FFMPEG-ALL - ffmpeg media converter - man(1) - [phpMan]

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FFMPEG-ALL(1) FFMPEG-ALL(1)

NAME
ffmpeg - ffmpeg media converter

SYNOPSIS
ffmpeg [global_options] {[input_file_options] -i input_url} ... {[output_file_options]
output_url} ...

DESCRIPTION
ffmpeg is a universal media converter. It can read a wide variety of inputs - including live
grabbing/recording devices - filter, and transcode them into a plethora of output formats.

ffmpeg reads from an arbitrary number of input "files" (which can be regular files, pipes,
network streams, grabbing devices, etc.), specified by the "-i" option, and writes to an
arbitrary number of output "files", which are specified by a plain output url. Anything found
on the command line which cannot be interpreted as an option is considered to be an output
url.

Each input or output url can, in principle, contain any number of streams of different types
(video/audio/subtitle/attachment/data). The allowed number and/or types of streams may be
limited by the container format. Selecting which streams from which inputs will go into which
output is either done automatically or with the "-map" option (see the Stream selection
chapter).

To refer to input files in options, you must use their indices (0-based). E.g. the first
input file is 0, the second is 1, etc. Similarly, streams within a file are referred to by
their indices. E.g. "2:3" refers to the fourth stream in the third input file. Also see the
Stream specifiers chapter.

As a general rule, options are applied to the next specified file. Therefore, order is
important, and you can have the same option on the command line multiple times. Each
occurrence is then applied to the next input or output file. Exceptions from this rule are
the global options (e.g. verbosity level), which should be specified first.

Do not mix input and output files -- first specify all input files, then all output files.
Also do not mix options which belong to different files. All options apply ONLY to the next
input or output file and are reset between files.

Some simple examples follow.

• Convert an input media file to a different format, by re-encoding media streams:

ffmpeg -i input.avi output.mp4

• Set the video bitrate of the output file to 64 kbit/s:

ffmpeg -i input.avi -b:v 64k -bufsize 64k output.mp4

• Force the frame rate of the output file to 24 fps:

ffmpeg -i input.avi -r 24 output.mp4

• Force the frame rate of the input file (valid for raw formats only) to 1 fps and the
frame rate of the output file to 24 fps:

ffmpeg -r 1 -i input.m2v -r 24 output.mp4

The format option may be needed for raw input files.

DETAILED DESCRIPTION
The transcoding process in ffmpeg for each output can be described by the following diagram:

_______ ______________
| | | |
| input | demuxer | encoded data | decoder
| file | ---------> | packets | -----+
|_______| |______________| |
v
_________
| |
| decoded |
| frames |
|_________|
________ ______________ |
| | | | |
| output | <-------- | encoded data | <----+
| file | muxer | packets | encoder
|________| |______________|

ffmpeg calls the libavformat library (containing demuxers) to read input files and get
packets containing encoded data from them. When there are multiple input files, ffmpeg tries
to keep them synchronized by tracking lowest timestamp on any active input stream.

Encoded packets are then passed to the decoder (unless streamcopy is selected for the stream,
see further for a description). The decoder produces uncompressed frames (raw video/PCM
audio/...) which can be processed further by filtering (see next section). After filtering,
the frames are passed to the encoder, which encodes them and outputs encoded packets. Finally
those are passed to the muxer, which writes the encoded packets to the output file.

Filtering
Before encoding, ffmpeg can process raw audio and video frames using filters from the
libavfilter library. Several chained filters form a filter graph. ffmpeg distinguishes
between two types of filtergraphs: simple and complex.

Simple filtergraphs

Simple filtergraphs are those that have exactly one input and output, both of the same type.
In the above diagram they can be represented by simply inserting an additional step between
decoding and encoding:

_________ ______________
| | | |
| decoded | | encoded data |
| frames |\ _ | packets |
|_________| \ /||______________|
\ __________ /
simple _\|| | / encoder
filtergraph | filtered |/
| frames |
|__________|

Simple filtergraphs are configured with the per-stream -filter option (with -vf and -af
aliases for video and audio respectively). A simple filtergraph for video can look for
example like this:

_______ _____________ _______ ________
| | | | | | | |
| input | ---> | deinterlace | ---> | scale | ---> | output |
|_______| |_____________| |_______| |________|

Note that some filters change frame properties but not frame contents. E.g. the "fps" filter
in the example above changes number of frames, but does not touch the frame contents. Another
example is the "setpts" filter, which only sets timestamps and otherwise passes the frames
unchanged.

Complex filtergraphs

Complex filtergraphs are those which cannot be described as simply a linear processing chain
applied to one stream. This is the case, for example, when the graph has more than one input
and/or output, or when output stream type is different from input. They can be represented
with the following diagram:

_________
| |
| input 0 |\ __________
|_________| \ | |
\ _________ /| output 0 |
\ | | / |__________|
_________ \| complex | /
| | | |/
| input 1 |---->| filter |\
|_________| | | \ __________
/| graph | \ | |
/ | | \| output 1 |
_________ / |_________| |__________|
| | /
| input 2 |/
|_________|

Complex filtergraphs are configured with the -filter_complex option. Note that this option
is global, since a complex filtergraph, by its nature, cannot be unambiguously associated
with a single stream or file.

The -lavfi option is equivalent to -filter_complex.

A trivial example of a complex filtergraph is the "overlay" filter, which has two video
inputs and one video output, containing one video overlaid on top of the other. Its audio
counterpart is the "amix" filter.

Stream copy
Stream copy is a mode selected by supplying the "copy" parameter to the -codec option. It
makes ffmpeg omit the decoding and encoding step for the specified stream, so it does only
demuxing and muxing. It is useful for changing the container format or modifying container-
level metadata. The diagram above will, in this case, simplify to this:

_______ ______________ ________
| | | | | |
| input | demuxer | encoded data | muxer | output |
| file | ---------> | packets | -------> | file |
|_______| |______________| |________|

Since there is no decoding or encoding, it is very fast and there is no quality loss.
However, it might not work in some cases because of many factors. Applying filters is
obviously also impossible, since filters work on uncompressed data.

STREAM SELECTION
ffmpeg provides the "-map" option for manual control of stream selection in each output file.
Users can skip "-map" and let ffmpeg perform automatic stream selection as described below.
The "-vn / -an / -sn / -dn" options can be used to skip inclusion of video, audio, subtitle
and data streams respectively, whether manually mapped or automatically selected, except for
those streams which are outputs of complex filtergraphs.

Description
The sub-sections that follow describe the various rules that are involved in stream
selection. The examples that follow next show how these rules are applied in practice.

While every effort is made to accurately reflect the behavior of the program, FFmpeg is under
continuous development and the code may have changed since the time of this writing.

Automatic stream selection

In the absence of any map options for a particular output file, ffmpeg inspects the output
format to check which type of streams can be included in it, viz. video, audio and/or
subtitles. For each acceptable stream type, ffmpeg will pick one stream, when available, from
among all the inputs.

It will select that stream based upon the following criteria:

• for video, it is the stream with the highest resolution,

• for audio, it is the stream with the most channels,

• for subtitles, it is the first subtitle stream found but there's a caveat. The output
format's default subtitle encoder can be either text-based or image-based, and only a
subtitle stream of the same type will be chosen.

In the case where several streams of the same type rate equally, the stream with the lowest
index is chosen.

Data or attachment streams are not automatically selected and can only be included using
"-map".

Manual stream selection

When "-map" is used, only user-mapped streams are included in that output file, with one
possible exception for filtergraph outputs described below.

Complex filtergraphs

If there are any complex filtergraph output streams with unlabeled pads, they will be added
to the first output file. This will lead to a fatal error if the stream type is not supported
by the output format. In the absence of the map option, the inclusion of these streams leads
to the automatic stream selection of their types being skipped. If map options are present,
these filtergraph streams are included in addition to the mapped streams.

Complex filtergraph output streams with labeled pads must be mapped once and exactly once.

Stream handling

Stream handling is independent of stream selection, with an exception for subtitles described
below. Stream handling is set via the "-codec" option addressed to streams within a specific
output file. In particular, codec options are applied by ffmpeg after the stream selection
process and thus do not influence the latter. If no "-codec" option is specified for a stream
type, ffmpeg will select the default encoder registered by the output file muxer.

An exception exists for subtitles. If a subtitle encoder is specified for an output file, the
first subtitle stream found of any type, text or image, will be included. ffmpeg does not
validate if the specified encoder can convert the selected stream or if the converted stream
is acceptable within the output format. This applies generally as well: when the user sets an
encoder manually, the stream selection process cannot check if the encoded stream can be
muxed into the output file. If it cannot, ffmpeg will abort and all output files will fail
to be processed.

Examples
The following examples illustrate the behavior, quirks and limitations of ffmpeg's stream
selection methods.

They assume the following three input files.

input file 'A.avi'
stream 0: video 640x360
stream 1: audio 2 channels

input file 'B.mp4'
stream 0: video 1920x1080
stream 1: audio 2 channels
stream 2: subtitles (text)
stream 3: audio 5.1 channels
stream 4: subtitles (text)

input file 'C.mkv'
stream 0: video 1280x720
stream 1: audio 2 channels
stream 2: subtitles (image)

Example: automatic stream selection

ffmpeg -i A.avi -i B.mp4 out1.mkv out2.wav -map 1:a -c:a copy out3.mov

There are three output files specified, and for the first two, no "-map" options are set, so
ffmpeg will select streams for these two files automatically.

out1.mkv is a Matroska container file and accepts video, audio and subtitle streams, so
ffmpeg will try to select one of each type.For video, it will select "stream 0" from B.mp4,
which has the highest resolution among all the input video streams.For audio, it will select
"stream 3" from B.mp4, since it has the greatest number of channels.For subtitles, it will
select "stream 2" from B.mp4, which is the first subtitle stream from among A.avi and B.mp4.

out2.wav accepts only audio streams, so only "stream 3" from B.mp4 is selected.

For out3.mov, since a "-map" option is set, no automatic stream selection will occur. The
"-map 1:a" option will select all audio streams from the second input B.mp4. No other streams
will be included in this output file.

For the first two outputs, all included streams will be transcoded. The encoders chosen will
be the default ones registered by each output format, which may not match the codec of the
selected input streams.

For the third output, codec option for audio streams has been set to "copy", so no decoding-
filtering-encoding operations will occur, or can occur. Packets of selected streams shall be
conveyed from the input file and muxed within the output file.

Example: automatic subtitles selection

ffmpeg -i C.mkv out1.mkv -c:s dvdsub -an out2.mkv

Although out1.mkv is a Matroska container file which accepts subtitle streams, only a video
and audio stream shall be selected. The subtitle stream of C.mkv is image-based and the
default subtitle encoder of the Matroska muxer is text-based, so a transcode operation for
the subtitles is expected to fail and hence the stream isn't selected. However, in out2.mkv,
a subtitle encoder is specified in the command and so, the subtitle stream is selected, in
addition to the video stream. The presence of "-an" disables audio stream selection for
out2.mkv.

Example: unlabeled filtergraph outputs

ffmpeg -i A.avi -i C.mkv -i B.mp4 -filter_complex "overlay" out1.mp4 out2.srt

A filtergraph is setup here using the "-filter_complex" option and consists of a single video
filter. The "overlay" filter requires exactly two video inputs, but none are specified, so
the first two available video streams are used, those of A.avi and C.mkv. The output pad of
the filter has no label and so is sent to the first output file out1.mp4. Due to this,
automatic selection of the video stream is skipped, which would have selected the stream in
B.mp4. The audio stream with most channels viz. "stream 3" in B.mp4, is chosen automatically.
No subtitle stream is chosen however, since the MP4 format has no default subtitle encoder
registered, and the user hasn't specified a subtitle encoder.

The 2nd output file, out2.srt, only accepts text-based subtitle streams. So, even though the
first subtitle stream available belongs to C.mkv, it is image-based and hence skipped. The
selected stream, "stream 2" in B.mp4, is the first text-based subtitle stream.

Example: labeled filtergraph outputs

ffmpeg -i A.avi -i B.mp4 -i C.mkv -filter_complex "[1:v]hue=s=0[outv];overlay;aresample" \
-map '[outv]' -an out1.mp4 \
out2.mkv \
-map '[outv]' -map 1:a:0 out3.mkv

The above command will fail, as the output pad labelled "[outv]" has been mapped twice. None
of the output files shall be processed.

ffmpeg -i A.avi -i B.mp4 -i C.mkv -filter_complex "[1:v]hue=s=0[outv];overlay;aresample" \
-an out1.mp4 \
out2.mkv \
-map 1:a:0 out3.mkv

This command above will also fail as the hue filter output has a label, "[outv]", and hasn't
been mapped anywhere.

The command should be modified as follows,

ffmpeg -i A.avi -i B.mp4 -i C.mkv -filter_complex "[1:v]hue=s=0,split=2[outv1][outv2];overlay;aresample" \
-map '[outv1]' -an out1.mp4 \
out2.mkv \
-map '[outv2]' -map 1:a:0 out3.mkv

The video stream from B.mp4 is sent to the hue filter, whose output is cloned once using the
split filter, and both outputs labelled. Then a copy each is mapped to the first and third
output files.

The overlay filter, requiring two video inputs, uses the first two unused video streams.
Those are the streams from A.avi and C.mkv. The overlay output isn't labelled, so it is sent
to the first output file out1.mp4, regardless of the presence of the "-map" option.

The aresample filter is sent the first unused audio stream, that of A.avi. Since this filter
output is also unlabelled, it too is mapped to the first output file. The presence of "-an"
only suppresses automatic or manual stream selection of audio streams, not outputs sent from
filtergraphs. Both these mapped streams shall be ordered before the mapped stream in
out1.mp4.

The video, audio and subtitle streams mapped to "out2.mkv" are entirely determined by
automatic stream selection.

out3.mkv consists of the cloned video output from the hue filter and the first audio stream
from B.mp4.

OPTIONS
All the numerical options, if not specified otherwise, accept a string representing a number
as input, which may be followed by one of the SI unit prefixes, for example: 'K', 'M', or
'G'.

If 'i' is appended to the SI unit prefix, the complete prefix will be interpreted as a unit
prefix for binary multiples, which are based on powers of 1024 instead of powers of 1000.
Appending 'B' to the SI unit prefix multiplies the value by 8. This allows using, for
example: 'KB', 'MiB', 'G' and 'B' as number suffixes.

Options which do not take arguments are boolean options, and set the corresponding value to
true. They can be set to false by prefixing the option name with "no". For example using
"-nofoo" will set the boolean option with name "foo" to false.

Stream specifiers
Some options are applied per-stream, e.g. bitrate or codec. Stream specifiers are used to
precisely specify which stream(s) a given option belongs to.

A stream specifier is a string generally appended to the option name and separated from it by
a colon. E.g. "-codec:a:1 ac3" contains the "a:1" stream specifier, which matches the second
audio stream. Therefore, it would select the ac3 codec for the second audio stream.

A stream specifier can match several streams, so that the option is applied to all of them.
E.g. the stream specifier in "-b:a 128k" matches all audio streams.

An empty stream specifier matches all streams. For example, "-codec copy" or "-codec: copy"
would copy all the streams without reencoding.

Possible forms of stream specifiers are:

stream_index
Matches the stream with this index. E.g. "-threads:1 4" would set the thread count for
the second stream to 4. If stream_index is used as an additional stream specifier (see
below), then it selects stream number stream_index from the matching streams. Stream
numbering is based on the order of the streams as detected by libavformat except when a
program ID is also specified. In this case it is based on the ordering of the streams in
the program.

stream_type[:additional_stream_specifier]
stream_type is one of following: 'v' or 'V' for video, 'a' for audio, 's' for subtitle,
'd' for data, and 't' for attachments. 'v' matches all video streams, 'V' only matches
video streams which are not attached pictures, video thumbnails or cover arts. If
additional_stream_specifier is used, then it matches streams which both have this type
and match the additional_stream_specifier. Otherwise, it matches all streams of the
specified type.

p:program_id[:additional_stream_specifier]
Matches streams which are in the program with the id program_id. If
additional_stream_specifier is used, then it matches streams which both are part of the
program and match the additional_stream_specifier.

#stream_id or i:stream_id
Match the stream by stream id (e.g. PID in MPEG-TS container).

m:key[:value]
Matches streams with the metadata tag key having the specified value. If value is not
given, matches streams that contain the given tag with any value.

u Matches streams with usable configuration, the codec must be defined and the essential
information such as video dimension or audio sample rate must be present.

Note that in ffmpeg, matching by metadata will only work properly for input files.

Generic options
These options are shared amongst the ff* tools.

-L Show license.

-h, -?, -help, --help [arg]
Show help. An optional parameter may be specified to print help about a specific item. If
no argument is specified, only basic (non advanced) tool options are shown.

Possible values of arg are:

long
Print advanced tool options in addition to the basic tool options.

full
Print complete list of options, including shared and private options for encoders,
decoders, demuxers, muxers, filters, etc.

decoder=decoder_name
Print detailed information about the decoder named decoder_name. Use the -decoders
option to get a list of all decoders.

encoder=encoder_name
Print detailed information about the encoder named encoder_name. Use the -encoders
option to get a list of all encoders.

demuxer=demuxer_name
Print detailed information about the demuxer named demuxer_name. Use the -formats
option to get a list of all demuxers and muxers.

muxer=muxer_name
Print detailed information about the muxer named muxer_name. Use the -formats option
to get a list of all muxers and demuxers.

filter=filter_name
Print detailed information about the filter named filter_name. Use the -filters
option to get a list of all filters.

bsf=bitstream_filter_name
Print detailed information about the bitstream filter named bitstream_filter_name.
Use the -bsfs option to get a list of all bitstream filters.

protocol=protocol_name
Print detailed information about the protocol named protocol_name. Use the
-protocols option to get a list of all protocols.

-version
Show version.

-buildconf
Show the build configuration, one option per line.

-formats
Show available formats (including devices).

-demuxers
Show available demuxers.

-muxers
Show available muxers.

-devices
Show available devices.

-codecs
Show all codecs known to libavcodec.

Note that the term 'codec' is used throughout this documentation as a shortcut for what
is more correctly called a media bitstream format.

-decoders
Show available decoders.

-encoders
Show all available encoders.

-bsfs
Show available bitstream filters.

-protocols
Show available protocols.

-filters
Show available libavfilter filters.

-pix_fmts
Show available pixel formats.

-sample_fmts
Show available sample formats.

-layouts
Show channel names and standard channel layouts.

-dispositions
Show stream dispositions.

-colors
Show recognized color names.

-sources device[,opt1=val1[,opt2=val2]...]
Show autodetected sources of the input device. Some devices may provide system-dependent
source names that cannot be autodetected. The returned list cannot be assumed to be
always complete.

ffmpeg -sources pulse,server=192.168.0.4

-sinks device[,opt1=val1[,opt2=val2]...]
Show autodetected sinks of the output device. Some devices may provide system-dependent
sink names that cannot be autodetected. The returned list cannot be assumed to be always
complete.

ffmpeg -sinks pulse,server=192.168.0.4

-loglevel [flags+]loglevel | -v [flags+]loglevel
Set logging level and flags used by the library.

The optional flags prefix can consist of the following values:

repeat
Indicates that repeated log output should not be compressed to the first line and the
"Last message repeated n times" line will be omitted.

level
Indicates that log output should add a "[level]" prefix to each message line. This
can be used as an alternative to log coloring, e.g. when dumping the log to file.

