DESCRIPTION

This document describes filters, sources, and sinks provided by the libavfilter library.

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.

                        [main]
        input --> split ---------------------> overlay --> output
                    |                             ^
                    |[tmp]                  [flip]|
                    +-----> crop --> vflip -------+

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:

        ffmpeg -i INPUT -vf "split [main][tmp]; [tmp] crop=iw:ih/2:0:0, vflip [flip]; [main][flip] overlay=0:H/2" OUTPUT

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, crop,vflip are in one linear chain, split and 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 [main] and [tmp].

The stream sent to the second output of split, labelled as [tmp], is processed through the crop filter, which crops away the lower half part of the video, and then vertically flipped. The overlay filter takes in input the first unchanged output of the split filter (which was labelled as [main]), and overlay on its lower half the output generated by the 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 source filters that do not have an audio/video input, and sink filters that will not have audio/video output.

GRAPH

The graph2dot program included in the FFmpeg tools directory can be used to parse a filtergraph description and issue a corresponding textual representation in the dot language.

Invoke the command:

        graph2dot -h

to see how to use graph2dot.

You can then pass the dot description to the dot program (from the graphviz suite of programs) and obtain a graphical representation of the filtergraph.

For example the sequence of commands:

        echo <GRAPH_DESCRIPTION> | \
        tools/graph2dot -o graph.tmp && \
        dot -Tpng graph.tmp -o graph.png && \
        display graph.png

can be used to create and display an image representing the graph described by the 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:

        ffmpeg -i infile -vf scale=640:360 outfile

your GRAPH_DESCRIPTION string will need to be of the form:

        nullsrc,scale=640:360,nullsink

you may also need to set the nullsrc parameters and add a format filter in order to simulate a specific input file.

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''.

Filtergraph syntax

A filtergraph has a textual representation, which is recognized by the -filter/-vf/-af and -filter_complex options in ffmpeg and -vf/-af in ffplay, and by the "avfilter_graph_parse_ptr()" function defined in 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: [in_link_1]...[in_link_N]filter_name=arguments[out_link_1]...[out_link_M]

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 "=arguments".

arguments is a string which contains the parameters used to initialize the filter instance. It may have one of two forms:

• A ':'-separated list of key=value pairs.
• A ':'-separated list of value. In this case, the keys are assumed to be the option names in the order they are declared. E.g. the "fade" filter declares three options in this order --- type, start_frame and nb_frames. Then the parameter list in:0:30 means that the value in is assigned to the option type, 0 to start_frame and 30 to nb_frames.
• A ':'-separated list of mixed direct value and long key=value pairs. The direct value must precede the key=value pairs, and follow the same constraints order of the previous point. The following key=value pairs can be set in any preferred order.

If the option value itself is a list of items (e.g. the "format" filter takes a list of pixel formats), the items in the list are usually separated by |.

The list of arguments can be quoted using the character ' as initial and ending mark, and the character \ for escaping the characters within the quoted text; otherwise the argument string is considered terminated when the next special character (belonging to the set []=;,) 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 in_link_1 ... in_link_N, are associated to the filter input pads, the following labels out_link_1 ... 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

        nullsrc, split[L1], [L2]overlay, nullsink

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 scale filters where format conversion is required. It is possible to specify swscale flags for those automatically inserted scalers by prepending "sws_flags=flags;" to the filtergraph description.

Here is a BNF description of the filtergraph syntax:

        <NAME>             ::= sequence of alphanumeric characters and '_'
        <LINKLABEL>        ::= "[" <NAME> "]"
        <LINKLABELS>       ::= <LINKLABEL> [<LINKLABELS>]
        <FILTER_ARGUMENTS> ::= sequence of chars (possibly quoted)
        <FILTER>           ::= [<LINKLABELS>] <NAME> ["=" <FILTER_ARGUMENTS>] [<LINKLABELS>]
        <FILTERCHAIN>      ::= <FILTER> [,<FILTERCHAIN>]
        <FILTERGRAPH>      ::= [sws_flags=<flags>;] <FILTERCHAIN> [;<FILTERGRAPH>]

Notes on filtergraph escaping

Filtergraph description composition entails several levels of escaping. See the ``Quoting and escaping'' section in the ffmpeg-utils(1) manual 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 ":" used to separate values, or one of the escaping characters "\'".

A second level escaping affects the whole filter description, which may contain the escaping characters "\'" or the special characters "[],;" 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 drawtext filter description text value:

        this is a 'string': may contain one, or more, special characters

This string contains the "'" special escaping character, and the ":" special character, so it needs to be escaped in this way:

        text=this is a \'string\'\: may contain one, or more, special characters

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:

        drawtext=text=this is a \\\'string\\\'\\: may contain one\, or more\, special characters

(note that in addition to the "\'" escaping special characters, also "," 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 "\" is special and needs to be escaped with another "\", the previous string will finally result in:

        -vf "drawtext=text=this is a \\\\\\'string\\\\\\'\\\\: may contain one\\, or more\\, special characters"

TIMELINE EDITING

Some filters support a generic 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:

t

timestamp expressed in seconds, NAN if the input timestamp is unknown

n

sequential number of the input frame, starting from 0

pos

the position in the file of the input frame, NAN if unknown

w

h

width and height of the input frame if video

Additionally, these filters support an enable command that can be used to re-define the expression.

Like any other filtering option, the enable option follows the same rules.

For example, to enable a blur filter (smartblur) from 10 seconds to 3 minutes, and a curves filter starting at 3 seconds:

        smartblur = enable='between(t,10,3*60)',
        curves    = enable='gte(t,3)' : preset=cross_process

AUDIO FILTERS

When you configure your FFmpeg build, you can disable any of the existing filters using "--disable-filters". The configure output will show the audio filters included in your build.

Below is a description of the currently available audio filters.

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 "threshold" and dividing it by the factor set with "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''. "attack" determines how long the signal has to rise above the threshold before any reduction will occur and "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 "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 "knee" flattens the hard edge at the threshold in the range of the chosen decibels.

The filter accepts the following options:

level_in

Set input gain. Default is 1. Range is between 0.015625 and 64.

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.

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.

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.

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.

makeup

Set the amount by how much signal will be amplified after processing. Default is 2. Range is from 1 and 64.

knee

Curve the sharp knee around the threshold to enter gain reduction more softly. Default is 2.82843. Range is between 1 and 8.

link

Choose if the "average" level between all channels of input stream or the louder("maximum") channel of input stream affects the reduction. Default is "average".

detection

Should the exact signal be taken in case of "peak" or an RMS one in case of "rms". Default is "rms" which is mostly smoother.

mix

How much to use compressed signal in output. Default is 1. Range is between 0 and 1.

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:

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.

duration, d

Specify the duration of the cross fade effect. See the Time duration section in the ffmpeg-utils(1) manual for the accepted syntax. By default the duration is determined by nb_samples. If set this option is used instead of nb_samples.

overlap, o

Should first stream end overlap with second stream start. Default is enabled.

curve1

Set curve for cross fade transition for first stream.

curve2

Set curve for cross fade transition for second stream.

For description of available curve types see afade filter description.

Examples

• Cross fade from one input to another:

          ffmpeg -i first.flac -i second.flac -filter_complex acrossfade=d=10:c1=exp:c2=exp output.flac
  

• Cross fade from one input to another but without overlapping:

          ffmpeg -i first.flac -i second.flac -filter_complex acrossfade=d=10:o=0:c1=exp:c2=exp output.flac
  

adelay

Delay one or more audio channels.

Samples in delayed channel are filled with silence.

The filter accepts the following option:

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.

Examples

• 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.

          adelay=1500|0|500
  

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 "delay", and the loudness of the reflected signal is the "decay". Multiple echoes can have different delays and decays.

A description of the accepted parameters follows.

in_gain

Set input gain of reflected signal. Default is 0.6.

out_gain

Set output gain of reflected signal. Default is 0.3.

delays

Set list of time intervals in milliseconds between original signal and reflections separated by '|'. Allowed range for each "delay" is "(0 - 90000.0]". Default is 1000.

decays

Set list of loudnesses of reflected signals separated by '|'. Allowed range for each "decay" is "(0 - 1.0]". Default is 0.5.

Examples

• Make it sound as if there are twice as many instruments as are actually playing:

          aecho=0.8:0.88:60:0.4
  

• If delay is very short, then it sound like a (metallic) robot playing music:

          aecho=0.8:0.88:6:0.4
  

• A longer delay will sound like an open air concert in the mountains:

          aecho=0.8:0.9:1000:0.3
  

• Same as above but with one more mountain:

          aecho=0.8:0.9:1000|1800:0.3|0.25
  

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:

level_in

Set input gain.

level_out

Set output gain.

mode

Set filter mode. For restoring material use "reproduction" mode, otherwise use "production" mode. Default is "reproduction" mode.

type

Set filter type. Selects medium. Can be one of the following:

col

select Columbia.

emi

select EMI.

bsi

select BSI (78RPM).

riaa

select RIAA.

cd

select Compact Disc (CD).

50fm

select 50Xs (FM).

75fm

select 75Xs (FM).

50kf

select 50Xs (FM-KF).

75kf

select 75Xs (FM-KF).

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:

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.

channel_layout, c

Set output channel layout. If not specified, the channel layout is specified by the number of expressions. If set to same, it will use by default the same input channel layout.

Each expression in exprs can contain the following constants and functions:

ch

channel number of the current expression

n

number of the evaluated sample, starting from 0

s

sample rate

t

time of the evaluated sample expressed in seconds

nb_in_channels

nb_out_channels

input and output number of channels

val(CH)

the value of input channel with number CH

Note: this filter is slow. For faster processing you should use a dedicated filter.

Examples

• Half volume:

          aeval=val(ch)/2:c=same
  

• Invert phase of the second channel:

          aeval=val(0)|-val(1)
  

afade

Apply fade-in/out effect to input audio.

A description of the accepted parameters follows.

type, t

Specify the effect type, can be either "in" for fade-in, or "out" for a fade-out effect. Default is "in".

start_sample, ss

Specify the number of the start sample for starting to apply the fade effect. Default is 0.

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.

start_time, st

Specify the start time of the fade effect. Default is 0. The value must be specified as a time duration; see the Time duration section in the ffmpeg-utils(1) manual for the accepted syntax. If set this option is used instead of start_sample.

duration, d

Specify the duration of the fade effect. See the Time duration section in the ffmpeg-utils(1) manual 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 nb_samples. If set this option is used instead of nb_samples.

curve

Set curve for fade transition.

It accepts the following values:

tri

select triangular, linear slope (default)

qsin

select quarter of sine wave

hsin

select half of sine wave

esin

select exponential sine wave

log

select logarithmic

ipar

select inverted parabola

qua

select quadratic

cub

select cubic

squ

select square root

cbr

select cubic root

par

select parabola

exp

select exponential

iqsin

select inverted quarter of sine wave

ihsin

select inverted half of sine wave

dese

select double-exponential seat

desi

select double-exponential sigmoid

Examples

• Fade in first 15 seconds of audio:

          afade=t=in:ss=0:d=15
  

• Fade out last 25 seconds of a 900 seconds audio:

          afade=t=out:st=875:d=25
  

afftfilt

Apply arbitrary expressions to samples in frequency domain.

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.

imag

Set frequency domain imaginary expression for each separate channel separated by '|'. If not set, real option is used.

Each expression in real and imag can contain the following constants:

sr

sample rate

b

current frequency bin number

nb

number of available bins

ch

channel number of the current expression

chs

number of channels

pts

current frame pts

win_size

Set window size.

It accepts the following values:

w16

w32

w64

w128

w256

w512

w1024

w2048

w4096

w8192

w16384

w32768

w65536

Default is "w4096"

win_func

Set window function. Default is "hann".

overlap

Set window overlap. If set to 1, the recommended overlap for selected window function will be picked. Default is 0.75.

Examples

• Leave almost only low frequencies in audio:

          afftfilt="1-clip((b/nb)*b,0,1)"
  

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:

sample_fmts

A '|'-separated list of requested sample formats.

sample_rates

A '|'-separated list of requested sample rates.

channel_layouts

A '|'-separated list of requested channel layouts.

See the Channel Layout section in the ffmpeg-utils(1) manual for the required syntax.

If a parameter is omitted, all values are allowed.

Force the output to either unsigned 8-bit or signed 16-bit stereo

        aformat=sample_fmts=u8|s16:channel_layouts=stereo

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 threshold and divide it by the factor set with ratio. The bottom of the noise floor is set via 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 attack and release.

attack determines how long the signal has to fall below the threshold before any reduction will occur and 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.

level_in

Set input level before filtering. Default is 1. Allowed range is from 0.015625 to 64.

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.

threshold

If a signal rises above this level the gain reduction is released. Default is 0.125. Allowed range is from 0 to 1.

ratio

Set a ratio about which the signal is reduced. Default is 2. Allowed range is from 1 to 9000.

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.

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.

makeup

Set amount of amplification of signal after processing. Default is 1. Allowed range is from 1 to 64.

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.

detection

Choose if exact signal should be taken for detection or an RMS like one. Default is rms. Can be peak or rms.

link

Choose if the average level between all channels or the louder channel affects the reduction. Default is average. Can be average or maximum.

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:

level_in

Set input gain. Default is 1.

level_out

Set output gain. Default is 1.

limit

Don't let signals above this level pass the limiter. Default is 1.

attack

The limiter will reach its attenuation level in this amount of time in milliseconds. Default is 5 milliseconds.

release

Come back from limiting to attenuation 1.0 in this amount of milliseconds. Default is 50 milliseconds.

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.

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.

level

Auto level output signal. Default is enabled. This normalizes audio back to 0dB if enabled.

Depending on picked setting it is recommended to upsample input 2x or 4x times with aresample before applying this filter.

allpass

Apply a two-pole all-pass filter with central frequency (in Hz) frequency, and filter-width 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:

frequency, f

Set frequency in Hz.

width_type

Set method to specify band-width of filter.

h

Hz

q

Q-Factor

o

octave

s

slope

width, w

Specify the band-width of a filter in width_type units.

amerge

Merge two or more audio streams into a single multi-channel stream.

The filter accepts the following options:

inputs

Set the number of inputs. Default is 2.

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.

Examples

• Merge two mono files into a stereo stream:

          amovie=left.wav [l] ; amovie=right.mp3 [r] ; [l] [r] amerge
  

• Multiple merges assuming 1 video stream and 6 audio streams in input.mkv:

          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
  

amix

Mixes multiple audio inputs into a single output.

Note that this filter only supports float samples (the amerge and pan audio filters support many formats). If the amix input has integer samples then aresample will be automatically inserted to perform the conversion to float samples.

For example

        ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex amix=inputs=3:duration=first:dropout_transition=3 OUTPUT

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:

inputs

The number of inputs. If unspecified, it defaults to 2.

duration

How to determine the end-of-stream.

longest

The duration of the longest input. (default)

shortest

The duration of the shortest input.

first

The duration of the first input.

dropout_transition

The transition time, in seconds, for volume renormalization when an input stream ends. The default value is 2 seconds.

anequalizer

High-order parametric multiband equalizer for each channel.

It accepts the following parameters:

params

This option string is in format: "cchn f=cf w=w g=g t=f | ..." Each equalizer band is separated by '|'.

chn

Set channel number to which equalization will be applied. If input doesn't have that channel the entry is ignored.

cf

Set central frequency for band. If input doesn't have that frequency the entry is ignored.

w

Set band width in hertz.

g

Set band gain in dB.

f

Set filter type for band, optional, can be:

0

Butterworth, this is default.

1

Chebyshev type 1.

2

Chebyshev type 2.

curves

With this option activated frequency response of anequalizer is displayed in video stream.

size

Set video stream size. Only useful if curves option is activated.

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.

fscale

Set frequency scale used to draw frequency response in video output. Can be linear or logarithmic. Default is logarithmic.

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.

Examples

• Lower gain by 10 of central frequency 200Hz and width 100 Hz for first 2 channels using Chebyshev type 1 filter:

          anequalizer=c0 f=200 w=100 g=-10 t=1|c1 f=200 w=100 g=-10 t=1
  

Commands

This filter supports the following commands:

change

Alter existing filter parameters. Syntax for the commands is : "fN|f=freq|w=width|g=gain"

fN is existing filter number, starting from 0, if no such filter is available error is returned. freq set new frequency parameter. width set new width parameter in herz. 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=...

anull

Pass the audio source unchanged to the output.

apad

Pad the end of an audio stream with silence.

This can be used together with ffmpeg -shortest to extend audio streams to the same length as the video stream.

A description of the accepted options follows.

packet_size

Set silence packet size. Default value is 4096.

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 whole_len.

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 pad_len.

If neither the pad_len nor the whole_len option is set, the filter will add silence to the end of the input stream indefinitely.

Examples

• Add 1024 samples of silence to the end of the input:

          apad=pad_len=1024
  

• Make sure the audio output will contain at least 10000 samples, pad the input with silence if required:

          apad=whole_len=10000
  

• Use 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 shortest option:

          ffmpeg -i VIDEO -i AUDIO -filter_complex "[1:0]apad" -shortest OUTPUT
  

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.

in_gain

Set input gain. Default is 0.4.

out_gain

Set output gain. Default is 0.74

delay

Set delay in milliseconds. Default is 3.0.

decay

Set decay. Default is 0.4.

speed

Set modulation speed in Hz. Default is 0.5.

type

Set modulation type. Default is triangular.

It accepts the following values:

triangular, t

sinusoidal, s

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:

level_in

Set input gain. By default it is 1. Range is [0.015625 - 64].

level_out

Set output gain. By default it is 1. Range is [0.015625 - 64].

mode

Set waveform shape the LFO will use. Can be one of: sine, triangle, square, sawup or sawdown. Default is sine.

amount

Set modulation. Define how much of original signal is affected by the LFO.

offset_l

Set left channel offset. Default is 0. Allowed range is [0 - 1].

offset_r

Set right channel offset. Default is 0.5. Allowed range is [0 - 1].

width

Set pulse width. Default is 1. Allowed range is [0 - 2].

timing

Set possible timing mode. Can be one of: bpm, ms or hz. Default is hz.

bpm

Set bpm. Default is 120. Allowed range is [30 - 300]. Only used if timing is set to bpm.

ms

Set ms. Default is 500. Allowed range is [10 - 2000]. Only used if timing is set to ms.

hz

Set frequency in Hz. Default is 2. Allowed range is [0.01 - 100]. Only used if timing is set to hz.

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 [sample_rate:]resampler_options, where sample_rate expresses a sample rate and resampler_options is a list of key=value pairs, separated by ``:''. See the ffmpeg-resampler manual for the complete list of supported options.

Examples

• Resample the input audio to 44100Hz:

          aresample=44100
  

• Stretch/squeeze samples to the given timestamps, with a maximum of 1000 samples per second compensation:

          aresample=async=1000
  

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:

nb_out_samples, n

Set the number of frames per each output audio frame. The number is intended as the number of samples per each channel. Default value is 1024.

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.

For example, to set the number of per-frame samples to 1234 and disable padding for the last frame, use:

        asetnsamples=n=1234:p=0

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:

sample_rate, r

Set the output sample rate. Default is 44100 Hz.

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 key:value.

The following values are shown in the output:

n

The (sequential) number of the input frame, starting from 0.

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/sample_rate.

pts_time

The presentation timestamp of the input frame in seconds.

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)

fmt

The sample format.

chlayout

The channel layout.

rate

The sample rate for the audio frame.

nb_samples

The number of samples (per channel) in the frame.

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.

plane_checksums

A list of Adler-32 checksums for each data plane.

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:

length

Short window length in seconds, used for peak and trough RMS measurement. Default is 0.05 (50 milliseconds). Allowed range is "[0.1 - 10]".

metadata

Set metadata injection. All the metadata keys are prefixed with "lavfi.astats.X", where "X" is channel number starting from 1 or string "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 "lavfi.astats.1.DC_offset" or this "lavfi.astats.Overall.Peak_count".

For description what each key means read below.

reset

Set number of frame after which stats are going to be recalculated. Default is disabled.

A description of each shown parameter follows:

DC offset

Mean amplitude displacement from zero.

Min level

Minimal sample level.

Max level

Maximal sample level.

Min difference

Minimal difference between two consecutive samples.

Max difference

Maximal difference between two consecutive samples.

