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776 commits

Author SHA1 Message Date
Mark Nauwelaerts
fa90dfc4df rtph264pay: avoid crashing on NULL access in debug message 2012-09-07 15:25:52 +02:00
Mark Nauwelaerts
8f4bfeb698 rtph263ppay: plug caps leak 2012-09-07 15:25:52 +02:00
Tim-Philipp Müller
9bf90f47cf video/x-xvid -> video/mpeg,mpegversion=4 2012-09-03 02:51:24 +01:00
Tim-Philipp Müller
4bb52bbadf docs: gst-launch -> gst-launch-1.0 and ffmpegcolorspace -> videoconvert 2012-08-27 21:20:30 +01:00
Olivier Crête
264bcf7d6f rtph264pay: Make it actually work after cleanups 2012-08-08 19:49:05 -07:00
Mark Nauwelaerts
1547fdbe5a rtpmparobustdepay: set correct data_size for generated dummy frame 
... which prevents getting stuck in a loop if such one is needed.
 2012-08-06 14:58:21 +02:00
Mark Nauwelaerts
3e1832f5a4 rtpmparobustdepay: improve and fix debug statement 
... so it really informs about next rather than past frame.
 2012-08-06 14:58:21 +02:00
Mark Nauwelaerts
31a1cb0a11 rtpmparobustdepay: update available bytewriter space when repositioning 
... and add some more assert to catch potential surprises early on.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680558
 2012-08-06 14:58:21 +02:00
Mark Nauwelaerts
0bf9d8c6a6 rtpmparobustdepay: modify buffer data rather than buffer itself 2012-07-26 16:34:52 +02:00
Mark Nauwelaerts
c40807f6aa rtpmparobustdepay: avoid leaking bytewriter instance 2012-07-26 16:34:52 +02:00
Wim Taymans
0ed9e07c5d h264depay: small cleanups 2012-07-25 12:49:07 +02:00
Wim Taymans
4b92022120 rtp: always use buffer lists 2012-07-23 16:42:56 +02:00
Patricia Muscalu
3dd99f06f4 rtpmp4vpay: always enable buffer-lists 2012-07-23 16:17:37 +02:00
Patricia Muscalu
15cce2dd26 rtpjpegpay: always enable buffer-lists 2012-07-23 16:15:59 +02:00
Wim Taymans
51371d26ee update for RTP buffer api changes 2012-07-17 16:38:27 +02:00
Patricia Muscalu
d38ac43a27 rtph264pay: use buffer lists 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679994
 2012-07-17 10:10:14 +02:00
Tim-Philipp Müller
c6224443a4 rtph264pay: avoid some relocations 2012-07-06 19:11:02 +01:00
Tim-Philipp Müller
3ef35ecdbc rtpmp4vpay: remove deprecated send-config property 
Use config-interval instead.
 2012-07-06 14:49:18 +01:00
Tim-Philipp Müller
cd1da84bcc rtph264depay: remove deprecated "byte-stream" and "access-unit" properties 
These will be picked automatically based on downstream caps now, so
if you want the depayloader to output a specific format, make sure
the element downstream advertises that preference or use a capsfilter
after the depayloader to force it.
 2012-07-06 14:46:22 +01:00
Tim-Philipp Müller
cffbf8cfc3 rtph264pay: remove deprecated and non-functional "profile-level-id" property 
This is now optionally taken from downstream caps, so can be
specified via a capsfilter after the payloader.
 2012-07-06 14:46:22 +01:00
Tim-Philipp Müller
48706beb70 rtph263ppay: accept any h263 input unless downstream forces specific requirements 
rtph263ppay should accept any input compatible with its sink template
caps if it just outputs to e.g. udpsink or fakesink.

