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Author SHA1 Message Date
Wim Taymans f8013487c9 rtspsrc: add support for NetClientClock 
When the server suggests a GstNetTimeProvider in the SDP, set up a
GstNetClientClock that slaves to the remote clock and suggest this clock in
provide_clock.
 2013-04-11 15:00:05 +01:00
Sebastian Dröge d80ff8e7f3 rtspsrc: Proxy the multicast-iface property of udpsrc 2013-04-03 17:53:13 +02:00
Wim Taymans 640de61740 rtspsrc: only EOS when our source sends BYE 
Only EOS when we receive a BYE event from the SSRC of our stream.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=675453
 2013-02-06 14:01:16 +01:00
Wim Taymans 0540492ab2 rtspsrc: save the stream SSRC 
Conflicts:
	gst/rtsp/gstrtspsrc.c
 2013-02-06 14:00:56 +01:00
Wim Taymans c8fb1c720c rtspsrc: flush connection when stopping 
When we stop, we can flush all pending commands so that we can stop and
join the task.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=684924
 2013-02-06 13:18:18 +01:00
Tim-Philipp Müller 95a37196b3 rtspsrc: add "proxy-id" and "proxy-pw" properties 
to match souphttpsrc. user/password passed via the URI
will still take precedence though.

https://bugzilla.gnome.org/show_bug.cgi?id=395427
 2012-12-31 00:22:27 +00:00
Wim Taymans 8cfec6a88d rtspsrc: fix cmd comparison 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=690476
 2012-12-20 17:12:30 +01:00
Wim Taymans 75616fac9a rtspsrc: add some more debug 2012-12-20 17:12:20 +01:00
Wim Taymans a858bf46db rtspsrc: fix TCP reconnect 
Ignore other commands when reconnecting, otherwise the loop function would pause
and the reconnection would not happen. Continue looping after doing a reconnect
so that we have a chance to actually read the new data.
 2012-12-13 09:30:59 +01:00
Wim Taymans b1dc816772 rtspsrc: timeout on udpsrc is in nanoseconds 2012-12-12 11:09:42 +01:00
Aleix Conchillo Flaque 3503aef946 rtspsrc: do not change state to PLAYING if currently chaning state 
* gst/rtsp/gstrtspsrc.c (gst_rtspsrc_play): state change might be
  happening in the application thread, so we don't change the state to
  PLAYING in the gstrtspsrc thread unless it is safe.

  A specific case is when chaning the state to NULL from the application
  thread. This will synchronously try to stop the task (with the element
  state lock acquired), but we will try a gst_element_set_state from
  gstrtspsrc thread which will block on the element state lock causing a
  deadlock.

  https://bugzilla.gnome.org/show_bug.cgi?id=684312
 2012-12-10 15:13:22 +01:00
Wim Taymans 64cdbb77a9 rtspsrc: use new option parser function 2012-11-27 11:13:37 +01:00
Wim Taymans 5d0507c09e rtspsrc: pause the task instead of spinning 
Actually pause the loop task instead of spinning forever.
 2012-11-22 11:34:31 +01:00
Wim Taymans c28bfa8902 rtspsrc: handle segment event 
Make a segment event when we send a new range header to a client (first PLAY
request or after a seek). Send the segment event in interleaved mode.
Clean the segment event on cleanup

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688382
 2012-11-16 15:38:29 +01:00
Wim Taymans bd91bd3193 rtspsrc: fix check for active streams 
A stream can be active without a srcpad yet and we want to send
events on those streams as well.
 2012-11-16 15:22:46 +01:00
Wim Taymans 11cf4d4fd3 rtspsrc: create and add pads outside of lock 
Create and add the ghostpad for the new stream outside of the lock because it
is not needed and causes deadlocks.
 2012-11-16 13:33:44 +01:00
Aleix Conchillo Flaque 6c855edf03 rtspsrc: allow client to disable reconnection 
* gst/rtsp/gstrtspsrc.[ch]: added new "udp-reconnect" property. Before,
  rtspsrc always tried to reconnect to the server when the RTSP
  connection was closed by the server. This property lets the user
  decide whether it wants rtspsrc to reconnect or not.