Mean difference

Mean difference between two consecutive samples. The average of each difference between two consecutive samples.

Peak level dB

RMS level dB

Standard peak and RMS level measured in dBFS.

RMS peak dB

RMS trough dB

Peak and trough values for RMS level measured over a short window.

Crest factor

Standard ratio of peak to RMS level (note: not in dB).

Flat factor

Flatness (i.e. consecutive samples with the same value) of the signal at its peak levels (i.e. either Min level or Max level).

Peak count

Number of occasions (not the number of samples) that the signal attained either Min level or Max level.

Bit depth

Overall bit depth of audio. Number of bits used for each sample.

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 aresample to do squeezing/stretching.

It accepts the following parameters:

compensate

Enable stretching/squeezing the data to make it match the timestamps. Disabled by default. When disabled, time gaps are covered with silence.

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.

max_comp

The maximum compensation in samples per second. Only relevant with compensate=1. The default value is 500.

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.

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.

Examples

• Slow down audio to 80% tempo:

          atempo=0.8
  

• To speed up audio to 125% tempo:

          atempo=1.25
  

atrim

Trim the input so that the output contains one continuous subpart of the input.

It accepts the following parameters:

start

Timestamp (in seconds) of the start of the section to keep. I.e. the audio sample with the timestamp start will be the first sample in the output.

end

Specify time of the first audio sample that will be dropped, i.e. the audio sample immediately preceding the one with the timestamp end will be the last sample in the output.

start_pts

Same as start, except this option sets the start timestamp in samples instead of seconds.

end_pts

Same as end, except this option sets the end timestamp in samples instead of seconds.

duration

The maximum duration of the output in seconds.

start_sample

The number of the first sample that should be output.

end_sample

The number of the first sample that should be dropped.

start, end, and duration are expressed as time duration specifications; see the Time duration section in the ffmpeg-utils(1) manual.

Note that the first two sets of the start/end options and the 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:

• Drop everything except the second minute of input:

          ffmpeg -i INPUT -af atrim=60:120
  

• Keep only the first 1000 samples:

          ffmpeg -i INPUT -af atrim=end_sample=1000
  

bandpass

Apply a two-pole Butterworth band-pass filter with central frequency frequency, and (3dB-point) band-width width. The 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:

frequency, f

Set the filter's central frequency. Default is 3000.

csg

Constant skirt gain if set to 1. Defaults to 0.

width_type

Set method to specify band-width of filter.

h

Hz

q

Q-Factor

o

octave

s

slope

width, w

Specify the band-width of a filter in width_type units.

bandreject

Apply a two-pole Butterworth band-reject filter with central frequency frequency, and (3dB-point) band-width width. The filter roll off at 6dB per octave (20dB per decade).

The filter accepts the following options:

frequency, f

Set the filter's central frequency. Default is 3000.

width_type

Set method to specify band-width of filter.

h

Hz

q

Q-Factor

o

octave

s

slope

width, w

Specify the band-width of a filter in width_type units.

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:

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.

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 100 Hz.

width_type

Set method to specify band-width of filter.

h

Hz

q

Q-Factor

o

octave

s

slope

width, w

Determine how steep is the filter's shelf transition.

biquad

Apply a biquad IIR filter with the given coefficients. Where b0, b1, b2 and a0, a1, a2 are the numerator and denominator coefficients respectively.

bs2b

Bauer stereo to binaural transformation, which improves headphone listening of stereo audio records.

It accepts the following parameters:

profile

Pre-defined crossfeed level.

default

Default level (fcut=700, feed=50).

cmoy

Chu Moy circuit (fcut=700, feed=60).

jmeier

Jan Meier circuit (fcut=650, feed=95).

fcut

Cut frequency (in Hz).

feed

Feed level (in Hz).

channelmap

Remap input channels to new locations.

It accepts the following parameters:

channel_layout

The channel layout of the output stream.

map

Map channels from input to output. The argument is a '|'-separated list of mappings, each in the "in_channel-out_channel" or in_channel form. 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. out_channel is the name of the output channel or its index in the output channel layout. If out_channel is not given then it is implicitly an index, starting with zero and increasing by one for each mapping.

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,

        ffmpeg -i in.mov -filter 'channelmap=map=DL-FL|DR-FR' out.wav

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

        ffmpeg -i in.wav -filter 'channelmap=1|2|0|5|3|4:5.1' out.wav

channelsplit

Split each channel from an input audio stream into a separate output stream.

It accepts the following parameters:

channel_layout

The channel layout of the input stream. The default is ``stereo''.

For example, assuming a stereo input MP3 file,

        ffmpeg -i in.mp3 -filter_complex channelsplit out.mkv

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:

        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

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:

in_gain

Set input gain. Default is 0.4.

out_gain

Set output gain. Default is 0.4.

delays

Set delays. A typical delay is around 40ms to 60ms.

decays

Set decays.

speeds

Set speeds.

depths

Set depths.

Examples

• A single delay:

          chorus=0.7:0.9:55:0.4:0.25:2
  

• Two delays:

          chorus=0.6:0.9:50|60:0.4|0.32:0.25|0.4:2|1.3
  

• Fuller sounding chorus with three delays:

          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
  

compand

Compress or expand the audio's dynamic range.

It accepts the following parameters:

attacks

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. attacks refers to increase of volume and 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.

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: "x0/y0|x1/y1|x2/y2|...." or "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 "0/0" is assumed but may be overridden (by "0/out-dBn"). Typical values for the transfer function are "-70/-70|-60/-20".

soft-knee

Set the curve radius in dB for all joints. It defaults to 0.01.

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.

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.

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.

Examples

• Make music with both quiet and loud passages suitable for listening to in a noisy environment:

          compand=.3|.3:1|1:-90/-60|-60/-40|-40/-30|-20/-20:6:0:-90:0.2
  

  Another example for audio with whisper and explosion parts:

          compand=0|0:1|1:-90/-900|-70/-70|-30/-9|0/-3:6:0:0:0
  

• A noise gate for when the noise is at a lower level than the signal:

          compand=.1|.1:.2|.2:-900/-900|-50.1/-900|-50/-50:.01:0:-90:.1
  

• 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):

          compand=.1|.1:.1|.1:-45.1/-45.1|-45/-900|0/-900:.01:45:-90:.1
  

• 2:1 compression starting at -6dB:

          compand=points=-80/-80|-6/-6|0/-3.8|20/3.5
  

• 2:1 compression starting at -9dB:

          compand=points=-80/-80|-9/-9|0/-5.3|20/2.9
  

• 2:1 compression starting at -12dB:

          compand=points=-80/-80|-12/-12|0/-6.8|20/1.9
  

• 2:1 compression starting at -18dB:

          compand=points=-80/-80|-18/-18|0/-9.8|20/0.7
  

• 3:1 compression starting at -15dB:

          compand=points=-80/-80|-15/-15|0/-10.8|20/-5.2
  

• Compressor/Gate:

          compand=points=-80/-105|-62/-80|-15.4/-15.4|0/-12|20/-7.6
  

• Expander:

          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
  

• Hard limiter at -6dB:

          compand=attacks=0:points=-80/-80|-6/-6|20/-6
  

• Hard limiter at -12dB:

          compand=attacks=0:points=-80/-80|-12/-12|20/-12
  

• Hard noise gate at -35 dB:

          compand=attacks=0:points=-80/-115|-35.1/-80|-35/-35|20/20
  

• Soft limiter:

          compand=attacks=0:points=-80/-80|-12.4/-12.4|-6/-8|0/-6.8|20/-2.8
  

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:

mm

Set millimeters distance. This is compensation distance for fine tuning. Default is 0.

cm

Set cm distance. This is compensation distance for tightening distance setup. Default is 0.

m

Set meters distance. This is compensation distance for hard distance setup. Default is 0.

dry

Set dry amount. Amount of unprocessed (dry) signal. Default is 0.

wet

Set wet amount. Amount of processed (wet) signal. Default is 1.

temp

Set temperature degree in Celsius. This is the temperature of the environment. Default is 20.

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 astats filter can be used to determine if a signal has a DC offset.

shift

Set the DC shift, allowed range is [-1, 1]. It indicates the amount to shift the audio.

limitergain

Optional. It should have a value much less than 1 (e.g. 0.05 or 0.02) and is used to prevent clipping.

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.

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.

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 "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.

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.

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.

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.

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.

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.

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.

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.

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.

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:

frequency, f

Set the filter's central frequency in Hz.

width_type

Set method to specify band-width of filter.

h

Hz

q

Q-Factor

o

octave

s

slope

width, w

Specify the band-width of a filter in width_type units.

gain, g

Set the required gain or attenuation in dB. Beware of clipping when using a positive gain.

Examples

• Attenuate 10 dB at 1000 Hz, with a bandwidth of 200 Hz:

          equalizer=f=1000:width_type=h:width=200:g=-10
  

• Apply 2 dB gain at 1000 Hz with Q 1 and attenuate 5 dB at 100 Hz with Q 2:

          equalizer=f=1000:width_type=q:width=1:g=2,equalizer=f=100:width_type=q:width=2:g=-5
  

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:

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.

c

Enable clipping. By default is enabled.

flanger

Apply a flanging effect to the audio.

The filter accepts the following options:

delay

Set base delay in milliseconds. Range from 0 to 30. Default value is 0.

depth

Set added swep delay in milliseconds. Range from 0 to 10. Default value is 2.

regen

Set percentage regeneration (delayed signal feedback). Range from -95 to 95. Default value is 0.

width

Set percentage of delayed signal mixed with original. Range from 0 to 100. Default value is 71.

speed

Set sweeps per second (Hz). Range from 0.1 to 10. Default value is 0.5.

shape

Set swept wave shape, can be triangular or sinusoidal. Default value is sinusoidal.

phase

Set swept wave percentage-shift for multi channel. Range from 0 to 100. Default value is 25.

interp

Set delay-line interpolation, linear or quadratic. Default is linear.

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:

frequency, f

Set frequency in Hz. Default is 3000.

poles, p

Set number of poles. Default is 2.

width_type

Set method to specify band-width of filter.

h

Hz

q

Q-Factor

o

octave

s

slope

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.

join

Join multiple input streams into one multi-channel stream.

It accepts the following parameters:

inputs

The number of input streams. It defaults to 2.

channel_layout

The desired output channel layout. It defaults to stereo.

map

Map channels from inputs to output. The argument is a '|'-separated list of mappings, each in the "input_idx.in_channel-out_channel" form. input_idx is the 0-based index of the input stream. 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. out_channel is the name of the output channel.

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):

        ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex join=inputs=3 OUTPUT

Build a 5.1 output from 6 single-channel streams:

        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

ladspa

Load a LADSPA (Linux Audio Developer's Simple Plugin API) plugin.

To enable compilation of this filter you need to configure FFmpeg with "--enable-ladspa".

file, f

Specifies the name of LADSPA plugin library to load. If the environment variable LADSPA_PATH is defined, the LADSPA plugin is searched in each one of the directories specified by the colon separated list in LADSPA_PATH, otherwise in the standard LADSPA paths, which are in this order: HOME/.ladspa/lib/, /usr/local/lib/ladspa/, /usr/lib/ladspa/.

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.

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=value0|c1=value1|c2=value2|..., where valuei is the value set on the i-th control. Alternatively they can be also defined using the following syntax: value0|value1|value2|..., where valuei is the value set on the i-th control. If controls is set to "help", all available controls and their valid ranges are printed.

sample_rate, s

Specify the sample rate, default to 44100. Only used if plugin have zero inputs.

nb_samples, n

Set the number of samples per channel per each output frame, default is 1024. Only used if plugin have zero inputs.

duration, d

Set the minimum duration of the sourced audio. See the Time duration section in the ffmpeg-utils(1) manual 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.

Examples

• List all available plugins within amp (LADSPA example plugin) library:

          ladspa=file=amp
  

• List all available controls and their valid ranges for "vcf_notch" plugin from "VCF" library:

          ladspa=f=vcf:p=vcf_notch:c=help
  

• Simulate low quality audio equipment using "Computer Music Toolkit" (CMT) plugin library:

          ladspa=file=cmt:plugin=lofi:controls=c0=22|c1=12|c2=12
  

• Add reverberation to the audio using TAP-plugins (Tom's Audio Processing plugins):

          ladspa=file=tap_reverb:tap_reverb
  

• Generate white noise, with 0.2 amplitude:

          ladspa=file=cmt:noise_source_white:c=c0=.2
  

• Generate 20 bpm clicks using plugin "C* Click - Metronome" from the "C* Audio Plugin Suite" (CAPS) library:

          ladspa=file=caps:Click:c=c1=20'
  

• Apply "C* Eq10X2 - Stereo 10-band equaliser" effect:

          ladspa=caps:Eq10X2:c=c0=-48|c9=-24|c3=12|c4=2
  

• Increase volume by 20dB using fast lookahead limiter from Steve Harris "SWH Plugins" collection:

          ladspa=fast_lookahead_limiter_1913:fastLookaheadLimiter:20|0|2
  

• Attenuate low frequencies using Multiband EQ from Steve Harris "SWH Plugins" collection:

          ladspa=mbeq_1197:mbeq:-24|-24|-24|0|0|0|0|0|0|0|0|0|0|0|0
  

Commands

This filter supports the following commands:

cN

Modify the N-th control value.

If the specified value is not valid, it is ignored and prior one is kept.

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:

frequency, f

Set frequency in Hz. Default is 500.

poles, p

Set number of poles. Default is 2.

width_type

Set method to specify band-width of filter.

h

Hz

q

Q-Factor

o

octave

s

slope

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.

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: "l|outdef|outdef|..."

l

output channel layout or number of channels

outdef

output channel specification, of the form: "out_name=[gain*]in_name[+[gain*]in_name...]"

out_name

output channel to define, either a channel name (FL, FR, etc.) or a channel number (c0, c1, etc.)

gain

multiplicative coefficient for the channel, 1 leaving the volume unchanged

in_name

input channel to use, see out_name for details; it is not possible to mix named and numbered input channels

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.

Mixing examples

For example, if you want to down-mix from stereo to mono, but with a bigger factor for the left channel:

        pan=1c|c0=0.9*c0+0.1*c1

A customized down-mix to stereo that works automatically for 3-, 4-, 5- and 7-channels surround:

        pan=stereo| FL < FL + 0.5*FC + 0.6*BL + 0.6*SL | FR < FR + 0.5*FC + 0.6*BR + 0.6*SR

Note that ffmpeg integrates a default down-mix (and up-mix) system that should be preferred (see ``-ac'' option) unless you have very specific needs.

Remapping examples

The channel remapping will be effective if, and only if:

*<gain coefficients are zeroes or ones,>

*<only one input per channel output,>

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:

        pan="stereo| c0=FL | c1=FR"

Given the same source, you can also switch front left and front right channels and keep the input channel layout:

        pan="5.1| c0=c1 | c1=c0 | c2=c2 | c3=c3 | c4=c4 | c5=c5"

If the input is a stereo audio stream, you can mute the front left channel (and still keep the stereo channel layout) with:

        pan="stereo|c1=c1"

Still with a stereo audio stream input, you can copy the right channel in both front left and right:

        pan="stereo| c0=FR | c1=FR"

replaygain

ReplayGain scanner filter. This filter takes an audio stream as an input and outputs it unchanged. At end of filtering it displays "track_gain" and "track_peak".

resample

Convert the audio sample format, sample rate and channel layout. It is not meant to be used directly.

rubberband

Apply time-stretching and pitch-shifting with librubberband.

The filter accepts the following options:

tempo

Set tempo scale factor.

pitch

Set pitch scale factor.

transients

Set transients detector. Possible values are:

crisp

mixed

smooth

detector

Set detector. Possible values are:

compound

percussive

soft

phase

Set phase. Possible values are:

laminar

independent

window

Set processing window size. Possible values are:

standard

short

long

smoothing

Set smoothing. Possible values are:

off

on

formant

Enable formant preservation when shift pitching. Possible values are:

shifted

preserved

pitchq

Set pitch quality. Possible values are:

quality

speed

consistency

channels

Set channels. Possible values are:

apart

together

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 pan and amerge filter.

The filter accepts the following options:

level_in

Set input gain. Default is 1. Range is between 0.015625 and 64.

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.

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.

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.

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.

makeup

Set the amount by how much signal will be amplified after processing. Default is 2. Range is from 1 and 64.

knee

Curve the sharp knee around the threshold to enter gain reduction more softly. Default is 2.82843. Range is between 1 and 8.

link

Choose if the "average" level between all channels of side-chain stream or the louder("maximum") channel of side-chain stream affects the reduction. Default is "average".

detection

Should the exact signal be taken in case of "peak" or an RMS one in case of "rms". Default is "rms" which is mainly smoother.

level_sc

Set sidechain gain. Default is 1. Range is between 0.015625 and 64.

mix

How much to use compressed signal in output. Default is 1. Range is between 0 and 1.

Examples

• 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:

          ffmpeg -i main.flac -i sidechain.flac -filter_complex "[1:a]asplit=2[sc][mix];[0:a][sc]sidechaincompress[compr];[compr][mix]amerge"
  

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:

level_in

Set input level before filtering. Default is 1. Allowed range is from 0.015625 to 64.

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.

threshold

If a signal rises above this level the gain reduction is released. Default is 0.125. Allowed range is from 0 to 1.

ratio

Set a ratio about which the signal is reduced. Default is 2. Allowed range is from 1 to 9000.

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.

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.

makeup

Set amount of amplification of signal after processing. Default is 1. Allowed range is from 1 to 64.

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.

detection

Choose if exact signal should be taken for detection or an RMS like one. Default is rms. Can be peak or rms.

link

Choose if the average level between all channels or the louder channel affects the reduction. Default is average. Can be average or maximum.

level_sc

Set sidechain gain. Default is 1. Range is from 0.015625 to 64.

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:

duration, d

Set silence duration until notification (default is 2 seconds).

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.

Examples

• Detect 5 seconds of silence with -50dB noise tolerance:

          silencedetect=n=-50dB:d=5
  

• Complete example with ffmpeg to detect silence with 0.0001 noise tolerance in silence.mp3:

          ffmpeg -i silence.mp3 -af silencedetect=noise=0.0001 -f null -
  

silenceremove

Remove silence from the beginning, middle or end of the audio.

The filter accepts the following options:

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 start_periods will be 1 but it can be increased to higher values to trim all audio up to specific count of non-silence periods. Default value is 0.

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 0.

start_threshold

This indicates what sample value should be treated as silence. For digital audio, a value of 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 0.

stop_periods

Set the count for trimming silence from the end of audio. To remove silence from the middle of a file, specify a 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 start_periods, making it suitable for removing periods of silence in the middle of the audio. Default value is 0.

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 0.

stop_threshold

This is the same as 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 0.

leave_silence

This indicate that 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 0.

detection

Set how is silence detected. Can be "rms" or "peak". Second is faster and works better with digital silence which is exactly 0. Default value is "rms".

window

Set ratio used to calculate size of window for detecting silence. Default value is 0.02. Allowed range is from 0 to 10.

Examples

• 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:

          silenceremove=1:5:0.02
  

• Trim all silence encountered from begining to end where there is more than 1 second of silence in audio:

          silenceremove=0:0:0:-1:1:-90dB
  

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 <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 "--enable-netcdf".

The filter accepts the following options:

sofa

Set the SOFA file used for rendering.

gain

Set gain applied to audio. Value is in dB. Default is 0.

rotation

Set rotation of virtual loudspeakers in deg. Default is 0.

elevation

Set elevation of virtual speakers in deg. Default is 0.

radius

Set distance in meters between loudspeakers and the listener with near-field HRTFs. Default is 1.

type

Set processing type. Can be time or freq. time is processing audio in time domain which is slow but gives high quality output. freq is processing audio in frequency domain which is fast but gives mediocre output. Default is freq.