Flags can also be used alone by adding a '+'/'-' prefix to set/reset a single flag
without affecting other flags or changing loglevel. When setting both flags and loglevel,
a '+' separator is expected between the last flags value and before loglevel.

loglevel is a string or a number containing one of the following values:

quiet, -8
Show nothing at all; be silent.

panic, 0
Only show fatal errors which could lead the process to crash, such as an assertion
failure. This is not currently used for anything.

fatal, 8
Only show fatal errors. These are errors after which the process absolutely cannot
continue.

error, 16
Show all errors, including ones which can be recovered from.

warning, 24
Show all warnings and errors. Any message related to possibly incorrect or unexpected
events will be shown.

info, 32
Show informative messages during processing. This is in addition to warnings and
errors. This is the default value.

verbose, 40
Same as "info", except more verbose.

debug, 48
Show everything, including debugging information.

trace, 56

For example to enable repeated log output, add the "level" prefix, and set loglevel to
"verbose":

ffmpeg -loglevel repeat+level+verbose -i input output

Another example that enables repeated log output without affecting current state of
"level" prefix flag or loglevel:

ffmpeg [...] -loglevel +repeat

By default the program logs to stderr. If coloring is supported by the terminal, colors
are used to mark errors and warnings. Log coloring can be disabled setting the
environment variable AV_LOG_FORCE_NOCOLOR, or can be forced setting the environment
variable AV_LOG_FORCE_COLOR.

-report
Dump full command line and log output to a file named "program-YYYYMMDD-HHMMSS.log" in
the current directory. This file can be useful for bug reports. It also implies
"-loglevel debug".

Setting the environment variable FFREPORT to any value has the same effect. If the value
is a ':'-separated key=value sequence, these options will affect the report; option
values must be escaped if they contain special characters or the options delimiter ':'
(see the ``Quoting and escaping'' section in the ffmpeg-utils manual).

The following options are recognized:

file
set the file name to use for the report; %p is expanded to the name of the program,
%t is expanded to a timestamp, "%%" is expanded to a plain "%"

level
set the log verbosity level using a numerical value (see "-loglevel").

For example, to output a report to a file named ffreport.log using a log level of 32
(alias for log level "info"):

FFREPORT=file=ffreport.log:level=32 ffmpeg -i input output

Errors in parsing the environment variable are not fatal, and will not appear in the
report.

-hide_banner
Suppress printing banner.

All FFmpeg tools will normally show a copyright notice, build options and library
versions. This option can be used to suppress printing this information.

-cpuflags flags (global)
Allows setting and clearing cpu flags. This option is intended for testing. Do not use it
unless you know what you're doing.

ffmpeg -cpuflags -sse+mmx ...
ffmpeg -cpuflags mmx ...
ffmpeg -cpuflags 0 ...

Possible flags for this option are:

x86
mmx
mmxext
sse
sse2
sse2slow
sse3
sse3slow
ssse3
atom
sse4.1
sse4.2
avx
avx2
xop
fma3
fma4
3dnow
3dnowext
bmi1
bmi2
cmov
ARM
armv5te
armv6
armv6t2
vfp
vfpv3
neon
setend
AArch64
armv8
vfp
neon
PowerPC
altivec
Specific Processors
pentium2
pentium3
pentium4
k6
k62
athlon
athlonxp
k8
-cpucount count (global)
Override detection of CPU count. This option is intended for testing. Do not use it
unless you know what you're doing.

ffmpeg -cpucount 2

-max_alloc bytes
Set the maximum size limit for allocating a block on the heap by ffmpeg's family of
malloc functions. Exercise extreme caution when using this option. Don't use if you do
not understand the full consequence of doing so. Default is INT_MAX.

AVOptions
These options are provided directly by the libavformat, libavdevice and libavcodec libraries.
To see the list of available AVOptions, use the -help option. They are separated into two
categories:

generic
These options can be set for any container, codec or device. Generic options are listed
under AVFormatContext options for containers/devices and under AVCodecContext options for
codecs.

private
These options are specific to the given container, device or codec. Private options are
listed under their corresponding containers/devices/codecs.

For example to write an ID3v2.3 header instead of a default ID3v2.4 to an MP3 file, use the
id3v2_version private option of the MP3 muxer:

ffmpeg -i input.flac -id3v2_version 3 out.mp3

All codec AVOptions are per-stream, and thus a stream specifier should be attached to them:

ffmpeg -i multichannel.mxf -map 0:v:0 -map 0:a:0 -map 0:a:0 -c:a:0 ac3 -b:a:0 640k -ac:a:1 2 -c:a:1 aac -b:2 128k out.mp4

In the above example, a multichannel audio stream is mapped twice for output. The first
instance is encoded with codec ac3 and bitrate 640k. The second instance is downmixed to 2
channels and encoded with codec aac. A bitrate of 128k is specified for it using absolute
index of the output stream.

Note: the -nooption syntax cannot be used for boolean AVOptions, use -option 0/-option 1.

Note: the old undocumented way of specifying per-stream AVOptions by prepending v/a/s to the
options name is now obsolete and will be removed soon.

Main options
-f fmt (input/output)
Force input or output file format. The format is normally auto detected for input files
and guessed from the file extension for output files, so this option is not needed in
most cases.

-i url (input)
input file url

-y (global)
Overwrite output files without asking.

-n (global)
Do not overwrite output files, and exit immediately if a specified output file already
exists.

-stream_loop number (input)
Set number of times input stream shall be looped. Loop 0 means no loop, loop -1 means
infinite loop.

-recast_media (global)
Allow forcing a decoder of a different media type than the one detected or designated by
the demuxer. Useful for decoding media data muxed as data streams.

-c[:stream_specifier] codec (input/output,per-stream)
-codec[:stream_specifier] codec (input/output,per-stream)
Select an encoder (when used before an output file) or a decoder (when used before an
input file) for one or more streams. codec is the name of a decoder/encoder or a special
value "copy" (output only) to indicate that the stream is not to be re-encoded.

For example

ffmpeg -i INPUT -map 0 -c:v libx264 -c:a copy OUTPUT

encodes all video streams with libx264 and copies all audio streams.

For each stream, the last matching "c" option is applied, so

ffmpeg -i INPUT -map 0 -c copy -c:v:1 libx264 -c:a:137 libvorbis OUTPUT

will copy all the streams except the second video, which will be encoded with libx264,
and the 138th audio, which will be encoded with libvorbis.

-t duration (input/output)
When used as an input option (before "-i"), limit the duration of data read from the
input file.

When used as an output option (before an output url), stop writing the output after its
duration reaches duration.

duration must be a time duration specification, see the Time duration section in the
ffmpeg-utils(1) manual.

-to and -t are mutually exclusive and -t has priority.

-to position (input/output)
Stop writing the output or reading the input at position. position must be a time
duration specification, see the Time duration section in the ffmpeg-utils(1) manual.

-to and -t are mutually exclusive and -t has priority.

-fs limit_size (output)
Set the file size limit, expressed in bytes. No further chunk of bytes is written after
the limit is exceeded. The size of the output file is slightly more than the requested
file size.

-ss position (input/output)
When used as an input option (before "-i"), seeks in this input file to position. Note
that in most formats it is not possible to seek exactly, so ffmpeg will seek to the
closest seek point before position. When transcoding and -accurate_seek is enabled (the
default), this extra segment between the seek point and position will be decoded and
discarded. When doing stream copy or when -noaccurate_seek is used, it will be preserved.

When used as an output option (before an output url), decodes but discards input until
the timestamps reach position.

position must be a time duration specification, see the Time duration section in the
ffmpeg-utils(1) manual.

-sseof position (input)
Like the "-ss" option but relative to the "end of file". That is negative values are
earlier in the file, 0 is at EOF.

-isync input_index (input)
Assign an input as a sync source.

This will take the difference between the start times of the target and reference inputs
and offset the timestamps of the target file by that difference. The source timestamps of
the two inputs should derive from the same clock source for expected results. If "copyts"
is set then "start_at_zero" must also be set. If either of the inputs has no starting
timestamp then no sync adjustment is made.

Acceptable values are those that refer to a valid ffmpeg input index. If the sync
reference is the target index itself or -1, then no adjustment is made to target
timestamps. A sync reference may not itself be synced to any other input.

Default value is -1.

-itsoffset offset (input)
Set the input time offset.

offset must be a time duration specification, see the Time duration section in the
ffmpeg-utils(1) manual.

The offset is added to the timestamps of the input files. Specifying a positive offset
means that the corresponding streams are delayed by the time duration specified in
offset.

-itsscale scale (input,per-stream)
Rescale input timestamps. scale should be a floating point number.

-timestamp date (output)
Set the recording timestamp in the container.

date must be a date specification, see the Date section in the ffmpeg-utils(1) manual.

-metadata[:metadata_specifier] key=value (output,per-metadata)
Set a metadata key/value pair.

An optional metadata_specifier may be given to set metadata on streams, chapters or
programs. See "-map_metadata" documentation for details.

This option overrides metadata set with "-map_metadata". It is also possible to delete
metadata by using an empty value.

For example, for setting the title in the output file:

ffmpeg -i in.avi -metadata title="my title" out.flv

To set the language of the first audio stream:

ffmpeg -i INPUT -metadata:s:a:0 language=eng OUTPUT

-disposition[:stream_specifier] value (output,per-stream)
Sets the disposition for a stream.

By default, the disposition is copied from the input stream, unless the output stream
this option applies to is fed by a complex filtergraph - in that case the disposition is
unset by default.

value is a sequence of items separated by '+' or '-'. The first item may also be prefixed
with '+' or '-', in which case this option modifies the default value. Otherwise (the
first item is not prefixed) this options overrides the default value. A '+' prefix adds
the given disposition, '-' removes it. It is also possible to clear the disposition by
setting it to 0.

If no "-disposition" options were specified for an output file, ffmpeg will automatically
set the 'default' disposition on the first stream of each type, when there are multiple
streams of this type in the output file and no stream of that type is already marked as
default.

The "-dispositions" option lists the known dispositions.

For example, to make the second audio stream the default stream:

ffmpeg -i in.mkv -c copy -disposition:a:1 default out.mkv

To make the second subtitle stream the default stream and remove the default disposition
from the first subtitle stream:

ffmpeg -i in.mkv -c copy -disposition:s:0 0 -disposition:s:1 default out.mkv

To add an embedded cover/thumbnail:

ffmpeg -i in.mp4 -i IMAGE -map 0 -map 1 -c copy -c:v:1 png -disposition:v:1 attached_pic out.mp4

Not all muxers support embedded thumbnails, and those who do, only support a few formats,
like JPEG or PNG.

-program [title=title:][program_num=program_num:]st=stream[:st=stream...] (output)
Creates a program with the specified title, program_num and adds the specified stream(s)
to it.

-target type (output)
Specify target file type ("vcd", "svcd", "dvd", "dv", "dv50"). type may be prefixed with
"pal-", "ntsc-" or "film-" to use the corresponding standard. All the format options
(bitrate, codecs, buffer sizes) are then set automatically. You can just type:

ffmpeg -i myfile.avi -target vcd /tmp/vcd.mpg

Nevertheless you can specify additional options as long as you know they do not conflict
with the standard, as in:

ffmpeg -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg

The parameters set for each target are as follows.

VCD

<pal>:
-f vcd -muxrate 1411200 -muxpreload 0.44 -packetsize 2324
-s 352x288 -r 25
-codec:v mpeg1video -g 15 -b:v 1150k -maxrate:v 1150k -minrate:v 1150k -bufsize:v 327680
-ar 44100 -ac 2
-codec:a mp2 -b:a 224k

<ntsc>:
-f vcd -muxrate 1411200 -muxpreload 0.44 -packetsize 2324
-s 352x240 -r 30000/1001
-codec:v mpeg1video -g 18 -b:v 1150k -maxrate:v 1150k -minrate:v 1150k -bufsize:v 327680
-ar 44100 -ac 2
-codec:a mp2 -b:a 224k

<film>:
-f vcd -muxrate 1411200 -muxpreload 0.44 -packetsize 2324
-s 352x240 -r 24000/1001
-codec:v mpeg1video -g 18 -b:v 1150k -maxrate:v 1150k -minrate:v 1150k -bufsize:v 327680
-ar 44100 -ac 2
-codec:a mp2 -b:a 224k

SVCD

<pal>:
-f svcd -packetsize 2324
-s 480x576 -pix_fmt yuv420p -r 25
-codec:v mpeg2video -g 15 -b:v 2040k -maxrate:v 2516k -minrate:v 0 -bufsize:v 1835008 -scan_offset 1
-ar 44100
-codec:a mp2 -b:a 224k

<ntsc>:
-f svcd -packetsize 2324
-s 480x480 -pix_fmt yuv420p -r 30000/1001
-codec:v mpeg2video -g 18 -b:v 2040k -maxrate:v 2516k -minrate:v 0 -bufsize:v 1835008 -scan_offset 1
-ar 44100
-codec:a mp2 -b:a 224k

<film>:
-f svcd -packetsize 2324
-s 480x480 -pix_fmt yuv420p -r 24000/1001
-codec:v mpeg2video -g 18 -b:v 2040k -maxrate:v 2516k -minrate:v 0 -bufsize:v 1835008 -scan_offset 1
-ar 44100
-codec:a mp2 -b:a 224k

DVD

<pal>:
-f dvd -muxrate 10080k -packetsize 2048
-s 720x576 -pix_fmt yuv420p -r 25
-codec:v mpeg2video -g 15 -b:v 6000k -maxrate:v 9000k -minrate:v 0 -bufsize:v 1835008
-ar 48000
-codec:a ac3 -b:a 448k

<ntsc>:
-f dvd -muxrate 10080k -packetsize 2048
-s 720x480 -pix_fmt yuv420p -r 30000/1001
-codec:v mpeg2video -g 18 -b:v 6000k -maxrate:v 9000k -minrate:v 0 -bufsize:v 1835008
-ar 48000
-codec:a ac3 -b:a 448k

<film>:
-f dvd -muxrate 10080k -packetsize 2048
-s 720x480 -pix_fmt yuv420p -r 24000/1001
-codec:v mpeg2video -g 18 -b:v 6000k -maxrate:v 9000k -minrate:v 0 -bufsize:v 1835008
-ar 48000
-codec:a ac3 -b:a 448k

<pal>:
-f dv
-s 720x576 -pix_fmt yuv420p -r 25
-ar 48000 -ac 2

<ntsc>:
-f dv
-s 720x480 -pix_fmt yuv411p -r 30000/1001
-ar 48000 -ac 2

<film>:
-f dv
-s 720x480 -pix_fmt yuv411p -r 24000/1001
-ar 48000 -ac 2

The "dv50" target is identical to the "dv" target except that the pixel format set is
"yuv422p" for all three standards.

Any user-set value for a parameter above will override the target preset value. In that
case, the output may not comply with the target standard.

-dn (input/output)
As an input option, blocks all data streams of a file from being filtered or being
automatically selected or mapped for any output. See "-discard" option to disable streams
individually.

As an output option, disables data recording i.e. automatic selection or mapping of any
data stream. For full manual control see the "-map" option.

-dframes number (output)
Set the number of data frames to output. This is an obsolete alias for "-frames:d", which
you should use instead.

-frames[:stream_specifier] framecount (output,per-stream)
Stop writing to the stream after framecount frames.

-q[:stream_specifier] q (output,per-stream)
-qscale[:stream_specifier] q (output,per-stream)
Use fixed quality scale (VBR). The meaning of q/qscale is codec-dependent. If qscale is
used without a stream_specifier then it applies only to the video stream, this is to
maintain compatibility with previous behavior and as specifying the same codec specific
value to 2 different codecs that is audio and video generally is not what is intended
when no stream_specifier is used.

-filter[:stream_specifier] filtergraph (output,per-stream)
Create the filtergraph specified by filtergraph and use it to filter the stream.

filtergraph is a description of the filtergraph to apply to the stream, and must have a
single input and a single output of the same type of the stream. In the filtergraph, the
input is associated to the label "in", and the output to the label "out". See the ffmpeg-
filters manual for more information about the filtergraph syntax.

See the -filter_complex option if you want to create filtergraphs with multiple inputs
and/or outputs.

-filter_script[:stream_specifier] filename (output,per-stream)
This option is similar to -filter, the only difference is that its argument is the name
of the file from which a filtergraph description is to be read.

-reinit_filter[:stream_specifier] integer (input,per-stream)
This boolean option determines if the filtergraph(s) to which this stream is fed gets
reinitialized when input frame parameters change mid-stream. This option is enabled by
default as most video and all audio filters cannot handle deviation in input frame
properties. Upon reinitialization, existing filter state is lost, like e.g. the frame
count "n" reference available in some filters. Any frames buffered at time of
reinitialization are lost. The properties where a change triggers reinitialization are,
for video, frame resolution or pixel format; for audio, sample format, sample rate,
channel count or channel layout.

-filter_threads nb_threads (global)
Defines how many threads are used to process a filter pipeline. Each pipeline will
produce a thread pool with this many threads available for parallel processing. The
default is the number of available CPUs.

-pre[:stream_specifier] preset_name (output,per-stream)
Specify the preset for matching stream(s).

-stats (global)
Print encoding progress/statistics. It is on by default, to explicitly disable it you
need to specify "-nostats".

-stats_period time (global)
Set period at which encoding progress/statistics are updated. Default is 0.5 seconds.

-progress url (global)
Send program-friendly progress information to url.

Progress information is written periodically and at the end of the encoding process. It
is made of "key=value" lines. key consists of only alphanumeric characters. The last key
of a sequence of progress information is always "progress".

The update period is set using "-stats_period".

-stdin
Enable interaction on standard input. On by default unless standard input is used as an
input. To explicitly disable interaction you need to specify "-nostdin".

Disabling interaction on standard input is useful, for example, if ffmpeg is in the
background process group. Roughly the same result can be achieved with "ffmpeg ... <
/dev/null" but it requires a shell.

-debug_ts (global)
Print timestamp information. It is off by default. This option is mostly useful for
testing and debugging purposes, and the output format may change from one version to
another, so it should not be employed by portable scripts.

See also the hash and framehash muxers.

tee
The tee muxer can be used to write the same data to several outputs, such as files or
streams. It can be used, for example, to stream a video over a network and save it to disk
at the same time.

It is different from specifying several outputs to the ffmpeg command-line tool. With the tee
muxer, the audio and video data will be encoded only once. With conventional multiple
outputs, multiple encoding operations in parallel are initiated, which can be a very
expensive process. The tee muxer is not useful when using the libavformat API directly
because it is then possible to feed the same packets to several muxers directly.

Since the tee muxer does not represent any particular output format, ffmpeg cannot auto-
select output streams. So all streams intended for output must be specified using "-map". See
the examples below.

Some encoders may need different options depending on the output format; the auto-detection
of this can not work with the tee muxer, so they need to be explicitly specified. The main
example is the global_header flag.

The slave outputs are specified in the file name given to the muxer, separated by '|'. If any
of the slave name contains the '|' separator, leading or trailing spaces or any special
character, those must be escaped (see the "Quoting and escaping" section in the
ffmpeg-utils(1) manual).

Options

use_fifo bool
If set to 1, slave outputs will be processed in separate threads using the fifo muxer.
This allows to compensate for different speed/latency/reliability of outputs and setup
transparent recovery. By default this feature is turned off.

fifo_options
Options to pass to fifo pseudo-muxer instances. See fifo.

Muxer options can be specified for each slave by prepending them as a list of key=value pairs
separated by ':', between square brackets. If the options values contain a special character
or the ':' separator, they must be escaped; note that this is a second level escaping.

The following special options are also recognized:

f Specify the format name. Required if it cannot be guessed from the output URL.

bsfs[/spec]
Specify a list of bitstream filters to apply to the specified output.

It is possible to specify to which streams a given bitstream filter applies, by appending
a stream specifier to the option separated by "/". spec must be a stream specifier (see
Format stream specifiers).

