mirror of
https://github.com/mpv-player/mpv
synced 2024-12-24 15:52:25 +00:00
633 lines
25 KiB
ReStructuredText
633 lines
25 KiB
ReStructuredText
AUDIO FILTERS
|
|
=============
|
|
|
|
Audio filters allow you to modify the audio stream and its properties. The
|
|
syntax is:
|
|
|
|
``--af=<filter1[=parameter1:parameter2:...],filter2,...>``
|
|
Setup a chain of audio filters.
|
|
|
|
.. note::
|
|
|
|
To get a full list of available audio filters, see ``--af=help``.
|
|
|
|
You can also set defaults for each filter. The defaults are applied before the
|
|
normal filter parameters.
|
|
|
|
``--af-defaults=<filter1[=parameter1:parameter2:...],filter2,...>``
|
|
Set defaults for each filter.
|
|
|
|
Audio filters are managed in lists. There are a few commands to manage the
|
|
filter list:
|
|
|
|
``--af-add=<filter1[,filter2,...]>``
|
|
Appends the filters given as arguments to the filter list.
|
|
|
|
``--af-pre=<filter1[,filter2,...]>``
|
|
Prepends the filters given as arguments to the filter list.
|
|
|
|
``--af-del=<index1[,index2,...]>``
|
|
Deletes the filters at the given indexes. Index numbers start at 0,
|
|
negative numbers address the end of the list (-1 is the last).
|
|
|
|
``--af-clr``
|
|
Completely empties the filter list.
|
|
|
|
Available filters are:
|
|
|
|
``lavrresample[=option1:option2:...]``
|
|
This filter uses libavresample (or libswresample, depending on the build)
|
|
to change sample rate, sample format, or channel layout of the audio stream.
|
|
This filter is automatically enabled if the audio output does not support
|
|
the audio configuration of the file being played.
|
|
|
|
It supports only the following sample formats: u8, s16, s32, float.
|
|
|
|
``filter-size=<length>``
|
|
Length of the filter with respect to the lower sampling rate. (default:
|
|
16)
|
|
``phase-shift=<count>``
|
|
Log2 of the number of polyphase entries. (..., 10->1024, 11->2048,
|
|
12->4096, ...) (default: 10->1024)
|
|
``cutoff=<cutoff>``
|
|
Cutoff frequency (0.0-1.0), default set depending upon filter length.
|
|
``linear``
|
|
If set then filters will be linearly interpolated between polyphase
|
|
entries. (default: no)
|
|
``no-detach``
|
|
Do not detach if input and output audio format/rate/channels match.
|
|
(If you just want to set defaults for this filter that will be used
|
|
even by automatically inserted lavrresample instances, you should
|
|
prefer setting them with ``--af-defaults=lavrresample:...``.)
|
|
``o=<string>``
|
|
Set AVOptions on the SwrContext or AVAudioResampleContext. These should
|
|
be documented by FFmpeg or Libav.
|
|
|
|
``lavcac3enc[=tospdif[:bitrate[:minch]]]``
|
|
Encode multi-channel audio to AC-3 at runtime using libavcodec. Supports
|
|
16-bit native-endian input format, maximum 6 channels. The output is
|
|
big-endian when outputting a raw AC-3 stream, native-endian when
|
|
outputting to S/PDIF. If the input sample rate is not 48 kHz, 44.1 kHz or
|
|
32 kHz, it will be resampled to 48 kHz.
|
|
|
|
``tospdif=<yes|no>``
|
|
Output raw AC-3 stream if ``no``, output to S/PDIF for
|
|
pass-through if ``yes`` (default).
|
|
|
|
``bitrate=<rate>``
|
|
The bitrate use for the AC-3 stream. Set it to 384 to get 384 kbps.
|
|
|
|
The default is 640. Some receivers might not be able to handle this.
|
|
|
|
Valid values: 32, 40, 48, 56, 64, 80, 96, 112, 128,
|
|
160, 192, 224, 256, 320, 384, 448, 512, 576, 640.