  https://bugzilla.gnome.org/show_bug.cgi?id=683912
 2012-11-16 12:55:10 +01:00
Wim Taymans e2a4d28c1f rtspsrc: clear variables before retrying 
Else we might unref an old udpsrc twice in cleanup.
 2012-11-16 12:17:37 +01:00
Wim Taymans cc9cb26be1 rtspsrc: propose ports in multicast 
When the user configured a port-range, propose ports from this range
as the multicast ports. The server is free to ignore this request but if it
honours it, increment our ports so that we suggest the next port pair for the
next stream.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=639420
 2012-11-16 12:17:37 +01:00
Wim Taymans 5025b3f1b3 rtspsrc: add more debug 2012-11-16 12:17:37 +01:00
Marc Leeman 7cbca3dcd1 rtsp: the RTCP port number is inclusive 
The configured port number pair has its upper bound set to the maximum
allowed RTCP port, inclusive.

See https://bugzilla.gnome.org/show_bug.cgi?id=639420
 2012-11-06 13:22:58 +01:00
Tim-Philipp Müller 230cf41cc9 Fix FSF address 
https://bugzilla.gnome.org/show_bug.cgi?id=687520
 2012-11-04 00:07:18 +00:00
Wim Taymans adb70e89f9 rtspsrc: remove unused include 2012-10-10 12:05:34 +02:00
Tim-Philipp Müller 8b20603f8b rtspsrc: answer URI query 
Without this, something also answered the query
with TRUE but without setting a uri, not sure
what that was..
 2012-09-21 23:33:47 +01:00
Daniela 03fbd7ec6e rtspsrc: avoid leak 
When setup fails, make sure to cleanup afterwards.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=673509
 2012-09-07 16:33:18 +02:00
Aleix Conchillo Flaque 4a200b670f rtp: make rtp packet probation configurable (bug #682512) 2012-08-30 21:49:57 +02:00
Tim-Philipp Müller 4bb52bbadf docs: gst-launch -> gst-launch-1.0 and ffmpegcolorspace -> videoconvert 2012-08-27 21:20:30 +01:00
Aleix Conchillo Flaque 8d864dbbfc rtspsrc: make jitterbuffer drop-on-latency available (fix #682055) 
Conflicts:

	gst/rtsp/gstrtspsrc.h
 2012-08-22 10:39:19 +02:00
Mark Nauwelaerts a549b0bf2c rtspsrc: manage race between connection closing and flushing 
... where the former can happen in task thread and the latter in mainloop
upon downward state change.
 2012-08-03 14:10:32 +02:00
Wim Taymans ef38efc2d7 rtsp: go and stay in the loop function on PLAY 
When we have a PLAY request, go into the LOOP function next. When we are
looping, keep on looping until we are told otherwise.
This fixed rtsp and TCP connections.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680551
 2012-07-25 12:50:01 +02:00
Wim Taymans 943b56ff8e rtsp: set caps after activating the pad 2012-07-25 12:49:35 +02:00
Maria Giovanna Chiossa 561b131e1a rtspsrc: also set UDP buffer size in multicast 
Also set the UDP buffer size in multicast mode.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=675448
 2012-07-19 15:26:36 +02:00
Wim Taymans 51371d26ee update for RTP buffer api changes 2012-07-17 16:38:27 +02:00
Sebastian Dröge aeafc3a093 gst: Implement segment-done event 2012-07-05 13:13:09 +02:00
Tim-Philipp Müller 456847c66b rtspsrc: update for gst_element_make_from_uri() changes 2012-06-23 14:57:28 +01:00
Wim Taymans 30d3dfee36 update for task api change 2012-06-20 10:33:42 +02:00
Wim Taymans 694be55c05 rtspsrc: Don't reset time in flush-stop 
Don't reset the time in flush-stop. Live sources can do this flush in the
playing state and so the pipeline will never have a chance to update the
base_time of the elements, which only happens when going from paused to
playing.
 2012-06-14 08:58:58 +02:00
Wim Taymans 935472aba7 rtspsrc: Rework the async state handling 
Always send the flushing events to the udp elements now that basesrc supports
this. This makes sure a segment event is sent correctly after a flush.
Keep track of the currently executing command and make it possible to specify
what command you want to cancel when starting a new async command.

