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5475 commits

Author SHA1 Message Date
Wim Taymans
7e43e99726 videocrop: fix compilation 2011-10-16 15:32:50 +02:00
Wim Taymans
ddd58a4035 Merge branch 'master' into 0.11 
Conflicts:
	gst/rtp/gstrtpvrawdepay.c
 2011-10-16 15:26:38 +02:00
Arun Raghavan
cc7aeb3f33 videomixer2: Fix a leak 
Buffers weren't being unref'ed in one case inside, causing memory usage
to blow up.
 2011-10-14 13:10:49 +05:30
Marc Leeman
98075ad70d set colour masks for video/x-raw-rgb in rtpvrawdepay 2011-10-14 09:32:47 +02:00
Thiago Santos
0196fb4668 aspectratiocrop: Port to 0.11 2011-10-13 15:37:47 -03:00
Thiago Santos
04080866aa videocrop: Port to 0.11 2011-10-13 15:37:47 -03:00
Arun Raghavan
4d3ee9005c videomixer2: Fix incorrect gst_buffer_replace() call 
This got exposed when gst_buffer_replace() was changed from a macro to a
function.
 2011-10-13 16:59:50 +05:30
Edward Hervey
d4a2a46606 rtpssrcdemux: Fix wrong usage of gst_iterator_filter 
It takes a GValue* as the user_data.

And don't forget to unref the demuxer before returning.
 2011-10-13 09:34:04 +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
René Stadler
26d0812543 matroskamux: fix segment handling, so we actually use running time 
gst_matroska_mux_best_pad adjusts the buffer timestamp to running time using
the segment stored in the pad's collect data. However, the event handler didn't
pass the newsegment event on to collectpads' handler, so this segment was never
updated at all.

Re-fixes bug #432612.
 2011-10-11 14:58:43 +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
Tim-Philipp Müller
ad245a0dc2 matroska-demux: don't leak audio codec_data buffer 2011-10-10 19:02:58 +01:00
Edward Hervey
919dcf405d alpha: Don't use start() vmethod 
The only thing we're doing is initializing parameters ...
* which won't work because we don't have upstream/downstream caps
* which will be initialized when ::set_caps() is called
 2011-10-10 17:42:02 +02:00
Wim Taymans
0577e069c2 id3demux: port to 0.11 2011-10-10 13:22:12 +02:00
Wim Taymans
31180790d6 icydemux: port to 0.11 2011-10-10 12:54:22 +02:00
Tim-Philipp Müller
8c762dabb2 Merge remote-tracking branch 'origin/master' into 0.11 2011-10-09 16:29:05 +01:00
Tim-Philipp Müller
309b5fa0c1 qtdemux: update for __gst_debug_min name change 2011-10-09 16:25:15 +01:00
Thiago Santos
ca417fd376 qtmux: Fix memory leak on atoms recovery function 
Remember to free the ftyp data after writing it to a file.

Fixes #660969
 2011-10-09 11:18:18 -03:00
Wim Taymans
94021224fc qtmux: report new bits 2011-10-06 12:26:33 +02:00
Wim Taymans
586ef0babd Merge branch 'master' into 0.11 
Conflicts:
	ext/speex/gstspeexdec.c
	ext/speex/gstspeexenc.c
	gst/isomp4/atoms.c
	gst/isomp4/gstqtmux.c
 2011-10-06 12:23:39 +02:00
Vincent Penquerc'h
be82dd8e3a matroskademux: improve segment handling with non-zero starting timestamp 
... as well as related items, such as seeking and position reporting.

