mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-18 22:36:33 +00:00
Code Issues 1 Projects Releases Wiki Activity
13543 commits 44 branches 1510 tags 460 MiB
Commit graph

152 commits

Author SHA1 Message Date
Sebastian Dröge
2ed8f2e503 audioresample: Don't left-shift into the sign bit, instead use unsigned integers 2014-04-22 18:28:10 +02:00
Sebastian Dröge
122446476f audioresample: Fix up indention 2014-04-15 19:31:28 +02:00
Sebastian Dröge
5826f79980 audioresample: Fix out of bounds memory accesses 2014-04-15 19:31:28 +02:00
Vincent Penquerc'h
f588d14cdc audioresample: reject 0 denominator when creating resampler 
Coverity 1195140, 1195139, 1195138
 2014-04-10 12:35:03 +01:00
Sebastian Dröge
4e3d101aa8 audioresample: It's HAVE_EMMINTRIN_H, not HAVE_XMMINTRIN_H for SSE2 2014-01-20 16:11:04 +01:00
Antoine Jacoutot
daa194b71e audioresample: Fix build on x86 if emmintrin.h is available but can't be used 
On i386, EMMINTRIN is defined but not usable without SSE so check for
__SSE__ and __SSE2__ as well.

https://bugzilla.gnome.org/show_bug.cgi?id=670690
 2014-01-20 16:08:41 +01:00
Tim-Philipp Müller
ba32b2e16b audioresample: make explicit that neon is disabled and why 
https://bugzilla.gnome.org/show_bug.cgi?id=703477
 2013-07-03 09:44:32 +01:00
Carlos Rafael Giani
1b48d431f4 audioresample: disable 16-bit integer NEON support 
it seems to be broken (produces no audio), plus the performance gain
is small

Signed-off-by: Carlos Rafael Giani <dv@pseudoterminal.org>
 2013-07-03 08:14:42 +02:00
Sebastian Dröge
948a4a3632 gst: Add better support for static plugins 2013-04-15 15:52:58 +02:00
Tim-Philipp Müller
5f59b4f7ee Fix FSF address 
https://bugzilla.gnome.org/show_bug.cgi?id=687520
 2012-11-03 23:05:09 +00:00
Sebastian Dröge
3864209f6e audioresample: Use auto sinc table mode by default 2012-10-25 14:03:52 +02:00
Carlos Rafael Giani
d793a2b560 audioresample: added ARM NEON support 
This adds ARM NEON accelerated code paths for 16-bit integer
and 32-bit floating point samples.

It is a modified combination of patches #3 and #5 from Jyri Sarha
( http://lists.xiph.org/pipermail/speex-dev/2011-September/008240.html &
http://lists.xiph.org/pipermail/speex-dev/2011-September/008238.html )

Signed-off-by: Carlos Rafael Giani <dv@pseudoterminal.org>
 2012-10-25 14:03:52 +02:00
Carlos Rafael Giani
19073ab8c4 audioresample: changed inner_product_single semantics 
This is an adaptation of patch #3 from Jyri Sarha
( http://lists.xiph.org/pipermail/speex-dev/2011-September/008240.html ),
but without the NEON optimizations (these come in a separate commit).
The idea is to replace SATURATE32(PSHR32(x, shift), a) operations with a
combined SATURATE32PSHR(x, shift, a) macro that can be optimized for
specific platforms (and also avoids rare rounding errors).

Signed-off-by: Carlos Rafael Giani <dv@pseudoterminal.org>
 2012-10-25 14:03:52 +02:00
Carlos Rafael Giani
c41faa3d8e audioresample: sinc filter performance improvements 
Original idea comes from Jyri Sarha
( http://lists.xiph.org/pipermail/speex-dev/2011-September/008243.html ).
Patch was discovered by Branislav Katreniak
( branislav.katreniak@streamunlimited.com ) for StreamUnlimited
( http://streamunlimited.com/ ). Tests showed up to 5x speed increase in
the resampler in the 44.1<->48kHz case.
I added the sinc-filter-mode and sinc-filter-auto-threshold properties
and the auto mode threshold tests, and adapted the code to GStreamer 1.0.

