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doc/filters: sort audio filters by name
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Stefano Sabatini
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doc/filters.texi
930
doc/filters.texi
@ -347,282 +347,6 @@ aconvert=u8:auto
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@end example
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@end itemize
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@section allpass
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Apply a two-pole all-pass filter with central frequency (in Hz)
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@var{frequency}, and filter-width @var{width}.
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An all-pass filter changes the audio's frequency to phase relationship
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without changing its frequency to amplitude relationship.
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The filter accepts the following options:
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@table @option
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@item frequency, f
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Set frequency in Hz.
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@item width_type
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Set method to specify band-width of filter.
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@table @option
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@item h
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Hz
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@item q
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Q-Factor
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@item o
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octave
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@item s
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slope
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@end table
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@item width, w
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Specify the band-width of a filter in width_type units.
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@end table
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@section highpass
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Apply a high-pass filter with 3dB point frequency.
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The filter can be either single-pole, or double-pole (the default).
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The filter roll off at 6dB per pole per octave (20dB per pole per decade).
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The filter accepts the following options:
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@table @option
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@item frequency, f
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Set frequency in Hz. Default is 3000.
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@item poles, p
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Set number of poles. Default is 2.
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@item width_type
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Set method to specify band-width of filter.
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@table @option
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@item h
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Hz
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@item q
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Q-Factor
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@item o
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octave
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@item s
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slope
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@end table
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@item width, w
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Specify the band-width of a filter in width_type units.
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Applies only to double-pole filter.
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The default is 0.707q and gives a Butterworth response.
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@end table
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@section lowpass
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Apply a low-pass filter with 3dB point frequency.
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The filter can be either single-pole or double-pole (the default).
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The filter roll off at 6dB per pole per octave (20dB per pole per decade).
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The filter accepts the following options:
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@table @option
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@item frequency, f
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Set frequency in Hz. Default is 500.
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@item poles, p
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Set number of poles. Default is 2.
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@item width_type
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Set method to specify band-width of filter.
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@table @option
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@item h
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Hz
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@item q
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Q-Factor
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@item o
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octave
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@item s
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slope
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@end table
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@item width, w
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Specify the band-width of a filter in width_type units.
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Applies only to double-pole filter.
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The default is 0.707q and gives a Butterworth response.
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@end table
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@section bass
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Boost or cut the bass (lower) frequencies of the audio using a two-pole
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shelving filter with a response similar to that of a standard
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hi-fi's tone-controls. This is also known as shelving equalisation (EQ).
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The filter accepts the following options:
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@table @option
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@item gain, g
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Give the gain at 0 Hz. Its useful range is about -20
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(for a large cut) to +20 (for a large boost).
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Beware of clipping when using a positive gain.
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@item frequency, f
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Set the filter's central frequency and so can be used
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to extend or reduce the frequency range to be boosted or cut.
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The default value is @code{100} Hz.
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@item width_type
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Set method to specify band-width of filter.
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@table @option
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@item h
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Hz
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@item q
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Q-Factor
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@item o
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octave
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@item s
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slope
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@end table
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@item width, w
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Determine how steep is the filter's shelf transition.
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@end table
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@section treble
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Boost or cut treble (upper) frequencies of the audio using a two-pole
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shelving filter with a response similar to that of a standard
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hi-fi's tone-controls. This is also known as shelving equalisation (EQ).
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The filter accepts the following options:
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@table @option
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@item gain, g
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Give the gain at whichever is the lower of ~22 kHz and the
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Nyquist frequency. Its useful range is about -20 (for a large cut)
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to +20 (for a large boost). Beware of clipping when using a positive gain.
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@item frequency, f
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Set the filter's central frequency and so can be used
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to extend or reduce the frequency range to be boosted or cut.
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The default value is @code{3000} Hz.
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@item width_type
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Set method to specify band-width of filter.
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@table @option
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@item h
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Hz
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@item q
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Q-Factor
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@item o
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octave
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@item s
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slope
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@end table
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@item width, w
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Determine how steep is the filter's shelf transition.
