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5377 commits

Author SHA1 Message Date
Wim Taymans
fc4684f4c6 fix compilation 2011-10-27 16:03:17 +02:00
Stefan Sauer
2468492f02 interfaces: clean up the use of iface and class/klass 2011-10-21 14:58:41 +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