https://bugzilla.gnome.org/show_bug.cgi?id=659808
 2011-10-05 14:34:55 +02:00
Thiago Santos
535f92a0a4 qtmux: update esds atom under wave atom for aac bitrates 
AAC in mov format puts an ESDS atom inside of a WAVE atom in
STSD atom, we need to update the bitrate on this ESDS. This patch
fixes it.
 2011-09-30 13:05:24 -03:00
Thiago Santos
31acc88b39 qtmux: Also update btrt atom 
When rewriting bitrates, also update the btrt atom under stsd
 2011-09-30 13:05:24 -03:00
Thiago Santos
7a143ea94f qtmux: Calculate average bitrate for streams 
Calculate and use average bitrate for streams when no
bitrate tag was received
 2011-09-30 12:43:13 -03:00
Thiago Santos
4737090594 qtmux: Avoid a buffer metadata copy if possible 
If first_ts is 0 there is no need to subtract, so we might
skip some copying to make the buffer metadata writable.
 2011-09-30 12:43:13 -03:00
Wim Taymans
2e069225b9 Merge branch 'master' into 0.11 2011-09-28 16:18:54 +02:00
Vincent Penquerc'h
671b56f9da matroskademux: ensure minimal alignment for audio/x-raw-* buffers 
Since matroskademux will attempt to push unaligned buffers,
downstream might have trouble with those, especially if downstream
uses ORC, such as audioconvert.

Ensure we push buffers aligned to the basic type at least for
those raw buffers.

https://bugzilla.gnome.org/show_bug.cgi?id=659798
 2011-09-28 12:49:42 +02:00
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
Raimo Järvi
827c3aa14b goom2k1: Fix compiler warnings on 64 bit mingw-w64 
Fixes bug #660294.
 2011-09-28 00:18:15 +01:00
Julien Isorce
2131a3b7f8 ac3parse: correctly check for ac3/e-ac3 switch 
https://bugzilla.gnome.org/show_bug.cgi?id=659943
 2011-09-23 16:26:50 +01:00
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
Ha Nguyen
931020158e rtpbin: Fix a leaked clock for each buffering message 
Fixes bug #659237.
 2011-09-19 14:05:26 +02:00
Mark Nauwelaerts
d959bb6041 qtdemux: parse embedded ID32 tags 2011-09-19 12:11:45 +02:00
Mark Nauwelaerts
e2179cbb74 rtpsession: avoid source premature timing out 
Use slightly adjusted sender interval to determine sender timeout rather than
our own sender side interval (which may have been forced small).
 2011-09-19 11:56:44 +02:00
Mark Nauwelaerts
f65d4c8300 rtpsession: avoid timing out source too quickly 
... following a PAUSE/PLAY cycle, particularly applicable when operating
with a short RTCP interval (possibly forced so server-side).
 2011-09-19 11:56:44 +02:00
Mark Nauwelaerts
77ebd33991 rtpjitterbuffer/rtpbin: relax dropping rtcp packets 
... to at least having it trigger a/v synchronization, possibly without
using provided values which are still not considered sane
(as previously dropped).
 2011-09-19 11:56:44 +02:00
Mark Nauwelaerts
adfe7d0467 rtpjitterbuffer: some more reset when clearing pt map 
... which in particular caters for some more reset following a possible
rtsp PLAY.
 2011-09-19 11:56:44 +02:00
Mark Nauwelaerts
81fc784163 rtspsrc: do not set elements to PLAYING when doing seek in PAUSED 2011-09-19 11:56:44 +02:00
Mark Nauwelaerts
915db26029 rtpjitterbuffer: only reset skew on gap if input ts available 2011-09-19 11:56:44 +02:00
Mark Nauwelaerts
1e17e10f75 rtpjitterbuffer: check some more for possible rtp timestamp discontinuity 
... when operating in non slave mode, and reset if detected.
This should avoid some (large) bogus outgoing timestamp due to jumps
in rtp time, as result of PAUSE/PLAY or seek or ...
 2011-09-19 11:56:40 +02:00
Mark Nauwelaerts
8599801cae rtspsrc: switch to rtp time based syncing when guessed appropriate 2011-09-19 11:52:08 +02:00
Mark Nauwelaerts
9c95072048 rtpbin: alternative inter-stream syncing methods 
... at least if not syncing to NPT time:
* either sync using RTCP SR data (as currently)
* only perform the above once using initial RTCP SR packets
* discard RTCP and sync by equating provided stream's clock-base rtptime,
  as provided by jitterbuffer (typically obtained from RTP-Info in RTSP).
 2011-09-19 11:52:03 +02:00
Mark Nauwelaerts
4b7301e4d1 rtpjitterbuffer: also provide clock-base to sync signal 2011-09-19 11:52:00 +02:00
Mark Nauwelaerts
f29c253934 rtpbin: allow configurable rtcp stream syncing interval 
... rather than necessarily syncing at each RTCP SR.
 2011-09-19 11:51:57 +02:00
Mark Nauwelaerts
afd26f0078 rtpsession: trigger reconsideration if rtcp interval set 2011-09-19 11:51:50 +02:00
Mark Nauwelaerts
3e33a7a09f rtspsrc: configure rtcp interval if provided 
... in PLAY response.
 2011-09-19 11:51:47 +02:00
Lasse Laukkanen
056e9188b1 isomp4: Fix allowing zero duration tracks 
https://bugzilla.gnome.org/show_bug.cgi?id=637486
 2011-09-19 11:18:27 +02:00
Vincent Penquerc'h
3319737e5c udpsrc: error out when no protocol is specified in the uri 
It is certainly better than to crash.