If the stream specifier is not specified, the bitstream filters will be applied to all
streams in the output. This will cause that output operation to fail if the output
contains streams to which the bitstream filter cannot be applied e.g. "h264_mp4toannexb"
being applied to an output containing an audio stream.

Options for a bitstream filter must be specified in the form of "opt=value".

Several bitstream filters can be specified, separated by ",".

use_fifo bool
This allows to override tee muxer use_fifo option for individual slave muxer.

fifo_options
This allows to override tee muxer fifo_options for individual slave muxer. See fifo.

select
Select the streams that should be mapped to the slave output, specified by a stream
specifier. If not specified, this defaults to all the mapped streams. This will cause
that output operation to fail if the output format does not accept all mapped streams.

You may use multiple stream specifiers separated by commas (",") e.g.: "a:0,v"

onfail
Specify behaviour on output failure. This can be set to either "abort" (which is default)
or "ignore". "abort" will cause whole process to fail in case of failure on this slave
output. "ignore" will ignore failure on this output, so other outputs will continue
without being affected.

Examples

• Encode something and both archive it in a WebM file and stream it as MPEG-TS over UDP:

ffmpeg -i ... -c:v libx264 -c:a mp2 -f tee -map 0:v -map 0:a
"archive-20121107.mkv|[f=mpegts]udp://10.0.1.255:1234/"

• As above, but continue streaming even if output to local file fails (for example local
drive fills up):

ffmpeg -i ... -c:v libx264 -c:a mp2 -f tee -map 0:v -map 0:a
"[onfail=ignore]archive-20121107.mkv|[f=mpegts]udp://10.0.1.255:1234/"

• Use ffmpeg to encode the input, and send the output to three different destinations. The
"dump_extra" bitstream filter is used to add extradata information to all the output
video keyframes packets, as requested by the MPEG-TS format. The select option is applied
to out.aac in order to make it contain only audio packets.

ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac
-f tee "[bsfs/v=dump_extra=freq=keyframe]out.ts|[movflags=+faststart]out.mp4|[select=a]out.aac"

• As above, but select only stream "a:1" for the audio output. Note that a second level
escaping must be performed, as ":" is a special character used to separate options.

ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac
-f tee "[bsfs/v=dump_extra=freq=keyframe]out.ts|[movflags=+faststart]out.mp4|[select=\'a:1\']out.aac"

webm_chunk
WebM Live Chunk Muxer.

This muxer writes out WebM headers and chunks as separate files which can be consumed by
clients that support WebM Live streams via DASH.

Options

This muxer supports the following options:

chunk_start_index
Index of the first chunk (defaults to 0).

header
Filename of the header where the initialization data will be written.

audio_chunk_duration
Duration of each audio chunk in milliseconds (defaults to 5000).

Example

ffmpeg -f v4l2 -i /dev/video0 \
-f alsa -i hw:0 \
-map 0:0 \
-c:v libvpx-vp9 \
-s 640x360 -keyint_min 30 -g 30 \
-f webm_chunk \
-header webm_live_video_360.hdr \
-chunk_start_index 1 \
webm_live_video_360_%d.chk \
-map 1:0 \
-c:a libvorbis \
-b:a 128k \
-f webm_chunk \
-header webm_live_audio_128.hdr \
-chunk_start_index 1 \
-audio_chunk_duration 1000 \
webm_live_audio_128_%d.chk

webm_dash_manifest
WebM DASH Manifest muxer.

This muxer implements the WebM DASH Manifest specification to generate the DASH manifest XML.
It also supports manifest generation for DASH live streams.

For more information see:

• WebM DASH Specification:
<https://sites.google.com/a/webmproject.org/wiki/adaptive-streaming/webm-dash-specification>

• ISO DASH Specification:
<http://standards.iso.org/ittf/PubliclyAvailableStandards/c065274_ISO_IEC_23009-1_2014.zip>

Options

This muxer supports the following options:

adaptation_sets
This option has the following syntax: "id=x,streams=a,b,c id=y,streams=d,e" where x and y
are the unique identifiers of the adaptation sets and a,b,c,d and e are the indices of
the corresponding audio and video streams. Any number of adaptation sets can be added
using this option.

live
Set this to 1 to create a live stream DASH Manifest. Default: 0.

chunk_start_index
Start index of the first chunk. This will go in the startNumber attribute of the
SegmentTemplate element in the manifest. Default: 0.

chunk_duration_ms
Duration of each chunk in milliseconds. This will go in the duration attribute of the
SegmentTemplate element in the manifest. Default: 1000.

utc_timing_url
URL of the page that will return the UTC timestamp in ISO format. This will go in the
value attribute of the UTCTiming element in the manifest. Default: None.

time_shift_buffer_depth
Smallest time (in seconds) shifting buffer for which any Representation is guaranteed to
be available. This will go in the timeShiftBufferDepth attribute of the MPD element.
Default: 60.

minimum_update_period
Minimum update period (in seconds) of the manifest. This will go in the
minimumUpdatePeriod attribute of the MPD element. Default: 0.

Example

ffmpeg -f webm_dash_manifest -i video1.webm \
-f webm_dash_manifest -i video2.webm \
-f webm_dash_manifest -i audio1.webm \
-f webm_dash_manifest -i audio2.webm \
-map 0 -map 1 -map 2 -map 3 \
-c copy \
-f webm_dash_manifest \
-adaptation_sets "id=0,streams=0,1 id=1,streams=2,3" \
manifest.xml

METADATA
FFmpeg is able to dump metadata from media files into a simple UTF-8-encoded INI-like text
file and then load it back using the metadata muxer/demuxer.

The file format is as follows:

1. A file consists of a header and a number of metadata tags divided into sections, each on
its own line.

2. The header is a ;FFMETADATA string, followed by a version number (now 1).

3. Metadata tags are of the form key=value

4. Immediately after header follows global metadata

5. After global metadata there may be sections with per-stream/per-chapter metadata.

6. A section starts with the section name in uppercase (i.e. STREAM or CHAPTER) in brackets
([, ]) and ends with next section or end of file.

7. At the beginning of a chapter section there may be an optional timebase to be used for
start/end values. It must be in form TIMEBASE=num/den, where num and den are integers. If
the timebase is missing then start/end times are assumed to be in nanoseconds.

Next a chapter section must contain chapter start and end times in form START=num,
END=num, where num is a positive integer.

8. Empty lines and lines starting with ; or # are ignored.

9. Metadata keys or values containing special characters (=, ;, #, \ and a newline) must be
escaped with a backslash \.

10. Note that whitespace in metadata (e.g. foo = bar) is considered to be a part of the tag
(in the example above key is foo , value is
bar).

A ffmetadata file might look like this:

;FFMETADATA1
title=bike\\shed
;this is a comment
artist=FFmpeg troll team

[CHAPTER]
TIMEBASE=1/1000
START=0
#chapter ends at 0:01:00
END=60000
title=chapter \#1
[STREAM]
title=multi\
line

By using the ffmetadata muxer and demuxer it is possible to extract metadata from an input
file to an ffmetadata file, and then transcode the file into an output file with the edited
ffmetadata file.

Extracting an ffmetadata file with ffmpeg goes as follows:

ffmpeg -i INPUT -f ffmetadata FFMETADATAFILE

Reinserting edited metadata information from the FFMETADATAFILE file can be done as:

ffmpeg -i INPUT -i FFMETADATAFILE -map_metadata 1 -codec copy OUTPUT

PROTOCOL OPTIONS
The libavformat library provides some generic global options, which can be set on all the
protocols. In addition each protocol may support so-called private options, which are
specific for that component.

Options may be set by specifying -option value in the FFmpeg tools, or by setting the value
explicitly in the "AVFormatContext" options or using the libavutil/opt.h API for programmatic
use.

The list of supported options follows:

protocol_whitelist list (input)
Set a ","-separated list of allowed protocols. "ALL" matches all protocols. Protocols
prefixed by "-" are disabled. All protocols are allowed by default but protocols used by
an another protocol (nested protocols) are restricted to a per protocol subset.

PROTOCOLS
Protocols are configured elements in FFmpeg that enable access to resources that require
specific protocols.

When you configure your FFmpeg build, all the supported protocols are enabled by default. You
can list all available ones using the configure option "--list-protocols".

You can disable all the protocols using the configure option "--disable-protocols", and
selectively enable a protocol using the option "--enable-protocol=PROTOCOL", or you can
disable a particular protocol using the option "--disable-protocol=PROTOCOL".

The option "-protocols" of the ff* tools will display the list of supported protocols.

All protocols accept the following options:

rw_timeout
Maximum time to wait for (network) read/write operations to complete, in microseconds.

A description of the currently available protocols follows.

amqp
Advanced Message Queueing Protocol (AMQP) version 0-9-1 is a broker based publish-subscribe
communication protocol.

FFmpeg must be compiled with --enable-librabbitmq to support AMQP. A separate AMQP broker
must also be run. An example open-source AMQP broker is RabbitMQ.

After starting the broker, an FFmpeg client may stream data to the broker using the command:

ffmpeg -re -i input -f mpegts amqp://[[user]:[password]@]hostname[:port][/vhost]

Where hostname and port (default is 5672) is the address of the broker. The client may also
set a user/password for authentication. The default for both fields is "guest". Name of
virtual host on broker can be set with vhost. The default value is "/".

Muliple subscribers may stream from the broker using the command:

ffplay amqp://[[user]:[password]@]hostname[:port][/vhost]

In RabbitMQ all data published to the broker flows through a specific exchange, and each
subscribing client has an assigned queue/buffer. When a packet arrives at an exchange, it may
be copied to a client's queue depending on the exchange and routing_key fields.

The following options are supported:

exchange
Sets the exchange to use on the broker. RabbitMQ has several predefined exchanges:
"amq.direct" is the default exchange, where the publisher and subscriber must have a
matching routing_key; "amq.fanout" is the same as a broadcast operation (i.e. the data is
forwarded to all queues on the fanout exchange independent of the routing_key); and
"amq.topic" is similar to "amq.direct", but allows for more complex pattern matching
(refer to the RabbitMQ documentation).

routing_key
Sets the routing key. The default value is "amqp". The routing key is used on the
"amq.direct" and "amq.topic" exchanges to decide whether packets are written to the queue
of a subscriber.

pkt_size
Maximum size of each packet sent/received to the broker. Default is 131072. Minimum is
4096 and max is any large value (representable by an int). When receiving packets, this
sets an internal buffer size in FFmpeg. It should be equal to or greater than the size of
the published packets to the broker. Otherwise the received message may be truncated
causing decoding errors.

connection_timeout
The timeout in seconds during the initial connection to the broker. The default value is
rw_timeout, or 5 seconds if rw_timeout is not set.

delivery_mode mode
Sets the delivery mode of each message sent to broker. The following values are
accepted:

persistent
Delivery mode set to "persistent" (2). This is the default value. Messages may be
written to the broker's disk depending on its setup.

non-persistent
Delivery mode set to "non-persistent" (1). Messages will stay in broker's memory
unless the broker is under memory pressure.

async
Asynchronous data filling wrapper for input stream.

Fill data in a background thread, to decouple I/O operation from demux thread.

async:<URL>
async:http://host/resource
async:cache:http://host/resource

bluray
Read BluRay playlist.

The accepted options are:

angle
BluRay angle

chapter
Start chapter (1...N)

playlist
Playlist to read (BDMV/PLAYLIST/?????.mpls)

Examples:

Read longest playlist from BluRay mounted to /mnt/bluray:

bluray:/mnt/bluray

Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2:

-playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray

cache
Caching wrapper for input stream.

Cache the input stream to temporary file. It brings seeking capability to live streams.

The accepted options are:

read_ahead_limit
Amount in bytes that may be read ahead when seeking isn't supported. Range is -1 to
INT_MAX. -1 for unlimited. Default is 65536.

URL Syntax is

cache:<URL>

concat
Physical concatenation protocol.

Read and seek from many resources in sequence as if they were a unique resource.

A URL accepted by this protocol has the syntax:

concat:<URL1>|<URL2>|...|<URLN>

where URL1, URL2, ..., URLN are the urls of the resource to be concatenated, each one
possibly specifying a distinct protocol.

For example to read a sequence of files split1.mpeg, split2.mpeg, split3.mpeg with ffplay use
the command:

ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg

Note that you may need to escape the character "|" which is special for many shells.

concatf
Physical concatenation protocol using a line break delimited list of resources.

Read and seek from many resources in sequence as if they were a unique resource.

A URL accepted by this protocol has the syntax:

concatf:<URL>

where URL is the url containing a line break delimited list of resources to be concatenated,
each one possibly specifying a distinct protocol. Special characters must be escaped with
backslash or single quotes. See the "Quoting and escaping" section in the ffmpeg-utils(1)
manual.

For example to read a sequence of files split1.mpeg, split2.mpeg, split3.mpeg listed in
separate lines within a file split.txt with ffplay use the command:

ffplay concatf:split.txt

Where split.txt contains the lines:

split1.mpeg
split2.mpeg
split3.mpeg

crypto
AES-encrypted stream reading protocol.

The accepted options are:

key Set the AES decryption key binary block from given hexadecimal representation.

iv Set the AES decryption initialization vector binary block from given hexadecimal
representation.

Accepted URL formats:

crypto:<URL>
crypto+<URL>

data
Data in-line in the URI. See <http://en.wikipedia.org/wiki/Data_URI_scheme>.

For example, to convert a GIF file given inline with ffmpeg:

ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png

fd
File descriptor access protocol.

The accepted syntax is:

fd: -fd <file_descriptor>

If fd is not specified, by default the stdout file descriptor will be used for writing, stdin
for reading. Unlike the pipe protocol, fd protocol has seek support if it corresponding to a
regular file. fd protocol doesn't support pass file descriptor via URL for security.

This protocol accepts the following options:

blocksize
Set I/O operation maximum block size, in bytes. Default value is "INT_MAX", which results
in not limiting the requested block size. Setting this value reasonably low improves
user termination request reaction time, which is valuable if data transmission is slow.

fd Set file descriptor.

file
File access protocol.

Read from or write to a file.

A file URL can have the form:

file:<filename>

where filename is the path of the file to read.

An URL that does not have a protocol prefix will be assumed to be a file URL. Depending on
the build, an URL that looks like a Windows path with the drive letter at the beginning will
also be assumed to be a file URL (usually not the case in builds for unix-like systems).

For example to read from a file input.mpeg with ffmpeg use the command:

ffmpeg -i file:input.mpeg output.mpeg

This protocol accepts the following options:

truncate
Truncate existing files on write, if set to 1. A value of 0 prevents truncating. Default
value is 1.

follow
If set to 1, the protocol will retry reading at the end of the file, allowing reading
files that still are being written. In order for this to terminate, you either need to
use the rw_timeout option, or use the interrupt callback (for API users).

seekable
Controls if seekability is advertised on the file. 0 means non-seekable, -1 means auto
(seekable for normal files, non-seekable for named pipes).

Many demuxers handle seekable and non-seekable resources differently, overriding this
might speed up opening certain files at the cost of losing some features (e.g. accurate
seeking).

ftp
FTP (File Transfer Protocol).

Read from or write to remote resources using FTP protocol.

Following syntax is required.

ftp://[user[:password]@]server[:port]/path/to/remote/resource.mpeg

This protocol accepts the following options.

timeout
Set timeout in microseconds of socket I/O operations used by the underlying low level
operation. By default it is set to -1, which means that the timeout is not specified.

ftp-user
Set a user to be used for authenticating to the FTP server. This is overridden by the
user in the FTP URL.

ftp-password
Set a password to be used for authenticating to the FTP server. This is overridden by the
password in the FTP URL, or by ftp-anonymous-password if no user is set.

ftp-anonymous-password
Password used when login as anonymous user. Typically an e-mail address should be used.

ftp-write-seekable
Control seekability of connection during encoding. If set to 1 the resource is supposed
to be seekable, if set to 0 it is assumed not to be seekable. Default value is 0.

NOTE: Protocol can be used as output, but it is recommended to not do it, unless special care
is taken (tests, customized server configuration etc.). Different FTP servers behave in
different way during seek operation. ff* tools may produce incomplete content due to server
limitations.

gopher
Gopher protocol.

gophers
Gophers protocol.

The Gopher protocol with TLS encapsulation.

hls
Read Apple HTTP Live Streaming compliant segmented stream as a uniform one. The M3U8
playlists describing the segments can be remote HTTP resources or local files, accessed using
the standard file protocol. The nested protocol is declared by specifying "+proto" after the
hls URI scheme name, where proto is either "file" or "http".

hls+http://host/path/to/remote/resource.m3u8
hls+file://path/to/local/resource.m3u8

Using this protocol is discouraged - the hls demuxer should work just as well (if not, please
report the issues) and is more complete. To use the hls demuxer instead, simply use the
direct URLs to the m3u8 files.

http
HTTP (Hyper Text Transfer Protocol).

This protocol accepts the following options:

seekable
Control seekability of connection. If set to 1 the resource is supposed to be seekable,
if set to 0 it is assumed not to be seekable, if set to -1 it will try to autodetect if
it is seekable. Default value is -1.

chunked_post
If set to 1 use chunked Transfer-Encoding for posts, default is 1.

content_type
Set a specific content type for the POST messages or for listen mode.

http_proxy
set HTTP proxy to tunnel through e.g. http://example.com:1234

headers
Set custom HTTP headers, can override built in default headers. The value must be a
string encoding the headers.

multiple_requests
Use persistent connections if set to 1, default is 0.

post_data
Set custom HTTP post data.

referer
Set the Referer header. Include 'Referer: URL' header in HTTP request.

user_agent
Override the User-Agent header. If not specified the protocol will use a string
describing the libavformat build. ("Lavf/<version>")

reconnect_at_eof
If set then eof is treated like an error and causes reconnection, this is useful for live
/ endless streams.

reconnect_streamed
If set then even streamed/non seekable streams will be reconnected on errors.

reconnect_on_network_error
Reconnect automatically in case of TCP/TLS errors during connect.

reconnect_on_http_error
A comma separated list of HTTP status codes to reconnect on. The list can include
specific status codes (e.g. '503') or the strings '4xx' / '5xx'.

reconnect_delay_max
Sets the maximum delay in seconds after which to give up reconnecting

mime_type
Export the MIME type.

http_version
Exports the HTTP response version number. Usually "1.0" or "1.1".

icy If set to 1 request ICY (SHOUTcast) metadata from the server. If the server supports
this, the metadata has to be retrieved by the application by reading the
icy_metadata_headers and icy_metadata_packet options. The default is 1.

icy_metadata_headers
If the server supports ICY metadata, this contains the ICY-specific HTTP reply headers,
separated by newline characters.

icy_metadata_packet
If the server supports ICY metadata, and icy was set to 1, this contains the last non-
empty metadata packet sent by the server. It should be polled in regular intervals by
applications interested in mid-stream metadata updates.

cookies
Set the cookies to be sent in future requests. The format of each cookie is the same as
the value of a Set-Cookie HTTP response field. Multiple cookies can be delimited by a
newline character.

offset
Set initial byte offset.

end_offset
Try to limit the request to bytes preceding this offset.

method
When used as a client option it sets the HTTP method for the request.

When used as a server option it sets the HTTP method that is going to be expected from
the client(s). If the expected and the received HTTP method do not match the client will
be given a Bad Request response. When unset the HTTP method is not checked for now. This
will be replaced by autodetection in the future.

listen
If set to 1 enables experimental HTTP server. This can be used to send data when used as
an output option, or read data from a client with HTTP POST when used as an input option.
If set to 2 enables experimental multi-client HTTP server. This is not yet implemented in
ffmpeg.c and thus must not be used as a command line option.

