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@chapter Filtering Introduction
@c man begin FILTERING INTRODUCTION
Filtering in FFmpeg is enabled through the libavfilter library.
In libavfilter, a filter can have multiple inputs and multiple
outputs.
To illustrate the sorts of things that are possible, we consider the
following filtergraph.
@verbatim
[main]
input --> split ---------------------> overlay --> output
| ^
|[tmp] [flip]|
+-----> crop --> vflip -------+
@end verbatim
This filtergraph splits the input stream in two streams, then sends one
stream through the crop filter and the vflip filter, before merging it
back with the other stream by overlaying it on top. You can use the
following command to achieve this:
@example
ffmpeg -i INPUT -vf "split [main][tmp]; [tmp] crop=iw:ih/2:0:0, vflip [flip]; [main][flip] overlay=0:H/2" OUTPUT
@end example
The result will be that the top half of the video is mirrored
onto the bottom half of the output video.
Filters in the same linear chain are separated by commas, and distinct
linear chains of filters are separated by semicolons. In our example,
@var{crop,vflip} are in one linear chain, @var{split} and
@var{overlay} are separately in another. The points where the linear
chains join are labelled by names enclosed in square brackets. In the
example, the split filter generates two outputs that are associated to
the labels @var{[main]} and @var{[tmp]}.
The stream sent to the second output of @var{split}, labelled as
@var{[tmp]}, is processed through the @var{crop} filter, which crops
away the lower half part of the video, and then vertically flipped. The
@var{overlay} filter takes in input the first unchanged output of the
split filter (which was labelled as @var{[main]}), and overlay on its
lower half the output generated by the @var{crop,vflip} filterchain.
Some filters take in input a list of parameters: they are specified
after the filter name and an equal sign, and are separated from each other
by a colon.
There exist so-called @var{source filters} that do not have an
audio/video input, and @var{sink filters} that will not have audio/video
output.
@c man end FILTERING INTRODUCTION
@chapter graph2dot
@c man begin GRAPH2DOT
The @file{graph2dot} program included in the FFmpeg @file{tools}
directory can be used to parse a filtergraph description and issue a
corresponding textual representation in the dot language.
Invoke the command:
@example
graph2dot -h
@end example
to see how to use @file{graph2dot}.
You can then pass the dot description to the @file{dot} program (from
the graphviz suite of programs) and obtain a graphical representation
of the filtergraph.
For example the sequence of commands:
@example
echo @var{GRAPH_DESCRIPTION} | \
tools/graph2dot -o graph.tmp && \
dot -Tpng graph.tmp -o graph.png && \
display graph.png
@end example
can be used to create and display an image representing the graph
described by the @var{GRAPH_DESCRIPTION} string. Note that this string must be
a complete self-contained graph, with its inputs and outputs explicitly defined.
For example if your command line is of the form:
@example
ffmpeg -i infile -vf scale=640:360 outfile
@end example
your @var{GRAPH_DESCRIPTION} string will need to be of the form:
@example
nullsrc,scale=640:360,nullsink
@end example
you may also need to set the @var{nullsrc} parameters and add a @var{format}
filter in order to simulate a specific input file.
@c man end GRAPH2DOT
@chapter Filtergraph description
@c man begin FILTERGRAPH DESCRIPTION
A filtergraph is a directed graph of connected filters. It can contain
cycles, and there can be multiple links between a pair of
filters. Each link has one input pad on one side connecting it to one
filter from which it takes its input, and one output pad on the other
side connecting it to one filter accepting its output.
Each filter in a filtergraph is an instance of a filter class
registered in the application, which defines the features and the
number of input and output pads of the filter.
A filter with no input pads is called a "source", and a filter with no
output pads is called a "sink".
@anchor{Filtergraph syntax}
@section Filtergraph syntax
A filtergraph has a textual representation, which is recognized by the
@option{-filter}/@option{-vf}/@option{-af} and
@option{-filter_complex} options in @command{ffmpeg} and
@option{-vf}/@option{-af} in @command{ffplay}, and by the
@code{avfilter_graph_parse_ptr()} function defined in
@file{libavfilter/avfilter.h}.
A filterchain consists of a sequence of connected filters, each one
connected to the previous one in the sequence. A filterchain is
represented by a list of ","-separated filter descriptions.
A filtergraph consists of a sequence of filterchains. A sequence of
filterchains is represented by a list of ";"-separated filterchain
descriptions.
A filter is represented by a string of the form:
[@var{in_link_1}]...[@var{in_link_N}]@var{filter_name}=@var{arguments}[@var{out_link_1}]...[@var{out_link_M}]
@var{filter_name} is the name of the filter class of which the
described filter is an instance of, and has to be the name of one of
the filter classes registered in the program.
The name of the filter class is optionally followed by a string
"=@var{arguments}".
@var{arguments} is a string which contains the parameters used to
initialize the filter instance. It may have one of two forms:
@itemize
@item
A ':'-separated list of @var{key=value} pairs.
@item
A ':'-separated list of @var{value}. In this case, the keys are assumed to be
the option names in the order they are declared. E.g. the @code{fade} filter
declares three options in this order -- @option{type}, @option{start_frame} and
@option{nb_frames}. Then the parameter list @var{in:0:30} means that the value
@var{in} is assigned to the option @option{type}, @var{0} to
@option{start_frame} and @var{30} to @option{nb_frames}.
@item
A ':'-separated list of mixed direct @var{value} and long @var{key=value}
pairs. The direct @var{value} must precede the @var{key=value} pairs, and
follow the same constraints order of the previous point. The following
@var{key=value} pairs can be set in any preferred order.
@end itemize
If the option value itself is a list of items (e.g. the @code{format} filter
takes a list of pixel formats), the items in the list are usually separated by
@samp{|}.
The list of arguments can be quoted using the character @samp{'} as initial
and ending mark, and the character @samp{\} for escaping the characters
within the quoted text; otherwise the argument string is considered
terminated when the next special character (belonging to the set
@samp{[]=;,}) is encountered.
The name and arguments of the filter are optionally preceded and
followed by a list of link labels.
A link label allows one to name a link and associate it to a filter output
or input pad. The preceding labels @var{in_link_1}
... @var{in_link_N}, are associated to the filter input pads,
the following labels @var{out_link_1} ... @var{out_link_M}, are
associated to the output pads.
When two link labels with the same name are found in the
filtergraph, a link between the corresponding input and output pad is
created.
If an output pad is not labelled, it is linked by default to the first
unlabelled input pad of the next filter in the filterchain.
For example in the filterchain
@example
nullsrc, split[L1], [L2]overlay, nullsink
@end example
the split filter instance has two output pads, and the overlay filter
instance two input pads. The first output pad of split is labelled
"L1", the first input pad of overlay is labelled "L2", and the second
output pad of split is linked to the second input pad of overlay,
which are both unlabelled.
In a filter description, if the input label of the first filter is not
specified, "in" is assumed; if the output label of the last filter is not
specified, "out" is assumed.
In a complete filterchain all the unlabelled filter input and output
pads must be connected. A filtergraph is considered valid if all the
filter input and output pads of all the filterchains are connected.
Libavfilter will automatically insert @ref{scale} filters where format
conversion is required. It is possible to specify swscale flags
for those automatically inserted scalers by prepending
@code{sws_flags=@var{flags};}
to the filtergraph description.
Here is a BNF description of the filtergraph syntax:
@example
@var{NAME} ::= sequence of alphanumeric characters and '_'
@var{LINKLABEL} ::= "[" @var{NAME} "]"
@var{LINKLABELS} ::= @var{LINKLABEL} [@var{LINKLABELS}]
@var{FILTER_ARGUMENTS} ::= sequence of chars (possibly quoted)
@var{FILTER} ::= [@var{LINKLABELS}] @var{NAME} ["=" @var{FILTER_ARGUMENTS}] [@var{LINKLABELS}]
@var{FILTERCHAIN} ::= @var{FILTER} [,@var{FILTERCHAIN}]
@var{FILTERGRAPH} ::= [sws_flags=@var{flags};] @var{FILTERCHAIN} [;@var{FILTERGRAPH}]
@end example
@section Notes on filtergraph escaping
Filtergraph description composition entails several levels of
escaping. See @ref{quoting_and_escaping,,the "Quoting and escaping"
section in the ffmpeg-utils(1) manual,ffmpeg-utils} for more
information about the employed escaping procedure.
A first level escaping affects the content of each filter option
value, which may contain the special character @code{:} used to
separate values, or one of the escaping characters @code{\'}.
A second level escaping affects the whole filter description, which
may contain the escaping characters @code{\'} or the special
characters @code{[],;} used by the filtergraph description.
Finally, when you specify a filtergraph on a shell commandline, you
need to perform a third level escaping for the shell special
characters contained within it.
For example, consider the following string to be embedded in
the @ref{drawtext} filter description @option{text} value:
@example
this is a 'string': may contain one, or more, special characters
@end example
This string contains the @code{'} special escaping character, and the
@code{:} special character, so it needs to be escaped in this way:
@example
text=this is a \'string\'\: may contain one, or more, special characters
@end example
A second level of escaping is required when embedding the filter
description in a filtergraph description, in order to escape all the
filtergraph special characters. Thus the example above becomes:
@example
drawtext=text=this is a \\\'string\\\'\\: may contain one\, or more\, special characters
@end example
(note that in addition to the @code{\'} escaping special characters,
also @code{,} needs to be escaped).
Finally an additional level of escaping is needed when writing the
filtergraph description in a shell command, which depends on the
escaping rules of the adopted shell. For example, assuming that
@code{\} is special and needs to be escaped with another @code{\}, the
previous string will finally result in:
@example
-vf "drawtext=text=this is a \\\\\\'string\\\\\\'\\\\: may contain one\\, or more\\, special characters"
@end example
@chapter Timeline editing
Some filters support a generic @option{enable} option. For the filters
supporting timeline editing, this option can be set to an expression which is
evaluated before sending a frame to the filter. If the evaluation is non-zero,
the filter will be enabled, otherwise the frame will be sent unchanged to the
next filter in the filtergraph.
The expression accepts the following values:
@table @samp
@item t
timestamp expressed in seconds, NAN if the input timestamp is unknown
@item n
sequential number of the input frame, starting from 0
@item pos
the position in the file of the input frame, NAN if unknown
@item w
@item h
width and height of the input frame if video
@end table
Additionally, these filters support an @option{enable} command that can be used
to re-define the expression.
Like any other filtering option, the @option{enable} option follows the same
rules.
For example, to enable a blur filter (@ref{smartblur}) from 10 seconds to 3
minutes, and a @ref{curves} filter starting at 3 seconds:
@example
smartblur = enable='between(t,10,3*60)',
curves = enable='gte(t,3)' : preset=cross_process
@end example
@c man end FILTERGRAPH DESCRIPTION
@chapter Audio Filters
@c man begin AUDIO FILTERS
When you configure your FFmpeg build, you can disable any of the
existing filters using @code{--disable-filters}.
The configure output will show the audio filters included in your
build.
Below is a description of the currently available audio filters.
@section acompressor
A compressor is mainly used to reduce the dynamic range of a signal.
Especially modern music is mostly compressed at a high ratio to
improve the overall loudness. It's done to get the highest attention
of a listener, "fatten" the sound and bring more "power" to the track.
If a signal is compressed too much it may sound dull or "dead"
afterwards or it may start to "pump" (which could be a powerful effect
but can also destroy a track completely).
The right compression is the key to reach a professional sound and is
the high art of mixing and mastering. Because of its complex settings
it may take a long time to get the right feeling for this kind of effect.
Compression is done by detecting the volume above a chosen level
@code{threshold} and dividing it by the factor set with @code{ratio}.
So if you set the threshold to -12dB and your signal reaches -6dB a ratio
of 2:1 will result in a signal at -9dB. Because an exact manipulation of
the signal would cause distortion of the waveform the reduction can be
levelled over the time. This is done by setting "Attack" and "Release".
@code{attack} determines how long the signal has to rise above the threshold
before any reduction will occur and @code{release} sets the time the signal
has to fall below the threshold to reduce the reduction again. Shorter signals
than the chosen attack time will be left untouched.
The overall reduction of the signal can be made up afterwards with the
@code{makeup} setting. So compressing the peaks of a signal about 6dB and
raising the makeup to this level results in a signal twice as loud than the
source. To gain a softer entry in the compression the @code{knee} flattens the
hard edge at the threshold in the range of the chosen decibels.
The filter accepts the following options:
@table @option
@item level_in
Set input gain. Default is 1. Range is between 0.015625 and 64.
@item threshold
If a signal of second stream rises above this level it will affect the gain
reduction of the first stream.
By default it is 0.125. Range is between 0.00097563 and 1.
@item ratio
Set a ratio by which the signal is reduced. 1:2 means that if the level
rose 4dB above the threshold, it will be only 2dB above after the reduction.
Default is 2. Range is between 1 and 20.
@item attack
Amount of milliseconds the signal has to rise above the threshold before gain
reduction starts. Default is 20. Range is between 0.01 and 2000.
@item release
Amount of milliseconds the signal has to fall below the threshold before
reduction is decreased again. Default is 250. Range is between 0.01 and 9000.
@item makeup
Set the amount by how much signal will be amplified after processing.
Default is 2. Range is from 1 and 64.
@item knee
Curve the sharp knee around the threshold to enter gain reduction more softly.
Default is 2.82843. Range is between 1 and 8.
