Commit graph

636 commits

Author SHA1 Message Date
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
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
Wim Taymans
0024300aa2 rtp: port some pay/depayloaders 2011-04-07 19:04:33 +02:00
David Schleef
e54ba41ff7 rtpvrawpay: Implement interlacing 2011-02-17 18:05:43 -08:00
Wim Taymans
4279aa6a68 theorapay: handle 0 sized packets
Handle 0 sized packets (repeat frame) in the payloader and depayloader.

Fixes #641827
2011-02-14 16:48:06 +01:00
Olivier Crête
8a7a327db7 rtptheoradepay: Request new keyframe on lost packets
Theora can only use the last frame (or the keyframe) as a reference, so in
practice. If we receive a buffer that references an unknown codebook, request
new headers. It probably means that headers were lost.
2011-02-01 18:28:51 +01:00
Wim Taymans
f95c30a413 j2kpay: skip EPH packets
Include EPH markers into the previous chunk of packets.
2011-02-01 16:39:10 +01:00
Olivier Crête
07ebec51f5 rtppcmapay: Rename the class to have the right name
It was name pmca instead of pcma and made debug logs hard to search.
2011-01-31 17:56:43 -05:00
Tim-Philipp Müller
693b3b7e0b h264depay: don't leak codec data buffer in byte-stream=true mode
https://bugzilla.gnome.org/show_bug.cgi?id=640063
2011-01-20 14:10:55 +00:00
Edward Hervey
4decc3aaea rtp: Fix unitialized variables on macosx 2011-01-06 12:29:21 +01:00
Wim Taymans
6b91c5f6e7 vrawdepay: fix length check
Add some more debugging.
Add the length check so we don't cause unneeded warnings.
2011-01-05 15:03:32 +01:00
Wim Taymans
5ed3701a2d mp4adepay: improve timestamps on outgoing packets
Improve parsing of the samplerate.
Parse the framelen so that we can calculate timestamps.
When interpollate the incomming timestamp on outgoing buffers when there are
multiple subframes.

fixes #625825
2010-12-31 13:57:05 +01:00
Wim Taymans
9c2393490f mp4adepay: fix timestamps on buffers 2010-12-30 16:24:46 +01:00
Wim Taymans
756869421c mpvpay: fix flushing and discont
Fix flushing and disconts.
Clean up in state changes.
2010-12-30 16:24:46 +01:00
Tim-Philipp Müller
fafd0b7bc3 rtpjpegdepay: fix framerate parsing for locales that use a comma as floating point
atof() converts strings according to the current locale, but the
framerate string will likely always use a dot as floating point
separator, so use g_ascii_strtod() instead (but also canonicalise
the string before, so we can handle both formats as input).
2010-12-29 14:59:30 +00:00
Wim Taymans
ef0bc7558d gstpay: fix klass, add RTP as a use case 2010-12-23 18:39:52 +01:00
Wim Taymans
5fe6046c20 gstdepay: cleanup the cache 2010-12-23 18:39:52 +01:00
Wim Taymans
7c9b91d2d8 gstpay/depay: add generic gstreamer payloader
Add the beginnings of a generic GStreamer buffers payloader.
2010-12-23 18:39:52 +01:00
Wim Taymans
e13340ccb5 mp4gpay: reset state on flush-stop 2010-12-23 17:06:58 +01:00
Wim Taymans
1dd71cc63f mp4gdepay: flush state on flush-stop 2010-12-23 16:26:07 +01:00
Wim Taymans
6db12cb003 rtpac3pay: add AC3 payloader 2010-12-21 22:34:49 +01:00
Wim Taymans
97993d3119 ac3depay: fix debug category description 2010-12-21 22:17:19 +01:00
Wim Taymans
e2f4fe8d3d mpapay: add debug category 2010-12-21 22:16:42 +01:00
Wim Taymans
f4155f3cf3 rtp: add RTP hint to the klass 2010-12-21 17:23:03 +01:00
Wim Taymans
f357e09ac1 rtp: fix rank of payloaders and depayloaders
Set the payloaders and depayloaders to a reasonable rank.
2010-12-21 17:22:58 +01:00
Wim Taymans
d5c8771b2b vrawdepay: reset depayloader state
Reset the depayloader state on flush-stop.
2010-12-21 15:24:18 +01:00
Wim Taymans
eb99eb5515 mp4pay: use vmethod for intercepting events 2010-12-21 15:23:08 +01:00
Wim Taymans
e47f4487b4 theorapay: clear packet on flush-stop 2010-12-21 13:55:40 +01:00
Wim Taymans
2c6e198157 vorbispay: clear packet on flush-stop 2010-12-21 13:49:41 +01:00
Wim Taymans
1eb0f65f39 mp4gdepay: reset depayloader state 2010-12-21 12:31:44 +01:00
Wim Taymans
e8b8753c90 h264pay: flush adapter on flush-stop 2010-12-21 12:29:58 +01:00
Wim Taymans
6a5e6eac55 mpapay: flush last packets on EOS 2010-12-20 18:50:25 +01:00
Wim Taymans
933a170898 mpapay: reset payloader on state change 2010-12-20 16:51:47 +01:00
Wim Taymans
984849f8fe mpapay: reset payloader on flush
Reset the payloader on a flush event.
Handle DISCONT better.
2010-12-20 16:06:26 +01:00
Mark Nauwelaerts
4c368242c0 rtph264depay: determine output h264 layout using caps negotiation
... thereby (partially) deprecating properties currently controlling whether
or not byte-stream output or NAL/AU alignment (though properties still determine
fallback if nothing specified in caps).

Fixes #606662.
2010-12-17 15:38:27 +01:00
Wim Taymans
b87ec0262b j2kpay: handle EOC correctly
Don't include the next 2 bytes when we are at the end of the data and there are
no more bytes left.
2010-12-16 18:57:27 +01:00
Edward Hervey
34222431aa rtpj2kpay: Initialize all fields
Makes sad compliers happy
2010-12-15 18:21:34 +01:00
Wim Taymans
744472d2ad j2kpay: cleanup header construction
Use a simpler way of constructing the header that doesn't depend on
the endianness.
2010-12-15 16:25:10 +01:00
Wim Taymans
184c4219a7 j2kdepay: add support for buffer lists 2010-12-15 13:12:09 +01:00
Wim Taymans
957eac9579 j2kpay: stop scanning when we reached the end
Stop scanning for markers when we reached the end of the data.
2010-12-14 15:28:40 +01:00
Wim Taymans
acc37e52a7 mp4vpay: we can also accept xvid caps 2010-12-12 15:14:40 +01:00