Commit graph

794 commits

Author SHA1 Message Date
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
Olivier Crête
773c48e22f client: Check client provided addresses against the address pool 2013-03-11 11:07:19 +01:00
Olivier Crête
cda75709bb address-pool: Add API to request a specific address from the pool
Also add relevant unit tests.
2013-03-11 11:07:19 +01:00
Olivier Crête
bb7a8af077 tests: Check the passing around of a RTSPAddressPool
Make sure the RTSPAddressPool is propagated from the MediaFactory all the
way down to the stream.
2013-03-11 11:07:19 +01:00
Olivier Crête
2581cc103c tests: Add more tests for the address pool 2013-03-11 11:07:19 +01:00
Olivier Crête
456f4367e3 address-pool: Fix off by one error
When splitting a port range, the port after a skip is not part of range.
2013-03-11 11:07:19 +01:00
Tim-Philipp Müller
82828852ee Automatic update of common submodule
From 2de221c to 04c7a1e
2013-03-07 00:04:19 +00:00
George McCollister
084141fb60 configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
AM_CONFIG_HEADER was removed in automake 1.13

https://bugzilla.gnome.org/show_bug.cgi?id=693368
2013-02-07 23:25:26 +00:00
Stefan Sauer
938cba2e28 Automatic update of common submodule
From a942293 to 2de221c
2013-01-28 20:45:44 +01:00
Wim Taymans
6db0dbc76c client: make sure the watch exists while sending data
Protect the send_func with a lock. This allows us to wait for sending
to complete before changing the send_func and user_data. We add an
extra ref to the watch to make sure that it remains valid during
sending.
When closing the connection, set the send_func to NULL

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
2013-01-28 11:11:46 +01:00