af.rst

AUDIO FILTERS

Audio filters allow you to modify the audio stream and its properties. The syntax is:

--af=...

Setup a chain of audio filters. See --vf (`VIDEO FILTERS`_) for the full syntax.

This is an object settings list option. See `List Options`_ for details.

Note

To get a full list of available audio filters, see --af=help.

Also, keep in mind that most actual filters are available via the lavfi wrapper, which gives you access to most of libavfilter's filters. This includes all filters that have been ported from MPlayer to libavfilter.

The --vf description describes how libavfilter can be used and how to workaround deprecated mpv filters.

See --vf group of options for info on how --af-add, --af-pre, --af-clr, and possibly others work.

Available filters are:

lavcac3enc[=options]

Encode multi-channel audio to AC-3 at runtime using libavcodec. Supports 16-bit native-endian input format, maximum 6 channels. The output is big-endian when outputting a raw AC-3 stream, native-endian when outputting to S/PDIF. If the input sample rate is not 48 kHz, 44.1 kHz or 32 kHz, it will be resampled to 48 kHz.

tospdif=<yes|no>

Output raw AC-3 stream if no, output to S/PDIF for pass-through if yes (default).

bitrate=<rate>

The bitrate use for the AC-3 stream. Set it to 384 to get 384 kbps.

The default is 640. Some receivers might not be able to handle this.

Valid values: 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384, 448, 512, 576, 640.

The special value auto selects a default bitrate based on the input channel number:

1ch:	96
2ch:	192
3ch:	224
4ch:	384
5ch:	448
6ch:	448

minch=<n>

If the input channel number is less than <minch>, the filter will detach itself (default: 3).

encoder=<name>

Select the libavcodec encoder used. Currently, this should be an AC-3 encoder, and using another codec will fail horribly.

format=format:srate:channels:out-srate:out-channels

Does not do any format conversion itself. Rather, it may cause the filter system to insert necessary conversion filters before or after this filter if needed. It is primarily useful for controlling the audio format going into other filters. To specify the format for audio output, see --audio-format, --audio-samplerate, and --audio-channels. This filter is able to force a particular format, whereas --audio-* may be overridden by the ao based on output compatibility.

All parameters are optional. The first 3 parameters restrict what the filter accepts as input. They will therefore cause conversion filters to be inserted before this one. The out- parameters tell the filters or audio outputs following this filter how to interpret the data without actually doing a conversion. Setting these will probably just break things unless you really know you want this for some reason, such as testing or dealing with broken media.

<format>

Force conversion to this format. Use --af=format=format=help to get a list of valid formats.

<srate>

Force conversion to a specific sample rate. The rate is an integer, 48000 for example.

<channels>

Force mixing to a specific channel layout. See --audio-channels option for possible values.

<out-srate>

<out-channels>

NOTE: this filter used to be named force. The old format filter used to do conversion itself, unlike this one which lets the filter system handle the conversion.

scaletempo[=option1:option2:...]

Scales audio tempo without altering pitch, optionally synced to playback speed.

This works by playing 'stride' ms of audio at normal speed then consuming 'stride*scale' ms of input audio. It pieces the strides together by blending 'overlap'% of stride with audio following the previous stride. It optionally performs a short statistical analysis on the next 'search' ms of audio to determine the best overlap position.

scale=<amount>

Nominal amount to scale tempo. Scales this amount in addition to speed. (default: 1.0)

stride=<amount>

Length in milliseconds to output each stride. Too high of a value will cause noticeable skips at high scale amounts and an echo at low scale amounts. Very low values will alter pitch. Increasing improves performance. (default: 60)

overlap=<factor>

Factor of stride to overlap. Decreasing improves performance. (default: .20)

search=<amount>

Length in milliseconds to search for best overlap position. Decreasing improves performance greatly. On slow systems, you will probably want to set this very low. (default: 14)

speed=<tempo|pitch|both|none>

Set response to speed change.

tempo

Scale tempo in sync with speed (default).

pitch

Reverses effect of filter. Scales pitch without altering tempo. Add this to your input.conf to step by musical semi-tones:

[ multiply speed 0.9438743126816935
] multiply speed 1.059463094352953

Warning

Loses sync with video.

both

Scale both tempo and pitch.

none

Ignore speed changes.

