Commit graph

34 commits

Author SHA1 Message Date
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
d4e8d800c9 stream: add method to check control url of stream 2013-07-03 15:13:45 +02:00
Wim Taymans
2ffb0f69d2 stream: add methods and property to set control string 2013-07-02 14:50:30 +02:00
Wim Taymans
a7fe63298c stream: add more support for IPv6
Rename _get_address to _get_multicast_address in GstRTSPStream to
make it clear that this function only deals with multicast.
Make it possible to have both an IPv4 and IPv6 multicast address on
a stream. Give the client an IPv4 or IPv6 address depending on the
address it used to connect to the server.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
2013-07-01 16:46:39 +02:00
Wim Taymans
82812988a6 stream: handle failed port allocation
Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
can't allocate any family at all. Also keep track of what port families we
allocated.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
2013-07-01 14:47:33 +02:00
Wim Taymans
284a0a5cd1 stream: improve docs 2013-07-01 12:20:50 +02:00
Aleix Conchillo Flaque
aeaadf0e5e stream: allow access to the rtp session
https://bugzilla.gnome.org/show_bug.cgi?id=703004
2013-06-24 23:42:58 +02:00
Alexander Schrab
c3f8673174 dscp qos support in gst-rtsp-stream
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
2013-06-24 14:51:44 +02:00
Alexander Schrab
3e119be829 rtspstream: handle both ipv4 and ipv6 clients
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
2013-06-03 11:23:40 +02:00
Wim Taymans
7b880231b1 stream: keep the transport object alive
Keep the transport object alive while we have it as qdata on the
source.
2013-05-30 06:49:20 +02:00
Wim Taymans
0ddd98bfa6 stream: set elements to NULL before removing
When removing a stream, set the elements to NULL first. This avoids
element-is-not-in-NULL-state errors when we dispose the elements.
2013-04-23 10:28:34 +02:00
Ognyan Tonchev
00291e5285 stream: add method to get the srcpad 2013-04-22 17:32:31 +02:00
Ognyan Tonchev
6081f91351 stream: clear session and caps for reuse
Set the session and caps to NULL after unref otherwise we might unref
them again later.

See https://bugzilla.gnome.org/show_bug.cgi?id=698376
2013-04-22 16:09:22 +02:00
Olivier Crête
5a39e25949 stream: Select unicast address from pool if appropriate 2013-03-11 11:07:20 +01:00
Olivier Crête
a797cbde06 stream: Properties are always there in Gst 1.0 2013-03-11 11:07:20 +01:00
Olivier Crête
d06e68abd1 address-pool: Add unicast addresses 2013-03-11 11:07:20 +01:00
Olivier Crête
773c48e22f client: Check client provided addresses against the address pool 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
d3d74ab77b stream: improve debug 2012-11-28 12:40:18 +01:00
Wim Taymans
1826844ee4 stream: set udp sources to PLAYING
Set the UDP sources to PLAYING and locked state before we add it to the pipeline
so that it doesn't cause our pipeline to produce ASYNC-DONE.
2012-11-20 12:24:13 +01:00
David Svensson Fors
0996266342 rtsp-stream: plug socket leak
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
2012-11-20 09:26:28 +01:00
Wim Taymans
ba21661ce4 rtsp: improve debug 2012-11-15 16:15:20 +01:00
Wim Taymans
44a2855eb3 stream: add methods to deal with address pool
Add methods to get and set the address pool for the stream
Add method to allocate and get the multicast addresses for this stream.
2012-11-15 15:36:21 +01:00
Wim Taymans
6c2947e68b stream: share src and sink sockets
the allocated socket is in the used-socket property, not socket.
2012-11-15 13:52:07 +01:00
Wim Taymans
4753588b09 stream: add locking 2012-11-13 11:14:49 +01:00
Wim Taymans
c7d20e5603 stream-transport: add keep-alive method 2012-11-12 17:11:18 +01:00
Wim Taymans
75473fc88d stream-transport: add method to handle RTP/RTCP
Call new methods instead of poking into the structures directly.
2012-11-12 17:06:42 +01:00
Tim-Philipp Müller
4dba434f16 Fix FSF address 2012-11-04 00:14:25 +00:00
Wim Taymans
6a838fd5c8 stream: transports must already have been removed 2012-10-26 17:29:30 +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
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