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458 lines
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ReStructuredText
458 lines
19 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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Also, keep in mind that most actual filters are available via the ``lavfi``
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wrapper, which gives you access to most of libavfilter's filters. This
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includes all filters that have been ported from MPlayer to libavfilter.
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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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``normalize=<yes|no|auto>``
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Whether to normalize when remixing channel layouts (default: auto).
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``auto`` uses the value set by ``--audio-normalize-downmix``.
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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[:minch]]]``
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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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``minch=<n>``
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If the input channel number is less than ``<minch>``, the filter will
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detach itself (default: 3).
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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:[0-1,1-0,2-2,3-3] 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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Does not do any format conversion itself. Rather, it may cause the
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filter system to insert necessary conversion filters before or after this
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filter if needed. It is primarily useful for controlling the audio format
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going into other filters. To specify the format for audio output, see
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``--audio-format``, ``--audio-samplerate``, and ``--audio-channels``. This
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filter is able to force a particular format, whereas ``--audio-*``
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may be overridden by the ao based on output compatibility.
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All parameters are optional. The first 3 parameters restrict what the filter
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accepts as input. They will therefore cause conversion filters to be
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inserted before this one. The ``out-`` parameters tell the filters or audio
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outputs following this filter how to interpret the data without actually
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doing a conversion. Setting these will probably just break things unless you
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really know you want this for some reason, such as testing or dealing with
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broken media.
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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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*NOTE*: this filter used to be named ``force``. The old ``format`` filter
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used to do conversion itself, unlike this one which lets the filter system
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handle the conversion.
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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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``replaygain-fallback``
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Gain in dB to apply if the file has no replay gain tags. This option
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is always applied if the replaygain logic is somehow inactive. If this
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is applied, no other replaygain options are applied.
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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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``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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``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).
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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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.. note::
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This filter can cause distortion with audio signals that have a very
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large dynamic range.
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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 a value will
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cause noticeable 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 this to your ``input.conf`` to step by musical semi-tones::
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[ multiply speed 0.9438743126816935
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] multiply speed 1.059463094352953
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.. warning::
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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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.. admonition:: Examples
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``mpv --af=scaletempo --speed=1.2 media.ogg``
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Would play media at 1.2x normal speed, with audio at normal
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pitch. Changing playback speed would change audio tempo to match.
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``mpv --af=scaletempo=scale=1.2:speed=none --speed=1.2 media.ogg``
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Would play media at 1.2x normal speed, with audio at normal
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pitch, but changing playback speed would have no effect on audio
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tempo.
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``mpv --af=scaletempo=stride=30:overlap=.50:search=10 media.ogg``
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Would tweak the quality and performance parameters.
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``mpv --af=format=float,scaletempo media.ogg``
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Would make scaletempo use float code. Maybe faster on some
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platforms.
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``mpv --af=scaletempo=scale=1.2:speed=pitch audio.ogg``
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Would play media at 1.2x normal speed, with audio at normal pitch.
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Changing playback speed would change pitch, leaving audio tempo at
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1.2x.
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``rubberband``
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High quality pitch correction with librubberband. This can be used in place
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of ``scaletempo``, and will be used to adjust audio pitch when playing
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at speed different from normal.
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This filter has a number of sub-options. You can list them with
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``mpv --af=rubberband=help``. This will also show the default values
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for each option. The options are not documented here, because they are
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merely passed to librubberband. Look at the librubberband documentation
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to learn what each option does:
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http://breakfastquay.com/rubberband/code-doc/classRubberBand_1_1RubberBandStretcher.html
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(The mapping of the mpv rubberband filter sub-option names and values to
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those of librubberband follows a simple pattern: ``"Option" + Name + Value``.)
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``lavfi=graph``
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Filter audio using FFmpeg's libavfilter.
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``<graph>``
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Libavfilter graph. See ``lavfi`` video filter for details - the graph
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syntax is the same.
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.. warning::
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Don't forget to quote libavfilter graphs as described in the lavfi
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video filter section.
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``o=<string>``
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AVOptions.
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