@item link
Choose if the @code{average} level between all channels of input stream
or the louder(@code{maximum}) channel of input stream affects the
reduction. Default is @code{average}.
@item detection
Should the exact signal be taken in case of @code{peak} or an RMS one in case
of @code{rms}. Default is @code{rms} which is mostly smoother.
@item mix
How much to use compressed signal in output. Default is 1.
Range is between 0 and 1.
@end table
@section acrossfade
Apply cross fade from one input audio stream to another input audio stream.
The cross fade is applied for specified duration near the end of first stream.
The filter accepts the following options:
@table @option
@item nb_samples, ns
Specify the number of samples for which the cross fade effect has to last.
At the end of the cross fade effect the first input audio will be completely
silent. Default is 44100.
@item duration, d
Specify the duration of the cross fade effect. See
@ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}
for the accepted syntax.
By default the duration is determined by @var{nb_samples}.
If set this option is used instead of @var{nb_samples}.
@item overlap, o
Should first stream end overlap with second stream start. Default is enabled.
@item curve1
Set curve for cross fade transition for first stream.
@item curve2
Set curve for cross fade transition for second stream.
For description of available curve types see @ref{afade} filter description.
@end table
@subsection Examples
@itemize
@item
Cross fade from one input to another:
@example
ffmpeg -i first.flac -i second.flac -filter_complex acrossfade=d=10:c1=exp:c2=exp output.flac
@end example
@item
Cross fade from one input to another but without overlapping:
@example
ffmpeg -i first.flac -i second.flac -filter_complex acrossfade=d=10:o=0:c1=exp:c2=exp output.flac
@end example
@end itemize
@section adelay
Delay one or more audio channels.
Samples in delayed channel are filled with silence.
The filter accepts the following option:
@table @option
@item delays
Set list of delays in milliseconds for each channel separated by '|'.
At least one delay greater than 0 should be provided.
Unused delays will be silently ignored. If number of given delays is
smaller than number of channels all remaining channels will not be delayed.
@end table
@subsection Examples
@itemize
@item
Delay first channel by 1.5 seconds, the third channel by 0.5 seconds and leave
the second channel (and any other channels that may be present) unchanged.
@example
adelay=1500|0|500
@end example
@end itemize
@section aecho
Apply echoing to the input audio.
Echoes are reflected sound and can occur naturally amongst mountains
(and sometimes large buildings) when talking or shouting; digital echo
effects emulate this behaviour and are often used to help fill out the
sound of a single instrument or vocal. The time difference between the
original signal and the reflection is the @code{delay}, and the
loudness of the reflected signal is the @code{decay}.
Multiple echoes can have different delays and decays.
A description of the accepted parameters follows.
@table @option
@item in_gain
Set input gain of reflected signal. Default is @code{0.6}.
@item out_gain
Set output gain of reflected signal. Default is @code{0.3}.
@item delays
Set list of time intervals in milliseconds between original signal and reflections
separated by '|'. Allowed range for each @code{delay} is @code{(0 - 90000.0]}.
Default is @code{1000}.
@item decays
Set list of loudnesses of reflected signals separated by '|'.
Allowed range for each @code{decay} is @code{(0 - 1.0]}.
Default is @code{0.5}.
@end table
@subsection Examples
@itemize
@item
Make it sound as if there are twice as many instruments as are actually playing:
@example
aecho=0.8:0.88:60:0.4
@end example
@item
If delay is very short, then it sound like a (metallic) robot playing music:
@example
aecho=0.8:0.88:6:0.4
@end example
@item
A longer delay will sound like an open air concert in the mountains:
@example
aecho=0.8:0.9:1000:0.3
@end example
@item
Same as above but with one more mountain:
@example
aecho=0.8:0.9:1000|1800:0.3|0.25
@end example
@end itemize
@section aemphasis
Audio emphasis filter creates or restores material directly taken from LPs or
emphased CDs with different filter curves. E.g. to store music on vinyl the
signal has to be altered by a filter first to even out the disadvantages of
this recording medium.
Once the material is played back the inverse filter has to be applied to
restore the distortion of the frequency response.
The filter accepts the following options:
@table @option
@item level_in
Set input gain.
@item level_out
Set output gain.
@item mode
Set filter mode. For restoring material use @code{reproduction} mode, otherwise
use @code{production} mode. Default is @code{reproduction} mode.
@item type
Set filter type. Selects medium. Can be one of the following:
@table @option
@item col
select Columbia.
@item emi
select EMI.
@item bsi
select BSI (78RPM).
@item riaa
select RIAA.
@item cd
select Compact Disc (CD).
@item 50fm
select 50µs (FM).
@item 75fm
select 75µs (FM).
@item 50kf
select 50µs (FM-KF).
@item 75kf
select 75µs (FM-KF).
@end table
@end table
@section aeval
Modify an audio signal according to the specified expressions.
This filter accepts one or more expressions (one for each channel),
which are evaluated and used to modify a corresponding audio signal.
It accepts the following parameters:
@table @option
@item exprs
Set the '|'-separated expressions list for each separate channel. If
the number of input channels is greater than the number of
expressions, the last specified expression is used for the remaining
output channels.
@item channel_layout, c
Set output channel layout. If not specified, the channel layout is
specified by the number of expressions. If set to @samp{same}, it will
use by default the same input channel layout.
@end table
Each expression in @var{exprs} can contain the following constants and functions:
@table @option
@item ch
channel number of the current expression
@item n
number of the evaluated sample, starting from 0
@item s
sample rate
@item t
time of the evaluated sample expressed in seconds
@item nb_in_channels
@item nb_out_channels
input and output number of channels
@item val(CH)
the value of input channel with number @var{CH}
@end table
Note: this filter is slow. For faster processing you should use a
dedicated filter.
@subsection Examples
@itemize
@item
Half volume:
@example
aeval=val(ch)/2:c=same
@end example
@item
Invert phase of the second channel:
@example
aeval=val(0)|-val(1)
@end example
@end itemize
@anchor{afade}
@section afade
Apply fade-in/out effect to input audio.
A description of the accepted parameters follows.
@table @option
@item type, t
Specify the effect type, can be either @code{in} for fade-in, or
@code{out} for a fade-out effect. Default is @code{in}.
@item start_sample, ss
Specify the number of the start sample for starting to apply the fade
effect. Default is 0.
@item nb_samples, ns
Specify the number of samples for which the fade effect has to last. At
the end of the fade-in effect the output audio will have the same
volume as the input audio, at the end of the fade-out transition
the output audio will be silence. Default is 44100.
@item start_time, st
Specify the start time of the fade effect. Default is 0.
The value must be specified as a time duration; see
@ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}
for the accepted syntax.
If set this option is used instead of @var{start_sample}.
@item duration, d
Specify the duration of the fade effect. See
@ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}
for the accepted syntax.
At the end of the fade-in effect the output audio will have the same
volume as the input audio, at the end of the fade-out transition
the output audio will be silence.
By default the duration is determined by @var{nb_samples}.
If set this option is used instead of @var{nb_samples}.
@item curve
Set curve for fade transition.
It accepts the following values:
@table @option
@item tri
select triangular, linear slope (default)
@item qsin
select quarter of sine wave
@item hsin
select half of sine wave
@item esin
select exponential sine wave
@item log
select logarithmic
@item ipar
select inverted parabola
@item qua
select quadratic
@item cub
select cubic
@item squ
select square root
@item cbr
select cubic root
@item par
select parabola
@item exp
select exponential
@item iqsin
select inverted quarter of sine wave
@item ihsin
select inverted half of sine wave
@item dese
select double-exponential seat
@item desi
select double-exponential sigmoid
@end table
@end table
@subsection Examples
@itemize
@item
Fade in first 15 seconds of audio:
@example
afade=t=in:ss=0:d=15
@end example
@item
Fade out last 25 seconds of a 900 seconds audio:
@example
afade=t=out:st=875:d=25
@end example
@end itemize
@section afftfilt
Apply arbitrary expressions to samples in frequency domain.
@table @option
@item real
Set frequency domain real expression for each separate channel separated
by '|'. Default is "1".
If the number of input channels is greater than the number of
expressions, the last specified expression is used for the remaining
output channels.
@item imag
Set frequency domain imaginary expression for each separate channel
separated by '|'. If not set, @var{real} option is used.
Each expression in @var{real} and @var{imag} can contain the following
constants:
@table @option
@item sr
sample rate
@item b
current frequency bin number
@item nb
number of available bins
@item ch
channel number of the current expression
@item chs
number of channels
@item pts
current frame pts
@end table
@item win_size
Set window size.
It accepts the following values:
@table @samp
@item w16
@item w32
@item w64
@item w128
@item w256
@item w512
@item w1024
@item w2048
@item w4096
@item w8192
@item w16384
@item w32768
@item w65536
@end table
Default is @code{w4096}
@item win_func
Set window function. Default is @code{hann}.
@item overlap
Set window overlap. If set to 1, the recommended overlap for selected
window function will be picked. Default is @code{0.75}.
@end table
@subsection Examples
@itemize
@item
Leave almost only low frequencies in audio:
@example
afftfilt="1-clip((b/nb)*b,0,1)"
@end example
@end itemize
@anchor{aformat}
@section aformat
Set output format constraints for the input audio. The framework will
negotiate the most appropriate format to minimize conversions.
It accepts the following parameters:
@table @option
@item sample_fmts
A '|'-separated list of requested sample formats.
@item sample_rates
A '|'-separated list of requested sample rates.
@item channel_layouts
A '|'-separated list of requested channel layouts.
See @ref{channel layout syntax,,the Channel Layout section in the ffmpeg-utils(1) manual,ffmpeg-utils}
for the required syntax.
@end table
If a parameter is omitted, all values are allowed.
Force the output to either unsigned 8-bit or signed 16-bit stereo
@example
aformat=sample_fmts=u8|s16:channel_layouts=stereo
@end example
@section agate
A gate is mainly used to reduce lower parts of a signal. This kind of signal
processing reduces disturbing noise between useful signals.
Gating is done by detecting the volume below a chosen level @var{threshold}
and divide it by the factor set with @var{ratio}. The bottom of the noise
floor is set via @var{range}. Because an exact manipulation of the signal
would cause distortion of the waveform the reduction can be levelled over
time. This is done by setting @var{attack} and @var{release}.
@var{attack} determines how long the signal has to fall below the threshold
before any reduction will occur and @var{release} sets the time the signal
has to raise above the threshold to reduce the reduction again.
Shorter signals than the chosen attack time will be left untouched.
@table @option
@item level_in
Set input level before filtering.
Default is 1. Allowed range is from 0.015625 to 64.
@item range
Set the level of gain reduction when the signal is below the threshold.
Default is 0.06125. Allowed range is from 0 to 1.
@item threshold
If a signal rises above this level the gain reduction is released.
Default is 0.125. Allowed range is from 0 to 1.
@item ratio
Set a ratio about which the signal is reduced.
Default is 2. Allowed range is from 1 to 9000.
@item attack
Amount of milliseconds the signal has to rise above the threshold before gain
reduction stops.
Default is 20 milliseconds. Allowed range is from 0.01 to 9000.
@item release
Amount of milliseconds the signal has to fall below the threshold before the
reduction is increased again. Default is 250 milliseconds.
Allowed range is from 0.01 to 9000.
@item makeup
Set amount of amplification of signal after processing.
Default is 1. Allowed range is from 1 to 64.
@item knee
Curve the sharp knee around the threshold to enter gain reduction more softly.
Default is 2.828427125. Allowed range is from 1 to 8.
@item detection
Choose if exact signal should be taken for detection or an RMS like one.
Default is rms. Can be peak or rms.
@item link
Choose if the average level between all channels or the louder channel affects
the reduction.
Default is average. Can be average or maximum.
@end table
@section alimiter
The limiter prevents input signal from raising over a desired threshold.
This limiter uses lookahead technology to prevent your signal from distorting.
It means that there is a small delay after signal is processed. Keep in mind
that the delay it produces is the attack time you set.
The filter accepts the following options:
@table @option
@item level_in
Set input gain. Default is 1.
@item level_out
Set output gain. Default is 1.
@item limit
Don't let signals above this level pass the limiter. Default is 1.
@item attack
The limiter will reach its attenuation level in this amount of time in
milliseconds. Default is 5 milliseconds.
@item release
Come back from limiting to attenuation 1.0 in this amount of milliseconds.
Default is 50 milliseconds.
@item asc
When gain reduction is always needed ASC takes care of releasing to an
average reduction level rather than reaching a reduction of 0 in the release
time.
@item asc_level
Select how much the release time is affected by ASC, 0 means nearly no changes
in release time while 1 produces higher release times.
@item level
Auto level output signal. Default is enabled.
This normalizes audio back to 0dB if enabled.
@end table
Depending on picked setting it is recommended to upsample input 2x or 4x times
with @ref{aresample} before applying this filter.
@section allpass
Apply a two-pole all-pass filter with central frequency (in Hz)
@var{frequency}, and filter-width @var{width}.
An all-pass filter changes the audio's frequency to phase relationship
without changing its frequency to amplitude relationship.
The filter accepts the following options:
@table @option
@item frequency, f
Set 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.
@end table
@anchor{amerge}
@section amerge
Merge two or more audio streams into a single multi-channel stream.
The filter accepts the following options:
@table @option
@item inputs
Set the number of inputs. Default is 2.
