsp-media: Use flags to distinguish between PLAY and RECORD media
+
+2015-01-28 17:49:16 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-record.c:
+ test-record: Set latency for playback-style example to 2s instead of 200ms
+
+2015-01-21 17:27:56 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: add some unit tests for ANNOUNCE and RECORD
+ https://bugzilla.gnome.org/show_bug.cgi?id=743175
+
+2015-01-21 16:32:44 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: fix a couple of leaks in handle_announce
+
+2015-01-19 13:20:39 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ rtsp-media: Expose latency setting for setting the rtpbin latency
+
+2015-01-17 10:28:13 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-record.c:
+ test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
+
+2015-01-16 20:48:42 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
+
+2015-01-09 12:40:47 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/Makefile.am:
+ * examples/test-record.c:
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ Add initial support for RECORD
+ We currently only support media that is RECORD or PLAY only, not both at once.
+ https://bugzilla.gnome.org/show_bug.cgi?id=743175
+
+2015-01-30 12:50:20 +0100 Anila Balavan <anilabn@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: RTCP and RTP transport cache cookies seperated
+ RTCP packets were not sent because the same tr_cache_cookie was used for
+ both RTP and RTCP. So only one of the tr_cache lists were populated
+ depending on which one was sent first. If the tr_cache list is not
+ populated then no packets can be sent. Most often this happened to be
+ RTCP. Now seperate RTCP and RTP transport cache cookies are added which
+ resulted in both the tr_cache_lists to be populated regardless of which
+ one was sent first.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
+
+2015-01-21 14:57:03 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: fix false compiler warning
+ rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
+
+2015-01-19 20:35:15 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: log interleaved data received
+
+2015-01-19 20:18:20 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: fix unintentional fallthrough to debug warning when receiving interleaved data
+
+2015-01-19 13:09:20 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: If we have a single-stream media and SETUP contains no control, use the one and only stream
+
+2015-01-18 19:08:36 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Use a random session ID in the SDP
+ RFC4566 Section 5.2 says that it should make the username, session id,
+ nettype, addrtype and unicast address tuple globally unique. Always using
+ 1188340656180883 is not going to guarantee that: https://xkcd.com/221/
+ Instead let's create a 64 bit random number, which at least brings us
+ closer to the goal of global uniqueness.
+ https://tools.ietf.org/html/rfc4566#section-5.2
+
+2015-01-17 10:29:36 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * examples/test-launch.c:
+ * examples/test-mp4.c:
+ * examples/test-ogg.c:
+ * examples/test-uri.c:
+ examples: Don't call gst_init() and gst_get_option_group()
+ The latter calls the former at the appropriate time.
+
+2015-01-16 20:04:01 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Drop trailing \0 of RTSP DATA messages
+ We add a trailing \0 in GstRTSPConnection to make parsing of
+ string message bodies easier (e.g. the SDP from DESCRIBE) but
+ for actual data this means we have to drop it or otherwise
+ create invalid data.
+
+2015-01-16 11:10:20 +0100 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Have one copy of the transports cache for RTP and RTCP each
+ Fixes crash when two threads access handle_new_sample() at the same
+ time, one for RTP, one for RTCP.
+ Otherwise, when iterating over the transports cache, it might be modified by
+ another thread at the same time if the transports cookie has changed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=742954
+
+2015-01-15 19:34:20 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Set format=TIME on our app sources for TCP
+
+2015-01-13 15:29:29 +0100 Sebastian Rasmussen <sebrn@axis.com>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ Revert "rtsp-session-pool: Make sure session IDs are properly URI-escaped"
+ This reverts commit 935e8f852d050b4939f1d0f44b38e9b55a2fbe36.
+ RFC 2326 states that session IDs may consist of alphanumeric as well as
+ the safe characters $-_.+ -- N.B. the percent character is not allowed.
+ Previously the session ID was URI-escaped, this meant that any character
+ which was not alphanumeric or any of the characters +-._~ would be
+ percent encoded. While the RFC (surprisingly) mentions that linear white
+ space in session IDs should be URI-escaped, it does not say anything
+ about other characters. Moreover no white space is allowed in the
+ session ID. Finally the percent character which is the result of
+ URI-escaping is not allowed in a session ID.