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@end table
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@section bandpass
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Apply a two-pole Butterworth band-pass filter with central
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frequency @var{frequency}, and (3dB-point) band-width width.
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The @var{csg} option selects a constant skirt gain (peak gain = Q)
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instead of the default: constant 0dB peak gain.
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The filter roll off at 6dB per octave (20dB per decade).
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The filter accepts the following options:
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@table @option
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@item frequency, f
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Set the filter's central frequency. Default is @code{3000}.
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@item csg
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Constant skirt gain if set to 1. Defaults to 0.
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@item width_type
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Set method to specify band-width of filter.
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@table @option
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@item h
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Hz
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@item q
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Q-Factor
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@item o
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octave
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@item s
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slope
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@end table
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@item width, w
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Specify the band-width of a filter in width_type units.
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@end table
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@section bandreject
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Apply a two-pole Butterworth band-reject filter with central
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frequency @var{frequency}, and (3dB-point) band-width @var{width}.
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The filter roll off at 6dB per octave (20dB per decade).
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The filter accepts the following options:
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@table @option
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@item frequency, f
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Set the filter's central frequency. Default is @code{3000}.
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@item width_type
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Set method to specify band-width of filter.
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@table @option
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@item h
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Hz
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@item q
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Q-Factor
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@item o
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octave
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@item s
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slope
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@end table
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@item width, w
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Specify the band-width of a filter in width_type units.
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@end table
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@section biquad
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Apply a biquad IIR filter with the given coefficients.
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Where @var{b0}, @var{b1}, @var{b2} and @var{a0}, @var{a1}, @var{a2}
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are the numerator and denominator coefficients respectively.
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@section equalizer
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Apply a two-pole peaking equalisation (EQ) filter. With this
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filter, the signal-level at and around a selected frequency can
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be increased or decreased, whilst (unlike bandpass and bandreject
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filters) that at all other frequencies is unchanged.
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In order to produce complex equalisation curves, this filter can
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be given several times, each with a different central frequency.
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The filter accepts the following options:
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@table @option
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@item frequency, f
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Set the filter's central frequency in Hz.
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@item width_type
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Set method to specify band-width of filter.
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@table @option
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@item h
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Hz
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@item q
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Q-Factor
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@item o
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octave
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@item s
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slope
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@end table
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@item width, w
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Specify the band-width of a filter in width_type units.
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@item gain, g
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Set the required gain or attenuation in dB.
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Beware of clipping when using a positive gain.
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@end table
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@section afade
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Apply fade-in/out effect to input audio.
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@ -738,6 +462,36 @@ For example to force the output to either unsigned 8-bit or signed 16-bit stereo
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aformat=sample_fmts=u8|s16:channel_layouts=stereo
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@end example
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@section allpass
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Apply a two-pole all-pass filter with central frequency (in Hz)
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@var{frequency}, and filter-width @var{width}.
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An all-pass filter changes the audio's frequency to phase relationship
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without changing its frequency to amplitude relationship.
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The filter accepts the following options:
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@table @option
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@item frequency, f
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Set frequency in Hz.
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@item width_type
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Set method to specify band-width of filter.
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@table @option
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@item h
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Hz
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@item q
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Q-Factor
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@item o
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octave
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@item s
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slope
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@end table
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@item width, w
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Specify the band-width of a filter in width_type units.
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@end table
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@section amerge
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Merge two or more audio streams into a single multi-channel stream.
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@ -1071,6 +825,39 @@ amovie=file.ogg [a] ; amovie=file.mp3 [b] ;
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[a2] [b2] amerge
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@end example
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@section asyncts
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Synchronize audio data with timestamps by squeezing/stretching it and/or
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dropping samples/adding silence when needed.
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This filter is not built by default, please use @ref{aresample} to do squeezing/stretching.
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The filter accepts the following named parameters:
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@table @option
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@item compensate
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Enable stretching/squeezing the data to make it match the timestamps. Disabled
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by default. When disabled, time gaps are covered with silence.
