mirror of https://github.com/mpv-player/mpv
618 lines
24 KiB
ReStructuredText
618 lines
24 KiB
ReStructuredText
AUDIO FILTERS
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=============
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Audio filters allow you to modify the audio stream and its properties. The
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syntax is:
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``--af=<filter1[=parameter1:parameter2:...],filter2,...>``
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Setup a chain of audio filters.
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.. note::
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To get a full list of available audio filters, see ``--af=help``.
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You can also set defaults for each filter. The defaults are applied before the
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normal filter parameters.
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``--af-defaults=<filter1[=parameter1:parameter2:...],filter2,...>``
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Set defaults for each filter.
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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,...]>``
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Appends the filters given as arguments to the filter list.
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``--af-pre=<filter1[,filter2,...]>``
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Prepends the filters given as arguments to the filter list.
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``--af-del=<index1[,index2,...]>``
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Deletes the filters at the given indexes. Index numbers start at 0,
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negative numbers address the end of the list (-1 is the last).
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``--af-clr``
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Completely empties the filter list.
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Available filters are:
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``lavrresample[=option1:option2:...]``
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This filter uses libavresample (or libswresample, depending on the build)
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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
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the audio configuration of the file being played.
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It supports only the following sample formats: u8, s16, s32, float.
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``filter-size=<length>``
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Length of the filter with respect to the lower sampling rate. (default:
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16)
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``phase-shift=<count>``
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Log2 of the number of polyphase entries. (..., 10->1024, 11->2048,
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12->4096, ...) (default: 10->1024)
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``cutoff=<cutoff>``
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Cutoff frequency (0.0-1.0), default set depending upon filter length.
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``linear``
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If set then filters will be linearly interpolated between polyphase
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entries. (default: no)
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``no-detach``
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Do not detach if input and output audio format/rate/channels match.
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(If you just want to set defaults for this filter that will be used
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even by automatically inserted lavrresample instances, you should
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prefer setting them with ``--af-defaults=lavrresample:...``.)
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``o=<string>``
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Set AVOptions on the SwrContext or AVAudioResampleContext. These should
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be documented by FFmpeg or Libav.
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``lavcac3enc[=tospdif[:bitrate[:minchn]]]``
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Encode multi-channel audio to AC-3 at runtime using libavcodec. Supports
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16-bit native-endian input format, maximum 6 channels. The output is
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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
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32 kHz, it will be resampled to 48 kHz.
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``tospdif=<yes|no>``
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Output raw AC-3 stream if ``no``, output to S/PDIF for
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pass-through if ``yes`` (default).
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``bitrate=<rate>``
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The bitrate use for the AC-3 stream. Set it to 384 to get 384 kbps.
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The default is 640. Some receivers might not be able to handle this.
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Valid values: 32, 40, 48, 56, 64, 80, 96, 112, 128,
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160, 192, 224, 256, 320, 384, 448, 512, 576, 640.
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The special value ``auto`` selects a default bitrate based on the
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input channel number:
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:1ch: 96
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:2ch: 192
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:3ch: 224
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:4ch: 384
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:5ch: 448
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:6ch: 448
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``minchn=<n>``
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If the input channel number is less than ``<minchn>``, the filter will
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detach itself (default: 3).
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``sweep[=speed]``
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Produces a sine sweep.
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``<0.0-1.0>``
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Sine function delta, use very low values to hear the sweep.
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``sinesuppress[=freq:decay]``
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Remove a sine at the specified frequency. Useful to get rid of the 50/60 Hz
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noise on low quality audio equipment. It only works on mono input.
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``<freq>``
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The frequency of the sine which should be removed (in Hz) (default:
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50)
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``<decay>``
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Controls the adaptivity (a larger value will make the filter adapt to
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amplitude and phase changes quicker, a smaller value will make the
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adaptation slower) (default: 0.0001). Reasonable values are around
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0.001.
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``bs2b[=option1:option2:...]``
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Bauer stereophonic to binaural transformation using libbs2b. Improves the
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headphone listening experience by making the sound similar to that from
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loudspeakers, allowing each ear to hear both channels and taking into
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account the distance difference and the head shadowing effect. It is
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applicable only to 2-channel audio.