# Server side (sending):
ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://<server>:<port>

# Client side (receiving):
ffmpeg -i http://<server>:<port> -c copy somefile.ogg

# Client can also be done with wget:
wget http://<server>:<port> -O somefile.ogg

# Server side (receiving):
ffmpeg -listen 1 -i http://<server>:<port> -c copy somefile.ogg

# Client side (sending):
ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://<server>:<port>

# Client can also be done with wget:
wget --post-file=somefile.ogg http://<server>:<port>

send_expect_100
Send an Expect: 100-continue header for POST. If set to 1 it will send, if set to 0 it
won't, if set to -1 it will try to send if it is applicable. Default value is -1.

auth_type
Set HTTP authentication type. No option for Digest, since this method requires getting
nonce parameters from the server first and can't be used straight away like Basic.

none
Choose the HTTP authentication type automatically. This is the default.

basic
Choose the HTTP basic authentication.

Basic authentication sends a Base64-encoded string that contains a user name and
password for the client. Base64 is not a form of encryption and should be considered
the same as sending the user name and password in clear text (Base64 is a reversible
encoding). If a resource needs to be protected, strongly consider using an
authentication scheme other than basic authentication. HTTPS/TLS should be used with
basic authentication. Without these additional security enhancements, basic
authentication should not be used to protect sensitive or valuable information.

HTTP Cookies

Some HTTP requests will be denied unless cookie values are passed in with the request. The
cookies option allows these cookies to be specified. At the very least, each cookie must
specify a value along with a path and domain. HTTP requests that match both the domain and
path will automatically include the cookie value in the HTTP Cookie header field. Multiple
cookies can be delimited by a newline.

The required syntax to play a stream specifying a cookie is:

ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8

Icecast
Icecast protocol (stream to Icecast servers)

This protocol accepts the following options:

ice_genre
Set the stream genre.

ice_name
Set the stream name.

ice_description
Set the stream description.

ice_url
Set the stream website URL.

ice_public
Set if the stream should be public. The default is 0 (not public).

user_agent
Override the User-Agent header. If not specified a string of the form "Lavf/<version>"
will be used.

password
Set the Icecast mountpoint password.

content_type
Set the stream content type. This must be set if it is different from audio/mpeg.

legacy_icecast
This enables support for Icecast versions < 2.4.0, that do not support the HTTP PUT
method but the SOURCE method.

tls Establish a TLS (HTTPS) connection to Icecast.

icecast://[<username>[:<password>]@]<server>:<port>/<mountpoint>

ipfs
InterPlanetary File System (IPFS) protocol support. One can access files stored on the IPFS
network through so-called gateways. These are http(s) endpoints. This protocol wraps the
IPFS native protocols (ipfs:// and ipns://) to be sent to such a gateway. Users can (and
should) host their own node which means this protocol will use one's local gateway to access
files on the IPFS network.

This protocol accepts the following options:

gateway
Defines the gateway to use. When not set, the protocol will first try locating the local
gateway by looking at $IPFS_GATEWAY, $IPFS_PATH and "$HOME/.ipfs/", in that order.

One can use this protocol in 2 ways. Using IPFS:

ffplay ipfs://<hash>

Or the IPNS protocol (IPNS is mutable IPFS):

ffplay ipns://<hash>

mmst
MMS (Microsoft Media Server) protocol over TCP.

mmsh
MMS (Microsoft Media Server) protocol over HTTP.

The required syntax is:

mmsh://<server>[:<port>][/<app>][/<playpath>]

md5
MD5 output protocol.

Computes the MD5 hash of the data to be written, and on close writes this to the designated
output or stdout if none is specified. It can be used to test muxers without writing an
actual file.

Some examples follow.

# Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
ffmpeg -i input.flv -f avi -y md5:output.avi.md5

# Write the MD5 hash of the encoded AVI file to stdout.
ffmpeg -i input.flv -f avi -y md5:

Note that some formats (typically MOV) require the output protocol to be seekable, so they
will fail with the MD5 output protocol.

pipe
UNIX pipe access protocol.

Read and write from UNIX pipes.

The accepted syntax is:

pipe:[<number>]

If fd isn't specified, number is the number corresponding to the file descriptor of the pipe
(e.g. 0 for stdin, 1 for stdout, 2 for stderr). If number is not specified, by default the
stdout file descriptor will be used for writing, stdin for reading.

For example to read from stdin with ffmpeg:

cat test.wav | ffmpeg -i pipe:0
# ...this is the same as...
cat test.wav | ffmpeg -i pipe:

For writing to stdout with ffmpeg:

ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
# ...this is the same as...
ffmpeg -i test.wav -f avi pipe: | cat > test.avi

This protocol accepts the following options:

fd Set file descriptor.

Note that some formats (typically MOV), require the output protocol to be seekable, so they
will fail with the pipe output protocol.

prompeg
Pro-MPEG Code of Practice #3 Release 2 FEC protocol.

The Pro-MPEG CoP#3 FEC is a 2D parity-check forward error correction mechanism for MPEG-2
Transport Streams sent over RTP.

This protocol must be used in conjunction with the "rtp_mpegts" muxer and the "rtp" protocol.

The required syntax is:

-f rtp_mpegts -fec prompeg=<option>=<val>... rtp://<hostname>:<port>

The destination UDP ports are "port + 2" for the column FEC stream and "port + 4" for the row
FEC stream.

This protocol accepts the following options:

l=n The number of columns (4-20, LxD <= 100)

d=n The number of rows (4-20, LxD <= 100)

Example usage:

-f rtp_mpegts -fec prompeg=l=8:d=4 rtp://<hostname>:<port>

rist
Reliable Internet Streaming Transport protocol

The accepted options are:

rist_profile
Supported values:

simple
main
This one is default.

advanced
buffer_size
Set internal RIST buffer size in milliseconds for retransmission of data. Default value
is 0 which means the librist default (1 sec). Maximum value is 30 seconds.

fifo_size
Size of the librist receiver output fifo in number of packets. This must be a power of 2.
Defaults to 8192 (vs the librist default of 1024).

overrun_nonfatal=1|0
Survive in case of librist fifo buffer overrun. Default value is 0.

pkt_size
Set maximum packet size for sending data. 1316 by default.

log_level
Set loglevel for RIST logging messages. You only need to set this if you explicitly want
to enable debug level messages or packet loss simulation, otherwise the regular loglevel
is respected.

secret
Set override of encryption secret, by default is unset.

encryption
Set encryption type, by default is disabled. Acceptable values are 128 and 256.

rtmp
Real-Time Messaging Protocol.

The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia content across a
TCP/IP network.

The required syntax is:

rtmp://[<username>:<password>@]<server>[:<port>][/<app>][/<instance>][/<playpath>]

The accepted parameters are:

username
An optional username (mostly for publishing).

password
An optional password (mostly for publishing).

server
The address of the RTMP server.

port
The number of the TCP port to use (by default is 1935).

app It is the name of the application to access. It usually corresponds to the path where the
application is installed on the RTMP server (e.g. /ondemand/, /flash/live/, etc.). You
can override the value parsed from the URI through the "rtmp_app" option, too.

playpath
It is the path or name of the resource to play with reference to the application
specified in app, may be prefixed by "mp4:". You can override the value parsed from the
URI through the "rtmp_playpath" option, too.

listen
Act as a server, listening for an incoming connection.

timeout
Maximum time to wait for the incoming connection. Implies listen.

Additionally, the following parameters can be set via command line options (or in code via
"AVOption"s):

rtmp_app
Name of application to connect on the RTMP server. This option overrides the parameter
specified in the URI.

rtmp_buffer
Set the client buffer time in milliseconds. The default is 3000.

rtmp_conn
Extra arbitrary AMF connection parameters, parsed from a string, e.g. like "B:1 S:authMe
O:1 NN:code:1.23 NS:flag:ok O:0". Each value is prefixed by a single character denoting
the type, B for Boolean, N for number, S for string, O for object, or Z for null,
followed by a colon. For Booleans the data must be either 0 or 1 for FALSE or TRUE,
respectively. Likewise for Objects the data must be 0 or 1 to end or begin an object,
respectively. Data items in subobjects may be named, by prefixing the type with 'N' and
specifying the name before the value (i.e. "NB:myFlag:1"). This option may be used
multiple times to construct arbitrary AMF sequences.

rtmp_enhanced_codecs
Specify the list of codecs the client advertises to support in an enhanced RTMP stream.
This option should be set to a comma separated list of fourcc values, like
"hvc1,av01,vp09" for multiple codecs or "hvc1" for only one codec. The specified list
will be presented in the "fourCcLive" property of the Connect Command Message.

rtmp_flashver
Version of the Flash plugin used to run the SWF player. The default is LNX 9,0,124,2.
(When publishing, the default is FMLE/3.0 (compatible; <libavformat version>).)

rtmp_flush_interval
Number of packets flushed in the same request (RTMPT only). The default is 10.

rtmp_live
Specify that the media is a live stream. No resuming or seeking in live streams is
possible. The default value is "any", which means the subscriber first tries to play the
live stream specified in the playpath. If a live stream of that name is not found, it
plays the recorded stream. The other possible values are "live" and "recorded".

rtmp_pageurl
URL of the web page in which the media was embedded. By default no value will be sent.

rtmp_playpath
Stream identifier to play or to publish. This option overrides the parameter specified in
the URI.

rtmp_subscribe
Name of live stream to subscribe to. By default no value will be sent. It is only sent
if the option is specified or if rtmp_live is set to live.

rtmp_swfhash
SHA256 hash of the decompressed SWF file (32 bytes).

rtmp_swfsize
Size of the decompressed SWF file, required for SWFVerification.

rtmp_swfurl
URL of the SWF player for the media. By default no value will be sent.

rtmp_swfverify
URL to player swf file, compute hash/size automatically.

rtmp_tcurl
URL of the target stream. Defaults to proto://host[:port]/app.

tcp_nodelay=1|0
Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.

Remark: Writing to the socket is currently not optimized to minimize system calls and
reduces the efficiency / effect of TCP_NODELAY.

For example to read with ffplay a multimedia resource named "sample" from the application
"vod" from an RTMP server "myserver":

ffplay rtmp://myserver/vod/sample

To publish to a password protected server, passing the playpath and app names separately:

ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@myserver/

rtmpe
Encrypted Real-Time Messaging Protocol.

The Encrypted Real-Time Messaging Protocol (RTMPE) is used for streaming multimedia content
within standard cryptographic primitives, consisting of Diffie-Hellman key exchange and
HMACSHA256, generating a pair of RC4 keys.

rtmps
Real-Time Messaging Protocol over a secure SSL connection.

The Real-Time Messaging Protocol (RTMPS) is used for streaming multimedia content across an
encrypted connection.

rtmpt
Real-Time Messaging Protocol tunneled through HTTP.

The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used for streaming
multimedia content within HTTP requests to traverse firewalls.

rtmpte
Encrypted Real-Time Messaging Protocol tunneled through HTTP.

The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE) is used for
streaming multimedia content within HTTP requests to traverse firewalls.

rtmpts
Real-Time Messaging Protocol tunneled through HTTPS.

The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used for streaming
multimedia content within HTTPS requests to traverse firewalls.

libsmbclient
libsmbclient permits one to manipulate CIFS/SMB network resources.

Following syntax is required.

smb://[[domain:]user[:password@]]server[/share[/path[/file]]]

This protocol accepts the following options.

timeout
Set timeout in milliseconds of socket I/O operations used by the underlying low level
operation. By default it is set to -1, which means that the timeout is not specified.

truncate
Truncate existing files on write, if set to 1. A value of 0 prevents truncating. Default
value is 1.

workgroup
Set the workgroup used for making connections. By default workgroup is not specified.

For more information see: <http://www.samba.org/>.

libssh
Secure File Transfer Protocol via libssh

Read from or write to remote resources using SFTP protocol.

Following syntax is required.

sftp://[user[:password]@]server[:port]/path/to/remote/resource.mpeg

This protocol accepts the following options.

timeout
Set timeout of socket I/O operations used by the underlying low level operation. By
default it is set to -1, which means that the timeout is not specified.

truncate
Truncate existing files on write, if set to 1. A value of 0 prevents truncating. Default
value is 1.

private_key
Specify the path of the file containing private key to use during authorization. By
default libssh searches for keys in the ~/.ssh/ directory.

Example: Play a file stored on remote server.

ffplay sftp://user:password@server_address:22/home/user/resource.mpeg

librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
Real-Time Messaging Protocol and its variants supported through librtmp.

Requires the presence of the librtmp headers and library during configuration. You need to
explicitly configure the build with "--enable-librtmp". If enabled this will replace the
native RTMP protocol.

This protocol provides most client functions and a few server functions needed to support
RTMP, RTMP tunneled in HTTP (RTMPT), encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and
tunneled variants of these encrypted types (RTMPTE, RTMPTS).

The required syntax is:

<rtmp_proto>://<server>[:<port>][/<app>][/<playpath>] <options>

where rtmp_proto is one of the strings "rtmp", "rtmpt", "rtmpe", "rtmps", "rtmpte", "rtmpts"
corresponding to each RTMP variant, and server, port, app and playpath have the same meaning
as specified for the RTMP native protocol. options contains a list of space-separated
options of the form key=val.

See the librtmp manual page (man 3 librtmp) for more information.

For example, to stream a file in real-time to an RTMP server using ffmpeg:

ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream

To play the same stream using ffplay:

ffplay "rtmp://myserver/live/mystream live=1"

rtp
Real-time Transport Protocol.

The required syntax for an RTP URL is: rtp://hostname[:port][?option=val...]

port specifies the RTP port to use.

The following URL options are supported:

ttl=n
Set the TTL (Time-To-Live) value (for multicast only).

rtcpport=n
Set the remote RTCP port to n.

localrtpport=n
Set the local RTP port to n.

localrtcpport=n'
Set the local RTCP port to n.

pkt_size=n
Set max packet size (in bytes) to n.

buffer_size=size
Set the maximum UDP socket buffer size in bytes.

connect=0|1
Do a connect() on the UDP socket (if set to 1) or not (if set to 0).

sources=ip[,ip]
List allowed source IP addresses.

block=ip[,ip]
List disallowed (blocked) source IP addresses.

write_to_source=0|1
Send packets to the source address of the latest received packet (if set to 1) or to a
default remote address (if set to 0).

localport=n
Set the local RTP port to n.

localaddr=addr
Local IP address of a network interface used for sending packets or joining multicast
groups.

timeout=n
Set timeout (in microseconds) of socket I/O operations to n.

This is a deprecated option. Instead, localrtpport should be used.

Important notes:

1. If rtcpport is not set the RTCP port will be set to the RTP port value plus 1.

2. If localrtpport (the local RTP port) is not set any available port will be used for the
local RTP and RTCP ports.

3. If localrtcpport (the local RTCP port) is not set it will be set to the local RTP port
value plus 1.

rtsp
Real-Time Streaming Protocol.

RTSP is not technically a protocol handler in libavformat, it is a demuxer and muxer. The
demuxer supports both normal RTSP (with data transferred over RTP; this is used by e.g. Apple
and Microsoft) and Real-RTSP (with data transferred over RDT).

The muxer can be used to send a stream using RTSP ANNOUNCE to a server supporting it
(currently Darwin Streaming Server and Mischa Spiegelmock's
<https://github.com/revmischa/rtsp-server>).

The required syntax for a RTSP url is:

rtsp://<hostname>[:<port>]/<path>

Options can be set on the ffmpeg/ffplay command line, or set in code via "AVOption"s or in
"avformat_open_input".

Muxer

The following options are supported.

rtsp_transport
Set RTSP transport protocols.

It accepts the following values:

udp Use UDP as lower transport protocol.

tcp Use TCP (interleaving within the RTSP control channel) as lower transport protocol.

Default value is 0.

rtsp_flags
Set RTSP flags.

The following values are accepted:

latm
Use MP4A-LATM packetization instead of MPEG4-GENERIC for AAC.

rfc2190
Use RFC 2190 packetization instead of RFC 4629 for H.263.

skip_rtcp
Don't send RTCP sender reports.

h264_mode0
Use mode 0 for H.264 in RTP.

send_bye
Send RTCP BYE packets when finishing.

Default value is 0.

min_port
Set minimum local UDP port. Default value is 5000.

max_port
Set maximum local UDP port. Default value is 65000.

buffer_size
Set the maximum socket buffer size in bytes.

pkt_size
Set max send packet size (in bytes). Default value is 1472.

Demuxer

The following options are supported.

initial_pause
Do not start playing the stream immediately if set to 1. Default value is 0.

rtsp_transport
Set RTSP transport protocols.

It accepts the following values:

udp Use UDP as lower transport protocol.

tcp Use TCP (interleaving within the RTSP control channel) as lower transport protocol.

udp_multicast
Use UDP multicast as lower transport protocol.

http
Use HTTP tunneling as lower transport protocol, which is useful for passing proxies.

https
Use HTTPs tunneling as lower transport protocol, which is useful for passing proxies
and widely used for security consideration.

Multiple lower transport protocols may be specified, in that case they are tried one at a
time (if the setup of one fails, the next one is tried). For the muxer, only the tcp and
udp options are supported.

rtsp_flags
Set RTSP flags.

The following values are accepted:

filter_src
Accept packets only from negotiated peer address and port.

listen
Act as a server, listening for an incoming connection.

prefer_tcp
Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.

satip_raw
Export raw MPEG-TS stream instead of demuxing. The flag will simply write out the raw
stream, with the original PAT/PMT/PIDs intact.

Default value is none.

allowed_media_types
Set media types to accept from the server.

The following flags are accepted:

video
audio
data
subtitle

By default it accepts all media types.

min_port
Set minimum local UDP port. Default value is 5000.

max_port
Set maximum local UDP port. Default value is 65000.

listen_timeout
Set maximum timeout (in seconds) to establish an initial connection. Setting
listen_timeout > 0 sets rtsp_flags to listen. Default is -1 which means an infinite
timeout when listen mode is set.

reorder_queue_size
Set number of packets to buffer for handling of reordered packets.

timeout
Set socket TCP I/O timeout in microseconds.

user_agent
Override User-Agent header. If not specified, it defaults to the libavformat identifier
string.

buffer_size
Set the maximum socket buffer size in bytes.

When receiving data over UDP, the demuxer tries to reorder received packets (since they may
arrive out of order, or packets may get lost totally). This can be disabled by setting the
maximum demuxing delay to zero (via the "max_delay" field of AVFormatContext).

When watching multi-bitrate Real-RTSP streams with ffplay, the streams to display can be
chosen with "-vst" n and "-ast" n for video and audio respectively, and can be switched on
the fly by pressing "v" and "a".

Examples

The following examples all make use of the ffplay and ffmpeg tools.

• Watch a stream over UDP, with a max reordering delay of 0.5 seconds:

ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4

• Watch a stream tunneled over HTTP:

ffplay -rtsp_transport http rtsp://server/video.mp4

• Send a stream in realtime to a RTSP server, for others to watch:

ffmpeg -re -i <input> -f rtsp -muxdelay 0.1 rtsp://server/live.sdp

• Receive a stream in realtime:

ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp <output>

sap
Session Announcement Protocol (RFC 2974). This is not technically a protocol handler in
libavformat, it is a muxer and demuxer. It is used for signalling of RTP streams, by
announcing the SDP for the streams regularly on a separate port.

Muxer

The syntax for a SAP url given to the muxer is:

sap://<destination>[:<port>][?<options>]

The RTP packets are sent to destination on port port, or to port 5004 if no port is
specified. options is a "&"-separated list. The following options are supported:

announce_addr=address
Specify the destination IP address for sending the announcements to. If omitted, the
announcements are sent to the commonly used SAP announcement multicast address
224.2.127.254 (sap.mcast.net), or ff0e::2:7ffe if destination is an IPv6 address.

announce_port=port
Specify the port to send the announcements on, defaults to 9875 if not specified.

ttl=ttl
Specify the time to live value for the announcements and RTP packets, defaults to 255.

same_port=0|1
If set to 1, send all RTP streams on the same port pair. If zero (the default), all
streams are sent on unique ports, with each stream on a port 2 numbers higher than the
previous. VLC/Live555 requires this to be set to 1, to be able to receive the stream.
The RTP stack in libavformat for receiving requires all streams to be sent on unique
ports.