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@item min_delta
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Minimum difference between timestamps and audio data (in seconds) to trigger
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adding/dropping samples. Default value is 0.1. If you get non-perfect sync with
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this filter, try setting this parameter to 0.
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@item max_comp
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Maximum compensation in samples per second. Relevant only with compensate=1.
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Default value 500.
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@item first_pts
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Assume the first pts should be this value. The time base is 1 / sample rate.
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This allows for padding/trimming at the start of stream. By default, no
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assumption is made about the first frame's expected pts, so no padding or
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trimming is done. For example, this could be set to 0 to pad the beginning with
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silence if an audio stream starts after the video stream or to trim any samples
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with a negative pts due to encoder delay.
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@end table
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@section atempo
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Adjust audio tempo.
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@ -1095,6 +882,236 @@ atempo=1.25
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@end example
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@end itemize
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@section atrim
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Trim the input so that the output contains one continuous subpart of the input.
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This filter accepts the following options:
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@table @option
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@item start
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Timestamp (in seconds) of the start of the kept section. I.e. the audio sample
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with the timestamp @var{start} will be the first sample in the output.
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@item end
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Timestamp (in seconds) of the first audio sample that will be dropped. I.e. the
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audio sample immediately preceding the one with the timestamp @var{end} will be
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the last sample in the output.
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@item start_pts
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Same as @var{start}, except this option sets the start timestamp in samples
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instead of seconds.
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@item end_pts
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Same as @var{end}, except this option sets the end timestamp in samples instead
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of seconds.
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@item duration
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Maximum duration of the output in seconds.
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@item start_sample
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Number of the first sample that should be passed to output.
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@item end_sample
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Number of the first sample that should be dropped.
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@end table
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Note that the first two sets of the start/end options and the @option{duration}
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option look at the frame timestamp, while the _sample options simply count the
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samples that pass through the filter. So start/end_pts and start/end_sample will
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give different results when the timestamps are wrong, inexact or do not start at
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zero. Also note that this filter does not modify the timestamps. If you wish
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that the output timestamps start at zero, insert the asetpts filter after the
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atrim filter.
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If multiple start or end options are set, this filter tries to be greedy and
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keep all samples that match at least one of the specified constraints. To keep
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only the part that matches all the constraints at once, chain multiple atrim
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filters.
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The defaults are such that all the input is kept. So it is possible to set e.g.
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just the end values to keep everything before the specified time.
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Examples:
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@itemize
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@item
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drop everything except the second minute of input
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@example
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ffmpeg -i INPUT -af atrim=60:120
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@end example
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@item
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keep only the first 1000 samples
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@example
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ffmpeg -i INPUT -af atrim=end_sample=1000
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@end example
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@end itemize
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@section bandpass
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Apply a two-pole Butterworth band-pass filter with central
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frequency @var{frequency}, and (3dB-point) band-width width.
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The @var{csg} option selects a constant skirt gain (peak gain = Q)
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instead of the default: constant 0dB peak gain.
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The filter roll off at 6dB per octave (20dB per decade).
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The filter accepts the following options:
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@table @option
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@item frequency, f
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Set the filter's central frequency. Default is @code{3000}.
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@item csg
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Constant skirt gain if set to 1. Defaults to 0.
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@item width_type
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Set method to specify band-width of filter.
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@table @option
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@item h
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Hz
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@item q
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Q-Factor
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@item o
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octave
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@item s
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slope
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@end table
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@item width, w
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Specify the band-width of a filter in width_type units.
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@end table
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@section bandreject
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Apply a two-pole Butterworth band-reject filter with central
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frequency @var{frequency}, and (3dB-point) band-width @var{width}.
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The filter roll off at 6dB per octave (20dB per decade).
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The filter accepts the following options:
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@table @option
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@item frequency, f
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Set the filter's central frequency. Default is @code{3000}.
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@item width_type
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Set method to specify band-width of filter.
