ffmpeg-filters(1)
NAME
ffmpeg-filters - FFmpeg filters
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
acrusher
Reduce audio bit resolution.
This filter is bit crusher with enhanced functionality. A bit crusher
is used to audibly reduce number of bits an audio signal is sampled
with. This doesn't change the bit depth at all, it just produces the
effect. Material reduced in bit depth sounds more harsh and "digital".
This filter is able to even round to continuous values instead of
discrete bit depths. Additionally it has a D/C offset which results in
different crushing of the lower and the upper half of the signal. An
Anti-Aliasing setting is able to produce "softer" crushing sounds.
Another feature of this filter is the logarithmic mode. This setting
switches from linear distances between bits to logarithmic ones. The
result is a much more "natural" sounding crusher which doesn't gate low
signals for example. The human ear has a logarithmic perception, too so
this kind of crushing is much more pleasant. Logarithmic crushing is
also able to get anti-aliased.
The filter accepts the following options:
level_in
Set level in.
level_out
Set level out.
bits
Set bit reduction.
mix Set mixing amount.
mode
Can be linear: "lin" or logarithmic: "log".
dc Set DC.
aa Set anti-aliasing.
samples
Set sample reduction.
lfo Enable LFO. By default disabled.
lforange
Set LFO range.
lforate
Set LFO rate.
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. If you want to delay exact number of samples, append 'S'
to number.
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
* Delay second channel by 500 samples, the third channel by 700
samples and leave the first channel (and any other channels that
may be present) unchanged.
adelay=0|500S|700S
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 dividing 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 rise 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 by 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 an input signal from rising over a desired
threshold. This limiter uses lookahead technology to prevent your
signal from distorting. It means that there is a small delay after the
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.
aloop
Loop audio samples.
The filter accepts the following options:
loop
Set the number of loops.
size
Set maximal number of samples.
start
Set first sample of loop.
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.
f 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.
t 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 a reasonable value makes it possible
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
areverse
Reverse an audio clip.
Warning: This filter requires memory to buffer the entire clip, so
trimming is suggested.
Examples
* Take the first 5 seconds of a clip, and reverse it.
atrim=end=5,areverse
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
signals 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.
crystalizer
Simple algorithm to expand audio dynamic range.
The filter accepts the following options:
i Sets the intensity of effect (default: 2.0). Must be in range
between 0.0 (unchanged sound) to 10.0 (maximum effect).
c Enable clipping. By default is enabled.
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 options:
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.
firequalizer
Apply FIR Equalization using arbitrary frequency response.
The filter accepts the following option:
gain
Set gain curve equation (in dB). The expression can contain
variables:
f the evaluated frequency
sr sample rate
ch channel number, set to 0 when multichannels evaluation is
disabled
chid
channel id, see libavutil/channel_layout.h, set to the first
channel id when multichannels evaluation is disabled
chs number of channels
chlayout
channel_layout, see libavutil/channel_layout.h
and functions:
gain_interpolate(f)
interpolate gain on frequency f based on gain_entry
cubic_interpolate(f)
same as gain_interpolate, but smoother
This option is also available as command. Default is
gain_interpolate(f).
gain_entry
Set gain entry for gain_interpolate function. The expression can
contain functions:
entry(f, g)
store gain entry at frequency f with value g
This option is also available as command.
delay
Set filter delay in seconds. Higher value means more accurate.
Default is 0.01.
accuracy
Set filter accuracy in Hz. Lower value means more accurate.
Default is 5.
wfunc
Set window function. Acceptable values are:
rectangular
rectangular window, useful when gain curve is already smooth
hann
hann window (default)
hamming
hamming window
blackman
blackman window
nuttall3
3-terms continuous 1st derivative nuttall window
mnuttall3
minimum 3-terms discontinuous nuttall window
nuttall
4-terms continuous 1st derivative nuttall window
bnuttall
minimum 4-terms discontinuous nuttall (blackman-nuttall) window
bharris
blackman-harris window
tukey
tukey window
fixed
If enabled, use fixed number of audio samples. This improves speed
when filtering with large delay. Default is disabled.
multi
Enable multichannels evaluation on gain. Default is disabled.
zero_phase
Enable zero phase mode by subtracting timestamp to compensate
delay. Default is disabled.
scale
Set scale used by gain. Acceptable values are:
linlin
linear frequency, linear gain
linlog
linear frequency, logarithmic (in dB) gain (default)
loglin
logarithmic (in octave scale where 20 Hz is 0) frequency,
linear gain
loglog
logarithmic frequency, logarithmic gain
dumpfile
Set file for dumping, suitable for gnuplot.
dumpscale
Set scale for dumpfile. Acceptable values are same with scale
option. Default is linlog.
Examples
* lowpass at 1000 Hz:
firequalizer=gain='if(lt(f,1000), 0, -INF)'
* lowpass at 1000 Hz with gain_entry:
firequalizer=gain_entry='entry(1000,0); entry(1001, -INF)'
* custom equalization:
firequalizer=gain_entry='entry(100,0); entry(400, -4); entry(1000, -6); entry(2000, 0)'
* higher delay with zero phase to compensate delay:
firequalizer=delay=0.1:fixed=on:zero_phase=on
* lowpass on left channel, highpass on right channel:
firequalizer=gain='if(eq(chid,1), gain_interpolate(f), if(eq(chid,2), gain_interpolate(1e6+f), 0))'
:gain_entry='entry(1000, 0); entry(1001,-INF); entry(1e6+1000,0)':multi=on
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.
hdcd
Decodes High Definition Compatible Digital (HDCD) data. A 16-bit PCM
stream with embedded HDCD codes is expanded into a 20-bit PCM stream.
The filter supports the Peak Extend and Low-level Gain Adjustment
features of HDCD, and detects the Transient Filter flag.
ffmpeg -i HDCD16.flac -af hdcd OUT24.flac
When using the filter with wav, note the default encoding for wav is
16-bit, so the resulting 20-bit stream will be truncated back to
16-bit. Use something like -acodec pcm_s24le after the filter to get
24-bit PCM output.
ffmpeg -i HDCD16.wav -af hdcd OUT16.wav
ffmpeg -i HDCD16.wav -af hdcd -acodec pcm_s24le OUT24.wav
The filter accepts the following options:
disable_autoconvert
Disable any automatic format conversion or resampling in the filter
graph.
process_stereo
Process the stereo channels together. If target_gain does not match
between channels, consider it invalid and use the last valid
target_gain.
cdt_ms
Set the code detect timer period in ms.
force_pe
Always extend peaks above -3dBFS even if PE isn't signaled.
analyze_mode
Replace audio with a solid tone and adjust the amplitude to signal
some specific aspect of the decoding process. The output file can
be loaded in an audio editor alongside the original to aid
analysis.
"analyze_mode=pe:force_pe=true" can be used to see all samples
above the PE level.
Modes are:
0, off
Disabled
1, lle
Gain adjustment level at each sample
2, pe
Samples where peak extend occurs
3, cdt
Samples where the code detect timer is active
4, tgm
Samples where the target gain does not match between channels
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.
loudnorm
EBU R128 loudness normalization. Includes both dynamic and linear
normalization modes. Support for both single pass (livestreams, files)
and double pass (files) modes. This algorithm can target IL, LRA, and
maximum true peak.
To enable compilation of this filter you need to configure FFmpeg with
"--enable-libebur128".
The filter accepts the following options:
I, i
Set integrated loudness target. Range is -70.0 - -5.0. Default
value is -24.0.
LRA, lra
Set loudness range target. Range is 1.0 - 20.0. Default value is
7.0.
TP, tp
Set maximum true peak. Range is -9.0 - +0.0. Default value is
-2.0.
measured_I, measured_i
Measured IL of input file. Range is -99.0 - +0.0.
measured_LRA, measured_lra
Measured LRA of input file. Range is 0.0 - 99.0.
measured_TP, measured_tp
Measured true peak of input file. Range is -99.0 - +99.0.
measured_thresh
Measured threshold of input file. Range is -99.0 - +0.0.
offset
Set offset gain. Gain is applied before the true-peak limiter.
Range is -99.0 - +99.0. Default is +0.0.
linear
Normalize linearly if possible. measured_I, measured_LRA,
measured_TP, and measured_thresh must also to be specified in order
to use this mode. Options are true or false. Default is true.
dual_mono
Treat mono input files as "dual-mono". If a mono file is intended
for playback on a stereo system, its EBU R128 measurement will be
perceptually incorrect. If set to "true", this option will
compensate for this effect. Multi-channel input files are not
affected by this option. Options are true or false. Default is
false.
print_format
Set print format for stats. Options are summary, json, or none.
Default value is none.
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 indicates 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 beginning 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. freq is processing audio in
frequency domain which is fast. Default is freq.
speakers
Set custom positions of virtual loudspeakers. Syntax for this
option is: <CH> <AZIM> <ELEV>[|<CH> <AZIM> <ELEV>|...]. Each
virtual loudspeaker is described with short channel name following
with azimuth and elevation in degreees. Each virtual loudspeaker
description is separated by '|'. For example to override front
left and front right channel positions use: 'speakers=FL 45 15|FR
345 15'. Descriptions with unrecognised channel names are ignored.
Examples
* Using ClubFritz6 sofa file:
sofalizer=sofa=/path/to/ClubFritz6.sofa:type=freq:radius=1
* Using ClubFritz12 sofa file and bigger radius with small rotation:
sofalizer=sofa=/path/to/ClubFritz12.sofa:type=freq:radius=2:rotation=5
* Similar as above but with custom speaker positions for front left,
front right, rear left and rear right and also with custom gain:
"sofalizer=sofa=/path/to/ClubFritz6.sofa:type=freq:radius=2:speakers=FL 45|FR 315|RL 135|RR 225:gain=28"
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.
Examples
* Apply karaoke like effect:
stereotools=mlev=0.015625
* Convert M/S signal to L/R:
"stereotools=mode=ms>lr"
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].
p Set what planes of frame filter will use for averaging. Default is
all.
avgblur
Apply average blur filter.
The filter accepts the following options:
sizeX
Set horizontal kernel size.
planes
Set which planes to filter. By default all planes are filtered.
sizeY
Set vertical kernel size, if zero it will be same as "sizeX".
Default is 0.
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.
bitplanenoise
Show and measure bit plane noise.
The filter accepts the following options:
bitplane
Set which plane to analyze. Default is 1.
filter
Filter out noisy pixels from "bitplane" set above. Default is
disabled.
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. By
default, the output terminates when the longest 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
freeze
exclusion
glow
hardlight
hardmix
heat
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
bwdif
Deinterlace the input video ("bwdif" stands for "Bob Weaver
Deinterlacing Filter").
Motion adaptive deinterlacing based on yadif with the use of w3fdif and
cubic interpolation algorithms. It accepts the following parameters:
mode
The interlacing mode to adopt. It accepts one of the following
values:
0, send_frame
Output one frame for each frame.
1, send_field
Output one frame for each field.
The default value is "send_field".
parity
The picture field parity assumed for the input interlaced video. It
accepts one of the following values:
0, tff
Assume the top field is first.
1, bff
Assume the bottom field is first.
-1, auto
Enable automatic detection of field parity.
The default value is "auto". If the interlacing is unknown or the
decoder does not export this information, top field first will be
assumed.
deint
Specify which frames to deinterlace. Accept one of the following
values:
0, all
Deinterlace all frames.
1, interlaced
Only deinterlace frames marked as interlaced.
The default value is "all".
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
ciescope
Display CIE color diagram with pixels overlaid onto it.
The filter accepts the following options:
system
Set color system.
ntsc, 470m
ebu, 470bg
smpte
240m
apple
widergb
cie1931
rec709, hdtv
uhdtv, rec2020
cie Set CIE system.
xyy
ucs
luv
gamuts
Set what gamuts to draw.
See "system" option for available values.
size, s
Set ciescope size, by default set to 512.
intensity, i
Set intensity used to map input pixel values to CIE diagram.
contrast
Set contrast used to draw tongue colors that are out of active
color system gamut.
corrgamma
Correct gamma displayed on scope, by default enabled.
showwhite
Show white point on CIE diagram, by default disabled.
gamma
Set input gamma. Used only with XYZ input color space.
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.
mv_type, mvt
Set motion vectors type to visualize. Includes MVs from all frames
unless specified by frame_type option.
Available flags for mv_type are:
fp forward predicted MVs
bp backward predicted MVs
frame_type, ft
Set frame type to visualize motion vectors of.
Available flags for frame_type are:
if intra-coded frames (I-frames)
pf predicted frames (P-frames)
bf bi-directionally predicted frames (B-frames)
Examples
* Visualize forward predicted MVs of all frames using ffplay:
ffplay -flags2 +export_mvs input.mp4 -vf codecview=mv_type=fp
* Visualize multi-directionals MVs of P and B-Frames using ffplay:
ffplay -flags2 +export_mvs input.mp4 -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
bt2020
BT.2020
For example to convert from BT.601 to SMPTE-240M, use the command:
colormatrix=bt601:smpte240m
colorspace
Convert colorspace, transfer characteristics or color primaries.
The filter accepts the following options:
all Specify all color properties at once.
The accepted values are:
bt470m
BT.470M
bt470bg
BT.470BG
bt601-6-525
BT.601-6 525
bt601-6-625
BT.601-6 625
bt709
BT.709
smpte170m
SMPTE-170M
smpte240m
SMPTE-240M
bt2020
BT.2020
space
Specify output colorspace.
The accepted values are:
bt709
BT.709
fcc FCC
bt470bg
BT.470BG or BT.601-6 625
smpte170m
SMPTE-170M or BT.601-6 525
smpte240m
SMPTE-240M
bt2020ncl
BT.2020 with non-constant luminance
trc Specify output transfer characteristics.
The accepted values are:
bt709
BT.709
gamma22
Constant gamma of 2.2
gamma28
Constant gamma of 2.8
smpte170m
SMPTE-170M, BT.601-6 625 or BT.601-6 525
smpte240m
SMPTE-240M
bt2020-10
BT.2020 for 10-bits content
bt2020-12
BT.2020 for 12-bits content
primaries
Specify output color primaries.
The accepted values are:
bt709
BT.709
bt470m
BT.470M
bt470bg
BT.470BG or BT.601-6 625
smpte170m
SMPTE-170M or BT.601-6 525
smpte240m
SMPTE-240M
bt2020
BT.2020
range
Specify output color range.
The accepted values are:
mpeg
MPEG (restricted) range
jpeg
JPEG (full) range
format
Specify output color format.
The accepted values are:
yuv420p
YUV 4:2:0 planar 8-bits
yuv420p10
YUV 4:2:0 planar 10-bits
yuv420p12
YUV 4:2:0 planar 12-bits
yuv422p
YUV 4:2:2 planar 8-bits
yuv422p10
YUV 4:2:2 planar 10-bits
yuv422p12
YUV 4:2:2 planar 12-bits
yuv444p
YUV 4:4:4 planar 8-bits
yuv444p10
YUV 4:4:4 planar 10-bits
yuv444p12
YUV 4:4:4 planar 12-bits
fast
Do a fast conversion, which skips gamma/primary correction. This
will take significantly less CPU, but will be mathematically
incorrect. To get output compatible with that produced by the
colormatrix filter, use fast=1.
dither
Specify dithering mode.
The accepted values are:
none
No dithering
fsb Floyd-Steinberg dithering
wpadapt
Whitepoint adaptation mode.
The accepted values are:
bradford
Bradford whitepoint adaptation
vonkries
von Kries whitepoint adaptation
identity
identity whitepoint adaptation (i.e. no whitepoint adaptation)
iall
Override all input properties at once. Same accepted values as all.
ispace
Override input colorspace. Same accepted values as space.
iprimaries
Override input color primaries. Same accepted values as primaries.
itrc
Override input transfer characteristics. Same accepted values as
trc.
irange
Override input color range. Same accepted values as range.
The filter converts the transfer characteristics, color space and color
primaries to the specified user values. The output value, if not
specified, is set to a default value based on the "all" property. If
that property is also not specified, the filter will log an error. The
output color range and format default to the same value as the input
color range and format. The input transfer characteristics, color
space, color primaries and color range should be set on the input data.
If any of these are missing, the filter will log an error and no
conversion will take place.
For example to convert the input to SMPTE-240M, use the command:
colorspace=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.
coreimage
Video filtering on GPU using Apple's CoreImage API on OSX.
Hardware acceleration is based on an OpenGL context. Usually, this
means it is processed by video hardware. However, software-based OpenGL
implementations exist which means there is no guarantee for hardware
processing. It depends on the respective OSX.
There are many filters and image generators provided by Apple that come
with a large variety of options. The filter has to be referenced by its
name along with its options.
The coreimage filter accepts the following options:
list_filters
List all available filters and generators along with all their
respective options as well as possible minimum and maximum values
along with the default values.
list_filters=true
filter
Specify all filters by their respective name and options. Use
list_filters to determine all valid filter names and options.
Numerical options are specified by a float value and are
automatically clamped to their respective value range. Vector and
color options have to be specified by a list of space separated
float values. Character escaping has to be done. A special option
name "default" is available to use default options for a filter.
It is required to specify either "default" or at least one of the
filter options. All omitted options are used with their default
values. The syntax of the filter string is as follows:
filter=<NAME>@<OPTION>=<VALUE>[@<OPTION>=<VALUE>][@...][#<NAME>@<OPTION>=<VALUE>[@<OPTION>=<VALUE>][@...]][#...]
output_rect
Specify a rectangle where the output of the filter chain is copied
into the input image. It is given by a list of space separated
float values:
output_rect=x\ y\ width\ height
If not given, the output rectangle equals the dimensions of the
input image. The output rectangle is automatically cropped at the
borders of the input image. Negative values are valid for each
component.
output_rect=25\ 25\ 100\ 100
Several filters can be chained for successive processing without GPU-
HOST transfers allowing for fast processing of complex filter chains.
Currently, only filters with zero (generators) or exactly one (filters)
input image and one output image are supported. Also, transition
filters are not yet usable as intended.
Some filters generate output images with additional padding depending
on the respective filter kernel. The padding is automatically removed
to ensure the filter output has the same size as the input image.
For image generators, the size of the output image is determined by the
previous output image of the filter chain or the input image of the
whole filterchain, respectively. The generators do not use the pixel
information of this image to generate their output. However, the
generated output is blended onto this image, resulting in partial or
complete coverage of the output image.
The coreimagesrc video source can be used for generating input images
which are directly fed into the filter chain. By using it, providing
input images by another video source or an input video is not required.
Examples
* List all filters available:
coreimage=list_filters=true
* Use the CIBoxBlur filter with default options to blur an image:
coreimage=filter=CIBoxBlur@default
* Use a filter chain with CISepiaTone at default values and
CIVignetteEffect with its center at 100x100 and a radius of 50
pixels:
coreimage=filter=CIBoxBlur@default#CIVignetteEffect@inputCenter=100\ 100@inputRadius=50
* Use nullsrc and CIQRCodeGenerator to create a QR code for the
FFmpeg homepage, given as complete and escaped command-line for
Apple's standard bash shell:
ffmpeg -f lavfi -i nullsrc=s=100x100,coreimage=filter=CIQRCodeGenerator@inputMessage=https\\\\\://FFmpeg.org/@inputCorrectionLevel=H -frames:v 1 QRCode.png
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.
exact
Enable exact cropping. If enabled, subsampled videos will be
cropped at exact width/height/x/y as specified and will not be
rounded to nearest smaller value. 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 8-bit 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.
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.
plot
Save Gnuplot script of the curves in specified file.
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='0/0 0.5/0.58 1/1'
* Vintage effect:
curves=r='0/0.11 .42/.51 1/0.95':g='0/0 0.50/0.48 1/1':b='0/0.22 .49/.44 1/0.8'
Here we obtain the following coordinates for each components:
red "(0;0.11) (0.42;0.51) (1;0.95)"
green
"(0;0) (0.50;0.48) (1;1)"
blue
"(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='0/0 0.45/0.53 1/1'
* Check out the curves of the "cross_process" profile using ffmpeg
and gnuplot:
ffmpeg -f lavfi -i color -vf curves=cross_process:plot=/tmp/curves.plt -frames:v 1 -f null -
gnuplot -p /tmp/curves.plt
datascope
Video data analysis filter.
This filter shows hexadecimal pixel values of part of video.
The filter accepts the following options:
size, s
Set output video size.
x Set x offset from where to pick pixels.
y Set y offset from where to pick pixels.
mode
Set scope mode, can be one of the following:
mono
Draw hexadecimal pixel values with white color on black
background.
color
Draw hexadecimal pixel values with input video pixel color on
black background.
color2
Draw hexadecimal pixel values on color background picked from
input video, the text color is picked in such way so its always
visible.
axis
Draw rows and columns numbers on left and top of video.
opacity
Set background opacity.
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:color=[email protected]
* Fill the box with pink color:
drawbox=x=10:y=10:w=100:h=100:color=[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
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:color=[email protected]
* Draw a white 3x3 grid with an opacity of 50%:
drawgrid=w=iw/3:h=ih/3:t=2:c=[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 a number
between 0.0 and 1.0. The expression accepts the same variables x,
y as well. 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.
Sequences 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. Takes one or two arguments.
The first argument is mandatory and specifies the metadata key.
The second argument is optional and specifies a default value, used
when the metadata key is not found or empty.
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: fontcolor=[email protected]: box=1: boxcolor=[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 the text at a random position, switching to a new position
every 30 seconds:
drawtext="fontsize=30:fontfile=FreeSerif.ttf:text='hello world':x=if(eq(mod(t\,30)\,0)\,rand(0\,(w-text_w))\,x):y=if(eq(mod(t\,30)\,0)\,rand(0\,(h-text_h))\,y)"
* 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").
fieldhint
Create new frames by copying the top and bottom fields from surrounding
frames supplied as numbers by the hint file.
hint
Set file containing hints: absolute/relative frame numbers.