@end table
If the channel layouts of the inputs are disjoint, and therefore compatible,
the channel layout of the output will be set accordingly and the channels
will be reordered as necessary. If the channel layouts of the inputs are not
disjoint, the output will have all the channels of the first input then all
the channels of the second input, in that order, and the channel layout of
the output will be the default value corresponding to the total number of
channels.
For example, if the first input is in 2.1 (FL+FR+LF) and the second input
is FC+BL+BR, then the output will be in 5.1, with the channels in the
following order: a1, a2, b1, a3, b2, b3 (a1 is the first channel of the
first input, b1 is the first channel of the second input).
On the other hand, if both input are in stereo, the output channels will be
in the default order: a1, a2, b1, b2, and the channel layout will be
arbitrarily set to 4.0, which may or may not be the expected value.
All inputs must have the same sample rate, and format.
If inputs do not have the same duration, the output will stop with the
shortest.
@subsection Examples
@itemize
@item
Merge two mono files into a stereo stream:
@example
amovie=left.wav [l] ; amovie=right.mp3 [r] ; [l] [r] amerge
@end example
@item
Multiple merges assuming 1 video stream and 6 audio streams in @file{input.mkv}:
@example
ffmpeg -i input.mkv -filter_complex "[0:1][0:2][0:3][0:4][0:5][0:6] amerge=inputs=6" -c:a pcm_s16le output.mkv
@end example
@end itemize
@section amix
Mixes multiple audio inputs into a single output.
Note that this filter only supports float samples (the @var{amerge}
and @var{pan} audio filters support many formats). If the @var{amix}
input has integer samples then @ref{aresample} will be automatically
inserted to perform the conversion to float samples.
For example
@example
ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex amix=inputs=3:duration=first:dropout_transition=3 OUTPUT
@end example
will mix 3 input audio streams to a single output with the same duration as the
first input and a dropout transition time of 3 seconds.
It accepts the following parameters:
@table @option
@item inputs
The number of inputs. If unspecified, it defaults to 2.
@item duration
How to determine the end-of-stream.
@table @option
@item longest
The duration of the longest input. (default)
@item shortest
The duration of the shortest input.
@item first
The duration of the first input.
@end table
@item dropout_transition
The transition time, in seconds, for volume renormalization when an input
stream ends. The default value is 2 seconds.
@end table
@section anequalizer
High-order parametric multiband equalizer for each channel.
It accepts the following parameters:
@table @option
@item params
This option string is in format:
"c@var{chn} f=@var{cf} w=@var{w} g=@var{g} t=@var{f} | ..."
Each equalizer band is separated by '|'.
@table @option
@item chn
Set channel number to which equalization will be applied.
If input doesn't have that channel the entry is ignored.
@item cf
Set central frequency for band.
If input doesn't have that frequency the entry is ignored.
@item w
Set band width in hertz.
@item g
Set band gain in dB.
@item f
Set filter type for band, optional, can be:
@table @samp
@item 0
Butterworth, this is default.
@item 1
Chebyshev type 1.
@item 2
Chebyshev type 2.
@end table
@end table
@item curves
With this option activated frequency response of anequalizer is displayed
in video stream.
@item size
Set video stream size. Only useful if curves option is activated.
@item mgain
Set max gain that will be displayed. Only useful if curves option is activated.
Setting this to reasonable value allows to display gain which is derived from
neighbour bands which are too close to each other and thus produce higher gain
when both are activated.
@item fscale
Set frequency scale used to draw frequency response in video output.
Can be linear or logarithmic. Default is logarithmic.
@item colors
Set color for each channel curve which is going to be displayed in video stream.
This is list of color names separated by space or by '|'.
Unrecognised or missing colors will be replaced by white color.
@end table
@subsection Examples
@itemize
@item
Lower gain by 10 of central frequency 200Hz and width 100 Hz
for first 2 channels using Chebyshev type 1 filter:
@example
anequalizer=c0 f=200 w=100 g=-10 t=1|c1 f=200 w=100 g=-10 t=1
@end example
@end itemize
@subsection Commands
This filter supports the following commands:
@table @option
@item change
Alter existing filter parameters.
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{gain} set new gain parameter in dB.
Full filter invocation with asendcmd may look like this:
asendcmd=c='4.0 anequalizer change 0|f=200|w=50|g=1',anequalizer=...
@end table
@section anull
Pass the audio source unchanged to the output.
@section apad
Pad the end of an audio stream with silence.
This can be used together with @command{ffmpeg} @option{-shortest} to
extend audio streams to the same length as the video stream.
A description of the accepted options follows.
@table @option
@item packet_size
Set silence packet size. Default value is 4096.
@item pad_len
Set the number of samples of silence to add to the end. After the
value is reached, the stream is terminated. This option is mutually
exclusive with @option{whole_len}.
@item whole_len
Set the minimum total number of samples in the output audio stream. If
the value is longer than the input audio length, silence is added to
the end, until the value is reached. This option is mutually exclusive
with @option{pad_len}.
@end table
If neither the @option{pad_len} nor the @option{whole_len} option is
set, the filter will add silence to the end of the input stream
indefinitely.
@subsection Examples
@itemize
@item
Add 1024 samples of silence to the end of the input:
@example
apad=pad_len=1024
@end example
@item
Make sure the audio output will contain at least 10000 samples, pad
the input with silence if required:
@example
apad=whole_len=10000
@end example
@item
Use @command{ffmpeg} to pad the audio input with silence, so that the
video stream will always result the shortest and will be converted
until the end in the output file when using the @option{shortest}
option:
@example
ffmpeg -i VIDEO -i AUDIO -filter_complex "[1:0]apad" -shortest OUTPUT
@end example
@end itemize
@section aphaser
Add a phasing effect to the input audio.
A phaser filter creates series of peaks and troughs in the frequency spectrum.
The position of the peaks and troughs are modulated so that they vary over time, creating a sweeping effect.
A description of the accepted parameters follows.
@table @option
@item in_gain
Set input gain. Default is 0.4.
@item out_gain
Set output gain. Default is 0.74
@item delay
Set delay in milliseconds. Default is 3.0.
@item decay
Set decay. Default is 0.4.
@item speed
Set modulation speed in Hz. Default is 0.5.
@item type
Set modulation type. Default is triangular.
It accepts the following values:
@table @samp
@item triangular, t
@item sinusoidal, s
@end table
@end table
@section apulsator
Audio pulsator is something between an autopanner and a tremolo.
But it can produce funny stereo effects as well. Pulsator changes the volume
of the left and right channel based on a LFO (low frequency oscillator) with
different waveforms and shifted phases.
This filter have the ability to define an offset between left and right
channel. An offset of 0 means that both LFO shapes match each other.
The left and right channel are altered equally - a conventional tremolo.
An offset of 50% means that the shape of the right channel is exactly shifted
in phase (or moved backwards about half of the frequency) - pulsator acts as
an autopanner. At 1 both curves match again. Every setting in between moves the
phase shift gapless between all stages and produces some "bypassing" sounds with
sine and triangle waveforms. The more you set the offset near 1 (starting from
the 0.5) the faster the signal passes from the left to the right speaker.
The filter accepts the following options:
@table @option
@item level_in
Set input gain. By default it is 1. Range is [0.015625 - 64].
@item level_out
Set output gain. By default it is 1. Range is [0.015625 - 64].
@item mode
Set waveform shape the LFO will use. Can be one of: sine, triangle, square,
sawup or sawdown. Default is sine.
@item amount
Set modulation. Define how much of original signal is affected by the LFO.
@item offset_l
Set left channel offset. Default is 0. Allowed range is [0 - 1].
@item offset_r
Set right channel offset. Default is 0.5. Allowed range is [0 - 1].
@item width
Set pulse width. Default is 1. Allowed range is [0 - 2].
@item timing
Set possible timing mode. Can be one of: bpm, ms or hz. Default is hz.
@item bpm
Set bpm. Default is 120. Allowed range is [30 - 300]. Only used if timing
is set to bpm.
@item ms
Set ms. Default is 500. Allowed range is [10 - 2000]. Only used if timing
is set to ms.
@item hz
Set frequency in Hz. Default is 2. Allowed range is [0.01 - 100]. Only used
if timing is set to hz.
@end table
@anchor{aresample}
@section aresample
Resample the input audio to the specified parameters, using the
libswresample library. If none are specified then the filter will
automatically convert between its input and output.
This filter is also able to stretch/squeeze the audio data to make it match
the timestamps or to inject silence / cut out audio to make it match the
timestamps, do a combination of both or do neither.
The filter accepts the syntax
[@var{sample_rate}:]@var{resampler_options}, where @var{sample_rate}
expresses a sample rate and @var{resampler_options} is a list of
@var{key}=@var{value} pairs, separated by ":". See the
ffmpeg-resampler manual for the complete list of supported options.
@subsection Examples
@itemize
@item
Resample the input audio to 44100Hz:
@example
aresample=44100
@end example
@item
Stretch/squeeze samples to the given timestamps, with a maximum of 1000
samples per second compensation:
@example
aresample=async=1000
@end example
@end itemize
@section asetnsamples
Set the number of samples per each output audio frame.
The last output packet may contain a different number of samples, as
the filter will flush all the remaining samples when the input audio
signal its end.
The filter accepts the following options:
@table @option
@item nb_out_samples, n
Set the number of frames per each output audio frame. The number is
intended as the number of samples @emph{per each channel}.
Default value is 1024.
@item pad, p
If set to 1, the filter will pad the last audio frame with zeroes, so
that the last frame will contain the same number of samples as the
previous ones. Default value is 1.
@end table
For example, to set the number of per-frame samples to 1234 and
disable padding for the last frame, use:
@example
asetnsamples=n=1234:p=0
@end example
@section asetrate
Set the sample rate without altering the PCM data.
This will result in a change of speed and pitch.
The filter accepts the following options:
@table @option
@item sample_rate, r
Set the output sample rate. Default is 44100 Hz.
@end table
@section ashowinfo
Show a line containing various information for each input audio frame.
The input audio is not modified.
The shown line contains a sequence of key/value pairs of the form
@var{key}:@var{value}.
The following values are shown in the output:
@table @option
@item n
The (sequential) number of the input frame, starting from 0.
@item pts
The presentation timestamp of the input frame, in time base units; the time base
depends on the filter input pad, and is usually 1/@var{sample_rate}.
@item pts_time
The presentation timestamp of the input frame in seconds.
@item pos
position of the frame in the input stream, -1 if this information in
unavailable and/or meaningless (for example in case of synthetic audio)
@item fmt
The sample format.
@item chlayout
The channel layout.
@item rate
The sample rate for the audio frame.
@item nb_samples
The number of samples (per channel) in the frame.
@item checksum
The Adler-32 checksum (printed in hexadecimal) of the audio data. For planar
audio, the data is treated as if all the planes were concatenated.
@item plane_checksums
A list of Adler-32 checksums for each data plane.
@end table
@anchor{astats}
@section astats
Display time domain statistical information about the audio channels.
Statistics are calculated and displayed for each audio channel and,
where applicable, an overall figure is also given.
It accepts the following option:
@table @option
@item length
Short window length in seconds, used for peak and trough RMS measurement.
Default is @code{0.05} (50 milliseconds). Allowed range is @code{[0.1 - 10]}.
@item metadata
Set metadata injection. All the metadata keys are prefixed with @code{lavfi.astats.X},
where @code{X} is channel number starting from 1 or string @code{Overall}. Default is
disabled.
Available keys for each channel are:
DC_offset
Min_level
Max_level
Min_difference
Max_difference
Mean_difference
Peak_level
RMS_peak
RMS_trough
Crest_factor
Flat_factor
Peak_count
Bit_depth
and for Overall:
DC_offset
Min_level
Max_level
Min_difference
Max_difference
Mean_difference
Peak_level
RMS_level
RMS_peak
RMS_trough
Flat_factor
Peak_count
Bit_depth
Number_of_samples
For example full key look like this @code{lavfi.astats.1.DC_offset} or
this @code{lavfi.astats.Overall.Peak_count}.
For description what each key means read below.
@item reset
Set number of frame after which stats are going to be recalculated.
Default is disabled.
@end table
A description of each shown parameter follows:
@table @option
@item DC offset
Mean amplitude displacement from zero.
@item Min level
Minimal sample level.
@item Max level
Maximal sample level.
@item Min difference
Minimal difference between two consecutive samples.
@item Max difference
Maximal difference between two consecutive samples.
@item Mean difference
Mean difference between two consecutive samples.
The average of each difference between two consecutive samples.
@item Peak level dB
@item RMS level dB
Standard peak and RMS level measured in dBFS.
@item RMS peak dB
@item RMS trough dB
Peak and trough values for RMS level measured over a short window.
@item Crest factor
Standard ratio of peak to RMS level (note: not in dB).
@item Flat factor
Flatness (i.e. consecutive samples with the same value) of the signal at its peak levels
(i.e. either @var{Min level} or @var{Max level}).
@item Peak count
Number of occasions (not the number of samples) that the signal attained either
@var{Min level} or @var{Max level}.
@item Bit depth
Overall bit depth of audio. Number of bits used for each sample.
@end table
@section asyncts
Synchronize audio data with timestamps by squeezing/stretching it and/or
dropping samples/adding silence when needed.
This filter is not built by default, please use @ref{aresample} to do squeezing/stretching.
It accepts the following parameters:
@table @option
@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
The minimum difference between timestamps and audio data (in seconds) to trigger
adding/dropping samples. The default value is 0.1. If you get an imperfect
sync with this filter, try setting this parameter to 0.