rtph263ppay ! rtph263pdepay should also work with any compatible input.
This would fail before with not-negotiated errors because the get_caps
function would see the encoding-name in the depayloader's template caps
and default to baseline H.263 because there's no profile/level information
in those caps, which is the right thing to do if downstream has filtercaps
from an SDP, but not if those fields are absent because they can be
anything like with the depayloader's template caps. Makes

  videotestsrc ! avenc_h263p ! rtph263ppay ! rtph263pdepay ! fakesink

work.
 2012-07-06 11:57:38 +01:00
Wim Taymans
8eadb9c12c update for query api changes 2012-07-06 11:26:46 +02:00
Javier Jardón
c740490c26 rtp: remove some outdated comments 
https://bugzilla.gnome.org/show_bug.cgi?id=679301
 2012-07-03 08:58:26 +01:00
Wim Taymans
6d158775bb rtph264pay: cleanups 
Use the caps properties for alignment and format.
Remove some old properties, we always want to use bufferlists when we can now.
 2012-06-28 12:00:09 +02:00
Wim Taymans
429bda6923 h264pay: prefer AVC, it's easier to parse etc 2012-06-28 11:32:03 +02:00
Wim Taymans
540245894f theoradepay: fix buffer memory 
The memory was added to the input buffer instead of the output buffer.
 2012-06-14 10:43:56 +02:00
Thiago Santos
78ec03e32f Some printf variable format fixes 
The osx compiler complains about those
 2012-06-05 17:53:57 -03:00
Edward Hervey
923be8a85b rtpmp2tdepay: Only output integral mpeg-ts packets 
From RFC 2250

2. Encapsulation of MPEG System and Transport Streams
...
   For MPEG2 Transport Streams the RTP payload will contain an integral
   number of MPEG transport packets.  To avoid end system
   inefficiencies, data from multiple small MTS packets (normally fixed
   in size at 188 bytes) are aggregated into a single RTP packet.  The
   number of transport packets contained is computed by dividing RTP
   payload length by the length of an MTS packet (188).
....

Since it needs to contain "an integral number of MPEG transport packets", a
simple fix is to check that's the case, and strip off any leftover data.

Fixes #676799

Conflicts:

	gst/rtp/gstrtpmp2tdepay.c
 2012-05-26 12:04:54 +02:00
Luis de Bethencourt
c81fff0471 rtp: fix build issue in gstrtph264pay.c 2012-05-24 09:29:25 +01:00
Jonas Holmberg
7bf3a1bf95 rtph264pay: Add unrestricted caps 
If there are no profile restrictions downstream, return caps with
profile=constrained-baseline in the first structure and append
unrestricted caps as the last structure.

Fixes bug #672019
 2012-05-24 10:01:19 +02:00
Mark Nauwelaerts
182596b3ab rtpmp2tpay: respect mtu and packet boundaries 
See #659915.
 2012-05-18 12:53:44 +02:00
Youness Alaoui
7703a11073 rtpjpegpay: Allow U and V components to use different quant tables if they contain the same data 
This allows some cameras (Logitech C920) that specify different quant
tables but both with the same data, to work.
Bug reported by Robert Krakora
 2012-05-16 09:49:08 +02:00
idc-dragon
e0945d0a2d celtdepay: calculate size correctly 
The summation was done wrong, causing the de-payloader to exit its loop too
early, before all frames are processed.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=674472
 2012-04-25 10:29:56 +02:00
Sebastian Dröge
04b70571e5 video: Update for libgstvideo API changes 2012-04-19 12:20:59 +02:00
Edward Hervey
71fc25849e rtp: Use unchecked variant of GstByteWriter where applicable 
The size was checked before
 2012-04-12 15:50:16 +02:00
Tim-Philipp Müller
e09ae5736d Use new gst_element_class_set_static_metadata() 2012-04-10 00:51:41 +01:00
Sebastian Dröge
aa2cd462da gst: Update for GST_PLUGIN_DEFINE() API changes 2012-04-05 17:36:38 +02:00
Sebastian Dröge
5cdd49bf25 gst: Update versioning 2012-04-04 14:37:47 +02:00
Wim Taymans
3d61d12e03 update for buffer api change 2012-03-30 18:15:34 +02:00
Wim Taymans
69002aa24f update for buffer changes 2012-03-28 12:53:05 +02:00
Wim Taymans
e310ee8218 caps: improve caps handling 
Avoid caps copy and leaks
 2012-03-27 16:42:41 +02:00
Mark Nauwelaerts
e5ab3cc0a0 rtph264pay: ensure output caps are set when pushing output data 
... even if some SPS/PPS has not passed by yet.
 2012-03-26 18:38:34 +02:00
Mark Nauwelaerts
4bbc2a7106 rtpL16(de)pay: fix raw audio format in template caps 2012-03-26 18:38:34 +02:00
Olivier Crête
06f1c1817e rtph264depay: Make output in AVC stream format work even without complete sprop-parameter-set 
This allows outputting streams in AVC format even if the SPS/PPS are sent inside
the RTP stream.

