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291 commits

Author SHA1 Message Date
Wim Taymans
7b5763179a rtsp-media: use element metadata to find payloader 
Use the element metadata to find the payloader instead of checking
for the base class.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
 2013-11-22 11:19:35 +01:00
Aleix Conchillo Flaque
e5332535a7 rtsp-stream: add getter for payload type 
* gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.

* gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
  element and create the stream with this one instead of the dynpay%d
  element.

  https://bugzilla.gnome.org/show_bug.cgi?id=712396
 2013-11-22 11:19:35 +01:00
Sebastian Rasmussen
08160e0913 rtsp-*: Refer to NULL as a constant in comments 
Plus one typo fix.

https://bugzilla.gnome.org/show_bug.cgi?id=714988
 2013-11-22 09:13:14 +00:00
Sebastian Pölsterl
e756324490 Fixed several GIR warnings 2013-11-12 11:15:58 +01:00
Wim Taymans
59b53c90c3 rtsp-media: remove old line 2013-10-04 05:48:52 +02:00
Youness Alaoui
917bbfcc20 media: Check dynamically if the pipeline supports seeking 
We should not depend on whether or not the pipeline state change
returned NO_PREROLL or not. A media could dynamically change its
element and switch from seekable to non seekable so it's best to test
the seekable nature of the pipeline dynamically when we try to do a seek.
 2013-10-02 06:00:10 +02:00
Youness Alaoui
33dc78209c media: Return FALSE if seeking is not supported 2013-10-02 05:57:15 +02:00
Wim Taymans
d138f79da0 rtsp-media: don't seek accurate by default 
Accurate seeking is perhaps a little overkill in the most common situation and
causes some formats (mp3) over slow media to seek extremely slowly.
 2013-10-01 17:16:11 +02:00
Jonas Holmberg
19178a413c auth, media, media-factory: unref permissions 
https://bugzilla.gnome.org/show_bug.cgi?id=707638
 2013-09-06 18:57:55 +01:00
Wim Taymans
f094256add media: set protocols on streams 2013-08-16 16:34:56 +02:00
Wim Taymans
04d2da4d03 media-factory: allow all protocols 2013-08-16 16:19:27 +02:00
Wim Taymans
cf96774e6d media: configure protocols in new streams 2013-08-16 16:10:43 +02:00
Youness Alaoui
cdbb6bcc15 rtsp-media: send state in "new-state" signal 
https://bugzilla.gnome.org/show_bug.cgi?id=705110
 2013-08-13 16:41:53 +01:00
Youness Alaoui
a95ab4b29e Add vmethod for rtsp-media subclass to access rtpbin 2013-08-02 16:59:04 +02:00
Wim Taymans
6ac547cc34 media: add docs for new method 2013-08-02 16:57:26 +02:00
Youness Alaoui
050b16ad84 Add API to rtsp-media set the pipeline's state 2013-08-02 16:53:07 +02:00
Youness Alaoui
5e642c7ef1 Update current position/duration when gst_rtsp_media_get_range_string is called 2013-08-02 16:51:15 +02:00
Wim Taymans
041b1b79a1 docs: improve docs 2013-07-16 12:32:51 +02:00
Wim Taymans
0b3644a21b docs: improve docs 2013-07-11 16:57:14 +02:00
Wim Taymans
ccceb1de11 docs: update docs 2013-07-11 12:18:26 +02:00
Wim Taymans
d1e4baab6c media: Accept a thread in _prepare 
Remove out own threadpool handling and use the provided thread and
maincontext for the bus messages and the state changes.
 2013-07-10 17:08:14 +02:00
Wim Taymans
c4c9c873b8 media: start media pipeline in context 
Start the media pipeline in the provided context (or our default one
when NULL). This makes sure that we run the bus thread in this context and that
all media threads are children of this context.
 2013-07-09 20:44:51 +02:00
Wim Taymans
