Sebastian Pölsterl
75598337a9
rtsp-server: added annotations to indicate type of ownership transfer of return values
...
https://bugzilla.gnome.org/show_bug.cgi?id=680777
2012-10-28 15:39:04 +00:00
Tim-Philipp Müller
9499be6232
No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
2012-10-28 15:37:51 +00:00
Tim-Philipp Müller
09718e294d
bindings: remove vala bindings
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They'll be reunited with the other GStreamer bindings
https://bugzilla.gnome.org/show_bug.cgi?id=680777
2012-10-28 15:09:04 +00:00
Wim Taymans
543aa383e7
rtsp: only create transport when needed
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Only create the StreamTransport when configured.
2012-10-28 00:23:57 +02:00
Wim Taymans
66a29c7ed9
client: small cleanup
2012-10-27 23:53:35 +02:00
Wim Taymans
fb117a4f75
rtsp: refactor configuration of transport
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Move the configuration of the transport to a place where it makes
more sense.
2012-10-27 23:49:24 +02:00
Wim Taymans
8c30d050fa
client: refactor transport parsing
2012-10-27 21:26:55 +02:00
Wim Taymans
fee8216513
client: refuse to change the MTU on shared media
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If we change the MTU of chared media, it changes for all clients.
We don't want to set the MTU to something large for clients that
stream over UDP.
2012-10-27 21:05:03 +02:00
Wim Taymans
0bb0e3733c
small fixes to docs and debug
2012-10-27 11:53:51 +02:00
Wim Taymans
6a838fd5c8
stream: transports must already have been removed
2012-10-26 17:29:30 +02:00
Wim Taymans
6f7d755894
stream: improve join and leave of the pipeline
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simplify code
Do the cleanup properly
Add some docs
2012-10-26 17:28:10 +02:00
Wim Taymans
693dd3cfc4
media: move unprepare below default implementation
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Makes it easier to find the default implementation
2012-10-26 15:23:16 +02:00
Wim Taymans
0d55e1f50c
media: signal unprepared when we actually finish
2012-10-26 15:21:50 +02:00
Wim Taymans
84b7cf1590
media: no need to unlock, unprepare does that when needed
2012-10-26 15:19:23 +02:00
Wim Taymans
348b7f9c21
docs: update docs
2012-10-26 12:35:20 +02:00
Wim Taymans
6b7ff45ca6
rtsp: fix MTU setting
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Fix setting of the MTU. There is no need for a vmethod.
2012-10-26 12:35:20 +02:00
Wim Taymans
0c8c9e3282
docs: update docs
2012-10-26 12:35:20 +02:00
Tim-Philipp Müller
a8186f90e8
configure: bump version number after refactoring
2012-10-26 11:24:55 +01:00
Wim Taymans
de7c72dec2
rtsp: massive refactoring
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Make GObjects from the remaining simple structures.
Remove GstRTSPSessionStream, it's not needed.
Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
a GstRTSPStream should be transported to a client.
Rename GstRTSPMediaFactory::get_element -> create_element because that
more accurately describes what it does.
Make nice methods instead of poking in the structures.
Move some methods inside the relevant object source code.
Use GPtrArray to store objects instead of plain arrays, it is more
natural and allows us to more easily clean up.
Move the allocation of udp ports to the Stream object. The Stream object
contains the elements needed to stream the media to a client.
Improve the prepare and unprepare methods. Unprepare should now undo
everything prepare did. Improve also async unprepare when doing EOS on
shutdown. Make sure we always unprepare correctly.
2012-10-25 21:29:58 +02:00
Sebastian Rasmussen
0de6262dc4
rtsp-client: Unref server address clients connected to
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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
2012-10-23 23:05:45 +01:00
Ognyan Tonchev
78bde6fa3e
rtsp-server: don't ref server socket if it is NULL
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Fixes test_bind_already_in_use unit test again after commit 6a497440
.
https://bugzilla.gnome.org/show_bug.cgi?id=686644
2012-10-22 18:11:28 +01:00
Sebastian Rasmussen
1240f98206
tests: Add libgio link dependency
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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
2012-10-22 18:07:52 +01:00
Sebastian Pölsterl
5cec59737b
rtsp-media-mapping: rename find_media vfunc to find_factory
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The virtual method and class method should have the same name
so it is correctly represented in GIR file
https://bugzilla.gnome.org/show_bug.cgi?id=680777
2012-10-18 19:31:23 +01:00
Sebastian Pölsterl
e11e855ac8
rtsp-server: fixed comments and GIR annotations
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https://bugzilla.gnome.org/show_bug.cgi?id=680777
2012-10-18 19:17:01 +01:00
Alessandro Decina
bc474a5b26
media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
2012-10-15 10:50:27 +02:00
Alessandro Decina
1e954a1a5e
rtsp-server: allow binding on port 0 (binds on a random port)
2012-10-15 10:50:27 +02:00
Alessandro Decina
6a49744088
rtsp-server: add bound-port property
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bound-port can be used to retrieve the port number when the server is bound on
port 0, which binds on a random port.