Signed-off-by: Carlos Rafael Giani <dv@pseudoterminal.org>
 2012-10-25 14:03:52 +02:00
Sebastian Dröge
3c1041d5eb Revert "gst: Add better support for static plugins" 
This reverts commit d2d79e3bc2,
which was accidentially pushed.
 2012-10-24 13:26:26 +02:00
Sebastian Dröge
d2d79e3bc2 gst: Add better support for static plugins 2012-10-24 12:10:44 +02:00
Mark Nauwelaerts
17e3dc3357 audioresample: mark semi-unused variable 
../../../gst-plugins-base/gst/audioresample/gstaudioresample.c: In function 'gst_audio_resample_dump_drain':
../../../gst-plugins-base/gst/audioresample/gstaudioresample.c:729:9: warning: variable 'in_len' set but not used [-Wunused-but-set-variable]
 2012-09-18 13:16:39 +02:00
Sebastian Rasmussen
6c2aea9551 Fix bug where debug category was declared inside a function 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676670
 2012-05-24 10:33:02 +02:00
Tim-Philipp Müller
3c6a3ad629 Use new gst_element_class_set_static_metadata() 2012-04-10 00:45:16 +01:00
Sebastian Dröge
ad42b16375 gst: Update for GST_PLUGIN_DEFINE() API change 2012-04-05 15:11:05 +02:00
Sebastian Dröge
65307dd132 gst: Update versioning 2012-04-04 14:55:15 +02:00
Wim Taymans
25137962ad fix for caps API changes 2012-03-11 19:04:41 +01:00
Wim Taymans
642ca2bd40 audioresample: remove transform lock 
In this particular case it was not sufficient anyways because the setcaps
function didn't take the transform lock.
 2012-02-23 11:19:52 +01:00
Wim Taymans
9212619549 update for new fixate_caps function 2012-02-22 12:32:44 +01:00
Wim Taymans
fcdc385aa1 port to new map API 2012-01-25 12:30:53 +01:00
Mark Nauwelaerts
97a4f7e1e5 audioresample: fix debug message format specifier 2012-01-06 16:15:45 +01:00
Sebastian Dröge
5bdf6b3383 gst: Add new layout field to the raw audio caps 2012-01-05 10:34:25 +01:00
Wim Taymans
8a9a0bf6da audioresample: truncate in fixation 2012-01-02 15:59:09 +01:00
Tim-Philipp Müller
177525f89f Merge remote-tracking branch 'origin/master' into 0.11 
Conflicts:
	gst-libs/gst/netbuffer/gstnetbuffer.c
	gst/ffmpegcolorspace/avcodec.h
	gst/ffmpegcolorspace/gstffmpegcodecmap.c
	gst/ffmpegcolorspace/imgconvert.c
	gst/ffmpegcolorspace/imgconvert_template.h
	gst/ffmpegcolorspace/mem.c
	gst/playback/README
	gst/playback/gstplaybasebin.c
	gst/playback/gstplaybasebin.h
	gst/playback/gstplaybin.c
	sys/v4l/v4lmjpegsrc_calls.c
	sys/v4l/videodev_mjpeg.h
	tests/check/elements/gnomevfssink.c
 2011-12-02 11:10:17 +00:00
Piotr Fusik
14644457b0 various: typo fixes 
Fix typos in code and docs. Fixes. #658984
 2011-12-02 12:03:27 +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
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
96374054ac various: fix pad template leaks 
https://bugzilla.gnome.org/show_bug.cgi?id=662664
 2011-11-28 13:09:02 +00:00
Wim Taymans
2202511e77 add parent to query function 2011-11-16 17:25:17 +01:00
Wim Taymans
372b9329b9 remove query types 2011-11-09 11:47:54 +01:00
Wim Taymans
33196cdd2c audio: change audio format syntax a little 
Remove the _ in front of the endianness prefix.
Remove the _3 postfix for the 24 bits formats.
Add a _32 postfix after the formats that occupy extra space beyond their
natural size.
The result is that the GST_AUDIO_NE() macro can simply append the endianness
after all formats and that we only specify a different sample width when it is
different from the natural size of the sample. This makes things more consistent
and follows the pulseaudio conventions instead of the alsa ones.
 2011-09-06 12:06:39 +02:00
Wim Taymans
ba41bb5ca7 Merge branch 'master' into 0.11 
Conflicts:
	ext/ogg/gstoggmux.c
	gst/playback/gstplaysink.c
 2011-08-18 19:36:50 +02:00
Wim Taymans
dae848818d audio: rework audio caps. 
Rework the audio caps similar to the video caps. Remove
width/depth/endianness/signed fields and replace with a simple string
format and media type audio/x-raw.
Create a GstAudioInfo and some helper methods to parse caps.
Remove duplicate code from the ringbuffer and replace with audio info.
Use AudioInfo in the base audio filter class.
Port elements to new API.
 2011-08-18 19:15:03 +02:00
Vincent Penquerc'h
30236ddfd3 audioresample: fix build without orc 
https://bugzilla.gnome.org/show_bug.cgi?id=656781
 2011-08-18 11:03:58 +02:00
Wim Taymans
d679dd2c54 audioresample: fix after merge 2011-08-17 10:47:38 +02:00
Wim Taymans
33467d9629 Merge branch 'master' into 0.11 
Conflicts:
	configure.ac
	ext/pango/gsttextoverlay.c
	ext/theora/gsttheoradec.c
	gst/adder/gstadder.c
	gst/adder/gstadder.h
	gst/audioresample/gstaudioresample.c
	gst/encoding/gstencodebin.c
	gst/playback/gstdecodebin.c
	gst/playback/gstdecodebin2.c
	tests/check/elements/decodebin2.c
	tests/check/elements/playbin-compressed.c
	win32/common/libgsttag.def
 2011-08-16 18:01:14 +02:00
Vincent Penquerc'h
49ec6899f4 audioresample: fix quality setting being ignored by the resampler state 
https://bugzilla.gnome.org/show_bug.cgi?id=636562
 2011-08-12 09:55:17 +02:00
Vincent Penquerc'h
746415a6e3 audioresample: use SSE/SSE2 when possible 
Compile in the code on i386 and x86_64, and use ORC to determine
when the runtime platform can run the code.