0499a1ec7d media: make it possible to set permissions 
Make it possible to set permissions on media and media factory objects
 2013-07-09 14:33:43 +02:00
Wim Taymans
12583e819c media: add optional context for bus messages 
Add an optional mainloop to _prepare that will handle the bus messages instead
of always using the shared mainloop.
 2013-07-08 11:10:20 +02:00
Wim Taymans
19cffc7999 auth: remove auth from media and factory 
Remove the auth object from media and factory. We want to have the RTSPClient
authenticate and authorize resources, there is no need to place another auth
manager on the media/factory.
 2013-07-05 20:53:19 +02:00
Wim Taymans
3999bd4e4e media: add method to find a stream by control url 2013-07-03 15:14:39 +02:00
Ognyan Tonchev
cd4120ef26 rtsp-media: Do not leak the query in default_query_stop 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
 2013-06-26 15:42:01 +02:00
Wim Taymans
81c3843ad1 media: don't unlock when conversion fails 
Don't unlock the state lock when conversion fails because it was not locked.
 2013-06-25 15:46:41 +02:00
Youness Alaoui
0b94f50eab Add query_position and query_stop vmethods to rtsp-media 2013-06-25 15:23:36 +02:00
Youness Alaoui
842f5ad9c4 Fix typo in property install for rtsp-media's time-provider 2013-06-25 15:12:36 +02:00
Wim Taymans
aab1198516 media: add _get_element() method 
Add method to get the element used when creating the media.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
 2013-06-24 23:56:57 +02:00
Wim Taymans
6d69a4ae80 media: fix docs 2013-06-24 23:51:38 +02:00
Wim Taymans
23ec78faea media: handle segment query format mismatch 
It's possible that the segment query returns with a different format than what
we asked for, handle this case also.
 2013-06-20 11:14:31 +02:00
David Svensson Fors
52eb796bec media: use segment stop in collect_media_stats 
Use segment stop instead of duration as range end point.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
 2013-06-20 10:17:32 +02:00
Ognyan Tonchev
d9e245e62e rtsp-media: Do not leak the element in take_pipeline 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
 2013-06-17 17:18:40 +02:00
David Svensson Fors
6151072a2e media: convert_range replaces get_range_times 
get_range_times worked for handling UTC ranges for seeks, but we also
need to convert back from NPT to the requested unit in
get_range_string. convert_range is now used for both.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
 2013-06-14 16:11:34 +02:00
David Svensson Fors
7efa871c1f media: possibility to override range time conversion 
Make it possible to override the conversion from GstRTSPTimeRange to
GstClockTimes, that is done before seeking on the media
pipeline. Overriding can be useful for UTC ranges, where the default
conversion gives nanoseconds since 1900.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
 2013-06-03 14:29:05 +02:00
Wim Taymans
573b10bc83 media: release lock when removing fakesink 2013-04-23 10:28:35 +02:00
Wim Taymans
b80b8824be media: listen to pad-removed signals 
Listen to the pad-removed signal and remove the stream associated with the
removed pad.
Add signal to be notified of the removed pad.
Remove the fakesink in unprepare()
Fix signatures of the signal methods
 2013-04-22 17:34:37 +02:00
Ognyan Tonchev
a26b06cc69 media: disconnect from signal handlers in unprepare() 
We connected to the pad-added and no-more-pads signals in prepare() so
we need to disconnect from them in unprepare().

