Commit graph

520 commits

Author SHA1 Message Date
Sebastian Dröge
76985c5e81 rtpbuffer: Fix gst_rtp_buffer_ext_timestamp() with clang 5 on iOS/ARM
The bitwise NOT operator is not defined on signed integers.
Thanks to Wim Taymans for finding the cause.

https://bugzilla.gnome.org/show_bug.cgi?id=711819
2013-11-13 20:15:02 +01:00
Wim Taymans
240c7234f6 rtpbuffer: check for valid payload type
The payload type can't be between 72 and 76 because with the marker bit set,
this could be mistaken for an RTCP packet then. We do a relaxed check and
only refuse 72-76 when the marker bit is set. The effect is that when
we try to map an RTCP packet as an RTP packet, we will certainly fail.
2013-09-13 16:05:58 +02:00
Wim Taymans
ca1dac6982 rtcpbuffer: do additional packet checks
Check the packet size and avoid crashing on malformed packets.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=655727
2013-08-26 11:47:40 +02:00
Wim Taymans
b848f38215 rtcpbuffer: improve bye parsing
It is an error to ask for a non-existing BYE SSRC, the caller should
check the SSRC count first.
2013-08-26 11:46:11 +02:00
Wim Taymans
121235511a rtpbasedepayload: mark DISCONT on buffer in all cases
Always mark discont on the input buffer when we detect a seqnum
discont and not only when we previously marked ourselves DISCONT.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706422
2013-08-21 12:38:10 +02:00
Olivier Crête
c6fd304eb6 rtpbaseaudiopayload: Avoid copying the data 2013-08-18 22:24:08 -04:00
Wim Taymans
c1da65da5e rtcpbuffer: calculate FB packet length correctly 2013-08-06 15:44:03 +02:00
Ognyan Tonchev
25fdde908a rtpbasepayload: Do not leak the event when segment is delayed
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703119
2013-06-26 15:45:30 +02:00
Branko Subasic
4dd5c5b808 rtpbuffer: add gst_rtp_buffer_get_payload_bytes
The function gst_rtp_buffer_get_payload can not be used in Python
because it lacks necessary length parameter. This patch adds a new
function, gst_rtp_buffer_get_payload_bytes, to use from Python
bindings. The new function has the advisory "Rename to:" annotation
so it can replace the gst_rtp_buffer_get_payload whan creating
bindings.

The function gst_rtp_buffer_get_extension_bytes is also added. It wraps
gst_rtp_buffer_get_extension_data which doesn't work in Python due to
incomplete annotation and because it returns the length as number of
32-bit words.