See https://bugzilla.gnome.org/show_bug.cgi?id=677905
 2012-06-12 16:05:40 +02:00
Sebastian Dröge a1948e34d2 elements: Use gst_pad_set_caps() instead of manual event fiddling 2012-06-08 15:54:42 +02:00
Wim Taymans eb982e4bbe rtspsrc: only reset the manager object when we did a seek 
Only reset the manager object when we used a Range header, ie. when we did a
seek. Otherwise we just paused and we can resume just fine.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677475
 2012-06-07 12:11:14 +02:00
Maria Giovanna Chiossa ff019d05f6 rtsp: add the Scale header when needed 
Setting GST_SEEK_FLAG_SKIP when sending a seek event in rtspsrc should
set the "Scale" field in the rtsp PLAY header.
Because the boolean "src->skip" is set after the call, "Speed" instead
of "Scale" is always set. Move the assignment before issuing the _play
request.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676618
 2012-05-24 09:57:31 +02:00
Sebastian Dröge d99eb6d2cb Update everything for the removal of the interface library and mixer/tuner interfaces 2012-04-13 13:15:11 +02:00
Tim-Philipp Müller e09ae5736d Use new gst_element_class_set_static_metadata() 2012-04-10 00:51:41 +01:00
Sebastian Dröge aa2cd462da gst: Update for GST_PLUGIN_DEFINE() API changes 2012-04-05 17:36:38 +02:00
Sebastian Dröge 5cdd49bf25 gst: Update versioning 2012-04-04 14:37:47 +02:00
Wim Taymans 3d61d12e03 update for buffer api change 2012-03-30 18:15:34 +02:00
Wim Taymans c44cd8f55b Merge branch 'master' into 0.11 
unport gdkpixbuf
not merged: https://bugzilla.gnome.org/show_bug.cgi?id=654850

Conflicts:
	docs/plugins/Makefile.am
	docs/plugins/gst-plugins-good-plugins-docs.sgml
	docs/plugins/gst-plugins-good-plugins-sections.txt
	docs/plugins/gst-plugins-good-plugins.hierarchy
	docs/plugins/inspect/plugin-avi.xml
	docs/plugins/inspect/plugin-png.xml
	ext/flac/gstflacdec.c
	ext/flac/gstflacdec.h
	ext/libpng/gstpngdec.c
	ext/libpng/gstpngenc.c
	ext/speex/gstspeexdec.c
	gst/audioparsers/gstflacparse.c
	gst/flv/gstflvmux.c
	gst/rtp/gstrtpdvdepay.c
	gst/rtp/gstrtph264depay.c
 2012-03-22 11:53:24 +01:00
Marc Leeman b4756db358 gstrtspsrc: disable RTSP keep-alive on request 2012-03-12 15:14:21 +01:00
Sebastian Dröge f2e569cde8 rtspsrc: Use correct enum for return values 2012-03-06 14:18:33 +01:00
Wim Taymans ca9532ccc5 update for new memory api 2012-02-22 02:10:33 +01:00
Wim Taymans 9365f12d6e GST_FLOW_WRONG_STATE -> GST_FLOW_FLUSHING 2012-02-08 16:43:30 +01:00
Sebastian Dröge 0b517ce9fb Merge branch '0.11' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-good into 0.11 2012-01-25 12:49:34 +01:00
Sebastian Dröge 10554b271f Merge branch 'master' into 0.11 
Conflicts:
	ext/flac/gstflacdec.c
	ext/jpeg/gstjpegenc.c
	ext/pulse/pulsesink.c
	sys/v4l2/gstv4l2src.c
 2012-01-25 12:49:11 +01:00
Wim Taymans b4630dd3e0 more memory API porting 2012-01-25 12:30:29 +01:00
Mark Nauwelaerts a224ffb971 rtspsrc: simplify internal src event debug logging 
... which avoids almost superfluous obtaining of rtsp element.
 2012-01-20 17:10:57 +01:00
Mark Nauwelaerts 018852ddc2 rtspsrc: avoid NULL string comparison 2012-01-20 17:10:54 +01:00
Wim Taymans 1584806634 port to new gthread API 2012-01-19 11:33:53 +01:00
Sebastian Dröge 305901c7cc rtspsrc: Update for the new GIO versions of the udp elements 2012-01-17 16:49:10 +01:00
Sebastian Dröge 93e3ed5a86 Merge branch 'master' into 0.11 
Conflicts:
	ext/cairo/gsttextoverlay.c
	ext/pulse/pulseaudiosink.c
	gst/audioparsers/gstaacparse.c
	gst/avi/gstavimux.c
	gst/flv/gstflvmux.c
	gst/interleave/interleave.c
	gst/isomp4/gstqtmux.c
	gst/matroska/matroska-demux.c
	gst/matroska/matroska-mux.c
	gst/matroska/matroska-mux.h
	gst/matroska/matroska-read-common.c
	gst/multifile/gstmultifilesink.c
	gst/multipart/multipartmux.c
	gst/shapewipe/gstshapewipe.c
	gst/smpte/gstsmpte.c
	gst/udp/gstmultiudpsink.c
	gst/videobox/gstvideobox.c
	gst/videocrop/gstaspectratiocrop.c
	gst/videomixer/videomixer.c
	gst/videomixer/videomixer2.c
	gst/wavparse/gstwavparse.c
	po/ja.po
	po/lv.po
	po/sr.po
	tests/check/Makefile.am
	tests/check/elements/qtmux.c
	tests/check/elements/rgvolume.c
 2012-01-10 14:32:32 +01:00
Wim Taymans 5fd2b7abe3 GST_FLOW_UNEXPECTED -> GST_FLOW_EOS 2012-01-03 15:26:21 +01:00
Tim-Philipp Müller 27ee8931dd autodetect, rtsp: gst_registry_get_default() -> gst_registry_get() 2012-01-02 14:32:40 +00:00
Tim-Philipp Müller b8b8454bcb Suppress deprecation warnings in selected files, for g_static_rec_mutex_* mostly 
GStaticRecMutex is part of our API/ABI, not much we can do here
in 0.10 for most of these.
 2011-12-12 09:46:27 +00:00
Wim Taymans d0b936acc7 rtspsrc: remove unused flush param 2011-12-06 13:59:52 +01:00
Wim Taymans 71b615515a update for clock provider API change 2011-11-28 17:52:06 +01:00
Wim Taymans ac849ec2b3 fix for element flag updates 2011-11-28 16:57:24 +01:00
Vincent Penquerc'h c0e101e93f various: fix pad template leaks 
https://bugzilla.gnome.org/show_bug.cgi?id=662664
 2011-11-28 13:30:27 +00:00
Tim-Philipp Müller 87aa29d2cf rtspsrc: make connection-speed property a guint64 2011-11-24 01:19:32 +00:00
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
Wim Taymans e6ec5cce2e rtspsrc: respect aggregate control attributes 
when the SDP specifies an aggregate control url, use that for playback
control.

