af.rst

AUDIO FILTERS

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

--af=<filter1[=parameter1:parameter2:...],filter2,...>

Setup a chain of audio filters.

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.

You can also set defaults for each filter. The defaults are applied before the normal filter parameters.

--af-defaults=<filter1[=parameter1:parameter2:...],filter2,...>

Set defaults for each filter.

Audio filters are managed in lists. There are a few commands to manage the filter list:

--af-add=<filter1[,filter2,...]>

Appends the filters given as arguments to the filter list.

--af-pre=<filter1[,filter2,...]>

Prepends the filters given as arguments to the filter list.

--af-del=<index1[,index2,...]>

Deletes the filters at the given indexes. Index numbers start at 0, negative numbers address the end of the list (-1 is the last).

--af-clr

Completely empties the filter list.

Available filters are:

lavrresample[=option1:option2:...]

This filter uses libavresample (or libswresample, depending on the build) to change sample rate, sample format, or channel layout of the audio stream. This filter is automatically enabled if the audio output does not support the audio configuration of the file being played.

It supports only the following sample formats: u8, s16, s32, float.

filter-size=<length>

Length of the filter with respect to the lower sampling rate. (default: 16)

phase-shift=<count>

Log2 of the number of polyphase entries. (..., 10->1024, 11->2048, 12->4096, ...) (default: 10->1024)

cutoff=<cutoff>

Cutoff frequency (0.0-1.0), default set depending upon filter length.

linear

If set then filters will be linearly interpolated between polyphase entries. (default: no)

no-detach

Do not detach if input and output audio format/rate/channels match. (If you just want to set defaults for this filter that will be used even by automatically inserted lavrresample instances, you should prefer setting them with --af-defaults=lavrresample:....)

normalize=<yes|no|auto>

Whether to normalize when remixing channel layouts (default: auto). auto uses the value set by --audio-normalize-downmix.

o=<string>

Set AVOptions on the SwrContext or AVAudioResampleContext. These should be documented by FFmpeg or Libav.

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.

equalizer=g1:g2:g3:...:g10

10 octave band graphic equalizer, implemented using 10 IIR band-pass filters. This means that it works regardless of what type of audio is being played back. The center frequencies for the 10 bands are:

No.	frequency
0	31.25 Hz
1	62.50 Hz
2	125.00 Hz
3	250.00 Hz
4	500.00 Hz
5	1.00 kHz
6	2.00 kHz
7	4.00 kHz
8	8.00 kHz
9	16.00 kHz

If the sample rate of the sound being played is lower than the center frequency for a frequency band, then that band will be disabled. A known bug with this filter is that the characteristics for the uppermost band are not completely symmetric if the sample rate is close to the center frequency of that band. This problem can be worked around by upsampling the sound using a resampling filter before it reaches this filter.

<g1>:<g2>:<g3>:...:<g10>

floating point numbers representing the gain in dB for each frequency band (-12-12)

Example

mpv --af=equalizer=11:11:10:5:0:-12:0:5:12:12 media.avi

Would amplify the sound in the upper and lower frequency region while canceling it almost completely around 1 kHz.

channels=nch[:routes]

Can be used for adding, removing, routing and copying audio channels. If only <nch> is given, the default routing is used. It works as follows: If the number of output channels is greater than the number of input channels, empty channels are inserted (except when mixing from mono to stereo; then the mono channel is duplicated). If the number of output channels is less than the number of input channels, the exceeding channels are truncated.

<nch>

number of output channels (1-8)

<routes>

List of , separated routes, in the form from1-to1,from2-to2,.... Each pair defines where to route each channel. There can be at most 8 routes. Without this argument, the default routing is used. Since , is also used to separate filters, you must quote this argument with [...] or similar.

Examples

mpv --af=channels=4:[0-1,1-0,2-2,3-3] media.avi

Would change the number of channels to 4 and set up 4 routes that swap channel 0 and channel 1 and leave channel 2 and 3 intact. Observe that if media containing two channels were played back, channels 2 and 3 would contain silence but 0 and 1 would still be swapped.

mpv --af=channels=6:[0-0,0-1,0-2,0-3] media.avi

Would change the number of channels to 6 and set up 4 routes that copy channel 0 to channels 0 to 3. Channel 4 and 5 will contain silence.

