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