There must be one line for each frame in a clip. Each line must
contain two numbers separated by the comma, optionally followed by
"-" or "+". Numbers supplied on each line of file can not be out
of [N-1,N+1] where N is current frame number for "absolute" mode or
out of [-1, 1] range for "relative" mode. First number tells from
which frame to pick up top field and second number tells from which
frame to pick up bottom field.
If optionally followed by "+" output frame will be marked as
interlaced, else if followed by "-" output frame will be marked as
progressive, else it will be marked same as input frame. If line
starts with "#" or ";" that line is skipped.
mode
Can be item "absolute" or "relative". Default is "absolute".
Example of first several lines of "hint" file for "relative" mode:
0,0 - # first frame
1,0 - # second frame, use third's frame top field and second's frame bottom field
1,0 - # third frame, use fourth's frame top field and third's frame bottom field
1,0 -
0,0 -
0,0 -
1,0 -
1,0 -
1,0 -
0,0 -
0,0 -
1,0 -
1,0 -
1,0 -
0,0 -
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).
gblur
Apply Gaussian blur filter.
The filter accepts the following options:
sigma
Set horizontal sigma, standard deviation of Gaussian blur. Default
is 0.5.
steps
Set number of steps for Gaussian approximation. Defauls is 1.
planes
Set which planes to filter. By default all planes are filtered.
sigmaV
Set vertical sigma, if negative it will be same as "sigma".
Default is "-1".
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
gradfun
Fix the banding artifacts that are sometimes introduced into nearly
flat regions by truncation to 8-bit 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.
fgopacity
Set foreground opacity. Default is 0.7.
bgopacity
Set background opacity. Default is 0.5.
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.
hwupload_cuda
Upload system memory frames to a CUDA device.
It accepts the following optional parameters:
device
The number of the CUDA device to use
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.
hysteresis
Grow first stream into second stream by connecting components. This
makes it possible to build more robust edge masks.
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.
threshold
Set threshold which is used in filtering. If pixel component value
is higher than this value filter algorithm for connecting
components is activated. By default value is 0.
idet
Detect video interlacing type.
This filter tries to detect if the input frames are interlaced,
progressive, top or bottom field first. It will also try to 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.
rep_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 its
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.
loop
Loop video frames.
The filter accepts the following options:
loop
Set the number of loops.
size
Set maximal size in number of frames.
start
Set first frame of loop.
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)'
* Technicolor like effect:
lutyuv=u='(val-maxval/2)*2+maxval/2':v='(val-maxval/2)*2+maxval/2'
lut2
Compute and apply a lookup table from two video inputs.
This filter accepts 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
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 inputs.
The expressions can contain the following constants:
w
h The input width and height.
x The first input value for the pixel component.
y The second input value for the pixel component.
bdx The first input video bit depth.
bdy The second input video bit depth.
All expressions default to "x".
Examples
* Highlight differences between two RGB video streams:
lut2='ifnot(x-y,0,pow(2,bdx)-1):ifnot(x-y,0,pow(2,bdx)-1):ifnot(x-y,0,pow(2,bdx)-1)'
* Highlight differences between two YUV video streams:
lut2='ifnot(x-y,0,pow(2,bdx)-1):ifnot(x-y,pow(2,bdx-1),pow(2,bdx)-1):ifnot(x-y,pow(2,bdx-1),pow(2,bdx)-1)'
maskedclamp
Clamp the first input stream with the second input and third input
stream.
Returns the value of first stream to be between second input stream -
"undershoot" and third input stream + "overshoot".
This filter accepts the following options:
undershoot
Default value is 0.
overshoot
Default value is 0.
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.
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
mestimate
Estimate and export motion vectors using block matching algorithms.
Motion vectors are stored in frame side data to be used by other
filters.
This filter accepts the following options:
method
Specify the motion estimation method. Accepts one of the following
values:
esa Exhaustive search algorithm.
tss Three step search algorithm.
tdls
Two dimensional logarithmic search algorithm.
ntss
New three step search algorithm.
fss Four step search algorithm.
ds Diamond search algorithm.
hexbs
Hexagon-based search algorithm.
epzs
Enhanced predictive zonal search algorithm.
umh Uneven multi-hexagon search algorithm.
Default value is esa.
mb_size
Macroblock size. Default 16.
search_param
Search parameter. Default 7.
minterpolate
Convert the video to specified frame rate using motion interpolation.
This filter accepts the following options:
fps Specify the output frame rate. This can be rational e.g.
"60000/1001". Frames are dropped if fps is lower than source fps.
Default 60.
mi_mode
Motion interpolation mode. Following values are accepted:
dup Duplicate previous or next frame for interpolating new ones.
blend
Blend source frames. Interpolated frame is mean of previous and
next frames.
mci Motion compensated interpolation. Following options are
effective when this mode is selected:
mc_mode
Motion compensation mode. Following values are accepted:
obmc
Overlapped block motion compensation.
aobmc
Adaptive overlapped block motion compensation. Window
weighting coefficients are controlled adaptively
according to the reliabilities of the neighboring
motion vectors to reduce oversmoothing.
Default mode is obmc.
me_mode
Motion estimation mode. Following values are accepted:
bidir
Bidirectional motion estimation. Motion vectors are
estimated for each source frame in both forward and
backward directions.
bilat
Bilateral motion estimation. Motion vectors are
estimated directly for interpolated frame.
Default mode is bilat.
me The algorithm to be used for motion estimation. Following
values are accepted:
esa Exhaustive search algorithm.
tss Three step search algorithm.
tdls
Two dimensional logarithmic search algorithm.
ntss
New three step search algorithm.
fss Four step search algorithm.
ds Diamond search algorithm.
hexbs
Hexagon-based search algorithm.
epzs
Enhanced predictive zonal search algorithm.
umh Uneven multi-hexagon search algorithm.
Default algorithm is epzs.
mb_size
Macroblock size. Default 16.
search_param
Motion estimation search parameter. Default 32.
vsmbc
Enable variable-size block motion compensation. Motion
estimation is applied with smaller block sizes at object
boundaries in order to make the them less blur. Default is
0 (disabled).
scd Scene change detection method. Scene change leads motion vectors to
be in random direction. Scene change detection replace interpolated
frames by duplicate ones. May not be needed for other modes.
Following values are accepted:
none
Disable scene change detection.
fdiff
Frame difference. Corresponding pixel values are compared and
if it satisfies scd_threshold scene change is detected.
Default method is fdiff.
scd_threshold
Scene change detection threshold. Default is 5.0.
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.
nlmeans
Denoise frames using Non-Local Means algorithm.
Each pixel is adjusted by looking for other pixels with similar
contexts. This context similarity is defined by comparing their
surrounding patches of size pxp. Patches are searched in an area of rxr
around the pixel.
Note that the research area defines centers for patches, which means
some patches will be made of pixels outside that research area.
The filter accepts the following options.
s Set denoising strength.
p Set patch size.
pc Same as p but for chroma planes.
The default value is 0 and means automatic.
r Set research size.
rc Same as r but for chroma planes.
The default value is 0 and means automatic.
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.
tf Use both fields, top first.
bf 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"
palettegen
Generate one palette for a whole video stream.
It accepts the following options:
max_colors
Set the maximum number of colors to quantize in the palette. Note:
the palette will still contain 256 colors; the unused palette
entries will be black.
reserve_transparent
Create a palette of 255 colors maximum and reserve the last one for
transparency. Reserving the transparency color is useful for GIF
optimization. If not set, the maximum of colors in the palette
will be 256. You probably want to disable this option for a
standalone image. Set by default.
stats_mode
Set statistics mode.
It accepts the following values:
full
Compute full frame histograms.
diff
Compute histograms only for the part that differs from previous
frame. This might be relevant to give more importance to the
moving part of your input if the background is static.
single
Compute new histogram for each frame.
Default value is full.
The filter also exports the frame metadata "lavfi.color_quant_ratio"
("nb_color_in / nb_color_out") which you can use to evaluate the degree
of color quantization of the palette. This information is also visible
at info logging level.
Examples
* Generate a representative palette of a given video using ffmpeg:
ffmpeg -i input.mkv -vf palettegen palette.png
paletteuse
Use a palette to downsample an input video stream.
The filter takes two inputs: one video stream and a palette. The
palette must be a 256 pixels image.
It accepts the following options:
dither
Select dithering mode. Available algorithms are:
bayer
Ordered 8x8 bayer dithering (deterministic)
heckbert
Dithering as defined by Paul Heckbert in 1982 (simple error
diffusion). Note: this dithering is sometimes considered
"wrong" and is included as a reference.
floyd_steinberg
Floyd and Steingberg dithering (error diffusion)
sierra2
Frankie Sierra dithering v2 (error diffusion)
sierra2_4a
Frankie Sierra dithering v2 "Lite" (error diffusion)
Default is sierra2_4a.
bayer_scale
When bayer dithering is selected, this option defines the scale of
the pattern (how much the crosshatch pattern is visible). A low
value means more visible pattern for less banding, and higher value
means less visible pattern at the cost of more banding.
The option must be an integer value in the range [0,5]. Default is
2.
diff_mode
If set, define the zone to process
rectangle
Only the changing rectangle will be reprocessed. This is
similar to GIF cropping/offsetting compression mechanism. This
option can be useful for speed if only a part of the image is
changing, and has use cases such as limiting the scope of the
error diffusal dither to the rectangle that bounds the moving
scene (it leads to more deterministic output if the scene
doesn't change much, and as a result less moving noise and
better GIF compression).
Default is none.
new Take new palette for each output frame.
Examples
* Use a palette (generated for example with palettegen) to encode a
GIF using ffmpeg:
ffmpeg -i input.mkv -i palette.png -lavfi paletteuse output.gif
perspective
Correct perspective of video not recorded perpendicular to the screen.
A description of the accepted parameters follows.
x0
y0
x1
y1
x2
y2
x3
y3 Set coordinates expression for top left, top right, bottom left and
bottom right corners. Default values are "0:0:W:0:0:H:W:H" with
which perspective will remain unchanged. If the "sense" option is
set to "source", then the specified points will be sent to the
corners of the destination. If the "sense" option is set to
"destination", then the corners of the source will be sent to the
specified coordinates.
The expressions can use the following variables:
W
H the width and height of video frame.
in Input frame count.
on Output frame count.
interpolation
Set interpolation for perspective correction.
It accepts the following values:
linear
cubic
Default value is linear.
sense
Set interpretation of coordinate options.
It accepts the following values:
0, source
Send point in the source specified by the given coordinates to
the corners of the destination.
1, destination
Send the corners of the source to the point in the destination
specified by the given coordinates.
Default value is source.
eval
Set when the expressions for coordinates x0,y0,...x3,y3 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.
phase
Delay interlaced video by one field time so that the field order
changes.
The intended use is to fix PAL movies that have been captured with the
opposite field order to the film-to-video transfer.
A description of the accepted parameters follows.
mode
Set phase mode.
It accepts the following values:
t Capture field order top-first, transfer bottom-first. Filter
will delay the bottom field.
b Capture field order bottom-first, transfer top-first. Filter
will delay the top field.
p Capture and transfer with the same field order. This mode only
exists for the documentation of the other options to refer to,
but if you actually select it, the filter will faithfully do
nothing.
a Capture field order determined automatically by field flags,
transfer opposite. Filter selects among t and b modes on a
frame by frame basis using field flags. If no field information
is available, then this works just like u.
u Capture unknown or varying, transfer opposite. Filter selects
among t and b on a frame by frame basis by analyzing the images
and selecting the alternative that produces best match between
the fields.
T Capture top-first, transfer unknown or varying. Filter selects
among t and p using image analysis.
B Capture bottom-first, transfer unknown or varying. Filter
selects among b and p using image analysis.
A Capture determined by field flags, transfer unknown or varying.
Filter selects among t, b and p using field flags and image
analysis. If no field information is available, then this works
just like U. This is the default mode.
U Both capture and transfer unknown or varying. Filter selects
among t, b and p using image analysis only.
pixdesctest
Pixel format descriptor test filter, mainly useful for internal
testing. The output video should be equal to the input video.
For example:
format=monow, pixdesctest
can be used to test the monowhite pixel format descriptor definition.
pp
Enable the specified chain of postprocessing subfilters using
libpostproc. This library should be automatically selected with a GPL
build ("--enable-gpl"). Subfilters must be separated by '/' and can be
disabled by prepending a '-'. Each subfilter and some options have a
short and a long name that can be used interchangeably, i.e. dr/dering
are the same.
The filters accept the following options:
subfilters
Set postprocessing subfilters string.
All subfilters share common options to determine their scope:
a/autoq
Honor the quality commands for this subfilter.
c/chrom
Do chrominance filtering, too (default).
y/nochrom
Do luminance filtering only (no chrominance).
n/noluma
Do chrominance filtering only (no luminance).
These options can be appended after the subfilter name, separated by a
'|'.
Available subfilters are:
hb/hdeblock[|difference[|flatness]]
Horizontal deblocking filter
difference
Difference factor where higher values mean more deblocking
(default: 32).
flatness
Flatness threshold where lower values mean more deblocking
(default: 39).
vb/vdeblock[|difference[|flatness]]
Vertical deblocking filter
difference
Difference factor where higher values mean more deblocking
(default: 32).
flatness
Flatness threshold where lower values mean more deblocking
(default: 39).
ha/hadeblock[|difference[|flatness]]
Accurate horizontal deblocking filter
difference
Difference factor where higher values mean more deblocking
(default: 32).
flatness
Flatness threshold where lower values mean more deblocking
(default: 39).
va/vadeblock[|difference[|flatness]]
Accurate vertical deblocking filter
difference
Difference factor where higher values mean more deblocking
(default: 32).
flatness
Flatness threshold where lower values mean more deblocking
(default: 39).
The horizontal and vertical deblocking filters share the difference and
flatness values so you cannot set different horizontal and vertical
thresholds.
h1/x1hdeblock
Experimental horizontal deblocking filter
v1/x1vdeblock
Experimental vertical deblocking filter
dr/dering
Deringing filter
tn/tmpnoise[|threshold1[|threshold2[|threshold3]]], temporal noise
reducer
threshold1
larger -> stronger filtering
threshold2
larger -> stronger filtering
threshold3
larger -> stronger filtering
al/autolevels[:f/fullyrange], automatic brightness / contrast
correction
f/fullyrange
Stretch luminance to "0-255".
lb/linblenddeint
Linear blend deinterlacing filter that deinterlaces the given block
by filtering all lines with a "(1 2 1)" filter.
li/linipoldeint
Linear interpolating deinterlacing filter that deinterlaces the
given block by linearly interpolating every second line.
ci/cubicipoldeint
Cubic interpolating deinterlacing filter deinterlaces the given
block by cubically interpolating every second line.
md/mediandeint
Median deinterlacing filter that deinterlaces the given block by
applying a median filter to every second line.
fd/ffmpegdeint
FFmpeg deinterlacing filter that deinterlaces the given block by
filtering every second line with a "(-1 4 2 4 -1)" filter.
l5/lowpass5
Vertically applied FIR lowpass deinterlacing filter that
deinterlaces the given block by filtering all lines with a "(-1 2 6
2 -1)" filter.
fq/forceQuant[|quantizer]
Overrides the quantizer table from the input with the constant
quantizer you specify.
quantizer
Quantizer to use
de/default
Default pp filter combination ("hb|a,vb|a,dr|a")
fa/fast
Fast pp filter combination ("h1|a,v1|a,dr|a")
ac High quality pp filter combination ("ha|a|128|7,va|a,dr|a")
Examples
* Apply horizontal and vertical deblocking, deringing and automatic
brightness/contrast:
pp=hb/vb/dr/al
* Apply default filters without brightness/contrast correction:
pp=de/-al
* Apply default filters and temporal denoiser:
pp=default/tmpnoise|1|2|3
* Apply deblocking on luminance only, and switch vertical deblocking
on or off automatically depending on available CPU time:
pp=hb|y/vb|a
pp7
Apply Postprocessing filter 7. It is variant of the spp filter, similar
to spp = 6 with 7 point DCT, where only the center sample is used after
IDCT.
The filter accepts the following options:
qp Force a constant quantization parameter. It accepts an integer in
range 0 to 63. If not set, the filter will use the QP from the
video stream (if available).
mode
Set thresholding mode. Available modes are:
hard
Set hard thresholding.
soft
Set soft thresholding (better de-ringing effect, but likely
blurrier).
medium
Set medium thresholding (good results, default).
prewitt
Apply prewitt operator to input video stream.
The filter accepts the following option:
planes
Set which planes will be processed, unprocessed planes will be
copied. By default value 0xf, all planes will be processed.
scale
Set value which will be multiplied with filtered result.
delta
Set value which will be added to filtered result.
psnr
Obtain the average, maximum and minimum PSNR (Peak Signal to Noise
Ratio) between two input videos.
This filter takes in input two input videos, the first input is
considered the "main" source and is passed unchanged to the output. The
second input is used as a "reference" video for computing the PSNR.
Both video inputs must have the same resolution and pixel format for
this filter to work correctly. Also it assumes that both inputs have
the same number of frames, which are compared one by one.
The obtained average PSNR is printed through the logging system.
The filter stores the accumulated MSE (mean squared error) of each
frame, and at the end of the processing it is averaged across all
frames equally, and the following formula is applied to obtain the
PSNR:
PSNR = 10*log10(MAX^2/MSE)
Where MAX is the average of the maximum values of each component of the
image.
The description of the accepted parameters follows.
stats_file, f
If specified the filter will use the named file to save the PSNR of
each individual frame. When filename equals "-" the data is sent to
standard output.
stats_version
Specifies which version of the stats file format to use. Details of
each format are written below. Default value is 1.
stats_add_max
Determines whether the max value is output to the stats log.
Default value is 0. Requires stats_version >= 2. If this is set
and stats_version < 2, the filter will return an error.
The file printed if stats_file is selected, contains a sequence of
key/value pairs of the form key:value for each compared couple of
frames.
If a stats_version greater than 1 is specified, a header line precedes
the list of per-frame-pair stats, with key value pairs following the
frame format with the following parameters:
psnr_log_version
The version of the log file format. Will match stats_version.
fields
A comma separated list of the per-frame-pair parameters included in
the log.
A description of each shown per-frame-pair parameter follows:
n sequential number of the input frame, starting from 1
mse_avg
Mean Square Error pixel-by-pixel average difference of the compared
frames, averaged over all the image components.
mse_y, mse_u, mse_v, mse_r, mse_g, mse_g, mse_a
Mean Square Error pixel-by-pixel average difference of the compared
frames for the component specified by the suffix.
psnr_y, psnr_u, psnr_v, psnr_r, psnr_g, psnr_b, psnr_a
Peak Signal to Noise ratio of the compared frames for the component
specified by the suffix.
max_avg, max_y, max_u, max_v
Maximum allowed value for each channel, and average over all
channels.
For example:
movie=ref_movie.mpg, setpts=PTS-STARTPTS [main];
[main][ref] psnr="stats_file=stats.log" [out]
On this example the input file being processed is compared with the
reference file ref_movie.mpg. The PSNR of each individual frame is
stored in stats.log.
pullup
Pulldown reversal (inverse telecine) filter, capable of handling mixed
hard-telecine, 24000/1001 fps progressive, and 30000/1001 fps
progressive content.
The pullup filter is designed to take advantage of future context in
making its decisions. This filter is stateless in the sense that it
does not lock onto a pattern to follow, but it instead looks forward to
the following fields in order to identify matches and rebuild
progressive frames.
To produce content with an even framerate, insert the fps filter after
pullup, use "fps=24000/1001" if the input frame rate is 29.97fps,
"fps=24" for 30fps and the (rare) telecined 25fps input.
The filter accepts the following options:
jl
jr
jt
jb These options set the amount of "junk" to ignore at the left,
right, top, and bottom of the image, respectively. Left and right
are in units of 8 pixels, while top and bottom are in units of 2
lines. The default is 8 pixels on each side.
sb Set the strict breaks. Setting this option to 1 will reduce the
chances of filter generating an occasional mismatched frame, but it
may also cause an excessive number of frames to be dropped during
high motion sequences. Conversely, setting it to -1 will make
filter match fields more easily. This may help processing of video
where there is slight blurring between the fields, but may also
cause there to be interlaced frames in the output. Default value
is 0.
mp Set the metric plane to use. It accepts the following values:
l Use luma plane.
u Use chroma blue plane.
v Use chroma red plane.
This option may be set to use chroma plane instead of the default
luma plane for doing filter's computations. This may improve
accuracy on very clean source material, but more likely will
decrease accuracy, especially if there is chroma noise (rainbow
effect) or any grayscale video. The main purpose of setting mp to
a chroma plane is to reduce CPU load and make pullup usable in
realtime on slow machines.
For best results (without duplicated frames in the output file) it is
necessary to change the output frame rate. For example, to inverse
telecine NTSC input:
ffmpeg -i input -vf pullup -r 24000/1001 ...
qp
Change video quantization parameters (QP).
The filter accepts the following option:
qp Set expression for quantization parameter.
The expression is evaluated through the eval API and can contain, among
others, the following constants:
known
1 if index is not 129, 0 otherwise.
qp Sequentional index starting from -129 to 128.
Examples
* Some equation like:
qp=2+2*sin(PI*qp)
random
Flush video frames from internal cache of frames into a random order.
No frame is discarded. Inspired by frei0r nervous filter.
frames
Set size in number of frames of internal cache, in range from 2 to
512. Default is 30.
seed
Set seed for random number generator, must be an integer included
between 0 and "UINT32_MAX". If not specified, or if explicitly set
to less than 0, the filter will try to use a good random seed on a
best effort basis.
readvitc
Read vertical interval timecode (VITC) information from the top lines
of a video frame.
The filter adds frame metadata key "lavfi.readvitc.tc_str" with the
timecode value, if a valid timecode has been detected. Further metadata
key "lavfi.readvitc.found" is set to 0/1 depending on whether timecode
data has been found or not.
