Discovered with:
find . -type f \( -name '*.md' -o -name '*.rst' \) -exec grep -n 'http://' {} +
All real, i.e. non-example, links found were moved to https. There are
some dead links and websites with no https available which were not
converted.
scaletempo2 is a new audio filter for playing back
audio at modified speed and is based on chromium
commit 51ed77e3f37a9a9b80d6d0a8259e84a8ca635259.
It sounds subjectively better than the existing
implementions scaletempo and rubberband.
This mode drops or repeats audio data to adapt to video speed, instead
of resampling it or such. It was added to deal with SPDIF. The
implementation was part of fill_audio_out_buffers() - the entire
function is something whose complexity exploded in my face, and which I
want to clean up, and this is hopefully a first step.
Put it in a filter, and mess with the shitty glue code. It's all sort of
roundabout and illogical, but that can be rectified later. The important
part is that it works much like the resample or scaletempo filters.
For PCM audio, this does not work on samples anymore. This makes it much
worse. But for PCM you can use saner mechanisms that sound better. Also,
something about PTS tracking is wrong. But not wasting more time on
this.
This filter wasn't referenced anywhere and thus was dead code. It should
have been in the audio filter list in user_filters.c. This was intended
as compatibility wrapper (to avoid breaking old command lines and config
files), and has no real use. Apparently I forgot to add it to the filter
list (did I even test this shit?), and so it was rotting around for 1.5
years doing nothing (just like myself).
Note that users can just use the libavfilter provided filter to force
resampling, just that it has a different name and different options.
There's also af_format to force inserting auto conversion through the
internal f_swsresample filter.
This commit introduces the multiply-pitch af-command. Users may bind
keys to this command in order to incrementally adjust the pitch of a
track. This will probably mostly be useful for musicians trying to
transpose up and down by semi tones without having to calculate
the correct ratio beforehand.
As an example, here is an input.conf to test this feature:
{ af-command all multiply-pitch 0.9438743126816935
} af-command all multiply-pitch 1.059463094352953
The future direction might be not having such a user-visible filter at
all, similar to how vf_scale went away (or actually, redirects to
libavfilter's vf_scale).
These couldn't be relicensed, and won't survive the LGPL transition. The
other existing filters are mostly LGPL (except libaf glue code).
This remove the deprecated pan option. I guess it could be restored by
inserting a libavfilter filter (if there's one), but for now let it be
gone.
This temporarily breaks volume control (and things related to it, like
replaygain).
af_volume is deprecated, and so are its replaygain sub-options. To make
it possible to use replaygain without deprecated options (and of course
to make it available at all after af_volume is dropped), reintroduce
them as top-level options.
This also means that they are easily changeable at runtime by using them
as properties. Change the "volume" property to use the new update
mechanism as well.
We don't actually bother sharing the implementation between new and
deprecated mechanisms, as the deprecated one will simply be deleted.
For the from_dB() functions, we mention anders' copyright, although I'm
not sure if a mere formula is copyrightable. This will have to be
determined later.
This whole change is mostly untested. Our distributed human CI will take
care of it.
Basically, see the example in input.rst.
This is better than the "old" vf-toggle method, because it doesn't
require the user to duplicate the filter string in mpv.conf and
input.conf.
Some aspects of this changes are untested, so enjoy your alpha testing.
Remove low quality drc filter. Anyone whishing to have dynamic range
compression should use the much more powerful acompressor ffmpeg filter:
mpv --af=lavfi=[acompressor] INPUT
Or with parameters:
mpv --af=lavfi=[acompressor=threshold=-25dB:ratio=3:makeup=8dB] INPUT
Refer to https://ffmpeg.org/ffmpeg-filters.html#acompressor for a full
list of supported parameters.
Signed-off-by: wm4 <wm4@nowhere>
Some users still use this filter, so the filter was going to be kept.
But I overlooked that libavfilter provides this filter. Remove the
redundant wrapper from mpv. Something like --af=lavfi=bs2b should work
and give exactly the same results.
All of these filters are considered not useful anymore by us. Some have
replacements in libavfilter (useable through af_lavfi).
af_center, af_extrastereo, af_karaoke, af_sinesuppress, af_sub,
af_surround, af_sweep: pretty simple and useless filters which probably
nobody ever wants.
af_ladspa: has a replacement in libavfilter.
af_hrtf: the algorithm doesn't work properly on most sources, and the
implementation was buggy and complicated. (The filter was inherited from
MPlayer; but even in mpv times we had to apply fixes that fixed major
issues with added noise.) There is a ladspa filter if you still want to
use it.
af_export: I'm not even sure what this is supposed to do. Possibly it
was meant for GUIs rendering audio visualizations, but it couldn't
really work well. For example, the size of the audio depended on the
samplerate (fixed number of samples only), and it couldn't retrieve the
complete audio, only fragments. If this is really needed for GUIs, mpv
should add native visualization, or a proper API for it.
librubberband exports a big load of options. Normally, the default
settings (whether they're librubberband defaults or our defaults) should
be sufficient, but since I'm not so sure about this, making it
configurable allows others to figure it out for me.
If "--af=rubberband" is used, librubberband will be used to speed up or
slow down audio with pitch correction.
This still has some problems: the audio delay is not calculated
correctly, so the audio position jitters around by a few milliseconds.
This will probably ruin video timing.
Until now, the audio chain could handle both little endian and big
endian formats. This actually doesn't make much sense, since the audio
API and the HW will most likely prefer native formats. Or at the very
least, it should be trivial for audio drivers to do the byte swapping
themselves.
From now on, the audio chain contains native-endian formats only. All
AOs and some filters are adjusted. af_convertsignendian.c is now wrongly
named, but the filter name is adjusted. In some cases, the audio
infrastructure was reused on the demuxer side, but that is relatively
easy to rectify.
This is a quite intrusive and radical change. It's possible that it will
break some things (especially if they're obscure or not Linux), so watch
out for regressions. It's probably still better to do it the bulldozer
way, since slow transition and researching foreign platforms would take
a lot of time and effort.