Note

You should probably not use this filter. If you want to change the output channel layout, try the format filter, which can make mpv automatically up- and downmix standard channel layouts.

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

<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.

volume[=<volumedb>[:...]]

Implements software volume control. Use this filter with caution since it can reduce the signal to noise ratio of the sound. In most cases it is best to use the Master volume control of your sound card or the volume knob on your amplifier.

NOTE: This filter is not reentrant and can therefore only be enabled once for every audio stream.

<volumedb>

Sets the desired gain in dB for all channels in the stream from -200 dB to +60 dB, where -200 dB mutes the sound completely and +60 dB equals a gain of 1000 (default: 0).

replaygain-track

Adjust volume gain according to the track-gain replaygain value stored in the file metadata.

replaygain-album

Like replaygain-track, but using the album-gain value instead.

replaygain-preamp

Pre-amplification gain in dB to apply to the selected replaygain gain (default: 0).

replaygain-clip=yes|no

Prevent clipping caused by replaygain by automatically lowering the gain (default). Use replaygain-clip=no to disable this.

replaygain-fallback

Gain in dB to apply if the file has no replay gain tags. This option is always applied if the replaygain logic is somehow inactive. If this is applied, no other replaygain options are applied.

softclip

Turns soft clipping on. Soft-clipping can make the sound more smooth if very high volume levels are used. Enable this option if the dynamic range of the loudspeakers is very low.

WARNING: This feature creates distortion and should be considered a last resort.

s16

Force S16 sample format if set. Lower quality, but might be faster in some situations.

detach

Remove the filter if the volume is not changed at audio filter config time. Useful with replaygain: if the current file has no replaygain tags, then the filter will be removed if this option is enabled. (If --softvol=yes is used and the player volume controls are used during playback, a different volume filter will be inserted.)

Example

mpv --af=volume=10.1 media.avi

Would amplify the sound by 10.1 dB and hard-clip if the sound level is too high.

pan=n:[<matrix>]

Mixes channels arbitrarily. Basically a combination of the volume and the channels filter that can be used to down-mix many channels to only a few, e.g. stereo to mono, or vary the "width" of the center speaker in a surround sound system. This filter is hard to use, and will require some tinkering before the desired result is obtained. The number of options for this filter depends on the number of output channels. An example how to downmix a six-channel file to two channels with this filter can be found in the examples section near the end.

<n>

Number of output channels (1-8).

<matrix>

A list of values [L00,L01,L02,...,L10,L11,L12,...,Ln0,Ln1,Ln2,...], where each element Lij means how much of input channel i is mixed into output channel j (range 0-1). So in principle you first have n numbers saying what to do with the first input channel, then n numbers that act on the second input channel etc. If you do not specify any numbers for some input channels, 0 is assumed. Note that the values are separated by ,, which is already used by the option parser to separate filters. This is why you must quote the value list with [...] or similar.

Examples

mpv --af=pan=1:[0.5,0.5] media.avi

Would downmix from stereo to mono.

mpv --af=pan=3:[1,0,0.5,0,1,0.5] media.avi

Would give 3 channel output leaving channels 0 and 1 intact, and mix channels 0 and 1 into output channel 2 (which could be sent to a subwoofer for example).

Note

If you just want to force remixing to a certain output channel layout, it is easier to use the format filter. For example, mpv '--af=format=channels=5.1' '--audio-channels=5.1' would always force remixing audio to 5.1 and output it like this.

This filter supports the following af-command commands:

set-matrix

Set the <matrix> argument dynamically. This can be used to change the mixing matrix at runtime, without reinitializing the entire filter chain.

scaletempo[=option1:option2:...]

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

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

scale=<amount>

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

stride=<amount>

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

overlap=<percent>

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

search=<amount>

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

speed=<tempo|pitch|both|none>

Set response to speed change.

tempo

Scale tempo in sync with speed (default).

pitch

Reverses effect of filter. Scales pitch without altering tempo. Add 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.

rubberband

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

<pitch-scale>

Sets the pitch scaling factor. Frequencies are multiplied by this value.

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: http://breakfastquay.com/rubberband/code-doc/classRubberBand_1_1RubberBandStretcher.html (The mapping of the mpv rubberband filter sub-option names and values to those of librubberband follows a simple pattern: "Option" + Name + Value.)

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.

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.