This filter accepts the following options:
scan_max
Set the maximum number of lines to scan for VITC data. If the value
is set to "-1" the full video frame is scanned. Default is 45.
thr_b
Set the luma threshold for black. Accepts float numbers in the
range [0.0,1.0], default value is 0.2. The value must be equal or
less than "thr_w".
thr_w
Set the luma threshold for white. Accepts float numbers in the
range [0.0,1.0], default value is 0.6. The value must be equal or
greater than "thr_b".
Examples
* Detect and draw VITC data onto the video frame; if no valid VITC is
detected, draw "--:--:--:--" as a placeholder:
ffmpeg -i input.avi -filter:v 'readvitc,drawtext=fontfile=FreeMono.ttf:text=%{metadata\\:lavfi.readvitc.tc_str\\:--\\\\\\:--\\\\\\:--\\\\\\:--}:x=(w-tw)/2:y=400-ascent'
remap
Remap pixels using 2nd: Xmap and 3rd: Ymap input video stream.
Destination pixel at position (X, Y) will be picked from source (x, y)
position where x = Xmap(X, Y) and y = Ymap(X, Y). If mapping values are
out of range, zero value for pixel will be used for destination pixel.
Xmap and Ymap input video streams must be of same dimensions. Output
video stream will have Xmap/Ymap video stream dimensions. Xmap and
Ymap input video streams are 16bit depth, single channel.
removegrain
The removegrain filter is a spatial denoiser for progressive video.
m0 Set mode for the first plane.
m1 Set mode for the second plane.
m2 Set mode for the third plane.
m3 Set mode for the fourth plane.
Range of mode is from 0 to 24. Description of each mode follows:
0 Leave input plane unchanged. Default.
1 Clips the pixel with the minimum and maximum of the 8 neighbour
pixels.
2 Clips the pixel with the second minimum and maximum of the 8
neighbour pixels.
3 Clips the pixel with the third minimum and maximum of the 8
neighbour pixels.
4 Clips the pixel with the fourth minimum and maximum of the 8
neighbour pixels. This is equivalent to a median filter.
5 Line-sensitive clipping giving the minimal change.
6 Line-sensitive clipping, intermediate.
7 Line-sensitive clipping, intermediate.
8 Line-sensitive clipping, intermediate.
9 Line-sensitive clipping on a line where the neighbours pixels are
the closest.
10 Replaces the target pixel with the closest neighbour.
11 [1 2 1] horizontal and vertical kernel blur.
12 Same as mode 11.
13 Bob mode, interpolates top field from the line where the neighbours
pixels are the closest.
14 Bob mode, interpolates bottom field from the line where the
neighbours pixels are the closest.
15 Bob mode, interpolates top field. Same as 13 but with a more
complicated interpolation formula.
16 Bob mode, interpolates bottom field. Same as 14 but with a more
complicated interpolation formula.
17 Clips the pixel with the minimum and maximum of respectively the
maximum and minimum of each pair of opposite neighbour pixels.
18 Line-sensitive clipping using opposite neighbours whose greatest
distance from the current pixel is minimal.
19 Replaces the pixel with the average of its 8 neighbours.
20 Averages the 9 pixels ([1 1 1] horizontal and vertical blur).
21 Clips pixels using the averages of opposite neighbour.
22 Same as mode 21 but simpler and faster.
23 Small edge and halo removal, but reputed useless.
24 Similar as 23.
removelogo
Suppress a TV station logo, using an image file to determine which
pixels comprise the logo. It works by filling in the pixels that
comprise the logo with neighboring pixels.
The filter accepts the following options:
filename, f
Set the filter bitmap file, which can be any image format supported
by libavformat. The width and height of the image file must match
those of the video stream being processed.
Pixels in the provided bitmap image with a value of zero are not
considered part of the logo, non-zero pixels are considered part of the
logo. If you use white (255) for the logo and black (0) for the rest,
you will be safe. For making the filter bitmap, it is recommended to
take a screen capture of a black frame with the logo visible, and then
using a threshold filter followed by the erode filter once or twice.
If needed, little splotches can be fixed manually. Remember that if
logo pixels are not covered, the filter quality will be much reduced.
Marking too many pixels as part of the logo does not hurt as much, but
it will increase the amount of blurring needed to cover over the image
and will destroy more information than necessary, and extra pixels will
slow things down on a large logo.
repeatfields
This filter uses the repeat_field flag from the Video ES headers and
hard repeats fields based on its value.
reverse
Reverse a video clip.
Warning: This filter requires memory to buffer the entire clip, so
trimming is suggested.
Examples
* Take the first 5 seconds of a clip, and reverse it.
trim=end=5,reverse
rotate
Rotate video by an arbitrary angle expressed in radians.
The filter accepts the following options:
A description of the optional parameters follows.
angle, a
Set an expression for the angle by which to rotate the input video
clockwise, expressed as a number of radians. A negative value will
result in a counter-clockwise rotation. By default it is set to
"0".
This expression is evaluated for each frame.
out_w, ow
Set the output width expression, default value is "iw". This
expression is evaluated just once during configuration.
out_h, oh
Set the output height expression, default value is "ih". This
expression is evaluated just once during configuration.
bilinear
Enable bilinear interpolation if set to 1, a value of 0 disables
it. Default value is 1.
fillcolor, c
Set the color used to fill the output area not covered by the
rotated image. For the general syntax of this option, check the
"Color" section in the ffmpeg-utils manual. If the special value
"none" is selected then no background is printed (useful for
example if the background is never shown).
Default value is "black".
The expressions for the angle and the output size can contain the
following constants and functions:
n sequential number of the input frame, starting from 0. It is always
NAN before the first frame is filtered.
t time in seconds of the input frame, it is set to 0 when the filter
is configured. It is always NAN before the first frame is filtered.
hsub
vsub
horizontal and vertical chroma subsample values. For example for
the pixel format "yuv422p" hsub is 2 and vsub is 1.
in_w, iw
in_h, ih
the input video width and height
out_w, ow
out_h, oh
the output width and height, that is the size of the padded area as
specified by the width and height expressions
rotw(a)
roth(a)
the minimal width/height required for completely containing the
input video rotated by a radians.
These are only available when computing the out_w and out_h
expressions.
Examples
* Rotate the input by PI/6 radians clockwise:
rotate=PI/6
* Rotate the input by PI/6 radians counter-clockwise:
rotate=-PI/6
* Rotate the input by 45 degrees clockwise:
rotate=45*PI/180
* Apply a constant rotation with period T, starting from an angle of
PI/3:
rotate=PI/3+2*PI*t/T
* Make the input video rotation oscillating with a period of T
seconds and an amplitude of A radians:
rotate=A*sin(2*PI/T*t)
* Rotate the video, output size is chosen so that the whole rotating
input video is always completely contained in the output:
rotate='2*PI*t:ow=hypot(iw,ih):oh=ow'
* Rotate the video, reduce the output size so that no background is
ever shown:
rotate=2*PI*t:ow='min(iw,ih)/sqrt(2)':oh=ow:c=none
Commands
The filter supports the following commands:
a, angle
Set the angle 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.
sab
Apply Shape Adaptive Blur.
The filter accepts the following options:
luma_radius, lr
Set luma blur filter strength, must be a value in range 0.1-4.0,
default value is 1.0. A greater value will result in a more blurred
image, and in slower processing.
luma_pre_filter_radius, lpfr
Set luma pre-filter radius, must be a value in the 0.1-2.0 range,
default value is 1.0.
luma_strength, ls
Set luma maximum difference between pixels to still be considered,
must be a value in the 0.1-100.0 range, default value is 1.0.
chroma_radius, cr
Set chroma blur filter strength, must be a value in range -0.9-4.0.
A greater value will result in a more blurred image, and in slower
processing.
chroma_pre_filter_radius, cpfr
Set chroma pre-filter radius, must be a value in the -0.9-2.0
range.
chroma_strength, cs
Set chroma maximum difference between pixels to still be
considered, must be a value in the -0.9-100.0 range.
Each chroma option value, if not explicitly specified, is set to the
corresponding luma option value.
scale
Scale (resize) the input video, using the libswscale library.
The scale filter forces the output display aspect ratio to be the same
of the input, by changing the output sample aspect ratio.
If the input image format is different from the format requested by the
next filter, the scale filter will convert the input to the requested
format.
Options
The filter accepts the following options, or any of the options
supported by the libswscale scaler.
See the ffmpeg-scaler manual for the complete list of scaler options.
width, w
height, h
Set the output video dimension expression. Default value is the
input dimension.
If the value is 0, the input width is used for the output.
If one of the values is -1, the scale filter will use a value that
maintains the aspect ratio of the input image, calculated from the
other specified dimension. If both of them are -1, the input size
is used
If one of the values is -n with n > 1, the scale filter will also
use a value that maintains the aspect ratio of the input image,
calculated from the other specified dimension. After that it will,
however, make sure that the calculated dimension is divisible by n
and adjust the value if necessary.
See below for the list of accepted constants for use in the
dimension expression.
eval
Specify when to evaluate width and height expression. 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.
interl
Set the interlacing mode. It accepts the following values:
1 Force interlaced aware scaling.
0 Do not apply interlaced scaling.
-1 Select interlaced aware scaling depending on whether the source
frames are flagged as interlaced or not.
Default value is 0.
flags
Set libswscale scaling flags. See the ffmpeg-scaler manual for the
complete list of values. If not explicitly specified the filter
applies the default flags.
param0, param1
Set libswscale input parameters for scaling algorithms that need
them. See the ffmpeg-scaler manual for the complete documentation.
If not explicitly specified the filter applies empty parameters.
size, s
Set the video size. For the syntax of this option, check the "Video
size" section in the ffmpeg-utils manual.
in_color_matrix
out_color_matrix
Set in/output YCbCr color space type.
This allows the autodetected value to be overridden as well as
allows forcing a specific value used for the output and encoder.
If not specified, the color space type depends on the pixel format.
Possible values:
auto
Choose automatically.
bt709
Format conforming to International Telecommunication Union
(ITU) Recommendation BT.709.
fcc Set color space conforming to the United States Federal
Communications Commission (FCC) Code of Federal Regulations
(CFR) Title 47 (2003) 73.682 (a).
bt601
Set color space conforming to:
* ITU Radiocommunication Sector (ITU-R) Recommendation BT.601
* ITU-R Rec. BT.470-6 (1998) Systems B, B1, and G
* Society of Motion Picture and Television Engineers (SMPTE)
ST 170:2004
smpte240m
Set color space conforming to SMPTE ST 240:1999.
in_range
out_range
Set in/output YCbCr sample range.
This allows the autodetected value to be overridden as well as
allows forcing a specific value used for the output and encoder. If
not specified, the range depends on the pixel format. Possible
values:
auto
Choose automatically.
jpeg/full/pc
Set full range (0-255 in case of 8-bit luma).
mpeg/tv
Set "MPEG" range (16-235 in case of 8-bit luma).
force_original_aspect_ratio
Enable decreasing or increasing output video width or height if
necessary to keep the original aspect ratio. Possible values:
disable
Scale the video as specified and disable this feature.
decrease
The output video dimensions will automatically be decreased if
needed.
increase
The output video dimensions will automatically be increased if
needed.
One useful instance of this option is that when you know a specific
device's maximum allowed resolution, you can use this to limit the
output video to that, while retaining the aspect ratio. For
example, device A allows 1280x720 playback, and your video is
1920x800. Using this option (set it to decrease) and specifying
1280x720 to the command line makes the output 1280x533.
Please note that this is a different thing than specifying -1 for w
or h, you still need to specify the output resolution for this
option to work.
The values of the w and h options are expressions containing the
following constants:
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 (scaled) width and height
ow
oh These are the same as out_w and out_h
a The same as iw / ih
sar input sample aspect ratio
dar The input display aspect ratio. Calculated from "(iw / ih) * sar".
hsub
vsub
horizontal and vertical input chroma subsample values. For example
for the pixel format "yuv422p" hsub is 2 and vsub is 1.
ohsub
ovsub
horizontal and vertical output chroma subsample values. For example
for the pixel format "yuv422p" hsub is 2 and vsub is 1.
Examples
* Scale the input video to a size of 200x100
scale=w=200:h=100
This is equivalent to:
scale=200:100
or:
scale=200x100
* Specify a size abbreviation for the output size:
scale=qcif
which can also be written as:
scale=size=qcif
* Scale the input to 2x:
scale=w=2*iw:h=2*ih
* The above is the same as:
scale=2*in_w:2*in_h
* Scale the input to 2x with forced interlaced scaling:
scale=2*iw:2*ih:interl=1
* Scale the input to half size:
scale=w=iw/2:h=ih/2
* Increase the width, and set the height to the same size:
scale=3/2*iw:ow
* Seek Greek harmony:
scale=iw:1/PHI*iw
scale=ih*PHI:ih
* Increase the height, and set the width to 3/2 of the height:
scale=w=3/2*oh:h=3/5*ih
* Increase the size, making the size a multiple of the chroma
subsample values:
scale="trunc(3/2*iw/hsub)*hsub:trunc(3/2*ih/vsub)*vsub"
* Increase the width to a maximum of 500 pixels, keeping the same
aspect ratio as the input:
scale=w='min(500\, iw*3/2):h=-1'
Commands
This filter supports the following commands:
width, w
height, h
Set the output video dimension 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.
scale_npp
Use the NVIDIA Performance Primitives (libnpp) to perform scaling
and/or pixel format conversion on CUDA video frames. Setting the output
width and height works in the same way as for the scale filter.
The following additional options are accepted:
format
The pixel format of the output CUDA frames. If set to the string
"same" (the default), the input format will be kept. Note that
automatic format negotiation and conversion is not yet supported
for hardware frames
interp_algo
The interpolation algorithm used for resizing. One of the
following:
nn Nearest neighbour.
linear
cubic
cubic2p_bspline
2-parameter cubic (B=1, C=0)
cubic2p_catmullrom
2-parameter cubic (B=0, C=1/2)
cubic2p_b05c03
2-parameter cubic (B=1/2, C=3/10)
super
Supersampling
lanczos
scale2ref
Scale (resize) the input video, based on a reference video.
See the scale filter for available options, scale2ref supports the same
but uses the reference video instead of the main input as basis.
Examples
* Scale a subtitle stream to match the main video in size before
overlaying
'scale2ref[b][a];[a][b]overlay'
selectivecolor
Adjust cyan, magenta, yellow and black (CMYK) to certain ranges of
colors (such as "reds", "yellows", "greens", "cyans", ...). The
adjustment range is defined by the "purity" of the color (that is, how
saturated it already is).
This filter is similar to the Adobe Photoshop Selective Color tool.
The filter accepts the following options:
correction_method
Select color correction method.
Available values are:
absolute
Specified adjustments are applied "as-is" (added/subtracted to
original pixel component value).
relative
Specified adjustments are relative to the original component
value.
Default is "absolute".
reds
Adjustments for red pixels (pixels where the red component is the
maximum)
yellows
Adjustments for yellow pixels (pixels where the blue component is
the minimum)
greens
Adjustments for green pixels (pixels where the green component is
the maximum)
cyans
Adjustments for cyan pixels (pixels where the red component is the
minimum)
blues
Adjustments for blue pixels (pixels where the blue component is the
maximum)
magentas
Adjustments for magenta pixels (pixels where the green component is
the minimum)
whites
Adjustments for white pixels (pixels where all components are
greater than 128)
neutrals
Adjustments for all pixels except pure black and pure white
blacks
Adjustments for black pixels (pixels where all components are
lesser than 128)
psfile
Specify a Photoshop selective color file (".asv") to import the
settings from.
All the adjustment settings (reds, yellows, ...) accept up to 4 space
separated floating point adjustment values in the [-1,1] range,
respectively to adjust the amount of cyan, magenta, yellow and black
for the pixels of its range.
Examples
* Increase cyan by 50% and reduce yellow by 33% in every green areas,
and increase magenta by 27% in blue areas:
selectivecolor=greens=.5 0 -.33 0:blues=0 .27
* Use a Photoshop selective color preset:
selectivecolor=psfile=MySelectiveColorPresets/Misty.asv
separatefields
The "separatefields" takes a frame-based video input and splits each
frame into its components fields, producing a new half height clip with
twice the frame rate and twice the frame count.
This filter use field-dominance information in frame to decide which of
each pair of fields to place first in the output. If it gets it wrong
use setfield filter before "separatefields" filter.
setdar, setsar
The "setdar" filter sets the Display Aspect Ratio for the filter output
video.
This is done by changing the specified Sample (aka Pixel) Aspect Ratio,
according to the following equation:
<DAR> = <HORIZONTAL_RESOLUTION> / <VERTICAL_RESOLUTION> * <SAR>
Keep in mind that the "setdar" filter does not modify the pixel
dimensions of the video frame. Also, the display aspect ratio set by
this filter may be changed by later filters in the filterchain, e.g. in
case of scaling or if another "setdar" or a "setsar" filter is applied.
The "setsar" filter sets the Sample (aka Pixel) Aspect Ratio for the
filter output video.
Note that as a consequence of the application of this filter, the
output display aspect ratio will change according to the equation
above.
Keep in mind that the sample aspect ratio set by the "setsar" filter
may be changed by later filters in the filterchain, e.g. if another
"setsar" or a "setdar" filter is applied.
It accepts the following parameters:
r, ratio, dar ("setdar" only), sar ("setsar" only)
Set the aspect ratio used by the filter.
The parameter can be a floating point number string, an expression,
or a string of the form num:den, where num and den are the
numerator and denominator of the aspect ratio. If the parameter is
not specified, it is assumed the value "0". In case the form
"num:den" is used, the ":" character should be escaped.
max Set the maximum integer value to use for expressing numerator and
denominator when reducing the expressed aspect ratio to a rational.
Default value is 100.
The parameter sar is an expression containing the following constants:
E, PI, PHI
These are approximated values for the mathematical constants e
(Euler's number), pi (Greek pi), and phi (the golden ratio).
w, h
The input width and height.
a These are the same as w / h.
sar The input sample aspect ratio.
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.
Examples
* To change the display aspect ratio to 16:9, specify one of the
following:
setdar=dar=1.77777
setdar=dar=16/9
* To change the sample aspect ratio to 10:11, specify:
setsar=sar=10/11
* To set a display aspect ratio of 16:9, and specify a maximum
integer value of 1000 in the aspect ratio reduction, use the
command:
setdar=ratio=16/9:max=1000
setfield
Force field for the output video frame.
The "setfield" filter marks the interlace type field for the output
frames. It does not change the input frame, but only sets the
corresponding property, which affects how the frame is treated by
following filters (e.g. "fieldorder" or "yadif").
The filter accepts the following options:
mode
Available values are:
auto
Keep the same field property.
bff Mark the frame as bottom-field-first.
tff Mark the frame as top-field-first.
prog
Mark the frame as progressive.
showinfo
Show a line containing various information for each input video frame.
The input video 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, expressed as a
number of time base units. The time base unit depends on the filter
input pad.
pts_time
The Presentation TimeStamp of the input frame, expressed as a
number of seconds.
pos The position of the frame in the input stream, or -1 if this
information is unavailable and/or meaningless (for example in case
of synthetic video).
fmt The pixel format name.
sar The sample aspect ratio of the input frame, expressed in the form
num/den.
s The size of the input frame. For the syntax of this option, check
the "Video size" section in the ffmpeg-utils manual.
i The type of interlaced mode ("P" for "progressive", "T" for top
field first, "B" for bottom field first).
iskey
This is 1 if the frame is a key frame, 0 otherwise.
type
The picture type of the input frame ("I" for an I-frame, "P" for a
P-frame, "B" for a B-frame, or "?" for an unknown type). Also
refer to the documentation of the "AVPictureType" enum and of the
"av_get_picture_type_char" function defined in libavutil/avutil.h.
checksum
The Adler-32 checksum (printed in hexadecimal) of all the planes of
the input frame.
plane_checksum
The Adler-32 checksum (printed in hexadecimal) of each plane of the
input frame, expressed in the form "[c0 c1 c2 c3]".
showpalette
Displays the 256 colors palette of each frame. This filter is only
relevant for pal8 pixel format frames.
It accepts the following option:
s Set the size of the box used to represent one palette color entry.
Default is 30 (for a "30x30" pixel box).
shuffleframes
Reorder and/or duplicate video frames.
It accepts the following parameters:
mapping
Set the destination indexes of input frames. This is space or '|'
separated list of indexes that maps input frames to output frames.
Number of indexes also sets maximal value that each index may have.
The first frame has the index 0. The default is to keep the input
unchanged.
Examples
* Swap second and third frame of every three frames of the input:
ffmpeg -i INPUT -vf "shuffleframes=0 2 1" OUTPUT
* Swap 10th and 1st frame of every ten frames of the input:
ffmpeg -i INPUT -vf "shuffleframes=9 1 2 3 4 5 6 7 8 0" OUTPUT
shuffleplanes
Reorder and/or duplicate video planes.
It accepts the following parameters:
map0
The index of the input plane to be used as the first output plane.
map1
The index of the input plane to be used as the second output plane.
map2
The index of the input plane to be used as the third output plane.
map3
The index of the input plane to be used as the fourth output plane.
The first plane has the index 0. The default is to keep the input
unchanged.
Examples
* Swap the second and third planes of the input:
ffmpeg -i INPUT -vf shuffleplanes=0:2:1:3 OUTPUT
signalstats
Evaluate various visual metrics that assist in determining issues
associated with the digitization of analog video media.
By default the filter will log these metadata values:
YMIN
Display the minimal Y value contained within the input frame.
Expressed in range of [0-255].
YLOW
Display the Y value at the 10% percentile within the input frame.
Expressed in range of [0-255].
YAVG
Display the average Y value within the input frame. Expressed in
range of [0-255].
YHIGH
Display the Y value at the 90% percentile within the input frame.
Expressed in range of [0-255].
