mirror of https://github.com/mpv-player/mpv.git
230 lines
9.5 KiB
ReStructuredText
230 lines
9.5 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=...``
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Setup a chain of audio filters. See ``--vf`` (`VIDEO FILTERS`_) for the
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full syntax.
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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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The ``--vf`` description describes how libavfilter can be used and how to
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workaround deprecated mpv filters.
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See ``--vf`` group of options for info on how ``--af-defaults``, ``--af-add``,
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``--af-pre``, ``--af-del``, ``--af-clr``, and possibly others work.
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Available filters are:
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``lavcac3enc[=options]``
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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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``encoder=<name>``
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Select the libavcodec encoder used. Currently, this should be an AC-3
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encoder, and using another codec will fail horribly.
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``format=format:srate:channels: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-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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``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=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. It can also be used to adjust audio pitch
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without changing playback speed.
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``<pitch-scale>``
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Sets the pitch scaling factor. Frequencies are multiplied by this value.
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This filter has a number of additional 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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This filter supports the following ``af-command`` commands:
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``set-pitch``
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Set the ``<pitch-scale>`` argument dynamically. This can be used to
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change the playback pitch at runtime. Note that speed is controlled
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using the standard ``speed`` property, not ``af-command``.
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``multiply-pitch <factor>``
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Multiply the current value of ``<pitch-scale>`` dynamically. For
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example: 0.5 to go down by an octave, 1.5 to go up by a perfect fifth.
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If you want to go up or down by semi-tones, use 1.059463094352953 and
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0.9438743126816935
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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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``fix-pts=<yes|no>``
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Determine PTS based on sample count (default: no). If this is enabled,
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the player won't rely on libavfilter passing through PTS accurately.
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Instead, it pass a sample count as PTS to libavfilter, and compute the
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PTS used by mpv based on that and the input PTS. This helps with filters
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which output a recomputed PTS instead of the original PTS (including
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filters which require the PTS to start at 0). mpv normally expects
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filters to not touch the PTS (or only to the extent of changing frame
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boundaries), so this is not the default, but it will be needed to use
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broken filters. In practice, these broken filters will either cause slow
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A/V desync over time (with some files), or break playback completely if
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you seek or start playback from the middle of a file.
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``drop``
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This filter drops or repeats audio frames to adapt to playback speed. It
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always operates on full audio frames, because it was made to handle SPDIF
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(compressed audio passthrough). This is used automatically if the
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``--video-sync=display-adrop`` option is used. Do not use this filter (or
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the given option); they are extremely low quality.
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