|
|
|
|
The special value ``auto`` selects a default bitrate based on the
|
|
input channel number:
|
|
|
|
:1ch: 96
|
|
:2ch: 192
|
|
:3ch: 224
|
|
:4ch: 384
|
|
:5ch: 448
|
|
:6ch: 448
|
|
|
|
``minch=<n>``
|
|
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
|
|
loudspeakers, allowing each ear to hear both channels and taking into
|
|
account the distance difference and the head shadowing effect. It is
|
|
applicable only to 2-channel audio.
|
|
|
|
``fcut=<300-1000>``
|
|
Set cut frequency in Hz.
|
|
``feed=<10-150>``
|
|
Set feed level for low frequencies in 0.1*dB.
|
|
``profile=<value>``
|
|
Several profiles are available for convenience:
|
|
|
|
:default: will be used if nothing else was specified (fcut=700,
|
|
feed=45)
|
|
:cmoy: Chu Moy circuit implementation (fcut=700, feed=60)
|
|
:jmeier: Jan Meier circuit implementation (fcut=650, feed=95)
|
|
|
|
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
|
|
being played back. The center frequencies for the 10 bands are:
|
|
|
|
=== ==========
|
|
No. frequency
|
|
=== ==========
|
|
0 31.25 Hz
|
|
1 62.50 Hz
|
|
2 125.00 Hz
|
|
3 250.00 Hz
|
|
4 500.00 Hz
|
|
5 1.00 kHz
|
|
6 2.00 kHz
|
|
7 4.00 kHz
|
|
8 8.00 kHz
|
|
9 16.00 kHz
|
|
=== ==========
|
|
|
|
If the sample rate of the sound being played is lower than the center
|
|
frequency for a frequency band, then that band will be disabled. A known
|
|
bug with this filter is that the characteristics for the uppermost band
|
|
are not completely symmetric if the sample rate is close to the center
|
|
frequency of that band. This problem can be worked around by upsampling
|
|
the sound using a resampling filter before it reaches this filter.
|
|
|
|
``<g1>:<g2>:<g3>:...:<g10>``
|
|
floating point numbers representing the gain in dB for each frequency
|
|
band (-12-12)
|
|
|
|
.. admonition:: Example
|
|
|
|
``mpv --af=equalizer=11:11:10:5:0:-12:0:5:12:12 media.avi``
|
|
Would amplify the sound in the upper and lower frequency region
|
|
while canceling it almost completely around 1 kHz.
|
|
|
|
``channels=nch[:routes]``
|
|
Can be used for adding, removing, routing and copying audio channels. If
|
|
only ``<nch>`` is given, the default routing is used. It works as follows:
|
|
If the number of output channels is greater than the number of input
|
|
channels, empty channels are inserted (except when mixing from mono to
|
|
stereo; then the mono channel is duplicated). If the number of output
|
|
channels is less than the number of input channels, the exceeding
|
|
channels are truncated.
|
|
|
|
``<nch>``
|
|
number of output channels (1-8)
|
|
``<routes>``
|
|
List of ``,`` separated routes, in the form ``from1-to1,from2-to2,...``.
|
|
Each pair defines where to route each channel. There can be at most
|
|
8 routes. Without this argument, the default routing is used. Since
|
|
``,`` is also used to separate filters, you must quote this argument
|
|
with ``[...]`` or similar.
|
|
|
|
.. admonition:: Examples
|
|
|
|
``mpv --af=channels=4:[0-1,1-0,0-2,1-3] media.avi``
|
|
Would change the number of channels to 4 and set up 4 routes that
|
|
swap channel 0 and channel 1 and leave channel 2 and 3 intact.
|
|
Observe that if media containing two channels were played back,
|
|
channels 2 and 3 would contain silence but 0 and 1 would still be
|
|
swapped.
|
|
|
|
``mpv --af=channels=6:[0-0,0-1,0-2,0-3] media.avi``
|
|
Would change the number of channels to 6 and set up 4 routes that
|
|
copy channel 0 to channels 0 to 3. Channel 4 and 5 will contain
|
|
silence.
