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@id=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 optionally followed by "@id". 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.
A special syntax implemented in the ffmpeg CLI tool allows loading option values from files. This is done be prepending a slash ’/’ to the option name, then the supplied value is interpreted as a path from which the actual value is loaded. E.g.
ffmpeg -i <INPUT> -vf drawtext=/text=/tmp/some_text <OUTPUT>
will load the text to be drawn from /tmp/some_text. API users wishing to implement a similar feature should use the "avfilter_graph_segment_*()" functions together with custom IO code.
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.
Leading and trailing whitespaces (space, tabs, or line feeds) separating tokens in the filtergraph specification are ignored. This means that the filtergraph can be expressed using empty lines and spaces to improve redability.
For example, the filtergraph:
testsrc,split[L1],hflip[L2];[L1][L2] hstack
can be represented as:
testsrc,
split [L1], hflip [L2];
[L1][L2] hstack
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 '_'
<FILTER_NAME> ::=
<NAME>["@"<NAME>]
<LINKLABEL> ::= "[" <NAME>
"]"
<LINKLABELS> ::= <LINKLABEL>
[<LINKLABELS>]
<FILTER_ARGUMENTS> ::= sequence of chars (possibly
quoted)
<FILTER> ::= [<LINKLABELS>] <FILTER_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"
In order to avoid cumbersome escaping when using a commandline tool accepting a filter specification as input, it is advisable to avoid direct inclusion of the filter or options specification in the shell.
For example, in case of the drawtext filter, you might prefer to use the textfile option in place of text to specify the text to render.
When using the ffmpeg tool, you might consider to use the -filter_script option or -filter_complex_script option.
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; deprecated, do not use | ||
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
See "ffmpeg -filters" to view which filters have timeline support.
CHANGING OPTIONS AT RUNTIME WITH A COMMAND
Some options can be changed during the operation of the filter using a command. These options are marked ’T’ on the output of ffmpeg -h filter=<name of filter>. The name of the command is the name of the option and the argument is the new value.
OPTIONS FOR FILTERS WITH SEVERAL INPUTS
Some filters
with several inputs support a common set of options. These
options can only be set by name, not with the short
notation.
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.
shortest
If set to 1, force the output to terminate when the shortest input terminates. Default value is 0.
repeatlast
If set to 1, force the filter to extend the last frame of secondary streams until the end of the primary stream. A value of 0 disables this behavior. Default value is 1.
ts_sync_mode
How strictly to sync streams
based on secondary input timestamps; it accepts one of the
following values:
default
Frame from secondary input with the nearest lower or equal timestamp to the primary input frame.
nearest
Frame from secondary input with the absolute nearest timestamp to the primary input frame.
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.
mode
Set mode of compressor operation. Can be "upward" or "downward". Default is "downward".
threshold
If a signal of stream rises above this level it will affect the gain reduction. 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 1. Range is from 1 to 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. |
Commands
This filter supports the all above options as commands.
acontrast
Simple audio dynamic range compression/expansion filter.
The filter
accepts the following options:
contrast
Set contrast. Default is 33. Allowed range is between 0 and 100.
acopy
Copy the input audio source unchanged to the output. This is
mainly useful for testing purposes.
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
acrossover
Split audio stream into several bands.
This filter splits audio stream into two or more frequency ranges. Summing all streams back will give flat output.
The filter
accepts the following options:
split
Set split frequencies. Those must be positive and increasing.
order
Set filter order for each band split. This controls filter roll-off or steepness of filter transfer function. Available values are:
2nd |
12 dB per octave. |
|||
4th |
24 dB per octave. |
|||
6th |
36 dB per octave. |
|||
8th |
48 dB per octave. |
10th
60 dB per octave.
12th
72 dB per octave.
14th
84 dB per octave.
16th
96 dB per octave.
18th
108 dB per octave.
20th
120 dB per octave.
Default is 4th.
level
Set input gain level. Allowed range is from 0 to 1. Default value is 1.
gains
Set output gain for each band. Default value is 1 for all bands.
precision
Set which precision to use when
processing samples.
auto
Auto pick internal sample format depending on other filters.
float
Always use single-floating point precision sample format.
double
Always use double-floating point precision sample format.
Default value is "auto".
Examples
• |
Split input audio stream into two bands (low and high) with split frequency of 1500 Hz, each band will be in separate stream: |
ffmpeg -i in.flac -filter_complex 'acrossover=split=1500[LOW][HIGH]' -map '[LOW]' low.wav -map '[HIGH]' high.wav
• |
Same as above, but with higher filter order: |
ffmpeg -i in.flac -filter_complex 'acrossover=split=1500:order=8th[LOW][HIGH]' -map '[LOW]' low.wav -map '[HIGH]' high.wav
• |
Same as above, but also with additional middle band (frequencies between 1500 and 8000): |
ffmpeg -i in.flac -filter_complex 'acrossover=split=1500 8000:order=8th[LOW][MID][HIGH]' -map '[LOW]' low.wav -map '[MID]' mid.wav -map '[HIGH]' high.wav
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, 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.
Commands
This filter supports the all above options as commands.
acue
Delay audio filtering until a given wallclock timestamp. See
the cue filter.
adeclick
Remove impulsive noise from input audio.
Samples
detected as impulsive noise are replaced by interpolated
samples using autoregressive modelling.
window, w
Set window size, in milliseconds. Allowed range is from 10 to 100. Default value is 55 milliseconds. This sets size of window which will be processed at once.
overlap, o
Set window overlap, in percentage of window size. Allowed range is from 50 to 95. Default value is 75 percent. Setting this to a very high value increases impulsive noise removal but makes whole process much slower.
arorder, a
Set autoregression order, in percentage of window size. Allowed range is from 0 to 25. Default value is 2 percent. This option also controls quality of interpolated samples using neighbour good samples.
threshold, t
Set threshold value. Allowed range is from 1 to 100. Default value is 2. This controls the strength of impulsive noise which is going to be removed. The lower value, the more samples will be detected as impulsive noise.
burst, b
Set burst fusion, in percentage of window size. Allowed range is 0 to 10. Default value is 2. If any two samples detected as noise are spaced less than this value then any sample between those two samples will be also detected as noise.
method, m
Set overlap method.
It accepts the
following values:
add, a
Select overlap-add method. Even not interpolated samples are slightly changed with this method.
save, s
Select overlap-save method. Not interpolated samples remain unchanged.
Default value is "a".
adeclip
Remove clipped samples from input audio.
Samples
detected as clipped are replaced by interpolated samples
using autoregressive modelling.
window, w
Set window size, in milliseconds. Allowed range is from 10 to 100. Default value is 55 milliseconds. This sets size of window which will be processed at once.
overlap, o
Set window overlap, in percentage of window size. Allowed range is from 50 to 95. Default value is 75 percent.
arorder, a
Set autoregression order, in percentage of window size. Allowed range is from 0 to 25. Default value is 8 percent. This option also controls quality of interpolated samples using neighbour good samples.
threshold, t
Set threshold value. Allowed range is from 1 to 100. Default value is 10. Higher values make clip detection less aggressive.
hsize, n
Set size of histogram used to detect clips. Allowed range is from 100 to 9999. Default value is 1000. Higher values make clip detection less aggressive.
method, m
Set overlap method.
It accepts the
following values:
add, a
Select overlap-add method. Even not interpolated samples are slightly changed with this method.
save, s
Select overlap-save method. Not interpolated samples remain unchanged.
Default value is "a".
adecorrelate
Apply decorrelation to input audio stream.
The filter
accepts the following options:
stages
Set decorrelation stages of filtering. Allowed range is from 1 to 16. Default value is 6.
seed
Set random seed used for setting delay in samples across channels.
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 ’|’. 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. If you want instead to delay in seconds, append ’s’ to number.
all |
Use last set delay for all remaining channels. By default is disabled. This option if enabled changes how option "delays" is interpreted. |
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
• |
Delay all channels by same number of samples: |
adelay=delays=64S:all=1
adenorm
Remedy denormals in audio by adding extremely low-level
noise.
This filter shall be placed before any filter that can produce denormals.
A description
of the accepted parameters follows.
level
Set level of added noise in dB. Default is -351. Allowed range is from -451 to -90.
type
Set type of added noise.
dc |
Add DC signal. |
|||
ac |
Add AC signal. |
square
Add square signal.
pulse
Add pulse signal.
Default is "dc".
Commands
This filter supports the all above options as commands.
aderivative,
aintegral
Compute derivative/integral of audio stream.
Applying both filters one after another produces original audio.
adrc
Apply spectral dynamic range controller filter to input
audio stream.
A description
of the accepted options follows.
transfer
Set the transfer expression.
The expression can contain the following constants:
ch |
current channel number |
|||
sn |
current sample number |
nb_channels
number of channels
t |
timestamp expressed in seconds |
|||
sr |
sample rate |
|||
p |
current frequency power value, in dB |
|||
f |
current frequency in Hz |
Default value is "p".
attack
Set the attack in milliseconds. Default is 50 milliseconds. Allowed range is from 1 to 1000 milliseconds.
release
Set the release in milliseconds. Default is 100 milliseconds. Allowed range is from 5 to 2000 milliseconds.
channels
Set which channels to filter, by default "all" channels in audio stream are filtered.
Commands
This filter supports the all above options as commands.
Examples
• |
Apply spectral compression to all frequencies with threshold of -50 dB and 1:6 ratio: |
adrc=transfer='if(gt(p,-50),-50+(p-(-50))/6,p)':attack=50:release=100
• |
Similar to above but with 1:2 ratio and filtering only front center channel: |
adrc=transfer='if(gt(p,-50),-50+(p-(-50))/2,p)':attack=50:release=100:channels=FC
• |
Apply spectral noise gate to all frequencies with threshold of -85 dB and with short attack time and short release time: |
adrc=transfer='if(lte(p,-85),p-800,p)':attack=1:release=5
• |
Apply spectral expansion to all frequencies with threshold of -10 dB and 1:2 ratio: |
adrc=transfer='if(lt(p,-10),-10+(p-(-10))*2,p)':attack=50:release=100
• |
Apply limiter to max -60 dB to all frequencies, with attack of 2 ms and release of 10 ms: |
adrc=transfer='min(p,-60)':attack=2:release=10
adynamicequalizer
Apply dynamic equalization to input audio stream.
A description
of the accepted options follows.
threshold
Set the detection threshold used to trigger equalization. Threshold detection is using detection filter. Default value is 0. Allowed range is from 0 to 100.
dfrequency
Set the detection frequency in Hz used for detection filter used to trigger equalization. Default value is 1000 Hz. Allowed range is between 2 and 1000000 Hz.
dqfactor
Set the detection resonance factor for detection filter used to trigger equalization. Default value is 1. Allowed range is from 0.001 to 1000.
tfrequency
Set the target frequency of equalization filter. Default value is 1000 Hz. Allowed range is between 2 and 1000000 Hz.
tqfactor
Set the target resonance factor for target equalization filter. Default value is 1. Allowed range is from 0.001 to 1000.
attack
Set the amount of milliseconds the signal from detection has to rise above the detection threshold before equalization starts. Default is 20. Allowed range is between 1 and 2000.
release
Set the amount of milliseconds the signal from detection has to fall below the detection threshold before equalization ends. Default is 200. Allowed range is between 1 and 2000.
ratio
Set the ratio by which the equalization gain is raised. Default is 1. Allowed range is between 0 and 30.
makeup
Set the makeup offset by which the equalization gain is raised. Default is 0. Allowed range is between 0 and 100.
range
Set the max allowed cut/boost amount. Default is 50. Allowed range is from 1 to 200.
mode
Set the mode of filter
operation, can be one of the following:
listen
Output only isolated detection signal.
cut |
Cut frequencies above detection threshold. |
boost
Boost frequencies bellow detection threshold.
Default mode is cut.
dftype
Set the type of detection
filter, can be one of the following:
bandpass
lowpass
highpass
peak
Default type is bandpass.
tftype
Set the type of target filter,
can be one of the following:
bell
lowshelf
highshelf
Default type is bell.
direction
Set processing direction
relative to threshold.
downward
Boost/Cut if threshold is higher/lower than detected volume.
upward
Boost/Cut if threshold is lower/higher than detected volume.
Default direction is downward.
auto
Automatically gather threshold from detection filter. By default is disabled. This option is useful to detect threshold in certain time frame of input audio stream, in such case option value is changed at runtime.
Available
values are:
disabled
Disable using automatically gathered threshold value.
off |
Stop picking threshold value. |
|||
on |
Start picking threshold value. |
precision
Set which precision to use when
processing samples.
auto
Auto pick internal sample format depending on other filters.
float
Always use single-floating point precision sample format.
double
Always use double-floating point precision sample format.
Commands
This filter supports the all above options as commands.
adynamicsmooth
Apply dynamic smoothing to input audio stream.
A description
of the accepted options follows.
sensitivity
Set an amount of sensitivity to frequency fluctations. Default is 2. Allowed range is from 0 to 1e+06.
basefreq
Set a base frequency for smoothing. Default value is 22050. Allowed range is from 2 to 1e+06.
Commands
This filter supports the all above options as commands.
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 loudness 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 sounds 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 50µs (FM).
75fm
select 75µs (FM).
50kf
select 50µs (FM-KF).
75kf
select 75µs (FM-KF).
Commands
This filter supports the all above options as commands.
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)
aexciter
An exciter is used to produce high sound that is not present
in the original signal. This is done by creating harmonic
distortions of the signal which are restricted in range and
added to the original signal. An Exciter raises the upper
end of an audio signal without simply raising the higher
frequencies like an equalizer would do to create a more
"crisp" or "brilliant" sound.
The filter
accepts the following options:
level_in
Set input level prior processing of signal. Allowed range is from 0 to 64. Default value is 1.
level_out
Set output level after processing of signal. Allowed range is from 0 to 64. Default value is 1.
amount
Set the amount of harmonics added to original signal. Allowed range is from 0 to 64. Default value is 1.
drive
Set the amount of newly created harmonics. Allowed range is from 0.1 to 10. Default value is 8.5.
blend
Set the octave of newly created harmonics. Allowed range is from -10 to 10. Default value is 0.
freq
Set the lower frequency limit of producing harmonics in Hz. Allowed range is from 2000 to 12000 Hz. Default is 7500 Hz.
ceil
Set the upper frequency limit of producing harmonics. Allowed range is from 9999 to 20000 Hz. If value is lower than 10000 Hz no limit is applied.
listen
Mute the original signal and output only added harmonics. By default is disabled.
Commands
This filter supports the all above options as commands.
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
losi
select logistic sigmoid
sinc
select sine cardinal function
isinc
select inverted sine cardinal function
quat
select quartic
quatr
select quartic root
qsin2
select squared quarter of sine wave
hsin2
select squared half of sine wave
nofade
no fade applied
silence
Set the initial gain for fade-in or final gain for fade-out. Default value is 0.0.
unity
Set the initial gain for fade-out or final gain for fade-in. Default value is 1.0.
Commands
This filter supports the all above options as commands.
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
afftdn
Denoise audio samples with FFT.
A description
of the accepted parameters follows.
noise_reduction, nr
Set the noise reduction in dB, allowed range is 0.01 to 97. Default value is 12 dB.
noise_floor, nf
Set the noise floor in dB, allowed range is -80 to -20. Default value is -50 dB.
noise_type, nt
Set the noise type.
It accepts the
following values:
white, w
Select white noise.
vinyl, v
Select vinyl noise.
shellac, s
Select shellac noise.
custom, c
Select custom noise, defined in "bn" option.
Default value is white noise.
band_noise, bn
Set custom band noise profile for every one of 15 bands. Bands are separated by ’ ’ or ’|’.
residual_floor, rf
Set the residual floor in dB, allowed range is -80 to -20. Default value is -38 dB.
track_noise, tn
Enable noise floor tracking. By default is disabled. With this enabled, noise floor is automatically adjusted.
track_residual, tr
Enable residual tracking. By default is disabled.
output_mode, om
Set the output mode.
It accepts the
following values:
input, i
Pass input unchanged.
output, o
Pass noise filtered out.
noise, n
Pass only noise.
Default value is output.
adaptivity, ad
Set the adaptivity factor, used how fast to adapt gains adjustments per each frequency bin. Value 0 enables instant adaptation, while higher values react much slower. Allowed range is from 0 to 1. Default value is 0.5.
floor_offset, fo
Set the noise floor offset factor. This option is used to adjust offset applied to measured noise floor. It is only effective when noise floor tracking is enabled. Allowed range is from -2.0 to 2.0. Default value is 1.0.
noise_link, nl
Set the noise link used for multichannel audio.
It accepts the
following values:
none
Use unchanged channel’s noise floor.
min |
Use measured min noise floor of all channels. |
|||
max |
Use measured max noise floor of all channels. |
average
Use measured average noise floor of all channels.
Default value is min.
band_multiplier, bm
Set the band multiplier factor, used how much to spread bands across frequency bins. Allowed range is from 0.2 to 5. Default value is 1.25.
sample_noise, sn
Toggle capturing and measurement of noise profile from input audio.
It accepts the
following values:
start, begin
Start sample noise capture.
stop, end
Stop sample noise capture and measure new noise band profile.
Default value is "none".
gain_smooth, gs
Set gain smooth spatial radius, used to smooth gains applied to each frequency bin. Useful to reduce random music noise artefacts. Higher values increases smoothing of gains. Allowed range is from 0 to 50. Default value is 0.
Commands
This filter supports the some above mentioned options as commands.
Examples
• |
Reduce white noise by 10dB, and use previously measured noise floor of -40dB: |
afftdn=nr=10:nf=-40
• |
Reduce white noise by 10dB, also set initial noise floor to -80dB and enable automatic tracking of noise floor so noise floor will gradually change during processing: |
afftdn=nr=10:nf=-80:tn=1
• |
Reduce noise by 20dB, using noise floor of -40dB and using commands to take noise profile of first 0.4 seconds of input audio: |
asendcmd=0.0 afftdn sn start,asendcmd=0.4 afftdn sn stop,afftdn=nr=20:nf=-40
afftfilt
Apply arbitrary expressions to samples in frequency domain.
real
Set frequency domain real expression for each separate channel separated by ’|’. Default is "re". 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 ’|’. Default is "im".
Each expression in real and imag can contain the following constants and functions:
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 | ||
re |
current real part of frequency bin of current channel | ||
im |
current imaginary part of frequency bin of current channel |
real(b, ch)
Return the value of real part of frequency bin at location (bin,channel)
imag(b, ch)
Return the value of imaginary part of frequency bin at location (bin,channel)
win_size
Set window size. Allowed range is from 16 to 131072. Default is 4096
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
bohman
kaiser
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="'real=re * (1-clip((b/nb)*b,0,1))':imag='im * (1-clip((b/nb)*b,0,1))'"
• |
Apply robotize effect: |
afftfilt="real='hypot(re,im)*sin(0)':imag='hypot(re,im)*cos(0)':win_size=512:overlap=0.75"
• |
Apply whisper effect: |
afftfilt="real='hypot(re,im)*cos((random(0)*2-1)*2*3.14)':imag='hypot(re,im)*sin((random(1)*2-1)*2*3.14)':win_size=128:overlap=0.8"
• |
Apply phase shift: |
afftfilt="real=re*cos(1)-im*sin(1):imag=re*sin(1)+im*cos(1)"
afir
Apply an arbitrary Finite Impulse Response filter.
This filter is designed for applying long FIR filters, up to 60 seconds long.
It can be used as component for digital crossover filters, room equalization, cross talk cancellation, wavefield synthesis, auralization, ambiophonics, ambisonics and spatialization.
This filter uses the streams higher than first one as FIR coefficients. If the non-first stream holds a single channel, it will be used for all input channels in the first stream, otherwise the number of channels in the non-first stream must be same as the number of channels in the first stream.
It accepts the following parameters:
dry |
Set dry gain. This sets input gain. |
|||
wet |
Set wet gain. This sets final output gain. |
length
Set Impulse Response filter length. Default is 1, which means whole IR is processed.
gtype
Enable applying gain measured from power of IR.
Set which
approach to use for auto gain measurement.
none
Do not apply any gain.
peak
select peak gain, very conservative approach. This is default value.
dc |
select DC gain, limited application. | ||
gn |
select gain to noise approach, this is most popular one. | ||
ac |
select AC gain. | ||
rms |
select RMS gain. |
irgain
Set gain to be applied to IR coefficients before filtering. Allowed range is 0 to 1. This gain is applied after any gain applied with gtype option.
irfmt
Set format of IR stream. Can be "mono" or "input". Default is "input".
maxir
Set max allowed Impulse Response filter duration in seconds. Default is 30 seconds. Allowed range is 0.1 to 60 seconds.
response
Show IR frequency response, magnitude(magenta), phase(green) and group delay(yellow) in additional video stream. By default it is disabled.
channel
Set for which IR channel to display frequency response. By default is first channel displayed. This option is used only when response is enabled.
size
Set video stream size. This option is used only when response is enabled.
rate
Set video stream frame rate. This option is used only when response is enabled.
minp
Set minimal partition size used for convolution. Default is 8192. Allowed range is from 1 to 65536. Lower values decreases latency at cost of higher CPU usage.
maxp
Set maximal partition size used for convolution. Default is 8192. Allowed range is from 8 to 65536. Lower values may increase CPU usage.
nbirs
Set number of input impulse responses streams which will be switchable at runtime. Allowed range is from 1 to 32. Default is 1.
ir |
Set IR stream which will be used for convolution, starting from 0, should always be lower than supplied value by "nbirs" option. Default is 0. This option can be changed at runtime via commands. |
precision
Set which precision to use when
processing samples.
auto
Auto pick internal sample format depending on other filters.
float
Always use single-floating point precision sample format.
double
Always use double-floating point precision sample format.
Default value is auto.
irload
Set when to load IR stream. Can be "init" or "access". First one load and prepares all IRs on initialization, second one once on first access of specific IR. Default is "init".
Examples
• |
Apply reverb to stream using mono IR file as second input, complete command using ffmpeg: |
ffmpeg -i input.wav -i middle_tunnel_1way_mono.wav -lavfi afir output.wav
• |
Apply true stereo processing given input stereo stream, and two stereo impulse responses for left and right channel, the impulse response files are files with names l_ir.wav and r_ir.wav: |
"pan=4C|c0=FL|c1=FL|c2=FR|c3=FR[a];amovie=l_ir.wav[LIR];amovie=r_ir.wav[RIR];[LIR][RIR]amerge[ir];[a][ir]afir=irfmt=input:gtype=gn:irgain=-5dB,pan=stereo|FL<c0+c2|FR<c1+c3"
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, f
A ’|’-separated list of requested sample formats.
sample_rates, r
A ’|’-separated list of requested sample rates.
channel_layouts, cl
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
afreqshift
Apply frequency shift to input audio samples.
The filter
accepts the following options:
shift
Specify frequency shift. Allowed range is -INT_MAX to INT_MAX. Default value is 0.0.
level
Set output gain applied to final output. Allowed range is from 0.0 to 1.0. Default value is 1.0.
order
Set filter order used for filtering. Allowed range is from 1 to 16. Default value is 8.
Commands
This filter supports the all above options as commands.
afwtdn
Reduce broadband noise from input samples using
Wavelets.
A description
of the accepted options follows.
sigma
Set the noise sigma, allowed range is from 0 to 1. Default value is 0. This option controls strength of denoising applied to input samples. Most useful way to set this option is via decibels, eg. -45dB.
levels
Set the number of wavelet levels of decomposition. Allowed range is from 1 to 12. Default value is 10. Setting this too low make denoising performance very poor.
wavet
Set wavelet type for
decomposition of input frame. They are sorted by number of
coefficients, from lowest to highest. More coefficients
means worse filtering speed, but overall better quality.
Available wavelets are:
sym2
sym4
rbior68
deb10
sym10
coif5
bl3 |
percent
Set percent of full denoising. Allowed range is from 0 to 100 percent. Default value is 85 percent or partial denoising.
profile
If enabled, first input frame will be used as noise profile. If first frame samples contain non-noise performance will be very poor.
adaptive
If enabled, input frames are analyzed for presence of noise. If noise is detected with high possibility then input frame profile will be used for processing following frames, until new noise frame is detected.
samples
Set size of single frame in number of samples. Allowed range is from 512 to 65536. Default frame size is 8192 samples.
softness
Set softness applied inside thresholding function. Allowed range is from 0 to 10. Default softness is 1.
Commands
This filter supports the all above options as commands.
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.
mode
Set the mode of operation. Can be "upward" or "downward". Default is "downward". If set to "upward" mode, higher parts of signal will be amplified, expanding dynamic range in upward direction. Otherwise, in case of "downward" lower parts of signal will be reduced.
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. Setting this to 0 disables reduction and then filter behaves like expander.
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".
Commands
This filter supports the all above options as commands.
aiir
Apply an arbitrary Infinite Impulse Response filter.
It accepts the
following parameters:
zeros, z
Set B/numerator/zeros/reflection coefficients.
poles, p
Set A/denominator/poles/ladder coefficients.
gains, k
Set channels gains.
dry_gain
Set input gain.
wet_gain
Set output gain.
format, f
Set coefficients format.
ll |
lattice-ladder function |
|||
sf |
analog transfer function |
|||
tf |
digital transfer function |
|||
zp |
Z-plane zeros/poles, cartesian (default) |
|||
pr |
Z-plane zeros/poles, polar radians |
|||
pd |
Z-plane zeros/poles, polar degrees |
|||
sp |
S-plane zeros/poles |
process, r
Set type of processing.
d |
direct processing |
|||
s |
serial processing |
|||
p |
parallel processing |
precision, e
Set filtering precision.
dbl |
double-precision floating-point (default) |
|||
flt |
single-precision floating-point |
|||
i32 |
32-bit integers |
|||
i16 |
16-bit integers |
normalize, n
Normalize filter coefficients, by default is enabled. Enabling it will normalize magnitude response at DC to 0dB.
mix |
How much to use filtered signal in output. Default is 1. Range is between 0 and 1. |
response
Show IR frequency response, magnitude(magenta), phase(green) and group delay(yellow) in additional video stream. By default it is disabled.
channel
Set for which IR channel to display frequency response. By default is first channel displayed. This option is used only when response is enabled.
size
Set video stream size. This option is used only when response is enabled.