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:

level_in

Set input level before filtering for both channels. Defaults is 1. Allowed range is from 0.015625 to 64.

level_out

Set output level after filtering for both channels. Defaults is 1. Allowed range is from 0.015625 to 64.

balance_in

Set input balance between both channels. Default is 0. Allowed range is from -1 to 1.

balance_out

Set output balance between both channels. Default is 0. Allowed range is from -1 to 1.

softclip

Enable softclipping. Results in analog distortion instead of harsh digital 0dB clipping. Disabled by default.

mutel

Mute the left channel. Disabled by default.

muter

Mute the right channel. Disabled by default.

phasel

Change the phase of the left channel. Disabled by default.

phaser

Change the phase of the right channel. Disabled by default.

mode

Set stereo mode. Available values are:

lr>lr

Left/Right to Left/Right, this is default.

lr>ms

Left/Right to Mid/Side.

ms>lr

Mid/Side to Left/Right.

lr>ll

Left/Right to Left/Left.

lr>rr

Left/Right to Right/Right.

lr>l+r

Left/Right to Left + Right.

lr>rl

Left/Right to Right/Left.

slev

Set level of side signal. Default is 1. Allowed range is from 0.015625 to 64.

sbal

Set balance of side signal. Default is 0. Allowed range is from -1 to 1.

mlev

Set level of the middle signal. Default is 1. Allowed range is from 0.015625 to 64.

mpan

Set middle signal pan. Default is 0. Allowed range is from -1 to 1.

base

Set stereo base between mono and inversed channels. Default is 0. Allowed range is from -1 to 1.

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.

sclevel

Set S/C level. Default is 1. Allowed range is from 1 to 100.

phase

Set the stereo phase in degrees. Default is 0. Allowed range is from 0 to 360.

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:

delay

Time in milliseconds of the delay of left signal into right and vice versa. Default is 20 milliseconds.

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.

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.

drymix

Set level of input signal of original channel. Default is 0.8.

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:

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.

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 3000 Hz.

width_type

Set method to specify band-width of filter.

h

Hz

q

Q-Factor

o

octave

s

slope

width, w

Determine how steep is the filter's shelf transition.

tremolo

Sinusoidal amplitude modulation.

The filter accepts the following options:

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.

d

Depth of modulation as a percentage. Range is 0.0 - 1.0. Default value is 0.5.

vibrato

Sinusoidal phase modulation.

The filter accepts the following options:

f

Modulation frequency in Hertz. Range is 0.1 - 20000.0. Default value is 5.0 Hz.

d

Depth of modulation as a percentage. Range is 0.0 - 1.0. Default value is 0.5.

volume

Adjust the input audio volume.

It accepts the following parameters:

volume

Set audio volume expression.

Output values are clipped to the maximum value.

The output audio volume is given by the relation:

        <output_volume> = <volume> * <input_volume>

The default value for volume is ``1.0''.

precision

This parameter represents the mathematical precision.

It determines which input sample formats will be allowed, which affects the precision of the volume scaling.

fixed

8-bit fixed-point; this limits input sample format to U8, S16, and S32.

float

32-bit floating-point; this limits input sample format to FLT. (default)

double

64-bit floating-point; this limits input sample format to DBL.

replaygain

Choose the behaviour on encountering ReplayGain side data in input frames.

drop

Remove ReplayGain side data, ignoring its contents (the default).

ignore

Ignore ReplayGain side data, but leave it in the frame.

track

Prefer the track gain, if present.

album

Prefer the album gain, if present.

replaygain_preamp

Pre-amplification gain in dB to apply to the selected replaygain gain.

Default value for replaygain_preamp is 0.0.

eval

Set when the volume expression is evaluated.

It accepts the following values:

once

only evaluate expression once during the filter initialization, or when the volume command is sent

frame

evaluate expression for each incoming frame

Default value is once.

The volume expression can contain the following parameters.

n

frame number (starting at zero)

nb_channels

number of channels

nb_consumed_samples

number of samples consumed by the filter

nb_samples

number of samples in the current frame

pos

original frame position in the file

pts

frame PTS

sample_rate

sample rate

startpts

PTS at start of stream

startt

time at start of stream

t

frame time

tb

timestamp timebase

volume

last set volume value

Note that when eval is set to once only the sample_rate and tb variables are available, all other variables will evaluate to NAN.

Commands

This filter supports the following commands:

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.

replaygain_noclip

Prevent clipping by limiting the gain applied.

Default value for replaygain_noclip is 1.

Examples

• Halve the input audio volume:

          volume=volume=0.5
          volume=volume=1/2
          volume=volume=-6.0206dB
  

  In all the above example the named key for volume can be omitted, for example like in:

          volume=0.5
  

• Increase input audio power by 6 decibels using fixed-point precision:

          volume=volume=6dB:precision=fixed
  

• Fade volume after time 10 with an annihilation period of 5 seconds:

          volume='if(lt(t,10),1,max(1-(t-10)/5,0))':eval=frame
  

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.

Examples

Here is an excerpt of the output:

        [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

It means that:

• The mean square energy is approximately -27 dB, or 10^-2.7.
• The largest sample is at -4 dB, or more precisely between -4 dB and -5 dB.
• There are 6 samples at -4 dB, 62 at -5 dB, 286 at -6 dB, etc.

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.

AUDIO SOURCES

Below is a description of the currently available audio sources.

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 libavfilter/asrc_abuffer.h.

It accepts the following parameters:

time_base

The timebase which will be used for timestamps of submitted frames. It must be either a floating-point number or in numerator/denominator form.

sample_rate

The sample rate of the incoming audio buffers.

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 libavutil/samplefmt.h

channel_layout

The channel layout of the incoming audio buffers. Either a channel layout name from channel_layout_map in libavutil/channel_layout.c or its corresponding integer representation from the AV_CH_LAYOUT_* macros in libavutil/channel_layout.h

channels

The number of channels of the incoming audio buffers. If both channels and channel_layout are specified, then they must be consistent.

Examples

        abuffer=sample_rate=44100:sample_fmt=s16p:channel_layout=stereo

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:

        abuffer=sample_rate=44100:sample_fmt=6:channel_layout=0x3

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:

exprs

Set the '|'-separated expressions list for each separate channel. In case the 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.

channel_layout, c

Set the channel layout. The number of channels in the specified layout must be equal to the number of specified expressions.

duration, d

Set the minimum duration of the sourced audio. See the Time duration section in the ffmpeg-utils(1) manual 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.

nb_samples, n

Set the number of samples per channel per each output frame, default to 1024.

sample_rate, s

Specify the sample rate, default to 44100.

Each expression in exprs can contain the following constants:

n

number of the evaluated sample, starting from 0

t

time of the evaluated sample expressed in seconds, starting from 0

s

sample rate

Examples

• Generate silence:

          aevalsrc=0
  

• Generate a sin signal with frequency of 440 Hz, set sample rate to 8000 Hz:

          aevalsrc="sin(440*2*PI*t):s=8000"
  

• Generate a two channels signal, specify the channel layout (Front Center + Back Center) explicitly:

          aevalsrc="sin(420*2*PI*t)|cos(430*2*PI*t):c=FC|BC"
  

• Generate white noise:

          aevalsrc="-2+random(0)"
  

• Generate an amplitude modulated signal:

          aevalsrc="sin(10*2*PI*t)*sin(880*2*PI*t)"
  

• Generate 2.5 Hz binaural beats on a 360 Hz carrier:

          aevalsrc="0.1*sin(2*PI*(360-2.5/2)*t) | 0.1*sin(2*PI*(360+2.5/2)*t)"
  

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:

channel_layout, cl

Specifies the channel layout, and can be either an integer or a string representing a channel layout. The default value of channel_layout is ``stereo''.

Check the channel_layout_map definition in libavutil/channel_layout.c for the mapping between strings and channel layout values.

sample_rate, r

Specifies the sample rate, and defaults to 44100.

nb_samples, n

Set the number of samples per requested frames.

Examples

• Set the sample rate to 48000 Hz and the channel layout to AV_CH_LAYOUT_MONO.

          anullsrc=r=48000:cl=4
  

• Do the same operation with a more obvious syntax:

          anullsrc=r=48000:cl=mono
  

All the parameters need to be explicitly defined.

flite

Synthesize a voice utterance using the libflite library.

To enable compilation of this filter you need to configure FFmpeg with "--enable-libflite".

Note that the flite library is not thread-safe.

The filter accepts the following options:

list_voices

If set to 1, list the names of the available voices and exit immediately. Default value is 0.

nb_samples, n

Set the maximum number of samples per frame. Default value is 512.

textfile

Set the filename containing the text to speak.

text

Set the text to speak.

voice, v

Set the voice to use for the speech synthesis. Default value is "kal". See also the list_voices option.

Examples

• Read from file speech.txt, and synthesize the text using the standard flite voice:

          flite=textfile=speech.txt
  

• Read the specified text selecting the "slt" voice:

          flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt
  

• Input text to ffmpeg:

          ffmpeg -f lavfi -i flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt
  

• Make ffplay speak the specified text, using "flite" and the "lavfi" device:

          ffplay -f lavfi flite=text='No more be grieved for which that thou hast done.'
  

For more information about libflite, check: <http://www.speech.cs.cmu.edu/flite/>

anoisesrc

Generate a noise audio signal.

The filter accepts the following options:

sample_rate, r

Specify the sample rate. Default value is 48000 Hz.

amplitude, a

Specify the amplitude (0.0 - 1.0) of the generated audio stream. Default value is 1.0.

duration, d

Specify the duration of the generated audio stream. Not specifying this option results in noise with an infinite length.

color, colour, c

Specify the color of noise. Available noise colors are white, pink, and brown. Default color is white.

seed, s

Specify a value used to seed the PRNG.

nb_samples, n

Set the number of samples per each output frame, default is 1024.

Examples

• Generate 60 seconds of pink noise, with a 44.1 kHz sampling rate and an amplitude of 0.5:

          anoisesrc=d=60:c=pink:r=44100:a=0.5
  

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:

frequency, f

Set the carrier frequency. Default is 440 Hz.

beep_factor, b

Enable a periodic beep every second with frequency beep_factor times the carrier frequency. Default is 0, meaning the beep is disabled.

sample_rate, r

Specify the sample rate, default is 44100.

duration, d

Specify the duration of the generated audio stream.

samples_per_frame

Set the number of samples per output frame.

The expression can contain the following constants:

n

The (sequential) number of the output audio frame, starting from 0.

pts

The PTS (Presentation TimeStamp) of the output audio frame, expressed in TB units.

t

The PTS of the output audio frame, expressed in seconds.

TB

The timebase of the output audio frames.

Default is 1024.

Examples

• Generate a simple 440 Hz sine wave:

          sine
  

• Generate a 220 Hz sine wave with a 880 Hz beep each second, for 5 seconds:

          sine=220:4:d=5
          sine=f=220:b=4:d=5
          sine=frequency=220:beep_factor=4:duration=5
  

• Generate a 1 kHz sine wave following "1602,1601,1602,1601,1602" NTSC pattern:

          sine=1000:samples_per_frame='st(0,mod(n,5)); 1602-not(not(eq(ld(0),1)+eq(ld(0),3)))'
  

AUDIO SINKS

Below is a description of the currently available audio sinks.

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 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 "avfilter_init_filter" for initialization.

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.

VIDEO FILTERS

When you configure your FFmpeg build, you can disable any of the existing filters using "--disable-filters". The configure output will show the video filters included in your build.

Below is a description of the currently available video filters.

alphaextract

Extract the alpha component from the input as a grayscale video. This is especially useful with the alphamerge filter.

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 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 alphaextract, you might use:

        movie=in_alpha.mkv [alpha]; [in][alpha] alphamerge [out]

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 overlay filter instead.

ass

Same as the 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 subtitles filter:

shaping

Set the shaping engine

Available values are:

auto

The default libass shaping engine, which is the best available.

simple

Fast, font-agnostic shaper that can do only substitutions

complex

Slower shaper using OpenType for substitutions and positioning

The default is "auto".

atadenoise

Apply an Adaptive Temporal Averaging Denoiser to the video input.

The filter accepts the following options:

0a

Set threshold A for 1st plane. Default is 0.02. Valid range is 0 to 0.3.

0b

Set threshold B for 1st plane. Default is 0.04. Valid range is 0 to 5.

1a

Set threshold A for 2nd plane. Default is 0.02. Valid range is 0 to 0.3.

1b

Set threshold B for 2nd plane. Default is 0.04. Valid range is 0 to 5.

2a

Set threshold A for 3rd plane. Default is 0.02. Valid range is 0 to 0.3.

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.

s

Set number of frames filter will use for averaging. Default is 33. Must be odd number in range [5, 129].

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:

min_val

Set the minimal luminance value. Default is 16.

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:

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.

picture_black_ratio_th, pic_th

Set the threshold for considering a picture ``black''. Express the minimum value for the ratio:

        <nb_black_pixels> / <nb_pixels>

for which a picture is considered black. Default value is 0.98.

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:

        <absolute_threshold> = <luminance_minimum_value> + <pixel_black_th> * <luminance_range_size>

luminance_range_size and 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.

The following example sets the maximum pixel threshold to the minimum value, and detects only black intervals of 2 or more seconds:

        blackdetect=d=2:pix_th=0.00

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:

amount

The percentage of the pixels that have to be below the threshold; it defaults to 98.

threshold, thresh

The threshold below which a pixel value is considered black; it defaults to 32.

blend, tblend

Blend two video frames into each other.

The "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 "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.

c0_mode

c1_mode

c2_mode

c3_mode

all_mode

Set blend mode for specific pixel component or all pixel components in case of all_mode. Default value is "normal".

Available values for component modes are:

addition

addition128

and

average

burn

darken

difference

difference128

divide

dodge

exclusion

glow

hardlight

hardmix

lighten

linearlight

multiply

multiply128

negation

normal

or

overlay

phoenix

pinlight

reflect

screen

softlight

subtract

vividlight

xor

c0_opacity

c1_opacity

c2_opacity

c3_opacity

all_opacity

Set blend opacity for specific pixel component or all pixel components in case of all_opacity. Only used in combination with pixel component blend modes.

c0_expr

c1_expr

c2_expr

c3_expr

all_expr

Set blend expression for specific pixel component or all pixel components in case of all_expr. Note that related mode options will be ignored if those are set.

The expressions can use the following variables:

N

The sequential number of the filtered frame, starting from 0.

X

Y

the coordinates of the current sample

W

H

the width and height of currently filtered plane

SW

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 "1,1" for the luma plane, and "0.5,0.5" for chroma planes.

T

Time of the current frame, expressed in seconds.

TOP, A

Value of pixel component at current location for first video frame (top layer).

BOTTOM, B

Value of pixel component at current location for second video frame (bottom layer).

shortest

Force termination when the shortest input terminates. Default is 0. This option is only defined for the "blend" filter.

repeatlast

Continue applying the last bottom frame after the end of the stream. A value of 0 disable the filter after the last frame of the bottom layer is reached. Default is 1. This option is only defined for the "blend" filter.

Examples

• Apply transition from bottom layer to top layer in first 10 seconds:

          blend=all_expr='A*(if(gte(T,10),1,T/10))+B*(1-(if(gte(T,10),1,T/10)))'
  

• Apply 1x1 checkerboard effect:

          blend=all_expr='if(eq(mod(X,2),mod(Y,2)),A,B)'
  

• Apply uncover left effect:

          blend=all_expr='if(gte(N*SW+X,W),A,B)'
  

• Apply uncover down effect:

          blend=all_expr='if(gte(Y-N*SH,0),A,B)'
  

• Apply uncover up-left effect:

          blend=all_expr='if(gte(T*SH*40+Y,H)*gte((T*40*SW+X)*W/H,W),A,B)'
  

• Split diagonally video and shows top and bottom layer on each side:

          blend=all_expr=if(gt(X,Y*(W/H)),A,B)
  

• Display differences between the current and the previous frame:

          tblend=all_mode=difference128
  

boxblur

Apply a boxblur algorithm to the input video.

It accepts the following parameters:

luma_radius, lr

luma_power, lp

chroma_radius, cr

chroma_power, cp

alpha_radius, ar

alpha_power, ap

A description of the accepted options follows.

luma_radius, lr

chroma_radius, cr

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 "min(w,h)/2" for the luma and alpha planes, and of "min(cw,ch)/2" for the chroma planes.

Default value for luma_radius is ``2''. If not specified, chroma_radius and alpha_radius default to the corresponding value set for luma_radius.

The expressions can contain the following constants:

w

h

The input width and height in pixels.

cw

ch

The input chroma image width and height in pixels.

hsub

vsub

The horizontal and vertical chroma subsample values. For example, for the pixel format ``yuv422p'', hsub is 2 and vsub is 1.

luma_power, lp

chroma_power, cp

alpha_power, ap

Specify how many times the boxblur filter is applied to the corresponding plane.

Default value for luma_power is 2. If not specified, chroma_power and alpha_power default to the corresponding value set for luma_power.

A value of 0 will disable the effect.

Examples

• Apply a boxblur filter with the luma, chroma, and alpha radii set to 2:

          boxblur=luma_radius=2:luma_power=1
          boxblur=2:1
  

• Set the luma radius to 2, and alpha and chroma radius to 0:

          boxblur=2:1:cr=0:ar=0
  

• Set the luma and chroma radii to a fraction of the video dimension:

          boxblur=luma_radius=min(h\,w)/10:luma_power=1:chroma_radius=min(cw\,ch)/10:chroma_power=1
  

chromakey

YUV colorspace color/chroma keying.

The filter accepts the following options:

color

The color which will be replaced with transparency.

similarity

Similarity percentage with the key color.

0.01 matches only the exact key color, while 1.0 matches everything.

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.

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.

Examples

• Make every green pixel in the input image transparent:

          ffmpeg -i input.png -vf chromakey=green out.png
  

• Overlay a greenscreen-video on top of a static black background.

          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
  

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 export_mvs flag in the codec flags2 option.

The filter accepts the following option:

mv

Set motion vectors to visualize.

Available flags for mv are:

pf

forward predicted MVs of P-frames

bf

forward predicted MVs of B-frames

bb

backward predicted MVs of B-frames

qp

Display quantization parameters using the chroma planes

Examples

• Visualizes multi-directionals MVs from P and B-Frames using ffplay:

          ffplay -flags2 +export_mvs input.mpg -vf codecview=mv=pf+bf+bb
  

colorbalance

Modify intensity of primary colors (red, green and blue) of input frames.

The filter allows an input frame to be adjusted in the shadows, midtones or highlights regions for the red-cyan, green-magenta or blue-yellow balance.

A positive adjustment value shifts the balance towards the primary color, a negative value towards the complementary color.

The filter accepts the following options:

rs

gs

bs

Adjust red, green and blue shadows (darkest pixels).

rm

gm

bm

Adjust red, green and blue midtones (medium pixels).

rh

gh

bh

Adjust red, green and blue highlights (brightest pixels).

Allowed ranges for options are "[-1.0, 1.0]". Defaults are 0.

Examples

• Add red color cast to shadows:

          colorbalance=rs=.3
  

colorkey

RGB colorspace color keying.

The filter accepts the following options:

color

The color which will be replaced with transparency.

similarity

Similarity percentage with the key color.

0.01 matches only the exact key color, while 1.0 matches everything.

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.

Examples

• Make every green pixel in the input image transparent:

          ffmpeg -i input.png -vf colorkey=green out.png
  

• Overlay a greenscreen-video on top of a static background image.

          ffmpeg -i background.png -i video.mp4 -filter_complex "[1:v]colorkey=0x3BBD1E:0.3:0.2[ckout];[0:v][ckout]overlay[out]" -map "[out]" output.flv
  

colorlevels

Adjust video input frames using levels.

The filter accepts the following options:

rimin

gimin

bimin

aimin

Adjust red, green, blue and alpha input black point. Allowed ranges for options are "[-1.0, 1.0]". Defaults are 0.

rimax

gimax

bimax

aimax

Adjust red, green, blue and alpha input white point. Allowed ranges for options are "[-1.0, 1.0]". Defaults are 1.

Input levels are used to lighten highlights (bright tones), darken shadows (dark tones), change the balance of bright and dark tones.

romin

gomin

bomin

aomin

Adjust red, green, blue and alpha output black point. Allowed ranges for options are "[0, 1.0]". Defaults are 0.

romax

gomax

bomax

aomax

Adjust red, green, blue and alpha output white point. Allowed ranges for options are "[0, 1.0]". Defaults are 1.

Output levels allows manual selection of a constrained output level range.

