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401 commits

Author SHA1 Message Date
Tim-Philipp Müller
5f59b4f7ee Fix FSF address 
https://bugzilla.gnome.org/show_bug.cgi?id=687520
 2012-11-03 23:05:09 +00:00
Miguel Angel Cabrera Moya
b1318c86e8 rtpbasedepay: remove unused variable 
https://bugzilla.gnome.org/show_bug.cgi?id=687146
 2012-10-29 21:20:35 +00:00
Tim-Philipp Müller
a4f2df6341 Revert "g-i: change g-ir-scanner arg --library=libgstfoo-X.la to --library=gstfoo-X" 
This reverts commit e39fbe6b7e.

Looks like we need to pass the full .la file after all in a setup
with libtool, or it might not find the library, e.g. like

  ERROR: can't resolve libraries to shared libraries: gstfft-1.0

Conflicts:
	gst-libs/gst/audio/Makefile.am
	gst-libs/gst/pbutils/Makefile.am

Also see https://bugzilla.gnome.org/show_bug.cgi?id=603710
 2012-10-29 12:47:05 +00:00
Tim-Philipp Müller
e39fbe6b7e g-i: change g-ir-scanner arg --library=libgstfoo-X.la to --library=gstfoo-X 
As it should be according to the man page.

https://bugzilla.gnome.org/show_bug.cgi?id=679315
 2012-10-28 17:35:57 +00:00
Tim-Philipp Müller
5e0dfec62c Remove -DGST_USE_UNSTABLE_API 2012-09-17 16:05:37 +01:00
Tim-Philipp Müller
21c61586ad rtpbasepayload: error out if no CAPS event was received before buffers 
Most payloaders set/send their own output format from the setcaps
function, so if we don't get input caps, things probably wont' work
right, even if the input format is fixed (as in the case of the mpeg-ts
payloader for example).

https://bugzilla.gnome.org/show_bug.cgi?id=683428
 2012-09-06 18:23:22 +01:00
Tim-Philipp Müller
3d006f6d2a rtpbasepayload: assume input caps are accepted if subclass has no set_caps vfunc 
Not that anyone should ascribe too much meaning to these return
values in the age of sticky caps.
 2012-09-06 17:47:01 +01:00
Mark Nauwelaerts
bd67736851 rtpbasedepay: indicate packet loss using GAP event 2012-09-05 12:02:32 +02:00
Tim-Philipp Müller
392d3225ce rtp: fix buffer leak when gst_rtp_buffer_map() fails because of broken data 
Makes libs/rtp unit test valgrind clean.
 2012-08-22 09:20:55 +01:00
Wim Taymans
1968127650 rtp: Fix extension data support 
Allocate header, payload and padding in separate memory blocks in
gst_rtp_buffer_allocate().
don't use part of the payload data as storage for the extension data but store
it in a separate memory block that can be enlarged when needed.
Rework the one and two-byte header extension to make it reserve space for the
extra extension first.
Fix RTP unit test. Don't map the complete buffer or make assumptions on the
memory layout of the underlaying implementation. We can now always add extension
data because we have a separate memory block for it.
 2012-08-22 09:56:39 +02:00
Wim Taymans
2d6fd0f72d rtp: fix extension length calculation 2012-08-22 09:56:39 +02:00
Wim Taymans
f548e58385 rtp: remove unused field 2012-08-22 09:56:39 +02:00
Andoni Morales Alastruey
d2aebc7f94 rtpbuffer: use proper format for gsize 2012-08-08 17:41:19 +02:00
Wim Taymans
11a494d5c9 rtp: Add support for multiple memory blocks in RTP 
Add support RTP buffers with multiple memory blocks. We allow one block for the
header, one for the extension data, N for data and one memory block for the
padding.
Remove the validate function, we validate now when we map because we need to
parse things in order to map multiple memory blocks.
 2012-07-17 16:41:36 +02:00
Evan Nemerson
f21c4667b9 rtp: add many missing annotations on RTP/RTCP buffer functions 2012-07-17 11:10:37 +02:00
Evan Nemerson
63579633f5 rtpbaseaudiopayload: add transfer annotation to get_adapter return 2012-07-17 11:10:04 +02:00
Edward Hervey
2817bdadc9 libs: Remove "Since" markers and minor doc fixups 2012-07-13 12:11:06 +02:00
Wim Taymans
baa2fac2f8 audiopayload: disable broken bufferlist handling 
The bufferlist handling is broken so make sure it is never enabled.
 2012-06-06 16:40:24 +02:00
Sebastian Rasmussen
b7b123964b gst-libs: make pkg-config get path to pkg-config dirs from configure 
When --with-pkg-config-path is supplied to configure this path is now
explicitly propagated to pkg-config.

