Set split frequencies. Those must be positive and increasing.
@item order
-Set filter order, can be @var{2nd}, @var{4th} or @var{8th}.
+Set filter order for each band split. This controls filter roll-off or steepness
+of filter transfer function.
+Available values are:
+
+@table @samp
+@item 2nd
+12 dB per octave.
+@item 4th
+24 dB per octave.
+@item 6th
+36 dB per octave.
+@item 8th
+48 dB per octave.
+@item 10th
+60 dB per octave.
+@item 12th
+72 dB per octave.
+@item 14th
+84 dB per octave.
+@item 16th
+96 dB per octave.
+@item 18th
+108 dB per octave.
+@item 20th
+120 dB per octave.
+@end table
+
Default is @var{4th}.
+
+@item level
+Set input gain level. Allowed range is from 0 to 1. Default value is 1.
+
+@item gains
+Set output gain for each band. Default value is 1 for all bands.
@end table
+@subsection Examples
+
+@itemize
+@item
+Split input audio stream into two bands (low and high) with split frequency of 1500 Hz,
+each band will be in separate stream:
+@example
+ffmpeg -i in.flac -filter_complex 'acrossover=split=1500[LOW][HIGH]' -map '[LOW]' low.wav -map '[HIGH]' high.wav
+@end example
+
+@item
+Same as above, but with higher filter order:
+@example
+ffmpeg -i in.flac -filter_complex 'acrossover=split=1500:order=8th[LOW][HIGH]' -map '[LOW]' low.wav -map '[HIGH]' high.wav
+@end example
+
+@item
+Same as above, but also with additional middle band (frequencies between 1500 and 8000):
+@example
+ffmpeg -i in.flac -filter_complex 'acrossover=split=1500 8000:order=8th[LOW][MID][HIGH]' -map '[LOW]' low.wav -map '[MID]' mid.wav -map '[HIGH]' high.wav
+@end example
+@end itemize
+
@section acrusher
Reduce audio bit resolution.
@end example
@end itemize
+@section adenorm
+Remedy denormals in audio by adding extremely low-level noise.
+
+This filter shall be placed before any filter that can produce denormals.
+
+A description of the accepted parameters follows.
+
+@table @option
+@item level
+Set level of added noise in dB. Default is @code{-351}.
+Allowed range is from -451 to -90.
+
+@item type
+Set type of added noise.
+
+@table @option
+@item dc
+Add DC signal.
+@item ac
+Add AC signal.
+@item square
+Add square signal.
+@item pulse
+Add pulse signal.
+@end table
+
+Default is @code{dc}.
+@end table
+
+@subsection Commands
+
+This filter supports the all above options as @ref{commands}.
+
@section aderivative, aintegral
Compute derivative/integral of audio stream.
@end table
@end table
+@subsection Commands
+
+This filter supports the all above options as @ref{commands}.
+
@section aeval
Modify an audio signal according to the specified expressions.
select double-exponential sigmoid
@item losi
select logistic sigmoid
+@item sinc
+select sine cardinal function
+@item isinc
+select inverted sine cardinal function
@item nofade
no fade applied
@end table
aformat=sample_fmts=u8|s16:channel_layouts=stereo
@end example
+@section afreqshift
+Apply frequency shift to input audio samples.
+
+The filter accepts the following options:
+
+@table @option
+@item shift
+Specify frequency shift. Allowed range is -INT_MAX to INT_MAX.
+Default value is 0.0.
+
+@item level
+Set output gain applied to final output. Allowed range is from 0.0 to 1.0.
+Default value is 1.0.
+@end table
+
+@subsection Commands
+
+This filter supports the all above options as @ref{commands}.
+
@section agate
A gate is mainly used to reduce lower parts of a signal. This kind of signal
Default is @code{average}. Can be @code{average} or @code{maximum}.
@end table
+@subsection Commands
+
+This filter supports the all above options as @ref{commands}.
+
@section aiir
Apply an arbitrary Infinite Impulse Response filter.
@table @option
@item zeros, z
-Set numerator/zeros coefficients.
+Set B/numerator/zeros/reflection coefficients.
@item poles, p
-Set denominator/poles coefficients.
+Set A/denominator/poles/ladder coefficients.
@item gains, k
Set channels gains.
Set coefficients format.
@table @samp
+@item ll
+lattice-ladder function
+@item sf
+analog transfer function
@item tf
digital transfer function
@item zp
@end table
@item process, r
-Set kind of processing.
