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Author SHA1 Message Date
Wim Taymans 105650127e add parent to pad functions 2011-11-17 15:02:55 +01:00
Wim Taymans 6190312214 add parent to query function 2011-11-16 17:27:13 +01:00
Tim-Philipp Müller c27bbe4be2 Update for GstURIHandler get_protocols() changes 2011-11-13 23:44:44 +00:00
Tim-Philipp Müller a150d1e734 soup, pushfile, rtsp, udp, v4l2: update for GstURIHandler API changes 2011-11-13 18:50:51 +00:00
Wim Taymans c48df77320 update for probe api changes 2011-11-08 11:18:06 +01:00
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
Mark Nauwelaerts 234609844e rtspsrc: perform post-flush state tricks downstream to upstream 
... so downstream is set when upstream resumes data flow.
 2011-04-04 11:49:00 +02:00
Mark Nauwelaerts 226a7cb32e rtspsrc: distribute new base_time to manager children following flush seek 
... by forcing a state changed to PLAYING, which should otherwise be a
no-op as elements should already be in that state.

In particular, jitterbuffer needs new base_time as soon as possible to perform
proper timing (e.g. eos timeout handling) and can't wait for the new base_time
that will be distributed when the whole pipeline returns to PLAYING.

See bug #646397.
 2011-04-04 11:49:00 +02:00
Wim Taymans 8f22a09dc4 Merge branch 'master' into 0.11-fdo 2011-03-28 20:50:59 +02:00
Mark Nauwelaerts 2738917852 rtspsrc: improve recovery from failed seek 
In case server-side fails to perform seek, i.e. PLAY at non-zero requested
position, recovery so far would arrange for streaming to continue, albeit
having lost position tracking in the process.  So, query position prior
to seek and use upon failed seek.
 2011-03-09 17:18:09 +01:00
Wim Taymans 759a3507d7 Merge branch 'master' into 0.11 
Conflicts:
	configure.ac
 2011-02-28 11:58:05 +01:00
Miguel Angel Cabrera Moya 3cca27ced1 rtspsrc: fix minor leaks when handling server requests. 
https://bugzilla.gnome.org/show_bug.cgi?id=640163
 2011-02-14 11:33:18 +01:00
Stefan Kost 6f6b2a7efc rtspsrc: strip trailing spaces 2011-02-07 17:08:47 +02:00
Stefan Kost 5e071d51f2 rtpsrc: set multiple properties in one go 
There is no need for separate g_object_set() calls here.
 2011-02-07 17:07:42 +02:00
Tim-Philipp Müller 08855b45b6 rtspsrc: don't leak url string 
https://bugzilla.gnome.org/show_bug.cgi?id=640064
 2011-01-20 13:46:44 +00:00
Wim Taymans bc0824181b rtspsrc: don't confuse return values 
Return a return value of the right type.
 2011-01-05 18:33:41 +01:00
Stefan Kost c9e0db6469 rtspsrc: remove unused variables when debug-logging disabled 2011-01-03 20:17:47 +02:00
Wim Taymans dc221c0219 rtspsrc: increase udp buffer size 
Set a bigger UDP buffer size by default to reduce packet loss with
high bitrate streams.
 2011-01-03 15:40:11 +01:00
Tim-Philipp Müller 96830324a5 rtspsrc: serialise/deserialise floats without changing locale 
Use g_ascii_dtostr() and g_ascii_strtod() to serialise/deserialise
floating point numbers, instead of ugly hacks that switch locale
before and after calling libc functions (which is not a good idea
in a multi-threaded application).
 2010-12-29 15:54:46 +00:00
Wim Taymans 2a49d34c3e rtspsrc: on-npt-stop is a manager signal 2010-12-23 16:25:15 +01:00
Wim Taymans 12bc7258b9 rtspsrc: improve RTP session handling 
Store the RTP session in the stream so that we can more efficiently
perform actions on the stream based on RTP signals.
 2010-12-23 15:24:29 +01:00
Tim-Philipp Müller 7759ad0db2 docs: update rtspsrc docs, rtpbin is not in -bad any more 2010-12-22 13:04:42 +00:00
Mark Nauwelaerts 287894a89a rtspsrc: mark DISCONT when resuming PLAY 
In particular, when streaming interleaved, this arranges for setting a new
timestamp on outgoing buffer so downstream can appropriate reset
to a change in (rtp)time.
 2010-12-10 12:11:15 +01:00
Mark Nauwelaerts c25625c31c rtspsrc: degrade gracefully upon failing seek and tweak QUERY_SEEKING response 2010-12-10 12:09:49 +01:00
Mark Nauwelaerts 52b5929a2b rtspsrc: add and use auto buffering mode 
... which selects BUFFER for a non-live stream, and otherwise SLAVE.

Fixes #633088.
 2010-12-10 12:09:32 +01:00
Wim Taymans 1d57ec6a6e rtspsrc: use _object_ref_sink() when we can 2010-12-07 11:42:15 +01:00
Mark Nauwelaerts 0f2373cbd1 rtspsrc: reset session manager base time when flushing 
... as rtpbin uses running time to handle rtpjitterbuffer's buffer mode pauses.
 2010-12-03 15:50:17 +01:00
Mark Nauwelaerts 148af2235e rtspsrc: include range request for all streams with non-aggregate control 2010-12-03 15:50:17 +01:00
Mark Nauwelaerts dedf145316 rtspsrc: fix debug statement 2010-12-03 15:50:17 +01:00
Wim Taymans 7ed250c793 rtspsrc: select multicast transports in a smarter way 
When we see a multicast address in the SDP connection, only try to negotiate a
multicast transport with the server.