|
|
|
|
.. note::
|
|
|
|
You should probably not use this filter. If you want to change the
|
|
output channel layout, try the ``format`` filter, which can make mpv
|
|
automatically up- and downmix standard channel layouts.
|
|
|
|
``format=format:srate:channels:out-format:out-srate:out-channels``
|
|
Force a specific audio format/configuration without actually changing the
|
|
audio data. Keep in mind that the filter system might auto-insert actual
|
|
conversion filters before or after this filter if needed.
|
|
|
|
All parameters are optional. The first 3 parameters restrict what the filter
|
|
accepts as input. The ``out-`` parameters change the audio format, without
|
|
actually doing a conversion. The data will be 'reinterpreted' by the
|
|
filters or audio outputs following this filter.
|
|
|
|
``<format>``
|
|
Force conversion to this format. Use ``--af=format=format=help`` to get
|
|
a list of valid formats.
|
|
|
|
``<srate>``
|
|
Force conversion to a specific sample rate. The rate is an integer,
|
|
48000 for example.
|
|
|
|
``<channels>``
|
|
Force mixing to a specific channel layout. See ``--audio-channels`` option
|
|
for possible values.
|
|
|
|
``<out-format>``
|
|
|
|
``<out-srate>``
|
|
|
|
``<out-channels>``
|
|
|
|
See also ``--audio-format``, ``--audio-samplerate``, and
|
|
``--audio-channels`` for related options. Keep in mind that
|
|
``--audio-channels`` does not actually force the number of
|
|
channels in most cases, while this filter can do this.
|
|
|
|
*NOTE*: this filter used to be named ``force``. Also, unlike the old
|
|
``format`` filter, this does not do any actual conversion anymore.
|
|
Conversion is done by other, automatically inserted filters.
|
|
|
|
``convert24``
|
|
Filter for internal use only. Converts between 24-bit and 32-bit sample
|
|
formats.
|
|
|
|
``convertsign``
|
|
Filter for internal use only. Converts between signed/unsigned formats.
|
|
|
|
``volume[=<volumedb>[:...]]``
|
|
Implements software volume control. Use this filter with caution since it
|
|
can reduce the signal to noise ratio of the sound. In most cases it is
|
|
best to use the *Master* volume control of your sound card or the volume
|
|
knob on your amplifier.
|
|
|
|
*NOTE*: This filter is not reentrant and can therefore only be enabled
|
|
once for every audio stream.
|
|
|
|
``<volumedb>``
|
|
Sets the desired gain in dB for all channels in the stream from -200 dB
|
|
to +60 dB, where -200 dB mutes the sound completely and +60 dB equals a
|
|
gain of 1000 (default: 0).
|
|
``replaygain-track``
|
|
Adjust volume gain according to the track-gain replaygain value stored
|
|
in the file metadata.
|
|
``replaygain-album``
|
|
Like replaygain-track, but using the album-gain value instead.
|
|
``replaygain-preamp``
|
|
Pre-amplification gain in dB to apply to the selected replaygain gain
|
|
(default: 0).
|
|
``replaygain-clip=yes|no``
|
|
Prevent clipping caused by replaygain by automatically lowering the
|
|
gain (default). Use ``replaygain-clip=no`` to disable this.
|
|
``softclip``
|
|
Turns soft clipping on. Soft-clipping can make the
|
|
sound more smooth if very high volume levels are used. Enable this
|
|
option if the dynamic range of the loudspeakers is very low.
|
|
|
|
*WARNING*: This feature creates distortion and should be considered a
|
|
last resort.
|
|
``s16``
|
|
Force S16 sample format if set. Lower quality, but might be faster
|
|
in some situations.
|
|
``detach``
|
|
Remove the filter if the volume is not changed at audio filter config
|
|
time. Useful with replaygain: if the current file has no replaygain
|
|
tags, then the filter will be removed if this option is enabled.