https://bugzilla.gnome.org/show_bug.cgi?id=654850

Ported from master
 2012-03-22 16:18:37 -04:00
Wim Taymans
c44cd8f55b Merge branch 'master' into 0.11 
unport gdkpixbuf
not merged: https://bugzilla.gnome.org/show_bug.cgi?id=654850

Conflicts:
	docs/plugins/Makefile.am
	docs/plugins/gst-plugins-good-plugins-docs.sgml
	docs/plugins/gst-plugins-good-plugins-sections.txt
	docs/plugins/gst-plugins-good-plugins.hierarchy
	docs/plugins/inspect/plugin-avi.xml
	docs/plugins/inspect/plugin-png.xml
	ext/flac/gstflacdec.c
	ext/flac/gstflacdec.h
	ext/libpng/gstpngdec.c
	ext/libpng/gstpngenc.c
	ext/speex/gstspeexdec.c
	gst/audioparsers/gstflacparse.c
	gst/flv/gstflvmux.c
	gst/rtp/gstrtpdvdepay.c
	gst/rtp/gstrtph264depay.c
 2012-03-22 11:53:24 +01:00
Wim Taymans
ced47580b7 update for bufferpool changes 2012-03-15 22:11:17 +01:00
Wim Taymans
f3a770a20c update for allocation query changes 2012-03-15 20:37:56 +01:00
Olivier Crête
053f33adc8 rtph264depay: Make output in AVC stream format work even without complete sprop-parameter-set 
This allows outputting streams in AVC format even if the SPS/PPS are sent inside
the RTP stream.

https://bugzilla.gnome.org/show_bug.cgi?id=654850
 2012-03-15 14:20:22 -04:00
Wim Taymans
751fcf035b take padding into account 2012-03-14 19:56:56 +01:00
Wim Taymans
734f11e4d3 mp4vpay: we can also handle x-divx 2012-03-14 11:26:35 +01:00
Wim Taymans
fba47d17e8 mp4vdepay: fix buffer handling 
Don't always output the payload subbuffer, use a separate variable to
make things clearer and without the error.
 2012-03-13 21:31:48 +01:00
Wim Taymans
745210e792 h264depay: unmap on empty packet 2012-03-13 19:26:23 +01:00
Wim Taymans
d65de434f5 rtph264pay: do DTS and PTS correctly 2012-03-13 18:07:18 +01:00
Wim Taymans
e4fed38f49 rtp: fix unmap calls 2012-03-13 17:27:32 +01:00
Wim Taymans
a32d944a38 fix for caps api changes 2012-03-11 19:06:37 +01:00
Sebastian Dröge
78079635a6 dvdepay: Fix 'comparison of unsigned expression >= 0 is always true' compiler warning 
This was an actual bug as it could've caused reading from
invalid memory areas when the input is broken.
 2012-03-06 14:16:21 +01:00
Wim Taymans
ca9532ccc5 update for new memory api 2012-02-22 02:10:33 +01:00
Olivier Crête
18899cf94d rtph264pay: Force baseline is profile-level-id is unspecified 2012-02-21 10:51:43 +01:00
Olivier Crête
1fe69911a4 rtph264pay: Force baseline is profile-level-id is unspecified 2012-02-20 14:30:55 -05:00
Wim Taymans
225e98d623 Merge branch 'master' into 0.11 
Conflicts:
	ext/flac/gstflacenc.c
	ext/jack/gstjackaudioclient.c
	ext/jack/gstjackaudiosink.c
	ext/jack/gstjackaudiosrc.c
	ext/pulse/plugin.c
	ext/shout2/gstshout2.c
	gst/matroska/matroska-mux.c
	gst/rtp/gstrtph264pay.c
 2012-02-10 16:23:14 +01:00
Tim-Philipp Müller
5b25f3737b rtph264pay: add stream-format and alignment to h264 sink caps 
We're happy to accept both byte-stream and avc, advertise
that on the sink caps and fix up _get_caps() function to
not just return "video/x-h264".

