Commit graph

288 commits

Author SHA1 Message Date
Tim-Philipp Müller
b0c6a9aa33 configure: fail if GStreamer core/base requirements are not met 2010-04-25 16:35:59 +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
4fdd2bf4d1 session: add support for prevent session timeouts
Add an atomix counter to prevent session timeouts when we are, for example,
streaming over TCP.
2010-04-06 17:07:27 +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
558c7fddd2 session: small cleanups 2010-04-06 15:44:45 +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
ac8343ea62 media: allow configuration of allowed lower transport 2010-03-19 15:15:29 +01:00
Wim Taymans
e866345f15 rtsp: keep track of server ip and ipv6
Keep track of how the client connected to the server and setup the udp ports
with the same protocol.
Copy the server ip address in the SDP so that clients can send RTCP back to
us.
2010-03-16 18:37:18 +01:00
Wim Taymans
4eccdd9dd7 session: indent 2010-03-16 18:34:43 +01:00
Wim Taymans
d749f1e7d5 client: use right size for malloc 2010-03-16 18:33:23 +01:00
Wim Taymans
0509aa1cbf server: comment ipv6 server listening address 2010-03-10 11:45:30 +01:00
Wim Taymans
6afa5be799 media: allow for ipv6 sockets 2010-03-10 11:45:06 +01:00
Wim Taymans
17bb89f1fc server: rework server part
Allow setting a bind address, make sure we can deal with ipv6.
Remove the port property and change with the service property.
2010-03-09 13:49:00 +01:00
Wim Taymans
1b0dc41534 media: update comments a little 2010-03-09 13:44:20 +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
Wim Taymans
68804ff984 test: catch server bind errors 2010-03-09 13:41:33 +01:00
Alessandro Decina
5f535ecf87 rtspmedia: emit "unprepared" if _prepare fails.
Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
media object is removed from its factory's cache.
2010-03-09 10:27:38 +01:00
Wim Taymans
2997806d43 media: collect media position when seek completes 2010-03-05 19:08:08 +01:00
Luca Ognibene
e19c382bbb client: call unlink_streams in client finalize
Fixes #599027
2010-03-05 18:37:17 +01:00
Wim Taymans
83ed258684 media: limit the time to wait to something huge
Avoid waiting forever but limit the timeout to 20 seconds.
2010-03-05 18:23:18 +01:00
Wim Taymans
f90c422e62 sdp: reindent and check for prepared status 2010-03-05 17:57:08 +01:00
Wim Taymans
c7ca9b74eb media: avoid doing _get_state() for state changes
When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
until the media is prerolled or in error. This avoids doing a blocking call of
gst_element_get_state() that can cause lockups when there is an error.

Fixes #611899
2010-03-05 17:54:09 +01:00
Wim Taymans
d45eae2edd media: reindent 2010-03-05 16:20:08 +01:00
Wim Taymans
851e8aa744 media-factory: better error handling
Improve the error handling a bit.
2010-03-05 13:34:15 +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
53f8350b36 media: set multicast sink parameters
Disable loop and automatic multicast join on the udpsink elements.
Add some more debug info.
Reset some state variables in the right place.
Use the right port numbers for multicast.
2010-03-05 13:28:58 +01:00
Wim Taymans
63addbc278 session: handle transport setup correctly
Handle UDP, MCAST and TCP transport negotiation more correctly.
Store the server session SSRC in the transport.
2010-03-05 13:27:18 +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
Nikolay Ivanov
92eb244215 sdp: make server work better when behind a proxy 2009-12-25 15:22:23 +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
Sebastian Pölsterl
6d227be7a9 Use GStreamer's debugging subsystem 2009-11-21 19:20:23 +01:00
Sebastian Pölsterl
87fbfa54a0 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams 2009-11-21 19:20:23 +01:00
Wim Taymans
f8604a6bc7 back to development 2009-11-05 11:22:44 +01:00
Wim Taymans
95797040eb release 0.10.5 2009-11-05 11:20:45 +01:00
Wim Taymans
07f6c4de5a configure: bump required versions 2009-10-14 12:11:31 +02:00
Luca Ognibene
745900dd48 client: call weak-unref on client->sessions from finalize
Fixes bug #596305
2009-10-13 10:57:35 +02:00
Sebastian Pölsterl
f8630c6c81 media: Fixed crasher where caps got unref'ed too often 2009-10-13 10:57:31 +02:00
Sebastian Pölsterl
7d38b37ae6 Added pkg-config file to use gst-rtsp-server uninstalled 2009-10-13 10:56:22 +02:00
Wim Taymans
297b6a755a media: add some docs 2009-09-11 13:52:27 +02:00
Peter Kjellerstedt
309f53a12b rtsp: Use gst_rtsp_watch_send_message().
Use gst_rtsp_watch_send_message() since the old API which used
gst_rtsp_watch_queue_message() has been deprecated.
2009-08-24 13:27:00 +02:00
Wim Taymans
02c60f3529 back to development 2009-08-05 11:53:56 +02:00
Wim Taymans
5ec236326c Release 0.10.4 2009-08-05 11:44:49 +02:00
Wim Taymans
7338ab81e1 rtsp: allocate channels in TCP mode
When the client does not provide us with channels in TCP mode, allocate channels
ourselves.
2009-07-27 19:42:44 +02:00
Wim Taymans
daccf6bc99 client: don't crash when tunnelid is missing
When a clients tries to open an HTTP tunnel but fails to provide a tunnelid,
don't crash but return an error response to the client.

Fixes #589489
2009-07-24 12:49:41 +02:00
Sebastian Pölsterl
748290b888 bindings: update vala bindings with new method 2009-07-13 11:31:23 +02:00
Wim Taymans
a4c90c28c7 sessionpool: add function to filter sessions
Add generic function to retrieve/remove sessions.
2009-06-30 21:27:53 +02:00
Tim-Philipp Müller
a403469a03 configure: bump core/base requirements to release 2009-06-22 18:57:25 +01:00