+ So there is no reason to do any URI-escaping, and now it is removed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=742869
+
+2015-01-12 16:14:12 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From f2c6b95 to bc76a8b
+
+2014-12-31 13:04:57 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * Makefile.am:
+ Fix 'make check' from top-level directory
+
+2014-12-30 18:13:49 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * examples/test-launch.c:
+ * examples/test-mp4.c:
+ * examples/test-ogg.c:
+ * examples/test-uri.c:
+ examples: Add command-line parsing and take a 'port' argument
+ This allows users to run multiple servers on different ports for testing.
+ Only done for examples that actually take arguments and hence are capable of
+ outputting different streams for each instance on each port.
+ https://bugzilla.gnome.org/show_bug.cgi?id=742115
+
+2014-12-29 12:06:50 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ rtsp-client: Add a send_message default signal handler
+ This allows subclasses to easily hook into the response sending
+ mechanism without doing everything from a signal, which seems
+ awkward from subclasses.
+
+2014-12-18 10:56:44 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From ef1ffdc to f2c6b95
+
+2014-12-17 20:02:05 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * Makefile.am:
+ * configure.ac:
+ configure: add --disable-examples switch
+ https://bugzilla.gnome.org/show_bug.cgi?id=741678
+
+2014-12-01 23:42:34 +1100 Matthew Waters <matthew@centricular.com>
+
+ * examples/.gitignore:
+ * examples/Makefile.am:
+ * examples/test-video-rtx.c:
+ examples: add a retransmisison example implementing RFC4588
+ Currently only SSRC-multiplexed rtx streams are supported
+
+2014-12-16 16:46:15 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Fix some minor memory leaks
+
+2014-12-16 16:46:06 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Some minor cleanup
+
+2014-12-16 16:42:13 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Fix compiler warnings
+ rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+ ^
+ rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
+ g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+ ^
+
+2014-11-27 01:12:36 +1100 Matthew Waters <matthew@centricular.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media-factory.h:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-media.h:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ media: implement ssrc-multiplexed retransmission support
+ based off RFC 4588 and the server-rtpaux example in -good
+
+2014-11-28 12:45:14 +0100 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp: Ref transports in hash table.
+ Also ref streams for transports.
+ This solves a crash when reciving a rtcp after teardown but before
+ client finalize.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
+
+2014-11-27 17:13:05 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * common:
+ Automatic update of common submodule
+ From 7bb2bce to ef1ffdc
+
+2014-11-07 12:48:53 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: refactor cleanup of cached media
+
+2014-10-23 13:39:10 +0200 Linus Svensson <linussn@axis.com>
+
+ * tests/check/gst/client.c:
+ tests: Remove FIXME
+ The session leak is now fixed, lets remove those FIXME comments.
+
+2014-10-23 17:54:37 +0200 Linus Svensson <linussn@axis.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: Test to setup two sessions on one connection
+ https://bugzilla.gnome.org/show_bug.cgi?id=739112
+
+2014-10-24 12:05:27 +0200 Linus Svensson <linussn@axis.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: Test setup with tcp transport
+ https://bugzilla.gnome.org/show_bug.cgi?id=739112
+
+2014-10-24 12:04:54 +0200 Linus Svensson <linussn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: Configure transport after creating session media
+ The default implementation of configure_client_transport() in
+ rtsp-client uses the session media when it chooses channels for
+ interleaved traffic.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739112
+
+2014-10-23 12:54:03 +0200 Linus Svensson <linussn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ client: Stop caching media in client when doing setup
+ If the media has been managed by a session media, it should not be
+ cached in the client any longer. The GstRTSPSessionMedia object is now
+ responsible for unpreparing the GstRTSPMedia object using
+ gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
+ session media.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739112
+
+2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: unref srtp decoder when leaving bin
+ https://bugzilla.gnome.org/show_bug.cgi?id=739481
+
+2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: mikey memory leaks
+ https://bugzilla.gnome.org/show_bug.cgi?id=739383
+
+2014-10-27 18:01:35 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From 84d06cd to 7bb2bce
+
+2014-10-24 17:48:04 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * Makefile.am:
+ Parallelise 'make check-valgrind'
+
+2014-10-21 13:04:14 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From a8c8939 to 84d06cd
+
+2014-10-21 13:00:49 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From 36388a1 to a8c8939
+
+2014-10-01 07:12:30 -0400 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: deactivate media when shutting down from paused
+ This was only done when going directly from playing.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737829