https://bugzilla.gnome.org/show_bug.cgi?id=606662
 2012-02-10 14:08:55 +00:00
Tim-Philipp Müller
6872b40873 rtph264depay: add stream-format and alignment fields to src template caps 
Because we can. And so we get a warning if we try to output avc with
nal alignment or somesuch.

https://bugzilla.gnome.org/show_bug.cgi?id=606662
 2012-02-10 14:08:55 +00:00
Vincent Penquerc'h
d651baf05a rtpmp2tpay: do not try to flush a packet when no data is available 
https://bugzilla.gnome.org/show_bug.cgi?id=668874
 2012-01-31 13:12:47 +00:00
Pascal Buhler
c16fed2ad9 rtph264depay: Exclude NALu size from payload length on truncated packets. 
https://bugzilla.gnome.org/show_bug.cgi?id=667846
 2012-01-30 15:49:07 +00:00
Sebastian Dröge
0b517ce9fb Merge branch '0.11' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-good into 0.11 2012-01-25 12:49:34 +01:00
Sebastian Dröge
10554b271f Merge branch 'master' into 0.11 
Conflicts:
	ext/flac/gstflacdec.c
	ext/jpeg/gstjpegenc.c
	ext/pulse/pulsesink.c
	sys/v4l2/gstv4l2src.c
 2012-01-25 12:49:11 +01:00
Wim Taymans
b4630dd3e0 more memory API porting 2012-01-25 12:30:29 +01:00
Wim Taymans
583d39dd8d update for new memory API 2012-01-25 12:30:28 +01:00
Mark Nauwelaerts
a3ea25bc88 rtpmp4adepay: prevent out-of-bound array access 2012-01-20 17:10:48 +01:00
Mark Nauwelaerts
ed94e01231 rtptheoradepay: remove dead code 2012-01-20 17:10:40 +01:00
Sebastian Dröge
59e08fa503 configure: Remove socket/winsock specific checks 
Not necessary anymore.
 2012-01-17 16:53:31 +01:00
Vincent Penquerc'h
2a7a38ca07 rtph263ppay: fix caps leak 2012-01-16 15:42:46 +00:00
Sebastian Dröge
4885f34458 rtp: Update for the new audio caps 2012-01-05 10:30:34 +01:00
Tim-Philipp Müller
668e15598b Merge remote-tracking branch 'origin/master' into 0.11 
Conflicts:
	sys/v4l2/gstv4l2object.c
 2011-12-08 01:28:26 +00:00
Wim Taymans
b1d771cf8c h263pay: fix invalid return value 2011-12-06 14:23:30 +01:00
Edward Hervey
04520cbe9a rtp: Initialize GstRTPBuffer before usage 2011-12-05 18:39:59 +01:00
Sebastian Rasmussen
c090201ca5 rtpjpegpay: Ceil jpeg dimensions, instead of floor 
A JPEG image inside an RTP stream has a preceeding RFC2435 header that
conveys width/height. The dimensions in this header are limited to be
multiples of 8. Since JPEG uses an MCU of 8x8 pixels any image must
already indirectly have image data dimensions that are rounded up in
order to contain enough data to render the image. Therefore this fix
safely rounds the image dimensions in the RFC2435 header up to the
closest multiple of 8.
 2011-12-05 10:48:54 +01:00
Vincent Penquerc'h
c0e101e93f various: fix pad template leaks 
https://bugzilla.gnome.org/show_bug.cgi?id=662664
 2011-11-28 13:30:27 +00:00
Matej Knopp
1e5dd9e315 Fix printf format compiler warnings on OS X / 64bit 
https://bugzilla.gnome.org/show_bug.cgi?id=662615
 2011-11-22 01:28:22 +00:00
Wim Taymans
105650127e add parent to pad functions 2011-11-17 15:02:55 +01:00
Wim Taymans
797523efbd _peer_get_caps() -> _peer_query_caps() 2011-11-15 18:04:44 +01:00
Wim Taymans
75dc9634eb change getcaps to query 
Chain up event function in payloaders.
 2011-11-15 18:04:44 +01:00
Wim Taymans
af1eec2ece rtp: fix for rtp header changes 2011-11-11 19:21:50 +01:00
Wim Taymans