https://bugzilla.gnome.org/show_bug.cgi?id=658178
 2011-09-19 10:16:38 +02:00
Branko Subasic
11b0a0effc matroskademux: Avoid sending EOS when in paused state 
Changed the ebml reader's gst_ebml_peek_id_length() function so
that it returns the actual reason for why the peek failed, instead
of (almost) always returning GST_FLOW_UNEXPECTED. This prevents
the pulling task from sending EOS when doing a flushing seek.
 2011-09-16 15:18:48 +02:00
Vincent Penquerc'h
26ae233035 matroskademux: fix stuttering A/V 
Someone got had by implicit promotion to unsigned in ops with
a signed and an unsigned value.

https://bugzilla.gnome.org/show_bug.cgi?id=659153
 2011-09-15 17:29:00 +01:00
Vincent Penquerc'h
352bab2ef7 navseek: toggle pause/play on space bar 
A useful thing to have.

https://bugzilla.gnome.org/show_bug.cgi?id=659065
 2011-09-14 21:32:42 +01:00
David Svensson Fors
682ae32f6f matroskademux: configurable timestamp gap handling 
matroskademux performs segment tricks to skip gaps in streams,
notably at start for non 0 based files.  There may however be
cases when full presentation (including intermediate gaps) is
desired, so a property allows to configure as of which gap
to act (or not at all).

API: GstMatroskaDemux::max-gap-time

Fixes #659009.
 2011-09-14 14:49:36 +02:00
Thiago Santos
261d11a6d7 qtmux: Fix ctts generation for streams that don't start at 0 timestamps 
Subtract the first timestamp of a stream from all input buffers to
get 0-based timestamps for creating a sane ctts table. Without this
patch the ctts could have larger values than needed, causing the
playback to have a delay at startup.

As the first timestamp is only found after a few buffers are queued
(due to possible reordered buffers), once we find the first timestamp
we subtract it from all buffers on the queue, from that point on,
all buffers have their timestamps subtract when they are collected.

https://bugzilla.gnome.org/show_bug.cgi?id=658659
 2011-09-12 07:37:10 -03:00
Alessandro Decina
aea09188dc flvmux: don't release request pads going PAUSED->READY 
Don't release request pads but just reset them. This makes pipelines using
flvmux reusable.
 2011-09-12 10:00:59 +02:00
Vincent Penquerc'h
d17d13219c ac3parse: use bsid 9 and 10 to control sample rate 
See http://matroska.org/technical/specs/codecid/index.html

The spec is silent about this though...

https://bugzilla.gnome.org/show_bug.cgi?id=658546
 2011-09-09 13:59:31 +02:00
Mark Nauwelaerts
95b5ece2c9 rtspsrc: ensure some initial state variable setup 
... which might otherwise be skipped if the PLAY command is issued before
the OPEN command had a chance to actually be acted upon.