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@table @option
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@item h
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Hz
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@item q
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Q-Factor
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@item o
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octave
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@item s
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slope
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@end table
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@item width, w
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Specify the band-width of a filter in width_type units.
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@end table
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@section bass
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Boost or cut the bass (lower) frequencies of the audio using a two-pole
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shelving filter with a response similar to that of a standard
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hi-fi's tone-controls. This is also known as shelving equalisation (EQ).
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The filter accepts the following options:
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@table @option
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@item gain, g
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Give the gain at 0 Hz. Its useful range is about -20
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(for a large cut) to +20 (for a large boost).
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Beware of clipping when using a positive gain.
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@item frequency, f
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Set the filter's central frequency and so can be used
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to extend or reduce the frequency range to be boosted or cut.
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The default value is @code{100} Hz.
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@item width_type
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Set method to specify band-width of filter.
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@table @option
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@item h
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Hz
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@item q
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Q-Factor
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@item o
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octave
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@item s
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slope
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@end table
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@item width, w
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Determine how steep is the filter's shelf transition.
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@end table
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@section biquad
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Apply a biquad IIR filter with the given coefficients.
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Where @var{b0}, @var{b1}, @var{b2} and @var{a0}, @var{a1}, @var{a2}
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are the numerator and denominator coefficients respectively.
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@section channelmap
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Remap input channels to new locations.
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This filter accepts the following named parameters:
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@table @option
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@item channel_layout
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Channel layout of the output stream.
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@item map
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Map channels from input to output. The argument is a '|'-separated list of
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mappings, each in the @code{@var{in_channel}-@var{out_channel}} or
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@var{in_channel} form. @var{in_channel} can be either the name of the input
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channel (e.g. FL for front left) or its index in the input channel layout.
|
||||
@var{out_channel} is the name of the output channel or its index in the output
|
||||
channel layout. If @var{out_channel} is not given then it is implicitly an
|
||||
index, starting with zero and increasing by one for each mapping.
|
||||
@end table
|
||||
|
||||
If no mapping is present, the filter will implicitly map input channels to
|
||||
output channels preserving index.
|
||||
|
||||
For example, assuming a 5.1+downmix input MOV file
|
||||
@example
|
||||
ffmpeg -i in.mov -filter 'channelmap=map=DL-FL|DR-FR' out.wav
|
||||
@end example
|
||||
will create an output WAV file tagged as stereo from the downmix channels of
|
||||
the input.
|
||||
|
||||
To fix a 5.1 WAV improperly encoded in AAC's native channel order
|
||||
@example
|
||||
ffmpeg -i in.wav -filter 'channelmap=1|2|0|5|3|4:channel_layout=5.1' out.wav
|
||||
@end example
|
||||
|
||||
@section channelsplit
|
||||
|
||||
Split each channel in input audio stream into a separate output stream.
|
||||
|
||||
This filter accepts the following named parameters:
|
||||
@table @option
|
||||
@item channel_layout
|
||||
Channel layout of the input stream. Default is "stereo".
|
||||
@end table
|
||||
|
||||
For example, assuming a stereo input MP3 file
|
||||
@example
|
||||
ffmpeg -i in.mp3 -filter_complex channelsplit out.mkv
|
||||
@end example
|
||||
will create an output Matroska file with two audio streams, one containing only
|
||||
the left channel and the other the right channel.
|
||||
|
||||
To split a 5.1 WAV file into per-channel files
|
||||
@example
|
||||
ffmpeg -i in.wav -filter_complex
|
||||
'channelsplit=channel_layout=5.1[FL][FR][FC][LFE][SL][SR]'
|
||||
-map '[FL]' front_left.wav -map '[FR]' front_right.wav -map '[FC]'
|
||||
front_center.wav -map '[LFE]' lfe.wav -map '[SL]' side_left.wav -map '[SR]'
|
||||
side_right.wav
|
||||
@end example
|
||||
|
||||
@section earwax
|
||||
|
||||
Make audio easier to listen to on headphones.