@item max_comp
The maximum compensation in samples per second. Only relevant with compensate=1.
The default value is 500.
@item first_pts
Assume that the first PTS should be this value. The time base is 1 / sample
rate. This allows for padding/trimming at the start of the 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.
@end table
@section atempo
Adjust audio tempo.
The filter accepts exactly one parameter, the audio tempo. If not
specified then the filter will assume nominal 1.0 tempo. Tempo must
be in the [0.5, 2.0] range.
@subsection Examples
@itemize
@item
Slow down audio to 80% tempo:
@example
atempo=0.8
@end example
@item
To speed up audio to 125% tempo:
@example
atempo=1.25
@end example
@end itemize
@section atrim
Trim the input so that the output contains one continuous subpart of the input.
It accepts the following parameters:
@table @option
@item start
Timestamp (in seconds) of the start of the section to keep. I.e. the audio
sample with the timestamp @var{start} will be the first sample in the output.
@item end
Specify time 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
The maximum duration of the output in seconds.
@item start_sample
The number of the first sample that should be output.
@item end_sample
The number of the first sample that should be dropped.
@end table
@option{start}, @option{end}, and @option{duration} are expressed as time
duration specifications; see
@ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
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
to have 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 bandpass
Apply a two-pole Butterworth band-pass filter with central
frequency @var{frequency}, and (3dB-point) band-width width.
The @var{csg} option selects a constant skirt gain (peak gain = Q)
instead of the default: constant 0dB peak gain.
The filter roll off at 6dB per octave (20dB per decade).
The filter accepts the following options:
@table @option
@item frequency, f
Set the filter's central frequency. Default is @code{3000}.
@item csg
Constant skirt gain if set to 1. Defaults to 0.
@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.
@end table
@section bandreject
Apply a two-pole Butterworth band-reject filter with central
frequency @var{frequency}, and (3dB-point) band-width @var{width}.
The filter roll off at 6dB per octave (20dB per decade).
The filter accepts the following options:
@table @option
@item frequency, f
Set the filter's central frequency. Default is @code{3000}.
@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.
@end table
@section bass
Boost or cut the bass (lower) 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:
@table @option
@item gain, g
Give the gain at 0 Hz. 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 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{100} 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
Determine how steep is the filter's shelf transition.
@end table
@section biquad
Apply a biquad IIR filter with the given coefficients.
Where @var{b0}, @var{b1}, @var{b2} and @var{a0}, @var{a1}, @var{a2}
are the numerator and denominator coefficients respectively.
@section bs2b
Bauer stereo to binaural transformation, which improves headphone listening of
stereo audio records.
It accepts the following parameters:
@table @option
@item profile
Pre-defined crossfeed level.
@table @option
@item default
Default level (fcut=700, feed=50).
@item cmoy
Chu Moy circuit (fcut=700, feed=60).
@item jmeier
Jan Meier circuit (fcut=650, feed=95).
@end table
@item fcut
Cut frequency (in Hz).
@item feed
Feed level (in Hz).
@end table
@section channelmap
Remap input channels to new locations.
It accepts the following parameters:
@table @option
@item channel_layout
The 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 indices.
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|4:5.1' out.wav
@end example
@section channelsplit
Split each channel from an input audio stream into a separate output stream.
It accepts the following parameters:
@table @option
@item channel_layout
The channel layout of the input stream. The 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.
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 chorus
Add a chorus effect to the audio.
Can make a single vocal sound like a chorus, but can also be applied to instrumentation.
Chorus resembles an echo effect with a short delay, but whereas with echo the delay is
constant, with chorus, it is varied using using sinusoidal or triangular modulation.
The modulation depth defines the range the modulated delay is played before or after
the delay. Hence the delayed sound will sound slower or faster, that is the delayed
sound tuned around the original one, like in a chorus where some vocals are slightly
off key.
It accepts the following parameters:
@table @option
@item in_gain
Set input gain. Default is 0.4.
@item out_gain
Set output gain. Default is 0.4.
@item delays
Set delays. A typical delay is around 40ms to 60ms.
@item decays
Set decays.
@item speeds
Set speeds.
@item depths
Set depths.
@end table
@subsection Examples
@itemize
@item
A single delay:
@example
chorus=0.7:0.9:55:0.4:0.25:2
@end example
@item
Two delays:
@example
chorus=0.6:0.9:50|60:0.4|0.32:0.25|0.4:2|1.3
@end example
@item
Fuller sounding chorus with three delays:
@example
chorus=0.5:0.9:50|60|40:0.4|0.32|0.3:0.25|0.4|0.3:2|2.3|1.3
@end example
@end itemize
@section compand
Compress or expand the audio's dynamic range.
It accepts the following parameters:
@table @option
@item attacks
@item decays
A list of times in seconds for each channel over which the instantaneous level
of the input signal is averaged to determine its volume. @var{attacks} refers to
increase of volume and @var{decays} refers to decrease of volume. For most
situations, the attack time (response to the audio getting louder) should be
shorter than the decay time, because the human ear is more sensitive to sudden
loud audio than sudden soft audio. A typical value for attack is 0.3 seconds and
a typical value for decay is 0.8 seconds.
If specified number of attacks & decays is lower than number of channels, the last
set attack/decay will be used for all remaining channels.
@item points
A list of points for the transfer function, specified in dB relative to the
maximum possible signal amplitude. Each key points list must be defined using
the following syntax: @code{x0/y0|x1/y1|x2/y2|....} or
@code{x0/y0 x1/y1 x2/y2 ....}
The input values must be in strictly increasing order but the transfer function
does not have to be monotonically rising. The point @code{0/0} is assumed but
may be overridden (by @code{0/out-dBn}). Typical values for the transfer
function are @code{-70/-70|-60/-20}.
@item soft-knee
Set the curve radius in dB for all joints. It defaults to 0.01.
@item gain
Set the additional gain in dB to be applied at all points on the transfer
function. This allows for easy adjustment of the overall gain.
It defaults to 0.
@item volume
Set an initial volume, in dB, to be assumed for each channel when filtering
starts. This permits the user to supply a nominal level initially, so that, for
example, a very large gain is not applied to initial signal levels before the
companding has begun to operate. A typical value for audio which is initially
quiet is -90 dB. It defaults to 0.
@item delay
Set a delay, in seconds. The input audio is analyzed immediately, but audio is
delayed before being fed to the volume adjuster. Specifying a delay
approximately equal to the attack/decay times allows the filter to effectively
operate in predictive rather than reactive mode. It defaults to 0.
@end table
@subsection Examples
@itemize
@item
Make music with both quiet and loud passages suitable for listening to in a
noisy environment:
@example
compand=.3|.3:1|1:-90/-60|-60/-40|-40/-30|-20/-20:6:0:-90:0.2
@end example
Another example for audio with whisper and explosion parts:
@example
compand=0|0:1|1:-90/-900|-70/-70|-30/-9|0/-3:6:0:0:0
@end example
@item
A noise gate for when the noise is at a lower level than the signal:
@example
compand=.1|.1:.2|.2:-900/-900|-50.1/-900|-50/-50:.01:0:-90:.1
@end example
@item
Here is another noise gate, this time for when the noise is at a higher level
than the signal (making it, in some ways, similar to squelch):
@example
compand=.1|.1:.1|.1:-45.1/-45.1|-45/-900|0/-900:.01:45:-90:.1
@end example
@item
2:1 compression starting at -6dB:
@example
compand=points=-80/-80|-6/-6|0/-3.8|20/3.5
@end example
@item
2:1 compression starting at -9dB:
@example
compand=points=-80/-80|-9/-9|0/-5.3|20/2.9
@end example
@item
2:1 compression starting at -12dB:
@example
compand=points=-80/-80|-12/-12|0/-6.8|20/1.9
@end example
@item
2:1 compression starting at -18dB:
@example
compand=points=-80/-80|-18/-18|0/-9.8|20/0.7
@end example
@item
3:1 compression starting at -15dB:
@example
compand=points=-80/-80|-15/-15|0/-10.8|20/-5.2
@end example
@item
Compressor/Gate:
@example
compand=points=-80/-105|-62/-80|-15.4/-15.4|0/-12|20/-7.6
@end example
@item
Expander:
@example
compand=attacks=0:points=-80/-169|-54/-80|-49.5/-64.6|-41.1/-41.1|-25.8/-15|-10.8/-4.5|0/0|20/8.3
@end example
@item
Hard limiter at -6dB:
@example
compand=attacks=0:points=-80/-80|-6/-6|20/-6
@end example
@item
Hard limiter at -12dB:
@example
compand=attacks=0:points=-80/-80|-12/-12|20/-12
@end example
@item
Hard noise gate at -35 dB:
@example
compand=attacks=0:points=-80/-115|-35.1/-80|-35/-35|20/20
@end example
@item
Soft limiter:
@example
compand=attacks=0:points=-80/-80|-12.4/-12.4|-6/-8|0/-6.8|20/-2.8
@end example
@end itemize
@section compensationdelay
Compensation Delay Line is a metric based delay to compensate differing
positions of microphones or speakers.
For example, you have recorded guitar with two microphones placed in
different location. Because the front of sound wave has fixed speed in
normal conditions, the phasing of microphones can vary and depends on
their location and interposition. The best sound mix can be achieved when
these microphones are in phase (synchronized). Note that distance of
~30 cm between microphones makes one microphone to capture signal in
antiphase to another microphone. That makes the final mix sounding moody.
This filter helps to solve phasing problems by adding different delays
to each microphone track and make them synchronized.
The best result can be reached when you take one track as base and
synchronize other tracks one by one with it.
Remember that synchronization/delay tolerance depends on sample rate, too.
Higher sample rates will give more tolerance.
It accepts the following parameters:
@table @option
@item mm
Set millimeters distance. This is compensation distance for fine tuning.
Default is 0.
@item cm
Set cm distance. This is compensation distance for tightening distance setup.
Default is 0.
@item m
Set meters distance. This is compensation distance for hard distance setup.
Default is 0.
@item dry
Set dry amount. Amount of unprocessed (dry) signal.
Default is 0.
@item wet
Set wet amount. Amount of processed (wet) signal.
Default is 1.
@item temp
Set temperature degree in Celsius. This is the temperature of the environment.
Default is 20.
@end table
@section dcshift
Apply a DC shift to the audio.
This can be useful to remove a DC offset (caused perhaps by a hardware problem
in the recording chain) from the audio. The effect of a DC offset is reduced
headroom and hence volume. The @ref{astats} filter can be used to determine if
a signal has a DC offset.
@table @option
@item shift
Set the DC shift, allowed range is [-1, 1]. It indicates the amount to shift
the audio.
@item limitergain
Optional. It should have a value much less than 1 (e.g. 0.05 or 0.02) and is
used to prevent clipping.
@end table
@section dynaudnorm
Dynamic Audio Normalizer.
This filter applies a certain amount of gain to the input audio in order
to bring its peak magnitude to a target level (e.g. 0 dBFS). However, in
contrast to more "simple" normalization algorithms, the Dynamic Audio
Normalizer *dynamically* re-adjusts the gain factor to the input audio.
This allows for applying extra gain to the "quiet" sections of the audio
while avoiding distortions or clipping the "loud" sections. In other words:
The Dynamic Audio Normalizer will "even out" the volume of quiet and loud
sections, in the sense that the volume of each section is brought to the
same target level. Note, however, that the Dynamic Audio Normalizer achieves
this goal *without* applying "dynamic range compressing". It will retain 100%
of the dynamic range *within* each section of the audio file.
@table @option
@item f
Set the frame length in milliseconds. In range from 10 to 8000 milliseconds.
Default is 500 milliseconds.
The Dynamic Audio Normalizer processes the input audio in small chunks,
referred to as frames. This is required, because a peak magnitude has no
meaning for just a single sample value. Instead, we need to determine the
peak magnitude for a contiguous sequence of sample values. While a "standard"
normalizer would simply use the peak magnitude of the complete file, the
Dynamic Audio Normalizer determines the peak magnitude individually for each
frame. The length of a frame is specified in milliseconds. By default, the
Dynamic Audio Normalizer uses a frame length of 500 milliseconds, which has
been found to give good results with most files.
Note that the exact frame length, in number of samples, will be determined
automatically, based on the sampling rate of the individual input audio file.
@item g
Set the Gaussian filter window size. In range from 3 to 301, must be odd
number. Default is 31.
Probably the most important parameter of the Dynamic Audio Normalizer is the
@code{window size} of the Gaussian smoothing filter. The filter's window size
is specified in frames, centered around the current frame. For the sake of
simplicity, this must be an odd number. Consequently, the default value of 31
takes into account the current frame, as well as the 15 preceding frames and
the 15 subsequent frames. Using a larger window results in a stronger
smoothing effect and thus in less gain variation, i.e. slower gain
adaptation. Conversely, using a smaller window results in a weaker smoothing
effect and thus in more gain variation, i.e. faster gain adaptation.
In other words, the more you increase this value, the more the Dynamic Audio
Normalizer will behave like a "traditional" normalization filter. On the
contrary, the more you decrease this value, the more the Dynamic Audio
Normalizer will behave like a dynamic range compressor.
@item p
Set the target peak value. This specifies the highest permissible magnitude
level for the normalized audio input. This filter will try to approach the
target peak magnitude as closely as possible, but at the same time it also
makes sure that the normalized signal will never exceed the peak magnitude.
