Ubuntu Manpage: ffmpeg - ffmpeg media converter

Provided by: ffmpeg_6.1.1-3ubuntu5_amd64

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.

EXAMPLES

   Video and Audio grabbing
       If you specify the input format and device then ffmpeg can grab video and audio directly.

               ffmpeg -f oss -i /dev/dsp -f video4linux2 -i /dev/video0 /tmp/out.mpg

       Or with an ALSA audio source (mono input, card id 1) instead of OSS:

               ffmpeg -f alsa -ac 1 -i hw:1 -f video4linux2 -i /dev/video0 /tmp/out.mpg

       Note that you must activate the right video source and channel before launching ffmpeg with any TV viewer
       such as <http://linux.bytesex.org/xawtv/> by Gerd Knorr. You also have to set the audio recording  levels
       correctly with a standard mixer.

   X11 grabbing
       Grab the X11 display with ffmpeg via

               ffmpeg -f x11grab -video_size cif -framerate 25 -i :0.0 /tmp/out.mpg

       0.0 is display.screen number of your X11 server, same as the DISPLAY environment variable.

               ffmpeg -f x11grab -video_size cif -framerate 25 -i :0.0+10,20 /tmp/out.mpg

       0.0  is  display.screen  number  of  your X11 server, same as the DISPLAY environment variable. 10 is the
       x-offset and 20 the y-offset for the grabbing.

   Video and Audio file format conversion
       Any supported file format and protocol can serve as input to ffmpeg:

       Examples:

       •   You can use YUV files as input:

                   ffmpeg -i /tmp/test%d.Y /tmp/out.mpg

           It will use the files:

                   /tmp/test0.Y, /tmp/test0.U, /tmp/test0.V,
                   /tmp/test1.Y, /tmp/test1.U, /tmp/test1.V, etc...

           The Y files use twice the resolution of the U and V files. They are raw files, without  header.  They
           can  be  generated  by  all decent video decoders. You must specify the size of the image with the -s
           option if ffmpeg cannot guess it.

       •   You can input from a raw YUV420P file:

                   ffmpeg -i /tmp/test.yuv /tmp/out.avi

           test.yuv is a file containing raw YUV planar data. Each frame is composed of the Y plane followed  by
           the U and V planes at half vertical and horizontal resolution.

       •   You can output to a raw YUV420P file:

                   ffmpeg -i mydivx.avi hugefile.yuv

       •   You can set several input files and output files:

                   ffmpeg -i /tmp/a.wav -s 640x480 -i /tmp/a.yuv /tmp/a.mpg

           Converts the audio file a.wav and the raw YUV video file a.yuv to MPEG file a.mpg.

       •   You can also do audio and video conversions at the same time:

                   ffmpeg -i /tmp/a.wav -ar 22050 /tmp/a.mp2

           Converts a.wav to MPEG audio at 22050 Hz sample rate.

       •   You  can  encode to several formats at the same time and define a mapping from input stream to output
           streams:

                   ffmpeg -i /tmp/a.wav -map 0:a -b:a 64k /tmp/a.mp2 -map 0:a -b:a 128k /tmp/b.mp2

           Converts a.wav to a.mp2 at 64 kbits and to b.mp2 at 128  kbits.  '-map  file:index'  specifies  which
           input stream is used for each output stream, in the order of the definition of output streams.

       •   You can transcode decrypted VOBs:

                   ffmpeg -i snatch_1.vob -f avi -c:v mpeg4 -b:v 800k -g 300 -bf 2 -c:a libmp3lame -b:a 128k snatch.avi

           This  is  a  typical DVD ripping example; the input is a VOB file, the output an AVI file with MPEG-4
           video and MP3 audio. Note that in this command  we  use  B-frames  so  the  MPEG-4  stream  is  DivX5
           compatible,  and  GOP  size  is  300  which means one intra frame every 10 seconds for 29.97fps input
           video. Furthermore, the audio stream is MP3-encoded so you need to enable  LAME  support  by  passing
           "--enable-libmp3lame"  to  configure.   The mapping is particularly useful for DVD transcoding to get
           the desired audio language.

           NOTE: To see the supported input formats, use "ffmpeg -demuxers".

       •   You can extract images from a video, or create a video from many images:

           For extracting images from a video:

                   ffmpeg -i foo.avi -r 1 -s WxH -f image2 foo-%03d.jpeg

           This will extract one video frame per second from the video and  will  output  them  in  files  named
           foo-001.jpeg, foo-002.jpeg, etc. Images will be rescaled to fit the new WxH values.

           If  you want to extract just a limited number of frames, you can use the above command in combination
           with the "-frames:v" or "-t" option, or in combination with -ss to start extracting  from  a  certain
           point in time.

           For creating a video from many images:

                   ffmpeg -f image2 -framerate 12 -i foo-%03d.jpeg -s WxH foo.avi

           The  syntax  "foo-%03d.jpeg"  specifies  to use a decimal number composed of three digits padded with
           zeroes to express the sequence number. It is the same syntax supported by the C printf function,  but
           only formats accepting a normal integer are suitable.

           When  importing an image sequence, -i also supports expanding shell-like wildcard patterns (globbing)
           internally, by selecting the image2-specific "-pattern_type glob" option.

           For example, for creating a video from filenames matching the glob pattern "foo-*.jpeg":

                   ffmpeg -f image2 -pattern_type glob -framerate 12 -i 'foo-*.jpeg' -s WxH foo.avi

       •   You can put many streams of the same type in the output:

                   ffmpeg -i test1.avi -i test2.avi -map 1:1 -map 1:0 -map 0:1 -map 0:0 -c copy -y test12.nut

           The resulting output file test12.nut will contain the first four streams  from  the  input  files  in
           reverse order.

       •   To force CBR video output:

                   ffmpeg -i myfile.avi -b 4000k -minrate 4000k -maxrate 4000k -bufsize 1835k out.m2v

       •   The  four  options  lmin,  lmax,  mblmin and mblmax use 'lambda' units, but you may use the QP2LAMBDA
           constant to easily convert from 'q' units:

                   ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext

SYNTAX

       This section documents the syntax and formats employed by the FFmpeg libraries and tools.

   Quoting and escaping
       FFmpeg adopts the following quoting and escaping mechanism, unless explicitly  specified.  The  following
       rules are applied:

       •   '  and  \  are  special characters (respectively used for quoting and escaping). In addition to them,
           there might be other special characters depending on the  specific  syntax  where  the  escaping  and
           quoting are employed.

       •   A special character is escaped by prefixing it with a \.

       •   All characters enclosed between '' are included literally in the parsed string. The quote character '
           itself cannot be quoted, so you may need to close the quote and escape it.

       •   Leading and trailing whitespaces, unless escaped or quoted, are removed from the parsed string.

       Note  that  you may need to add a second level of escaping when using the command line or a script, which
       depends on the syntax of the adopted shell language.

       The function "av_get_token" defined in libavutil/avstring.h can be  used  to  parse  a  token  quoted  or
       escaped according to the rules defined above.

       The  tool  tools/ffescape in the FFmpeg source tree can be used to automatically quote or escape a string
       in a script.

       Examples

       •   Escape the string "Crime d'Amour" containing the "'" special character:

                   Crime d\'Amour

       •   The string above contains a quote, so the "'" needs to be escaped when quoting it:

                   'Crime d'\''Amour'

       •   Include leading or trailing whitespaces using quoting:

                   '  this string starts and ends with whitespaces  '

       •   Escaping and quoting can be mixed together:

                   ' The string '\'string\'' is a string '

       •   To include a literal \ you can use either escaping or quoting:

                   'c:\foo' can be written as c:\\foo

   Date
       The accepted syntax is:

               [(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH:MM:SS[.m...]]])|(HHMMSS[.m...]]]))[Z]
               now

       If the value is "now" it takes the current time.

       Time is local time unless Z is appended, in which case it is interpreted as UTC.  If  the  year-month-day
       part is not specified it takes the current year-month-day.

   Time duration
       There are two accepted syntaxes for expressing time duration.

               [-][<HH>:]<MM>:<SS>[.<m>...]

       HH  expresses  the number of hours, MM the number of minutes for a maximum of 2 digits, and SS the number
       of seconds for a maximum of 2 digits. The m at the end expresses decimal value for SS.

       or

               [-]<S>+[.<m>...][s|ms|us]

       S expresses the number of seconds, with the optional decimal part m.  The optional literal suffixes s, ms
       or us indicate to interpret the value as seconds, milliseconds or microseconds, respectively.

       In both expressions, the optional - indicates negative duration.

       Examples

       The following examples are all valid time duration:

       55  55 seconds

       0.2 0.2 seconds

       200ms
           200 milliseconds, that's 0.2s

       200000us
           200000 microseconds, that's 0.2s

       12:03:45
           12 hours, 03 minutes and 45 seconds

       23.189
           23.189 seconds

   Video size
       Specify the size of the sourced video, it may be a string of the form widthxheight, or the name of a size
       abbreviation.

       The following abbreviations are recognized:

       ntsc
           720x480

       pal 720x576

       qntsc
           352x240

       qpal
           352x288

       sntsc
           640x480

       spal
           768x576

       film
           352x240

       ntsc-film
           352x240

       sqcif
           128x96

       qcif
           176x144

       cif 352x288

       4cif
           704x576

       16cif
           1408x1152

       qqvga
           160x120

       qvga
           320x240

       vga 640x480

       svga
           800x600

       xga 1024x768

       uxga
           1600x1200

       qxga
           2048x1536

       sxga
           1280x1024

       qsxga
           2560x2048

       hsxga
           5120x4096

       wvga
           852x480

       wxga
           1366x768

       wsxga
           1600x1024

       wuxga
           1920x1200

       woxga
           2560x1600

       wqsxga
           3200x2048

       wquxga
           3840x2400

       whsxga
           6400x4096

       whuxga
           7680x4800

       cga 320x200

       ega 640x350

       hd480
           852x480

       hd720
           1280x720

       hd1080
           1920x1080

       2k  2048x1080

       2kflat
           1998x1080

       2kscope
           2048x858

       4k  4096x2160

       4kflat
           3996x2160

       4kscope
           4096x1716

       nhd 640x360

       hqvga
           240x160

       wqvga
           400x240

       fwqvga
           432x240

       hvga
           480x320

       qhd 960x540

       2kdci
           2048x1080

       4kdci
           4096x2160

       uhd2160
           3840x2160

       uhd4320
           7680x4320

   Video rate
       Specify the frame rate of a video, expressed as the number of frames generated per second. It has to be a
       string in the format frame_rate_num/frame_rate_den, an integer number, a float number or  a  valid  video
       frame rate abbreviation.

       The following abbreviations are recognized:

       ntsc
           30000/1001

       pal 25/1

       qntsc
           30000/1001

       qpal
           25/1

       sntsc
           30000/1001

       spal
           25/1

       film
           24/1

       ntsc-film
           24000/1001

   Ratio
       A ratio can be expressed as an expression, or in the form numerator:denominator.

       Note  that  a ratio with infinite (1/0) or negative value is considered valid, so you should check on the
       returned value if you want to exclude those values.

       The undefined value can be expressed using the "0:0" string.

   Color
       It can be the name of a color as defined below (case insensitive match) or a "[0x|#]RRGGBB[AA]" sequence,
       possibly followed by @ and a string representing the alpha component.

       The alpha component may be a string composed by "0x" followed by  an  hexadecimal  number  or  a  decimal
       number between 0.0 and 1.0, which represents the opacity value (0x00 or 0.0 means completely transparent,
       0xff or 1.0 completely opaque). If the alpha component is not specified then 0xff is assumed.

       The string random will result in a random color.

       The following names of colors are recognized:

       AliceBlue
           0xF0F8FF

       AntiqueWhite
           0xFAEBD7

       Aqua
           0x00FFFF

       Aquamarine
           0x7FFFD4

       Azure
           0xF0FFFF

       Beige
           0xF5F5DC

       Bisque
           0xFFE4C4

       Black
           0x000000

       BlanchedAlmond
           0xFFEBCD

       Blue
           0x0000FF

       BlueViolet
           0x8A2BE2

       Brown
           0xA52A2A

       BurlyWood
           0xDEB887

       CadetBlue
           0x5F9EA0

       Chartreuse
           0x7FFF00

       Chocolate
           0xD2691E

       Coral
           0xFF7F50

       CornflowerBlue
           0x6495ED

       Cornsilk
           0xFFF8DC

       Crimson
           0xDC143C

       Cyan
           0x00FFFF

       DarkBlue
           0x00008B

       DarkCyan
           0x008B8B

       DarkGoldenRod
           0xB8860B

       DarkGray
           0xA9A9A9

       DarkGreen
           0x006400

       DarkKhaki
           0xBDB76B

       DarkMagenta
           0x8B008B

       DarkOliveGreen
           0x556B2F

       Darkorange
           0xFF8C00

       DarkOrchid
           0x9932CC

       DarkRed
           0x8B0000

       DarkSalmon
           0xE9967A

       DarkSeaGreen
           0x8FBC8F

       DarkSlateBlue
           0x483D8B

       DarkSlateGray
           0x2F4F4F

       DarkTurquoise
           0x00CED1

       DarkViolet
           0x9400D3

       DeepPink
           0xFF1493

       DeepSkyBlue
           0x00BFFF

       DimGray
           0x696969

       DodgerBlue
           0x1E90FF

       FireBrick
           0xB22222

       FloralWhite
           0xFFFAF0

       ForestGreen
           0x228B22

       Fuchsia
           0xFF00FF

       Gainsboro
           0xDCDCDC

       GhostWhite
           0xF8F8FF

       Gold
           0xFFD700

       GoldenRod
           0xDAA520

       Gray
           0x808080

       Green
           0x008000

       GreenYellow
           0xADFF2F

       HoneyDew
           0xF0FFF0

       HotPink
           0xFF69B4

       IndianRed
           0xCD5C5C

       Indigo
           0x4B0082

       Ivory
           0xFFFFF0

       Khaki
           0xF0E68C

       Lavender
           0xE6E6FA

       LavenderBlush
           0xFFF0F5

       LawnGreen
           0x7CFC00

       LemonChiffon
           0xFFFACD

       LightBlue
           0xADD8E6

       LightCoral
           0xF08080

       LightCyan
           0xE0FFFF

       LightGoldenRodYellow
           0xFAFAD2

       LightGreen
           0x90EE90

       LightGrey
           0xD3D3D3

       LightPink
           0xFFB6C1

       LightSalmon
           0xFFA07A

       LightSeaGreen
           0x20B2AA

       LightSkyBlue
           0x87CEFA

       LightSlateGray
           0x778899

       LightSteelBlue
           0xB0C4DE

       LightYellow
           0xFFFFE0

       Lime
           0x00FF00

       LimeGreen
           0x32CD32

       Linen
           0xFAF0E6

       Magenta
           0xFF00FF

       Maroon
           0x800000

       MediumAquaMarine
           0x66CDAA

       MediumBlue
           0x0000CD

       MediumOrchid
           0xBA55D3

       MediumPurple
           0x9370D8

       MediumSeaGreen
           0x3CB371

       MediumSlateBlue
           0x7B68EE

       MediumSpringGreen
           0x00FA9A

       MediumTurquoise
           0x48D1CC

       MediumVioletRed
           0xC71585

       MidnightBlue
           0x191970

       MintCream
           0xF5FFFA

       MistyRose
           0xFFE4E1

       Moccasin
           0xFFE4B5

       NavajoWhite
           0xFFDEAD

       Navy
           0x000080

       OldLace
           0xFDF5E6

       Olive
           0x808000

       OliveDrab
           0x6B8E23

       Orange
           0xFFA500

       OrangeRed
           0xFF4500

       Orchid
           0xDA70D6

       PaleGoldenRod
           0xEEE8AA

       PaleGreen
           0x98FB98

       PaleTurquoise
           0xAFEEEE

       PaleVioletRed
           0xD87093

       PapayaWhip
           0xFFEFD5

       PeachPuff
           0xFFDAB9

       Peru
           0xCD853F

       Pink
           0xFFC0CB

       Plum
           0xDDA0DD

       PowderBlue
           0xB0E0E6

       Purple
           0x800080

       Red 0xFF0000

       RosyBrown
           0xBC8F8F

       RoyalBlue
           0x4169E1

       SaddleBrown
           0x8B4513

       Salmon
           0xFA8072

       SandyBrown
           0xF4A460

       SeaGreen
           0x2E8B57

       SeaShell
           0xFFF5EE

       Sienna
           0xA0522D

       Silver
           0xC0C0C0

       SkyBlue
           0x87CEEB

       SlateBlue
           0x6A5ACD

       SlateGray
           0x708090

       Snow
           0xFFFAFA

       SpringGreen
           0x00FF7F

       SteelBlue
           0x4682B4

       Tan 0xD2B48C

       Teal
           0x008080

       Thistle
           0xD8BFD8

       Tomato
           0xFF6347

       Turquoise
           0x40E0D0

       Violet
           0xEE82EE

       Wheat
           0xF5DEB3

       White
           0xFFFFFF

       WhiteSmoke
           0xF5F5F5

       Yellow
           0xFFFF00

       YellowGreen
           0x9ACD32

   Channel Layout
       A  channel  layout  specifies the spatial disposition of the channels in a multi-channel audio stream. To
       specify a channel layout, FFmpeg makes use of a special syntax.

       Individual channels are identified by an id, as given by the table below:

       FL  front left

       FR  front right

       FC  front center

       LFE low frequency

       BL  back left

       BR  back right

       FLC front left-of-center

       FRC front right-of-center

       BC  back center

       SL  side left

       SR  side right

       TC  top center

       TFL top front left

       TFC top front center

       TFR top front right

       TBL top back left

       TBC top back center

       TBR top back right

       DL  downmix left

       DR  downmix right

       WL  wide left

       WR  wide right

       SDL surround direct left

       SDR surround direct right

       LFE2
           low frequency 2

       Standard channel layout compositions can be specified by using the following identifiers:

       mono
           FC

       stereo
           FL+FR

       2.1 FL+FR+LFE

       3.0 FL+FR+FC

       3.0(back)
           FL+FR+BC

       4.0 FL+FR+FC+BC

       quad
           FL+FR+BL+BR

       quad(side)
           FL+FR+SL+SR

       3.1 FL+FR+FC+LFE

       5.0 FL+FR+FC+BL+BR

       5.0(side)
           FL+FR+FC+SL+SR

       4.1 FL+FR+FC+LFE+BC

       5.1 FL+FR+FC+LFE+BL+BR

       5.1(side)
           FL+FR+FC+LFE+SL+SR

       6.0 FL+FR+FC+BC+SL+SR

       6.0(front)
           FL+FR+FLC+FRC+SL+SR

       3.1.2
           FL+FR+FC+LFE+TFL+TFR

       hexagonal
           FL+FR+FC+BL+BR+BC

       6.1 FL+FR+FC+LFE+BC+SL+SR

       6.1 FL+FR+FC+LFE+BL+BR+BC

       6.1(front)
           FL+FR+LFE+FLC+FRC+SL+SR

       7.0 FL+FR+FC+BL+BR+SL+SR

       7.0(front)
           FL+FR+FC+FLC+FRC+SL+SR

       7.1 FL+FR+FC+LFE+BL+BR+SL+SR

       7.1(wide)
           FL+FR+FC+LFE+BL+BR+FLC+FRC

       7.1(wide-side)
           FL+FR+FC+LFE+FLC+FRC+SL+SR

       5.1.2
           FL+FR+FC+LFE+BL+BR+TFL+TFR

       octagonal
           FL+FR+FC+BL+BR+BC+SL+SR

       cube
           FL+FR+BL+BR+TFL+TFR+TBL+TBR

       5.1.4
           FL+FR+FC+LFE+BL+BR+TFL+TFR+TBL+TBR

       7.1.2
           FL+FR+FC+LFE+BL+BR+SL+SR+TFL+TFR

       7.1.4
           FL+FR+FC+LFE+BL+BR+SL+SR+TFL+TFR+TBL+TBR

       hexadecagonal
           FL+FR+FC+BL+BR+BC+SL+SR+WL+WR+TBL+TBR+TBC+TFC+TFL+TFR

       downmix
           DL+DR

       22.2
           FL+FR+FC+LFE+BL+BR+FLC+FRC+BC+SL+SR+TC+TFL+TFC+TFR+TBL+TBC+TBR+LFE2+TSL+TSR+BFC+BFL+BFR

       A custom channel layout can be specified as a sequence of terms, separated by '+'.  Each term can be:

       •   the name of a single channel (e.g. FL, FR, FC, LFE, etc.), each optionally containing a  custom  name
           after a '@', (e.g. FL@Left, FR@Right, FC@Center, LFE@Low_Frequency, etc.)

       A standard channel layout can be specified by the following:

       •   the name of a single channel (e.g. FL, FR, FC, LFE, etc.)

       •   the name of a standard channel layout (e.g. mono, stereo, 4.0, quad, 5.0, etc.)

       •   a  number  of  channels,  in  decimal,  followed by 'c', yielding the default channel layout for that
           number of channels (see the function "av_channel_layout_default"). Note that not all  channel  counts
           have a default layout.

       •   a  number  of  channels,  in  decimal,  followed  by 'C', yielding an unknown channel layout with the
           specified number of channels. Note that not all channel layout specification strings support  unknown
           channel layouts.

       •   a   channel   layout   mask,  in  hexadecimal  starting  with  "0x"  (see  the  "AV_CH_*"  macros  in
           libavutil/channel_layout.h.

       Before libavutil version 53 the trailing character "c" to specify a number of channels was optional,  but
       now it is required, while a channel layout mask can also be specified as a decimal number (if and only if
       not followed by "c" or "C").

       See also the function "av_channel_layout_from_string" defined in libavutil/channel_layout.h.

EXPRESSION EVALUATION

       When  evaluating an arithmetic expression, FFmpeg uses an internal formula evaluator, implemented through
       the libavutil/eval.h interface.

       An expression may contain unary, binary operators, constants, and functions.

       Two expressions expr1 and expr2 can be combined to form  another  expression  "expr1;expr2".   expr1  and
       expr2 are evaluated in turn, and the new expression evaluates to the value of expr2.

       The following binary operators are available: "+", "-", "*", "/", "^".

       The following unary operators are available: "+", "-".

       The following functions are available:

       abs(x)
           Compute absolute value of x.

       acos(x)
           Compute arccosine of x.

       asin(x)
           Compute arcsine of x.

       atan(x)
           Compute arctangent of x.

       atan2(x, y)
           Compute principal value of the arc tangent of y/x.

       between(x, min, max)
           Return 1 if x is greater than or equal to min and lesser than or equal to max, 0 otherwise.

       bitand(x, y)
       bitor(x, y)
           Compute bitwise and/or operation on x and y.

           The  results  of  the  evaluation  of  x and y are converted to integers before executing the bitwise
           operation.

           Note that both the conversion to  integer  and  the  conversion  back  to  floating  point  can  lose
           precision. Beware of unexpected results for large numbers (usually 2^53 and larger).

       ceil(expr)
           Round the value of expression expr upwards to the nearest integer. For example, "ceil(1.5)" is "2.0".

       clip(x, min, max)
           Return the value of x clipped between min and max.

       cos(x)
           Compute cosine of x.

       cosh(x)
           Compute hyperbolic cosine of x.

       eq(x, y)
           Return 1 if x and y are equivalent, 0 otherwise.

       exp(x)
           Compute exponential of x (with base "e", the Euler's number).

       floor(expr)
           Round  the  value  of expression expr downwards to the nearest integer. For example, "floor(-1.5)" is
           "-2.0".

       gauss(x)
           Compute Gauss function of x, corresponding to "exp(-x*x/2) / sqrt(2*PI)".

       gcd(x, y)
           Return the greatest common divisor of x and y. If both x and y are 0 or either or both are less  than
           zero then behavior is undefined.

       gt(x, y)
           Return 1 if x is greater than y, 0 otherwise.

       gte(x, y)
           Return 1 if x is greater than or equal to y, 0 otherwise.

       hypot(x, y)
           This  function  is  similar  to  the C function with the same name; it returns "sqrt(x*x + y*y)", the
           length of the hypotenuse of a right triangle with sides of length x and y, or  the  distance  of  the
           point (x, y) from the origin.

       if(x, y)
           Evaluate  x,  and  if  the  result  is  non-zero  return  the result of the evaluation of y, return 0
           otherwise.

       if(x, y, z)
           Evaluate x, and if the result is non-zero return the evaluation result of y, otherwise the evaluation
           result of z.

       ifnot(x, y)
           Evaluate x, and if the result is zero return the result of the evaluation of y, return 0 otherwise.

       ifnot(x, y, z)
           Evaluate x, and if the result is zero return the evaluation result of  y,  otherwise  the  evaluation
           result of z.

       isinf(x)
           Return 1.0 if x is +/-INFINITY, 0.0 otherwise.

       isnan(x)
           Return 1.0 if x is NAN, 0.0 otherwise.

       ld(var)
           Load  the  value  of  the internal variable with number var, which was previously stored with st(var,
           expr).  The function returns the loaded value.

       lerp(x, y, z)
           Return linear interpolation between x and y by amount of z.

       log(x)
           Compute natural logarithm of x.

       lt(x, y)
           Return 1 if x is lesser than y, 0 otherwise.

       lte(x, y)
           Return 1 if x is lesser than or equal to y, 0 otherwise.

       max(x, y)
           Return the maximum between x and y.

       min(x, y)
           Return the minimum between x and y.

       mod(x, y)
           Compute the remainder of division of x by y.

       not(expr)
           Return 1.0 if expr is zero, 0.0 otherwise.

       pow(x, y)
           Compute the power of x elevated y, it is equivalent to "(x)^(y)".

       print(t)
       print(t, l)
           Print the value of expression t with loglevel l. If l is not specified then a default  log  level  is
           used.  Returns the value of the expression printed.