Examples

• Make video output darker:

          colorlevels=rimin=0.058:gimin=0.058:bimin=0.058
  

• Increase contrast:

          colorlevels=rimin=0.039:gimin=0.039:bimin=0.039:rimax=0.96:gimax=0.96:bimax=0.96
  

• Make video output lighter:

          colorlevels=rimax=0.902:gimax=0.902:bimax=0.902
  

• Increase brightness:

          colorlevels=romin=0.5:gomin=0.5:bomin=0.5
  

colorchannelmixer

Adjust video input frames by re-mixing color channels.

This filter modifies a color channel by adding the values associated to the other channels of the same pixels. For example if the value to modify is red, the output value will be:

        <red>=<red>*<rr> + <blue>*<rb> + <green>*<rg> + <alpha>*<ra>

The filter accepts the following options:

rr

rg

rb

ra

Adjust contribution of input red, green, blue and alpha channels for output red channel. Default is 1 for rr, and 0 for rg, rb and ra.

gr

gg

gb

ga

Adjust contribution of input red, green, blue and alpha channels for output green channel. Default is 1 for gg, and 0 for gr, gb and ga.

br

bg

bb

ba

Adjust contribution of input red, green, blue and alpha channels for output blue channel. Default is 1 for bb, and 0 for br, bg and ba.

ar

ag

ab

aa

Adjust contribution of input red, green, blue and alpha channels for output alpha channel. Default is 1 for aa, and 0 for ar, ag and ab.

Allowed ranges for options are "[-2.0, 2.0]".

Examples

• Convert source to grayscale:

          colorchannelmixer=.3:.4:.3:0:.3:.4:.3:0:.3:.4:.3
  

• Simulate sepia tones:

          colorchannelmixer=.393:.769:.189:0:.349:.686:.168:0:.272:.534:.131
  

colormatrix

Convert color matrix.

The filter accepts the following options:

src

dst

Specify the source and destination color matrix. Both values must be specified.

The accepted values are:

bt709

BT.709

bt601

BT.601

smpte240m

SMPTE-240M

fcc

FCC

For example to convert from BT.601 to SMPTE-240M, use the command:

        colormatrix=bt601:smpte240m

convolution

Apply convolution 3x3 or 5x5 filter.

The filter accepts the following options:

0m

1m

2m

3m

Set matrix for each plane. Matrix is sequence of 9 or 25 signed integers.

0rdiv

1rdiv

2rdiv

3rdiv

Set multiplier for calculated value for each plane.

0bias

1bias

2bias

3bias

Set bias for each plane. This value is added to the result of the multiplication. Useful for making the overall image brighter or darker. Default is 0.0.

Examples

• Apply sharpen:

          convolution="0 -1 0 -1 5 -1 0 -1 0:0 -1 0 -1 5 -1 0 -1 0:0 -1 0 -1 5 -1 0 -1 0:0 -1 0 -1 5 -1 0 -1 0"
  

• Apply blur:

          convolution="1 1 1 1 1 1 1 1 1:1 1 1 1 1 1 1 1 1:1 1 1 1 1 1 1 1 1:1 1 1 1 1 1 1 1 1:1/9:1/9:1/9:1/9"
  

• Apply edge enhance:

          convolution="0 0 0 -1 1 0 0 0 0:0 0 0 -1 1 0 0 0 0:0 0 0 -1 1 0 0 0 0:0 0 0 -1 1 0 0 0 0:5:1:1:1:0:128:128:128"
  

• Apply edge detect:

          convolution="0 1 0 1 -4 1 0 1 0:0 1 0 1 -4 1 0 1 0:0 1 0 1 -4 1 0 1 0:0 1 0 1 -4 1 0 1 0:5:5:5:1:0:128:128:128"
  

• Apply emboss:

          convolution="-2 -1 0 -1 1 1 0 1 2:-2 -1 0 -1 1 1 0 1 2:-2 -1 0 -1 1 1 0 1 2:-2 -1 0 -1 1 1 0 1 2"
  

copy

Copy the input source unchanged to the output. This is mainly useful for testing purposes.

crop

Crop the input video to given dimensions.

It accepts the following parameters:

w, out_w

The width of the output video. It defaults to "iw". This expression is evaluated only once during the filter configuration, or when the w or out_w command is sent.

h, out_h

The height of the output video. It defaults to "ih". This expression is evaluated only once during the filter configuration, or when the h or out_h command is sent.

x

The horizontal position, in the input video, of the left edge of the output video. It defaults to "(in_w-out_w)/2". This expression is evaluated per-frame.

y

The vertical position, in the input video, of the top edge of the output video. It defaults to "(in_h-out_h)/2". This expression is evaluated per-frame.

keep_aspect

If set to 1 will force the output display aspect ratio to be the same of the input, by changing the output sample aspect ratio. It defaults to 0.

The out_w, out_h, x, y parameters are expressions containing the following constants:

x

y

The computed values for x and y. They are evaluated for each new frame.

in_w

in_h

The input width and height.

iw

ih

These are the same as in_w and in_h.

out_w

out_h

The output (cropped) width and height.

ow

oh

These are the same as out_w and out_h.

a

same as iw / ih

sar

input sample aspect ratio

dar

input display aspect ratio, it is the same as (iw / ih) * sar

hsub

vsub

horizontal and vertical chroma subsample values. For example for the pixel format ``yuv422p'' hsub is 2 and vsub is 1.

n

The number of the input frame, starting from 0.

pos

the position in the file of the input frame, NAN if unknown

t

The timestamp expressed in seconds. It's NAN if the input timestamp is unknown.

The expression for out_w may depend on the value of out_h, and the expression for out_h may depend on out_w, but they cannot depend on x and y, as x and y are evaluated after out_w and out_h.

The x and y parameters specify the expressions for the position of the top-left corner of the output (non-cropped) area. They are evaluated for each frame. If the evaluated value is not valid, it is approximated to the nearest valid value.

The expression for x may depend on y, and the expression for y may depend on x.

Examples

• Crop area with size 100x100 at position (12,34).

          crop=100:100:12:34
  

  Using named options, the example above becomes:

          crop=w=100:h=100:x=12:y=34
  

• Crop the central input area with size 100x100:

          crop=100:100
  

• Crop the central input area with size 2/3 of the input video:

          crop=2/3*in_w:2/3*in_h
  

• Crop the input video central square:

          crop=out_w=in_h
          crop=in_h
  

• Delimit the rectangle with the top-left corner placed at position 100:100 and the right-bottom corner corresponding to the right-bottom corner of the input image.

          crop=in_w-100:in_h-100:100:100
  

• Crop 10 pixels from the left and right borders, and 20 pixels from the top and bottom borders

          crop=in_w-2*10:in_h-2*20
  

• Keep only the bottom right quarter of the input image:

          crop=in_w/2:in_h/2:in_w/2:in_h/2
  

• Crop height for getting Greek harmony:

          crop=in_w:1/PHI*in_w
  

• Apply trembling effect:

          crop=in_w/2:in_h/2:(in_w-out_w)/2+((in_w-out_w)/2)*sin(n/10):(in_h-out_h)/2 +((in_h-out_h)/2)*sin(n/7)
  

• Apply erratic camera effect depending on timestamp:

          crop=in_w/2:in_h/2:(in_w-out_w)/2+((in_w-out_w)/2)*sin(t*10):(in_h-out_h)/2 +((in_h-out_h)/2)*sin(t*13)"
  

• Set x depending on the value of y:

          crop=in_w/2:in_h/2:y:10+10*sin(n/10)
  

Commands

This filter supports the following commands:

w, out_w

h, out_h

x

y

Set width/height of the output video and the horizontal/vertical position in the input video. The command accepts the same syntax of the corresponding option.

If the specified expression is not valid, it is kept at its current value.

cropdetect

Auto-detect the crop size.

It calculates the necessary cropping parameters and prints the recommended parameters via the logging system. The detected dimensions correspond to the non-black area of the input video.

It accepts the following parameters:

limit

Set higher black value threshold, which can be optionally specified from nothing (0) to everything (255 for 8bit based formats). An intensity value greater to the set value is considered non-black. It defaults to 24. You can also specify a value between 0.0 and 1.0 which will be scaled depending on the bitdepth of the pixel format.

round

The value which the width/height should be divisible by. It defaults to 16. The offset is automatically adjusted to center the video. Use 2 to get only even dimensions (needed for 4:2:2 video). 16 is best when encoding to most video codecs.

reset_count, reset

Set the counter that determines after how many frames cropdetect will reset the previously detected largest video area and start over to detect the current optimal crop area. Default value is 0.

This can be useful when channel logos distort the video area. 0 indicates 'never reset', and returns the largest area encountered during playback.

curves

Apply color adjustments using curves.

This filter is similar to the Adobe Photoshop and GIMP curves tools. Each component (red, green and blue) has its values defined by N key points tied from each other using a smooth curve. The x-axis represents the pixel values from the input frame, and the y-axis the new pixel values to be set for the output frame.

By default, a component curve is defined by the two points (0;0) and (1;1). This creates a straight line where each original pixel value is ``adjusted'' to its own value, which means no change to the image.

The filter allows you to redefine these two points and add some more. A new curve (using a natural cubic spline interpolation) will be define to pass smoothly through all these new coordinates. The new defined points needs to be strictly increasing over the x-axis, and their x and y values must be in the [0;1] interval. If the computed curves happened to go outside the vector spaces, the values will be clipped accordingly.

If there is no key point defined in "x=0", the filter will automatically insert a (0;0) point. In the same way, if there is no key point defined in "x=1", the filter will automatically insert a (1;1) point.

The filter accepts the following options:

preset

Select one of the available color presets. This option can be used in addition to the r, g, b parameters; in this case, the later options takes priority on the preset values. Available presets are:

none

color_negative

cross_process

darker

increase_contrast

lighter

linear_contrast

medium_contrast

negative

strong_contrast

vintage

Default is "none".

master, m

Set the master key points. These points will define a second pass mapping. It is sometimes called a ``luminance'' or ``value'' mapping. It can be used with r, g, b or all since it acts like a post-processing LUT.

red, r

Set the key points for the red component.

green, g

Set the key points for the green component.

blue, b

Set the key points for the blue component.

all

Set the key points for all components (not including master). Can be used in addition to the other key points component options. In this case, the unset component(s) will fallback on this all setting.

psfile

Specify a Photoshop curves file (".acv") to import the settings from.

To avoid some filtergraph syntax conflicts, each key points list need to be defined using the following syntax: "x0/y0 x1/y1 x2/y2 ...".

Examples

• Increase slightly the middle level of blue:

          curves=blue='0.5/0.58'
  

• Vintage effect:

          curves=r='0/0.11 .42/.51 1/0.95':g='0.50/0.48':b='0/0.22 .49/.44 1/0.8'
  

  Here we obtain the following coordinates for each components:

  "(0;0.11) (0.42;0.51) (1;0.95)"

  "(0;0) (0.50;0.48) (1;1)"

  "(0;0.22) (0.49;0.44) (1;0.80)"

• The previous example can also be achieved with the associated built-in preset:

          curves=preset=vintage
  

• Or simply:

          curves=vintage
  

• Use a Photoshop preset and redefine the points of the green component:

          curves=psfile='MyCurvesPresets/purple.acv':green='0.45/0.53'
  

dctdnoiz

Denoise frames using 2D DCT (frequency domain filtering).

This filter is not designed for real time.

The filter accepts the following options:

sigma, s

Set the noise sigma constant.

This sigma defines a hard threshold of "3 * sigma"; every DCT coefficient (absolute value) below this threshold with be dropped.

If you need a more advanced filtering, see expr.

Default is 0.

overlap

Set number overlapping pixels for each block. Since the filter can be slow, you may want to reduce this value, at the cost of a less effective filter and the risk of various artefacts.

If the overlapping value doesn't permit processing the whole input width or height, a warning will be displayed and according borders won't be denoised.

Default value is blocksize-1, which is the best possible setting.

expr, e

Set the coefficient factor expression.

For each coefficient of a DCT block, this expression will be evaluated as a multiplier value for the coefficient.

If this is option is set, the sigma option will be ignored.

The absolute value of the coefficient can be accessed through the c variable.

n

Set the blocksize using the number of bits. "1<<n" defines the blocksize, which is the width and height of the processed blocks.

The default value is 3 (8x8) and can be raised to 4 for a blocksize of 16x16. Note that changing this setting has huge consequences on the speed processing. Also, a larger block size does not necessarily means a better de-noising.

Examples

Apply a denoise with a sigma of 4.5:

        dctdnoiz=4.5

The same operation can be achieved using the expression system:

        dctdnoiz=e='gte(c, 4.5*3)'

Violent denoise using a block size of "16x16":

        dctdnoiz=15:n=4

deband

Remove banding artifacts from input video. It works by replacing banded pixels with average value of referenced pixels.

The filter accepts the following options:

1thr

2thr

3thr

4thr

Set banding detection threshold for each plane. Default is 0.02. Valid range is 0.00003 to 0.5. If difference between current pixel and reference pixel is less than threshold, it will be considered as banded.

range, r

Banding detection range in pixels. Default is 16. If positive, random number in range 0 to set value will be used. If negative, exact absolute value will be used. The range defines square of four pixels around current pixel.

direction, d

Set direction in radians from which four pixel will be compared. If positive, random direction from 0 to set direction will be picked. If negative, exact of absolute value will be picked. For example direction 0, -PI or -2*PI radians will pick only pixels on same row and -PI/2 will pick only pixels on same column.

blur

If enabled, current pixel is compared with average value of all four surrounding pixels. The default is enabled. If disabled current pixel is compared with all four surrounding pixels. The pixel is considered banded if only all four differences with surrounding pixels are less than threshold.

decimate

Drop duplicated frames at regular intervals.

The filter accepts the following options:

cycle

Set the number of frames from which one will be dropped. Setting this to N means one frame in every batch of N frames will be dropped. Default is 5.

dupthresh

Set the threshold for duplicate detection. If the difference metric for a frame is less than or equal to this value, then it is declared as duplicate. Default is 1.1

scthresh

Set scene change threshold. Default is 15.

blockx

blocky

Set the size of the x and y-axis blocks used during metric calculations. Larger blocks give better noise suppression, but also give worse detection of small movements. Must be a power of two. Default is 32.

ppsrc

Mark main input as a pre-processed input and activate clean source input stream. This allows the input to be pre-processed with various filters to help the metrics calculation while keeping the frame selection lossless. When set to 1, the first stream is for the pre-processed input, and the second stream is the clean source from where the kept frames are chosen. Default is 0.

chroma

Set whether or not chroma is considered in the metric calculations. Default is 1.

deflate

Apply deflate effect to the video.

This filter replaces the pixel by the local(3x3) average by taking into account only values lower than the pixel.

It accepts the following options:

threshold0

threshold1

threshold2

threshold3

Limit the maximum change for each plane, default is 65535. If 0, plane will remain unchanged.

dejudder

Remove judder produced by partially interlaced telecined content.

Judder can be introduced, for instance, by pullup filter. If the original source was partially telecined content then the output of "pullup,dejudder" will have a variable frame rate. May change the recorded frame rate of the container. Aside from that change, this filter will not affect constant frame rate video.

The option available in this filter is:

cycle

Specify the length of the window over which the judder repeats.

Accepts any integer greater than 1. Useful values are:

4

If the original was telecined from 24 to 30 fps (Film to NTSC).

5

If the original was telecined from 25 to 30 fps (PAL to NTSC).

20

If a mixture of the two.

The default is 4.

delogo

Suppress a TV station logo by a simple interpolation of the surrounding pixels. Just set a rectangle covering the logo and watch it disappear (and sometimes something even uglier appear - your mileage may vary).

It accepts the following parameters:

x

y

Specify the top left corner coordinates of the logo. They must be specified.

w

h

Specify the width and height of the logo to clear. They must be specified.

band, t

Specify the thickness of the fuzzy edge of the rectangle (added to w and h). The default value is 1. This option is deprecated, setting higher values should no longer be necessary and is not recommended.

show

When set to 1, a green rectangle is drawn on the screen to simplify finding the right x, y, w, and h parameters. The default value is 0.

The rectangle is drawn on the outermost pixels which will be (partly) replaced with interpolated values. The values of the next pixels immediately outside this rectangle in each direction will be used to compute the interpolated pixel values inside the rectangle.

Examples

• Set a rectangle covering the area with top left corner coordinates 0,0 and size 100x77, and a band of size 10:

          delogo=x=0:y=0:w=100:h=77:band=10
  

deshake

Attempt to fix small changes in horizontal and/or vertical shift. This filter helps remove camera shake from hand-holding a camera, bumping a tripod, moving on a vehicle, etc.

The filter accepts the following options:

x

y

w

h

Specify a rectangular area where to limit the search for motion vectors. If desired the search for motion vectors can be limited to a rectangular area of the frame defined by its top left corner, width and height. These parameters have the same meaning as the drawbox filter which can be used to visualise the position of the bounding box.

This is useful when simultaneous movement of subjects within the frame might be confused for camera motion by the motion vector search.

If any or all of x, y, w and h are set to -1 then the full frame is used. This allows later options to be set without specifying the bounding box for the motion vector search.

Default - search the whole frame.

rx

ry

Specify the maximum extent of movement in x and y directions in the range 0-64 pixels. Default 16.

edge

Specify how to generate pixels to fill blanks at the edge of the frame. Available values are:

blank, 0

Fill zeroes at blank locations

original, 1

Original image at blank locations

clamp, 2

Extruded edge value at blank locations

mirror, 3

Mirrored edge at blank locations

Default value is mirror.

blocksize

Specify the blocksize to use for motion search. Range 4-128 pixels, default 8.

contrast

Specify the contrast threshold for blocks. Only blocks with more than the specified contrast (difference between darkest and lightest pixels) will be considered. Range 1-255, default 125.

search

Specify the search strategy. Available values are:

exhaustive, 0

Set exhaustive search

less, 1

Set less exhaustive search.

Default value is exhaustive.

filename

If set then a detailed log of the motion search is written to the specified file.

opencl

If set to 1, specify using OpenCL capabilities, only available if FFmpeg was configured with "--enable-opencl". Default value is 0.

detelecine

Apply an exact inverse of the telecine operation. It requires a predefined pattern specified using the pattern option which must be the same as that passed to the telecine filter.

This filter accepts the following options:

first_field

top, t

top field first

bottom, b

bottom field first The default value is "top".

pattern

A string of numbers representing the pulldown pattern you wish to apply. The default value is 23.

start_frame

A number representing position of the first frame with respect to the telecine pattern. This is to be used if the stream is cut. The default value is 0.

dilation

Apply dilation effect to the video.

This filter replaces the pixel by the local(3x3) maximum.

It accepts the following options:

threshold0

threshold1

threshold2

threshold3

Limit the maximum change for each plane, default is 65535. If 0, plane will remain unchanged.

coordinates

Flag which specifies the pixel to refer to. Default is 255 i.e. all eight pixels are used.

Flags to local 3x3 coordinates maps like this:

    1 2 3
    4   5
    6 7 8

displace

Displace pixels as indicated by second and third input stream.

It takes three input streams and outputs one stream, the first input is the source, and second and third input are displacement maps.

The second input specifies how much to displace pixels along the x-axis, while the third input specifies how much to displace pixels along the y-axis. If one of displacement map streams terminates, last frame from that displacement map will be used.

Note that once generated, displacements maps can be reused over and over again.

A description of the accepted options follows.

edge

Set displace behavior for pixels that are out of range.

Available values are:

blank

Missing pixels are replaced by black pixels.

smear

Adjacent pixels will spread out to replace missing pixels.

wrap

Out of range pixels are wrapped so they point to pixels of other side.

Default is smear.

Examples

• Add ripple effect to rgb input of video size hd720:

          ffmpeg -i INPUT -f lavfi -i nullsrc=s=hd720,lutrgb=128:128:128 -f lavfi -i nullsrc=s=hd720,geq='r=128+30*sin(2*PI*X/400+T):g=128+30*sin(2*PI*X/400+T):b=128+30*sin(2*PI*X/400+T)' -lavfi '[0][1][2]displace' OUTPUT
  

• Add wave effect to rgb input of video size hd720:

          ffmpeg -i INPUT -f lavfi -i nullsrc=hd720,geq='r=128+80*(sin(sqrt((X-W/2)*(X-W/2)+(Y-H/2)*(Y-H/2))/220*2*PI+T)):g=128+80*(sin(sqrt((X-W/2)*(X-W/2)+(Y-H/2)*(Y-H/2))/220*2*PI+T)):b=128+80*(sin(sqrt((X-W/2)*(X-W/2)+(Y-H/2)*(Y-H/2))/220*2*PI+T))' -lavfi '[1]split[x][y],[0][x][y]displace' OUTPUT
  

drawbox

Draw a colored box on the input image.