A frame's maximum local gain factor is imposed directly by the target peak
magnitude. The default value is 0.95 and thus leaves a headroom of 5%*.
It is not recommended to go above this value.
@item m
Set the maximum gain factor. In range from 1.0 to 100.0. Default is 10.0.
The Dynamic Audio Normalizer determines the maximum possible (local) gain
factor for each input frame, i.e. the maximum gain factor that does not
result in clipping or distortion. The maximum gain factor is determined by
the frame's highest magnitude sample. However, the Dynamic Audio Normalizer
additionally bounds the frame's maximum gain factor by a predetermined
(global) maximum gain factor. This is done in order to avoid excessive gain
factors in "silent" or almost silent frames. By default, the maximum gain
factor is 10.0, For most inputs the default value should be sufficient and
it usually is not recommended to increase this value. Though, for input
with an extremely low overall volume level, it may be necessary to allow even
higher gain factors. Note, however, that the Dynamic Audio Normalizer does
not simply apply a "hard" threshold (i.e. cut off values above the threshold).
Instead, a "sigmoid" threshold function will be applied. This way, the
gain factors will smoothly approach the threshold value, but never exceed that
value.
@item r
Set the target RMS. In range from 0.0 to 1.0. Default is 0.0 - disabled.
By default, the Dynamic Audio Normalizer performs "peak" normalization.
This means that the maximum local gain factor for each frame is defined
(only) by the frame's highest magnitude sample. This way, the samples can
be amplified as much as possible without exceeding the maximum signal
level, i.e. without clipping. Optionally, however, the Dynamic Audio
Normalizer can also take into account the frame's root mean square,
abbreviated RMS. In electrical engineering, the RMS is commonly used to
determine the power of a time-varying signal. It is therefore considered
that the RMS is a better approximation of the "perceived loudness" than
just looking at the signal's peak magnitude. Consequently, by adjusting all
frames to a constant RMS value, a uniform "perceived loudness" can be
established. If a target RMS value has been specified, a frame's local gain
factor is defined as the factor that would result in exactly that RMS value.
Note, however, that the maximum local gain factor is still restricted by the
frame's highest magnitude sample, in order to prevent clipping.
@item n
Enable channels coupling. By default is enabled.
By default, the Dynamic Audio Normalizer will amplify all channels by the same
amount. This means the same gain factor will be applied to all channels, i.e.
the maximum possible gain factor is determined by the "loudest" channel.
However, in some recordings, it may happen that the volume of the different
channels is uneven, e.g. one channel may be "quieter" than the other one(s).
In this case, this option can be used to disable the channel coupling. This way,
the gain factor will be determined independently for each channel, depending
only on the individual channel's highest magnitude sample. This allows for
harmonizing the volume of the different channels.
@item c
Enable DC bias correction. By default is disabled.
An audio signal (in the time domain) is a sequence of sample values.
In the Dynamic Audio Normalizer these sample values are represented in the
-1.0 to 1.0 range, regardless of the original input format. Normally, the
audio signal, or "waveform", should be centered around the zero point.
That means if we calculate the mean value of all samples in a file, or in a
single frame, then the result should be 0.0 or at least very close to that
value. If, however, there is a significant deviation of the mean value from
0.0, in either positive or negative direction, this is referred to as a
DC bias or DC offset. Since a DC bias is clearly undesirable, the Dynamic
Audio Normalizer provides optional DC bias correction.
With DC bias correction enabled, the Dynamic Audio Normalizer will determine
the mean value, or "DC correction" offset, of each input frame and subtract
that value from all of the frame's sample values which ensures those samples
are centered around 0.0 again. Also, in order to avoid "gaps" at the frame
boundaries, the DC correction offset values will be interpolated smoothly
between neighbouring frames.
@item b
Enable alternative boundary mode. By default is disabled.
The Dynamic Audio Normalizer takes into account a certain neighbourhood
around each frame. This includes the preceding frames as well as the
subsequent frames. However, for the "boundary" frames, located at the very
beginning and at the very end of the audio file, not all neighbouring
frames are available. In particular, for the first few frames in the audio
file, the preceding frames are not known. And, similarly, for the last few
frames in the audio file, the subsequent frames are not known. Thus, the
question arises which gain factors should be assumed for the missing frames
in the "boundary" region. The Dynamic Audio Normalizer implements two modes
to deal with this situation. The default boundary mode assumes a gain factor
of exactly 1.0 for the missing frames, resulting in a smooth "fade in" and
"fade out" at the beginning and at the end of the input, respectively.
@item s
Set the compress factor. In range from 0.0 to 30.0. Default is 0.0.
By default, the Dynamic Audio Normalizer does not apply "traditional"
compression. This means that signal peaks will not be pruned and thus the
full dynamic range will be retained within each local neighbourhood. However,
in some cases it may be desirable to combine the Dynamic Audio Normalizer's
normalization algorithm with a more "traditional" compression.
For this purpose, the Dynamic Audio Normalizer provides an optional compression
(thresholding) function. If (and only if) the compression feature is enabled,
all input frames will be processed by a soft knee thresholding function prior
to the actual normalization process. Put simply, the thresholding function is
going to prune all samples whose magnitude exceeds a certain threshold value.
However, the Dynamic Audio Normalizer does not simply apply a fixed threshold
value. Instead, the threshold value will be adjusted for each individual
frame.
In general, smaller parameters result in stronger compression, and vice versa.
Values below 3.0 are not recommended, because audible distortion may appear.
@end table
@section earwax
Make audio easier to listen to on headphones.
This filter adds `cues' to 44.1kHz stereo (i.e. audio CD format) audio
so that when listened to on headphones the stereo image is moved from
inside your head (standard for headphones) to outside and in front of
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
@subsection Examples
@itemize
@item
Attenuate 10 dB at 1000 Hz, with a bandwidth of 200 Hz:
@example
equalizer=f=1000:width_type=h:width=200:g=-10
@end example
@item
Apply 2 dB gain at 1000 Hz with Q 1 and attenuate 5 dB at 100 Hz with Q 2:
@example
equalizer=f=1000:width_type=q:width=1:g=2,equalizer=f=100:width_type=q:width=2:g=-5
@end example
@end itemize
@section extrastereo
Linearly increases the difference between left and right channels which
adds some sort of "live" effect to playback.
The filter accepts the following option:
@table @option
@item m
Sets the difference coefficient (default: 2.5). 0.0 means mono sound
(average of both channels), with 1.0 sound will be unchanged, with
-1.0 left and right channels will be swapped.
@item c
Enable clipping. By default is enabled.
@end table
@section flanger
Apply a flanging effect to the audio.
The filter accepts the following options:
@table @option
@item delay
Set base delay in milliseconds. Range from 0 to 30. Default value is 0.
@item depth
Set added swep delay in milliseconds. Range from 0 to 10. Default value is 2.
@item regen
Set percentage regeneration (delayed signal feedback). Range from -95 to 95.
Default value is 0.
@item width
Set percentage of delayed signal mixed with original. Range from 0 to 100.
Default value is 71.
@item speed
Set sweeps per second (Hz). Range from 0.1 to 10. Default value is 0.5.
@item shape
Set swept wave shape, can be @var{triangular} or @var{sinusoidal}.
Default value is @var{sinusoidal}.
@item phase
Set swept wave percentage-shift for multi channel. Range from 0 to 100.
Default value is 25.
@item interp
Set delay-line interpolation, @var{linear} or @var{quadratic}.
Default is @var{linear}.
@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.
It accepts the following parameters:
@table @option
@item inputs
The number of input streams. It defaults to 2.
@item channel_layout
The desired output channel layout. It 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 they 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.
Join 3 inputs (with properly set channel layouts):
@example
ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex join=inputs=3 OUTPUT
@end example
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 ladspa
Load a LADSPA (Linux Audio Developer's Simple Plugin API) plugin.
To enable compilation of this filter you need to configure FFmpeg with
@code{--enable-ladspa}.
@table @option
@item file, f
Specifies the name of LADSPA plugin library to load. If the environment
variable @env{LADSPA_PATH} is defined, the LADSPA plugin is searched in
each one of the directories specified by the colon separated list in
@env{LADSPA_PATH}, otherwise in the standard LADSPA paths, which are in
this order: @file{HOME/.ladspa/lib/}, @file{/usr/local/lib/ladspa/},
@file{/usr/lib/ladspa/}.
@item plugin, p
Specifies the plugin within the library. Some libraries contain only
one plugin, but others contain many of them. If this is not set filter
will list all available plugins within the specified library.
@item controls, c
Set the '|' separated list of controls which are zero or more floating point
values that determine the behavior of the loaded plugin (for example delay,
threshold or gain).
Controls need to be defined using the following syntax:
c0=@var{value0}|c1=@var{value1}|c2=@var{value2}|..., where
@var{valuei} is the value set on the @var{i}-th control.
Alternatively they can be also defined using the following syntax:
@var{value0}|@var{value1}|@var{value2}|..., where
@var{valuei} is the value set on the @var{i}-th control.
If @option{controls} is set to @code{help}, all available controls and
their valid ranges are printed.
@item sample_rate, s
Specify the sample rate, default to 44100. Only used if plugin have
zero inputs.
@item nb_samples, n
Set the number of samples per channel per each output frame, default
is 1024. Only used if plugin have zero inputs.
@item duration, d
Set the minimum 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.
Note that the resulting duration may be greater than the specified duration,
as the generated audio is always cut at the end of a complete frame.
If not specified, or the expressed duration is negative, the audio is
supposed to be generated forever.
Only used if plugin have zero inputs.
@end table
@subsection Examples
@itemize
@item
List all available plugins within amp (LADSPA example plugin) library:
@example
ladspa=file=amp
@end example
@item
List all available controls and their valid ranges for @code{vcf_notch}
plugin from @code{VCF} library:
@example
ladspa=f=vcf:p=vcf_notch:c=help
@end example
@item
Simulate low quality audio equipment using @code{Computer Music Toolkit} (CMT)
plugin library:
@example
ladspa=file=cmt:plugin=lofi:controls=c0=22|c1=12|c2=12
@end example
@item
Add reverberation to the audio using TAP-plugins
(Tom's Audio Processing plugins):
@example
ladspa=file=tap_reverb:tap_reverb
@end example
@item
Generate white noise, with 0.2 amplitude:
@example
ladspa=file=cmt:noise_source_white:c=c0=.2
@end example
@item
Generate 20 bpm clicks using plugin @code{C* Click - Metronome} from the
@code{C* Audio Plugin Suite} (CAPS) library:
@example
ladspa=file=caps:Click:c=c1=20'
@end example
@item
Apply @code{C* Eq10X2 - Stereo 10-band equaliser} effect:
@example
ladspa=caps:Eq10X2:c=c0=-48|c9=-24|c3=12|c4=2
@end example
@item
Increase volume by 20dB using fast lookahead limiter from Steve Harris
@code{SWH Plugins} collection:
@example
ladspa=fast_lookahead_limiter_1913:fastLookaheadLimiter:20|0|2
@end example
@item
Attenuate low frequencies using Multiband EQ from Steve Harris
@code{SWH Plugins} collection:
@example
ladspa=mbeq_1197:mbeq:-24|-24|-24|0|0|0|0|0|0|0|0|0|0|0|0
@end example
@end itemize
@subsection Commands
This filter supports the following commands:
@table @option
@item cN
Modify the @var{N}-th control value.
If the specified value is not valid, it is ignored and prior one is kept.
@end table
@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
@anchor{pan}
@section pan
Mix channels with specific gain levels. The filter accepts the output
channel layout followed by a set of channels definitions.
This filter is also designed to efficiently remap the channels of an audio
stream.
The filter accepts parameters of the form:
"@var{l}|@var{outdef}|@var{outdef}|..."
@table @option
@item l
output channel layout or number of channels
@item outdef
output channel specification, of the form:
"@var{out_name}=[@var{gain}*]@var{in_name}[+[@var{gain}*]@var{in_name}...]"
@item out_name
output channel to define, either a channel name (FL, FR, etc.) or a channel
number (c0, c1, etc.)
@item gain
multiplicative coefficient for the channel, 1 leaving the volume unchanged
@item in_name
input channel to use, see out_name for details; it is not possible to mix
named and numbered input channels
@end table
If the `=' in a channel specification is replaced by `<', then the gains for
that specification will be renormalized so that the total is 1, thus
avoiding clipping noise.
@subsection Mixing examples
For example, if you want to down-mix from stereo to mono, but with a bigger
factor for the left channel:
@example
pan=1c|c0=0.9*c0+0.1*c1
@end example
A customized down-mix to stereo that works automatically for 3-, 4-, 5- and
7-channels surround:
@example
pan=stereo| FL < FL + 0.5*FC + 0.6*BL + 0.6*SL | FR < FR + 0.5*FC + 0.6*BR + 0.6*SR
@end example
Note that @command{ffmpeg} integrates a default down-mix (and up-mix) system
that should be preferred (see "-ac" option) unless you have very specific
needs.
@subsection Remapping examples
The channel remapping will be effective if, and only if:
@itemize
@item gain coefficients are zeroes or ones,
@item only one input per channel output,
@end itemize
If all these conditions are satisfied, the filter will notify the user ("Pure
channel mapping detected"), and use an optimized and lossless method to do the
remapping.