YMAX
Display the maximum Y value contained within the input frame.
Expressed in range of [0-255].
UMIN
Display the minimal U value contained within the input frame.
Expressed in range of [0-255].
ULOW
Display the U value at the 10% percentile within the input frame.
Expressed in range of [0-255].
UAVG
Display the average U value within the input frame. Expressed in
range of [0-255].
UHIGH
Display the U value at the 90% percentile within the input frame.
Expressed in range of [0-255].
UMAX
Display the maximum U value contained within the input frame.
Expressed in range of [0-255].
VMIN
Display the minimal V value contained within the input frame.
Expressed in range of [0-255].
VLOW
Display the V value at the 10% percentile within the input frame.
Expressed in range of [0-255].
VAVG
Display the average V value within the input frame. Expressed in
range of [0-255].
VHIGH
Display the V value at the 90% percentile within the input frame.
Expressed in range of [0-255].
VMAX
Display the maximum V value contained within the input frame.
Expressed in range of [0-255].
SATMIN
Display the minimal saturation value contained within the input
frame. Expressed in range of [0-~181.02].
SATLOW
Display the saturation value at the 10% percentile within the input
frame. Expressed in range of [0-~181.02].
SATAVG
Display the average saturation value within the input frame.
Expressed in range of [0-~181.02].
SATHIGH
Display the saturation value at the 90% percentile within the input
frame. Expressed in range of [0-~181.02].
SATMAX
Display the maximum saturation value contained within the input
frame. Expressed in range of [0-~181.02].
HUEMED
Display the median value for hue within the input frame. Expressed
in range of [0-360].
HUEAVG
Display the average value for hue within the input frame. Expressed
in range of [0-360].
YDIF
Display the average of sample value difference between all values
of the Y plane in the current frame and corresponding values of the
previous input frame. Expressed in range of [0-255].
UDIF
Display the average of sample value difference between all values
of the U plane in the current frame and corresponding values of the
previous input frame. Expressed in range of [0-255].
VDIF
Display the average of sample value difference between all values
of the V plane in the current frame and corresponding values of the
previous input frame. Expressed in range of [0-255].
YBITDEPTH
Display bit depth of Y plane in current frame. Expressed in range
of [0-16].
UBITDEPTH
Display bit depth of U plane in current frame. Expressed in range
of [0-16].
VBITDEPTH
Display bit depth of V plane in current frame. Expressed in range
of [0-16].
The filter accepts the following options:
stat
out stat specify an additional form of image analysis. out output
video with the specified type of pixel highlighted.
Both options accept the following values:
tout
Identify temporal outliers pixels. A temporal outlier is a
pixel unlike the neighboring pixels of the same field. Examples
of temporal outliers include the results of video dropouts,
head clogs, or tape tracking issues.
vrep
Identify vertical line repetition. Vertical line repetition
includes similar rows of pixels within a frame. In born-digital
video vertical line repetition is common, but this pattern is
uncommon in video digitized from an analog source. When it
occurs in video that results from the digitization of an analog
source it can indicate concealment from a dropout compensator.
brng
Identify pixels that fall outside of legal broadcast range.
color, c
Set the highlight color for the out option. The default color is
yellow.
Examples
* Output data of various video metrics:
ffprobe -f lavfi movie=example.mov,signalstats="stat=tout+vrep+brng" -show_frames
* Output specific data about the minimum and maximum values of the Y
plane per frame:
ffprobe -f lavfi movie=example.mov,signalstats -show_entries frame_tags=lavfi.signalstats.YMAX,lavfi.signalstats.YMIN
* Playback video while highlighting pixels that are outside of
broadcast range in red.
ffplay example.mov -vf signalstats="out=brng:color=red"
* Playback video with signalstats metadata drawn over the frame.
ffplay example.mov -vf signalstats=stat=brng+vrep+tout,drawtext=fontfile=FreeSerif.ttf:textfile=signalstat_drawtext.txt
The contents of signalstat_drawtext.txt used in the command are:
time %{pts:hms}
Y (%{metadata:lavfi.signalstats.YMIN}-%{metadata:lavfi.signalstats.YMAX})
U (%{metadata:lavfi.signalstats.UMIN}-%{metadata:lavfi.signalstats.UMAX})
V (%{metadata:lavfi.signalstats.VMIN}-%{metadata:lavfi.signalstats.VMAX})
saturation maximum: %{metadata:lavfi.signalstats.SATMAX}
smartblur
Blur the input video without impacting the outlines.
It accepts the following options:
luma_radius, lr
Set the luma radius. The option value must be a float number in the
range [0.1,5.0] that specifies the variance of the gaussian filter
used to blur the image (slower if larger). Default value is 1.0.
luma_strength, ls
Set the luma strength. The option value must be a float number in
the range [-1.0,1.0] that configures the blurring. A value included
in [0.0,1.0] will blur the image whereas a value included in
[-1.0,0.0] will sharpen the image. Default value is 1.0.
luma_threshold, lt
Set the luma threshold used as a coefficient to determine whether a
pixel should be blurred or not. The option value must be an integer
in the range [-30,30]. A value of 0 will filter all the image, a
value included in [0,30] will filter flat areas and a value
included in [-30,0] will filter edges. Default value is 0.
chroma_radius, cr
Set the chroma radius. The option value must be a float number in
the range [0.1,5.0] that specifies the variance of the gaussian
filter used to blur the image (slower if larger). Default value is
1.0.
chroma_strength, cs
Set the chroma strength. The option value must be a float number in
the range [-1.0,1.0] that configures the blurring. A value included
in [0.0,1.0] will blur the image whereas a value included in
[-1.0,0.0] will sharpen the image. Default value is 1.0.
chroma_threshold, ct
Set the chroma threshold used as a coefficient to determine whether
a pixel should be blurred or not. The option value must be an
integer in the range [-30,30]. A value of 0 will filter all the
image, a value included in [0,30] will filter flat areas and a
value included in [-30,0] will filter edges. Default value is 0.
If a chroma option is not explicitly set, the corresponding luma value
is set.
ssim
Obtain the SSIM (Structural SImilarity Metric) between two input
videos.
This filter takes in input two input videos, the first input is
considered the "main" source and is passed unchanged to the output. The
second input is used as a "reference" video for computing the SSIM.
Both video inputs must have the same resolution and pixel format for
this filter to work correctly. Also it assumes that both inputs have
the same number of frames, which are compared one by one.
The filter stores the calculated SSIM of each frame.
The description of the accepted parameters follows.
stats_file, f
If specified the filter will use the named file to save the SSIM of
each individual frame. When filename equals "-" the data is sent to
standard output.
The file printed if stats_file is selected, contains a sequence of
key/value pairs of the form key:value for each compared couple of
frames.
A description of each shown parameter follows:
n sequential number of the input frame, starting from 1
Y, U, V, R, G, B
SSIM of the compared frames for the component specified by the
suffix.
All SSIM of the compared frames for the whole frame.
dB Same as above but in dB representation.
For example:
movie=ref_movie.mpg, setpts=PTS-STARTPTS [main];
[main][ref] ssim="stats_file=stats.log" [out]
On this example the input file being processed is compared with the
reference file ref_movie.mpg. The SSIM of each individual frame is
stored in stats.log.
Another example with both psnr and ssim at same time:
ffmpeg -i main.mpg -i ref.mpg -lavfi "ssim;[0:v][1:v]psnr" -f null -
stereo3d
Convert between different stereoscopic image formats.
The filters accept the following options:
in Set stereoscopic image format of input.
Available values for input image formats are:
sbsl
side by side parallel (left eye left, right eye right)
sbsr
side by side crosseye (right eye left, left eye right)
sbs2l
side by side parallel with half width resolution (left eye
left, right eye right)
sbs2r
side by side crosseye with half width resolution (right eye
left, left eye right)
abl above-below (left eye above, right eye below)
abr above-below (right eye above, left eye below)
ab2l
above-below with half height resolution (left eye above, right
eye below)
ab2r
above-below with half height resolution (right eye above, left
eye below)
al alternating frames (left eye first, right eye second)
ar alternating frames (right eye first, left eye second)
irl interleaved rows (left eye has top row, right eye starts on
next row)
irr interleaved rows (right eye has top row, left eye starts on
next row)
icl interleaved columns, left eye first
icr interleaved columns, right eye first
Default value is sbsl.
out Set stereoscopic image format of output.
sbsl
side by side parallel (left eye left, right eye right)
sbsr
side by side crosseye (right eye left, left eye right)
sbs2l
side by side parallel with half width resolution (left eye
left, right eye right)
sbs2r
side by side crosseye with half width resolution (right eye
left, left eye right)
abl above-below (left eye above, right eye below)
abr above-below (right eye above, left eye below)
ab2l
above-below with half height resolution (left eye above, right
eye below)
ab2r
above-below with half height resolution (right eye above, left
eye below)
al alternating frames (left eye first, right eye second)
ar alternating frames (right eye first, left eye second)
irl interleaved rows (left eye has top row, right eye starts on
next row)
irr interleaved rows (right eye has top row, left eye starts on
next row)
arbg
anaglyph red/blue gray (red filter on left eye, blue filter on
right eye)
argg
anaglyph red/green gray (red filter on left eye, green filter
on right eye)
arcg
anaglyph red/cyan gray (red filter on left eye, cyan filter on
right eye)
arch
anaglyph red/cyan half colored (red filter on left eye, cyan
filter on right eye)
arcc
anaglyph red/cyan color (red filter on left eye, cyan filter on
right eye)
arcd
anaglyph red/cyan color optimized with the least squares
projection of dubois (red filter on left eye, cyan filter on
right eye)
agmg
anaglyph green/magenta gray (green filter on left eye, magenta
filter on right eye)
agmh
anaglyph green/magenta half colored (green filter on left eye,
magenta filter on right eye)
agmc
anaglyph green/magenta colored (green filter on left eye,
magenta filter on right eye)
agmd
anaglyph green/magenta color optimized with the least squares
projection of dubois (green filter on left eye, magenta filter
on right eye)
aybg
anaglyph yellow/blue gray (yellow filter on left eye, blue
filter on right eye)
aybh
anaglyph yellow/blue half colored (yellow filter on left eye,
blue filter on right eye)
aybc
anaglyph yellow/blue colored (yellow filter on left eye, blue
filter on right eye)
aybd
anaglyph yellow/blue color optimized with the least squares
projection of dubois (yellow filter on left eye, blue filter on
right eye)
ml mono output (left eye only)
mr mono output (right eye only)
chl checkerboard, left eye first
chr checkerboard, right eye first
icl interleaved columns, left eye first
icr interleaved columns, right eye first
hdmi
HDMI frame pack
Default value is arcd.
Examples
* Convert input video from side by side parallel to anaglyph
yellow/blue dubois:
stereo3d=sbsl:aybd
* Convert input video from above below (left eye above, right eye
below) to side by side crosseye.
stereo3d=abl:sbsr
streamselect, astreamselect
Select video or audio streams.
The filter accepts the following options:
inputs
Set number of inputs. Default is 2.
map Set input indexes to remap to outputs.
Commands
The "streamselect" and "astreamselect" filter supports the following
commands:
map Set input indexes to remap to outputs.
Examples
* Select first 5 seconds 1st stream and rest of time 2nd stream:
sendcmd='5.0 streamselect map 1',streamselect=inputs=2:map=0
* Same as above, but for audio:
asendcmd='5.0 astreamselect map 1',astreamselect=inputs=2:map=0
sobel
Apply sobel operator to input video stream.
The filter accepts the following option:
planes
Set which planes will be processed, unprocessed planes will be
copied. By default value 0xf, all planes will be processed.
scale
Set value which will be multiplied with filtered result.
delta
Set value which will be added to filtered result.
spp
Apply a simple postprocessing filter that compresses and decompresses
the image at several (or - in the case of quality level 6 - all) shifts
and average the results.
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 0-6. If set to 0, the
filter will have no effect. A value of 6 means the higher quality.
For each increment of that value the speed drops by a factor of
approximately 2. Default value is 3.
qp Force a constant quantization parameter. If not set, the filter
will use the QP from the video stream (if available).
mode
Set thresholding mode. Available modes are:
hard
Set hard thresholding (default).
soft
Set soft thresholding (better de-ringing effect, but likely
blurrier).
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).
subtitles
Draw subtitles on top of input video using the libass library.
To enable compilation of this filter you need to configure FFmpeg with
"--enable-libass". This filter also requires a build with libavcodec
and libavformat to convert the passed subtitles file to ASS (Advanced
Substation Alpha) subtitles format.
The filter accepts the following options:
filename, f
Set the filename of the subtitle file to read. It must be
specified.
original_size
Specify the size of the original video, the video for which the ASS
file was composed. For the syntax of this option, check the "Video
size" section in the ffmpeg-utils manual. Due to a misdesign in
ASS aspect ratio arithmetic, this is necessary to correctly scale
the fonts if the aspect ratio has been changed.
fontsdir
Set a directory path containing fonts that can be used by the
filter. These fonts will be used in addition to whatever the font
provider uses.
charenc
Set subtitles input character encoding. "subtitles" filter only.
Only useful if not UTF-8.
stream_index, si
Set subtitles stream index. "subtitles" filter only.
force_style
Override default style or script info parameters of the subtitles.
It accepts a string containing ASS style format "KEY=VALUE" couples
separated by ",".
If the first key is not specified, it is assumed that the first value
specifies the filename.
For example, to render the file sub.srt on top of the input video, use
the command:
subtitles=sub.srt
which is equivalent to:
subtitles=filename=sub.srt
To render the default subtitles stream from file video.mkv, use:
subtitles=video.mkv
To render the second subtitles stream from that file, use:
subtitles=video.mkv:si=1
To make the subtitles stream from sub.srt appear in transparent green
"DejaVu Serif", use:
subtitles=sub.srt:force_style='FontName=DejaVu Serif,PrimaryColour=&HAA00FF00'
super2xsai
Scale the input by 2x and smooth using the Super2xSaI (Scale and
Interpolate) pixel art scaling algorithm.
Useful for enlarging pixel art images without reducing sharpness.
swaprect
Swap two rectangular objects in video.
This filter accepts the following options:
w Set object width.
h Set object height.
x1 Set 1st rect x coordinate.
y1 Set 1st rect y coordinate.
x2 Set 2nd rect x coordinate.
y2 Set 2nd rect y coordinate.
All expressions are evaluated once for each frame.
The all options are expressions containing the following constants:
w
h The input width and height.
a same as w / h
sar input sample aspect ratio
dar input display aspect ratio, it is the same as (w / h) * sar
n The number of the input frame, starting from 0.
t The timestamp expressed in seconds. It's NAN if the input timestamp
is unknown.
pos the position in the file of the input frame, NAN if unknown
swapuv
Swap U & V plane.
telecine
Apply telecine process to the video.
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.
Some typical patterns:
NTSC output (30i):
27.5p: 32222
24p: 23 (classic)
24p: 2332 (preferred)
20p: 33
18p: 334
16p: 3444
PAL output (25i):
27.5p: 12222
24p: 222222222223 ("Euro pulldown")
16.67p: 33
16p: 33333334
thumbnail
Select the most representative frame in a given sequence of consecutive
frames.
The filter accepts the following options:
n Set the frames batch size to analyze; in a set of n frames, the
filter will pick one of them, and then handle the next batch of n
frames until the end. Default is 100.
Since the filter keeps track of the whole frames sequence, a bigger n
value will result in a higher memory usage, so a high value is not
recommended.
Examples
* Extract one picture each 50 frames:
thumbnail=50
* Complete example of a thumbnail creation with ffmpeg:
ffmpeg -i in.avi -vf thumbnail,scale=300:200 -frames:v 1 out.png
tile
Tile several successive frames together.
The filter accepts the following options:
layout
Set the grid size (i.e. the number of lines and columns). For the
syntax of this option, check the "Video size" section in the
ffmpeg-utils manual.
nb_frames
Set the maximum number of frames to render in the given area. It
must be less than or equal to wxh. The default value is 0, meaning
all the area will be used.
margin
Set the outer border margin in pixels.
padding
Set the inner border thickness (i.e. the number of pixels between
frames). For more advanced padding options (such as having
different values for the edges), refer to the pad video filter.
color
Specify the color of the unused area. For the syntax of this
option, check the "Color" section in the ffmpeg-utils manual. The
default value of color is "black".
Examples
* Produce 8x8 PNG tiles of all keyframes (-skip_frame nokey) in a
movie:
ffmpeg -skip_frame nokey -i file.avi -vf 'scale=128:72,tile=8x8' -an -vsync 0 keyframes%03d.png
The -vsync 0 is necessary to prevent ffmpeg from duplicating each
output frame to accommodate the originally detected frame rate.
* Display 5 pictures in an area of "3x2" frames, with 7 pixels
between them, and 2 pixels of initial margin, using mixed flat and
named options:
tile=3x2:nb_frames=5:padding=7:margin=2
tinterlace
Perform various types of temporal field interlacing.
Frames are counted starting from 1, so the first input frame is
considered odd.
The filter accepts the following options:
mode
Specify the mode of the interlacing. This option can also be
specified as a value alone. See below for a list of values for this
option.
Available values are:
merge, 0
Move odd frames into the upper field, even into the lower
field, generating a double height frame at half frame rate.
------> time
Input:
Frame 1 Frame 2 Frame 3 Frame 4
11111 22222 33333 44444
11111 22222 33333 44444
11111 22222 33333 44444
11111 22222 33333 44444
Output:
11111 33333
22222 44444
11111 33333
22222 44444
11111 33333
22222 44444
11111 33333
22222 44444
drop_even, 1
Only output odd frames, even frames are dropped, generating a
frame with unchanged height at half frame rate.
------> time
Input:
Frame 1 Frame 2 Frame 3 Frame 4
11111 22222 33333 44444
11111 22222 33333 44444
11111 22222 33333 44444
11111 22222 33333 44444
Output:
11111 33333
11111 33333
11111 33333
11111 33333
drop_odd, 2
Only output even frames, odd frames are dropped, generating a
frame with unchanged height at half frame rate.
------> time
Input:
Frame 1 Frame 2 Frame 3 Frame 4
11111 22222 33333 44444
11111 22222 33333 44444
11111 22222 33333 44444
11111 22222 33333 44444
Output:
22222 44444
22222 44444
22222 44444
22222 44444
pad, 3
Expand each frame to full height, but pad alternate lines with
black, generating a frame with double height at the same input
frame rate.
------> time
Input:
Frame 1 Frame 2 Frame 3 Frame 4
11111 22222 33333 44444
11111 22222 33333 44444
11111 22222 33333 44444
11111 22222 33333 44444
Output:
11111 ..... 33333 .....
..... 22222 ..... 44444
11111 ..... 33333 .....
..... 22222 ..... 44444
11111 ..... 33333 .....
..... 22222 ..... 44444
11111 ..... 33333 .....
..... 22222 ..... 44444
interleave_top, 4
Interleave the upper field from odd frames with the lower field
from even frames, generating a frame with unchanged height at
half frame rate.
------> time
Input:
Frame 1 Frame 2 Frame 3 Frame 4
11111<- 22222 33333<- 44444
11111 22222<- 33333 44444<-
11111<- 22222 33333<- 44444
11111 22222<- 33333 44444<-
Output:
11111 33333
22222 44444
11111 33333
22222 44444
interleave_bottom, 5
Interleave the lower field from odd frames with the upper field
from even frames, generating a frame with unchanged height at
half frame rate.
------> time
Input:
Frame 1 Frame 2 Frame 3 Frame 4
11111 22222<- 33333 44444<-
11111<- 22222 33333<- 44444
11111 22222<- 33333 44444<-
11111<- 22222 33333<- 44444
Output:
22222 44444
11111 33333
22222 44444
11111 33333
interlacex2, 6
Double frame rate with unchanged height. Frames are inserted
each containing the second temporal field from the previous
input frame and the first temporal field from the next input
frame. This mode relies on the top_field_first flag. Useful for
interlaced video displays with no field synchronisation.
------> time
Input:
Frame 1 Frame 2 Frame 3 Frame 4
11111 22222 33333 44444
11111 22222 33333 44444
11111 22222 33333 44444
11111 22222 33333 44444
Output:
11111 22222 22222 33333 33333 44444 44444
11111 11111 22222 22222 33333 33333 44444
11111 22222 22222 33333 33333 44444 44444
11111 11111 22222 22222 33333 33333 44444
mergex2, 7
Move odd frames into the upper field, even into the lower
field, generating a double height frame at same frame rate.
------> time
Input:
Frame 1 Frame 2 Frame 3 Frame 4
11111 22222 33333 44444
11111 22222 33333 44444
11111 22222 33333 44444
11111 22222 33333 44444
Output:
11111 33333 33333 55555
22222 22222 44444 44444
11111 33333 33333 55555
22222 22222 44444 44444
11111 33333 33333 55555
22222 22222 44444 44444
11111 33333 33333 55555
22222 22222 44444 44444
Numeric values are deprecated but are accepted for backward
compatibility reasons.
Default mode is "merge".
flags
Specify flags influencing the filter process.
Available value for flags is:
low_pass_filter, vlfp
Enable vertical low-pass filtering in the filter. Vertical
low-pass filtering is required when creating an interlaced
destination from a progressive source which contains high-
frequency vertical detail. Filtering will reduce interlace
'twitter' and Moire patterning.
Vertical low-pass filtering can only be enabled for mode
interleave_top and interleave_bottom.
transpose
Transpose rows with columns in the input video and optionally flip it.
It accepts the following parameters:
dir Specify the transposition direction.
Can assume the following values:
0, 4, cclock_flip
Rotate by 90 degrees counterclockwise and vertically flip
(default), that is:
L.R L.l
. . -> . .
l.r R.r
1, 5, clock
Rotate by 90 degrees clockwise, that is:
L.R l.L
. . -> . .
l.r r.R
2, 6, cclock
Rotate by 90 degrees counterclockwise, that is:
L.R R.r
. . -> . .
l.r L.l
3, 7, clock_flip
Rotate by 90 degrees clockwise and vertically flip, that is:
L.R r.R
. . -> . .
l.r l.L
For values between 4-7, the transposition is only done if the input
video geometry is portrait and not landscape. These values are
deprecated, the "passthrough" option should be used instead.