Fixes #619531
 2010-06-14 19:24:14 +02:00
Wim Taymans cb8252275d rtsp: try all ranges from the sdp 
Try all ranges in the SDP before giving up.
 2010-06-04 13:58:38 +02:00
Wim Taymans 6fbca707bb rtspsrc: make parse_range return result 
Make the parse_range function return if the parsing succeeded or failed.
 2010-06-04 13:58:38 +02:00
Wim Taymans a50cd7c27d rtspsrc: don't leak the session 2010-05-07 19:02:21 +02:00
Wim Taymans bc72d8250c rtsp: configure bandwidth properties in the session 2010-05-07 18:59:42 +02:00
Wim Taymans db3c4e7f46 rtspsrc: fall back to SDP ports instead of server_port 
In multicast, fall back to the ports in the SDP instead of the server_port
attribute as this is more in line with the RFC.
 2010-05-07 12:51:05 +02:00
Wim Taymans 4e1ced0a77 rtspsrc: refactor collecting the transport info 
Make a method to collect the ports and destination address.
 2010-05-07 12:24:51 +02:00
Wim Taymans 05352d7ea8 rtspsrc: handle servers that send broken Transports 
Handle servers that send their port pairs with the wrong name.

Fixes #617537
 2010-05-07 11:28:36 +02:00
Wim Taymans ef4d2901aa rtspsrc: use the SDP connection info in multicast 
Parse the connection info from the SDP.
When we need to configure the multicast destination, fall back to the SDP
connection info when the transport did not specify a destination and ttl.