Coefficients in "tf" and "sf" format are separated by spaces and are in ascending order.
Coefficients in "zp" format are separated by spaces and order of coefficients doesn’t matter. Coefficients in "zp" format are complex numbers with i imaginary unit.
Different coefficients and gains can be provided for every channel, in such case use ’|’ to separate coefficients or gains. Last provided coefficients will be used for all remaining channels.
Examples
• |
Apply 2 pole elliptic notch at around 5000Hz for 48000 Hz sample rate: |
aiir=k=1:z=7.957584807809675810E-1 -2.575128568908332300 3.674839853930788710 -2.57512875289799137 7.957586296317130880E-1:p=1 -2.86950072432325953 3.63022088054647218 -2.28075678147272232 6.361362326477423500E-1:f=tf:r=d
• |
Same as above but in "zp" format: |
aiir=k=0.79575848078096756:z=0.80918701+0.58773007i 0.80918701-0.58773007i 0.80884700+0.58784055i 0.80884700-0.58784055i:p=0.63892345+0.59951235i 0.63892345-0.59951235i 0.79582691+0.44198673i 0.79582691-0.44198673i:f=zp:r=s
• |
Apply 3-rd order analog normalized Butterworth low-pass filter, using analog transfer function format: |
aiir=z=1.3057 0 0 0:p=1.3057 2.3892 2.1860 1:f=sf:r=d
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.
latency
Compensate the delay introduced by using the lookahead buffer set with attack parameter. Also flush the valid audio data in the lookahead buffer when the stream hits EOF.
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, t
Set method to specify band-width of filter.
h |
Hz |
|||
q |
Q-Factor |
|||
o |
octave |
|||
s |
slope |
|||
k |
kHz |
width, w
Specify the band-width of a filter in width_type units.
mix, m
How much to use filtered signal in output. Default is 1. Range is between 0 and 1.
channels, c
Specify which channels to filter, by default all available are filtered.
normalize, n
Normalize biquad coefficients, by default is disabled. Enabling it will normalize magnitude response at DC to 0dB.
order, o
Set the filter order, can be 1 or 2. Default is 2.
transform, a
Set transform type of IIR filter.
di |
||
dii |
||
tdi |
tdii
latt
svf |
||
zdf |
precision, r
Set precison of filtering.
auto
Pick automatic sample format depending on surround filters.
s16 |
Always use signed 16-bit. |
|||
s32 |
Always use signed 32-bit. |
|||
f32 |
Always use float 32-bit. |
|||
f64 |
Always use float 64-bit. |
Commands
This filter
supports the following commands:
frequency, f
Change allpass frequency. Syntax for the command is : "frequency"
width_type, t
Change allpass width_type. Syntax for the command is : "width_type"
width, w
Change allpass width. Syntax for the command is : "width"
mix, m
Change allpass mix. Syntax for the command is : "mix"
aloop
Loop audio samples.
The filter
accepts the following options:
loop
Set the number of loops. Setting this value to -1 will result in infinite loops. Default is 0.
size
Set maximal number of samples. Default is 0.
start
Set first sample of loop. Default is 0.
time
Set the time of loop start in seconds. Only used if option named start is set to -1.
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.
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.
weights
Specify weight of each input audio stream as a sequence of numbers separated by a space. If fewer weights are specified compared to number of inputs, the last weight is assigned to the remaining inputs. Default weight for each input is 1.
normalize
Always scale inputs instead of only doing summation of samples. Beware of heavy clipping if inputs are not normalized prior or after filtering by this filter if this option is disabled. By default is enabled.
Examples
• |
This 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: |
ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex amix=inputs=3:duration=first:dropout_transition=3 OUTPUT
• |
This will mix one vocal and one music input audio stream to a single output with the same duration as the longest input. The music will have quarter the weight as the vocals, and the inputs are not normalized: |
ffmpeg -i VOCALS -i MUSIC -filter_complex amix=inputs=2:duration=longest:dropout_transition=0:weights="1 0.25":normalize=0 OUTPUT
Commands
This filter
supports the following commands:
weights
normalize
Syntax is same as option with same name.
amultiply
Multiply first audio stream with second audio stream and
store result in output audio stream. Multiplication is done
by multiplying each sample from first stream with sample at
same position from second stream.
With this element-wise multiplication one can create amplitude fades and amplitude modulations.
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 Hertz. 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=...
anlmdn
Reduce broadband noise in audio samples using Non-Local
Means algorithm.
Each sample is adjusted by looking for other samples with similar contexts. This context similarity is defined by comparing their surrounding patches of size p. Patches are searched in an area of r around the sample.
The filter
accepts the following options:
strength, s
Set denoising strength. Allowed range is from 0.00001 to 10000. Default value is 0.00001.
patch, p
Set patch radius duration. Allowed range is from 1 to 100 milliseconds. Default value is 2 milliseconds.
research, r
Set research radius duration. Allowed range is from 2 to 300 milliseconds. Default value is 6 milliseconds.
output, o
Set the output mode.
It accepts the following values:
i |
Pass input unchanged. |
|||
o |
Pass noise filtered out. |
|||
n |
Pass only noise. |
Default value is o.
smooth, m
Set smooth factor. Default value is 11. Allowed range is from 1 to 1000.
Commands
This filter supports the all above options as commands.
anlmf,
anlms
Apply Normalized Least-Mean-(Squares|Fourth) algorithm to
the first audio stream using the second audio stream.
This adaptive filter is used to mimic a desired filter by finding the filter coefficients that relate to producing the least mean square of the error signal (difference between the desired, 2nd input audio stream and the actual signal, the 1st input audio stream).
A description
of the accepted options follows.
order
Set filter order.
mu |
Set filter mu. |
|||
eps |
Set the filter eps. |
leakage
Set the filter leakage.
out_mode
It accepts the following values:
i |
Pass the 1st input. | ||
d |
Pass the 2nd input. | ||
o |
Pass difference between desired, 2nd input and error signal estimate. | ||
n |
Pass difference between input, 1st input and error signal estimate. | ||
e |
Pass error signal estimated samples. |
Default value is o.
Examples
• |
One of many usages of this filter is noise reduction, input audio is filtered with same samples that are delayed by fixed amount, one such example for stereo audio is: |
asplit[a][b],[a]adelay=32S|32S[a],[b][a]anlms=order=128:leakage=0.0005:mu=.5:out_mode=o
Commands
This filter supports the same commands as options, excluding option "order".
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.
pad_dur
Specify the duration of samples of silence to add. See the Time duration section in the ffmpeg-utils(1) manual for the accepted syntax. Used only if set to non-negative value.
whole_dur
Specify the minimum total duration in the output audio stream. See the Time duration section in the ffmpeg-utils(1) manual for the accepted syntax. Used only if set to non-negative value. 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_dur
If neither the pad_len nor the whole_len nor pad_dur nor whole_dur option is set, the filter will add silence to the end of the input stream indefinitely.
Note that for ffmpeg 4.4 and earlier a zero pad_dur or whole_dur also caused the filter to add silence 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
aphaseshift
Apply phase shift to input audio samples.
The filter
accepts the following options:
shift
Specify phase shift. Allowed range is from -1.0 to 1.0. Default value is 0.0.
level
Set output gain applied to final output. Allowed range is from 0.0 to 1.0. Default value is 1.0.
order
Set filter order used for filtering. Allowed range is from 1 to 16. Default value is 8.
Commands
This filter supports the all above options as commands.
apsnr
Measure Audio Peak Signal-to-Noise Ratio.
This filter takes two audio streams for input, and outputs first audio stream. Results are in dB per channel at end of either input.
apsyclip
Apply Psychoacoustic clipper to input audio stream.
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].
clip
Set the clipping start value. Default value is 0dBFS or 1.
diff
Output only difference samples, useful to hear introduced distortions. By default is disabled.
adaptive
Set strength of adaptive distortion applied. Default value is 0.5. Allowed range is from 0 to 1.
iterations
Set number of iterations of psychoacoustic clipper. Allowed range is from 1 to 20. Default value is 10.
level
Auto level output signal. Default is disabled. This normalizes audio back to 0dBFS if enabled.
Commands
This filter supports the all above options as commands.
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 "Resampler Options" section in the ffmpeg-resampler(1) 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
arls
Apply Recursive Least Squares algorithm to the first audio
stream using the second audio stream.
This adaptive filter is used to mimic a desired filter by recursively finding the filter coefficients that relate to producing the minimal weighted linear least squares cost function of the error signal (difference between the desired, 2nd input audio stream and the actual signal, the 1st input audio stream).
A description
of the accepted options follows.
order
Set the filter order.
lambda
Set the forgetting factor.
delta
Set the coefficient to initialize internal covariance matrix.
out_mode
Set the filter output samples. It accepts the following values:
i |
Pass the 1st input. | ||
d |
Pass the 2nd input. | ||
o |
Pass difference between desired, 2nd input and error signal estimate. | ||
n |
Pass difference between input, 1st input and error signal estimate. | ||
e |
Pass error signal estimated samples. |
Default value is o.
arnndn
Reduce noise from speech using Recurrent Neural
Networks.
This filter
accepts the following options:
model, m
Set train model file to load. This option is always required.
mix |
Set how much to mix filtered samples into final output. Allowed range is from -1 to 1. Default value is 1. Negative values are special, they set how much to keep filtered noise in the final filter output. Set this option to -1 to hear actual noise removed from input signal. |
Commands
This filter supports the all above options as commands.
asdr
Measure Audio Signal-to-Distortion Ratio.
This filter takes two audio streams for input, and outputs first audio stream. Results are in dB per channel at end of either input.
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.
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.
asisdr
Measure Audio Scaled-Invariant Signal-to-Distortion
Ratio.
This filter takes two audio streams for input, and outputs first audio stream. Results are in dB per channel at end of either input.
asoftclip
Apply audio soft clipping.
Soft clipping is a type of distortion effect where the amplitude of a signal is saturated along a smooth curve, rather than the abrupt shape of hard-clipping.
This filter
accepts the following options:
type
Set type of soft-clipping.
It accepts the
following values:
hard
tanh
atan
cubic
exp |
||
alg |
quintic
sin |
||
erf |
threshold
Set threshold from where to start clipping. Default value is 0dB or 1.
output
Set gain applied to output. Default value is 0dB or 1.
param
Set additional parameter which controls sigmoid function.
oversample
Set oversampling factor.
Commands
This filter supports the all above options as commands.
aspectralstats
Display frequency domain statistical information about the
audio channels. Statistics are calculated and stored as
metadata for each audio channel and for each audio
frame.
It accepts the
following option:
win_size
Set the window length in samples. Default value is 2048. Allowed range is from 32 to 65536.
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
bohman
kaiser
Default is "hann".
overlap
Set window overlap. Allowed range is from 0 to 1. Default value is 0.5.
measure
Select the parameters which are measured. The metadata keys can be used as flags, default is all which measures everything. none disables all measurement.
A list of each
metadata key follows:
mean
variance
centroid
spread
skewness
kurtosis
entropy
flatness
crest
flux
slope
decrease
rolloff
asr
Automatic Speech Recognition
This filter uses PocketSphinx for speech recognition. To enable compilation of this filter, you need to configure FFmpeg with "--enable-pocketsphinx".
It accepts the
following options:
rate
Set sampling rate of input audio. Defaults is 16000. This need to match speech models, otherwise one will get poor results.
hmm |
Set dictionary containing acoustic model files. |
dict
Set pronunciation dictionary.
lm |
Set language model file. |
lmctl
Set language model set.
lmname
Set which language model to use.
logfn
Set output for log messages.
The filter exports recognized speech as the frame metadata "lavfi.asr.text".
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 - 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: Bit_depth Crest_factor DC_offset Dynamic_range Entropy Flat_factor Max_difference Max_level Mean_difference Min_difference Min_level Noise_floor Noise_floor_count Number_of_Infs Number_of_NaNs Number_of_denormals Peak_count Abs_Peak_count Peak_level RMS_difference RMS_peak RMS_trough Zero_crossings Zero_crossings_rate
and for "Overall": Bit_depth DC_offset Entropy Flat_factor Max_difference Max_level Mean_difference Min_difference Min_level Noise_floor Noise_floor_count Number_of_Infs Number_of_NaNs Number_of_denormals Number_of_samples Peak_count Abs_Peak_count Peak_level RMS_difference RMS_level RMS_peak RMS_trough
For example, a full key looks like "lavfi.astats.1.DC_offset" or "lavfi.astats.Overall.Peak_count".
Read below for the description of the keys.
reset
Set the number of frames over which cumulative stats are calculated before being reset. Default is disabled.
measure_perchannel
Select the parameters which are measured per channel. The metadata keys can be used as flags, default is all which measures everything. none disables all per channel measurement.
measure_overall
Select the parameters which are measured overall. The metadata keys can be used as flags, default is all which measures everything. none disables all overall measurement.
A description
of the measure keys follow:
none
no measures
all |
all measures |
Bit_depth
overall bit depth of audio, i.e. number of bits used for each sample
Crest_factor
standard ratio of peak to RMS level (note: not in dB)
DC_offset
mean amplitude displacement from zero
Dynamic_range
measured dynamic range of audio in dB
Entropy
entropy measured across whole audio, entropy of value near 1.0 is typically measured for white noise
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)
Max_difference
maximal difference between two consecutive samples
Max_level
maximal sample level
Mean_difference
mean difference between two consecutive samples, i.e. the average of each difference between two consecutive samples
Min_difference
minimal difference between two consecutive samples
Min_level
minimal sample level
Noise_floor
minimum local peak measured in dBFS over a short window
Noise_floor_count
number of occasions (not the number of samples) that the signal attained Noise floor
Number_of_Infs
number of samples with an infinite value
Number_of_NaNs
number of samples with a NaN (not a number) value
Number_of_denormals
number of samples with a subnormal value
Number_of_samples
number of samples
Peak_count
number of occasions (not the number of samples) that the signal attained either Min_level or Max_level
Abs_Peak_count
number of occasions that the absolute samples taken from the signal attained max absolute value of Min_level and Max_level
Peak_level
standard peak level measured in dBFS
RMS_difference
Root Mean Square difference between two consecutive samples
RMS_level
standard RMS level measured in dBFS
RMS_peak
RMS_trough
peak and trough values for RMS level measured over a short window, measured in dBFS.
Zero crossings
number of points where the waveform crosses the zero level axis
Zero crossings rate
rate of Zero crossings and number of audio samples
asubboost
Boost subwoofer frequencies.
The filter accepts the following options:
dry |
Set dry gain, how much of original signal is kept. Allowed range is from 0 to 1. Default value is 1.0. | ||
wet |
Set wet gain, how much of filtered signal is kept. Allowed range is from 0 to 1. Default value is 1.0. |
boost
Set max boost factor. Allowed range is from 1 to 12. Default value is 2.
decay
Set delay line decay gain value. Allowed range is from 0 to 1. Default value is 0.0.
feedback
Set delay line feedback gain value. Allowed range is from 0 to 1. Default value is 0.9.
cutoff
Set cutoff frequency in Hertz. Allowed range is 50 to 900. Default value is 100.
slope
Set slope amount for cutoff frequency. Allowed range is 0.0001 to 1. Default value is 0.5.
delay
Set delay. Allowed range is from 1 to 100. Default value is 20.
channels
Set the channels to process. Default value is all available.
Commands
This filter supports the all above options as commands.
asubcut
Cut subwoofer frequencies.
This filter allows to set custom, steeper roll off than highpass filter, and thus is able to more attenuate frequency content in stop-band.
The filter
accepts the following options:
cutoff
Set cutoff frequency in Hertz. Allowed range is 2 to 200. Default value is 20.
order
Set filter order. Available values are from 3 to 20. Default value is 10.
level
Set input gain level. Allowed range is from 0 to 1. Default value is 1.
Commands
This filter supports the all above options as commands.
asupercut
Cut super frequencies.
The filter
accepts the following options:
cutoff
Set cutoff frequency in Hertz. Allowed range is 20000 to 192000. Default value is 20000.
order
Set filter order. Available values are from 3 to 20. Default value is 10.
level
Set input gain level. Allowed range is from 0 to 1. Default value is 1.
Commands
This filter supports the all above options as commands.
asuperpass
Apply high order Butterworth band-pass filter.
The filter
accepts the following options:
centerf
Set center frequency in Hertz. Allowed range is 2 to 999999. Default value is 1000.
order
Set filter order. Available values are from 4 to 20. Default value is 4.
qfactor
Set Q-factor. Allowed range is from 0.01 to 100. Default value is 1.
level
Set input gain level. Allowed range is from 0 to 2. Default value is 1.
Commands
This filter supports the all above options as commands.
asuperstop
Apply high order Butterworth band-stop filter.
The filter
accepts the following options:
centerf
Set center frequency in Hertz. Allowed range is 2 to 999999. Default value is 1000.
order
Set filter order. Available values are from 4 to 20. Default value is 4.
qfactor
Set Q-factor. Allowed range is from 0.01 to 100. Default value is 1.
level
Set input gain level. Allowed range is from 0 to 2. Default value is 1.
Commands
This filter supports the all above options as commands.
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, 100.0] range.
Note that tempo greater than 2 will skip some samples rather than blend them in. If for any reason this is a concern it is always possible to daisy-chain several instances of atempo to achieve the desired product tempo.
Examples
• |
Slow down audio to 80% tempo: |
atempo=0.8
• |
To speed up audio to 300% tempo: |
atempo=3
• |
To speed up audio to 300% tempo by daisy-chaining two atempo instances: |
atempo=sqrt(3),atempo=sqrt(3)
Commands
This filter
supports the following commands:
tempo
Change filter tempo scale factor. Syntax for the command is : "tempo"
atilt
Apply spectral tilt filter to audio stream.
This filter apply any spectral roll-off slope over any specified frequency band.
The filter
accepts the following options:
freq
Set central frequency of tilt in Hz. Default is 10000 Hz.
slope
Set slope direction of tilt. Default is 0. Allowed range is from -1 to 1.
width
Set width of tilt. Default is 1000. Allowed range is from 100 to 10000.
order
Set order of tilt filter.
level
Set input volume level. Allowed range is from 0 to 4. Defalt is 1.
Commands
This filter supports the all above options as commands.
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
axcorrelate
Calculate normalized windowed cross-correlation between two
input audio streams.
Resulted samples are always between -1 and 1 inclusive. If result is 1 it means two input samples are highly correlated in that selected segment. Result 0 means they are not correlated at all. If result is -1 it means two input samples are out of phase, which means they cancel each other.
The filter
accepts the following options:
size
Set size of segment over which cross-correlation is calculated. Default is 256. Allowed range is from 2 to 131072.
algo
Set algorithm for cross-correlation. Can be "slow" or "fast" or "best". Default is "best". Fast algorithm assumes mean values over any given segment are always zero and thus need much less calculations to make. This is generally not true, but is valid for typical audio streams.
Examples
• |
Calculate correlation between channels in stereo audio stream: |
ffmpeg -i stereo.wav -af channelsplit,axcorrelate=size=1024:algo=fast correlation.wav
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, t
Set method to specify band-width of filter.
h |
Hz |
|||
q |
Q-Factor |
|||
o |
octave |
|||
s |
slope |
|||
k |
kHz |
width, w
Specify the band-width of a filter in width_type units.
mix, m
How much to use filtered signal in output. Default is 1. Range is between 0 and 1.
channels, c
Specify which channels to filter, by default all available are filtered.
normalize, n
Normalize biquad coefficients, by default is disabled. Enabling it will normalize magnitude response at DC to 0dB.
transform, a
Set transform type of IIR filter.
di |
||
dii |
||
tdi |
tdii
latt
svf |
||
zdf |
precision, r
Set precison of filtering.
auto
Pick automatic sample format depending on surround filters.
s16 |
Always use signed 16-bit. |
|||
s32 |
Always use signed 32-bit. |
|||
f32 |
Always use float 32-bit. |
|||
f64 |
Always use float 64-bit. |
block_size, b
Set block size used for reverse IIR processing. If this value is set to high enough value (higher than impulse response length truncated when reaches near zero values) filtering will become linear phase otherwise if not big enough it will just produce nasty artifacts.
Note that filter delay will be exactly this many samples when set to non-zero value.
Commands
This filter
supports the following commands:
frequency, f
Change bandpass frequency. Syntax for the command is : "frequency"
width_type, t
Change bandpass width_type. Syntax for the command is : "width_type"
width, w
Change bandpass width. Syntax for the command is : "width"
mix, m
Change bandpass mix. Syntax for the command is : "mix"
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, t
Set method to specify band-width of filter.
h |
Hz |
|||
q |
Q-Factor |
|||
o |
octave |
|||
s |
slope |
|||
k |
kHz |
width, w
Specify the band-width of a filter in width_type units.
mix, m
How much to use filtered signal in output. Default is 1. Range is between 0 and 1.
channels, c
Specify which channels to filter, by default all available are filtered.
normalize, n
Normalize biquad coefficients, by default is disabled. Enabling it will normalize magnitude response at DC to 0dB.
transform, a
Set transform type of IIR filter.
di |
||
dii |
||
tdi |
tdii
latt
svf |
||
zdf |
precision, r
Set precison of filtering.
auto
Pick automatic sample format depending on surround filters.
s16 |
Always use signed 16-bit. |
|||
s32 |
Always use signed 32-bit. |
|||
f32 |
Always use float 32-bit. |
|||
f64 |
Always use float 64-bit. |
block_size, b
Set block size used for reverse IIR processing. If this value is set to high enough value (higher than impulse response length truncated when reaches near zero values) filtering will become linear phase otherwise if not big enough it will just produce nasty artifacts.
Note that filter delay will be exactly this many samples when set to non-zero value.
Commands
This filter
supports the following commands:
frequency, f
Change bandreject frequency. Syntax for the command is : "frequency"
width_type, t
Change bandreject width_type. Syntax for the command is : "width_type"
width, w
Change bandreject width. Syntax for the command is : "width"
mix, m
Change bandreject mix. Syntax for the command is : "mix"
bass,
lowshelf
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, t
Set method to specify band-width of filter.
h |
Hz |
|||
q |
Q-Factor |
|||
o |
octave |
|||
s |
slope |
|||
k |
kHz |
width, w
Determine how steep is the filter’s shelf transition.
poles, p
Set number of poles. Default is 2.
mix, m
How much to use filtered signal in output. Default is 1. Range is between 0 and 1.
channels, c
Specify which channels to filter, by default all available are filtered.
normalize, n
Normalize biquad coefficients, by default is disabled. Enabling it will normalize magnitude response at DC to 0dB.
transform, a
Set transform type of IIR filter.
di |
||
dii |
||
tdi |
tdii
latt
svf |
||
zdf |
precision, r
Set precison of filtering.
auto
Pick automatic sample format depending on surround filters.
s16 |
Always use signed 16-bit. |
|||
s32 |
Always use signed 32-bit. |
|||
f32 |
Always use float 32-bit. |
|||
f64 |
Always use float 64-bit. |
block_size, b
Set block size used for reverse IIR processing. If this value is set to high enough value (higher than impulse response length truncated when reaches near zero values) filtering will become linear phase otherwise if not big enough it will just produce nasty artifacts.
Note that filter delay will be exactly this many samples when set to non-zero value.
Commands
This filter
supports the following commands:
frequency, f
Change bass frequency. Syntax for the command is : "frequency"
width_type, t
Change bass width_type. Syntax for the command is : "width_type"
width, w
Change bass width. Syntax for the command is : "width"
gain, g
Change bass gain. Syntax for the command is : "gain"
mix, m
Change bass mix. Syntax for the command is : "mix"
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. and channels, c specify which
channels to filter, by default all available are
filtered.
Commands
This filter supports the following commands:
a0 |
|||
a1 |
|||
a2 |
|||
b0 |
|||
b1 |
|||
b2 |
Change biquad parameter. Syntax for the command is : "value" |
mix, m
How much to use filtered signal in output. Default is 1. Range is between 0 and 1.
channels, c
Specify which channels to filter, by default all available are filtered.
normalize, n
Normalize biquad coefficients, by default is disabled. Enabling it will normalize magnitude response at DC to 0dB.
transform, a
Set transform type of IIR filter.
di |
||
dii |
||
tdi |
tdii
latt
svf |
||
zdf |
precision, r
Set precison of filtering.
auto
Pick automatic sample format depending on surround filters.
s16 |
Always use signed 16-bit. |
|||
s32 |
Always use signed 32-bit. |
|||
f32 |
Always use float 32-bit. |
|||
f64 |
Always use float 64-bit. |
block_size, b
Set block size used for reverse IIR processing. If this value is set to high enough value (higher than impulse response length truncated when reaches near zero values) filtering will become linear phase otherwise if not big enough it will just produce nasty artifacts.
Note that filter delay will be exactly this many samples when set to non-zero value.
bs2b
Bauer stereo to binaural transformation, which improves
headphone listening of stereo audio records.
To enable compilation of this filter you need to configure FFmpeg with "--enable-libbs2b".
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:
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. |
channel_layout
The channel layout of the output stream.
If no mapping is present, the filter will implicitly map input channels to output channels, preserving indices.
Examples
• |
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".
channels
A channel layout describing the channels to be extracted as separate output streams or "all" to extract each input channel as a separate stream. The default is "all".
Choosing channels not present in channel layout in the input will result in an error.
Examples
• |
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
• |
Extract only LFE from a 5.1 WAV file: |
ffmpeg -i
in.wav -filter_complex
'channelsplit=channel_layout=5.1:channels=LFE[LFE]'
-map '[LFE]' lfe.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|1/0".