https://bugzilla.gnome.org/show_bug.cgi?id=673377
 2012-05-05 23:26:20 +01:00
Sebastian Dröge
65307dd132 gst: Update versioning 2012-04-04 14:55:15 +02:00
Wim Taymans
296e1bf3dd rtpbuffer: removed old memory 
Ensure writability of rtp buffer and remove old memory first
Fix some docs
 2012-04-04 09:34:00 +02:00
Wim Taymans
6e9d28eef6 rtp: fix initializer 2012-04-02 11:05:38 +02:00
Wim Taymans
92f46c07fe rtpbuffer: keep more state 
Prepare for the future, make it possible to map multiple buffer regions, like
the header and the payload.
 2012-04-02 10:31:18 +02:00
Wim Taymans
9ef519d99a Improve buffer allocation of wrapped memory 2012-04-01 18:11:23 +02:00
Wim Taymans
345dc31f20 update for buffer api change 2012-03-30 18:15:30 +02:00
Mark Nauwelaerts
6039266c43 rtpbasepayload: plug caps leak 2012-03-29 17:15:43 +02:00
Wim Taymans
df5253b22c update for memory api changes 2012-03-15 13:32:08 +01:00
Wim Taymans
28034226c6 update for memory api changes 2012-03-14 21:35:45 +01:00
Wim Taymans
37e940df83 rtpbasepay: add support for DTS and PTS 2012-03-13 18:15:04 +01:00
Wim Taymans
25137962ad fix for caps API changes 2012-03-11 19:04:41 +01:00
Wim Taymans
63f3f27164 update for new memory api 2012-02-22 02:05:24 +01:00
Olivier Crête
cb044668d3 rtcpbuffer: Set the map.size to the current size of the RTCP packet 
maxsize is the maximum size
 2012-01-27 19:01:55 +01:00
Olivier Crête
b993b8457d rtpcbuffer: To write inside a RTCP buffer, you must be able to read 
So always require read
 2012-01-27 19:01:55 +01:00
Olivier Crête
6b559a50fb rtcpbuffer: Return errors if the map mode doesn't match the actions 2012-01-27 19:01:55 +01:00
Olivier Crête
ab359d36d5 rtcpbuffer: Don't try to modify read-only buffers 2012-01-27 19:01:55 +01:00
Wim Taymans
e7575bc525 rtp: improve structures 
Remove flags that is in the mapinfo now
 2012-01-25 12:30:53 +01:00
Wim Taymans
fcdc385aa1 port to new map API 2012-01-25 12:30:53 +01:00
Olivier Crête
1a592199e9 rtpbasepayload: Port to group-less GstBufferList 2012-01-25 11:55:02 +01:00
Wim Taymans
d9ef75b799 rtcp: handle size update correctly 
Do explicit resize to set the size of a buffer instead of setting a value in
unmap.
 2012-01-19 15:20:01 +01:00
Wim Taymans
5872bcc33a Update for memory API changes 2012-01-19 12:15:18 +01:00
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
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 
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 
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
Sebastian Dröge
0a1d85c233 rtp: Unref events if the parent element disappeared or has no event handler implemented 2011-04-08 15:10:02 +02:00
Ole André Vadla Ravnås
f59b985698 rtp: fix pad callbacks so they handle when parent goes away 
1) We need to lock and get a strong ref to the parent, if still there.
2) If it has gone away, we need to handle that gracefully.

This is necessary in order to safely modify a running pipeline. Has been
observed when a streaming thread is doing a buffer_alloc() while an
application thread sends an event on a pad further downstream, and from
within a pad probe (holding STREAM_LOCK) carries out the pipeline plumbing
while the streaming thread has its buffer_alloc() in progress.
 2011-04-08 15:05:23 +02:00
Wim Taymans
3ea2bc3ab0 Merge branch 'master' into 0.11 
Conflicts:
	gst-libs/gst/rtp/gstbasertpdepayload.c
 2011-04-07 16:19:08 +02:00
Bastien Nocera
96463bb8df rtp: Remove unused variables 
https://bugzilla.gnome.org/show_bug.cgi?id=646924
 2011-04-07 10:16:39 +02:00
Wim Taymans
4007076b55 Merge branch 'master' into 0.11 
Conflicts:
	ext/theora/gsttheoraenc.c
 2011-04-06 16:33:56 +02:00
Pascal Buhler
1ad98b0d98 rtcpbuffer: Round to next 32bit word, not current 32bit word at end of SDES chunk 2011-04-05 15:27:03 +02:00