Example command lines follow.

To broadcast a stream on the local subnet, for watching in VLC:

ffmpeg -re -i <input> -f sap sap://224.0.0.255?same_port=1

Similarly, for watching in ffplay:

ffmpeg -re -i <input> -f sap sap://224.0.0.255

And for watching in ffplay, over IPv6:

ffmpeg -re -i <input> -f sap sap://[ff0e::1:2:3:4]

Demuxer

The syntax for a SAP url given to the demuxer is:

sap://[<address>][:<port>]

address is the multicast address to listen for announcements on, if omitted, the default
224.2.127.254 (sap.mcast.net) is used. port is the port that is listened on, 9875 if omitted.

The demuxers listens for announcements on the given address and port. Once an announcement
is received, it tries to receive that particular stream.

Example command lines follow.

To play back the first stream announced on the normal SAP multicast address:

ffplay sap://

To play back the first stream announced on one the default IPv6 SAP multicast address:

ffplay sap://[ff0e::2:7ffe]

sctp
Stream Control Transmission Protocol.

The accepted URL syntax is:

sctp://<host>:<port>[?<options>]

The protocol accepts the following options:

listen
If set to any value, listen for an incoming connection. Outgoing connection is done by
default.

max_streams
Set the maximum number of streams. By default no limit is set.

srt
Haivision Secure Reliable Transport Protocol via libsrt.

The supported syntax for a SRT URL is:

srt://<hostname>:<port>[?<options>]

options contains a list of &-separated options of the form key=val.

options contains a list of '-key val' options.

This protocol accepts the following options.

connect_timeout=milliseconds
Connection timeout; SRT cannot connect for RTT > 1500 msec (2 handshake exchanges) with
the default connect timeout of 3 seconds. This option applies to the caller and
rendezvous connection modes. The connect timeout is 10 times the value set for the
rendezvous mode (which can be used as a workaround for this connection problem with
earlier versions).

ffs=bytes
Flight Flag Size (Window Size), in bytes. FFS is actually an internal parameter and you
should set it to not less than recv_buffer_size and mss. The default value is relatively
large, therefore unless you set a very large receiver buffer, you do not need to change
this option. Default value is 25600.

inputbw=bytes/seconds
Sender nominal input rate, in bytes per seconds. Used along with oheadbw, when maxbw is
set to relative (0), to calculate maximum sending rate when recovery packets are sent
along with the main media stream: inputbw * (100 + oheadbw) / 100 if inputbw is not set
while maxbw is set to relative (0), the actual input rate is evaluated inside the
library. Default value is 0.

iptos=tos
IP Type of Service. Applies to sender only. Default value is 0xB8.

ipttl=ttl
IP Time To Live. Applies to sender only. Default value is 64.

latency=microseconds
Timestamp-based Packet Delivery Delay. Used to absorb bursts of missed packet
retransmissions. This flag sets both rcvlatency and peerlatency to the same value. Note
that prior to version 1.3.0 this is the only flag to set the latency, however this is
effectively equivalent to setting peerlatency, when side is sender and rcvlatency when
side is receiver, and the bidirectional stream sending is not supported.

listen_timeout=microseconds
Set socket listen timeout.

maxbw=bytes/seconds
Maximum sending bandwidth, in bytes per seconds. -1 infinite (CSRTCC limit is 30mbps) 0
relative to input rate (see inputbw) >0 absolute limit value Default value is 0
(relative)

mode=caller|listener|rendezvous
Connection mode. caller opens client connection. listener starts server to listen for
incoming connections. rendezvous use Rendez-Vous connection mode. Default value is
caller.

mss=bytes
Maximum Segment Size, in bytes. Used for buffer allocation and rate calculation using a
packet counter assuming fully filled packets. The smallest MSS between the peers is used.
This is 1500 by default in the overall internet. This is the maximum size of the UDP
packet and can be only decreased, unless you have some unusual dedicated network
settings. Default value is 1500.

nakreport=1|0
If set to 1, Receiver will send `UMSG_LOSSREPORT` messages periodically until a lost
packet is retransmitted or intentionally dropped. Default value is 1.

oheadbw=percents
Recovery bandwidth overhead above input rate, in percents. See inputbw. Default value is
25%.

passphrase=string
HaiCrypt Encryption/Decryption Passphrase string, length from 10 to 79 characters. The
passphrase is the shared secret between the sender and the receiver. It is used to
generate the Key Encrypting Key using PBKDF2 (Password-Based Key Derivation Function). It
is used only if pbkeylen is non-zero. It is used on the receiver only if the received
data is encrypted. The configured passphrase cannot be recovered (write-only).

enforced_encryption=1|0
If true, both connection parties must have the same password set (including empty, that
is, with no encryption). If the password doesn't match or only one side is unencrypted,
the connection is rejected. Default is true.

kmrefreshrate=packets
The number of packets to be transmitted after which the encryption key is switched to a
new key. Default is -1. -1 means auto (0x1000000 in srt library). The range for this
option is integers in the 0 - "INT_MAX".

kmpreannounce=packets
The interval between when a new encryption key is sent and when switchover occurs. This
value also applies to the subsequent interval between when switchover occurs and when the
old encryption key is decommissioned. Default is -1. -1 means auto (0x1000 in srt
library). The range for this option is integers in the 0 - "INT_MAX".

snddropdelay=microseconds
The sender's extra delay before dropping packets. This delay is added to the default drop
delay time interval value.

Special value -1: Do not drop packets on the sender at all.

payload_size=bytes
Sets the maximum declared size of a packet transferred during the single call to the
sending function in Live mode. Use 0 if this value isn't used (which is default in file
mode). Default is -1 (automatic), which typically means MPEG-TS; if you are going to use
SRT to send any different kind of payload, such as, for example, wrapping a live stream
in very small frames, then you can use a bigger maximum frame size, though not greater
than 1456 bytes.

pkt_size=bytes
Alias for payload_size.

peerlatency=microseconds
The latency value (as described in rcvlatency) that is set by the sender side as a
minimum value for the receiver.

pbkeylen=bytes
Sender encryption key length, in bytes. Only can be set to 0, 16, 24 and 32. Enable
sender encryption if not 0. Not required on receiver (set to 0), key size obtained from
sender in HaiCrypt handshake. Default value is 0.

rcvlatency=microseconds
The time that should elapse since the moment when the packet was sent and the moment when
it's delivered to the receiver application in the receiving function. This time should
be a buffer time large enough to cover the time spent for sending, unexpectedly extended
RTT time, and the time needed to retransmit the lost UDP packet. The effective latency
value will be the maximum of this options' value and the value of peerlatency set by the
peer side. Before version 1.3.0 this option is only available as latency.

recv_buffer_size=bytes
Set UDP receive buffer size, expressed in bytes.

send_buffer_size=bytes
Set UDP send buffer size, expressed in bytes.

timeout=microseconds
Set raise error timeouts for read, write and connect operations. Note that the SRT
library has internal timeouts which can be controlled separately, the value set here is
only a cap on those.

tlpktdrop=1|0
Too-late Packet Drop. When enabled on receiver, it skips missing packets that have not
been delivered in time and delivers the following packets to the application when their
time-to-play has come. It also sends a fake ACK to the sender. When enabled on sender and
enabled on the receiving peer, the sender drops the older packets that have no chance of
being delivered in time. It was automatically enabled in the sender if the receiver
supports it.

sndbuf=bytes
Set send buffer size, expressed in bytes.

rcvbuf=bytes
Set receive buffer size, expressed in bytes.

Receive buffer must not be greater than ffs.

lossmaxttl=packets
The value up to which the Reorder Tolerance may grow. When Reorder Tolerance is > 0, then
packet loss report is delayed until that number of packets come in. Reorder Tolerance
increases every time a "belated" packet has come, but it wasn't due to retransmission
(that is, when UDP packets tend to come out of order), with the difference between the
latest sequence and this packet's sequence, and not more than the value of this option.
By default it's 0, which means that this mechanism is turned off, and the loss report is
always sent immediately upon experiencing a "gap" in sequences.

minversion
The minimum SRT version that is required from the peer. A connection to a peer that does
not satisfy the minimum version requirement will be rejected.

The version format in hex is 0xXXYYZZ for x.y.z in human readable form.

streamid=string
A string limited to 512 characters that can be set on the socket prior to connecting.
This stream ID will be able to be retrieved by the listener side from the socket that is
returned from srt_accept and was connected by a socket with that set stream ID. SRT does
not enforce any special interpretation of the contents of this string. This option
doesn’t make sense in Rendezvous connection; the result might be that simply one side
will override the value from the other side and it’s the matter of luck which one would
win

srt_streamid=string
Alias for streamid to avoid conflict with ffmpeg command line option.

smoother=live|file
The type of Smoother used for the transmission for that socket, which is responsible for
the transmission and congestion control. The Smoother type must be exactly the same on
both connecting parties, otherwise the connection is rejected.

messageapi=1|0
When set, this socket uses the Message API, otherwise it uses Buffer API. Note that in
live mode (see transtype) there’s only message API available. In File mode you can chose
to use one of two modes:

Stream API (default, when this option is false). In this mode you may send as many data
as you wish with one sending instruction, or even use dedicated functions that read
directly from a file. The internal facility will take care of any speed and congestion
control. When receiving, you can also receive as many data as desired, the data not
extracted will be waiting for the next call. There is no boundary between data portions
in the Stream mode.

Message API. In this mode your single sending instruction passes exactly one piece of
data that has boundaries (a message). Contrary to Live mode, this message may span across
multiple UDP packets and the only size limitation is that it shall fit as a whole in the
sending buffer. The receiver shall use as large buffer as necessary to receive the
message, otherwise the message will not be given up. When the message is not complete
(not all packets received or there was a packet loss) it will not be given up.

transtype=live|file
Sets the transmission type for the socket, in particular, setting this option sets
multiple other parameters to their default values as required for a particular
transmission type.

live: Set options as for live transmission. In this mode, you should send by one sending
instruction only so many data that fit in one UDP packet, and limited to the value
defined first in payload_size (1316 is default in this mode). There is no speed control
in this mode, only the bandwidth control, if configured, in order to not exceed the
bandwidth with the overhead transmission (retransmitted and control packets).

file: Set options as for non-live transmission. See messageapi for further explanations

linger=seconds
The number of seconds that the socket waits for unsent data when closing. Default is -1.
-1 means auto (off with 0 seconds in live mode, on with 180 seconds in file mode). The
range for this option is integers in the 0 - "INT_MAX".

tsbpd=1|0
When true, use Timestamp-based Packet Delivery mode. The default behavior depends on the
transmission type: enabled in live mode, disabled in file mode.

For more information see: <https://github.com/Haivision/srt>.

srtp
Secure Real-time Transport Protocol.

The accepted options are:

srtp_in_suite
srtp_out_suite
Select input and output encoding suites.

Supported values:

AES_CM_128_HMAC_SHA1_80
SRTP_AES128_CM_HMAC_SHA1_80
AES_CM_128_HMAC_SHA1_32
SRTP_AES128_CM_HMAC_SHA1_32
srtp_in_params
srtp_out_params
Set input and output encoding parameters, which are expressed by a base64-encoded
representation of a binary block. The first 16 bytes of this binary block are used as
master key, the following 14 bytes are used as master salt.

subfile
Virtually extract a segment of a file or another stream. The underlying stream must be
seekable.

Accepted options:

start
Start offset of the extracted segment, in bytes.

end End offset of the extracted segment, in bytes. If set to 0, extract till end of file.

Examples:

Extract a chapter from a DVD VOB file (start and end sectors obtained externally and
multiplied by 2048):

subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB

Play an AVI file directly from a TAR archive:

subfile,,start,183241728,end,366490624,,:archive.tar

Play a MPEG-TS file from start offset till end:

subfile,,start,32815239,end,0,,:video.ts

tee
Writes the output to multiple protocols. The individual outputs are separated by |

tee:file://path/to/local/this.avi|file://path/to/local/that.avi

tcp
Transmission Control Protocol.

The required syntax for a TCP url is:

tcp://<hostname>:<port>[?<options>]

options contains a list of &-separated options of the form key=val.

The list of supported options follows.

listen=2|1|0
Listen for an incoming connection. 0 disables listen, 1 enables listen in single client
mode, 2 enables listen in multi-client mode. Default value is 0.

local_addr=addr
Local IP address of a network interface used for tcp socket connect.

local_port=port
Local port used for tcp socket connect.

timeout=microseconds
Set raise error timeout, expressed in microseconds.

This option is only relevant in read mode: if no data arrived in more than this time
interval, raise error.

listen_timeout=milliseconds
Set listen timeout, expressed in milliseconds.

recv_buffer_size=bytes
Set receive buffer size, expressed bytes.

send_buffer_size=bytes
Set send buffer size, expressed bytes.

tcp_nodelay=1|0
Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.

Remark: Writing to the socket is currently not optimized to minimize system calls and
reduces the efficiency / effect of TCP_NODELAY.

tcp_mss=bytes
Set maximum segment size for outgoing TCP packets, expressed in bytes.

The following example shows how to setup a listening TCP connection with ffmpeg, which is
then accessed with ffplay:

ffmpeg -i <input> -f <format> tcp://<hostname>:<port>?listen
ffplay tcp://<hostname>:<port>

tls
Transport Layer Security (TLS) / Secure Sockets Layer (SSL)

The required syntax for a TLS/SSL url is:

tls://<hostname>:<port>[?<options>]

The following parameters can be set via command line options (or in code via "AVOption"s):

ca_file, cafile=filename
A file containing certificate authority (CA) root certificates to treat as trusted. If
the linked TLS library contains a default this might not need to be specified for
verification to work, but not all libraries and setups have defaults built in. The file
must be in OpenSSL PEM format.

tls_verify=1|0
If enabled, try to verify the peer that we are communicating with. Note, if using
OpenSSL, this currently only makes sure that the peer certificate is signed by one of the
root certificates in the CA database, but it does not validate that the certificate
actually matches the host name we are trying to connect to. (With other backends, the
host name is validated as well.)

This is disabled by default since it requires a CA database to be provided by the caller
in many cases.

cert_file, cert=filename
A file containing a certificate to use in the handshake with the peer. (When operating
as server, in listen mode, this is more often required by the peer, while client
certificates only are mandated in certain setups.)

key_file, key=filename
A file containing the private key for the certificate.

listen=1|0
If enabled, listen for connections on the provided port, and assume the server role in
the handshake instead of the client role.

http_proxy
The HTTP proxy to tunnel through, e.g. "http://example.com:1234". The proxy must support
the CONNECT method.

Example command lines:

To create a TLS/SSL server that serves an input stream.

ffmpeg -i <input> -f <format> tls://<hostname>:<port>?listen&cert=<server.crt>&key=<server.key>

To play back a stream from the TLS/SSL server using ffplay:

ffplay tls://<hostname>:<port>

udp
User Datagram Protocol.

The required syntax for an UDP URL is:

udp://<hostname>:<port>[?<options>]

options contains a list of &-separated options of the form key=val.

In case threading is enabled on the system, a circular buffer is used to store the incoming
data, which allows one to reduce loss of data due to UDP socket buffer overruns. The
fifo_size and overrun_nonfatal options are related to this buffer.

The list of supported options follows.

buffer_size=size
Set the UDP maximum socket buffer size in bytes. This is used to set either the receive
or send buffer size, depending on what the socket is used for. Default is 32 KB for
output, 384 KB for input. See also fifo_size.

bitrate=bitrate
If set to nonzero, the output will have the specified constant bitrate if the input has
enough packets to sustain it.

burst_bits=bits
When using bitrate this specifies the maximum number of bits in packet bursts.

localport=port
Override the local UDP port to bind with.

localaddr=addr
Local IP address of a network interface used for sending packets or joining multicast
groups.

pkt_size=size
Set the size in bytes of UDP packets.

reuse=1|0
Explicitly allow or disallow reusing UDP sockets.

ttl=ttl
Set the time to live value (for multicast only).

connect=1|0
Initialize the UDP socket with connect(). In this case, the destination address can't be
changed with ff_udp_set_remote_url later. If the destination address isn't known at the
start, this option can be specified in ff_udp_set_remote_url, too. This allows finding
out the source address for the packets with getsockname, and makes writes return with
AVERROR(ECONNREFUSED) if "destination unreachable" is received. For receiving, this
gives the benefit of only receiving packets from the specified peer address/port.

sources=address[,address]
Only receive packets sent from the specified addresses. In case of multicast, also
subscribe to multicast traffic coming from these addresses only.

block=address[,address]
Ignore packets sent from the specified addresses. In case of multicast, also exclude the
source addresses in the multicast subscription.

fifo_size=units
Set the UDP receiving circular buffer size, expressed as a number of packets with size of
188 bytes. If not specified defaults to 7*4096.

overrun_nonfatal=1|0
Survive in case of UDP receiving circular buffer overrun. Default value is 0.

timeout=microseconds
Set raise error timeout, expressed in microseconds.

This option is only relevant in read mode: if no data arrived in more than this time
interval, raise error.

broadcast=1|0
Explicitly allow or disallow UDP broadcasting.

Note that broadcasting may not work properly on networks having a broadcast storm
protection.

Examples

• Use ffmpeg to stream over UDP to a remote endpoint:

ffmpeg -i <input> -f <format> udp://<hostname>:<port>

• Use ffmpeg to stream in mpegts format over UDP using 188 sized UDP packets, using a large
input buffer:

ffmpeg -i <input> -f mpegts udp://<hostname>:<port>?pkt_size=188&buffer_size=65535

• Use ffmpeg to receive over UDP from a remote endpoint:

ffmpeg -i udp://[<multicast-address>]:<port> ...

unix
Unix local socket

The required syntax for a Unix socket URL is:

unix://<filepath>

The following parameters can be set via command line options (or in code via "AVOption"s):

timeout
Timeout in ms.

listen
Create the Unix socket in listening mode.

zmq
ZeroMQ asynchronous messaging using the libzmq library.

This library supports unicast streaming to multiple clients without relying on an external
server.

The required syntax for streaming or connecting to a stream is:

zmq:tcp://ip-address:port

Example: Create a localhost stream on port 5555:

ffmpeg -re -i input -f mpegts zmq:tcp://127.0.0.1:5555

Multiple clients may connect to the stream using:

ffplay zmq:tcp://127.0.0.1:5555

Streaming to multiple clients is implemented using a ZeroMQ Pub-Sub pattern. The server side
binds to a port and publishes data. Clients connect to the server (via IP address/port) and
subscribe to the stream. The order in which the server and client start generally does not
matter.

ffmpeg must be compiled with the --enable-libzmq option to support this protocol.

Options can be set on the ffmpeg/ffplay command line. The following options are supported:

pkt_size
Forces the maximum packet size for sending/receiving data. The default value is 131,072
bytes. On the server side, this sets the maximum size of sent packets via ZeroMQ. On the
clients, it sets an internal buffer size for receiving packets. Note that pkt_size on the
clients should be equal to or greater than pkt_size on the server. Otherwise the received
message may be truncated causing decoding errors.

DEVICE OPTIONS
The libavdevice library provides the same interface as libavformat. Namely, an input device
is considered like a demuxer, and an output device like a muxer, and the interface and
generic device options are the same provided by libavformat (see the ffmpeg-formats manual).

In addition each input or output device may support so-called private options, which are
specific for that component.

Options may be set by specifying -option value in the FFmpeg tools, or by setting the value
explicitly in the device "AVFormatContext" options or using the libavutil/opt.h API for
programmatic use.

