mirror of
https://github.com/mpv-player/mpv
synced 2024-12-25 00:02:13 +00:00
df5548a754
librubberband exports a big load of options. Normally, the default settings (whether they're librubberband defaults or our defaults) should be sufficient, but since I'm not so sure about this, making it configurable allows others to figure it out for me.
632 lines
25 KiB
ReStructuredText
632 lines
25 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``.)
|
|
|
|
``<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 ``[ multiply speed 0.9438743126816935`` and
|
|
``] multiply speed 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.
|
|
|
|
``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.
|
|
|