Commit graph

9845 commits

Author SHA1 Message Date
Tim-Philipp Müller
4828234639 alsamixer: use GRectMutext instead of GStaticRecMutex with newer glib versions 2011-12-04 20:38:19 +00:00
Tim-Philipp Müller
9c307bccc5 alsamixer: embed static mutexes into the mixer structure
instead of allocating them dynamically
2011-12-04 20:21:26 +00:00
Tim-Philipp Müller
ff6cc8af82 tools, tests: g_thread_init() is deprecated in glib master
It's not needed any longer.
2011-12-04 17:16:36 +00:00
Tim-Philipp Müller
0d98aa25b8 Work around deprecated thread API in glib master
Add private replacements for deprecated functions such as
g_mutex_new(), g_mutex_free(), g_cond_new() etc., mostly
to avoid the deprecation warnings. We'll change these
over to the new API once we depend on glib >= 2.32.

Replace g_thread_create() with g_thread_try_new().
2011-12-04 17:16:30 +00:00
Tim-Philipp Müller
6098442bd0 xmpwriter: update for thread API deprecations in glib master 2011-12-04 15:23:21 +00:00
Stefan Sauer
5398950012 fft-example: re-add Makefile.am 2011-12-04 13:43:06 +01:00
Stefan Sauer
f4213c5e85 configure: trim trailing whitespace 2011-12-02 23:35:50 +01:00
Stefan Sauer
3fb41e2cc3 tests: add a test for fft result value-ranges
Add a small example that uses ffts of various types and parameters and check the
result value ranges.
2011-12-02 23:34:47 +01:00
Piotr Fusik
14644457b0 various: typo fixes
Fix typos in code and docs. Fixes. #658984
2011-12-02 12:03:27 +01:00
Stefan Sauer
0cce8ab97d adder: be more graceful in the clipfunction
Doing dynamic pipelines is hard in 0.10. As we don't have the sticky events in
0.10 and sending such events in special elements like adder and tee was outvoted
on last attempt, be graceful to the misbehaviour instead.
2011-12-01 12:03:17 +01:00
Tim-Philipp Müller
3c87d7dc77 tests: fix caps leak in audioresample tests 2011-12-01 01:22:19 +00:00
Tim-Philipp Müller
c58d4f54d6 tests: fix memory leak in basetime test 2011-12-01 01:07:26 +00:00
Tim-Philipp Müller
1bf8fa1e5f playbin2: tone down debug message about file URIs with spaces
Complain a bit less loudly about URIs that have not been
escaped properly.
2011-11-30 23:58:19 +00:00
Tim-Philipp Müller
e88e47cd24 Revert "alsasrc: Improve timestamp accuracy"
This reverts commit 0b774e0b7c.
2011-11-30 23:15:35 +00:00
Tim-Philipp Müller
e5ae553850 Revert "alsasrc: Fix some compilation errors"
This reverts commit 2b84f5bd74.
2011-11-30 23:15:22 +00:00
Tim-Philipp Müller
4cc8920db4 Revert "alsa: Remove unused but set variable"
This reverts commit e9aed7f31c.
2011-11-30 23:15:12 +00:00
Tim-Philipp Müller
1290f7de0e Revert "alsasrc: fail gracefully when ALSA does not give timestamps"
This reverts commit c7282a5718.
2011-11-30 23:15:03 +00:00
Tim-Philipp Müller
d11849114c Revert "alsasrc: handle the case where the drivers don't supply timestamps"
This reverts commit 8154b69112.
2011-11-30 23:14:54 +00:00
Stefan Sauer
6d167abdfa Revert "alsasrc: style fix"
This reverts commit f70ca6d4cb.
2011-11-30 23:14:44 +00:00
Sebastian Dröge
21b252727d playsinkconvertbin: Don't send undefined NEWSEGMENT events to the internal elements
This happens when the internal elements are added before any NEWSEGMENT
event arrived and in that case we shouldn't send a NEWSEGMENT event
to the internal elements at all. They will get the NEWSEGMENT event
from upstream later.
2011-11-30 14:25:11 +01:00
Sebastian Dröge
e7853d3a3d playbin2: Fix decoder-sink compatibility check for raw audio/video formats
If the sink supports raw audio/video, we first check
if the decoder could output any raw audio/video format
and assume it is compatible with the sink then. We don't
do a complete compatibility check here if converters
are plugged between the decoder and the sink because
the converters will convert between raw formats and
even if the decoder format is not supported by the decoder
a converter will convert it.

We assume here that the converters can convert between
any raw format.

Fixes bug #665120.
2011-11-29 14:15:45 +01:00
Alessandro Decina
ab921eec11 oggdemux: fix compiler warning 2011-11-29 09:16:20 +01:00
Alessandro Decina
848711706b libgstvideo: minor fixes to key unit events
Make out args to gst_video_event_parse_{downstream|upstream}_force_key_unit
optional, update libgstvideo.def and fix docs a bit.

API: gst_video_event_new_upstream_force_key_unit
API: gst_video_event_new_downstream_force_key_unit
API: gst_video_event_is_force_key_unit
API: gst_video_event_parse_upstream_force_key_unit
API: gst_video_event_parse_downstream_force_key_unit

https://bugzilla.gnome.org/show_bug.cgi?id=607742
2011-11-29 09:15:59 +01:00
Andoni Morales Alastruey
df44e771f1 libgstvideo: Add force key unit events 2011-11-29 08:58:28 +01:00
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
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
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
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
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
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