David Schleef
17f3810f7b
video: remove // comments
2009-05-15 16:21:15 -07:00
David Schleef
45cf881f39
video: Add Y444, v210, v216 formats
2009-05-15 16:18:59 -07:00
David Schleef
4ec34e83d5
video: Copy BaseVideo classes from Schroedinger
2009-05-15 16:18:58 -07:00
Tim-Philipp Müller
f2031e1313
pbutils: add descriptions for 3GP, JPEG 2000 and Motion JPEG 2000
2009-05-15 20:50:06 +01:00
Wim Taymans
b9723f6e1c
audioclock: make our internal time monotonic
...
Make the internal time increase monotonically.
2009-05-13 21:38:56 +02:00
Sebastian Dröge
ab75db1653
propertyprobe: Fix typo in the docs
2009-05-12 15:53:07 +02:00
Wim Taymans
0a09632396
rtpdepay: add some more comments
2009-05-12 10:39:49 +02:00
Wim Taymans
d655120ee6
audioclock: make sure values are ever increasing
2009-05-12 10:39:41 +02:00
Sebastian Dröge
24dd91b1f0
interfaces: Seperate some more struct definitions from typedefs
2009-05-12 09:03:25 +02:00
Sebastian Dröge
e057414049
interfaces: Seperate some more struct definitions from typedefs
2009-05-12 09:03:25 +02:00
Sebastian Dröge
59aa1251d9
interfaces: API: Add gst_mixer_get_mixer_type()
...
This is a convenience function that returns the mixer_type
of the interface struct.
2009-05-12 09:03:24 +02:00
Sebastian Dröge
29b063b39b
interfaces: Add docs for gst_color_balance_get_balance_type()
2009-05-12 09:03:24 +02:00
Sebastian Dröge
9fc4d195e1
vorbistag: Use text/uri-list as mimetype instead of ---> for URI lists
2009-05-12 09:03:22 +02:00
John Millikin
ef473dd0ae
vorbistag: Store cover art in vorbiscomments
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Fixes bug #513373 .
2009-05-12 09:03:22 +02:00
Sebastian Dröge
e1875bf25f
interfaces: API: Add gst_color_balance_get_balance_type()
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This is a convenience function that returns the balance_type
of the interface struct.
2009-05-12 09:03:22 +02:00
Sebastian Dröge
b6c3567b41
interfaces: Separate struct definitions from typedefs
2009-05-12 09:03:22 +02:00
Tim-Philipp Müller
279b996d20
pbutils: add description for APE tag caps
2009-05-12 01:59:01 +01:00
Tim-Philipp Müller
3d33e2a873
tagdemux: cache events from upstream and re-send them once we have a source pad
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Makes sure tags don't get dropped when we have multiple tag demuxers in a row.
Fixes #580318 .
2009-05-12 01:15:21 +01:00
Michael Smith
8f6399f109
riff: support UYVY raw 4:2:2 in riff.
2009-05-11 14:04:16 -07:00
Andy Wingo
9f74ce745f
Revert "add can-activate-pull property to baseaudiosink"
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This reverts commit c4074a2ee4
.
2009-04-29 11:18:42 +02:00
Andy Wingo
219a31fa3c
Revert "[baseaudiosink] add docs for can-activate-pull"
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This reverts commit 416ce16f26
.
2009-04-29 11:18:33 +02:00
Andy Wingo
416ce16f26
[baseaudiosink] add docs for can-activate-pull
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* gst-libs/gst/audio/gstbaseaudiosink.c: Add documentation for
can-activate-pull.
2009-04-28 18:48:33 +02:00
Andy Wingo
c4074a2ee4
add can-activate-pull property to baseaudiosink
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* gst-libs/gst/audio/gstbaseaudiosink.c: Add can-activate-pull property
to baseaudiosink.
2009-04-28 18:28:50 +02:00
Tim-Philipp Müller
8efe6108c4
cddabasesrc: fix posting of discid tags after MERGE_MODE_REPLACE_ALL changes in core
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Don't use REPLACE_ALL merge mode when that's not really what we want,
as now that REPLACE_ALL actually does what it's supposed to do in
core, we drop tags we wanted to keep, such as the various disc id
tags. Add unit test for this as well. Fixes #579463 .
2009-04-19 18:15:28 +01:00
Tim-Philipp Müller
418760cf74
rtspconnection: don't use GLib-2.16 API, we require only 2.14
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Fixes #579267 .
2009-04-17 10:35:34 +01:00
Wim Taymans
32904de58f
baseaudiosink: don't unparent the ringbuffer
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when going to NULL, don't unparent the ringbuffer because we don't support going
back to 0 very well yet.
Fixes #579203
2009-04-17 11:03:32 +02:00
Olivier Crete
d927114ef8
RTCP: don't fail when retrieving invalid PT
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We can't meaningfully assert on valid packet types so just return the type as it
is. Update the comments to reflect this.
Fixes #579192 .
2009-04-17 10:53:10 +02:00
Wim Taymans
f83f57b648
app: add trivial cast macros
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Add trivial cast macros for appsrc and appsink. Mark them as being since 0.10.23
and add the macros to the standard macros in the docs.