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``fcut=<300-1000>``
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Set cut frequency in Hz.
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``feed=<10-150>``
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Set feed level for low frequencies in 0.1*dB.
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``profile=<value>``
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Several profiles are available for convenience:
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:default: will be used if nothing else was specified (fcut=700,
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feed=45)
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:cmoy: Chu Moy circuit implementation (fcut=700, feed=60)
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:jmeier: Jan Meier circuit implementation (fcut=650, feed=95)
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If ``fcut`` or ``feed`` options are specified together with a profile, they
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will be applied on top of the selected profile.
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``hrtf[=flag]``
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Head-related transfer function: Converts multichannel audio to 2-channel
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output for headphones, preserving the spatiality of the sound.
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==== ===================================
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Flag Meaning
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==== ===================================
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m matrix decoding of the rear channel
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s 2-channel matrix decoding
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0 no matrix decoding (default)
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==== ===================================
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``equalizer=g1:g2:g3:...:g10``
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10 octave band graphic equalizer, implemented using 10 IIR band-pass
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filters. This means that it works regardless of what type of audio is
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being played back. The center frequencies for the 10 bands are:
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=== ==========
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No. frequency
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=== ==========
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0 31.25 Hz
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1 62.50 Hz
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2 125.00 Hz
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3 250.00 Hz
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4 500.00 Hz
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5 1.00 kHz
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6 2.00 kHz
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7 4.00 kHz
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8 8.00 kHz
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9 16.00 kHz
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=== ==========
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If the sample rate of the sound being played is lower than the center
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frequency for a frequency band, then that band will be disabled. A known
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bug with this filter is that the characteristics for the uppermost band
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are not completely symmetric if the sample rate is close to the center
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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>``
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floating point numbers representing the gain in dB for each frequency
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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``
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Would amplify the sound in the upper and lower frequency region
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while canceling it almost completely around 1 kHz.
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``channels=nch[:routes]``
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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:
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If the number of output channels is greater than the number of input
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channels, empty channels are inserted (except when mixing from mono to
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stereo; then the mono channel is duplicated). If the number of output
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channels is less than the number of input channels, the exceeding
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channels are truncated.
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``<nch>``
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number of output channels (1-8)
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``<routes>``
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List of ``,`` separated routes, in the form ``from1-to1,from2-to2,...``.
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Each pair defines where to route each channel. There can be at most
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8 routes. Without this argument, the default routing is used. Since
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``,`` is also used to separate filters, you must quote this argument
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with ``[...]`` or similar.
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.. admonition:: Examples
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``mpv --af=channels=4:[0-1,1-0,0-2,1-3] media.avi``
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Would change the number of channels to 4 and set up 4 routes that
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swap channel 0 and channel 1 and leave channel 2 and 3 intact.
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Observe that if media containing two channels were played back,
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channels 2 and 3 would contain silence but 0 and 1 would still be
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swapped.
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``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
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copy channel 0 to channels 0 to 3. Channel 4 and 5 will contain
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silence.
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.. note::
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You should probably not use this filter. If you want to change the
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output channel layout, try the ``format`` filter, which can make mpv
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automatically up- and downmix standard channel layouts.
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``format=format:srate:channels:out-format:out-srate:out-channels``
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Force a specific audio format/configuration without actually changing the
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audio data. Keep in mind that the filter system might auto-insert actual
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conversion filters before or after this filter if needed.
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All parameters are optional. The first 3 parameters restrict what the filter
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accepts as input. The ``out-`` parameters change the audio format, without
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actually doing a conversion. The data will be 'reinterpreted' by the
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filters or audio outputs following this filter.
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``<format>``
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Force conversion to this format. Use ``--af=format=format=help`` to get
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a list of valid formats.
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``<srate>``
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Force conversion to a specific sample rate. The rate is an integer,
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48000 for example.
