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5444 commits

Author SHA1 Message Date
Sebastian Dröge
c603bf9891 videomixer2: Use the clip function instead of the prepare_buffer function 2011-10-28 10:40:36 +02:00
Tim-Philipp Müller
d18a578ba4 rtpmanager, v4l2: fix compiler warnings after gst_caps_new_simple() change 2011-10-28 09:06:41 +01:00
Tim-Philipp Müller
b0f43f43d4 qtdemux: fix compiler warnings after gst_caps_new_simple() change 2011-10-28 09:06:28 +01:00
Sebastian Dröge
ae07ebedc9 videomixer2: Use collectpads2 from core 2011-10-28 09:36:17 +02:00
René Stadler
70420b0661 wavenc: port to 0.11 raw audio caps 2011-10-27 19:40:09 +02:00
Wim Taymans
facda96308 Merge branch 'master' into 0.11 
Conflicts:
	gst/flv/gstflvmux.c
 2011-10-27 19:06:06 +02:00
Wim Taymans
9c14280b1d make some more things compile again 2011-10-27 19:00:52 +02:00
Wim Taymans
4b6a226263 Merge branch 'master' into 0.11 
Conflicts:
	ext/pulse/pulseaudiosink.c
	ext/pulse/pulsesink.c
 2011-10-27 16:08:22 +02:00
Wim Taymans
fc4684f4c6 fix compilation 2011-10-27 16:03:17 +02:00
Jan Schmidt
3b03db5e40 deinterlace: Don't pointlessly hold object lock over caps operations 
Avoids a deadlock when getcaps is recursive due to the getcaps being
reflected upstream/downstream. The lock isn't actually protecting
anything here.
 2011-10-28 00:41:45 +11:00
Jan Schmidt
f6255bb8d0 flvmux: add some comments and defines to clarify code. 2011-10-27 00:41:29 +11:00
René Stadler
1b213d905a matroska: refactor ebml-write to be more 0.11 friendly 
Switching to a more 0.11-friendly pattern, where getting the buffer's data
pointer and setting the size many times is less natural. This is of course in
preparation to the upcoming port of the plugin.
 2011-10-21 23:17:16 +02:00
René Stadler
003a14e15a matroska: remove stale floatcast include 
GDOUBLE_TO_BE was moved to core a long time ago.
 2011-10-21 22:41:57 +02:00
René Stadler
6fc85e6c9d matroskamux: fix possible crash with malformed dirac codec_data 
Since size is unsigned, we need to safeguard against wrapping below zero.
 2011-10-21 22:41:57 +02:00
René Stadler
6082f27d48 equalizer: remove avoidable call to gst_object_set_name 2011-10-21 22:33:34 +02:00
René Stadler
dcd493279d deinterlace: remove avoidable call to gst_object_set_name 2011-10-21 22:32:38 +02:00
Stefan Sauer
2468492f02 interfaces: clean up the use of iface and class/klass 2011-10-21 14:58:41 +02:00
René Stadler
5baca05ec3 ebml: small correction to previous commit 
Signal a short read with UNEXPECTED, exactly like the peek_bytes function.
 2011-10-21 10:03:14 +02:00
Edward Hervey
4b5f8ff8d1 ebml: Fix push-based behaviour 
The 'peek' method was completely wrong (!?)
 2011-10-19 13:10:47 +02:00
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