Fixes #634093
 2010-12-02 19:16:47 +01:00
Mark Nauwelaerts b6b0de0c49 rtspsrc: handle stale digest authentication session data 
In particular, handle Unauthorized server response when trying to convey
keep-alive.

Fixes #635532.
 2010-11-29 17:34:28 +00:00
Mark Nauwelaerts ca7870de49 rtspsrc: fix duration reporting 
Init segment prior to storing duration info in it.

Fixes #632548.
 2010-10-19 16:47:20 +02:00
Stefan Kost d8167e3071 various (gst): add a missing G_PARAM_STATIC_STRINGS flags 2010-10-13 18:00:28 +03:00
Wim Taymans ee7207aa3e rtspsrc: mark as a source 
Mark the rtspsrc element as a source.
Requires 0.10.31.1 now
 2010-10-11 15:12:51 +02:00
René Stadler 0cfe24d132 rtspsrc: fix missing null-terminator in protocols array 
Fixes random crash regression from commit ae84ae.
 2010-09-28 16:21:48 +03:00
Wim Taymans ef29a59903 rtspsrc: don't add /UDP in the transport, it's the default 
don't add the default UDP lower-transport, some servers don't seem to like it.

Fixes #630500
 2010-09-24 16:26:20 +02:00
Wim Taymans 8f2d254e24 rtspsrc: don't clear sdp when set as uri 
when we set the SDP with an uri, don't clear it when we go to READY.
 2010-09-10 18:06:48 +02:00
Wim Taymans 7698d8bc4a rtspsrc: use sdp uri parse method 
Use the sdp parse method that does proper uri escaping.
 2010-09-10 18:02:04 +02:00
Wim Taymans ae84ae1b36 rtspsrc: add rtsp-sdp protocol support 
Allow setting an SDP with the rtsp-sdp:// url.

Based on patch from Marco Ballesio.

See #628214
 2010-09-10 12:14:21 +02:00
American Dynamics 5999e8e716 rtspsrc: Add property to configure udpsrc buffer size 
Add a new udp-buffer-size property to configure the buffer-size on the udpsrc
elements.

Fixes #628058
 2010-09-06 12:22:11 +02:00
Wim Taymans 3bae70ceea rtspext: stop configuration on first failure 
Stop the configuration of a stream as soon as some of the extensions return
FALSE.

Fixes #581294
 2010-09-06 11:01:57 +02:00
Wim Taymans e4f8144bbf rtspsrc: implement custom event handler 
Extend the _push_event() function so that it can also send events to the udp
sources when asked.
Implement a custum send_event function that correctly dispatches the downstream
events in TCP mode. This fixes sending EOS to rtspsrc and have it push the EOS
downstream.
 2010-09-06 10:45:23 +02:00
Sebastian Dröge d224251df4 rtspsrc: Don't use GST_FLOW_IS_FATAL() and GST_FLOW_IS_SUCCESS() 2010-09-04 14:52:10 +02:00
Wim Taymans 9dcfed0a5b rtspsrc: don't reuse udp sockets 
Don't reuse sockets but make the udpsrc element fail the state change when the
socket is already in use. If we don't prevent reuse, we might end up using the same
port for different streams in some cases.

Fixes #622017
 2010-08-04 10:40:23 +02:00
Wim Taymans e39d7f7359 rtspsrc: improve error and warning message 
Improve error and warning message.

Fixes #622577
 2010-08-04 10:39:44 +02:00
Arnaud Vrac c6f47c34fb rtspsrc: add port-range property to rtspsrc 
To support setups with firewall/ipsec, it is useful for an rtsp client to be
able to set the range of ports that can be used for rtp/rtcp reception.
Allows this by adding a "port-range" property to the rtspsrc element.

Fixes #625153
 2010-07-26 17:47:35 +02:00
Wim Taymans 8696d10a5b rtspsrc: fix memory leak in server request reply 
The RTSP server rtspsrc is communicating with, sends a GET_PARAMETER request
periodically as a ping.  The code in gst_rtspsrc_handle_request forms an OK
response and sends, but doesn't call gst_rtsp_message_unset to free the memory
after sending the response.  This results in a constant slow memory leak.

Fixes #624770
 2010-07-26 15:33:44 +02:00
Wim Taymans 5534c7d91d rtspsrc: fix locking after moving things around 2010-06-18 20:04:08 +02:00
Wim Taymans 651c82a01f rtspsrc: make some errors as warnings 
Avoid spamming the testsuite with these error debug lines.
 2010-06-18 16:56:19 +02:00
Wim Taymans 966ced2208 rtspsrc: factor out the connections 
Keep a global connection for aggregate control but also keep stream connections
for non-aggregate control.
Add some helper methods to connect/close/flush the connections.
 2010-06-18 15:13:06 +02:00
Wim Taymans ddc214d322 rtspsrc: add non-aggregate control 
Add non-aggregate control.
Separate retrieving thr SDP from parsing and setting up the streaming from the
SDP.
 2010-06-18 15:13:06 +02:00