           Prints t with loglevel l

       random(x)
           Return  a pseudo random value between 0.0 and 1.0. x is the index of the internal variable which will
           be used to save the seed/state.

       root(expr, max)
           Find an input value for which the function represented by expr  with  argument  ld(0)  is  0  in  the
           interval 0..max.

           The expression in expr must denote a continuous function or the result is undefined.

           ld(0)  is  used  to represent the function input value, which means that the given expression will be
           evaluated multiple times with various input values that the expression can access through ld(0). When
           the expression evaluates to 0 then the corresponding input value will be returned.

       round(expr)
           Round the value of expression expr to the nearest integer. For example, "round(1.5)" is "2.0".

       sgn(x)
           Compute sign of x.

       sin(x)
           Compute sine of x.

       sinh(x)
           Compute hyperbolic sine of x.

       sqrt(expr)
           Compute the square root of expr. This is equivalent to "(expr)^.5".

       squish(x)
           Compute expression "1/(1 + exp(4*x))".

       st(var, expr)
           Store the value of the expression expr in an internal variable.  var  specifies  the  number  of  the
           variable  where  to  store the value, and it is a value ranging from 0 to 9. The function returns the
           value stored in the internal variable.  Note, Variables are currently not shared between expressions.

       tan(x)
           Compute tangent of x.

       tanh(x)
           Compute hyperbolic tangent of x.

       taylor(expr, x)
       taylor(expr, x, id)
           Evaluate a Taylor series at x, given  an  expression  representing  the  ld(id)-th  derivative  of  a
           function at 0.

           When the series does not converge the result is undefined.

           ld(id)  is used to represent the derivative order in expr, which means that the given expression will
           be evaluated multiple times with various input values that the expression can access through  ld(id).
           If id is not specified then 0 is assumed.

           Note, when you have the derivatives at y instead of 0, "taylor(expr, x-y)" can be used.

       time(0)
           Return the current (wallclock) time in seconds.

       trunc(expr)
           Round the value of expression expr towards zero to the nearest integer. For example, "trunc(-1.5)" is
           "-1.0".

       while(cond, expr)
           Evaluate  expression  expr  while  the expression cond is non-zero, and returns the value of the last
           expr evaluation, or NAN if cond was always false.

       The following constants are available:

       PI  area of the unit disc, approximately 3.14

       E   exp(1) (Euler's number), approximately 2.718

       PHI golden ratio (1+sqrt(5))/2, approximately 1.618

       Assuming that an expression is considered "true" if it has a non-zero value, note that:

       "*" works like AND

       "+" works like OR

       For example the construct:

               if (A AND B) then C

       is equivalent to:

               if(A*B, C)

       In your C code, you can extend the list of unary and binary functions, and define  recognized  constants,
       so that they are available for your expressions.

       The  evaluator  also  recognizes  the  International  System unit prefixes.  If 'i' is appended after the
       prefix, binary prefixes are used, which are based on powers of 1024 instead of powers of 1000.   The  'B'
       postfix  multiplies  the  value  by 8, and can be appended after a unit prefix or used alone. This allows
       using for example 'KB', 'MiB', 'G' and 'B' as number postfix.

       The list of available International System prefixes follows, with indication of the corresponding  powers
       of 10 and of 2.

       y   10^-24 / 2^-80

       z   10^-21 / 2^-70

       a   10^-18 / 2^-60

       f   10^-15 / 2^-50

       p   10^-12 / 2^-40

       n   10^-9 / 2^-30

       u   10^-6 / 2^-20

       m   10^-3 / 2^-10

       c   10^-2

       d   10^-1

       h   10^2

       k   10^3 / 2^10

       K   10^3 / 2^10

       M   10^6 / 2^20

       G   10^9 / 2^30

       T   10^12 / 2^40

       P   10^15 / 2^50

       E   10^18 / 2^60

       Z   10^21 / 2^70

       Y   10^24 / 2^80

CODEC OPTIONS

libavcodec provides some generic global options, which can be set on all the encoders and decoders. In
addition each codec may support so-called private options, which are specific for a given codec.

Sometimes, a global option may only affect a specific kind of codec, and may be nonsensical or ignored by
another, so you need to be aware of the meaning of the specified options. Also some options are meant
only for decoding or encoding.

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

The list of supported options follow:

b integer (encoding,audio,video)
Set bitrate in bits/s. Default value is 200K.

ab integer (encoding,audio)
Set audio bitrate (in bits/s). Default value is 128K.

bt integer (encoding,video)
Set video bitrate tolerance (in bits/s). In 1-pass mode, bitrate tolerance specifies how far
ratecontrol is willing to deviate from the target average bitrate value. This is not related to
min/max bitrate. Lowering tolerance too much has an adverse effect on quality.

flags flags (decoding/encoding,audio,video,subtitles)
Set generic flags.

Possible values:

mv4 Use four motion vector by macroblock (mpeg4).

qpel
Use 1/4 pel motion compensation.

loop
Use loop filter.

qscale
Use fixed qscale.

pass1
Use internal 2pass ratecontrol in first pass mode.

pass2
Use internal 2pass ratecontrol in second pass mode.

gray
Only decode/encode grayscale.

psnr
Set error[?] variables during encoding.

truncated
Input bitstream might be randomly truncated.

drop_changed
Don't output frames whose parameters differ from first decoded frame in stream. Error
AVERROR_INPUT_CHANGED is returned when a frame is dropped.

ildct
Use interlaced DCT.

low_delay
Force low delay.

global_header
Place global headers in extradata instead of every keyframe.

bitexact
Only write platform-, build- and time-independent data. (except (I)DCT). This ensures that file
and data checksums are reproducible and match between platforms. Its primary use is for
regression testing.

aic Apply H263 advanced intra coding / mpeg4 ac prediction.

ilme
Apply interlaced motion estimation.

cgop
Use closed gop.

output_corrupt
Output even potentially corrupted frames.

time_base rational number
Set codec time base.

It is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented.
For fixed-fps content, timebase should be "1 / frame_rate" and timestamp increments should be
identically 1.

g integer (encoding,video)
Set the group of picture (GOP) size. Default value is 12.

ar integer (decoding/encoding,audio)
Set audio sampling rate (in Hz).

ac integer (decoding/encoding,audio)
Set number of audio channels.

cutoff integer (encoding,audio)
Set cutoff bandwidth. (Supported only by selected encoders, see their respective documentation
sections.)

frame_size integer (encoding,audio)
Set audio frame size.

Each submitted frame except the last must contain exactly frame_size samples per channel. May be 0
when the codec has CODEC_CAP_VARIABLE_FRAME_SIZE set, in that case the frame size is not restricted.
It is set by some decoders to indicate constant frame size.

frame_number integer
Set the frame number.

delay integer
qcomp float (encoding,video)
Set video quantizer scale compression (VBR). It is used as a constant in the ratecontrol equation.
Recommended range for default rc_eq: 0.0-1.0.

qblur float (encoding,video)
Set video quantizer scale blur (VBR).

qmin integer (encoding,video)
Set min video quantizer scale (VBR). Must be included between -1 and 69, default value is 2.

qmax integer (encoding,video)
Set max video quantizer scale (VBR). Must be included between -1 and 1024, default value is 31.

qdiff integer (encoding,video)
Set max difference between the quantizer scale (VBR).

bf integer (encoding,video)
Set max number of B frames between non-B-frames.

Must be an integer between -1 and 16. 0 means that B-frames are disabled. If a value of -1 is used,
it will choose an automatic value depending on the encoder.

Default value is 0.

b_qfactor float (encoding,video)
Set qp factor between P and B frames.

codec_tag integer
bug flags (decoding,video)
Workaround not auto detected encoder bugs.

Possible values:

autodetect
xvid_ilace
Xvid interlacing bug (autodetected if fourcc==XVIX)

ump4
(autodetected if fourcc==UMP4)

no_padding
padding bug (autodetected)

amv
qpel_chroma
std_qpel
old standard qpel (autodetected per fourcc/version)

qpel_chroma2
direct_blocksize
direct-qpel-blocksize bug (autodetected per fourcc/version)

edge
edge padding bug (autodetected per fourcc/version)

hpel_chroma
dc_clip
ms Workaround various bugs in microsoft broken decoders.

trunc
trancated frames

strict integer (decoding/encoding,audio,video)
Specify how strictly to follow the standards.

Possible values:

very
strictly conform to an older more strict version of the spec or reference software

strict
strictly conform to all the things in the spec no matter what consequences

normal
unofficial
allow unofficial extensions

experimental
allow non standardized experimental things, experimental (unfinished/work in progress/not well
tested) decoders and encoders. Note: experimental decoders can pose a security risk, do not use
this for decoding untrusted input.

b_qoffset float (encoding,video)
Set QP offset between P and B frames.

err_detect flags (decoding,audio,video)
Set error detection flags.

Possible values:

crccheck
verify embedded CRCs

bitstream
detect bitstream specification deviations

buffer
detect improper bitstream length

explode
abort decoding on minor error detection

ignore_err
ignore decoding errors, and continue decoding. This is useful if you want to analyze the content
of a video and thus want everything to be decoded no matter what. This option will not result in
a video that is pleasing to watch in case of errors.

careful
consider things that violate the spec and have not been seen in the wild as errors

compliant
consider all spec non compliancies as errors

aggressive
consider things that a sane encoder should not do as an error

has_b_frames integer
block_align integer
rc_override_count integer
maxrate integer (encoding,audio,video)
Set max bitrate tolerance (in bits/s). Requires bufsize to be set.

minrate integer (encoding,audio,video)
Set min bitrate tolerance (in bits/s). Most useful in setting up a CBR encode. It is of little use
elsewise.

bufsize integer (encoding,audio,video)
Set ratecontrol buffer size (in bits).

i_qfactor float (encoding,video)
Set QP factor between P and I frames.

i_qoffset float (encoding,video)
Set QP offset between P and I frames.

dct integer (encoding,video)
Set DCT algorithm.

Possible values:

auto
autoselect a good one (default)

fastint
fast integer

int accurate integer

mmx
altivec
faan
floating point AAN DCT

lumi_mask float (encoding,video)
Compress bright areas stronger than medium ones.

tcplx_mask float (encoding,video)
Set temporal complexity masking.

scplx_mask float (encoding,video)
Set spatial complexity masking.

p_mask float (encoding,video)
Set inter masking.

dark_mask float (encoding,video)
Compress dark areas stronger than medium ones.

idct integer (decoding/encoding,video)
Select IDCT implementation.

Possible values:

auto
int
simple
simplemmx
simpleauto
Automatically pick a IDCT compatible with the simple one

arm
altivec
sh4
simplearm
simplearmv5te
simplearmv6
simpleneon
xvid
faani
floating point AAN IDCT

slice_count integer
ec flags (decoding,video)
Set error concealment strategy.

Possible values:

guess_mvs
iterative motion vector (MV) search (slow)

deblock
use strong deblock filter for damaged MBs

favor_inter
favor predicting from the previous frame instead of the current

bits_per_coded_sample integer
aspect rational number (encoding,video)
Set sample aspect ratio.

sar rational number (encoding,video)
Set sample aspect ratio. Alias to aspect.

debug flags (decoding/encoding,audio,video,subtitles)
Print specific debug info.

Possible values:

pict
picture info

rc rate control

bitstream
mb_type
macroblock (MB) type

qp per-block quantization parameter (QP)

dct_coeff
green_metadata
display complexity metadata for the upcoming frame, GoP or for a given duration.

skip
startcode
er error recognition

mmco
memory management control operations (H.264)

bugs
buffers
picture buffer allocations

thread_ops
threading operations

nomc
skip motion compensation

cmp integer (encoding,video)
Set full pel me compare function.

Possible values:

sad sum of absolute differences, fast (default)

sse sum of squared errors

satd
sum of absolute Hadamard transformed differences

dct sum of absolute DCT transformed differences

psnr
sum of squared quantization errors (avoid, low quality)

bit number of bits needed for the block

rd rate distortion optimal, slow

zero
0

vsad
sum of absolute vertical differences

vsse
sum of squared vertical differences

nsse
noise preserving sum of squared differences

w53 5/3 wavelet, only used in snow

w97 9/7 wavelet, only used in snow

dctmax
chroma
subcmp integer (encoding,video)
Set sub pel me compare function.

Possible values:

sad sum of absolute differences, fast (default)

sse sum of squared errors

satd
sum of absolute Hadamard transformed differences

dct sum of absolute DCT transformed differences

psnr
sum of squared quantization errors (avoid, low quality)

bit number of bits needed for the block

rd rate distortion optimal, slow

zero
0

vsad
sum of absolute vertical differences

vsse
sum of squared vertical differences

nsse
noise preserving sum of squared differences

w53 5/3 wavelet, only used in snow

w97 9/7 wavelet, only used in snow

dctmax
chroma
mbcmp integer (encoding,video)
Set macroblock compare function.

Possible values:

sad sum of absolute differences, fast (default)

sse sum of squared errors

satd
sum of absolute Hadamard transformed differences

dct sum of absolute DCT transformed differences

psnr
sum of squared quantization errors (avoid, low quality)

bit number of bits needed for the block

rd rate distortion optimal, slow

zero
0

vsad
sum of absolute vertical differences

vsse
sum of squared vertical differences

nsse
noise preserving sum of squared differences

w53 5/3 wavelet, only used in snow

w97 9/7 wavelet, only used in snow

dctmax
chroma
ildctcmp integer (encoding,video)
Set interlaced dct compare function.

Possible values:

sad sum of absolute differences, fast (default)

sse sum of squared errors

satd
sum of absolute Hadamard transformed differences

dct sum of absolute DCT transformed differences

psnr
sum of squared quantization errors (avoid, low quality)

bit number of bits needed for the block

rd rate distortion optimal, slow

zero
0

vsad
sum of absolute vertical differences

vsse
sum of squared vertical differences

nsse
noise preserving sum of squared differences

w53 5/3 wavelet, only used in snow

w97 9/7 wavelet, only used in snow

dctmax
chroma
dia_size integer (encoding,video)
Set diamond type & size for motion estimation.

(1024, INT_MAX)
full motion estimation(slowest)

(768, 1024]
umh motion estimation

(512, 768]
hex motion estimation

(256, 512]
l2s diamond motion estimation

[2,256]
var diamond motion estimation

(-1, 2)
small diamond motion estimation

-1 funny diamond motion estimation

(INT_MIN, -1)
sab diamond motion estimation

last_pred integer (encoding,video)
Set amount of motion predictors from the previous frame.

precmp integer (encoding,video)
Set pre motion estimation compare function.

Possible values:

sad sum of absolute differences, fast (default)

sse sum of squared errors

satd
sum of absolute Hadamard transformed differences

dct sum of absolute DCT transformed differences

psnr
sum of squared quantization errors (avoid, low quality)

bit number of bits needed for the block

rd rate distortion optimal, slow

zero
0

vsad
sum of absolute vertical differences

vsse
sum of squared vertical differences

nsse
noise preserving sum of squared differences

w53 5/3 wavelet, only used in snow

w97 9/7 wavelet, only used in snow

dctmax
chroma
pre_dia_size integer (encoding,video)
Set diamond type & size for motion estimation pre-pass.

subq integer (encoding,video)
Set sub pel motion estimation quality.

me_range integer (encoding,video)
Set limit motion vectors range (1023 for DivX player).

global_quality integer (encoding,audio,video)
slice_flags integer
mbd integer (encoding,video)
Set macroblock decision algorithm (high quality mode).

Possible values:

simple
use mbcmp (default)

bits
use fewest bits

rd use best rate distortion

rc_init_occupancy integer (encoding,video)
Set number of bits which should be loaded into the rc buffer before decoding starts.

flags2 flags (decoding/encoding,audio,video,subtitles)
Possible values:

fast
Allow non spec compliant speedup tricks.

noout
Skip bitstream encoding.

ignorecrop
Ignore cropping information from sps.

local_header
Place global headers at every keyframe instead of in extradata.

chunks
Frame data might be split into multiple chunks.

showall
Show all frames before the first keyframe.

export_mvs
Export motion vectors into frame side-data (see "AV_FRAME_DATA_MOTION_VECTORS") for codecs that
support it. See also doc/examples/export_mvs.c.

skip_manual
Do not skip samples and export skip information as frame side data.

ass_ro_flush_noop
Do not reset ASS ReadOrder field on flush.

icc_profiles
Generate/parse embedded ICC profiles from/to colorimetry tags.

export_side_data flags (decoding/encoding,audio,video,subtitles)
Possible values:

mvs Export motion vectors into frame side-data (see "AV_FRAME_DATA_MOTION_VECTORS") for codecs that
support it. See also doc/examples/export_mvs.c.

prft
Export encoder Producer Reference Time into packet side-data (see "AV_PKT_DATA_PRFT") for codecs
that support it.

venc_params
Export video encoding parameters through frame side data (see "AV_FRAME_DATA_VIDEO_ENC_PARAMS")
for codecs that support it. At present, those are H.264 and VP9.

film_grain
Export film grain parameters through frame side data (see "AV_FRAME_DATA_FILM_GRAIN_PARAMS").
Supported at present by AV1 decoders.

threads integer (decoding/encoding,video)
Set the number of threads to be used, in case the selected codec implementation supports multi-
threading.

Possible values:

auto, 0
automatically select the number of threads to set

Default value is auto.

dc integer (encoding,video)
Set intra_dc_precision.

nssew integer (encoding,video)
Set nsse weight.

skip_top integer (decoding,video)
Set number of macroblock rows at the top which are skipped.

skip_bottom integer (decoding,video)
Set number of macroblock rows at the bottom which are skipped.

profile integer (encoding,audio,video)
Set encoder codec profile. Default value is unknown. Encoder specific profiles are documented in the
relevant encoder documentation.

level integer (encoding,audio,video)
Set the encoder level. This level depends on the specific codec, and might correspond to the profile
level. It is set by default to unknown.

Possible values:

unknown
lowres integer (decoding,audio,video)
Decode at 1= 1/2, 2=1/4, 3=1/8 resolutions.

mblmin integer (encoding,video)
Set min macroblock lagrange factor (VBR).

mblmax integer (encoding,video)
Set max macroblock lagrange factor (VBR).

skip_loop_filter integer (decoding,video)
skip_idct integer (decoding,video)
skip_frame integer (decoding,video)
Make decoder discard processing depending on the frame type selected by the option value.

skip_loop_filter skips frame loop filtering, skip_idct skips frame IDCT/dequantization, skip_frame
skips decoding.

Possible values:

none
Discard no frame.

default
Discard useless frames like 0-sized frames.

noref
Discard all non-reference frames.

bidir
Discard all bidirectional frames.

nokey
Discard all frames excepts keyframes.

nointra
Discard all frames except I frames.

all Discard all frames.

Default value is default.

bidir_refine integer (encoding,video)
Refine the two motion vectors used in bidirectional macroblocks.

keyint_min integer (encoding,video)
Set minimum interval between IDR-frames.

refs integer (encoding,video)
Set reference frames to consider for motion compensation.

trellis integer (encoding,audio,video)
Set rate-distortion optimal quantization.

mv0_threshold integer (encoding,video)
compression_level integer (encoding,audio,video)
bits_per_raw_sample integer
channel_layout integer (decoding/encoding,audio)
Possible values:

request_channel_layout integer (decoding,audio)
Possible values:

rc_max_vbv_use float (encoding,video)
rc_min_vbv_use float (encoding,video)
color_primaries integer (decoding/encoding,video)
Possible values:

bt709
BT.709

bt470m
BT.470 M

bt470bg
BT.470 BG

smpte170m
SMPTE 170 M

smpte240m
SMPTE 240 M

film
Film

bt2020
BT.2020

smpte428
smpte428_1
SMPTE ST 428-1

smpte431
SMPTE 431-2

smpte432
SMPTE 432-1

jedec-p22
JEDEC P22

color_trc integer (decoding/encoding,video)
Possible values:

bt709
BT.709

gamma22
BT.470 M

gamma28
BT.470 BG

smpte170m
SMPTE 170 M

smpte240m
SMPTE 240 M

linear
Linear

log
log100
Log

log_sqrt
log316
Log square root

iec61966_2_4
iec61966-2-4
IEC 61966-2-4

bt1361
bt1361e
BT.1361

iec61966_2_1
iec61966-2-1
IEC 61966-2-1

bt2020_10
bt2020_10bit
BT.2020 - 10 bit

bt2020_12
bt2020_12bit
BT.2020 - 12 bit

smpte2084
SMPTE ST 2084

smpte428
smpte428_1
SMPTE ST 428-1

arib-std-b67
ARIB STD-B67

colorspace integer (decoding/encoding,video)
Possible values:

rgb RGB

bt709
BT.709

fcc FCC

bt470bg
BT.470 BG

smpte170m
SMPTE 170 M

smpte240m
SMPTE 240 M

ycocg
YCOCG

bt2020nc
bt2020_ncl
BT.2020 NCL

bt2020c
bt2020_cl
BT.2020 CL

smpte2085
SMPTE 2085

chroma-derived-nc
Chroma-derived NCL

chroma-derived-c
Chroma-derived CL

ictcp
ICtCp

color_range integer (decoding/encoding,video)
If used as input parameter, it serves as a hint to the decoder, which color_range the input has.
Possible values:

tv
mpeg
limited
MPEG (219*2^(n-8))

pc
jpeg
full
JPEG (2^n-1)

chroma_sample_location integer (decoding/encoding,video)
Possible values:

left
center
topleft
top
bottomleft
bottom
log_level_offset integer
Set the log level offset.

slices integer (encoding,video)
Number of slices, used in parallelized encoding.

thread_type flags (decoding/encoding,video)
Select which multithreading methods to use.

Use of frame will increase decoding delay by one frame per thread, so clients which cannot provide
future frames should not use it.

Possible values:

slice
Decode more than one part of a single frame at once.

Multithreading using slices works only when the video was encoded with slices.

frame
Decode more than one frame at once.

Default value is slice+frame.

audio_service_type integer (encoding,audio)
Set audio service type.

Possible values:

ma Main Audio Service

ef Effects

vi Visually Impaired

hi Hearing Impaired

di Dialogue

co Commentary

em Emergency

vo Voice Over

ka Karaoke

request_sample_fmt sample_fmt (decoding,audio)
Set sample format audio decoders should prefer. Default value is "none".

pkt_timebase rational number
sub_charenc encoding (decoding,subtitles)
Set the input subtitles character encoding.

field_order field_order (video)
Set/override the field order of the video. Possible values:

progressive
Progressive video

tt Interlaced video, top field coded and displayed first

bb Interlaced video, bottom field coded and displayed first

tb Interlaced video, top coded first, bottom displayed first

bt Interlaced video, bottom coded first, top displayed first

skip_alpha bool (decoding,video)
Set to 1 to disable processing alpha (transparency). This works like the gray flag in the flags
option which skips chroma information instead of alpha. Default is 0.

codec_whitelist list (input)
"," separated list of allowed decoders. By default all are allowed.

dump_separator string (input)
Separator used to separate the fields printed on the command line about the Stream parameters. For
example, to separate the fields with newlines and indentation:

ffprobe -dump_separator "
" -i ~/videos/matrixbench_mpeg2.mpg

max_pixels integer (decoding/encoding,video)
Maximum number of pixels per image. This value can be used to avoid out of memory failures due to
large images.

apply_cropping bool (decoding,video)
Enable cropping if cropping parameters are multiples of the required alignment for the left and top
parameters. If the alignment is not met the cropping will be partially applied to maintain alignment.
Default is 1 (enabled). Note: The required alignment depends on if "AV_CODEC_FLAG_UNALIGNED" is set
and the CPU. "AV_CODEC_FLAG_UNALIGNED" cannot be changed from the command line. Also hardware
decoders will not apply left/top Cropping.

DECODERS

       Decoders are configured elements in FFmpeg which allow the decoding of multimedia streams.

       When you configure your FFmpeg build, all the supported native decoders are enabled by default.  Decoders
       requiring  an  external library must be enabled manually via the corresponding "--enable-lib" option. You
       can list all available decoders using the configure option "--list-decoders".

       You can disable all the decoders with the configure option "--disable-decoders" and selectively enable  /
       disable single decoders with the options "--enable-decoder=DECODER" / "--disable-decoder=DECODER".

       The option "-decoders" of the ff* tools will display the list of enabled decoders.

VIDEO DECODERS

A description of some of the currently available video decoders follows.

av1
AOMedia Video 1 (AV1) decoder.

Options

operating_point
Select an operating point of a scalable AV1 bitstream (0 - 31). Default is 0.

rawvideo
Raw video decoder.

This decoder decodes rawvideo streams.

Options

top top_field_first
Specify the assumed field type of the input video.

-1 the video is assumed to be progressive (default)

0 bottom-field-first is assumed

1 top-field-first is assumed

libdav1d
dav1d AV1 decoder.

libdav1d allows libavcodec to decode the AOMedia Video 1 (AV1) codec. Requires the presence of the
libdav1d headers and library during configuration. You need to explicitly configure the build with
"--enable-libdav1d".