|
|
(If ``--softvol=yes`` is used and the player volume controls are used
|
|
during playback, a different volume filter will be inserted.)
|
|
|
|
.. admonition:: Example
|
|
|
|
``mpv --af=volume=10.1 media.avi``
|
|
Would amplify the sound by 10.1 dB and hard-clip if the sound level
|
|
is too high.
|
|
|
|
``pan=n:[<matrix>]``
|
|
Mixes channels arbitrarily. Basically a combination of the volume and the
|
|
channels filter that can be used to down-mix many channels to only a few,
|
|
e.g. stereo to mono, or vary the "width" of the center speaker in a
|
|
surround sound system. This filter is hard to use, and will require some
|
|
tinkering before the desired result is obtained. The number of options for
|
|
this filter depends on the number of output channels. An example how to
|
|
downmix a six-channel file to two channels with this filter can be found
|
|
in the examples section near the end.
|
|
|
|
``<n>``
|
|
Number of output channels (1-8).
|
|
``<matrix>``
|
|
A list of values ``[L00,L01,L02,...,L10,L11,L12,...,Ln0,Ln1,Ln2,...]``,
|
|
where each element ``Lij`` means how much of input channel i is mixed
|
|
into output channel j (range 0-1). So in principle you first have n
|
|
numbers saying what to do with the first input channel, then n numbers
|
|
that act on the second input channel etc. If you do not specify any
|
|
numbers for some input channels, 0 is assumed.
|
|
Note that the values are separated by ``,``, which is already used
|
|
by the option parser to separate filters. This is why you must quote
|
|
the value list with ``[...]`` or similar.
|
|
|
|
.. admonition:: Examples
|
|
|
|
``mpv --af=pan=1:[0.5,0.5] media.avi``
|
|
Would downmix from stereo to mono.
|
|
|
|
``mpv --af=pan=3:[1,0,0.5,0,1,0.5] media.avi``
|
|
Would give 3 channel output leaving channels 0 and 1 intact, and mix
|
|
channels 0 and 1 into output channel 2 (which could be sent to a
|
|
subwoofer for example).
|
|
|
|
.. note::
|
|
|
|
If you just want to force remixing to a certain output channel layout,
|
|
it is easier to use the ``format`` filter. For example,
|
|
``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
|
|
only useful if you have more than 2 loudspeakers.
|
|
|
|
``[ch1,ch2,...]``
|
|
The delay in ms that should be imposed on each channel (floating point
|
|
number between 0 and 1000).
|
|
|
|
To calculate the required delay for the different channels, do as follows:
|
|
|
|
1. Measure the distance to the loudspeakers in meters in relation to your
|
|
listening position, giving you the distances s1 to s5 (for a 5.1
|
|
system). There is no point in compensating for the subwoofer (you will
|
|
not hear the difference anyway).
|
|
|
|
2. Subtract the distances s1 to s5 from the maximum distance, i.e.
|
|
``s[i] = max(s) - s[i]; i = 1...5``.
|
|
|
|
3. Calculate the required delays in ms as ``d[i] = 1000*s[i]/342; i =
|
|
1...5``.
|
|
|
|
.. admonition:: Example
|
|
|
|
``mpv --af=delay=[10.5,10.5,0,0,7,0] media.avi``
|
|
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``.)
|
|
|
|
``<method>``
|
|
Sets the used method.
|
|
|
|
1
|
|
Use a single sample to smooth the variations via the standard
|
|
weighted mean over past samples (default).
|
|
2
|
|
Use several samples to smooth the variations via the standard
|
|
weighted mean over past samples.
|
|
|
|
``<target>``
|
|
Sets the target amplitude as a fraction of the maximum for the sample
|
|
type (default: 0.25).
|
|
|
|
.. note::
|
|
|
|
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).