Numerical values are deprecated, and should be dropped in favor of
symbolic constants.
passthrough
Do not apply the transposition if the input geometry matches the
one specified by the specified value. It accepts the following
values:
none
Always apply transposition.
portrait
Preserve portrait geometry (when height >= width).
landscape
Preserve landscape geometry (when width >= height).
Default value is "none".
For example to rotate by 90 degrees clockwise and preserve portrait
layout:
transpose=dir=1:passthrough=portrait
The command above can also be specified as:
transpose=1:portrait
trim
Trim the input so that the output contains one continuous subpart of
the input.
It accepts the following parameters:
start
Specify the time of the start of the kept section, i.e. the frame
with the timestamp start will be the first frame in the output.
end Specify the time of the first frame that will be dropped, i.e. the
frame immediately preceding the one with the timestamp end will be
the last frame in the output.
start_pts
This is the same as start, except this option sets the start
timestamp in timebase units instead of seconds.
end_pts
This is the same as end, except this option sets the end timestamp
in timebase units instead of seconds.
duration
The maximum duration of the output in seconds.
start_frame
The number of the first frame that should be passed to the output.
end_frame
The number of the first frame 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 for the
accepted syntax.
Note that the first two sets of the start/end options and the duration
option look at the frame timestamp, while the _frame variants simply
count the frames that pass through the filter. Also note that this
filter does not modify the timestamps. If you wish for the output
timestamps to start at zero, insert a setpts filter after the trim
filter.
If multiple start or end options are set, this filter tries to be
greedy and keep all the frames that match at least one of the specified
constraints. To keep only the part that matches all the constraints at
once, chain multiple trim 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 -vf trim=60:120
* Keep only the first second:
ffmpeg -i INPUT -vf trim=duration=1
unsharp
Sharpen or blur the input video.
It accepts the following parameters:
luma_msize_x, lx
Set the luma matrix horizontal size. It must be an odd integer
between 3 and 23. The default value is 5.
luma_msize_y, ly
Set the luma matrix vertical size. It must be an odd integer
between 3 and 23. The default value is 5.
luma_amount, la
Set the luma effect strength. It must be a floating point number,
reasonable values lay between -1.5 and 1.5.
Negative values will blur the input video, while positive values
will sharpen it, a value of zero will disable the effect.
Default value is 1.0.
chroma_msize_x, cx
Set the chroma matrix horizontal size. It must be an odd integer
between 3 and 23. The default value is 5.
chroma_msize_y, cy
Set the chroma matrix vertical size. It must be an odd integer
between 3 and 23. The default value is 5.
chroma_amount, ca
Set the chroma effect strength. It must be a floating point number,
reasonable values lay between -1.5 and 1.5.
Negative values will blur the input video, while positive values
will sharpen it, a value of zero will disable the effect.
Default value is 0.0.
opencl
If set to 1, specify using OpenCL capabilities, only available if
FFmpeg was configured with "--enable-opencl". Default value is 0.
All parameters are optional and default to the equivalent of the string
'5:5:1.0:5:5:0.0'.
Examples
* Apply strong luma sharpen effect:
unsharp=luma_msize_x=7:luma_msize_y=7:luma_amount=2.5
* Apply a strong blur of both luma and chroma parameters:
unsharp=7:7:-2:7:7:-2
uspp
Apply ultra slow/simple postprocessing filter that compresses and
decompresses the image at several (or - in the case of quality level 8
- all) shifts and average the results.
The way this differs from the behavior of spp is that uspp actually
encodes & decodes each case with libavcodec Snow, whereas spp uses a
simplified intra only 8x8 DCT similar to MJPEG.
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 0-8. If set to 0, the
filter will have no effect. A value of 8 means the higher quality.
For each increment of that value the speed drops by a factor of
approximately 2. Default value is 3.
qp Force a constant quantization parameter. If not set, the filter
will use the QP from the video stream (if available).
vaguedenoiser
Apply a wavelet based denoiser.
It transforms each frame from the video input into the wavelet domain,
using Cohen-Daubechies-Feauveau 9/7. Then it applies some filtering to
the obtained coefficients. It does an inverse wavelet transform after.
Due to wavelet properties, it should give a nice smoothed result, and
reduced noise, without blurring picture features.
This filter accepts the following options:
threshold
The filtering strength. The higher, the more filtered the video
will be. Hard thresholding can use a higher threshold than soft
thresholding before the video looks overfiltered.
method
The filtering method the filter will use.
It accepts the following values:
hard
All values under the threshold will be zeroed.
soft
All values under the threshold will be zeroed. All values above
will be reduced by the threshold.
garrote
Scales or nullifies coefficients - intermediary between (more)
soft and (less) hard thresholding.
nsteps
Number of times, the wavelet will decompose the picture. Picture
can't be decomposed beyond a particular point (typically, 8 for a
640x480 frame - as 2^9 = 512 > 480)
percent
Partial of full denoising (limited coefficients shrinking), from 0
to 100.
planes
A list of the planes to process. By default all planes are
processed.
vectorscope
Display 2 color component values in the two dimensional graph (which is
called a vectorscope).
This filter accepts the following options:
mode, m
Set vectorscope mode.
It accepts the following values:
gray
Gray values are displayed on graph, higher brightness means
more pixels have same component color value on location in
graph. This is the default mode.
color
Gray values are displayed on graph. Surrounding pixels values
which are not present in video frame are drawn in gradient of 2
color components which are set by option "x" and "y". The 3rd
color component is static.
color2
Actual color components values present in video frame are
displayed on graph.
color3
Similar as color2 but higher frequency of same values "x" and
"y" on graph increases value of another color component, which
is luminance by default values of "x" and "y".
color4
Actual colors present in video frame are displayed on graph. If
two different colors map to same position on graph then color
with higher value of component not present in graph is picked.
color5
Gray values are displayed on graph. Similar to "color" but with
3rd color component picked from radial gradient.
x Set which color component will be represented on X-axis. Default is
1.
y Set which color component will be represented on Y-axis. Default is
2.
intensity, i
Set intensity, used by modes: gray, color, color3 and color5 for
increasing brightness of color component which represents frequency
of (X, Y) location in graph.
envelope, e
none
No envelope, this is default.
instant
Instant envelope, even darkest single pixel will be clearly
highlighted.
peak
Hold maximum and minimum values presented in graph over time.
This way you can still spot out of range values without
constantly looking at vectorscope.
peak+instant
Peak and instant envelope combined together.
graticule, g
Set what kind of graticule to draw.
none
green
color
opacity, o
Set graticule opacity.
flags, f
Set graticule flags.
white
Draw graticule for white point.
black
Draw graticule for black point.
name
Draw color points short names.
bgopacity, b
Set background opacity.
lthreshold, l
Set low threshold for color component not represented on X or Y
axis. Values lower than this value will be ignored. Default is 0.
Note this value is multiplied with actual max possible value one
pixel component can have. So for 8-bit input and low threshold
value of 0.1 actual threshold is 0.1 * 255 = 25.
hthreshold, h
Set high threshold for color component not represented on X or Y
axis. Values higher than this value will be ignored. Default is 1.
Note this value is multiplied with actual max possible value one
pixel component can have. So for 8-bit input and high threshold
value of 0.9 actual threshold is 0.9 * 255 = 230.
colorspace, c
Set what kind of colorspace to use when drawing graticule.
auto
601
709
Default is auto.
vidstabdetect
Analyze video stabilization/deshaking. Perform pass 1 of 2, see
vidstabtransform for pass 2.
This filter generates a file with relative translation and rotation
transform information about subsequent frames, which is then used by
the vidstabtransform filter.
To enable compilation of this filter you need to configure FFmpeg with
"--enable-libvidstab".
This filter accepts the following options:
result
Set the path to the file used to write the transforms information.
Default value is transforms.trf.
shakiness
Set how shaky the video is and how quick the camera is. It accepts
an integer in the range 1-10, a value of 1 means little shakiness,
a value of 10 means strong shakiness. Default value is 5.
accuracy
Set the accuracy of the detection process. It must be a value in
the range 1-15. A value of 1 means low accuracy, a value of 15
means high accuracy. Default value is 15.
stepsize
Set stepsize of the search process. The region around minimum is
scanned with 1 pixel resolution. Default value is 6.
mincontrast
Set minimum contrast. Below this value a local measurement field is
discarded. Must be a floating point value in the range 0-1. Default
value is 0.3.
tripod
Set reference frame number for tripod mode.
If enabled, the motion of the frames is compared to a reference
frame in the filtered stream, identified by the specified number.
The idea is to compensate all movements in a more-or-less static
scene and keep the camera view absolutely still.
If set to 0, it is disabled. The frames are counted starting from
1.
show
Show fields and transforms in the resulting frames. It accepts an
integer in the range 0-2. Default value is 0, which disables any
visualization.
Examples
* Use default values:
vidstabdetect
* Analyze strongly shaky movie and put the results in file
mytransforms.trf:
vidstabdetect=shakiness=10:accuracy=15:result="mytransforms.trf"
* Visualize the result of internal transformations in the resulting
video:
vidstabdetect=show=1
* Analyze a video with medium shakiness using ffmpeg:
ffmpeg -i input -vf vidstabdetect=shakiness=5:show=1 dummy.avi
vidstabtransform
Video stabilization/deshaking: pass 2 of 2, see vidstabdetect for pass
1.
Read a file with transform information for each frame and
apply/compensate them. Together with the vidstabdetect filter this can
be used to deshake videos. See also
<http://public.hronopik.de/vid.stab>. It is important to also use the
unsharp filter, see below.
To enable compilation of this filter you need to configure FFmpeg with
"--enable-libvidstab".
Options
input
Set path to the file used to read the transforms. Default value is
transforms.trf.
smoothing
Set the number of frames (value*2 + 1) used for lowpass filtering
the camera movements. Default value is 10.
For example a number of 10 means that 21 frames are used (10 in the
past and 10 in the future) to smoothen the motion in the video. A
larger value leads to a smoother video, but limits the acceleration
of the camera (pan/tilt movements). 0 is a special case where a
static camera is simulated.
optalgo
Set the camera path optimization algorithm.
Accepted values are:
gauss
gaussian kernel low-pass filter on camera motion (default)
avg averaging on transformations
maxshift
Set maximal number of pixels to translate frames. Default value is
-1, meaning no limit.
maxangle
Set maximal angle in radians (degree*PI/180) to rotate frames.
Default value is -1, meaning no limit.
crop
Specify how to deal with borders that may be visible due to
movement compensation.
Available values are:
keep
keep image information from previous frame (default)
black
fill the border black
invert
Invert transforms if set to 1. Default value is 0.
relative
Consider transforms as relative to previous frame if set to 1,
absolute if set to 0. Default value is 0.
zoom
Set percentage to zoom. A positive value will result in a zoom-in
effect, a negative value in a zoom-out effect. Default value is 0
(no zoom).
optzoom
Set optimal zooming to avoid borders.
Accepted values are:
0 disabled
1 optimal static zoom value is determined (only very strong
movements will lead to visible borders) (default)
2 optimal adaptive zoom value is determined (no borders will be
visible), see zoomspeed
Note that the value given at zoom is added to the one calculated
here.
zoomspeed
Set percent to zoom maximally each frame (enabled when optzoom is
set to 2). Range is from 0 to 5, default value is 0.25.
interpol
Specify type of interpolation.
Available values are:
no no interpolation
linear
linear only horizontal
bilinear
linear in both directions (default)
bicubic
cubic in both directions (slow)
tripod
Enable virtual tripod mode if set to 1, which is equivalent to
"relative=0:smoothing=0". Default value is 0.
Use also "tripod" option of vidstabdetect.
debug
Increase log verbosity if set to 1. Also the detected global
motions are written to the temporary file global_motions.trf.
Default value is 0.
Examples
* Use ffmpeg for a typical stabilization with default values:
ffmpeg -i inp.mpeg -vf vidstabtransform,unsharp=5:5:0.8:3:3:0.4 inp_stabilized.mpeg
Note the use of the unsharp filter which is always recommended.
* Zoom in a bit more and load transform data from a given file:
vidstabtransform=zoom=5:input="mytransforms.trf"
* Smoothen the video even more:
vidstabtransform=smoothing=30
vflip
Flip the input video vertically.
For example, to vertically flip a video with ffmpeg:
ffmpeg -i in.avi -vf "vflip" out.avi
vignette
Make or reverse a natural vignetting effect.
The filter accepts the following options:
angle, a
Set lens angle expression as a number of radians.
The value is clipped in the "[0,PI/2]" range.
Default value: "PI/5"
x0
y0 Set center coordinates expressions. Respectively "w/2" and "h/2" by
default.
mode
Set forward/backward mode.
Available modes are:
forward
The larger the distance from the central point, the darker the
image becomes.
backward
The larger the distance from the central point, the brighter
the image becomes. This can be used to reverse a vignette
effect, though there is no automatic detection to extract the
lens angle and other settings (yet). It can also be used to
create a burning effect.
Default value is forward.
eval
Set evaluation mode for the expressions (angle, x0, y0).
It accepts the following values:
init
Evaluate expressions only once during the filter
initialization.
frame
Evaluate expressions for each incoming frame. This is way
slower than the init mode since it requires all the scalers to
be re-computed, but it allows advanced dynamic expressions.
Default value is init.
dither
Set dithering to reduce the circular banding effects. Default is 1
(enabled).
aspect
Set vignette aspect. This setting allows one to adjust the shape of
the vignette. Setting this value to the SAR of the input will make
a rectangular vignetting following the dimensions of the video.
Default is "1/1".
Expressions
The alpha, x0 and y0 expressions can contain the following parameters.
w
h input width and height
n the number of input frame, starting from 0
pts the PTS (Presentation TimeStamp) time of the filtered video frame,
expressed in TB units, NAN if undefined
r frame rate of the input video, NAN if the input frame rate is
unknown
t the PTS (Presentation TimeStamp) of the filtered video frame,
expressed in seconds, NAN if undefined
tb time base of the input video
Examples
* Apply simple strong vignetting effect:
vignette=PI/4
* Make a flickering vignetting:
vignette='PI/4+random(1)*PI/50':eval=frame
vstack
Stack input videos vertically.
All streams must be of same pixel format and of same width.
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.
w3fdif
Deinterlace the input video ("w3fdif" stands for "Weston 3 Field
Deinterlacing Filter").
Based on the process described by Martin Weston for BBC R&D, and
implemented based on the de-interlace algorithm written by Jim
Easterbrook for BBC R&D, the Weston 3 field deinterlacing filter uses
filter coefficients calculated by BBC R&D.
There are two sets of filter coefficients, so called "simple": and
"complex". Which set of filter coefficients is used can be set by
passing an optional parameter:
filter
Set the interlacing filter coefficients. Accepts one of the
following values:
simple
Simple filter coefficient set.
complex
More-complex filter coefficient set.
Default value is complex.
deint
Specify which frames to deinterlace. Accept one of the following
values:
all Deinterlace all frames,
interlaced
Only deinterlace frames marked as interlaced.
Default value is all.
waveform
Video waveform monitor.
The waveform monitor plots color component intensity. By default
luminance only. Each column of the waveform corresponds to a column of
pixels in the source video.
It accepts the following options:
mode, m
Can be either "row", or "column". Default is "column". In row
mode, the graph on the left side represents color component value 0
and the right side represents value = 255. In column mode, the top
side represents color component value = 0 and bottom side
represents value = 255.
intensity, i
Set intensity. Smaller values are useful to find out how many
values of the same luminance are distributed across input
rows/columns. Default value is 0.04. Allowed range is [0, 1].
mirror, r
Set mirroring mode. 0 means unmirrored, 1 means mirrored. In
mirrored mode, higher values will be represented on the left side
for "row" mode and at the top for "column" mode. Default is 1
(mirrored).
display, d
Set display mode. It accepts the following values:
overlay
Presents information identical to that in the "parade", except
that the graphs representing color components are superimposed
directly over one another.
This display mode makes it easier to spot relative differences
or similarities in overlapping areas of the color components
that are supposed to be identical, such as neutral whites,
grays, or blacks.
stack
Display separate graph for the color components side by side in
"row" mode or one below the other in "column" mode.
parade
Display separate graph for the color components side by side in
"column" mode or one below the other in "row" mode.
Using this display mode makes it easy to spot color casts in
the highlights and shadows of an image, by comparing the
contours of the top and the bottom graphs of each waveform.
Since whites, grays, and blacks are characterized by exactly
equal amounts of red, green, and blue, neutral areas of the
picture should display three waveforms of roughly equal
width/height. If not, the correction is easy to perform by
making level adjustments the three waveforms.
Default is "stack".
components, c
Set which color components to display. Default is 1, which means
only luminance or red color component if input is in RGB
colorspace. If is set for example to 7 it will display all 3 (if)
available color components.
envelope, e
none
No envelope, this is default.
instant
Instant envelope, minimum and maximum values presented in graph
will be easily visible even with small "step" value.
peak
Hold minimum and maximum values presented in graph across time.
This way you can still spot out of range values without
constantly looking at waveforms.
peak+instant
Peak and instant envelope combined together.
filter, f
lowpass
No filtering, this is default.
flat
Luma and chroma combined together.
aflat
Similar as above, but shows difference between blue and red
chroma.
chroma
Displays only chroma.
color
Displays actual color value on waveform.
acolor
Similar as above, but with luma showing frequency of chroma
values.
graticule, g
Set which graticule to display.
none
Do not display graticule.
green
Display green graticule showing legal broadcast ranges.
opacity, o
Set graticule opacity.
flags, fl
Set graticule flags.
numbers
Draw numbers above lines. By default enabled.
dots
Draw dots instead of lines.
scale, s
Set scale used for displaying graticule.
digital
millivolts
ire
Default is digital.
bgopacity, b
Set background opacity.
weave
The "weave" takes a field-based video input and join each two
sequential fields into single frame, producing a new double height clip
with half the frame rate and half the frame count.
It accepts the following option:
first_field
Set first field. Available values are:
top, t
Set the frame as top-field-first.
bottom, b
Set the frame as bottom-field-first.
Examples
* Interlace video using select and separatefields filter:
separatefields,select=eq(mod(n,4),0)+eq(mod(n,4),3),weave
xbr
Apply the xBR high-quality magnification filter which is designed for
pixel art. It follows a set of edge-detection rules, see
<http://www.libretro.com/forums/viewtopic.php?f=6&t=134>.
It accepts the following option:
n Set the scaling dimension: 2 for "2xBR", 3 for "3xBR" and 4 for
"4xBR". Default is 3.
yadif
Deinterlace the input video ("yadif" means "yet another deinterlacing
filter").
It accepts the following parameters:
mode
The interlacing mode to adopt. It accepts one of the following
values:
0, send_frame
Output one frame for each frame.
1, send_field
Output one frame for each field.
2, send_frame_nospatial
Like "send_frame", but it skips the spatial interlacing check.
3, send_field_nospatial
Like "send_field", but it skips the spatial interlacing check.
The default value is "send_frame".
parity
The picture field parity assumed for the input interlaced video. It
accepts one of the following values:
0, tff
Assume the top field is first.
1, bff
Assume the bottom field is first.
-1, auto
Enable automatic detection of field parity.
The default value is "auto". If the interlacing is unknown or the
decoder does not export this information, top field first will be
assumed.
deint
Specify which frames to deinterlace. Accept one of the following
values:
0, all
Deinterlace all frames.
1, interlaced
Only deinterlace frames marked as interlaced.
The default value is "all".
zoompan
Apply Zoom & Pan effect.
This filter accepts the following options:
zoom, z
Set the zoom expression. Default is 1.
x
y Set the x and y expression. Default is 0.
d Set the duration expression in number of frames. This sets for how
many number of frames effect will last for single input image.
s Set the output image size, default is 'hd720'.
fps Set the output frame rate, default is '25'.
Each expression can contain the following constants:
in_w, iw
Input width.
in_h, ih
Input height.
out_w, ow
Output width.
out_h, oh
Output height.
in Input frame count.
on Output frame count.
x
y Last calculated 'x' and 'y' position from 'x' and 'y' expression
for current input frame.
px
py 'x' and 'y' of last output frame of previous input frame or 0 when
there was not yet such frame (first input frame).
zoom
Last calculated zoom from 'z' expression for current input frame.
pzoom
Last calculated zoom of last output frame of previous input frame.
duration
Number of output frames for current input frame. Calculated from
'd' expression for each input frame.
pduration
number of output frames created for previous input frame
a Rational number: input width / input height
sar sample aspect ratio
dar display aspect ratio
Examples
* Zoom-in up to 1.5 and pan at same time to some spot near center of
picture:
zoompan=z='min(zoom+0.0015,1.5)':d=700:x='if(gte(zoom,1.5),x,x+1/a)':y='if(gte(zoom,1.5),y,y+1)':s=640x360
* Zoom-in up to 1.5 and pan always at center of picture:
zoompan=z='min(zoom+0.0015,1.5)':d=700:x='iw/2-(iw/zoom/2)':y='ih/2-(ih/zoom/2)'
* Same as above but without pausing:
zoompan=z='min(max(zoom,pzoom)+0.0015,1.5)':d=1:x='iw/2-(iw/zoom/2)':y='ih/2-(ih/zoom/2)'
zscale
Scale (resize) the input video, using the z.lib library:
https://github.com/sekrit-twc/zimg.
The zscale filter forces the output display aspect ratio to be the same
as the input, by changing the output sample aspect ratio.
If the input image format is different from the format requested by the
next filter, the zscale filter will convert the input to the requested
format.
Options
The filter accepts the following options.
width, w
height, h
Set the output video dimension expression. Default value is the
input dimension.
