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This changes the name of this project to mpv. Most user-visible mentions of "MPlayer" and "mplayer" are changed to "mpv". The binary name and the default config file location are changed as well. The new default config file location is: ~/.mpv/ Remove etc/mplayer.desktop. Apparently this was for the MPlayer GUI, which has been removed from mplayer2 ages ago. We don't have a logo, and the MS Windows resource files sort-of require one, so leave etc/mplayer.ico/.xpm as-is. Remove the debian and rpm packaging scripts. These contained outdated dependencies and likely were more harmful than useful. (Patches which add working and well-tested packaging are welcome.)
543 lines
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ReStructuredText
543 lines
22 KiB
ReStructuredText
.. _audio_filters:
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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*: To get a full list of available audio filters, see ``--af=help``.
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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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resample[=srate[:sloppy[:type]]]
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Changes the sample rate of the audio stream. Can be used if you have a
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fixed frequency sound card or if you are stuck with an old sound card that
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is only capable of max 44.1kHz. This filter is automatically enabled if
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necessary. It only supports 16-bit integer and float in native-endian
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format as input.
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<srate>
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output sample frequency in Hz. The valid range for this parameter is
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8000 to 192000. If the input and output sample frequency are the same
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or if this parameter is omitted the filter is automatically unloaded.
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A high sample frequency normally improves the audio quality,
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especially when used in combination with other filters.
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<sloppy>
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Allow (1) or disallow (0) the output frequency to differ slightly from
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the frequency given by <srate> (default: 1). Can be used if the
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startup of the playback is extremely slow.
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<type>
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Select which resampling method to use.
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:0: linear interpolation (fast, poor quality especially when
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upsampling)
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:1: polyphase filterbank and integer processing
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:2: polyphase filterbank and floating point processing
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(slow, best quality)
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*EXAMPLE*:
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``mpv --af=resample=44100:0:0``
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would set the output frequency of the resample filter to 44100Hz using
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exact output frequency scaling and linear interpolation.
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lavcresample[=srate[:length[:linear[:count[:cutoff]]]]]
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Changes the sample rate of the audio stream to an integer <srate> in Hz.
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It only supports the 16-bit native-endian format.
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<srate>
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the output sample rate
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<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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<linear>
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if 1 then filters will be linearly interpolated between polyphase
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entries
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<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>
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cutoff frequency (0.0-1.0), default set depending upon filter length
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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. The output sample rate of this filter is same with
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the input sample rate. When input sample rate is 48kHz, 44.1kHz, or 32kHz,
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this filter directly use it. Otherwise a resampling filter is
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auto-inserted before this filter to make the input and output sample rate
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be 48kHz. You need to specify ``--channels=N`` to make the decoder decode
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audio into N-channel, then the filter can encode the N-channel input to
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AC-3.
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<tospdif>
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Output raw AC-3 stream if zero or not set, output to S/PDIF for
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passthrough when <tospdif> is set non-zero.
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<bitrate>
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The bitrate to encode the AC-3 stream. Set it to either 384 or 384000
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to get 384kbits.
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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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Default bitrate is based on the 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>
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If the input channel number is less than <minchn>, the filter will
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detach itself (default: 5).
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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 50/60Hz
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noise on low quality audio equipment. It probably 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
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the headphone listening experience by making the sound similar to that
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from 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 will
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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 the resample 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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*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 while
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canceling it almost completely around 1kHz.
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channels=nch[:nr:from1:to1:from2:to2:from3:to3:...]
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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: If
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the number of output channels is bigger than the number of input channels
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empty channels are inserted (except mixing from mono to stereo, then the
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mono channel is repeated in both of the output channels). If the number of
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output channels is smaller 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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<nr>
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number of routes (1-8)
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<from1:to1:from2:to2:from3:to3:...>
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Pairs of numbers between 0 and 7 that define where to route each
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channel.
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*EXAMPLE*:
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``mpv --af=channels=4:4:0:1:1:0:2:2:3:3 media.avi``
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Would change the number of channels to 4 and set up 4 routes that swap
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channel 0 and channel 1 and leave channel 2 and 3 intact. Observe that
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if media containing two channels was played back, channels 2 and 3
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would contain silence but 0 and 1 would still be swapped.
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``mpv --af=channels=6:4:0:0:0:1:0:2:0:3 media.avi``
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Would change the number of channels to 6 and set up 4 routes that copy
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channel 0 to channels 0 to 3. Channel 4 and 5 will contain silence.
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format[=format]
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Convert between different sample formats. Automatically enabled when
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needed by the sound card or another filter. See also ``--format``.
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<format>
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Sets the desired format. The general form is 'sbe', where 's' denotes
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the sign (either 's' for signed or 'u' for unsigned), 'b' denotes the
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number of bits per sample (16, 24 or 32) and 'e' denotes the
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endianness ('le' means little-endian, 'be' big-endian and 'ne' the
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endianness of the computer mpv is running on). Valid values
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(amongst others) are: 's16le', 'u32be' and 'u24ne'. Exceptions to this
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rule that are also valid format specifiers: u8, s8, floatle, floatbe,
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floatne, mulaw, alaw, mpeg2, ac3 and imaadpcm.
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volume[=v[:sc]]
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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 set the level for the PCM sound to max, leave this filter out and
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control the output level to your speakers with the master volume control
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of the mixer. In case your sound card has a digital PCM mixer instead of
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an analog one, and you hear distortion, use the MASTER mixer instead. If
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there is an external amplifier connected to the computer (this is almost
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always the case), the noise level can be minimized by adjusting the master
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level and the volume knob on the amplifier until the hissing noise in the
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background is gone.