+
+2014-10-20 15:40:59 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-context.h:
+ rtsp-client: add stream transport to context
+ We add the stream transport to the context so we can get the configured
+ client stream transport in the setup request signal.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738905
+
+2014-10-02 12:02:48 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: release lock even not all transports have been removed
+ We don't want to keep the lock even we return FALSE because not all the
+ transports have been removed. This could lead into a deadlock.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737797
+
+2014-10-10 18:43:00 -0400 Olivier Crête <olivier.crete@ocrete.ca>
+
+ * gst/rtsp-server/rtsp-sdp.c:
+ rtsp-sdp: Rename clock-base and seqnum-base to timestamp-offset and seqnum-offset
+ These were renamed in GstRTPBasePayload in 1.0
+
+2014-09-30 16:36:51 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: set session media to NULL without the lock
+ We need to set session medias to NULL without the client lock otherwise
+ we can end up in a deadlock if another thread is waiting for the lock
+ and media unprepare is also waiting for that thread to end.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737690
+
+2014-09-30 23:22:45 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ rtsp-media: Set state to UNPREPARING in all cases
+
+2014-09-30 19:17:04 +0200 Ognyan Tonchev <otonchev@gmail.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ media: set state to unpreparing when unprepare is initiated
+ https://bugzilla.gnome.org/show_bug.cgi?id=737675
+
+2014-09-30 01:35:02 +0200 Sebastian Rasmussen <sebrn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Remove backlog limit while processings requests
+ If the backlog limit is kept two cases of deadlocks may be
+ encountered when streaming over TCP. Without the backlog
+ limit this deadlocks can not happen, at the expence of
+ memory usage.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737631
+
+2014-09-22 13:32:06 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: do not free main context before rtsp watch
+ https://bugzilla.gnome.org/show_bug.cgi?id=737110
+
+2014-09-19 18:29:00 +0200 Branko Subasic <branko@axis.com>
+
+ * tests/check/gst/rtspserver.c:
+ tests: Extend unit test timeout to accomodate for valgrind
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
+
+2014-09-19 18:28:50 +0200 Branko Subasic <branko@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ rtsp-*: Treat sending packets to clients as keepalive
+ As long as gst-rtsp-server can successfully send RTP/RTCP data to
+ clients then the client must be reading. This change makes the server
+ timeout the connection if the client stops reading.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
+
+2014-09-19 18:28:30 +0200 Branko Subasic <branko@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Allow backlog to grow while expiring session
+ Allow the send backlog in the RTSP watch to grow to unlimited size while
+ attempting to bring the media pipeline to NULL due to a session
+ expiring. Without this change the appsink element cannot change state
+ because it is blocked while rendering data in the new_sample callback.
+ This callback will block until it has successfully put the data into the
+ send backlog. There is a chance that the send backlog is full at this
+ point which means that the callback may block for a long time, possibly
+ forever. Therefore the media pipeline may also be prevented from
+ changing state for a long time.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
+
+2014-09-22 09:30:39 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Make old compilers happy
+ rtsp-client.c:2553:50: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
+ Just in case that guint8 doesn't fit in a pointer. Just in case ...
+
+2014-09-16 11:41:52 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: raise the backlog limits before pausing
+ We need to raise the backlog limits before pausing the pipeline or else
+ the appsink might be blocking in the render method in wait_backlog() and
+ we would deadlock waiting for paused.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736322
+
+2014-09-16 11:29:38 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: make define for the WATCH_BACKLOG
+ See https://bugzilla.gnome.org/show_bug.cgi?id=736322
+
+2014-09-09 18:11:39 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: simplify session transport handling
+ link/unlink of the transport in a session was done to keep track of all
+ TCP transports and to send RTP/RTCP data to the streams. We can simplify
+ that by putting all the TCP transports in a hashtable indexed with the
+ channel number.
+ We also don't need to link/unlink the transports when we pause/resume
+ the streams. The same effect is already achieved when we pause/play the
+ media. Indeed, when we pause the media, the transport is removed from
+ the media and the callbacks will not be called anymore.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=736041
+
+2014-09-09 18:10:12 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream-transport.h:
+ stream-transport: make method to handle received data
+ Make a method to handle the data received on a channel. It sends the
+ data to the stream of the transport on the RTP or RTCP pads based on
+ the channel number.