If the width or w is 0, the input width is used for the output. If
the height or h is 0, the input height is used for the output.
If one of the values is -1, the zscale filter will use a value that
maintains the aspect ratio of the input image, calculated from the
other specified dimension. If both of them are -1, the input size
is used
If one of the values is -n with n > 1, the zscale filter will also
use a value that maintains the aspect ratio of the input image,
calculated from the other specified dimension. After that it will,
however, make sure that the calculated dimension is divisible by n
and adjust the value if necessary.
See below for the list of accepted constants for use in the
dimension expression.
size, s
Set the video size. For the syntax of this option, check the "Video
size" section in the ffmpeg-utils manual.
dither, d
Set the dither type.
Possible values are:
none
ordered
random
error_diffusion
Default is none.
filter, f
Set the resize filter type.
Possible values are:
point
bilinear
bicubic
spline16
spline36
lanczos
Default is bilinear.
range, r
Set the color range.
Possible values are:
input
limited
full
Default is same as input.
primaries, p
Set the color primaries.
Possible values are:
input
709
unspecified
170m
240m
2020
Default is same as input.
transfer, t
Set the transfer characteristics.
Possible values are:
input
709
unspecified
601
linear
2020_10
2020_12
Default is same as input.
matrix, m
Set the colorspace matrix.
Possible value are:
input
709
unspecified
470bg
170m
2020_ncl
2020_cl
Default is same as input.
rangein, rin
Set the input color range.
Possible values are:
input
limited
full
Default is same as input.
primariesin, pin
Set the input color primaries.
Possible values are:
input
709
unspecified
170m
240m
2020
Default is same as input.
transferin, tin
Set the input transfer characteristics.
Possible values are:
input
709
unspecified
601
linear
2020_10
2020_12
Default is same as input.
matrixin, min
Set the input colorspace matrix.
Possible value are:
input
709
unspecified
470bg
170m
2020_ncl
2020_cl
chromal, c
Set the output chroma location.
Possible values are:
input
left
center
topleft
top
bottomleft
bottom
chromalin, cin
Set the input chroma location.
Possible values are:
input
left
center
topleft
top
bottomleft
bottom
The values of the w and h options are expressions containing the
following constants:
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 (scaled) width and height
ow
oh These are the same as out_w and out_h
a The same as iw / ih
sar input sample aspect ratio
dar The input display aspect ratio. Calculated from "(iw / ih) * sar".
hsub
vsub
horizontal and vertical input chroma subsample values. For example
for the pixel format "yuv422p" hsub is 2 and vsub is 1.
ohsub
ovsub
horizontal and vertical output chroma subsample values. For example
for the pixel format "yuv422p" hsub is 2 and vsub is 1.
VIDEO SOURCES
Below is a description of the currently available video sources.
buffer
Buffer video frames, and make them available to the filter chain.
This source is mainly intended for a programmatic use, in particular
through the interface defined in libavfilter/vsrc_buffer.h.
It accepts the following parameters:
video_size
Specify the size (width and height) of the buffered video frames.
For the syntax of this option, check the "Video size" section in
the ffmpeg-utils manual.
width
The input video width.
height
The input video height.
pix_fmt
A string representing the pixel format of the buffered video
frames. It may be a number corresponding to a pixel format, or a
pixel format name.
time_base
Specify the timebase assumed by the timestamps of the buffered
frames.
frame_rate
Specify the frame rate expected for the video stream.
pixel_aspect, sar
The sample (pixel) aspect ratio of the input video.
sws_param
Specify the optional parameters to be used for the scale filter
which is automatically inserted when an input change is detected in
the input size or format.
hw_frames_ctx
When using a hardware pixel format, this should be a reference to
an AVHWFramesContext describing input frames.
For example:
buffer=width=320:height=240:pix_fmt=yuv410p:time_base=1/24:sar=1
will instruct the source to accept video frames with size 320x240 and
with format "yuv410p", assuming 1/24 as the timestamps timebase and
square pixels (1:1 sample aspect ratio). Since the pixel format with
name "yuv410p" corresponds to the number 6 (check the enum
AVPixelFormat definition in libavutil/pixfmt.h), this example
corresponds to:
buffer=size=320x240:pixfmt=6:time_base=1/24:pixel_aspect=1/1
Alternatively, the options can be specified as a flat string, but this
syntax is deprecated:
width:height:pix_fmt:time_base.num:time_base.den:pixel_aspect.num:pixel_aspect.den[:sws_param]
cellauto
Create a pattern generated by an elementary cellular automaton.
The initial state of the cellular automaton can be defined through the
filename and pattern options. If such options are not specified an
initial state is created randomly.
At each new frame a new row in the video is filled with the result of
the cellular automaton next generation. The behavior when the whole
frame is filled is defined by the scroll option.
This source accepts the following options:
filename, f
Read the initial cellular automaton state, i.e. the starting row,
from the specified file. In the file, each non-whitespace
character is considered an alive cell, a newline will terminate the
row, and further characters in the file will be ignored.
pattern, p
Read the initial cellular automaton state, i.e. the starting row,
from the specified string.
Each non-whitespace character in the string is considered an alive
cell, a newline will terminate the row, and further characters in
the string will be ignored.
rate, r
Set the video rate, that is the number of frames generated per
second. Default is 25.
random_fill_ratio, ratio
Set the random fill ratio for the initial cellular automaton row.
It is a floating point number value ranging from 0 to 1, defaults
to 1/PHI.
This option is ignored when a file or a pattern is specified.
random_seed, seed
Set the seed for filling randomly the initial row, 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.
rule
Set the cellular automaton rule, it is a number ranging from 0 to
255. Default value is 110.
size, s
Set the size of the output video. For the syntax of this option,
check the "Video size" section in the ffmpeg-utils manual.
If filename or pattern is specified, the size is set by default to
the width of the specified initial state row, and the height is set
to width * PHI.
If size is set, it must contain the width of the specified pattern
string, and the specified pattern will be centered in the larger
row.
If a filename or a pattern string is not specified, the size value
defaults to "320x518" (used for a randomly generated initial
state).
scroll
If set to 1, scroll the output upward when all the rows in the
output have been already filled. If set to 0, the new generated row
will be written over the top row just after the bottom row is
filled. Defaults to 1.
start_full, full
If set to 1, completely fill the output with generated rows before
outputting the first frame. This is the default behavior, for
disabling set the value to 0.
stitch
If set to 1, stitch the left and right row edges together. This is
the default behavior, for disabling set the value to 0.
Examples
* Read the initial state from pattern, and specify an output of size
200x400.
cellauto=f=pattern:s=200x400
* Generate a random initial row with a width of 200 cells, with a
fill ratio of 2/3:
cellauto=ratio=2/3:s=200x200
* Create a pattern generated by rule 18 starting by a single alive
cell centered on an initial row with width 100:
cellauto=p=@s=100x400:full=0:rule=18
* Specify a more elaborated initial pattern:
cellauto=p='@@ @ @@':s=100x400:full=0:rule=18
coreimagesrc
Video source generated on GPU using Apple's CoreImage API on OSX.
This video source is a specialized version of the coreimage video
filter. Use a core image generator at the beginning of the applied
filterchain to generate the content.
The coreimagesrc video source accepts the following options:
list_generators
List all available generators along with all their respective
options as well as possible minimum and maximum values along with
the default values.
list_generators=true
size, s
Specify the size of the sourced video. For the syntax of this
option, check the "Video size" section in the ffmpeg-utils manual.
The default value is "320x240".
rate, r
Specify the frame rate of the sourced video, as the number of
frames generated per second. It has to be a string in the format
frame_rate_num/frame_rate_den, an integer number, a floating point
number or a valid video frame rate abbreviation. The default value
is "25".
sar Set the sample aspect ratio of the sourced video.
duration, d
Set the duration of the sourced video. See the Time duration
section in the ffmpeg-utils(1) manual for the accepted syntax.
If not specified, or the expressed duration is negative, the video
is supposed to be generated forever.
Additionally, all options of the coreimage video filter are accepted.
A complete filterchain can be used for further processing of the
generated input without CPU-HOST transfer. See coreimage documentation
and examples for details.
Examples
* Use CIQRCodeGenerator to create a QR code for the FFmpeg homepage,
given as complete and escaped command-line for Apple's standard
bash shell:
ffmpeg -f lavfi -i coreimagesrc=s=100x100:filter=CIQRCodeGenerator@inputMessage=https\\\\\://FFmpeg.org/@inputCorrectionLevel=H -frames:v 1 QRCode.png
This example is equivalent to the QRCode example of coreimage
without the need for a nullsrc video source.
mandelbrot
Generate a Mandelbrot set fractal, and progressively zoom towards the
point specified with start_x and start_y.
This source accepts the following options:
end_pts
Set the terminal pts value. Default value is 400.
end_scale
Set the terminal scale value. Must be a floating point value.
Default value is 0.3.
inner
Set the inner coloring mode, that is the algorithm used to draw the
Mandelbrot fractal internal region.
It shall assume one of the following values:
black
Set black mode.
convergence
Show time until convergence.
mincol
Set color based on point closest to the origin of the
iterations.
period
Set period mode.
Default value is mincol.
bailout
Set the bailout value. Default value is 10.0.
maxiter
Set the maximum of iterations performed by the rendering algorithm.
Default value is 7189.
outer
Set outer coloring mode. It shall assume one of following values:
iteration_count
Set iteration cound mode.
normalized_iteration_count
set normalized iteration count mode.
Default value is normalized_iteration_count.
rate, r
Set frame rate, expressed as number of frames per second. Default
value is "25".
size, s
Set frame size. For the syntax of this option, check the "Video
size" section in the ffmpeg-utils manual. Default value is
"640x480".
start_scale
Set the initial scale value. Default value is 3.0.
start_x
Set the initial x position. Must be a floating point value between
-100 and 100. Default value is
-0.743643887037158704752191506114774.
start_y
Set the initial y position. Must be a floating point value between
-100 and 100. Default value is
-0.131825904205311970493132056385139.
mptestsrc
Generate various test patterns, as generated by the MPlayer test
filter.
The size of the generated video is fixed, and is 256x256. This source
is useful in particular for testing encoding features.
This source accepts the following options:
rate, r
Specify the frame rate of the sourced video, as the number of
frames generated per second. It has to be a string in the format
frame_rate_num/frame_rate_den, an integer number, a floating point
number or a valid video frame rate abbreviation. The default value
is "25".
duration, d
Set the duration of the sourced video. See the Time duration
section in the ffmpeg-utils(1) manual for the accepted syntax.
If not specified, or the expressed duration is negative, the video
is supposed to be generated forever.
test, t
Set the number or the name of the test to perform. Supported tests
are:
dc_luma
dc_chroma
freq_luma
freq_chroma
amp_luma
amp_chroma
cbp
mv
ring1
ring2
all
Default value is "all", which will cycle through the list of all
tests.
Some examples:
mptestsrc=t=dc_luma
will generate a "dc_luma" test pattern.
frei0r_src
Provide a frei0r source.
To enable compilation of this filter you need to install the frei0r
header and configure FFmpeg with "--enable-frei0r".
This source accepts the following parameters:
size
The size of the video to generate. For the syntax of this option,
check the "Video size" section in the ffmpeg-utils manual.
framerate
The framerate of the generated video. It may be a string of the
form num/den or a frame rate abbreviation.
filter_name
The name to the frei0r source to load. For more information
regarding frei0r and how to set the parameters, read the frei0r
section in the video filters documentation.
filter_params
A '|'-separated list of parameters to pass to the frei0r source.
For example, to generate a frei0r partik0l source with size 200x200 and
frame rate 10 which is overlaid on the overlay filter main input:
frei0r_src=size=200x200:framerate=10:filter_name=partik0l:filter_params=1234 [overlay]; [in][overlay] overlay
life
Generate a life pattern.
This source is based on a generalization of John Conway's life game.
The sourced input represents a life grid, each pixel represents a cell
which can be in one of two possible states, alive or dead. Every cell
interacts with its eight neighbours, which are the cells that are
horizontally, vertically, or diagonally adjacent.
At each interaction the grid evolves according to the adopted rule,
which specifies the number of neighbor alive cells which will make a
cell stay alive or born. The rule option allows one to specify the rule
to adopt.
This source accepts the following options:
filename, f
Set the file from which to read the initial grid state. In the
file, each non-whitespace character is considered an alive cell,
and newline is used to delimit the end of each row.
If this option is not specified, the initial grid is generated
randomly.
rate, r
Set the video rate, that is the number of frames generated per
second. Default is 25.
random_fill_ratio, ratio
Set the random fill ratio for the initial random grid. It is a
floating point number value ranging from 0 to 1, defaults to 1/PHI.
It is ignored when a file is specified.
random_seed, seed
Set the seed for filling the initial random grid, 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.
rule
Set the life rule.
A rule can be specified with a code of the kind "SNS/BNB", where NS
and NB are sequences of numbers in the range 0-8, NS specifies the
number of alive neighbor cells which make a live cell stay alive,
and NB the number of alive neighbor cells which make a dead cell to
become alive (i.e. to "born"). "s" and "b" can be used in place of
"S" and "B", respectively.
Alternatively a rule can be specified by an 18-bits integer. The 9
high order bits are used to encode the next cell state if it is
alive for each number of neighbor alive cells, the low order bits
specify the rule for "borning" new cells. Higher order bits encode
for an higher number of neighbor cells. For example the number
6153 = "(12<<9)+9" specifies a stay alive rule of 12 and a born
rule of 9, which corresponds to "S23/B03".
Default value is "S23/B3", which is the original Conway's game of
life rule, and will keep a cell alive if it has 2 or 3 neighbor
alive cells, and will born a new cell if there are three alive
cells around a dead cell.
size, s
Set the size of the output video. For the syntax of this option,
check the "Video size" section in the ffmpeg-utils manual.
If filename is specified, the size is set by default to the same
size of the input file. If size is set, it must contain the size
specified in the input file, and the initial grid defined in that
file is centered in the larger resulting area.
If a filename is not specified, the size value defaults to
"320x240" (used for a randomly generated initial grid).
stitch
If set to 1, stitch the left and right grid edges together, and the
top and bottom edges also. Defaults to 1.
mold
Set cell mold speed. If set, a dead cell will go from death_color
to mold_color with a step of mold. mold can have a value from 0 to
255.
life_color
Set the color of living (or new born) cells.
death_color
Set the color of dead cells. If mold is set, this is the first
color used to represent a dead cell.
mold_color
Set mold color, for definitely dead and moldy cells.
For the syntax of these 3 color options, check the "Color" section
in the ffmpeg-utils manual.
Examples
* Read a grid from pattern, and center it on a grid of size 300x300
pixels:
life=f=pattern:s=300x300
* Generate a random grid of size 200x200, with a fill ratio of 2/3:
life=ratio=2/3:s=200x200
* Specify a custom rule for evolving a randomly generated grid:
life=rule=S14/B34
* Full example with slow death effect (mold) using ffplay:
ffplay -f lavfi life=s=300x200:mold=10:r=60:ratio=0.1:death_color=#C83232:life_color=#00ff00,scale=1200:800:flags=16
allrgb, allyuv, color, haldclutsrc, nullsrc, rgbtestsrc, smptebars,
smptehdbars, testsrc, testsrc2, yuvtestsrc
The "allrgb" source returns frames of size 4096x4096 of all rgb colors.
The "allyuv" source returns frames of size 4096x4096 of all yuv colors.
The "color" source provides an uniformly colored input.
The "haldclutsrc" source provides an identity Hald CLUT. See also
haldclut filter.
The "nullsrc" source returns unprocessed video frames. It is mainly
useful to be employed in analysis / debugging tools, or as the source
for filters which ignore the input data.
The "rgbtestsrc" source generates an RGB test pattern useful for
detecting RGB vs BGR issues. You should see a red, green and blue
stripe from top to bottom.
The "smptebars" source generates a color bars pattern, based on the
SMPTE Engineering Guideline EG 1-1990.
The "smptehdbars" source generates a color bars pattern, based on the
SMPTE RP 219-2002.
The "testsrc" source generates a test video pattern, showing a color
pattern, a scrolling gradient and a timestamp. This is mainly intended
for testing purposes.
The "testsrc2" source is similar to testsrc, but supports more pixel
formats instead of just "rgb24". This allows using it as an input for
other tests without requiring a format conversion.
The "yuvtestsrc" source generates an YUV test pattern. You should see a
y, cb and cr stripe from top to bottom.
The sources accept the following parameters:
color, c
Specify the color of the source, only available in the "color"
source. For the syntax of this option, check the "Color" section in
the ffmpeg-utils manual.
level
Specify the level of the Hald CLUT, only available in the
"haldclutsrc" source. A level of "N" generates a picture of "N*N*N"
by "N*N*N" pixels to be used as identity matrix for 3D lookup
tables. Each component is coded on a "1/(N*N)" scale.
size, s
Specify the size of the sourced video. For the syntax of this
option, check the "Video size" section in the ffmpeg-utils manual.
The default value is "320x240".
This option is not available with the "haldclutsrc" filter.
rate, r
Specify the frame rate of the sourced video, as the number of
frames generated per second. It has to be a string in the format
frame_rate_num/frame_rate_den, an integer number, a floating point
number or a valid video frame rate abbreviation. The default value
is "25".
sar Set the sample aspect ratio of the sourced video.
duration, d
Set the duration of the sourced video. See the Time duration
section in the ffmpeg-utils(1) manual for the accepted syntax.
If not specified, or the expressed duration is negative, the video
is supposed to be generated forever.
decimals, n
Set the number of decimals to show in the timestamp, only available
in the "testsrc" source.
The displayed timestamp value will correspond to the original
timestamp value multiplied by the power of 10 of the specified
value. Default value is 0.
For example the following:
testsrc=duration=5.3:size=qcif:rate=10
will generate a video with a duration of 5.3 seconds, with size 176x144
and a frame rate of 10 frames per second.
The following graph description will generate a red source with an
opacity of 0.2, with size "qcif" and a frame rate of 10 frames per
second.
color=c=[email protected]:s=qcif:r=10
If the input content is to be ignored, "nullsrc" can be used. The
following command generates noise in the luminance plane by employing
the "geq" filter:
nullsrc=s=256x256, geq=random(1)*255:128:128
Commands
The "color" source supports the following commands:
c, color
Set the color of the created image. Accepts the same syntax of the
corresponding color option.
VIDEO SINKS
Below is a description of the currently available video sinks. buffersink Buffer video frames, and make them available to the end of the filter graph. 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 AVBufferSinkContext structure, which defines the incoming buffers' formats, to be passed as the opaque parameter to "avfilter_init_filter" for initialization. nullsink Null video sink: do absolutely nothing with the input video. It is mainly useful as a template and for use in analysis / debugging tools.
MULTIMEDIA FILTERS
Below is a description of the currently available multimedia filters.
ahistogram
Convert input audio to a video output, displaying the volume histogram.
The filter accepts the following options:
dmode
Specify how histogram is calculated.
It accepts the following values:
single
Use single histogram for all channels.
separate
Use separate histogram for each channel.
Default is "single".
rate, r
Set frame rate, expressed as number of frames per second. Default
value is "25".
size, s
Specify the video size for the output. For the syntax of this
option, check the "Video size" section in the ffmpeg-utils manual.
Default value is "hd720".
scale
Set display scale.
It accepts the following values:
log logarithmic
sqrt
square root
cbrt
cubic root
lin linear
rlog
reverse logarithmic
Default is "log".
ascale
Set amplitude scale.
It accepts the following values:
log logarithmic
lin linear
Default is "log".
acount
Set how much frames to accumulate in histogram. Defauls is 1.
Setting this to -1 accumulates all frames.
rheight
Set histogram ratio of window height.
slide
Set sonogram sliding.
It accepts the following values:
replace
replace old rows with new ones.
scroll
scroll from top to bottom.
Default is "replace".
aphasemeter
Convert input audio to a video output, displaying the audio phase.
The filter accepts the following options:
rate, r
Set the output frame rate. Default value is 25.
size, s
Set the video size for the output. For the syntax of this option,
check the "Video size" section in the ffmpeg-utils manual. Default
value is "800x400".
rc
gc
bc Specify the red, green, blue contrast. Default values are 2, 7 and
1. Allowed range is "[0, 255]".
mpc Set color which will be used for drawing median phase. If color is
"none" which is default, no median phase value will be drawn.
The filter also exports the frame metadata "lavfi.aphasemeter.phase"
which represents mean phase of current audio frame. Value is in range
"[-1, 1]". The "-1" means left and right channels are completely out
of phase and 1 means channels are in phase.
avectorscope
Convert input audio to a video output, representing the audio vector
scope.
The filter is used to measure the difference between channels of stereo
audio stream. A monoaural signal, consisting of identical left and
right signal, results in straight vertical line. Any stereo separation
is visible as a deviation from this line, creating a Lissajous figure.
If the straight (or deviation from it) but horizontal line appears this
indicates that the left and right channels are out of phase.
The filter accepts the following options:
mode, m
Set the vectorscope mode.
Available values are:
lissajous
Lissajous rotated by 45 degrees.
lissajous_xy
Same as above but not rotated.
polar
Shape resembling half of circle.
Default value is lissajous.
size, s
Set the video size for the output. For the syntax of this option,
check the "Video size" section in the ffmpeg-utils manual. Default
value is "400x400".
rate, r
Set the output frame rate. Default value is 25.
rc
gc
bc
ac Specify the red, green, blue and alpha contrast. Default values are
40, 160, 80 and 255. Allowed range is "[0, 255]".
rf
gf
bf
af Specify the red, green, blue and alpha fade. Default values are 15,
10, 5 and 5. Allowed range is "[0, 255]".
zoom
Set the zoom factor. Default value is 1. Allowed range is "[1,
10]".
draw
Set the vectorscope drawing mode.
Available values are:
dot Draw dot for each sample.
line
Draw line between previous and current sample.
