Commit graph

32 commits

Author SHA1 Message Date
Patricia Muscalu
768fb5695c Get payloader stats only for the sending streams
Get/set payloader properties only for streams that actually
contain a payloader element.

https://bugzilla.gnome.org/show_bug.cgi?id=796523
2018-06-13 10:13:12 +03:00
Mathieu Duponchelle
c725ef01a4 All around: add annotations and API guards 2018-02-12 19:16:11 +01:00
Edward Hervey
64a46d47ba rtsp-server: Minor doc fixes
Mostly for g-i
2017-12-07 16:08:50 +01:00
Edward Hervey
4d98bc5e55 Run gst-indent 2017-11-21 09:53:08 +01:00
Patricia Muscalu
5ec1b80989 rtsp-session-media: add function to get a list of transports
Change-Id: I817e10624da0f3200f24d1b232cff481099278e3

https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-11-15 17:46:00 +02:00
Sebastian Dröge
ccf6c6eb53 Add initial support for RECORD
We currently only support media that is RECORD or PLAY only, not both at once.

https://bugzilla.gnome.org/show_bug.cgi?id=743175
2015-02-06 09:42:42 +01:00
Linus Svensson
a455181aff client: Stop caching media in client when doing setup
If the media has been managed by a session media, it should not be
cached in the client any longer. The GstRTSPSessionMedia object is now
responsible for unpreparing the GstRTSPMedia object using
gst_rtsp_media_unprepare(). Unprepare the media when finalizing the
session media.

https://bugzilla.gnome.org/show_bug.cgi?id=739112
2014-11-07 12:34:23 +01:00
Evan Nemerson
34e6ac3b9f introspection: add (nullable) annotations to return values
https://bugzilla.gnome.org/show_bug.cgi?id=730952
2014-06-26 19:08:16 +02:00
Sebastian Rasmussen
b1b5301577 gobject-introspection: Add annotations to support language bindings
In addition a few cosmetic changes:

 * Adjust the order of arguments
 * Fix typo: occured -> occurred
 * Fix indentation after Return:-clauses

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
2014-03-24 00:36:42 +00:00
Wim Taymans
8bd53dcf9c session: improve RTP-Info
Ignore streams that can't generate RTP-Info instead of failing.
Don't return the empty string when all streams are unconfigured but
return NULL so that we don't generate and empty RTP-Info header.
Improve docs a little.
2014-02-07 16:39:49 +01:00
Andrey Utkin
271f533098 Don't free rtpinfo GString when it is NULL
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
2014-02-07 16:24:08 +01:00
Wim Taymans
8aaa432d58 stream: return clock-rate from get_rtpinfo
And use it to correct the rtptime to the requested start-time.

See https://bugzilla.gnome.org/show_bug.cgi?id=712198
2013-12-26 17:14:06 +01:00
Wim Taymans
037e21b578 session-media: calculate start-time 2013-12-26 16:29:39 +01:00
Wim Taymans
4ca0b23a3f session-media: let the session-media make the RTPInfo
Add method to create the RTPInfo for a stream-transport.
Add method to create the RTPInfo for all stream-transports in a
session-media.
Use the session-media RTPInfo code in client. This allows us to refactor
another method to link the TCP callbacks.
2013-12-26 16:29:38 +01:00
Wim Taymans
2f17369e9d media: add suspend modes
Add support for different suspend modes. The stream is suspended right after
producing the SDP and after PAUSE. Different suspend modes are available that
affect the state of the pipeline. NONE leaves the pipeline state unchanged and
is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
state and RESET will bring the pipeline to the NULL state.
A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
this means that the pipeline needs to be prerolled again.

Base on patches by Ognyan Tonchev <ognyan@axis.com>

See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2013-11-28 16:18:39 +01:00
Sebastian Rasmussen
d1a2853659 rtsp-*: Fix type name typos in comments
* rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
  * rtsp-auth: Refer to part of constant name as text
  * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
  * rtsp-session-media: Fix GstRTSPSessionMedia typo
  * rtsp-stream: Fix typo when refering to GstBin

https://bugzilla.gnome.org/show_bug.cgi?id=714988
2013-11-22 09:13:08 +00:00
Sebastian Pölsterl
e756324490 Fixed several GIR warnings 2013-11-12 11:15:58 +01:00
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
5a833f503e session: use path matching for session media
Use a path string instead of a uri to lookup session media in the sessions. Also
use path matching to find the largest possible path that matches.
2013-07-03 12:37:48 +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
Olivier Crête
b9d111372e Document locking and its order 2013-03-11 11:07:19 +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
1d53c46d23 MediaMapping -> MountPoints
Describes better what the object manages.
2012-11-26 12:37:55 +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
45b6693b39 rtsp: make address-pool return an address object
Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
store more info in the structure and allows us to more easily return the address
to the right pool when no longer needed.
Pass the address to the StreamTransport so that we can return it to the pool
when the stream transport is freed or changed.
2012-11-15 13:25:14 +01:00
Wim Taymans
883cf794e4 session-media: add locking 2012-11-12 16:51:03 +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
543aa383e7 rtsp: only create transport when needed
Only create the StreamTransport when configured.
2012-10-28 00:23:57 +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