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Author SHA1 Message Date
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 
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 
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 
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 
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 
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 
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 
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 
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 
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 
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 
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 
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 
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 
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 
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 
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 
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 
From 6c0b52c to 6bb6951
 2012-10-06 15:02:27 +01:00
Patricia Muscalu
870b8db279 rtsp-client: do not destroy the rtsp watch 
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 
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 
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) 
* 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 
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 
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 
From 94ccf4c to 668acee
 2012-08-05 16:43:53 +01:00
Patricia Muscalu
228e2ccc2d rtsp-client: make create_sdp virtual method 
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 
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 
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 
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 
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
Tim-Philipp Müller
217a46e4c1 rtsp-media: update for gst_element_make_from_uri() changes 2012-06-23 15:06:11 +01:00