X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=doc%2Ffilters.texi;h=62e34b99ee334ece748bee21c76814f5794863d0;hb=32586a42da8c9dedd30c6e4c0417c9da91c1f283;hp=c13b4efacf770e8e417397adf25d5856be79a457;hpb=9bcf2aa47756c253238903f3e5ef6c96d927cfe6;p=ffmpeg

diff --git a/doc/filters.texi b/doc/filters.texi
index c13b4efacf7..62e34b99ee3 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -523,10 +523,65 @@ The filter accepts the following options:
 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.
@@ -731,6 +786,39 @@ adelay=delays=64S:all=1
 @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.
@@ -843,6 +931,10 @@ select 75µs (FM-KF).
 @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.
@@ -985,11 +1077,19 @@ select double-exponential seat
 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
 @end table
 
+@subsection Commands
+
+This filter supports the all above options as @ref{commands}.
+
 @subsection Examples
 
 @itemize
@@ -1314,6 +1414,25 @@ Force the output to either unsigned 8-bit or signed 16-bit stereo
 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
@@ -1382,6 +1501,10 @@ the reduction.
 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.
@@ -1390,10 +1513,10 @@ It accepts the following parameters:
 
 @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.
@@ -1408,6 +1531,10 @@ Set output gain.
 Set coefficients format.
 
 @table @samp
+@item ll
+lattice-ladder function
+@item sf
+analog transfer function
 @item tf
 digital transfer function
 @item zp
@@ -1421,8 +1548,16 @@ S-plane zeros/poles
 @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.
@@ -1458,7 +1593,7 @@ displayed. This option is used only when @var{response} is enabled.
 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
@@ -1483,6 +1618,12 @@ Same as above but in @code{zp} format:
 @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
@@ -1573,6 +1714,30 @@ Enabling it will normalize magnitude response at DC to 0dB.
 
 @item order, o
 Set the filter order, can be 1 or 2. Default is 2.
+
+@item transform, a
+Set transform type of IIR filter.
+@table @option
+@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
 
 @subsection Commands
@@ -1751,7 +1916,7 @@ Set central frequency for band.
 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.
@@ -1816,7 +1981,7 @@ Syntax for the commands is : "@var{fN}|f=@var{freq}|w=@var{width}|g=@var{gain}"
 @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:
@@ -1868,16 +2033,7 @@ Set smooth factor. Default value is @var{11}. Allowed range is from @var{1} to @
 
 @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.
@@ -2040,6 +2196,25 @@ It accepts the following values:
 @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.
@@ -2159,6 +2334,13 @@ This filter accepts the following options:
 @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
@@ -2261,6 +2443,7 @@ Set type of soft-clipping.
 
 It accepts the following values:
 @table @option
+@item hard
 @item tanh
 @item atan
 @item cubic
@@ -2268,10 +2451,14 @@ It accepts the following values:
 @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
@@ -2471,11 +2658,11 @@ The filter accepts the following options:
 @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.
@@ -2483,10 +2670,10 @@ Default value is 0.7.
 
 @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
@@ -2502,6 +2689,104 @@ Default value is 20.
 
 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 asuperpass
+Apply high order Butterworth band-pass filter.
+
+The filter accepts the following options:
+
+@table @option
+@item centerf
+Set center frequency in Hertz. Allowed range is 2 to 999999.
+Default value is 1000.
+
+@item order
+Set filter order. Available values are from 4 to 20.
+Default value is 4.
+
+@item qfactor
+Set Q-factor. Allowed range is from 0.01 to 100. Default value is 1.
+
+@item level
+Set input gain level. Allowed range is from 0 to 2. Default value is 1.
+@end table
+
+@subsection Commands
+
+This filter supports the all above options as @ref{commands}.
+
+@section asuperstop
+Apply high order Butterworth band-stop filter.
+
+The filter accepts the following options:
+
+@table @option
+@item centerf
+Set center frequency in Hertz. Allowed range is 2 to 999999.
+Default value is 1000.
+
+@item order
+Set filter order. Available values are from 4 to 20.
+Default value is 4.
+
+@item qfactor
+Set Q-factor. Allowed range is from 0.01 to 100. Default value is 1.
+
+@item level
+Set input gain level. Allowed range is from 0 to 2. Default value is 1.
+@end table
+
+@subsection Commands
+
+This filter supports the all above options as @ref{commands}.
+
 @section atempo
 
