After 3b26b782ee, `ebur128->true_peak` was only set to the maximum of the
current "true peak per frame" values, when it should report the true peak for
the entire stream.
Fixes: 3b26b782ee
Easier to read, less convoluted, and ~30% faster. Most importantly, this
avoids repeating the redundant recalculation of the true peak on every
single sample, by moving the FIND_PEAK() loop out of the main loop. (Note
that FIND_PEAK() does not depend on the current sample index at all, so
there is no reason for it to ever be recomputed here)
Processes two channels in parallel, using 128-bit XMM registers.
In theory, we could go up to YMM registers to process 4 channels, but this is
not a gain except for relatively high channel counts (e.g. 7.1), and also
complicates the sample load/store operations considerably.
I decided to only add an AVX variant, since the C code is not substantially
slower enough to justify a separate function just for ancient CPUs.
Instead of having a planar array for each channel, use a single packed array.
This will help processing multiple channels in parallel, as we can directly
load all channels' data in a single load instruction.
Also improves memory locality of data, as the loop order is:
for (samples) {
for (channels) {
process sample
}
}
This patch is analogous to 20f9727018:
It hides the internal part of AVFilter by adding a new internal
structure FFFilter (declared in filters.h) that has an AVFilter
as its first member; the internal part of AVFilter is moved to
this new structure.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
There are lots of files that don't need it: The number of object
files that actually need it went down from 2011 to 884 here.
Keep it for external users in order to not cause breakages.
Also improve the other headers a bit while just at it.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Makes it robust against adding fields before it, which will be useful in
following commits.
Majority of the patch generated by the following Coccinelle script:
@@
typedef AVOption;
identifier arr_name;
initializer list il;
initializer list[8] il1;
expression tail;
@@
AVOption arr_name[] = { il, { il1,
- tail
+ .unit = tail
}, ... };
with some manual changes, as the script:
* has trouble with options defined inside macros
* sometimes does not handle options under an #else branch
* sometimes swallows whitespace
internal.h does not depend on video.h (and should not depend on it)
and therefore should not include video.h at all; instead all users
of video.h should include it directly.
Doing so also avoids unnecessary video.h inclusions in files that
don't need it, like most audio filters.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
If one looks at the many query_formats callbacks in existence,
one will immediately recognize that there is one type of default
callback for video and a slightly different default callback for
audio: It is "return ff_set_common_formats_from_list(ctx, pix_fmts);"
for video with a filter-specific pix_fmts list. For audio, it is
the same with a filter-specific sample_fmts list together with
ff_set_common_all_samplerates() and ff_set_common_all_channel_counts().
This commit allows to remove the boilerplate query_formats callbacks
by replacing said callback with a union consisting the old callback
and pointers for pixel and sample format arrays. For the not uncommon
case in which these lists only contain a single entry (besides the
sentinel) enum AVPixelFormat and enum AVSampleFormat fields are also
added to the union to store them directly in the AVFilter,
thereby avoiding a relocation.
The state of said union will be contained in a new, dedicated AVFilter
field (the nb_inputs and nb_outputs fields have been shrunk to uint8_t
in order to create a hole for this new field; this is no problem, as
the maximum of all the nb_inputs is four; for nb_outputs it is only
two).
The state's default value coincides with the earlier default of
query_formats being unset, namely that the filter accepts all formats
(and also sample rates and channel counts/layouts for audio)
provided that these properties agree coincide for all inputs and
outputs.
By using different union members for audio and video filters
the type-unsafety of using the same functions for audio and video
lists will furthermore be more confined to formats.c than before.
When the new fields are used, they will also avoid allocations:
Currently something nearly equivalent to ff_default_query_formats()
is called after every successful call to a query_formats callback;
yet in the common case that the newly allocated AVFilterFormats
are not used at all (namely if there are no free links) these newly
allocated AVFilterFormats are freed again without ever being used.
Filters no longer using the callback will not exhibit this any more.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Up until now, an AVFilter's lists of input and output AVFilterPads
were terminated by a sentinel and the only way to get the length
of these lists was by using avfilter_pad_count(). This has two
drawbacks: first, sizeof(AVFilterPad) is not negligible
(i.e. 64B on 64bit systems); second, getting the size involves
a function call instead of just reading the data.
This commit therefore changes this. The sentinels are removed and new
private fields nb_inputs and nb_outputs are added to AVFilter that
contain the number of elements of the respective AVFilterPad array.
Given that AVFilter.(in|out)puts are the only arrays of zero-terminated
AVFilterPads an API user has access to (AVFilterContext.(in|out)put_pads
are not zero-terminated and they already have a size field) the argument
to avfilter_pad_count() is always one of these lists, so it just has to
find the filter the list belongs to and read said number. This is slower
than before, but a replacement function that just reads the internal numbers
that users are expected to switch to will be added soon; and furthermore,
avfilter_pad_count() is probably never called in hot loops anyway.
This saves about 49KiB from the binary; notice that these sentinels are
not in .bss despite being zeroed: they are in .data.rel.ro due to the
non-sentinels.
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
The magic constants come from the unofficial "ITU-R BS.1770-1 filter
specifications"¹ by Raiden (libebur128) which relies on "Parameter
Quantization in Direct-Form Recursive Audio Filters"² by Brian
Neunaber.
The constants seem to include a quantization bias, for example:
- Vb is supposed to be exactly √Vh in a high shelf filter
- the Pre-filter Gain should likely be 4dB
- Pre Q and RLB Q are respectively very close to √½ and ½
Those are not adjusted to prevent the values from drifting away from
the official specifications.
An alternative to this approach would be to requantize on the fly as
proposed by pbelkner³, where the 48kHz code path would use the exact
specifications constants while derivating constants for other
frequencies.
[1]: https://www.scribd.com/document/49991813/ITU-R-BS-1770-1-filters
[2]: https://www.scribd.com/document/6531763/Direct-Form-Filter-Parameter-Quantization
[3]: https://hydrogenaud.io/index.php?topic=86116.msg740092#msg740092
It will allow to refernce it as a whole without clunky macros.
Most of the changes have been automatically made with sed:
sed -i '
s/-> *in_formats/->incfg.formats/g;
s/-> *out_formats/->outcfg.formats/g;
s/-> *in_channel_layouts/->incfg.channel_layouts/g;
s/-> *out_channel_layouts/->outcfg.channel_layouts/g;
s/-> *in_samplerates/->incfg.samplerates/g;
s/-> *out_samplerates/->outcfg.samplerates/g;
' src/libavfilter/*(.)
These names are mostly the same, so not using duplicates saves
allocations, checks for the allocations as well as frees.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