For example, if you have a 5.1 source and want a stereo audio stream by
dropping the extra channels:
@example
pan="stereo| c0=FL | c1=FR"
@end example
Given the same source, you can also switch front left and front right channels
and keep the input channel layout:
@example
pan="5.1| c0=c1 | c1=c0 | c2=c2 | c3=c3 | c4=c4 | c5=c5"
@end example
If the input is a stereo audio stream, you can mute the front left channel (and
still keep the stereo channel layout) with:
@example
pan="stereo|c1=c1"
@end example
Still with a stereo audio stream input, you can copy the right channel in both
front left and right:
@example
pan="stereo| c0=FR | c1=FR"
@end example
@section replaygain
ReplayGain scanner filter. This filter takes an audio stream as an input and
outputs it unchanged.
At end of filtering it displays @code{track_gain} and @code{track_peak}.
@section resample
Convert the audio sample format, sample rate and channel layout. It is
not meant to be used directly.
@section rubberband
Apply time-stretching and pitch-shifting with librubberband.
The filter accepts the following options:
@table @option
@item tempo
Set tempo scale factor.
@item pitch
Set pitch scale factor.
@item transients
Set transients detector.
Possible values are:
@table @var
@item crisp
@item mixed
@item smooth
@end table
@item detector
Set detector.
Possible values are:
@table @var
@item compound
@item percussive
@item soft
@end table
@item phase
Set phase.
Possible values are:
@table @var
@item laminar
@item independent
@end table
@item window
Set processing window size.
Possible values are:
@table @var
@item standard
@item short
@item long
@end table
@item smoothing
Set smoothing.
Possible values are:
@table @var
@item off
@item on
@end table
@item formant
Enable formant preservation when shift pitching.
Possible values are:
@table @var
@item shifted
@item preserved
@end table
@item pitchq
Set pitch quality.
Possible values are:
@table @var
@item quality
@item speed
@item consistency
@end table
@item channels
Set channels.
Possible values are:
@table @var
@item apart
@item together
@end table
@end table
@section sidechaincompress
This filter acts like normal compressor but has the ability to compress
detected signal using second input signal.
It needs two input streams and returns one output stream.
First input stream will be processed depending on second stream signal.
The filtered signal then can be filtered with other filters in later stages of
processing. See @ref{pan} and @ref{amerge} filter.
The filter accepts the following options:
@table @option
@item level_in
Set input gain. Default is 1. Range is between 0.015625 and 64.
@item threshold
If a signal of second stream raises above this level it will affect the gain
reduction of first stream.
By default is 0.125. Range is between 0.00097563 and 1.
@item ratio
Set a ratio about which the signal is reduced. 1:2 means that if the level
raised 4dB above the threshold, it will be only 2dB above after the reduction.
Default is 2. Range is between 1 and 20.
@item attack
Amount of milliseconds the signal has to rise above the threshold before gain
reduction starts. Default is 20. Range is between 0.01 and 2000.
@item release
Amount of milliseconds the signal has to fall below the threshold before
reduction is decreased again. Default is 250. Range is between 0.01 and 9000.
@item makeup
Set the amount by how much signal will be amplified after processing.
Default is 2. Range is from 1 and 64.
@item knee
Curve the sharp knee around the threshold to enter gain reduction more softly.
Default is 2.82843. Range is between 1 and 8.
@item link
Choose if the @code{average} level between all channels of side-chain stream
or the louder(@code{maximum}) channel of side-chain stream affects the
reduction. Default is @code{average}.
@item detection
Should the exact signal be taken in case of @code{peak} or an RMS one in case
of @code{rms}. Default is @code{rms} which is mainly smoother.
@item level_sc
Set sidechain gain. Default is 1. Range is between 0.015625 and 64.
@item mix
How much to use compressed signal in output. Default is 1.
Range is between 0 and 1.
@end table
@subsection Examples
@itemize
@item
Full ffmpeg example taking 2 audio inputs, 1st input to be compressed
depending on the signal of 2nd input and later compressed signal to be
merged with 2nd input:
@example
ffmpeg -i main.flac -i sidechain.flac -filter_complex "[1:a]asplit=2[sc][mix];[0:a][sc]sidechaincompress[compr];[compr][mix]amerge"
@end example
@end itemize
@section sidechaingate
A sidechain gate acts like a normal (wideband) gate but has the ability to
filter the detected signal before sending it to the gain reduction stage.
Normally a gate uses the full range signal to detect a level above the
threshold.
For example: If you cut all lower frequencies from your sidechain signal
the gate will decrease the volume of your track only if not enough highs
appear. With this technique you are able to reduce the resonation of a
natural drum or remove "rumbling" of muted strokes from a heavily distorted
guitar.
It needs two input streams and returns one output stream.
First input stream will be processed depending on second stream signal.
The filter accepts the following options:
@table @option
@item level_in
Set input level before filtering.
Default is 1. Allowed range is from 0.015625 to 64.
@item range
Set the level of gain reduction when the signal is below the threshold.
Default is 0.06125. Allowed range is from 0 to 1.
@item threshold
If a signal rises above this level the gain reduction is released.
Default is 0.125. Allowed range is from 0 to 1.
@item ratio
Set a ratio about which the signal is reduced.
Default is 2. Allowed range is from 1 to 9000.
@item attack
Amount of milliseconds the signal has to rise above the threshold before gain
reduction stops.
Default is 20 milliseconds. Allowed range is from 0.01 to 9000.
@item release
Amount of milliseconds the signal has to fall below the threshold before the
reduction is increased again. Default is 250 milliseconds.
Allowed range is from 0.01 to 9000.
@item makeup
Set amount of amplification of signal after processing.
Default is 1. Allowed range is from 1 to 64.
@item knee
Curve the sharp knee around the threshold to enter gain reduction more softly.
Default is 2.828427125. Allowed range is from 1 to 8.
@item detection
Choose if exact signal should be taken for detection or an RMS like one.
Default is rms. Can be peak or rms.
@item link
Choose if the average level between all channels or the louder channel affects
the reduction.
Default is average. Can be average or maximum.
@item level_sc
Set sidechain gain. Default is 1. Range is from 0.015625 to 64.
@end table
@section silencedetect
Detect silence in an audio stream.
This filter logs a message when it detects that the input audio volume is less
or equal to a noise tolerance value for a duration greater or equal to the
minimum detected noise duration.
The printed times and duration are expressed in seconds.
The filter accepts the following options:
@table @option
@item duration, d
Set silence duration until notification (default is 2 seconds).
@item noise, n
Set noise tolerance. Can be specified in dB (in case "dB" is appended to the
specified value) or amplitude ratio. Default is -60dB, or 0.001.
@end table
@subsection Examples
@itemize
@item
Detect 5 seconds of silence with -50dB noise tolerance:
@example
silencedetect=n=-50dB:d=5
@end example
@item
Complete example with @command{ffmpeg} to detect silence with 0.0001 noise
tolerance in @file{silence.mp3}:
@example
ffmpeg -i silence.mp3 -af silencedetect=noise=0.0001 -f null -
@end example
@end itemize
@section silenceremove
Remove silence from the beginning, middle or end of the audio.
The filter accepts the following options:
@table @option
@item start_periods
This value is used to indicate if audio should be trimmed at beginning of
the audio. A value of zero indicates no silence should be trimmed from the
beginning. When specifying a non-zero value, it trims audio up until it
finds non-silence. Normally, when trimming silence from beginning of audio
the @var{start_periods} will be @code{1} but it can be increased to higher
values to trim all audio up to specific count of non-silence periods.
Default value is @code{0}.
@item start_duration
Specify the amount of time that non-silence must be detected before it stops
trimming audio. By increasing the duration, bursts of noises can be treated
as silence and trimmed off. Default value is @code{0}.
@item start_threshold
This indicates what sample value should be treated as silence. For digital
audio, a value of @code{0} may be fine but for audio recorded from analog,
you may wish to increase the value to account for background noise.
Can be specified in dB (in case "dB" is appended to the specified value)
or amplitude ratio. Default value is @code{0}.
@item stop_periods
Set the count for trimming silence from the end of audio.
To remove silence from the middle of a file, specify a @var{stop_periods}
that is negative. This value is then treated as a positive value and is
used to indicate the effect should restart processing as specified by
@var{start_periods}, making it suitable for removing periods of silence
in the middle of the audio.
Default value is @code{0}.
@item stop_duration
Specify a duration of silence that must exist before audio is not copied any
more. By specifying a higher duration, silence that is wanted can be left in
the audio.
Default value is @code{0}.
@item stop_threshold
This is the same as @option{start_threshold} but for trimming silence from
the end of audio.
Can be specified in dB (in case "dB" is appended to the specified value)
or amplitude ratio. Default value is @code{0}.
@item leave_silence
This indicate that @var{stop_duration} length of audio should be left intact
at the beginning of each period of silence.
For example, if you want to remove long pauses between words but do not want
to remove the pauses completely. Default value is @code{0}.
@item detection
Set how is silence detected. Can be @code{rms} or @code{peak}. Second is faster
and works better with digital silence which is exactly 0.
Default value is @code{rms}.
@item window
Set ratio used to calculate size of window for detecting silence.
Default value is @code{0.02}. Allowed range is from @code{0} to @code{10}.
@end table
@subsection Examples
@itemize
@item
The following example shows how this filter can be used to start a recording
that does not contain the delay at the start which usually occurs between
pressing the record button and the start of the performance:
@example
silenceremove=1:5:0.02
@end example
@item
Trim all silence encountered from begining to end where there is more than 1
second of silence in audio:
@example
silenceremove=0:0:0:-1:1:-90dB
@end example
@end itemize
@section sofalizer
SOFAlizer uses head-related transfer functions (HRTFs) to create virtual
loudspeakers around the user for binaural listening via headphones (audio
formats up to 9 channels supported).
The HRTFs are stored in SOFA files (see @url{http://www.sofacoustics.org/} for a database).
SOFAlizer is developed at the Acoustics Research Institute (ARI) of the
Austrian Academy of Sciences.
To enable compilation of this filter you need to configure FFmpeg with
@code{--enable-netcdf}.
The filter accepts the following options:
@table @option
@item sofa
Set the SOFA file used for rendering.
@item gain
Set gain applied to audio. Value is in dB. Default is 0.
@item rotation
Set rotation of virtual loudspeakers in deg. Default is 0.
@item elevation
Set elevation of virtual speakers in deg. Default is 0.
@item radius
Set distance in meters between loudspeakers and the listener with near-field
HRTFs. Default is 1.
@item type
Set processing type. Can be @var{time} or @var{freq}. @var{time} is
processing audio in time domain which is slow but gives high quality output.
@var{freq} is processing audio in frequency domain which is fast but gives
mediocre output. Default is @var{freq}.
@end table
@section stereotools
This filter has some handy utilities to manage stereo signals, for converting
M/S stereo recordings to L/R signal while having control over the parameters
or spreading the stereo image of master track.
The filter accepts the following options:
@table @option
@item level_in
Set input level before filtering for both channels. Defaults is 1.
Allowed range is from 0.015625 to 64.
@item level_out
Set output level after filtering for both channels. Defaults is 1.
Allowed range is from 0.015625 to 64.
@item balance_in
Set input balance between both channels. Default is 0.
Allowed range is from -1 to 1.
@item balance_out
Set output balance between both channels. Default is 0.
Allowed range is from -1 to 1.
@item softclip
Enable softclipping. Results in analog distortion instead of harsh digital 0dB
clipping. Disabled by default.
@item mutel
Mute the left channel. Disabled by default.
@item muter
Mute the right channel. Disabled by default.
@item phasel
Change the phase of the left channel. Disabled by default.
@item phaser
Change the phase of the right channel. Disabled by default.
@item mode
Set stereo mode. Available values are:
@table @samp
@item lr>lr
Left/Right to Left/Right, this is default.
@item lr>ms
Left/Right to Mid/Side.
@item ms>lr
Mid/Side to Left/Right.
@item lr>ll
Left/Right to Left/Left.
@item lr>rr
Left/Right to Right/Right.
@item lr>l+r
Left/Right to Left + Right.
@item lr>rl
Left/Right to Right/Left.
@end table
@item slev
Set level of side signal. Default is 1.
Allowed range is from 0.015625 to 64.
@item sbal
Set balance of side signal. Default is 0.
Allowed range is from -1 to 1.
@item mlev
Set level of the middle signal. Default is 1.
Allowed range is from 0.015625 to 64.
@item mpan
Set middle signal pan. Default is 0. Allowed range is from -1 to 1.
@item base
Set stereo base between mono and inversed channels. Default is 0.
Allowed range is from -1 to 1.
@item delay
Set delay in milliseconds how much to delay left from right channel and
vice versa. Default is 0. Allowed range is from -20 to 20.
@item sclevel
Set S/C level. Default is 1. Allowed range is from 1 to 100.
@item phase
Set the stereo phase in degrees. Default is 0. Allowed range is from 0 to 360.
@end table
@section stereowiden
This filter enhance the stereo effect by suppressing signal common to both
channels and by delaying the signal of left into right and vice versa,
thereby widening the stereo effect.
The filter accepts the following options:
@table @option
@item delay
Time in milliseconds of the delay of left signal into right and vice versa.
Default is 20 milliseconds.
@item feedback
Amount of gain in delayed signal into right and vice versa. Gives a delay
effect of left signal in right output and vice versa which gives widening
effect. Default is 0.3.
@item crossfeed
Cross feed of left into right with inverted phase. This helps in suppressing
the mono. If the value is 1 it will cancel all the signal common to both
channels. Default is 0.3.