Fixes #657376.
 2011-09-09 10:53:08 +02:00
Mark Nauwelaerts
ef1ad78eee matroskademux: tweak gap handling 
... so as to avoid buffers before and after gap to have identical running time.
 2011-09-08 15:10:43 +02:00
Thiago Santos
ed3adece77 qtmux: remove one G_UNLIKELY for user property 
Using G_UNLIKELY on user properties isn't nice, specially when
that is the default option.
 2011-09-07 11:46:07 -03:00
Andoni Morales Alastruey
782fc78d57 matroskamux: handle GstForceKeyUnit event 
... by starting a new cluster after forwarding event.

Fixes #644154.
 2011-09-07 14:51:56 +02:00
Sebastian Dröge
c29069fd11 ac3parse: Add Converter to the classification because it can convert between different alignments 
This allows decodebin2 to let it negotiate properly.
 2011-09-07 12:11:39 +02:00
Sebastian Dröge
786d35f53f audioparsers: Improve src template caps 
Remove the parsed/framed fields and add all fields to the template
caps that always exist.
 2011-09-07 12:10:48 +02:00
Mark Nauwelaerts
625e7a6143 aacparse: parse codec_data to determine number of samples per frame 
Fixes #656734.
 2011-09-07 11:20:03 +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
aa0ae490d0 matroskamux: make default duration check less sensitive 
Frame duration might vary for 1 usecond, in this case matroskamux
decides to create BLOCKGROUP instead of SIMPLEBLOCK.

Convert duration to timecodescale which is (typically) less precise, and
then also allow the difference of 1/-1 to arrange for less sensitive check.

Based on patch by Alexey Fisher <bug-track@fisher-privat.net>

Fixes #653080.
 2011-09-06 15:09:13 +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
Wim Taymans
e204c5934c -good: port to new audio caps 2011-09-06 13:16:27 +02:00
Mark Nauwelaerts
b9a54a38b0 amrparse: fix and streamline valid frame checking 
... to handle various combinations of sync or not, and sufficient data
or not as might be expected.

Fixes #650714.
 2011-09-05 15:51:48 +02:00
Mark Nauwelaerts
4b8ead4340 qtdemux: fragmented support; avoid adjustment for keyframe seek 
... since all index data may not yet be available at that time.
 2011-09-05 14:56:18 +02:00
Mark Nauwelaerts
08d25a69d5 qtdemux: fragmented support; mark all audio track samples as keyframe 2011-09-05 14:56:18 +02:00
Brian Li
a3e9b676c0 qtdemux: fragmented support; properly init return variable value 
Fixes #655918.
 2011-09-05 14:56:08 +02:00
Mark Nauwelaerts
2603c2079d rtspsrc: add gtk-doc for new short-header property 2011-09-05 13:32:17 +02:00
Marc Leeman
ce276d903c rtspsrc: allow sending short RTSP requests to a server 
Some encoders (Arecont) do not like the long OPTIONS sent at startup as sent by
GStreamer, but do accept the short header as sent by Live555.

This patch makes the extending the request optional by adding a property
(short-header).

Fixes #655805.

API: GstRTSPSrc:short-header
 2011-09-05 13:26:06 +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
Olivier Crête
b2e8362767 rtpsession: Initialise the last_keyframe_request variable 2011-09-02 19:24:46 -04:00
Peter Korsgaard
d73410c4af multiudpsink: make add/remove/clear/get-stats action signals 
http://bugzilla.gnome.org/show_bug.cgi?id=657830

