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241 commits

Author SHA1 Message Date
Wim Taymans
6d6ba1ee61 server: separate create and accept 
Create separate create and accept methods so that subclasses can create custom
client object.
Configure the server in the client object and prepare for keeping track of
connected clients.
 2011-01-12 10:57:08 +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
9f52f281ba auth: fix memleak and add some docs 
Fix a memleak of the basic auth token.
Add docs for the helper function
 2011-01-12 10:41:42 +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
61bee9985a server: move includes back 
the includes are needed for sockaddr_in.
 2011-01-12 00:20:36 +01:00
Wim Taymans
da35feb1aa rtsp: move network includes where they are needed 2011-01-11 22:42:25 +01:00
Sreerenj Balachandran
28597c913d rtsp-media.h: Minor corrections in comments. 
Fixes #638944
 2011-01-11 21:32:45 +01:00
Edward Hervey
2cc9eee3e6 gitignore: updates 2011-01-11 13:04:31 +01:00
Wim Taymans
e1787e0776 funnel: rename fsfunnel to rtspfunnel 
Rename the funnel to avoid conflicts with the farsight one.
 2011-01-10 15:10:53 +01:00
Wim Taymans
7b3cbfde1b rtsp-media: add and use fsfunnel 
Add a copy of fsfunnel to the build because input-selector removed the (broken)
select-all property that we need.
 2011-01-10 13:43:10 +01:00
Tim-Philipp Müller
c19eb8fb4e gobject-introspection: use PKG_CONFIG_PATH specified at configure time 
Use PKG_CONFIG_PATH specified at configure time (if any) as well
for the g-ir-compiler, rather than just assuming the env var has
been set.
 2011-01-08 02:00:12 +00:00
Tim-Philipp Müller
8b1ec41d08 gobject-introspection: fix g-i build for uninstalled setup 
Requires gst-plugins-base git (> 0.10.31.2).
 2011-01-08 01:15:35 +00:00
Wim Taymans
186089ff1e factory-uri: use right property type 2011-01-07 11:24:39 +01:00
Wim Taymans
257bac1bab factory-uri: attempt to configure buffer-lists 
Attempt to configure buffer lists in the payloader for improved performance.
 2011-01-05 12:07:42 +01:00
Wim Taymans
790c067919 media: attempt to configure bigger UDP buffers 
Attempt to configure bigger udp kernel send buffers to avoid overflowing the
send buffers with high bitrate streams.
 2011-01-05 12:06:23 +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
915cd708ea media: add signal to notify of state changes 2010-12-28 18:34:10 +01:00
Wim Taymans
43f4696f78 client: cleanup headers 2010-12-28 18:31:26 +01:00
Wim Taymans
899f624845 client: fix typo 2010-12-28 12:18:41 +01:00
Wim Taymans
50a71b9d86 factory-uri: add support for gstpay 
Add an option to prefer gstpay over decoder + raw payloader.
 2010-12-23 18:53:01 +01:00
Wim Taymans
9ce4ea165b factory-uri: rework the autoplugger. 
Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
before payloaders.
 2010-12-23 15:58:14 +01:00
Wim Taymans
1053860112 factory-uri: use better factory filter 
Make better payloader filter based on autoplug rank and RTP use case.
 2010-12-21 17:39:09 +01:00
Wim Taymans
ad2e0edee5 server: set SO_REUSEADDR before bind 
Set the SO_REUSEADDR _before_ bind() to make it actually work.
 2010-12-18 11:24:48 +01:00
Wim Taymans
1ea450179e media: emit prepared signal when prepared 
Make a 'prepared' signal and emit it when we successfully prepared the element.
This signal can be used to configure the media object after it has been prepared
for streaming.
 2010-12-17 18:45:10 +01:00
Wim Taymans
fc12ade012 Merge branch 'master' into 0.11 
Conflicts:
	common
	configure.ac
 2010-12-13 11:43:13 +01:00
Wim Taymans
ca76a73ca0 media: update range when active clients changed 
When we changed the number of active clients, update the current range
information because we want the second client connecting to a shared resource
continue from where the stream currently.
 2010-12-12 15:48:47 +01:00
Wim Taymans
d99a448f79 factory-uri: add colorspace and fix pt 
Rework the way we pass data to the autoplugger.
When we have raw caps, plug a converter element to make pluggin to raw
payloaders more successful.
Make sure all dynamically plugged payloaders have a unique payload types.
 2010-12-12 04:06:41 +01:00
Wim Taymans
7ef0bf98da factory-uri: add a factory to stream any URI 
Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
when we have one.
 2010-12-11 18:04:34 +01:00
Wim Taymans
34f0973831 media: ignore spurious ASYNC_DONE messages 
When we are dynamically adding pads, the addition of the udpsrc elements will
trigger an ASYNC_DONE. We have to ignore this because we only want to react to
the real ASYNC_DONE when everything is prerolled.
 2010-12-11 18:04:34 +01:00
Wim Taymans
75a7cda97d media-factory: make lock macro 2010-12-11 18:04:29 +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
Edward Hervey
b95165fcff rtsp-server: Some more doc fixups 2010-12-11 10:48:25 +01:00
Edward Hervey
bdd477f2bf Makefile.am: Use standard GIR make behaviour 2010-12-07 18:14:39 +01:00
Wim Taymans
422fea478c media: warn and fail when gstrtpbin is not found 2010-12-06 19:29:53 +01:00
Sebastian Pölsterl
347e10e1f9 Added initial gobject-introspection support 2010-09-23 13:39:42 +02:00
Wim Taymans
c310f0032c media-factory: don't use host for shared hash key 
When we generate the key to share made between connections, don't include the
host used to connect so that we can share media even if between clients that
connected with localhost and ones with the ip address.
 2010-09-23 11:35:40 +02:00
Wim Taymans
ed473f6f26 media: help the compiler a little 2010-09-22 16:15:56 +02:00
Wim Taymans
450b68252f media: cleanup media transport before freeing 
Cleanup the media transport data before freeing. In particular, remove the qdata
from the rtpsource object.
 2010-08-24 16:47:30 +02:00
Wim Taymans
dc33070da3 media-factory: add eos-shutdown property 
Add an eos-shutdown property that will send an EOS to the pipeline before
shutting it down. This allows for nice cleanup in case of a muxer.

