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Author SHA1 Message Date
Sebastian Rasmussen
d6a4dee036 rtsp-sdp: Parse framerate caps field and set SDP attribute 
The SDP attribute and its format is described in RFC4566.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
 2013-05-23 21:02:58 +02:00
Sebastian Rasmussen
5fd034ff1a rtsp-sdp: Parse width/height from caps and set SDP attribute 
The SDP attribute and its format is described in RFC6064.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
 2013-05-23 21:02:50 +02:00
Patricia Muscalu
0951aa37e1 rtsp-sdp: add bandwidth line 
https://bugzilla.gnome.org/show_bug.cgi?id=699220
 2013-05-15 12:36:32 +02:00
Sebastian Dröge
be193ceb86 Automatic update of common submodule 
From 5edcd85 to 098c0d7
 2013-05-15 10:55:09 +02:00
Ognyan Tonchev
6065400a62 tests: add dynamic payloader prepare/unprepare check 2013-04-23 11:28:39 +02:00
Wim Taymans
573b10bc83 media: release lock when removing fakesink 2013-04-23 10:28:35 +02:00
Wim Taymans
0ddd98bfa6 stream: set elements to NULL before removing 
When removing a stream, set the elements to NULL first. This avoids
element-is-not-in-NULL-state errors when we dispose the elements.
 2013-04-23 10:28:34 +02:00
Tim-Philipp Müller
a09210b648 Automatic update of common submodule 
From 3cb3d3c to 5edcd85
 2013-04-22 23:55:48 +01:00
Wim Taymans
b80b8824be media: listen to pad-removed signals 
Listen to the pad-removed signal and remove the stream associated with the
removed pad.
Add signal to be notified of the removed pad.
Remove the fakesink in unprepare()
Fix signatures of the signal methods
 2013-04-22 17:34:37 +02:00
Wim Taymans
b614319622 tests: add example of reusable pipelines 2013-04-22 17:33:30 +02:00
Ognyan Tonchev
00291e5285 stream: add method to get the srcpad 2013-04-22 17:32:31 +02:00
Ognyan Tonchev
f15288259e check: add media prepare/unprepare test 
See https://bugzilla.gnome.org/show_bug.cgi?id=698376
 2013-04-22 16:49:39 +02:00
Ognyan Tonchev
a26b06cc69 media: disconnect from signal handlers in unprepare() 
We connected to the pad-added and no-more-pads signals in prepare() so
we need to disconnect from them in unprepare().

See https://bugzilla.gnome.org/show_bug.cgi?id=698376
 2013-04-22 16:40:48 +02:00
Ognyan Tonchev
9b31fcc7f8 media: don't free streams array 
Don't free the streams array in the unprepare() method, they were not
added in prepare().

See https://bugzilla.gnome.org/show_bug.cgi?id=698376
 2013-04-22 16:25:17 +02:00
Ognyan Tonchev
0bdff0161c media: don't unref the pipeline in unprepare 
Unprepare() should undo what prepare() does. Because the pipeline is
not created in prepare(), we should not unref it in unprepare()
 2013-04-22 16:19:35 +02:00
Ognyan Tonchev
6081f91351 stream: clear session and caps for reuse 
Set the session and caps to NULL after unref otherwise we might unref
them again later.

See https://bugzilla.gnome.org/show_bug.cgi?id=698376
 2013-04-22 16:09:22 +02:00
David Svensson Fors
bba7c4042d client: send out teardown signal before tearing down 
The advantage is that in the signal handler you get direct access to
information about what streams are about to get torn down (in the
GstRTSPClientState).

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
 2013-04-15 12:21:54 +02:00
David Svensson Fors
825d6f0b51 client: expose connection 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
 2013-04-15 12:17:34 +02:00
Tim-Philipp Müller
3ba1342906 Automatic update of common submodule 
From aed87ae to 3cb3d3c
 2013-04-14 17:58:22 +01:00
Wim Taymans
a64cb68164 media: add method to get the base_time of the pipeline 
Together with a shared clock, this base-time could eventually be sent to
the client so that it can reconstruct the exact running-time of the clock
on the server.
 2013-04-12 11:34:38 +01:00
Wim Taymans
36ff679558 media: add GstNetTimeProvider support 
Add a property to let the media provide a GstNetTimeProvider for its clock.
Make methods to get the clock and nettimeprovider
Add a x-gst-clock property to the SDP with the IP and port number of the nettime
provider and also the current time of the clock. This should make it possible
for (GStreamer) clients to slave their clock to the server clock.
 2013-04-09 22:38:44 +02:00
Stefan Sauer
1704018d5d Automatic update of common submodule 
From 04c7a1e to aed87ae
 2013-04-09 21:02:47 +02:00
Wim Taymans
95bf53513f media: wait for buffering to complete 
Wait for buffering to complete before changing the state to the target state.
 2013-04-09 20:39:58 +02:00
Wim Taymans
ec0718d7c9 media: small cleanup 2013-04-09 20:11:35 +02:00
David Svensson Fors
d728d59a00 tests: remove extra unref in test_setup_non_existing_stream 
The unref is not needed anymore, teardown runs without it.