Options

The following options are supported by the libdav1d wrapper.

framethreads
Set amount of frame threads to use during decoding. The default value is 0 (autodetect). This option
is deprecated for libdav1d >= 1.0 and will be removed in the future. Use the option "max_frame_delay"
and the global option "threads" instead.

tilethreads
Set amount of tile threads to use during decoding. The default value is 0 (autodetect). This option
is deprecated for libdav1d >= 1.0 and will be removed in the future. Use the global option "threads"
instead.

max_frame_delay
Set max amount of frames the decoder may buffer internally. The default value is 0 (autodetect).

filmgrain
Apply film grain to the decoded video if present in the bitstream. Defaults to the internal default
of the library. This option is deprecated and will be removed in the future. See the global option
"export_side_data" to export Film Grain parameters instead of applying it.

oppoint
Select an operating point of a scalable AV1 bitstream (0 - 31). Defaults to the internal default of
the library.

alllayers
Output all spatial layers of a scalable AV1 bitstream. The default value is false.

libdavs2
AVS2-P2/IEEE1857.4 video decoder wrapper.

This decoder allows libavcodec to decode AVS2 streams with davs2 library.

libuavs3d
AVS3-P2/IEEE1857.10 video decoder.

libuavs3d allows libavcodec to decode AVS3 streams. Requires the presence of the libuavs3d headers and
library during configuration. You need to explicitly configure the build with "--enable-libuavs3d".

Options

The following option is supported by the libuavs3d wrapper.

frame_threads
Set amount of frame threads to use during decoding. The default value is 0 (autodetect).

QSV Decoders
The family of Intel QuickSync Video decoders (VC1, MPEG-2, H.264, HEVC, JPEG/MJPEG, VP8, VP9, AV1).

Common Options

The following options are supported by all qsv decoders.

async_depth
Internal parallelization depth, the higher the value the higher the latency.

gpu_copy
A GPU-accelerated copy between video and system memory

default
on
off

HEVC Options

Extra options for hevc_qsv.

load_plugin
A user plugin to load in an internal session

none
hevc_sw
hevc_hw
load_plugins
A :-separate list of hexadecimal plugin UIDs to load in an internal session

v210
Uncompressed 4:2:2 10-bit decoder.

Options

custom_stride
Set the line size of the v210 data in bytes. The default value is 0 (autodetect). You can use the
special -1 value for a strideless v210 as seen in BOXX files.

AUDIO DECODERS

A description of some of the currently available audio decoders follows.

ac3
AC-3 audio decoder.

This decoder implements part of ATSC A/52:2010 and ETSI TS 102 366, as well as the undocumented RealAudio
3 (a.k.a. dnet).

AC-3 Decoder Options

-drc_scale value
Dynamic Range Scale Factor. The factor to apply to dynamic range values from the AC-3 stream. This
factor is applied exponentially. The default value is 1. There are 3 notable scale factor ranges:

drc_scale == 0
DRC disabled. Produces full range audio.

0 < drc_scale <= 1
DRC enabled. Applies a fraction of the stream DRC value. Audio reproduction is between full
range and full compression.

drc_scale > 1
DRC enabled. Applies drc_scale asymmetrically. Loud sounds are fully compressed. Soft sounds
are enhanced.

flac
FLAC audio decoder.

This decoder aims to implement the complete FLAC specification from Xiph.

FLAC Decoder options

-use_buggy_lpc
The lavc FLAC encoder used to produce buggy streams with high lpc values (like the default value).
This option makes it possible to decode such streams correctly by using lavc's old buggy lpc logic
for decoding.

ffwavesynth
Internal wave synthesizer.

This decoder generates wave patterns according to predefined sequences. Its use is purely internal and
the format of the data it accepts is not publicly documented.

libcelt
libcelt decoder wrapper.

libcelt allows libavcodec to decode the Xiph CELT ultra-low delay audio codec. Requires the presence of
the libcelt headers and library during configuration. You need to explicitly configure the build with
"--enable-libcelt".

libgsm
libgsm decoder wrapper.

libgsm allows libavcodec to decode the GSM full rate audio codec. Requires the presence of the libgsm
headers and library during configuration. You need to explicitly configure the build with
"--enable-libgsm".

This decoder supports both the ordinary GSM and the Microsoft variant.

libilbc
libilbc decoder wrapper.

libilbc allows libavcodec to decode the Internet Low Bitrate Codec (iLBC) audio codec. Requires the
presence of the libilbc headers and library during configuration. You need to explicitly configure the
build with "--enable-libilbc".

Options

The following option is supported by the libilbc wrapper.

enhance
Enable the enhancement of the decoded audio when set to 1. The default value is 0 (disabled).

libopencore-amrnb
libopencore-amrnb decoder wrapper.

libopencore-amrnb allows libavcodec to decode the Adaptive Multi-Rate Narrowband audio codec. Using it
requires the presence of the libopencore-amrnb headers and library during configuration. You need to
explicitly configure the build with "--enable-libopencore-amrnb".

An FFmpeg native decoder for AMR-NB exists, so users can decode AMR-NB without this library.

libopencore-amrwb
libopencore-amrwb decoder wrapper.

libopencore-amrwb allows libavcodec to decode the Adaptive Multi-Rate Wideband audio codec. Using it
requires the presence of the libopencore-amrwb headers and library during configuration. You need to
explicitly configure the build with "--enable-libopencore-amrwb".

An FFmpeg native decoder for AMR-WB exists, so users can decode AMR-WB without this library.

libopus
libopus decoder wrapper.

libopus allows libavcodec to decode the Opus Interactive Audio Codec. Requires the presence of the
libopus headers and library during configuration. You need to explicitly configure the build with
"--enable-libopus".

An FFmpeg native decoder for Opus exists, so users can decode Opus without this library.

SUBTITLES DECODERS

libaribb24
ARIB STD-B24 caption decoder.

Implements profiles A and C of the ARIB STD-B24 standard.

libaribb24 Decoder Options

-aribb24-base-path path
Sets the base path for the libaribb24 library. This is utilized for reading of configuration files
(for custom unicode conversions), and for dumping of non-text symbols as images under that location.

Unset by default.

-aribb24-skip-ruby-text boolean
Tells the decoder wrapper to skip text blocks that contain half-height ruby text.

Enabled by default.

libaribcaption
Yet another ARIB STD-B24 caption decoder using external libaribcaption library.

Implements profiles A and C of the Japanse ARIB STD-B24 standard, Brazilian ABNT NBR 15606-1, and
Philippines version of ISDB-T.

Requires the presence of the libaribcaption headers and library (<https://github.com/xqq/libaribcaption>)
during configuration. You need to explicitly configure the build with "--enable-libaribcaption". If
both libaribb24 and libaribcaption are enabled, libaribcaption decoder precedes.

libaribcaption Decoder Options

-sub_type subtitle_type
Specifies the format of the decoded subtitles.

bitmap
Graphical image.

ass ASS formatted text.

text
Simple text based output without formatting.

The default is ass as same as libaribb24 decoder. Some present players (e.g., mpv) expect ASS format
for ARIB caption.

-caption_encoding encoding_scheme
Specifies the encoding scheme of input subtitle text.

auto
Automatically detect text encoding (default).

jis 8bit-char JIS encoding defined in ARIB STD B24. This encoding used in Japan for ISDB captions.

utf8
UTF-8 encoding defined in ARIB STD B24. This encoding is used in Philippines for ISDB-T
captions.

latin
Latin character encoding defined in ABNT NBR 15606-1. This encoding is used in South America for
SBTVD / ISDB-Tb captions.

-font font_name[,font_name2,...]
Specify comma-separated list of font family names to be used for bitmap or ass type subtitle
rendering. Only first font name is used for ass type subtitle.

If not specified, use internaly defined default font family.

-ass_single_rect boolean
ARIB STD-B24 specifies that some captions may be displayed at different positions at a time (multi-
rectangle subtitle). Since some players (e.g., old mpv) can't handle multiple ASS rectangles in a
single AVSubtitle, or multiple ASS rectangles of indeterminate duration with the same start
timestamp, this option can change the behavior so that all the texts are displayed in a single ASS
rectangle.

The default is false.

If your player cannot handle AVSubtitles with multiple ASS rectangles properly, set this option to
true or define ASS_SINGLE_RECT=1 to change default behavior at compilation.

-force_outline_text boolean
Specify whether always render outline text for all characters regardless of the indication by
charactor style.

The default is false.

-outline_width number (0.0 - 3.0)
Specify width for outline text, in dots (relative).

The default is 1.5.

-ignore_background boolean
Specify whether to ignore background color rendering.

The default is false.

-ignore_ruby boolean
Specify whether to ignore rendering for ruby-like (furigana) characters.

The default is false.

-replace_drcs boolean
Specify whether to render replaced DRCS characters as Unicode characters.

The default is true.

-replace_msz_ascii boolean
Specify whether to replace MSZ (Middle Size; half width) fullwidth alphanumerics with halfwidth
alphanumerics.

The default is true.

-replace_msz_japanese boolean
Specify whether to replace some MSZ (Middle Size; half width) fullwidth japanese special characters
with halfwidth ones.

The default is true.

-replace_msz_glyph boolean
Specify whether to replace MSZ (Middle Size; half width) characters with halfwidth glyphs if the
fonts supports it. This option works under FreeType or DirectWrite renderer with Adobe-Japan1
compliant fonts. e.g., IBM Plex Sans JP, Morisawa BIZ UDGothic, Morisawa BIZ UDMincho, Yu Gothic, Yu
Mincho, and Meiryo.

The default is true.

-canvas_size image_size
Specify the resolution of the canvas to render subtitles to; usually, this should be frame size of
input video. This only applies when "-subtitle_type" is set to bitmap.

The libaribcaption decoder assumes input frame size for bitmap rendering as below:

1. PROFILE_A : 1440 x 1080 with SAR (PAR) 4:3

2. PROFILE_C : 320 x 180 with SAR (PAR) 1:1

If actual frame size of input video does not match above assumption, the rendered captions may be
distorted. To make the captions undistorted, add "-canvas_size" option to specify actual input video
size.

Note that the "-canvas_size" option is not required for video with different size but same aspect
ratio. In such cases, the caption will be stretched or shrunk to actual video size if "-canvas_size"
option is not specified. If "-canvas_size" option is specified with different size, the caption will
be stretched or shrunk as specified size with calculated SAR.

libaribcaption decoder usage examples

Display MPEG-TS file with ARIB subtitle by "ffplay" tool:

ffplay -sub_type bitmap MPEG.TS

Display MPEG-TS file with input frame size 1920x1080 by "ffplay" tool:

ffplay -sub_type bitmap -canvas_size 1920x1080 MPEG.TS

Embed ARIB subtitle in transcoded video:

ffmpeg -sub_type bitmap -i src.m2t -filter_complex "[0:v][0:s]overlay" -vcodec h264 dest.mp4

dvbsub
Options

compute_clut
-2 Compute clut once if no matching CLUT is in the stream.

-1 Compute clut if no matching CLUT is in the stream.

0 Never compute CLUT

1 Always compute CLUT and override the one provided in the stream.

dvb_substream
Selects the dvb substream, or all substreams if -1 which is default.

dvdsub
This codec decodes the bitmap subtitles used in DVDs; the same subtitles can also be found in VobSub file
pairs and in some Matroska files.

Options

palette
Specify the global palette used by the bitmaps. When stored in VobSub, the palette is normally
specified in the index file; in Matroska, the palette is stored in the codec extra-data in the same
format as in VobSub. In DVDs, the palette is stored in the IFO file, and therefore not available when
reading from dumped VOB files.

The format for this option is a string containing 16 24-bits hexadecimal numbers (without 0x prefix)
separated by commas, for example "0d00ee, ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a,
7b7b7b, d1d1d1, 7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b".

ifo_palette
Specify the IFO file from which the global palette is obtained. (experimental)

forced_subs_only
Only decode subtitle entries marked as forced. Some titles have forced and non-forced subtitles in
the same track. Setting this flag to 1 will only keep the forced subtitles. Default value is 0.

libzvbi-teletext
Libzvbi allows libavcodec to decode DVB teletext pages and DVB teletext subtitles. Requires the presence
of the libzvbi headers and library during configuration. You need to explicitly configure the build with
"--enable-libzvbi".

Options

txt_page
List of teletext page numbers to decode. Pages that do not match the specified list are dropped. You
may use the special "*" string to match all pages, or "subtitle" to match all subtitle pages.
Default value is *.

txt_default_region
Set default character set used for decoding, a value between 0 and 87 (see ETS 300 706, Section 15,
Table 32). Default value is -1, which does not override the libzvbi default. This option is needed
for some legacy level 1.0 transmissions which cannot signal the proper charset.

txt_chop_top
Discards the top teletext line. Default value is 1.

txt_format
Specifies the format of the decoded subtitles.

bitmap
The default format, you should use this for teletext pages, because certain graphics and colors
cannot be expressed in simple text or even ASS.

text
Simple text based output without formatting.

ass Formatted ASS output, subtitle pages and teletext pages are returned in different styles,
subtitle pages are stripped down to text, but an effort is made to keep the text alignment and
the formatting.

txt_left
X offset of generated bitmaps, default is 0.

txt_top
Y offset of generated bitmaps, default is 0.

txt_chop_spaces
Chops leading and trailing spaces and removes empty lines from the generated text. This option is
useful for teletext based subtitles where empty spaces may be present at the start or at the end of
the lines or empty lines may be present between the subtitle lines because of double-sized teletext
characters. Default value is 1.

txt_duration
Sets the display duration of the decoded teletext pages or subtitles in milliseconds. Default value
is -1 which means infinity or until the next subtitle event comes.

txt_transparent
Force transparent background of the generated teletext bitmaps. Default value is 0 which means an
opaque background.

txt_opacity
Sets the opacity (0-255) of the teletext background. If txt_transparent is not set, it only affects
characters between a start box and an end box, typically subtitles. Default value is 0 if
txt_transparent is set, 255 otherwise.

ENCODERS

       Encoders are configured elements in FFmpeg which allow the encoding of multimedia streams.

       When  you configure your FFmpeg build, all the supported native encoders are enabled by default. Encoders
       requiring an external library must be enabled manually via the corresponding "--enable-lib"  option.  You
       can list all available encoders using the configure option "--list-encoders".

       You  can disable all the encoders with the configure option "--disable-encoders" and selectively enable /
       disable single encoders with the options "--enable-encoder=ENCODER" / "--disable-encoder=ENCODER".

       The option "-encoders" of the ff* tools will display the list of enabled encoders.

AUDIO ENCODERS

A description of some of the currently available audio encoders follows.

aac
Advanced Audio Coding (AAC) encoder.

This encoder is the default AAC encoder, natively implemented into FFmpeg.

Options

b Set bit rate in bits/s. Setting this automatically activates constant bit rate (CBR) mode. If this
option is unspecified it is set to 128kbps.

q Set quality for variable bit rate (VBR) mode. This option is valid only using the ffmpeg command-line
tool. For library interface users, use global_quality.

cutoff
Set cutoff frequency. If unspecified will allow the encoder to dynamically adjust the cutoff to
improve clarity on low bitrates.

aac_coder
Set AAC encoder coding method. Possible values:

twoloop
Two loop searching (TLS) method. This is the default method.

This method first sets quantizers depending on band thresholds and then tries to find an optimal
combination by adding or subtracting a specific value from all quantizers and adjusting some
individual quantizer a little. Will tune itself based on whether aac_is, aac_ms and aac_pns are
enabled.

anmr
Average noise to mask ratio (ANMR) trellis-based solution.

This is an experimental coder which currently produces a lower quality, is more unstable and is
slower than the default twoloop coder but has potential. Currently has no support for the aac_is
or aac_pns options. Not currently recommended.

fast
Constant quantizer method.

Uses a cheaper version of twoloop algorithm that doesn't try to do as many clever adjustments.
Worse with low bitrates (less than 64kbps), but is better and much faster at higher bitrates.

aac_ms
Sets mid/side coding mode. The default value of "auto" will automatically use M/S with bands which
will benefit from such coding. Can be forced for all bands using the value "enable", which is mainly
useful for debugging or disabled using "disable".

aac_is
Sets intensity stereo coding tool usage. By default, it's enabled and will automatically toggle IS
for similar pairs of stereo bands if it's beneficial. Can be disabled for debugging by setting the
value to "disable".

aac_pns
Uses perceptual noise substitution to replace low entropy high frequency bands with imperceptible
white noise during the decoding process. By default, it's enabled, but can be disabled for debugging
purposes by using "disable".

aac_tns
Enables the use of a multitap FIR filter which spans through the high frequency bands to hide
quantization noise during the encoding process and is reverted by the decoder. As well as decreasing
unpleasant artifacts in the high range this also reduces the entropy in the high bands and allows for
more bits to be used by the mid-low bands. By default it's enabled but can be disabled for debugging
by setting the option to "disable".

aac_ltp
Enables the use of the long term prediction extension which increases coding efficiency in very low
bandwidth situations such as encoding of voice or solo piano music by extending constant harmonic
peaks in bands throughout frames. This option is implied by profile:a aac_low and is incompatible
with aac_pred. Use in conjunction with -ar to decrease the samplerate.

aac_pred
Enables the use of a more traditional style of prediction where the spectral coefficients transmitted
are replaced by the difference of the current coefficients minus the previous "predicted"
coefficients. In theory and sometimes in practice this can improve quality for low to mid bitrate
audio. This option implies the aac_main profile and is incompatible with aac_ltp.

profile
Sets the encoding profile, possible values:

aac_low
The default, AAC "Low-complexity" profile. Is the most compatible and produces decent quality.

mpeg2_aac_low
Equivalent to "-profile:a aac_low -aac_pns 0". PNS was introduced with the MPEG4 specifications.

aac_ltp
Long term prediction profile, is enabled by and will enable the aac_ltp option. Introduced in
MPEG4.

aac_main
Main-type prediction profile, is enabled by and will enable the aac_pred option. Introduced in
MPEG2.

If this option is unspecified it is set to aac_low.

ac3 and ac3_fixed
AC-3 audio encoders.

These encoders implement part of ATSC A/52:2010 and ETSI TS 102 366, as well as the undocumented
RealAudio 3 (a.k.a. dnet).

The ac3 encoder uses floating-point math, while the ac3_fixed encoder only uses fixed-point integer math.
This does not mean that one is always faster, just that one or the other may be better suited to a
particular system. The ac3_fixed encoder is not the default codec for any of the output formats, so it
must be specified explicitly using the option "-acodec ac3_fixed" in order to use it.

AC-3 Metadata

The AC-3 metadata options are used to set parameters that describe the audio, but in most cases do not
affect the audio encoding itself. Some of the options do directly affect or influence the decoding and
playback of the resulting bitstream, while others are just for informational purposes. A few of the
options will add bits to the output stream that could otherwise be used for audio data, and will thus
affect the quality of the output. Those will be indicated accordingly with a note in the option list
below.

These parameters are described in detail in several publicly-available documents.

*<<http://www.atsc.org/cms/standards/a_52-2010.pdf>>
*<<http://www.atsc.org/cms/standards/a_54a_with_corr_1.pdf>>
*<<http://www.dolby.com/uploadedFiles/zz-_Shared_Assets/English_PDFs/Professional/18_Metadata.Guide.pdf>>
*<<http://www.dolby.com/uploadedFiles/zz-_Shared_Assets/English_PDFs/Professional/46_DDEncodingGuidelines.pdf>>

Metadata Control Options

-per_frame_metadata boolean
Allow Per-Frame Metadata. Specifies if the encoder should check for changing metadata for each frame.

0 The metadata values set at initialization will be used for every frame in the stream. (default)

1 Metadata values can be changed before encoding each frame.

Downmix Levels

-center_mixlev level
Center Mix Level. The amount of gain the decoder should apply to the center channel when downmixing
to stereo. This field will only be written to the bitstream if a center channel is present. The value
is specified as a scale factor. There are 3 valid values:

0.707
Apply -3dB gain

0.595
Apply -4.5dB gain (default)

0.500
Apply -6dB gain

-surround_mixlev level
Surround Mix Level. The amount of gain the decoder should apply to the surround channel(s) when
downmixing to stereo. This field will only be written to the bitstream if one or more surround
channels are present. The value is specified as a scale factor. There are 3 valid values:

0.707
Apply -3dB gain

0.500
Apply -6dB gain (default)

0.000
Silence Surround Channel(s)

Audio Production Information

Audio Production Information is optional information describing the mixing environment. Either none or
both of the fields are written to the bitstream.

-mixing_level number
Mixing Level. Specifies peak sound pressure level (SPL) in the production environment when the mix
was mastered. Valid values are 80 to 111, or -1 for unknown or not indicated. The default value is
-1, but that value cannot be used if the Audio Production Information is written to the bitstream.
Therefore, if the "room_type" option is not the default value, the "mixing_level" option must not be
-1.

-room_type type
Room Type. Describes the equalization used during the final mixing session at the studio or on the
dubbing stage. A large room is a dubbing stage with the industry standard X-curve equalization; a
small room has flat equalization. This field will not be written to the bitstream if both the
"mixing_level" option and the "room_type" option have the default values.

0
notindicated
Not Indicated (default)

1
large
Large Room

2
small
Small Room

Other Metadata Options

-copyright boolean
Copyright Indicator. Specifies whether a copyright exists for this audio.

0
off No Copyright Exists (default)

1
on Copyright Exists

-dialnorm value
Dialogue Normalization. Indicates how far the average dialogue level of the program is below digital
100% full scale (0 dBFS). This parameter determines a level shift during audio reproduction that sets
the average volume of the dialogue to a preset level. The goal is to match volume level between
program sources. A value of -31dB will result in no volume level change, relative to the source
volume, during audio reproduction. Valid values are whole numbers in the range -31 to -1, with -31
being the default.

-dsur_mode mode
Dolby Surround Mode. Specifies whether the stereo signal uses Dolby Surround (Pro Logic). This field
will only be written to the bitstream if the audio stream is stereo. Using this option does NOT mean
the encoder will actually apply Dolby Surround processing.

0
notindicated
Not Indicated (default)

1
off Not Dolby Surround Encoded

2
on Dolby Surround Encoded

-original boolean
Original Bit Stream Indicator. Specifies whether this audio is from the original source and not a
copy.

0
off Not Original Source

1
on Original Source (default)

Extended Bitstream Information

The extended bitstream options are part of the Alternate Bit Stream Syntax as specified in Annex D of the
A/52:2010 standard. It is grouped into 2 parts. If any one parameter in a group is specified, all values
in that group will be written to the bitstream. Default values are used for those that are written but
have not been specified. If the mixing levels are written, the decoder will use these values instead of
the ones specified in the "center_mixlev" and "surround_mixlev" options if it supports the Alternate Bit
Stream Syntax.

Extended Bitstream Information - Part 1

-dmix_mode mode
Preferred Stereo Downmix Mode. Allows the user to select either Lt/Rt (Dolby Surround) or Lo/Ro
(normal stereo) as the preferred stereo downmix mode.

0
notindicated
Not Indicated (default)

1
ltrt
Lt/Rt Downmix Preferred

2
loro
Lo/Ro Downmix Preferred

-ltrt_cmixlev level
Lt/Rt Center Mix Level. The amount of gain the decoder should apply to the center channel when
downmixing to stereo in Lt/Rt mode.

1.414
Apply +3dB gain

1.189
Apply +1.5dB gain

1.000
Apply 0dB gain

0.841
Apply -1.5dB gain

0.707
Apply -3.0dB gain

0.595
Apply -4.5dB gain (default)

0.500
Apply -6.0dB gain

0.000
Silence Center Channel

-ltrt_surmixlev level
Lt/Rt Surround Mix Level. The amount of gain the decoder should apply to the surround channel(s) when
downmixing to stereo in Lt/Rt mode.

0.841
Apply -1.5dB gain

0.707
Apply -3.0dB gain

0.595
Apply -4.5dB gain

0.500
Apply -6.0dB gain (default)

0.000
Silence Surround Channel(s)

-loro_cmixlev level
Lo/Ro Center Mix Level. The amount of gain the decoder should apply to the center channel when
downmixing to stereo in Lo/Ro mode.

1.414
Apply +3dB gain

1.189
Apply +1.5dB gain

1.000
Apply 0dB gain

0.841
Apply -1.5dB gain

0.707
Apply -3.0dB gain

0.595
Apply -4.5dB gain (default)

0.500
Apply -6.0dB gain

0.000
Silence Center Channel

-loro_surmixlev level
Lo/Ro Surround Mix Level. The amount of gain the decoder should apply to the surround channel(s) when
downmixing to stereo in Lo/Ro mode.

0.841
Apply -1.5dB gain

0.707
Apply -3.0dB gain

0.595
Apply -4.5dB gain

0.500
Apply -6.0dB gain (default)

0.000
Silence Surround Channel(s)

Extended Bitstream Information - Part 2

-dsurex_mode mode
Dolby Surround EX Mode. Indicates whether the stream uses Dolby Surround EX (7.1 matrixed to 5.1).
Using this option does NOT mean the encoder will actually apply Dolby Surround EX processing.

0
notindicated
Not Indicated (default)

1
on Dolby Surround EX Off

2
off Dolby Surround EX On

-dheadphone_mode mode
Dolby Headphone Mode. Indicates whether the stream uses Dolby Headphone encoding (multi-channel
matrixed to 2.0 for use with headphones). Using this option does NOT mean the encoder will actually
apply Dolby Headphone processing.

0
notindicated
Not Indicated (default)

1
on Dolby Headphone Off

2
off Dolby Headphone On

-ad_conv_type type
A/D Converter Type. Indicates whether the audio has passed through HDCD A/D conversion.

0
standard
Standard A/D Converter (default)

1
hdcd
HDCD A/D Converter

Other AC-3 Encoding Options

-stereo_rematrixing boolean
Stereo Rematrixing. Enables/Disables use of rematrixing for stereo input. This is an optional AC-3
feature that increases quality by selectively encoding the left/right channels as mid/side. This
option is enabled by default, and it is highly recommended that it be left as enabled except for
testing purposes.

cutoff frequency
Set lowpass cutoff frequency. If unspecified, the encoder selects a default determined by various
other encoding parameters.

