Commit graph

189 commits

Author SHA1 Message Date
Havard Graff
0826fb95b7 audio: video: Optimize by using cached quark for meta tag
Avoid taking the global quark lock for every single buffer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/295>
2020-06-27 09:23:10 +00:00
Mathieu Duponchelle
cc516695b0 plugins: uddate gst_type_mark_as_plugin_api() calls 2020-06-06 00:41:17 +02:00
Mathieu Duponchelle
e666c9ec04 plugins: Use gst_type_mark_as_plugin_api() for all non-element plugin types 2020-06-03 22:44:23 -04:00
Sebastian Dröge
44cd1c7a65 audioresample: Drain resampler on discontinuities
Otherwise we would lose the last few samples when resetting the
resampler.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/670>
2020-05-27 17:06:08 +00:00
Sebastian Dröge
bf0cffc474 audioresample: Drain resampler and reset timestamp tracking on stream-start event too
And also reset timestamp tracking on EOS events as more data might come
afterwards with a new stream-start event. This keeps the code the same.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/670>
2020-05-27 17:06:08 +00:00
Sebastian Dröge
6d423cbba2 audioresample: Drain the resampler and reset timestamp tracking on caps changes
Especially when changing the sample rate our timestamp tracking will be
completely off, but even otherwise we would usually lose the last few
samples if we don't drain here as the resampler gets reset if anything
but the sample rate changes.

This is usually not a problem as the first buffer after a caps event
usually has the discont flag set, but can cause problems if
 - the caps event is followed by a segment event, which then causes
   draining according to the new sample rate
 - the caps were changed because of rengotiation due to a reconfigure
   event and there is not discontinuity from upstream

In both cases we would output buffers with completely wrong timestamps.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/670>
2020-05-27 17:06:08 +00:00
Niels De Graef
a1ef6a1179 audioresample: Use G_DECLARE_FINAL_TYPE 2020-03-16 15:47:58 +00:00
Tim-Philipp Müller
289d8e53e2 Remove autotools build system 2019-10-13 14:15:43 +01:00
Thibault Saunier
909baa2360 Pass the code through codespell 2019-08-30 13:05:36 +00:00
Thibault Saunier
27ba8d24ec doc: Port to hotdoc 2019-05-13 11:34:08 -04:00
Johan Bjäreholt
448ed0f244 Removed unused GST_CAT_PERFORMANCE
https://bugzilla.gnome.org/show_bug.cgi?id=797175
2018-09-19 15:06:19 +03:00
George Kiagiadakis
114f0abbd7 audioresample: implement support for non-interleaved buffers
https://bugzilla.gnome.org/show_bug.cgi?id=705986
2018-07-11 16:26:13 +03:00
Xavier Claessens
201e7c7803 Meson: Generate pc file for all plugins in base
https://bugzilla.gnome.org/show_bug.cgi?id=794568
2018-04-25 11:05:56 +01:00
Nicolas Dufresne
8e6c6266d7 Remove plugin specific static build option
Static and dynamic plugins now have the same interface. The standard
--enable-static/--enable-shared toggle are sufficient.
2017-05-16 13:42:07 -04:00
Thibault Saunier
099ac9faf2 docs: Convert gtkdoc comments to markdown
Modernizing the documentation, making it simpler to read an
modify and allowing us to possibly switch to hotdoc in the
future.
2017-03-10 18:19:17 -03:00
Nirbheek Chauhan
5c4f4ac1bd Add support for Meson as alternative/parallel build system
https://github.com/mesonbuild/meson

With contributions from:

Tim-Philipp Müller <tim@centricular.com>
Jussi Pakkanen <jpakkane@gmail.com> (original port)

Highlights of the features provided are:
* Faster builds on Linux (~40-50% faster)
* The ability to build with MSVC on Windows
* Generate Visual Studio project files
* Generate XCode project files
* Much faster builds on Windows (on-par with Linux)
* Seriously fast configure and building on embedded

... and many more. For more details see:

http://blog.nirbheek.in/2016/05/gstreamer-and-meson-new-hope.html
http://blog.nirbheek.in/2016/07/building-and-developing-gstreamer-using.html

