Wim Taymans
2a0aaa1019
Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
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This reverts commit d6a4dee036
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We already have a way to place extra attributes in the SDP, just make a string
property in the payloader with a- or x- prefix.
2013-05-31 15:43:11 +02:00
Wim Taymans
cfdf2e6db5
rtsp: place a- and x- properties as attributes
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application/x-rtp has properties with a- and x- prefixes that should be
placed as attributes in the SDP for the media instead of being added to the
fmtp.
2013-05-31 15:43:10 +02:00
Wim Taymans
d1339f60f4
example: add TLS example
2013-05-31 12:11:12 +02:00
Wim Taymans
0a285290cb
server: add support for TLS
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Add methods to set and get a TLS certificate.
Add vmethod to configure a new connection. By default, configure the TLS
certificate in a new connection if needed.
2013-05-31 11:42:36 +02:00
Wim Taymans
531ffca018
server: remove accept_client vmethod
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This vmethod is not very useful so remove it.
2013-05-31 11:14:17 +02:00
Wim Taymans
0091339254
server: don't crash on NULL GError
2013-05-30 17:23:51 +02:00
Patricia Muscalu
aa50aac669
rtsp-session-pool: corrected session timeout detection
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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
2013-05-30 13:13:05 +02:00
Wim Taymans
7526178a09
client: improve debug
2013-05-30 10:52:46 +02:00
Wim Taymans
d638b03ff9
server: refactor connection setup
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Let the server accept the socket connection and construct a GstRTSPConnection
from it. Remove the code from the client and let the client only deal with
a fully configure GstRTSPConnection object.
We will need this later when the server will configure the connection for
TLS.
2013-05-30 07:18:22 +02:00
Wim Taymans
7b880231b1
stream: keep the transport object alive
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Keep the transport object alive while we have it as qdata on the
source.
2013-05-30 06:49:20 +02:00
Alexander Schrab
c75e1c6b47
rtsp-server: Do not crash on nmapping of server
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* generate error when gst_rtsp_connection_accept fails
* do not stop accepting incoming connections because
accepting a client fails
https://bugzilla.gnome.org/show_bug.cgi?id=701072
2013-05-27 13:20:36 +02:00
Alexander Schrab
e047c9fec1
rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
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https://bugzilla.gnome.org/show_bug.cgi?id=700953
2013-05-27 11:15:50 +02:00
Sebastian Rasmussen
d6a4dee036
rtsp-sdp: Parse framerate caps field and set SDP attribute
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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
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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
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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
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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
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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
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From 3cb3d3c to 5edcd85
2013-04-22 23:55:48 +01:00
Wim Taymans
b80b8824be
media: listen to pad-removed signals
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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
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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()
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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
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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
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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
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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
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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
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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
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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
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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
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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
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From 04c7a1e to aed87ae
2013-04-09 21:02:47 +02:00
Wim Taymans
95bf53513f
media: wait for buffering to complete
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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
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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
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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
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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
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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
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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