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Author SHA1 Message Date
Wim Taymans
47cbb230e9 audio: move audio interfaces 
Move the audio related interfaces to the audio library.
 2011-11-30 07:57:02 +01:00
Wim Taymans
552e825b4f fix includes for moved interfaces 2011-11-30 07:23:47 +01:00
Wim Taymans
4fb0f98bb9 encoding-profile: small cleanup in docs 2011-11-30 07:23:07 +01:00
Edward Hervey
5bc6ffcd8b video: Don't forget to install moved header files 2011-11-29 19:49:50 +01:00
Edward Hervey
a3b272f0a3 tests: More fixes for moved interfaces 2011-11-29 19:31:55 +01:00
Wim Taymans
871b306fce video: move some interfaces 
Move some interfaces to the video library
 2011-11-29 19:10:01 +01:00
Stefan Sauer
089c760993 adder: fill the audio-info that we use and not some random other one 2011-11-29 14:47:37 +01:00
Stefan Sauer
1cea9c851c adder: unbreak adder 
There was one line too much removed when porting.
 2011-11-29 14:22:19 +01:00
Stefan Sauer
9debd13665 adder: fix deadly setcaps recursion 
Use a flag to avoid calling setcaps until our stack is exhausted. I don't see how this would be useful.
 2011-11-29 10:42:16 +01:00
Tim-Philipp Müller
0d87fd7146 Merge remote-tracking branch 'origin/master' into 0.11 
Conflicts:
	gst-libs/gst/fft/gstffts16.h
 2011-11-28 21:25:11 +00:00
Tim-Philipp Müller
6fe4d31961 Merge commit 'c5544630250ec434e4dafaf17274e83865415120' into 0.11 2011-11-28 21:20:38 +00:00
Tim-Philipp Müller
0c056a04fe Merge commit '4a58223e4c824fedc024af435337a769e8ce593e' into 0.11 2011-11-28 21:20:10 +00:00
Philippe Normand
0a841f6712 fft: Bracket public headers 
This is especially needed if the gstfftw library is used from C++
code.

Fixes #665074
 2011-11-28 20:28:19 +01:00
Philippe Normand
ed5279e3c5 typefindfunctions: Fix compiler warning 2011-11-28 20:10:49 +01:00
Alexey Fisher
36434c20eb typefind: fix build error 
fix build errors:
gsttypefindfunctions.c:248:25: error: 'low' may be used uninitialized in this function [-Werror=uninitialized]
gsttypefindfunctions.c:239:24: error: 'high' may be used uninitialized in this function [-Werror=uninitialized]

Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>
 2011-11-28 18:10:55 +00:00
Sebastian Dröge
f179213aa0 playsinkconvertbin: Fix stupid mistake in last commit 2011-11-28 19:06:57 +01:00
Sebastian Dröge
c1b1e2b44e playsinkconvertbin: Only return the converter caps if we actually have raw caps 
Fixes bug #664818 (hopefully).
 2011-11-28 19:03:54 +01:00
Wim Taymans
5b868bd424 Update for indexable change 2011-11-28 18:24:03 +01:00
Kipp Cannon
4c52f4e625 audioresample: Don't emit DISCONT buffers if no discontinuity happened 
audioresample is derived from GstBaseTransform, and one of
GstBaseTransform's traits is that if the derived element does not
produce an output buffer from some input buffer then the first output
buffer after that gets flaged as a discontinuity, whether or not the
buffer actually is discontinuous from the output buffer that preceded
it. When downsampling, the audioresample element requires more than
one input sample for each output sample, and if the ratio of input to
output sample rates is high enough and the input buffers short enough
it can come to pass that the resampler does not receive enough samples
on its input to produce any output.  Currently the resampler returns
GST_BASE_TRANSFORM_FLOW_DROPPED from the transform() method in this case,
causing the next buffer to be flagged as a discontinuity. If subsequent
elements in the pipeline reset themselves on disconts, this can cause
clicks and other undesireable behaviour.

Fixes bug #665004.
 2011-11-28 18:03:22 +01:00
Wim Taymans
468d1dde89 audio: update for clock provider API change 2011-11-28 17:51:41 +01:00
Vincent Penquerc'h
e67aa28de9 typefind: typefind UTF-16 and UTF-32 
This avoids the MP3 typefinder from getting the highest score
every time it thinks there's something it might possibly be
able to parse.

https://bugzilla.gnome.org/show_bug.cgi?id=607619
 2011-11-28 15:58:29 +00:00
Wim Taymans
b4cdf008dd fix for element flag cleanups 2011-11-28 16:55:32 +01:00
Vincent Penquerc'h
c554463025 Revert "theoradec: move the QoS logic to libgstvideo" 
This reverts commit 149a4ce390.

*grumble* I managed to merge something I did not mean to.
 2011-11-28 13:27:29 +00:00
Vincent Penquerc'h
ea78b060a7 Revert "libgstvideo: add a new API to handle QoS events and dropping logic" 
This reverts commit eb03323fb6.

