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AUDIO FILTERS
=============
Audio filters allow you to modify the audio stream and its properties. The
syntax is:
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``--af=<filter1[=parameter1:parameter2:...],filter2,...>``
Setup a chain of audio filters.
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.. note::
To get a full list of available audio filters, see ``--af=help``.
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Also, keep in mind that most actual filters are available via the ``lavfi``
wrapper, which gives you access to most of libavfilter's filters. This
includes all filters that have been ported from MPlayer to libavfilter.
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
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filter list:
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``--af-add=<filter1[,filter2,...]>``
Appends the filters given as arguments to the filter list.
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``--af-pre=<filter1[,filter2,...]>``
Prepends the filters given as arguments to the filter list.
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``--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).
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``--af-clr``
Completely empties the filter list.
Available filters are:
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``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.
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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>``
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Length of the filter with respect to the lower sampling rate. (default:
16)
``phase-shift=<count>``
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Log2 of the number of polyphase entries. (..., 10->1024, 11->2048,
12->4096, ...) (default: 10->1024)
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``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:...``.)
``normalize=<yes|no>``
Whether to normalize when remixing channel layouts (default: yes). This
is e.g. applied when downmixing surround audio to stereo. The advantage
is that this guarantees that no clipping can happen. Unfortunately,
this can also lead to too low volume levels. Whether you enable or
disable this is essentially a matter of taste, but the default uses
the safer choice.
``o=<string>``
Set AVOptions on the SwrContext or AVAudioResampleContext. These should
be documented by FFmpeg or Libav.
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``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
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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
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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).
``equalizer=g1:g2:g3:...:g10``
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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
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the sound using a resampling filter before it reaches this filter.
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``<g1>:<g2>:<g3>:...:<g10>``
floating point numbers representing the gain in dB for each frequency
band (-12-12)
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.. admonition:: Example
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``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
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while canceling it almost completely around 1 kHz.
``channels=nch[:routes]``
Can be used for adding, removing, routing and copying audio channels. If
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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.
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``<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.
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.. admonition:: Examples
``mpv --af=channels=4:[0-1,1-0,2-2,3-3] media.avi``
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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``
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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.
audio/filter: split af_format into separate filters, rename af_force af_format is the old audio conversion filter. It could do all possible conversions supported by the audio chain. However, ever since the addition of af_lavrresample, most conversions are done by libav/swresample, and af_format is used as fallback. Separate out the fallback cases and remove af_format. af_convert24 does 24 bit <-> 32 bit conversions, while af_convertsignendian does sign and endian conversions. Maybe the way the conversions are split sounds a bit odd. But the former changes the size of the audio data, while the latter is fully in-place, so there's at least different buffer management. This requires a quite complicated algorithm to make sure all these "partial" conversion filters can actually get from one format to another. E.g. s24le->s32be always requires convertsignendian and convert24, but af.c has no idea what the intermediate format should be. So I added a graph search (trying every possible format and filter) to determine required format and filter. When I wrote this, it seemed this was still better than messing everything into af_lavrresample, but maybe this is overkill and I'll change my opinion. For now, it seems nice to get rid of af_format though. The AC3->IEC61937 conversion isn't supported anymore, but I don't think this is needed anywhere. Most AOs test all formats explicitly, or use the AF_FORMAT_IS_IEC61937() macro (which includes AC3). One positive consequence of this change is that conversions always include dithering (done by libav/swresample), instead of possibly going through af_format, which doesn't do anything fancy. Rename af_force to af_format. It's essentially compatible with command line uses of af_format. We retain a compatibility alias for af_force.
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``format=format:srate:channels:out-format:out-srate:out-channels``
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Does not do any format conversion itself. Rather, it may cause the
filter system to insert necessary conversion filters before or after this
filter if needed. It is primarily useful for controlling the audio format
going into other filters. To specify the format for audio output, see
``--audio-format``, ``--audio-samplerate``, and ``--audio-channels``. This
filter is able to force a particular format, whereas ``--audio-*``
may be overridden by the ao based on output compatibility.