-Can be @code{d} - direct or @code{s} - serial cascading. Default is @code{s}.
+Set type of processing.
+
+@table @samp
+@item d
+direct processing
+@item s
+serial processing
+@item p
+parallel processing
+@end table
@item precision, e
Set filtering precision.
Set video stream size. This option is used only when @var{response} is enabled.
@end table
-Coefficients in @code{tf} format are separated by spaces and are in ascending
+Coefficients in @code{tf} and @code{sf} format are separated by spaces and are in ascending
order.
Coefficients in @code{zp} format are separated by spaces and order of coefficients
@example
aiir=k=0.79575848078096756:z=0.80918701+0.58773007i 0.80918701-0.58773007i 0.80884700+0.58784055i 0.80884700-0.58784055i:p=0.63892345+0.59951235i 0.63892345-0.59951235i 0.79582691+0.44198673i 0.79582691-0.44198673i:f=zp:r=s
@end example
+
+@item
+Apply 3-rd order analog normalized Butterworth low-pass filter, using analog transfer function format:
+@example
+aiir=z=1.3057 0 0 0:p=1.3057 2.3892 2.1860 1:f=sf:r=d
+@end example
@end itemize
@section alimiter
@item di
@item dii
@item tdii
+@item latt
+@end table
+
+@item precision, r
+Set precison of filtering.
+@table @option
+@item auto
+Pick automatic sample format depending on surround filters.
+@item s16
+Always use signed 16-bit.
+@item s32
+Always use signed 32-bit.
+@item f32
+Always use float 32-bit.
+@item f64
+Always use float 64-bit.
@end table
@end table
If input doesn't have that frequency the entry is ignored.
@item w
-Set band width in hertz.
+Set band width in Hertz.
@item g
Set band gain in dB.
@var{fN} is existing filter number, starting from 0, if no such filter is available
error is returned.
@var{freq} set new frequency parameter.
-@var{width} set new width parameter in herz.
+@var{width} set new width parameter in Hertz.
@var{gain} set new gain parameter in dB.
Full filter invocation with asendcmd may look like this:
@subsection Commands
-This filter supports the following commands:
-@table @option
-@item s
-Change denoise strength. Argument is single float number.
-Syntax for the command is : "@var{s}"
-
-@item o
-Change output mode.
-Syntax for the command is : "i", "o" or "n" string.
-@end table
+This filter supports the all above options as @ref{commands}.
@section anlms
Apply Normalized Least-Mean-Squares algorithm to the first audio stream using the second audio stream.
@end table
@end table
+@section aphaseshift
+Apply phase shift to input audio samples.
+
+The filter accepts the following options:
+
+@table @option
+@item shift
+Specify phase shift. Allowed range is from -1.0 to 1.0.
+Default value is 0.0.
+
+@item level
+Set output gain applied to final output. Allowed range is from 0.0 to 1.0.
+Default value is 1.0.
+@end table
+
+@subsection Commands
+
+This filter supports the all above options as @ref{commands}.
+
@section apulsator
Audio pulsator is something between an autopanner and a tremolo.
@table @option
@item model, m
Set train model file to load. This option is always required.
+
+@item mix
+Set how much to mix filtered samples into final output.
+Allowed range is from -1 to 1. Default value is 1.
+Negative values are special, they set how much to keep filtered noise
+in the final filter output. Set this option to -1 to hear actual
+noise removed from input signal.
@end table
@section asetnsamples
It accepts the following values:
@table @option
+@item hard
@item tanh
@item atan
@item cubic
@item alg
@item quintic
@item sin
+@item erf
@end table
@item param
Set additional parameter which controls sigmoid function.
+
+@item oversample
+Set oversampling factor.
@end table
@subsection Commands
@table @option
@item dry
Set dry gain, how much of original signal is kept. Allowed range is from 0 to 1.
-Default value is 0.5.
+Default value is 0.7.
@item wet
Set wet gain, how much of filtered signal is kept. Allowed range is from 0 to 1.
-Default value is 0.8.
+Default value is 0.7.
@item decay
Set delay line decay gain value. Allowed range is from 0 to 1.
@item feedback
Set delay line feedback gain value. Allowed range is from 0 to 1.
-Default value is 0.5.
+Default value is 0.9.
@item cutoff
-Set cutoff frequency in herz. Allowed range is 50 to 900.
+Set cutoff frequency in Hertz. Allowed range is 50 to 900.
Default value is 100.
@item slope
This filter supports the all above options as @ref{commands}.
+@section asubcut
+Cut subwoofer frequencies.