https://bugzilla.gnome.org/show_bug.cgi?id=698562
2013-06-18 11:23:40 +02:00
Nicolas Dufresne
94b7ae7767 rtpbasepayload: Delay segment event after caps
https://bugzilla.gnome.org/show_bug.cgi?id=700222
2013-05-14 09:50:22 +02:00
Tom Greenwood
789ddf42a9 rtpbasedepayload: Ignore caps events if the caps did not change
https://bugzilla.gnome.org/show_bug.cgi?id=697672
2013-04-15 10:00:05 +02:00
Tim-Philipp Müller
664adc6e19 gst-libs: use GST_*_1_0 environment variables everywhere
The _1_0 suffixed environment variables override the
non-suffixed ones, so if we're in an environment that
sets the _1_0 suffixed ones, such as jhbuild, we need
to set those to make sure ours actually always get
used.
2013-01-16 10:16:27 +00:00
Thijs Vermeir
2887485358 rtp: fix compiler warning
comparison is always true due to limited range of data type
2012-12-18 15:27:48 +01:00
Sebastian Dröge
3f82e919dd libs: Use foo/foo.h as single-include header consistently everywhere
https://bugzilla.gnome.org/show_bug.cgi?id=688785
2012-12-12 17:13:10 +00:00
Evan Nemerson
4d77fba46c libs: Add missing single include headers and use them in GIRs 2012-11-21 11:01:24 +01:00
Tim-Philipp Müller
71e46b2478 gst_adapter_prev_timestamp -> gst_adapter_prev_pts
https://bugzilla.gnome.org/show_bug.cgi?id=675598
2012-11-14 00:03:15 +00:00
Wim Taymans
af3f75f3a9 rtpbuffer: protect against empty buffers 2012-11-12 11:18:16 +01:00
Wim Taymans
4463df5b0d rtp: fix ntp56 parsing 2012-11-06 09:18:54 +01:00
Wim Taymans
82d327fb91 rtp: add helpers for header extensions
Add helpers and defines for the NTP-64 and NTP-56 header extensions.
2012-11-06 09:18:54 +01:00
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
Wim Taymans
da1c863711 Merge branch 'master' into 0.11
Conflicts:
	gst-libs/gst/tag/gstvorbistag.c
2011-04-04 11:31:33 +02:00
Trond Andersen
cec628a414 rtcpbuffer: fix invalid read in validation of padding in rtcp packet 2011-04-04 09:43:06 +02:00
Wim Taymans
730b87271c bufferlist: fixes for new API 2011-03-31 17:47:43 +02:00
Tim-Philipp Müller
45b6bda76c libs: make sure gobject-introspection scanner calls gst_init()
Cherry-picked from 0.11, since it's the right thing to do (we
now silently rely on various _get_type() working without
gst_init() having been called).
2011-03-30 21:08:29 +01:00
Tim-Philipp Müller
a818fe7381 libs: replace 0.10 with @GST_MAJORMINOR@ in Makefile.am
For easier cherry-picking/merging later.
2011-03-30 20:57:32 +01:00
Wim Taymans
248ab2d064 Fix for latest API changes 2011-03-30 16:50:45 +02:00
Wim Taymans
e1869fa267 Merge branch 'master' into 0.11-fdo 2011-03-28 20:13:59 +02:00
Wim Taymans
e33b73f9df tests: fix RTP and RTCP unit tests 2011-03-28 18:42:09 +02:00
Wim Taymans
3d25a4b470 libs: port to new data API 2011-03-27 13:55:15 +02:00
Olivier Crête
103fb67d20 rtpbuffer: Off-by-one error when creating RTP header extensions with a two-byte header 2011-03-17 21:50:24 -04:00
Tim-Philipp Müller
842911d241 libs: make sure gobject-introspection scanner calls gst_init()
Fixes introspection failures caused by type assertions/warnings.
Since we now moved from _get_type() functions to external GType
variables in a couple of places, we actually have to call gst_init()
to make sure these are set when we use GST_TYPE_FOO.
2011-03-09 12:17:14 +00:00
Wim Taymans
c6dd11981d Merge branch 'master' into 0.11
Conflicts:
	configure.ac
	gst-libs/gst/pbutils/Makefile.am
2011-02-28 11:47:44 +01:00
Tim-Philipp Müller
0ed757db33 gobject-introspection: use same PKG_CONFIG_PATH for g-ir-compiler as for g-ir-scanner
Make sure to use the PKG_CONFIG_PATH set at configure time instead of
just relying on an env-var set one. This makes sure both g-ir-compiler
and g-ir-scanner use the same PKG_CONFIG_PATH for determining include
paths etc.
2011-01-08 02:10:03 +00:00
Wim Taymans
678753b325 baseaudiopay: fix timestamps on buffer lists
Fix the outgoing timestamps and RTP timestamps on outgoing buffers when using
buffer lists.
2010-12-30 18:09:58 +01:00
Wim Taymans
c665034742 basedepay: fix refcounting issue
Make sure that when _make_writable() returns a new buffer, we actually push that
one instead of the old one.
2010-12-28 11:44:09 +01:00
Wim Taymans
6ffabccf04 depay: update some docs 2010-12-21 15:11:10 +01:00
Wim Taymans
9e787a0211 rtpdepayloade: add support for getting events
Add support for intercepting sink events in the depayloader by adding a new
vmethod.
2010-12-21 15:02:18 +01:00
Wim Taymans
d51ff7e4eb basertppay: use RTP base time when invalid timestamps
When we have an invalid running-time (because we clipped, for example) use the
RTP base time for timestamping instead of generating wrong RTP timestamps.
2010-12-21 13:39:26 +01:00
Wim Taymans
bc63334503 rtppayload: copy applied rate to segment
Use set_segment_full to copy all segment values to the segment structure.
2010-12-21 13:39:26 +01:00
Wim Taymans
34ea5bdd06 rtpbuffer: relax arrangement for RTP bufferlists
Don't assume there are exactly 2 buffers but allow cases where the header and
payload are in 1 buffer or where the payload is in more buffers.
2010-12-15 16:37:29 +01:00
Wim Taymans
1c2c9c10b5 basedepay: add support for buffer lists in the depayloader
Add support for buffer lists in the depayloader.
2010-12-15 13:09:17 +01:00
Stefan Kost
ecb164675d docs: fix wrong use of Since: keyword 2010-12-08 12:11:23 +02:00
Wim Taymans
eee6bc7dc9 more 0.10 -> 0.11 changes 2010-12-06 17:09:10 +01:00
Evan Nemerson
8fb2c27ed0 introspection: Add information on exported packages to GIRs
https://bugzilla.gnome.org/show_bug.cgi?id=635392
2010-11-21 00:44:37 +00:00
Olivier Crête
582417e031 rtcpbuffer: Add function to manipulation the data in RTCP feedback packets
Add methods to get/set the length of the Feedback Control Information (FCI) as
well as getting a pointer to the FCI itself.
2010-10-05 16:19:14 +02:00
Olivier Crête
7536d96d7c rtpbuffer: Add function to transform a GstBuffer into a GstBufferList
Add a new function called gst_rtp_buffer_list_from_buffer() that takes
a GstBuffer containing a RTP packets and spits out a GstBufferList
containing two buffers, one with the header and the other with the payload.
2010-10-05 16:19:14 +02:00
Olivier Crête
f6b7ea3d39 rtpbuffer: Add functions to add RFC 5285 header extensions to GstBufferLists
Add functions to add header extensions to buffer lists, these functions only modify
the header part of the buffer lists, so the data is not copied.
2010-10-05 16:19:14 +02:00
Olivier Crête
fb770ca5e5 rtpbuffer: Add function to read RFC 5285 header extensions from GstBufferLists 2010-10-05 16:19:14 +02:00
Olivier Crête
484871b495 rtpbuffer: Add function to add RTP header extensions with a two bytes header 2010-10-05 16:19:14 +02:00
Olivier Crête
02a0139451 rtpbuffer: Add function to append RFC 5285 one byte header extensions 2010-10-05 16:19:14 +02:00
Olivier Crête
e4c06debb2 rtpbuffer: Add function to parse RFC 5285 header extensions
RFC 5285 describes a generic method to add multiple header extensions to RTP packets.
These functions parse these headers and return them, both for the one-byte header and the
two bytes headers.
2010-10-05 16:19:14 +02:00
Thijs Vermeir
a0fa0ff8bf basertpdepay: ensure metadata is writable 2010-09-29 16:53:21 +02:00
Tim-Philipp Müller
b550eabdac rtp: improve basertpdepayload's error message when no input caps were set
This is pretty much an FAQ, so try to make the error message a bit
more helpful. Also, don't tell people to file a bug in bugzilla
about this (which is what happens if the default error message for
CORE_NEGOTIATION is used).
2010-09-06 18:19:44 +01:00
Wim Taymans
9fd1c48267 rtppayload: notify of first timestamp/seqnum
Notify of the first timestamp/seqnum pushed out by the payloader.