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 locations. 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 a distance of ~30 cm between microphones makes one microphone capture the signal in antiphase to the other microphone. That makes the final mix sound 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.
The filter 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 in degrees Celsius. This is the temperature of the environment. Default is 20.
Commands
This filter supports the all above options as commands.
crossfeed
Apply headphone crossfeed filter.
Crossfeed is the process of blending the left and right channels of stereo audio recording. It is mainly used to reduce extreme stereo separation of low frequencies.
The intent is to produce more speaker like sound to the listener.
The filter
accepts the following options:
strength
Set strength of crossfeed. Default is 0.2. Allowed range is from 0 to 1. This sets gain of low shelf filter for side part of stereo image. Default is -6dB. Max allowed is -30db when strength is set to 1.
range
Set soundstage wideness. Default is 0.5. Allowed range is from 0 to 1. This sets cut off frequency of low shelf filter. Default is cut off near 1550 Hz. With range set to 1 cut off frequency is set to 2100 Hz.
slope
Set curve slope of low shelf filter. Default is 0.5. Allowed range is from 0.01 to 1.
level_in
Set input gain. Default is 0.9.
level_out
Set output gain. Default is 1.
block_size
Set block size used for reverse IIR processing. If this value is set to high enough value (higher than impulse response length truncated when reaches near zero values) filtering will become linear phase otherwise if not big enough it will just produce nasty artifacts.
Note that filter delay will be exactly this many samples when set to non-zero value.
Commands
This filter supports the all above options as commands.
crystalizer
Simple algorithm for audio noise sharpening.
This filter linearly increases differences betweeen each audio sample.
The filter accepts the following options:
i |
Sets the intensity of effect (default: 2.0). Must be in range between -10.0 to 0 (unchanged sound) to 10.0 (maximum effect). To inverse filtering use negative value. | ||
c |
Enable clipping. By default is enabled. |
Commands
This filter supports the all above options as commands.
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.
deesser
Apply de-essing to the audio samples.
i |
Set intensity for triggering de-essing. Allowed range is from 0 to 1. Default is 0. | ||
m |
Set amount of ducking on treble part of sound. Allowed range is from 0 to 1. Default is 0.5. | ||
f |
How much of original frequency content to keep when de-essing. Allowed range is from 0 to 1. Default is 0.5. | ||
s |
Set the output mode. |
It accepts the following values:
i |
Pass input unchanged. |
|||
o |
Pass ess filtered out. |
|||
e |
Pass only ess. |
Default value is o.
dialoguenhance
Enhance dialogue in stereo audio.
This filter accepts stereo input and produce surround (3.0) channels output. The newly produced front center channel have enhanced speech dialogue originally available in both stereo channels. This filter outputs front left and front right channels same as available in stereo input.
The filter
accepts the following options:
original
Set the original center factor to keep in front center channel output. Allowed range is from 0 to 1. Default value is 1.
enhance
Set the dialogue enhance factor to put in front center channel output. Allowed range is from 0 to 3. Default value is 1.
voice
Set the voice detection factor. Allowed range is from 2 to 32. Default value is 2.
Commands
This filter supports the all above options as commands.
drmeter
Measure audio dynamic range.
DR values of 14 and higher is found in very dynamic material. DR of 8 to 13 is found in transition material. And anything less that 8 have very poor dynamics and is very compressed.
The filter
accepts the following options:
length
Set window length in seconds used to split audio into segments of equal length. Default is 3 seconds.
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.
framelen, 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.
gausssize, 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.
peak, 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.
maxgain, 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.
targetrms, 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.
coupling, 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.
correctdc, 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.
altboundary, 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.
compress, 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.
threshold, t
Set the target threshold value. This specifies the lowest permissible magnitude level for the audio input which will be normalized. If input frame volume is above this value frame will be normalized. Otherwise frame may not be normalized at all. The default value is set to 0, which means all input frames will be normalized. This option is mostly useful if digital noise is not wanted to be amplified.
channels, h
Specify which channels to filter, by default all available channels are filtered.
overlap, o
Specify overlap for frames. If set to 0 (default) no frame overlapping is done. Using >0 and <1 values will make less conservative gain adjustments, like when framelen option is set to smaller value, if framelen option value is compensated for non-zero overlap then gain adjustments will be smoother across time compared to zero overlap case.
curve, v
Specify the peak mapping curve expression which is going to be used when calculating gain applied to frames. The max output frame gain will still be limited by other options mentioned previously for this filter.
The expression can contain the following constants:
ch |
current channel number |
|||
sn |
current sample number |
nb_channels
number of channels
t |
timestamp expressed in seconds |
|||
sr |
sample rate |
|||
p |
current frame peak value |
Commands
This filter supports the all above options as commands.
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, t
Set method to specify band-width of filter.
h |
Hz |
|||
q |
Q-Factor |
|||
o |
octave |
|||
s |
slope |
|||
k |
kHz |
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.
mix, m
How much to use filtered signal in output. Default is 1. Range is between 0 and 1.
channels, c
Specify which channels to filter, by default all available are filtered.
normalize, n
Normalize biquad coefficients, by default is disabled. Enabling it will normalize magnitude response at DC to 0dB.
transform, a
Set transform type of IIR filter.
di |
||
dii |
||
tdi |
tdii
latt
svf |
||
zdf |
precision, r
Set precison of filtering.
auto
Pick automatic sample format depending on surround filters.
s16 |
Always use signed 16-bit. |
|||
s32 |
Always use signed 32-bit. |
|||
f32 |
Always use float 32-bit. |
|||
f64 |
Always use float 64-bit. |
block_size, b
Set block size used for reverse IIR processing. If this value is set to high enough value (higher than impulse response length truncated when reaches near zero values) filtering will become linear phase otherwise if not big enough it will just produce nasty artifacts.
Note that filter delay will be exactly this many samples when set to non-zero value.
Examples
• |
Attenuate 10 dB at 1000 Hz, with a bandwidth of 200 Hz: |
equalizer=f=1000:t=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:t=q:w=1:g=2,equalizer=f=100:t=q:w=2:g=-5
Commands
This filter
supports the following commands:
frequency, f
Change equalizer frequency. Syntax for the command is : "frequency"
width_type, t
Change equalizer width_type. Syntax for the command is : "width_type"
width, w
Change equalizer width. Syntax for the command is : "width"
gain, g
Change equalizer gain. Syntax for the command is : "gain"
mix, m
Change equalizer mix. Syntax for the command is : "mix"
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. |
Commands
This filter supports the all above options as commands.
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.
fft2
Enable 2-channel convolution using complex FFT. This improves speed significantly. Default is disabled.
min_phase
Enable minimum phase impulse response. Default is disabled.
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 sweep 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.
haas
Apply Haas effect to audio.
Note that this makes most sense to apply on mono signals. With this filter applied to mono signals it give some directionality and stretches its stereo image.
The filter
accepts the following options:
level_in
Set input level. By default is 1, or 0dB
level_out
Set output level. By default is 1, or 0dB.
side_gain
Set gain applied to side part of signal. By default is 1.
middle_source
Set kind of middle source. Can
be one of the following:
left
Pick left channel.
right
Pick right channel.
mid |
Pick middle part signal of stereo image. |
side
Pick side part signal of stereo image.
middle_phase
Change middle phase. By default is disabled.
left_delay
Set left channel delay. By default is 2.05 milliseconds.
left_balance
Set left channel balance. By default is -1.
left_gain
Set left channel gain. By default is 1.
left_phase
Change left phase. By default is disabled.
right_delay
Set right channel delay. By defaults is 2.12 milliseconds.
right_balance
Set right channel balance. By default is 1.
right_gain
Set right channel gain. By default is 1.
right_phase
Change right phase. By default is enabled.
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 -c:a 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
headphone
Apply head-related transfer functions (HRTFs) to create
virtual loudspeakers around the user for binaural listening
via headphones. The HRIRs are provided via additional
streams, for each channel one stereo input stream is
needed.
The filter accepts the following options:
map |
Set mapping of input streams for convolution. The argument is a ’|’-separated list of channel names in order as they are given as additional stream inputs for filter. This also specify number of input streams. Number of input streams must be not less than number of channels in first stream plus one. |
gain
Set gain applied to audio. Value is in dB. Default is 0.
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.
lfe |
Set custom gain for LFE channels. Value is in dB. Default is 0. |
size
Set size of frame in number of samples which will be processed at once. Default value is 1024. Allowed range is from 1024 to 96000.
hrir
Set format of hrir stream. Default value is stereo. Alternative value is multich. If value is set to stereo, number of additional streams should be greater or equal to number of input channels in first input stream. Also each additional stream should have stereo number of channels. If value is set to multich, number of additional streams should be exactly one. Also number of input channels of additional stream should be equal or greater than twice number of channels of first input stream.
Examples
• |
Full example using wav files as coefficients with amovie filters for 7.1 downmix, each amovie filter use stereo file with IR coefficients as input. The files give coefficients for each position of virtual loudspeaker: |
ffmpeg -i
input.wav
-filter_complex
"amovie=azi_270_ele_0_DFC.wav[sr];amovie=azi_90_ele_0_DFC.wav[sl];amovie=azi_225_ele_0_DFC.wav[br];amovie=azi_135_ele_0_DFC.wav[bl];amovie=azi_0_ele_0_DFC.wav,asplit[fc][lfe];amovie=azi_35_ele_0_DFC.wav[fl];amovie=azi_325_ele_0_DFC.wav[fr];[0:a][fl][fr][fc][lfe][bl][br][sl][sr]headphone=FL|FR|FC|LFE|BL|BR|SL|SR"
output.wav
• |
Full example using wav files as coefficients with amovie filters for 7.1 downmix, but now in multich hrir format. |
ffmpeg -i
input.wav -filter_complex
"amovie=minp.wav[hrirs];[0:a][hrirs]headphone=map=FL|FR|FC|LFE|BL|BR|SL|SR:hrir=multich"
output.wav
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, t
Set method to specify band-width of filter.
h |
Hz |
|||
q |
Q-Factor |
|||
o |
octave |
|||
s |
slope |
|||
k |
kHz |
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.
mix, m
How much to use filtered signal in output. Default is 1. Range is between 0 and 1.
channels, c
Specify which channels to filter, by default all available are filtered.
normalize, n
Normalize biquad coefficients, by default is disabled. Enabling it will normalize magnitude response at DC to 0dB.
transform, a
Set transform type of IIR filter.
di |
||
dii |
||
tdi |
tdii
latt
svf |
||
zdf |
precision, r
Set precison of filtering.
auto
Pick automatic sample format depending on surround filters.
s16 |
Always use signed 16-bit. |
|||
s32 |
Always use signed 32-bit. |
|||
f32 |
Always use float 32-bit. |
|||
f64 |
Always use float 64-bit. |
block_size, b
Set block size used for reverse IIR processing. If this value is set to high enough value (higher than impulse response length truncated when reaches near zero values) filtering will become linear phase otherwise if not big enough it will just produce nasty artifacts.
Note that filter delay will be exactly this many samples when set to non-zero value.
Commands
This filter
supports the following commands:
frequency, f
Change highpass frequency. Syntax for the command is : "frequency"
width_type, t
Change highpass width_type. Syntax for the command is : "width_type"
width, w
Change highpass width. Syntax for the command is : "width"
mix, m
Change highpass mix. Syntax for the command is : "mix"
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.
latency, l
Enable latency compensation, by default is disabled. Only used if plugin have 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
• |
Reduce stereo image using "Narrower" from the "C* Audio Plugin Suite" (CAPS) library: |
ladspa=caps:Narrower
• |
Another white noise, now using "C* Audio Plugin Suite" (CAPS) library: |
ladspa=caps:White:.2
• |
Some fractal noise, using "C* Audio Plugin Suite" (CAPS) library: |
ladspa=caps:Fractal:c=c1=1
• |
Dynamic volume normalization using "VLevel" plugin: |
ladspa=vlevel-ladspa:vlevel_mono
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. In
dynamic mode, to accurately detect true peaks, the audio
stream will be upsampled to 192 kHz. Use the "-ar"
option or "aresample" filter to explicitly set an
output sample rate.
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 - 50.0. 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 by linearly scaling the source audio. "measured_I", "measured_LRA", "measured_TP", and "measured_thresh" must all be specified. Target LRA shouldn’t be lower than source LRA and the change in integrated loudness shouldn’t result in a true peak which exceeds the target TP. If any of these conditions aren’t met, normalization mode will revert to dynamic. 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, t
Set method to specify band-width of filter.
h |
Hz |
|||
q |
Q-Factor |
|||
o |
octave |
|||
s |
slope |
|||
k |
kHz |
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.
mix, m
How much to use filtered signal in output. Default is 1. Range is between 0 and 1.
channels, c
Specify which channels to filter, by default all available are filtered.
normalize, n
Normalize biquad coefficients, by default is disabled. Enabling it will normalize magnitude response at DC to 0dB.
transform, a
Set transform type of IIR filter.
di |
||
dii |
||
tdi |
tdii
latt
svf |
||
zdf |
precision, r
Set precison of filtering.
auto
Pick automatic sample format depending on surround filters.
s16 |
Always use signed 16-bit. |
|||
s32 |
Always use signed 32-bit. |
|||
f32 |
Always use float 32-bit. |
|||
f64 |
Always use float 64-bit. |
block_size, b
Set block size used for reverse IIR processing. If this value is set to high enough value (higher than impulse response length truncated when reaches near zero values) filtering will become linear phase otherwise if not big enough it will just produce nasty artifacts.
Note that filter delay will be exactly this many samples when set to non-zero value.
Examples
• |
Lowpass only LFE channel, it LFE is not present it does nothing: |
lowpass=c=LFE
Commands
This filter
supports the following commands:
frequency, f
Change lowpass frequency. Syntax for the command is : "frequency"
width_type, t
Change lowpass width_type. Syntax for the command is : "width_type"
width, w
Change lowpass width. Syntax for the command is : "width"
mix, m
Change lowpass mix. Syntax for the command is : "mix"
lv2
Load a LV2 (LADSPA Version 2) plugin.
To enable
compilation of this filter you need to configure FFmpeg with
"--enable-lv2".
plugin, p
Specifies the plugin URI. You may need to escape ’:’.
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). 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
• |
Apply bass enhancer plugin from Calf: |
lv2=p=http\\\\://calf.sourceforge.net/plugins/BassEnhancer:c=amount=2
• |
Apply vinyl plugin from Calf: |
lv2=p=http\\\\://calf.sourceforge.net/plugins/Vinyl:c=drone=0.2|aging=0.5
• |
Apply bit crusher plugin from ArtyFX: |
lv2=p=http\\\\://www.openavproductions.com/artyfx#bitta:c=crush=0.3
Commands
This filter supports all options that are exported by plugin as commands.
mcompand
Multiband Compress or expand the audio’s dynamic
range.
The input audio is divided into bands using 4th order Linkwitz-Riley IIRs. This is akin to the crossover of a loudspeaker, and results in flat frequency response when absent compander action.
It accepts the
following parameters:
args
This option syntax is: attack,decay,[attack,decay..] soft-knee points crossover_frequency [delay [initial_volume [gain]]] | attack,decay ... For explanation of each item refer to compand filter documentation.
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".
The filter
accepts the following exported read-only options:
track_gain
Exported track gain in dB at end of stream.
track_peak
Exported track peak at end of stream.
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.
To enable compilation of this filter, you need to configure FFmpeg with "--enable-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
Commands
This filter
supports the following commands:
tempo
Change filter tempo scale factor. Syntax for the command is : "tempo"
pitch
Change filter pitch scale factor. Syntax for the command is : "pitch"
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.
mode
Set mode of compressor operation. Can be "upward" or "downward". Default is "downward".
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 1. Range is from 1 to 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. |
Commands
This filter supports the all above options as commands.
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.
mode
Set the mode of operation. Can be "upward" or "downward". Default is "downward". If set to "upward" mode, higher parts of signal will be amplified, expanding dynamic range in upward direction. Otherwise, in case of "downward" lower parts of signal will be reduced.
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. Setting this to 0 disables reduction and then filter behaves like expander.
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.
Commands
This filter supports the all above options as commands.
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 "lavfi.silence_start" or "lavfi.silence_start.X" metadata key is set on the first frame whose timestamp equals or exceeds the detection duration and it contains the timestamp of the first frame of the silence.
The "lavfi.silence_duration" or "lavfi.silence_duration.X" and "lavfi.silence_end" or "lavfi.silence_end.X" metadata keys are set on the first frame after the silence. If mono is enabled, and each channel is evaluated separately, the ".X" suffixed keys are used, and "X" corresponds to the channel number.
The filter
accepts the following options:
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.
duration, d
Set silence duration until notification (default is 2 seconds). See the Time duration section in the ffmpeg-utils(1) manual for the accepted syntax.
mono, m
Process each channel separately, instead of combined. By default is disabled.
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.
start_silence
Specify max duration of silence at beginning that will be kept after trimming. Default is 0, which is equal to trimming all samples detected as silence.
start_mode
Specify mode of detection of silence end at start of multi-channel audio. Can be any or all. Default is any. With any, any sample from any channel that is detected as non-silence will trigger end of silence trimming at start of audio stream. With all, only if every sample from every channel is detected as non-silence will trigger end of silence trimming at start of audio stream, limited usage.
stop_periods
Set the count for trimming silence from the end of audio. When specifying a positive value, it trims audio after it finds specified silence period. 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 stop_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.
stop_silence
Specify max duration of silence at end that will be kept after trimming. Default is 0, which is equal to trimming all samples detected as silence.
stop_mode
Specify mode of detection of silence start after start of multi-channel audio. Can be any or all. Default is all. With any, any sample from any channel that is detected as silence will trigger start of silence trimming after start of audio stream, limited usage. With all, only if every sample from every channel is detected as silence will trigger start of silence trimming after start of audio stream.
detection
Set how is silence detected.
avg |
Mean of absolute values of samples in moving window. | ||
rms |
Root squared mean of absolute values of samples in moving window. |
peak
Maximum of absolute values of samples in moving window.
median
Median of absolute values of samples in moving window.
ptp |
Absolute of max peak to min peak difference of samples in moving window. | ||
dev |
Standard deviation of values of samples in moving window. |
Default value is "rms".
window
Set duration in number of seconds used to calculate size of window in number of samples for detecting silence. Using 0 will effectively disable any windowing and use only single sample per channel for silence detection. In that case it may be needed to also set start_silence and/or stop_silence to nonzero values with also start_duration and/or stop_duration to nonzero values. Default value is 0.02. Allowed range is from 0 to 10.
timestamp
Set processing mode of every
audio frame output timestamp.
write
Full timestamps rewrite, keep only the start time for the first output frame.
copy
Non-dropped frames are left with same timestamp as input audio frame.
Defaults value is "write".
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=start_periods=1:start_duration=5:start_threshold=0.02
• |
Trim all silence encountered from beginning to end where there is more than 1 second of silence in audio: |
silenceremove=stop_periods=-1:stop_duration=1:stop_threshold=-90dB
• |
Trim all digital silence samples, using peak detection, from beginning to end where there is more than 0 samples of digital silence in audio and digital silence is detected in all channels at same positions in stream: |
silenceremove=window=0:detection=peak:stop_mode=all:start_mode=all:stop_periods=-1:stop_threshold=0
• |
Trim every 2nd encountered silence period from beginning to end where there is more than 1 second of silence per silence period in audio: |
silenceremove=stop_periods=-2:stop_duration=1:stop_threshold=-90dB
• |
Similar as above, but keep maximum of 0.5 seconds of silence from each trimmed period: |
silenceremove=stop_periods=-2:stop_duration=1:stop_threshold=-90dB:stop_silence=0.5
• |
Similar as above, but keep maximum of 1.5 seconds of silence from start of audio: |
silenceremove=stop_periods=-2:stop_duration=1:stop_threshold=-90dB:stop_silence=0.5:start_periods=1:start_duration=1:start_silence=1.5:stop_threshold=-90dB
Commands
This filter supports some above options as commands.
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-libmysofa".
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 degrees. 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.
lfegain
Set custom gain for LFE channels. Value is in dB. Default is 0.
framesize
Set custom frame size in number of samples. Default is 1024. Allowed range is from 1024 to 96000. Only used if option type is set to freq.
normalize
Should all IRs be normalized upon importing SOFA file. By default is enabled.
interpolate
Should nearest IRs be interpolated with neighbor IRs if exact position does not match. By default is disabled.
minphase
Minphase all IRs upon loading of SOFA file. By default is disabled.
anglestep
Set neighbor search angle step. Only used if option interpolate is enabled.
radstep
Set neighbor search radius step. Only used if option interpolate is enabled.
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, back left and back right and also with custom gain: |
"sofalizer=sofa=/path/to/ClubFritz6.sofa:type=freq:radius=2:speakers=FL 45|FR 315|BL 135|BR 225:gain=28"
speechnorm
Speech Normalizer.
This filter expands or compresses each half-cycle of audio samples (local set of samples all above or all below zero and between two nearest zero crossings) depending on threshold value, so audio reaches target peak value under conditions controlled by below options.
The filter
accepts the following options:
peak, p
Set the expansion target peak value. This specifies the highest allowed absolute amplitude level for the normalized audio input. Default value is 0.95. Allowed range is from 0.0 to 1.0.
expansion, e
Set the maximum expansion factor. Allowed range is from 1.0 to 50.0. Default value is 2.0. This option controls maximum local half-cycle of samples expansion. The maximum expansion would be such that local peak value reaches target peak value but never to surpass it and that ratio between new and previous peak value does not surpass this option value.
compression, c
Set the maximum compression factor. Allowed range is from 1.0 to 50.0. Default value is 2.0. This option controls maximum local half-cycle of samples compression. This option is used only if threshold option is set to value greater than 0.0, then in such cases when local peak is lower or same as value set by threshold all samples belonging to that peak’s half-cycle will be compressed by current compression factor.
threshold, t
Set the threshold value. Default value is 0.0. Allowed range is from 0.0 to 1.0. This option specifies which half-cycles of samples will be compressed and which will be expanded. Any half-cycle samples with their local peak value below or same as this option value will be compressed by current compression factor, otherwise, if greater than threshold value they will be expanded with expansion factor so that it could reach peak target value but never surpass it.
raise, r
Set the expansion raising amount per each half-cycle of samples. Default value is 0.001. Allowed range is from 0.0 to 1.0. This controls how fast expansion factor is raised per each new half-cycle until it reaches expansion value. Setting this options too high may lead to distortions.
fall, f
Set the compression raising amount per each half-cycle of samples. Default value is 0.001. Allowed range is from 0.0 to 1.0. This controls how fast compression factor is raised per each new half-cycle until it reaches compression value.
channels, h
Specify which channels to filter, by default all available channels are filtered.
invert, i
Enable inverted filtering, by default is disabled. This inverts interpretation of threshold option. When enabled any half-cycle of samples with their local peak value below or same as threshold option will be expanded otherwise it will be compressed.
link, l
Link channels when calculating gain applied to each filtered channel sample, by default is disabled. When disabled each filtered channel gain calculation is independent, otherwise when this option is enabled the minimum of all possible gains for each filtered channel is used.
rms, m
Set the expansion target RMS value. This specifies the highest allowed RMS level for the normalized audio input. Default value is 0.0, thus disabled. Allowed range is from 0.0 to 1.0.
Commands
This filter supports the all above options as commands.
Examples
• |
Weak and slow amplification: |
speechnorm=e=3:r=0.00001:l=1
• |
Moderate and slow amplification: |
speechnorm=e=6.25:r=0.00001:l=1
• |
Strong and fast amplification: |
speechnorm=e=12.5:r=0.0001:l=1
• |
Very strong and fast amplification: |
speechnorm=e=25:r=0.0001:l=1
• |
Extreme and fast amplification: |
speechnorm=e=50:r=0.0001:l=1
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.
ms>ll
Mid/Side to Left/Left.
ms>rr
Mid/Side to Right/Right.
ms>rl
Mid/Side to Right/Left.
lr>l-r
Left/Right to Left - Right.
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.
bmode_in, bmode_out
Set balance mode for balance_in/balance_out option.
Can be one of
the following:
balance
Classic balance mode. Attenuate one channel at time. Gain is raised up to 1.
amplitude
Similar as classic mode above but gain is raised up to 2.
power
Equal power distribution, from -6dB to +6dB range.
Commands
This filter supports the all above options as commands.
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.
Commands
This filter supports the all above options except "delay" as commands.
superequalizer
Apply 18 band equalizer.
The filter accepts the following options:
1b |
Set 65Hz band gain. |
|||
2b |
Set 92Hz band gain. |
|||
3b |
Set 131Hz band gain. |
|||
4b |
Set 185Hz band gain. |
|||
5b |
Set 262Hz band gain. |
|||
6b |
Set 370Hz band gain. |
|||
7b |
Set 523Hz band gain. |
|||
8b |
Set 740Hz band gain. |
|||
9b |
Set 1047Hz band gain. |
|||
10b |
Set 1480Hz band gain. |
|||
11b |
Set 2093Hz band gain. |
|||
12b |
Set 2960Hz band gain. |
|||
13b |
Set 4186Hz band gain. |
|||
14b |
Set 5920Hz band gain. |
|||
15b |
Set 8372Hz band gain. |
|||
16b |
Set 11840Hz band gain. |
|||
17b |
Set 16744Hz band gain. |
|||
18b |
Set 20000Hz band gain. |
surround
Apply audio surround upmix filter.
This filter allows to produce multichannel output from audio stream.
The filter
accepts the following options:
chl_out
Set output channel layout. By default, this is 5.1.
See the Channel Layout section in the ffmpeg-utils(1) manual for the required syntax.
chl_in
Set input channel layout. By default, this is stereo.