Fixes #617537
 2010-05-06 16:52:26 +02:00
Wim Taymans d6579912cb rtspsrc: make setup url in a smarter way 
Make sure we always separate the base and control url parts with a / when
creating the setup url.
 2010-05-04 16:36:15 +02:00
Alessandro Decina c8a02a91a6 rtspsrc: handle SEEKING queries. 2010-05-04 16:05:13 +02:00
Stefan Kost 0e048803b9 rtsp: remove obsolete google extension 
This was not build for a while and can be removed.
 2010-04-08 17:50:49 +03:00
Wim Taymans b84bf10455 rtspsrc: add property to control the buffering method 
Add a property to control how the jitterbuffer performs timestamping and
buffering.
 2010-04-05 15:26:03 +02:00
Benjamin Otte 3f511ec361 Add -Wwrite-strings to the configure flags 
... and fix all warnings
 2010-03-21 14:17:47 +01:00
Wim Taymans ef804589ca rtsp: use GType from -base and bump required version 
Use the transport flags GType from -base and bump the required version of -base
because of this.
 2010-03-19 15:03:43 +01:00
Benjamin Otte cccfeaa59c gst_element_class_set_details => gst_element_class_set_details_simple 2010-03-18 14:32:00 +01:00
Benjamin Otte 1055aaa9cb Add -Wredundant-decls warning flag 
Also fix compile issues
 2010-03-17 19:35:10 +01:00
Benjamin Otte 3342b1679e Add -Wmissing-declarations -Wmissing-prototypes warning flags 
And fix all the warnings.
 2010-03-17 18:23:28 +01:00
Wim Taymans ba6dbaecfc rtspsrc: don't forget to send keepalive messages 
When we operate in TCP mode, still send keepalive messages when we
need to.

Fixes #612696
 2010-03-15 11:38:23 +01:00
Wim Taymans d29fa60f97 rtspsrc: check for NULL before doing strcmp 
Check the connection and address type for NULL before doing strcmp and
crashing.

Fixes #612553
 2010-03-11 12:56:11 +01:00
Wim Taymans 821096c4f1 rtspsrc: parse connection information 
Parse the connection information from the SDP and use it to figure out if we are
dealing with ipv4 or ipv6 connections.
 2010-03-10 11:28:22 +01:00
Wim Taymans 8eb5c2c794 rtspsrc: require a destination for multicast 
When setting up the multicast sockets, we need a destination address to listen
on or else we error.
 2010-03-10 11:21:20 +01:00
Wim Taymans 574447b092 rtspsrc: handle ipv6 listening ports when needed 
Add some code to make udpsrc listen on an ipv6 address when needed. The
detection of IPV6 is not yet implemented.
 2010-03-10 11:21:20 +01:00
Wim Taymans 38f2b4735d rtspsrc: send keep alive when paused 
When we are paused, send keep alive messages to the server so that our session
doesn't time out when we go back to playing later.
 2010-03-10 11:21:18 +01:00
Wim Taymans 66709a7a68 rtspsrc: configure multicast correctly 
Take the transport destination for multicast.
Disable loop and autojoin for multicast on the udpsinks.
 2010-03-08 17:48:46 +01:00
Wim Taymans a0b651bf5b rtspsrc: avoid stopping NULL tasks 
Check the task for NULL, it could be paused and set to NULL before.
 2010-02-16 19:54:32 +01:00
Mark Nauwelaerts 87e80aab57 rtspsrc: fix typo in debug message 2010-02-16 16:07:21 +01:00
Wim Taymans 8d814f3782 rtpbin: pass running_time to jitterbuffer pause 
Pass the current running time to the jitterbuffer when pausing or resuming so
that it calculate the right offsets.
Small cleanups and comments.
Set the default rtspsrc latency to 2 seconds.
 2010-02-12 17:22:54 +01:00
Wim Taymans c2dfc94b1d rtspsrc: cleanup properties 
Use more default constants.
Use static strings param flag.
Init properties explicitly instead of letting gobject do this.
 2010-02-12 15:20:07 +01:00
Wim Taymans c35a984801 rtspsrc: free transports on errors 
See #608564
 2010-02-01 19:32:11 +01:00
Wim Taymans 8c5a822250 rtspsrc: fix on-npt-stop signal warnings for RDT 
The RDT manager does not implement this signal so we need to check for it before
trying to connect to it.
 2010-01-05 12:23:16 +01:00
Wim Taymans a65240d1c1 rtspsrc: fix some comments, remove property check 
Fix some comments, clarify some FIXMEs
Remove the on-ntp-stop signal check now that the jitterbuffer is in
-good and we know that it supports this signal.
 2009-12-24 22:23:01 +01:00
Thiago Santos ac03ad782a rtspsrc: Parse all rtpinfo entries 
Do not forget to parse all rtp-info entries, instead of
parsing the first one only.

Fixes #605222
 2009-12-24 17:08:22 -03:00