Floating-Point-Only AC-3 Encoding Options

These options are only valid for the floating-point encoder and do not exist for the fixed-point encoder
due to the corresponding features not being implemented in fixed-point.

-channel_coupling boolean
Enables/Disables use of channel coupling, which is an optional AC-3 feature that increases quality by
combining high frequency information from multiple channels into a single channel. The per-channel
high frequency information is sent with less accuracy in both the frequency and time domains. This
allows more bits to be used for lower frequencies while preserving enough information to reconstruct
the high frequencies. This option is enabled by default for the floating-point encoder and should
generally be left as enabled except for testing purposes or to increase encoding speed.

-1
auto
Selected by Encoder (default)

0
off Disable Channel Coupling

1
on Enable Channel Coupling

-cpl_start_band number
Coupling Start Band. Sets the channel coupling start band, from 1 to 15. If a value higher than the
bandwidth is used, it will be reduced to 1 less than the coupling end band. If auto is used, the
start band will be determined by the encoder based on the bit rate, sample rate, and channel layout.
This option has no effect if channel coupling is disabled.

-1
auto
Selected by Encoder (default)

flac
FLAC (Free Lossless Audio Codec) Encoder

Options

The following options are supported by FFmpeg's flac encoder.

compression_level
Sets the compression level, which chooses defaults for many other options if they are not set
explicitly. Valid values are from 0 to 12, 5 is the default.

frame_size
Sets the size of the frames in samples per channel.

lpc_coeff_precision
Sets the LPC coefficient precision, valid values are from 1 to 15, 15 is the default.

lpc_type
Sets the first stage LPC algorithm

none
LPC is not used

fixed
fixed LPC coefficients

levinson
cholesky
lpc_passes
Number of passes to use for Cholesky factorization during LPC analysis

min_partition_order
The minimum partition order

max_partition_order
The maximum partition order

prediction_order_method
estimation
2level
4level
8level
search
Bruteforce search

log
ch_mode
Channel mode

auto
The mode is chosen automatically for each frame

indep
Channels are independently coded

left_side
right_side
mid_side
exact_rice_parameters
Chooses if rice parameters are calculated exactly or approximately. if set to 1 then they are chosen
exactly, which slows the code down slightly and improves compression slightly.

multi_dim_quant
Multi Dimensional Quantization. If set to 1 then a 2nd stage LPC algorithm is applied after the first
stage to finetune the coefficients. This is quite slow and slightly improves compression.

opus
Opus encoder.

This is a native FFmpeg encoder for the Opus format. Currently its in development and only implements the
CELT part of the codec. Its quality is usually worse and at best is equal to the libopus encoder.

Options

b Set bit rate in bits/s. If unspecified it uses the number of channels and the layout to make a good
guess.

opus_delay
Sets the maximum delay in milliseconds. Lower delays than 20ms will very quickly decrease quality.

libfdk_aac
libfdk-aac AAC (Advanced Audio Coding) encoder wrapper.

The libfdk-aac library is based on the Fraunhofer FDK AAC code from the Android project.

Requires the presence of the libfdk-aac headers and library during configuration. You need to explicitly
configure the build with "--enable-libfdk-aac". The library is also incompatible with GPL, so if you
allow the use of GPL, you should configure with "--enable-gpl --enable-nonfree --enable-libfdk-aac".

This encoder has support for the AAC-HE profiles.

VBR encoding, enabled through the vbr or flags +qscale options, is experimental and only works with some
combinations of parameters.

Support for encoding 7.1 audio is only available with libfdk-aac 0.1.3 or higher.

For more information see the fdk-aac project at <http://sourceforge.net/p/opencore-amr/fdk-aac/>.

Options

The following options are mapped on the shared FFmpeg codec options.

b Set bit rate in bits/s. If the bitrate is not explicitly specified, it is automatically set to a
suitable value depending on the selected profile.

In case VBR mode is enabled the option is ignored.

ar Set audio sampling rate (in Hz).

channels
Set the number of audio channels.

flags +qscale
Enable fixed quality, VBR (Variable Bit Rate) mode. Note that VBR is implicitly enabled when the vbr
value is positive.

cutoff
Set cutoff frequency. If not specified (or explicitly set to 0) it will use a value automatically
computed by the library. Default value is 0.

profile
Set audio profile.

The following profiles are recognized:

aac_low
Low Complexity AAC (LC)

aac_he
High Efficiency AAC (HE-AAC)

aac_he_v2
High Efficiency AAC version 2 (HE-AACv2)

aac_ld
Low Delay AAC (LD)

aac_eld
Enhanced Low Delay AAC (ELD)

If not specified it is set to aac_low.

The following are private options of the libfdk_aac encoder.

afterburner
Enable afterburner feature if set to 1, disabled if set to 0. This improves the quality but also the
required processing power.

Default value is 1.

eld_sbr
Enable SBR (Spectral Band Replication) for ELD if set to 1, disabled if set to 0.

Default value is 0.

eld_v2
Enable ELDv2 (LD-MPS extension for ELD stereo signals) for ELDv2 if set to 1, disabled if set to 0.

Note that option is available when fdk-aac version
(AACENCODER_LIB_VL0.AACENCODER_LIB_VL1.AACENCODER_LIB_VL2) > (4.0.0).

Default value is 0.

signaling
Set SBR/PS signaling style.

It can assume one of the following values:

default
choose signaling implicitly (explicit hierarchical by default, implicit if global header is
disabled)

implicit
implicit backwards compatible signaling

explicit_sbr
explicit SBR, implicit PS signaling

explicit_hierarchical
explicit hierarchical signaling

Default value is default.

latm
Output LATM/LOAS encapsulated data if set to 1, disabled if set to 0.

Default value is 0.

header_period
Set StreamMuxConfig and PCE repetition period (in frames) for sending in-band configuration buffers
within LATM/LOAS transport layer.

Must be a 16-bits non-negative integer.

Default value is 0.

vbr Set VBR mode, from 1 to 5. 1 is lowest quality (though still pretty good) and 5 is highest quality. A
value of 0 will disable VBR, and CBR (Constant Bit Rate) is enabled.

Currently only the aac_low profile supports VBR encoding.

VBR modes 1-5 correspond to roughly the following average bit rates:

1 32 kbps/channel

2 40 kbps/channel

3 48-56 kbps/channel

4 64 kbps/channel

5 about 80-96 kbps/channel

Default value is 0.

frame_length
Set the audio frame length in samples. Default value is the internal default of the library. Refer to
the library's documentation for information about supported values.

Examples

• Use ffmpeg to convert an audio file to VBR AAC in an M4A (MP4) container:

ffmpeg -i input.wav -codec:a libfdk_aac -vbr 3 output.m4a

• Use ffmpeg to convert an audio file to CBR 64k kbps AAC, using the High-Efficiency AAC profile:

ffmpeg -i input.wav -c:a libfdk_aac -profile:a aac_he -b:a 64k output.m4a

libmp3lame
LAME (Lame Ain't an MP3 Encoder) MP3 encoder wrapper.

Requires the presence of the libmp3lame headers and library during configuration. You need to explicitly
configure the build with "--enable-libmp3lame".

See libshine for a fixed-point MP3 encoder, although with a lower quality.

Options

The following options are supported by the libmp3lame wrapper. The lame-equivalent of the options are
listed in parentheses.

b (-b)
Set bitrate expressed in bits/s for CBR or ABR. LAME "bitrate" is expressed in kilobits/s.

q (-V)
Set constant quality setting for VBR. This option is valid only using the ffmpeg command-line tool.
For library interface users, use global_quality.

compression_level (-q)
Set algorithm quality. Valid arguments are integers in the 0-9 range, with 0 meaning highest quality
but slowest, and 9 meaning fastest while producing the worst quality.

cutoff (--lowpass)
Set lowpass cutoff frequency. If unspecified, the encoder dynamically adjusts the cutoff.

reservoir
Enable use of bit reservoir when set to 1. Default value is 1. LAME has this enabled by default, but
can be overridden by use --nores option.

joint_stereo (-m j)
Enable the encoder to use (on a frame by frame basis) either L/R stereo or mid/side stereo. Default
value is 1.

abr (--abr)
Enable the encoder to use ABR when set to 1. The lame --abr sets the target bitrate, while this
options only tells FFmpeg to use ABR still relies on b to set bitrate.

original (-o)
Set MPEG audio original flag when set to 1. The default value is 1 (enabled).

libopencore-amrnb
OpenCORE Adaptive Multi-Rate Narrowband encoder.

Requires the presence of the libopencore-amrnb headers and library during configuration. You need to
explicitly configure the build with "--enable-libopencore-amrnb --enable-version3".

This is a mono-only encoder. Officially it only supports 8000Hz sample rate, but you can override it by
setting strict to unofficial or lower.

Options

b Set bitrate in bits per second. Only the following bitrates are supported, otherwise libavcodec will
round to the nearest valid bitrate.

4750
5150
5900
6700
7400
7950
10200
12200
dtx Allow discontinuous transmission (generate comfort noise) when set to 1. The default value is 0
(disabled).

libopus
libopus Opus Interactive Audio Codec encoder wrapper.

Requires the presence of the libopus headers and library during configuration. You need to explicitly
configure the build with "--enable-libopus".

Option Mapping

Most libopus options are modelled after the opusenc utility from opus-tools. The following is an option
mapping chart describing options supported by the libopus wrapper, and their opusenc-equivalent in
parentheses.

b (bitrate)
Set the bit rate in bits/s. FFmpeg's b option is expressed in bits/s, while opusenc's bitrate in
kilobits/s.

vbr (vbr, hard-cbr, and cvbr)
Set VBR mode. The FFmpeg vbr option has the following valid arguments, with the opusenc equivalent
options in parentheses:

off (hard-cbr)
Use constant bit rate encoding.

on (vbr)
Use variable bit rate encoding (the default).

constrained (cvbr)
Use constrained variable bit rate encoding.

compression_level (comp)
Set encoding algorithm complexity. Valid options are integers in the 0-10 range. 0 gives the fastest
encodes but lower quality, while 10 gives the highest quality but slowest encoding. The default is
10.

frame_duration (framesize)
Set maximum frame size, or duration of a frame in milliseconds. The argument must be exactly the
following: 2.5, 5, 10, 20, 40, 60. Smaller frame sizes achieve lower latency but less quality at a
given bitrate. Sizes greater than 20ms are only interesting at fairly low bitrates. The default is
20ms.

packet_loss (expect-loss)
Set expected packet loss percentage. The default is 0.

fec (n/a)
Enable inband forward error correction. packet_loss must be non-zero to take advantage - frequency of
FEC 'side-data' is proportional to expected packet loss. Default is disabled.

application (N.A.)
Set intended application type. Valid options are listed below:

voip
Favor improved speech intelligibility.

audio
Favor faithfulness to the input (the default).

lowdelay
Restrict to only the lowest delay modes by disabling voice-optimized modes.

cutoff (N.A.)
Set cutoff bandwidth in Hz. The argument must be exactly one of the following: 4000, 6000, 8000,
12000, or 20000, corresponding to narrowband, mediumband, wideband, super wideband, and fullband
respectively. The default is 0 (cutoff disabled). Note that libopus forces a wideband cutoff for
bitrates < 15 kbps, unless CELT-only (application set to lowdelay) mode is used.

mapping_family (mapping_family)
Set channel mapping family to be used by the encoder. The default value of -1 uses mapping family 0
for mono and stereo inputs, and mapping family 1 otherwise. The default also disables the surround
masking and LFE bandwidth optimzations in libopus, and requires that the input contains 8 channels or
fewer.

Other values include 0 for mono and stereo, 1 for surround sound with masking and LFE bandwidth
optimizations, and 255 for independent streams with an unspecified channel layout.

apply_phase_inv (N.A.) (requires libopus >= 1.2)
If set to 0, disables the use of phase inversion for intensity stereo, improving the quality of mono
downmixes, but slightly reducing normal stereo quality. The default is 1 (phase inversion enabled).

libshine
Shine Fixed-Point MP3 encoder wrapper.

Shine is a fixed-point MP3 encoder. It has a far better performance on platforms without an FPU, e.g.
armel CPUs, and some phones and tablets. However, as it is more targeted on performance than quality, it
is not on par with LAME and other production-grade encoders quality-wise. Also, according to the
project's homepage, this encoder may not be free of bugs as the code was written a long time ago and the
project was dead for at least 5 years.

This encoder only supports stereo and mono input. This is also CBR-only.

The original project (last updated in early 2007) is at <http://sourceforge.net/projects/libshine-fxp/>.
We only support the updated fork by the Savonet/Liquidsoap project at <https://github.com/savonet/shine>.

Requires the presence of the libshine headers and library during configuration. You need to explicitly
configure the build with "--enable-libshine".

VIDEO ENCODERS

A description of some of the currently available video encoders follows.

a64_multi, a64_multi5
A64 / Commodore 64 multicolor charset encoder. "a64_multi5" is extended with 5th color (colram).

Cinepak
Cinepak aka CVID encoder. Compatible with Windows 3.1 and vintage MacOS.

Options

g integer
Keyframe interval. A keyframe is inserted at least every "-g" frames, sometimes sooner.

q:v integer
Quality factor. Lower is better. Higher gives lower bitrate. The following table lists bitrates when
encoding akiyo_cif.y4m for various values of "-q:v" with "-g 100":

"-q:v 1" 1918 kb/s
"-q:v 2" 1735 kb/s
"-q:v 4" 1500 kb/s
"-q:v 10" 1041 kb/s
"-q:v 20" 826 kb/s
"-q:v 40" 553 kb/s
"-q:v 100" 394 kb/s
"-q:v 200" 312 kb/s
"-q:v 400" 266 kb/s
"-q:v 1000" 237 kb/s
max_extra_cb_iterations integer
Max extra codebook recalculation passes, more is better and slower.

skip_empty_cb boolean
Avoid wasting bytes, ignore vintage MacOS decoder.

max_strips integer
min_strips integer
The minimum and maximum number of strips to use. Wider range sometimes improves quality. More
strips is generally better quality but costs more bits. Fewer strips tend to yield more keyframes.
Vintage compatible is 1..3.

strip_number_adaptivity integer
How much number of strips is allowed to change between frames. Higher is better but slower.

GIF
GIF image/animation encoder.

Options

gifflags integer
Sets the flags used for GIF encoding.

offsetting
Enables picture offsetting.

Default is enabled.

transdiff
Enables transparency detection between frames.

Default is enabled.

gifimage integer
Enables encoding one full GIF image per frame, rather than an animated GIF.

Default value is 0.

global_palette integer
Writes a palette to the global GIF header where feasible.

If disabled, every frame will always have a palette written, even if there is a global palette
supplied.

Default value is 1.

Hap
Vidvox Hap video encoder.

Options

format integer
Specifies the Hap format to encode.

hap
hap_alpha
hap_q

Default value is hap.

chunks integer
Specifies the number of chunks to split frames into, between 1 and 64. This permits multithreaded
decoding of large frames, potentially at the cost of data-rate. The encoder may modify this value to
divide frames evenly.

Default value is 1.

compressor integer
Specifies the second-stage compressor to use. If set to none, chunks will be limited to 1, as chunked
uncompressed frames offer no benefit.

none
snappy

Default value is snappy.

jpeg2000
The native jpeg 2000 encoder is lossy by default, the "-q:v" option can be used to set the encoding
quality. Lossless encoding can be selected with "-pred 1".

Options

format integer
Can be set to either "j2k" or "jp2" (the default) that makes it possible to store non-rgb pix_fmts.

tile_width integer
Sets tile width. Range is 1 to 1073741824. Default is 256.

tile_height integer
Sets tile height. Range is 1 to 1073741824. Default is 256.

pred integer
Allows setting the discrete wavelet transform (DWT) type

dwt97int (Lossy)
dwt53 (Lossless)

Default is "dwt97int"

sop boolean
Enable this to add SOP marker at the start of each packet. Disabled by default.

eph boolean
Enable this to add EPH marker at the end of each packet header. Disabled by default.

prog integer
Sets the progression order to be used by the encoder. Possible values are:

lrcp
rlcp
rpcl
pcrl
cprl

Set to "lrcp" by default.

layer_rates string
By default, when this option is not used, compression is done using the quality metric. This option
allows for compression using compression ratio. The compression ratio for each level could be
specified. The compression ratio of a layer "l" species the what ratio of total file size is
contained in the first "l" layers.

Example usage:

ffmpeg -i input.bmp -c:v jpeg2000 -layer_rates "100,10,1" output.j2k

This would compress the image to contain 3 layers, where the data contained in the first layer would
be compressed by 1000 times, compressed by 100 in the first two layers, and shall contain all data
while using all 3 layers.

librav1e
rav1e AV1 encoder wrapper.

Requires the presence of the rav1e headers and library during configuration. You need to explicitly
configure the build with "--enable-librav1e".

Options

qmax
Sets the maximum quantizer to use when using bitrate mode.

qmin
Sets the minimum quantizer to use when using bitrate mode.

qp Uses quantizer mode to encode at the given quantizer (0-255).

speed
Selects the speed preset (0-10) to encode with.

tiles
Selects how many tiles to encode with.

tile-rows
Selects how many rows of tiles to encode with.

tile-columns
Selects how many columns of tiles to encode with.

rav1e-params
Set rav1e options using a list of key=value pairs separated by ":". See rav1e --help for a list of
options.

For example to specify librav1e encoding options with -rav1e-params:

ffmpeg -i input -c:v librav1e -b:v 500K -rav1e-params speed=5:low_latency=true output.mp4

libaom-av1
libaom AV1 encoder wrapper.

Requires the presence of the libaom headers and library during configuration. You need to explicitly
configure the build with "--enable-libaom".

Options

The wrapper supports the following standard libavcodec options:

b Set bitrate target in bits/second. By default this will use variable-bitrate mode. If maxrate and
minrate are also set to the same value then it will use constant-bitrate mode, otherwise if crf is
set as well then it will use constrained-quality mode.

g keyint_min
Set key frame placement. The GOP size sets the maximum distance between key frames; if zero the
output stream will be intra-only. The minimum distance is ignored unless it is the same as the GOP
size, in which case key frames will always appear at a fixed interval. Not set by default, so
without this option the library has completely free choice about where to place key frames.

qmin qmax
Set minimum/maximum quantisation values. Valid range is from 0 to 63 (warning: this does not match
the quantiser values actually used by AV1 - divide by four to map real quantiser values to this
range). Defaults to min/max (no constraint).

minrate maxrate bufsize rc_init_occupancy
Set rate control buffering parameters. Not used if not set - defaults to unconstrained variable
bitrate.

threads
Set the number of threads to use while encoding. This may require the tiles or row-mt options to
also be set to actually use the specified number of threads fully. Defaults to the number of hardware
threads supported by the host machine.

profile
Set the encoding profile. Defaults to using the profile which matches the bit depth and chroma
subsampling of the input.

The wrapper also has some specific options:

cpu-used
Set the quality/encoding speed tradeoff. Valid range is from 0 to 8, higher numbers indicating
greater speed and lower quality. The default value is 1, which will be slow and high quality.

auto-alt-ref
Enable use of alternate reference frames. Defaults to the internal default of the library.

arnr-max-frames (frames)
Set altref noise reduction max frame count. Default is -1.

arnr-strength (strength)
Set altref noise reduction filter strength. Range is -1 to 6. Default is -1.

aq-mode (aq-mode)
Set adaptive quantization mode. Possible values:

none (0)
Disabled.

variance (1)
Variance-based.

complexity (2)
Complexity-based.

cyclic (3)
Cyclic refresh.

tune (tune)
Set the distortion metric the encoder is tuned with. Default is "psnr".

psnr (0)
ssim (1)
lag-in-frames
Set the maximum number of frames which the encoder may keep in flight at any one time for lookahead
purposes. Defaults to the internal default of the library.

error-resilience
Enable error resilience features:

default
Improve resilience against losses of whole frames.

Not enabled by default.

crf Set the quality/size tradeoff for constant-quality (no bitrate target) and constrained-quality (with
maximum bitrate target) modes. Valid range is 0 to 63, higher numbers indicating lower quality and
smaller output size. Only used if set; by default only the bitrate target is used.

static-thresh
Set a change threshold on blocks below which they will be skipped by the encoder. Defined in
arbitrary units as a nonnegative integer, defaulting to zero (no blocks are skipped).

drop-threshold
Set a threshold for dropping frames when close to rate control bounds. Defined as a percentage of
the target buffer - when the rate control buffer falls below this percentage, frames will be dropped
until it has refilled above the threshold. Defaults to zero (no frames are dropped).

denoise-noise-level (level)
Amount of noise to be removed for grain synthesis. Grain synthesis is disabled if this option is not
set or set to 0.

denoise-block-size (pixels)
Block size used for denoising for grain synthesis. If not set, AV1 codec uses the default value of
32.

undershoot-pct (pct)
Set datarate undershoot (min) percentage of the target bitrate. Range is -1 to 100. Default is -1.

overshoot-pct (pct)
Set datarate overshoot (max) percentage of the target bitrate. Range is -1 to 1000. Default is -1.

minsection-pct (pct)
Minimum percentage variation of the GOP bitrate from the target bitrate. If minsection-pct is not
set, the libaomenc wrapper computes it as follows: "(minrate * 100 / bitrate)". Range is -1 to 100.
Default is -1 (unset).

maxsection-pct (pct)
Maximum percentage variation of the GOP bitrate from the target bitrate. If maxsection-pct is not
set, the libaomenc wrapper computes it as follows: "(maxrate * 100 / bitrate)". Range is -1 to 5000.
Default is -1 (unset).

frame-parallel (boolean)
Enable frame parallel decodability features. Default is true.

tiles
Set the number of tiles to encode the input video with, as columns x rows. Larger numbers allow
greater parallelism in both encoding and decoding, but may decrease coding efficiency. Defaults to
the minimum number of tiles required by the size of the input video (this is 1x1 (that is, a single
tile) for sizes up to and including 4K).

tile-columns tile-rows
Set the number of tiles as log2 of the number of tile rows and columns. Provided for compatibility
with libvpx/VP9.

row-mt (Requires libaom >= 1.0.0-759-g90a15f4f2)
Enable row based multi-threading. Disabled by default.

enable-cdef (boolean)
Enable Constrained Directional Enhancement Filter. The libaom-av1 encoder enables CDEF by default.

enable-restoration (boolean)
Enable Loop Restoration Filter. Default is true for libaom-av1.

enable-global-motion (boolean)
Enable the use of global motion for block prediction. Default is true.

enable-intrabc (boolean)
Enable block copy mode for intra block prediction. This mode is useful for screen content. Default is
true.

enable-rect-partitions (boolean) (Requires libaom >= v2.0.0)
Enable rectangular partitions. Default is true.

enable-1to4-partitions (boolean) (Requires libaom >= v2.0.0)
Enable 1:4/4:1 partitions. Default is true.

enable-ab-partitions (boolean) (Requires libaom >= v2.0.0)
Enable AB shape partitions. Default is true.

enable-angle-delta (boolean) (Requires libaom >= v2.0.0)
Enable angle delta intra prediction. Default is true.

enable-cfl-intra (boolean) (Requires libaom >= v2.0.0)
Enable chroma predicted from luma intra prediction. Default is true.

enable-filter-intra (boolean) (Requires libaom >= v2.0.0)
Enable filter intra predictor. Default is true.

enable-intra-edge-filter (boolean) (Requires libaom >= v2.0.0)
Enable intra edge filter. Default is true.

enable-smooth-intra (boolean) (Requires libaom >= v2.0.0)
Enable smooth intra prediction mode. Default is true.

enable-paeth-intra (boolean) (Requires libaom >= v2.0.0)
Enable paeth predictor in intra prediction. Default is true.

enable-palette (boolean) (Requires libaom >= v2.0.0)
Enable palette prediction mode. Default is true.

enable-flip-idtx (boolean) (Requires libaom >= v2.0.0)
Enable extended transform type, including FLIPADST_DCT, DCT_FLIPADST, FLIPADST_FLIPADST,
ADST_FLIPADST, FLIPADST_ADST, IDTX, V_DCT, H_DCT, V_ADST, H_ADST, V_FLIPADST, H_FLIPADST. Default is
true.

enable-tx64 (boolean) (Requires libaom >= v2.0.0)
Enable 64-pt transform. Default is true.

reduced-tx-type-set (boolean) (Requires libaom >= v2.0.0)
Use reduced set of transform types. Default is false.

use-intra-dct-only (boolean) (Requires libaom >= v2.0.0)
Use DCT only for INTRA modes. Default is false.

use-inter-dct-only (boolean) (Requires libaom >= v2.0.0)
Use DCT only for INTER modes. Default is false.

use-intra-default-tx-only (boolean) (Requires libaom >= v2.0.0)
Use Default-transform only for INTRA modes. Default is false.

enable-ref-frame-mvs (boolean) (Requires libaom >= v2.0.0)
Enable temporal mv prediction. Default is true.

enable-reduced-reference-set (boolean) (Requires libaom >= v2.0.0)
Use reduced set of single and compound references. Default is false.

enable-obmc (boolean) (Requires libaom >= v2.0.0)
Enable obmc. Default is true.

enable-dual-filter (boolean) (Requires libaom >= v2.0.0)
Enable dual filter. Default is true.

enable-diff-wtd-comp (boolean) (Requires libaom >= v2.0.0)
Enable difference-weighted compound. Default is true.

enable-dist-wtd-comp (boolean) (Requires libaom >= v2.0.0)
Enable distance-weighted compound. Default is true.

enable-onesided-comp (boolean) (Requires libaom >= v2.0.0)
Enable one sided compound. Default is true.

enable-interinter-wedge (boolean) (Requires libaom >= v2.0.0)
Enable interinter wedge compound. Default is true.

enable-interintra-wedge (boolean) (Requires libaom >= v2.0.0)
Enable interintra wedge compound. Default is true.

enable-masked-comp (boolean) (Requires libaom >= v2.0.0)
Enable masked compound. Default is true.

enable-interintra-comp (boolean) (Requires libaom >= v2.0.0)
Enable interintra compound. Default is true.

enable-smooth-interintra (boolean) (Requires libaom >= v2.0.0)
Enable smooth interintra mode. Default is true.

aom-params
Set libaom options using a list of key=value pairs separated by ":". For a list of supported options,
see aomenc --help under the section "AV1 Specific Options".

