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10668 commits 44 branches 1510 tags 469 MiB
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384 commits

Author SHA1 Message Date
Wim Taymans
de020130e6 fix for probe updates 2011-11-07 17:14:17 +01:00
Wim Taymans
768e3826ab more template fixes 2011-11-04 17:39:15 +01:00
Wim Taymans
a95acb7122 make %u in all request pad templates 2011-11-04 11:58:22 +01:00
Wim Taymans
0560ab53c0 update for new task api 2011-11-02 09:06:37 +01:00
Wim Taymans
9a8a8e72c8 structure: fix for api update 2011-11-02 09:06:37 +01:00
Tim-Philipp Müller
9f77b02b15 Update for pad API changes 
GstProbeType, GstProbeReturn and GstActivateMode -> GstPad*
 2011-11-01 00:52:28 +00:00
Wim Taymans
87fbd1e784 Merge branch 'master' into 0.11 
Conflicts:
	common
	ext/pulse/pulsesink.c
	ext/soup/gstsouphttpclientsink.c
	gst/audioparsers/gstaacparse.c
	gst/audioparsers/gstac3parse.c
	gst/rtp/gstrtph264depay.c
	gst/rtpmanager/gstrtpjitterbuffer.c
	gst/rtpmanager/rtpjitterbuffer.c
	gst/rtsp/gstrtspsrc.c
	sys/ximage/gstximagesrc.c
 2011-09-28 12:44:59 +02:00
Mark Nauwelaerts
81fc784163 rtspsrc: do not set elements to PLAYING when doing seek in PAUSED 2011-09-19 11:56:44 +02:00
Mark Nauwelaerts
8599801cae rtspsrc: switch to rtp time based syncing when guessed appropriate 2011-09-19 11:52:08 +02:00
Mark Nauwelaerts
3e33a7a09f rtspsrc: configure rtcp interval if provided 
... in PLAY response.
 2011-09-19 11:51:47 +02:00
Mark Nauwelaerts
95b5ece2c9 rtspsrc: ensure some initial state variable setup 
... which might otherwise be skipped if the PLAY command is issued before
the OPEN command had a chance to actually be acted upon.

Fixes #657376.
 2011-09-09 10:53:08 +02:00
Wim Taymans
33f18b8ea4 Merge branch 'master' into 0.11 
Conflicts:
	gst/audioparsers/gstamrparse.c
	gst/isomp4/qtdemux.c
 2011-09-06 16:06:25 +02:00
Mark Nauwelaerts
2603c2079d rtspsrc: add gtk-doc for new short-header property 2011-09-05 13:32:17 +02:00
Marc Leeman
ce276d903c rtspsrc: allow sending short RTSP requests to a server 
Some encoders (Arecont) do not like the long OPTIONS sent at startup as sent by
GStreamer, but do accept the short header as sent by Live555.

This patch makes the extending the request optional by adding a property
(short-header).

Fixes #655805.

API: GstRTSPSrc:short-header
 2011-09-05 13:26:06 +02:00
Wim Taymans
4bb2b140e9 Merge branch 'master' into 0.11 
Conflicts:
	sys/v4l2/v4l2src_calls.c
 2011-08-16 18:35:53 +02:00
Edward Hervey
d08e0ccc48 rtspsrc: Properly error out if SDP contains no streams 
Also fixes unitialized variable error on macosx.
 2011-08-09 11:28:17 +02:00
Wim Taymans
4121021bb2 Merge branch 'master' into 0.11 
Conflicts:
	ext/pulse/pulsesink.c
	ext/pulse/pulsesrc.c
	gst/audioparsers/gstac3parse.c
	gst/rtp/gstrtph264depay.c
	gst/rtp/gstrtph264pay.c
	gst/rtpmanager/gstrtpssrcdemux.c
 2011-08-03 18:25:30 +02:00
Mark Nauwelaerts
9764b57b0a rtspsrc: set SOURCE flag at init time 
Fixes #654816.
 2011-07-25 12:44:38 +02:00
Wim Taymans
9c087d7d85 Merge branch 'master' into 0.11 2011-07-15 17:06:39 +02:00
Mark Nauwelaerts
b98585df82 rtspsrc: fix seeking regression 
... introduced when shuffling around code for the async implementation
by setting state of source (and udp sources) in _play before downstream
flushing is undone.
 2011-07-12 15:13:25 +02:00
Wim Taymans
f0749ed617 rtsp: fix for uri changes 2011-06-22 16:41:13 +02:00
Wim Taymans
e221908169 rtsp: fix for flush_stop API change 2011-06-13 17:14:51 +02:00
Wim Taymans
eed80e2dd3 -good: update for buffer API change 2011-06-13 16:33:57 +02:00
Wim Taymans
c731cd3d95 rtsp: port to 0.11 2011-06-09 17:52:34 +02:00
Wim Taymans
710fa239d5 Merge branch 'master' into 0.11 2011-06-08 18:06:56 +02:00
Mark Nauwelaerts
785247cfb3 rtspsrc: reset state tracking variable when appropriate 
... so we don't end up interrupting an operation that should not be interrupted
based on the indication of a previous interruptable operation.
 2011-06-06 12:59:23 +02:00
Wim Taymans
0b1bdcf7cb Merge branch 'master' into 0.11 
Conflicts:
	sys/ximage/ximageutil.c
 2011-06-02 18:51:29 +02:00
Miguel Angel Cabrera Moya
c39b7a5359 rtspsrc: uniform unknown message handling 
Do the same processing in all the cases when an unknown message is received.
That is, give a warning.