Fixes #579130
2009-04-16 12:14:43 +02:00
Sebastian Dröge
a6cf0c8f06
video: Fix typo in the docs
2009-04-15 15:35:59 +02:00
Sebastian Dröge
a1d8cfde9d
video: Add support for YVYU YUV colorspace
2009-04-15 14:53:47 +02:00
Tim-Philipp Müller
75acca2835
docs: fix hyperlink and move fft attribution to the right place
2009-04-15 00:19:19 +01:00
Stefan Kost
ab24d9d65c
log: use G_GUINT64_FORMAT instead of llu
2009-04-15 00:02:39 +03:00
Josep Torra
71ab187355
RTSP: add missing headers for WMS RTSP
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Add missing headers related to Windows Media RTSP extension.
Fixes #578942
2009-04-14 18:31:52 +02:00
Tim-Philipp Müller
9f23b82b2c
Give credit to Mark Borgerding (kissfft author)
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and add myself to AUTHORS as well. Fixes #575638 .
2009-04-14 17:11:19 +01:00
Johann Prieur
86edcadc43
RTCP: add beginnings of Feedback messages
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Add the beginnings of parsing and constructing Feedback messages.
Fixes #577610 .
2009-04-14 16:45:20 +02:00
Wim Taymans
dffd1bcc97
baseaudiosrc: adjust the internal timestamp
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Adjust the internal timestamp before comparing it against the adjusted clock
time.
Fixes #578506
2009-04-14 13:16:14 +02:00
Wim Taymans
0c4c1410f9
baseaudiosink: use new clock time methods
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Use the unadjusted internal clock times to calculate the internal/external
offset when calibrating the clock.
When going to NULL, unparent and free the ringbuffer, like we do in the source
element.
See #578506
2009-04-14 13:12:59 +02:00
Wim Taymans
4231d54823
audioclock: add methods for the internal offset
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Add two methods for getting the unadjusted time of the clock and one for
adjusting an internal time. We will need these methods for correctly handling
the time after a gst_audio_clock_reset().
Add a debug category and some debug lines to the audio clock.
API: gst_audio_clock_get_time()
API: gst_audio_clock_adjust()
API: GST_AUDIO_CLOCK_CAST()
2009-04-14 13:08:52 +02:00
Wim Taymans
251f152c20
baseaudiosink: use the internal clock time
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We can't assume that the internal clock time is the same as the function we
installed on our provided clock because somebody might have changed it.
2009-04-10 21:50:55 +02:00
Martin Samuelsson
ee03bf5379
appsink: make callbacks return GstFlowReturn
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Make the new_buffer and new_preroll callbacks return a GstFlowReturn so that
errors can be reported properly.
Fixes #577827 .
2009-04-09 23:46:17 +02:00
Wim Taymans
e6798c5cce
ringbuffer: allow for custom commit functions
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Allow subclasses to override the commit method.
2009-04-09 18:04:44 +02:00
Wim Taymans
cae2981f83
baseaudiosink: fix a small glitch after pause
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After we pause the stream and interrupt the writeout to the ringbuffer, also adjust
the amount of output samples we consumed. We can't do this reliably with the
current API when we are doing trick modes but we can do the right thing for
normal playback.
2009-04-08 18:06:54 +02:00
Stefan Kost
ff9ee1dc5a
audiofilter: don't leak pad-template
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gst_element_class_add_pad_template() does not take ownership.
2009-04-07 22:39:07 +03:00
Edward Hervey
2555eeb737
navigation/v4l: Don't use g_return_val_if_fail for computed/used values.
2009-04-04 16:28:14 +02:00
Wim Taymans
88110ea67e
rtsp: use fully qualified urls when using a proxy
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Use a fully qualified url when specifying the url for tunneled requests through
a proxy.
See #573173
2009-04-02 22:28:55 +02:00
Jan Schmidt
033e654172
navigation: Extend the navigation interface
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Add support for a set of standard commands that can be queried and executed to
support applications like DVD. Add query construction and parsing functions.
Add new messages that can be sent on the bus to provide notifications related
to commands, multiangle changes, and button highlight activity.
Add some helper functions to parse the existing GstNavigation events that
elements might receive.
Document it all and add unit tests.
2009-04-02 12:21:18 +01:00
Wim Taymans
eed784b372
rtsp: fix little typo in the comments
2009-04-01 09:03:35 +02:00
Tim-Philipp Müller
fc8c5cba15
rtspconnection: make gst_rtsp_watch_queue_message() thread-safe
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People might queue messages from a thread other than the thread in which
the main context which this watch is attached is iterated from, so use
a GAsyncQueue instead of a GList, so g_list_append() doesn't trample
over list nodes just freed in the other thread. This just fixes issues
I've had with gst-rtsp-server. We might need more locking in various
places here.
2009-03-31 18:30:57 +01:00
Tim-Philipp Müller
dfe96ce618
rtsp: clear the entire builder structure
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And use structure instead of variable with sizeof when
clearing the rtsp message structure, for clarity.
2009-03-31 18:30:48 +01:00
Tim-Philipp Müller
dd9f077177
docs: fix typo in gst_rtsp_message_unset() API docs
2009-03-31 18:30:48 +01:00