Building with Meson should work on both Linux and Windows, but may
need a few more tweaks on other operating systems.
2016-08-20 11:09:51 +01:00
Wim Taymans
c1cb395f66 audioresample: after a reset, recalculate the ouput size
After we reset the resampler, there is no history anymore in the resampler
and the previously calculated output size is no longer valid.
Recalculate the new output size after a reset to make sure we don't try
to convert too much.
2016-07-19 18:20:57 +02:00
Wim Taymans
f8e4c801eb audioresample: remove last ORC remains 2016-03-28 13:25:55 +02:00
Wim Taymans
524ea147cc audio-resampler: improve filter construction
Remove some unused variables from the inner product functions.
Make filter coefficients by interpolating if required.
Rename some fields.
Try hard to not recalculate filters when just chaging the rate.
Add more proprties to audioresample.
2016-03-28 13:25:52 +02:00
Wim Taymans
ed747492ef audio-resampler: add reset function
Add a function to reset the audio-resampler.
Use new function in audio-converter
Use the new functions in gstaudioresample and fixup drain functions.
2016-03-28 13:25:51 +02:00
Wim Taymans
05eb109c0d audio-resampler: handle filter length changes
Update the buffer with history samples when the filter length changes
because of an update of the parameters or sample rates.
2016-03-28 13:25:51 +02:00
Wim Taymans
d348fbb9b9 audio-converter: make some optimized functions
Make an optimized function that just calls the resampler when possible.
Optimize the resampler transform_size function a little.
2016-03-28 13:25:50 +02:00
Wim Taymans
de37491662 audio-converter: simplify API
Remove the consumed/produced output fields from the resampler and
converter. Let the caler specify the right number of input/output
samples so we can be more optimal.
Use just one function to update the converter configuration.
Simplify some things internally.
Make it possible to use writable input as temp space in audioconvert.
2016-03-28 13:25:50 +02:00
Wim Taymans
1d9a793545 audio-converter: more work on resampling
- Fix the resampler in the audio converter
- fix memory leaks
2016-03-28 13:13:59 +02:00
Wim Taymans
75d668e152 audio-converter: add resampler
Add a resampler to the processing chain when needed.
port the audio resampler to the new audioconverter library
2016-03-28 13:13:59 +02:00
Vineeth TM
44b70ca3a1 base: use new gst_element_class_add_static_pad_template()
https://bugzilla.gnome.org/show_bug.cgi?id=763075
2016-03-24 14:25:41 +02:00
Wim Taymans
9d66b7cdd2 resample: avoid overflows
Avoid overflow in rate calculation. This can cause the resampler to
start on the wrong phase after a rate change.
Avoid overflow in cubic fraction calculation. This can cause noise when
dealing with higher samplerates.
2016-02-11 19:55:08 +01:00
Wim Taymans
188c0811de resample: fix double interpolation sse code
We were only reading 2 filter taps and we need to read 4 to do cubic
interpolation.
2016-02-11 18:03:59 +01:00
Sebastian Dröge
e51c9a3dad audioresample: Clip input buffers to the segment before handling them
https://bugzilla.gnome.org/show_bug.cgi?id=757068
2015-11-02 10:20:37 +02:00
Sebastian Dröge
c5dbee33b0 audioresample: Also copy metas if their API has no tags attached to it
This is the default basetransform behaviour, being more strict than that
is not really useful.
2015-06-29 13:06:59 +02:00
Mathieu Duponchelle
2ad27e4c13 audioresample: copy metadata that only has the "audio" tag.
https://bugzilla.gnome.org/show_bug.cgi?id=750406
2015-06-04 19:16:40 +02:00
Tim-Philipp Müller
ec5c93f169 docs: update element example pipelines
- gst-launch -> gst-launch-1.0
- use autoaudiosink and audiovideosink more often
- review pipeline examples and descriptions
2015-05-10 11:38:19 +01:00
Tim-Philipp Müller
c680e324bc Remove obsolete Android build cruft
This is not needed any longer.
2015-04-26 18:42:34 +01:00
Sebastian Dröge
2bd4ea6e8e Constify some static arrays everywhere 2015-01-21 09:49:47 +01:00
Jan Alexander Steffens (heftig)
a636c39638 audioresample: Try to prevent endless looping
Speex may decide not to consume any samples because it can't write any. I've
seen a hang during draining caused by the resample loop never terminating.
In that case, resampling happened as normal until olen was 0 but ilen was
still 1. _process_native then reduced ichunk to 0, so ilen never decreased
below 1 and the loop never terminated.

Instead of reverting 684cf44 ({audioresample: don't skip input samples),
break only if all output samples have been produced and speex refuses
to consume any more input samples.

https://bugzilla.gnome.org/show_bug.cgi?id=732908
2015-01-19 19:36:13 +01:00
Peter G. Baum
0b4abc267e audioresample: remove unused variables
https://bugzilla.gnome.org/show_bug.cgi?id=738026
2014-10-07 14:59:10 +03:00
Kipp Cannon
684cf44ee3 audioresample: don't skip input samples
when downsampling, the output buffer can be filled before all the input
samples are consumed.  this is correct:  when downsampling, several input
samples are needed for each output sample, so when only a small number of
input samples are available the number of output samples produced can be 0.

the resampler, however, was discarding those extra input samples instead of
clocking them into its filter history for the next iteration.  this patch
fixes this by removing the check that the output buffer is full.  the code
now always loops until all input samples are consumed, and relies on the
calling code to have provided a suitably sized location for the output.
note that there are already other checks in place in the calling code to
ensure that this is the case.

https://bugzilla.gnome.org/show_bug.cgi?id=732908
2014-09-05 11:17:43 +03:00
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