https://bugzilla.gnome.org/show_bug.cgi?id=696542
 2013-03-28 12:54:10 +00:00
David Svensson Fors
75221ac8e3 tests: GSocketService cleanup in test_bind_already_in_use 
Use g_socket_service_stop so the rtspserver test stops listening for
incoming connections in test_bind_already_in_use.

https://bugzilla.gnome.org/show_bug.cgi?id=696541
 2013-03-28 12:48:46 +00:00
Olivier Crête
91210f40f2 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here 
Instead use a GWeakRef which is safe to use

This is a known GLib bug, see:
https://bugzilla.gnome.org/show_bug.cgi?id=667145
 2013-03-22 18:59:50 -04:00
Olivier Crête
c18eafbb24 rtsp-media/client: Reply to PLAY request with same type of Range 
Remember the type of Range from the PLAY request and use the same type for
the reply.
 2013-03-22 15:53:06 +01:00
Patricia Muscalu
8a08fddb41 rtsp-client: expose uri 2013-03-18 23:44:38 +00:00
Olivier Crête
4a99e1cf56 tests: Hold ref while creating second media 
To test if the media aren't shared, make sure we keep the first one while creating a second
otherwise the same memory address may be reused.
 2013-03-13 17:47:44 -04:00
Tim-Philipp Müller
025ac34580 configure: remove out-of-date comment 2013-03-12 00:10:18 +00:00
Tim-Philipp Müller
9da40095c3 .gitignore: ignore more build files 2013-03-12 00:05:49 +00:00
Tim-Philipp Müller
fba09126a8 tests: use right _LIBS variable for gst-plugins-base libs 2013-03-12 00:03:36 +00:00
Wim Taymans
4a2276c0e6 check: add librtp to libs 2013-03-11 11:35:14 +01:00
Olivier Crête
6a2238b2fb tests: Add test to check selecting a port the server will send from 2013-03-11 11:07:20 +01:00
Olivier Crête
d3c70d4d51 tests: Make sure packets are actually received 2013-03-11 11:07:20 +01:00
Olivier Crête
5a39e25949 stream: Select unicast address from pool if appropriate 2013-03-11 11:07:20 +01:00
Olivier Crête
a797cbde06 stream: Properties are always there in Gst 1.0 2013-03-11 11:07:20 +01:00
Olivier Crête
444c5892f7 tests: Add tests for unicast addresses in pool 2013-03-11 11:07:20 +01:00
Olivier Crête
27a057962c address-pool: Verify that multicast addresses are used for multicast and vice-versa 2013-03-11 11:07:20 +01:00
Olivier Crête
d06e68abd1 address-pool: Add unicast addresses 2013-03-11 11:07:20 +01:00
Olivier Crête
4c61c6d308 rtsp-server: Limit the number of threads per server instance 
If we exceed the maximum, just round robin the clients over the existing
threads.
 2013-03-11 11:07:20 +01:00
Olivier Crête
4071e1b999 rtsp-server: No need to store the GMainContext in the client context 2013-03-11 11:07:20 +01:00
Olivier Crête
dcc92cbde1 tests: Add test for client disconnection 2013-03-11 11:07:20 +01:00
Olivier Crête
2e11184171 tests: Test client and session timeouts with multiple threads 2013-03-11 11:07:19 +01:00
Olivier Crête
b9d111372e Document locking and its order 2013-03-11 11:07:19 +01:00
Olivier Crête
176f5dd0be tests: Test that slow DESCRIBE don't block other clients 2013-03-11 11:07:19 +01:00
Olivier Crête
29d9878536 tests: Add tests for client-requested multicast address 2013-03-11 11:07:19 +01:00
Olivier Crête
41951c4afd docs: Put the various functions in the right sections 2013-03-11 11:07:19 +01:00
Olivier Crête
f0ab7ce1bf docs: Generate docs for GstRTSPAddressPool 2013-03-11 11:07:19 +01:00