@item drymix
Set level of input signal of original channel. Default is 0.8.
@end table
@section treble
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:
@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 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
@item width, w
Determine how steep is the filter's shelf transition.
@end table
@section tremolo
Sinusoidal amplitude modulation.
The filter accepts the following options:
@table @option
@item f
Modulation frequency in Hertz. Modulation frequencies in the subharmonic range
(20 Hz or lower) will result in a tremolo effect.
This filter may also be used as a ring modulator by specifying
a modulation frequency higher than 20 Hz.
Range is 0.1 - 20000.0. Default value is 5.0 Hz.
@item d
Depth of modulation as a percentage. Range is 0.0 - 1.0.
Default value is 0.5.
@end table
@section vibrato
Sinusoidal phase modulation.
The filter accepts the following options:
@table @option
@item f
Modulation frequency in Hertz.
Range is 0.1 - 20000.0. Default value is 5.0 Hz.
@item d
Depth of modulation as a percentage. Range is 0.0 - 1.0.
Default value is 0.5.
@end table
@section volume
Adjust the input audio volume.
It accepts the following parameters:
@table @option
@item volume
Set audio volume expression.
Output values are clipped to the maximum value.
The output audio volume is given by the relation:
@example
@var{output_volume} = @var{volume} * @var{input_volume}
@end example
The default value for @var{volume} is "1.0".
@item precision
This parameter represents the mathematical precision.
It determines which input sample formats will be allowed, which affects the
precision of the volume scaling.
@table @option
@item fixed
8-bit fixed-point; this limits input sample format to U8, S16, and S32.
@item float
32-bit floating-point; this limits input sample format to FLT. (default)
@item double
64-bit floating-point; this limits input sample format to DBL.
@end table
@item replaygain
Choose the behaviour on encountering ReplayGain side data in input frames.
@table @option
@item drop
Remove ReplayGain side data, ignoring its contents (the default).
@item ignore
Ignore ReplayGain side data, but leave it in the frame.
@item track
Prefer the track gain, if present.
@item album
Prefer the album gain, if present.
@end table
@item replaygain_preamp
Pre-amplification gain in dB to apply to the selected replaygain gain.
Default value for @var{replaygain_preamp} is 0.0.
@item eval
Set when the volume expression is evaluated.
It accepts the following values:
@table @samp
@item once
only evaluate expression once during the filter initialization, or
when the @samp{volume} command is sent
@item frame
evaluate expression for each incoming frame
@end table
Default value is @samp{once}.
@end table
The volume expression can contain the following parameters.
@table @option
@item n
frame number (starting at zero)
@item nb_channels
number of channels
@item nb_consumed_samples
number of samples consumed by the filter
@item nb_samples
number of samples in the current frame
@item pos
original frame position in the file
@item pts
frame PTS
@item sample_rate
sample rate
@item startpts
PTS at start of stream
@item startt
time at start of stream
@item t
frame time
@item tb
timestamp timebase
@item volume
last set volume value
@end table
Note that when @option{eval} is set to @samp{once} only the
@var{sample_rate} and @var{tb} variables are available, all other
variables will evaluate to NAN.
@subsection Commands
This filter supports the following commands:
@table @option
@item volume
Modify the volume expression.
The command accepts the same syntax of the corresponding option.
If the specified expression is not valid, it is kept at its current
value.
@item replaygain_noclip
Prevent clipping by limiting the gain applied.
Default value for @var{replaygain_noclip} is 1.
@end table
@subsection Examples
@itemize
@item
Halve the input audio volume:
@example
volume=volume=0.5
volume=volume=1/2
volume=volume=-6.0206dB
@end example
In all the above example the named key for @option{volume} can be
omitted, for example like in:
@example
volume=0.5
@end example
@item
Increase input audio power by 6 decibels using fixed-point precision:
@example
volume=volume=6dB:precision=fixed
@end example
@item
Fade volume after time 10 with an annihilation period of 5 seconds:
@example
volume='if(lt(t,10),1,max(1-(t-10)/5,0))':eval=frame
@end example
@end itemize
@section volumedetect
Detect the volume of the input video.
The filter has no parameters. The input is not modified. Statistics about
the volume will be printed in the log when the input stream end is reached.
In particular it will show the mean volume (root mean square), maximum
volume (on a per-sample basis), and the beginning of a histogram of the
registered volume values (from the maximum value to a cumulated 1/1000 of
the samples).
All volumes are in decibels relative to the maximum PCM value.
@subsection Examples
Here is an excerpt of the output:
@example
[Parsed_volumedetect_0 @ 0xa23120] mean_volume: -27 dB
[Parsed_volumedetect_0 @ 0xa23120] max_volume: -4 dB
[Parsed_volumedetect_0 @ 0xa23120] histogram_4db: 6
[Parsed_volumedetect_0 @ 0xa23120] histogram_5db: 62
[Parsed_volumedetect_0 @ 0xa23120] histogram_6db: 286
[Parsed_volumedetect_0 @ 0xa23120] histogram_7db: 1042
[Parsed_volumedetect_0 @ 0xa23120] histogram_8db: 2551
[Parsed_volumedetect_0 @ 0xa23120] histogram_9db: 4609
[Parsed_volumedetect_0 @ 0xa23120] histogram_10db: 8409
@end example
It means that:
@itemize
@item
The mean square energy is approximately -27 dB, or 10^-2.7.
@item
The largest sample is at -4 dB, or more precisely between -4 dB and -5 dB.
@item
There are 6 samples at -4 dB, 62 at -5 dB, 286 at -6 dB, etc.
@end itemize
In other words, raising the volume by +4 dB does not cause any clipping,
raising it by +5 dB causes clipping for 6 samples, etc.
@c man end AUDIO FILTERS
@chapter Audio Sources
@c man begin AUDIO SOURCES
Below is a description of the currently available audio sources.
@section abuffer
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}.
It accepts the following parameters:
@table @option
@item time_base
The timebase which will be used for timestamps of submitted frames. It must be
either a floating-point number or in @var{numerator}/@var{denominator} form.
@item sample_rate
The sample rate of the incoming audio buffers.
@item sample_fmt
The sample format of the incoming audio buffers.
Either a sample format name or its corresponding integer representation from
the enum AVSampleFormat in @file{libavutil/samplefmt.h}
@item channel_layout
The channel layout of the incoming audio buffers.
Either a channel layout name from channel_layout_map in
@file{libavutil/channel_layout.c} or its corresponding integer representation
from the AV_CH_LAYOUT_* macros in @file{libavutil/channel_layout.h}
@item channels
The number of channels of the incoming audio buffers.
If both @var{channels} and @var{channel_layout} are specified, then they
must be consistent.
@end table
@subsection Examples
@example
abuffer=sample_rate=44100:sample_fmt=s16p:channel_layout=stereo
@end example
will instruct the source to accept planar 16bit signed stereo at 44100Hz.
Since the sample format with name "s16p" corresponds to the number
6 and the "stereo" channel layout corresponds to the value 0x3, this is
equivalent to:
@example
abuffer=sample_rate=44100:sample_fmt=6:channel_layout=0x3
@end example
@section aevalsrc
Generate an audio signal specified by an expression.
This source accepts in input one or more expressions (one for each
channel), which are evaluated and used to generate a corresponding
audio signal.
This source accepts the following options:
@table @option
@item exprs
Set the '|'-separated expressions list for each separate channel. In case the
@option{channel_layout} option is not specified, the selected channel layout
depends on the number of provided expressions. Otherwise the last
specified expression is applied to the remaining output channels.
@item channel_layout, c
Set the channel layout. The number of channels in the specified layout
must be equal to the number of specified expressions.
@item duration, d
Set the minimum 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.
Note that the resulting duration may be greater than the specified
duration, as the generated audio is always cut at the end of a
complete frame.
If not specified, or the expressed duration is negative, the audio is
supposed to be generated forever.
@item nb_samples, n
Set the number of samples per channel per each output frame,
default to 1024.
@item sample_rate, s
Specify the sample rate, default to 44100.
@end table
Each expression in @var{exprs} can contain the following constants:
@table @option
@item n
number of the evaluated sample, starting from 0
@item t
time of the evaluated sample expressed in seconds, starting from 0
@item s
sample rate
@end table
@subsection Examples
@itemize
@item
Generate silence:
@example
aevalsrc=0
@end example
@item
Generate a sin signal with frequency of 440 Hz, set sample rate to
8000 Hz:
@example
aevalsrc="sin(440*2*PI*t):s=8000"
@end example
@item
Generate a two channels signal, specify the channel layout (Front
Center + Back Center) explicitly:
@example
aevalsrc="sin(420*2*PI*t)|cos(430*2*PI*t):c=FC|BC"
@end example
@item
Generate white noise:
@example
aevalsrc="-2+random(0)"
@end example
@item
Generate an amplitude modulated signal:
@example
aevalsrc="sin(10*2*PI*t)*sin(880*2*PI*t)"
@end example
@item
Generate 2.5 Hz binaural beats on a 360 Hz carrier:
@example
aevalsrc="0.1*sin(2*PI*(360-2.5/2)*t) | 0.1*sin(2*PI*(360+2.5/2)*t)"
@end example
@end itemize
@section anullsrc
The null audio source, return unprocessed audio frames. It is mainly useful
as a template and to be employed in analysis / debugging tools, or as
the source for filters which ignore the input data (for example the sox
synth filter).
This source accepts the following options:
@table @option
@item channel_layout, cl
Specifies the channel layout, and can be either an integer or a string
representing a channel layout. The default value of @var{channel_layout}
is "stereo".
Check the channel_layout_map definition in
@file{libavutil/channel_layout.c} for the mapping between strings and
channel layout values.
@item sample_rate, r
Specifies the sample rate, and defaults to 44100.
@item nb_samples, n
Set the number of samples per requested frames.
@end table
@subsection Examples
@itemize
@item
Set the sample rate to 48000 Hz and the channel layout to AV_CH_LAYOUT_MONO.
@example
anullsrc=r=48000:cl=4
@end example
@item
Do the same operation with a more obvious syntax:
@example
anullsrc=r=48000:cl=mono
@end example
@end itemize
All the parameters need to be explicitly defined.
@section flite
Synthesize a voice utterance using the libflite library.
To enable compilation of this filter you need to configure FFmpeg with
@code{--enable-libflite}.
Note that the flite library is not thread-safe.
The filter accepts the following options:
@table @option
@item list_voices
If set to 1, list the names of the available voices and exit
immediately. Default value is 0.
@item nb_samples, n
Set the maximum number of samples per frame. Default value is 512.
@item textfile
Set the filename containing the text to speak.
@item text
Set the text to speak.
@item voice, v
Set the voice to use for the speech synthesis. Default value is
@code{kal}. See also the @var{list_voices} option.
@end table
@subsection Examples
@itemize
@item
Read from file @file{speech.txt}, and synthesize the text using the
standard flite voice:
@example
flite=textfile=speech.txt
@end example
@item
Read the specified text selecting the @code{slt} voice:
@example
flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt
@end example
@item
Input text to ffmpeg:
@example
ffmpeg -f lavfi -i flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt
@end example
@item
Make @file{ffplay} speak the specified text, using @code{flite} and
the @code{lavfi} device:
@example
ffplay -f lavfi flite=text='No more be grieved for which that thou hast done.'
@end example
@end itemize
For more information about libflite, check:
@url{http://www.speech.cs.cmu.edu/flite/}
@section anoisesrc
Generate a noise audio signal.
The filter accepts the following options:
@table @option
@item sample_rate, r
Specify the sample rate. Default value is 48000 Hz.
@item amplitude, a
Specify the amplitude (0.0 - 1.0) of the generated audio stream. Default value
is 1.0.
@item duration, d
Specify the duration of the generated audio stream. Not specifying this option
results in noise with an infinite length.
@item color, colour, c
Specify the color of noise. Available noise colors are white, pink, and brown.
Default color is white.
@item seed, s
Specify a value used to seed the PRNG.
@item nb_samples, n
Set the number of samples per each output frame, default is 1024.
@end table
@subsection Examples
@itemize
@item
Generate 60 seconds of pink noise, with a 44.1 kHz sampling rate and an amplitude of 0.5:
@example
anoisesrc=d=60:c=pink:r=44100:a=0.5
@end example
@end itemize
@section sine
Generate an audio signal made of a sine wave with amplitude 1/8.
The audio signal is bit-exact.
The filter accepts the following options:
@table @option
@item frequency, f
Set the carrier frequency. Default is 440 Hz.
@item beep_factor, b
Enable a periodic beep every second with frequency @var{beep_factor} times
the carrier frequency. Default is 0, meaning the beep is disabled.
@item sample_rate, r
Specify the sample rate, default is 44100.
@item duration, d
Specify the duration of the generated audio stream.
@item samples_per_frame
Set the number of samples per output frame.
The expression can contain the following constants:
@table @option
@item n
The (sequential) number of the output audio frame, starting from 0.
@item pts
The PTS (Presentation TimeStamp) of the output audio frame,
expressed in @var{TB} units.
@item t
The PTS of the output audio frame, expressed in seconds.
@item TB
The timebase of the output audio frames.
@end table
Default is @code{1024}.