2012-10-15 10:50:27 +02:00
Alessandro Decina
8f507e4512
rtsp-media-factory: make ::get_element overridable by GI bindings
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The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
as the invoker for ::get_element(), making it overridable by GI generated
bindings.
2012-10-15 10:50:26 +02:00
Alessandro Decina
3a49b8e783
rtsp-media-factory-uri: don't autoplug parsers in a loop
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Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
h264parse forever.
2012-10-15 10:50:26 +02:00
Alessandro Decina
8da18a85ef
Explicitly link against gio. Fix link error on mac.
2012-10-15 10:50:26 +02:00
Ognyan Tonchev
4f0ef292f0
session: add ttl to the transport header in SETUP
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See https://bugzilla.gnome.org/show_bug.cgi?id=685561
2012-10-10 11:13:58 +02:00
Ognyan Tonchev
d581b7bd4e
client: Use client transport settings for multicast if allowed.
...
This patch makes it possible for the client to send transport settings for
multicast (destination && ttl). Client settings must be explicitly allowed or
the server will use its own settings.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
2012-10-10 11:07:59 +02:00
Tim-Philipp Müller
1a5cf33f0f
Automatic update of common submodule
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From 6c0b52c to 6bb6951
2012-10-06 15:02:27 +01:00
Patricia Muscalu
870b8db279
rtsp-client: do not destroy the rtsp watch
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Don't destroy the client watch while dispatching. The rtsp watch is
automatically destroyed after the rtsp watch function closed() has
been called.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
2012-10-05 11:44:32 +02:00
Tim-Philipp Müller
7164532536
Automatic update of common submodule
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From 4f962f7 to 6c0b52c
2012-09-22 16:11:48 +01:00
Ognyan Tonchev
f4a0a371b7
media: fix check for seekability
2012-09-10 16:29:35 +02:00
Wim Taymans
3e55e0e467
client: use more GIO
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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
2012-09-07 17:14:30 +02:00
Wim Taymans
87c73c06fb
server: remove obsolete includes
2012-09-07 17:14:10 +02:00
Aleix Conchillo Flaque
c6cce4a6b9
rtsp-media: also initialize transports in on_ssrc_active (bug #683304 )
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* gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
be available in "on_new_ssrc". The transports are added in
gst_rtsp_media_set_state when going to PLAYING state. However,
"on_new_ssrc" might be called before this happens.
https://bugzilla.gnome.org/show_bug.cgi?id=683304
2012-09-07 16:45:17 +02:00
Aleix Conchillo Flaque
bef57648b8
rtsp-client: add signals for rtsp requests ( fixes #683287 )
2012-09-07 16:41:29 +02:00
Aleix Conchillo Flaque
ebc4ce4de1
add new-session signal to rtsp-client ( fixes #683058 )
2012-08-30 22:00:30 +02:00
Stefan Sauer
02b17d6baa
Automatic update of common submodule
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From 668acee to 4f962f7
2012-08-22 13:34:55 +02:00
Patricia Muscalu
50e4c7e8c4
rtsp-server: fixed segfault in gst_rtsp_server_create_socket
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Do not assume that *error is set in g_socket_address_enumerator_next.
Added test_bind_already_in_use unit-test.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
2012-08-20 11:49:27 +02:00
Tim-Philipp Müller
41f9875366
Automatic update of common submodule
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From 94ccf4c to 668acee
2012-08-05 16:43:53 +01:00
Patricia Muscalu
228e2ccc2d
rtsp-client: make create_sdp virtual method
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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
2012-07-24 12:52:53 +02:00
Sebastian Dröge
d803c8e29f
Automatic update of common submodule
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From 98e386f to 94ccf4c
2012-07-23 08:48:25 +02:00
Wim Taymans
f305020636
client: fix docs
2012-07-10 11:39:58 +02:00
Ognyan Tonchev
ed66f974dd
rtsp-server: use an existing socket to establish HTTP tunnel
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Make it possible to transfer a socket from an HTTP server to be used as
an RTSP over HTTP tunnel.
2012-07-10 11:38:05 +02:00
Ognyan Tonchev
86e53af34a
rtsp: Handle the blocksize parameter
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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
2012-07-10 11:26:01 +02:00
Sebastian Rasmussen
4743624bb5
Have unit test get header from source dir, not installed dir
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This makes compilation of unit tests work in a build directory other
than the source directory.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
2012-06-27 11:42:21 +02:00