|
||||
@ -1106,6 +1123,149 @@ the listener (standard for speakers).
|
||||
|
||||
Ported from SoX.
|
||||
|
||||
@section equalizer
|
||||
|
||||
Apply a two-pole peaking equalisation (EQ) filter. With this
|
||||
filter, the signal-level at and around a selected frequency can
|
||||
be increased or decreased, whilst (unlike bandpass and bandreject
|
||||
filters) that at all other frequencies is unchanged.
|
||||
|
||||
In order to produce complex equalisation curves, this filter can
|
||||
be given several times, each with a different central frequency.
|
||||
|
||||
The filter accepts the following options:
|
||||
|
||||
@table @option
|
||||
@item frequency, f
|
||||
Set the filter's central frequency in Hz.
|
||||
|
||||
@item width_type
|
||||
Set method to specify band-width of filter.
|
||||
@table @option
|
||||
@item h
|
||||
Hz
|
||||
@item q
|
||||
Q-Factor
|
||||
@item o
|
||||
octave
|
||||
@item s
|
||||
slope
|
||||
@end table
|
||||
|
||||
@item width, w
|
||||
Specify the band-width of a filter in width_type units.
|
||||
|
||||
@item gain, g
|
||||
Set the required gain or attenuation in dB.
|
||||
Beware of clipping when using a positive gain.
|
||||
@end table
|
||||
|
||||
@section highpass
|
||||
|
||||
Apply a high-pass filter with 3dB point frequency.
|
||||
The filter can be either single-pole, or double-pole (the default).
|
||||
The filter roll off at 6dB per pole per octave (20dB per pole per decade).
|
||||
|
||||
The filter accepts the following options:
|
||||
|
||||
@table @option
|
||||
@item frequency, f
|
||||
Set frequency in Hz. Default is 3000.
|
||||
|
||||
@item poles, p
|
||||
Set number of poles. Default is 2.
|
||||
|
||||
@item width_type
|
||||
Set method to specify band-width of filter.
|
||||
@table @option
|
||||
@item h
|
||||
Hz
|
||||
@item q
|
||||
Q-Factor
|
||||
@item o
|
||||
octave
|
||||
@item s
|
||||
slope
|
||||
@end table
|
||||
|
||||
@item width, w
|
||||
Specify the band-width of a filter in width_type units.
|
||||
Applies only to double-pole filter.
|
||||
The default is 0.707q and gives a Butterworth response.
|
||||
@end table
|
||||
|
||||
@section join
|
||||
|
||||
Join multiple input streams into one multi-channel stream.
|
||||
|
||||
The filter accepts the following named parameters:
|
||||
@table @option
|
||||
|
||||
@item inputs
|
||||
Number of input streams. Defaults to 2.
|
||||
|
||||
@item channel_layout
|
||||
Desired output channel layout. Defaults to stereo.
|
||||
|
||||
@item map
|
||||
Map channels from inputs to output. The argument is a '|'-separated list of
|
||||
mappings, each in the @code{@var{input_idx}.@var{in_channel}-@var{out_channel}}
|
||||
form. @var{input_idx} is the 0-based index of the input stream. @var{in_channel}
|
||||
can be either the name of the input channel (e.g. FL for front left) or its
|
||||
index in the specified input stream. @var{out_channel} is the name of the output
|
||||
channel.
|
||||
@end table
|
||||
|
||||
The filter will attempt to guess the mappings when those are not specified
|
||||
explicitly. It does so by first trying to find an unused matching input channel
|
||||
and if that fails it picks the first unused input channel.
|
||||
|
||||
E.g. to join 3 inputs (with properly set channel layouts)
|
||||
@example
|
||||
ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex join=inputs=3 OUTPUT
|
||||
@end example
|
||||
|
||||
To build a 5.1 output from 6 single-channel streams:
|
||||
@example
|
||||
ffmpeg -i fl -i fr -i fc -i sl -i sr -i lfe -filter_complex
|
||||
'join=inputs=6:channel_layout=5.1:map=0.0-FL|1.0-FR|2.0-FC|3.0-SL|4.0-SR|5.0-LFE'
|
||||
out
|
||||
@end example
|
||||
|
||||
@section lowpass
|
||||
|
||||
Apply a low-pass filter with 3dB point frequency.
