mirror of
https://github.com/mpv-player/mpv
synced 2024-12-24 07:42:17 +00:00
60a7f3b8bc
Mostly copied from vf_lavfi. The parts that could be shared are minor, because most code is about setting up audio and video, which are too different. This won't work with Libav. I used ffplay.c as guide, and noticed too late that their setup methods are incompatible with Libav's. Trying to make it work with both would be too much effort. The configure test for av_opt_set_int_list() should disable af_lavfi gracefully when compiling with Libav. Due to option parser chaos, you currently can't have a "," as part of the filter graph string - not even with quoting or escaping. This will probably be fixed later. The audio filter chain is not PTS aware. So we have to do some hacks to make up a fake PTS, and we have to map the output PTS back to the filter chain's method of tracking PTS changes and buffering, by adjusting af->delay.
563 lines
22 KiB
ReStructuredText
563 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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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 doesn't 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, s16ne, s32ne, floatne.
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srate=<srate>
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the output sample rate
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length=<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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don't detach if input and output audio format/rate/channels are the
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same. You should add this option if you specify additional parameters,
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as automatically inserted lavrresample instances will use the
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default settings.
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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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force=in-format:in-srate:in-channels:out-format:out-srate:out-channels
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Force a specific audio format/configuration without actually changing the
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audio data. Keep in mind that the filter system might auto-insert actual
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conversion filters before or after this filter if needed.
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All parameters are optional. The ``in-`` variants restrict what the filter
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accepts as input. The ``out-`` variants change the audio format, without
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actually doing a conversion. The data will be 'reinterpreted' by the
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filters or audio outputs following this filter.
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<in-format>
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Force conversion to this format. See ``format`` filter for valid audio
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format values.
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<in-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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<in-channels>
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Force mixing to a specific channel layout. See ``--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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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, mpeg2, and ac3.
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volume[=v[:sc[:fast]]]
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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.
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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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<fast>
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Force S16 sample format if set to 1. Lower quality, but might be faster
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in some situations.
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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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*NOTE*: if you just want to force remixing to a certain output channel
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layout, it's easier to use the ``force`` filter. For example,
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``mpv '--af=force=channels=5.1' '--channels=5.1'`` would always force
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remixing audio to 5.1 and output it like this.
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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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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.
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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*: This filter can cause distortion with audio signals that have a
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very large dynamic range.
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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
|
|
a fully specified pathname.
|
|
<label>
|
|
Specifies the filter within the library. Some libraries contain only
|
|
one filter, but others contain many of them. Entering 'help' here,
|
|
will list all available filters within the specified library, which
|
|
eliminates the use of 'listplugins' from the LADSPA SDK.
|
|
<controls>
|
|
Controls are zero or more floating point values that determine the
|
|
behavior of the loaded plugin (for example delay, threshold or gain).
|
|
In verbose mode (add ``-v`` to the mpv command line), all
|
|
available controls and their valid ranges are printed. This eliminates
|
|
the use of 'analyseplugin' from the LADSPA SDK.
|
|
|
|
karaoke
|
|
Simple voice removal filter exploiting the fact that voice is usually
|
|
recorded with mono gear and later 'center' mixed onto the final audio
|
|
stream. Beware that this filter will turn your signal into mono. Works
|
|
well for 2 channel tracks; do not bother trying it on anything but 2
|
|
channel stereo.
|
|
|
|
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 value will
|
|
cause noticable 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 ``[ speed_mult 0.9438743126816935`` and ``] speed_mult
|
|
1.059463094352953`` to your ``input.conf`` to step by musical
|
|
semi-tones.
|
|
|
|
*WARNING*: Loses sync with video.
|
|
both
|
|
Scale both tempo and pitch.
|
|
none
|
|
Ignore speed changes.
|
|
|
|
*EXAMPLE*:
|
|
|
|
``mpv --af=scaletempo --speed=1.2 media.ogg``
|
|
Would playback 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 playback media at 1.2x normal speed, with audio at normal pitch,
|
|
but changing playback speed has no effect on audio tempo.
|
|
|
|
``mpv --af=scaletempo=stride=30:overlap=.50:search=10 media.ogg``
|
|
Would tweak the quality and performace parameters.
|
|
|
|
``mpv --af=format=floatne,scaletempo media.ogg``
|
|
Would make scaletempo use float code. Maybe faster on some platforms.
|
|
|
|
``mpv --af=scaletempo=scale=1.2:speed=pitch audio.ogg``
|
|
Would playback audio file at 1.2x normal speed, with audio at normal
|
|
pitch. Changing playback speed, would change pitch, leaving audio
|
|
tempo at 1.2x.
|
|
|
|
lavfi=graph
|
|
Filter audio using ffmpeg's libavfilter.
|
|
|
|
<graph>
|
|
Libavfilter graph. See ``lavfi`` video filter for details - the graph
|
|
syntax is the same.
|
|
|
|
Warning: due to shortcomings in the current ``-af`` option parser code,
|
|
the filter graph must not contain any ``,``.
|