For example to specify libaom encoding options with -aom-params:

ffmpeg -i input -c:v libaom-av1 -b:v 500K -aom-params tune=psnr:enable-tpl-model=1 output.mp4

libsvtav1
SVT-AV1 encoder wrapper.

Requires the presence of the SVT-AV1 headers and library during configuration. You need to explicitly
configure the build with "--enable-libsvtav1".

Options

profile
Set the encoding profile.

main
high
professional
level
Set the operating point level. For example: '4.0'

hielevel
Set the Hierarchical prediction levels.

3level
4level
This is the default.

tier
Set the operating point tier.

main
This is the default.

high
qmax
Set the maximum quantizer to use when using a bitrate mode.

qmin
Set the minimum quantizer to use when using a bitrate mode.

crf Constant rate factor value used in crf rate control mode (0-63).

qp Set the quantizer used in cqp rate control mode (0-63).

sc_detection
Enable scene change detection.

la_depth
Set number of frames to look ahead (0-120).

preset
Set the quality-speed tradeoff, in the range 0 to 13. Higher values are faster but lower quality.

tile_rows
Set log2 of the number of rows of tiles to use (0-6).

tile_columns
Set log2 of the number of columns of tiles to use (0-4).

svtav1-params
Set SVT-AV1 options using a list of key=value pairs separated by ":". See the SVT-AV1 encoder user
guide for a list of accepted parameters.

libjxl
libjxl JPEG XL encoder wrapper.

Requires the presence of the libjxl headers and library during configuration. You need to explicitly
configure the build with "--enable-libjxl".

Options

The libjxl wrapper supports the following options:

distance
Set the target Butteraugli distance. This is a quality setting: lower distance yields higher quality,
with distance=1.0 roughly comparable to libjpeg Quality 90 for photographic content. Setting
distance=0.0 yields true lossless encoding. Valid values range between 0.0 and 15.0, and sane values
rarely exceed 5.0. Setting distance=0.1 usually attains transparency for most input. The default is
1.0.

effort
Set the encoding effort used. Higher effort values produce more consistent quality and usually
produces a better quality/bpp curve, at the cost of more CPU time required. Valid values range from 1
to 9, and the default is 7.

modular
Force the encoder to use Modular mode instead of choosing automatically. The default is to use VarDCT
for lossy encoding and Modular for lossless. VarDCT is generally superior to Modular for lossy
encoding but does not support lossless encoding.

libkvazaar
Kvazaar H.265/HEVC encoder.

Requires the presence of the libkvazaar headers and library during configuration. You need to explicitly
configure the build with --enable-libkvazaar.

Options

b Set target video bitrate in bit/s and enable rate control.

kvazaar-params
Set kvazaar parameters as a list of name=value pairs separated by commas (,). See kvazaar
documentation for a list of options.

libopenh264
Cisco libopenh264 H.264/MPEG-4 AVC encoder wrapper.

This encoder requires the presence of the libopenh264 headers and library during configuration. You need
to explicitly configure the build with "--enable-libopenh264". The library is detected using pkg-config.

For more information about the library see <http://www.openh264.org>.

Options

The following FFmpeg global options affect the configurations of the libopenh264 encoder.

b Set the bitrate (as a number of bits per second).

g Set the GOP size.

maxrate
Set the max bitrate (as a number of bits per second).

flags +global_header
Set global header in the bitstream.

slices
Set the number of slices, used in parallelized encoding. Default value is 0. This is only used when
slice_mode is set to fixed.

loopfilter
Enable loop filter, if set to 1 (automatically enabled). To disable set a value of 0.

profile
Set profile restrictions. If set to the value of main enable CABAC (set the
"SEncParamExt.iEntropyCodingModeFlag" flag to 1).

max_nal_size
Set maximum NAL size in bytes.

allow_skip_frames
Allow skipping frames to hit the target bitrate if set to 1.

libtheora
libtheora Theora encoder wrapper.

Requires the presence of the libtheora headers and library during configuration. You need to explicitly
configure the build with "--enable-libtheora".

For more information about the libtheora project see <http://www.theora.org/>.

Options

The following global options are mapped to internal libtheora options which affect the quality and the
bitrate of the encoded stream.

b Set the video bitrate in bit/s for CBR (Constant Bit Rate) mode. In case VBR (Variable Bit Rate)
mode is enabled this option is ignored.

flags
Used to enable constant quality mode (VBR) encoding through the qscale flag, and to enable the
"pass1" and "pass2" modes.

g Set the GOP size.

global_quality
Set the global quality as an integer in lambda units.

Only relevant when VBR mode is enabled with "flags +qscale". The value is converted to QP units by
dividing it by "FF_QP2LAMBDA", clipped in the [0 - 10] range, and then multiplied by 6.3 to get a
value in the native libtheora range [0-63]. A higher value corresponds to a higher quality.

q Enable VBR mode when set to a non-negative value, and set constant quality value as a double floating
point value in QP units.

The value is clipped in the [0-10] range, and then multiplied by 6.3 to get a value in the native
libtheora range [0-63].

This option is valid only using the ffmpeg command-line tool. For library interface users, use
global_quality.

Examples

• Set maximum constant quality (VBR) encoding with ffmpeg:

ffmpeg -i INPUT -codec:v libtheora -q:v 10 OUTPUT.ogg

• Use ffmpeg to convert a CBR 1000 kbps Theora video stream:

ffmpeg -i INPUT -codec:v libtheora -b:v 1000k OUTPUT.ogg

libvpx
VP8/VP9 format supported through libvpx.

Requires the presence of the libvpx headers and library during configuration. You need to explicitly
configure the build with "--enable-libvpx".

Options

The following options are supported by the libvpx wrapper. The vpxenc-equivalent options or values are
listed in parentheses for easy migration.

To reduce the duplication of documentation, only the private options and some others requiring special
attention are documented here. For the documentation of the undocumented generic options, see the Codec
Options chapter.

To get more documentation of the libvpx options, invoke the command ffmpeg -h encoder=libvpx, ffmpeg -h
encoder=libvpx-vp9 or vpxenc --help. Further information is available in the libvpx API documentation.

b (target-bitrate)
Set bitrate in bits/s. Note that FFmpeg's b option is expressed in bits/s, while vpxenc's target-
bitrate is in kilobits/s.

g (kf-max-dist)
keyint_min (kf-min-dist)
qmin (min-q)
Minimum (Best Quality) Quantizer.

qmax (max-q)
Maximum (Worst Quality) Quantizer. Can be changed per-frame.

bufsize (buf-sz, buf-optimal-sz)
Set ratecontrol buffer size (in bits). Note vpxenc's options are specified in milliseconds, the
libvpx wrapper converts this value as follows: "buf-sz = bufsize * 1000 / bitrate", "buf-optimal-sz =
bufsize * 1000 / bitrate * 5 / 6".

rc_init_occupancy (buf-initial-sz)
Set number of bits which should be loaded into the rc buffer before decoding starts. Note vpxenc's
option is specified in milliseconds, the libvpx wrapper converts this value as follows:
"rc_init_occupancy * 1000 / bitrate".

undershoot-pct
Set datarate undershoot (min) percentage of the target bitrate.

overshoot-pct
Set datarate overshoot (max) percentage of the target bitrate.

skip_threshold (drop-frame)
qcomp (bias-pct)
maxrate (maxsection-pct)
Set GOP max bitrate in bits/s. Note vpxenc's option is specified as a percentage of the target
bitrate, the libvpx wrapper converts this value as follows: "(maxrate * 100 / bitrate)".

minrate (minsection-pct)
Set GOP min bitrate in bits/s. Note vpxenc's option is specified as a percentage of the target
bitrate, the libvpx wrapper converts this value as follows: "(minrate * 100 / bitrate)".

minrate, maxrate, b end-usage=cbr
"(minrate == maxrate == bitrate)".

crf (end-usage=cq, cq-level)
tune (tune)
psnr (psnr)
ssim (ssim)
quality, deadline (deadline)
best
Use best quality deadline. Poorly named and quite slow, this option should be avoided as it may
give worse quality output than good.

good
Use good quality deadline. This is a good trade-off between speed and quality when used with the
cpu-used option.

realtime
Use realtime quality deadline.

speed, cpu-used (cpu-used)
Set quality/speed ratio modifier. Higher values speed up the encode at the cost of quality.

nr (noise-sensitivity)
static-thresh
Set a change threshold on blocks below which they will be skipped by the encoder.

slices (token-parts)
Note that FFmpeg's slices option gives the total number of partitions, while vpxenc's token-parts is
given as log2(partitions).

max-intra-rate
Set maximum I-frame bitrate as a percentage of the target bitrate. A value of 0 means unlimited.

force_key_frames
"VPX_EFLAG_FORCE_KF"

Alternate reference frame related
auto-alt-ref
Enable use of alternate reference frames (2-pass only). Values greater than 1 enable multi-layer
alternate reference frames (VP9 only).

arnr-maxframes
Set altref noise reduction max frame count.

arnr-type
Set altref noise reduction filter type: backward, forward, centered.

arnr-strength
Set altref noise reduction filter strength.

rc-lookahead, lag-in-frames (lag-in-frames)
Set number of frames to look ahead for frametype and ratecontrol.

min-gf-interval
Set minimum golden/alternate reference frame interval (VP9 only).

error-resilient
Enable error resiliency features.

sharpness integer
Increase sharpness at the expense of lower PSNR. The valid range is [0, 7].

ts-parameters
Sets the temporal scalability configuration using a :-separated list of key=value pairs. For example,
to specify temporal scalability parameters with "ffmpeg":

ffmpeg -i INPUT -c:v libvpx -ts-parameters ts_number_layers=3:\
ts_target_bitrate=250,500,1000:ts_rate_decimator=4,2,1:\
ts_periodicity=4:ts_layer_id=0,2,1,2:ts_layering_mode=3 OUTPUT

Below is a brief explanation of each of the parameters, please refer to "struct vpx_codec_enc_cfg" in
"vpx/vpx_encoder.h" for more details.

ts_number_layers
Number of temporal coding layers.

ts_target_bitrate
Target bitrate for each temporal layer (in kbps). (bitrate should be inclusive of the lower
temporal layer).

ts_rate_decimator
Frame rate decimation factor for each temporal layer.

ts_periodicity
Length of the sequence defining frame temporal layer membership.

ts_layer_id
Template defining the membership of frames to temporal layers.

ts_layering_mode
(optional) Selecting the temporal structure from a set of pre-defined temporal layering modes.
Currently supports the following options.

0 No temporal layering flags are provided internally, relies on flags being passed in using
"metadata" field in "AVFrame" with following keys.

vp8-flags
Sets the flags passed into the encoder to indicate the referencing scheme for the current
frame. Refer to function "vpx_codec_encode" in "vpx/vpx_encoder.h" for more details.

temporal_id
Explicitly sets the temporal id of the current frame to encode.

2 Two temporal layers. 0-1...

3 Three temporal layers. 0-2-1-2...; with single reference frame.

4 Same as option "3", except there is a dependency between the two temporal layer 2 frames
within the temporal period.

VP9-specific options
lossless
Enable lossless mode.

tile-columns
Set number of tile columns to use. Note this is given as log2(tile_columns). For example, 8 tile
columns would be requested by setting the tile-columns option to 3.

tile-rows
Set number of tile rows to use. Note this is given as log2(tile_rows). For example, 4 tile rows
would be requested by setting the tile-rows option to 2.

frame-parallel
Enable frame parallel decodability features.

aq-mode
Set adaptive quantization mode (0: off (default), 1: variance 2: complexity, 3: cyclic refresh,
4: equator360).

colorspace color-space
Set input color space. The VP9 bitstream supports signaling the following colorspaces:

rgb sRGB
bt709 bt709
unspecified unknown
bt470bg bt601
smpte170m smpte170
smpte240m smpte240
bt2020_ncl bt2020
row-mt boolean
Enable row based multi-threading.

tune-content
Set content type: default (0), screen (1), film (2).

corpus-complexity
Corpus VBR mode is a variant of standard VBR where the complexity distribution midpoint is passed
in rather than calculated for a specific clip or chunk.

The valid range is [0, 10000]. 0 (default) uses standard VBR.

enable-tpl boolean
Enable temporal dependency model.

ref-frame-config
Using per-frame metadata, set members of the structure "vpx_svc_ref_frame_config_t" in
"vpx/vp8cx.h" to fine-control referencing schemes and frame buffer management. Use a :-separated
list of key=value pairs. For example,

av_dict_set(&av_frame->metadata, "ref-frame-config", \
"rfc_update_buffer_slot=7:rfc_lst_fb_idx=0:rfc_gld_fb_idx=1:rfc_alt_fb_idx=2:rfc_reference_last=0:rfc_reference_golden=0:rfc_reference_alt_ref=0");

rfc_update_buffer_slot
Indicates the buffer slot number to update

rfc_update_last
Indicates whether to update the LAST frame

rfc_update_golden
Indicates whether to update GOLDEN frame

rfc_update_alt_ref
Indicates whether to update ALT_REF frame

rfc_lst_fb_idx
LAST frame buffer index

rfc_gld_fb_idx
GOLDEN frame buffer index

rfc_alt_fb_idx
ALT_REF frame buffer index

rfc_reference_last
Indicates whether to reference LAST frame

rfc_reference_golden
Indicates whether to reference GOLDEN frame

rfc_reference_alt_ref
Indicates whether to reference ALT_REF frame

rfc_reference_duration
Indicates frame duration

For more information about libvpx see: <http://www.webmproject.org/>

libwebp
libwebp WebP Image encoder wrapper

libwebp is Google's official encoder for WebP images. It can encode in either lossy or lossless mode.
Lossy images are essentially a wrapper around a VP8 frame. Lossless images are a separate codec developed
by Google.

Pixel Format

Currently, libwebp only supports YUV420 for lossy and RGB for lossless due to limitations of the format
and libwebp. Alpha is supported for either mode. Because of API limitations, if RGB is passed in when
encoding lossy or YUV is passed in for encoding lossless, the pixel format will automatically be
converted using functions from libwebp. This is not ideal and is done only for convenience.

Options

-lossless boolean
Enables/Disables use of lossless mode. Default is 0.

-compression_level integer
For lossy, this is a quality/speed tradeoff. Higher values give better quality for a given size at
the cost of increased encoding time. For lossless, this is a size/speed tradeoff. Higher values give
smaller size at the cost of increased encoding time. More specifically, it controls the number of
extra algorithms and compression tools used, and varies the combination of these tools. This maps to
the method option in libwebp. The valid range is 0 to 6. Default is 4.

-quality float
For lossy encoding, this controls image quality. For lossless encoding, this controls the effort and
time spent in compression. Range is 0 to 100. Default is 75.

-preset type
Configuration preset. This does some automatic settings based on the general type of the image.

none
Do not use a preset.

default
Use the encoder default.

picture
Digital picture, like portrait, inner shot

photo
Outdoor photograph, with natural lighting

drawing
Hand or line drawing, with high-contrast details

icon
Small-sized colorful images

text
Text-like

libx264, libx264rgb
x264 H.264/MPEG-4 AVC encoder wrapper.

This encoder requires the presence of the libx264 headers and library during configuration. You need to
explicitly configure the build with "--enable-libx264".

libx264 supports an impressive number of features, including 8x8 and 4x4 adaptive spatial transform,
adaptive B-frame placement, CAVLC/CABAC entropy coding, interlacing (MBAFF), lossless mode, psy
optimizations for detail retention (adaptive quantization, psy-RD, psy-trellis).

Many libx264 encoder options are mapped to FFmpeg global codec options, while unique encoder options are
provided through private options. Additionally the x264opts and x264-params private options allows one to
pass a list of key=value tuples as accepted by the libx264 "x264_param_parse" function.

The x264 project website is at <http://www.videolan.org/developers/x264.html>.

The libx264rgb encoder is the same as libx264, except it accepts packed RGB pixel formats as input
instead of YUV.

Supported Pixel Formats

x264 supports 8- to 10-bit color spaces. The exact bit depth is controlled at x264's configure time.

Options

The following options are supported by the libx264 wrapper. The x264-equivalent options or values are
listed in parentheses for easy migration.

To get a more accurate and extensive documentation of the libx264 options, invoke the command x264
--fullhelp or consult the libx264 documentation.

b (bitrate)
Set bitrate in bits/s. Note that FFmpeg's b option is expressed in bits/s, while x264's bitrate is in
kilobits/s.

bf (bframes)
g (keyint)
qmin (qpmin)
Minimum quantizer scale.

qmax (qpmax)
Maximum quantizer scale.

qdiff (qpstep)
Maximum difference between quantizer scales.

qblur (qblur)
Quantizer curve blur

qcomp (qcomp)
Quantizer curve compression factor

refs (ref)
Number of reference frames each P-frame can use. The range is from 0-16.

level (level)
Set the "x264_param_t.i_level_idc" value in case the value is positive, it is ignored otherwise.

This value can be set using the "AVCodecContext" API (e.g. by setting the "AVCodecContext" value
directly), and is specified as an integer mapped on a corresponding level (e.g. the value 31 maps to
H.264 level IDC "3.1", as defined in the "x264_levels" table). It is ignored when set to a non
positive value.

Alternatively it can be set as a private option, overriding the value set in "AVCodecContext", and in
this case must be specified as the level IDC identifier (e.g. "3.1"), as defined by H.264 Annex A.

sc_threshold (scenecut)
Sets the threshold for the scene change detection.

trellis (trellis)
Performs Trellis quantization to increase efficiency. Enabled by default.

nr (nr)
me_range (merange)
Maximum range of the motion search in pixels.

me_method (me)
Set motion estimation method. Possible values in the decreasing order of speed:

dia (dia)
epzs (dia)
Diamond search with radius 1 (fastest). epzs is an alias for dia.

hex (hex)
Hexagonal search with radius 2.

umh (umh)
Uneven multi-hexagon search.

esa (esa)
Exhaustive search.

tesa (tesa)
Hadamard exhaustive search (slowest).

forced-idr
Normally, when forcing a I-frame type, the encoder can select any type of I-frame. This option forces
it to choose an IDR-frame.

subq (subme)
Sub-pixel motion estimation method.

b_strategy (b-adapt)
Adaptive B-frame placement decision algorithm. Use only on first-pass.

keyint_min (min-keyint)
Minimum GOP size.

coder
Set entropy encoder. Possible values:

ac Enable CABAC.

vlc Enable CAVLC and disable CABAC. It generates the same effect as x264's --no-cabac option.

cmp Set full pixel motion estimation comparison algorithm. Possible values:

chroma
Enable chroma in motion estimation.

sad Ignore chroma in motion estimation. It generates the same effect as x264's --no-chroma-me option.

threads (threads)
Number of encoding threads.

thread_type
Set multithreading technique. Possible values:

slice
Slice-based multithreading. It generates the same effect as x264's --sliced-threads option.

frame
Frame-based multithreading.

flags
Set encoding flags. It can be used to disable closed GOP and enable open GOP by setting it to
"-cgop". The result is similar to the behavior of x264's --open-gop option.

rc_init_occupancy (vbv-init)
preset (preset)
Set the encoding preset.

tune (tune)
Set tuning of the encoding params.

profile (profile)
Set profile restrictions.

fastfirstpass
Enable fast settings when encoding first pass, when set to 1. When set to 0, it has the same effect
of x264's --slow-firstpass option.

crf (crf)
Set the quality for constant quality mode.

crf_max (crf-max)
In CRF mode, prevents VBV from lowering quality beyond this point.

qp (qp)
Set constant quantization rate control method parameter.

aq-mode (aq-mode)
Set AQ method. Possible values:

none (0)
Disabled.

variance (1)
Variance AQ (complexity mask).

autovariance (2)
Auto-variance AQ (experimental).

aq-strength (aq-strength)
Set AQ strength, reduce blocking and blurring in flat and textured areas.

psy Use psychovisual optimizations when set to 1. When set to 0, it has the same effect as x264's
--no-psy option.

psy-rd (psy-rd)
Set strength of psychovisual optimization, in psy-rd:psy-trellis format.

rc-lookahead (rc-lookahead)
Set number of frames to look ahead for frametype and ratecontrol.

weightb
Enable weighted prediction for B-frames when set to 1. When set to 0, it has the same effect as
x264's --no-weightb option.

weightp (weightp)
Set weighted prediction method for P-frames. Possible values:

none (0)
Disabled

simple (1)
Enable only weighted refs

smart (2)
Enable both weighted refs and duplicates

ssim (ssim)
Enable calculation and printing SSIM stats after the encoding.

intra-refresh (intra-refresh)
Enable the use of Periodic Intra Refresh instead of IDR frames when set to 1.

avcintra-class (class)
Configure the encoder to generate AVC-Intra. Valid values are 50,100 and 200

bluray-compat (bluray-compat)
Configure the encoder to be compatible with the bluray standard. It is a shorthand for setting
"bluray-compat=1 force-cfr=1".

b-bias (b-bias)
Set the influence on how often B-frames are used.

b-pyramid (b-pyramid)
Set method for keeping of some B-frames as references. Possible values:

none (none)
Disabled.

strict (strict)
Strictly hierarchical pyramid.

normal (normal)
Non-strict (not Blu-ray compatible).

mixed-refs
Enable the use of one reference per partition, as opposed to one reference per macroblock when set to
1. When set to 0, it has the same effect as x264's --no-mixed-refs option.

8x8dct
Enable adaptive spatial transform (high profile 8x8 transform) when set to 1. When set to 0, it has
the same effect as x264's --no-8x8dct option.

fast-pskip
Enable early SKIP detection on P-frames when set to 1. When set to 0, it has the same effect as
x264's --no-fast-pskip option.

aud (aud)
Enable use of access unit delimiters when set to 1.

mbtree
Enable use macroblock tree ratecontrol when set to 1. When set to 0, it has the same effect as x264's
--no-mbtree option.

deblock (deblock)
Set loop filter parameters, in alpha:beta form.

cplxblur (cplxblur)
Set fluctuations reduction in QP (before curve compression).

partitions (partitions)
Set partitions to consider as a comma-separated list of. Possible values in the list:

p8x8
8x8 P-frame partition.

p4x4
4x4 P-frame partition.

b8x8
4x4 B-frame partition.

i8x8
8x8 I-frame partition.

i4x4
4x4 I-frame partition. (Enabling p4x4 requires p8x8 to be enabled. Enabling i8x8 requires
adaptive spatial transform (8x8dct option) to be enabled.)

none (none)
Do not consider any partitions.

all (all)
Consider every partition.

direct-pred (direct)
Set direct MV prediction mode. Possible values:

none (none)
Disable MV prediction.

spatial (spatial)
Enable spatial predicting.

temporal (temporal)
Enable temporal predicting.

auto (auto)
Automatically decided.

slice-max-size (slice-max-size)
Set the limit of the size of each slice in bytes. If not specified but RTP payload size (ps) is
specified, that is used.

stats (stats)
Set the file name for multi-pass stats.

nal-hrd (nal-hrd)
Set signal HRD information (requires vbv-bufsize to be set). Possible values:

none (none)
Disable HRD information signaling.

vbr (vbr)
Variable bit rate.

cbr (cbr)
Constant bit rate (not allowed in MP4 container).

x264opts (N.A.)
Set any x264 option, see x264 --fullhelp for a list.

Argument is a list of key=value couples separated by ":". In filter and psy-rd options that use ":"
as a separator themselves, use "," instead. They accept it as well since long ago but this is kept
undocumented for some reason.

For example to specify libx264 encoding options with ffmpeg:

ffmpeg -i foo.mpg -c:v libx264 -x264opts keyint=123:min-keyint=20 -an out.mkv

a53cc boolean
Import closed captions (which must be ATSC compatible format) into output. Only the mpeg2 and h264
decoders provide these. Default is 1 (on).

udu_sei boolean
Import user data unregistered SEI if available into output. Default is 0 (off).

mb_info boolean
Set mb_info data through AVFrameSideData, only useful when used from the API. Default is 0 (off).

x264-params (N.A.)
Override the x264 configuration using a :-separated list of key=value parameters.

This option is functionally the same as the x264opts, but is duplicated for compatibility with the
Libav fork.

For example to specify libx264 encoding options with ffmpeg:

ffmpeg -i INPUT -c:v libx264 -x264-params level=30:bframes=0:weightp=0:\
cabac=0:ref=1:vbv-maxrate=768:vbv-bufsize=2000:analyse=all:me=umh:\
no-fast-pskip=1:subq=6:8x8dct=0:trellis=0 OUTPUT

Encoding ffpresets for common usages are provided so they can be used with the general presets system
(e.g. passing the pre option).

libx265
x265 H.265/HEVC encoder wrapper.

This encoder requires the presence of the libx265 headers and library during configuration. You need to
explicitly configure the build with --enable-libx265.