Fixes #612264
2010-09-06 13:15:41 +02:00
American Dynamics
1f19649695 basertpdepay: don't clear the discont flag too early
Set the discont flag when we receive a DISCONT buffer and only clear the discont
state when we pushed out a DISCONT buffer.

Fixes #626869
2010-08-18 12:43:48 +02:00
Tim-Philipp Müller
e776699036 build: use new AG_GST_PKG_CONFIG_PATH m4 macro from common
Sets up a GST_PKG_CONFIG_PATH variable for use in Makefile.am
(avoids trailing ':' in PKG_CONFIG_PATH used).
2010-08-14 19:12:37 +01:00
Tim-Philipp Müller
b61b83376a introspection: set PKG_CONFIG_PATH so that our in-tree libs come first when calling scanner
When calling gobject-introspection scanner, make sure our own
freshly-built libs within the source tree (well, build dir) come
first in the PKG_CONFIG_PATH. May or may not help to make sure
that it doesn't pick up older external plugins-base libs (or
.gir files) from outside the source tree / build directory as
dependencies of the introspected lib instead of using the
stuff we just built in a sibling directory.

https://bugzilla.gnome.org/show_bug.cgi?id=623698
2010-08-14 19:11:48 +01:00
Olivier Crête
0a24137100 basertpaudiopayload: Add extra frame for non-complete frame lengths
Some payloaders like rtpg729pay can add a shorter frame at the end of a
RTP packet. We need to count it like a full frame for timestamps.

https://bugzilla.gnome.org/show_bug.cgi?id=618324
2010-05-13 11:03:12 +02:00
Olivier Crête
8a2b81a576 basertpaudiopayload: Set duration on buffers
Set the duration of the buffers from their size
2010-05-13 10:54:08 +02:00
Mark Nauwelaerts
ed71d802fc basertpdepayload: ensure writable metadata 2010-04-30 19:38:40 +02:00
Olivier Crête
66cc2faba7 audiopayload: use ptime-multiple
Based on patch by Olivier Crête <olivier.crete@collabora.co.uk>

Fixes #613248
2010-04-09 16:17:31 +02:00
Wim Taymans
1fa171d396 audiopayload: add property to control packet duration
Add a property to specify that the amount of data in a packet should be a
multiple of ptime-multiple.

