Edward Hervey
ea0ed511f8
rtp: Initialize GstRTPBuffer before usage
2011-12-05 18:42:24 +01:00
Edward Hervey
94230af7a3
rtp: Don't forget to initialize GstRTPBuffer
2011-12-05 18:30:37 +01:00
Tim-Philipp Müller
177525f89f
Merge remote-tracking branch 'origin/master' into 0.11
...
Conflicts:
gst-libs/gst/netbuffer/gstnetbuffer.c
gst/ffmpegcolorspace/avcodec.h
gst/ffmpegcolorspace/gstffmpegcodecmap.c
gst/ffmpegcolorspace/imgconvert.c
gst/ffmpegcolorspace/imgconvert_template.h
gst/ffmpegcolorspace/mem.c
gst/playback/README
gst/playback/gstplaybasebin.c
gst/playback/gstplaybasebin.h
gst/playback/gstplaybin.c
sys/v4l/v4lmjpegsrc_calls.c
sys/v4l/videodev_mjpeg.h
tests/check/elements/gnomevfssink.c
2011-12-02 11:10:17 +00:00
Piotr Fusik
14644457b0
various: typo fixes
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Fix typos in code and docs. Fixes. #658984
2011-12-02 12:03:27 +01:00
Edward Hervey
d94535832b
gst-libs: Add --warn-all to introspection scanner
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And let's get fixing those docs :)
2011-11-25 10:31:38 +01:00
Wim Taymans
7afdff3575
Merge branch 'master' into 0.11
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Conflicts:
gst-libs/gst/audio/gstaudiodecoder.c
2011-11-17 17:07:41 +01:00
Wim Taymans
e302833e65
add parent to pad functions
2011-11-17 12:48:25 +01:00
Wim Taymans
2202511e77
add parent to query function
2011-11-16 17:25:17 +01:00
Wim Taymans
026ec68f75
_peer_get_caps() -> _peer_query_caps()
2011-11-15 18:04:17 +01:00
Wim Taymans
ab9ffa93f5
change getcaps to query
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Add sink and src event functions in rtpbasepayload
Add query vmethod to rtpbasepayload.
2011-11-15 18:04:16 +01:00
Olivier Crête
82827df405
rtcpbuffer: Add feedback message types from RFC 5104
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These are Codec Control messages (CCM)
https://bugzilla.gnome.org/show_bug.cgi?id=658419
2011-11-14 12:24:16 +01:00
Wim Taymans
fc04bcecbe
fix docs
2011-11-14 10:46:56 +01:00
Wim Taymans
107d5a3d05
rtp: fix headers
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indent, add padding, remove old abidata
2011-11-11 19:21:09 +01:00
Wim Taymans
5f1312b5d8
rename files to match object names
2011-11-11 12:32:23 +01:00
Wim Taymans
ccf511a5d4
rename BaseRTP -> RTPBase
2011-11-11 12:24:08 +01:00
Wim Taymans
ad8f694ec6
remove bogus files
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They got somehow commited in 7012e88090
2011-11-11 10:39:52 +01:00
Wim Taymans
24347217a5
rtp: fix de/payloaders
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gst_basertppayload -> gst_base_rtp_payload
Add pts/dts support in the depayloader
Remove old timestamp code
Add a default getcaps function so subclasses can chain up to it instead of
relying on the return value of the getcaps function.
2011-11-10 17:18:00 +01:00
Edward Hervey
771cbbb17c
rtpbuffer: Fix compilation issues with gcc 4.6.1
2011-11-04 10:36:15 +01:00
Wim Taymans
df4999aeb1
bufferlist: update for new API
2011-11-02 09:04:27 +01:00
Wim Taymans
01854cca80
basertppay: rename caps fields
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Make the caps fields for timestamp and seqnum match the element
properties.
See #628773
2011-10-27 18:54:50 +02:00
Wim Taymans
9555229e79
basedepay: remove old fields
2011-10-27 18:50:32 +02:00
Wim Taymans
06311362e9
fix compilation
2011-10-27 17:26:58 +02:00
Wim Taymans
7012e88090
Merge branch 'master' into 0.11
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Conflicts:
gst-libs/gst/audio/audio.h
gst-libs/gst/audio/gstaudiodecoder.c
gst-libs/gst/audio/gstaudiodecoder.h
gst-libs/gst/audio/gstaudioencoder.c
gst-libs/gst/audio/gstbaseaudioencoder.h
gst/playback/Makefile.am
gst/playback/gstplaybin.c
gst/playback/gstplaysink.c
gst/playback/gstplaysinkvideoconvert.c
gst/playback/gstsubtitleoverlay.c
gst/videorate/gstvideorate.c
gst/videoscale/gstvideoscale.c
win32/common/libgstaudio.def
2011-09-06 15:24:32 +02:00
Wim Taymans
e1287b97ab
Merge branch 'master' into 0.11
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Conflicts:
ext/ogg/gstoggmux.c
gst-libs/gst/audio/audio.c
gst-libs/gst/audio/audio.h
gst-libs/gst/audio/multichannel.h
gst-libs/gst/pbutils/Makefile.am
gst-libs/gst/pbutils/gstdiscoverer.c
gst/playback/gstplaysinkaudioconvert.c
gst/playback/gstplaysinkvideoconvert.c
win32/common/libgstaudio.def
2011-08-29 11:37:36 +02:00
Olivier Crête
791eeeb1a6
basertppayload: Make perfect timestamps reproducible across element restart
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Without the perfect timestamp machinery, the RTP timestamp can be
computed directly from the running time of a buffer, but the perfect
timestamp patch broke that assumption. This patch restores it by
having the first perfect timestamp be the running time of that buffer
and counting from there.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=654434
2011-08-25 14:16:48 +02:00
Wim Taymans
3fab57b5cf
Merge branch 'master' into 0.11
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Conflicts:
gst-libs/gst/interfaces/videooverlay.c
gst-libs/gst/rtp/gstrtpbuffer.c
po/af.po
po/az.po
po/bg.po
po/ca.po
po/cs.po
po/da.po
po/de.po
po/el.po
po/en_GB.po
po/es.po
po/eu.po
po/fi.po
po/fr.po
po/gl.po
po/hu.po
po/id.po
po/it.po
po/ja.po
po/lt.po
po/lv.po
po/nb.po
po/nl.po
po/or.po
po/pl.po
po/pt_BR.po
po/ro.po
po/ru.po
po/sk.po
po/sl.po
po/sq.po
po/sr.po
po/sv.po
po/tr.po
po/uk.po
po/vi.po
po/zh_CN.po
2011-08-22 13:06:27 +02:00
Stefan Kost
01bbdd6bdf
docs: handle warnings emitted by gtk-doc
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This is useful and in most cases someone had put arbitrary markup into the docs,
misspelled xref'ed symbols, forgot to add stuff to the docs etc..