See the Channel Layout section in the ffmpeg-utils(1) manual for the required syntax.
level_in
Set input volume level. By default, this is 1.
level_out
Set output volume level. By default, this is 1.
lfe |
Enable LFE channel output if output channel layout has it. By default, this is enabled. |
lfe_low
Set LFE low cut off frequency. By default, this is 128 Hz.
lfe_high
Set LFE high cut off frequency. By default, this is 256 Hz.
lfe_mode
Set LFE mode, can be add or sub. Default is add. In add mode, LFE channel is created from input audio and added to output. In sub mode, LFE channel is created from input audio and added to output but also all non-LFE output channels are subtracted with output LFE channel.
smooth
Set temporal smoothness strength, used to gradually change factors when transforming stereo sound in time. Allowed range is from 0.0 to 1.0. Useful to improve output quality with focus option values greater than 0.0. Default is 0.0. Only values inside this range and without edges are effective.
angle
Set angle of stereo surround transform, Allowed range is from 0 to 360. Default is 90.
focus
Set focus of stereo surround transform, Allowed range is from -1 to 1. Default is 0.
fc_in
Set front center input volume. By default, this is 1.
fc_out
Set front center output volume. By default, this is 1.
fl_in
Set front left input volume. By default, this is 1.
fl_out
Set front left output volume. By default, this is 1.
fr_in
Set front right input volume. By default, this is 1.
fr_out
Set front right output volume. By default, this is 1.
sl_in
Set side left input volume. By default, this is 1.
sl_out
Set side left output volume. By default, this is 1.
sr_in
Set side right input volume. By default, this is 1.
sr_out
Set side right output volume. By default, this is 1.
bl_in
Set back left input volume. By default, this is 1.
bl_out
Set back left output volume. By default, this is 1.
br_in
Set back right input volume. By default, this is 1.
br_out
Set back right output volume. By default, this is 1.
bc_in
Set back center input volume. By default, this is 1.
bc_out
Set back center output volume. By default, this is 1.
lfe_in
Set LFE input volume. By default, this is 1.
lfe_out
Set LFE output volume. By default, this is 1.
allx
Set spread usage of stereo image across X axis for all channels. Allowed range is from -1 to 15. By default this value is negative -1, and thus unused.
ally
Set spread usage of stereo image across Y axis for all channels. Allowed range is from -1 to 15. By default this value is negative -1, and thus unused.
fcx, flx, frx, blx, brx, slx, srx, bcx
Set spread usage of stereo image across X axis for each channel. Allowed range is from 0.06 to 15. By default this value is 0.5.
fcy, fly, fry, bly, bry, sly, sry, bcy
Set spread usage of stereo image across Y axis for each channel. Allowed range is from 0.06 to 15. By default this value is 0.5.
win_size
Set window size. Allowed range is from 1024 to 65536. Default size is 4096.
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
bohman
kaiser
Default is "hann".
overlap
Set window overlap. If set to 1, the recommended overlap for selected window function will be picked. Default is 0.5.
tiltshelf
Boost or cut the lower frequencies and cut or boost higher
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 3000 Hz.
width_type, t
Set method to specify band-width of filter.
h |
Hz |
|||
q |
Q-Factor |
|||
o |
octave |
|||
s |
slope |
|||
k |
kHz |
width, w
Determine how steep is the filter’s shelf transition.
poles, p
Set number of poles. Default is 2.
mix, m
How much to use filtered signal in output. Default is 1. Range is between 0 and 1.
channels, c
Specify which channels to filter, by default all available are filtered.
normalize, n
Normalize biquad coefficients, by default is disabled. Enabling it will normalize magnitude response at DC to 0dB.
transform, a
Set transform type of IIR filter.
di |
||
dii |
||
tdi |
tdii
latt
svf |
||
zdf |
precision, r
Set precison of filtering.
auto
Pick automatic sample format depending on surround filters.
s16 |
Always use signed 16-bit. |
|||
s32 |
Always use signed 32-bit. |
|||
f32 |
Always use float 32-bit. |
|||
f64 |
Always use float 64-bit. |
block_size, b
Set block size used for reverse IIR processing. If this value is set to high enough value (higher than impulse response length truncated when reaches near zero values) filtering will become linear phase otherwise if not big enough it will just produce nasty artifacts.
Note that filter delay will be exactly this many samples when set to non-zero value.
Commands
This filter supports some options as commands.
treble,
highshelf
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, t
Set method to specify band-width of filter.
h |
Hz |
|||
q |
Q-Factor |
|||
o |
octave |
|||
s |
slope |
|||
k |
kHz |
width, w
Determine how steep is the filter’s shelf transition.
poles, p
Set number of poles. Default is 2.
mix, m
How much to use filtered signal in output. Default is 1. Range is between 0 and 1.
channels, c
Specify which channels to filter, by default all available are filtered.
normalize, n
Normalize biquad coefficients, by default is disabled. Enabling it will normalize magnitude response at DC to 0dB.
transform, a
Set transform type of IIR filter.
di |
||
dii |
||
tdi |
tdii
latt
svf |
||
zdf |
precision, r
Set precison of filtering.
auto
Pick automatic sample format depending on surround filters.
s16 |
Always use signed 16-bit. |
|||
s32 |
Always use signed 32-bit. |
|||
f32 |
Always use float 32-bit. |
|||
f64 |
Always use float 64-bit. |
block_size, b
Set block size used for reverse IIR processing. If this value is set to high enough value (higher than impulse response length truncated when reaches near zero values) filtering will become linear phase otherwise if not big enough it will just produce nasty artifacts.
Note that filter delay will be exactly this many samples when set to non-zero value.
Commands
This filter
supports the following commands:
frequency, f
Change treble frequency. Syntax for the command is : "frequency"
width_type, t
Change treble width_type. Syntax for the command is : "width_type"
width, w
Change treble width. Syntax for the command is : "width"
gain, g
Change treble gain. Syntax for the command is : "gain"
mix, m
Change treble mix. Syntax for the command is : "mix"
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. |
virtualbass
Apply audio Virtual Bass filter.
This filter accepts stereo input and produce stereo with LFE (2.1) channels output. The newly produced LFE channel have enhanced virtual bass originally obtained from both stereo channels. This filter outputs front left and front right channels unchanged as available in stereo input.
The filter
accepts the following options:
cutoff
Set the virtual bass cutoff frequency. Default value is 250 Hz. Allowed range is from 100 to 500 Hz.
strength
Set the virtual bass strength. Allowed range is from 0.5 to 3. Default value is 3.
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.
replaygain_noclip
Prevent clipping by limiting the gain applied.
Default value for replaygain_noclip is 1.
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; deprecated, do not use | ||
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.
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. It supports only 16-bit signed integer samples, so the input will be converted when needed. 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/buffersrc.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)"
afdelaysrc
Generate a fractional delay FIR coefficients.
The resulting stream can be used with afir filter for filtering the audio signal.
The filter
accepts the following options:
delay, d
Set the fractional delay. Default is 0.
sample_rate, r
Set the sample rate, default is 44100.
nb_samples, n
Set the number of samples per each frame. Default is 1024.
taps, t
Set the number of filter coefficents in output audio stream. Default value is 0.
channel_layout, c
Specifies the channel layout, and can be a string representing a channel layout. The default value of channel_layout is "stereo".
afireqsrc
Generate a FIR equalizer coefficients.
The resulting stream can be used with afir filter for filtering the audio signal.
The filter
accepts the following options:
preset, p
Set equalizer preset. Default preset is "flat".
Available
presets are:
custom
flat
acoustic
bass
beats
classic
clear
deep bass
dubstep
electronic
hard-style
hip-hop
jazz
metal
movie
pop |
||
r&b |
rock
vocal booster
gains, g
Set custom gains for each band. Only used if the preset option is set to "custom". Gains are separated by white spaces and each gain is set in dBFS. Default is "0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0".
bands, b
Set the custom bands from where custon equalizer gains are set. This must be in strictly increasing order. Only used if the preset option is set to "custom". Bands are separated by white spaces and each band represent frequency in Hz. Default is "25 40 63 100 160 250 400 630 1000 1600 2500 4000 6300 10000 16000 24000".
taps, t
Set number of filter coefficents in output audio stream. Default value is 4096.
sample_rate, r
Set sample rate of output audio stream, default is 44100.
nb_samples, n
Set number of samples per each frame in output audio stream. Default is 1024.
interp, i
Set interpolation method for FIR equalizer coefficients. Can be "linear" or "cubic".
phase, h
Set phase type of FIR filter. Can be "linear" or "min": minimum-phase. Default is minimum-phase filter.
afirsrc
Generate a FIR coefficients using frequency sampling
method.
The resulting stream can be used with afir filter for filtering the audio signal.
The filter
accepts the following options:
taps, t
Set number of filter coefficents in output audio stream. Default value is 1025.
frequency, f
Set frequency points from where magnitude and phase are set. This must be in non decreasing order, and first element must be 0, while last element must be 1. Elements are separated by white spaces.
magnitude, m
Set magnitude value for every frequency point set by frequency. Number of values must be same as number of frequency points. Values are separated by white spaces.
phase, p
Set phase value for every frequency point set by frequency. Number of values must be same as number of frequency points. Values are separated by white spaces.
sample_rate, r
Set sample rate, default is 44100.
nb_samples, n
Set number of samples per each frame. Default is 1024.
win_func, w
Set window function. Default is blackman.
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.
duration, d
Set the duration of the sourced audio. 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 audio is supposed to be generated forever.
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 versions of the flite library prior to 2.0 are 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.festvox.org/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, brown, blue, violet and velvet. 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.
density
Set the density (0.0 - 1.0) for the velvet noise generator, default is 0.05.
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
hilbert
Generate odd-tap Hilbert transform FIR coefficients.
The resulting stream can be used with afir filter for phase-shifting the signal by 90 degrees.
This is used in many matrix coding schemes and for analytic signal generation. The process is often written as a multiplication by i (or j), the imaginary unit.
The filter
accepts the following options:
sample_rate, s
Set sample rate, default is 44100.
taps, t
Set length of FIR filter, default is 22051.
nb_samples, n
Set number of samples per each frame.
win_func, w
Set window function to be used when generating FIR coefficients.
sinc
Generate a sinc kaiser-windowed low-pass, high-pass,
band-pass, or band-reject FIR coefficients.
The resulting stream can be used with afir filter for filtering the audio signal.
The filter
accepts the following options:
sample_rate, r
Set sample rate, default is 44100.
nb_samples, n
Set number of samples per each frame. Default is 1024.
hp |
Set high-pass frequency. Default is 0. | ||
lp |
Set low-pass frequency. Default is 0. If high-pass frequency is lower than low-pass frequency and low-pass frequency is higher than 0 then filter will create band-pass filter coefficients, otherwise band-reject filter coefficients. |
phase
Set filter phase response. Default is 50. Allowed range is from 0 to 100.
beta
Set Kaiser window beta.
att |
Set stop-band attenuation. Default is 120dB, allowed range is from 40 to 180 dB. |
round
Enable rounding, by default is disabled.
hptaps
Set number of taps for high-pass filter.
lptaps
Set number of taps for low-pass filter.
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.
addroi
Mark a region of interest in a video frame.
The frame data is passed through unchanged, but metadata is attached to the frame indicating regions of interest which can affect the behaviour of later encoding. Multiple regions can be marked by applying the filter multiple times.
x |
Region distance in pixels from the left edge of the frame. | ||
y |
Region distance in pixels from the top edge of the frame. | ||
w |
Region width in pixels. | ||
h |
Region height in pixels. |
The parameters x, y, w and h are expressions, and may contain the following variables:
iw |
Width of the input frame. |
|||
ih |
Height of the input frame. |
qoffset
Quantisation offset to apply within the region.
This must be a real value in the range -1 to +1. A value of zero indicates no quality change. A negative value asks for better quality (less quantisation), while a positive value asks for worse quality (greater quantisation).
The range is calibrated so that the extreme values indicate the largest possible offset - if the rest of the frame is encoded with the worst possible quality, an offset of -1 indicates that this region should be encoded with the best possible quality anyway. Intermediate values are then interpolated in some codec-dependent way.
For example, in 10-bit H.264 the quantisation parameter varies between -12 and 51. A typical qoffset value of -1/10 therefore indicates that this region should be encoded with a QP around one-tenth of the full range better than the rest of the frame. So, if most of the frame were to be encoded with a QP of around 30, this region would get a QP of around 24 (an offset of approximately -1/10 * (51 - -12) = -6.3). An extreme value of -1 would indicate that this region should be encoded with the best possible quality regardless of the treatment of the rest of the frame - that is, should be encoded at a QP of -12.
clear
If set to true, remove any existing regions of interest marked on the frame before adding the new one.
Examples
• |
Mark the centre quarter of the frame as interesting. |
addroi=iw/4:ih/4:iw/2:ih/2:-1/10
• |
Mark the 100-pixel-wide region on the left edge of the frame as very uninteresting (to be encoded at much lower quality than the rest of the frame). |
addroi=0:0:100:ih:+1/5
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]
amplify
Amplify differences between current pixel and pixels of
adjacent frames in same pixel location.
This filter
accepts the following options:
radius
Set frame radius. Default is 2. Allowed range is from 1 to 63. For example radius of 3 will instruct filter to calculate average of 7 frames.
factor
Set factor to amplify difference. Default is 2. Allowed range is from 0 to 65535.
threshold
Set threshold for difference amplification. Any difference greater or equal to this value will not alter source pixel. Default is 10. Allowed range is from 0 to 65535.
tolerance
Set tolerance for difference amplification. Any difference lower to this value will not alter source pixel. Default is 0. Allowed range is from 0 to 65535.
low |
Set lower limit for changing source pixel. Default is 65535. Allowed range is from 0 to 65535. This option controls maximum possible value that will decrease source pixel value. |
high
Set high limit for changing source pixel. Default is 65535. Allowed range is from 0 to 65535. This option controls maximum possible value that will increase source pixel value.
planes
Set which planes to filter. Default is all. Allowed range is from 0 to 15.
Commands
This filter
supports the following commands that corresponds to
option of same name:
factor
threshold
tolerance
low |
high
planes
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 9. Must be odd number in range [5, 129]. | ||
p |
Set what planes of frame filter will use for averaging. Default is all. | ||
a |
Set what variant of algorithm filter will use for averaging. Default is "p" parallel. Alternatively can be set to "s" serial. |
Parallel can be faster then serial, while other way around is never true. Parallel will abort early on first change being greater then thresholds, while serial will continue processing other side of frames if they are equal or below thresholds.
0s |
|||
1s |
|||
2s |
Set sigma for 1st plane, 2nd plane or 3rd plane. Default is 32767. Valid range is from 0 to 32767. This options controls weight for each pixel in radius defined by size. Default value means every pixel have same weight. Setting this option to 0 effectively disables filtering. |
Commands
This filter supports same commands as options except option "s". The command accepts the same syntax of the corresponding option.
avgblur
Apply average blur filter.
The filter
accepts the following options:
sizeX
Set horizontal radius size.
planes
Set which planes to filter. By default all planes are filtered.
sizeY
Set vertical radius size, if zero it will be same as "sizeX". Default is 0.
Commands
This filter supports same commands as options. The command accepts the same syntax of the corresponding option.
If the specified expression is not valid, it is kept at its current value.
backgroundkey
Turns a static background into transparency.
The filter
accepts the following option:
threshold
Threshold for scene change detection.
similarity
Similarity percentage with the background.
blend
Set the blend amount for pixels that are not similar.
Commands
This filter supports the all above options as commands.
bbox
Compute the bounding box for the non-black pixels in the
input frame luma plane.
This filter computes the bounding box containing all the pixels with a luma 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 luma value. Default is 16.
Commands
This filter supports the all above options as commands.
bilateral
Apply bilateral filter, spatial smoothing while preserving
edges.
The filter
accepts the following options:
sigmaS
Set sigma of gaussian function to calculate spatial weight. Allowed range is 0 to 512. Default is 0.1.
sigmaR
Set sigma of gaussian function to calculate range weight. Allowed range is 0 to 1. Default is 0.1.
planes
Set planes to filter. Default is first only.
Commands
This filter supports the all above options as commands.
bilateral_cuda
CUDA accelerated bilateral filter, an edge preserving
filter. This filter is mathematically accurate thanks to the
use of GPU acceleration. For best output quality, use one to
one chroma subsampling, i.e. yuv444p format.
The filter
accepts the following options:
sigmaS
Set sigma of gaussian function to calculate spatial weight, also called sigma space. Allowed range is 0.1 to 512. Default is 0.1.
sigmaR
Set sigma of gaussian function to calculate color range weight, also called sigma color. Allowed range is 0.1 to 512. Default is 0.1.
window_size
Set window size of the bilateral function to determine the number of neighbours to loop on. If the number entered is even, one will be added automatically. Allowed range is 1 to 255. Default is 1.
Examples
• |
Apply the bilateral filter on a video. |
./ffmpeg -v
verbose \
-hwaccel cuda -hwaccel_output_format cuda -i input.mp4 \
-init_hw_device cuda \
-filter_complex \
" \
[0:v]scale_cuda=format=yuv444p[scaled_video];
[scaled_video]bilateral_cuda=window_size=9:sigmaS=3.0:sigmaR=50.0"
\
-an -sn -c:v h264_nvenc -cq 20 out.mp4
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.
The filter outputs its detection analysis to both the log as well as frame metadata. If a black segment of at least the specified minimum duration is found, a line with the start and end timestamps as well as duration is printed to the log with level "info". In addition, a log line with level "debug" is printed per frame showing the black amount detected for that frame.
The filter also attaches metadata to the first frame of a black segment with key "lavfi.black_start" and to the first frame after the black segment ends with key "lavfi.black_end". The value is the frame’s timestamp. This metadata is added regardless of the minimum duration specified.
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 luma value for which a pixel is considered "black". The provided value is scaled according to the following equation:
<absolute_threshold> = <luma_minimum_value> + <pixel_black_th> * <luma_range_size>
luma_range_size and luma_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.
This filter exports frame metadata "lavfi.blackframe.pblack". The value represents the percentage of pixels in the picture that are below the threshold 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
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
and |
average
bleach
burn
darken
difference
divide
dodge
exclusion
extremity
freeze
geometric
glow
grainextract
grainmerge
hardlight
hardmix
hardoverlay
harmonic
heat
interpolate
lighten
linearlight
multiply
multiply128
negation
normal
or |
overlay
phoenix
pinlight
reflect
screen
softdifference
softlight
stain
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 for the plane being filtered. It is the ratio between the dimensions of the current plane to the luma plane, e.g. for a "yuv420p" frame, the values are "1,1" for the luma plane and "0.5,0.5" for the 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).
The "blend" filter also supports the framesync options.
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 linear horizontal transition from top layer to bottom layer: |
blend=all_expr='A*(X/W)+B*(1-X/W)'
• |
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=grainextract
Commands
This filter supports same commands as options.
blockdetect
Determines blockiness of frames without altering the input
frames.
Based on Remco Muijs and Ihor Kirenko: "A no-reference blocking artifact measure for adaptive video processing." 2005 13th European signal processing conference.
The filter
accepts the following options:
period_min
period_max
Set minimum and maximum values for determining pixel grids (periods). Default values are [3,24].
planes
Set planes to filter. Default is first only.
Examples
• |
Determine blockiness for the first plane and search for periods within [8,32]: |
blockdetect=period_min=8:period_max=32:planes=1
blurdetect
Determines blurriness of frames without altering the input
frames.
Based on Marziliano, Pina, et al. "A no-reference perceptual blur metric." Allows for a block-based abbreviation.
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".
radius
Define the radius to search around an edge pixel for local maxima.
block_pct
Determine blurriness only for the most significant blocks, given in percentage.
block_width
Determine blurriness for blocks of width block_width. If set to any value smaller 1, no blocks are used and the whole image is processed as one no matter of block_height.
block_height
Determine blurriness for blocks of height block_height. If set to any value smaller 1, no blocks are used and the whole image is processed as one no matter of block_width.
planes
Set planes to filter. Default is first only.
Examples
• |
Determine blur for 80% of most significant 32x32 blocks: |
blurdetect=block_width=32:block_height=32:block_pct=80
bm3d
Denoise frames using Block-Matching 3D algorithm.
The filter
accepts the following options.
sigma
Set denoising strength. Default value is 1. Allowed range is from 0 to 999.9. The denoising algorithm is very sensitive to sigma, so adjust it according to the source.
block
Set local patch size. This sets dimensions in 2D.
bstep
Set sliding step for processing blocks. Default value is 4. Allowed range is from 1 to 64. Smaller values allows processing more reference blocks and is slower.
group
Set maximal number of similar blocks for 3rd dimension. Default value is 1. When set to 1, no block matching is done. Larger values allows more blocks in single group. Allowed range is from 1 to 256.
range
Set radius for search block matching. Default is 9. Allowed range is from 1 to INT32_MAX.
mstep
Set step between two search locations for block matching. Default is 1. Allowed range is from 1 to 64. Smaller is slower.
thmse
Set threshold of mean square error for block matching. Valid range is 0 to INT32_MAX.
hdthr
Set thresholding parameter for hard thresholding in 3D transformed domain. Larger values results in stronger hard-thresholding filtering in frequency domain.
estim
Set filtering estimation mode. Can be "basic" or "final". Default is "basic".
ref |
If enabled, filter will use 2nd stream for block matching. Default is disabled for "basic" value of estim option, and always enabled if value of estim is "final". |
planes
Set planes to filter. Default is all available except alpha.
Examples
• |
Basic filtering with bm3d: |
bm3d=sigma=3:block=4:bstep=2:group=1:estim=basic
• |
Same as above, but filtering only luma: |
bm3d=sigma=3:block=4:bstep=2:group=1:estim=basic:planes=1
• |
Same as above, but with both estimation modes: |
split[a][b],[a]bm3d=sigma=3:block=4:bstep=2:group=1:estim=basic[a],[b][a]bm3d=sigma=3:block=4:bstep=2:group=16:estim=final:ref=1
• |
Same as above, but prefilter with nlmeans filter instead: |
split[a][b],[a]nlmeans=s=3:r=7:p=3[a],[b][a]bm3d=sigma=3:block=4:bstep=2:group=16:estim=final:ref=1
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. Accepts one of the following values:
0, all
Deinterlace all frames.
1, interlaced
Only deinterlace frames marked as interlaced.
The default value is "all".
bwdif_cuda
Deinterlace the input video using the bwdif
algorithm, but implemented in CUDA so that it can work as
part of a GPU accelerated pipeline with nvdec and/or
nvenc.
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. Accepts one of the following values:
0, all
Deinterlace all frames.
1, interlaced
Only deinterlace frames marked as interlaced.
The default value is "all".
ccrepack
Repack CEA-708 closed captioning side data
This filter fixes various issues seen with commerical encoders related to upstream malformed CEA-708 payloads, specifically incorrect number of tuples (wrong cc_count for the target FPS), and incorrect ordering of tuples (i.e. the CEA-608 tuples are not at the first entries in the payload).
cas
Apply Contrast Adaptive Sharpen filter to video stream.
The filter
accepts the following options:
strength
Set the sharpening strength. Default value is 0.
planes
Set planes to filter. Default value is to filter all planes except alpha plane.
Commands
This filter supports same commands as options.
chromahold
Remove all color information for all colors except for
certain one.
The filter
accepts the following options:
color
The color which will not be replaced with neutral chroma.
similarity
Similarity percentage with the above color. 0.01 matches only the exact key color, while 1.0 matches everything.
blend
Blend percentage. 0.0 makes pixels either fully gray, or not gray at all. Higher values result in more preserved color.
yuv |
Signals that the color passed is already in YUV instead of RGB. |
Literal 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.
Commands
This filter supports same commands as options. The command accepts the same syntax of the corresponding option.
If the specified expression is not valid, it is kept at its current value.
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. |
Literal 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.
Commands
This filter supports same commands as options. 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
• |
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
chromakey_cuda
CUDA accelerated YUV colorspace color/chroma keying.
This filter works like normal chromakey filter but operates on CUDA frames. for more details and parameters see chromakey.
Examples
• |
Make all the green pixels in the input video transparent and use it as an overlay for another video: |
./ffmpeg \
-hwaccel cuda -hwaccel_output_format cuda -i input_green.mp4
\
-hwaccel cuda -hwaccel_output_format cuda -i base_video.mp4
\
-init_hw_device cuda \
-filter_complex \
" \
[0:v]chromakey_cuda=0x25302D:0.1:0.12:1[overlay_video]; \
[1:v]scale_cuda=format=yuv420p[base]; \
[base][overlay_video]overlay_cuda" \
-an -sn -c:v h264_nvenc -cq 20 output.mp4
• |
Process two software sources, explicitly uploading the frames: |
./ffmpeg
-init_hw_device cuda=cuda -filter_hw_device cuda \
-f lavfi -i color=size=800x600:color=white,format=yuv420p \
-f lavfi -i yuvtestsrc=size=200x200,format=yuv420p \
-filter_complex \
" \
[0]hwupload[under]; \
[1]hwupload,chromakey_cuda=green:0.1:0.12[over]; \
[under][over]overlay_cuda" \
-c:v hevc_nvenc -cq 18 -preset slow output.mp4
chromanr
Reduce chrominance noise.