+
+2014-09-15 16:54:05 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * examples/test-mp4.c:
+ test: add example of dumping RTCP reports
+
+2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-media: Make sure that sequence numbers are monotonic after pause
+ The sequence number is not monotonic for RTP packets after pause. The
+ reason is basepayloader generates a randon sequence number when the
+ pipeline goes from ready to pause. With this fix generation of sequence
+ number will be monotonic when going from pause to play request.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736017
+
+2014-08-28 13:35:15 +0200 Göran Jönsson <goranjn@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ rtsp-client: Protect saved clients watch with a mutex
+ Fixes a crash when close() is called while merging clients
+ in handle_tunnel(). In that case close() would destroy the
+ watch while it is still being used in handle_tunnel().
+ https://bugzilla.gnome.org/show_bug.cgi?id=735570
+
+2014-08-13 17:22:16 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Remove the multicast group udp sources when removing from the bin
+
+2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-stream.h:
+ rtsp-media: Query position and stop time only on the RTP parts of the pipeline
+ The RTCP parts, in specific the RTCP udpsinks, are not flushed when
+ seeking and will always continue counting the time. This leads to
+ the NPT after a backwards seek to be something completely different
+ to the actual seek position.
+ https://bugzilla.gnome.org/show_bug.cgi?id=732644
+
+2014-08-09 14:41:35 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/test-appsrc.c:
+ examples: fix another reference leak
+ gst_rtsp_media_get_element() returns a new ref.
+
+2014-07-17 01:34:17 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * examples/test-appsrc.c:
+ examples: unref element after usage
+ gst_bin_get_by_name_recurse_up() returns an element
+ reference that must be unreffed after usage.
+ https://bugzilla.gnome.org/show_bug.cgi?id=734546
+
+2014-07-02 22:45:07 +0530 Arun Raghavan <arun@accosted.net>
+
+ * gst/rtsp-server/rtsp-media.c:
+ signals: Fix copy-pasto in target-state signal offset
+
+2014-08-01 10:46:44 +0200 Edward Hervey <edward@collabora.com>
+
+ * Makefile.am:
+ * common:
+ Makefile: Add usage of build-checks step
+ Allows building checks without running them
+
+2014-06-25 18:23:10 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ rtsp-stream: Listen on the multicast group for RTP/RTCP packets
+ When a UDP multicast transport is used it is expected that the server listens
+ for RTP and RTCP packets on the multicast group with the corresponding port.
+ Without this we will never get RTCP packets from clients in multicast mode.
+ https://bugzilla.gnome.org/show_bug.cgi?id=732238
+
+2014-07-19 18:04:52 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
+=== release 1.4.0 ===
+
+2014-07-19 17:56:31 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.4.0
+
+2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
+
+ * gst/rtsp-server/rtsp-media.h:
+ media: correct misspelled words in description
+ https://bugzilla.gnome.org/show_bug.cgi?id=733244
+
+=== release 1.3.91 ===
+
+2014-07-11 12:19:08 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.3.91
+
+2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * docs/libs/gst-rtsp-server-sections.txt:
+ docs: update docs
+
+2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-server.c:
+ server: implement client REMOVE filter
+
+2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-client.h:
+ client: expose _close() method
+ Expose a previously internal close method to close the client
+ connection.
+
+2014-07-10 12:20:15 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ session-pool: signal session-removed outside of the lock
+ Release the lock before emiting the session-removed signal.
+
+2014-07-10 11:32:20 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-server.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ filter: Release lock in filter functions
+ Release the object lock before calling the filter functions. We need to
+ keep a cookie to detect when the list changed during the filter
+ callback. We also keep a hashtable to make sure we only call the filter
+ function once for each object in case of concurrent modification.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
+
+2014-07-09 15:16:08 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: check if watch is set in handle_teardown()
+ The unit tests run without a watch
+
+2014-07-09 14:19:10 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * tests/check/gst/client.c:
+ client tests: send teardown to cleanup session
+
+2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * tests/check/gst/rtspserver.c:
+ server tests: send teardown to cleanup session
+
+2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: keep ref to client for the session removed handler
+ This extra ref will be dropped when all client sessions have been
+ removed. A session is removed when a client sends teardown, closes its
+ endpoint of the TCP connection or the sessions expires.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
+
+2014-07-08 12:36:12 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-session.c:
+ * tests/check/gst/client.c:
+ client: manage media in session as a last step
+ Once we manage a media in a session, we can't unmanage it anymore
+ without destroying it. Therefore, first check everything before we
+ manage the media, otherwise if something is wrong we have no way to
+ unmanage the media.