e84b8dbe94 update for base class rename 2011-11-11 12:32:41 +01:00
Wim Taymans
249d0083cc update for base class rename 2011-11-11 12:25:01 +01:00
Wim Taymans
7e12b58e37 update for adapter api changes 2011-11-10 18:32:58 +01:00
Wim Taymans
fbaf216d25 update for changed base classes 2011-11-10 17:23:47 +01:00
Wim Taymans
85e73e0818 h263ppay: report to 0.11 2011-11-09 12:25:01 +01:00
Wim Taymans
95f3987332 Merge branch 'master' into 0.11 
Conflicts:
	ext/flac/gstflacdec.c
	gst/audioparsers/gstflacparse.c
	gst/isomp4/qtdemux.c
 2011-11-09 12:18:01 +01:00
Olivier Crête
e15c293f13 rtph263ppay: Return the sink pad template as sink caps, not the src's 
https://bugzilla.gnome.org/show_bug.cgi?id=577784
 2011-11-08 15:53:39 +01:00
Olivier Crête
4b28d9d44e rtph263ppay: Also implement size/framerate restrictions in getcaps 
https://bugzilla.gnome.org/show_bug.cgi?id=577784
 2011-11-08 15:53:18 +01:00
Olivier Crête
ff31090671 rtph263ppay: Implement getcaps following RFC 4629, picks the right annexes 
https://bugzilla.gnome.org/show_bug.cgi?id=577784
 2011-11-08 15:52:57 +01:00
Tim-Philipp Müller
d65490dfad rtp: use GLib's G_BIG_ENDIAN define instead of BIG_ENDIAN 
Fixes compiler warning on mingw32
 2011-11-03 23:28:31 +00:00
Wim Taymans
b1ef7e8a86 update for meta api change 2011-11-02 09:06:37 +01:00
Wim Taymans
9a8a8e72c8 structure: fix for api update 2011-11-02 09:06:37 +01:00
Wim Taymans
9c14280b1d make some more things compile again 2011-10-27 19:00:52 +02:00
Wim Taymans
fc4684f4c6 fix compilation 2011-10-27 16:03:17 +02:00
Marc Leeman
98075ad70d set colour masks for video/x-raw-rgb in rtpvrawdepay 2011-10-14 09:32:47 +02:00
Wim Taymans
a5cc912140 Merge branch 'master' into 0.11 
Conflicts:
	ext/jpeg/gstjpegdec.c
	gst/rtp/gstrtpvrawpay.c
 2011-10-13 08:58:06 +02:00
Edward Hervey
1b56d40170 rtpvrawpay: Only use 24 LSB for depth=24 RGB caps 
... and indent the masks for clarity
 2011-10-12 11:26:50 +02:00
Sjoerd Simons
bf65acf11f gstrtpg722pay: Compensate for clockrate vs. samplerate difference 
The RTP clock-rate used for G722 is 8000, even though the samplerate is
16000. Compensate for this by pretending G722 has 8 bits per sample
instead of the 4 bits as if it were a codec that ran at half the speed,
but with twice the number of bits. Fixes #661376
 2011-10-10 21:50:28 +01:00
Wim Taymans
87fbd1e784 Merge branch 'master' into 0.11 
Conflicts:
	common
	ext/pulse/pulsesink.c
	ext/soup/gstsouphttpclientsink.c
	gst/audioparsers/gstaacparse.c
	gst/audioparsers/gstac3parse.c
	gst/rtp/gstrtph264depay.c
	gst/rtpmanager/gstrtpjitterbuffer.c
	gst/rtpmanager/rtpjitterbuffer.c
	gst/rtsp/gstrtspsrc.c
	sys/ximage/gstximagesrc.c
 2011-09-28 12:44:59 +02:00
Mark Nauwelaerts
fd757890eb rtph264depay: improve downstream flow return feedback to upstream 
... although basertpdepay does not really make it easy/possible to do so
all the way.
 2011-09-20 14:14:39 +02:00
Wim Taymans
83ea243000 Merge branch 'master' into 0.11 
Conflicts:
	common
 2011-09-06 16:37:03 +02:00
Wim Taymans
33f18b8ea4 Merge branch 'master' into 0.11 
Conflicts:
	gst/audioparsers/gstamrparse.c
	gst/isomp4/qtdemux.c
 2011-09-06 16:06:25 +02:00
Mark Nauwelaerts
06f8e356a6 rtpmp4gdepay: improve bogus interleaved index compensating 
Patch by <gudake@gmail.com>