INPUT DEVICES
Input devices are configured elements in FFmpeg which enable accessing the data coming from a
multimedia device attached to your system.

When you configure your FFmpeg build, all the supported input devices are enabled by default.
You can list all available ones using the configure option "--list-indevs".

You can disable all the input devices using the configure option "--disable-indevs", and
selectively enable an input device using the option "--enable-indev=INDEV", or you can
disable a particular input device using the option "--disable-indev=INDEV".

The option "-devices" of the ff* tools will display the list of supported input devices.

A description of the currently available input devices follows.

alsa
ALSA (Advanced Linux Sound Architecture) input device.

To enable this input device during configuration you need libasound installed on your system.

This device allows capturing from an ALSA device. The name of the device to capture has to be
an ALSA card identifier.

An ALSA identifier has the syntax:

hw:<CARD>[,<DEV>[,<SUBDEV>]]

where the DEV and SUBDEV components are optional.

The three arguments (in order: CARD,DEV,SUBDEV) specify card number or identifier, device
number and subdevice number (-1 means any).

To see the list of cards currently recognized by your system check the files
/proc/asound/cards and /proc/asound/devices.

For example to capture with ffmpeg from an ALSA device with card id 0, you may run the
command:

ffmpeg -f alsa -i hw:0 alsaout.wav

For more information see: <http://www.alsa-project.org/alsa-doc/alsa-lib/pcm.html>

Options

sample_rate
Set the sample rate in Hz. Default is 48000.

channels
Set the number of channels. Default is 2.

android_camera
Android camera input device.

This input devices uses the Android Camera2 NDK API which is available on devices with API
level 24+. The availability of android_camera is autodetected during configuration.

This device allows capturing from all cameras on an Android device, which are integrated into
the Camera2 NDK API.

The available cameras are enumerated internally and can be selected with the camera_index
parameter. The input file string is discarded.

Generally the back facing camera has index 0 while the front facing camera has index 1.

Options

video_size
Set the video size given as a string such as 640x480 or hd720. Falls back to the first
available configuration reported by Android if requested video size is not available or
by default.

framerate
Set the video framerate. Falls back to the first available configuration reported by
Android if requested framerate is not available or by default (-1).

camera_index
Set the index of the camera to use. Default is 0.

input_queue_size
Set the maximum number of frames to buffer. Default is 5.

avfoundation
AVFoundation input device.

AVFoundation is the currently recommended framework by Apple for streamgrabbing on OSX >=
10.7 as well as on iOS.

The input filename has to be given in the following syntax:

-i "[[VIDEO]:[AUDIO]]"

The first entry selects the video input while the latter selects the audio input. The stream
has to be specified by the device name or the device index as shown by the device list.
Alternatively, the video and/or audio input device can be chosen by index using the

B<-video_device_index E<lt>INDEXE<gt>>

and/or

B<-audio_device_index E<lt>INDEXE<gt>>

, overriding any device name or index given in the input filename.

All available devices can be enumerated by using -list_devices true, listing all device names
and corresponding indices.

There are two device name aliases:

"default"
Select the AVFoundation default device of the corresponding type.

"none"
Do not record the corresponding media type. This is equivalent to specifying an empty
device name or index.

Options

AVFoundation supports the following options:

-list_devices <TRUE|FALSE>
If set to true, a list of all available input devices is given showing all device names
and indices.

-video_device_index <INDEX>
Specify the video device by its index. Overrides anything given in the input filename.

-audio_device_index <INDEX>
Specify the audio device by its index. Overrides anything given in the input filename.

-pixel_format <FORMAT>
Request the video device to use a specific pixel format. If the specified format is not
supported, a list of available formats is given and the first one in this list is used
instead. Available pixel formats are: "monob, rgb555be, rgb555le, rgb565be, rgb565le,
rgb24, bgr24, 0rgb, bgr0, 0bgr, rgb0,
bgr48be, uyvy422, yuva444p, yuva444p16le, yuv444p, yuv422p16, yuv422p10, yuv444p10,
yuv420p, nv12, yuyv422, gray"

-framerate
Set the grabbing frame rate. Default is "ntsc", corresponding to a frame rate of
"30000/1001".

-video_size
Set the video frame size.

-capture_cursor
Capture the mouse pointer. Default is 0.

-capture_mouse_clicks
Capture the screen mouse clicks. Default is 0.

-capture_raw_data
Capture the raw device data. Default is 0. Using this option may result in receiving the
underlying data delivered to the AVFoundation framework. E.g. for muxed devices that
sends raw DV data to the framework (like tape-based camcorders), setting this option to
false results in extracted video frames captured in the designated pixel format only.
Setting this option to true results in receiving the raw DV stream untouched.

Examples

• Print the list of AVFoundation supported devices and exit:

$ ffmpeg -f avfoundation -list_devices true -i ""

• Record video from video device 0 and audio from audio device 0 into out.avi:

$ ffmpeg -f avfoundation -i "0:0" out.avi

• Record video from video device 2 and audio from audio device 1 into out.avi:

$ ffmpeg -f avfoundation -video_device_index 2 -i ":1" out.avi

• Record video from the system default video device using the pixel format bgr0 and do not
record any audio into out.avi:

$ ffmpeg -f avfoundation -pixel_format bgr0 -i "default:none" out.avi

• Record raw DV data from a suitable input device and write the output into out.dv:

$ ffmpeg -f avfoundation -capture_raw_data true -i "zr100:none" out.dv

bktr
BSD video input device.

Options

framerate
Set the frame rate.

video_size
Set the video frame size. Default is "vga".

standard
Available values are:

pal
ntsc
secam
paln
palm
ntscj

decklink
The decklink input device provides capture capabilities for Blackmagic DeckLink devices.

To enable this input device, you need the Blackmagic DeckLink SDK and you need to configure
with the appropriate "--extra-cflags" and "--extra-ldflags". On Windows, you need to run the
IDL files through widl.

DeckLink is very picky about the formats it supports. Pixel format of the input can be set
with raw_format. Framerate and video size must be determined for your device with
-list_formats 1. Audio sample rate is always 48 kHz and the number of channels can be 2, 8 or
16. Note that all audio channels are bundled in one single audio track.

Options

list_devices
If set to true, print a list of devices and exit. Defaults to false. This option is
deprecated, please use the "-sources" option of ffmpeg to list the available input
devices.

list_formats
If set to true, print a list of supported formats and exit. Defaults to false.

format_code <FourCC>
This sets the input video format to the format given by the FourCC. To see the supported
values of your device(s) use list_formats. Note that there is a FourCC 'pal ' that can
also be used as pal (3 letters). Default behavior is autodetection of the input video
format, if the hardware supports it.

raw_format
Set the pixel format of the captured video. Available values are:

auto
This is the default which means 8-bit YUV 422 or 8-bit ARGB if format autodetection
is used, 8-bit YUV 422 otherwise.

uyvy422
8-bit YUV 422.

yuv422p10
10-bit YUV 422.

argb
8-bit RGB.

bgra
8-bit RGB.

rgb10
10-bit RGB.

teletext_lines
If set to nonzero, an additional teletext stream will be captured from the vertical
ancillary data. Both SD PAL (576i) and HD (1080i or 1080p) sources are supported. In case
of HD sources, OP47 packets are decoded.

This option is a bitmask of the SD PAL VBI lines captured, specifically lines 6 to 22,
and lines 318 to 335. Line 6 is the LSB in the mask. Selected lines which do not contain
teletext information will be ignored. You can use the special all constant to select all
possible lines, or standard to skip lines 6, 318 and 319, which are not compatible with
all receivers.

For SD sources, ffmpeg needs to be compiled with "--enable-libzvbi". For HD sources, on
older (pre-4K) DeckLink card models you have to capture in 10 bit mode.

channels
Defines number of audio channels to capture. Must be 2, 8 or 16. Defaults to 2.

duplex_mode
Sets the decklink device duplex/profile mode. Must be unset, half, full,
one_sub_device_full, one_sub_device_half, two_sub_device_full, four_sub_device_half
Defaults to unset.

Note: DeckLink SDK 11.0 have replaced the duplex property by a profile property. For the
DeckLink Duo 2 and DeckLink Quad 2, a profile is shared between any 2 sub-devices that
utilize the same connectors. For the DeckLink 8K Pro, a profile is shared between all 4
sub-devices. So DeckLink 8K Pro support four profiles.

Valid profile modes for DeckLink 8K Pro(with DeckLink SDK >= 11.0): one_sub_device_full,
one_sub_device_half, two_sub_device_full, four_sub_device_half

Valid profile modes for DeckLink Quad 2 and DeckLink Duo 2: half, full

timecode_format
Timecode type to include in the frame and video stream metadata. Must be none, rp188vitc,
rp188vitc2, rp188ltc, rp188hfr, rp188any, vitc, vitc2, or serial. Defaults to none (not
included).

In order to properly support 50/60 fps timecodes, the ordering of the queried timecode
types for rp188any is HFR, VITC1, VITC2 and LTC for >30 fps content. Note that this is
slightly different to the ordering used by the DeckLink API, which is HFR, VITC1, LTC,
VITC2.

video_input
Sets the video input source. Must be unset, sdi, hdmi, optical_sdi, component, composite
or s_video. Defaults to unset.

audio_input
Sets the audio input source. Must be unset, embedded, aes_ebu, analog, analog_xlr,
analog_rca or microphone. Defaults to unset.

video_pts
Sets the video packet timestamp source. Must be video, audio, reference, wallclock or
abs_wallclock. Defaults to video.

audio_pts
Sets the audio packet timestamp source. Must be video, audio, reference, wallclock or
abs_wallclock. Defaults to audio.

draw_bars
If set to true, color bars are drawn in the event of a signal loss. Defaults to true.

queue_size
Sets maximum input buffer size in bytes. If the buffering reaches this value, incoming
frames will be dropped. Defaults to 1073741824.

audio_depth
Sets the audio sample bit depth. Must be 16 or 32. Defaults to 16.

decklink_copyts
If set to true, timestamps are forwarded as they are without removing the initial offset.
Defaults to false.

timestamp_align
Capture start time alignment in seconds. If set to nonzero, input frames are dropped till
the system timestamp aligns with configured value. Alignment difference of up to one
frame duration is tolerated. This is useful for maintaining input synchronization across
N different hardware devices deployed for 'N-way' redundancy. The system time of
different hardware devices should be synchronized with protocols such as NTP or PTP,
before using this option. Note that this method is not foolproof. In some border cases
input synchronization may not happen due to thread scheduling jitters in the OS. Either
sync could go wrong by 1 frame or in a rarer case timestamp_align seconds. Defaults to
0.

wait_for_tc (bool)
Drop frames till a frame with timecode is received. Sometimes serial timecode isn't
received with the first input frame. If that happens, the stored stream timecode will be
inaccurate. If this option is set to true, input frames are dropped till a frame with
timecode is received. Option timecode_format must be specified. Defaults to false.

enable_klv(bool)
If set to true, extracts KLV data from VANC and outputs KLV packets. KLV VANC packets
are joined based on MID and PSC fields and aggregated into one KLV packet. Defaults to
false.

Examples

• List input devices:

ffmpeg -sources decklink

• List supported formats:

ffmpeg -f decklink -list_formats 1 -i 'Intensity Pro'

• Capture video clip at 1080i50:

ffmpeg -format_code Hi50 -f decklink -i 'Intensity Pro' -c:a copy -c:v copy output.avi

• Capture video clip at 1080i50 10 bit:

ffmpeg -raw_format yuv422p10 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder' -c:a copy -c:v copy output.avi

• Capture video clip at 1080i50 with 16 audio channels:

ffmpeg -channels 16 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder' -c:a copy -c:v copy output.avi

dshow
Windows DirectShow input device.

DirectShow support is enabled when FFmpeg is built with the mingw-w64 project. Currently
only audio and video devices are supported.

Multiple devices may be opened as separate inputs, but they may also be opened on the same
input, which should improve synchronism between them.

The input name should be in the format:

<TYPE>=<NAME>[:<TYPE>=<NAME>]

where TYPE can be either audio or video, and NAME is the device's name or alternative name..

Options

If no options are specified, the device's defaults are used. If the device does not support
the requested options, it will fail to open.

video_size
Set the video size in the captured video.

framerate
Set the frame rate in the captured video.

sample_rate
Set the sample rate (in Hz) of the captured audio.

sample_size
Set the sample size (in bits) of the captured audio.

channels
Set the number of channels in the captured audio.

list_devices
If set to true, print a list of devices and exit.

list_options
If set to true, print a list of selected device's options and exit.

video_device_number
Set video device number for devices with the same name (starts at 0, defaults to 0).

audio_device_number
Set audio device number for devices with the same name (starts at 0, defaults to 0).

pixel_format
Select pixel format to be used by DirectShow. This may only be set when the video codec
is not set or set to rawvideo.

audio_buffer_size
Set audio device buffer size in milliseconds (which can directly impact latency,
depending on the device). Defaults to using the audio device's default buffer size
(typically some multiple of 500ms). Setting this value too low can degrade performance.
See also <http://msdn.microsoft.com/en-us/library/windows/desktop/dd377582(v=vs.85).aspx>

video_pin_name
Select video capture pin to use by name or alternative name.

audio_pin_name
Select audio capture pin to use by name or alternative name.

crossbar_video_input_pin_number
Select video input pin number for crossbar device. This will be routed to the crossbar
device's Video Decoder output pin. Note that changing this value can affect future
invocations (sets a new default) until system reboot occurs.

crossbar_audio_input_pin_number
Select audio input pin number for crossbar device. This will be routed to the crossbar
device's Audio Decoder output pin. Note that changing this value can affect future
invocations (sets a new default) until system reboot occurs.

show_video_device_dialog
If set to true, before capture starts, popup a display dialog to the end user, allowing
them to change video filter properties and configurations manually. Note that for
crossbar devices, adjusting values in this dialog may be needed at times to toggle
between PAL (25 fps) and NTSC (29.97) input frame rates, sizes, interlacing, etc.
Changing these values can enable different scan rates/frame rates and avoiding green bars
at the bottom, flickering scan lines, etc. Note that with some devices, changing these
properties can also affect future invocations (sets new defaults) until system reboot
occurs.

show_audio_device_dialog
If set to true, before capture starts, popup a display dialog to the end user, allowing
them to change audio filter properties and configurations manually.

show_video_crossbar_connection_dialog
If set to true, before capture starts, popup a display dialog to the end user, allowing
them to manually modify crossbar pin routings, when it opens a video device.

show_audio_crossbar_connection_dialog
If set to true, before capture starts, popup a display dialog to the end user, allowing
them to manually modify crossbar pin routings, when it opens an audio device.

show_analog_tv_tuner_dialog
If set to true, before capture starts, popup a display dialog to the end user, allowing
them to manually modify TV channels and frequencies.

show_analog_tv_tuner_audio_dialog
If set to true, before capture starts, popup a display dialog to the end user, allowing
them to manually modify TV audio (like mono vs. stereo, Language A,B or C).

audio_device_load
Load an audio capture filter device from file instead of searching it by name. It may
load additional parameters too, if the filter supports the serialization of its
properties to. To use this an audio capture source has to be specified, but it can be
anything even fake one.

audio_device_save
Save the currently used audio capture filter device and its parameters (if the filter
supports it) to a file. If a file with the same name exists it will be overwritten.

video_device_load
Load a video capture filter device from file instead of searching it by name. It may load
additional parameters too, if the filter supports the serialization of its properties to.
To use this a video capture source has to be specified, but it can be anything even fake
one.

video_device_save
Save the currently used video capture filter device and its parameters (if the filter
supports it) to a file. If a file with the same name exists it will be overwritten.

use_video_device_timestamps
If set to false, the timestamp for video frames will be derived from the wallclock
instead of the timestamp provided by the capture device. This allows working around
devices that provide unreliable timestamps.

Examples

• Print the list of DirectShow supported devices and exit:

$ ffmpeg -list_devices true -f dshow -i dummy

• Open video device Camera:

$ ffmpeg -f dshow -i video="Camera"

• Open second video device with name Camera:

$ ffmpeg -f dshow -video_device_number 1 -i video="Camera"

• Open video device Camera and audio device Microphone:

$ ffmpeg -f dshow -i video="Camera":audio="Microphone"

• Print the list of supported options in selected device and exit:

$ ffmpeg -list_options true -f dshow -i video="Camera"

• Specify pin names to capture by name or alternative name, specify alternative device
name:

$ ffmpeg -f dshow -audio_pin_name "Audio Out" -video_pin_name 2 -i video=video="@device_pnp_\\?\pci#ven_1a0a&dev_6200&subsys_62021461&rev_01#4&e2c7dd6&0&00e1#{65e8773d-8f56-11d0-a3b9-00a0c9223196}\{ca465100-deb0-4d59-818f-8c477184adf6}":audio="Microphone"

• Configure a crossbar device, specifying crossbar pins, allow user to adjust video capture
properties at startup:

$ ffmpeg -f dshow -show_video_device_dialog true -crossbar_video_input_pin_number 0
-crossbar_audio_input_pin_number 3 -i video="AVerMedia BDA Analog Capture":audio="AVerMedia BDA Analog Capture"

fbdev
Linux framebuffer input device.

The Linux framebuffer is a graphic hardware-independent abstraction layer to show graphics on
a computer monitor, typically on the console. It is accessed through a file device node,
usually /dev/fb0.

For more detailed information read the file Documentation/fb/framebuffer.txt included in the
Linux source tree.

See also <http://linux-fbdev.sourceforge.net/>, and fbset(1).

To record from the framebuffer device /dev/fb0 with ffmpeg:

ffmpeg -f fbdev -framerate 10 -i /dev/fb0 out.avi

You can take a single screenshot image with the command:

ffmpeg -f fbdev -framerate 1 -i /dev/fb0 -frames:v 1 screenshot.jpeg

Options

framerate
Set the frame rate. Default is 25.

gdigrab
Win32 GDI-based screen capture device.

This device allows you to capture a region of the display on Windows.

There are two options for the input filename:

desktop

title=<window_title>

The first option will capture the entire desktop, or a fixed region of the desktop. The
second option will instead capture the contents of a single window, regardless of its
position on the screen.

For example, to grab the entire desktop using ffmpeg:

ffmpeg -f gdigrab -framerate 6 -i desktop out.mpg

Grab a 640x480 region at position "10,20":

ffmpeg -f gdigrab -framerate 6 -offset_x 10 -offset_y 20 -video_size vga -i desktop out.mpg

Grab the contents of the window named "Calculator"

ffmpeg -f gdigrab -framerate 6 -i title=Calculator out.mpg

Options

draw_mouse
Specify whether to draw the mouse pointer. Use the value 0 to not draw the pointer.
Default value is 1.

framerate
Set the grabbing frame rate. Default value is "ntsc", corresponding to a frame rate of
"30000/1001".

show_region
Show grabbed region on screen.

If show_region is specified with 1, then the grabbing region will be indicated on screen.
With this option, it is easy to know what is being grabbed if only a portion of the
screen is grabbed.

Note that show_region is incompatible with grabbing the contents of a single window.

For example:

ffmpeg -f gdigrab -show_region 1 -framerate 6 -video_size cif -offset_x 10 -offset_y 20 -i desktop out.mpg

video_size
Set the video frame size. The default is to capture the full screen if desktop is
selected, or the full window size if title=window_title is selected.

offset_x
When capturing a region with video_size, set the distance from the left edge of the
screen or desktop.

Note that the offset calculation is from the top left corner of the primary monitor on
Windows. If you have a monitor positioned to the left of your primary monitor, you will
need to use a negative offset_x value to move the region to that monitor.

offset_y
When capturing a region with video_size, set the distance from the top edge of the screen
or desktop.