The filter
accepts the following options:
thres
Set threshold for averaging chrominance values. Sum of absolute difference of Y, U and V pixel components of current pixel and neighbour pixels lower than this threshold will be used in averaging. Luma component is left unchanged and is copied to output. Default value is 30. Allowed range is from 1 to 200.
sizew
Set horizontal radius of rectangle used for averaging. Allowed range is from 1 to 100. Default value is 5.
sizeh
Set vertical radius of rectangle used for averaging. Allowed range is from 1 to 100. Default value is 5.
stepw
Set horizontal step when averaging. Default value is 1. Allowed range is from 1 to 50. Mostly useful to speed-up filtering.
steph
Set vertical step when averaging. Default value is 1. Allowed range is from 1 to 50. Mostly useful to speed-up filtering.
threy
Set Y threshold for averaging chrominance values. Set finer control for max allowed difference between Y components of current pixel and neigbour pixels. Default value is 200. Allowed range is from 1 to 200.
threu
Set U threshold for averaging chrominance values. Set finer control for max allowed difference between U components of current pixel and neigbour pixels. Default value is 200. Allowed range is from 1 to 200.
threv
Set V threshold for averaging chrominance values. Set finer control for max allowed difference between V components of current pixel and neigbour pixels. Default value is 200. Allowed range is from 1 to 200.
distance
Set distance type used in
calculations.
manhattan
Absolute difference.
euclidean
Difference squared.
Default distance type is manhattan.
Commands
This filter supports same commands as options. The command accepts the same syntax of the corresponding option.
chromashift
Shift chroma pixels horizontally and/or vertically.
The filter accepts the following options:
cbh |
Set amount to shift chroma-blue horizontally. |
|||
cbv |
Set amount to shift chroma-blue vertically. |
|||
crh |
Set amount to shift chroma-red horizontally. |
|||
crv |
Set amount to shift chroma-red vertically. |
edge
Set edge mode, can be smear, default, or warp.
Commands
This filter supports the all above options as commands.
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
dcip3
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.
fill
Fill with CIE colors. By default is enabled.
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:
block
Display block partition structure using the luma plane.
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.
pl |
Preserve lightness when changing color balance. Default is disabled. |
Examples
• |
Add red color cast to shadows: |
colorbalance=rs=.3
Commands
This filter supports the all above options as commands.
colorcontrast
Adjust color contrast between RGB components.
The filter accepts the following options:
rc |
Set the red-cyan contrast. Defaults is 0.0. Allowed range is from -1.0 to 1.0. | ||
gm |
Set the green-magenta contrast. Defaults is 0.0. Allowed range is from -1.0 to 1.0. | ||
by |
Set the blue-yellow contrast. Defaults is 0.0. Allowed range is from -1.0 to 1.0. | ||
rcw |
|||
gmw |
|||
byw |
Set the weight of each "rc", "gm", "by" option value. Default value is 0.0. Allowed range is from 0.0 to 1.0. If all weights are 0.0 filtering is disabled. | ||
pl |
Set the amount of preserving lightness. Default value is 0.0. Allowed range is from 0.0 to 1.0. |
Commands
This filter supports the all above options as commands.
colorcorrect
Adjust color white balance selectively for blacks and
whites. This filter operates in YUV colorspace.
The filter accepts the following options:
rl |
Set the red shadow spot. Allowed range is from -1.0 to 1.0. Default value is 0. | ||
bl |
Set the blue shadow spot. Allowed range is from -1.0 to 1.0. Default value is 0. | ||
rh |
Set the red highlight spot. Allowed range is from -1.0 to 1.0. Default value is 0. | ||
bh |
Set the blue highlight spot. Allowed range is from -1.0 to 1.0. Default value is 0. |
saturation
Set the amount of saturation. Allowed range is from -3.0 to 3.0. Default value is 1.
analyze
If set to anything other than "manual" it will analyze every frame and use derived parameters for filtering output frame.
Possible values
are:
manual
average
minmax
median
Default value is "manual".
Commands
This filter supports the all above options as commands.
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]".
pc |
Set preserve color mode. The accepted values are: |
none
Disable color preserving, this is default.
lum |
Preserve luminance. |
|||
max |
Preserve max value of RGB triplet. |
|||
avg |
Preserve average value of RGB triplet. |
|||
sum |
Preserve sum value of RGB triplet. |
|||
nrm |
Preserve normalized value of RGB triplet. |
|||
pwr |
Preserve power value of RGB triplet. |
|||
pa |
Set the preserve color amount when changing colors. Allowed range is from "[0.0, 1.0]". Default is 0.0, thus disabled.
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
Commands
This filter supports the all above options as commands.
colorize
Overlay a solid color on the video stream.
The filter accepts the following options:
hue |
Set the color hue. Allowed range is from 0 to 360. Default value is 0. |
saturation
Set the color saturation. Allowed range is from 0 to 1. Default value is 0.5.
lightness
Set the color lightness. Allowed range is from 0 to 1. Default value is 0.5.
mix |
Set the mix of source lightness. By default is set to 1.0. Allowed range is from 0.0 to 1.0. |
Commands
This filter supports the all above options as commands.
colorkey
RGB colorspace color keying. This filter operates on 8-bit
RGB format frames by setting the alpha component of each
pixel which falls within the similarity radius of the key
color to 0. The alpha value for pixels outside the
similarity radius depends on the value of the blend
option.
The filter
accepts the following options:
color
Set the color for which alpha will be set to 0 (full transparency). See "Color" section in the ffmpeg-utils manual. Default is "black".
similarity
Set the radius from the key color within which other colors also have full transparency. The computed distance is related to the unit fractional distance in 3D space between the RGB values of the key color and the pixel’s color. Range is 0.01 to 1.0. 0.01 matches within a very small radius around the exact key color, while 1.0 matches everything. Default is 0.01.
blend
Set how the alpha value for pixels that fall outside the similarity radius is computed. 0.0 makes pixels either fully transparent or fully opaque. Higher values result in semi-transparent pixels, with greater transparency the more similar the pixel color is to the key color. Range is 0.0 to 1.0. Default is 0.0.
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
Commands
This filter supports same commands as options. The command accepts the same syntax of the corresponding option.
If the specified expression is not valid, it is kept at its current value.
colorhold
Remove all color information for all RGB colors except for
certain one.
The filter
accepts the following options:
color
The color which will not be replaced with neutral gray.
similarity
Similarity percentage with the above color. 0.01 matches only the exact key color, while 1.0 matches everything.
blend
Blend percentage. 0.0 makes pixels fully gray. Higher values result in more preserved color.
Commands
This filter supports same commands as options. The command accepts the same syntax of the corresponding option.
If the specified expression is not valid, it is kept at its current value.
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.
preserve
Set preserve color mode. The
accepted values are:
none
Disable color preserving, this is default.
lum |
Preserve luminance. |
|||
max |
Preserve max value of RGB triplet. |
|||
avg |
Preserve average value of RGB triplet. |
|||
sum |
Preserve sum value of RGB triplet. |
|||
nrm |
Preserve normalized value of RGB triplet. |
|||
pwr |
Preserve power value of RGB triplet. |
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
Commands
This filter supports the all above options as commands.
colormap
Apply custom color maps to video stream.
This filter needs three input video streams. First stream is video stream that is going to be filtered out. Second and third video stream specify color patches for source color to target color mapping.
The filter
accepts the following options:
patch_size
Set the source and target video stream patch size in pixels.
nb_patches
Set the max number of used patches from source and target video stream. Default value is number of patches available in additional video streams. Max allowed number of patches is 64.
type
Set the adjustments used for target colors. Can be "relative" or "absolute". Defaults is "absolute".
kernel
Set the kernel used to measure color differences between mapped colors.
The accepted
values are:
euclidean
weuclidean
Default is "euclidean".
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
fcc |
FCC |
bt601
BT.601
bt470
BT.470
bt470bg
BT.470BG
smpte170m
SMPTE-170M
smpte240m
SMPTE-240M
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. Input video needs to have an even size.
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
ycgco
YCgCo
bt2020ncl
BT.2020 with non-constant luminance
trc |
Specify output transfer characteristics. |
The accepted
values are:
bt709
BT.709
bt470m
BT.470M
bt470bg
BT.470BG
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
srgb
SRGB
iec61966-2-1
iec61966-2-1
iec61966-2-4
iec61966-2-4
xvycc
xvycc
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
film
film
smpte431
SMPTE-431
smpte432
SMPTE-432
bt2020
BT.2020
jedec-p22
JEDEC P22 phosphors
range
Specify output color range.
The accepted values are:
tv |
TV (restricted) range |
mpeg
MPEG (restricted) range
pc |
PC (full) 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
colorspace_cuda
CUDA accelerated implementation of the colorspace
filter.
It is by no means feature complete compared to the software colorspace filter, and at the current time only supports color range conversion between jpeg/full and mpeg/limited range.
The filter
accepts the following options:
range
Specify output color range.
The accepted values are:
tv |
TV (restricted) range |
mpeg
MPEG (restricted) range
pc |
PC (full) range |
jpeg
JPEG (full) range
colortemperature
Adjust color temperature in video to simulate variations in
ambient color temperature.
The filter
accepts the following options:
temperature
Set the temperature in Kelvin. Allowed range is from 1000 to 40000. Default value is 6500 K.
mix |
Set mixing with filtered output. Allowed range is from 0 to 1. Default value is 1. | ||
pl |
Set the amount of preserving lightness. Allowed range is from 0 to 1. Default value is 0. |
Commands
This filter supports same commands as options.
convolution
Apply convolution of 3x3, 5x5, 7x7 or horizontal/vertical up
to 49 elements.
The filter accepts the following options:
0m |
|||
1m |
|||
2m |
|||
3m |
Set matrix for each plane. Matrix is sequence of 9, 25 or 49 signed integers in square mode, and from 1 to 49 odd number of signed integers in row mode. |
0rdiv
1rdiv
2rdiv
3rdiv
Set multiplier for calculated value for each plane. If unset or 0, it will be sum of all matrix elements.
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.
0mode
1mode
2mode
3mode
Set matrix mode for each plane. Can be square, row or column. Default is square.
Commands
This filter supports the all above options as commands.
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 laplacian edge detector which includes diagonals: |
convolution="1 1 1 1 -8 1 1 1 1:1 1 1 1 -8 1 1 1 1:1 1 1 1 -8 1 1 1 1:1 1 1 1 -8 1 1 1 1:5:5:5:1:0:128:128:0"
• |
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"
convolve
Apply 2D convolution of video stream in frequency domain
using second stream as impulse.
The filter
accepts the following options:
planes
Set which planes to process.
impulse
Set which impulse video frames will be processed, can be first or all. Default is all.
The "convolve" filter also supports the framesync options.
copy
Copy the input video 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
corr
Obtain the correlation between two input videos.
This filter takes two input videos.
Both input videos 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 per component, average, min and max correlation is printed through the logging system.
The filter stores the calculated correlation of each frame in frame metadata.
This filter also supports the framesync options.
In the below example the input file main.mpg being processed is compared with the reference file ref.mpg.
ffmpeg -i main.mpg -i ref.mpg -lavfi corr -f null -
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
• |
Cover a rectangular object by the supplied image of a given video using ffmpeg: |
ffmpeg -i file.ts -vf find_rect=newref.pgm,cover_rect=cover.jpg:mode=cover new.mkv
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; deprecated, do not use | ||
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 or video area of the input video according to mode.
It accepts the
following parameters:
mode
Depending on mode crop
detection is based on either the mere black value of
surrounding pixels or a combination of motion vectors and
edge pixels.
black
Detect black pixels surrounding the playing video. For fine control use option limit.
mvedges
Detect the playing video by the motion vectors inside the video and scanning for edge pixels typically forming the border of a playing video.
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.
skip
Set the number of initial frames for which evaluation is skipped. Default is 2. Range is 0 to INT_MAX.
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.
mv_threshold
Set motion in pixel units as threshold for motion detection. It defaults to 8.
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 "5/255", and default value for high is "15/255".
Examples
• |
Find video area surrounded by black borders: |
ffmpeg -i file.mp4 -vf cropdetect,metadata=mode=print -f null -
• |
Find an embedded video area, generate motion vectors beforehand: |
ffmpeg -i file.mp4 -vf mestimate,cropdetect=mode=mvedges,metadata=mode=print -f null -
• |
Find an embedded video area, use motion vectors from decoder: |
ffmpeg -flags2 +export_mvs -i file.mp4 -vf cropdetect=mode=mvedges,metadata=mode=print -f null -
Commands
This filter
supports the following commands:
limit
The command accepts the same syntax of the corresponding option. If the specified expression is not valid, it is kept at its current value.
cue
Delay video filtering until a given wallclock timestamp. The
filter first passes on preroll amount of frames, then
it buffers at most buffer amount of frames and waits
for the cue. After reaching the cue it forwards the buffered
frames and also any subsequent frames coming in its
input.
The filter can be used synchronize the output of multiple ffmpeg processes for realtime output devices like decklink. By putting the delay in the filtering chain and pre-buffering frames the process can pass on data to output almost immediately after the target wallclock timestamp is reached.
Perfect frame accuracy cannot be guaranteed, but the result is good enough for some use cases.
cue |
The cue timestamp expressed in a UNIX timestamp in microseconds. Default is 0. |
preroll
The duration of content to pass on as preroll expressed in seconds. Default is 0.
buffer
The maximum duration of content to buffer before waiting for the cue expressed in seconds. Default is 0.
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 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. The curve is formed by using a natural or monotonic cubic spline interpolation, depending on the interp option (default: "natural"). The "natural" spline produces a smoother curve in general while the monotonic ("pchip") spline guarantees the transitions between the specified points to be monotonic. 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.
interp
Specify the kind of
interpolation. Available algorithms are:
natural
Natural cubic spline using a piece-wise cubic polynomial that is twice continuously differentiable.
pchip
Monotonic cubic spline using a piecewise cubic Hermite interpolating polynomial (PCHIP).
To avoid some filtergraph syntax conflicts, each key points list need to be defined using the following syntax: "x0/y0 x1/y1 x2/y2 ...".
Commands
This filter supports same commands as options.
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.
format
Set display number format. Can be "hex", or "dec". Default is "hex".
components
Set pixel components to display. By default all pixel components are displayed.
Commands
This filter supports same commands as options excluding "size" option.
dblur
Apply Directional blur filter.
The filter
accepts the following options:
angle
Set angle of directional blur. Default is 45.
radius
Set radius of directional blur. Default is 5.
planes
Set which planes to filter. By default all planes are filtered.
Commands
This filter supports same commands as options. The command accepts the same syntax of the corresponding option.
If the specified expression is not valid, it is kept at its current value.
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, b
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.
coupling, c
If enabled, current pixel is changed if and only if all pixel components are banded, e.g. banding detection threshold is triggered for all color components. The default is disabled.
Commands
This filter supports the all above options as commands.
deblock
Remove blocking artifacts from input video.
The filter
accepts the following options:
filter
Set filter type, can be weak or strong. Default is strong. This controls what kind of deblocking is applied.
block
Set size of block, allowed range is from 4 to 512. Default is 8.
alpha
beta
gamma
delta
Set blocking detection thresholds. Allowed range is 0 to 1. Defaults are: 0.098 for alpha and 0.05 for the rest. Using higher threshold gives more deblocking strength. Setting alpha controls threshold detection at exact edge of block. Remaining options controls threshold detection near the edge. Each one for below/above or left/right. Setting any of those to 0 disables deblocking.
planes
Set planes to filter. Default is to filter all available planes.
Examples
• |
Deblock using weak filter and block size of 4 pixels. |
deblock=filter=weak:block=4
• |
Deblock using strong filter, block size of 4 pixels and custom thresholds for deblocking more edges. |
deblock=filter=strong:block=4:alpha=0.12:beta=0.07:gamma=0.06:delta=0.05
• |
Similar as above, but filter only first plane. |
deblock=filter=strong:block=4:alpha=0.12:beta=0.07:gamma=0.06:delta=0.05:planes=1
• |
Similar as above, but filter only second and third plane. |
deblock=filter=strong:block=4:alpha=0.12:beta=0.07:gamma=0.06:delta=0.05:planes=6
Commands
This filter supports the all above options as commands.
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.
mixed
Set whether or not the input only partially contains content to be decimated. Default is "false". If enabled video output stream will be in variable frame rate.
deconvolve
Apply 2D deconvolution of video stream in frequency domain
using second stream as impulse.
The filter
accepts the following options:
planes
Set which planes to process.
impulse
Set which impulse video frames will be processed, can be first or all. Default is all.
noise
Set noise when doing divisions. Default is 0.0000001. Useful when width and height are not same and not power of 2 or if stream prior to convolving had noise.
The "deconvolve" filter also supports the framesync options.
dedot
Reduce cross-luminance (dot-crawl) and cross-color
(rainbows) from video.
It accepts the following options:
m |
Set mode of operation. Can be combination of dotcrawl for cross-luminance reduction and/or rainbows for cross-color reduction. | ||
lt |
Set spatial luma threshold. Lower values increases reduction of cross-luminance. | ||
tl |
Set tolerance for temporal luma. Higher values increases reduction of cross-luminance. | ||
tc |
Set tolerance for chroma temporal variation. Higher values increases reduction of cross-color. | ||
ct |
Set temporal chroma threshold. Lower values increases reduction of cross-color. |
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.
Commands
This filter supports the all above options as commands.
deflicker
Remove temporal frame luminance variations.
It accepts the
following options:
size, s
Set moving-average filter size in frames. Default is 5. Allowed range is 2 - 129.
mode, m
Set averaging mode to smooth temporal luminance variations.
Available values are:
am |
Arithmetic mean |
|||
gm |
Geometric mean |
|||
hm |
Harmonic mean |
|||
qm |
Quadratic mean |
|||
cm |
Cubic mean |
|||
pm |
Power mean |
median
Median
bypass
Do not actually modify frame. Useful when one only wants metadata.
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. |
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: |
delogo=x=0:y=0:w=100:h=77
derain
Remove the rain in the input image/video by applying the
derain methods based on convolutional neural networks.
Supported models:
• |
Recurrent Squeeze-and-Excitation Context Aggregation Net (RESCAN). See <http://openaccess.thecvf.com/content_ECCV_2018/papers/Xia_Li_Recurrent_Squeeze-and-Excitation_Context_ECCV_2018_paper.pdf>. |
Training as well as model generation scripts are provided in the repository at <https://github.com/XueweiMeng/derain_filter.git>.
The filter
accepts the following options:
filter_type
Specify which filter to use.
This option accepts the following values:
derain
Derain filter. To conduct derain filter, you need to use a derain model.
dehaze
Dehaze filter. To conduct dehaze filter, you need to use a dehaze model.
Default value is derain.
dnn_backend
Specify which DNN backend to
use for model loading and execution. This option accepts the
following values:
tensorflow
TensorFlow backend. To enable this backend you need to install the TensorFlow for C library (see <https://www.tensorflow.org/install/lang_c>) and configure FFmpeg with "--enable-libtensorflow"
model
Set path to model file specifying network architecture and its parameters. Note that different backends use different file formats. TensorFlow can load files for only its format.
To get full functionality (such as async execution), please use the dnn_processing filter.
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.
despill
Remove unwanted contamination of foreground colors, caused
by reflected color of greenscreen or bluescreen.
This filter
accepts the following options:
type
Set what type of despill to use.
mix |
Set how spillmap will be generated. |
expand
Set how much to get rid of still remaining spill.
red |
Controls amount of red in spill area. |
green
Controls amount of green in spill area. Should be -1 for greenscreen.
blue
Controls amount of blue in spill area. Should be -1 for bluescreen.
brightness
Controls brightness of spill area, preserving colors.
alpha
Modify alpha from generated spillmap.
Commands
This filter supports the all above options as commands.
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
Commands
This filter supports the all above options as commands.
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.
mirror
Out of range pixels will be replaced with mirrored pixels.
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
dnn_classify
Do classification with deep neural networks based on
bounding boxes.
The filter
accepts the following options:
dnn_backend
Specify which DNN backend to use for model loading and execution. This option accepts only openvino now, tensorflow backends will be added.
model
Set path to model file specifying network architecture and its parameters. Note that different backends use different file formats.
input
Set the input name of the dnn network.
output
Set the output name of the dnn network.
confidence
Set the confidence threshold (default: 0.5).
labels
Set path to label file specifying the mapping between label id and name. Each label name is written in one line, tailing spaces and empty lines are skipped. The first line is the name of label id 0, and the second line is the name of label id 1, etc. The label id is considered as name if the label file is not provided.
backend_configs
Set the configs to be passed into backend
For tensorflow backend, you can set its configs with sess_config options, please use tools/python/tf_sess_config.py to get the configs for your system.
dnn_detect
Do object detection with deep neural networks.
The filter
accepts the following options:
dnn_backend
Specify which DNN backend to use for model loading and execution. This option accepts only openvino now, tensorflow backends will be added.
model
Set path to model file specifying network architecture and its parameters. Note that different backends use different file formats.
input
Set the input name of the dnn network.
output
Set the output name of the dnn network.
confidence
Set the confidence threshold (default: 0.5).
labels
Set path to label file specifying the mapping between label id and name. Each label name is written in one line, tailing spaces and empty lines are skipped. The first line is the name of label id 0 (usually it is ’background’), and the second line is the name of label id 1, etc. The label id is considered as name if the label file is not provided.
backend_configs
Set the configs to be passed into backend. To use async execution, set async (default: set). Roll back to sync execution if the backend does not support async.
dnn_processing
Do image processing with deep neural networks. It works
together with another filter which converts the pixel format
of the Frame to what the dnn network requires.
The filter
accepts the following options:
dnn_backend
Specify which DNN backend to
use for model loading and execution. This option accepts the
following values:
tensorflow
TensorFlow backend. To enable this backend you need to install the TensorFlow for C library (see <https://www.tensorflow.org/install/lang_c>) and configure FFmpeg with "--enable-libtensorflow"
openvino
OpenVINO backend. To enable this backend you need to build and install the OpenVINO for C library (see <https://github.com/openvinotoolkit/openvino/blob/master/build-instruction.md>) and configure FFmpeg with "--enable-libopenvino" (--extra-cflags=-I... --extra-ldflags=-L... might be needed if the header files and libraries are not installed into system path)
model
Set path to model file specifying network architecture and its parameters. Note that different backends use different file formats. TensorFlow, OpenVINO backend can load files for only its format.
input
Set the input name of the dnn network.
output
Set the output name of the dnn network.
backend_configs
Set the configs to be passed into backend. To use async execution, set async (default: set). Roll back to sync execution if the backend does not support async.
For tensorflow backend, you can set its configs with sess_config options, please use tools/python/tf_sess_config.py to get the configs of TensorFlow backend for your system.
Examples
• |
Remove rain in rgb24 frame with can.pb (see derain filter): |
./ffmpeg -i rain.jpg -vf format=rgb24,dnn_processing=dnn_backend=tensorflow:model=can.pb:input=x:output=y derain.jpg
• |
Handle the Y channel with srcnn.pb (see sr filter) for frame with yuv420p (planar YUV formats supported): |
./ffmpeg -i 480p.jpg -vf format=yuv420p,scale=w=iw*2:h=ih*2,dnn_processing=dnn_backend=tensorflow:model=srcnn.pb:input=x:output=y -y srcnn.jpg
• |
Handle the Y channel with espcn.pb (see sr filter), which changes frame size, for format yuv420p (planar YUV formats supported), please use tools/python/tf_sess_config.py to get the configs of TensorFlow backend for your system. |
./ffmpeg -i 480p.jpg -vf format=yuv420p,dnn_processing=dnn_backend=tensorflow:model=espcn.pb:input=x:output=y:backend_configs=sess_config=0x10022805320e09cdccccccccccec3f20012a01303801 -y tmp.espcn.jpg
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. A value of "fill" will create a filled box. Default value is 3.
See below for the list of accepted constants.
replace
Applicable if the input has alpha. With value 1, the pixels of the painted box will overwrite the video’s color and alpha pixels. Default is 0, which composites the box onto the input, leaving the video’s alpha intact.
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. |
box_source
Box source can be set as side_data_detection_bboxes if you want to use box data in detection bboxes of side data.
If box_source is set, the x, y, width and height will be ignored and still use box data in detection bboxes of side data. So please do not use this parameter if you were not sure about the box source.
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:red [AT] 0.5
The previous example can be specified as:
drawbox=x=10:y=20:w=200:h=60:color=red [AT] 0.5
• |
Fill the box with pink color: |
drawbox=x=10:y=10:w=100:h=100:color=pink [AT] 0.5:t=fill
• |
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
Commands
This filter supports same commands as options. The command accepts the same syntax of the corresponding option.
If the specified expression is not valid, it is kept at its current value.
drawgraph
Draw a graph using input video 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".
rate, r
Set the output frame rate. Default value is 25.
The foreground color expressions can use the following variables:
MIN |
Minimal value of metadata value. |
|||
MAX |
Maximal value of metadata value. |
|||
VAL |
Current metadata key value. |
The color is defined as 0xAABBGGRR.
Example using metadata from signalstats filter:
signalstats,drawgraph=lavfi.signalstats.YAVG:min=0:max=255
Example using metadata from ebur128 filter:
ebur128=metadata=1,adrawgraph=lavfi.r128.M:min=-120:max=5
drawgrid
Draw a grid on the input image.
It accepts the following parameters:
x |
|||
y |
The expressions which specify the coordinates of some point of grid intersection (meant to configure offset). Both default to 0. |
width, w
height, h
The expressions which specify the width and height of the grid cell, if 0 they are interpreted as the input width and height, respectively, minus "thickness", so image gets framed. Default to 0.
color, c
Specify the color of the grid. For the general syntax of this option, check the "Color" section in the ffmpeg-utils manual. If the special value "invert" is used, the grid color is the same as the video with inverted luma.
thickness, t
The expression which sets the thickness of the grid line. Default value is 1.
See below for the list of accepted constants.
replace
Applicable if the input has alpha. With 1 the pixels of the painted grid will overwrite the video’s color and alpha pixels. Default is 0, which composites the grid onto the input, leaving the video’s alpha intact.
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=red [AT] 0.5
• |
Draw a white 3x3 grid with an opacity of 50%: |
drawgrid=w=iw/3:h=ih/3:t=2:c=white [AT] 0.5
Commands
This filter supports same commands as options. The command accepts the same syntax of the corresponding option.
If the specified expression is not valid, it is kept at its current value.
