audio/filter: remove some useless filters

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.
This commit is contained in:
wm4
2015-09-03 23:55:36 +02:00
parent e1fbd3b790
commit 091bfa3abf
24 changed files with 0 additions and 4132 deletions

View File

@@ -96,25 +96,6 @@ Available filters are:
If the input channel number is less than ``<minch>``, the filter will
detach itself (default: 3).
``sweep[=speed]``
Produces a sine sweep.
``<0.0-1.0>``
Sine function delta, use very low values to hear the sweep.
``sinesuppress[=freq:decay]``
Remove a sine at the specified frequency. Useful to get rid of the 50/60 Hz
noise on low quality audio equipment. It only works on mono input.
``<freq>``
The frequency of the sine which should be removed (in Hz) (default:
50)
``<decay>``
Controls the adaptivity (a larger value will make the filter adapt to
amplitude and phase changes quicker, a smaller value will make the
adaptation slower) (default: 0.0001). Reasonable values are around
0.001.
``bs2b[=option1:option2:...]``
Bauer stereophonic to binaural transformation using libbs2b. Improves the
headphone listening experience by making the sound similar to that from
@@ -137,18 +118,6 @@ Available filters are:
If ``fcut`` or ``feed`` options are specified together with a profile, they
will be applied on top of the selected profile.
``hrtf[=flag]``
Head-related transfer function: Converts multichannel audio to 2-channel
output for headphones, preserving the spatiality of the sound.
==== ===================================
Flag Meaning
==== ===================================
m matrix decoding of the rear channel
s 2-channel matrix decoding
0 no matrix decoding (default)
==== ===================================
``equalizer=g1:g2:g3:...:g10``
10 octave band graphic equalizer, implemented using 10 IIR band-pass
filters. This means that it works regardless of what type of audio is
@@ -354,64 +323,6 @@ Available filters are:
``mpv '--af=format=channels=5.1' '--audio-channels=5.1'`` would always force
remixing audio to 5.1 and output it like this.
``sub[=fc:ch]``
Adds a subwoofer channel to the audio stream. The audio data used for
creating the subwoofer channel is an average of the sound in channel 0 and
channel 1. The resulting sound is then low-pass filtered by a 4th order
Butterworth filter with a default cutoff frequency of 60Hz and added to a
separate channel in the audio stream.
.. warning::
Disable this filter when you are playing media with an LFE channel
(e.g. 5.1 surround sound), otherwise this filter will disrupt the sound
to the subwoofer.
``<fc>``
cutoff frequency in Hz for the low-pass filter (20 Hz to 300 Hz)
(default: 60 Hz) For the best result try setting the cutoff frequency
as low as possible. This will improve the stereo or surround sound
experience.
``<ch>``
Determines the channel number in which to insert the sub-channel
audio. Channel number can be between 0 and 7 (default: 5). Observe
that the number of channels will automatically be increased to <ch> if
necessary.
.. admonition:: Example
``mpv --af=sub=100:4 --audio-channels=5 media.avi``
Would add a subwoofer channel with a cutoff frequency of 100 Hz to
output channel 4.
``center``
Creates a center channel from the front channels. May currently be low
quality as it does not implement a high-pass filter for proper extraction
yet, but averages and halves the channels instead.
``<ch>``
Determines the channel number in which to insert the center channel.
Channel number can be between 0 and 7 (default: 5). Observe that the
number of channels will automatically be increased to ``<ch>`` if
necessary.
``surround[=delay]``
Decoder for matrix encoded surround sound like Dolby Surround. Some files
with 2-channel audio actually contain matrix encoded surround sound.
``<delay>``
delay time in ms for the rear speakers (0 to 1000) (default: 20) This
delay should be set as follows: If d1 is the distance from the
listening position to the front speakers and d2 is the distance from
the listening position to the rear speakers, then the delay should be
set to 15ms if d1 <= d2 and to 15 + 5*(d1-d2) if d1 > d2.
.. admonition:: Example
``mpv --af=surround=15 --audio-channels=4 media.avi``
Would add surround sound decoding with 15 ms delay for the sound to
the rear speakers.
``delay[=[ch1,ch2,...]]``
Delays the sound to the loudspeakers such that the sound from the
different channels arrives at the listening position simultaneously. It is
@@ -440,36 +351,6 @@ Available filters are:
Would delay front left and right by 10.5 ms, the two rear channels
and the subwoofer by 0 ms and the center channel by 7 ms.
``export=mmapped_file:nsamples]``
Exports the incoming signal to other processes using memory mapping
(``mmap()``). Memory mapped areas contain a header::
int nch /* number of channels */
int size /* buffer size */
unsigned long long counter /* Used to keep sync, updated every time
new data is exported. */
The rest is payload (non-interleaved) 16-bit data.
``<mmapped_file>``
File to map data to (required)
``<nsamples>``
number of samples per channel (default: 512).
.. admonition:: Example
``mpv --af=export=/tmp/mpv-af_export:1024 media.avi``
Would export 1024 samples per channel to ``/tmp/mpv-af_export``.
``extrastereo[=mul]``
(Linearly) increases the difference between left and right channels which
adds some sort of "live" effect to playback.
``<mul>``
Sets the difference coefficient (default: 2.5). 0.0 means mono sound
(average of both channels), with 1.0 sound will be unchanged, with
-1.0 left and right channels will be swapped.
``drc[=method:target]``
Applies dynamic range compression. This maximizes the volume by compressing
the audio signal's dynamic range. (Formerly called ``volnorm``.)
@@ -493,45 +374,6 @@ Available filters are:
This filter can cause distortion with audio signals that have a very
large dynamic range.
``ladspa=file:label:[<control0>,<control1>,...]``
Load a LADSPA (Linux Audio Developer's Simple Plugin API) plugin. This
filter is reentrant, so multiple LADSPA plugins can be used at once.
``<file>``
Specifies the LADSPA plugin library file.
.. note::
See also the note about the ``LADSPA_PATH`` variable in the
`ENVIRONMENT VARIABLES`_ section.
``<label>``
Specifies the filter within the library. Some libraries contain only
one filter, but others contain many of them. Entering 'help' here
will list all available filters within the specified library, which
eliminates the use of 'listplugins' from the LADSPA SDK.
``[<control0>,<control1>,...]``
Controls are zero or more ``,`` separated floating point values that
determine the behavior of the loaded plugin (for example delay,
threshold or gain).
In verbose mode (add ``-v`` to the mpv command line), all
available controls and their valid ranges are printed. This eliminates
the use of 'analyseplugin' from the LADSPA SDK.
Note that ``,`` is already used by the option parser to separate
filters, so you must quote the list of values with ``[...]`` or
similar.
.. admonition:: Example
``mpv --af=ladspa='/usr/lib/ladspa/delay.so':delay_5s:[0.5,0.2] media.avi``
Does something.
``karaoke``
Simple voice removal filter exploiting the fact that voice is usually
recorded with mono gear and later 'center' mixed onto the final audio
stream. Beware that this filter will turn your signal into mono. Works
well for 2 channel tracks; do not bother trying it on anything but 2
channel stereo.
``scaletempo[=option1:option2:...]``
Scales audio tempo without altering pitch, optionally synced to playback
speed (default).