Wim Taymans
84e72262d0
client: add generic close-after-send support
...
Add a property to send_response() to close the connection after the response has
been sent to the client.
2012-11-26 13:19:06 +01:00
Wim Taymans
1d53c46d23
MediaMapping -> MountPoints
...
Describes better what the object manages.
2012-11-26 12:37:55 +01:00
Wim Taymans
0f93879b2c
media: fix seeking
2012-11-21 17:21:28 +01:00
Wim Taymans
5eb5fd45f3
media: support more Range formats
...
Use the new -base methods to convert the Range string into a seek start and stop
value.
2012-11-21 16:41:56 +01:00
Wim Taymans
37a7ec8033
factory: keep ref to factory while media active
...
While the media from a factory is alive, keep a ref to the factory.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
2012-11-20 12:29:55 +01:00
Wim Taymans
8fcdca987d
factory-uri: add some debug
2012-11-20 12:29:26 +01:00
Wim Taymans
1826844ee4
stream: set udp sources to PLAYING
...
Set the UDP sources to PLAYING and locked state before we add it to the pipeline
so that it doesn't cause our pipeline to produce ASYNC-DONE.
2012-11-20 12:24:13 +01:00
Wim Taymans
8211cdfdc2
factory-uri: take ref to factory
...
Take a ref to the factory that we place in our list.
2012-11-20 12:10:16 +01:00
David Svensson Fors
0eeb4a5c73
server: start and stop multiple times
...
Stop listening on the RTSP port when the GSource is removed, so clients
can't connect and the server can be started again.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
2012-11-20 11:30:37 +01:00
Wim Taymans
8a7197f078
server: fix small leak
2012-11-20 11:24:35 +01:00
Wim Taymans
989f004e24
media: unref source in finish_unprepare
...
The source is created in prepare, unref it in finish_unprepare.
See https://bugzilla.gnome.org/show_bug.cgi?id=688707
2012-11-20 09:46:40 +01:00
David Svensson Fors
01973c924d
rtsp-media: remove bus watch before finalizing
...
* A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
* An extra media ref is added for the bus watch. This extra ref is unreffed by
the GDestroyNotify function.
* gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
* GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
gst_rtsp_media_unprepare before unreffing the media.
This way, the bus watch will be removed before the media is finalized.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
2012-11-20 09:46:00 +01:00
Alessandro Decina
65042a9551
client: wait until the TEARDOWN response is sent to close the connection
...
Responses can be sent async so we need to wait until the TEARDOWN response has
been written before we close the connection to the client. This avoids the risk
of writing/polling closed sockets.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
2012-11-20 09:32:19 +01:00
David Svensson Fors
0996266342
rtsp-stream: plug socket leak
...
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
2012-11-20 09:26:28 +01:00
Tim-Philipp Müller
0006ca6d60
rtsp-server: don't use deprecated API
2012-11-17 00:11:27 +00:00
Tim-Philipp Müller
290968eb8c
rtsp-client: fix unused-but-set-variable compiler warning
...
rtsp-client.c:1260:21: error: variable 'protocols' set but not used
2012-11-17 00:03:42 +00:00
Wim Taymans
26ff5fc073
rtsp: cleanups
2012-11-15 17:11:16 +01:00
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
44a2855eb3
stream: add methods to deal with address pool
...
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
...
It is a stream property
2012-11-15 15:32:43 +01:00
Wim Taymans
2160d6dbd3
client: set blocksize only on stream
...
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
...
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
...
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
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
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
b30202b174
address-pool: add object to manage multicast addresses
...
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
...
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
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
...
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
...
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
...
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
...
https://bugzilla.gnome.org/show_bug.cgi?id=680777
2012-10-28 15:39:04 +00:00