Signed-off-by: Peter Korsgaard <jacmet@sunsite.dk>
 2011-09-01 22:54:27 +01:00
Wim Taymans
24df106272 mp2t: fix encoding name according to RFC3551 2011-08-31 18:45:15 +02:00
Mark Nauwelaerts
e15d29ffe4 qtdemux: push mode; perform some extra checks prior to upstream seeking 2011-08-30 14:24:04 +02:00
Mark Nauwelaerts
9de9d7e4d4 qtdemux: push mode; fix buffered streaming 
That is, in case where no seek is peformed to moov, but preceding
limited mdat is buffered.
 2011-08-30 14:23:49 +02:00
Wim Taymans
9521326703 shapewipe: port to 0.11 2011-08-30 14:06:12 +02:00
Wim Taymans
ef560b86e2 law: port to 0.11 2011-08-30 12:25:35 +02:00
Wim Taymans
5ad41c7292 alaw: port to 0.11 2011-08-29 19:11:25 +02:00
Wim Taymans
e62d326dc2 goom: fix comment 2011-08-29 19:10:35 +02:00
Mark Nauwelaerts
5ea19b0696 qtdemux: avoid overflow wraparound in timestamp when adding durations 
Do some type juggling to avoid overflow, while still allowing for 'negative'
durations (which would need a wraparound effect).
 2011-08-29 15:16:16 +02:00
Wim Taymans
75e153bb13 allocation: fix for vmethod changes 2011-08-26 14:20:49 +02:00
Wim Taymans
18065ac823 port to new video flags 2011-08-25 16:41:23 +02:00
Wim Taymans
e9df54819c Merge branch 'master' into 0.11 2011-08-24 14:16:44 +02:00
Vincent Penquerc'h
f3fc3e1f69 aacparse: only require two frames in a row when we do not have sync 
This avoids a single bit error dropping two frames unnecessarily.
The two consecutive frames check is still required when we don't
have sync.

https://bugzilla.gnome.org/show_bug.cgi?id=657080
 2011-08-24 08:26:31 +02:00
Wim Taymans
60f0e44bf6 video: port to new colorimetry info 2011-08-23 19:09:31 +02:00
Wim Taymans
d6908f1a2d Merge branch 'master' into 0.11 2011-08-22 13:10:07 +02:00
Wim Taymans
9d6371405e fourcc: remove fourcc from caps 2011-08-22 12:24:15 +02:00
David Schleef
88557c4792 breakmydata: element is not passthrough 2011-08-21 15:15:14 -07:00
David Schleef
2a83da13fc multifilesrc: quiet debugging 2011-08-21 15:15:14 -07:00
David Schleef
0446787e65 deinterlace: change field handling through methods 
This likely breaks stuff.  The good: all of the methods now create
field images aligned with input frames, without timestamp mangling.
The bad: this touches a lot of code, much of which is hairy and in
need of cleanup.  However, at this point we can reasonably create a
PSNR-based test.
 2011-08-21 15:15:14 -07:00
Alessandro Decina
ad996feb28 multifilesink: reset ->streamheaders to NULL on _stop 
Fixes invalid memory access reusing multifilesink
 2011-08-21 14:41:59 +02:00
Wim Taymans
5e3a52f1d9 cutter: bring cutter somewhat into this millennium 2011-08-20 10:46:18 +02:00
Wim Taymans
c1abdd7626 rg: fix caps 2011-08-19 16:27:20 +02:00
Wim Taymans
445bf71bd1 port to more audio api changes 2011-08-19 16:09:48 +02:00
Wim Taymans
77ad0a1363 port more elements to new audio caps and API 2011-08-19 14:01:45 +02:00
Wim Taymans
90f5b31b4b port to new audio API and caps 2011-08-19 11:49:44 +02:00
Wim Taymans
135b41cb5b Merge branch 'master' into 0.11 2011-08-18 19:37:39 +02:00
Wim Taymans
09b15d7dfe port to new audio caps. 2011-08-18 19:21:07 +02:00
Vincent Penquerc'h
e032d26674 matroskademux: ensure no-more-pads is always emitted 
In particular, do so even if failing to read while prerolling,
such as when reading from a partial file (eg, while it is being
downloaded).

