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10562 commits 44 branches 1510 tags 329 MiB
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575 commits

Author SHA1 Message Date
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
Alexey Fisher
9b15f9c6a1 rtpspeexpay: Do not transmitt samples with GAP flag 
If we get GAP samples, there is no need to transmitt it.
In some situations, microphone is muted, we can drop net traffick
usage to ~1 kbit/s. Without patch it will stay ~20 kbit/s
 2011-04-08 13:56:13 +02:00