Fixes #654585.
 2011-09-06 13:20:23 +02:00
Olivier Crête
d4778dbe43 rtph263ppay: Set H263-2000 if thats what the other side wants 
The static caps states this element supports H263-2000, but setcaps never
sets it, so it was lie.

See https://bugzilla.gnome.org/show_bug.cgi?id=577784
 2011-09-05 12:58:55 +02:00
Wim Taymans
24df106272 mp2t: fix encoding name according to RFC3551 2011-08-31 18:45:15 +02:00
Wim Taymans
18065ac823 port to new video flags 2011-08-25 16:41:23 +02:00
Wim Taymans
60f0e44bf6 video: port to new colorimetry info 2011-08-23 19:09:31 +02:00
Wim Taymans
9d6371405e fourcc: remove fourcc from caps 2011-08-22 12:24:15 +02:00
Wim Taymans
77ad0a1363 port more elements to new audio caps and API 2011-08-19 14:01:45 +02:00
Wim Taymans
ee2aa25e04 port to new API 2011-08-03 18:37:27 +02:00
Wim Taymans
4121021bb2 Merge branch 'master' into 0.11 
Conflicts:
	ext/pulse/pulsesink.c
	ext/pulse/pulsesrc.c
	gst/audioparsers/gstac3parse.c
	gst/rtp/gstrtph264depay.c
	gst/rtp/gstrtph264pay.c
	gst/rtpmanager/gstrtpssrcdemux.c
 2011-08-03 18:25:30 +02:00
Robert Krakora
f7893b8721 rtpjpegpay: Add support for H.264 payload in MJPEG container 
See http://www.quickcamteam.net/uvc-h264/USB_Video_Payload_H.264_0.87.pdf