See #613248
2010-04-09 16:17:31 +02:00
Vincent Untz
764c899215 libs: point gobject-introspection scanner to .la files
Point g-ir-scanner to the .la file of our library, which hopefully
makes it find the right dependencies in all cases (ie. our locally
built libgstreamer and not the system-installed one). This is also
how it's done in Gtk+ and how it's documented in the wiki, see
http://live.gnome.org/GObjectIntrospection/AutotoolsIntegration

Fixes #603710.
2010-04-03 14:03:45 +01:00
Tim-Philipp Müller
b37c993e4e gst-libs: more gobject-introspection fixes
Use right .pc file variable for compiler includes this time:
g-ir-compiler wants the girdirs not the typelibdirs as includes.
2010-03-30 23:46:10 +01:00
Tim-Philipp Müller
64cfa6bf73 gst-libs: fix up gobject-introspection some more
Use new girdir and typlibdir from core .pc files, so we can figure
out the right includes to pass to the gobject-introspection tools,
whether core is installed in the same prefix as gobject-introspection
or in a different prefix or uninstalled. This also keeps us from adding
bogus paths to the includes that only work if core is uninstalled.

Also add some missing includes/pkgs where needed.
2010-03-30 19:56:56 +01:00
Benjamin Otte
90f24667d7 Constify some strings in the API
Needed by plugins-good
2010-03-19 22:34:36 +01:00
Tim-Philipp Müller
e836151009 docs: more helper libraries docs fixes
Quieten gtk-doc a bit more.
2010-03-16 00:44:50 +00:00
Sebastian Dröge
d5a4ca9962 build: Make some more rules silent if requested 2010-03-09 21:01:38 +00:00
Wim Taymans
92a474b18c basedepay: clarify some documentation 2010-03-08 12:11:01 +01:00
Olivier Crête
6c6d0e32cf basertppayload: ptime/maxptime should be unsigned
https://bugzilla.gnome.org/show_bug.cgi?id=607403
2010-01-21 10:46:31 +01:00
Olivier Crête
8d2ac0b2ec basertppayload: ptime should be in nanoseconds
https://bugzilla.gnome.org/show_bug.cgi?id=607403
2010-01-21 10:46:17 +01:00
Olivier Crête
ad399c8069 basertppayload: Reject empty caps
https://bugzilla.gnome.org/show_bug.cgi?id=607353
2010-01-19 13:29:19 +01:00
Wim Taymans
73d5ae1107 audiopayload: add support for buffer-lists 2010-01-06 13:39:14 +01:00
Olivier Crête
bc6179952b basertpaudiopayload: Respect ptime if it is given
If the ptime is given in the caps, respect it and force the minimum
and maximum sizes to be exactly the requested ptime.

https://bugzilla.gnome.org/show_bug.cgi?id=606050
2010-01-05 13:20:49 -05:00
Olivier Crête
a4b0f2a1bd rtpbasepayload: Store ptime from caps
https://bugzilla.gnome.org/show_bug.cgi?id=606050
2010-01-05 13:20:49 -05:00
Olivier Crête
21151ba940 basertppayload: Accept maxptime from caps
https://bugzilla.gnome.org/show_bug.cgi?id=606050
2010-01-05 13:20:49 -05:00
Wim Taymans
f7070b6bc6 rtcpbuffer: add helper functions for SDES types
Add functions to convert SDES names to their types and back. Will be used later
to set SDES items using a GstStructure.

See #595265
2009-12-22 20:15:28 +01:00
Stefan Kost
f3db4e01b5 rtp: dump packets which we reject 2009-10-28 11:30:58 +02:00
Tim-Philipp Müller
6f4c1ac583 Remove GST_DEBUG_FUNCPTR where they're pointless
There's not much point in using GST_DEBUG_FUNCPTR with GObject
virtual functions such as get_property, set_propery, finalize and
dispose, since they'll never be used by anyone anyway. Saves a
few bytes and possibly a sixteenth of a polar bear.
2009-10-28 00:59:35 +00:00
Olivier Crête
e27c24b200 rtpaudiopayload: Only sent exact multiple of the frame size
Also align the maximum size with the frame size, not only the minimum
2009-10-23 13:56:05 +03:00
Wim Taymans
a87811f49a basertppayload: small comment fix 2009-10-16 10:59:39 +02:00
Peter Kjellerstedt
7bca2a0019 rtp: Correct timestamping of buffers when buffer_lists are used
The timestamping of buffers when buffer_lists are used failed if
a buffer did not have both a timestamp and an offset.
2009-10-16 10:51:22 +02:00
Sebastian Dröge
df9b8b57b3 introspection: Build pkgconfig before all libraries and set PKG_CONFIG_PATH
This way g-ir-scanner can find the gstreamer-*-0.10 pkg-config files.
2009-09-13 11:19:50 +02:00
Wim Taymans
e2e7ae0129 basertppay: don't print RTP timestamps as clocktime
Don't try to print the RTP timestamp as a GstClockTime, it's just a guint32.