It accepts the following parameters:

x

y

The expressions which specify the top left corner coordinates of the box. It defaults to 0.

width, w

height, h

The expressions which specify the width and height of the box; if 0 they are interpreted as the input width and height. It defaults to 0.

color, c

Specify the color of the box to write. For the general syntax of this option, check the ``Color'' section in the ffmpeg-utils manual. If the special value "invert" is used, the box edge color is the same as the video with inverted luma.

thickness, t

The expression which sets the thickness of the box edge. Default value is 3.

See below for the list of accepted constants.

The parameters for x, y, w and h and t are expressions containing the following constants:

dar

The input display aspect ratio, it is the same as (w / h) * sar.

hsub

vsub

horizontal and vertical chroma subsample values. For example for the pixel format ``yuv422p'' hsub is 2 and vsub is 1.

in_h, ih

in_w, iw

The input width and height.

sar

The input sample aspect ratio.

x

y

The x and y offset coordinates where the box is drawn.

w

h

The width and height of the drawn box.

t

The thickness of the drawn box.

These constants allow the x, y, w, h and t expressions to refer to each other, so you may for example specify "y=x/dar" or "h=w/dar".

Examples

• Draw a black box around the edge of the input image:

          drawbox
  

• Draw a box with color red and an opacity of 50%:

          drawbox=10:20:200:60:[email protected]
  

  The previous example can be specified as:

          drawbox=x=10:y=20:w=200:h=60:[email protected]
  

• Fill the box with pink color:

          drawbox=x=10:y=10:w=100:h=100:[email protected]:t=max
  

• Draw a 2-pixel red 2.40:1 mask:

          drawbox=x=-t:y=0.5*(ih-iw/2.4)-t:w=iw+t*2:h=iw/2.4+t*2:t=2:c=red
  

drawgraph, adrawgraph

Draw a graph using input video or audio metadata.

It accepts the following parameters:

m1

Set 1st frame metadata key from which metadata values will be used to draw a graph.

fg1

Set 1st foreground color expression.

m2

Set 2nd frame metadata key from which metadata values will be used to draw a graph.

fg2

Set 2nd foreground color expression.

m3

Set 3rd frame metadata key from which metadata values will be used to draw a graph.

fg3

Set 3rd foreground color expression.

m4

Set 4th frame metadata key from which metadata values will be used to draw a graph.

fg4

Set 4th foreground color expression.

min

Set minimal value of metadata value.

max

Set maximal value of metadata value.

bg

Set graph background color. Default is white.

mode

Set graph mode.

Available values for mode is:

bar

dot

line

Default is "line".

slide

Set slide mode.

Available values for slide is:

frame

Draw new frame when right border is reached.

replace

Replace old columns with new ones.

scroll

Scroll from right to left.

rscroll

Scroll from left to right.

Default is "frame".

size

Set size of graph video. For the syntax of this option, check the ``Video size'' section in the ffmpeg-utils manual. The default value is "900x256".

The foreground color expressions can use the following variables:

MIN

Minimal value of metadata value.

MAX

Maximal value of metadata value.

VAL

Current metadata key value.

The color is defined as 0xAABBGGRR.

Example using metadata from signalstats filter:

        signalstats,drawgraph=lavfi.signalstats.YAVG:min=0:max=255

Example using metadata from ebur128 filter:

        ebur128=metadata=1,adrawgraph=lavfi.r128.M:min=-120:max=5

drawgrid

Draw a grid on the input image.

It accepts the following parameters:

x

y

The expressions which specify the coordinates of some point of grid intersection (meant to configure offset). Both default to 0.

width, w

height, h

The expressions which specify the width and height of the grid cell, if 0 they are interpreted as the input width and height, respectively, minus "thickness", so image gets framed. Default to 0.

color, c

Specify the color of the grid. For the general syntax of this option, check the ``Color'' section in the ffmpeg-utils manual. If the special value "invert" is used, the grid color is the same as the video with inverted luma.

thickness, t

The expression which sets the thickness of the grid line. Default value is 1.

See below for the list of accepted constants.

The parameters for x, y, w and h and t are expressions containing the following constants:

dar

The input display aspect ratio, it is the same as (w / h) * sar.

hsub

vsub

horizontal and vertical chroma subsample values. For example for the pixel format ``yuv422p'' hsub is 2 and vsub is 1.

in_h, ih

in_w, iw

The input grid cell width and height.

sar

The input sample aspect ratio.

x

y

The x and y coordinates of some point of grid intersection (meant to configure offset).

w

h

The width and height of the drawn cell.

t

The thickness of the drawn cell.

These constants allow the x, y, w, h and t expressions to refer to each other, so you may for example specify "y=x/dar" or "h=w/dar".

Examples

• Draw a grid with cell 100x100 pixels, thickness 2 pixels, with color red and an opacity of 50%:

          drawgrid=width=100:height=100:thickness=2:[email protected]
  

• Draw a white 3x3 grid with an opacity of 50%:

          drawgrid=w=iw/3:h=ih/3:t=2:[email protected]
  

drawtext

Draw a text string or text from a specified file on top of a video, using the libfreetype library.

To enable compilation of this filter, you need to configure FFmpeg with "--enable-libfreetype". To enable default font fallback and the font option you need to configure FFmpeg with "--enable-libfontconfig". To enable the text_shaping option, you need to configure FFmpeg with "--enable-libfribidi".

Syntax

It accepts the following parameters:

box

Used to draw a box around text using the background color. The value must be either 1 (enable) or 0 (disable). The default value of box is 0.

boxborderw

Set the width of the border to be drawn around the box using boxcolor. The default value of boxborderw is 0.

boxcolor

The color to be used for drawing box around text. For the syntax of this option, check the ``Color'' section in the ffmpeg-utils manual.

The default value of boxcolor is ``white''.

borderw

Set the width of the border to be drawn around the text using bordercolor. The default value of borderw is 0.

bordercolor

Set the color to be used for drawing border around text. For the syntax of this option, check the ``Color'' section in the ffmpeg-utils manual.

The default value of bordercolor is ``black''.

expansion

Select how the text is expanded. Can be either "none", "strftime" (deprecated) or "normal" (default). See the drawtext_expansion, Text expansion section below for details.

fix_bounds

If true, check and fix text coords to avoid clipping.

fontcolor

The color to be used for drawing fonts. For the syntax of this option, check the ``Color'' section in the ffmpeg-utils manual.

The default value of fontcolor is ``black''.

fontcolor_expr

String which is expanded the same way as text to obtain dynamic fontcolor value. By default this option has empty value and is not processed. When this option is set, it overrides fontcolor option.

font

The font family to be used for drawing text. By default Sans.

fontfile

The font file to be used for drawing text. The path must be included. This parameter is mandatory if the fontconfig support is disabled.

draw

This option does not exist, please see the timeline system

alpha

Draw the text applying alpha blending. The value can be either a number between 0.0 and 1.0 The expression accepts the same variables x, y do. The default value is 1. Please see fontcolor_expr

fontsize

The font size to be used for drawing text. The default value of fontsize is 16.

text_shaping

If set to 1, attempt to shape the text (for example, reverse the order of right-to-left text and join Arabic characters) before drawing it. Otherwise, just draw the text exactly as given. By default 1 (if supported).

ft_load_flags

The flags to be used for loading the fonts.

The flags map the corresponding flags supported by libfreetype, and are a combination of the following values:

default

no_scale

no_hinting

render

no_bitmap

vertical_layout

force_autohint

crop_bitmap

pedantic

ignore_global_advance_width

no_recurse

ignore_transform

monochrome

linear_design

no_autohint

Default value is ``default''.

For more information consult the documentation for the FT_LOAD_* libfreetype flags.

shadowcolor

The color to be used for drawing a shadow behind the drawn text. For the syntax of this option, check the ``Color'' section in the ffmpeg-utils manual.

The default value of shadowcolor is ``black''.

shadowx

shadowy

The x and y offsets for the text shadow position with respect to the position of the text. They can be either positive or negative values. The default value for both is ``0''.

start_number

The starting frame number for the n/frame_num variable. The default value is ``0''.

tabsize

The size in number of spaces to use for rendering the tab. Default value is 4.

timecode

Set the initial timecode representation in ``hh:mm:ss[:;.]ff'' format. It can be used with or without text parameter. timecode_rate option must be specified.

timecode_rate, rate, r

Set the timecode frame rate (timecode only).

text

The text string to be drawn. The text must be a sequence of UTF-8 encoded characters. This parameter is mandatory if no file is specified with the parameter textfile.

textfile

A text file containing text to be drawn. The text must be a sequence of UTF-8 encoded characters.

This parameter is mandatory if no text string is specified with the parameter text.

If both text and textfile are specified, an error is thrown.

reload

If set to 1, the textfile will be reloaded before each frame. Be sure to update it atomically, or it may be read partially, or even fail.

x

y

The expressions which specify the offsets where text will be drawn within the video frame. They are relative to the top/left border of the output image.

The default value of x and y is ``0''.

See below for the list of accepted constants and functions.

The parameters for x and y are expressions containing the following constants and functions:

dar

input display aspect ratio, it is the same as (w / h) * sar

hsub

vsub

horizontal and vertical chroma subsample values. For example for the pixel format ``yuv422p'' hsub is 2 and vsub is 1.

line_h, lh

the height of each text line

main_h, h, H

the input height

main_w, w, W

the input width

max_glyph_a, ascent

the maximum distance from the baseline to the highest/upper grid coordinate used to place a glyph outline point, for all the rendered glyphs. It is a positive value, due to the grid's orientation with the Y axis upwards.

max_glyph_d, descent

the maximum distance from the baseline to the lowest grid coordinate used to place a glyph outline point, for all the rendered glyphs. This is a negative value, due to the grid's orientation, with the Y axis upwards.

max_glyph_h

maximum glyph height, that is the maximum height for all the glyphs contained in the rendered text, it is equivalent to ascent - descent.

max_glyph_w

maximum glyph width, that is the maximum width for all the glyphs contained in the rendered text

n

the number of input frame, starting from 0

rand(min, max)

return a random number included between min and max

sar

The input sample aspect ratio.

t

timestamp expressed in seconds, NAN if the input timestamp is unknown

text_h, th

the height of the rendered text

text_w, tw

the width of the rendered text

x

y

the x and y offset coordinates where the text is drawn.

These parameters allow the x and y expressions to refer each other, so you can for example specify "y=x/dar".

Text expansion

If expansion is set to "strftime", the filter recognizes strftime() sequences in the provided text and expands them accordingly. Check the documentation of strftime(). This feature is deprecated.

If expansion is set to "none", the text is printed verbatim.

If expansion is set to "normal" (which is the default), the following expansion mechanism is used.

The backslash character \, followed by any character, always expands to the second character.

Sequence of the form "%{...}" are expanded. The text between the braces is a function name, possibly followed by arguments separated by ':'. If the arguments contain special characters or delimiters (':' or '}'), they should be escaped.

Note that they probably must also be escaped as the value for the text option in the filter argument string and as the filter argument in the filtergraph description, and possibly also for the shell, that makes up to four levels of escaping; using a text file avoids these problems.

The following functions are available:

expr, e

The expression evaluation result.

It must take one argument specifying the expression to be evaluated, which accepts the same constants and functions as the x and y values. Note that not all constants should be used, for example the text size is not known when evaluating the expression, so the constants text_w and text_h will have an undefined value.

expr_int_format, eif

Evaluate the expression's value and output as formatted integer.

The first argument is the expression to be evaluated, just as for the expr function. The second argument specifies the output format. Allowed values are x, X, d and u. They are treated exactly as in the "printf" function. The third parameter is optional and sets the number of positions taken by the output. It can be used to add padding with zeros from the left.

gmtime

The time at which the filter is running, expressed in UTC. It can accept an argument: a strftime() format string.

localtime

The time at which the filter is running, expressed in the local time zone. It can accept an argument: a strftime() format string.

metadata

Frame metadata. It must take one argument specifying metadata key.

n, frame_num

The frame number, starting from 0.

pict_type

A 1 character description of the current picture type.

pts

The timestamp of the current frame. It can take up to three arguments.

The first argument is the format of the timestamp; it defaults to "flt" for seconds as a decimal number with microsecond accuracy; "hms" stands for a formatted [-]HH:MM:SS.mmm timestamp with millisecond accuracy. "gmtime" stands for the timestamp of the frame formatted as UTC time; "localtime" stands for the timestamp of the frame formatted as local time zone time.

The second argument is an offset added to the timestamp.

If the format is set to "localtime" or "gmtime", a third argument may be supplied: a strftime() format string. By default, YYYY-MM-DD HH:MM:SS format will be used.

Examples

• Draw ``Test Text'' with font FreeSerif, using the default values for the optional parameters.

          drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf: text='Test Text'"
  

• Draw 'Test Text' with font FreeSerif of size 24 at position x=100 and y=50 (counting from the top-left corner of the screen), text is yellow with a red box around it. Both the text and the box have an opacity of 20%.

          drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf: text='Test Text':\
                    x=100: y=50: fontsize=24: [email protected]: box=1: [email protected]"
  

  Note that the double quotes are not necessary if spaces are not used within the parameter list.

• Show the text at the center of the video frame:

          drawtext="fontsize=30:fontfile=FreeSerif.ttf:text='hello world':x=(w-text_w)/2:y=(h-text_h)/2"
  

• Show a text line sliding from right to left in the last row of the video frame. The file LONG_LINE is assumed to contain a single line with no newlines.

          drawtext="fontsize=15:fontfile=FreeSerif.ttf:text=LONG_LINE:y=h-line_h:x=-50*t"
  

• Show the content of file CREDITS off the bottom of the frame and scroll up.

          drawtext="fontsize=20:fontfile=FreeSerif.ttf:textfile=CREDITS:y=h-20*t"
  

• Draw a single green letter ``g'', at the center of the input video. The glyph baseline is placed at half screen height.

          drawtext="fontsize=60:fontfile=FreeSerif.ttf:fontcolor=green:text=g:x=(w-max_glyph_w)/2:y=h/2-ascent"
  

• Show text for 1 second every 3 seconds:

          drawtext="fontfile=FreeSerif.ttf:fontcolor=white:x=100:y=x/dar:enable=lt(mod(t\,3)\,1):text='blink'"
  

• Use fontconfig to set the font. Note that the colons need to be escaped.

          drawtext='fontfile=Linux Libertine O-40\:style=Semibold:text=FFmpeg'
  

• Print the date of a real-time encoding (see strftime(3)):

          drawtext='fontfile=FreeSans.ttf:text=%{localtime\:%a %b %d %Y}'
  

• Show text fading in and out (appearing/disappearing):

          #!/bin/sh
          DS=1.0 # display start
          DE=10.0 # display end
          FID=1.5 # fade in duration
          FOD=5 # fade out duration
          ffplay -f lavfi "color,drawtext=text=TEST:fontsize=50:fontfile=FreeSerif.ttf:fontcolor_expr=ff0000%{eif\\\\: clip(255*(1*between(t\\, $DS + $FID\\, $DE - $FOD) + ((t - $DS)/$FID)*between(t\\, $DS\\, $DS + $FID) + (-(t - $DE)/$FOD)*between(t\\, $DE - $FOD\\, $DE) )\\, 0\\, 255) \\\\: x\\\\: 2 }"
  

For more information about libfreetype, check: <http://www.freetype.org/>.

For more information about fontconfig, check: <http://freedesktop.org/software/fontconfig/fontconfig-user.html>.

For more information about libfribidi, check: <http://fribidi.org/>.

edgedetect

Detect and draw edges. The filter uses the Canny Edge Detection algorithm.

The filter accepts the following options:

low

high

Set low and high threshold values used by the Canny thresholding algorithm.

The high threshold selects the ``strong'' edge pixels, which are then connected through 8-connectivity with the ``weak'' edge pixels selected by the low threshold.

low and high threshold values must be chosen in the range [0,1], and low should be lesser or equal to high.

Default value for low is "20/255", and default value for high is "50/255".

mode

Define the drawing mode.

wires

Draw white/gray wires on black background.

colormix

Mix the colors to create a paint/cartoon effect.

Default value is wires.

Examples

• Standard edge detection with custom values for the hysteresis thresholding:

          edgedetect=low=0.1:high=0.4
  

• Painting effect without thresholding:

          edgedetect=mode=colormix:high=0
  

eq

Set brightness, contrast, saturation and approximate gamma adjustment.

The filter accepts the following options:

contrast

Set the contrast expression. The value must be a float value in range "-2.0" to 2.0. The default value is ``1''.

brightness

Set the brightness expression. The value must be a float value in range "-1.0" to 1.0. The default value is ``0''.

saturation

Set the saturation expression. The value must be a float in range 0.0 to 3.0. The default value is ``1''.

gamma

Set the gamma expression. The value must be a float in range 0.1 to 10.0. The default value is ``1''.

gamma_r

Set the gamma expression for red. The value must be a float in range 0.1 to 10.0. The default value is ``1''.

gamma_g

Set the gamma expression for green. The value must be a float in range 0.1 to 10.0. The default value is ``1''.

gamma_b

Set the gamma expression for blue. The value must be a float in range 0.1 to 10.0. The default value is ``1''.

gamma_weight

Set the gamma weight expression. It can be used to reduce the effect of a high gamma value on bright image areas, e.g. keep them from getting overamplified and just plain white. The value must be a float in range 0.0 to 1.0. A value of 0.0 turns the gamma correction all the way down while 1.0 leaves it at its full strength. Default is ``1''.

eval

Set when the expressions for brightness, contrast, saturation and gamma expressions are evaluated.

It accepts the following values:

init

only evaluate expressions once during the filter initialization or when a command is processed

frame

evaluate expressions for each incoming frame

Default value is init.

The expressions accept the following parameters:

n

frame count of the input frame starting from 0

pos

byte position of the corresponding packet in the input file, NAN if unspecified

r

frame rate of the input video, NAN if the input frame rate is unknown

t

timestamp expressed in seconds, NAN if the input timestamp is unknown

Commands

The filter supports the following commands:

contrast

Set the contrast expression.

brightness

Set the brightness expression.

saturation

Set the saturation expression.

gamma

Set the gamma expression.

gamma_r

Set the gamma_r expression.

gamma_g

Set gamma_g expression.

gamma_b

Set gamma_b expression.

gamma_weight

Set gamma_weight 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.

erosion

Apply erosion effect to the video.

This filter replaces the pixel by the local(3x3) minimum.

It accepts the following options:

threshold0

threshold1

threshold2

threshold3

Limit the maximum change for each plane, default is 65535. If 0, plane will remain unchanged.

coordinates

Flag which specifies the pixel to refer to. Default is 255 i.e. all eight pixels are used.

Flags to local 3x3 coordinates maps like this:

    1 2 3
    4   5
    6 7 8

extractplanes

Extract color channel components from input video stream into separate grayscale video streams.

The filter accepts the following option:

planes

Set plane(s) to extract.

Available values for planes are:

y

u

v

a

r

g

b

Choosing planes not available in the input will result in an error. That means you cannot select "r", "g", "b" planes with "y", "u", "v" planes at same time.

Examples

• Extract luma, u and v color channel component from input video frame into 3 grayscale outputs:

          ffmpeg -i video.avi -filter_complex 'extractplanes=y+u+v[y][u][v]' -map '[y]' y.avi -map '[u]' u.avi -map '[v]' v.avi
  

elbg

Apply a posterize effect using the ELBG (Enhanced LBG) algorithm.

For each input image, the filter will compute the optimal mapping from the input to the output given the codebook length, that is the number of distinct output colors.

This filter accepts the following options.

codebook_length, l

Set codebook length. The value must be a positive integer, and represents the number of distinct output colors. Default value is 256.

nb_steps, n

Set the maximum number of iterations to apply for computing the optimal mapping. The higher the value the better the result and the higher the computation time. Default value is 1.

seed, s

Set a random seed, must be an integer included between 0 and UINT32_MAX. If not specified, or if explicitly set to -1, the filter will try to use a good random seed on a best effort basis.

pal8

Set pal8 output pixel format. This option does not work with codebook length greater than 256.

fade

Apply a fade-in/out effect to the input video.