Examples

mpv --af=scaletempo --speed=1.2 media.ogg

Would play media at 1.2x normal speed, with audio at normal pitch. Changing playback speed would change audio tempo to match.

mpv --af=scaletempo=scale=1.2:speed=none --speed=1.2 media.ogg

Would play media at 1.2x normal speed, with audio at normal pitch, but changing playback speed would have no effect on audio tempo.

mpv --af=scaletempo=stride=30:overlap=.50:search=10 media.ogg

Would tweak the quality and performance parameters.

mpv --af=scaletempo=scale=1.2:speed=pitch audio.ogg

Would play media at 1.2x normal speed, with audio at normal pitch. Changing playback speed would change pitch, leaving audio tempo at 1.2x.

scaletempo2[=option1:option2:...]

Scales audio tempo without altering pitch. The algorithm is ported from chromium and uses the Waveform Similarity Overlap-and-add (WSOLA) method. It seems to achieves higher audio quality than scaletempo, and rubberband R2 engine, or engine=faster. This filter is inserted automatically if audio-pitch-correction option is used (on by default) when the playback speed is changed.

By default, the search-interval and window-size parameters have the same values as in chromium.

min-speed=<speed>

Mute audio if the playback speed is below <speed>. (default: 0.25)

max-speed=<speed>

Mute audio if the playback speed is above <speed> and <speed> != 0. (default: 8.0)

search-interval=<amount>

Length in milliseconds to search for best overlap position. (default: 40)

window-size=<amount>

Length in milliseconds of the overlap-and-add window. (default: 12)

rubberband

High quality pitch correction with librubberband. This can be used in place of scaletempo and scaletempo2, and will be used to adjust audio pitch when playing at speed different from normal. It can also be used to adjust audio pitch without changing playback speed.

pitch-scale=<amount>

Sets the pitch scaling factor. Frequencies are multiplied by this value. (default: 1.0)

engine=<faster|finer>

Select the core Rubberband engine to be used. There are two available:

Faster:	This is the Rubberband R2 engine. It uses significantly less CPU than the Finer (R3) engine.
Finer:	This is the Rubberband R3 engine. This engine is only available with librubberband version 3 or newer. This produces significantly higher quality output, at the cost of higher CPU usage. (Default if available)

This filter has a number of additional sub-options. You can list them with mpv --af=rubberband=help. This will also show the default values for each option. The options are not documented here, because they are merely passed to librubberband. Look at the librubberband documentation to learn what each option does: https://breakfastquay.com/rubberband/code-doc/classRubberBand_1_1RubberBandStretcher.html Do note that certain options are only applicable to one of R2 (faster) and R3 (finer) engines. (The mapping of the mpv rubberband filter sub-option names and values to those of librubberband follows a simple pattern: "Option" + Name + Value.)

This filter supports the following af-command commands:

set-pitch

Set the <pitch-scale> argument dynamically. This can be used to change the playback pitch at runtime. Note that speed is controlled using the standard speed property, not af-command.

multiply-pitch <factor>

Multiply the current value of <pitch-scale> dynamically.

lavfi=graph

Filter audio using FFmpeg's libavfilter.

<graph>

Libavfilter graph. See lavfi video filter for details - the graph syntax is the same.

Warning

Don't forget to quote libavfilter graphs as described in the lavfi video filter section.

o=<string>

AVOptions.

fix-pts=<yes|no>

Determine PTS based on sample count (default: no). If this is enabled, the player won't rely on libavfilter passing through PTS accurately. Instead, it pass a sample count as PTS to libavfilter, and compute the PTS used by mpv based on that and the input PTS. This helps with filters which output a recomputed PTS instead of the original PTS (including filters which require the PTS to start at 0). mpv normally expects filters to not touch the PTS (or only to the extent of changing frame boundaries), so this is not the default, but it will be needed to use broken filters. In practice, these broken filters will either cause slow A/V desync over time (with some files), or break playback completely if you seek or start playback from the middle of a file.

drop

This filter drops or repeats audio frames to adapt to playback speed. It always operates on full audio frames, because it was made to handle SPDIF (compressed audio passthrough). This is used automatically if the --video-sync=display-adrop option is used. Do not use this filter (or the given option); they are extremely low quality.