mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-08-01 11:15:10 +00:00
Code Issues 1 Projects Releases Wiki Activity
13005 commits 44 branches 1506 tags 390 MiB
Commit graph

469 commits

Author SHA1 Message Date
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