 Adjust audio tempo.
@@ -2693,6 +2978,30 @@ Specify which channels to filter, by default all available are filtered.
 @item normalize, n
 Normalize biquad coefficients, by default is disabled.
 Enabling it will normalize magnitude response at DC to 0dB.
+
+@item transform, a
+Set transform type of IIR filter.
+@table @option
+@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
 
 @subsection Commands
@@ -2756,6 +3065,30 @@ Specify which channels to filter, by default all available are filtered.
 @item normalize, n
 Normalize biquad coefficients, by default is disabled.
 Enabling it will normalize magnitude response at DC to 0dB.
+
+@item transform, a
+Set transform type of IIR filter.
+@table @option
+@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
 
 @subsection Commands
@@ -2826,6 +3159,30 @@ Specify which channels to filter, by default all available are filtered.
 @item normalize, n
 Normalize biquad coefficients, by default is disabled.
 Enabling it will normalize magnitude response at DC to 0dB.
+
+@item transform, a
+Set transform type of IIR filter.
+@table @option
+@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
 
 @subsection Commands
@@ -2884,6 +3241,30 @@ Specify which channels to filter, by default all available are filtered.
 @item normalize, n
 Normalize biquad coefficients, by default is disabled.
 Enabling it will normalize magnitude response at DC to 0dB.
+
+@item transform, a
+Set transform type of IIR filter.
+@table @option
+@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
 
 @section bs2b
@@ -3607,6 +3988,30 @@ Specify which channels to filter, by default all available are filtered.
 @item normalize, n
 Normalize biquad coefficients, by default is disabled.
 Enabling it will normalize magnitude response at DC to 0dB.
+
+@item transform, a
+Set transform type of IIR filter.
+@table @option
+@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
 
 @subsection Examples
@@ -4084,6 +4489,30 @@ Specify which channels to filter, by default all available are filtered.
 @item normalize, n
 Normalize biquad coefficients, by default is disabled.
 Enabling it will normalize magnitude response at DC to 0dB.
+
+@item transform, a
+Set transform type of IIR filter.
+@table @option
+@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
 
 @subsection Commands
@@ -4410,6 +4839,30 @@ Specify which channels to filter, by default all available are filtered.
 @item normalize, n
 Normalize biquad coefficients, by default is disabled.
 Enabling it will normalize magnitude response at DC to 0dB.
+
+@item transform, a
+Set transform type of IIR filter.
+@table @option
+@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
 
 @subsection Examples
@@ -4865,6 +5318,10 @@ Default is average. Can be average or maximum.
 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.
@@ -5126,6 +5583,69 @@ and also with custom gain:
 @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
@@ -5197,6 +5717,12 @@ Mid/Side to Left/Left.
 
 @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
@@ -5246,6 +5772,10 @@ Equal power distribution, from -6dB to +6dB range.
 @end table
 @end table
 
+@subsection Commands
+
+This filter supports the all above options as @ref{commands}.
+
 @subsection Examples
 
 @itemize
@@ -5530,6 +6060,30 @@ Specify which channels to filter, by default all available are filtered.
 @item normalize, n
 Normalize biquad coefficients, by default is disabled.
 Enabling it will normalize magnitude response at DC to 0dB.
+
+@item transform, a
+Set transform type of IIR filter.
+@table @option
+@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
 
 @subsection Commands
@@ -5798,7 +6352,7 @@ Below is a description of the currently available audio sources.
 Buffer audio frames, and make them available to the filter chain.
 
 This source is mainly intended for a programmatic use, in particular
-through the interface defined in @file{libavfilter/asrc_abuffer.h}.
+through the interface defined in @file{libavfilter/buffersrc.h}.
 
 It accepts the following parameters:
 @table @option
@@ -6004,6 +6558,13 @@ Specifies the sample rate, and defaults to 44100.
 @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
@@ -6398,12 +6959,6 @@ and a separate video created with @var{alphaextract}, you might use:
 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
@@ -6525,7 +7080,7 @@ Alternatively can be set to @code{s} serial.
 