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" and "--enable-libharfbuzz". 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 value
must be specified using one of the following formats:
*<"boxborderw=10" set the width of all the
borders to 10>
*<"boxborderw=10|20" set the width of the top
and bottom borders to
10>
and the width of the left and right borders to 20
*<"boxborderw=10|20|30"
set the width of the top border to 10, the
width>
of the bottom border to 30 and the width of the left and right borders to 20
*<"boxborderw=10|20|30|40"
set the borders width to 10 (top), 20
(right),>
30 (bottom), 40 (left)
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".
line_spacing
Set the line spacing in pixels. The default value of line_spacing is 0.
text_align
Set the vertical and horizontal alignment of the text with respect to the box boundaries. The value is combination of flags, one for the vertical alignment (T=top, M=middle, B=bottom) and one for the horizontal alignment (L=left, C=center, R=right). Please note that tab characters are only supported with the left horizontal alignment.
y_align
Specify what the y value
is referred to. Possible values are:
*<"text" the top of the highest glyph of the
first text line is
placed at y>
*<"baseline" the baseline of the first text
line is placed at y>
*<"font" the baseline of the first text line is
placed at y plus
the>
ascent (in pixels) defined in the font metrics
The default value of y_align is "text" for backward compatibility.
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.
basetime
Set a start time for the count. Value is in microseconds. Only applied in the deprecated "strftime" expansion mode. To emulate in normal expansion mode use the "pts" function, supplying the start time (in seconds) as the second argument.
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.
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".
boxw
Set the width of the box to be drawn around text. The default value of boxw is computed automatically to match the text width
boxh
Set the height of the box to be drawn around text. The default value of boxh is computed automatically to match the text height
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). Value will be rounded to nearest integer. Minimum value is "1". Drop-frame timecode is supported for frame rates 30 & 60.
tc24hmax
If set to 1, the output of the timecode option will wrap around at 24 hours. Default is 0 (disabled).
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.
text_source
Text source should be set as side_data_detection_bboxes if you want to use text data in detection bboxes of side data.
If text source is set, text and textfile will be ignored and still use text data in detection bboxes of side data. So please do not use this parameter if you are not sure about the text source.
reload
The textfile will be reloaded at specified frame interval. Be sure to update textfile atomically, or it may be read partially, or even fail. Range is 0 to INT_MAX. Default is 0.
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
font_a
the ascent size defined in the font metrics
font_d
the descent size defined in the font metrics
top_a
the maximum ascender of the glyphs of the first text line
bottom_d
the maximum descender of the glyphs of the last text line
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 to each other, so you can for example specify "y=x/dar".
pict_type
A one character description of the current frame’s picture type.
pkt_pos
The current packet’s position in the input file or stream (in bytes, from the start of the input). A value of -1 indicates this info is not available.
duration
The current packet’s duration, in seconds.
pkt_size
The current packet’s size (in bytes).
Text expansion
If expansion is set to "strftime", the filter recognizes sequences accepted by the "strftime" C function in the provided text and expands them accordingly. Check the documentation of "strftime". This feature is deprecated in favor of "normal" expansion with the "gmtime" or "localtime" expansion functions.
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 with the textfile option 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" C function format string. The format string is extended to support the variable %[1-6]N which prints fractions of the second with optionally specified number of digits.
localtime
The time at which the filter is running, expressed in the local time zone. It can accept an argument: a "strftime" C function format string. The format string is extended to support the variable %[1-6]N which prints fractions of the second with optionally specified number of digits.
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.
Available metadata can be identified by inspecting entries starting with TAG included within each frame section printed by running "ffprobe -show_frames".
String metadata generated in filters leading to the drawtext filter are also available.
n, frame_num
The frame number, starting from 0.
pict_type
A one 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 "hms", a third argument "24HH" may be supplied to present the hour part of the formatted timestamp in 24h format (00-23).
If the format is set to "localtime" or "gmtime", a third argument may be supplied: a "strftime" C function format string. By default, YYYY-MM-DD HH:MM:SS format will be used.
Commands
This filter
supports altering parameters via commands:
reinit
Alter existing filter parameters.
Syntax for the argument is the same as for filter invocation, e.g.
fontsize=56:fontcolor=green:text='Hello World'
Full filter invocation with sendcmd would look like this:
sendcmd=c='56.0 drawtext reinit fontsize=56\:fontcolor=green\:text=Hello\\ World'
If the entire argument can’t be parsed or applied as valid values then the filter will continue with its existing parameters.
The following
options are also supported as commands:
*<x>
*<y>
*<alpha>
*<fontsize>
*<fontcolor>
*<boxcolor>
*<bordercolor>
*<shadowcolor>
*<box>
*<boxw>
*<boxh>
*<boxborderw>
*<line_spacing>
*<text_align>
*<shadowx>
*<shadowy>
*<borderw>
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=yellow [AT] 0.2: box=1:
boxcolor=red [AT] 0.2"
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'
• |
Draw "Test Text" with font size dependent on height of the video. |
drawtext="text='Test Text': fontsize=h/30: x=(w-text_w)/2: y=(h-text_h*2)"
• |
Print the date of a real-time encoding (see documentation for the "strftime" C function): |
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 }"
• |
Horizontally align multiple separate texts. Note that max_glyph_a and the fontsize value are included in the y offset. |
drawtext=fontfile=FreeSans.ttf:text=DOG:fontsize=24:x=10:y=20+24-max_glyph_a,
drawtext=fontfile=FreeSans.ttf:text=cow:fontsize=24:x=80:y=20+24-max_glyph_a
• |
Plot special lavf.image2dec.source_basename metadata onto each frame if such metadata exists. Otherwise, plot the string "NA". Note that image2 demuxer must have option -export_path_metadata 1 for the special metadata fields to be available for filters. |
drawtext="fontsize=20:fontcolor=white:fontfile=FreeSans.ttf:text='%{metadata\:lavf.image2dec.source_basename\:NA}':x=10:y=10"
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/>.
For more information about libharfbuzz, check: <https://github.com/harfbuzz/harfbuzz>.
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.
canny
Apply Canny edge detector on all selected planes.
Default value is wires.
planes
Select planes for filtering. By default all available planes are filtered.
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
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. Default is disabled.
use_alpha
Include alpha values in the quantization calculation. Allows creating palettized output images (e.g. PNG8) with multiple alpha smooth blending.
entropy
Measure graylevel entropy in histogram of color channels of
video frames.
It accepts the
following parameters:
mode
Can be either normal or diff. Default is normal.
diff mode measures entropy of histogram delta values, absolute differences between neighbour histogram values.
epx
Apply the EPX magnification filter which is designed for
pixel art.
It accepts the following option:
n |
Set the scaling dimension: 2 for "2xEPX", 3 for "3xEPX". Default is 3. |
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 -1000.0 to 1000.0. 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; deprecated, do not use | ||
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
Commands
This filter supports the all above options as commands.
estdif
Deinterlace the input video ("estdif" stands for
"Edge Slope Tracing Deinterlacing Filter").
Spatial only
filter that uses edge slope tracing algorithm to interpolate
missing lines. It accepts the following parameters:
mode
The interlacing mode to adopt.
It accepts one of the following values:
frame
Output one frame for each frame.
field
Output one frame for each field.
The default value is "field".
parity
The picture field parity assumed for the input interlaced video. It accepts one of the following values:
tff |
Assume the top field is first. |
|||
bff |
Assume the bottom field is first. |
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. Accepts one of the following values:
all |
Deinterlace all frames. |
interlaced
Only deinterlace frames marked as interlaced.
The default value is "all".
rslope
Specify the search radius for edge slope tracing. Default value is 1. Allowed range is from 1 to 15.
redge
Specify the search radius for best edge matching. Default value is 2. Allowed range is from 0 to 15.
ecost
Specify the edge cost for edge matching. Default value is 2. Allowed range is from 0 to 50.
mcost
Specify the middle cost for edge matching. Default value is 1. Allowed range is from 0 to 50.
dcost
Specify the distance cost for edge matching. Default value is 1. Allowed range is from 0 to 50.
interp
Specify the interpolation used. Default is 4-point interpolation. It accepts one of the following values:
2p |
Two-point interpolation. |
|||
4p |
Four-point interpolation. |
|||
6p |
Six-point interpolation. |
Commands
This filter supports same commands as options.
exposure
Adjust exposure of the video stream.
The filter
accepts the following options:
exposure
Set the exposure correction in EV. Allowed range is from -3.0 to 3.0 EV Default value is 0 EV.
black
Set the black level correction. Allowed range is from -1.0 to 1.0. Default value is 0.
Commands
This filter supports same commands as options.
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
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
feedback
Apply feedback video filter.
This filter pass cropped input frames to 2nd output. From there it can be filtered with other video filters. After filter receives frame from 2nd input, that frame is combined on top of original frame from 1st input and passed to 1st output.
The typical usage is filter only part of frame.
The filter accepts the following options:
x |
||||
y |
Set the top left crop position. |
|||
w |
||||
h |
Set the crop size. |
Examples
• |
Blur only top left rectangular part of video frame size 100x100 with gblur filter. |
[in][blurin]feedback=x=0:y=0:w=100:h=100[out][blurout];[blurout]gblur=8[blurin]
• |
Draw black box on top left part of video frame of size 100x100 with drawbox filter. |
[in][blurin]feedback=x=0:y=0:w=100:h=100[out][blurout];[blurout]drawbox=x=0:y=0:w=100:h=100:t=100[blurin]
fftdnoiz
Denoise frames using 3D FFT (frequency domain
filtering).
The filter
accepts the following options:
sigma
Set the noise sigma constant. This sets denoising strength. Default value is 1. Allowed range is from 0 to 30. Using very high sigma with low overlap may give blocking artifacts.
amount
Set amount of denoising. By default all detected noise is reduced. Default value is 1. Allowed range is from 0 to 1.
block
Set size of block in pixels, Default is 32, can be 8 to 256.
overlap
Set block overlap. Default is 0.5. Allowed range is from 0.2 to 0.8.
method
Set denoising method. Default is "wiener", can also be "hard".
prev
Set number of previous frames to use for denoising. By default is set to 0.
next
Set number of next frames to to use for denoising. By default is set to 0.
planes
Set planes which will be filtered, by default are all available filtered except alpha.
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.
eval
Set when the expressions are evaluated.
It accepts the
following values:
init
Only evaluate expressions once during the filter initialization.
frame
Evaluate expressions for each incoming frame.
Default value is init.
The filter accepts the following variables:
X |
|||
Y |
The coordinates of the current sample. | ||
W |
|||
H |
The width and height of the image. | ||
N |
The number of input frame, starting from 0. | ||
WS |
|||
HS |
The size of FFT array for horizontal and vertical processing. |
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 optionally followed by "t" output frame will use only top field, or in case of "b" it will use only bottom field. If line starts with "#" or ";" that line is skipped.
mode
Can be item "absolute" or "relative" or "pattern". Default is "absolute". The "pattern" mode is same as "relative" mode, except at last entry of file if there are more frames to process than "hint" file is seek back to start.
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.
fillborders
Fill borders of the input video, without changing video
stream dimensions. Sometimes video can have garbage at the
four edges and you may not want to crop video input to keep
size multiple of some number.
This filter
accepts the following options:
left
Number of pixels to fill from left border.
right
Number of pixels to fill from right border.
top |
Number of pixels to fill from top border. |
bottom
Number of pixels to fill from bottom border.
mode
Set fill mode.
It accepts the
following values:
smear
fill pixels using outermost pixels
mirror
fill pixels using mirroring (half sample symmetric)
fixed
fill pixels with constant value
reflect
fill pixels using reflecting (whole sample symmetric)
wrap
fill pixels using wrapping
fade
fade pixels to constant value
margins
fill pixels at top and bottom with weighted averages pixels near borders
Default is smear.
color
Set color for pixels in fixed or fade mode. Default is black.
Commands
This filter supports same commands as options. The command accepts the same syntax of the corresponding option.
If the specified expression is not valid, it is kept at its current value.
find_rect
Find a rectangular object in the input video.
The object to search for must be specified as a gray8 image specified with the object option.
For each possible match, a score is computed. If the score reaches the specified threshold, the object is considered found.
If the input video contains multiple instances of the object, the filter will find only one of them.
When an object
is found, the following metadata entries are set in the
matching frame:
lavfi.rect.w
width of object
lavfi.rect.h
height of object
lavfi.rect.x
x position of object
lavfi.rect.y
y position of object
lavfi.rect.score
match score of the found object
It accepts the
following options:
object
Filepath of the object image, needs to be in gray8.
threshold
Detection threshold, expressed as a decimal number in the range 0-1.
A threshold value of 0.01 means only exact matches, a threshold of 0.99 means almost everything matches.
Default value is 0.5.
mipmaps
Number of mipmaps, default is 3.
xmin, ymin, xmax, ymax
Specifies the rectangle in which to search.
discard
Discard frames where object is not detected. Default is disabled.
Examples
• |
Cover a rectangular object by the supplied image of a given video using ffmpeg: |
ffmpeg -i file.ts -vf find_rect=newref.pgm,cover_rect=cover.jpg:mode=cover new.mkv
• |
Find the position of an object in each frame using ffprobe and write it to a log file: |
ffprobe -f
lavfi
movie=test.mp4,find_rect=object=object.pgm:threshold=0.3 \
-show_entries
frame=pkt_pts_time:frame_tags=lavfi.rect.x,lavfi.rect.y \
-of csv -o find_rect.csv
floodfill
Flood area with values of same pixel components with another
values.
It accepts the following options:
x |
Set pixel x coordinate. |
|||
y |
Set pixel y coordinate. |
|||
s0 |
Set source #0 component value. |
|||
s1 |
Set source #1 component value. |
|||
s2 |
Set source #2 component value. |
|||
s3 |
Set source #3 component value. |
|||
d0 |
Set destination #0 component value. |
|||
d1 |
Set destination #1 component value. |
|||
d2 |
Set destination #2 component value. |
|||
d3 |
Set destination #3 component value. |
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. It accepts expressions containing the following constants: |
source_fps
The input’s frame rate
ntsc
NTSC frame rate of "30000/1001"
pal |
PAL frame rate of 25.0 |
film
Film frame rate of 24.0
ntsc_film
NTSC-film frame rate of "24000/1001"
The default is 25.
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.
round
Timestamp (PTS) rounding method.
Possible values
are:
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".
eof_action
Action performed when reading the last frame.
Possible values
are:
round
Use same timestamp rounding method as used for other frames.
pass
Pass through last frame if input duration has not been reached yet.
The default is "round".
Alternatively, the options can be specified as a flat string: fps[:start_time[: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 8.2.
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.
freezedetect
Detect frozen video.
This filter logs a message and sets frame metadata when it detects that the input video has no significant change in content during a specified duration. Video freeze detection calculates the mean average absolute difference of all the components of video frames and compares it to a noise floor.
The printed times and duration are expressed in seconds. The "lavfi.freezedetect.freeze_start" metadata key is set on the first frame whose timestamp equals or exceeds the detection duration and it contains the timestamp of the first frame of the freeze. The "lavfi.freezedetect.freeze_duration" and "lavfi.freezedetect.freeze_end" metadata keys are set on the first frame after the freeze.
The filter
accepts the following options:
noise, n
Set noise tolerance. Can be specified in dB (in case "dB" is appended to the specified value) or as a difference ratio between 0 and 1. Default is -60dB, or 0.001.
duration, d
Set freeze duration until notification (default is 2 seconds).
freezeframes
Freeze video frames.
This filter freezes video frames using frame from 2nd input.
The filter
accepts the following options:
first
Set number of first frame from which to start freeze.
last
Set number of last frame from which to end freeze.
replace
Set number of frame from 2nd input which will be used instead of replaced frames.
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 FREI0R_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 a color description as 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>
Commands
This filter supports the filter_params option as commands.
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 − 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. Default 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.
Commands
This filter supports same commands as options. The command accepts the same syntax of the corresponding option.
If the specified expression is not valid, it is kept at its current value.
geq
Apply generic equation to each pixel.
The filter
accepts the following options:
lum_expr, lum
Set the luma 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 luma 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 luma 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.
psum(x,y), lumsum(x, y),
cbsum(x,y), crsum(x,y), rsum(x,y), gsum(x,y),
bsum(x,y), alphasum(x,y)
Sum of sample values in the rectangle from (0,0) to (x,y), this allows obtaining sums of samples within a rectangle. See the functions without the sum postfix.
interpolation
Set one of interpolation
methods:
nearest, n
bilinear, b
Default is bilinear.
For functions, if x and y are outside the area, the value will be automatically clipped to the closer edge.
Please note that this filter can use multiple threads in which case each slice will have its own expression state. If you want to use only a single expression state because your expressions depend on previous state then you should limit the number of filter threads to 1.
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
graphmonitor
Show various filtergraph stats.
With this filter one can debug complete filtergraph. Especially issues with links filling with queued frames.
The filter
accepts the following options:
size, s
Set video output size. Default is hd720.
opacity, o
Set video opacity. Default is 0.9. Allowed range is from 0 to 1.
mode, m
Set output mode flags.
Available
values for flags are:
full
No any filtering. Default.
compact
Show only filters with queued frames.
nozero
Show only filters with non-zero stats.
noeof
Show only filters with non-eof stat.
nodisabled
Show only filters that are enabled in timeline.
flags, f
Set flags which enable which stats are shown in video.
Available
values for flags are:
none
All flags turned off.
all |
All flags turned on. |
queue
Display number of queued frames in each link.
frame_count_in
Display number of frames taken from filter.
frame_count_out
Display number of frames given out from filter.
frame_count_delta
Display delta number of frames between above two values.
pts |
Display current filtered frame pts. |
pts_delta
Display pts delta between current and previous frame.
time
Display current filtered frame time.
time_delta
Display time delta between current and previous frame.
timebase
Display time base for filter link.
format
Display used format for filter link.
size
Display video size or number of audio channels in case of audio used by filter link.
rate
Display video frame rate or sample rate in case of audio used by filter link.
eof |
Display link output status. |
sample_count_in
Display number of samples taken from filter.
sample_count_out
Display number of samples given out from filter.
sample_count_delta
Display delta number of samples between above two values.
disabled
Show the timeline filter status.
rate, r
Set upper limit for video rate of output stream, Default value is 25. This guarantee that output video frame rate will not be higher than this value.
grayworld
A color constancy filter that applies color correction based
on the grayworld assumption
The algorithm uses linear light, so input data should be linearized beforehand (and possibly correctly tagged).
ffmpeg -i INPUT -vf zscale=transfer=linear,grayworld,zscale=transfer=bt709,format=yuv420p OUTPUT
greyedge
A color constancy variation filter which estimates scene
illumination via grey edge algorithm and corrects the scene
colors accordingly.
See: <https://staff.science.uva.nl/th.gevers/pub/GeversTIP07.pdf>
The filter
accepts the following options:
difford
The order of differentiation to be applied on the scene. Must be chosen in the range [0,2] and default value is 1.
minknorm
The Minkowski parameter to be used for calculating the Minkowski distance. Must be chosen in the range [0,20] and default value is 1. Set to 0 for getting max value instead of calculating Minkowski distance.
sigma
The standard deviation of Gaussian blur to be applied on the scene. Must be chosen in the range [0,1024.0] and default value = 1. floor( sigma * break_off_sigma(3) ) can’t be equal to 0 if difford is greater than 0.
Examples
• |
Grey Edge: |
greyedge=difford=1:minknorm=5:sigma=2
• |
Max Edge: |
greyedge=difford=1:minknorm=0:sigma=2
guided
Apply guided filter for edge-preserving smoothing, dehazing
and so on.
The filter
accepts the following options:
radius
Set the box radius in pixels. Allowed range is 1 to 20. Default is 3.
eps |
Set regularization parameter (with square). Allowed range is 0 to 1. Default is 0.01. |
mode
Set filter mode. Can be "basic" or "fast". Default is "basic".
sub |
Set subsampling ratio for "fast" mode. Range is 2 to 64. Default is 4. No subsampling occurs in "basic" mode. |
guidance
Set guidance mode. Can be "off" or "on". Default is "off". If "off", single input is required. If "on", two inputs of the same resolution and pixel format are required. The second input serves as the guidance.
planes
Set planes to filter. Default is first only.
Commands
This filter supports the all above options as commands.
Examples
• |
Edge-preserving smoothing with guided filter: |
ffmpeg -i in.png -vf guided out.png
• |
Dehazing, structure-transferring filtering, detail enhancement with guided filter. For the generation of guidance image, refer to paper "Guided Image Filtering". See: <http://kaiminghe.com/publications/pami12guidedfilter.pdf>. |
ffmpeg -i in.png -i guidance.png -filter_complex guided=guidance=on out.png
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:
clut
Set which CLUT video frames will be processed from second input stream, can be first or all. Default is all.
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).
This filter also supports the framesync options.
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>.
Commands
This filter supports the "interp" option as commands.
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:
stack
Per color component graphs are placed below each other.
parade
Per color component graphs are placed side by side.
overlay
Presents information identical to that in the "parade", except that the graphs representing color components are superimposed directly over one another.
Default is "stack".
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.
colors_mode
Set colors mode. It accepts the
following values:
whiteonblack
blackonwhite
whiteongray
blackongray
coloronblack
coloronwhite
colorongray
blackoncolor
whiteoncolor
grayoncolor
Default is "whiteonblack".
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.
Commands
This filter supports same commands as options. The command accepts the same syntax of the corresponding option.
If the specified expression is not valid, it is kept at its current value.
hwdownload
Download hardware frames to system memory.
The input must be in hardware frames, and the output a non-hardware format. Not all formats will be supported on the output - it may be necessary to insert an additional format filter immediately following in the graph to get the output in a supported format.
hwmap
Map hardware frames to system memory or to another
device.
This filter has several different modes of operation; which one is used depends on the input and output formats:
• |
Hardware frame input, normal frame output |
Map the input frames to system memory and pass them to the output. If the original hardware frame is later required (for example, after overlaying something else on part of it), the hwmap filter can be used again in the next mode to retrieve it.
• |
Normal frame input, hardware frame output |
If the input is actually a software-mapped hardware frame, then unmap it - that is, return the original hardware frame.
Otherwise, a device must be provided. Create new hardware surfaces on that device for the output, then map them back to the software format at the input and give those frames to the preceding filter. This will then act like the hwupload filter, but may be able to avoid an additional copy when the input is already in a compatible format.
• |
Hardware frame input and output |
A device must be supplied for the output, either directly or with the derive_device option. The input and output devices must be of different types and compatible - the exact meaning of this is system-dependent, but typically it means that they must refer to the same underlying hardware context (for example, refer to the same graphics card).
If the input frames were originally created on the output device, then unmap to retrieve the original frames.
Otherwise, map the frames to the output device - create new hardware frames on the output corresponding to the frames on the input.
The following
additional parameters are accepted:
mode
Set the frame mapping mode.
Some combination of:
read
The mapped frame should be readable.
write
The mapped frame should be writeable.
overwrite
The mapping will always overwrite the entire frame.
This may improve performance in some cases, as the original contents of the frame need not be loaded.
direct
The mapping must not involve any copying.
Indirect mappings to copies of frames are created in some cases where either direct mapping is not possible or it would have unexpected properties. Setting this flag ensures that the mapping is direct and will fail if that is not possible.
Defaults to read+write if not specified.
derive_device type
Rather than using the device supplied at initialisation, instead derive a new device of type type from the device the input frames exist on.
reverse
In a hardware to hardware mapping, map in reverse - create frames in the sink and map them back to the source. This may be necessary in some cases where a mapping in one direction is required but only the opposite direction is supported by the devices being used.
This option is dangerous - it may break the preceding filter in undefined ways if there are any additional constraints on that filter’s output. Do not use it without fully understanding the implications of its use.
hwupload
Upload system memory frames to hardware surfaces.
The device to upload to must be supplied when the filter is initialised. If using ffmpeg, select the appropriate device with the -filter_hw_device option or with the derive_device option. The input and output devices must be of different types and compatible - the exact meaning of this is system-dependent, but typically it means that they must refer to the same underlying hardware context (for example, refer to the same graphics card).
The following
additional parameters are accepted:
derive_device type
Rather than using the device supplied at initialisation, instead derive a new device of type type from the device the input frames exist on.
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
accepts 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.
hsvhold
Turns a certain HSV range into gray values.
This filter measures color difference between set HSV color in options and ones measured in video stream. Depending on options, output colors can be changed to be gray or not.
The filter accepts the following options:
hue |
Set the hue value which will be used in color difference calculation. Allowed range is from -360 to 360. Default value is 0. | ||
sat |
Set the saturation value which will be used in color difference calculation. Allowed range is from -1 to 1. Default value is 0. | ||
val |
Set the value which will be used in color difference calculation. Allowed range is from -1 to 1. Default value is 0. |
similarity
Set similarity percentage with the key color. Allowed range is from 0 to 1. Default value is 0.01.
0.00001 matches only the exact key color, while 1.0 matches everything.
blend
Blend percentage. Allowed range is from 0 to 1. Default value is 0.
0.0 makes pixels either fully gray, or not gray at all.
Higher values result in more gray pixels, with a higher gray pixel the more similar the pixels color is to the key color.
hsvkey
Turns a certain HSV range into transparency.
This filter measures color difference between set HSV color in options and ones measured in video stream. Depending on options, output colors can be changed to transparent by adding alpha channel.
The filter accepts the following options:
hue |
Set the hue value which will be used in color difference calculation. Allowed range is from -360 to 360. Default value is 0. | ||
sat |
Set the saturation value which will be used in color difference calculation. Allowed range is from -1 to 1. Default value is 0. | ||
val |
Set the value which will be used in color difference calculation. Allowed range is from -1 to 1. Default value is 0. |
similarity
Set similarity percentage with the key color. Allowed range is from 0 to 1. Default value is 0.01.
0.00001 matches only the exact key color, while 1.0 matches everything.
blend
Blend percentage. Allowed range is from 0 to 1. Default value is 0.