All parameters are optional. The first 3 parameters restrict what the filter
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accepts as input. They will therefore cause conversion filters to be
inserted before this one. The ``out-`` parameters tell the filters or audio
outputs following this filter how to interpret the data without actually
doing a conversion. Setting these will probably just break things unless you
really know you want this for some reason, such as testing or dealing with
broken media.
audio/filter: split af_format into separate filters, rename af_force af_format is the old audio conversion filter. It could do all possible conversions supported by the audio chain. However, ever since the addition of af_lavrresample, most conversions are done by libav/swresample, and af_format is used as fallback. Separate out the fallback cases and remove af_format. af_convert24 does 24 bit <-> 32 bit conversions, while af_convertsignendian does sign and endian conversions. Maybe the way the conversions are split sounds a bit odd. But the former changes the size of the audio data, while the latter is fully in-place, so there's at least different buffer management. This requires a quite complicated algorithm to make sure all these "partial" conversion filters can actually get from one format to another. E.g. s24le->s32be always requires convertsignendian and convert24, but af.c has no idea what the intermediate format should be. So I added a graph search (trying every possible format and filter) to determine required format and filter. When I wrote this, it seemed this was still better than messing everything into af_lavrresample, but maybe this is overkill and I'll change my opinion. For now, it seems nice to get rid of af_format though. The AC3->IEC61937 conversion isn't supported anymore, but I don't think this is needed anywhere. Most AOs test all formats explicitly, or use the AF_FORMAT_IS_IEC61937() macro (which includes AC3). One positive consequence of this change is that conversions always include dithering (done by libav/swresample), instead of possibly going through af_format, which doesn't do anything fancy. Rename af_force to af_format. It's essentially compatible with command line uses of af_format. We retain a compatibility alias for af_force.
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``<format>``
Force conversion to this format. Use ``--af=format=format=help`` to get
a list of valid formats.
audio/filter: split af_format into separate filters, rename af_force af_format is the old audio conversion filter. It could do all possible conversions supported by the audio chain. However, ever since the addition of af_lavrresample, most conversions are done by libav/swresample, and af_format is used as fallback. Separate out the fallback cases and remove af_format. af_convert24 does 24 bit <-> 32 bit conversions, while af_convertsignendian does sign and endian conversions. Maybe the way the conversions are split sounds a bit odd. But the former changes the size of the audio data, while the latter is fully in-place, so there's at least different buffer management. This requires a quite complicated algorithm to make sure all these "partial" conversion filters can actually get from one format to another. E.g. s24le->s32be always requires convertsignendian and convert24, but af.c has no idea what the intermediate format should be. So I added a graph search (trying every possible format and filter) to determine required format and filter. When I wrote this, it seemed this was still better than messing everything into af_lavrresample, but maybe this is overkill and I'll change my opinion. For now, it seems nice to get rid of af_format though. The AC3->IEC61937 conversion isn't supported anymore, but I don't think this is needed anywhere. Most AOs test all formats explicitly, or use the AF_FORMAT_IS_IEC61937() macro (which includes AC3). One positive consequence of this change is that conversions always include dithering (done by libav/swresample), instead of possibly going through af_format, which doesn't do anything fancy. Rename af_force to af_format. It's essentially compatible with command line uses of af_format. We retain a compatibility alias for af_force.
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``<srate>``
Force conversion to a specific sample rate. The rate is an integer,
48000 for example.
audio/filter: split af_format into separate filters, rename af_force af_format is the old audio conversion filter. It could do all possible conversions supported by the audio chain. However, ever since the addition of af_lavrresample, most conversions are done by libav/swresample, and af_format is used as fallback. Separate out the fallback cases and remove af_format. af_convert24 does 24 bit <-> 32 bit conversions, while af_convertsignendian does sign and endian conversions. Maybe the way the conversions are split sounds a bit odd. But the former changes the size of the audio data, while the latter is fully in-place, so there's at least different buffer management. This requires a quite complicated algorithm to make sure all these "partial" conversion filters can actually get from one format to another. E.g. s24le->s32be always requires convertsignendian and convert24, but af.c has no idea what the intermediate format should be. So I added a graph search (trying every possible format and filter) to determine required format and filter. When I wrote this, it seemed this was still better than messing everything into af_lavrresample, but maybe this is overkill and I'll change my opinion. For now, it seems nice to get rid of af_format though. The AC3->IEC61937 conversion isn't supported anymore, but I don't think this is needed anywhere. Most AOs test all formats explicitly, or use the AF_FORMAT_IS_IEC61937() macro (which includes AC3). One positive consequence of this change is that conversions always include dithering (done by libav/swresample), instead of possibly going through af_format, which doesn't do anything fancy. Rename af_force to af_format. It's essentially compatible with command line uses of af_format. We retain a compatibility alias for af_force.
