Wim Taymans
853128e1c7
client: don't leak transports
2012-06-14 10:14:06 +02:00
David Svensson Fors
3f49c2d8f4
rtsp-client: free transport on no_stream in SETUP handler
2012-06-14 10:14:06 +02:00
David Svensson Fors
8f5d82be6d
rtsp-client: changed session media iteration
...
In client_unlink_session: now don't iterate in session->medias
list where items are removed by gst_rtsp_session_release_media.
Instead, repeatedly remove the first item.
2012-06-14 10:14:06 +02:00
David Svensson Fors
dc796bf075
rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
...
GstRTSPSessionMedia is not a GObject type. When the
GstRTSPSession is freed, it will free the media.
2012-06-14 10:14:06 +02:00
David Svensson Fors
7b145aeeab
client: fix GSocketAddress leak in gst_rtsp_client_accept
...
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
2012-06-06 14:49:40 +02:00
David Svensson Fors
ffa3166fbd
rtsp: fix compiler warnings
...
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
2012-05-22 15:37:25 +02:00
Wim Taymans
6cc2fb9bfc
rtsp-server: port to new thread API
2012-05-11 09:42:47 +02:00
Sebastian Dröge
e2f10f5ba5
rtsp-server: Fix compilation and compiler warnings
2012-04-13 15:27:22 +02:00
Wim Taymans
4c59e211e2
rtsp-server: port to GIO
...
Port to GIO
2012-03-07 15:04:29 +01:00
Tim-Philipp Müller
e67a1c664c
rtsp-client: update for new map API
2012-02-13 11:06:33 +00:00
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
6fa73b2552
client: use method to access property
2011-08-16 16:07:04 +02:00
Wim Taymans
9573058f54
client: use media multicast group
2011-08-16 13:43:44 +02:00
Robert Krakora
ae67971cde
sdp: copy and free the server ip address
...
Copy and free the server ip address to make memory management easier later.
2011-08-16 13:31:52 +02:00
Wim Taymans
b0e22d6861
client: do configuration of transport in one place
...
Move the configuration of the transport destination address to where we also
configure the other bits.
2011-08-16 12:51:44 +02:00
Wim Taymans
8749b1e08f
Merge branch 'master' into 0.11
2011-08-16 12:11:59 +02:00
Robert Krakora
f7223cfdab
client: destroy pipeline on client disconnect with no prior TEARDOWN.
...
The problem occurs when the client abruptly closes the connection without
issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP
server is where the pipeline gets torn down. Since this handler is not called,
the pipeline remains and is up and running. Subsequent clients get their own
pipelines and if the do not issue TEARDOWNs then those pipelines will also
remain up and running. This is a resource leak.
2011-08-16 12:09:48 +02:00
Wim Taymans
b5aa7628bf
Merge branch 'master' into 0.11
2011-08-16 11:12:33 +02:00
David Schleef
aa128813fe
client: fix reference counting
2011-07-27 15:02:08 -07:00
Thijs Vermeir
93fb73b46f
fix compiler warnings about unused variables
2011-07-20 17:16:42 +02:00
Wim Taymans
bd8eb8f3d9
client: update for buffer API change
2011-06-13 19:05:57 +02:00
Wim Taymans
914b481e42
rtsp-server: port to 0.11
2011-04-26 19:22:50 +02:00
Wim Taymans
df0e2c2859
client: use the response from the clientstate
...
Create the response object only once and store in the client state.
Make all methods use the state response,
2011-01-12 15:37:39 +01:00
Wim Taymans
4a4a15077b
client: emit signal when closing
2011-01-12 15:35:51 +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
5773df1d52
rtsp-server: Pass ClientState structure arround
...
Pass the collected information for the ongoing request in a GstRTSPClientState
structure that we can then pass around to simplify the method arguments. This
will also be handy when we implement logging functionality.
2011-01-12 13:16:08 +01:00
Wim Taymans
748d044b62
client: unref auth in finalize
2011-01-12 12:07:20 +01:00
Wim Taymans
8ccebd90b4
client: add support for setting the server.
...
