Wim Taymans
e4ea72ccdf
stream: use the address managed by the stream
...
Use the address managed by the stream for multicast. This allows us to have 1
multicast address for each stream.
Because the address is now managed by the stream we don't have to pass it around
anymore.
Set the address pool on the streams.
2012-11-15 16:18:29 +01:00
Wim Taymans
ba21661ce4
rtsp: improve debug
2012-11-15 16:15:20 +01:00
Wim Taymans
c34f5d1c1a
media: add signal for new streams
...
This allows applications to listen for new streams and configure properties on
them, like the address pool.
2012-11-15 15:41:42 +01:00
Wim Taymans
4168a67992
media: configure address pool in new streams
2012-11-15 15:41:19 +01:00
Wim Taymans
1b4ac6e5b0
media: remove MTU property
...
It is a stream property
2012-11-15 15:32:43 +01:00
Wim Taymans
f15ffb521c
rtsp: use AddressPool
...
Remove the multicast_group property.
Use the configured addresspool to allocate multicast addresses.
2012-11-14 17:23:59 +01:00
Wim Taymans
dfe3efef74
media: wait for concurrent _prepare
...
If a prepare is busy, wait for the result.
2012-11-13 11:54:17 +01:00
Wim Taymans
47127bd270
media: add lock around message handler
...
We don't want to dispatch messages while we are still processing the result of
the state change.
2012-11-13 11:49:08 +01:00
Wim Taymans
9a97de88ea
media: add lock to protect state changes
2012-11-13 11:15:35 +01:00
Tim-Philipp Müller
4dba434f16
Fix FSF address
2012-11-04 00:14:25 +00:00
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
Wim Taymans
0bb0e3733c
small fixes to docs and debug
2012-10-27 11:53:51 +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
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
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 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
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
Ognyan Tonchev
f4a0a371b7
media: fix check for seekability
2012-09-10 16:29:35 +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
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
Tim-Philipp Müller
217a46e4c1
rtsp-media: update for gst_element_make_from_uri() changes
2012-06-23 15:06:11 +01:00
David Svensson Fors
36df0dd8be
rtsp-media: don't collect media stats when going to NULL
...
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
2012-06-14 10:14:06 +02:00
Wim Taymans
6cc2fb9bfc
rtsp-server: port to new thread API
2012-05-11 09:42:47 +02:00
Wim Taymans
e0be150e91
media: fix state of the appqueue
2012-03-13 18:10:53 +01:00
Wim Taymans
4c59e211e2
rtsp-server: port to GIO
...
Port to GIO
2012-03-07 15:04:29 +01:00
Wim Taymans
fde25cd9c3
rtsp-server: port some more to 0.11
...
Fix caps.
Remove bufferlist stuff
Update for new API.
Add queue before appsink now that preroll-queue-len is gone.
Update for request pad changes.
2011-12-09 10:53:30 +01:00
Wim Taymans
bace3995d5
Merge branch 'master' into 0.11
2011-11-03 12:58:42 +01:00
Wim Taymans
a701e8595e
media: add a seekable boolean
...
Maintain the seekable state with a new variable instead of reusing the
is_live variable.
2011-11-03 12:55:24 +01:00
Victor Gottardi
526bbb5a8f
Disallow seek in live media
2011-11-03 12:45:18 +01:00
Wim Taymans
c079325169
media: add property for multicast group
...
Add a property to configure the multicast group in the media.
Based on patches from Marc Leeman and Robert Krakora.
2011-08-16 14:50:05 +02:00
Wim Taymans
5b53335873
media: add property for multicast group
...
Add a property to configure the multicast group in the media.
Based on patches from Marc Leeman and Robert Krakora.
2011-08-16 13:25:16 +02:00
Wim Taymans
b5aa7628bf
Merge branch 'master' into 0.11
2011-08-16 11:12:33 +02:00
Wim Taymans
bbab01747d
media: use new api
2011-08-04 08:59:17 +02:00
Thijs Vermeir
93fb73b46f
fix compiler warnings about unused variables
2011-07-20 17:16:42 +02:00
Wim Taymans
80e0b0b19a
media: port to new caps API
2011-05-17 09:48:13 +02:00
Wim Taymans
914b481e42
rtsp-server: port to 0.11
2011-04-26 19:22:50 +02:00
Wim Taymans
6959ebd8e8
Merge branch 'master' into 0.11
...
Conflicts:
common
configure.ac
2011-04-26 19:07:13 +02:00
Wim Taymans
a924e90c79
media: remove more unused code
2011-02-02 15:37:03 +01:00
Wim Taymans
ec2201a3a8
media: remove duplicate filtering
...
Remove the duplicate filtering code now that we have a released -good version.
Give a warning instead.
2011-02-02 15:30:45 +01:00
Wim Taymans
8477fdbf43
media: fix default buffer size
2011-01-31 17:38:47 +01:00
Wim Taymans
88b4c02dff
media: add property to configure kernel buffer sizes
...
Add a property to configure the kernel UDP buffer size.
2011-01-31 17:37:41 +01:00
Wim Taymans
44b418b346
media: init debug category before starting thread
2011-01-13 18:57:15 +01:00
Wim Taymans
9e97faf2db
server: improve debugging in various objects
2011-01-12 18:14:48 +01:00
Wim Taymans
7797023fda
media: enable per factory authorisations
...
Allow for adding a GstRTSPAuth on the factory and media level and check
permissions when accessing the factory.
Add hints to the auth methods for future more fine grained authorisation.
Add example application for per factory authentication.
2011-01-12 13:57:09 +01:00
Wim Taymans
e1787e0776
funnel: rename fsfunnel to rtspfunnel
...
Rename the funnel to avoid conflicts with the farsight one.
2011-01-10 15:10:53 +01:00
Wim Taymans
7b3cbfde1b
rtsp-media: add and use fsfunnel
...
Add a copy of fsfunnel to the build because input-selector removed the (broken)
select-all property that we need.
2011-01-10 13:43:10 +01:00
Wim Taymans
790c067919
media: attempt to configure bigger UDP buffers
...
Attempt to configure bigger udp kernel send buffers to avoid overflowing the
send buffers with high bitrate streams.
2011-01-05 12:06:23 +01:00