mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-09-21 19:40:15 +00:00
Code Issues 1 Projects Releases Wiki Activity
164 commits 44 branches 1508 tags 419 MiB
Commit graph

118 commits

Author SHA1 Message Date
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
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
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
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
Wim Taymans
a4c90c28c7 sessionpool: add function to filter sessions 
Add generic function to retrieve/remove sessions.
 2009-06-30 21:27:53 +02:00
Wim Taymans
5d4c0e20c0 media: fix indentation 2009-06-18 16:05:18 +02:00
Sebastian Pölsterl
f384231ca3 Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often. 2009-06-18 15:54:15 +02:00
Sebastian Pölsterl
036550bf60 set state and remove elements of media in for loop 2009-06-18 15:54:11 +02:00
Sebastian
3bd2d36b1b Added gst_rtsp_media_remove_elements function 2009-06-18 15:54:04 +02:00
Sebastian
1a3e5b369c Don't use name for gstrtpbin so we can add multiple instances to the pipeline 2009-06-18 15:54:01 +02:00
Sebastian Pölsterl
749765b921 Added vmethod unprepare to GstRTSPMedia 
The default implementation sets the state of the pipeline to GST_STATE_NULL
 2009-06-18 15:53:49 +02:00
Sebastian Pölsterl
045875ecbe Made collect_streams function public 2009-06-18 15:53:42 +02:00
Sebastian Pölsterl
e417d83dce Added vmethod create_pipeline to GstRTSPMediaFactory 
The pipeline is created in this method and the GstRTSPMedia's element is added to it
 2009-06-18 15:53:34 +02:00
Wim Taymans
a697d16c75 client: use g_source_destroy() 
We need to use g_source_destroy() because we might have added the source to a
different main context than the default one.
 2009-06-11 11:27:47 +02:00
Wim Taymans
5e4757eff6 rtsp: prepare for handling GET/SET_PARAMETER 
Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there
is a body now.
Fix return codes of handlers.
 2009-06-10 00:01:07 +02:00
Wim Taymans
94b6da045a media: don't leak session pads 2009-06-04 19:20:26 +02:00
Wim Taymans
9a38f95417 media: clean up the messages a bit 2009-06-04 18:32:15 +02:00
Wim Taymans
e1765dec13 sdp: warn and skip streams without media 2009-06-03 12:13:21 +02:00
Wim Taymans
03ae66062b media: fix message 
Fix a debug message
Make dumping RTCP stats configurable
 2009-05-27 11:15:22 +02:00
Wim Taymans
3fc1439965 media: be less verbose and leak less 2009-05-26 19:20:07 +02:00
Wim Taymans
1340e21239 media: don't leak the destination address 2009-05-26 19:07:33 +02:00
Wim Taymans
9bed89c3b7 rtsp: use RTCP to keep the session alive 
Use the RTCP rtcp-from stats field to find the associated session and use this
to keep the session alive.
 2009-05-26 19:01:10 +02:00
Wim Taymans
7bbdf7bf97 session: add 5sec to the real session timeout 
Allow the session to live 5sec longer before really timing out. This should give
clients some extra time to keep the session active.
 2009-05-26 17:27:07 +02:00
Wim Taymans
461169537b client: replay OK to GET/SET_PARAMETER 
Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it
so that we return OK for those requests.
 2009-05-26 17:25:59 +02:00
Wim Taymans
5955fc7d12 media: keep track of active transports 
Keep track of which transport is active to avoid closing the connection too
soon.
Remove the destination transport also when going to NULL.
Print some stats about the SDES and other RTCP messages we receive from the
clients.
 2009-05-26 11:42:41 +02:00
Wim Taymans
7a8b931a83 media: also count active TCP connections 2009-05-24 19:56:45 +02:00
Wim Taymans
fab65082da rtsp: add support for dynamic elements 
Add support for dynamic elements.
Don't set live pipelines back to paused.
 2009-05-24 19:34:52 +02:00
Wim Taymans
415e5e674b sdp: don't add encoding name when absent in caps 2009-05-24 19:33:22 +02:00
Wim Taymans
740d71bd50 client: warn when we can't do RTP-Info 2009-05-23 16:30:55 +02:00
Wim Taymans
e5dc7c3719 factory: factor out the stream construction 2009-05-23 16:18:04 +02:00
Wim Taymans
8fcbe501dc client: only add RTP-Info when we have the info 
Only add RTP-Info for a stream when we can get the seqnum and timestamp from the
depayloader.
 2009-05-23 16:17:02 +02:00
Wim Taymans
b83f54f159 media: link the RTP udpsrc to the session manager 
Link the RTP udpsrc and the appsrc to the session manager so that they don't
shut down when the client sends a packet to open firewalls.
 2009-05-15 17:58:44 +02:00
Wim Taymans
5f19d4b09e media: seek to key frames 2009-04-29 17:25:04 +02:00
Wim Taymans
6ffd7432a5 media: emit the unprepared signal by id 
Emit the unprepared signal by id instead of name and set the media as
reused.
 2009-04-21 22:44:05 +02:00
Sebastian Pölsterl
708c8daaec Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare 2009-04-21 22:40:01 +02:00
Sebastian Pölsterl
9b7cb2a4ef Added finalize function to GstRTPSPServer to unref session pool and media mapping 2009-04-21 00:14:41 +02:00
Wim Taymans
3f1f38f479 server: use appsink and appsrc with the API 
Use the appsink/appsrc API instead of the signals for higher
performance.
 2009-04-14 23:38:58 +02:00