Default value is dot.
scale
Specify amplitude scale of audio samples.
Available values are:
lin Linear.
sqrt
Square root.
cbrt
Cubic root.
log Logarithmic.
Examples
* Complete example using ffplay:
ffplay -f lavfi 'amovie=input.mp3, asplit [a][out1];
[a] avectorscope=zoom=1.3:rc=2:gc=200:bc=10:rf=1:gf=8:bf=7 [out0]'
bench, abench
Benchmark part of a filtergraph.
The filter accepts the following options:
action
Start or stop a timer.
Available values are:
start
Get the current time, set it as frame metadata (using the key
"lavfi.bench.start_time"), and forward the frame to the next
filter.
stop
Get the current time and fetch the "lavfi.bench.start_time"
metadata from the input frame metadata to get the time
difference. Time difference, average, maximum and minimum time
(respectively "t", "avg", "max" and "min") are then printed.
The timestamps are expressed in seconds.
Examples
* Benchmark selectivecolor filter:
bench=start,selectivecolor=reds=-.2 .12 -.49,bench=stop
concat
Concatenate audio and video streams, joining them together one after
the other.
The filter works on segments of synchronized video and audio streams.
All segments must have the same number of streams of each type, and
that will also be the number of streams at output.
The filter accepts the following options:
n Set the number of segments. Default is 2.
v Set the number of output video streams, that is also the number of
video streams in each segment. Default is 1.
a Set the number of output audio streams, that is also the number of
audio streams in each segment. Default is 0.
unsafe
Activate unsafe mode: do not fail if segments have a different
format.
The filter has v+a outputs: first v video outputs, then a audio
outputs.
There are nx(v+a) inputs: first the inputs for the first segment, in
the same order as the outputs, then the inputs for the second segment,
etc.
Related streams do not always have exactly the same duration, for
various reasons including codec frame size or sloppy authoring. For
that reason, related synchronized streams (e.g. a video and its audio
track) should be concatenated at once. The concat filter will use the
duration of the longest stream in each segment (except the last one),
and if necessary pad shorter audio streams with silence.
For this filter to work correctly, all segments must start at timestamp
0.
All corresponding streams must have the same parameters in all
segments; the filtering system will automatically select a common pixel
format for video streams, and a common sample format, sample rate and
channel layout for audio streams, but other settings, such as
resolution, must be converted explicitly by the user.
Different frame rates are acceptable but will result in variable frame
rate at output; be sure to configure the output file to handle it.
Examples
* Concatenate an opening, an episode and an ending, all in bilingual
version (video in stream 0, audio in streams 1 and 2):
ffmpeg -i opening.mkv -i episode.mkv -i ending.mkv -filter_complex \
'[0:0] [0:1] [0:2] [1:0] [1:1] [1:2] [2:0] [2:1] [2:2]
concat=n=3:v=1:a=2 [v] [a1] [a2]' \
-map '[v]' -map '[a1]' -map '[a2]' output.mkv
* Concatenate two parts, handling audio and video separately, using
the (a)movie sources, and adjusting the resolution:
movie=part1.mp4, scale=512:288 [v1] ; amovie=part1.mp4 [a1] ;
movie=part2.mp4, scale=512:288 [v2] ; amovie=part2.mp4 [a2] ;
[v1] [v2] concat [outv] ; [a1] [a2] concat=v=0:a=1 [outa]
Note that a desync will happen at the stitch if the audio and video
streams do not have exactly the same duration in the first file.
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.
picture
Draw single picture.
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
ebur128
EBU R128 scanner filter. This filter takes an audio stream as input and
outputs it unchanged. By default, it logs a message at a frequency of
10Hz with the Momentary loudness (identified by "M"), Short-term
loudness ("S"), Integrated loudness ("I") and Loudness Range ("LRA").
The filter also has a video output (see the video option) with a real
time graph to observe the loudness evolution. The graphic contains the
logged message mentioned above, so it is not printed anymore when this
option is set, unless the verbose logging is set. The main graphing
area contains the short-term loudness (3 seconds of analysis), and the
gauge on the right is for the momentary loudness (400 milliseconds).
More information about the Loudness Recommendation EBU R128 on
<http://tech.ebu.ch/loudness>.
The filter accepts the following options:
video
Activate the video output. The audio stream is passed unchanged
whether this option is set or no. The video stream will be the
first output stream if activated. Default is 0.
size
Set the video size. This option is for video only. For the syntax
of this option, check the "Video size" section in the ffmpeg-utils
manual. Default and minimum resolution is "640x480".
meter
Set the EBU scale meter. Default is 9. Common values are 9 and 18,
respectively for EBU scale meter +9 and EBU scale meter +18. Any
other integer value between this range is allowed.
metadata
Set metadata injection. If set to 1, the audio input will be
segmented into 100ms output frames, each of them containing various
loudness information in metadata. All the metadata keys are
prefixed with "lavfi.r128.".
Default is 0.
framelog
Force the frame logging level.
Available values are:
info
information logging level
verbose
verbose logging level
By default, the logging level is set to info. If the video or the
metadata options are set, it switches to verbose.
peak
Set peak mode(s).
Available modes can be cumulated (the option is a "flag" type).
Possible values are:
none
Disable any peak mode (default).
sample
Enable sample-peak mode.
Simple peak mode looking for the higher sample value. It logs a
message for sample-peak (identified by "SPK").
true
Enable true-peak mode.
If enabled, the peak lookup is done on an over-sampled version
of the input stream for better peak accuracy. It logs a message
for true-peak. (identified by "TPK") and true-peak per frame
(identified by "FTPK"). This mode requires a build with
"libswresample".
dualmono
Treat mono input files as "dual mono". If a mono file is intended
for playback on a stereo system, its EBU R128 measurement will be
perceptually incorrect. If set to "true", this option will
compensate for this effect. Multi-channel input files are not
affected by this option.
panlaw
Set a specific pan law to be used for the measurement of dual mono
files. This parameter is optional, and has a default value of
-3.01dB.
Examples
* Real-time graph using ffplay, with a EBU scale meter +18:
ffplay -f lavfi -i "amovie=input.mp3,ebur128=video=1:meter=18 [out0][out1]"
* Run an analysis with ffmpeg:
ffmpeg -nostats -i input.mp3 -filter_complex ebur128 -f null -
interleave, ainterleave
Temporally interleave frames from several inputs.
"interleave" works with video inputs, "ainterleave" with audio.
These filters read frames from several inputs and send the oldest
queued frame to the output.
Input streams must have well defined, monotonically increasing frame
timestamp values.
In order to submit one frame to output, these filters need to enqueue
at least one frame for each input, so they cannot work in case one
input is not yet terminated and will not receive incoming frames.
For example consider the case when one input is a "select" filter which
always drops input frames. The "interleave" filter will keep reading
from that input, but it will never be able to send new frames to output
until the input sends an end-of-stream signal.
Also, depending on inputs synchronization, the filters will drop frames
in case one input receives more frames than the other ones, and the
queue is already filled.
These filters accept the following options:
nb_inputs, n
Set the number of different inputs, it is 2 by default.
Examples
* Interleave frames belonging to different streams using ffmpeg:
ffmpeg -i bambi.avi -i pr0n.mkv -filter_complex "[0:v][1:v] interleave" out.avi
* Add flickering blur effect:
select='if(gt(random(0), 0.2), 1, 2)':n=2 [tmp], boxblur=2:2, [tmp] interleave
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. If "key" is not set, delete all metadata
values in the frame.
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" and "delete".
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. Instead of plain filename any writable url can be
specified. Filename ``-'' is a shorthand for 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)'
* Print silencedetect output to file metadata.txt.
silencedetect,ametadata=mode=print:file=metadata.txt
* Direct all metadata to a pipe with file descriptor 4.
metadata=mode=print:file='pipe\:4'
perms, aperms
Set read/write permissions for the output frames.
These filters are mainly aimed at developers to test direct path in the
following filter in the filtergraph.
The filters accept the following options:
mode
Select the permissions mode.
It accepts the following values:
none
Do nothing. This is the default.
ro Set all the output frames read-only.
rw Set all the output frames directly writable.
toggle
Make the frame read-only if writable, and writable if read-
only.
random
Set each output frame read-only or writable randomly.
seed
Set the seed for the random mode, 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.
Note: in case of auto-inserted filter between the permission filter and
the following one, the permission might not be received as expected in
that following filter. Inserting a format or aformat filter before the
perms/aperms filter can avoid this problem.
realtime, arealtime
Slow down filtering to match real time approximatively.
These filters will pause the filtering for a variable amount of time to
match the output rate with the input timestamps. They are similar to
the re option to "ffmpeg".
They accept the following options:
limit
Time limit for the pauses. Any pause longer than that will be
considered a timestamp discontinuity and reset the timer. Default
is 2 seconds.
select, aselect
Select frames to pass in output.
This filter accepts the following options:
expr, e
Set expression, which is evaluated for each input frame.
If the expression is evaluated to zero, the frame is discarded.
If the evaluation result is negative or NaN, the frame is sent to
the first output; otherwise it is sent to the output with index
"ceil(val)-1", assuming that the input index starts from 0.
For example a value of 1.2 corresponds to the output with index
"ceil(1.2)-1 = 2-1 = 1", that is the second output.
outputs, n
Set the number of outputs. The output to which to send the selected
frame is based on the result of the evaluation. Default value is 1.
The expression can contain the following constants:
n The (sequential) number of the filtered frame, starting from 0.
selected_n
The (sequential) number of the selected frame, starting from 0.
prev_selected_n
The sequential number of the last selected frame. It's NAN if
undefined.
TB The timebase of the input timestamps.
pts The PTS (Presentation TimeStamp) of the filtered video frame,
expressed in TB units. It's NAN if undefined.
t The PTS of the filtered video frame, expressed in seconds. It's NAN
if undefined.
prev_pts
The PTS of the previously filtered video frame. It's NAN if
undefined.
prev_selected_pts
The PTS of the last previously filtered video frame. It's NAN if
undefined.
prev_selected_t
The PTS of the last previously selected video frame. It's NAN if
undefined.
start_pts
The PTS of the first video frame in the video. It's NAN if
undefined.
start_t
The time of the first video frame in the video. It's NAN if
undefined.
pict_type (video only)
The type of the filtered frame. It can assume one of the following
values:
I
P
B
S
SI
SP
BI
interlace_type (video only)
The frame interlace type. It can assume one of the following
values:
PROGRESSIVE
The frame is progressive (not interlaced).
TOPFIRST
The frame is top-field-first.
BOTTOMFIRST
The frame is bottom-field-first.
consumed_sample_n (audio only)
the number of selected samples before the current frame
samples_n (audio only)
the number of samples in the current frame
sample_rate (audio only)
the input sample rate
key This is 1 if the filtered frame is a key-frame, 0 otherwise.
pos the position in the file of the filtered frame, -1 if the
information is not available (e.g. for synthetic video)
scene (video only)
value between 0 and 1 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 (see the example below)
concatdec_select
The concat demuxer can select only part of a concat input file by
setting an inpoint and an outpoint, but the output packets may not
be entirely contained in the selected interval. By using this
variable, it is possible to skip frames generated by the concat
demuxer which are not exactly contained in the selected interval.
This works by comparing the frame pts against the
lavf.concat.start_time and the lavf.concat.duration packet metadata
values which are also present in the decoded frames.
The concatdec_select variable is -1 if the frame pts is at least
start_time and either the duration metadata is missing or the frame
pts is less than start_time + duration, 0 otherwise, and NaN if the
start_time metadata is missing.
That basically means that an input frame is selected if its pts is
within the interval set by the concat demuxer.
The default value of the select expression is "1".
Examples
* Select all frames in input:
select
The example above is the same as:
select=1
* Skip all frames:
select=0
* Select only I-frames:
select='eq(pict_type\,I)'
* Select one frame every 100:
select='not(mod(n\,100))'
* Select only frames contained in the 10-20 time interval:
select=between(t\,10\,20)
* Select only I-frames contained in the 10-20 time interval:
select=between(t\,10\,20)*eq(pict_type\,I)
* Select frames with a minimum distance of 10 seconds:
select='isnan(prev_selected_t)+gte(t-prev_selected_t\,10)'
* Use aselect to select only audio frames with samples number > 100:
aselect='gt(samples_n\,100)'
* Create a mosaic of the first scenes:
ffmpeg -i video.avi -vf select='gt(scene\,0.4)',scale=160:120,tile -frames:v 1 preview.png
Comparing scene against a value between 0.3 and 0.5 is generally a
sane choice.
* Send even and odd frames to separate outputs, and compose them:
select=n=2:e='mod(n, 2)+1' [odd][even]; [odd] pad=h=2*ih [tmp]; [tmp][even] overlay=y=h
* Select useful frames from an ffconcat file which is using inpoints
and outpoints but where the source files are not intra frame only.
ffmpeg -copyts -vsync 0 -segment_time_metadata 1 -i input.ffconcat -vf select=concatdec_select -af aselect=concatdec_select output.avi
sendcmd, asendcmd
Send commands to filters in the filtergraph.
These filters read commands to be sent to other filters in the
filtergraph.
"sendcmd" must be inserted between two video filters, "asendcmd" must
be inserted between two audio filters, but apart from that they act the
same way.
The specification of commands can be provided in the filter arguments
with the commands option, or in a file specified by the filename
option.
These filters accept the following options:
commands, c
Set the commands to be read and sent to the other filters.
filename, f
Set the filename of the commands to be read and sent to the other
filters.
Commands syntax
A commands description consists of a sequence of interval
specifications, comprising a list of commands to be executed when a
particular event related to that interval occurs. The occurring event
is typically the current frame time entering or leaving a given time
interval.
An interval is specified by the following syntax:
<START>[-<END>] <COMMANDS>;
The time interval is specified by the START and END times. END is
optional and defaults to the maximum time.
The current frame time is considered within the specified interval if
it is included in the interval [START, END), that is when the time is
greater or equal to START and is lesser than END.
COMMANDS consists of a sequence of one or more command specifications,
separated by ",", relating to that interval. The syntax of a command
specification is given by:
[<FLAGS>] <TARGET> <COMMAND> <ARG>
FLAGS is optional and specifies the type of events relating to the time
interval which enable sending the specified command, and must be a non-
null sequence of identifier flags separated by "+" or "|" and enclosed
between "[" and "]".
The following flags are recognized:
enter
The command is sent when the current frame timestamp enters the
specified interval. In other words, the command is sent when the
previous frame timestamp was not in the given interval, and the
current is.
leave
The command is sent when the current frame timestamp leaves the
specified interval. In other words, the command is sent when the
previous frame timestamp was in the given interval, and the current
is not.
If FLAGS is not specified, a default value of "[enter]" is assumed.
TARGET specifies the target of the command, usually the name of the
filter class or a specific filter instance name.
COMMAND specifies the name of the command for the target filter.
ARG is optional and specifies the optional list of argument for the
given COMMAND.
Between one interval specification and another, whitespaces, or
sequences of characters starting with "#" until the end of line, are
ignored and can be used to annotate comments.
A simplified BNF description of the commands specification syntax
follows:
<COMMAND_FLAG> ::= "enter" | "leave"
<COMMAND_FLAGS> ::= <COMMAND_FLAG> [(+|"|")<COMMAND_FLAG>]
<COMMAND> ::= ["[" <COMMAND_FLAGS> "]"] <TARGET> <COMMAND> [<ARG>]
<COMMANDS> ::= <COMMAND> [,<COMMANDS>]
<INTERVAL> ::= <START>[-<END>] <COMMANDS>
<INTERVALS> ::= <INTERVAL>[;<INTERVALS>]
Examples
* Specify audio tempo change at second 4:
asendcmd=c='4.0 atempo tempo 1.5',atempo
* Specify a list of drawtext and hue commands in a file.
# show text in the interval 5-10
5.0-10.0 [enter] drawtext reinit 'fontfile=FreeSerif.ttf:text=hello world',
[leave] drawtext reinit 'fontfile=FreeSerif.ttf:text=';
# desaturate the image in the interval 15-20
15.0-20.0 [enter] hue s 0,
[enter] drawtext reinit 'fontfile=FreeSerif.ttf:text=nocolor',
[leave] hue s 1,
[leave] drawtext reinit 'fontfile=FreeSerif.ttf:text=color';
# apply an exponential saturation fade-out effect, starting from time 25
25 [enter] hue s exp(25-t)
A filtergraph allowing to read and process the above command list
stored in a file test.cmd, can be specified with:
sendcmd=f=test.cmd,drawtext=fontfile=FreeSerif.ttf:text='',hue
setpts, asetpts
Change the PTS (presentation timestamp) of the input frames.
"setpts" works on video frames, "asetpts" on audio frames.
This filter accepts the following options:
expr
The expression which is evaluated for each frame to construct its
timestamp.
The expression is evaluated through the eval API and can contain the
following constants:
FRAME_RATE
frame rate, only defined for constant frame-rate video
PTS The presentation timestamp in input
N The count of the input frame for video or the number of consumed
samples, not including the current frame for audio, starting from
0.
NB_CONSUMED_SAMPLES
The number of consumed samples, not including the current frame
(only audio)
NB_SAMPLES, S
The number of samples in the current frame (only audio)
SAMPLE_RATE, SR
The audio sample rate.
STARTPTS
The PTS of the first frame.
STARTT
the time in seconds of the first frame
INTERLACED
State whether the current frame is interlaced.
T the time in seconds of the current frame
POS original position in the file of the frame, or undefined if
undefined for the current frame
PREV_INPTS
The previous input PTS.
PREV_INT
previous input time in seconds
PREV_OUTPTS
The previous output PTS.
PREV_OUTT
previous output time in seconds
RTCTIME
The wallclock (RTC) time in microseconds. This is deprecated, use
time(0) instead.
RTCSTART
The wallclock (RTC) time at the start of the movie in microseconds.
TB The timebase of the input timestamps.
Examples
* Start counting PTS from zero
setpts=PTS-STARTPTS
* Apply fast motion effect:
setpts=0.5*PTS
* Apply slow motion effect:
setpts=2.0*PTS
* Set fixed rate of 25 frames per second:
setpts=N/(25*TB)
* Set fixed rate 25 fps with some jitter:
setpts='1/(25*TB) * (N + 0.05 * sin(N*2*PI/25))'
* Apply an offset of 10 seconds to the input PTS:
setpts=PTS+10/TB
* Generate timestamps from a "live source" and rebase onto the
current timebase:
setpts='(RTCTIME - RTCSTART) / (TB * 1000000)'
* Generate timestamps by counting samples:
asetpts=N/SR/TB
settb, asettb
Set the timebase to use for the output frames timestamps. It is mainly
useful for testing timebase configuration.
It accepts the following parameters:
expr, tb
The expression which is evaluated into the output timebase.
The value for tb is an arithmetic expression representing a rational.
The expression can contain the constants "AVTB" (the default timebase),
"intb" (the input timebase) and "sr" (the sample rate, audio only).
Default value is "intb".
Examples
* Set the timebase to 1/25:
settb=expr=1/25
* Set the timebase to 1/10:
settb=expr=0.1
* Set the timebase to 1001/1000:
settb=1+0.001
* Set the timebase to 2*intb:
settb=2*intb
* Set the default timebase value:
settb=AVTB
showcqt
Convert input audio to a video output representing frequency spectrum
logarithmically using Brown-Puckette constant Q transform algorithm
with direct frequency domain coefficient calculation (but the transform
itself is not really constant Q, instead the Q factor is actually
variable/clamped), with musical tone scale, from E0 to D#10.
The filter accepts the following options:
size, s
Specify the video size for the output. It must be even. For the
syntax of this option, check the "Video size" section in the
ffmpeg-utils manual. Default value is "1920x1080".
fps, rate, r
Set the output frame rate. Default value is 25.
bar_h
Set the bargraph height. It must be even. Default value is "-1"
which computes the bargraph height automatically.
axis_h
Set the axis height. It must be even. Default value is "-1" which
computes the axis height automatically.
sono_h
Set the sonogram height. It must be even. Default value is "-1"
which computes the sonogram height automatically.
fullhd
Set the fullhd resolution. This option is deprecated, use size, s
instead. Default value is 1.
sono_v, volume
Specify the sonogram volume expression. It can contain variables:
bar_v
the bar_v evaluated expression
frequency, freq, f
the frequency where it is evaluated
timeclamp, tc
the value of timeclamp option
and functions:
a_weighting(f)
A-weighting of equal loudness
b_weighting(f)
B-weighting of equal loudness
c_weighting(f)
C-weighting of equal loudness.
Default value is 16.
bar_v, volume2
Specify the bargraph volume expression. It can contain variables:
sono_v
the sono_v evaluated expression
frequency, freq, f
the frequency where it is evaluated
timeclamp, tc
the value of timeclamp option
and functions:
a_weighting(f)
A-weighting of equal loudness
b_weighting(f)
B-weighting of equal loudness
c_weighting(f)
C-weighting of equal loudness.
Default value is "sono_v".
sono_g, gamma
Specify the sonogram gamma. Lower gamma makes the spectrum more
contrast, higher gamma makes the spectrum having more range.
Default value is 3. Acceptable range is "[1, 7]".
bar_g, gamma2
Specify the bargraph gamma. Default value is 1. Acceptable range is
"[1, 7]".
timeclamp, tc
Specify the transform timeclamp. At low frequency, there is trade-
off between accuracy in time domain and frequency domain. If
timeclamp is lower, event in time domain is represented more
accurately (such as fast bass drum), otherwise event in frequency
domain is represented more accurately (such as bass guitar).
Acceptable range is "[0.1, 1]". Default value is 0.17.
basefreq
Specify the transform base frequency. Default value is
20.01523126408007475, which is frequency 50 cents below E0.
Acceptable range is "[10, 100000]".
endfreq
Specify the transform end frequency. Default value is
20495.59681441799654, which is frequency 50 cents above D#10.
