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10613 commits 44 branches 1507 tags 456 MiB
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5390 commits

Author SHA1 Message Date
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