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``<channels>``
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Force mixing to a specific channel layout. See ``--audio-channels`` option
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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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See also ``--audio-format``, ``--audio-samplerate``, and
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``--audio-channels`` for related options. Keep in mind that
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``--audio-channels`` does not actually force the number of
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channels in most cases, while this filter can do this.
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*NOTE*: this filter used to be named ``force``. Also, unlike the old
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``format`` filter, this does not do any actual conversion anymore.
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Conversion is done by other, automatically inserted filters.
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``convert24``
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Filter for internal use only. Converts between 24-bit and 32-bit sample
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formats.
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``convertsign``
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Filter for internal use only. Converts between signed/unsigned formats.
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``volume[=<volumedb>[:...]]``
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Implements software volume control. Use this filter with caution since it
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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
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knob on your amplifier.
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*NOTE*: This filter is not reentrant and can therefore only be enabled
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once for every audio stream.
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``<volumedb>``
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Sets the desired gain in dB for all channels in the stream from -200 dB
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to +60 dB, where -200 dB mutes the sound completely and +60 dB equals a
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gain of 1000 (default: 0).
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``replaygain-track``
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Adjust volume gain according to the track-gain replaygain value stored
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in the file metadata.
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``replaygain-album``
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Like replaygain-track, but using the album-gain value instead.
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``replaygain-preamp``
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Pre-amplification gain in dB to apply to the selected replaygain gain
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(default: 0).
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``replaygain-clip=yes|no``
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Prevent clipping caused by replaygain by automatically lowering the
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gain (default). Use ``replaygain-clip=no`` to disable this.
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``softclip``
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Turns soft clipping on. Soft-clipping can make the
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sound more smooth if very high volume levels are used. Enable this
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option if the dynamic range of the loudspeakers is very low.
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*WARNING*: This feature creates distortion and should be considered a
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last resort.
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``s16``
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Force S16 sample format if set. Lower quality, but might be faster
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in some situations.
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``detach``
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Remove the filter if the volume is not changed at audio filter config
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time. Useful with replaygain: if the current file has no replaygain
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tags, then the filter will be removed if this option is enabled.
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(If ``--softvol=yes`` is used and the player volume controls are used
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during playback, a different volume filter will be inserted.)
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.. admonition:: Example
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``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>]``
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Mixes channels arbitrarily. Basically a combination of the volume and the
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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
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surround sound system. This filter is hard to use, and will require some
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tinkering before the desired result is obtained. The number of options for
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this filter depends on the number of output channels. An example how to
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downmix a six-channel file to two channels with this filter can be found
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in the examples section near the end.
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``<n>``
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Number of output channels (1-8).
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``<matrix>``
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A list of values ``[L00,L01,L02,...,L10,L11,L12,...,Ln0,Ln1,Ln2,...]``,
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where each element ``Lij`` means how much of input channel i is mixed
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into output channel j (range 0-1). So in principle you first have n
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numbers saying what to do with the first input channel, then n numbers
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that act on the second input channel etc. If you do not specify any
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numbers for some input channels, 0 is assumed.
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Note that the values are separated by ``,``, which is already used
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by the option parser to separate filters. This is why you must quote
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the value list with ``[...]`` or similar.
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.. admonition:: Examples
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``mpv --af=pan=1:[0.5,0.5] media.avi``
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Would downmix from stereo to mono.
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``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
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channels 0 and 1 into output channel 2 (which could be sent to a
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subwoofer for example).
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.. note::
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If you just want to force remixing to a certain output channel layout,
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it is easier to use the ``format`` filter. For example,
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``mpv '--af=format=channels=5.1' '--audio-channels=5.1'`` would always force
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remixing audio to 5.1 and output it like this.
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``sub[=fc:ch]``
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Adds a subwoofer channel to the audio stream. The audio data used for
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creating the subwoofer channel is an average of the sound in channel 0 and
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channel 1. The resulting sound is then low-pass filtered by a 4th order
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Butterworth filter with a default cutoff frequency of 60Hz and added to a
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separate channel in the audio stream.
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.. warning::
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Disable this filter when you are playing media with an LFE channel
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(e.g. 5.1 surround sound), otherwise this filter will disrupt the sound
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to the subwoofer.