Acceptable range is "[10, 100000]".
coeffclamp
This option is deprecated and ignored.
tlength
Specify the transform length in time domain. Use this option to
control accuracy trade-off between time domain and frequency domain
at every frequency sample. It can contain variables:
frequency, freq, f
the frequency where it is evaluated
timeclamp, tc
the value of timeclamp option.
Default value is "384*tc/(384+tc*f)".
count
Specify the transform count for every video frame. Default value is
6. Acceptable range is "[1, 30]".
fcount
Specify the transform count for every single pixel. Default value
is 0, which makes it computed automatically. Acceptable range is
"[0, 10]".
fontfile
Specify font file for use with freetype to draw the axis. If not
specified, use embedded font. Note that drawing with font file or
embedded font is not implemented with custom basefreq and endfreq,
use axisfile option instead.
font
Specify fontconfig pattern. This has lower priority than fontfile.
The : in the pattern may be replaced by | to avoid unnecessary
escaping.
fontcolor
Specify font color expression. This is arithmetic expression that
should return integer value 0xRRGGBB. It can contain variables:
frequency, freq, f
the frequency where it is evaluated
timeclamp, tc
the value of timeclamp option
and functions:
midi(f)
midi number of frequency f, some midi numbers: E0(16), C1(24),
C2(36), A4(69)
r(x), g(x), b(x)
red, green, and blue value of intensity x.
Default value is "st(0, (midi(f)-59.5)/12); st(1,
if(between(ld(0),0,1), 0.5-0.5*cos(2*PI*ld(0)), 0)); r(1-ld(1)) +
b(ld(1))".
axisfile
Specify image file to draw the axis. This option override fontfile
and fontcolor option.
axis, text
Enable/disable drawing text to the axis. If it is set to 0, drawing
to the axis is disabled, ignoring fontfile and axisfile option.
Default value is 1.
csp Set colorspace. The accepted values are:
unspecified
Unspecified (default)
bt709
BT.709
fcc FCC
bt470bg
BT.470BG or BT.601-6 625
smpte170m
SMPTE-170M or BT.601-6 525
smpte240m
SMPTE-240M
bt2020ncl
BT.2020 with non-constant luminance
cscheme
Set spectrogram color scheme. This is list of floating point values
with format "left_r|left_g|left_b|right_r|right_g|right_b". The
default is "1|0.5|0|0|0.5|1".
Examples
* Playing audio while showing the spectrum:
ffplay -f lavfi 'amovie=a.mp3, asplit [a][out1]; [a] showcqt [out0]'
* Same as above, but with frame rate 30 fps:
ffplay -f lavfi 'amovie=a.mp3, asplit [a][out1]; [a] showcqt=fps=30:count=5 [out0]'
* Playing at 1280x720:
ffplay -f lavfi 'amovie=a.mp3, asplit [a][out1]; [a] showcqt=s=1280x720:count=4 [out0]'
* Disable sonogram display:
sono_h=0
* A1 and its harmonics: A1, A2, (near)E3, A3:
ffplay -f lavfi 'aevalsrc=0.1*sin(2*PI*55*t)+0.1*sin(4*PI*55*t)+0.1*sin(6*PI*55*t)+0.1*sin(8*PI*55*t),
asplit[a][out1]; [a] showcqt [out0]'
* Same as above, but with more accuracy in frequency domain:
ffplay -f lavfi 'aevalsrc=0.1*sin(2*PI*55*t)+0.1*sin(4*PI*55*t)+0.1*sin(6*PI*55*t)+0.1*sin(8*PI*55*t),
asplit[a][out1]; [a] showcqt=timeclamp=0.5 [out0]'
* Custom volume:
bar_v=10:sono_v=bar_v*a_weighting(f)
* Custom gamma, now spectrum is linear to the amplitude.
bar_g=2:sono_g=2
* Custom tlength equation:
tc=0.33:tlength='st(0,0.17); 384*tc / (384 / ld(0) + tc*f /(1-ld(0))) + 384*tc / (tc*f / ld(0) + 384 /(1-ld(0)))'
* Custom fontcolor and fontfile, C-note is colored green, others are
colored blue:
fontcolor='if(mod(floor(midi(f)+0.5),12), 0x0000FF, g(1))':fontfile=myfont.ttf
* Custom font using fontconfig:
font='Courier New,Monospace,mono|bold'
* Custom frequency range with custom axis using image file:
axisfile=myaxis.png:basefreq=40:endfreq=10000
showfreqs
Convert input audio to video output representing the audio power
spectrum. Audio amplitude is on Y-axis while frequency is on X-axis.
The filter accepts the following options:
size, s
Specify size of video. For the syntax of this option, check the
"Video size" section in the ffmpeg-utils manual. Default is
"1024x512".
mode
Set display mode. This set how each frequency bin will be
represented.
It accepts the following values:
line
bar
dot
Default is "bar".
ascale
Set amplitude scale.
It accepts the following values:
lin Linear scale.
sqrt
Square root scale.
cbrt
Cubic root scale.
log Logarithmic scale.
Default is "log".
fscale
Set frequency scale.
It accepts the following values:
lin Linear scale.
log Logarithmic scale.
rlog
Reverse logarithmic scale.
Default is "lin".
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 "w2048"
win_func
Set windowing function.
It accepts the following values:
rect
bartlett
hanning
hamming
blackman
welch
flattop
bharris
bnuttall
bhann
sine
nuttall
lanczos
gauss
tukey
dolph
cauchy
parzen
poisson
Default is "hanning".
overlap
Set window overlap. In range "[0, 1]". Default is 1, which means
optimal overlap for selected window function will be picked.
averaging
Set time averaging. Setting this to 0 will display current maximal
peaks. Default is 1, which means time averaging is disabled.
colors
Specify list of colors separated by space or by '|' which will be
used to draw channel frequencies. Unrecognized or missing colors
will be replaced by white color.
cmode
Set channel display mode.
It accepts the following values:
combined
separate
Default is "combined".
minamp
Set minimum amplitude used in "log" amplitude scaler.
showspectrum
Convert input audio to a video output, representing the audio frequency
spectrum.
The filter accepts the following options:
size, s
Specify the video size for the output. For the syntax of this
option, check the "Video size" section in the ffmpeg-utils manual.
Default value is "640x512".
slide
Specify how the spectrum should slide along the window.
It accepts the following values:
replace
the samples start again on the left when they reach the right
scroll
the samples scroll from right to left
fullframe
frames are only produced when the samples reach the right
rscroll
the samples scroll from left to right
Default value is "replace".
mode
Specify display mode.
It accepts the following values:
combined
all channels are displayed in the same row
separate
all channels are displayed in separate rows
Default value is combined.
color
Specify display color mode.
It accepts the following values:
channel
each channel is displayed in a separate color
intensity
each channel is displayed using the same color scheme
rainbow
each channel is displayed using the rainbow color scheme
moreland
each channel is displayed using the moreland color scheme
nebulae
each channel is displayed using the nebulae color scheme
fire
each channel is displayed using the fire color scheme
fiery
each channel is displayed using the fiery color scheme
fruit
each channel is displayed using the fruit color scheme
cool
each channel is displayed using the cool color scheme
Default value is channel.
scale
Specify scale used for calculating intensity color values.
It accepts the following values:
lin linear
sqrt
square root, default
cbrt
cubic root
log logarithmic
4thrt
4th root
5thrt
5th root
Default value is sqrt.
saturation
Set saturation modifier for displayed colors. Negative values
provide alternative color scheme. 0 is no saturation at all.
Saturation must be in [-10.0, 10.0] range. Default value is 1.
win_func
Set window function.
It accepts the following values:
rect
bartlett
hann
hanning
hamming
blackman
welch
flattop
bharris
bnuttall
bhann
sine
nuttall
lanczos
gauss
tukey
dolph
cauchy
parzen
poisson
Default value is "hann".
orientation
Set orientation of time vs frequency axis. Can be "vertical" or
"horizontal". Default is "vertical".
overlap
Set ratio of overlap window. Default value is 0. When value is 1
overlap is set to recommended size for specific window function
currently used.
gain
Set scale gain for calculating intensity color values. Default
value is 1.
data
Set which data to display. Can be "magnitude", default or "phase".
rotation
Set color rotation, must be in [-1.0, 1.0] range. Default value is
0.
The usage is very similar to the showwaves filter; see the examples in
that section.
Examples
* Large window with logarithmic color scaling:
showspectrum=s=1280x480:scale=log
* Complete example for a colored and sliding spectrum per channel
using ffplay:
ffplay -f lavfi 'amovie=input.mp3, asplit [a][out1];
[a] showspectrum=mode=separate:color=intensity:slide=1:scale=cbrt [out0]'
showspectrumpic
Convert input audio to a single video frame, representing the audio
frequency spectrum.
The filter accepts the following options:
size, s
Specify the video size for the output. For the syntax of this
option, check the "Video size" section in the ffmpeg-utils manual.
Default value is "4096x2048".
mode
Specify display mode.
It accepts the following values:
combined
all channels are displayed in the same row
separate
all channels are displayed in separate rows
Default value is combined.
color
Specify display color mode.
It accepts the following values:
channel
each channel is displayed in a separate color
intensity
each channel is displayed using the same color scheme
rainbow
each channel is displayed using the rainbow color scheme
moreland
each channel is displayed using the moreland color scheme
nebulae
each channel is displayed using the nebulae color scheme
fire
each channel is displayed using the fire color scheme
fiery
each channel is displayed using the fiery color scheme
fruit
each channel is displayed using the fruit color scheme
cool
each channel is displayed using the cool color scheme
Default value is intensity.
scale
Specify scale used for calculating intensity color values.
It accepts the following values:
lin linear
sqrt
square root, default
cbrt
cubic root
log logarithmic
4thrt
4th root
5thrt
5th root
Default value is log.
saturation
Set saturation modifier for displayed colors. Negative values
provide alternative color scheme. 0 is no saturation at all.
Saturation must be in [-10.0, 10.0] range. Default value is 1.
win_func
Set window function.
It accepts the following values:
rect
bartlett
hann
hanning
hamming
blackman
welch
flattop
bharris
bnuttall
bhann
sine
nuttall
lanczos
gauss
tukey
dolph
cauchy
parzen
poisson
Default value is "hann".
orientation
Set orientation of time vs frequency axis. Can be "vertical" or
"horizontal". Default is "vertical".
gain
Set scale gain for calculating intensity color values. Default
value is 1.
legend
Draw time and frequency axes and legends. Default is enabled.
rotation
Set color rotation, must be in [-1.0, 1.0] range. Default value is
0.
Examples
* Extract an audio spectrogram of a whole audio track in a 1024x1024
picture using ffmpeg:
ffmpeg -i audio.flac -lavfi showspectrumpic=s=1024x1024 spectrogram.png
showvolume
Convert input audio volume to a video output.
The filter accepts the following options:
rate, r
Set video rate.
b Set border width, allowed range is [0, 5]. Default is 1.
w Set channel width, allowed range is [80, 8192]. Default is 400.
h Set channel height, allowed range is [1, 900]. Default is 20.
f Set fade, allowed range is [0.001, 1]. Default is 0.95.
c Set volume color expression.
The expression can use the following variables:
VOLUME
Current max volume of channel in dB.
PEAK
Current peak.
CHANNEL
Current channel number, starting from 0.
t If set, displays channel names. Default is enabled.
v If set, displays volume values. Default is enabled.
o Set orientation, can be "horizontal" or "vertical", default is
"horizontal".
s Set step size, allowed range s [0, 5]. Default is 0, which means
step is disabled.
showwaves
Convert input audio to a video output, representing the samples waves.
The filter accepts the following options:
size, s
Specify the video size for the output. For the syntax of this
option, check the "Video size" section in the ffmpeg-utils manual.
Default value is "600x240".
mode
Set display mode.
Available values are:
point
Draw a point for each sample.
line
Draw a vertical line for each sample.
p2p Draw a point for each sample and a line between them.
cline
Draw a centered vertical line for each sample.
Default value is "point".
n Set the number of samples which are printed on the same column. A
larger value will decrease the frame rate. Must be a positive
integer. This option can be set only if the value for rate is not
explicitly specified.
rate, r
Set the (approximate) output frame rate. This is done by setting
the option n. Default value is "25".
split_channels
Set if channels should be drawn separately or overlap. Default
value is 0.
colors
Set colors separated by '|' which are going to be used for drawing
of each channel.
scale
Set amplitude scale.
Available values are:
lin Linear.
log Logarithmic.
sqrt
Square root.
cbrt
Cubic root.
Default is linear.
Examples
* Output the input file audio and the corresponding video
representation at the same time:
amovie=a.mp3,asplit[out0],showwaves[out1]
* Create a synthetic signal and show it with showwaves, forcing a
frame rate of 30 frames per second:
aevalsrc=sin(1*2*PI*t)*sin(880*2*PI*t):cos(2*PI*200*t),asplit[out0],showwaves=r=30[out1]
showwavespic
Convert input audio to a single video frame, representing the samples
waves.
The filter accepts the following options:
size, s
Specify the video size for the output. For the syntax of this
option, check the "Video size" section in the ffmpeg-utils manual.
Default value is "600x240".
split_channels
Set if channels should be drawn separately or overlap. Default
value is 0.
colors
Set colors separated by '|' which are going to be used for drawing
of each channel.
scale
Set amplitude scale. Can be linear "lin" or logarithmic "log".
Default is linear.
Examples
* Extract a channel split representation of the wave form of a whole
audio track in a 1024x800 picture using ffmpeg:
ffmpeg -i audio.flac -lavfi showwavespic=split_channels=1:s=1024x800 waveform.png
sidedata, asidedata
Delete frame side data, or select frames based on it.
This filter accepts the following options:
mode
Set mode of operation of the filter.
Can be one of the following:
select
Select every frame with side data of "type".
delete
Delete side data of "type". If "type" is not set, delete all
side data in the frame.
type
Set side data type used with all modes. Must be set for "select"
mode. For the list of frame side data types, refer to the
"AVFrameSideDataType" enum in libavutil/frame.h. For example, to
choose "AV_FRAME_DATA_PANSCAN" side data, you must specify
"PANSCAN".
spectrumsynth
Sythesize audio from 2 input video spectrums, first input stream
represents magnitude across time and second represents phase across
time. The filter will transform from frequency domain as displayed in
videos back to time domain as presented in audio output.
This filter is primarily created for reversing processed showspectrum
filter outputs, but can synthesize sound from other spectrograms too.
But in such case results are going to be poor if the phase data is not
available, because in such cases phase data need to be recreated,
usually its just recreated from random noise. For best results use
gray only output ("channel" color mode in showspectrum filter) and
"log" scale for magnitude video and "lin" scale for phase video. To
produce phase, for 2nd video, use "data" option. Inputs videos should
generally use "fullframe" slide mode as that saves resources needed for
decoding video.
The filter accepts the following options:
sample_rate
Specify sample rate of output audio, the sample rate of audio from
which spectrum was generated may differ.
channels
Set number of channels represented in input video spectrums.
scale
Set scale which was used when generating magnitude input spectrum.
Can be "lin" or "log". Default is "log".
slide
Set slide which was used when generating inputs spectrums. Can be
"replace", "scroll", "fullframe" or "rscroll". Default is
"fullframe".
win_func
Set window function used for resynthesis.
overlap
Set window overlap. In range "[0, 1]". Default is 1, which means
optimal overlap for selected window function will be picked.
orientation
Set orientation of input videos. Can be "vertical" or "horizontal".
Default is "vertical".
Examples
* First create magnitude and phase videos from audio, assuming audio
is stereo with 44100 sample rate, then resynthesize videos back to
audio with spectrumsynth:
ffmpeg -i input.flac -lavfi showspectrum=mode=separate:scale=log:overlap=0.875:color=channel:slide=fullframe:data=magnitude -an -c:v rawvideo magnitude.nut
ffmpeg -i input.flac -lavfi showspectrum=mode=separate:scale=lin:overlap=0.875:color=channel:slide=fullframe:data=phase -an -c:v rawvideo phase.nut
ffmpeg -i magnitude.nut -i phase.nut -lavfi spectrumsynth=channels=2:sample_rate=44100:win_func=hann:overlap=0.875:slide=fullframe output.flac
split, asplit
Split input into several identical outputs.
"asplit" works with audio input, "split" with video.
The filter accepts a single parameter which specifies the number of
outputs. If unspecified, it defaults to 2.
Examples
* Create two separate outputs from the same input:
[in] split [out0][out1]
* To create 3 or more outputs, you need to specify the number of
outputs, like in:
[in] asplit=3 [out0][out1][out2]
* Create two separate outputs from the same input, one cropped and
one padded:
[in] split [splitout1][splitout2];
[splitout1] crop=100:100:0:0 [cropout];
[splitout2] pad=200:200:100:100 [padout];
* Create 5 copies of the input audio with ffmpeg:
ffmpeg -i INPUT -filter_complex asplit=5 OUTPUT
zmq, azmq
Receive commands sent through a libzmq client, and forward them to
filters in the filtergraph.
"zmq" and "azmq" work as a pass-through filters. "zmq" must be inserted
between two video filters, "azmq" between two audio filters.
To enable these filters you need to install the libzmq library and
headers and configure FFmpeg with "--enable-libzmq".
For more information about libzmq see: <http://www.zeromq.org/>
The "zmq" and "azmq" filters work as a libzmq server, which receives
messages sent through a network interface defined by the bind_address
option.
The received message must be in the form:
<TARGET> <COMMAND> [<ARG>]
TARGET specifies the target of the command, usually the name of the
filter class or a specific filter instance name.
COMMAND specifies the name of the command for the target filter.
ARG is optional and specifies the optional argument list for the given
COMMAND.
Upon reception, the message is processed and the corresponding command
is injected into the filtergraph. Depending on the result, the filter
will send a reply to the client, adopting the format:
<ERROR_CODE> <ERROR_REASON>
<MESSAGE>
MESSAGE is optional.
Examples
Look at tools/zmqsend for an example of a zmq client which can be used
to send commands processed by these filters.
Consider the following filtergraph generated by ffplay
ffplay -dumpgraph 1 -f lavfi "
color=s=100x100:c=red [l];
color=s=100x100:c=blue [r];
nullsrc=s=200x100, zmq [bg];
[bg][l] overlay [bg+l];
[bg+l][r] overlay=x=100 "
To change the color of the left side of the video, the following
command can be used:
echo Parsed_color_0 c yellow | tools/zmqsend
To change the right side:
echo Parsed_color_1 c pink | tools/zmqsend
MULTIMEDIA SOURCES
Below is a description of the currently available multimedia sources.
amovie
This is the same as movie source, except it selects an audio stream by
default.
movie
Read audio and/or video stream(s) from a movie container.
It accepts the following parameters:
filename
The name of the resource to read (not necessarily a file; it can
also be a device or a stream accessed through some protocol).
format_name, f
Specifies the format assumed for the movie to read, and can be
either the name of a container or an input device. If not
specified, the format is guessed from movie_name or by probing.
seek_point, sp
Specifies the seek point in seconds. The frames will be output
starting from this seek point. The parameter is evaluated with
"av_strtod", so the numerical value may be suffixed by an IS
postfix. The default value is "0".
streams, s
Specifies the streams to read. Several streams can be specified,
separated by "+". The source will then have as many outputs, in the
same order. The syntax is explained in the ``Stream specifiers''
section in the ffmpeg manual. Two special names, "dv" and "da"
specify respectively the default (best suited) video and audio
stream. Default is "dv", or "da" if the filter is called as
"amovie".
stream_index, si
Specifies the index of the video stream to read. If the value is
-1, the most suitable video stream will be automatically selected.
The default value is "-1". Deprecated. If the filter is called
"amovie", it will select audio instead of video.
loop
Specifies how many times to read the stream in sequence. If the
value is less than 1, the stream will be read again and again.
Default value is "1".
Note that when the movie is looped the source timestamps are not
changed, so it will generate non monotonically increasing
timestamps.
discontinuity
Specifies the time difference between frames above which the point
is considered a timestamp discontinuity which is removed by
adjusting the later timestamps.
It allows overlaying a second video on top of the main input of a
filtergraph, as shown in this graph:
input -----------> deltapts0 --> overlay --> output
^
|
movie --> scale--> deltapts1 -------+
Examples
* Skip 3.2 seconds from the start of the AVI file in.avi, and overlay
it on top of the input labelled "in":
movie=in.avi:seek_point=3.2, scale=180:-1, setpts=PTS-STARTPTS [over];
[in] setpts=PTS-STARTPTS [main];
[main][over] overlay=16:16 [out]
* Read from a video4linux2 device, and overlay it on top of the input
labelled "in":
movie=/dev/video0:f=video4linux2, scale=180:-1, setpts=PTS-STARTPTS [over];
[in] setpts=PTS-STARTPTS [main];
[main][over] overlay=16:16 [out]
* Read the first video stream and the audio stream with id 0x81 from
dvd.vob; the video is connected to the pad named "video" and the
audio is connected to the pad named "audio":
movie=dvd.vob:s=v:0+#0x81 [video] [audio]
Commands
Both movie and amovie support the following commands:
seek
Perform seek using "av_seek_frame". The syntax is: seek
stream_index|timestamp|flags
* stream_index: If stream_index is -1, a default stream is
selected, and timestamp is automatically converted from
AV_TIME_BASE units to the stream specific time_base.
* timestamp: Timestamp in AVStream.time_base units or, if no
stream is specified, in AV_TIME_BASE units.
* flags: Flags which select direction and seeking mode.
get_duration
Get movie duration in AV_TIME_BASE units.
SEE ALSO
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavfilter(3)
AUTHORS
The FFmpeg developers.
For details about the authorship, see the Git history of the project
(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command git log in
the FFmpeg source directory, or browsing the online repository at
<http://source.ffmpeg.org>.
Maintainers for the specific components are listed in the file
MAINTAINERS in the source code tree.
FFMPEG-FILTERS(1)
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