It accepts the following parameters:

type, t

The effect type can be either ``in'' for a fade-in, or ``out'' for a fade-out effect. Default is "in".

start_frame, s

Specify the number of the frame to start applying the fade effect at. Default is 0.

nb_frames, n

The number of frames that the fade effect lasts. At the end of the fade-in effect, the output video will have the same intensity as the input video. At the end of the fade-out transition, the output video will be filled with the selected color. Default is 25.

alpha

If set to 1, fade only alpha channel, if one exists on the input. Default value is 0.

start_time, st

Specify the timestamp (in seconds) of the frame to start to apply the fade effect. If both start_frame and start_time are specified, the fade will start at whichever comes last. Default is 0.

duration, d

The number of seconds for which the fade effect has to last. At the end of the fade-in effect the output video will have the same intensity as the input video, at the end of the fade-out transition the output video will be filled with the selected color. If both duration and nb_frames are specified, duration is used. Default is 0 (nb_frames is used by default).

color, c

Specify the color of the fade. Default is ``black''.

Examples

• Fade in the first 30 frames of video:

          fade=in:0:30
  

  The command above is equivalent to:

          fade=t=in:s=0:n=30
  

• Fade out the last 45 frames of a 200-frame video:

          fade=out:155:45
          fade=type=out:start_frame=155:nb_frames=45
  

• Fade in the first 25 frames and fade out the last 25 frames of a 1000-frame video:

          fade=in:0:25, fade=out:975:25
  

• Make the first 5 frames yellow, then fade in from frame 5-24:

          fade=in:5:20:color=yellow
  

• Fade in alpha over first 25 frames of video:

          fade=in:0:25:alpha=1
  

• Make the first 5.5 seconds black, then fade in for 0.5 seconds:

          fade=t=in:st=5.5:d=0.5
  

fftfilt

Apply arbitrary expressions to samples in frequency domain

dc_Y

Adjust the dc value (gain) of the luma plane of the image. The filter accepts an integer value in range 0 to 1000. The default value is set to 0.

dc_U

Adjust the dc value (gain) of the 1st chroma plane of the image. The filter accepts an integer value in range 0 to 1000. The default value is set to 0.

dc_V

Adjust the dc value (gain) of the 2nd chroma plane of the image. The filter accepts an integer value in range 0 to 1000. The default value is set to 0.

weight_Y

Set the frequency domain weight expression for the luma plane.

weight_U

Set the frequency domain weight expression for the 1st chroma plane.

weight_V

Set the frequency domain weight expression for the 2nd chroma plane.

The filter accepts the following variables:

X

Y

The coordinates of the current sample.

W

H

The width and height of the image.

Examples

• High-pass:

          fftfilt=dc_Y=128:weight_Y='squish(1-(Y+X)/100)'
  

• Low-pass:

          fftfilt=dc_Y=0:weight_Y='squish((Y+X)/100-1)'
  

• Sharpen:

          fftfilt=dc_Y=0:weight_Y='1+squish(1-(Y+X)/100)'
  

• Blur:

          fftfilt=dc_Y=0:weight_Y='exp(-4 * ((Y+X)/(W+H)))'
  

field

Extract a single field from an interlaced image using stride arithmetic to avoid wasting CPU time. The output frames are marked as non-interlaced.

The filter accepts the following options:

type

Specify whether to extract the top (if the value is 0 or "top") or the bottom field (if the value is 1 or "bottom").

fieldmatch

Field matching filter for inverse telecine. It is meant to reconstruct the progressive frames from a telecined stream. The filter does not drop duplicated frames, so to achieve a complete inverse telecine "fieldmatch" needs to be followed by a decimation filter such as decimate in the filtergraph.

The separation of the field matching and the decimation is notably motivated by the possibility of inserting a de-interlacing filter fallback between the two. If the source has mixed telecined and real interlaced content, "fieldmatch" will not be able to match fields for the interlaced parts. But these remaining combed frames will be marked as interlaced, and thus can be de-interlaced by a later filter such as yadif before decimation.

In addition to the various configuration options, "fieldmatch" can take an optional second stream, activated through the ppsrc option. If enabled, the frames reconstruction will be based on the fields and frames from this second stream. This allows the first input to be pre-processed in order to help the various algorithms of the filter, while keeping the output lossless (assuming the fields are matched properly). Typically, a field-aware denoiser, or brightness/contrast adjustments can help.

Note that this filter uses the same algorithms as TIVTC/TFM (AviSynth project) and VIVTC/VFM (VapourSynth project). The later is a light clone of TFM from which "fieldmatch" is based on. While the semantic and usage are very close, some behaviour and options names can differ.

The decimate filter currently only works for constant frame rate input. If your input has mixed telecined (30fps) and progressive content with a lower framerate like 24fps use the following filterchain to produce the necessary cfr stream: "dejudder,fps=30000/1001,fieldmatch,decimate".

The filter accepts the following options:

order

Specify the assumed field order of the input stream. Available values are:

auto

Auto detect parity (use FFmpeg's internal parity value).

bff

Assume bottom field first.

tff

Assume top field first.

Note that it is sometimes recommended not to trust the parity announced by the stream.

Default value is auto.

mode

Set the matching mode or strategy to use. pc mode is the safest in the sense that it won't risk creating jerkiness due to duplicate frames when possible, but if there are bad edits or blended fields it will end up outputting combed frames when a good match might actually exist. On the other hand, pcn_ub mode is the most risky in terms of creating jerkiness, but will almost always find a good frame if there is one. The other values are all somewhere in between pc and pcn_ub in terms of risking jerkiness and creating duplicate frames versus finding good matches in sections with bad edits, orphaned fields, blended fields, etc.

More details about p/c/n/u/b are available in p/c/n/u/b meaning section.

Available values are:

pc

2-way matching (p/c)

pc_n

2-way matching, and trying 3rd match if still combed (p/c + n)

pc_u

2-way matching, and trying 3rd match (same order) if still combed (p/c + u)

pc_n_ub

2-way matching, trying 3rd match if still combed, and trying 4th/5th matches if still combed (p/c + n + u/b)

pcn

3-way matching (p/c/n)

pcn_ub

3-way matching, and trying 4th/5th matches if all 3 of the original matches are detected as combed (p/c/n + u/b)

The parenthesis at the end indicate the matches that would be used for that mode assuming order=tff (and field on auto or top).

In terms of speed pc mode is by far the fastest and pcn_ub is the slowest.

Default value is pc_n.

ppsrc

Mark the main input stream as a pre-processed input, and enable the secondary input stream as the clean source to pick the fields from. See the filter introduction for more details. It is similar to the clip2 feature from VFM/TFM.

Default value is 0 (disabled).

field

Set the field to match from. It is recommended to set this to the same value as order unless you experience matching failures with that setting. In certain circumstances changing the field that is used to match from can have a large impact on matching performance. Available values are:

auto

Automatic (same value as order).

bottom

Match from the bottom field.

top

Match from the top field.

Default value is auto.

mchroma

Set whether or not chroma is included during the match comparisons. In most cases it is recommended to leave this enabled. You should set this to 0 only if your clip has bad chroma problems such as heavy rainbowing or other artifacts. Setting this to 0 could also be used to speed things up at the cost of some accuracy.

Default value is 1.

y0

y1

These define an exclusion band which excludes the lines between y0 and y1 from being included in the field matching decision. An exclusion band can be used to ignore subtitles, a logo, or other things that may interfere with the matching. y0 sets the starting scan line and y1 sets the ending line; all lines in between y0 and y1 (including y0 and y1) will be ignored. Setting y0 and y1 to the same value will disable the feature. y0 and y1 defaults to 0.

scthresh

Set the scene change detection threshold as a percentage of maximum change on the luma plane. Good values are in the "[8.0, 14.0]" range. Scene change detection is only relevant in case combmatch=sc. The range for scthresh is "[0.0, 100.0]".

Default value is 12.0.

combmatch

When combatch is not none, "fieldmatch" will take into account the combed scores of matches when deciding what match to use as the final match. Available values are:

none

No final matching based on combed scores.

sc

Combed scores are only used when a scene change is detected.

full

Use combed scores all the time.

Default is sc.

combdbg

Force "fieldmatch" to calculate the combed metrics for certain matches and print them. This setting is known as micout in TFM/VFM vocabulary. Available values are:

none

No forced calculation.

pcn

Force p/c/n calculations.

pcnub

Force p/c/n/u/b calculations.

Default value is none.

cthresh

This is the area combing threshold used for combed frame detection. This essentially controls how ``strong'' or ``visible'' combing must be to be detected. Larger values mean combing must be more visible and smaller values mean combing can be less visible or strong and still be detected. Valid settings are from "-1" (every pixel will be detected as combed) to 255 (no pixel will be detected as combed). This is basically a pixel difference value. A good range is "[8, 12]".

Default value is 9.

chroma

Sets whether or not chroma is considered in the combed frame decision. Only disable this if your source has chroma problems (rainbowing, etc.) that are causing problems for the combed frame detection with chroma enabled. Actually, using chroma=0 is usually more reliable, except for the case where there is chroma only combing in the source.

Default value is 0.

blockx

blocky

Respectively set the x-axis and y-axis size of the window used during combed frame detection. This has to do with the size of the area in which combpel pixels are required to be detected as combed for a frame to be declared combed. See the combpel parameter description for more info. Possible values are any number that is a power of 2 starting at 4 and going up to 512.

Default value is 16.

combpel

The number of combed pixels inside any of the blocky by blockx size blocks on the frame for the frame to be detected as combed. While cthresh controls how ``visible'' the combing must be, this setting controls ``how much'' combing there must be in any localized area (a window defined by the blockx and blocky settings) on the frame. Minimum value is 0 and maximum is "blocky x blockx" (at which point no frames will ever be detected as combed). This setting is known as MI in TFM/VFM vocabulary.

Default value is 80.

p/c/n/u/b meaning

p/c/n

We assume the following telecined stream:

        Top fields:     1 2 2 3 4
        Bottom fields:  1 2 3 4 4

The numbers correspond to the progressive frame the fields relate to. Here, the first two frames are progressive, the 3rd and 4th are combed, and so on.

When "fieldmatch" is configured to run a matching from bottom (field=bottom) this is how this input stream get transformed:

        Input stream:
                        T     1 2 2 3 4
                        B     1 2 3 4 4   <-- matching reference
        
        Matches:              c c n n c
        
        Output stream:
                        T     1 2 3 4 4
                        B     1 2 3 4 4

As a result of the field matching, we can see that some frames get duplicated. To perform a complete inverse telecine, you need to rely on a decimation filter after this operation. See for instance the decimate filter.

The same operation now matching from top fields (field=top) looks like this:

        Input stream:
                        T     1 2 2 3 4   <-- matching reference
                        B     1 2 3 4 4
        
        Matches:              c c p p c
        
        Output stream:
                        T     1 2 2 3 4
                        B     1 2 2 3 4

In these examples, we can see what p, c and n mean; basically, they refer to the frame and field of the opposite parity:

*<p matches the field of the opposite parity in the previous frame>

*<c matches the field of the opposite parity in the current frame>

*<n matches the field of the opposite parity in the next frame>

u/b

The u and b matching are a bit special in the sense that they match from the opposite parity flag. In the following examples, we assume that we are currently matching the 2nd frame (Top:2, bottom:2). According to the match, a 'x' is placed above and below each matched fields.

With bottom matching (field=bottom):

        Match:           c         p           n          b          u
        
                         x       x               x        x          x
          Top          1 2 2     1 2 2       1 2 2      1 2 2      1 2 2
          Bottom       1 2 3     1 2 3       1 2 3      1 2 3      1 2 3
                         x         x           x        x              x
        
        Output frames:
                         2          1          2          2          2
                         2          2          2          1          3

With top matching (field=top):

        Match:           c         p           n          b          u
        
                         x         x           x        x              x
          Top          1 2 2     1 2 2       1 2 2      1 2 2      1 2 2
          Bottom       1 2 3     1 2 3       1 2 3      1 2 3      1 2 3
                         x       x               x        x          x
        
        Output frames:
                         2          2          2          1          2
                         2          1          3          2          2

Examples

Simple IVTC of a top field first telecined stream:

        fieldmatch=order=tff:combmatch=none, decimate

Advanced IVTC, with fallback on yadif for still combed frames:

        fieldmatch=order=tff:combmatch=full, yadif=deint=interlaced, decimate

fieldorder

Transform the field order of the input video.

It accepts the following parameters:

order

The output field order. Valid values are tff for top field first or bff for bottom field first.

The default value is tff.

The transformation is done by shifting the picture content up or down by one line, and filling the remaining line with appropriate picture content. This method is consistent with most broadcast field order converters.

If the input video is not flagged as being interlaced, or it is already flagged as being of the required output field order, then this filter does not alter the incoming video.

It is very useful when converting to or from PAL DV material, which is bottom field first.

For example:

        ffmpeg -i in.vob -vf "fieldorder=bff" out.dv

fifo, afifo

Buffer input images and send them when they are requested.

It is mainly useful when auto-inserted by the libavfilter framework.

It does not take parameters.

find_rect

Find a rectangular object

It accepts the following options:

object

Filepath of the object image, needs to be in gray8.

threshold

Detection threshold, default is 0.5.

mipmaps

Number of mipmaps, default is 3.

xmin, ymin, xmax, ymax

Specifies the rectangle in which to search.

Examples

• Generate a representative palette of a given video using ffmpeg:

          ffmpeg -i file.ts -vf find_rect=newref.pgm,cover_rect=cover.jpg:mode=cover new.mkv
  

cover_rect

Cover a rectangular object

It accepts the following options:

cover

Filepath of the optional cover image, needs to be in yuv420.

mode

Set covering mode.

It accepts the following values:

cover

cover it by the supplied image

blur

cover it by interpolating the surrounding pixels

Default value is blur.

Examples

• Generate a representative palette of a given video using ffmpeg:

          ffmpeg -i file.ts -vf find_rect=newref.pgm,cover_rect=cover.jpg:mode=cover new.mkv
  

format

Convert the input video to one of the specified pixel formats. Libavfilter will try to pick one that is suitable as input to the next filter.

It accepts the following parameters:

pix_fmts

A '|'-separated list of pixel format names, such as ``pix_fmts=yuv420p|monow|rgb24''.

Examples

• Convert the input video to the yuv420p format

          format=pix_fmts=yuv420p
  

  Convert the input video to any of the formats in the list

          format=pix_fmts=yuv420p|yuv444p|yuv410p
  

fps

Convert the video to specified constant frame rate by duplicating or dropping frames as necessary.

It accepts the following parameters:

fps

The desired output frame rate. The default is 25.

round

Rounding method.

Possible values are:

zero

zero round towards 0

inf

round away from 0

down

round towards -infinity

up

round towards +infinity

near

round to nearest

The default is "near".

start_time

Assume the first PTS should be the given value, in seconds. This allows for padding/trimming at the start of stream. By default, no assumption is made about the first frame's expected PTS, so no padding or trimming is done. For example, this could be set to 0 to pad the beginning with duplicates of the first frame if a video stream starts after the audio stream or to trim any frames with a negative PTS.

Alternatively, the options can be specified as a flat string: fps[:round].

See also the setpts filter.

Examples

• A typical usage in order to set the fps to 25:

          fps=fps=25
  

• Sets the fps to 24, using abbreviation and rounding method to round to nearest:

          fps=fps=film:round=near
  

framepack

Pack two different video streams into a stereoscopic video, setting proper metadata on supported codecs. The two views should have the same size and framerate and processing will stop when the shorter video ends. Please note that you may conveniently adjust view properties with the scale and fps filters.

It accepts the following parameters:

format

The desired packing format. Supported values are:

sbs

The views are next to each other (default).

tab

The views are on top of each other.

lines

The views are packed by line.

columns

The views are packed by column.

frameseq

The views are temporally interleaved.

Some examples:

        # Convert left and right views into a frame-sequential video
        ffmpeg -i LEFT -i RIGHT -filter_complex framepack=frameseq OUTPUT
        
        # Convert views into a side-by-side video with the same output resolution as the input
        ffmpeg -i LEFT -i RIGHT -filter_complex [0:v]scale=w=iw/2[left],[1:v]scale=w=iw/2[right],[left][right]framepack=sbs OUTPUT

framerate

Change the frame rate by interpolating new video output frames from the source frames.

This filter is not designed to function correctly with interlaced media. If you wish to change the frame rate of interlaced media then you are required to deinterlace before this filter and re-interlace after this filter.

A description of the accepted options follows.

fps

Specify the output frames per second. This option can also be specified as a value alone. The default is 50.

interp_start

Specify the start of a range where the output frame will be created as a linear interpolation of two frames. The range is [0-255], the default is 15.

interp_end

Specify the end of a range where the output frame will be created as a linear interpolation of two frames. The range is [0-255], the default is 240.

scene

Specify the level at which a scene change is detected as a value between 0 and 100 to indicate a new scene; a low value reflects a low probability for the current frame to introduce a new scene, while a higher value means the current frame is more likely to be one. The default is 7.

flags

Specify flags influencing the filter process.

Available value for flags is:

scene_change_detect, scd

Enable scene change detection using the value of the option scene. This flag is enabled by default.

framestep

Select one frame every N-th frame.

This filter accepts the following option:

step

Select frame after every "step" frames. Allowed values are positive integers higher than 0. Default value is 1.

frei0r

Apply a frei0r effect to the input video.

To enable the compilation of this filter, you need to install the frei0r header and configure FFmpeg with "--enable-frei0r".

It accepts the following parameters:

filter_name

The name of the frei0r effect to load. If the environment variable FREI0R_PATH is defined, the frei0r effect is searched for in each of the directories specified by the colon-separated list in FREIOR_PATH. Otherwise, the standard frei0r paths are searched, in this order: HOME/.frei0r-1/lib/, /usr/local/lib/frei0r-1/, /usr/lib/frei0r-1/.

filter_params

A '|'-separated list of parameters to pass to the frei0r effect.

A frei0r effect parameter can be a boolean (its value is either ``y'' or ``n''), a double, a color (specified as R/G/B, where R, G, and B are floating point numbers between 0.0 and 1.0, inclusive) or by a color description specified in the ``Color'' section in the ffmpeg-utils manual), a position (specified as X/Y, where X and Y are floating point numbers) and/or a string.

The number and types of parameters depend on the loaded effect. If an effect parameter is not specified, the default value is set.

Examples

• Apply the distort0r effect, setting the first two double parameters:

          frei0r=filter_name=distort0r:filter_params=0.5|0.01
  

• Apply the colordistance effect, taking a color as the first parameter:

          frei0r=colordistance:0.2/0.3/0.4
          frei0r=colordistance:violet
          frei0r=colordistance:0x112233
  

• Apply the perspective effect, specifying the top left and top right image positions:

          frei0r=perspective:0.2/0.2|0.8/0.2
  

For more information, see <http://frei0r.dyne.org>

fspp

Apply fast and simple postprocessing. It is a faster version of spp.

It splits (I)DCT into horizontal/vertical passes. Unlike the simple post- processing filter, one of them is performed once per block, not per pixel. This allows for much higher speed.

The filter accepts the following options:

quality

Set quality. This option defines the number of levels for averaging. It accepts an integer in the range 4-5. Default value is 4.

qp

Force a constant quantization parameter. It accepts an integer in range 0-63. If not set, the filter will use the QP from the video stream (if available).

strength

Set filter strength. It accepts an integer in range -15 to 32. Lower values mean more details but also more artifacts, while higher values make the image smoother but also blurrier. Default value is 0 X PSNR optimal.

use_bframe_qp

Enable the use of the QP from the B-Frames if set to 1. Using this option may cause flicker since the B-Frames have often larger QP. Default is 0 (not enabled).

geq

The filter accepts the following options:

lum_expr, lum

Set the luminance expression.

cb_expr, cb

Set the chrominance blue expression.

cr_expr, cr

Set the chrominance red expression.

alpha_expr, a

Set the alpha expression.

red_expr, r

Set the red expression.

green_expr, g

Set the green expression.

blue_expr, b

Set the blue expression.

