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311 commits

Author SHA1 Message Date
Sebastian Dröge
dc8984d76c Merge branch 'master' into 0.11 
Conflicts:
	gst-libs/gst/app/gstappsrc.c
	gst-libs/gst/audio/multichannel.h
	gst-libs/gst/video/videooverlay.c
	gst/playback/gstplaysink.c
	gst/playback/gststreamsynchronizer.c
	tests/check/Makefile.am
	win32/common/libgstvideo.def
 2012-01-10 13:15:12 +01:00
Pascal Buhler
0febae7443 rtcpbuffer: prevent overflow of 16bit header length. 
RTCP header can be  (2^16 + 1) * 4 bytes long, so when validating a bogus
packet it was possible to get a 16bit overflow resulting in a length of 0.
This would put the gst_rtcp_buffer_validate_data function in a endless loop.

https://bugzilla.gnome.org/show_bug.cgi?id=667313
 2012-01-05 11:12:25 +00:00
Wim Taymans
6be9a67148 rtp: add INIT macros 2011-12-09 19:22:21 +01:00
Tim-Philipp Müller
54c5cd8c3f rtpbuffer: add GST_RTP_BUFFER_INIT to initialize RTP buffers on the stack 
Fixes build of -good.
 2011-12-09 15:03:41 +00:00
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 
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 
And let's get fixing those docs :)
 2011-11-25 10:31:38 +01:00
Wim Taymans
7afdff3575 Merge branch 'master' into 0.11 
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 
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 
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 
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 
They got somehow commited in 7012e88090
 2011-11-11 10:39:52 +01:00
Wim Taymans
24347217a5 rtp: fix de/payloaders 
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 
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 
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 
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 
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 
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 
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 
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 
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 
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 
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 
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 
Fix for gst_*_segment_full rename.
 2011-05-09 18:16:46 +02:00
Wim Taymans
ec57868488 -base: don't use buffer caps 
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