This fixes a wedge in playbin2.

https://bugzilla.gnome.org/show_bug.cgi?id=651965
 2011-08-18 11:30:07 +02:00
Wim Taymans
48e47ad702 Merge branch 'master' into 0.11 2011-08-17 11:17:38 +02:00
Vincent Penquerc'h
f8a9f5bc1c spectrum: avoid crashing by resetting the correct number of channels 
https://bugzilla.gnome.org/show_bug.cgi?id=656606
 2011-08-16 22:44:07 +01:00
Wim Taymans
4bb2b140e9 Merge branch 'master' into 0.11 
Conflicts:
	sys/v4l2/v4l2src_calls.c
 2011-08-16 18:35:53 +02:00
Vincent Penquerc'h
6ac7ad8a2c flacparse: fix off by one in frame size check 
Yes, I was tracking another bug and the small test file I generated
to test with improbably just happened to trigger this, with a second
and last frame of 1615 bytes.

https://bugzilla.gnome.org/show_bug.cgi?id=656649
 2011-08-16 13:25:30 +01:00
Tim-Philipp Müller
5866c3a413 id3demux: remove specs from git as well now that parsing code is in -base 2011-08-14 20:46:01 +01:00
Mark Nauwelaerts
1ca89389e4 id3demux: use -base provided id3 tag parsing 
https://bugzilla.gnome.org/show_bug.cgi?id=654388
 2011-08-13 23:19:32 +01:00
Stefan Kost
a1b1d19105 qtdemux: initialize bitrate variable and reset for each loop 
Don't check eventually unset variable and don't accidentially use values from last
cycle.
 2011-08-12 16:32:58 +02:00
Edward Hervey
d08e0ccc48 rtspsrc: Properly error out if SDP contains no streams 
Also fixes unitialized variable error on macosx.
 2011-08-09 11:28:17 +02:00
Wim Taymans
5fa0c28bc4 isomp4: fixup after small api changes 
Port to recently changed api so that it compiles again.
 2011-08-05 12:07:50 +02:00
Edward Hervey
eb8075111d y4menc: Now depends on libgstvideo 2011-08-05 11:32:45 +02:00
Wim Taymans
84371e4066 goom: port to new caps 2011-08-04 16:32:39 +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
Jan Schmidt
1438bf26ac matroska: Register new debug category 
Register the matroskareadcommon debug category when the
plugin is loaded to avoid assertion output when debug is turned on.
 2011-08-03 22:52:07 +10:00
Philippe Normand
0424368cfc qtdemux: soften assertion check on stream size 
https://bugzilla.gnome.org/show_bug.cgi?id=655570
 2011-08-03 10:11:59 +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
Sebastian Dröge
4aa5485cfc effectv: Fix unused but set variable compiler warnings 2011-08-03 08:51:19 +02:00
Tim-Philipp Müller
a1712ad87c docs: fix two more Since: tags 2011-08-02 23:42:58 +01:00
Mart Raudsepp
62cd1215c7 deinterlace: Fix Since tags for fieldanalysis related new properties 
commit c1b100cf9c is after 0.10.29 and 0.10.30 was a branched release.
So fix Since tags from 0.10.29 to 0.10.31 for the new properties.
 2011-08-02 23:38:13 +01:00
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
Mark Nauwelaerts
c03648c8bb rtpsession: properly init rtcp_min_interval 2011-07-29 12:08:42 +02:00
Mark Nauwelaerts
3a98f6f0fd rtpssrcdemux: keep a ref on the src pad while using it 
Prevent a possible race if clear_ssrc() is called between getting the pad and
doing the push.