The colorspace is selected according to the specified options. If one of the lum_expr, cb_expr, or cr_expr options is specified, the filter will automatically select a YCbCr colorspace. If one of the red_expr, green_expr, or blue_expr options is specified, it will select an RGB colorspace.

If one of the chrominance expression is not defined, it falls back on the other one. If no alpha expression is specified it will evaluate to opaque value. If none of chrominance expressions are specified, they will evaluate to the luminance expression.

The expressions can use the following variables and functions:

N

The sequential number of the filtered frame, starting from 0.

X

Y

The coordinates of the current sample.

W

H

The width and height of the image.

SW

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 "1,1" for the luma plane, and "0.5,0.5" for chroma planes.

T

Time of the current frame, expressed in seconds.

p(x, y)

Return the value of the pixel at location (x,y) of the current plane.

lum(x, y)

Return the value of the pixel at location (x,y) of the luminance plane.

cb(x, y)

Return the value of the pixel at location (x,y) of the blue-difference chroma plane. Return 0 if there is no such plane.

cr(x, y)

Return the value of the pixel at location (x,y) of the red-difference chroma plane. Return 0 if there is no such plane.

r(x, y)

g(x, y)

b(x, y)

Return the value of the pixel at location (x,y) of the red/green/blue component. Return 0 if there is no such component.

alpha(x, y)

Return the value of the pixel at location (x,y) of the alpha plane. Return 0 if there is no such plane.

For functions, if x and y are outside the area, the value will be automatically clipped to the closer edge.

Examples

• Flip the image horizontally:

          geq=p(W-X\,Y)
  

• Generate a bidimensional sine wave, with angle "PI/3" and a wavelength of 100 pixels:

          geq=128 + 100*sin(2*(PI/100)*(cos(PI/3)*(X-50*T) + sin(PI/3)*Y)):128:128
  

• Generate a fancy enigmatic moving light:

          nullsrc=s=256x256,geq=random(1)/hypot(X-cos(N*0.07)*W/2-W/2\,Y-sin(N*0.09)*H/2-H/2)^2*1000000*sin(N*0.02):128:128
  

• Generate a quick emboss effect:

          format=gray,geq=lum_expr='(p(X,Y)+(256-p(X-4,Y-4)))/2'
  

• Modify RGB components depending on pixel position:

          geq=r='X/W*r(X,Y)':g='(1-X/W)*g(X,Y)':b='(H-Y)/H*b(X,Y)'
  

• Create a radial gradient that is the same size as the input (also see the vignette filter):

          geq=lum=255*gauss((X/W-0.5)*3)*gauss((Y/H-0.5)*3)/gauss(0)/gauss(0),format=gray
  

• Diagonal split screen to compare filter effect:

          ffmpeg -i input -filter_complex "[0:v]geq=lum=if(gt(X\,Y*(W/H))\,255),format=gray[alpha];[0:v][alpha]alphamerge,curves=preset=color_negative[filtered];[0:v][filtered]overlay" output
  

gradfun

Fix the banding artifacts that are sometimes introduced into nearly flat regions by truncation to 8bit color depth. Interpolate the gradients that should go where the bands are, and dither them.

It is designed for playback only. Do not use it prior to lossy compression, because compression tends to lose the dither and bring back the bands.

It accepts the following parameters:

strength

The maximum amount by which the filter will change any one pixel. This is also the threshold for detecting nearly flat regions. Acceptable values range from .51 to 64; the default value is 1.2. Out-of-range values will be clipped to the valid range.

radius

The neighborhood to fit the gradient to. A larger radius makes for smoother gradients, but also prevents the filter from modifying the pixels near detailed regions. Acceptable values are 8-32; the default value is 16. Out-of-range values will be clipped to the valid range.

Alternatively, the options can be specified as a flat string: strength[:radius]

Examples

• Apply the filter with a 3.5 strength and radius of 8:

          gradfun=3.5:8
  

• Specify radius, omitting the strength (which will fall-back to the default value):

          gradfun=radius=8
  

haldclut

Apply a Hald CLUT to a video stream.

First input is the video stream to process, and second one is the Hald CLUT. The Hald CLUT input can be a simple picture or a complete video stream.

The filter accepts the following options:

shortest

Force termination when the shortest input terminates. Default is 0.

repeatlast

Continue applying the last CLUT after the end of the stream. A value of 0 disable the filter after the last frame of the CLUT is reached. Default is 1.

"haldclut" also has the same interpolation options as lut3d (both filters share the same internals).

More information about the Hald CLUT can be found on Eskil Steenberg's website (Hald CLUT author) at <http://www.quelsolaar.com/technology/clut.html>.

Workflow examples

Hald CLUT video stream

Generate an identity Hald CLUT stream altered with various effects:

        ffmpeg -f lavfi -i B<haldclutsrc>=8 -vf "hue=H=2*PI*t:s=sin(2*PI*t)+1, curves=cross_process" -t 10 -c:v ffv1 clut.nut

Note: make sure you use a lossless codec.

Then use it with "haldclut" to apply it on some random stream:

        ffmpeg -f lavfi -i mandelbrot -i clut.nut -filter_complex '[0][1] haldclut' -t 20 mandelclut.mkv

The Hald CLUT will be applied to the 10 first seconds (duration of clut.nut), then the latest picture of that CLUT stream will be applied to the remaining frames of the "mandelbrot" stream.

Hald CLUT with preview

A Hald CLUT is supposed to be a squared image of "Level*Level*Level" by "Level*Level*Level" pixels. For a given Hald CLUT, FFmpeg will select the biggest possible square starting at the top left of the picture. The remaining padding pixels (bottom or right) will be ignored. This area can be used to add a preview of the Hald CLUT.

Typically, the following generated Hald CLUT will be supported by the "haldclut" filter:

        ffmpeg -f lavfi -i B<haldclutsrc>=8 -vf "
           pad=iw+320 [padded_clut];
           smptebars=s=320x256, split [a][b];
           [padded_clut][a] overlay=W-320:h, curves=color_negative [main];
           [main][b] overlay=W-320" -frames:v 1 clut.png

It contains the original and a preview of the effect of the CLUT: SMPTE color bars are displayed on the right-top, and below the same color bars processed by the color changes.

Then, the effect of this Hald CLUT can be visualized with:

        ffplay input.mkv -vf "movie=clut.png, [in] haldclut"

hflip

Flip the input video horizontally.

For example, to horizontally flip the input video with ffmpeg:

        ffmpeg -i in.avi -vf "hflip" out.avi

histeq

This filter applies a global color histogram equalization on a per-frame basis.

It can be used to correct video that has a compressed range of pixel intensities. The filter redistributes the pixel intensities to equalize their distribution across the intensity range. It may be viewed as an ``automatically adjusting contrast filter''. This filter is useful only for correcting degraded or poorly captured source video.

The filter accepts the following options:

strength

Determine the amount of equalization to be applied. As the strength is reduced, the distribution of pixel intensities more-and-more approaches that of the input frame. The value must be a float number in the range [0,1] and defaults to 0.200.

intensity

Set the maximum intensity that can generated and scale the output values appropriately. The strength should be set as desired and then the intensity can be limited if needed to avoid washing-out. The value must be a float number in the range [0,1] and defaults to 0.210.

antibanding

Set the antibanding level. If enabled the filter will randomly vary the luminance of output pixels by a small amount to avoid banding of the histogram. Possible values are "none", "weak" or "strong". It defaults to "none".

histogram

Compute and draw a color distribution histogram for the input video.

The computed histogram is a representation of the color component distribution in an image.

Standard histogram displays the color components distribution in an image. Displays color graph for each color component. Shows distribution of the Y, U, V, A or R, G, B components, depending on input format, in the current frame. Below each graph a color component scale meter is shown.

The filter accepts the following options:

level_height

Set height of level. Default value is 200. Allowed range is [50, 2048].

scale_height

Set height of color scale. Default value is 12. Allowed range is [0, 40].

display_mode

Set display mode. It accepts the following values:

parade

Per color component graphs are placed below each other.

overlay

Presents information identical to that in the "parade", except that the graphs representing color components are superimposed directly over one another.

Default is "parade".

levels_mode

Set mode. Can be either "linear", or "logarithmic". Default is "linear".

components

Set what color components to display. Default is 7.

Examples

• Calculate and draw histogram:

          ffplay -i input -vf histogram
  

hqdn3d

This is a high precision/quality 3d denoise filter. It aims to reduce image noise, producing smooth images and making still images really still. It should enhance compressibility.

It accepts the following optional parameters:

luma_spatial

A non-negative floating point number which specifies spatial luma strength. It defaults to 4.0.

chroma_spatial

A non-negative floating point number which specifies spatial chroma strength. It defaults to 3.0*luma_spatial/4.0.

luma_tmp

A floating point number which specifies luma temporal strength. It defaults to 6.0*luma_spatial/4.0.

chroma_tmp

A floating point number which specifies chroma temporal strength. It defaults to luma_tmp*chroma_spatial/luma_spatial.

hqx

Apply a high-quality magnification filter designed for pixel art. This filter was originally created by Maxim Stepin.

It accepts the following option:

n

Set the scaling dimension: 2 for "hq2x", 3 for "hq3x" and 4 for "hq4x". Default is 3.

hstack

Stack input videos horizontally.

All streams must be of same pixel format and of same height.

Note that this filter is faster than using overlay and pad filter to create same output.

The filter accept the following option:

inputs

Set number of input streams. Default is 2.

shortest

If set to 1, force the output to terminate when the shortest input terminates. Default value is 0.

hue

Modify the hue and/or the saturation of the input.

It accepts the following parameters:

h

Specify the hue angle as a number of degrees. It accepts an expression, and defaults to ``0''.

s

Specify the saturation in the [-10,10] range. It accepts an expression and defaults to ``1''.

H

Specify the hue angle as a number of radians. It accepts an expression, and defaults to ``0''.

b

Specify the brightness in the [-10,10] range. It accepts an expression and defaults to ``0''.

h and H are mutually exclusive, and can't be specified at the same time.

The b, h, H and s option values are expressions containing the following constants:

n

frame count of the input frame starting from 0

pts

presentation timestamp of the input frame expressed in time base units

r

frame rate of the input video, NAN if the input frame rate is unknown

t

timestamp expressed in seconds, NAN if the input timestamp is unknown

tb

time base of the input video

Examples

• Set the hue to 90 degrees and the saturation to 1.0:

          hue=h=90:s=1
  

• Same command but expressing the hue in radians:

          hue=H=PI/2:s=1
  

• Rotate hue and make the saturation swing between 0 and 2 over a period of 1 second:

          hue="H=2*PI*t: s=sin(2*PI*t)+1"
  

• Apply a 3 seconds saturation fade-in effect starting at 0:

          hue="s=min(t/3\,1)"
  

  The general fade-in expression can be written as:

          hue="s=min(0\, max((t-START)/DURATION\, 1))"
  

• Apply a 3 seconds saturation fade-out effect starting at 5 seconds:

          hue="s=max(0\, min(1\, (8-t)/3))"
  

  The general fade-out expression can be written as:

          hue="s=max(0\, min(1\, (START+DURATION-t)/DURATION))"
  

Commands

This filter supports the following commands:

b

s

h

H

Modify the hue and/or the saturation and/or brightness of the input video. The command accepts the same syntax of the corresponding option.

If the specified expression is not valid, it is kept at its current value.

idet

Detect video interlacing type.

This filter tries to detect if the input frames as interlaced, progressive, top or bottom field first. It will also try and detect fields that are repeated between adjacent frames (a sign of telecine).

Single frame detection considers only immediately adjacent frames when classifying each frame. Multiple frame detection incorporates the classification history of previous frames.

The filter will log these metadata values:

single.current_frame

Detected type of current frame using single-frame detection. One of: ``tff'' (top field first), ``bff'' (bottom field first), ``progressive'', or ``undetermined''

single.tff

Cumulative number of frames detected as top field first using single-frame detection.

multiple.tff

Cumulative number of frames detected as top field first using multiple-frame detection.

single.bff

Cumulative number of frames detected as bottom field first using single-frame detection.

multiple.current_frame

Detected type of current frame using multiple-frame detection. One of: ``tff'' (top field first), ``bff'' (bottom field first), ``progressive'', or ``undetermined''

multiple.bff

Cumulative number of frames detected as bottom field first using multiple-frame detection.

single.progressive

Cumulative number of frames detected as progressive using single-frame detection.

multiple.progressive

Cumulative number of frames detected as progressive using multiple-frame detection.

single.undetermined

Cumulative number of frames that could not be classified using single-frame detection.

multiple.undetermined

Cumulative number of frames that could not be classified using multiple-frame detection.

repeated.current_frame

Which field in the current frame is repeated from the last. One of ``neither'', ``top'', or ``bottom''.

repeated.neither

Cumulative number of frames with no repeated field.

repeated.top

Cumulative number of frames with the top field repeated from the previous frame's top field.

repeated.bottom

Cumulative number of frames with the bottom field repeated from the previous frame's bottom field.

The filter accepts the following options:

intl_thres

Set interlacing threshold.

prog_thres

Set progressive threshold.

repeat_thres

Threshold for repeated field detection.

half_life

Number of frames after which a given frame's contribution to the statistics is halved (i.e., it contributes only 0.5 to it's classification). The default of 0 means that all frames seen are given full weight of 1.0 forever.

analyze_interlaced_flag

When this is not 0 then idet will use the specified number of frames to determine if the interlaced flag is accurate, it will not count undetermined frames. If the flag is found to be accurate it will be used without any further computations, if it is found to be inaccurate it will be cleared without any further computations. This allows inserting the idet filter as a low computational method to clean up the interlaced flag

il

Deinterleave or interleave fields.

This filter allows one to process interlaced images fields without deinterlacing them. Deinterleaving splits the input frame into 2 fields (so called half pictures). Odd lines are moved to the top half of the output image, even lines to the bottom half. You can process (filter) them independently and then re-interleave them.

The filter accepts the following options:

luma_mode, l

chroma_mode, c

alpha_mode, a

Available values for luma_mode, chroma_mode and alpha_mode are:

none

Do nothing.

deinterleave, d

Deinterleave fields, placing one above the other.

interleave, i

Interleave fields. Reverse the effect of deinterleaving.

Default value is "none".

luma_swap, ls

chroma_swap, cs

alpha_swap, as

Swap luma/chroma/alpha fields. Exchange even & odd lines. Default value is 0.

inflate

Apply inflate effect to the video.

This filter replaces the pixel by the local(3x3) average by taking into account only values higher than the pixel.

It accepts the following options:

threshold0

threshold1

threshold2

threshold3

Limit the maximum change for each plane, default is 65535. If 0, plane will remain unchanged.

interlace

Simple interlacing filter from progressive contents. This interleaves upper (or lower) lines from odd frames with lower (or upper) lines from even frames, halving the frame rate and preserving image height.

           Original        Original             New Frame
           Frame 'j'      Frame 'j+1'             (tff)
          ==========      ===========       ==================
            Line 0  -------------------->    Frame 'j' Line 0
            Line 1          Line 1  ---->   Frame 'j+1' Line 1
            Line 2 --------------------->    Frame 'j' Line 2
            Line 3          Line 3  ---->   Frame 'j+1' Line 3
             ...             ...                   ...
        New Frame + 1 will be generated by Frame 'j+2' and Frame 'j+3' and so on

It accepts the following optional parameters:

scan

This determines whether the interlaced frame is taken from the even (tff - default) or odd (bff) lines of the progressive frame.

lowpass

Enable (default) or disable the vertical lowpass filter to avoid twitter interlacing and reduce moire patterns.

kerndeint

Deinterlace input video by applying Donald Graft's adaptive kernel deinterling. Work on interlaced parts of a video to produce progressive frames.

The description of the accepted parameters follows.

thresh

Set the threshold which affects the filter's tolerance when determining if a pixel line must be processed. It must be an integer in the range [0,255] and defaults to 10. A value of 0 will result in applying the process on every pixels.

map

Paint pixels exceeding the threshold value to white if set to 1. Default is 0.

order

Set the fields order. Swap fields if set to 1, leave fields alone if 0. Default is 0.

sharp

Enable additional sharpening if set to 1. Default is 0.

twoway

Enable twoway sharpening if set to 1. Default is 0.

Examples

• Apply default values:

          kerndeint=thresh=10:map=0:order=0:sharp=0:twoway=0
  

• Enable additional sharpening:

          kerndeint=sharp=1
  

• Paint processed pixels in white:

          kerndeint=map=1
  

lenscorrection

Correct radial lens distortion

This filter can be used to correct for radial distortion as can result from the use of wide angle lenses, and thereby re-rectify the image. To find the right parameters one can use tools available for example as part of opencv or simply trial-and-error. To use opencv use the calibration sample (under samples/cpp) from the opencv sources and extract the k1 and k2 coefficients from the resulting matrix.

Note that effectively the same filter is available in the open-source tools Krita and Digikam from the KDE project.

In contrast to the vignette filter, which can also be used to compensate lens errors, this filter corrects the distortion of the image, whereas vignette corrects the brightness distribution, so you may want to use both filters together in certain cases, though you will have to take care of ordering, i.e. whether vignetting should be applied before or after lens correction.

Options

The filter accepts the following options:

cx

Relative x-coordinate of the focal point of the image, and thereby the center of the distortion. This value has a range [0,1] and is expressed as fractions of the image width.

cy

Relative y-coordinate of the focal point of the image, and thereby the center of the distortion. This value has a range [0,1] and is expressed as fractions of the image height.

k1

Coefficient of the quadratic correction term. 0.5 means no correction.

k2

Coefficient of the double quadratic correction term. 0.5 means no correction.

The formula that generates the correction is:

r_src = r_tgt * (1 + k1 * (r_tgt / r_0)^2 + k2 * (r_tgt / r_0)^4)

where r_0 is halve of the image diagonal and r_src and r_tgt are the distances from the focal point in the source and target images, respectively.

lut3d

Apply a 3D LUT to an input video.

The filter accepts the following options:

file

Set the 3D LUT file name.

Currently supported formats:

3dl

AfterEffects

cube

Iridas

dat

DaVinci

m3d

Pandora

interp

Select interpolation mode.

Available values are:

nearest

Use values from the nearest defined point.

trilinear

Interpolate values using the 8 points defining a cube.

tetrahedral

Interpolate values using a tetrahedron.

lut, lutrgb, lutyuv

Compute a look-up table for binding each pixel component input value to an output value, and apply it to the input video.

lutyuv applies a lookup table to a YUV input video, lutrgb to an RGB input video.

These filters accept the following parameters:

c0

set first pixel component expression

c1

set second pixel component expression

c2

set third pixel component expression

c3

set fourth pixel component expression, corresponds to the alpha component

r

set red component expression

g

set green component expression

b

set blue component expression

a

alpha component expression

y

set Y/luminance component expression

u

set U/Cb component expression

v

set V/Cr component expression

Each of them specifies the expression to use for computing the lookup table for the corresponding pixel component values.

The exact component associated to each of the c* options depends on the format in input.

The lut filter requires either YUV or RGB pixel formats in input, lutrgb requires RGB pixel formats in input, and lutyuv requires YUV.

The expressions can contain the following constants and functions:

w

h

The input width and height.

val

The input value for the pixel component.

clipval

The input value, clipped to the minval-maxval range.

maxval

The maximum value for the pixel component.

minval

The minimum value for the pixel component.

negval

The negated value for the pixel component value, clipped to the minval-maxval range; it corresponds to the expression ``maxval-clipval+minval''.

clip(val)

The computed value in val, clipped to the minval-maxval range.

gammaval(gamma)

The computed gamma correction value of the pixel component value, clipped to the minval-maxval range. It corresponds to the expression "pow((clipval-minval)/(maxval-minval)\,gamma)*(maxval-minval)+minval"

All expressions default to ``val''.