+
+This filter allows to set custom, steeper
+roll off than highpass filter, and thus is able to more attenuate
+frequency content in stop-band.
+
+The filter accepts the following options:
+
+@table @option
+@item cutoff
+Set cutoff frequency in Hertz. Allowed range is 2 to 200.
+Default value is 20.
+
+@item order
+Set filter order. Available values are from 3 to 20.
+Default value is 10.
+
+@item level
+Set input gain level. Allowed range is from 0 to 1. Default value is 1.
+@end table
+
+@subsection Commands
+
+This filter supports the all above options as @ref{commands}.
+
+@section asupercut
+Cut super frequencies.
+
+The filter accepts the following options:
+
+@table @option
+@item cutoff
+Set cutoff frequency in Hertz. Allowed range is 20000 to 192000.
+Default value is 20000.
+
+@item order
+Set filter order. Available values are from 3 to 20.
+Default value is 10.
+
+@item level
+Set input gain level. Allowed range is from 0 to 1. Default value is 1.
+@end table
+
+@subsection Commands
+
+This filter supports the all above options as @ref{commands}.
+
@section atempo
Adjust audio tempo.
@item di
@item dii
@item tdii
+@item latt
+@end table
+
+@item precision, r
+Set precison of filtering.
+@table @option
+@item auto
+Pick automatic sample format depending on surround filters.
+@item s16
+Always use signed 16-bit.
+@item s32
+Always use signed 32-bit.
+@item f32
+Always use float 32-bit.
+@item f64
+Always use float 64-bit.
@end table
@end table
@item di
@item dii
@item tdii
+@item latt
+@end table
+
+@item precision, r
+Set precison of filtering.
+@table @option
+@item auto
+Pick automatic sample format depending on surround filters.
+@item s16
+Always use signed 16-bit.
+@item s32
+Always use signed 32-bit.
+@item f32
+Always use float 32-bit.
+@item f64
+Always use float 64-bit.
@end table
@end table
@item di
@item dii
@item tdii
+@item latt
+@end table
+
+@item precision, r
+Set precison of filtering.
+@table @option
+@item auto
+Pick automatic sample format depending on surround filters.
+@item s16
+Always use signed 16-bit.
+@item s32
+Always use signed 32-bit.
+@item f32
+Always use float 32-bit.
+@item f64
+Always use float 64-bit.
@end table
@end table
@item di
@item dii
@item tdii
+@item latt
+@end table
+
+@item precision, r
+Set precison of filtering.
+@table @option
+@item auto
+Pick automatic sample format depending on surround filters.
+@item s16
+Always use signed 16-bit.
+@item s32
+Always use signed 32-bit.
+@item f32
+Always use float 32-bit.
+@item f64
+Always use float 64-bit.
@end table
@end table
@item di
@item dii
@item tdii
+@item latt
+@end table
+
+@item precision, r
+Set precison of filtering.
+@table @option
+@item auto
+Pick automatic sample format depending on surround filters.
+@item s16
+Always use signed 16-bit.
+@item s32
+Always use signed 32-bit.
+@item f32
+Always use float 32-bit.
+@item f64
+Always use float 64-bit.
@end table
@end table
@item di
@item dii
@item tdii
+@item latt
+@end table
+
+@item precision, r
+Set precison of filtering.
+@table @option
+@item auto
+Pick automatic sample format depending on surround filters.
+@item s16
+Always use signed 16-bit.
+@item s32
+Always use signed 32-bit.
+@item f32
+Always use float 32-bit.
+@item f64
+Always use float 64-bit.
@end table
@end table
@item di
@item dii
@item tdii
+@item latt
+@end table
+
+@item precision, r
+Set precison of filtering.
+@table @option
+@item auto
+Pick automatic sample format depending on surround filters.
+@item s16
+Always use signed 16-bit.
+@item s32
+Always use signed 32-bit.
+@item f32
+Always use float 32-bit.
+@item f64
+Always use float 64-bit.
@end table
@end table
Set sidechain gain. Default is 1. Range is from 0.015625 to 64.
@end table
+@subsection Commands
+
+This filter supports the all above options as @ref{commands}.
+
@section silencedetect
Detect silence in an audio stream.
@end example
@end itemize
+@section speechnorm
+Speech Normalizer.
+
+This filter expands or compresses each half-cycle of audio samples
+(local set of samples all above or all below zero and between two nearest zero crossings) depending
+on threshold value, so audio reaches target peak value under conditions controlled by below options.