Options

b Sets target video bitrate.

bf
g Set the GOP size.

keyint_min
Minimum GOP size.

refs
Number of reference frames each P-frame can use. The range is from 1-16.

preset
Set the x265 preset.

tune
Set the x265 tune parameter.

profile
Set profile restrictions.

crf Set the quality for constant quality mode.

qp Set constant quantization rate control method parameter.

qmin
Minimum quantizer scale.

qmax
Maximum quantizer scale.

qdiff
Maximum difference between quantizer scales.

qblur
Quantizer curve blur

qcomp
Quantizer curve compression factor

i_qfactor
b_qfactor
forced-idr
Normally, when forcing a I-frame type, the encoder can select any type of I-frame. This option forces
it to choose an IDR-frame.

udu_sei boolean
Import user data unregistered SEI if available into output. Default is 0 (off).

x265-params
Set x265 options using a list of key=value couples separated by ":". See x265 --help for a list of
options.

For example to specify libx265 encoding options with -x265-params:

ffmpeg -i input -c:v libx265 -x265-params crf=26:psy-rd=1 output.mp4

libxavs2
xavs2 AVS2-P2/IEEE1857.4 encoder wrapper.

This encoder requires the presence of the libxavs2 headers and library during configuration. You need to
explicitly configure the build with --enable-libxavs2.

The following standard libavcodec options are used:

• b / bit_rate

• g / gop_size

• bf / max_b_frames

The encoder also has its own specific options:

Options

lcu_row_threads
Set the number of parallel threads for rows from 1 to 8 (default 5).

initial_qp
Set the xavs2 quantization parameter from 1 to 63 (default 34). This is used to set the initial qp
for the first frame.

qp Set the xavs2 quantization parameter from 1 to 63 (default 34). This is used to set the qp value
under constant-QP mode.

max_qp
Set the max qp for rate control from 1 to 63 (default 55).

min_qp
Set the min qp for rate control from 1 to 63 (default 20).

speed_level
Set the Speed level from 0 to 9 (default 0). Higher is better but slower.

log_level
Set the log level from -1 to 3 (default 0). -1: none, 0: error, 1: warning, 2: info, 3: debug.

xavs2-params
Set xavs2 options using a list of key=value couples separated by ":".

For example to specify libxavs2 encoding options with -xavs2-params:

ffmpeg -i input -c:v libxavs2 -xavs2-params RdoqLevel=0 output.avs2

libxvid
Xvid MPEG-4 Part 2 encoder wrapper.

This encoder requires the presence of the libxvidcore headers and library during configuration. You need
to explicitly configure the build with "--enable-libxvid --enable-gpl".

The native "mpeg4" encoder supports the MPEG-4 Part 2 format, so users can encode to this format without
this library.

Options

The following options are supported by the libxvid wrapper. Some of the following options are listed but
are not documented, and correspond to shared codec options. See the Codec Options chapter for their
documentation. The other shared options which are not listed have no effect for the libxvid encoder.

b
g
qmin
qmax
mpeg_quant
threads
bf
b_qfactor
b_qoffset
flags
Set specific encoding flags. Possible values:

mv4 Use four motion vector by macroblock.

aic Enable high quality AC prediction.

gray
Only encode grayscale.

gmc Enable the use of global motion compensation (GMC).

qpel
Enable quarter-pixel motion compensation.

cgop
Enable closed GOP.

global_header
Place global headers in extradata instead of every keyframe.

trellis
me_quality
Set motion estimation quality level. Possible values in decreasing order of speed and increasing
order of quality:

0 Use no motion estimation (default).

1, 2
Enable advanced diamond zonal search for 16x16 blocks and half-pixel refinement for 16x16 blocks.

3, 4
Enable all of the things described above, plus advanced diamond zonal search for 8x8 blocks and
half-pixel refinement for 8x8 blocks, also enable motion estimation on chroma planes for P and
B-frames.

5, 6
Enable all of the things described above, plus extended 16x16 and 8x8 blocks search.

mbd Set macroblock decision algorithm. Possible values in the increasing order of quality:

simple
Use macroblock comparing function algorithm (default).

bits
Enable rate distortion-based half pixel and quarter pixel refinement for 16x16 blocks.

rd Enable all of the things described above, plus rate distortion-based half pixel and quarter pixel
refinement for 8x8 blocks, and rate distortion-based search using square pattern.

lumi_aq
Enable lumi masking adaptive quantization when set to 1. Default is 0 (disabled).

variance_aq
Enable variance adaptive quantization when set to 1. Default is 0 (disabled).

When combined with lumi_aq, the resulting quality will not be better than any of the two specified
individually. In other words, the resulting quality will be the worse one of the two effects.

ssim
Set structural similarity (SSIM) displaying method. Possible values:

off Disable displaying of SSIM information.

avg Output average SSIM at the end of encoding to stdout. The format of showing the average SSIM is:

Average SSIM: %f

For users who are not familiar with C, %f means a float number, or a decimal (e.g. 0.939232).

frame
Output both per-frame SSIM data during encoding and average SSIM at the end of encoding to
stdout. The format of per-frame information is:

SSIM: avg: %1.3f min: %1.3f max: %1.3f

For users who are not familiar with C, %1.3f means a float number rounded to 3 digits after the
dot (e.g. 0.932).

ssim_acc
Set SSIM accuracy. Valid options are integers within the range of 0-4, while 0 gives the most
accurate result and 4 computes the fastest.

MediaFoundation
This provides wrappers to encoders (both audio and video) in the MediaFoundation framework. It can access
both SW and HW encoders. Video encoders can take input in either of nv12 or yuv420p form (some encoders
support both, some support only either - in practice, nv12 is the safer choice, especially among HW
encoders).

Microsoft RLE
Microsoft RLE aka MSRLE encoder. Only 8-bit palette mode supported. Compatible with Windows 3.1 and
Windows 95.

Options

g integer
Keyframe interval. A keyframe is inserted at least every "-g" frames, sometimes sooner.

mpeg2
MPEG-2 video encoder.

Options

profile
Select the mpeg2 profile to encode:

422
high
ss Spatially Scalable

snr SNR Scalable

main
simple
level
Select the mpeg2 level to encode:

high
high1440
main
low
seq_disp_ext integer
Specifies if the encoder should write a sequence_display_extension to the output.

-1
auto
Decide automatically to write it or not (this is the default) by checking if the data to be
written is different from the default or unspecified values.

0
never
Never write it.

1
always
Always write it.

video_format integer
Specifies the video_format written into the sequence display extension indicating the source of the
video pictures. The default is unspecified, can be component, pal, ntsc, secam or mac. For maximum
compatibility, use component.

a53cc boolean
Import closed captions (which must be ATSC compatible format) into output. Default is 1 (on).

png
PNG image encoder.

Private options

dpi integer
Set physical density of pixels, in dots per inch, unset by default

dpm integer
Set physical density of pixels, in dots per meter, unset by default

ProRes
Apple ProRes encoder.

FFmpeg contains 2 ProRes encoders, the prores-aw and prores-ks encoder. The used encoder can be chosen
with the "-vcodec" option.

Private Options for prores-ks

profile integer
Select the ProRes profile to encode

proxy
lt
standard
hq
4444
4444xq
quant_mat integer
Select quantization matrix.

auto
default
proxy
lt
standard
hq

If set to auto, the matrix matching the profile will be picked. If not set, the matrix providing the
highest quality, default, will be picked.

bits_per_mb integer
How many bits to allot for coding one macroblock. Different profiles use between 200 and 2400 bits
per macroblock, the maximum is 8000.

mbs_per_slice integer
Number of macroblocks in each slice (1-8); the default value (8) should be good in almost all
situations.

vendor string
Override the 4-byte vendor ID. A custom vendor ID like apl0 would claim the stream was produced by
the Apple encoder.

alpha_bits integer
Specify number of bits for alpha component. Possible values are 0, 8 and 16. Use 0 to disable alpha
plane coding.

Speed considerations

In the default mode of operation the encoder has to honor frame constraints (i.e. not produce frames with
size bigger than requested) while still making output picture as good as possible. A frame containing a
lot of small details is harder to compress and the encoder would spend more time searching for
appropriate quantizers for each slice.

Setting a higher bits_per_mb limit will improve the speed.

For the fastest encoding speed set the qscale parameter (4 is the recommended value) and do not set a
size constraint.

QSV Encoders
The family of Intel QuickSync Video encoders (MPEG-2, H.264, HEVC, JPEG/MJPEG, VP9, AV1)

Ratecontrol Method

The ratecontrol method is selected as follows:

• When global_quality is specified, a quality-based mode is used. Specifically this means either

- CQP - constant quantizer scale, when the qscale codec flag is also set (the -qscale ffmpeg
option).

- LA_ICQ - intelligent constant quality with lookahead, when the look_ahead option is also set.

- ICQ -- intelligent constant quality otherwise. For the ICQ modes, global quality range is 1 to
51, with 1 being the best quality.

• Otherwise, a bitrate-based mode is used. For all of those, you should specify at least the desired
average bitrate with the b option.

- LA - VBR with lookahead, when the look_ahead option is specified.

- VCM - video conferencing mode, when the vcm option is set.

- CBR - constant bitrate, when maxrate is specified and equal to the average bitrate.

- VBR - variable bitrate, when maxrate is specified, but is higher than the average bitrate.

- AVBR - average VBR mode, when maxrate is not specified, both avbr_accuracy and avbr_convergence
are set to non-zero. This mode is available for H264 and HEVC on Windows.

Note that depending on your system, a different mode than the one you specified may be selected by the
encoder. Set the verbosity level to verbose or higher to see the actual settings used by the QSV runtime.

Global Options -> MSDK Options

Additional libavcodec global options are mapped to MSDK options as follows:

• g/gop_size -> GopPicSize

• bf/max_b_frames+1 -> GopRefDist

• rc_init_occupancy/rc_initial_buffer_occupancy -> InitialDelayInKB

• slices -> NumSlice

• refs -> NumRefFrame

• b_strategy/b_frame_strategy -> BRefType

• cgop/CLOSED_GOP codec flag -> GopOptFlag

• For the CQP mode, the i_qfactor/i_qoffset and b_qfactor/b_qoffset set the difference between QPP and
QPI, and QPP and QPB respectively.

• Setting the coder option to the value vlc will make the H.264 encoder use CAVLC instead of CABAC.

Common Options

Following options are used by all qsv encoders.

async_depth
Specifies how many asynchronous operations an application performs before the application explicitly
synchronizes the result. If zero, the value is not specified.

preset
This option itemizes a range of choices from veryfast (best speed) to veryslow (best quality).

veryfast
faster
fast
medium
slow
slower
veryslow
forced_idr
Forcing I frames as IDR frames.

low_power
For encoders set this flag to ON to reduce power consumption and GPU usage.

Runtime Options

Following options can be used durning qsv encoding.

global_quality
i_quant_factor
i_quant_offset
b_quant_factor
b_quant_offset
Supported in h264_qsv and hevc_qsv. Change these value to reset qsv codec's qp configuration.

max_frame_size
Supported in h264_qsv and hevc_qsv. Change this value to reset qsv codec's MaxFrameSize
configuration.

gop_size
Change this value to reset qsv codec's gop configuration.

int_ref_type
int_ref_cycle_size
int_ref_qp_delta
int_ref_cycle_dist
Supported in h264_qsv and hevc_qsv. Change these value to reset qsv codec's Intra Refresh
configuration.

qmax
qmin
max_qp_i
min_qp_i
max_qp_p
min_qp_p
max_qp_b
min_qp_b
Supported in h264_qsv. Change these value to reset qsv codec's max/min qp configuration.

low_delay_brc
Supported in h264_qsv, hevc_qsv and av1_qsv. Change this value to reset qsv codec's low_delay_brc
configuration.

framerate
Change this value to reset qsv codec's framerate configuration.

bit_rate
rc_buffer_size
rc_initial_buffer_occupancy
rc_max_rate
Change these value to reset qsv codec's bitrate control configuration.

pic_timing_sei
Supported in h264_qsv and hevc_qsv. Change this value to reset qsv codec's pic_timing_sei
configuration.

H264 options

These options are used by h264_qsv

extbrc
Extended bitrate control.

recovery_point_sei
Set this flag to insert the recovery point SEI message at the beginning of every intra refresh cycle.

rdo Enable rate distortion optimization.

max_frame_size
Maximum encoded frame size in bytes.

max_frame_size_i
Maximum encoded frame size for I frames in bytes. If this value is set as larger than zero, then for
I frames the value set by max_frame_size is ignored.

max_frame_size_p
Maximum encoded frame size for P frames in bytes. If this value is set as larger than zero, then for
P frames the value set by max_frame_size is ignored.

max_slice_size
Maximum encoded slice size in bytes.

bitrate_limit
Toggle bitrate limitations. Modifies bitrate to be in the range imposed by the QSV encoder. Setting
this flag off may lead to violation of HRD conformance. Mind that specifying bitrate below the QSV
encoder range might significantly affect quality. If on this option takes effect in non CQP modes: if
bitrate is not in the range imposed by the QSV encoder, it will be changed to be in the range.

mbbrc
Setting this flag enables macroblock level bitrate control that generally improves subjective visual
quality. Enabling this flag may have negative impact on performance and objective visual quality
metric.

low_delay_brc
Setting this flag turns on or off LowDelayBRC feautre in qsv plugin, which provides more accurate
bitrate control to minimize the variance of bitstream size frame by frame. Value: -1-default 0-off
1-on

adaptive_i
This flag controls insertion of I frames by the QSV encoder. Turn ON this flag to allow changing of
frame type from P and B to I.

adaptive_b
This flag controls changing of frame type from B to P.

p_strategy
Enable P-pyramid: 0-default 1-simple 2-pyramid(bf need to be set to 0).

b_strategy
This option controls usage of B frames as reference.

dblk_idc
This option disable deblocking. It has value in range 0~2.

cavlc
If set, CAVLC is used; if unset, CABAC is used for encoding.

vcm Video conferencing mode, please see ratecontrol method.

idr_interval
Distance (in I-frames) between IDR frames.

pic_timing_sei
Insert picture timing SEI with pic_struct_syntax element.

single_sei_nal_unit
Put all the SEI messages into one NALU.

max_dec_frame_buffering
Maximum number of frames buffered in the DPB.

look_ahead
Use VBR algorithm with look ahead.

look_ahead_depth
Depth of look ahead in number frames.

look_ahead_downsampling
Downscaling factor for the frames saved for the lookahead analysis.

unknown
auto
off
2x
4x
int_ref_type
Specifies intra refresh type. The major goal of intra refresh is improvement of error resilience
without significant impact on encoded bitstream size caused by I frames. The SDK encoder achieves
this by encoding part of each frame in refresh cycle using intra MBs. none means no refresh. vertical
means vertical refresh, by column of MBs. horizontal means horizontal refresh, by rows of MBs. slice
means horizontal refresh by slices without overlapping. In case of slice, in_ref_cycle_size is
ignored. To enable intra refresh, B frame should be set to 0.

int_ref_cycle_size
Specifies number of pictures within refresh cycle starting from 2. 0 and 1 are invalid values.

int_ref_qp_delta
Specifies QP difference for inserted intra MBs. This is signed value in [-51, 51] range if target
encoding bit-depth for luma samples is 8 and this range is [-63, 63] for 10 bit-depth or [-75, 75]
for 12 bit-depth respectively.

int_ref_cycle_dist
Distance between the beginnings of the intra-refresh cycles in frames.

profile
unknown
baseline
main
high
a53cc
Use A53 Closed Captions (if available).

aud Insert the Access Unit Delimiter NAL.

mfmode
Multi-Frame Mode.

off
auto
repeat_pps
Repeat pps for every frame.

max_qp_i
Maximum video quantizer scale for I frame.

min_qp_i
Minimum video quantizer scale for I frame.

max_qp_p
Maximum video quantizer scale for P frame.

min_qp_p
Minimum video quantizer scale for P frame.

max_qp_b
Maximum video quantizer scale for B frame.

min_qp_b
Minimum video quantizer scale for B frame.

scenario
Provides a hint to encoder about the scenario for the encoding session.

unknown
displayremoting
videoconference
archive
livestreaming
cameracapture
videosurveillance
gamestreaming
remotegaming
avbr_accuracy
Accuracy of the AVBR ratecontrol (unit of tenth of percent).

avbr_convergence
Convergence of the AVBR ratecontrol (unit of 100 frames)

The parameters avbr_accuracy and avbr_convergence are for the average variable bitrate control (AVBR)
algorithm. The algorithm focuses on overall encoding quality while meeting the specified bitrate,
target_bitrate, within the accuracy range avbr_accuracy, after a avbr_Convergence period. This method
does not follow HRD and the instant bitrate is not capped or padded.

skip_frame
Use per-frame metadata "qsv_skip_frame" to skip frame when encoding. This option defines the usage of
this metadata.

no_skip
Frame skipping is disabled.

insert_dummy
Encoder inserts into bitstream frame where all macroblocks are encoded as skipped.

insert_nothing
Similar to insert_dummy, but encoder inserts nothing into bitstream. The skipped frames are still
used in brc. For example, gop still include skipped frames, and the frames after skipped frames
will be larger in size.

brc_only
skip_frame metadata indicates the number of missed frames before the current frame.

HEVC Options

These options are used by hevc_qsv

extbrc
Extended bitrate control.

recovery_point_sei
Set this flag to insert the recovery point SEI message at the beginning of every intra refresh cycle.

rdo Enable rate distortion optimization.

max_frame_size
Maximum encoded frame size in bytes.

max_frame_size_i
Maximum encoded frame size for I frames in bytes. If this value is set as larger than zero, then for
I frames the value set by max_frame_size is ignored.

max_frame_size_p
Maximum encoded frame size for P frames in bytes. If this value is set as larger than zero, then for
P frames the value set by max_frame_size is ignored.

max_slice_size
Maximum encoded slice size in bytes.

adaptive_i
This flag controls insertion of I frames by the QSV encoder. Turn ON this flag to allow changing of
frame type from P and B to I.

adaptive_b
This flag controls changing of frame type from B to P.

p_strategy
Enable P-pyramid: 0-default 1-simple 2-pyramid(bf need to be set to 0).

b_strategy
This option controls usage of B frames as reference.

dblk_idc
This option disable deblocking. It has value in range 0~2.

idr_interval
Distance (in I-frames) between IDR frames.

begin_only
Output an IDR-frame only at the beginning of the stream.

load_plugin
A user plugin to load in an internal session.

none
hevc_sw
hevc_hw
load_plugins
A :-separate list of hexadecimal plugin UIDs to load in an internal session.

look_ahead_depth
Depth of look ahead in number frames, available when extbrc option is enabled.

profile
Set the encoding profile (scc requires libmfx >= 1.32).

unknown
main
main10
mainsp
rext
scc
tier
Set the encoding tier (only level >= 4 can support high tier). This option only takes effect when
the level option is specified.

main
high
gpb 1: GPB (generalized P/B frame)

0: regular P frame.

tile_cols
Number of columns for tiled encoding.

tile_rows
Number of rows for tiled encoding.

aud Insert the Access Unit Delimiter NAL.

pic_timing_sei
Insert picture timing SEI with pic_struct_syntax element.

transform_skip
Turn this option ON to enable transformskip. It is supported on platform equal or newer than ICL.

int_ref_type
Specifies intra refresh type. The major goal of intra refresh is improvement of error resilience
without significant impact on encoded bitstream size caused by I frames. The SDK encoder achieves
this by encoding part of each frame in refresh cycle using intra MBs. none means no refresh. vertical
means vertical refresh, by column of MBs. horizontal means horizontal refresh, by rows of MBs. slice
means horizontal refresh by slices without overlapping. In case of slice, in_ref_cycle_size is
ignored. To enable intra refresh, B frame should be set to 0.

int_ref_cycle_size
Specifies number of pictures within refresh cycle starting from 2. 0 and 1 are invalid values.

int_ref_cycle_dist
Distance between the beginnings of the intra-refresh cycles in frames.

max_qp_i
Maximum video quantizer scale for I frame.

min_qp_i
Minimum video quantizer scale for I frame.

max_qp_p
Maximum video quantizer scale for P frame.

min_qp_p
Minimum video quantizer scale for P frame.

max_qp_b
Maximum video quantizer scale for B frame.

min_qp_b
Minimum video quantizer scale for B frame.

scenario
Provides a hint to encoder about the scenario for the encoding session.

unknown
displayremoting
videoconference
archive
livestreaming
cameracapture
videosurveillance
gamestreaming
remotegaming
avbr_accuracy
Accuracy of the AVBR ratecontrol (unit of tenth of percent).

avbr_convergence
Convergence of the AVBR ratecontrol (unit of 100 frames)

skip_frame
Use per-frame metadata "qsv_skip_frame" to skip frame when encoding. This option defines the usage of
this metadata.

no_skip
Frame skipping is disabled.

insert_dummy
Encoder inserts into bitstream frame where all macroblocks are encoded as skipped.

brc_only
skip_frame metadata indicates the number of missed frames before the current frame.

MPEG2 Options

These options are used by mpeg2_qsv

profile
unknown
simple
main
high

VP9 Options

These options are used by vp9_qsv

profile
unknown
profile0
profile1
profile2
profile3
tile_cols
Number of columns for tiled encoding (requires libmfx >= 1.29).

tile_rows
Number of rows for tiled encoding (requires libmfx >= 1.29).

AV1 Options

These options are used by av1_qsv (requires libvpl).

profile
unknown
main
tile_cols
Number of columns for tiled encoding.

tile_rows
Number of rows for tiled encoding.

adaptive_i
This flag controls insertion of I frames by the QSV encoder. Turn ON this flag to allow changing of
frame type from P and B to I.

adaptive_b
This flag controls changing of frame type from B to P.

b_strategy
This option controls usage of B frames as reference.

extbrc
Extended bitrate control.

look_ahead_depth
Depth of look ahead in number frames, available when extbrc option is enabled.

max_frame_size
Set the allowed max size in bytes for each frame. If the frame size exceeds the limitation, encoder
will adjust the QP value to control the frame size. Invalid in CQP rate control mode.

snow
Options

iterative_dia_size
dia size for the iterative motion estimation

VAAPI encoders
Wrappers for hardware encoders accessible via VAAPI.

These encoders only accept input in VAAPI hardware surfaces. If you have input in software frames, use
the hwupload filter to upload them to the GPU.

The following standard libavcodec options are used:

• g / gop_size

• bf / max_b_frames

• profile

If not set, this will be determined automatically from the format of the input frames and the
profiles supported by the driver.

• level

• b / bit_rate

• maxrate / rc_max_rate

• bufsize / rc_buffer_size

• rc_init_occupancy / rc_initial_buffer_occupancy

• compression_level

Speed / quality tradeoff: higher values are faster / worse quality.

• q / global_quality

Size / quality tradeoff: higher values are smaller / worse quality.

• qmin

• qmax

• i_qfactor / i_quant_factor

• i_qoffset / i_quant_offset

• b_qfactor / b_quant_factor

• b_qoffset / b_quant_offset

• slices

All encoders support the following options:

low_power
Some drivers/platforms offer a second encoder for some codecs intended to use less power than the
default encoder; setting this option will attempt to use that encoder. Note that it may support a
reduced feature set, so some other options may not be available in this mode.

idr_interval
Set the number of normal intra frames between full-refresh (IDR) frames in open-GOP mode. The intra
frames are still IRAPs, but will not include global headers and may have non-decodable leading
pictures.

b_depth
Set the B-frame reference depth. When set to one (the default), all B-frames will refer only to P-
or I-frames. When set to greater values multiple layers of B-frames will be present, frames in each
layer only referring to frames in higher layers.

async_depth
Maximum processing parallelism. Increase this to improve single channel performance. This option
doesn't work if driver doesn't implement vaSyncBuffer function. Please make sure there are enough
hw_frames allocated if a large number of async_depth is used.

rc_mode
Set the rate control mode to use. A given driver may only support a subset of modes.

Possible modes:

auto
Choose the mode automatically based on driver support and the other options. This is the
default.

CQP Constant-quality.

CBR Constant-bitrate.

VBR Variable-bitrate.

ICQ Intelligent constant-quality.

QVBR
Quality-defined variable-bitrate.

AVBR
Average variable bitrate.

Each encoder also has its own specific options:

av1_vaapi
profile sets the value of seq_profile. tier sets the value of seq_tier. level sets the value of
seq_level_idx.

tiles
Set the number of tiles to encode the input video with, as columns x rows. (default is auto,
which means use minimal tile column/row number).

tile_groups
Set tile groups number. All the tiles will be distributed as evenly as possible to each tile
group. (default is 1).

h264_vaapi
profile sets the value of profile_idc and the constraint_set*_flags. level sets the value of
level_idc.

coder
Set entropy encoder (default is cabac). Possible values:

ac
cabac
Use CABAC.

vlc
cavlc
Use CAVLC.

aud Include access unit delimiters in the stream (not included by default).

sei Set SEI message types to include. Some combination of the following values:

identifier
Include a user_data_unregistered message containing information about the encoder.

timing
Include picture timing parameters (buffering_period and pic_timing messages).

recovery_point
Include recovery points where appropriate (recovery_point messages).

hevc_vaapi
profile and level set the values of general_profile_idc and general_level_idc respectively.

aud Include access unit delimiters in the stream (not included by default).

tier
Set general_tier_flag. This may affect the level chosen for the stream if it is not explicitly
specified.

sei Set SEI message types to include. Some combination of the following values:

hdr Include HDR metadata if the input frames have it (mastering_display_colour_volume and
content_light_level messages).

tiles
Set the number of tiles to encode the input video with, as columns x rows. Larger numbers allow
greater parallelism in both encoding and decoding, but may decrease coding efficiency.

mjpeg_vaapi
Only baseline DCT encoding is supported. The encoder always uses the standard quantisation and
huffman tables - global_quality scales the standard quantisation table (range 1-100).