Examples

• Negate input video:

          lutrgb="r=maxval+minval-val:g=maxval+minval-val:b=maxval+minval-val"
          lutyuv="y=maxval+minval-val:u=maxval+minval-val:v=maxval+minval-val"
  

  The above is the same as:

          lutrgb="r=negval:g=negval:b=negval"
          lutyuv="y=negval:u=negval:v=negval"
  

• Negate luminance:

          lutyuv=y=negval
  

• Remove chroma components, turning the video into a graytone image:

          lutyuv="u=128:v=128"
  

• Apply a luma burning effect:

          lutyuv="y=2*val"
  

• Remove green and blue components:

          lutrgb="g=0:b=0"
  

• Set a constant alpha channel value on input:

          format=rgba,lutrgb=a="maxval-minval/2"
  

• Correct luminance gamma by a factor of 0.5:

          lutyuv=y=gammaval(0.5)
  

• Discard least significant bits of luma:

          lutyuv=y='bitand(val, 128+64+32)'
  

maskedmerge

Merge the first input stream with the second input stream using per pixel weights in the third input stream.

A value of 0 in the third stream pixel component means that pixel component from first stream is returned unchanged, while maximum value (eg. 255 for 8-bit videos) means that pixel component from second stream is returned unchanged. Intermediate values define the amount of merging between both input stream's pixel components.

This filter accepts the following options:

planes

Set which planes will be processed as bitmap, unprocessed planes will be copied from first stream. By default value 0xf, all planes will be processed.

mcdeint

Apply motion-compensation deinterlacing.

It needs one field per frame as input and must thus be used together with yadif=1/3 or equivalent.

This filter accepts the following options:

mode

Set the deinterlacing mode.

It accepts one of the following values:

fast

medium

slow

use iterative motion estimation

extra_slow

like slow, but use multiple reference frames.

Default value is fast.

parity

Set the picture field parity assumed for the input video. It must be one of the following values:

0, tff

assume top field first

1, bff

assume bottom field first

Default value is bff.

qp

Set per-block quantization parameter (QP) used by the internal encoder.

Higher values should result in a smoother motion vector field but less optimal individual vectors. Default value is 1.

mergeplanes

Merge color channel components from several video streams.

The filter accepts up to 4 input streams, and merge selected input planes to the output video.

This filter accepts the following options:

mapping

Set input to output plane mapping. Default is 0.

The mappings is specified as a bitmap. It should be specified as a hexadecimal number in the form 0xAa[Bb[Cc[Dd]]]. 'Aa' describes the mapping for the first plane of the output stream. 'A' sets the number of the input stream to use (from 0 to 3), and 'a' the plane number of the corresponding input to use (from 0 to 3). The rest of the mappings is similar, 'Bb' describes the mapping for the output stream second plane, 'Cc' describes the mapping for the output stream third plane and 'Dd' describes the mapping for the output stream fourth plane.

format

Set output pixel format. Default is "yuva444p".

Examples

• Merge three gray video streams of same width and height into single video stream:

          [a0][a1][a2]mergeplanes=0x001020:yuv444p
  

• Merge 1st yuv444p stream and 2nd gray video stream into yuva444p video stream:

          [a0][a1]mergeplanes=0x00010210:yuva444p
  

• Swap Y and A plane in yuva444p stream:

          format=yuva444p,mergeplanes=0x03010200:yuva444p
  

• Swap U and V plane in yuv420p stream:

          format=yuv420p,mergeplanes=0x000201:yuv420p
  

• Cast a rgb24 clip to yuv444p:

          format=rgb24,mergeplanes=0x000102:yuv444p
  

metadata, ametadata

Manipulate frame metadata.

This filter accepts the following options:

mode

Set mode of operation of the filter.

Can be one of the following:

select

If both "value" and "key" is set, select frames which have such metadata. If only "key" is set, select every frame that has such key in metadata.

add

Add new metadata "key" and "value". If key is already available do nothing.

modify

Modify value of already present key.

delete

If "value" is set, delete only keys that have such value. Otherwise, delete key.

print

Print key and its value if metadata was found. If "key" is not set print all metadata values available in frame.

key

Set key used with all modes. Must be set for all modes except "print".

value

Set metadata value which will be used. This option is mandatory for "modify" and "add" mode.

function

Which function to use when comparing metadata value and "value".

Can be one of following:

same_str

Values are interpreted as strings, returns true if metadata value is same as "value".

starts_with

Values are interpreted as strings, returns true if metadata value starts with the "value" option string.

less

Values are interpreted as floats, returns true if metadata value is less than "value".

equal

Values are interpreted as floats, returns true if "value" is equal with metadata value.

greater

Values are interpreted as floats, returns true if metadata value is greater than "value".

expr

Values are interpreted as floats, returns true if expression from option "expr" evaluates to true.

expr

Set expression which is used when "function" is set to "expr". The expression is evaluated through the eval API and can contain the following constants:

VALUE1

Float representation of "value" from metadata key.

VALUE2

Float representation of "value" as supplied by user in "value" option.

file

If specified in "print" mode, output is written to the named file. When filename equals ``-'' data is written to standard output. If "file" option is not set, output is written to the log with AV_LOG_INFO loglevel.

Examples

• Print all metadata values for frames with key "lavfi.singnalstats.YDIF" with values between 0 and 1.

  signalstats,metadata=print:key=lavfi.signalstats.YDIF:value=0:function=expr:expr='between(VALUE1,0,1)'

mpdecimate

Drop frames that do not differ greatly from the previous frame in order to reduce frame rate.

The main use of this filter is for very-low-bitrate encoding (e.g. streaming over dialup modem), but it could in theory be used for fixing movies that were inverse-telecined incorrectly.

A description of the accepted options follows.

max

Set the maximum number of consecutive frames which can be dropped (if positive), or the minimum interval between dropped frames (if negative). If the value is 0, the frame is dropped unregarding the number of previous sequentially dropped frames.

Default value is 0.

hi

lo

frac

Set the dropping threshold values.

Values for hi and lo are for 8x8 pixel blocks and represent actual pixel value differences, so a threshold of 64 corresponds to 1 unit of difference for each pixel, or the same spread out differently over the block.

A frame is a candidate for dropping if no 8x8 blocks differ by more than a threshold of hi, and if no more than frac blocks (1 meaning the whole image) differ by more than a threshold of lo.

Default value for hi is 64*12, default value for lo is 64*5, and default value for frac is 0.33.

negate

Negate input video.

It accepts an integer in input; if non-zero it negates the alpha component (if available). The default value in input is 0.

nnedi

Deinterlace video using neural network edge directed interpolation.

This filter accepts the following options:

weights

Mandatory option, without binary file filter can not work. Currently file can be found here: https://github.com/dubhater/vapoursynth-nnedi3/blob/master/src/nnedi3_weights.bin

deint

Set which frames to deinterlace, by default it is "all". Can be "all" or "interlaced".

field

Set mode of operation.

Can be one of the following:

af

Use frame flags, both fields.

a

Use frame flags, single field.

t

Use top field only.

b

Use bottom field only.

ft

Use both fields, top first.

fb

Use both fields, bottom first.

planes

Set which planes to process, by default filter process all frames.

nsize

Set size of local neighborhood around each pixel, used by the predictor neural network.

Can be one of the following:

s8x6

s16x6

s32x6

s48x6

s8x4

s16x4

s32x4

nns

Set the number of neurons in predicctor neural network. Can be one of the following:

n16

n32

n64

n128

n256

qual

Controls the number of different neural network predictions that are blended together to compute the final output value. Can be "fast", default or "slow".

etype

Set which set of weights to use in the predictor. Can be one of the following:

a

weights trained to minimize absolute error

s

weights trained to minimize squared error

pscrn

Controls whether or not the prescreener neural network is used to decide which pixels should be processed by the predictor neural network and which can be handled by simple cubic interpolation. The prescreener is trained to know whether cubic interpolation will be sufficient for a pixel or whether it should be predicted by the predictor nn. The computational complexity of the prescreener nn is much less than that of the predictor nn. Since most pixels can be handled by cubic interpolation, using the prescreener generally results in much faster processing. The prescreener is pretty accurate, so the difference between using it and not using it is almost always unnoticeable.

Can be one of the following:

none

original

new

Default is "new".

fapprox

Set various debugging flags.

noformat

Force libavfilter not to use any of the specified pixel formats for the input to the next filter.

It accepts the following parameters:

pix_fmts

A '|'-separated list of pixel format names, such as apix_fmts=yuv420p|monow|rgb24".

Examples

• Force libavfilter to use a format different from yuv420p for the input to the vflip filter:

          noformat=pix_fmts=yuv420p,vflip
  

• Convert the input video to any of the formats not contained in the list:

          noformat=yuv420p|yuv444p|yuv410p
  

noise

Add noise on video input frame.

The filter accepts the following options:

all_seed

c0_seed

c1_seed

c2_seed

c3_seed

Set noise seed for specific pixel component or all pixel components in case of all_seed. Default value is 123457.

all_strength, alls

c0_strength, c0s

c1_strength, c1s

c2_strength, c2s

c3_strength, c3s

Set noise strength for specific pixel component or all pixel components in case all_strength. Default value is 0. Allowed range is [0, 100].

all_flags, allf

c0_flags, c0f

c1_flags, c1f

c2_flags, c2f

c3_flags, c3f

Set pixel component flags or set flags for all components if all_flags. Available values for component flags are:

a

averaged temporal noise (smoother)

p

mix random noise with a (semi)regular pattern

t

temporal noise (noise pattern changes between frames)

u

uniform noise (gaussian otherwise)

Examples

Add temporal and uniform noise to input video:

        noise=alls=20:allf=t+u

null

Pass the video source unchanged to the output.

ocr

Optical Character Recognition

This filter uses Tesseract for optical character recognition.

It accepts the following options:

datapath

Set datapath to tesseract data. Default is to use whatever was set at installation.

language

Set language, default is ``eng''.

whitelist

Set character whitelist.

blacklist

Set character blacklist.

The filter exports recognized text as the frame metadata "lavfi.ocr.text".

ocv

Apply a video transform using libopencv.

To enable this filter, install the libopencv library and headers and configure FFmpeg with "--enable-libopencv".

It accepts the following parameters:

filter_name

The name of the libopencv filter to apply.

filter_params

The parameters to pass to the libopencv filter. If not specified, the default values are assumed.

Refer to the official libopencv documentation for more precise information: <http://docs.opencv.org/master/modules/imgproc/doc/filtering.html>

Several libopencv filters are supported; see the following subsections.

dilate

Dilate an image by using a specific structuring element. It corresponds to the libopencv function "cvDilate".

It accepts the parameters: struct_el|nb_iterations.

struct_el represents a structuring element, and has the syntax: colsxrows+anchor_xxanchor_y/shape

cols and rows represent the number of columns and rows of the structuring element, anchor_x and anchor_y the anchor point, and shape the shape for the structuring element. shape must be ``rect'', ``cross'', ``ellipse'', or ``custom''.

If the value for shape is ``custom'', it must be followed by a string of the form "=filename". The file with name filename is assumed to represent a binary image, with each printable character corresponding to a bright pixel. When a custom shape is used, cols and rows are ignored, the number or columns and rows of the read file are assumed instead.

The default value for struct_el is ``3x3+0x0/rect''.

nb_iterations specifies the number of times the transform is applied to the image, and defaults to 1.

Some examples:

        # Use the default values
        ocv=dilate
        
        # Dilate using a structuring element with a 5x5 cross, iterating two times
        ocv=filter_name=dilate:filter_params=5x5+2x2/cross|2
        
        # Read the shape from the file diamond.shape, iterating two times.
        # The file diamond.shape may contain a pattern of characters like this
        #   *
        #  ***
        # *****
        #  ***
        #   *
        # The specified columns and rows are ignored
        # but the anchor point coordinates are not
        ocv=dilate:0x0+2x2/custom=diamond.shape|2

erode

Erode an image by using a specific structuring element. It corresponds to the libopencv function "cvErode".

It accepts the parameters: struct_el:nb_iterations, with the same syntax and semantics as the dilate filter.

smooth

Smooth the input video.

The filter takes the following parameters: type|param1|param2|param3|param4.

type is the type of smooth filter to apply, and must be one of the following values: ``blur'', ``blur_no_scale'', ``median'', ``gaussian'', or ``bilateral''. The default value is ``gaussian''.

The meaning of param1, param2, param3, and param4 depend on the smooth type. param1 and param2 accept integer positive values or 0. param3 and param4 accept floating point values.

The default value for param1 is 3. The default value for the other parameters is 0.

These parameters correspond to the parameters assigned to the libopencv function "cvSmooth".

overlay

Overlay one video on top of another.

It takes two inputs and has one output. The first input is the ``main'' video on which the second input is overlaid.

It accepts the following parameters:

A description of the accepted options follows.

x

y

Set the expression for the x and y coordinates of the overlaid video on the main video. Default value is ``0'' for both expressions. In case the expression is invalid, it is set to a huge value (meaning that the overlay will not be displayed within the output visible area).

eof_action

The action to take when EOF is encountered on the secondary input; it accepts one of the following values:

repeat

Repeat the last frame (the default).

endall

End both streams.

pass

Pass the main input through.

eval

Set when the expressions for x, and y are evaluated.

It accepts the following values:

init

only evaluate expressions once during the filter initialization or when a command is processed

frame

evaluate expressions for each incoming frame

Default value is frame.

shortest

If set to 1, force the output to terminate when the shortest input terminates. Default value is 0.

format

Set the format for the output video.

It accepts the following values:

yuv420

force YUV420 output

yuv422

force YUV422 output

yuv444

force YUV444 output

rgb

force RGB output

Default value is yuv420.

rgb (deprecated)

If set to 1, force the filter to accept inputs in the RGB color space. Default value is 0. This option is deprecated, use format instead.

repeatlast

If set to 1, force the filter to draw the last overlay frame over the main input until the end of the stream. A value of 0 disables this behavior. Default value is 1.

The x, and y expressions can contain the following parameters.

main_w, W

main_h, H

The main input width and height.

overlay_w, w

overlay_h, h

The overlay input width and height.

x

y

The computed values for x and y. They are evaluated for each new frame.

hsub

vsub

horizontal and vertical chroma subsample values of the output format. For example for the pixel format ``yuv422p'' hsub is 2 and vsub is 1.

n

the number of input frame, starting from 0

pos

the position in the file of the input frame, NAN if unknown

t

The timestamp, expressed in seconds. It's NAN if the input timestamp is unknown.

Note that the n, pos, t variables are available only when evaluation is done per frame, and will evaluate to NAN when eval is set to init.

Be aware that frames are taken from each input video in timestamp order, hence, if their initial timestamps differ, it is a good idea to pass the two inputs through a setpts=PTS-STARTPTS filter to have them begin in the same zero timestamp, as the example for the movie filter does.

You can chain together more overlays but you should test the efficiency of such approach.

Commands

This filter supports the following commands:

x

y

Modify the x and y of the overlay input. The command accepts the same syntax of the corresponding option.

If the specified expression is not valid, it is kept at its current value.

Examples

• Draw the overlay at 10 pixels from the bottom right corner of the main video:

          overlay=main_w-overlay_w-10:main_h-overlay_h-10
  

  Using named options the example above becomes:

          overlay=x=main_w-overlay_w-10:y=main_h-overlay_h-10
  

• Insert a transparent PNG logo in the bottom left corner of the input, using the ffmpeg tool with the "-filter_complex" option:

          ffmpeg -i input -i logo -filter_complex 'overlay=10:main_h-overlay_h-10' output
  

• Insert 2 different transparent PNG logos (second logo on bottom right corner) using the ffmpeg tool:

          ffmpeg -i input -i logo1 -i logo2 -filter_complex 'overlay=x=10:y=H-h-10,overlay=x=W-w-10:y=H-h-10' output
  

• Add a transparent color layer on top of the main video; "WxH" must specify the size of the main input to the overlay filter:

          [email protected]:size=WxH [over]; [in][over] overlay [out]
  

• Play an original video and a filtered version (here with the deshake filter) side by side using the ffplay tool:

          ffplay input.avi -vf 'split[a][b]; [a]pad=iw*2:ih[src]; [b]deshake[filt]; [src][filt]overlay=w'
  

  The above command is the same as:

          ffplay input.avi -vf 'split[b], pad=iw*2[src], [b]deshake, [src]overlay=w'
  

• Make a sliding overlay appearing from the left to the right top part of the screen starting since time 2:

          overlay=x='if(gte(t,2), -w+(t-2)*20, NAN)':y=0
  

• Compose output by putting two input videos side to side:

          ffmpeg -i left.avi -i right.avi -filter_complex "
          nullsrc=size=200x100 [background];
          [0:v] setpts=PTS-STARTPTS, scale=100x100 [left];
          [1:v] setpts=PTS-STARTPTS, scale=100x100 [right];
          [background][left]       overlay=shortest=1       [background+left];
          [background+left][right] overlay=shortest=1:x=100 [left+right]
          "
  

• Mask 10-20 seconds of a video by applying the delogo filter to a section

          ffmpeg -i test.avi -codec:v:0 wmv2 -ar 11025 -b:v 9000k
          -vf '[in]split[split_main][split_delogo];[split_delogo]trim=start=360:end=371,delogo=0:0:640:480[delogoed];[split_main][delogoed]overlay=eof_action=pass[out]'
          masked.avi
  

• Chain several overlays in cascade:

          nullsrc=s=200x200 [bg];
          testsrc=s=100x100, split=4 [in0][in1][in2][in3];
          [in0] lutrgb=r=0, [bg]   overlay=0:0     [mid0];
          [in1] lutrgb=g=0, [mid0] overlay=100:0   [mid1];
          [in2] lutrgb=b=0, [mid1] overlay=0:100   [mid2];
          [in3] null,       [mid2] overlay=100:100 [out0]
  

owdenoise

Apply Overcomplete Wavelet denoiser.

The filter accepts the following options:

depth

Set depth.

Larger depth values will denoise lower frequency components more, but slow down filtering.

Must be an int in the range 8-16, default is 8.

luma_strength, ls

Set luma strength.

Must be a double value in the range 0-1000, default is 1.0.

chroma_strength, cs

Set chroma strength.

Must be a double value in the range 0-1000, default is 1.0.

pad

Add paddings to the input image, and place the original input at the provided x, y coordinates.

It accepts the following parameters:

width, w

height, h

Specify an expression for the size of the output image with the paddings added. If the value for width or height is 0, the corresponding input size is used for the output.

The width expression can reference the value set by the height expression, and vice versa.

The default value of width and height is 0.

x

y

Specify the offsets to place the input image at within the padded area, with respect to the top/left border of the output image.

The x expression can reference the value set by the y expression, and vice versa.

The default value of x and y is 0.

color

Specify the color of the padded area. For the syntax of this option, check the ``Color'' section in the ffmpeg-utils manual.

The default value of color is ``black''.

The value for the width, height, x, and y options are expressions containing the following constants:

in_w

in_h

The input video width and height.

iw

ih

These are the same as in_w and in_h.

out_w

out_h

The output width and height (the size of the padded area), as specified by the width and height expressions.

ow

oh

These are the same as out_w and out_h.

x

y

The x and y offsets as specified by the x and y expressions, or NAN if not yet specified.

a

same as iw / ih

sar

input sample aspect ratio

dar

input display aspect ratio, it is the same as (iw / ih) * sar

hsub

vsub

The horizontal and vertical chroma subsample values. For example for the pixel format ``yuv422p'' hsub is 2 and vsub is 1.

Examples

• Add paddings with the color ``violet'' to the input video. The output video size is 640x480, and the top-left corner of the input video is placed at column 0, row 40

          pad=640:480:0:40:violet
  

  The example above is equivalent to the following command:

          pad=width=640:height=480:x=0:y=40:color=violet
  

• Pad the input to get an output with dimensions increased by 3/2, and put the input video at the center of the padded area:

          pad="3/2*iw:3/2*ih:(ow-iw)/2:(oh-ih)/2"
  

• Pad the input to get a squared output with size equal to the maximum value between the input width and height, and put the input video at the center of the padded area:

          pad="max(iw\,ih):ow:(ow-iw)/2:(oh-ih)/2"
  

• Pad the input to get a final w/h ratio of 16:9:

          pad="ih*16/9:ih:(ow-iw)/2:(oh-ih)/2"
  

• In case of anamorphic video, in order to set the output display aspect correctly, it is necessary to use sar in the expression, according to the relation:

          (ih * X / ih) * sar = output_dar
          X = output_dar / sar
  

  Thus the previous example needs to be modified to:

          pad="ih*16/9/sar:ih:(ow-iw)/2:(oh-ih)/2"
  

• Double the output size and put the input video in the bottom-right corner of the output padded area:

          pad="2*iw:2*ih:ow-iw:oh-ih"