+
+The filter accepts the following options:
+
+@table @option
+@item peak, p
+Set the expansion target peak value. This specifies the highest allowed absolute amplitude
+level for the normalized audio input. Default value is 0.95. Allowed range is from 0.0 to 1.0.
+
+@item expansion, e
+Set the maximum expansion factor. Allowed range is from 1.0 to 50.0. Default value is 2.0.
+This option controls maximum local half-cycle of samples expansion. The maximum expansion
+would be such that local peak value reaches target peak value but never to surpass it and that
+ratio between new and previous peak value does not surpass this option value.
+
+@item compression, c
+Set the maximum compression factor. Allowed range is from 1.0 to 50.0. Default value is 2.0.
+This option controls maximum local half-cycle of samples compression. This option is used
+only if @option{threshold} option is set to value greater than 0.0, then in such cases
+when local peak is lower or same as value set by @option{threshold} all samples belonging to
+that peak's half-cycle will be compressed by current compression factor.
+
+@item threshold, t
+Set the threshold value. Default value is 0.0. Allowed range is from 0.0 to 1.0.
+This option specifies which half-cycles of samples will be compressed and which will be expanded.
+Any half-cycle samples with their local peak value below or same as this option value will be
+compressed by current compression factor, otherwise, if greater than threshold value they will be
+expanded with expansion factor so that it could reach peak target value but never surpass it.
+
+@item raise, r
+Set the expansion raising amount per each half-cycle of samples. Default value is 0.001.
+Allowed range is from 0.0 to 1.0. This controls how fast expansion factor is raised per
+each new half-cycle until it reaches @option{expansion} value.
+Setting this options too high may lead to distortions.
+
+@item fall, f
+Set the compression raising amount per each half-cycle of samples. Default value is 0.001.
+Allowed range is from 0.0 to 1.0. This controls how fast compression factor is raised per
+each new half-cycle until it reaches @option{compression} value.
+
+@item channels, h
+Specify which channels to filter, by default all available channels are filtered.
+
+@item invert, i
+Enable inverted filtering, by default is disabled. This inverts interpretation of @option{threshold}
+option. When enabled any half-cycle of samples with their local peak value below or same as
+@option{threshold} option will be expanded otherwise it will be compressed.
+
+@item link, l
+Link channels when calculating gain applied to each filtered channel sample, by default is disabled.
+When disabled each filtered channel gain calculation is independent, otherwise when this option
+is enabled the minimum of all possible gains for each filtered channel is used.
+@end table
+
+@subsection Commands
+
+This filter supports the all above options as @ref{commands}.
+
@section stereotools
This filter has some handy utilities to manage stereo signals, for converting
@item ms>rr
Mid/Side to Right/Right.
+
+@item ms>rl
+Mid/Side to Right/Left.
+
+@item lr>l-r
+Left/Right to Left - Right.
@end table
@item slev
@end table
@end table
+@subsection Commands
+
+This filter supports the all above options as @ref{commands}.
+
@subsection Examples
@itemize
@item di
@item dii
@item tdii
+@item latt
+@end table
+
+@item precision, r
+Set precison of filtering.
+@table @option
+@item auto
+Pick automatic sample format depending on surround filters.
+@item s16
+Always use signed 16-bit.
+@item s32
+Always use signed 32-bit.
+@item f32
+Always use float 32-bit.
+@item f64
+Always use float 64-bit.
@end table
@end table
@item nb_samples, n
Set the number of samples per requested frames.
+@item duration, d
+Set the duration of the sourced audio. See
+@ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}
+for the accepted syntax.
+
+If not specified, or the expressed duration is negative, the audio is
+supposed to be generated forever.
@end table
@subsection Examples
movie=in_alpha.mkv [alpha]; [in][alpha] alphamerge [out]
@end example
-Since this filter is designed for reconstruction, it operates on frame
-sequences without considering timestamps, and terminates when either
-input reaches end of stream. This will cause problems if your encoding
-pipeline drops frames. If you're trying to apply an image as an
-overlay to a video stream, consider the @var{overlay} filter instead.
-
@section amplify
Amplify differences between current pixel and pixels of adjacent frames in
Parallel can be faster then serial, while other way around is never true.
Parallel will abort early on first change being greater then thresholds, while serial
-will continue processing other side of frames if they are equal or bellow thresholds.
+will continue processing other side of frames if they are equal or below thresholds.
@end table
@subsection Commands
Modify alpha from generated spillmap.
@end table
+@subsection Commands
+
+This filter supports the all above options as @ref{commands}.