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``<channels>``
Force mixing to a specific channel layout. See ``--audio-channels`` option
for possible values.
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``<out-format>``
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``<out-srate>``
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``<out-channels>``
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*NOTE*: this filter used to be named ``force``. The old ``format`` filter
used to do conversion itself, unlike this one which lets the filter system
handle the conversion.
audio/filter: split af_format into separate filters, rename af_force af_format is the old audio conversion filter. It could do all possible conversions supported by the audio chain. However, ever since the addition of af_lavrresample, most conversions are done by libav/swresample, and af_format is used as fallback. Separate out the fallback cases and remove af_format. af_convert24 does 24 bit <-> 32 bit conversions, while af_convertsignendian does sign and endian conversions. Maybe the way the conversions are split sounds a bit odd. But the former changes the size of the audio data, while the latter is fully in-place, so there's at least different buffer management. This requires a quite complicated algorithm to make sure all these "partial" conversion filters can actually get from one format to another. E.g. s24le->s32be always requires convertsignendian and convert24, but af.c has no idea what the intermediate format should be. So I added a graph search (trying every possible format and filter) to determine required format and filter. When I wrote this, it seemed this was still better than messing everything into af_lavrresample, but maybe this is overkill and I'll change my opinion. For now, it seems nice to get rid of af_format though. The AC3->IEC61937 conversion isn't supported anymore, but I don't think this is needed anywhere. Most AOs test all formats explicitly, or use the AF_FORMAT_IS_IEC61937() macro (which includes AC3). One positive consequence of this change is that conversions always include dithering (done by libav/swresample), instead of possibly going through af_format, which doesn't do anything fancy. Rename af_force to af_format. It's essentially compatible with command line uses of af_format. We retain a compatibility alias for af_force.
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``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
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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>``
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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.
``replaygain-fallback``
Gain in dB to apply if the file has no replay gain tags. This option
is always applied if the replaygain logic is somehow inactive. If this
is applied, no other replaygain options are applied.
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``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.
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``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.)
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.. admonition:: Example
``mpv --af=volume=10.1 media.avi``
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Would amplify the sound by 10.1 dB and hard-clip if the sound level
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is too high.
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``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,
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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.
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``<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.
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.. admonition:: Examples
``mpv --af=pan=1:[0.5,0.5] media.avi``
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Would downmix from stereo to mono.
``mpv --af=pan=3:[1,0,0.5,0,1,0.5] media.avi``
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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.
``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).
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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``.
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.. admonition:: Example
``mpv --af=delay=[10.5,10.5,0,0,7,0] media.avi``
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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.
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``drc[=method:target]``
Applies dynamic range compression. This maximizes the volume by compressing
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the audio signal's dynamic range. (Formerly called ``volnorm``.)
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``<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.
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``<target>``
Sets the target amplitude as a fraction of the maximum for the sample
type (default: 0.25).
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.. note::
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This filter can cause distortion with audio signals that have a very
large dynamic range.
``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.
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``scale=<amount>``
Nominal amount to scale tempo. Scales this amount in addition to
speed. (default: 1.0)
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``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)
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``overlap=<percent>``
Percentage of stride to overlap. Decreasing improves performance.
(default: .20)
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``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)
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``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.
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Add this to your ``input.conf`` to step by musical semi-tones::
[ multiply speed 0.9438743126816935
] multiply speed 1.059463094352953
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.. warning::
Loses sync with video.
both
Scale both tempo and pitch.
none
Ignore speed changes.
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.. admonition:: Examples
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``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.
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``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.
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``mpv --af=scaletempo=stride=30:overlap=.50:search=10 media.ogg``
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Would tweak the quality and performance parameters.
``mpv --af=format=float,scaletempo media.ogg``
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Would make scaletempo use float code. Maybe faster on some
platforms.
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``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``.)
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``lavfi=graph``
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Filter audio using FFmpeg's libavfilter.
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``<graph>``
Libavfilter graph. See ``lavfi`` video filter for details - the graph
syntax is the same.
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.. warning::
Don't forget to quote libavfilter graphs as described in the lavfi
video filter section.
``o=<string>``
AVOptions.