|
||||
The filter can be either single-pole or double-pole (the default).
|
||||
The filter roll off at 6dB per pole per octave (20dB per pole per decade).
|
||||
|
||||
The filter accepts the following options:
|
||||
|
||||
@table @option
|
||||
@item frequency, f
|
||||
Set frequency in Hz. Default is 500.
|
||||
|
||||
@item poles, p
|
||||
Set number of poles. Default is 2.
|
||||
|
||||
@item width_type
|
||||
Set method to specify band-width of filter.
|
||||
@table @option
|
||||
@item h
|
||||
Hz
|
||||
@item q
|
||||
Q-Factor
|
||||
@item o
|
||||
octave
|
||||
@item s
|
||||
slope
|
||||
@end table
|
||||
|
||||
@item width, w
|
||||
Specify the band-width of a filter in width_type units.
|
||||
Applies only to double-pole filter.
|
||||
The default is 0.707q and gives a Butterworth response.
|
||||
@end table
|
||||
|
||||
@section pan
|
||||
|
||||
Mix channels with specific gain levels. The filter accepts the output
|
||||
@ -1196,6 +1356,11 @@ front left and right:
|
||||
pan="stereo: c0=FR : c1=FR"
|
||||
@end example
|
||||
|
||||
@section resample
|
||||
|
||||
Convert the audio sample format, sample rate and channel layout. This filter is
|
||||
not meant to be used directly.
|
||||
|
||||
@section silencedetect
|
||||
|
||||
Detect silence in an audio stream.
|
||||
@ -1234,201 +1399,42 @@ ffmpeg -i silence.mp3 -af silencedetect=noise=0.0001 -f null -
|
||||
@end example
|
||||
@end itemize
|
||||
|
||||
@section asyncts
|
||||
Synchronize audio data with timestamps by squeezing/stretching it and/or
|
||||
dropping samples/adding silence when needed.
|
||||
@section treble
|
||||
|
||||
This filter is not built by default, please use @ref{aresample} to do squeezing/stretching.
|
||||
Boost or cut treble (upper) frequencies of the audio using a two-pole
|
||||
shelving filter with a response similar to that of a standard
|
||||
hi-fi's tone-controls. This is also known as shelving equalisation (EQ).
|
||||
|
||||
The filter accepts the following options:
|
||||
|
||||
The filter accepts the following named parameters:
|
||||
@table @option
|
||||
@item gain, g
|
||||
Give the gain at whichever is the lower of ~22 kHz and the
|
||||
Nyquist frequency. Its useful range is about -20 (for a large cut)
|
||||
to +20 (for a large boost). Beware of clipping when using a positive gain.
|
||||
|
||||
@item compensate
|
||||
Enable stretching/squeezing the data to make it match the timestamps. Disabled
|
||||
by default. When disabled, time gaps are covered with silence.
|
||||
|
||||
@item min_delta
|
||||
Minimum difference between timestamps and audio data (in seconds) to trigger
|
||||
adding/dropping samples. Default value is 0.1. If you get non-perfect sync with
|
||||
this filter, try setting this parameter to 0.
|
||||
|
||||
@item max_comp
|
||||
Maximum compensation in samples per second. Relevant only with compensate=1.
|
||||
Default value 500.
|
||||
|
||||
@item first_pts
|
||||
Assume the first pts should be this value. The time base is 1 / sample rate.
|
||||
This allows for padding/trimming at the start of stream. By default, no
|
||||
assumption is made about the first frame's expected pts, so no padding or
|
||||
trimming is done. For example, this could be set to 0 to pad the beginning with
|
||||
silence if an audio stream starts after the video stream or to trim any samples
|
||||
with a negative pts due to encoder delay.