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.
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 |
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s |
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h |
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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.
huesaturation
Apply hue-saturation-intensity adjustments to input video
stream.
This filter operates in RGB colorspace.
This filter accepts the following options:
hue |
Set the hue shift in degrees to apply. Default is 0. Allowed range is from -180 to 180. |
saturation
Set the saturation shift. Default is 0. Allowed range is from -1 to 1.
intensity
Set the intensity shift. Default is 0. Allowed range is from -1 to 1.
colors
Set which primary and complementary colors are going to be adjusted. This options is set by providing one or multiple values. This can select multiple colors at once. By default all colors are selected.
r |
Adjust reds. |
|||
y |
Adjust yellows. |
|||
g |
Adjust greens. |
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c |
Adjust cyans. |
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b |
Adjust blues. |
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m |
Adjust magentas. |
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a |
Adjust all colors. |
strength
Set strength of filtering. Allowed range is from 0 to 100. Default value is 1.
rw, gw, bw
Set weight for each RGB component. Allowed range is from 0 to 1. By default is set to 0.333, 0.334, 0.333. Those options are used in saturation and lightess processing.
lightness
Set preserving lightness, by default is disabled. Adjusting hues can change lightness from original RGB triplet, with this option enabled lightness is kept at same 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.
The "hysteresis" filter also supports the framesync options.
iccdetect
Detect the colorspace from an embedded ICC profile (if
present), and update the frame’s tags accordingly.
This filter
accepts the following options:
force
If true, the frame’s existing colorspace tags will always be overridden by values detected from an ICC profile. Otherwise, they will only be assigned if they contain "unknown". Enabled by default.
iccgen
Generate ICC profiles and attach them to frames.
This filter
accepts the following options:
color_primaries
color_trc
Configure the colorspace that the ICC profile will be generated for. The default value of "auto" infers the value from the input frame’s metadata, defaulting to BT.709/sRGB as appropriate.
See the setparams filter for a list of possible values, but note that "unknown" are not valid values for this filter.
force
If true, an ICC profile will be generated even if it would overwrite an already existing ICC profile. Disabled by default.
identity
Obtain the identity score between two input videos.
This filter takes two input videos.
Both input videos 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 per component, average, min and max identity score is printed through the logging system.
The filter stores the calculated identity scores of each frame in frame metadata.
This filter also supports the framesync options.
In the below example the input file main.mpg being processed is compared with the reference file ref.mpg.
ffmpeg -i main.mpg -i ref.mpg -lavfi identity -f null -
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.
Commands
This filter supports the all above options as commands.
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.
Commands
This filter supports the all above options as commands.
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
Vertical lowpass filter to
avoid twitter interlacing and reduce moire patterns.
0, off
Disable vertical lowpass filter
1, linear
Enable linear filter (default)
2, complex
Enable complex filter. This will slightly less reduce twitter and moire but better retain detail and subjective sharpness impression.
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
kirsch
Apply kirsch 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.
Commands
This filter supports the all above options as commands.
lagfun
Slowly update darker pixels.
This filter
makes short flashes of light appear longer. This filter
accepts the following options:
decay
Set factor for decaying. Default is .95. Allowed range is from 0 to 1.
planes
Set which planes to filter. Default is all. Allowed range is from 0 to 15.
Commands
This filter supports the all above options as commands.
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. Default is 0.5. | ||
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. Default is 0.5. | ||
k1 |
Coefficient of the quadratic correction term. This value has a range [-1,1]. 0 means no correction. Default is 0. | ||
k2 |
Coefficient of the double quadratic correction term. This value has a range [-1,1]. 0 means no correction. Default is 0. | ||
i |
Set interpolation type. Can be "nearest" or "bilinear". Default is "nearest". | ||
fc |
Specify the color of the unmapped pixels. For the syntax of this option, check the "Color" section in the ffmpeg-utils manual. Default color is "black@0". |
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.
Commands
This filter supports the all above options as commands.
lensfun
Apply lens correction via the lensfun library
(<http://lensfun.sourceforge.net/>).
The "lensfun" filter requires the camera make, camera model, and lens model to apply the lens correction. The filter will load the lensfun database and query it to find the corresponding camera and lens entries in the database. As long as these entries can be found with the given options, the filter can perform corrections on frames. Note that incomplete strings will result in the filter choosing the best match with the given options, and the filter will output the chosen camera and lens models (logged with level "info"). You must provide the make, camera model, and lens model as they are required.
To obtain a list of available makes and models, leave out one or both of "make" and "model" options. The filter will send the full list to the log with level "INFO". The first column is the make and the second column is the model. To obtain a list of available lenses, set any values for make and model and leave out the "lens_model" option. The filter will send the full list of lenses in the log with level "INFO". The ffmpeg tool will exit after the list is printed.
The filter
accepts the following options:
make
The make of the camera (for example, "Canon"). This option is required.
model
The model of the camera (for example, "Canon EOS 100D"). This option is required.
lens_model
The model of the lens (for example, "Canon EF-S 18-55mm f/3.5-5.6 IS STM"). This option is required.
db_path
The full path to the lens database folder. If not set, the filter will attempt to load the database from the install path when the library was built. Default is unset.
mode
The type of correction to
apply. The following values are valid options:
vignetting
Enables fixing lens vignetting.
geometry
Enables fixing lens geometry. This is the default.
subpixel
Enables fixing chromatic aberrations.
vig_geo
Enables fixing lens vignetting and lens geometry.
vig_subpixel
Enables fixing lens vignetting and chromatic aberrations.
distortion
Enables fixing both lens geometry and chromatic aberrations.
all |
Enables all possible corrections. |
focal_length
The focal length of the image/video (zoom; expected constant for video). For example, a 18--55mm lens has focal length range of [18--55], so a value in that range should be chosen when using that lens. Default 18.
aperture
The aperture of the image/video (expected constant for video). Note that aperture is only used for vignetting correction. Default 3.5.
focus_distance
The focus distance of the image/video (expected constant for video). Note that focus distance is only used for vignetting and only slightly affects the vignetting correction process. If unknown, leave it at the default value (which is 1000).
scale
The scale factor which is applied after transformation. After correction the video is no longer necessarily rectangular. This parameter controls how much of the resulting image is visible. The value 0 means that a value will be chosen automatically such that there is little or no unmapped area in the output image. 1.0 means that no additional scaling is done. Lower values may result in more of the corrected image being visible, while higher values may avoid unmapped areas in the output.
target_geometry
The target geometry of the
output image/video. The following values are valid options:
rectilinear (default)
fisheye
panoramic
equirectangular
fisheye_orthographic
fisheye_stereographic
fisheye_equisolid
fisheye_thoby
reverse
Apply the reverse of image correction (instead of correcting distortion, apply it).
interpolation
The type of interpolation used
when correcting distortion. The following values are valid
options:
nearest
linear (default)
lanczos
Examples
• |
Apply lens correction with make "Canon", camera model "Canon EOS 100D", and lens model "Canon EF-S 18-55mm f/3.5-5.6 IS STM" with focal length of "18" and aperture of "8.0". |
ffmpeg -i input.mov -vf lensfun=make=Canon:model="Canon EOS 100D":lens_model="Canon EF-S 18-55mm f/3.5-5.6 IS STM":focal_length=18:aperture=8 -c:v h264 -b:v 8000k output.mov
• |
Apply the same as before, but only for the first 5 seconds of video. |
ffmpeg -i input.mov -vf lensfun=make=Canon:model="Canon EOS 100D":lens_model="Canon EF-S 18-55mm f/3.5-5.6 IS STM":focal_length=18:aperture=8:enable='lte(t\,5)' -c:v h264 -b:v 8000k output.mov
libplacebo
Flexible GPU-accelerated processing filter based on
libplacebo
(<https://code.videolan.org/videolan/libplacebo>).
Options
The options for this filter are divided into the following sections:
Output mode
These options
control the overall output mode. By default, libplacebo will
try to preserve the source colorimetry and size as best as
it can, but it will apply any embedded film grain, dolby
vision metadata or anamorphic SAR present in source frames.
inputs
Set the number of inputs. This can be used, alongside the "idx" variable, to allow placing/blending multiple inputs inside the output frame. This effectively enables functionality similar to hstack, overlay, etc.
w |
|||
h |
Set the output video dimension expression. Default values are "iw" and "ih". |
Allows for the same expressions as the scale filter.
crop_x
crop_y
Set the input crop x/y expressions, default values are "(iw-cw)/2" and "(ih-ch)/2".
crop_w
crop_h
Set the input crop width/height expressions, default values are "iw" and "ih".
pos_x
pos_y
Set the output placement x/y expressions, default values are "(ow-pw)/2" and "(oh-ph)/2".
pos_w
pos_h
Set the output placement width/height expressions, default values are "ow" and "oh".
fps |
Set the output frame rate. This can be rational, e.g. "60000/1001". If set to the special string "none" (the default), input timestamps will instead be passed through to the output unmodified. Otherwise, the input video frames will be interpolated as necessary to rescale the video to the specified target framerate, in a manner as determined by the frame_mixer option. |
format
Set the output format override. If unset (the default), frames will be output in the same format as the respective input frames. Otherwise, format conversion will be performed.
force_original_aspect_ratio
force_divisible_by
Work the same as the identical scale filter options.
normalize_sar
If enabled, output frames will always have a pixel aspect ratio of 1:1. This will introduce additional padding/cropping as necessary. If disabled (the default), any aspect ratio mismatches, including those from e.g. anamorphic video sources, are forwarded to the output pixel aspect ratio.
pad_crop_ratio
Specifies a ratio (between 0.0 and 1.0) between padding and cropping when the input aspect ratio does not match the output aspect ratio and normalize_sar is in effect. The default of 0.0 always pads the content with black borders, while a value of 1.0 always crops off parts of the content. Intermediate values are possible, leading to a mix of the two approaches.
fillcolor
Set the color used to fill the output area not covered by the output image, for example as a result of normalize_sar. For the general syntax of this option, check the "Color" section in the ffmpeg-utils manual. Defaults to "black".
corner_rounding
Render frames with rounded corners. The value, given as a float ranging from 0.0 to 1.0, indicates the relative degree of rounding, from fully square to fully circular. In other words, it gives the radius divided by half the smaller side length. Defaults to 0.0.
extra_opts
Pass extra libplacebo internal configuration options. These can be specified as a list of key=value pairs separated by ’:’. The following example shows how to configure a custom filter kernel ("EWA LanczosSharp") and use it to double the input image resolution:
-vf "libplacebo=w=iw*2:h=ih*2:extra_opts='upscaler=custom\:upscaler_preset=ewa_lanczos\:upscaler_blur=0.9812505644269356'"
colorspace
color_primaries
color_trc
range
Configure the colorspace that output frames will be delivered in. The default value of "auto" outputs frames in the same format as the input frames, leading to no change. For any other value, conversion will be performed.
See the setparams filter for a list of possible values.
apply_filmgrain
Apply film grain (e.g. AV1 or H.274) if present in source frames, and strip it from the output. Enabled by default.
apply_dolbyvision
Apply Dolby Vision RPU metadata if present in source frames, and strip it from the output. Enabled by default. Note that Dolby Vision will always output BT.2020+PQ, overriding the usual input frame metadata. These will also be picked as the values of "auto" for the respective frame output options.
In addition to
the expression constants documented for the scale
filter, the crop_w, crop_h, crop_x,
crop_y, pos_w, pos_h, pos_x and
pos_y options can also contain the following
constants:
in_idx, idx
The (0-based) numeric index of the currently active input stream.
crop_w, cw
crop_h, ch
The computed values of crop_w and crop_h.
pos_w, pw
pos_h, ph
The computed values of pos_w and pos_h.
in_t, t
The input frame timestamp, in seconds. NAN if input timestamp is unknown.
out_t, ot
The input frame timestamp, in seconds. NAN if input timestamp is unknown.
n |
The input frame number, starting with 0. |
Scaling
The options in
this section control how libplacebo performs upscaling and
(if necessary) downscaling. Note that libplacebo will always
internally operate on 4:4:4 content, so any sub-sampled
chroma formats such as "yuv420p" will necessarily
be upsampled and downsampled as part of the rendering
process. That means scaling might be in effect even if the
source and destination resolution are the same.
upscaler
downscaler
Configure the filter kernel
used for upscaling and downscaling. The respective defaults
are "spline36" and "mitchell". For a
full list of possible values, pass "help" to these
options. The most important values are:
none
Forces the use of built-in GPU texture sampling (typically bilinear). Extremely fast but poor quality, especially when downscaling.
bilinear
Bilinear interpolation. Can generally be done for free on GPUs, except when doing so would lead to aliasing. Fast and low quality.
nearest
Nearest-neighbour interpolation. Sharp but highly aliasing.
oversample
Algorithm that looks visually similar to nearest-neighbour interpolation but tries to preserve pixel aspect ratio. Good for pixel art, since it results in minimal distortion of the artistic appearance.
lanczos
Standard sinc-sinc interpolation kernel.
spline36
Cubic spline approximation of lanczos. No difference in performance, but has very slightly less ringing.
ewa_lanczos
Elliptically weighted average version of lanczos, based on a jinc-sinc kernel. This is also popularly referred to as just "Jinc scaling". Slow but very high quality.
gaussian
Gaussian kernel. Has certain ideal mathematical properties, but subjectively very blurry.
mitchell
Cubic BC spline with parameters recommended by Mitchell and Netravali. Very little ringing.
frame_mixer
Controls the kernel used for
mixing frames temporally. The default value is
"none", which disables frame mixing. For a full
list of possible values, pass "help" to this
option. The most important values are:
none
Disables frame mixing, giving a result equivalent to "nearest neighbour" semantics.
oversample
Oversamples the input video to create a "Smooth Motion"-type effect: if an output frame would exactly fall on the transition between two video frames, it is blended according to the relative overlap. This is the recommended option whenever preserving the original subjective appearance is desired.
mitchell_clamp
Larger filter kernel that smoothly interpolates multiple frames in a manner designed to eliminate ringing and other artefacts as much as possible. This is the recommended option wherever maximum visual smoothness is desired.
linear
Linear blend/fade between frames. Especially useful for constructing e.g. slideshows.
lut_entries
Configures the size of scaler LUTs, ranging from 1 to 256. The default of 0 will pick libplacebo’s internal default, typically 64.
antiringing
Enables anti-ringing (for non-EWA filters). The value (between 0.0 and 1.0) configures the strength of the anti-ringing algorithm. May increase aliasing if set too high. Disabled by default.
sigmoid
Enable sigmoidal compression during upscaling. Reduces ringing slightly. Enabled by default.
Debanding
Libplacebo
comes with a built-in debanding filter that is good at
counteracting many common sources of banding and blocking.
Turning this on is highly recommended whenever quality is
desired.
deband
Enable (fast) debanding algorithm. Disabled by default.
deband_iterations
Number of deband iterations of the debanding algorithm. Each iteration is performed with progressively increased radius (and diminished threshold). Recommended values are in the range 1 to 4. Defaults to 1.
deband_threshold
Debanding filter strength. Higher numbers lead to more aggressive debanding. Defaults to 4.0.
deband_radius
Debanding filter radius. A higher radius is better for slow gradients, while a lower radius is better for steep gradients. Defaults to 16.0.
deband_grain
Amount of extra output grain to add. Helps hide imperfections. Defaults to 6.0.
Color adjustment
A collection of
subjective color controls. Not very rigorous, so the exact
effect will vary somewhat depending on the input primaries
and colorspace.
brightness
Brightness boost, between -1.0 and 1.0. Defaults to 0.0.
contrast
Contrast gain, between 0.0 and 16.0. Defaults to 1.0.
saturation
Saturation gain, between 0.0 and 16.0. Defaults to 1.0.
hue |
Hue shift in radians, between -3.14 and 3.14. Defaults to 0.0. This will rotate the UV subvector, defaulting to BT.709 coefficients for RGB inputs. |
gamma
Gamma adjustment, between 0.0 and 16.0. Defaults to 1.0.
cones
Cone model to use for color blindness simulation. Accepts any combination of "l", "m" and "s". Here are some examples:
m |
Deuteranomaly / deuteranopia (affecting 3%-4% of the population) | ||
l |
Protanomaly / protanopia (affecting 1%-2% of the population) | ||
l+m |
Monochromacy (very rare) |
l+m+s
Achromatopsy (complete loss of daytime vision, extremely rare)
cone-strength
Gain factor for the cones specified by "cones", between 0.0 and 10.0. A value of 1.0 results in no change to color vision. A value of 0.0 (the default) simulates complete loss of those cones. Values above 1.0 result in exaggerating the differences between cones, which may help compensate for reduced color vision.
Peak detection
To help deal
with sources that only have static HDR10 metadata (or no
tagging whatsoever), libplacebo uses its own internal frame
analysis compute shader to analyze source frames and adapt
the tone mapping function in realtime. If this is too slow,
or if exactly reproducible frame-perfect results are needed,
it’s recommended to turn this feature off.
peak_detect
Enable HDR peak detection. Ignores static MaxCLL/MaxFALL values in favor of dynamic detection from the input. Note that the detected values do not get written back to the output frames, they merely guide the internal tone mapping process. Enabled by default.
smoothing_period
Peak detection smoothing period, between 0.0 and 1000.0. Higher values result in peak detection becoming less responsive to changes in the input. Defaults to 100.0.
minimum_peak
Lower bound on the detected peak (relative to SDR white), between 0.0 and 100.0. Defaults to 1.0.
scene_threshold_low
scene_threshold_high
Lower and upper thresholds for scene change detection. Expressed in a logarithmic scale between 0.0 and 100.0. Default to 5.5 and 10.0, respectively. Setting either to a negative value disables this functionality.
percentile
Which percentile of the frame brightness histogram to use as the source peak for tone-mapping. Defaults to 99.995, a fairly conservative value. Setting this to 100.0 disables frame histogram measurement and instead uses the true peak brightness for tone-mapping.
Tone mapping
The options in
this section control how libplacebo performs tone-mapping
and gamut-mapping when dealing with mismatches between
wide-gamut or HDR content. In general, libplacebo relies on
accurate source tagging and mastering display gamut
information to produce the best results.
gamut_mode
How to handle out-of-gamut
colors that can occur as a result of colorimetric gamut
mapping.
clip
Do nothing, simply clip out-of-range colors to the RGB volume. Low quality but extremely fast.
perceptual
Perceptually soft-clip colors to the gamut volume. This is the default.
relative
Relative colorimetric hard-clip. Similar to "perceptual" but without the soft knee.
saturation
Saturation mapping, maps primaries directly to primaries in RGB space. Not recommended except for artificial computer graphics for which a bright, saturated display is desired.
absolute
Absolute colorimetric hard-clip. Performs no adjustment of the white point.
desaturate
Hard-desaturates out-of-gamut colors towards white, while preserving the luminance. Has a tendency to distort the visual appearance of bright objects.
darken
Linearly reduces content brightness to preserves saturated details, followed by clipping the remaining out-of-gamut colors.
warn
Highlight out-of-gamut pixels (by inverting/marking them).
linear
Linearly reduces chromaticity of the entire image to make it fit within the target color volume. Be careful when using this on BT.2020 sources without proper mastering metadata, as doing so will lead to excessive desaturation.
tonemapping
Tone-mapping algorithm to use.
Available values are:
auto
Automatic selection based on internal heuristics. This is the default.
clip
Performs no tone-mapping, just clips out-of-range colors. Retains perfect color accuracy for in-range colors but completely destroys out-of-range information. Does not perform any black point adaptation. Not configurable.
st2094-40
EETF from SMPTE ST 2094-40 Annex B, which applies the Bezier curves from HDR10+ dynamic metadata based on Bezier curves to perform tone-mapping. The OOTF used is adjusted based on the ratio between the targeted and actual display peak luminances.
st2094-10
EETF from SMPTE ST 2094-10 Annex B.2, which takes into account the input signal average luminance in addition to the maximum/minimum. The configurable contrast parameter influences the slope of the linear output segment, defaulting to 1.0 for no increase/decrease in contrast. Note that this does not currently include the subjective gain/offset/gamma controls defined in Annex B.3.
bt.2390
EETF from the ITU-R Report BT.2390, a hermite spline roll-off with linear segment. The knee point offset is configurable. Note that this parameter defaults to 1.0, rather than the value of 0.5 from the ITU-R spec.
bt.2446a
EETF from ITU-R Report BT.2446, method A. Designed for well-mastered HDR sources. Can be used for both forward and inverse tone mapping. Not configurable.
spline
Simple spline consisting of two polynomials, joined by a single pivot point. The parameter gives the pivot point (in PQ space), defaulting to 0.30. Can be used for both forward and inverse tone mapping.
reinhard
Simple non-linear, global tone mapping algorithm. The parameter specifies the local contrast coefficient at the display peak. Essentially, a parameter of 0.5 implies that the reference white will be about half as bright as when clipping. Defaults to 0.5, which results in the simplest formulation of this function.
mobius
Generalization of the reinhard tone mapping algorithm to support an additional linear slope near black. The tone mapping parameter indicates the trade-off between the linear section and the non-linear section. Essentially, for a given parameter x, every color value below x will be mapped linearly, while higher values get non-linearly tone-mapped. Values near 1.0 make this curve behave like "clip", while values near 0.0 make this curve behave like "reinhard". The default value is 0.3, which provides a good balance between colorimetric accuracy and preserving out-of-gamut details.
hable
Piece-wise, filmic tone-mapping algorithm developed by John Hable for use in Uncharted 2, inspired by a similar tone-mapping algorithm used by Kodak. Popularized by its use in video games with HDR rendering. Preserves both dark and bright details very well, but comes with the drawback of changing the average brightness quite significantly. This is sort of similar to "reinhard" with parameter 0.24.
gamma
Fits a gamma (power) function to transfer between the source and target color spaces, effectively resulting in a perceptual hard-knee joining two roughly linear sections. This preserves details at all scales fairly accurately, but can result in an image with a muted or dull appearance. The parameter is used as the cutoff point, defaulting to 0.5.
linear
Linearly stretches the input range to the output range, in PQ space. This will preserve all details accurately, but results in a significantly different average brightness. Can be used for inverse tone-mapping in addition to regular tone-mapping. The parameter can be used as an additional linear gain coefficient (defaulting to 1.0).
tonemapping_param
For tunable tone mapping functions, this parameter can be used to fine-tune the curve behavior. Refer to the documentation of "tonemapping". The default value of 0.0 is replaced by the curve’s preferred default setting.
inverse_tonemapping
If enabled, this filter will also attempt stretching SDR signals to fill HDR output color volumes. Disabled by default.
tonemapping_lut_size
Size of the tone-mapping LUT, between 2 and 1024. Defaults to 256. Note that this figure is squared when combined with "peak_detect".
contrast_recovery
Contrast recovery strength. If set to a value above 0.0, the source image will be divided into high-frequency and low-frequency components, and a portion of the high-frequency image is added back onto the tone-mapped output. May cause excessive ringing artifacts for some HDR sources, but can improve the subjective sharpness and detail left over in the image after tone-mapping. Defaults to 0.30.
contrast_smoothness
Contrast recovery lowpass kernel size. Defaults to 3.5. Increasing or decreasing this will affect the visual appearance substantially. Has no effect when "contrast_recovery" is disabled.
Dithering
By default,
libplacebo will dither whenever necessary, which includes
rendering to any integer format below 16-bit precision.
It’s recommended to always leave this on, since not
doing so may result in visible banding in the output, even
if the "debanding" filter is enabled. If maximum
performance is needed, use "ordered_fixed" instead
of disabling dithering.
dithering
Dithering method to use.
Accepts the following values:
none
Disables dithering completely. May result in visible banding.
blue
Dither with pseudo-blue noise. This is the default.
ordered
Tunable ordered dither pattern.
ordered_fixed
Faster ordered dither with a fixed size of 6. Texture-less.
white
Dither with white noise. Texture-less.
dither_lut_size
Dither LUT size, as log base2 between 1 and 8. Defaults to 6, corresponding to a LUT size of "64x64".
dither_temporal
Enables temporal dithering. Disabled by default.
Custom shaders
libplacebo supports a number of custom shaders based on the mpv .hook GLSL syntax. A collection of such shaders can be found here: <https://github.com/mpv-player/mpv/wiki/User-Scripts#user-shaders>
A full
description of the mpv shader format is beyond the scope of
this section, but a summary can be found here:
<https://mpv.io/manual/master/#options-glsl-shader>
custom_shader_path
Specifies a path to a custom shader file to load at runtime.
custom_shader_bin
Specifies a complete custom shader as a raw string.
Debugging / performance
All of the
options in this section default off. They may be of
assistance when attempting to squeeze the maximum
performance at the cost of quality.
skip_aa
Disable anti-aliasing when downscaling.
polar_cutoff
Truncate polar (EWA) scaler kernels below this absolute magnitude, between 0.0 and 1.0.
disable_linear
Disable linear light scaling.
disable_builtin
Disable built-in GPU sampling (forces LUT).
disable_fbos
Forcibly disable FBOs, resulting in loss of almost all functionality, but offering the maximum possible speed.
Commands
This filter supports almost all of the above options as commands.