Note that the offset calculation is from the top left corner of the primary monitor on
Windows. If you have a monitor positioned above your primary monitor, you will need to
use a negative offset_y value to move the region to that monitor.

iec61883
FireWire DV/HDV input device using libiec61883.

To enable this input device, you need libiec61883, libraw1394 and libavc1394 installed on
your system. Use the configure option "--enable-libiec61883" to compile with the device
enabled.

The iec61883 capture device supports capturing from a video device connected via IEEE1394
(FireWire), using libiec61883 and the new Linux FireWire stack (juju). This is the default
DV/HDV input method in Linux Kernel 2.6.37 and later, since the old FireWire stack was
removed.

Specify the FireWire port to be used as input file, or "auto" to choose the first port
connected.

Options

dvtype
Override autodetection of DV/HDV. This should only be used if auto detection does not
work, or if usage of a different device type should be prohibited. Treating a DV device
as HDV (or vice versa) will not work and result in undefined behavior. The values auto,
dv and hdv are supported.

dvbuffer
Set maximum size of buffer for incoming data, in frames. For DV, this is an exact value.
For HDV, it is not frame exact, since HDV does not have a fixed frame size.

dvguid
Select the capture device by specifying its GUID. Capturing will only be performed from
the specified device and fails if no device with the given GUID is found. This is useful
to select the input if multiple devices are connected at the same time. Look at
/sys/bus/firewire/devices to find out the GUIDs.

Examples

• Grab and show the input of a FireWire DV/HDV device.

ffplay -f iec61883 -i auto

• Grab and record the input of a FireWire DV/HDV device, using a packet buffer of 100000
packets if the source is HDV.

ffmpeg -f iec61883 -i auto -dvbuffer 100000 out.mpg

jack
JACK input device.

To enable this input device during configuration you need libjack installed on your system.

A JACK input device creates one or more JACK writable clients, one for each audio channel,
with name client_name:input_N, where client_name is the name provided by the application, and
N is a number which identifies the channel. Each writable client will send the acquired data
to the FFmpeg input device.

Once you have created one or more JACK readable clients, you need to connect them to one or
more JACK writable clients.

To connect or disconnect JACK clients you can use the jack_connect and jack_disconnect
programs, or do it through a graphical interface, for example with qjackctl.

To list the JACK clients and their properties you can invoke the command jack_lsp.

Follows an example which shows how to capture a JACK readable client with ffmpeg.

# Create a JACK writable client with name "ffmpeg".
$ ffmpeg -f jack -i ffmpeg -y out.wav

# Start the sample jack_metro readable client.
$ jack_metro -b 120 -d 0.2 -f 4000

# List the current JACK clients.
$ jack_lsp -c
system:capture_1
system:capture_2
system:playback_1
system:playback_2
ffmpeg:input_1
metro:120_bpm

# Connect metro to the ffmpeg writable client.
$ jack_connect metro:120_bpm ffmpeg:input_1

For more information read: <http://jackaudio.org/>

Options

channels
Set the number of channels. Default is 2.

kmsgrab
KMS video input device.

Captures the KMS scanout framebuffer associated with a specified CRTC or plane as a DRM
object that can be passed to other hardware functions.

Requires either DRM master or CAP_SYS_ADMIN to run.

If you don't understand what all of that means, you probably don't want this. Look at
x11grab instead.

Options

device
DRM device to capture on. Defaults to /dev/dri/card0.

format
Pixel format of the framebuffer. This can be autodetected if you are running Linux 5.7
or later, but needs to be provided for earlier versions. Defaults to bgr0, which is the
most common format used by the Linux console and Xorg X server.

format_modifier
Format modifier to signal on output frames. This is necessary to import correctly into
some APIs. It can be autodetected if you are running Linux 5.7 or later, but will need
to be provided explicitly when needed in earlier versions. See the libdrm documentation
for possible values.

crtc_id
KMS CRTC ID to define the capture source. The first active plane on the given CRTC will
be used.

plane_id
KMS plane ID to define the capture source. Defaults to the first active plane found if
neither crtc_id nor plane_id are specified.

framerate
Framerate to capture at. This is not synchronised to any page flipping or framebuffer
changes - it just defines the interval at which the framebuffer is sampled. Sampling
faster than the framebuffer update rate will generate independent frames with the same
content. Defaults to 30.

Examples

• Capture from the first active plane, download the result to normal frames and encode.
This will only work if the framebuffer is both linear and mappable - if not, the result
may be scrambled or fail to download.

ffmpeg -f kmsgrab -i - -vf 'hwdownload,format=bgr0' output.mp4

• Capture from CRTC ID 42 at 60fps, map the result to VAAPI, convert to NV12 and encode as
H.264.

ffmpeg -crtc_id 42 -framerate 60 -f kmsgrab -i - -vf 'hwmap=derive_device=vaapi,scale_vaapi=w=1920:h=1080:format=nv12' -c:v h264_vaapi output.mp4

• To capture only part of a plane the output can be cropped - this can be used to capture a
single window, as long as it has a known absolute position and size. For example, to
capture and encode the middle quarter of a 1920x1080 plane:

ffmpeg -f kmsgrab -i - -vf 'hwmap=derive_device=vaapi,crop=960:540:480:270,scale_vaapi=960:540:nv12' -c:v h264_vaapi output.mp4

lavfi
Libavfilter input virtual device.

This input device reads data from the open output pads of a libavfilter filtergraph.

For each filtergraph open output, the input device will create a corresponding stream which
is mapped to the generated output. The filtergraph is specified through the option graph.

Options

graph
Specify the filtergraph to use as input. Each video open output must be labelled by a
unique string of the form "outN", where N is a number starting from 0 corresponding to
the mapped input stream generated by the device. The first unlabelled output is
automatically assigned to the "out0" label, but all the others need to be specified
explicitly.

The suffix "+subcc" can be appended to the output label to create an extra stream with
the closed captions packets attached to that output (experimental; only for EIA-608 /
CEA-708 for now). The subcc streams are created after all the normal streams, in the
order of the corresponding stream. For example, if there is "out19+subcc", "out7+subcc"
and up to "out42", the stream #43 is subcc for stream #7 and stream #44 is subcc for
stream #19.

If not specified defaults to the filename specified for the input device.

graph_file
Set the filename of the filtergraph to be read and sent to the other filters. Syntax of
the filtergraph is the same as the one specified by the option graph.

dumpgraph
Dump graph to stderr.

Examples

• Create a color video stream and play it back with ffplay:

ffplay -f lavfi -graph "color=c=pink [out0]" dummy

• As the previous example, but use filename for specifying the graph description, and omit
the "out0" label:

ffplay -f lavfi color=c=pink

• Create three different video test filtered sources and play them:

ffplay -f lavfi -graph "testsrc [out0]; testsrc,hflip [out1]; testsrc,negate [out2]" test3

• Read an audio stream from a file using the amovie source and play it back with ffplay:

ffplay -f lavfi "amovie=test.wav"

• Read an audio stream and a video stream and play it back with ffplay:

ffplay -f lavfi "movie=test.avi[out0];amovie=test.wav[out1]"

• Dump decoded frames to images and closed captions to a file (experimental):

ffmpeg -f lavfi -i "movie=test.ts[out0+subcc]" -map v frame%08d.png -map s -c copy -f rawvideo subcc.bin

libcdio
Audio-CD input device based on libcdio.

To enable this input device during configuration you need libcdio installed on your system.
It requires the configure option "--enable-libcdio".

This device allows playing and grabbing from an Audio-CD.

For example to copy with ffmpeg the entire Audio-CD in /dev/sr0, you may run the command:

ffmpeg -f libcdio -i /dev/sr0 cd.wav

Options

speed
Set drive reading speed. Default value is 0.

The speed is specified CD-ROM speed units. The speed is set through the libcdio
"cdio_cddap_speed_set" function. On many CD-ROM drives, specifying a value too large will
result in using the fastest speed.

paranoia_mode
Set paranoia recovery mode flags. It accepts one of the following values:

disable
verify
overlap
neverskip
full

Default value is disable.

For more information about the available recovery modes, consult the paranoia project
documentation.

libdc1394
IIDC1394 input device, based on libdc1394 and libraw1394.

Requires the configure option "--enable-libdc1394".

Options

framerate
Set the frame rate. Default is "ntsc", corresponding to a frame rate of "30000/1001".

pixel_format
Select the pixel format. Default is "uyvy422".

video_size
Set the video size given as a string such as "640x480" or "hd720". Default is "qvga".

openal
The OpenAL input device provides audio capture on all systems with a working OpenAL 1.1
implementation.

To enable this input device during configuration, you need OpenAL headers and libraries
installed on your system, and need to configure FFmpeg with "--enable-openal".

OpenAL headers and libraries should be provided as part of your OpenAL implementation, or as
an additional download (an SDK). Depending on your installation you may need to specify
additional flags via the "--extra-cflags" and "--extra-ldflags" for allowing the build system
to locate the OpenAL headers and libraries.

An incomplete list of OpenAL implementations follows:

Creative
The official Windows implementation, providing hardware acceleration with supported
devices and software fallback. See <http://openal.org/>.

OpenAL Soft
Portable, open source (LGPL) software implementation. Includes backends for the most
common sound APIs on the Windows, Linux, Solaris, and BSD operating systems. See
<http://kcat.strangesoft.net/openal.html>.

Apple
OpenAL is part of Core Audio, the official Mac OS X Audio interface. See
<http://developer.apple.com/technologies/mac/audio-and-video.html>

This device allows one to capture from an audio input device handled through OpenAL.

You need to specify the name of the device to capture in the provided filename. If the empty
string is provided, the device will automatically select the default device. You can get the
list of the supported devices by using the option list_devices.

Options

channels
Set the number of channels in the captured audio. Only the values 1 (monaural) and 2
(stereo) are currently supported. Defaults to 2.

sample_size
Set the sample size (in bits) of the captured audio. Only the values 8 and 16 are
currently supported. Defaults to 16.

sample_rate
Set the sample rate (in Hz) of the captured audio. Defaults to 44.1k.

list_devices
If set to true, print a list of devices and exit. Defaults to false.

Examples

Print the list of OpenAL supported devices and exit:

$ ffmpeg -list_devices true -f openal -i dummy out.ogg

Capture from the OpenAL device DR-BT101 via PulseAudio:

$ ffmpeg -f openal -i 'DR-BT101 via PulseAudio' out.ogg

Capture from the default device (note the empty string '' as filename):

$ ffmpeg -f openal -i '' out.ogg

Capture from two devices simultaneously, writing to two different files, within the same
ffmpeg command:

$ ffmpeg -f openal -i 'DR-BT101 via PulseAudio' out1.ogg -f openal -i 'ALSA Default' out2.ogg

Note: not all OpenAL implementations support multiple simultaneous capture - try the latest
OpenAL Soft if the above does not work.

oss
Open Sound System input device.

The filename to provide to the input device is the device node representing the OSS input
device, and is usually set to /dev/dsp.

For example to grab from /dev/dsp using ffmpeg use the command:

ffmpeg -f oss -i /dev/dsp /tmp/oss.wav

For more information about OSS see: <http://manuals.opensound.com/usersguide/dsp.html>

Options

sample_rate
Set the sample rate in Hz. Default is 48000.

channels
Set the number of channels. Default is 2.

pulse
PulseAudio input device.

To enable this output device you need to configure FFmpeg with "--enable-libpulse".

The filename to provide to the input device is a source device or the string "default"

To list the PulseAudio source devices and their properties you can invoke the command pactl
list sources.

More information about PulseAudio can be found on <http://www.pulseaudio.org>.

Options

server
Connect to a specific PulseAudio server, specified by an IP address. Default server is
used when not provided.

name
Specify the application name PulseAudio will use when showing active clients, by default
it is the "LIBAVFORMAT_IDENT" string.

stream_name
Specify the stream name PulseAudio will use when showing active streams, by default it is
"record".

sample_rate
Specify the samplerate in Hz, by default 48kHz is used.

channels
Specify the channels in use, by default 2 (stereo) is set.

frame_size
This option does nothing and is deprecated.

fragment_size
Specify the size in bytes of the minimal buffering fragment in PulseAudio, it will affect
the audio latency. By default it is set to 50 ms amount of data.

wallclock
Set the initial PTS using the current time. Default is 1.

Examples

Record a stream from default device:

ffmpeg -f pulse -i default /tmp/pulse.wav

sndio
sndio input device.

To enable this input device during configuration you need libsndio installed on your system.

The filename to provide to the input device is the device node representing the sndio input
device, and is usually set to /dev/audio0.

For example to grab from /dev/audio0 using ffmpeg use the command:

ffmpeg -f sndio -i /dev/audio0 /tmp/oss.wav

Options

sample_rate
Set the sample rate in Hz. Default is 48000.

channels
Set the number of channels. Default is 2.

video4linux2, v4l2
Video4Linux2 input video device.

"v4l2" can be used as alias for "video4linux2".

If FFmpeg is built with v4l-utils support (by using the "--enable-libv4l2" configure option),
it is possible to use it with the "-use_libv4l2" input device option.

The name of the device to grab is a file device node, usually Linux systems tend to
automatically create such nodes when the device (e.g. an USB webcam) is plugged into the
system, and has a name of the kind /dev/videoN, where N is a number associated to the device.

Video4Linux2 devices usually support a limited set of widthxheight sizes and frame rates. You
can check which are supported using -list_formats all for Video4Linux2 devices. Some
devices, like TV cards, support one or more standards. It is possible to list all the
supported standards using -list_standards all.

The time base for the timestamps is 1 microsecond. Depending on the kernel version and
configuration, the timestamps may be derived from the real time clock (origin at the Unix
Epoch) or the monotonic clock (origin usually at boot time, unaffected by NTP or manual
changes to the clock). The -timestamps abs or -ts abs option can be used to force conversion
into the real time clock.

Some usage examples of the video4linux2 device with ffmpeg and ffplay:

• List supported formats for a video4linux2 device:

ffplay -f video4linux2 -list_formats all /dev/video0

• Grab and show the input of a video4linux2 device:

ffplay -f video4linux2 -framerate 30 -video_size hd720 /dev/video0

• Grab and record the input of a video4linux2 device, leave the frame rate and size as
previously set:

ffmpeg -f video4linux2 -input_format mjpeg -i /dev/video0 out.mpeg

For more information about Video4Linux, check <http://linuxtv.org/>.

Options

standard
Set the standard. Must be the name of a supported standard. To get a list of the
supported standards, use the list_standards option.

channel
Set the input channel number. Default to -1, which means using the previously selected
channel.

video_size
Set the video frame size. The argument must be a string in the form WIDTHxHEIGHT or a
valid size abbreviation.

pixel_format
Select the pixel format (only valid for raw video input).

input_format
Set the preferred pixel format (for raw video) or a codec name. This option allows one
to select the input format, when several are available.

framerate
Set the preferred video frame rate.

list_formats
List available formats (supported pixel formats, codecs, and frame sizes) and exit.

Available values are:

all Show all available (compressed and non-compressed) formats.

raw Show only raw video (non-compressed) formats.

compressed
Show only compressed formats.

list_standards
List supported standards and exit.

Available values are:

all Show all supported standards.

timestamps, ts
Set type of timestamps for grabbed frames.

Available values are:

default
Use timestamps from the kernel.

abs Use absolute timestamps (wall clock).

mono2abs
Force conversion from monotonic to absolute timestamps.

Default value is "default".

use_libv4l2
Use libv4l2 (v4l-utils) conversion functions. Default is 0.

vfwcap
VfW (Video for Windows) capture input device.

The filename passed as input is the capture driver number, ranging from 0 to 9. You may use
"list" as filename to print a list of drivers. Any other filename will be interpreted as
device number 0.

Options

video_size
Set the video frame size.

framerate
Set the grabbing frame rate. Default value is "ntsc", corresponding to a frame rate of
"30000/1001".

x11grab
X11 video input device.

To enable this input device during configuration you need libxcb installed on your system. It
will be automatically detected during configuration.

This device allows one to capture a region of an X11 display.

The filename passed as input has the syntax:

[<hostname>]:<display_number>.<screen_number>[+<x_offset>,<y_offset>]

hostname:display_number.screen_number specifies the X11 display name of the screen to grab
from. hostname can be omitted, and defaults to "localhost". The environment variable DISPLAY
contains the default display name.

x_offset and y_offset specify the offsets of the grabbed area with respect to the top-left
border of the X11 screen. They default to 0.

Check the X11 documentation (e.g. man X) for more detailed information.

Use the xdpyinfo program for getting basic information about the properties of your X11
display (e.g. grep for "name" or "dimensions").

For example to grab from :0.0 using ffmpeg:

ffmpeg -f x11grab -framerate 25 -video_size cif -i :0.0 out.mpg

Grab at position "10,20":

ffmpeg -f x11grab -framerate 25 -video_size cif -i :0.0+10,20 out.mpg

Options

select_region
Specify whether to select the grabbing area graphically using the pointer. A value of 1
prompts the user to select the grabbing area graphically by clicking and dragging. A
single click with no dragging will select the whole screen. A region with zero width or
height will also select the whole screen. This option overwrites the video_size, grab_x,
and grab_y options. Default value is 0.

draw_mouse
Specify whether to draw the mouse pointer. A value of 0 specifies not to draw the
pointer. Default value is 1.

follow_mouse
Make the grabbed area follow the mouse. The argument can be "centered" or a number of
pixels PIXELS.

When it is specified with "centered", the grabbing region follows the mouse pointer and
keeps the pointer at the center of region; otherwise, the region follows only when the
mouse pointer reaches within PIXELS (greater than zero) to the edge of region.

For example:

ffmpeg -f x11grab -follow_mouse centered -framerate 25 -video_size cif -i :0.0 out.mpg

To follow only when the mouse pointer reaches within 100 pixels to edge:

ffmpeg -f x11grab -follow_mouse 100 -framerate 25 -video_size cif -i :0.0 out.mpg

framerate
Set the grabbing frame rate. Default value is "ntsc", corresponding to a frame rate of
"30000/1001".

show_region
Show grabbed region on screen.

If show_region is specified with 1, then the grabbing region will be indicated on screen.
With this option, it is easy to know what is being grabbed if only a portion of the
screen is grabbed.

region_border
Set the region border thickness if -show_region 1 is used. Range is 1 to 128 and default
is 3 (XCB-based x11grab only).

For example:

ffmpeg -f x11grab -show_region 1 -framerate 25 -video_size cif -i :0.0+10,20 out.mpg

With follow_mouse:

ffmpeg -f x11grab -follow_mouse centered -show_region 1 -framerate 25 -video_size cif -i :0.0 out.mpg

window_id
Grab this window, instead of the whole screen. Default value is 0, which maps to the
whole screen (root window).

The id of a window can be found using the xwininfo program, possibly with options -tree
and -root.

If the window is later enlarged, the new area is not recorded. Video ends when the window
is closed, unmapped (i.e., iconified) or shrunk beyond the video size (which defaults to
the initial window size).

This option disables options follow_mouse and select_region.

video_size
Set the video frame size. Default is the full desktop or window.

grab_x
grab_y
Set the grabbing region coordinates. They are expressed as offset from the top left
corner of the X11 window and correspond to the x_offset and y_offset parameters in the
device name. The default value for both options is 0.

OUTPUT DEVICES
Output devices are configured elements in FFmpeg that can write multimedia data to an output
device attached to your system.

When you configure your FFmpeg build, all the supported output devices are enabled by
default. You can list all available ones using the configure option "--list-outdevs".

You can disable all the output devices using the configure option "--disable-outdevs", and
selectively enable an output device using the option "--enable-outdev=OUTDEV", or you can
disable a particular input device using the option "--disable-outdev=OUTDEV".