Add support for keeping a ref to the server that started this client
connection.
2011-01-12 10:42:52 +01:00
Wim Taymans
c59d9e2970
client: delegate setup of auth to the manager
...
Delegate the configuration of the authentication tokens to the manager object
when configured.
2011-01-12 00:35:28 +01:00
Wim Taymans
5fb5f75020
auth: add authentication object
...
Add an object that can check the authorization of requests.
Implement basic authentication.
Add example authentication to test-video
2011-01-12 00:22:27 +01:00
Wim Taymans
da35feb1aa
rtsp: move network includes where they are needed
2011-01-11 22:42:25 +01:00
Jonas Larsson
b5a1719e89
client: use the socket length from getsockname
...
Use the length returned by getsockname to perform the getnameinfo call because
the size can depend on the socket type and platform.
Fixes #638723
2011-01-05 11:26:30 +01:00
Wim Taymans
160fc25867
docs: improve docs
2010-12-30 12:41:31 +01:00
Wim Taymans
50b4c8de98
rtsp-server: add support for buffer lists
...
Add support for sending bufferlists received from appsink.
Fixes #635832
2010-12-29 16:26:41 +01:00
Wim Taymans
4234d96314
media: make method to retrieve the play range
...
Make a method to retrieve the playback range so that we can conditionally create
a different range for the SDP and the PLAY requests.
2010-12-28 18:35:01 +01:00
Wim Taymans
899f624845
client: fix typo
2010-12-28 12:18:41 +01:00
Edward Hervey
a6556551e3
rtsp-server: Remove unused variable and dead assignment
2010-12-11 10:53:28 +01:00
Edward Hervey
eb83fc6318
rtsp-server: Run gst-indent
...
Since it wasn't using the upstream common previously, there was no
indentation check before commiting.
2010-12-11 10:48:42 +01:00
Wim Taymans
336ffc0941
client: improve client cleanups
...
Make sure the session does not timeout when using TCP. We need to do this
because quicktime player does not send RTCP for some reason in tunneled
mode.
Refactor some cleanup code.
Fixes #612915
2010-04-06 17:08:40 +02:00
Wim Taymans
48a54054e7
client: fix unlink on session timeouts
...
When our session times out, make sure we unlink all streams in this
session.
Remove the tunnelid when closing the connection.
2010-04-06 15:45:56 +02:00
Wim Taymans
30c31a65eb
client: handle lost_tunnel callbacks
...
Handle lost_tunnel callbacks and use it to store the tunnelid back into the
hashtable so that we can reuse it for when the client reopens the POST
socket.
Close the connection after a TEARDOWN.
Make sure or watchid is cleared when the watch is removed.
Fixes #612915
2010-04-06 11:13:51 +02:00
Wim Taymans
09b97dd4ac
rtsp-server: add more support for multicast
2010-03-19 18:03:40 +01:00
Wim Taymans
d749f1e7d5
client: use right size for malloc
2010-03-16 18:33:23 +01:00
Wim Taymans
b3814d4646
client: make content-base better
...
Use the URI formatting functions to make a content-base. Also make sure that
there is a trailing / at the end.
2010-03-09 13:43:29 +01:00
Wim Taymans
171e89c63a
client: guard against invalid paths
2010-03-09 13:42:50 +01:00
Luca Ognibene
e19c382bbb
client: call unlink_streams in client finalize
...
Fixes #599027
2010-03-05 18:37:17 +01:00
Wim Taymans
73e8d6c69a
client: rework transport parsing
...
Rework the transport parsing code so that we can ignore transports we don't
support instead of just picking the first one we can parse.
Configure a (for now hardcoded) destination for multicast transports.
2010-03-05 13:31:37 +01:00
Wim Taymans
ce6724f788
rtsp-client: implement error_full
...
Implement error_full to avoid some segfaults when the rtspconnection calls it.
See #608245
2010-01-27 18:38:27 +01:00
Wim Taymans
996112db95
docs: update docs and comments
2009-12-25 18:24:10 +01:00
Sebastian Pölsterl
3d7610b033
client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG
2009-11-21 19:20:39 +01:00