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``<fc>``
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cutoff frequency in Hz for the low-pass filter (20 Hz to 300 Hz)
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(default: 60 Hz) For the best result try setting the cutoff frequency
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as low as possible. This will improve the stereo or surround sound
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experience.
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``<ch>``
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Determines the channel number in which to insert the sub-channel
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audio. Channel number can be between 0 and 7 (default: 5). Observe
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that the number of channels will automatically be increased to <ch> if
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necessary.
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.. admonition:: Example
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``mpv --af=sub=100:4 --audio-channels=5 media.avi``
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Would add a subwoofer channel with a cutoff frequency of 100 Hz to
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output channel 4.
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``center``
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Creates a center channel from the front channels. May currently be low
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quality as it does not implement a high-pass filter for proper extraction
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yet, but averages and halves the channels instead.
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``<ch>``
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Determines the channel number in which to insert the center channel.
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Channel number can be between 0 and 7 (default: 5). Observe that the
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number of channels will automatically be increased to ``<ch>`` if
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necessary.
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``surround[=delay]``
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Decoder for matrix encoded surround sound like Dolby Surround. Some files
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with 2-channel audio actually contain matrix encoded surround sound.
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``<delay>``
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delay time in ms for the rear speakers (0 to 1000) (default: 20) This
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delay should be set as follows: If d1 is the distance from the
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listening position to the front speakers and d2 is the distance from
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the listening position to the rear speakers, then the delay should be
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set to 15ms if d1 <= d2 and to 15 + 5*(d1-d2) if d1 > d2.
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.. admonition:: Example
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``mpv --af=surround=15 --audio-channels=4 media.avi``
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Would add surround sound decoding with 15 ms delay for the sound to
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the rear speakers.
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``delay[=[ch1,ch2,...]]``
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Delays the sound to the loudspeakers such that the sound from the
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different channels arrives at the listening position simultaneously. It is
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only useful if you have more than 2 loudspeakers.
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``[ch1,ch2,...]``
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The delay in ms that should be imposed on each channel (floating point
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number between 0 and 1000).
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To calculate the required delay for the different channels, do as follows:
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1. Measure the distance to the loudspeakers in meters in relation to your
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listening position, giving you the distances s1 to s5 (for a 5.1
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system). There is no point in compensating for the subwoofer (you will
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not hear the difference anyway).
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2. Subtract the distances s1 to s5 from the maximum distance, i.e.
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``s[i] = max(s) - s[i]; i = 1...5``.
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3. Calculate the required delays in ms as ``d[i] = 1000*s[i]/342; i =
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1...5``.
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.. admonition:: Example
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``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
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and the subwoofer by 0 ms and the center channel by 7 ms.
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``export=mmapped_file:nsamples]``
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Exports the incoming signal to other processes using memory mapping
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(``mmap()``). Memory mapped areas contain a header::
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int nch /* number of channels */
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int size /* buffer size */
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unsigned long long counter /* Used to keep sync, updated every time
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new data is exported. */
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The rest is payload (non-interleaved) 16-bit data.
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``<mmapped_file>``
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File to map data to (required)
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``<nsamples>``
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number of samples per channel (default: 512).
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.. admonition:: Example
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``mpv --af=export=/tmp/mpv-af_export:1024 media.avi``
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Would export 1024 samples per channel to ``/tmp/mpv-af_export``.
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``extrastereo[=mul]``
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(Linearly) increases the difference between left and right channels which
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adds some sort of "live" effect to playback.
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``<mul>``
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Sets the difference coefficient (default: 2.5). 0.0 means mono sound
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(average of both channels), with 1.0 sound will be unchanged, with
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-1.0 left and right channels will be swapped.
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``drc[=method:target]``
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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>``
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Sets the used method.
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1
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Use a single sample to smooth the variations via the standard
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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 ``[ speed_mult 0.9438743126816935`` and ``] speed_mult
|
|
1.059463094352953`` to your ``input.conf`` to step by musical
|
|
semi-tones.
|
|
|
|
.. 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.
|
|
|
|
``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.
|
|
|