The option "-devices" of the ff* tools will display the list of enabled output devices.

A description of the currently available output devices follows.

alsa
ALSA (Advanced Linux Sound Architecture) output device.

Examples

• Play a file on default ALSA device:

ffmpeg -i INPUT -f alsa default

• Play a file on soundcard 1, audio device 7:

ffmpeg -i INPUT -f alsa hw:1,7

AudioToolbox
AudioToolbox output device.

Allows native output to CoreAudio devices on OSX.

The output filename can be empty (or "-") to refer to the default system output device or a
number that refers to the device index as shown using: "-list_devices true".

Alternatively, the audio input device can be chosen by index using the

B<-audio_device_index E<lt>INDEXE<gt>>

, overriding any device name or index given in the input filename.

All available devices can be enumerated by using -list_devices true, listing all device
names, UIDs and corresponding indices.

Options

AudioToolbox supports the following options:

-audio_device_index <INDEX>
Specify the audio device by its index. Overrides anything given in the output filename.

Examples

• Print the list of supported devices and output a sine wave to the default device:

$ ffmpeg -f lavfi -i sine=r=44100 -f audiotoolbox -list_devices true -

• Output a sine wave to the device with the index 2, overriding any output filename:

$ ffmpeg -f lavfi -i sine=r=44100 -f audiotoolbox -audio_device_index 2 -

caca
CACA output device.

This output device allows one to show a video stream in CACA window. Only one CACA window is
allowed per application, so you can have only one instance of this output device in an
application.

To enable this output device you need to configure FFmpeg with "--enable-libcaca". libcaca
is a graphics library that outputs text instead of pixels.

For more information about libcaca, check: <http://caca.zoy.org/wiki/libcaca>

Options

window_title
Set the CACA window title, if not specified default to the filename specified for the
output device.

window_size
Set the CACA window size, can be a string of the form widthxheight or a video size
abbreviation. If not specified it defaults to the size of the input video.

driver
Set display driver.

algorithm
Set dithering algorithm. Dithering is necessary because the picture being rendered has
usually far more colours than the available palette. The accepted values are listed with
"-list_dither algorithms".

antialias
Set antialias method. Antialiasing smoothens the rendered image and avoids the commonly
seen staircase effect. The accepted values are listed with "-list_dither antialiases".

charset
Set which characters are going to be used when rendering text. The accepted values are
listed with "-list_dither charsets".

color
Set color to be used when rendering text. The accepted values are listed with
"-list_dither colors".

list_drivers
If set to true, print a list of available drivers and exit.

list_dither
List available dither options related to the argument. The argument must be one of
"algorithms", "antialiases", "charsets", "colors".

Examples

• The following command shows the ffmpeg output is an CACA window, forcing its size to
80x25:

ffmpeg -i INPUT -c:v rawvideo -pix_fmt rgb24 -window_size 80x25 -f caca -

• Show the list of available drivers and exit:

ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_drivers true -

• Show the list of available dither colors and exit:

ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_dither colors -

decklink
The decklink output device provides playback capabilities for Blackmagic DeckLink devices.

To enable this output device, you need the Blackmagic DeckLink SDK and you need to configure
with the appropriate "--extra-cflags" and "--extra-ldflags". On Windows, you need to run the
IDL files through widl.

DeckLink is very picky about the formats it supports. Pixel format is always uyvy422,
framerate, field order and video size must be determined for your device with -list_formats
1. Audio sample rate is always 48 kHz.

Options

list_devices
If set to true, print a list of devices and exit. Defaults to false. This option is
deprecated, please use the "-sinks" option of ffmpeg to list the available output
devices.

list_formats
If set to true, print a list of supported formats and exit. Defaults to false.

preroll
Amount of time to preroll video in seconds. Defaults to 0.5.

duplex_mode
Sets the decklink device duplex/profile mode. Must be unset, half, full,
one_sub_device_full, one_sub_device_half, two_sub_device_full, four_sub_device_half
Defaults to unset.

Valid profile modes for DeckLink 8K Pro(with DeckLink SDK >= 11.0): one_sub_device_full,
one_sub_device_half, two_sub_device_full, four_sub_device_half

Valid profile modes for DeckLink Quad 2 and DeckLink Duo 2: half, full

timing_offset
Sets the genlock timing pixel offset on the used output. Defaults to unset.

link
Sets the SDI video link configuration on the used output. Must be unset, single link SDI,
dual link SDI or quad link SDI. Defaults to unset.

sqd Enable Square Division Quad Split mode for Quad-link SDI output. Must be unset, true or
false. Defaults to unset.

level_a
Enable SMPTE Level A mode on the used output. Must be unset, true or false. Defaults to
unset.

vanc_queue_size
Sets maximum output buffer size in bytes for VANC data. If the buffering reaches this
value, outgoing VANC data will be dropped. Defaults to 1048576.

Examples

• List output devices:

ffmpeg -sinks decklink

• List supported formats:

ffmpeg -i test.avi -f decklink -list_formats 1 'DeckLink Mini Monitor'

• Play video clip:

ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 'DeckLink Mini Monitor'

• Play video clip with non-standard framerate or video size:

ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 -s 720x486 -r 24000/1001 'DeckLink Mini Monitor'

fbdev
Linux framebuffer output device.

For more detailed information read the file Documentation/fb/framebuffer.txt included in the
Linux source tree.

Options

xoffset
yoffset
Set x/y coordinate of top left corner. Default is 0.

Examples

Play a file on framebuffer device /dev/fb0. Required pixel format depends on current
framebuffer settings.

ffmpeg -re -i INPUT -c:v rawvideo -pix_fmt bgra -f fbdev /dev/fb0

See also <http://linux-fbdev.sourceforge.net/>, and fbset(1).

opengl
OpenGL output device.

To enable this output device you need to configure FFmpeg with "--enable-opengl".

This output device allows one to render to OpenGL context. Context may be provided by
application or default SDL window is created.

When device renders to external context, application must implement handlers for following
messages: "AV_DEV_TO_APP_CREATE_WINDOW_BUFFER" - create OpenGL context on current thread.
"AV_DEV_TO_APP_PREPARE_WINDOW_BUFFER" - make OpenGL context current.
"AV_DEV_TO_APP_DISPLAY_WINDOW_BUFFER" - swap buffers. "AV_DEV_TO_APP_DESTROY_WINDOW_BUFFER"
- destroy OpenGL context. Application is also required to inform a device about current
resolution by sending "AV_APP_TO_DEV_WINDOW_SIZE" message.

Options

background
Set background color. Black is a default.

no_window
Disables default SDL window when set to non-zero value. Application must provide OpenGL
context and both "window_size_cb" and "window_swap_buffers_cb" callbacks when set.

window_title
Set the SDL window title, if not specified default to the filename specified for the
output device. Ignored when no_window is set.

window_size
Set preferred window size, can be a string of the form widthxheight or a video size
abbreviation. If not specified it defaults to the size of the input video, downscaled
according to the aspect ratio. Mostly usable when no_window is not set.

Examples

Play a file on SDL window using OpenGL rendering:

ffmpeg -i INPUT -f opengl "window title"

oss
OSS (Open Sound System) output device.

pulse
PulseAudio output device.

To enable this output device you need to configure FFmpeg with "--enable-libpulse".

More information about PulseAudio can be found on <http://www.pulseaudio.org>

Options

server
Connect to a specific PulseAudio server, specified by an IP address. Default server is
used when not provided.

name
Specify the application name PulseAudio will use when showing active clients, by default
it is the "LIBAVFORMAT_IDENT" string.

stream_name
Specify the stream name PulseAudio will use when showing active streams, by default it is
set to the specified output name.

device
Specify the device to use. Default device is used when not provided. List of output
devices can be obtained with command pactl list sinks.

buffer_size
buffer_duration
Control the size and duration of the PulseAudio buffer. A small buffer gives more
control, but requires more frequent updates.

buffer_size specifies size in bytes while buffer_duration specifies duration in
milliseconds.

When both options are provided then the highest value is used (duration is recalculated
to bytes using stream parameters). If they are set to 0 (which is default), the device
will use the default PulseAudio duration value. By default PulseAudio set buffer duration
to around 2 seconds.

prebuf
Specify pre-buffering size in bytes. The server does not start with playback before at
least prebuf bytes are available in the buffer. By default this option is initialized to
the same value as buffer_size or buffer_duration (whichever is bigger).

minreq
Specify minimum request size in bytes. The server does not request less than minreq bytes
from the client, instead waits until the buffer is free enough to request more bytes at
once. It is recommended to not set this option, which will initialize this to a value
that is deemed sensible by the server.

Examples

Play a file on default device on default server:

ffmpeg -i INPUT -f pulse "stream name"

sdl
SDL (Simple DirectMedia Layer) output device.

"sdl2" can be used as alias for "sdl".

This output device allows one to show a video stream in an SDL window. Only one SDL window is
allowed per application, so you can have only one instance of this output device in an
application.

To enable this output device you need libsdl installed on your system when configuring your
build.

For more information about SDL, check: <http://www.libsdl.org/>

Options

window_borderless
Set SDL window border off. Default value is 0 (enable window border).

window_enable_quit
Enable quit action (using window button or keyboard key) when non-zero value is provided.
Default value is 1 (enable quit action).

window_fullscreen
Set fullscreen mode when non-zero value is provided. Default value is zero.

window_size
Set the SDL window size, can be a string of the form widthxheight or a video size
abbreviation. If not specified it defaults to the size of the input video, downscaled
according to the aspect ratio.

window_title
Set the SDL window title, if not specified default to the filename specified for the
output device.

window_x
window_y
Set the position of the window on the screen.

Interactive commands

The window created by the device can be controlled through the following interactive
commands.

q, ESC
Quit the device immediately.

Examples

The following command shows the ffmpeg output is an SDL window, forcing its size to the qcif
format:

ffmpeg -i INPUT -c:v rawvideo -pix_fmt yuv420p -window_size qcif -f sdl "SDL output"

sndio
sndio audio output device.

v4l2
Video4Linux2 output device.

xv
XV (XVideo) output device.

This output device allows one to show a video stream in a X Window System window.

Options

display_name
Specify the hardware display name, which determines the display and communications domain
to be used.

The display name or DISPLAY environment variable can be a string in the format
hostname[:number[.screen_number]].

hostname specifies the name of the host machine on which the display is physically
attached. number specifies the number of the display server on that host machine.
screen_number specifies the screen to be used on that server.

If unspecified, it defaults to the value of the DISPLAY environment variable.

For example, "dual-headed:0.1" would specify screen 1 of display 0 on the machine named
``dual-headed''.

Check the X11 specification for more detailed information about the display name format.

window_id
When set to non-zero value then device doesn't create new window, but uses existing one
with provided window_id. By default this options is set to zero and device creates its
own window.

window_size
Set the created window size, can be a string of the form widthxheight or a video size
abbreviation. If not specified it defaults to the size of the input video. Ignored when
window_id is set.

window_x
window_y
Set the X and Y window offsets for the created window. They are both set to 0 by default.
The values may be ignored by the window manager. Ignored when window_id is set.

window_title
Set the window title, if not specified default to the filename specified for the output
device. Ignored when window_id is set.

For more information about XVideo see <http://www.x.org/>.

Examples

• Decode, display and encode video input with ffmpeg at the same time:

ffmpeg -i INPUT OUTPUT -f xv display

• Decode and display the input video to multiple X11 windows:

ffmpeg -i INPUT -f xv normal -vf negate -f xv negated

RESAMPLER OPTIONS
The audio resampler supports the following named options.

Options may be set by specifying -option value in the FFmpeg tools, option=value for the
aresample filter, by setting the value explicitly in the "SwrContext" options or using the
libavutil/opt.h API for programmatic use.

uchl, used_chlayout
Set used input channel layout. Default is unset. This option is only used for special
remapping.

isr, in_sample_rate
Set the input sample rate. Default value is 0.

osr, out_sample_rate
Set the output sample rate. Default value is 0.

isf, in_sample_fmt
Specify the input sample format. It is set by default to "none".

osf, out_sample_fmt
Specify the output sample format. It is set by default to "none".

tsf, internal_sample_fmt
Set the internal sample format. Default value is "none". This will automatically be
chosen when it is not explicitly set.

ichl, in_chlayout
ochl, out_chlayout
Set the input/output channel layout.

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

clev, center_mix_level
Set the center mix level. It is a value expressed in deciBel, and must be in the interval
[-32,32].

slev, surround_mix_level
Set the surround mix level. It is a value expressed in deciBel, and must be in the
interval [-32,32].

lfe_mix_level
Set LFE mix into non LFE level. It is used when there is a LFE input but no LFE output.
It is a value expressed in deciBel, and must be in the interval [-32,32].

rmvol, rematrix_volume
Set rematrix volume. Default value is 1.0.

rematrix_maxval
Set maximum output value for rematrixing. This can be used to prevent clipping vs.
preventing volume reduction. A value of 1.0 prevents clipping.

flags, swr_flags
Set flags used by the converter. Default value is 0.

It supports the following individual flags:

res force resampling, this flag forces resampling to be used even when the input and
output sample rates match.

dither_scale
Set the dither scale. Default value is 1.

dither_method
Set dither method. Default value is 0.

Supported values:

rectangular
select rectangular dither

triangular
select triangular dither

triangular_hp
select triangular dither with high pass

lipshitz
select Lipshitz noise shaping dither.

shibata
select Shibata noise shaping dither.

low_shibata
select low Shibata noise shaping dither.

high_shibata
select high Shibata noise shaping dither.

f_weighted
select f-weighted noise shaping dither

modified_e_weighted
select modified-e-weighted noise shaping dither

improved_e_weighted
select improved-e-weighted noise shaping dither

resampler
Set resampling engine. Default value is swr.

Supported values:

swr select the native SW Resampler; filter options precision and cheby are not applicable
in this case.

soxr
select the SoX Resampler (where available); compensation, and filter options
filter_size, phase_shift, exact_rational, filter_type & kaiser_beta, are not
applicable in this case.

filter_size
For swr only, set resampling filter size, default value is 32.

phase_shift
For swr only, set resampling phase shift, default value is 10, and must be in the
interval [0,30].

linear_interp
Use linear interpolation when enabled (the default). Disable it if you want to preserve
speed instead of quality when exact_rational fails.

exact_rational
For swr only, when enabled, try to use exact phase_count based on input and output sample
rate. However, if it is larger than "1 << phase_shift", the phase_count will be "1 <<
phase_shift" as fallback. Default is enabled.

cutoff
Set cutoff frequency (swr: 6dB point; soxr: 0dB point) ratio; must be a float value
between 0 and 1. Default value is 0.97 with swr, and 0.91 with soxr (which, with a
sample-rate of 44100, preserves the entire audio band to 20kHz).

precision
For soxr only, the precision in bits to which the resampled signal will be calculated.
The default value of 20 (which, with suitable dithering, is appropriate for a destination
bit-depth of 16) gives SoX's 'High Quality'; a value of 28 gives SoX's 'Very High
Quality'.

cheby
For soxr only, selects passband rolloff none (Chebyshev) & higher-precision approximation
for 'irrational' ratios. Default value is 0.

async
For swr only, simple 1 parameter audio sync to timestamps using stretching, squeezing,
filling and trimming. Setting this to 1 will enable filling and trimming, larger values
represent the maximum amount in samples that the data may be stretched or squeezed for
each second. Default value is 0, thus no compensation is applied to make the samples
match the audio timestamps.

first_pts
For swr only, assume the first pts should be this value. The time unit is 1 / sample
rate. 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 silence if an audio stream
starts after the video stream or to trim any samples with a negative pts due to encoder
delay.

min_comp
For swr only, set the minimum difference between timestamps and audio data (in seconds)
to trigger stretching/squeezing/filling or trimming of the data to make it match the
timestamps. The default is that stretching/squeezing/filling and trimming is disabled
(min_comp = "FLT_MAX").

min_hard_comp
For swr only, set the minimum difference between timestamps and audio data (in seconds)
to trigger adding/dropping samples to make it match the timestamps. This option
effectively is a threshold to select between hard (trim/fill) and soft (squeeze/stretch)
compensation. Note that all compensation is by default disabled through min_comp. The
default is 0.1.

comp_duration
For swr only, set duration (in seconds) over which data is stretched/squeezed to make it
match the timestamps. Must be a non-negative double float value, default value is 1.0.

max_soft_comp
For swr only, set maximum factor by which data is stretched/squeezed to make it match the
timestamps. Must be a non-negative double float value, default value is 0.

matrix_encoding
Select matrixed stereo encoding.

It accepts the following values:

none
select none

dolby
select Dolby

dplii
select Dolby Pro Logic II

Default value is "none".

filter_type
For swr only, select resampling filter type. This only affects resampling operations.

It accepts the following values:

cubic
select cubic

blackman_nuttall
select Blackman Nuttall windowed sinc

kaiser
select Kaiser windowed sinc

kaiser_beta
For swr only, set Kaiser window beta value. Must be a double float value in the interval
[2,16], default value is 9.

output_sample_bits
For swr only, set number of used output sample bits for dithering. Must be an integer in
the interval [0,64], default value is 0, which means it's not used.

SCALER OPTIONS
The video scaler supports the following named options.

Options may be set by specifying -option value in the FFmpeg tools, with a few API-only
exceptions noted below. For programmatic use, they can be set explicitly in the "SwsContext"
options or through the libavutil/opt.h API.

sws_flags
Set the scaler flags. This is also used to set the scaling algorithm. Only a single
algorithm should be selected. Default value is bicubic.

It accepts the following values:

fast_bilinear
Select fast bilinear scaling algorithm.

bilinear
Select bilinear scaling algorithm.

bicubic
Select bicubic scaling algorithm.

experimental
Select experimental scaling algorithm.

neighbor
Select nearest neighbor rescaling algorithm.

area
Select averaging area rescaling algorithm.

bicublin
Select bicubic scaling algorithm for the luma component, bilinear for chroma
components.

gauss
Select Gaussian rescaling algorithm.

sinc
Select sinc rescaling algorithm.

lanczos
Select Lanczos rescaling algorithm. The default width (alpha) is 3 and can be changed
by setting "param0".

spline
Select natural bicubic spline rescaling algorithm.

print_info
Enable printing/debug logging.

accurate_rnd
Enable accurate rounding.

full_chroma_int
Enable full chroma interpolation.

full_chroma_inp
Select full chroma input.

bitexact
Enable bitexact output.

srcw (API only)
Set source width.

srch (API only)
Set source height.

dstw (API only)
Set destination width.

dsth (API only)
Set destination height.

src_format (API only)
Set source pixel format (must be expressed as an integer).

dst_format (API only)
Set destination pixel format (must be expressed as an integer).

src_range (boolean)
If value is set to 1, indicates source is full range. Default value is 0, which indicates
source is limited range.

dst_range (boolean)
If value is set to 1, enable full range for destination. Default value is 0, which
enables limited range.

param0, param1
Set scaling algorithm parameters. The specified values are specific of some scaling
algorithms and ignored by others. The specified values are floating point number values.

sws_dither
Set the dithering algorithm. Accepts one of the following values. Default value is auto.

auto
automatic choice

none
no dithering

bayer
bayer dither

ed error diffusion dither

a_dither
arithmetic dither, based using addition

x_dither
arithmetic dither, based using xor (more random/less apparent patterning that
a_dither).

alphablend
Set the alpha blending to use when the input has alpha but the output does not. Default
value is none.

uniform_color
Blend onto a uniform background color

checkerboard
Blend onto a checkerboard

none
No blending

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.

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.

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.

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.

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.

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.

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 ':'.

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.

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.

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.

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.

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.

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:

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:

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

See:
<https://www.researchgate.net/publication/275213614_A_New_Color_Correction_Method_for_Underwater_Imaging>

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.