|
||||
@item frequency, f
|
||||
Set the filter's central frequency and so can be used
|
||||
to extend or reduce the frequency range to be boosted or cut.
|
||||
The default value is @code{3000} Hz.
|
||||
|
||||
@item width_type
|
||||
Set method to specify band-width of filter.
|
||||
@table @option
|
||||
@item h
|
||||
Hz
|
||||
@item q
|
||||
Q-Factor
|
||||
@item o
|
||||
octave
|
||||
@item s
|
||||
slope
|
||||
@end table
|
||||
|
||||
@section atrim
|
||||
Trim the input so that the output contains one continuous subpart of the input.
|
||||
|
||||
This filter accepts the following options:
|
||||
@table @option
|
||||
@item start
|
||||
Timestamp (in seconds) of the start of the kept section. I.e. the audio sample
|
||||
with the timestamp @var{start} will be the first sample in the output.
|
||||
|
||||
@item end
|
||||
Timestamp (in seconds) of the first audio sample that will be dropped. I.e. the
|
||||
audio sample immediately preceding the one with the timestamp @var{end} will be
|
||||
the last sample in the output.
|
||||
|
||||
@item start_pts
|
||||
Same as @var{start}, except this option sets the start timestamp in samples
|
||||
instead of seconds.
|
||||
|
||||
@item end_pts
|
||||
Same as @var{end}, except this option sets the end timestamp in samples instead
|
||||
of seconds.
|
||||
|
||||
@item duration
|
||||
Maximum duration of the output in seconds.
|
||||
|
||||
@item start_sample
|
||||
Number of the first sample that should be passed to output.
|
||||
|
||||
@item end_sample
|
||||
Number of the first sample that should be dropped.
|
||||
@item width, w
|
||||
Determine how steep is the filter's shelf transition.
|
||||
@end table
|
||||
|
||||
Note that the first two sets of the start/end options and the @option{duration}
|
||||
option look at the frame timestamp, while the _sample options simply count the
|
||||
samples that pass through the filter. So start/end_pts and start/end_sample will
|
||||
give different results when the timestamps are wrong, inexact or do not start at
|
||||
zero. Also note that this filter does not modify the timestamps. If you wish
|
||||
that the output timestamps start at zero, insert the asetpts filter after the
|
||||
atrim filter.
|
||||
|
||||
If multiple start or end options are set, this filter tries to be greedy and
|
||||
keep all samples that match at least one of the specified constraints. To keep
|
||||
only the part that matches all the constraints at once, chain multiple atrim
|
||||
filters.
|
||||
|
||||
The defaults are such that all the input is kept. So it is possible to set e.g.
|
||||
just the end values to keep everything before the specified time.
|
||||
|
||||
Examples:
|
||||
@itemize
|
||||
@item
|
||||
drop everything except the second minute of input
|
||||
@example
|
||||
ffmpeg -i INPUT -af atrim=60:120
|
||||
@end example
|
||||
|
||||
@item
|
||||
keep only the first 1000 samples
|
||||
@example
|
||||
ffmpeg -i INPUT -af atrim=end_sample=1000
|
||||
@end example
|
||||
|
||||
@end itemize
|
||||
|
||||
@section channelsplit
|
||||
Split each channel in input audio stream into a separate output stream.
|
||||
|
||||
This filter accepts the following named parameters:
|
||||
@table @option
|
||||
@item channel_layout
|
||||
Channel layout of the input stream. Default is "stereo".
|
||||
@end table
|
||||
|
||||
For example, assuming a stereo input MP3 file
|
||||
@example
|
||||
ffmpeg -i in.mp3 -filter_complex channelsplit out.mkv
|
||||
@end example
|
||||
will create an output Matroska file with two audio streams, one containing only
|
||||
the left channel and the other the right channel.