https://bugzilla.gnome.org/show_bug.cgi?id=651059
 2011-05-25 20:06:16 +02:00
Wim Taymans
d89790d545 Merge branch 'master' into 0.11 
Conflicts:
	gst/avi/gstavidemux.c
	gst/rtp/gstrtpac3depay.c
	gst/rtp/gstrtpg726depay.c
	gst/rtp/gstrtpmpvdepay.c
	gst/videofilter/gstgamma.c
 2011-05-24 17:34:19 +02:00
Stefan Kost
be413185d0 rtspsrc: use EINVAL for missing url parameter 
Fixes gcc warning about using uninitialized variable 'res'.
 2011-05-18 10:22:27 +03:00
Wim Taymans
e15651816e Merge branch 'master' into 0.11 2011-05-17 16:13:59 +02:00
Mark Nauwelaerts
dc2ddea91b rtspsrc: also allow PAUSE to be interrupted 
... as it is on the way out to NULL.

See #632504.
 2011-05-17 11:56:47 +02:00
Mark Nauwelaerts
283e4e4afd rtspsrc: ensure proper closing and cleanup 
... since the TEARDOWN sequence might not have had a chance to even start,
but at least connections should be closed (synchronously) and state cleaned up.

See #632504.
 2011-05-17 11:56:38 +02:00
Mark Nauwelaerts
f7ddf811d7 rtspsrc: fix and improve async handling 
Simplify the command handling; passing a command to thread means we really
want it to get the message, which means to always flush provided the command
can handle being interrupted.  Command thread indicates whether command
allows interruption and ensure non-flushing connection as it subsequently
needs it.

In particular, this also makes the TEARDOWN sequence interruptable
and also prevents races where _loop_ could miss a command and would
continue receiving (or at least trying to).

See #632504.
 2011-05-17 11:56:22 +02:00
Mark Nauwelaerts
e6798ad54c rtspsrc: tweak post-seek loop handling 2011-05-17 11:55:40 +02:00
Wim Taymans
ddfcd8bbfd rtspsrc: open on play and pause when not done yet 
With the async state changes, it is possible that we need to open the stream
before play and pause.
Also make sure we remember a previous open failure so that we don't keep trying
again.
 2011-05-17 11:55:34 +02:00
Wim Taymans
6fe680934a rtspsrc: improve async handling 
Simplify the command handling, only continue looping when we have not received
another command or when the previous loop was successfull.
Avoid looping on a disconnected socket.
 2011-05-17 11:55:32 +02:00
Wim Taymans
2513207433 rtspsrc: rework reconnect code 
Use the same async code path to implement reconnects.
Make sure we only post progress messages when doing async things.
 2011-05-17 11:55:29 +02:00
Wim Taymans
c27c10f8f4 rtspsrc: small cleanups 
Make sure we cancel the previous task when queuing a new one.
Move the messages to a central place so we can more easily post them.
 2011-05-17 11:55:27 +02:00
Wim Taymans
852c6e11cd rtspsrc: don't post errors when interrupting 2011-05-17 11:55:24 +02:00
Wim Taymans
220e47adcf rtspsrc: implement more async handling 
Remove some old locks.
Make sure we never go into the loop function when flushing.
 2011-05-17 11:55:20 +02:00
Wim Taymans
2873585238 rtspsrc: first attempt at async implementation 2011-05-17 11:55:18 +02:00
Wim Taymans
dae679e560 rtspsrc: small header cleanups 2011-05-17 11:55:15 +02:00
Wim Taymans
77acc618e1 use G_DEFINE_TYPE some more 2011-04-19 17:35:47 +02:00
Wim Taymans
7555d0949f Merge branch 'master' into 0.11 
Conflicts:
	android/apetag.mk
	android/avi.mk
	android/flv.mk
	android/icydemux.mk
	android/id3demux.mk
	android/qtdemux.mk
	android/rtp.mk
	android/rtpmanager.mk
	android/rtsp.mk
	android/soup.mk
	android/udp.mk
	android/wavenc.mk
	android/wavparse.mk
	configure.ac
 2011-04-18 10:23:45 +02:00
Thibault Saunier
b541208b77 android: Make it ready for androgenizer 
Remove the android/ top dir
Fixe the Makefile.am to be androgenized

To build gstreamer for android we are now using androgenizer which generates the needed Android.mk files.
Androgenizer can be found here: http://git.collabora.co.uk/?p=user/derek/androgenizer.git
 2011-04-11 01:20:11 +02:00
Wim Taymans
4e7f1633e4 rtpdec: reset structure before use 2011-04-05 17:26:44 +02:00
Wim Taymans
c124ba1489 Merge branch 'master' into 0.11 
Conflicts:
	gst/rtsp/gstrtspsrc.c
 2011-04-05 17:20:08 +02:00
Wim Taymans
547c97f590 rtspsrc: handle * control correctly 
Parse session control attributes when no media control attribute is
present. Threat * control attributes as an empty string, just like the
spec says.

Fixes #646800
 2011-04-05 17:12:28 +02:00
Wim Taymans
f67c95d826 rtsp/udp: port to 0.11 2011-04-05 17:06:41 +02:00