Fixes bug #655530.
 2011-08-03 10:09:42 +02:00
Wim Taymans
5771056ed5 rtpvorbispay: fix porting error 2011-08-02 11:51:45 +02:00
Wim Taymans
49af68ebf4 -good: fix for bufferpool API change 2011-07-29 17:27:07 +02:00
Sjoerd Simons
4c73439ee3 rtph264depay: Cope with FU-A E bit not being set 
Some h264 payloaders are unfortunately buggy and don't correctly set the
E bit in FU-A NAL when they have ended. Work around this by assuming
such a fragmentation unit has ended when there was no packet loss and a
new NAL is started
 2011-07-27 18:18:13 +01:00
Wim Taymans
3e089bd7a9 rtp: fix compilation 2011-07-26 17:45:01 +02:00
Olivier Crête
2591a882ae rtph264depay: Complete merged AU on marker bit 
The marker bit on a RTP packet means the AU has been completed, so push it out
immediately to reduce the latency.

https://bugzilla.gnome.org/show_bug.cgi?id=654850
 2011-07-21 17:11:08 +02:00
Olivier Crête
118a7cc36a rtph264pay: Only set the marker bit on the last NALU of a multi-NALU access unit 
An access unit could contain multiple NAL units, in that case, only the last
RTP packet of the last NALU should have its marker bit set.

https://bugzilla.gnome.org/show_bug.cgi?id=654850
 2011-07-21 17:11:06 +02:00
Mark Nauwelaerts
471904032d rtph264depay: reset upon FLUSH_STOP 
... which is particularly needed when merging NAL units, where not resetting
would lead to output of an older (pre-flush) AU (with unintended timestamp).
 2011-07-18 14:32:26 +02:00
Wim Taymans
9c087d7d85 Merge branch 'master' into 0.11 2011-07-15 17:06:39 +02:00
Olivier Crête
87c7f303b0 rtppcmApay/depay: Static clock rates on static payloads, dynamic on dynamic 
Partially reverts 397dc60b
 2011-07-14 20:13:01 -04:00
Olivier Crête
57a832cbb1 rtph264pay: Implement getcaps 
Convert profile-level-id from RTP caps into video/x-h264 style caps (with profile and level)
 2011-07-13 14:10:35 -04:00
Mark Nauwelaerts
eb82a50bd1 rtp: port remaining to 0.11 2011-07-10 21:50:19 +02:00
Wim Taymans
cc65bff7c1 Merge branch 'master' into 0.11 
Conflicts:
	configure.ac
	docs/plugins/inspect/plugin-esdsink.xml
	docs/plugins/inspect/plugin-gconfelements.xml
 2011-06-21 18:24:41 +02:00
Mark Nauwelaerts
3daf1ecc21 rtpmp4adepay: fix output buffer timestamps in case of multiple frames 2011-06-21 15:15:33 +02:00
Wim Taymans
3c889415a3 rtp: port some more (de)payloader 2011-06-13 17:14:00 +02:00
Wim Taymans
9a54175e9f rtp: port to 0.11 2011-06-13 16:33:46 +02:00
Wim Taymans
b0fbb1725f rtp: fix for API changes in the base classes 2011-06-13 13:25:49 +02:00
Wim Taymans
0b1bdcf7cb Merge branch 'master' into 0.11 
Conflicts:
	sys/ximage/ximageutil.c
 2011-06-02 18:51:29 +02:00
Marc Leeman
ff1c05d876 rtpmp4vpay: Deprecated send-config property and replace by config-interval 
Fixes bug #622412.
 2011-05-26 12:22:52 +02:00
Wim Taymans
d89790d545 Merge branch 'master' into 0.11 
Conflicts:
	gst/avi/gstavidemux.c
	gst/rtp/gstrtpac3depay.c
	gst/rtp/gstrtpg726depay.c
	gst/rtp/gstrtpmpvdepay.c
	gst/videofilter/gstgamma.c
 2011-05-24 17:34:19 +02:00
Mark Nauwelaerts
397dc60b71 pcmudepay: allow variable sample rate 2011-05-24 13:13:55 +02:00
Mark Nauwelaerts
f335fee99e pcmadepay: allow variable sample rate 2011-05-24 13:13:52 +02:00
Stefan Kost
d122ea0122 rtp: fix static array overruns in a nicer way 
Use G_N_ELEMENTS instead of hard-coding the array size.
 2011-05-20 10:34:47 +03:00
Stefan Kost
5792d3b9c0 rtp: fix static array overruns 
Yes array[10] has elements from 0...9.
 2011-05-20 00:53:44 +03:00
Jose Antonio Santos Cadenas
9d32243671 rtp: Fix segmentation fault processing payload buffers 
This commit checks if the value returned by
gst_rtp_buffer_get_payload_buffer and
gst_rtp_buffer_get_payload_subbuffer is NULL before using it.
 2011-05-18 15:25:24 +02:00
Wim Taymans
31ffc671f2 rtpgstpay: fix buffer leak 2011-04-26 16:04:07 +01:00
Wim Taymans
eb84592cad rtpgstpay: fix buffer leak 2011-04-26 15:58:12 +02:00
Wim Taymans
9a96783abb rtp: port some more elements 2011-04-25 18:14:45 +02:00
Wim Taymans
bf9b4f8362 rtp: port more to 0.11 2011-04-25 17:27:40 +02:00
Wim Taymans
60db07b4bb rtp: port some more (de)payloaders 2011-04-25 13:16:58 +02:00
Wim Taymans
4aa6ca5578 port more plugins to 0.11 2011-04-18 10:54:43 +02:00
Wim Taymans
7555d0949f Merge branch 'master' into 0.11 
Conflicts:
	android/apetag.mk
	android/avi.mk
	android/flv.mk
	android/icydemux.mk
	android/id3demux.mk
	android/qtdemux.mk
	android/rtp.mk
	android/rtpmanager.mk
	android/rtsp.mk
	android/soup.mk
	android/udp.mk
	android/wavenc.mk
	android/wavparse.mk
	configure.ac
 2011-04-18 10:23:45 +02:00
Tim-Philipp Müller
f325935314 pulse, speexenc, rtpgsmpay: don't use g_assert() for error handling 
Don't use g_assert() for error handling, even if they're highly unlikely.
Either we *know* that something can't happen, in which case we
should just not handle it, or we think something can happen, but it is
very very unlikely that it will ever happen, in which case we should
handle it like any other error instead of asserting.