2011-08-20 19:16:42 +02:00
Josep Torra
5629ed74b3
Fix debug statements
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Fixes build on MacOSX
Signed-off-by: Edward Hervey <edward.hervey@collabora.co.uk>
2011-08-10 11:15:41 +02:00
Mark Nauwelaerts
06557739ab
rtcpbuffer: provide a WRITE map with maximum available size
...
... which allows adding additional packets and may be needed to counteract
the shrink that implicitly occurred during a map/unmap cycle when adding
a previous packet.
2011-07-09 18:23:18 +02:00
Tim-Philipp Müller
4bf26ba5d2
Add -DGST_USE_UNSTABLE_API to the compiler flags to avoid warnings
2011-07-05 10:07:08 +01:00
Wim Taymans
a8ffd4e28c
rtp: fix for allocator name change
2011-06-22 11:45:58 +02:00
Debarshi Ray
2c6dbae423
Remove unused but set variables
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This is needed to satisfy the new -Wunused-but-set-variable added in
GCC 4.6: http://gcc.gnu.org/gcc-4.6/changes.html
2011-06-14 22:40:13 +01:00
Wim Taymans
9c54ca5254
-base: update for buffer API change
2011-06-13 16:32:56 +02:00
Wim Taymans
7538dffaa0
basertppayload: cleanup header
2011-06-13 16:28:58 +02:00
Wim Taymans
2a94b0eb04
rtp: use new memory alloc API
2011-06-07 16:18:40 +02:00
Wim Taymans
28f67f4847
event: fix some event leaks
2011-06-07 12:06:22 +02:00
Wim Taymans
81ebc0a82e
basertp: use caps event instead of setcaps function
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Use the caps event instead of the setcaps function to configure caps.
Use a default event handler for the base rtp payloader instead of the awkward
way of handling the return value.
2011-06-02 19:21:24 +02:00
Wim Taymans
a87c021237
Merge branch 'master' into 0.11
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Conflicts:
gst-libs/gst/video/convertframe.c
2011-05-24 09:47:15 +02:00
Stefan Kost
269205b1ad
docs: rtp library docs update
2011-05-23 23:56:09 +03:00
Sebastian Dröge
884213b8b8
base: Update for SEGMENT event parse API changes
2011-05-18 17:23:18 +02:00
Sebastian Dröge
97f18beaeb
basertppayload: Change ::get_caps to include the filter caps
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And improve downstream negotiation a bit by passing our proposed
caps to the peer as a filter.
2011-05-16 15:35:40 +02:00
Wim Taymans
94dfe80f71
-base: port to new SEGMENT API
2011-05-16 13:48:11 +02:00
Wim Taymans
816f4e791d
segment: fix for new core API
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Fix for gst_*_segment_full rename.
2011-05-09 18:16:46 +02:00
Wim Taymans
ec57868488
-base: don't use buffer caps
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Port to newest 0.11 core API, remove GST_PAD_CAPS and GST_BUFFER_CAPS.
2011-05-09 13:05:12 +02:00
Sebastian Dröge
a7e8c8debe
gstbasertppayload: Use g_once_init_{enter,leave}() in the _get_type() function
2011-04-18 18:30:41 +02:00
Sebastian Dröge
5d4fd722f0
rtp: Use G_DEFINE_TYPE instead of GST_BOILERPLATE
2011-04-18 18:29:35 +02:00
Sebastian Dröge
c8792778f8
Merge branch 'master' into 0.11
2011-04-16 16:06:26 +02:00
Tim-Philipp Müller
1d05e81435
libs: gobject-introspection scanner doesn't need to scan or update plugin info
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Make sure the scanner doesn't load or introspect or check any plugins,
(especially not outside the build directory).
2011-04-16 11:01:53 +01:00
Wim Taymans
6e160bed3d
Merge branch 'master' into 0.11
...
Conflicts:
android/alsa.mk
android/app.mk
android/app_plugin.mk
android/audio.mk
android/audioconvert.mk
android/decodebin.mk
android/decodebin2.mk
android/gdp.mk
android/interfaces.mk
android/netbuffer.mk
android/pbutils.mk
android/playbin.mk
android/queue2.mk
android/riff.mk
android/rtp.mk
android/rtsp.mk
android/sdp.mk
android/tag.mk
android/tcp.mk
android/typefindfunctions.mk
android/video.mk
2011-04-11 11:37:51 +02:00
Alessandro Decina
030f639a8e
android: make it ready for androgenizer
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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 07:23:21 +02:00