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This filter has a second feature: It measures the overall maximum sound
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level and prints out that level when mpv exits. This feature currently
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only works with floating-point data, use e.g. ``--af-adv=force=5``, or use
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``--af=stats``.
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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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<v>
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Sets the desired gain in dB for all channels in the stream from -200dB
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to +60dB, where -200dB mutes the sound completely and +60dB equals a
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gain of 1000 (default: 0).
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<sc>
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Turns soft clipping on (1) or off (0). 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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*EXAMPLE*:
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``mpv --af=volume=10.1:0 media.avi``
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Would amplify the sound by 10.1dB and hard-clip if the sound level is
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too high.
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pan=n[:L00:L01:L02:...L10:L11:L12:...Ln0:Ln1:Ln2:...]
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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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<Lij>
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How much of input channel i is mixed into output channel j (0-1). So
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in principle you first have n numbers saying what to do with the first
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input channel, then n numbers that act on the second input channel
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etc. If you do not specify any numbers for some input channels, 0 is
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assumed.
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*EXAMPLE*:
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``mpv --af=pan=1:0.5:0.5 media.avi``
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Would down-mix 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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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*: Disable this filter when you are playing DVDs with Dolby
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Digital 5.1 sound, otherwise this filter will disrupt the sound to the
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subwoofer.
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<fc>
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cutoff frequency in Hz for the low-pass filter (20Hz to 300Hz)
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(default: 60Hz) 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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*EXAMPLE*:
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``mpv --af=sub=100:4 --channels=5 media.avi``
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Would add a sub-woofer channel with a cutoff frequency of 100Hz 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. Many files
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with 2 channel audio actually contain matrixed surround sound. Requires a
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sound card supporting at least 4 channels.
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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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*EXAMPLE*:
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``mpv --af=surround=15 --channels=4 media.avi``
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Would add surround sound decoding with 15ms delay for the sound to the
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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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*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 10.5ms, the two rear channels and
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the sub by 0ms and the center channel by 7ms.
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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 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 (default: ``~/.mpv/mpv-af_export``)
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<nsamples>
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number of samples per channel (default: 512)
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*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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volnorm[=method:target]
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Maximizes the volume without distorting the sound.
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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).
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2
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Use several samples to smooth the variations via the standard
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weighted mean over past samples.
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<target>
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Sets the target amplitude as a fraction of the maximum for the sample
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type (default: 0.25).
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ladspa=file:label[:controls...]
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Load a LADSPA (Linux Audio Developer's Simple Plugin API) plugin. This
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filter is reentrant, so multiple LADSPA plugins can be used at once.
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<file>
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Specifies the LADSPA plugin library file. If ``LADSPA_PATH`` is set,
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it searches for the specified file. If it is not set, you must supply
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a fully specified pathname.
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<label>
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Specifies the filter within the library. Some libraries contain only
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one filter, but others contain many of them. Entering 'help' here,
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will list all available filters within the specified library, which
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eliminates the use of 'listplugins' from the LADSPA SDK.
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<controls>
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Controls are zero or more floating point values that determine the
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behavior of the loaded plugin (for example delay, threshold or gain).
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In verbose mode (add ``-v`` to the mpv command line), all
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available controls and their valid ranges are printed. This eliminates
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the use of 'analyseplugin' from the LADSPA SDK.
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karaoke
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Simple voice removal filter exploiting the fact that voice is usually
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recorded with mono gear and later 'center' mixed onto the final audio
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stream. Beware that this filter will turn your signal into mono. Works
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well for 2 channel tracks; do not bother trying it on anything but 2
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channel stereo.
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scaletempo[=option1:option2:...]
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Scales audio tempo without altering pitch, optionally synced to playback
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speed (default).
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This works by playing 'stride' ms of audio at normal speed then consuming
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'stride*scale' ms of input audio. It pieces the strides together by
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blending 'overlap'% of stride with audio following the previous stride. It
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optionally performs a short statistical analysis on the next 'search' ms
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of audio to determine the best overlap position.
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scale=<amount>
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Nominal amount to scale tempo. Scales this amount in addition to
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speed. (default: 1.0)
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stride=<amount>
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Length in milliseconds to output each stride. Too high of value will
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cause noticable skips at high scale amounts and an echo at low scale
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amounts. Very low values will alter pitch. Increasing improves
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performance. (default: 60)
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overlap=<percent>
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Percentage of stride to overlap. Decreasing improves performance.
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(default: .20)
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search=<amount>
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Length in milliseconds to search for best overlap position. Decreasing
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improves performance greatly. On slow systems, you will probably want
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to set this very low. (default: 14)
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speed=<tempo|pitch|both|none>
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Set response to speed change.
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tempo
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Scale tempo in sync with speed (default).
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pitch
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Reverses effect of filter. Scales pitch without altering tempo.
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Add ``[ speed_mult 0.9438743126816935`` and ``] speed_mult
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1.059463094352953`` to your ``input.conf`` to step by musical
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semi-tones.
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*WARNING*: Loses sync with video.
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both
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Scale both tempo and pitch.
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none
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Ignore speed changes.
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*EXAMPLE*:
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``mpv --af=scaletempo --speed=1.2 media.ogg``
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Would playback media at 1.2x normal speed, with audio at normal pitch.
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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``
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Would playback media at 1.2x normal speed, with audio at normal pitch,
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but changing playback speed has 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 performace parameters.
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``mpv --af=format=floatne,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``
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Would playback audio file at 1.2x normal speed, with audio at normal
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pitch. Changing playback speed, would change pitch, leaving audio
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tempo at 1.2x.
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stats
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Collects and prints statistics about the audio stream, especially the
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volume. These statistics are especially intended to help adjusting the
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volume while avoiding clipping. The volumes are printed in dB and
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compatible with the volume audio filter.
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