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Author SHA1 Message Date
Sebastian Dröge
5c5850b6b1 test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization 2015-06-10 17:14:44 +02:00
Xavier Claessens
6ec8fe44b2 GstRTSPAuth: Add client certificate authentication support 
https://bugzilla.gnome.org/show_bug.cgi?id=750471
 2015-06-09 19:51:46 -04:00
Sebastian Dröge
af2cb6445a test-netclock-client: Use new GstClock API to wait for clock synchronization 2015-06-09 13:53:47 +02:00
Sebastian Dröge
6219766555 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 13:52:05 +02:00
Edward Hervey
2c935a7884 Automatic update of common submodule 
From d9a3353 to 6015d26
 2015-06-09 11:30:54 +02:00
Stefan Sauer
7436fee689 Automatic update of common submodule 
From d37af32 to d9a3353
 2015-06-08 23:08:34 +02:00
Stefan Sauer
1541d3dd8a Automatic update of common submodule 
From 21ba2e5 to d37af32
 2015-06-07 23:07:31 +02:00
Stefan Sauer
68dbad0967 Automatic update of common submodule 
From c408583 to 21ba2e5
 2015-06-07 17:32:29 +02:00
Stefan Sauer
260e577b9c docs: remove variables that we define in the snippet from common 
This is syncing our Makefile.am with upstream gtkdoc.
 2015-06-07 17:19:10 +02:00
Stefan Sauer
4aaab390c4 Automatic update of common submodule 
From 44a3517 to c408583
 2015-06-07 17:16:47 +02:00
Sebastian Dröge
9c75932b16 Back to development 2015-06-07 16:44:55 +02:00
Sebastian Dröge
e86bbbb66c Release 1.5.1 2015-06-07 11:20:01 +02:00
Göran Jönsson
08e0c79cee 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
 2015-06-03 15:09:10 +02:00
Tim-Philipp Müller
550348738c tests: Use AM_TESTS_ENVIRONMENT 
Needed by the new automake test runner and the
current version of the common submodule.
 2015-05-27 17:04:41 +01:00
Sebastian Dröge
8700468499 rtsp-server: Use single-include rtsp header to make sure we get all definitions 2015-05-20 17:05:47 +03:00
Sebastian Dröge
1c30c60e64 rtsp-media: Mark some more functions static 2015-05-05 16:46:57 +02:00
Sebastian Dröge
bbdf0a47d1 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.
 2015-05-05 16:46:19 +02:00
Sebastian Dröge
e4264381a5 examples: Use AVPF profile for the RTX example 2015-05-04 16:33:08 +02:00
Sebastian Dröge
ec2c500a9d rtsp-sdp: Only add RTX to the SDP when using a feedback profile 2015-05-04 16:31:20 +02:00
Hyunjun Ko
4ff22ef6d2 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
 2015-04-27 12:41:59 +02:00
Tim-Philipp Müller
f777de7d7f 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 15:07:52 +01:00
Tim-Philipp Müller
226fbbc8f8 Revert "configure.ac: uncomment gettext version setup" 
This reverts commit 1545d8fef7.

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
 2015-04-26 14:59:09 +01:00
Alistair Buxton
eb7705a48d Fix timeout function signatures across tests and examples 2015-04-23 20:12:18 +02:00
Tim-Philipp Müller
753f8a8ac9 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 17:27:40 +01:00
Tim-Philipp Müller
bdbc6f24ce 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
 2015-04-23 17:22:59 +01:00
Vincent Penquerc'h
1545d8fef7 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
 2015-04-20 08:49:57 +01:00
Hyunjun Ko
fabde79bc3 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 15:14:15 +02:00
Hyunjun Ko
de590b4b2a 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
 2015-04-15 15:14:04 +02:00
Tim-Philipp Müller
bff66c0004 Update autogen.sh to latest version from common 
Fixes build after aclocal_check etc. helpers have been removed.
 2015-04-06 10:32:52 +01:00
Tim-Philipp Müller
a54d8733b2 Automatic update of common submodule 
From bc76a8b to c8fb372
 2015-04-03 18:58:26 +01:00
Sebastian Dröge
ef3bfd757b 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 21:04:43 +01:00
Sebastian Dröge
357af7aea6 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.
 2015-03-23 20:59:52 +01:00
Nicolas Dufresne
dfb053add3 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
 2015-03-21 11:04:05 -04:00
Nicolas Dufresne
01562286ba rtsp-stream: Don't leave buffer mapped 
If the seq is NULL, the RTP buffer was left mapped. We should always
unmap the buffer.
 2015-03-18 16:44:19 -04:00
Sebastian Dröge
01ae7c01f3 Fix typo in README 2015-03-15 12:27:39 +00:00
Tim-Philipp Müller
896767b041 Fix double semicolons 2015-03-10 09:39:22 +00:00
Sebastian Dröge
852cc09f54 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 16:00:38 +01:00
Sebastian Dröge
b58af93d83 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 13:00:25 +01:00
Sebastian Dröge
93bdbb6acd 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>
 2015-03-09 10:21:49 +01:00
Linus Svensson
9dadaed2fd 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
 2015-03-09 09:26:38 +01:00
Luis de Bethencourt
d92ff17026 examples: test-uri: fix tainted variable 
Insignificant but this keeps Coverity happy.

CID #1268404
 2015-03-03 13:53:11 +00:00
Jan Schmidt
b04856f0cf 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 11:53:16 +11:00
Jan Schmidt
db42945c2c 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.
 2015-03-03 11:53:16 +11:00
Gregor Boirie
bc7765eee7 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
 2015-03-02 10:50:57 +00:00
Kent-Inge Ingesson
d2f1997c4b 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
 2015-02-19 10:43:30 +02:00
Sebastian Dröge
51ed357597 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.
 2015-02-13 12:21:16 +02:00
Andreas Frisch
bac59c52f1 rtsp-stream: Add necessary queues between tee and multiudpsink 
https://bugzilla.gnome.org/show_bug.cgi?id=744379
 2015-02-13 11:28:43 +02:00
Sebastian Dröge
98b162f54b 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.
 2015-02-12 16:53:27 +02:00
Tim-Philipp Müller
dc43f427a9 rtsp-stream: minor code formatting fix 2015-02-11 17:25:35 +00:00
Luis de Bethencourt
ec7bf5379e 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
 2015-02-10 16:45:23 +00:00