Wim Taymans
44a2855eb3
stream: add methods to deal with address pool
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Add methods to get and set the address pool for the stream
Add method to allocate and get the multicast addresses for this stream.
2012-11-15 15:36:21 +01:00
Wim Taymans
1b4ac6e5b0
media: remove MTU property
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It is a stream property
2012-11-15 15:32:43 +01:00
Wim Taymans
2160d6dbd3
client: set blocksize only on stream
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Set the blocksize only on the current stream.
2012-11-15 15:29:35 +01:00
Wim Taymans
6c2947e68b
stream: share src and sink sockets
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the allocated socket is in the used-socket property, not socket.
2012-11-15 13:52:07 +01:00
Wim Taymans
45b6693b39
rtsp: make address-pool return an address object
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Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
store more info in the structure and allows us to more easily return the address
to the right pool when no longer needed.
Pass the address to the StreamTransport so that we can return it to the pool
when the stream transport is freed or changed.
2012-11-15 13:25:14 +01:00
Wim Taymans
fde8c01a85
examples: add multicast example
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Show how to set up the multicast address pool so that media can be
server with multicast.
2012-11-15 13:22:54 +01:00
Wim Taymans
f15ffb521c
rtsp: use AddressPool
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Remove the multicast_group property.
Use the configured addresspool to allocate multicast addresses.
2012-11-14 17:23:59 +01:00
Wim Taymans
d0ffc8e679
address-pool: add clear method
2012-11-14 16:20:36 +01:00
Wim Taymans
6085b1fcc1
address-pool: small cleanups
2012-11-14 16:10:45 +01:00
Wim Taymans
d6fcf92601
tests: add addresspool unit test
2012-11-14 15:50:42 +01:00
Wim Taymans
b30202b174
address-pool: add object to manage multicast addresses
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Make an object that can manage a rage of multicast addresses and ports.
2012-11-14 15:49:06 +01:00
Wim Taymans
7d6e4606fa
server: set default max-threads property
2012-11-13 12:05:42 +01:00
Wim Taymans
dfe3efef74
media: wait for concurrent _prepare
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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
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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
Wim Taymans
4753588b09
stream: add locking
2012-11-13 11:14:49 +01:00
Wim Taymans
c7d20e5603
stream-transport: add keep-alive method
2012-11-12 17:11:18 +01:00
Wim Taymans
75473fc88d
stream-transport: add method to handle RTP/RTCP
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Call new methods instead of poking into the structures directly.
2012-11-12 17:06:42 +01:00
Wim Taymans
883cf794e4
session-media: add locking
2012-11-12 16:51:03 +01:00
Wim Taymans
11cf3f3ccb
session: add locking
2012-11-12 16:42:37 +01:00
Wim Taymans
65fa516677
server: free old socket
2012-11-12 16:30:16 +01:00
Wim Taymans
04881bd632
mapping: add locking
2012-11-12 16:18:57 +01:00
Wim Taymans
b8cba7719c
media-factory: add locking
2012-11-12 16:14:19 +01:00
Wim Taymans
e61c84c9bb
auth: add locking
2012-11-12 16:03:21 +01:00
Wim Taymans
06cadebe71
server: add max-thread property
2012-11-12 15:53:28 +01:00
Wim Taymans
8523c9ca92
server: use a threadpool for the mainloops
2012-11-12 15:29:39 +01:00
Wim Taymans
c431592976
client: rename method
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gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
don't really create the client from the socket, we use the socket for the
client.
2012-11-12 15:01:13 +01:00
Wim Taymans
a58d404e1f
server: rework maincontext handling in clients
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Make a separate method to attach a client to a MainContext.
Let the server decide in what GMainContext the client will operate and give this
context to the client in attach. Then the server can later decide to use a
separate thread for each client or just use the mainthread.
2012-11-12 15:01:09 +01:00
Wim Taymans
5b4340067a
session: move session header code in session object
2012-11-12 12:40:34 +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
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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