 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
@@ -6581,7 +7136,7 @@ The filter accepts the following options:
 @table @option
 @item sigmaS
 Set sigma of gaussian function to calculate spatial weight.
-Allowed range is 0 to 10. Default is 0.1.
+Allowed range is 0 to 512. Default is 0.1.
 
 @item sigmaR
 Set sigma of gaussian function to calculate range weight.
@@ -7183,6 +7738,60 @@ ffmpeg -f lavfi -i color=c=black:s=1280x720 -i video.mp4 -shortest -filter_compl
 @end example
 @end itemize
 
+@section chromanr
+Reduce chrominance noise.
+
+The filter accepts the following options:
+
+@table @option
+@item thres
+Set threshold for averaging chrominance values.
+Sum of absolute difference of Y, U and V pixel components of current
+pixel and neighbour pixels lower than this threshold will be used in
+averaging. Luma component is left unchanged and is copied to output.
+Default value is 30. Allowed range is from 1 to 200.
+
+@item sizew
+Set horizontal radius of rectangle used for averaging.
+Allowed range is from 1 to 100. Default value is 5.
+
+@item sizeh
+Set vertical radius of rectangle used for averaging.
+Allowed range is from 1 to 100. Default value is 5.
+
+@item stepw
+Set horizontal step when averaging. Default value is 1.
+Allowed range is from 1 to 50.
+Mostly useful to speed-up filtering.
+
+@item steph
+Set vertical step when averaging. Default value is 1.
+Allowed range is from 1 to 50.
+Mostly useful to speed-up filtering.
+
+@item threy
+Set Y threshold for averaging chrominance values.
+Set finer control for max allowed difference between Y components
+of current pixel and neigbour pixels.
+Default value is 200. Allowed range is from 1 to 200.
+
+@item threu
+Set U threshold for averaging chrominance values.
+Set finer control for max allowed difference between U components
+of current pixel and neigbour pixels.
+Default value is 200. Allowed range is from 1 to 200.
+
+@item threv
+Set V threshold for averaging chrominance values.
+Set finer control for max allowed difference between V components
+of current pixel and neigbour pixels.
+Default value is 200. Allowed range is from 1 to 200.
+@end table
+
+@subsection Commands
+This filter supports same @ref{commands} as options.
+The command accepts the same syntax of the corresponding option.
+
 @section chromashift
 Shift chroma pixels horizontally and/or vertically.
 
@@ -7939,6 +8548,10 @@ Set matrix mode for each plane. Can be @var{square}, @var{row} or @var{column}.
 Default is @var{square}.
 @end table
 
+@subsection Commands
+
+This filter supports the all above options as @ref{commands}.
+
 @subsection Examples
 
 @itemize
@@ -8362,6 +8975,10 @@ The value which the width/height should be divisible by. It defaults to
 get only even dimensions (needed for 4:2:2 video). 16 is best when
 encoding to most video codecs.
 
+@item skip
+Set the number of initial frames for which evaluation is skipped.
+Default is 2. Range is 0 to INT_MAX.
+
 @item reset_count, reset
 Set the counter that determines after how many frames cropdetect will
 reset the previously detected largest video area and start over to
@@ -9156,6 +9773,10 @@ Controls brightness of spill area, preserving colors.
 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
@@ -10125,6 +10746,12 @@ Use fontconfig to set the font. Note that the colons need to be escaped.
 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
@@ -11520,6 +12147,10 @@ frei0r=perspective:0.2/0.2|0.8/0.2
 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}.
@@ -11828,6 +12459,9 @@ Display video size or number of audio channels in case of audio used by filter l
 
 @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
@@ -12820,7 +13454,7 @@ The obtained VMAF score is printed through the logging system.
 
 It requires Netflix's vmaf library (libvmaf) as a pre-requisite.
 After installing the library it can be enabled using:
-@code{./configure --enable-libvmaf --enable-version3}.
+@code{./configure --enable-libvmaf}.
 If no model path is specified it uses the default model: @code{vmaf_v0.6.1.pkl}.
 