Fixes #625597
 2010-08-20 18:17:08 +02:00
Wim Taymans
a900866570 media: use multiudpsink send-duplicates when we can 
If we have a new enough multiudpsink with the send-duplicates property, use this
instead of doing our own filtering. Our custom filtering code should eventually
be removed when we can depend on a released -good.
 2010-08-20 15:58:39 +02:00
Wim Taymans
c89d17ca26 media: don't leak destinations 
Refactor and cleanup the destinations array when the stream is destroyed.
 2010-08-20 13:19:56 +02:00
Wim Taymans
7c0f8a77ec media: don't add udp addresses multiple times 
Keep track of the udp addresses we added to udpsink and never add the same udp
destination twice. This avoids duplicate packets when using multicast.
 2010-08-20 13:09:12 +02:00
Wim Taymans
af732fa749 server: disable use of SO_LINGER 
SO_LINGER cause the client to fail to receive a TEARDOWN message because the
server close()s the connection.
 2010-08-20 10:18:34 +02:00
Wim Taymans
2607ff079d server: use 5 second linger period in SO_LINGER 
Wait 5 seconds before clearing the send buffers and reseting the connection with
the client when we do a close. This should be enough time to get the message to
the client.

See #622757
 2010-08-19 18:52:47 +02:00
Robert Krakora
8f6fd32065 server: use SO_LINGER 
SO_LINGER on the socket will make sure that any pending data on the socket is
flushed ASAP and that the socket connection is reset. This makes sure that the
socket can be reused immediately.

Fixes 622757
 2010-08-16 12:45:24 +02:00
David Schleef
6a880e53df Add stdlib.h for atoi() 2010-08-09 12:56:23 -07:00
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
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
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