https://bugzilla.gnome.org/show_bug.cgi?id=636562
 2011-08-12 09:55:11 +02:00
Vincent Penquerc'h
58fd202b7d audioresample: fix SSE2 building with double precision 
The full double implementation was missing.

https://bugzilla.gnome.org/show_bug.cgi?id=636562
 2011-08-12 09:53:12 +02:00
Josep Torra
5629ed74b3 Fix debug statements 
Fixes build on MacOSX

Signed-off-by: Edward Hervey <edward.hervey@collabora.co.uk>
 2011-08-10 11:15:41 +02:00
Wim Taymans
4fb67fb0da audioresample: fix for event handler change 2011-07-22 21:19:08 +02:00
Wim Taymans
40d567153a Merge branch 'master' into 0.11 2011-06-13 19:09:05 +02:00
David Schleef
4db89c82bb convert M_PI to G_PI, for msvc 2011-06-10 23:56:34 -07:00
Wim Taymans
0ac9bb7d99 Merge branch 'master' into 0.11 
Conflicts:
	tests/examples/audio/Makefile.am
	tests/examples/v4l/Makefile.am
 2011-06-10 12:14:57 +02:00
Tim-Philipp Müller
c692191c33 GST_PLUGINS_BASE_LIBS is not defined in -base. 2011-06-08 12:21:43 +01:00