Examples
• |
Tone-map input to standard gamut BT.709 output: |
libplacebo=colorspace=bt709:color_primaries=bt709:color_trc=bt709:range=tv
• |
Rescale input to fit into standard 1080p, with high quality scaling: |
libplacebo=w=1920:h=1080:force_original_aspect_ratio=decrease:normalize_sar=true:upscaler=ewa_lanczos:downscaler=ewa_lanczos
• |
Interpolate low FPS / VFR input to smoothed constant 60 fps output: |
libplacebo=fps=60:frame_mixer=mitchell_clamp
• |
Convert input to standard sRGB JPEG: |
libplacebo=format=yuv420p:colorspace=bt470bg:color_primaries=bt709:color_trc=iec61966-2-1:range=pc
• |
Use higher quality debanding settings: |
libplacebo=deband=true:deband_iterations=3:deband_radius=8:deband_threshold=6
• |
Run this filter on the CPU, on systems with Mesa installed (and with the most expensive options disabled): |
ffmpeg ... -init_hw_device vulkan:llvmpipe ... -vf libplacebo=upscaler=none:downscaler=none:peak_detect=false
• |
Suppress CPU-based AV1/H.274 film grain application in the decoder, in favor of doing it with this filter. Note that this is only a gain if the frames are either already on the GPU, or if you’re using libplacebo for other purposes, since otherwise the VRAM roundtrip will more than offset any expected speedup. |
ffmpeg -export_side_data +film_grain ... -vf libplacebo=apply_filmgrain=true
• |
Interop with VAAPI hwdec to avoid round-tripping through RAM: |
ffmpeg -init_hw_device vulkan -hwaccel vaapi -hwaccel_output_format vaapi ... -vf libplacebo
libvmaf
Calulate the VMAF (Video Multi-Method Assessment Fusion)
score for a reference/distorted pair of input videos.
The first input is the distorted video, and the second input is the reference video.
The obtained VMAF score is printed through the logging system.
It requires Netflix’s vmaf library (libvmaf) as a pre-requisite. After installing the library it can be enabled using: "./configure --enable-libvmaf".
The filter has
following options:
model
A ’|’ delimited list of vmaf models. Each model can be configured with a number of parameters. Default value: "version=vmaf_v0.6.1"
feature
A ’|’ delimited list of features. Each feature can be configured with a number of parameters.
log_path
Set the file path to be used to store log files.
log_fmt
Set the format of the log file (xml, json, csv, or sub).
n_threads
Set number of threads to be used when initializing libvmaf. Default value: 0, no threads.
n_subsample
Set frame subsampling interval to be used.
This filter also supports the framesync options.
Examples
• |
In the examples below, a distorted video distorted.mpg is compared with a reference file reference.mpg. | ||
• |
Basic usage: |
ffmpeg -i distorted.mpg -i reference.mpg -lavfi libvmaf=log_path=output.xml -f null -
• |
Example with multiple models: |
ffmpeg -i distorted.mpg -i reference.mpg -lavfi libvmaf='model=version=vmaf_v0.6.1\\:name=vmaf|version=vmaf_v0.6.1neg\\:name=vmaf_neg' -f null -
• |
Example with multiple addtional features: |
ffmpeg -i distorted.mpg -i reference.mpg -lavfi libvmaf='feature=name=psnr|name=ciede' -f null -
• |
Example with options and different containers: |
ffmpeg -i distorted.mpg -i reference.mkv -lavfi "[0:v]settb=AVTB,setpts=PTS-STARTPTS[main];[1:v]settb=AVTB,setpts=PTS-STARTPTS[ref];[main][ref]libvmaf=log_fmt=json:log_path=output.json" -f null -
libvmaf_cuda
This is the CUDA variant of the libvmaf filter. It
only accepts CUDA frames.
It requires Netflix’s vmaf library (libvmaf) as a pre-requisite. After installing the library it can be enabled using: "./configure --enable-nonfree --enable-ffnvcodec --enable-libvmaf".
Examples
• |
Basic usage showing CUVID hardware decoding and CUDA scaling with scale_cuda: |
ffmpeg \
-hwaccel cuda -hwaccel_output_format cuda -codec:v av1_cuvid
-i dis.obu \
-hwaccel cuda -hwaccel_output_format cuda -codec:v av1_cuvid
-i ref.obu \
-filter_complex "
[0:v]scale_cuda=format=yuv420p[ref]; \
[1:v]scale_cuda=format=yuv420p[dis]; \
[dis][ref]libvmaf_cuda=log_fmt=json:log_path=output.json
" \
-f null -
limitdiff
Apply limited difference filter using second and optionally
third video stream.
The filter
accepts the following options:
threshold
Set the threshold to use when allowing certain differences between video streams. Any absolute difference value lower or exact than this threshold will pick pixel components from first video stream.
elasticity
Set the elasticity of soft thresholding when processing video streams. This value multiplied with first one sets second threshold. Any absolute difference value greater or exact than second threshold will pick pixel components from second video stream. For values between those two threshold linear interpolation between first and second video stream will be used.
reference
Enable the reference (third) video stream processing. By default is disabled. If set, this video stream will be used for calculating absolute difference with first video stream.
planes
Specify which planes will be processed. Defaults to all available.
Commands
This filter supports the all above options as commands except option reference.
limiter
Limits the pixel components values to the specified range
[min, max].
The filter accepts the following options:
min |
Lower bound. Defaults to the lowest allowed value for the input. | ||
max |
Upper bound. Defaults to the highest allowed value for the input. |
planes
Specify which planes will be processed. Defaults to all available.
Commands
This filter supports the all above options as commands.
loop
Loop video frames.
The filter
accepts the following options:
loop
Set the number of loops. Setting this value to -1 will result in infinite loops. Default is 0.
size
Set maximal size in number of frames. Default is 0.
start
Set first frame of loop. Default is 0.
time
Set the time of loop start in seconds. Only used if option named start is set to -1.
Examples
• |
Loop single first frame infinitely: |
loop=loop=-1:size=1:start=0
• |
Loop single first frame 10 times: |
loop=loop=10:size=1:start=0
• |
Loop 10 first frames 5 times: |
loop=loop=5:size=10:start=0
lut1d
Apply a 1D LUT to an input video.
The filter
accepts the following options:
file
Set the 1D LUT file name.
Currently
supported formats:
cube
Iridas
csp |
cineSpace |
interp
Select interpolation mode.
Available
values are:
nearest
Use values from the nearest defined point.
linear
Interpolate values using the linear interpolation.
cosine
Interpolate values using the cosine interpolation.
cubic
Interpolate values using the cubic interpolation.
spline
Interpolate values using the spline interpolation.
Commands
This filter supports the all above options as commands.
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 |
|||
csp |
cineSpace |
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.
pyramid
Interpolate values using a pyramid.
prism
Interpolate values using a prism.
Commands
This filter supports the "interp" option as commands.
lumakey
Turn certain luma values into transparency.
The filter
accepts the following options:
threshold
Set the luma which will be used as base for transparency. Default value is 0.
tolerance
Set the range of luma values to be keyed out. Default value is 0.01.
softness
Set the range of softness. Default value is 0. Use this to control gradual transition from zero to full transparency.
Commands
This filter supports same commands as options. The command accepts the same syntax of the corresponding option.
If the specified expression is not valid, it is kept at its current value.
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/luma 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 "clipval".
Commands
This filter supports same commands as options.
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 luma: |
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 luma 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,
tlut2
The "lut2" filter takes two input streams and
outputs one stream.
The "tlut2" (time lut2) filter takes two consecutive frames from one single stream.
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 | ||
d |
set output bit depth, only available for "lut2" filter. By default is 0, which means bit depth is automatically picked from first input format. |
The "lut2" filter also supports the framesync options.
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".
Commands
This filter supports the all above options as commands except option "d".
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)'
• |
Show max difference between two video streams: |
lut2='if(lt(x,y),0,if(gt(x,y),pow(2,bdx)-1,pow(2,bdx-1))):if(lt(x,y),0,if(gt(x,y),pow(2,bdx)-1,pow(2,bdx-1))):if(lt(x,y),0,if(gt(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.
Commands
This filter supports the all above options as commands.
maskedmax
Merge the second and third input stream into output stream
using absolute differences between second input stream and
first input stream and absolute difference between third
input stream and first input stream. The picked value will
be from second input stream if second absolute difference is
greater than first one or from third input stream
otherwise.
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.
Commands
This filter supports the all above options as commands.
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.
Commands
This filter supports the all above options as commands.
maskedmin
Merge the second and third input stream into output stream
using absolute differences between second input stream and
first input stream and absolute difference between third
input stream and first input stream. The picked value will
be from second input stream if second absolute difference is
less than first one or from third input stream
otherwise.
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.
Commands
This filter supports the all above options as commands.
maskedthreshold
Pick pixels comparing absolute difference of two video
streams with fixed threshold.
If absolute difference between pixel component of first and second video stream is equal or lower than user supplied threshold than pixel component from first video stream is picked, otherwise pixel component from second video stream is picked.
This filter
accepts the following options:
threshold
Set threshold used when picking pixels from absolute difference from two input video streams.
planes
Set which planes will be processed as bitmap, unprocessed planes will be copied from second stream. By default value 0xf, all planes will be processed.
mode
Set mode of filter operation. Can be "abs" or "diff". Default is "abs".
Commands
This filter supports the all above options as commands.
maskfun
Create mask from input video.
For example it is useful to create motion masks after "tblend" filter.
This filter accepts the following options:
low |
Set low threshold. Any pixel component lower or exact than this value will be set to 0. |
high
Set high threshold. Any pixel component higher than this value will be set to max value allowed for current pixel format.
planes
Set planes to filter, by default all available planes are filtered.
fill
Fill all frame pixels with this value.
sum |
Set max average pixel value for frame. If sum of all pixel components is higher that this average, output frame will be completely filled with value set by fill option. Typically useful for scene changes when used in combination with "tblend" filter. |
Commands
This filter supports the all above options as commands.
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.
median
Pick median pixel from certain rectangle defined by
radius.
This filter
accepts the following options:
radius
Set horizontal radius size. Default value is 1. Allowed range is integer from 1 to 127.
planes
Set which planes to process. Default is 15, which is all available planes.
radiusV
Set vertical radius size. Default value is 0. Allowed range is integer from 0 to 127. If it is 0, value will be picked from horizontal "radius" option.
percentile
Set median percentile. Default value is 0.5. Default value of 0.5 will pick always median values, while 0 will pick minimum values, and 1 maximum values.
Commands
This filter supports same commands as options. The command accepts the same syntax of the corresponding option.
If the specified expression is not valid, it is kept at its current value.
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".
map0s
map1s
map2s
map3s
Set input to output stream mapping for output Nth plane. Default is 0.
map0p
map1p
map2p
map3p
Set input to output plane mapping for output Nth plane. Default is 0.
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.
midequalizer
Apply Midway Image Equalization effect using two video
streams.
Midway Image Equalization adjusts a pair of images to have the same histogram, while maintaining their dynamics as much as possible. It’s useful for e.g. matching exposures from a pair of stereo cameras.
This filter has two inputs and one output, which must be of same pixel format, but may be of different sizes. The output of filter is first input adjusted with midway histogram of both inputs.
This filter
accepts the following option:
planes
Set which planes to process. Default is 15, which is all available planes.
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.
vsbmc
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 10..
mix
Mix several video input streams into one video stream.
A description
of the accepted options follows.
inputs
The number of inputs. If unspecified, it defaults to 2.
weights
Specify weight of each input video stream as sequence. Each weight is separated by space. If number of weights is smaller than number of frames last specified weight will be used for all remaining unset weights.
scale
Specify scale, if it is set it will be multiplied with sum of each weight multiplied with pixel values to give final destination pixel value. By default scale is auto scaled to sum of weights.
planes
Set which planes to filter. Default is all. Allowed range is from 0 to 15.
duration
Specify how end of stream is
determined.
longest
The duration of the longest input. (default)
shortest
The duration of the shortest input.
first
The duration of the first input.
Commands
This filter
supports the following commands:
weights
scale
planes
Syntax is same as option with same name.
monochrome
Convert video to gray using custom color filter.
A description of the accepted options follows.
cb |
Set the chroma blue spot. Allowed range is from -1 to 1. Default value is 0. | ||
cr |
Set the chroma red spot. Allowed range is from -1 to 1. Default value is 0. |
size
Set the color filter size. Allowed range is from .1 to 10. Default value is 1.
high
Set the highlights strength. Allowed range is from 0 to 1. Default value is 0.
Commands
This filter supports the all above options as commands.
morpho
This filter allows to apply main morphological grayscale
transforms, erode and dilate with arbitrary structures set
in second input stream.
Unlike naive implementation and much slower performance in erosion and dilation filters, when speed is critical "morpho" filter should be used instead.
A description
of accepted options follows,
mode
Set morphological transform to
apply, can be:
erode
dilate
open
close
gradient
tophat
blackhat
Default is "erode".
planes
Set planes to filter, by default all planes except alpha are filtered.
structure
Set which structure video frames will be processed from second input stream, can be first or all. Default is all.
The "morpho" filter also supports the framesync options.
Commands
This filter supports same commands as options.
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 disregarding the number of previous sequentially dropped frames. |
Default value is 0.
keep
Set the maximum number of consecutive similar frames to ignore before to start dropping them. If the value is 0, the frame is dropped disregarding the number of previous sequentially similar 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.
msad
Obtain the MSAD (Mean Sum of Absolute Differences) between
two input videos.
This filter takes two input videos.
Both input videos 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 per component, average, min and max MSAD is printed through the logging system.
The filter stores the calculated MSAD of each frame in frame metadata.
This filter also supports the framesync options.
In the below example the input file main.mpg being processed is compared with the reference file ref.mpg.
ffmpeg -i main.mpg -i ref.mpg -lavfi msad -f null -
multiply
Multiply first video stream pixels values with second video
stream pixels values.
The filter
accepts the following options:
scale
Set the scale applied to second video stream. By default is 1. Allowed range is from 0 to 9.
offset
Set the offset applied to second video stream. By default is 0.5. Allowed range is from -1 to 1.
planes
Specify planes from input video stream that will be processed. By default all planes are processed.
Commands
This filter supports same commands as options.
negate
Negate (invert) the input video.
It accepts the
following option:
components
Set components to negate.
Available values for components are:
y |
||
u |
||
v |
||
a |
||
r |
||
g |
||
b |
negate_alpha
With value 1, it negates the alpha component, if present. Default value is 0.
Commands
This filter supports same commands as options.
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. Default is 1.0. Must be in range [1.0, 30.0]. | ||
p |
Set patch size. Default is 7. Must be odd number in range [0, 99]. | ||
pc |
Same as p but for chroma planes. |
The default value is 0 and means automatic.
r |
Set research size. Default is 15. Must be odd number in range [0, 99]. | ||
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 predictor 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, abs
weights trained to minimize absolute error
s, mse
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 |
new2
new3
Default is "new".
Commands
This filter supports same commands as options, excluding weights option.
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 pix_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
normalize
Normalize RGB video (aka histogram stretching, contrast
stretching). See:
https://en.wikipedia.org/wiki/Normalization_(image_processing)
For each channel of each frame, the filter computes the input range and maps it linearly to the user-specified output range. The output range defaults to the full dynamic range from pure black to pure white.
Temporal smoothing can be used on the input range to reduce flickering (rapid changes in brightness) caused when small dark or bright objects enter or leave the scene. This is similar to the auto-exposure (automatic gain control) on a video camera, and, like a video camera, it may cause a period of over- or under-exposure of the video.
The R,G,B channels can be normalized independently, which may cause some color shifting, or linked together as a single channel, which prevents color shifting. Linked normalization preserves hue. Independent normalization does not, so it can be used to remove some color casts. Independent and linked normalization can be combined in any ratio.
The normalize
filter accepts the following options:
blackpt
whitept
Colors which define the output range. The minimum input value is mapped to the blackpt. The maximum input value is mapped to the whitept. The defaults are black and white respectively. Specifying white for blackpt and black for whitept will give color-inverted, normalized video. Shades of grey can be used to reduce the dynamic range (contrast). Specifying saturated colors here can create some interesting effects.
smoothing
The number of previous frames to use for temporal smoothing. The input range of each channel is smoothed using a rolling average over the current frame and the smoothing previous frames. The default is 0 (no temporal smoothing).
independence
Controls the ratio of independent (color shifting) channel normalization to linked (color preserving) normalization. 0.0 is fully linked, 1.0 is fully independent. Defaults to 1.0 (fully independent).
strength
Overall strength of the filter. 1.0 is full strength. 0.0 is a rather expensive no-op. Defaults to 1.0 (full strength).
Commands
This filter supports same commands as options, excluding smoothing option. 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
Stretch video contrast to use the full dynamic range, with no temporal smoothing; may flicker depending on the source content:
normalize=blackpt=black:whitept=white:smoothing=0
As above, but with 50 frames of temporal smoothing; flicker should be reduced, depending on the source content:
normalize=blackpt=black:whitept=white:smoothing=50
As above, but with hue-preserving linked channel normalization:
normalize=blackpt=black:whitept=white:smoothing=50:independence=0
As above, but with half strength:
normalize=blackpt=black:whitept=white:smoothing=50:independence=0:strength=0.5
Map the darkest input color to red, the brightest input color to cyan:
normalize=blackpt=red:whitept=cyan
null
Pass the video source unchanged to the output.
ocr
Optical Character Recognition
This filter uses Tesseract for optical character recognition. To enable compilation of this filter, you need to configure FFmpeg with "--enable-libtesseract".
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". The filter exports confidence of recognized words as the frame metadata "lavfi.ocr.confidence".
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 depends 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".
oscilloscope
2D Video Oscilloscope.
Useful to measure spatial impulse, step responses, chroma delays, etc.
It accepts the following parameters:
x |
Set scope center x position. | ||
y |
Set scope center y position. | ||
s |
Set scope size, relative to frame diagonal. | ||
t |
Set scope tilt/rotation. | ||
o |
Set trace opacity. | ||
tx |
Set trace center x position. | ||
ty |
Set trace center y position. | ||
tw |
Set trace width, relative to width of frame. | ||
th |
Set trace height, relative to height of frame. | ||
c |
Set which components to trace. By default it traces first three components. | ||
g |
Draw trace grid. By default is enabled. | ||
st |
Draw some statistics. By default is enabled. | ||
sc |
Draw scope. By default is enabled. |
Commands
This filter supports same commands as options. 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
• |
Inspect full first row of video frame. |
oscilloscope=x=0.5:y=0:s=1
• |
Inspect full last row of video frame. |
oscilloscope=x=0.5:y=1:s=1
• |
Inspect full 5th line of video frame of height 1080. |
oscilloscope=x=0.5:y=5/1080:s=1
• |
Inspect full last column of video frame. |
oscilloscope=x=1:y=0.5:s=1:t=1
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
See framesync.
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
See framesync.
format
Set the format for the output video.
It accepts the
following values:
yuv420
force YUV 4:2:0 8-bit planar output
yuv420p10
force YUV 4:2:0 10-bit planar output
yuv422
force YUV 4:2:2 8-bit planar output
yuv422p10
force YUV 4:2:2 10-bit planar output
yuv444
force YUV 4:4:4 8-bit planar output
yuv444p10
force YUV 4:4:4 10-bit planar output
rgb |
force RGB 8-bit packed output |
gbrp
force RGB 8-bit planar output
auto
automatically pick format
Default value is yuv420.
repeatlast
See framesync.
alpha
Set format of alpha of the overlaid video, it can be straight or premultiplied. Default is straight.
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; deprecated, do not use | ||
t |
The timestamp, expressed in seconds. It’s NAN if the input timestamp is unknown. |
This filter also supports the framesync options.
Note that the n, 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: |
color=color=red@.3: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]
overlay_cuda
Overlay one video on top of another.
This is the CUDA variant of the overlay filter. It only accepts CUDA frames. The underlying input pixel formats have to match.
It takes two inputs and has one output. The first input is the "main" video on which the second input is overlaid.
It accepts the following parameters:
x |
|||
y |
Set expressions for the x and y coordinates of the overlaid video on the main video. |
They 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. | ||
n |
The ordinal index of the main input frame, starting from 0. | ||
pos |
The byte offset position in the file of the main input frame, NAN if unknown. Deprecated, do not use. | ||
t |
The timestamp of the main input frame, expressed in seconds, NAN if unknown. |
Default value is "0" for both expressions.
eval
Set when the expressions for x and y are evaluated.
It accepts the
following values:
init
Evaluate expressions once during filter initialization or when a command is processed.
frame
Evaluate expressions for each incoming frame
Default value is frame.
eof_action
See framesync.
shortest
See framesync.
repeatlast
See framesync.
This filter also supports the framesync options.
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.
If x or y evaluate to a negative number, they’ll be changed so the input image is centered on the padded area.
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".
eval
Specify when to evaluate width, height, x and y 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.
aspect
Pad to aspect instead to a resolution.
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.
transparency_color
Set the color that will be used as background for transparency.
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)
sierra3
Frankie Sierra dithering v3 (error diffusion)
burkes
Burkes dithering (error diffusion)
atkinson
Atkinson dithering by Bill Atkinson at Apple Computer (error diffusion)
none
Disable dithering.
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. |
alpha_threshold
Sets the alpha threshold for transparency. Alpha values above this threshold will be treated as completely opaque, and values below this threshold will be treated as completely transparent.
The option must be an integer value in the range [0,255]. Default is 128.
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. |
Commands
This filter supports the all above options as commands.
photosensitivity
Reduce various flashes in video, so to help users with
epilepsy.
It accepts the
following options:
frames, f
Set how many frames to use when filtering. Default is 30.
threshold, t
Set detection threshold factor. Default is 1. Lower is stricter.
skip
Set how many pixels to skip when sampling frames. Default is 1. Allowed range is from 1 to 1024.
bypass
Leave frames unchanged. Default is disabled.
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.
pixelize
Apply pixelization to video stream.
The filter
accepts the following options:
width, w
height, h
Set block dimensions that will be used for pixelization. Default value is 16.
mode, m
Set the mode of pixelization used.
Possible values are:
avg |
||
min |
||
max |
Default value is "avg".
planes, p
Set what planes to filter. Default is to filter all planes.
Commands
This filter supports all options as commands.
pixscope
Display sample values of color channels. Mainly useful for
checking color and levels. Minimum supported resolution is
640x480.
The filters accept the following options:
x |
Set scope X position, relative offset on X axis. | ||
y |
Set scope Y position, relative offset on Y axis. | ||
w |
Set scope width. | ||
h |
Set scope height. | ||
o |
Set window opacity. This window also holds statistics about pixel area. | ||
wx |
Set window X position, relative offset on X axis. | ||
wy |
Set window Y position, relative offset on Y axis. |
Commands
This filter supports same commands as options.
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 luma filtering only (no chrominance).
n/noluma
Do chrominance filtering only (no luma).
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 luma 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 luma 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).
premultiply
Apply alpha premultiply effect to input video stream using
first plane of second stream as alpha.
Both streams must have same dimensions and same pixel format.
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.
inplace
Do not require 2nd input for processing, instead use alpha plane from input stream.
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.
Commands
This filter supports the all above options as commands.
pseudocolor
Alter frame colors in video with pseudocolors.
This filter accepts the following options:
c0 |
set pixel first component expression | ||
c1 |
set pixel second component expression | ||
c2 |
set pixel third component expression | ||
c3 |
set pixel fourth component expression, corresponds to the alpha component |
index, i
set component to use as base for altering colors
preset, p
Pick one of built-in LUTs. By default is set to none.
Available LUTs:
magma
inferno
plasma
viridis
turbo
cividis
range1
range2
shadows
highlights
solar
nominal
preferred
total
spectral
cool
heat
fiery
blues
green
helix
opacity
Set opacity of output colors. Allowed range is from 0 to 1. Default value is set to 1.
Each of the expression options specifies the expression to use for computing the lookup table for the corresponding pixel component values.
The expressions can contain the following constants and functions:
w |
||||
h |
The input width and height. |
|||
val |
The input value for the pixel component. |
ymin, umin, vmin, amin
The minimum allowed component value.
ymax, umax, vmax, amax
The maximum allowed component value.
All expressions default to "val".
Commands
This filter supports the all above options as commands.
Examples
• |
Change too high luma values to gradient: |
pseudocolor="'if(between(val,ymax,amax),lerp(ymin,ymax,(val-ymax)/(amax-ymax)),-1):if(between(val,ymax,amax),lerp(umax,umin,(val-ymax)/(amax-ymax)),-1):if(between(val,ymax,amax),lerp(vmin,vmax,(val-ymax)/(amax-ymax)),-1):-1'"
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.
This filter also supports the framesync options.
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_b, 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.
Examples
• |
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.
• |
Another example with different containers: |
ffmpeg -i main.mpg -i ref.mkv -lavfi "[0:v]settb=AVTB,setpts=PTS-STARTPTS[main];[1:v]settb=AVTB,setpts=PTS-STARTPTS[ref];[main][ref]psnr" -f null -
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 |
Sequential 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.
readeia608
Read closed captioning (EIA-608) information from the top
lines of a video frame.
This filter
adds frame metadata for "lavfi.readeia608.X.cc"
and "lavfi.readeia608.X.line", where "X"
is the number of the identified line with EIA-608 data
(starting from 0). A description of each metadata value
follows:
lavfi.readeia608.X.cc
The two bytes stored as EIA-608 data (printed in hexadecimal).
lavfi.readeia608.X.line
The number of the line on which the EIA-608 data was identified and read.
This filter
accepts the following options:
scan_min
Set the line to start scanning for EIA-608 data. Default is 0.
scan_max
Set the line to end scanning for EIA-608 data. Default is 29.
spw |
Set the ratio of width reserved for sync code detection. Default is 0.27. Allowed range is "[0.1 - 0.7]". | ||
chp |
Enable checking the parity bit. In the event of a parity error, the filter will output 0x00 for that character. Default is false. | ||
lp |
Lowpass lines prior to further processing. Default is enabled. |
Commands
This filter supports the all above options as commands.
Examples
• |
Output a csv with presentation time and the first two lines of identified EIA-608 captioning data. |
ffprobe -f lavfi -i movie=captioned_video.mov,readeia608 -show_entries frame=pts_time:frame_tags=lavfi.readeia608.0.cc,lavfi.readeia608.1.cc -of csv
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.
format
Specify pixel format of output from this filter. Can be "color" or "gray". Default is "color".
fill
Specify the color of the unmapped pixels. For the syntax of this option, check the "Color" section in the ffmpeg-utils manual. Default color is "black".
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 res |