For YUV, 4:2:0, 4:2:2 and 4:4:4 subsampling modes are supported. RGB is also supported, and will
create an RGB JPEG.

jfif
Include JFIF header in each frame (not included by default).

huffman
Include standard huffman tables (on by default). Turning this off will save a few hundred bytes
in each output frame, but may lose compatibility with some JPEG decoders which don't fully handle
MJPEG.

mpeg2_vaapi
profile and level set the value of profile_and_level_indication.

vp8_vaapi
B-frames are not supported.

global_quality sets the q_idx used for non-key frames (range 0-127).

loop_filter_level
loop_filter_sharpness
Manually set the loop filter parameters.

vp9_vaapi
global_quality sets the q_idx used for P-frames (range 0-255).

loop_filter_level
loop_filter_sharpness
Manually set the loop filter parameters.

B-frames are supported, but the output stream is always in encode order rather than display order.
If B-frames are enabled, it may be necessary to use the vp9_raw_reorder bitstream filter to modify
the output stream to display frames in the correct order.

Only normal frames are produced - the vp9_superframe bitstream filter may be required to produce a
stream usable with all decoders.

vbn
Vizrt Binary Image encoder.

This format is used by the broadcast vendor Vizrt for quick texture streaming. Advanced features of the
format such as LZW compression of texture data or generation of mipmaps are not supported.

Options

format string
Sets the texture compression used by the VBN file. Can be dxt1, dxt5 or raw. Default is dxt5.

vc2
SMPTE VC-2 (previously BBC Dirac Pro). This codec was primarily aimed at professional broadcasting but
since it supports yuv420, yuv422 and yuv444 at 8 (limited range or full range), 10 or 12 bits, this makes
it suitable for other tasks which require low overhead and low compression (like screen recording).

Options

b Sets target video bitrate. Usually that's around 1:6 of the uncompressed video bitrate (e.g. for
1920x1080 50fps yuv422p10 that's around 400Mbps). Higher values (close to the uncompressed bitrate)
turn on lossless compression mode.

field_order
Enables field coding when set (e.g. to tt - top field first) for interlaced inputs. Should increase
compression with interlaced content as it splits the fields and encodes each separately.

wavelet_depth
Sets the total amount of wavelet transforms to apply, between 1 and 5 (default). Lower values reduce
compression and quality. Less capable decoders may not be able to handle values of wavelet_depth over
3.

wavelet_type
Sets the transform type. Currently only 5_3 (LeGall) and 9_7 (Deslauriers-Dubuc) are implemented,
with 9_7 being the one with better compression and thus is the default.

slice_width
slice_height
Sets the slice size for each slice. Larger values result in better compression. For compatibility
with other more limited decoders use slice_width of 32 and slice_height of 8.

tolerance
Sets the undershoot tolerance of the rate control system in percent. This is to prevent an expensive
search from being run.

qm Sets the quantization matrix preset to use by default or when wavelet_depth is set to 5

- default Uses the default quantization matrix from the specifications, extended with values for
the fifth level. This provides a good balance between keeping detail and omitting artifacts.

- flat Use a completely zeroed out quantization matrix. This increases PSNR but might reduce
perception. Use in bogus benchmarks.

- color Reduces detail but attempts to preserve color at extremely low bitrates.

SUBTITLES ENCODERS

   dvdsub
       This  codec encodes the bitmap subtitle format that is used in DVDs.  Typically they are stored in VOBSUB
       file pairs (*.idx + *.sub), and they can also be used in Matroska files.

       Options

       palette
           Specify the global palette used by the bitmaps.

           The format for this option is a string containing 16 24-bits hexadecimal numbers (without 0x  prefix)
           separated  by  commas,  for  example "0d00ee, ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a,
           7b7b7b, d1d1d1, 7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b".

       even_rows_fix
           When set to 1, enable a work-around that makes the number of pixel rows even in all subtitles.   This
           fixes  a problem with some players that cut off the bottom row if the number is odd.  The work-around
           just adds a fully transparent row if needed.  The overhead is low, typically one byte per subtitle on
           average.

           By default, this work-around is disabled.

BITSTREAM FILTERS

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

You can disable all the bitstream filters using the configure option "--disable-bsfs", and selectively
enable any bitstream filter using the option "--enable-bsf=BSF", or you can disable a particular
bitstream filter using the option "--disable-bsf=BSF".

The option "-bsfs" of the ff* tools will display the list of all the supported bitstream filters included
in your build.

The ff* tools have a -bsf option applied per stream, taking a comma-separated list of filters, whose
parameters follow the filter name after a '='.

ffmpeg -i INPUT -c:v copy -bsf:v filter1[=opt1=str1:opt2=str2][,filter2] OUTPUT

Below is a description of the currently available bitstream filters, with their parameters, if any.

aac_adtstoasc
Convert MPEG-2/4 AAC ADTS to an MPEG-4 Audio Specific Configuration bitstream.

This filter creates an MPEG-4 AudioSpecificConfig from an MPEG-2/4 ADTS header and removes the ADTS
header.

This filter is required for example when copying an AAC stream from a raw ADTS AAC or an MPEG-TS
container to MP4A-LATM, to an FLV file, or to MOV/MP4 files and related formats such as 3GP or M4A.
Please note that it is auto-inserted for MP4A-LATM and MOV/MP4 and related formats.

av1_metadata
Modify metadata embedded in an AV1 stream.

td Insert or remove temporal delimiter OBUs in all temporal units of the stream.

insert
Insert a TD at the beginning of every TU which does not already have one.

remove
Remove the TD from the beginning of every TU which has one.

color_primaries
transfer_characteristics
matrix_coefficients
Set the color description fields in the stream (see AV1 section 6.4.2).

color_range
Set the color range in the stream (see AV1 section 6.4.2; note that this cannot be set for streams
using BT.709 primaries, sRGB transfer characteristic and identity (RGB) matrix coefficients).

tv Limited range.

pc Full range.

chroma_sample_position
Set the chroma sample location in the stream (see AV1 section 6.4.2). This can only be set for 4:2:0
streams.

vertical
Left position (matching the default in MPEG-2 and H.264).

colocated
Top-left position.

tick_rate
Set the tick rate (time_scale / num_units_in_display_tick) in the timing info in the sequence header.

num_ticks_per_picture
Set the number of ticks in each picture, to indicate that the stream has a fixed framerate. Ignored
if tick_rate is not also set.

delete_padding
Deletes Padding OBUs.

chomp
Remove zero padding at the end of a packet.

dca_core
Extract the core from a DCA/DTS stream, dropping extensions such as DTS-HD.

dump_extra
Add extradata to the beginning of the filtered packets except when said packets already exactly begin
with the extradata that is intended to be added.

freq
The additional argument specifies which packets should be filtered. It accepts the values:

k
keyframe
add extradata to all key packets

e
all add extradata to all packets

If not specified it is assumed k.

For example the following ffmpeg command forces a global header (thus disabling individual packet
headers) in the H.264 packets generated by the "libx264" encoder, but corrects them by adding the header
stored in extradata to the key packets:

ffmpeg -i INPUT -map 0 -flags:v +global_header -c:v libx264 -bsf:v dump_extra out.ts

dv_error_marker
Blocks in DV which are marked as damaged are replaced by blocks of the specified color.

color
The color to replace damaged blocks by

sta A 16 bit mask which specifies which of the 16 possible error status values are to be replaced by
colored blocks. 0xFFFE is the default which replaces all non 0 error status values.

ok No error, no concealment

err Error, No concealment

res Reserved

notok
Error or concealment

notres
Not reserved

Aa, Ba, Ca, Ab, Bb, Cb, A, B, C, a, b, erri, erru
The specific error status code

see page 44-46 or section 5.5 of
<http://web.archive.org/web/20060927044735/http://www.smpte.org/smpte_store/standards/pdf/s314m.pdf>

eac3_core
Extract the core from a E-AC-3 stream, dropping extra channels.

extract_extradata
Extract the in-band extradata.

Certain codecs allow the long-term headers (e.g. MPEG-2 sequence headers, or H.264/HEVC (VPS/)SPS/PPS) to
be transmitted either "in-band" (i.e. as a part of the bitstream containing the coded frames) or "out of
band" (e.g. on the container level). This latter form is called "extradata" in FFmpeg terminology.

This bitstream filter detects the in-band headers and makes them available as extradata.

remove
When this option is enabled, the long-term headers are removed from the bitstream after extraction.

filter_units
Remove units with types in or not in a given set from the stream.

pass_types
List of unit types or ranges of unit types to pass through while removing all others. This is
specified as a '|'-separated list of unit type values or ranges of values with '-'.

remove_types
Identical to pass_types, except the units in the given set removed and all others passed through.

Extradata is unchanged by this transformation, but note that if the stream contains inline parameter sets
then the output may be unusable if they are removed.

For example, to remove all non-VCL NAL units from an H.264 stream:

ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=pass_types=1-5' OUTPUT

To remove all AUDs, SEI and filler from an H.265 stream:

ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=remove_types=35|38-40' OUTPUT

hapqa_extract
Extract Rgb or Alpha part of an HAPQA file, without recompression, in order to create an HAPQ or an
HAPAlphaOnly file.

texture
Specifies the texture to keep.

color
alpha

Convert HAPQA to HAPQ

ffmpeg -i hapqa_inputfile.mov -c copy -bsf:v hapqa_extract=texture=color -tag:v HapY -metadata:s:v:0 encoder="HAPQ" hapq_file.mov

Convert HAPQA to HAPAlphaOnly

ffmpeg -i hapqa_inputfile.mov -c copy -bsf:v hapqa_extract=texture=alpha -tag:v HapA -metadata:s:v:0 encoder="HAPAlpha Only" hapalphaonly_file.mov

h264_metadata
Modify metadata embedded in an H.264 stream.

aud Insert or remove AUD NAL units in all access units of the stream.

pass
insert
remove

Default is pass.

sample_aspect_ratio
Set the sample aspect ratio of the stream in the VUI parameters. See H.264 table E-1.

overscan_appropriate_flag
Set whether the stream is suitable for display using overscan or not (see H.264 section E.2.1).

video_format
video_full_range_flag
Set the video format in the stream (see H.264 section E.2.1 and table E-2).

colour_primaries
transfer_characteristics
matrix_coefficients
Set the colour description in the stream (see H.264 section E.2.1 and tables E-3, E-4 and E-5).

chroma_sample_loc_type
Set the chroma sample location in the stream (see H.264 section E.2.1 and figure E-1).

tick_rate
Set the tick rate (time_scale / num_units_in_tick) in the VUI parameters. This is the smallest time
unit representable in the stream, and in many cases represents the field rate of the stream (double
the frame rate).

fixed_frame_rate_flag
Set whether the stream has fixed framerate - typically this indicates that the framerate is exactly
half the tick rate, but the exact meaning is dependent on interlacing and the picture structure (see
H.264 section E.2.1 and table E-6).

zero_new_constraint_set_flags
Zero constraint_set4_flag and constraint_set5_flag in the SPS. These bits were reserved in a previous
version of the H.264 spec, and thus some hardware decoders require these to be zero. The result of
zeroing this is still a valid bitstream.

crop_left
crop_right
crop_top
crop_bottom
Set the frame cropping offsets in the SPS. These values will replace the current ones if the stream
is already cropped.

These fields are set in pixels. Note that some sizes may not be representable if the chroma is
subsampled or the stream is interlaced (see H.264 section 7.4.2.1.1).

sei_user_data
Insert a string as SEI unregistered user data. The argument must be of the form UUID+string, where
the UUID is as hex digits possibly separated by hyphens, and the string can be anything.

For example, 086f3693-b7b3-4f2c-9653-21492feee5b8+hello will insert the string ``hello'' associated
with the given UUID.

delete_filler
Deletes both filler NAL units and filler SEI messages.

display_orientation
Insert, extract or remove Display orientation SEI messages. See H.264 section D.1.27 and D.2.27 for
syntax and semantics.

pass
insert
remove
extract

Default is pass.

Insert mode works in conjunction with "rotate" and "flip" options. Any pre-existing Display
orientation messages will be removed in insert or remove mode. Extract mode attaches the display
matrix to the packet as side data.

rotate
Set rotation in display orientation SEI (anticlockwise angle in degrees). Range is -360 to +360.
Default is NaN.

flip
Set flip in display orientation SEI.

horizontal
vertical

Default is unset.

level
Set the level in the SPS. Refer to H.264 section A.3 and tables A-1 to A-5.

The argument must be the name of a level (for example, 4.2), a level_idc value (for example, 42), or
the special name auto indicating that the filter should attempt to guess the level from the input
stream properties.

h264_mp4toannexb
Convert an H.264 bitstream from length prefixed mode to start code prefixed mode (as defined in the Annex
B of the ITU-T H.264 specification).

This is required by some streaming formats, typically the MPEG-2 transport stream format (muxer
"mpegts").

For example to remux an MP4 file containing an H.264 stream to mpegts format with ffmpeg, you can use the
command:

ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts

Please note that this filter is auto-inserted for MPEG-TS (muxer "mpegts") and raw H.264 (muxer "h264")
output formats.

h264_redundant_pps
This applies a specific fixup to some Blu-ray streams which contain redundant PPSs modifying irrelevant
parameters of the stream which confuse other transformations which require correct extradata.

hevc_metadata
Modify metadata embedded in an HEVC stream.

aud Insert or remove AUD NAL units in all access units of the stream.

insert
remove
sample_aspect_ratio
Set the sample aspect ratio in the stream in the VUI parameters.

video_format
video_full_range_flag
Set the video format in the stream (see H.265 section E.3.1 and table E.2).

colour_primaries
transfer_characteristics
matrix_coefficients
Set the colour description in the stream (see H.265 section E.3.1 and tables E.3, E.4 and E.5).

chroma_sample_loc_type
Set the chroma sample location in the stream (see H.265 section E.3.1 and figure E.1).

tick_rate
Set the tick rate in the VPS and VUI parameters (time_scale / num_units_in_tick). Combined with
num_ticks_poc_diff_one, this can set a constant framerate in the stream. Note that it is likely to
be overridden by container parameters when the stream is in a container.

num_ticks_poc_diff_one
Set poc_proportional_to_timing_flag in VPS and VUI and use this value to set
num_ticks_poc_diff_one_minus1 (see H.265 sections 7.4.3.1 and E.3.1). Ignored if tick_rate is not
also set.

crop_left
crop_right
crop_top
crop_bottom
Set the conformance window cropping offsets in the SPS. These values will replace the current ones
if the stream is already cropped.

These fields are set in pixels. Note that some sizes may not be representable if the chroma is
subsampled (H.265 section 7.4.3.2.1).

level
Set the level in the VPS and SPS. See H.265 section A.4 and tables A.6 and A.7.

The argument must be the name of a level (for example, 5.1), a general_level_idc value (for example,
153 for level 5.1), or the special name auto indicating that the filter should attempt to guess the
level from the input stream properties.

hevc_mp4toannexb
Convert an HEVC/H.265 bitstream from length prefixed mode to start code prefixed mode (as defined in the
Annex B of the ITU-T H.265 specification).

This is required by some streaming formats, typically the MPEG-2 transport stream format (muxer
"mpegts").

For example to remux an MP4 file containing an HEVC stream to mpegts format with ffmpeg, you can use the
command:

ffmpeg -i INPUT.mp4 -codec copy -bsf:v hevc_mp4toannexb OUTPUT.ts

Please note that this filter is auto-inserted for MPEG-TS (muxer "mpegts") and raw HEVC/H.265 (muxer
"h265" or "hevc") output formats.

imxdump
Modifies the bitstream to fit in MOV and to be usable by the Final Cut Pro decoder. This filter only
applies to the mpeg2video codec, and is likely not needed for Final Cut Pro 7 and newer with the
appropriate -tag:v.

For example, to remux 30 MB/sec NTSC IMX to MOV:

ffmpeg -i input.mxf -c copy -bsf:v imxdump -tag:v mx3n output.mov

mjpeg2jpeg
Convert MJPEG/AVI1 packets to full JPEG/JFIF packets.

MJPEG is a video codec wherein each video frame is essentially a JPEG image. The individual frames can be
extracted without loss, e.g. by

ffmpeg -i ../some_mjpeg.avi -c:v copy frames_%d.jpg

Unfortunately, these chunks are incomplete JPEG images, because they lack the DHT segment required for
decoding. Quoting from <http://www.digitalpreservation.gov/formats/fdd/fdd000063.shtml>:

Avery Lee, writing in the rec.video.desktop newsgroup in 2001, commented that "MJPEG, or at least the
MJPEG in AVIs having the MJPG fourcc, is restricted JPEG with a fixed -- and *omitted* -- Huffman table.
The JPEG must be YCbCr colorspace, it must be 4:2:2, and it must use basic Huffman encoding, not
arithmetic or progressive. . . . You can indeed extract the MJPEG frames and decode them with a regular
JPEG decoder, but you have to prepend the DHT segment to them, or else the decoder won't have any idea
how to decompress the data. The exact table necessary is given in the OpenDML spec."

This bitstream filter patches the header of frames extracted from an MJPEG stream (carrying the AVI1
header ID and lacking a DHT segment) to produce fully qualified JPEG images.

ffmpeg -i mjpeg-movie.avi -c:v copy -bsf:v mjpeg2jpeg frame_%d.jpg
exiftran -i -9 frame*.jpg
ffmpeg -i frame_%d.jpg -c:v copy rotated.avi

mjpegadump
Add an MJPEG A header to the bitstream, to enable decoding by Quicktime.

mov2textsub
Extract a representable text file from MOV subtitles, stripping the metadata header from each subtitle
packet.

FORMAT OPTIONS

The libavformat library provides some generic global options, which can be set on all the muxers and
demuxers. In addition each muxer or demuxer 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:

avioflags flags (input/output)
Possible values:

direct
Reduce buffering.

probesize integer (input)
Set probing size in bytes, i.e. the size of the data to analyze to get stream information. A higher
value will enable detecting more information in case it is dispersed into the stream, but will
increase latency. Must be an integer not lesser than 32. It is 5000000 by default.

max_probe_packets integer (input)
Set the maximum number of buffered packets when probing a codec. Default is 2500 packets.

packetsize integer (output)
Set packet size.

fflags flags
Set format flags. Some are implemented for a limited number of formats.

Possible values for input files:

discardcorrupt
Discard corrupted packets.

fastseek
Enable fast, but inaccurate seeks for some formats.

genpts
Generate missing PTS if DTS is present.

igndts
Ignore DTS if PTS is set. Inert when nofillin is set.

ignidx
Ignore index.

nobuffer
Reduce the latency introduced by buffering during initial input streams analysis.

nofillin
Do not fill in missing values in packet fields that can be exactly calculated.

noparse
Disable AVParsers, this needs "+nofillin" too.

sortdts
Try to interleave output packets by DTS. At present, available only for AVIs with an index.

Possible values for output files:

autobsf
Automatically apply bitstream filters as required by the output format. Enabled by default.

bitexact
Only write platform-, build- and time-independent data. This ensures that file and data
checksums are reproducible and match between platforms. Its primary use is for regression
testing.

flush_packets
Write out packets immediately.

shortest
Stop muxing at the end of the shortest stream. It may be needed to increase max_interleave_delta
to avoid flushing the longer streams before EOF.

seek2any integer (input)
Allow seeking to non-keyframes on demuxer level when supported if set to 1. Default is 0.

analyzeduration integer (input)
Specify how many microseconds are analyzed to probe the input. A higher value will enable detecting
more accurate information, but will increase latency. It defaults to 5,000,000 microseconds = 5
seconds.

cryptokey hexadecimal string (input)
Set decryption key.

indexmem integer (input)
Set max memory used for timestamp index (per stream).

rtbufsize integer (input)
Set max memory used for buffering real-time frames.

fdebug flags (input/output)
Print specific debug info.

Possible values:

ts
max_delay integer (input/output)
Set maximum muxing or demuxing delay in microseconds.

fpsprobesize integer (input)
Set number of frames used to probe fps.

audio_preload integer (output)
Set microseconds by which audio packets should be interleaved earlier.

chunk_duration integer (output)
Set microseconds for each chunk.

chunk_size integer (output)
Set size in bytes for each chunk.

err_detect, f_err_detect flags (input)
Set error detection flags. "f_err_detect" is deprecated and should be used only via the ffmpeg tool.

Possible values:

crccheck
Verify embedded CRCs.

bitstream
Detect bitstream specification deviations.

buffer
Detect improper bitstream length.

explode
Abort decoding on minor error detection.

careful
Consider things that violate the spec and have not been seen in the wild as errors.

compliant
Consider all spec non compliancies as errors.

aggressive
Consider things that a sane encoder should not do as an error.

max_interleave_delta integer (output)
Set maximum buffering duration for interleaving. The duration is expressed in microseconds, and
defaults to 10000000 (10 seconds).

To ensure all the streams are interleaved correctly, libavformat will wait until it has at least one
packet for each stream before actually writing any packets to the output file. When some streams are
"sparse" (i.e. there are large gaps between successive packets), this can result in excessive
buffering.

This field specifies the maximum difference between the timestamps of the first and the last packet
in the muxing queue, above which libavformat will output a packet regardless of whether it has queued
a packet for all the streams.

If set to 0, libavformat will continue buffering packets until it has a packet for each stream,
regardless of the maximum timestamp difference between the buffered packets.

use_wallclock_as_timestamps integer (input)
Use wallclock as timestamps if set to 1. Default is 0.

avoid_negative_ts integer (output)
Possible values:

make_non_negative
Shift timestamps to make them non-negative. Also note that this affects only leading negative
timestamps, and not non-monotonic negative timestamps.

make_zero
Shift timestamps so that the first timestamp is 0.

auto (default)
Enables shifting when required by the target format.

disabled
Disables shifting of timestamp.

When shifting is enabled, all output timestamps are shifted by the same amount. Audio, video, and
subtitles desynching and relative timestamp differences are preserved compared to how they would have
been without shifting.

skip_initial_bytes integer (input)
Set number of bytes to skip before reading header and frames if set to 1. Default is 0.

correct_ts_overflow integer (input)
Correct single timestamp overflows if set to 1. Default is 1.

flush_packets integer (output)
Flush the underlying I/O stream after each packet. Default is -1 (auto), which means that the
underlying protocol will decide, 1 enables it, and has the effect of reducing the latency, 0 disables
it and may increase IO throughput in some cases.

output_ts_offset offset (output)
Set the output time offset.

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

The offset is added by the muxer to the output timestamps.

Specifying a positive offset means that the corresponding streams are delayed bt the time duration
specified in offset. Default value is 0 (meaning that no offset is applied).

format_whitelist list (input)
"," separated list of allowed demuxers. By default all are allowed.

dump_separator string (input)
Separator used to separate the fields printed on the command line about the Stream parameters. For
example, to separate the fields with newlines and indentation:

ffprobe -dump_separator "
" -i ~/videos/matrixbench_mpeg2.mpg

max_streams integer (input)
Specifies the maximum number of streams. This can be used to reject files that would require too many
resources due to a large number of streams.

skip_estimate_duration_from_pts bool (input)
Skip estimation of input duration when calculated using PTS. At present, applicable for MPEG-PS and
MPEG-TS.

strict, f_strict integer (input/output)
Specify how strictly to follow the standards. "f_strict" is deprecated and should be used only via
the ffmpeg tool.

Possible values:

very
strictly conform to an older more strict version of the spec or reference software

strict
strictly conform to all the things in the spec no matter what consequences

normal
unofficial
allow unofficial extensions

Format stream specifiers
Format stream specifiers allow selection of one or more streams that match specific properties.

The exact semantics of stream specifiers is defined by the avformat_match_stream_specifier() function
declared in the libavformat/avformat.h header and documented in the Stream specifiers section in the
ffmpeg(1) manual.

DEMUXERS

Demuxers are configured elements in FFmpeg that can read the multimedia streams from a particular type of
file.

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

You can disable all the demuxers using the configure option "--disable-demuxers", and selectively enable
a single demuxer with the option "--enable-demuxer=DEMUXER", or disable it with the option
"--disable-demuxer=DEMUXER".

The option "-demuxers" of the ff* tools will display the list of enabled demuxers. Use "-formats" to view
a combined list of enabled demuxers and muxers.

The description of some of the currently available demuxers follows.

aa
Audible Format 2, 3, and 4 demuxer.

This demuxer is used to demux Audible Format 2, 3, and 4 (.aa) files.

aac
Raw Audio Data Transport Stream AAC demuxer.

This demuxer is used to demux an ADTS input containing a single AAC stream alongwith any ID3v1/2 or APE
tags in it.

apng
Animated Portable Network Graphics demuxer.

This demuxer is used to demux APNG files. All headers, but the PNG signature, up to (but not including)
the first fcTL chunk are transmitted as extradata. Frames are then split as being all the chunks between
two fcTL ones, or between the last fcTL and IEND chunks.

-ignore_loop bool
Ignore the loop variable in the file if set. Default is enabled.

-max_fps int
Maximum framerate in frames per second. Default of 0 imposes no limit.

-default_fps int
Default framerate in frames per second when none is specified in the file (0 meaning as fast as
possible). Default is 15.

asf
Advanced Systems Format demuxer.

This demuxer is used to demux ASF files and MMS network streams.

-no_resync_search bool
Do not try to resynchronize by looking for a certain optional start code.

concat
Virtual concatenation script demuxer.

This demuxer reads a list of files and other directives from a text file and demuxes them one after the
other, as if all their packets had been muxed together.