|
||||
|
||||
To split a 5.1 WAV file into per-channel files
|
||||
@example
|
||||
ffmpeg -i in.wav -filter_complex
|
||||
'channelsplit=channel_layout=5.1[FL][FR][FC][LFE][SL][SR]'
|
||||
-map '[FL]' front_left.wav -map '[FR]' front_right.wav -map '[FC]'
|
||||
front_center.wav -map '[LFE]' lfe.wav -map '[SL]' side_left.wav -map '[SR]'
|
||||
side_right.wav
|
||||
@end example
|
||||
|
||||
@section channelmap
|
||||
Remap input channels to new locations.
|
||||
|
||||
This filter accepts the following named parameters:
|
||||
@table @option
|
||||
@item channel_layout
|
||||
Channel layout of the output stream.
|
||||
|
||||
@item map
|
||||
Map channels from input to output. The argument is a '|'-separated list of
|
||||
mappings, each in the @code{@var{in_channel}-@var{out_channel}} or
|
||||
@var{in_channel} form. @var{in_channel} can be either the name of the input
|
||||
channel (e.g. FL for front left) or its index in the input channel layout.
|
||||
@var{out_channel} is the name of the output channel or its index in the output
|
||||
channel layout. If @var{out_channel} is not given then it is implicitly an
|
||||
index, starting with zero and increasing by one for each mapping.
|
||||
@end table
|
||||
|
||||
If no mapping is present, the filter will implicitly map input channels to
|
||||
output channels preserving index.
|
||||
|
||||
For example, assuming a 5.1+downmix input MOV file
|
||||
@example
|
||||
ffmpeg -i in.mov -filter 'channelmap=map=DL-FL|DR-FR' out.wav
|
||||
@end example
|
||||
will create an output WAV file tagged as stereo from the downmix channels of
|
||||
the input.
|
||||
|
||||
To fix a 5.1 WAV improperly encoded in AAC's native channel order
|
||||
@example
|
||||
ffmpeg -i in.wav -filter 'channelmap=1|2|0|5|3|4:channel_layout=5.1' out.wav
|
||||
@end example
|
||||
|
||||
@section join
|
||||
Join multiple input streams into one multi-channel stream.
|
||||
|
||||
The filter accepts the following named parameters:
|
||||
@table @option
|
||||
|
||||
@item inputs
|
||||
Number of input streams. Defaults to 2.
|
||||
|
||||
@item channel_layout
|
||||
Desired output channel layout. Defaults to stereo.
|
||||
|
||||
@item map
|
||||
Map channels from inputs to output. The argument is a '|'-separated list of
|
||||
mappings, each in the @code{@var{input_idx}.@var{in_channel}-@var{out_channel}}
|
||||
form. @var{input_idx} is the 0-based index of the input stream. @var{in_channel}
|
||||
can be either the name of the input channel (e.g. FL for front left) or its
|
||||
index in the specified input stream. @var{out_channel} is the name of the output
|
||||
channel.
|
||||
@end table
|
||||
|
||||
The filter will attempt to guess the mappings when those are not specified
|
||||
explicitly. It does so by first trying to find an unused matching input channel
|
||||
and if that fails it picks the first unused input channel.
|
||||
|
||||
E.g. to join 3 inputs (with properly set channel layouts)
|
||||
@example
|
||||
ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex join=inputs=3 OUTPUT
|
||||
@end example
|
||||
|
||||
To build a 5.1 output from 6 single-channel streams:
|
||||
@example
|
||||
ffmpeg -i fl -i fr -i fc -i sl -i sr -i lfe -filter_complex
|
||||
'join=inputs=6:channel_layout=5.1:map=0.0-FL|1.0-FR|2.0-FC|3.0-SL|4.0-SR|5.0-LFE'
|
||||
out
|
||||
@end example
|
||||
|
||||
@section resample
|
||||
Convert the audio sample format, sample rate and channel layout. This filter is
|
||||
not meant to be used directly.
|
||||
|
||||
@section volume
|
||||
|
||||
Adjust the input audio volume.
|
||||
|
||||
Loading…
Reference in New Issue
Block a user