g_assert() is best left for conditions we have control of, like checking
internal consistency of our code, not checking return values of external
code.

Fixes a bunch of warnings when compiling with -DG_DISABLE_ASSERT:
gstrtpgsmpay.c: In function 'gst_rtp_gsm_pay_handle_buffer':
gstrtpgsmpay.c:130:17: warning: variable 'rtpgsmpay' set but not used
gstspeexenc.c: In function 'gst_speex_enc_encode':
gstspeexenc.c:904:19: warning: variable 'written' set but not used
pulsesink.c: In function 'gst_pulsesink_change_state':
pulsesink.c:2725:9: warning: variable 'res' set but not used
pulsesrc.c: In function 'gst_pulsesrc_change_state':
pulsesrc.c:1253:7: warning: variable 'e' set but not used
 2011-04-16 18:15:43 +01:00
Robert Swain
5b18c652fb rtp, rtpmanager: Address unused but set variables 
GCC 4.6.x spits warnings about variables that are unused but set. Such
variables have been removed where trivial but with comments left behind
for informational purposes in some cases.

gst_rtp_session_chain_recv_rtcp () was changed in commit 490113d4
to always return GST_FLOW_OK instead of the return value of
rtp_session_process_rtcp (), so we'll keep it that way.
 2011-04-16 12:49:16 +01:00
Thibault Saunier
b541208b77 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 01:20:11 +02:00
Haakon Sporsheim
fd545e260d rtpgstpay: declare frag_offset to hold 32bits. 
As specified in documenation above and below.

https://bugzilla.gnome.org/show_bug.cgi?id=646954
 2011-04-09 23:14:18 +01:00