*grumble* I managed to merge something I did not mean to.
 2011-11-28 13:26:53 +00:00
Vincent Penquerc'h
96374054ac various: fix pad template leaks 
https://bugzilla.gnome.org/show_bug.cgi?id=662664
 2011-11-28 13:09:02 +00:00
Vincent Penquerc'h
149a4ce390 theoradec: move the QoS logic to libgstvideo 
https://bugzilla.gnome.org/show_bug.cgi?id=658241
 2011-11-28 12:34:43 +00:00
Vincent Penquerc'h
eb03323fb6 libgstvideo: add a new API to handle QoS events and dropping logic 
https://bugzilla.gnome.org/show_bug.cgi?id=658241
 2011-11-28 12:34:43 +00:00
Mark Nauwelaerts
4a58223e4c audioencoder: elaborate some documentation 2011-11-28 11:37:33 +01:00
Mark Nauwelaerts
9f57d91137 audiodecoder: add some documentation 2011-11-28 11:37:27 +01:00
Mark Nauwelaerts
856a5dd581 audiodecoder: really discard NULL decoded frame altogether 
... including any timestamp, rather than having that one influence base_ts.
 2011-11-28 11:37:23 +01:00
Stefan Sauer
f70ca6d4cb alsasrc: style fix 
Use timestamp==0 instead of mixing it with !timestamp style checks.
 2011-11-28 10:55:39 +01:00
Stefan Sauer
8154b69112 alsasrc: handle the case where the drivers don't supply timestamps 
If highres-timestamp is 0, try lowres and if that fails fallback to system clock
timestamps.
 2011-11-28 09:13:29 +01:00
Matej Knopp
2c55cc7bcb uridecodebin: fix debug message printf format compiler warning 
https://bugzilla.gnome.org/show_bug.cgi?id=662607
 2011-11-27 22:43:20 +00:00
Tim-Philipp Müller
32b14c6ed3 Merge remote-tracking branch 'origin/master' into 0.11 
Conflicts:
	ext/vorbis/gstvorbisenc.c
	gst/playback/gstdecodebin2.c
	gst/playback/gstplaysinkconvertbin.c
	gst/videorate/gstvideorate.c
 2011-11-26 12:12:59 +00:00
Vincent Penquerc'h
c6b9145630 oggmux: set collectpads2 not to wait on sparse streams 
https://bugzilla.gnome.org/show_bug.cgi?id=663174
 2011-11-25 16:11:01 +00:00
Josep Torra
05ecdc1246 playsinkconvertbin: make identiy silent 2011-11-25 15:35:39 +01:00
Tim-Philipp Müller
a0639dad38 audio: remove unstable API guards from the audio decoder and encoder base classes 2011-11-25 13:11:54 +00:00
Tim-Philipp Müller
2dc7c2f676 docs: mention explicitly that playbin2 signals are emitted from a streaming thread 2011-11-25 13:01:47 +00:00
Sebastian Dröge
a5535e76e0 decodebin2: Set the multiqueue limits to the playing limits after overrun too 
We don't expect any new pads anymore and prerolling is finished now.
 2011-11-25 11:12:10 +01:00
Sebastian Dröge
494b2cb1a7 decodebin2: Cache the upstream seekability for demuxer decode chains and use it for the non-preroll multiqueue limits 
After preroll the multiqueue limits are still set to the preroll
limits if use-buffering is set to TRUE. In that case we only want
time limits on the multiqueue if upstream is seekable.
 2011-11-25 11:12:10 +01:00
Vincent Penquerc'h
59f5d980f6 decodebin2: fix prerolling for low bitrate streams from hlsdemux 
Such streams were detected as seekable, as the query on the typefind
element was testing the m3u8 file listing the actual streams, and
not going through the demuxer(s).

We now check for seekability for each multiqueue following a demuxer,
so the query will flow through the elements which might prevent seeking.

https://bugzilla.gnome.org/show_bug.cgi?id=647769
 2011-11-25 11:12:10 +01:00
Edward Hervey
d94535832b gst-libs: Add --warn-all to introspection scanner 
And let's get fixing those docs :)
 2011-11-25 10:31:38 +01:00
René Stadler
2094171aae tests: update for gstcheck API change 2011-11-24 21:47:02 +01:00
Vincent Penquerc'h
a5c64d5b97 oggdemux: minor cleanup 2011-11-24 17:12:56 +00:00
Vincent Penquerc'h
44ec58b41b libgstriff: add a couple tags that need skipping 
Found in a sample in the wild, appears to be ID3 tag.

https://bugzilla.gnome.org/show_bug.cgi?id=660249
 2011-11-24 17:04:19 +00:00
Sebastian Dröge
c6cffcfa19 videorate: Rename ARG_ enums to PROP_ 
This is more consistent with other code and these are
properties anyway, not arguments
 2011-11-24 14:41:56 +01:00
Sebastian Dröge
ec062ef3f2 videorate: Add property to force an output framerate 
API: GstVideoRate:force-fps

Changing the framerate during playback is not possible
with a capsfilter downstream if upstream is not using
gst_pad_alloc_buffer(). In that case there's no way in
0.10 to signal to videorate that the preferred framerate
has changed.

This new property will force the output framerate to
a specific value and can be changed during playback.
 2011-11-24 14:40:38 +01:00
Sebastian Dröge
683735a01e playsinkconvertbin: Reconfigure if we switch from raw to incompatible raw caps 
We might need to add converters and worked in passthrough mode before.
 2011-11-24 12:38:54 +01:00
Sebastian Dröge
113546b777 playsinkconvertbin: Override acceptcaps function for the two ghostpads 
The ghostpad acceptcaps functions are not valid in this case because
we don't only accept the caps accepted by the target but could also
insert converters. Fixes bug #663892.
 2011-11-24 12:37:58 +01:00
Sebastian Dröge
8f165b6206 playsinkaudioconvert: use-volume and use-converters are no construct-only properties anymore 
Fixes bug #663893.
 2011-11-24 11:34:12 +01:00