+
@section detelecine
Apply an exact inverse of the telecine operation. It requires a predefined
drawtext='fontfile=Linux Libertine O-40\:style=Semibold:text=FFmpeg'
@end example
+@item
+Draw "Test Text" with font size dependent on height of the video.
+@example
+drawtext="text='Test Text': fontsize=h/30: x=(w-text_w)/2: y=(h-text_h*2)"
+@end example
+
@item
Print the date of a real-time encoding (see strftime(3)):
@example
For more information, see
@url{http://frei0r.dyne.org}
+@subsection Commands
+
+This filter supports the @option{filter_params} option as @ref{commands}.
+
@section fspp
Apply fast and simple postprocessing. It is a faster version of @ref{spp}.
@item rate
Display video frame rate or sample rate in case of audio used by filter link.
+
+@item eof
+Display link output status.
@end table
@item rate, r
Overlay one video on top of another.
-This is the CUDA cariant of the @ref{overlay} filter.
+This is the CUDA variant of the @ref{overlay} filter.
It only accepts CUDA frames. The underlying input pixel formats have to match.
It takes two inputs and has one output. The first input is the "main"
Lowpass lines prior to further processing. Default is enabled.
@end table
+@subsection Commands
+
+This filter supports the all above options as @ref{commands}.
+
@subsection Examples
@itemize
@item ecolor, ec
Set envelope color. Default is @code{gold}.
+
+@item slide
+Set slide mode.
+
+Available values for slide is:
+@table @samp
+@item frame
+Draw new frame when right border is reached.
+
+@item replace
+Replace old columns with new ones.
+
+@item scroll
+Scroll from right to left.
+
+@item rscroll
+Scroll from left to right.
+
+@item picture
+Draw single picture.
+@end table
+
+Default is @code{replace}.
@end table
@section threshold
If diagonal field of view is set it overrides horizontal and vertical field of view.
@end table
+
+@item octahedron
+Octahedron projection.
@end table
@item interp
@item gauss
@item gaussian
Gaussian interpolation.
+@item mitchell
+Mitchell interpolation.
@end table
Default value is @b{@samp{line}}.
@item vdslice
@item hblur
@item fadegrays
+@item wipetl
+@item wipetr
+@item wipebl
+@item wipebr
+@item squeezeh
+@item squeezev
@end table
Default transition effect is fade.
@item seed
Set seed for picking gradient line points.
+
+@item duration, d
+Set the duration of the sourced video. See
+@ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}
+for the accepted syntax.
+
+If not specified, or the expressed duration is negative, the video is
+supposed to be generated forever.
+
+@item speed
+Set speed of gradients rotation.
@end table
If not specified, or the expressed duration is negative, the video is
supposed to be generated forever.
+Since the frame rate is used as time base, all frames including the last one
+will have their full duration. If the specified duration is not a multiple
+of the frame duration, it will be rounded up.
+
@item sar
Set the sample aspect ratio of the sourced video.
Enable video output. Default is enabled.
@end table
+@subsection phasing detection
+
+The filter also detects out of phase and mono sequences in stereo streams.
+It logs the sequence start, end and duration when it lasts longer or as long as the minimum set.
+
+The filter accepts the following options for this detection:
+
+@table @option
+@item phasing
+Enable mono and out of phase detection. Default is disabled.
+
+@item tolerance, t
+Set phase tolerance for mono detection, in amplitude ratio. Default is @code{0}.
+Allowed range is @code{[0, 1]}.
+
+@item angle, a
+Set angle threshold for out of phase detection, in degree. Default is @code{170}.
+Allowed range is @code{[90, 180]}.
+
+@item duration, d
+Set mono or out of phase duration until notification, expressed in seconds. Default is @code{2}.
+@end table
+
+@subsection Examples
+
+@itemize
+@item
+Complete example with @command{ffmpeg} to detect 1 second of mono with 0.001 phase tolerance:
+@example
+ffmpeg -i stereo.wav -af aphasemeter=video=0:phasing=1:duration=1:tolerance=0.001 -f null -
+@end example
+@end itemize
+
@section avectorscope
Convert input audio to a video output, representing the audio vector
@item minamp
Set minimum amplitude used in @code{log} amplitude scaler.
+@item data
+Set data display mode.
+
+It accepts the following values:
+@table @samp
+@item magnitude
+@item phase
+@item delay
+@end table
+Default is @code{magnitude}.
@end table
@section showspatial
projects / ffmpeg / blobdiff