|
|
|
|
This works by playing 'stride' ms of audio at normal speed then consuming
|
|
'stride*scale' ms of input audio. It pieces the strides together by
|
|
blending 'overlap'% of stride with audio following the previous stride. It
|
|
optionally performs a short statistical analysis on the next 'search' ms
|
|
of audio to determine the best overlap position.
|
|
|
|
``scale=<amount>``
|
|
Nominal amount to scale tempo. Scales this amount in addition to
|
|
speed. (default: 1.0)
|
|
``stride=<amount>``
|
|
Length in milliseconds to output each stride. Too high of a value will
|
|
cause noticeable skips at high scale amounts and an echo at low scale
|
|
amounts. Very low values will alter pitch. Increasing improves
|
|
performance. (default: 60)
|
|
``overlap=<percent>``
|
|
Percentage of stride to overlap. Decreasing improves performance.
|
|
(default: .20)
|
|
``search=<amount>``
|
|
Length in milliseconds to search for best overlap position. Decreasing
|
|
improves performance greatly. On slow systems, you will probably want
|
|
to set this very low. (default: 14)
|
|
``speed=<tempo|pitch|both|none>``
|
|
Set response to speed change.
|
|
|
|
tempo
|
|
Scale tempo in sync with speed (default).
|
|
pitch
|
|
Reverses effect of filter. Scales pitch without altering tempo.
|
|
Add this to your ``input.conf`` to step by musical semi-tones::
|
|
|
|
[ multiply speed 0.9438743126816935
|
|
] multiply speed 1.059463094352953
|
|
|
|
.. warning::
|
|
|
|
Loses sync with video.
|
|
both
|
|
Scale both tempo and pitch.
|
|
none
|
|
Ignore speed changes.
|
|
|
|
.. admonition:: Examples
|
|
|
|
``mpv --af=scaletempo --speed=1.2 media.ogg``
|
|
Would play media at 1.2x normal speed, with audio at normal
|
|
pitch. Changing playback speed would change audio tempo to match.
|
|
|
|
``mpv --af=scaletempo=scale=1.2:speed=none --speed=1.2 media.ogg``
|
|
Would play media at 1.2x normal speed, with audio at normal
|
|
pitch, but changing playback speed would have no effect on audio
|
|
tempo.
|
|
|
|
``mpv --af=scaletempo=stride=30:overlap=.50:search=10 media.ogg``
|
|
Would tweak the quality and performance parameters.
|
|
|
|
``mpv --af=format=float,scaletempo media.ogg``
|
|
Would make scaletempo use float code. Maybe faster on some
|
|
platforms.
|
|
|
|
``mpv --af=scaletempo=scale=1.2:speed=pitch audio.ogg``
|
|
Would play media at 1.2x normal speed, with audio at normal pitch.
|
|
Changing playback speed would change pitch, leaving audio tempo at
|
|
1.2x.
|
|
|
|
``rubberband``
|
|
High quality pitch correction with librubberband. This can be used in place
|
|
of ``scaletempo``, and will be used to adjust audio pitch when playing
|
|
at speed different from normal.
|
|
|
|
This filter has a number of sub-options. You can list them with
|
|
``mpv --af=rubberband=help``. This will also show the default values
|
|
for each option. The options are not documented here, because they are
|
|
merely passed to librubberband. Look at the librubberband documentation
|
|
to learn what each option does:
|
|
http://breakfastquay.com/rubberband/code-doc/classRubberBand_1_1RubberBandStretcher.html
|
|
(The mapping of the mpv rubberband filter sub-option names and values to
|
|
those of librubberband follows a simple pattern: ``"Option" + Name + Value``.)
|
|
|
|
``lavfi=graph``
|
|
Filter audio using FFmpeg's libavfilter.
|
|
|
|
``<graph>``
|
|
Libavfilter graph. See ``lavfi`` video filter for details - the graph
|
|
syntax is the same.
|
|
|
|
.. warning::
|
|
|
|
Don't forget to quote libavfilter graphs as described in the lavfi
|
|
video filter section.
|
|
|
|
``o=<string>``
|
|
AVOptions.
|
|
|