See https://bugzilla.gnome.org/show_bug.cgi?id=698376
 2013-04-22 16:40:48 +02:00
Ognyan Tonchev
9b31fcc7f8 media: don't free streams array 
Don't free the streams array in the unprepare() method, they were not
added in prepare().

See https://bugzilla.gnome.org/show_bug.cgi?id=698376
 2013-04-22 16:25:17 +02:00
Ognyan Tonchev
0bdff0161c media: don't unref the pipeline in unprepare 
Unprepare() should undo what prepare() does. Because the pipeline is
not created in prepare(), we should not unref it in unprepare()
 2013-04-22 16:19:35 +02:00
Wim Taymans
a64cb68164 media: add method to get the base_time of the pipeline 
Together with a shared clock, this base-time could eventually be sent to
the client so that it can reconstruct the exact running-time of the clock
on the server.
 2013-04-12 11:34:38 +01:00
Wim Taymans
36ff679558 media: add GstNetTimeProvider support 
Add a property to let the media provide a GstNetTimeProvider for its clock.
Make methods to get the clock and nettimeprovider
Add a x-gst-clock property to the SDP with the IP and port number of the nettime
provider and also the current time of the clock. This should make it possible
for (GStreamer) clients to slave their clock to the server clock.
 2013-04-09 22:38:44 +02:00
Wim Taymans
95bf53513f media: wait for buffering to complete 
Wait for buffering to complete before changing the state to the target state.
 2013-04-09 20:39:58 +02:00
Wim Taymans
ec0718d7c9 media: small cleanup 2013-04-09 20:11:35 +02:00
Olivier Crête
c18eafbb24 rtsp-media/client: Reply to PLAY request with same type of Range 
Remember the type of Range from the PLAY request and use the same type for
the reply.
 2013-03-22 15:53:06 +01:00
Olivier Crête
b9d111372e Document locking and its order 2013-03-11 11:07:19 +01:00
Ognyan Tonchev
0844e8afbc rtsp-media: Make the element a constructor parameter 
https://bugzilla.gnome.org/show_bug.cgi?id=689594
 2012-12-10 10:25:57 +01:00
Wim Taymans
6beabf1ed4 media: match prepare with unprepare 
Really unprepare when there were an equal amount of prepare calls.
 2012-11-30 15:03:15 +01:00
Wim Taymans
ca26588c7e media: media has to be unprepared in finalize 
Because unprepare takes away the last ref on the media.
 2012-11-30 14:58:46 +01:00
Wim Taymans
119674a828 media: let the source unref the last media ref 
the last ref to the media is held by the source so we don't need to add more ref
and unrefs, we simply destroy the media when the source is gone.
 2012-11-30 13:35:05 +01:00
Wim Taymans
339ea9b085 media: improve debug 2012-11-30 12:54:10 +01:00
Wim Taymans
241baba20a media: check state 
Make sure we are in the right state when collecting the position and duration.
Only make ourselves PREPARED when we were previously PREPARING.
 2012-11-30 12:53:02 +01:00
Wim Taymans
edf2ef4f0b media: use g_object_ref/unref for GObjects 2012-11-30 10:05:48 +01:00
Alessandro Decina
00d9a94e1a Fix compiler warning 2012-11-30 06:17:46 +01:00
Wim Taymans
28fd887547 media: avoid element leak 2012-11-29 17:20:56 +01:00
Wim Taymans
4eb010824e media: require an element in media constructor 2012-11-29 17:20:26 +01:00
Wim Taymans
ad00c5e792 rtsp: make object details private 
Make all object details private
Add methods to access private bits
 2012-11-29 11:11:05 +01:00
Wim Taymans
e11287eb7c media: check if prepared for some methods 
Check that the media object is prepared before doing seek and getting the
current position etc.
Add some g_return checks.
 2012-11-28 14:45:30 +01:00
Wim Taymans
fe71114a7d media: unref pipeline in finalize to avoid leaking it 2012-11-28 12:39:37 +01:00
Wim Taymans
d43a31055e rtsp: use gst_object_unref on GstObjects 2012-11-28 12:10:47 +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
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
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
1b4ac6e5b0 media: remove MTU property 
It is a stream property
 2012-11-15 15:32:43 +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
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
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
Wim Taymans
0bb0e3733c small fixes to docs and debug 2012-10-27 11:53:51 +02:00
Wim Taymans
6f7d755894 stream: improve join and leave of the pipeline 
simplify code
Do the cleanup properly
Add some docs
 2012-10-26 17:28:10 +02:00
Wim Taymans
693dd3cfc4 media: move unprepare below default implementation 
Makes it easier to find the default implementation
 2012-10-26 15:23:16 +02:00
Wim Taymans
0d55e1f50c media: signal unprepared when we actually finish 2012-10-26 15:21:50 +02:00
Wim Taymans
84b7cf1590 media: no need to unlock, unprepare does that when needed 2012-10-26 15:19:23 +02:00
Wim Taymans
6b7ff45ca6 rtsp: fix MTU setting 
Fix setting of the MTU. There is no need for a vmethod.
 2012-10-26 12:35:20 +02:00
Wim Taymans
de7c72dec2 rtsp: massive refactoring 
Make GObjects from the remaining simple structures.
Remove GstRTSPSessionStream, it's not needed.
Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
  a GstRTSPStream should be transported to a client.
Rename GstRTSPMediaFactory::get_element -> create_element because that
  more accurately describes what it does.
Make nice methods instead of poking in the structures.
Move some methods inside the relevant object source code.
Use GPtrArray to store objects instead of plain arrays, it is more
  natural and allows us to more easily clean up.
Move the allocation of udp ports to the Stream object. The Stream object
  contains the elements needed to stream the media to a client.
Improve the prepare and unprepare methods. Unprepare should now undo
  everything prepare did. Improve also async unprepare when doing EOS on
  shutdown. Make sure we always unprepare correctly.
 2012-10-25 21:29:58 +02:00
Sebastian Pölsterl
e11e855ac8 rtsp-server: fixed comments and GIR annotations 
https://bugzilla.gnome.org/show_bug.cgi?id=680777
 2012-10-18 19:17:01 +01:00
Ognyan Tonchev
d581b7bd4e client: Use client transport settings for multicast if allowed. 
This patch makes it possible for the client to send transport settings for
multicast (destination && ttl). Client settings must be explicitly allowed or
the server will use its own settings.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
 2012-10-10 11:07:59 +02:00
Ognyan Tonchev
f4a0a371b7 media: fix check for seekability 2012-09-10 16:29:35 +02:00
Aleix Conchillo Flaque
c6cce4a6b9 rtsp-media: also initialize transports in on_ssrc_active (bug #683304) 
* gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
  be available in "on_new_ssrc". The transports are added in
  gst_rtsp_media_set_state when going to PLAYING state. However,
  "on_new_ssrc" might be called before this happens.