@end table
@subsection Examples
@itemize
@item
Generate a simple 440 Hz sine wave:
@example
sine
@end example
@item
Generate a 220 Hz sine wave with a 880 Hz beep each second, for 5 seconds:
@example
sine=220:4:d=5
sine=f=220:b=4:d=5
sine=frequency=220:beep_factor=4:duration=5
@end example
@item
Generate a 1 kHz sine wave following @code{1602,1601,1602,1601,1602} NTSC
pattern:
@example
sine=1000:samples_per_frame='st(0,mod(n,5)); 1602-not(not(eq(ld(0),1)+eq(ld(0),3)))'
@end example
@end itemize
@c man end AUDIO SOURCES
@chapter Audio Sinks
@c man begin AUDIO SINKS
Below is a description of the currently available audio sinks.
@section abuffersink
Buffer audio frames, and make them available to the end of filter chain.
This sink is mainly intended for programmatic use, in particular
through the interface defined in @file{libavfilter/buffersink.h}
or the options system.
It accepts a pointer to an AVABufferSinkContext structure, which
defines the incoming buffers' formats, to be passed as the opaque
parameter to @code{avfilter_init_filter} for initialization.
@section anullsink
Null audio sink; do absolutely nothing with the input audio. It is
mainly useful as a template and for use in analysis / debugging
tools.
@c man end AUDIO SINKS
@chapter Video Filters
@c man begin VIDEO FILTERS
When you configure your FFmpeg build, you can disable any of the
existing filters using @code{--disable-filters}.
The configure output will show the video filters included in your
build.
Below is a description of the currently available video filters.
@section alphaextract
Extract the alpha component from the input as a grayscale video. This
is especially useful with the @var{alphamerge} filter.
@section alphamerge
Add or replace the alpha component of the primary input with the
grayscale value of a second input. This is intended for use with
@var{alphaextract} to allow the transmission or storage of frame
sequences that have alpha in a format that doesn't support an alpha
channel.
For example, to reconstruct full frames from a normal YUV-encoded video
and a separate video created with @var{alphaextract}, you might use:
@example
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 ass
Same as the @ref{subtitles} filter, except that it doesn't require libavcodec
and libavformat to work. On the other hand, it is limited to ASS (Advanced
Substation Alpha) subtitles files.
This filter accepts the following option in addition to the common options from
the @ref{subtitles} filter:
@table @option
@item shaping
Set the shaping engine
Available values are:
@table @samp
@item auto
The default libass shaping engine, which is the best available.
@item simple
Fast, font-agnostic shaper that can do only substitutions
@item complex
Slower shaper using OpenType for substitutions and positioning
@end table
The default is @code{auto}.
@end table
@section atadenoise
Apply an Adaptive Temporal Averaging Denoiser to the video input.
The filter accepts the following options:
@table @option
@item 0a
Set threshold A for 1st plane. Default is 0.02.
Valid range is 0 to 0.3.
@item 0b
Set threshold B for 1st plane. Default is 0.04.
Valid range is 0 to 5.
@item 1a
Set threshold A for 2nd plane. Default is 0.02.
Valid range is 0 to 0.3.
@item 1b
Set threshold B for 2nd plane. Default is 0.04.
Valid range is 0 to 5.
@item 2a
Set threshold A for 3rd plane. Default is 0.02.
Valid range is 0 to 0.3.
@item 2b
Set threshold B for 3rd plane. Default is 0.04.
Valid range is 0 to 5.
Threshold A is designed to react on abrupt changes in the input signal and
threshold B is designed to react on continuous changes in the input signal.
@item s
Set number of frames filter will use for averaging. Default is 33. Must be odd
number in range [5, 129].
@end table
@section bbox
Compute the bounding box for the non-black pixels in the input frame
luminance plane.
This filter computes the bounding box containing all the pixels with a
luminance value greater than the minimum allowed value.
The parameters describing the bounding box are printed on the filter
log.
The filter accepts the following option:
@table @option
@item min_val
Set the minimal luminance value. Default is @code{16}.
@end table
@section blackdetect
Detect video intervals that are (almost) completely black. Can be
useful to detect chapter transitions, commercials, or invalid
recordings. Output lines contains the time for the start, end and
duration of the detected black interval expressed in seconds.
In order to display the output lines, you need to set the loglevel at
least to the AV_LOG_INFO value.
The filter accepts the following options:
@table @option
@item black_min_duration, d
Set the minimum detected black duration expressed in seconds. It must
be a non-negative floating point number.
Default value is 2.0.
@item picture_black_ratio_th, pic_th
Set the threshold for considering a picture "black".
Express the minimum value for the ratio:
@example
@var{nb_black_pixels} / @var{nb_pixels}
@end example
for which a picture is considered black.
Default value is 0.98.
@item pixel_black_th, pix_th
Set the threshold for considering a pixel "black".
The threshold expresses the maximum pixel luminance value for which a
pixel is considered "black". The provided value is scaled according to
the following equation:
@example
@var{absolute_threshold} = @var{luminance_minimum_value} + @var{pixel_black_th} * @var{luminance_range_size}
@end example
@var{luminance_range_size} and @var{luminance_minimum_value} depend on
the input video format, the range is [0-255] for YUV full-range
formats and [16-235] for YUV non full-range formats.
Default value is 0.10.
@end table
The following example sets the maximum pixel threshold to the minimum
value, and detects only black intervals of 2 or more seconds:
@example
blackdetect=d=2:pix_th=0.00
@end example
@section blackframe
Detect frames that are (almost) completely black. Can be useful to
detect chapter transitions or commercials. Output lines consist of
the frame number of the detected frame, the percentage of blackness,
the position in the file if known or -1 and the timestamp in seconds.
In order to display the output lines, you need to set the loglevel at
least to the AV_LOG_INFO value.
It accepts the following parameters:
@table @option
@item amount
The percentage of the pixels that have to be below the threshold; it defaults to
@code{98}.
@item threshold, thresh
The threshold below which a pixel value is considered black; it defaults to
@code{32}.
@end table
@section blend, tblend
Blend two video frames into each other.
The @code{blend} filter takes two input streams and outputs one
stream, the first input is the "top" layer and second input is
"bottom" layer. Output terminates when shortest input terminates.
The @code{tblend} (time blend) filter takes two consecutive frames
from one single stream, and outputs the result obtained by blending
the new frame on top of the old frame.
A description of the accepted options follows.
@table @option
@item c0_mode
@item c1_mode
@item c2_mode
@item c3_mode
@item all_mode
Set blend mode for specific pixel component or all pixel components in case
of @var{all_mode}. Default value is @code{normal}.
Available values for component modes are:
@table @samp
@item addition
@item addition128
@item and
@item average
@item burn
@item darken
@item difference
@item difference128
@item divide
@item dodge
@item exclusion
@item glow
@item hardlight
@item hardmix
@item lighten
@item linearlight
@item multiply
@item multiply128
@item negation
@item normal
@item or
@item overlay
@item phoenix
@item pinlight
@item reflect
@item screen
@item softlight
@item subtract
@item vividlight
@item xor
@end table
@item c0_opacity
@item c1_opacity
@item c2_opacity
@item c3_opacity
@item all_opacity
Set blend opacity for specific pixel component or all pixel components in case
of @var{all_opacity}. Only used in combination with pixel component blend modes.
@item c0_expr
@item c1_expr
@item c2_expr
@item c3_expr
@item all_expr
Set blend expression for specific pixel component or all pixel components in case
of @var{all_expr}. Note that related mode options will be ignored if those are set.
The expressions can use the following variables:
@table @option
@item N
The sequential number of the filtered frame, starting from @code{0}.
@item X
@item Y
the coordinates of the current sample
@item W
@item H
the width and height of currently filtered plane
@item SW
@item SH
Width and height scale depending on the currently filtered plane. It is the
ratio between the corresponding luma plane number of pixels and the current
plane ones. E.g. for YUV4:2:0 the values are @code{1,1} for the luma plane, and
@code{0.5,0.5} for chroma planes.
@item T
Time of the current frame, expressed in seconds.
@item TOP, A
Value of pixel component at current location for first video frame (top layer).
@item BOTTOM, B
Value of pixel component at current location for second video frame (bottom layer).
@end table
@item shortest
Force termination when the shortest input terminates. Default is
@code{0}. This option is only defined for the @code{blend} filter.
@item repeatlast
Continue applying the last bottom frame after the end of the stream. A value of
@code{0} disable the filter after the last frame of the bottom layer is reached.
Default is @code{1}. This option is only defined for the @code{blend} filter.
@end table
@subsection Examples
@itemize
@item
Apply transition from bottom layer to top layer in first 10 seconds:
@example
blend=all_expr='A*(if(gte(T,10),1,T/10))+B*(1-(if(gte(T,10),1,T/10)))'
@end example
@item
Apply 1x1 checkerboard effect:
@example
blend=all_expr='if(eq(mod(X,2),mod(Y,2)),A,B)'
@end example
@item
Apply uncover left effect:
@example
blend=all_expr='if(gte(N*SW+X,W),A,B)'
@end example
@item
Apply uncover down effect:
@example
blend=all_expr='if(gte(Y-N*SH,0),A,B)'
@end example
@item
Apply uncover up-left effect:
@example
blend=all_expr='if(gte(T*SH*40+Y,H)*gte((T*40*SW+X)*W/H,W),A,B)'
@end example
@item
Split diagonally video and shows top and bottom layer on each side:
@example
blend=all_expr=if(gt(X,Y*(W/H)),A,B)
@end example
@item
Display differences between the current and the previous frame:
@example
tblend=all_mode=difference128
@end example
@end itemize
@section boxblur
Apply a boxblur algorithm to the input video.
It accepts the following parameters:
@table @option
@item luma_radius, lr
@item luma_power, lp
@item chroma_radius, cr
@item chroma_power, cp
@item alpha_radius, ar
@item alpha_power, ap
@end table
A description of the accepted options follows.
@table @option
@item luma_radius, lr
@item chroma_radius, cr
@item alpha_radius, ar
Set an expression for the box radius in pixels used for blurring the
corresponding input plane.
The radius value must be a non-negative number, and must not be
greater than the value of the expression @code{min(w,h)/2} for the
luma and alpha planes, and of @code{min(cw,ch)/2} for the chroma
planes.
Default value for @option{luma_radius} is "2". If not specified,
@option{chroma_radius} and @option{alpha_radius} default to the
corresponding value set for @option{luma_radius}.
The expressions can contain the following constants:
@table @option
@item w
@item h
The input width and height in pixels.
@item cw
@item ch
The input chroma image width and height in pixels.
@item hsub
@item vsub
The horizontal and vertical chroma subsample values. For example, for the
pixel format "yuv422p", @var{hsub} is 2 and @var{vsub} is 1.
@end table
@item luma_power, lp
@item chroma_power, cp
@item alpha_power, ap
Specify how many times the boxblur filter is applied to the
corresponding plane.
Default value for @option{luma_power} is 2. If not specified,
@option{chroma_power} and @option{alpha_power} default to the
corresponding value set for @option{luma_power}.
A value of 0 will disable the effect.
@end table
@subsection Examples
@itemize
@item
Apply a boxblur filter with the luma, chroma, and alpha radii
set to 2:
@example
boxblur=luma_radius=2:luma_power=1
boxblur=2:1
@end example
@item
Set the luma radius to 2, and alpha and chroma radius to 0:
@example
boxblur=2:1:cr=0:ar=0
@end example
@item
Set the luma and chroma radii to a fraction of the video dimension:
@example
boxblur=luma_radius=min(h\,w)/10:luma_power=1:chroma_radius=min(cw\,ch)/10:chroma_power=1
@end example
@end itemize
@section chromakey
YUV colorspace color/chroma keying.
The filter accepts the following options:
@table @option
@item color
The color which will be replaced with transparency.
@item similarity
Similarity percentage with the key color.
0.01 matches only the exact key color, while 1.0 matches everything.
@item blend
Blend percentage.
0.0 makes pixels either fully transparent, or not transparent at all.
Higher values result in semi-transparent pixels, with a higher transparency
the more similar the pixels color is to the key color.
@item yuv
Signals that the color passed is already in YUV instead of RGB.
Litteral colors like "green" or "red" don't make sense with this enabled anymore.
This can be used to pass exact YUV values as hexadecimal numbers.
@end table
@subsection Examples
@itemize
@item
Make every green pixel in the input image transparent:
@example
ffmpeg -i input.png -vf chromakey=green out.png
@end example
@item
Overlay a greenscreen-video on top of a static black background.
@example
ffmpeg -f lavfi -i color=c=black:s=1280x720 -i video.mp4 -shortest -filter_complex "[1:v]chromakey=0x70de77:0.1:0.2[ckout];[0:v][ckout]overlay[out]" -map "[out]" output.mkv
@end example
@end itemize
@section codecview
Visualize information exported by some codecs.
Some codecs can export information through frames using side-data or other
means. For example, some MPEG based codecs export motion vectors through the
@var{export_mvs} flag in the codec @option{flags2} option.
The filter accepts the following option:
@table @option
@item mv
Set motion vectors to visualize.
Available flags for @var{mv} are:
@table @samp
@item pf
forward predicted MVs of P-frames
@item bf
forward predicted MVs of B-frames
@item bb
backward predicted MVs of B-frames
@end table
@item qp
Display quantization parameters using the chroma planes
@end table
@subsection Examples
@itemize
@item
Visualizes multi-directionals MVs from P and B-Frames using @command{ffplay}:
@example
ffplay -flags2 +export_mvs input.mpg -vf codecview=mv=pf+bf+bb
@end example