+ If we created a new session and something went wrong, remove the session
+ again. Fixes a leak in the unit test.
+
+2014-07-03 19:52:42 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * examples/test-mp4.c:
+ * examples/test-ogg.c:
+ examples: print 'stream ready at url' for mp4 and ogg example
+
+2014-07-02 16:04:53 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ rtsp: fix for MIKEY api change
+
+2014-07-01 16:12:13 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: free watch context only once
+ The watch context is freed when the source is destroyed. Avoids
+ a CRITICAL when we try to unref the context twice.
+
+2014-07-01 15:02:15 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: fix build
+
+2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ client: protect sessions with lock
+ Protect the list of sessions with the lock.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226
+
+2014-07-01 12:13:47 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ Client: keep a ref to the session
+ Don't just keep a weak ref to the session objects but use a hard ref. We
+ will be notified when a session is removed from the pool (expired) with
+ the new session-removed signal.
+ Don't automatically close the RTSP connection when all the sessions of
+ a client are removed, a client can continue to operate and it can create
+ a new session if it wants. If you want to remove the client from the
+ server, you have to use gst_rtsp_server_client_filter() now.
+ Based on patch from Ognyan Tonchev <ognyan.tonchev at axis.com>
+ See https://bugzilla.gnome.org/show_bug.cgi?id=732226
+
+2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-session-pool.h:
+ session-pool: add session-removed signal
+ Add a signal to be notified when a session is removed from the pool.
+
+2014-06-30 00:37:59 -0700 Evan Nemerson <evan@nemerson.com>
+
+ * gst/rtsp-server/Makefile.am:
+ * gst/rtsp-server/rtsp-server.h:
+ Make rtsp-server.h a single-include header, use it for G-I
+ https://bugzilla.gnome.org/show_bug.cgi?id=732411
+
+=== release 1.3.90 ===
+
+2014-06-28 11:48:29 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * configure.ac:
+ * gst-rtsp-server.doap:
+ Release 1.3.90
+
+2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-stream.c:
+ stream: crypto can be NULL
+
+2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-mount-points.c:
+ introspection: add missing allow-none annotations
+ https://bugzilla.gnome.org/show_bug.cgi?id=730952
+
+2014-06-11 16:38:36 -0700 Evan Nemerson <evan@nemerson.com>
+
+ * gst/rtsp-server/rtsp-address-pool.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-session-media.c:
+ * gst/rtsp-server/rtsp-session-pool.c:
+ * gst/rtsp-server/rtsp-stream-transport.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ * gst/rtsp-server/rtsp-token.c:
+ introspection: add (nullable) annotations to return values
+ https://bugzilla.gnome.org/show_bug.cgi?id=730952
+
+2014-06-24 09:48:45 +0200 Evan Nemerson <evan@nemerson.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-stream.c:
+ gi: improve annotations
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
+
+2014-06-24 09:43:44 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-media-factory.c:
+ * gst/rtsp-server/rtsp-media.c:
+ * gst/rtsp-server/rtsp-server.c:
+ signals: use generic marshal function
+ Use the generic C marshal function.
+ Use more explicit type instead of G_TYPE_POINTER
+
+2014-06-24 09:42:47 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-context.h:
+ context: add type macro
+
+2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp-server/rtsp-client.c:
+ * gst/rtsp-server/rtsp-sdp.c:
+ * gst/rtsp-server/rtsp-sdp.h:
+ sdp: hide key length defines
+ They don't have a namespace.
+
+2014-06-22 19:37:31 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
=== release 1.3.3 ===
-2014-06-22 Sebastian Dröge <slomo@coaxion.net>
+2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
* configure.ac:
- releasing 1.3.3
+ * gst-rtsp-server.doap:
+ Release 1.3.3
2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
projects / platform / upstream / gstreamer.git / blobdiff