Fixes #594757
2009-09-10 18:21:08 +02:00
Havard Graff
f710bec408 basertpdepayload: fix event forwarding 2009-09-08 15:10:59 +02:00
Havard Graff
f0f72088bc rtcpbuffer: add missing break in handling of GST_RTCP_TYPE_PSFB
Fixes #594258
2009-09-08 13:03:21 +02:00
Sebastian Dröge
7e90e0846c introspection: Strip Gst prefix from all types/functions 2009-09-05 12:31:47 +02:00
Sebastian Dröge
7794caf9f8 introspection: Fix build if gir-repository is not installed 2009-09-05 11:49:41 +02:00
Sebastian Dröge
8001b380b1 rtp: Add gobject-introspection support 2009-09-05 11:25:42 +02:00
Wim Taymans
7a7663476f audiortppay: add some debugging 2009-09-03 18:53:19 +02:00
Wim Taymans
c1db9ebb20 audiortppay: handle gaps
Add various conversion functions between time<->bytes<->rtptime that will be
used later on.
Refactor the min/max packet length code so that it can be used for both
sample/frame based payloaders. Cache the returned values.
code cleanups.
When we discover a DISCONT buffer, make the outgoing RTP timestamps have the
same gap as the GStreamer timestamps gap.
2009-09-03 17:59:00 +02:00
Wim Taymans
3a3c6f309c audiortppay: fix frame duration calculations
Fix the calculation of the frame duration and rtp timestamps.
Add some debugging
2009-09-03 17:59:00 +02:00
Wim Taymans
bfc19462bb rtppay: add some debugging 2009-09-03 17:59:00 +02:00
Wim Taymans
bb91a7b47c audiortppay: use offsets for RTP timestamps
Have a custom sample/frame function to generate an offset that the base class
will use for generating RTP timestamps. This results in perfect RTP timestamps
on the output buffers.
Refactor setting metadata on output buffers.
Add some more functionality to _flush().
Handle DISCONT on the input buffers and set the marker bit and DISCONT flag on
the next outgoing buffer.
Flush the pending data on EOS.
2009-09-03 17:58:59 +02:00
Wim Taymans
c1ae0a2003 audiortppay: move function around 2009-09-03 17:58:59 +02:00
Wim Taymans
5808041f44 audiortppay: fix sample duration calculation 2009-09-03 17:58:59 +02:00
Wim Taymans
299ab7be0e audiortppay: more refactoring
Unify the sample/frame buffer handling code by making the functions plugable.
2009-09-03 17:58:59 +02:00
Wim Taymans
fb5037f727 audiortppayload: refactor some more
Refactor getting the packet min/max size and alignment code.
Refactor converting bytes to time.
change some variable to something shorter.
2009-09-03 17:58:59 +02:00
Wim Taymans
1c6b71af03 audiortppayload: refactor and cleanup
Always use the adapter when we need to fragment the incomming buffer. Use more
modern adapter functions to avoid malloc and memcpy. The overall result is that
the code looks cleaner while it should be equally fast and in some case avoid a
memcpy and malloc.
Use the adapter timestamping functions for more precise timestamps in case of
weird disconts.
Cache some values instead of recalculating them.
Add gst_base_rtp_audio_payload_flush() to flush a certain amount of bytes from
the internal adapter.

API: GstBaseRTPAudioPayload::gst_base_rtp_audio_payload_flush()
2009-09-03 17:58:59 +02:00
Wim Taymans
50b9640d01 basertppay: add property to disable perfect RTP time
Add a property to disable the generation of perfect RTP timestamps. By default
it is active.

API: GstBaseRTPPayload::perfect-rtptime
2009-09-03 11:29:23 +02:00
Wim Taymans
3a4edea56d basertppay: allow subclasses to influence RTP time
Allow subclasses to use the OFFSET field on RTP buffers to influence the way in
which RTP timestamps are generated. Usually timestamps are created from the
GStreamer timestamps on the buffer, which could result in imperfect RTP
timestamps.
2009-09-03 11:15:20 +02:00