  https://bugzilla.gnome.org/show_bug.cgi?id=683304
 2012-09-07 16:45:17 +02:00
Ognyan Tonchev
86e53af34a rtsp: Handle the blocksize parameter 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
 2012-07-10 11:26:01 +02:00
Tim-Philipp Müller
217a46e4c1 rtsp-media: update for gst_element_make_from_uri() changes 2012-06-23 15:06:11 +01:00
David Svensson Fors
36df0dd8be rtsp-media: don't collect media stats when going to NULL 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
 2012-06-14 10:14:06 +02:00
Wim Taymans
6cc2fb9bfc rtsp-server: port to new thread API 2012-05-11 09:42:47 +02:00
Wim Taymans
e0be150e91 media: fix state of the appqueue 2012-03-13 18:10:53 +01:00
Wim Taymans
4c59e211e2 rtsp-server: port to GIO 
Port to GIO
 2012-03-07 15:04:29 +01:00
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
bace3995d5 Merge branch 'master' into 0.11 2011-11-03 12:58:42 +01:00
Wim Taymans
a701e8595e media: add a seekable boolean 
Maintain the seekable state with a new variable instead of reusing the
is_live variable.
 2011-11-03 12:55:24 +01:00
Victor Gottardi
526bbb5a8f Disallow seek in live media 2011-11-03 12:45:18 +01:00
Wim Taymans
c079325169 media: add property for multicast group 
Add a property to configure the multicast group in the media.

Based on patches from Marc Leeman and Robert Krakora.
 2011-08-16 14:50:05 +02:00
Wim Taymans
5b53335873 media: add property for multicast group 
Add a property to configure the multicast group in the media.

Based on patches from Marc Leeman and Robert Krakora.
 2011-08-16 13:25:16 +02:00
Wim Taymans
b5aa7628bf Merge branch 'master' into 0.11 2011-08-16 11:12:33 +02:00
Wim Taymans
bbab01747d media: use new api 2011-08-04 08:59:17 +02:00
Thijs Vermeir
93fb73b46f fix compiler warnings about unused variables 2011-07-20 17:16:42 +02:00
Wim Taymans
80e0b0b19a media: port to new caps API 2011-05-17 09:48:13 +02:00
Wim Taymans
914b481e42 rtsp-server: port to 0.11 2011-04-26 19:22:50 +02:00
Wim Taymans
6959ebd8e8 Merge branch 'master' into 0.11 
Conflicts:
	common
	configure.ac
 2011-04-26 19:07:13 +02:00
Wim Taymans
a924e90c79 media: remove more unused code 2011-02-02 15:37:03 +01:00
Wim Taymans
ec2201a3a8 media: remove duplicate filtering 
Remove the duplicate filtering code now that we have a released -good version.
Give a warning instead.
 2011-02-02 15:30:45 +01:00
Wim Taymans
8477fdbf43 media: fix default buffer size 2011-01-31 17:38:47 +01:00
Wim Taymans
88b4c02dff media: add property to configure kernel buffer sizes 
Add a property to configure the kernel UDP buffer size.
 2011-01-31 17:37:41 +01:00
Wim Taymans
44b418b346 media: init debug category before starting thread 2011-01-13 18:57:15 +01:00
Wim Taymans
9e97faf2db server: improve debugging in various objects 2011-01-12 18:14:48 +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
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
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
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
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
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
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
Wim Taymans
422fea478c media: warn and fail when gstrtpbin is not found 2010-12-06 19:29:53 +01:00
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
David Schleef
6a880e53df Add stdlib.h for atoi() 2010-08-09 12:56:23 -07:00
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
6afa5be799 media: allow for ipv6 sockets 2010-03-10 11:45:06 +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
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
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
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
Sebastian Pölsterl
6d227be7a9 Use GStreamer's debugging subsystem 2009-11-21 19:20:23 +01:00
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
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