Based on patch by <olivier.crete@collabora.com>

https://bugzilla.gnome.org/show_bug.cgi?id=650916
 2011-07-28 14:51:01 +02:00
Olivier Crête
c7b9b98648 rtpssrcdemux: Make the pads lock recursive and hold it across the signal emit 
We need to keep the lock held because we don't want a push before the "new-ssrc-pad"
handler has completed. But we may want to push an event from inside that handler, hence
the recursive mutex.

https://bugzilla.gnome.org/show_bug.cgi?id=650916
 2011-07-28 14:50:59 +02:00
Olivier Crête
e26b5391c2 rtpssrcdemux: Use PADs lock 
https://bugzilla.gnome.org/show_bug.cgi?id=650916
 2011-07-28 14:50:57 +02:00
Tim-Philipp Müller
b843f8f99c gst: udpate for position/duration/convert query API changes 2011-07-28 11:38:31 +01:00
Tim-Philipp Müller
f94ea7299a avidemux: fix compiler warning 
gstavidemux.c: In function 'gst_avi_demux_parse_stream':
gstavidemux.c:1261:24: error: 'data' may be used uninitialized in this function [-Werror=uninitialized]
gstavidemux.c:1204:11: note: 'data' was declared here
 2011-07-28 11:38:31 +01:00
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
Arun Raghavan
89564fcb69 ac3parse: Support switching alignment on-the-fly 
This allows switching of alignment for E-AC3 streams at run-time. This
is requested by downstream elements via a custom event.

https://bugzilla.gnome.org/show_bug.cgi?id=650313
 2011-07-27 20:43:56 +05:30
Wim Taymans
13d0ad188f warp: add stride support 2011-07-27 12:42:21 +01:00
Wim Taymans
3e089bd7a9 rtp: fix compilation 2011-07-26 17:45:01 +02:00
Arun Raghavan
96972eb462 ac3parse: Add support for IEC 61937 alignment 
When pushing out buffers over S/PDIF or HDMI, IEC 61937 payloading
requires each buffer to contain 6 blocks from each substream. This adds
code to collect all the frames needed to meet this requirement before
pushing out a buffer.

https://bugzilla.gnome.org/show_bug.cgi?id=650313
 2011-07-26 10:40:00 +05:30
Olivier Crête
6095d2a3f0 rtpsession: Always send application requested feedback in immediate mode 
Send as many application requested feedback messages in immediate mode, even if they
have already been sent.

https://bugzilla.gnome.org/show_bug.cgi?id=654583
 2011-07-25 17:20:59 +02:00
Olivier Crête
354faabda0 rtpsession: Don't let the computed RTP bandwidth fall too low 
If it falls too low, the computed RTCP bandwidth will be near zero and
the RTCP thread will be stopped.

https://bugzilla.gnome.org/show_bug.cgi?id=654583
 2011-07-25 16:19:00 +02:00
Olivier Crête
4d48109f9d rtpsession: Wait longer to timeout SSRC collision 
Using the current RTCP interval to timeout SSRC collision can lead to
collisions being timed out immediately if a BYE packet is sent because
it is sent immediately, so the interval is 0. This is not what we
want. So just set a static 10 times the default RTCP interval, it
should be enough

https://bugzilla.gnome.org/show_bug.cgi?id=648642
 2011-07-25 16:18:58 +02:00
Mark Nauwelaerts
9764b57b0a rtspsrc: set SOURCE flag at init time 
Fixes #654816.
 2011-07-25 12:44:38 +02:00
Wim Taymans
7ca40d7a53 vertigotv: add stride support 2011-07-25 10:24:33 +01:00
Wim Taymans
83bc5e0765 replay: fix for event handler 2011-07-22 21:26:32 +02:00
Wim Taymans
984a0b54eb fixes for event handler changes 2011-07-22 21:19:45 +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
Alessandro Decina
216dc602c3 multipart: fix compiler warning 2011-07-20 08:53:25 +02:00
Mark Nauwelaerts
1880c4145e auparse: avoid hanging on invalid short input 
... as in such case there is no srcpad yet on which to forward EOS.
 2011-07-19 12:05:51 +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
Mark Nauwelaerts
6c0aec783a multifilesink: do not use g_slist_free_full 
... as that is only in GLib 2.28, which is not yet required at this time.
 2011-07-18 14:31:40 +02:00
Alessandro Decina
072bd74cc4 multifilesink: add max-files property 
Add max-files property to limit the number of files saved on disk.

API: multifilesink::max-files
 2011-07-18 10:21:41 +02:00