 The filter has following options:
@@ -12834,7 +13468,7 @@ Default value: @code{"/usr/local/share/model/vmaf_v0.6.1.pkl"}
 Set the file path to be used to store logs.
 
 @item log_fmt
-Set the format of the log file (xml or json).
+Set the format of the log file (csv, json or xml).
 
 @item enable_transform
 This option can enable/disable the @code{score_transform} applied to the final predicted VMAF score,
@@ -12917,6 +13551,10 @@ Upper bound. Defaults to the highest allowed value for the input.
 Specify which planes will be processed. Defaults to all available.
 @end table
 
+@subsection Commands
+
+This filter supports the all above options as @ref{commands}.
+
 @section loop
 
 Loop video frames.
@@ -14540,7 +15178,7 @@ testsrc=s=100x100, split=4 [in0][in1][in2][in3];
 
 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"
@@ -15321,6 +15959,10 @@ Set value which will be multiplied with filtered result.
 Set value which will be added to filtered result.
 @end table
 
+@subsection Commands
+
+This filter supports the all above options as @ref{commands}.
+
 @section pseudocolor
 
 Alter frame colors in video with pseudocolors.
@@ -15637,6 +16279,10 @@ Enable checking the parity bit. In the event of a parity error, the filter will
 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
@@ -15909,6 +16555,10 @@ Set value which will be multiplied with filtered result.
 Set value which will be added to filtered result.
 @end table
 
+@subsection Commands
+
+This filter supports the all above options as @ref{commands}.
+
 @section rotate
 
 Rotate video by an arbitrary angle expressed in radians.
@@ -17495,6 +18145,10 @@ Set value which will be multiplied with filtered result.
 Set value which will be added to filtered result.
 @end table
 
+@subsection Commands
+
+This filter supports the all above options as @ref{commands}.
+
 @anchor{spp}
 @section spp
 
@@ -17988,7 +18642,7 @@ subtitles=video.mkv:si=1
 To make the subtitles stream from @file{sub.srt} appear in 80% transparent blue
 @code{DejaVu Serif}, use:
 @example
-subtitles=sub.srt:force_style='FontName=DejaVu Serif,PrimaryColour=&HCCFF0000'
+subtitles=sub.srt:force_style='Fontname=DejaVu Serif,PrimaryColour=&HCCFF0000'
 @end example
 
 @section super2xsai
@@ -18152,6 +18806,29 @@ Show envelope. Default is disabled.
 
 @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
@@ -19370,6 +20047,9 @@ Set input horizontal/vertical/diagonal field of view. Values in degrees.
 
 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
@@ -19399,6 +20079,8 @@ Spline16 interpolation.
 @item gauss
 @item gaussian
 Gaussian interpolation.
+@item mitchell
+Mitchell interpolation.
 @end table
 
 Default value is @b{@samp{line}}.
@@ -20411,6 +21093,14 @@ Set one of available transition effects:
 @item hrslice
 @item vuslice
 @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.
 
@@ -22195,7 +22885,7 @@ Below is a description of the currently available video sources.
 Buffer video frames, and make them available to the filter chain.
 
 This source is mainly intended for a programmatic use, in particular
-through the interface defined in @file{libavfilter/vsrc_buffer.h}.
+through the interface defined in @file{libavfilter/buffersrc.h}.
 
 It accepts the following parameters:
 
@@ -22455,6 +23145,17 @@ Set number of colors to use at once. Allowed range is from 2 to 8. Default value
 
 @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
 
 
@@ -22844,6 +23545,10 @@ for the accepted syntax.
 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.
 
@@ -23173,6 +23878,39 @@ Set color which will be used for drawing median phase. If color is
 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
@@ -24797,6 +25535,16 @@ Default is @code{combined}.
 @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
@@ -25447,6 +26195,20 @@ Draw every sample directly.
 @end table
 
 Default value is @code{scale}.
+
+@item filter
+Set the filter mode.
+
+Available values are:
+@table @samp
+@item average
+Use average samples values for each drawn sample.
+
+@item peak
+Use peak samples values for each drawn sample.
+@end table
+
+Default value is @code{average}.
 @end table
 
 @subsection Examples