The timestamps in the files are adjusted so that the first file starts at 0 and each next file starts
where the previous one finishes. Note that it is done globally and may cause gaps if all streams do not
have exactly the same length.

All files must have the same streams (same codecs, same time base, etc.).

The duration of each file is used to adjust the timestamps of the next file: if the duration is incorrect
(because it was computed using the bit-rate or because the file is truncated, for example), it can cause
artifacts. The "duration" directive can be used to override the duration stored in each file.

Syntax

The script is a text file in extended-ASCII, with one directive per line. Empty lines, leading spaces
and lines starting with '#' are ignored. The following directive is recognized:

"file path"
Path to a file to read; special characters and spaces must be escaped with backslash or single
quotes.

All subsequent file-related directives apply to that file.

"ffconcat version 1.0"
Identify the script type and version.

To make FFmpeg recognize the format automatically, this directive must appear exactly as is (no extra
space or byte-order-mark) on the very first line of the script.

"duration dur"
Duration of the file. This information can be specified from the file; specifying it here may be more
efficient or help if the information from the file is not available or accurate.

If the duration is set for all files, then it is possible to seek in the whole concatenated video.

"inpoint timestamp"
In point of the file. When the demuxer opens the file it instantly seeks to the specified timestamp.
Seeking is done so that all streams can be presented successfully at In point.

This directive works best with intra frame codecs, because for non-intra frame ones you will usually
get extra packets before the actual In point and the decoded content will most likely contain frames
before In point too.

For each file, packets before the file In point will have timestamps less than the calculated start
timestamp of the file (negative in case of the first file), and the duration of the files (if not
specified by the "duration" directive) will be reduced based on their specified In point.

Because of potential packets before the specified In point, packet timestamps may overlap between two
concatenated files.

"outpoint timestamp"
Out point of the file. When the demuxer reaches the specified decoding timestamp in any of the
streams, it handles it as an end of file condition and skips the current and all the remaining
packets from all streams.

Out point is exclusive, which means that the demuxer will not output packets with a decoding
timestamp greater or equal to Out point.

This directive works best with intra frame codecs and formats where all streams are tightly
interleaved. For non-intra frame codecs you will usually get additional packets with presentation
timestamp after Out point therefore the decoded content will most likely contain frames after Out
point too. If your streams are not tightly interleaved you may not get all the packets from all
streams before Out point and you may only will be able to decode the earliest stream until Out point.

The duration of the files (if not specified by the "duration" directive) will be reduced based on
their specified Out point.

"file_packet_metadata key=value"
Metadata of the packets of the file. The specified metadata will be set for each file packet. You can
specify this directive multiple times to add multiple metadata entries. This directive is
deprecated, use "file_packet_meta" instead.

"file_packet_meta key value"
Metadata of the packets of the file. The specified metadata will be set for each file packet. You can
specify this directive multiple times to add multiple metadata entries.

"option key value"
Option to access, open and probe the file. Can be present multiple times.

"stream"
Introduce a stream in the virtual file. All subsequent stream-related directives apply to the last
introduced stream. Some streams properties must be set in order to allow identifying the matching
streams in the subfiles. If no streams are defined in the script, the streams from the first file
are copied.

"exact_stream_id id"
Set the id of the stream. If this directive is given, the string with the corresponding id in the
subfiles will be used. This is especially useful for MPEG-PS (VOB) files, where the order of the
streams is not reliable.

"stream_meta key value"
Metadata for the stream. Can be present multiple times.

"stream_codec value"
Codec for the stream.

"stream_extradata hex_string"
Extradata for the string, encoded in hexadecimal.

"chapter id start end"
Add a chapter. id is an unique identifier, possibly small and consecutive.

Options

This demuxer accepts the following option:

safe
If set to 1, reject unsafe file paths and directives. A file path is considered safe if it does not
contain a protocol specification and is relative and all components only contain characters from the
portable character set (letters, digits, period, underscore and hyphen) and have no period at the
beginning of a component.

If set to 0, any file name is accepted.

The default is 1.

auto_convert
If set to 1, try to perform automatic conversions on packet data to make the streams concatenable.
The default is 1.

Currently, the only conversion is adding the h264_mp4toannexb bitstream filter to H.264 streams in
MP4 format. This is necessary in particular if there are resolution changes.

segment_time_metadata
If set to 1, every packet will contain the lavf.concat.start_time and the lavf.concat.duration packet
metadata values which are the start_time and the duration of the respective file segments in the
concatenated output expressed in microseconds. The duration metadata is only set if it is known based
on the concat file. The default is 0.

Examples

• Use absolute filenames and include some comments:

# my first filename
file /mnt/share/file-1.wav
# my second filename including whitespace
file '/mnt/share/file 2.wav'
# my third filename including whitespace plus single quote
file '/mnt/share/file 3'\''.wav'

• Allow for input format auto-probing, use safe filenames and set the duration of the first file:

ffconcat version 1.0

file file-1.wav
duration 20.0

file subdir/file-2.wav

dash
Dynamic Adaptive Streaming over HTTP demuxer.

This demuxer presents all AVStreams found in the manifest. By setting the discard flags on AVStreams the
caller can decide which streams to actually receive. Each stream mirrors the "id" and "bandwidth"
properties from the "<Representation>" as metadata keys named "id" and "variant_bitrate" respectively.

Options

This demuxer accepts the following option:

cenc_decryption_key
16-byte key, in hex, to decrypt files encrypted using ISO Common Encryption (CENC/AES-128 CTR;
ISO/IEC 23001-7).

ea
Electronic Arts Multimedia format demuxer.

This format is used by various Electronic Arts games.

Options

merge_alpha bool
Normally the VP6 alpha channel (if exists) is returned as a secondary video stream, by setting this
option you can make the demuxer return a single video stream which contains the alpha channel in
addition to the ordinary video.

imf
Interoperable Master Format demuxer.

This demuxer presents audio and video streams found in an IMF Composition, as specified in
<https://doi.org/10.5594/SMPTE.ST2067-2.2020>.

ffmpeg [-assetmaps <path of ASSETMAP1>,<path of ASSETMAP2>,...] -i <path of CPL> ...

If "-assetmaps" is not specified, the demuxer looks for a file called ASSETMAP.xml in the same directory
as the CPL.

flv, live_flv, kux
Adobe Flash Video Format demuxer.

This demuxer is used to demux FLV files and RTMP network streams. In case of live network streams, if you
force format, you may use live_flv option instead of flv to survive timestamp discontinuities. KUX is a
flv variant used on the Youku platform.

ffmpeg -f flv -i myfile.flv ...
ffmpeg -f live_flv -i rtmp://<any.server>/anything/key ....

-flv_metadata bool
Allocate the streams according to the onMetaData array content.

-flv_ignore_prevtag bool
Ignore the size of previous tag value.

-flv_full_metadata bool
Output all context of the onMetadata.

gif
Animated GIF demuxer.

It accepts the following options:

min_delay
Set the minimum valid delay between frames in hundredths of seconds. Range is 0 to 6000. Default
value is 2.

max_gif_delay
Set the maximum valid delay between frames in hundredth of seconds. Range is 0 to 65535. Default
value is 65535 (nearly eleven minutes), the maximum value allowed by the specification.

default_delay
Set the default delay between frames in hundredths of seconds. Range is 0 to 6000. Default value is
10.

ignore_loop
GIF files can contain information to loop a certain number of times (or infinitely). If ignore_loop
is set to 1, then the loop setting from the input will be ignored and looping will not occur. If set
to 0, then looping will occur and will cycle the number of times according to the GIF. Default value
is 1.

For example, with the overlay filter, place an infinitely looping GIF over another video:

ffmpeg -i input.mp4 -ignore_loop 0 -i input.gif -filter_complex overlay=shortest=1 out.mkv

Note that in the above example the shortest option for overlay filter is used to end the output video at
the length of the shortest input file, which in this case is input.mp4 as the GIF in this example loops
infinitely.

hls
HLS demuxer

Apple HTTP Live Streaming demuxer.

This demuxer presents all AVStreams from all variant streams. The id field is set to the bitrate variant
index number. By setting the discard flags on AVStreams (by pressing 'a' or 'v' in ffplay), the caller
can decide which variant streams to actually receive. The total bitrate of the variant that the stream
belongs to is available in a metadata key named "variant_bitrate".

It accepts the following options:

live_start_index
segment index to start live streams at (negative values are from the end).

prefer_x_start
prefer to use #EXT-X-START if it's in playlist instead of live_start_index.

allowed_extensions
',' separated list of file extensions that hls is allowed to access.

max_reload
Maximum number of times a insufficient list is attempted to be reloaded. Default value is 1000.

m3u8_hold_counters
The maximum number of times to load m3u8 when it refreshes without new segments. Default value is
1000.

http_persistent
Use persistent HTTP connections. Applicable only for HTTP streams. Enabled by default.

http_multiple
Use multiple HTTP connections for downloading HTTP segments. Enabled by default for HTTP/1.1
servers.

http_seekable
Use HTTP partial requests for downloading HTTP segments. 0 = disable, 1 = enable, -1 = auto, Default
is auto.

seg_format_options
Set options for the demuxer of media segments using a list of key=value pairs separated by ":".

seg_max_retry
Maximum number of times to reload a segment on error, useful when segment skip on network error is
not desired. Default value is 0.

image2
Image file demuxer.

This demuxer reads from a list of image files specified by a pattern. The syntax and meaning of the
pattern is specified by the option pattern_type.

The pattern may contain a suffix which is used to automatically determine the format of the images
contained in the files.

The size, the pixel format, and the format of each image must be the same for all the files in the
sequence.

This demuxer accepts the following options:

framerate
Set the frame rate for the video stream. It defaults to 25.

loop
If set to 1, loop over the input. Default value is 0.

pattern_type
Select the pattern type used to interpret the provided filename.

pattern_type accepts one of the following values.

none
Disable pattern matching, therefore the video will only contain the specified image. You should
use this option if you do not want to create sequences from multiple images and your filenames
may contain special pattern characters.

sequence
Select a sequence pattern type, used to specify a sequence of files indexed by sequential
numbers.

A sequence pattern may contain the string "%d" or "%0Nd", which specifies the position of the
characters representing a sequential number in each filename matched by the pattern. If the form
"%d0Nd" is used, the string representing the number in each filename is 0-padded and N is the
total number of 0-padded digits representing the number. The literal character '%' can be
specified in the pattern with the string "%%".

If the sequence pattern contains "%d" or "%0Nd", the first filename of the file list specified by
the pattern must contain a number inclusively contained between start_number and
start_number+start_number_range-1, and all the following numbers must be sequential.

For example the pattern "img-%03d.bmp" will match a sequence of filenames of the form
img-001.bmp, img-002.bmp, ..., img-010.bmp, etc.; the pattern "i%%m%%g-%d.jpg" will match a
sequence of filenames of the form i%m%g-1.jpg, i%m%g-2.jpg, ..., i%m%g-10.jpg, etc.

Note that the pattern must not necessarily contain "%d" or "%0Nd", for example to convert a
single image file img.jpeg you can employ the command:

ffmpeg -i img.jpeg img.png

glob
Select a glob wildcard pattern type.

The pattern is interpreted like a glob() pattern. This is only selectable if libavformat was
compiled with globbing support.

glob_sequence (deprecated, will be removed)
Select a mixed glob wildcard/sequence pattern.

If your version of libavformat was compiled with globbing support, and the provided pattern
contains at least one glob meta character among "%*?[]{}" that is preceded by an unescaped "%",
the pattern is interpreted like a glob() pattern, otherwise it is interpreted like a sequence
pattern.

All glob special characters "%*?[]{}" must be prefixed with "%". To escape a literal "%" you
shall use "%%".

For example the pattern "foo-%*.jpeg" will match all the filenames prefixed by "foo-" and
terminating with ".jpeg", and "foo-%?%?%?.jpeg" will match all the filenames prefixed with
"foo-", followed by a sequence of three characters, and terminating with ".jpeg".

This pattern type is deprecated in favor of glob and sequence.

Default value is glob_sequence.

pixel_format
Set the pixel format of the images to read. If not specified the pixel format is guessed from the
first image file in the sequence.

start_number
Set the index of the file matched by the image file pattern to start to read from. Default value is
0.

start_number_range
Set the index interval range to check when looking for the first image file in the sequence, starting
from start_number. Default value is 5.

ts_from_file
If set to 1, will set frame timestamp to modification time of image file. Note that monotonity of
timestamps is not provided: images go in the same order as without this option. Default value is 0.
If set to 2, will set frame timestamp to the modification time of the image file in nanosecond
precision.

video_size
Set the video size of the images to read. If not specified the video size is guessed from the first
image file in the sequence.

export_path_metadata
If set to 1, will add two extra fields to the metadata found in input, making them also available for
other filters (see drawtext filter for examples). Default value is 0. The extra fields are described
below:

lavf.image2dec.source_path
Corresponds to the full path to the input file being read.

lavf.image2dec.source_basename
Corresponds to the name of the file being read.

Examples

• Use ffmpeg for creating a video from the images in the file sequence img-001.jpeg, img-002.jpeg, ...,
assuming an input frame rate of 10 frames per second:

ffmpeg -framerate 10 -i 'img-%03d.jpeg' out.mkv

• As above, but start by reading from a file with index 100 in the sequence:

ffmpeg -framerate 10 -start_number 100 -i 'img-%03d.jpeg' out.mkv

• Read images matching the "*.png" glob pattern , that is all the files terminating with the ".png"
suffix:

ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv

libgme
The Game Music Emu library is a collection of video game music file emulators.

See <https://bitbucket.org/mpyne/game-music-emu/overview> for more information.

It accepts the following options:

track_index
Set the index of which track to demux. The demuxer can only export one track. Track indexes start at
0. Default is to pick the first track. Number of tracks is exported as tracks metadata entry.

sample_rate
Set the sampling rate of the exported track. Range is 1000 to 999999. Default is 44100.

max_size (bytes)
The demuxer buffers the entire file into memory. Adjust this value to set the maximum buffer size,
which in turn, acts as a ceiling for the size of files that can be read. Default is 50 MiB.

libmodplug
ModPlug based module demuxer

See <https://github.com/Konstanty/libmodplug>

It will export one 2-channel 16-bit 44.1 kHz audio stream. Optionally, a "pal8" 16-color video stream
can be exported with or without printed metadata.

It accepts the following options:

noise_reduction
Apply a simple low-pass filter. Can be 1 (on) or 0 (off). Default is 0.

reverb_depth
Set amount of reverb. Range 0-100. Default is 0.

reverb_delay
Set delay in ms, clamped to 40-250 ms. Default is 0.

bass_amount
Apply bass expansion a.k.a. XBass or megabass. Range is 0 (quiet) to 100 (loud). Default is 0.

bass_range
Set cutoff i.e. upper-bound for bass frequencies. Range is 10-100 Hz. Default is 0.

surround_depth
Apply a Dolby Pro-Logic surround effect. Range is 0 (quiet) to 100 (heavy). Default is 0.

surround_delay
Set surround delay in ms, clamped to 5-40 ms. Default is 0.

max_size
The demuxer buffers the entire file into memory. Adjust this value to set the maximum buffer size,
which in turn, acts as a ceiling for the size of files that can be read. Range is 0 to 100 MiB. 0
removes buffer size limit (not recommended). Default is 5 MiB.

video_stream_expr
String which is evaluated using the eval API to assign colors to the generated video stream.
Variables which can be used are "x", "y", "w", "h", "t", "speed", "tempo", "order", "pattern" and
"row".

video_stream
Generate video stream. Can be 1 (on) or 0 (off). Default is 0.

video_stream_w
Set video frame width in 'chars' where one char indicates 8 pixels. Range is 20-512. Default is 30.

video_stream_h
Set video frame height in 'chars' where one char indicates 8 pixels. Range is 20-512. Default is 30.

video_stream_ptxt
Print metadata on video stream. Includes "speed", "tempo", "order", "pattern", "row" and "ts" (time
in ms). Can be 1 (on) or 0 (off). Default is 1.

libopenmpt
libopenmpt based module demuxer

See <https://lib.openmpt.org/libopenmpt/> for more information.

Some files have multiple subsongs (tracks) this can be set with the subsong option.

It accepts the following options:

subsong
Set the subsong index. This can be either 'all', 'auto', or the index of the subsong. Subsong
indexes start at 0. The default is 'auto'.

The default value is to let libopenmpt choose.

layout
Set the channel layout. Valid values are 1, 2, and 4 channel layouts. The default value is STEREO.

sample_rate
Set the sample rate for libopenmpt to output. Range is from 1000 to INT_MAX. The value default is
48000.

mov/mp4/3gp
Demuxer for Quicktime File Format & ISO/IEC Base Media File Format (ISO/IEC 14496-12 or MPEG-4 Part 12,
ISO/IEC 15444-12 or JPEG 2000 Part 12).

Registered extensions: mov, mp4, m4a, 3gp, 3g2, mj2, psp, m4b, ism, ismv, isma, f4v

Options

This demuxer accepts the following options:

enable_drefs
Enable loading of external tracks, disabled by default. Enabling this can theoretically leak
information in some use cases.

use_absolute_path
Allows loading of external tracks via absolute paths, disabled by default. Enabling this poses a
security risk. It should only be enabled if the source is known to be non-malicious.

seek_streams_individually
When seeking, identify the closest point in each stream individually and demux packets in that stream
from identified point. This can lead to a different sequence of packets compared to demuxing linearly
from the beginning. Default is true.

ignore_editlist
Ignore any edit list atoms. The demuxer, by default, modifies the stream index to reflect the
timeline described by the edit list. Default is false.

advanced_editlist
Modify the stream index to reflect the timeline described by the edit list. "ignore_editlist" must be
set to false for this option to be effective. If both "ignore_editlist" and this option are set to
false, then only the start of the stream index is modified to reflect initial dwell time or starting
timestamp described by the edit list. Default is true.

ignore_chapters
Don't parse chapters. This includes GoPro 'HiLight' tags/moments. Note that chapters are only parsed
when input is seekable. Default is false.

use_mfra_for
For seekable fragmented input, set fragment's starting timestamp from media fragment random access
box, if present.

Following options are available:

auto
Auto-detect whether to set mfra timestamps as PTS or DTS (default)

dts Set mfra timestamps as DTS

pts Set mfra timestamps as PTS

0 Don't use mfra box to set timestamps

use_tfdt
For fragmented input, set fragment's starting timestamp to "baseMediaDecodeTime" from the "tfdt" box.
Default is enabled, which will prefer to use the "tfdt" box to set DTS. Disable to use the
"earliest_presentation_time" from the "sidx" box. In either case, the timestamp from the "mfra" box
will be used if it's available and "use_mfra_for" is set to pts or dts.

export_all
Export unrecognized boxes within the udta box as metadata entries. The first four characters of the
box type are set as the key. Default is false.

export_xmp
Export entire contents of XMP_ box and uuid box as a string with key "xmp". Note that if "export_all"
is set and this option isn't, the contents of XMP_ box are still exported but with key "XMP_".
Default is false.

activation_bytes
4-byte key required to decrypt Audible AAX and AAX+ files. See Audible AAX subsection below.

audible_fixed_key
Fixed key used for handling Audible AAX/AAX+ files. It has been pre-set so should not be necessary to
specify.

decryption_key
16-byte key, in hex, to decrypt files encrypted using ISO Common Encryption (CENC/AES-128 CTR;
ISO/IEC 23001-7).

max_stts_delta
Very high sample deltas written in a trak's stts box may occasionally be intended but usually they
are written in error or used to store a negative value for dts correction when treated as signed
32-bit integers. This option lets the user set an upper limit, beyond which the delta is clamped to
1. Values greater than the limit if negative when cast to int32 are used to adjust onward dts.

Unit is the track time scale. Range is 0 to UINT_MAX. Default is "UINT_MAX - 48000*10" which allows
upto a 10 second dts correction for 48 kHz audio streams while accommodating 99.9% of "uint32" range.

interleaved_read
Interleave packets from multiple tracks at demuxer level. For badly interleaved files, this prevents
playback issues caused by large gaps between packets in different tracks, as MOV/MP4 do not have
packet placement requirements. However, this can cause excessive seeking on very badly interleaved
files, due to seeking between tracks, so disabling it may prevent I/O issues, at the expense of
playback.

Audible AAX

Audible AAX files are encrypted M4B files, and they can be decrypted by specifying a 4 byte activation
secret.

ffmpeg -activation_bytes 1CEB00DA -i test.aax -vn -c:a copy output.mp4

mpegts
MPEG-2 transport stream demuxer.

This demuxer accepts the following options:

resync_size
Set size limit for looking up a new synchronization. Default value is 65536.

skip_unknown_pmt
Skip PMTs for programs not defined in the PAT. Default value is 0.

fix_teletext_pts
Override teletext packet PTS and DTS values with the timestamps calculated from the PCR of the first
program which the teletext stream is part of and is not discarded. Default value is 1, set this
option to 0 if you want your teletext packet PTS and DTS values untouched.

ts_packetsize
Output option carrying the raw packet size in bytes. Show the detected raw packet size, cannot be
set by the user.

scan_all_pmts
Scan and combine all PMTs. The value is an integer with value from -1 to 1 (-1 means automatic
setting, 1 means enabled, 0 means disabled). Default value is -1.

merge_pmt_versions
Re-use existing streams when a PMT's version is updated and elementary streams move to different
PIDs. Default value is 0.

max_packet_size
Set maximum size, in bytes, of packet emitted by the demuxer. Payloads above this size are split
across multiple packets. Range is 1 to INT_MAX/2. Default is 204800 bytes.

mpjpeg
MJPEG encapsulated in multi-part MIME demuxer.

This demuxer allows reading of MJPEG, where each frame is represented as a part of
multipart/x-mixed-replace stream.

strict_mime_boundary
Default implementation applies a relaxed standard to multi-part MIME boundary detection, to prevent
regression with numerous existing endpoints not generating a proper MIME MJPEG stream. Turning this
option on by setting it to 1 will result in a stricter check of the boundary value.

rawvideo
Raw video demuxer.

This demuxer allows one to read raw video data. Since there is no header specifying the assumed video
parameters, the user must specify them in order to be able to decode the data correctly.

This demuxer accepts the following options:

framerate
Set input video frame rate. Default value is 25.

pixel_format
Set the input video pixel format. Default value is "yuv420p".

video_size
Set the input video size. This value must be specified explicitly.

For example to read a rawvideo file input.raw with ffplay, assuming a pixel format of "rgb24", a video
size of "320x240", and a frame rate of 10 images per second, use the command:

ffplay -f rawvideo -pixel_format rgb24 -video_size 320x240 -framerate 10 input.raw

sbg
SBaGen script demuxer.

This demuxer reads the script language used by SBaGen <http://uazu.net/sbagen/> to generate binaural
beats sessions. A SBG script looks like that:

-SE
a: 300-2.5/3 440+4.5/0
b: 300-2.5/0 440+4.5/3
off: -
NOW == a
+0:07:00 == b
+0:14:00 == a
+0:21:00 == b
+0:30:00 off

A SBG script can mix absolute and relative timestamps. If the script uses either only absolute timestamps
(including the script start time) or only relative ones, then its layout is fixed, and the conversion is
straightforward. On the other hand, if the script mixes both kind of timestamps, then the NOW reference
for relative timestamps will be taken from the current time of day at the time the script is read, and
the script layout will be frozen according to that reference. That means that if the script is directly
played, the actual times will match the absolute timestamps up to the sound controller's clock accuracy,
but if the user somehow pauses the playback or seeks, all times will be shifted accordingly.

tedcaptions
JSON captions used for <http://www.ted.com/>.

TED does not provide links to the captions, but they can be guessed from the page. The file
tools/bookmarklets.html from the FFmpeg source tree contains a bookmarklet to expose them.

This demuxer accepts the following option:

start_time
Set the start time of the TED talk, in milliseconds. The default is 15000 (15s). It is used to sync
the captions with the downloadable videos, because they include a 15s intro.

Example: convert the captions to a format most players understand:

ffmpeg -i http://www.ted.com/talks/subtitles/id/1/lang/en talk1-en.srt

vapoursynth
Vapoursynth wrapper.

Due to security concerns, Vapoursynth scripts will not be autodetected so the input format has to be
forced. For ff* CLI tools, add "-f vapoursynth" before the input "-i yourscript.vpy".

This demuxer accepts the following option:

max_script_size
The demuxer buffers the entire script into memory. Adjust this value to set the maximum buffer size,
which in turn, acts as a ceiling for the size of scripts that can be read. Default is 1 MiB.

MUXERS

Muxers are configured elements in FFmpeg which allow writing multimedia streams to a particular type of
file.

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

You can disable all the muxers with the configure option "--disable-muxers" and selectively enable /
disable single muxers with the options "--enable-muxer=MUXER" / "--disable-muxer=MUXER".

The option "-muxers" of the ff* tools will display the list of enabled muxers. Use "-formats" to view a
combined list of enabled demuxers and muxers.

A description of some of the currently available muxers follows.

a64
A64 muxer for Commodore 64 video. Accepts a single "a64_multi" or "a64_multi5" codec video stream.

adts
Audio Data Transport Stream muxer. It accepts a single AAC stream.

Options

It accepts the following options:

write_id3v2 bool
Enable to write ID3v2.4 tags at the start of the stream. Default is disabled.

write_apetag bool
Enable to write APE tags at the end of the stream. Default is disabled.

write_mpeg2 bool
Enable to set MPEG version bit in the ADTS frame header to 1 which indicates MPEG-2. Default is 0,
which indicates MPEG-4.

aiff
Audio Interchange File Format muxer.

Options

It accepts the following options:

write_id3v2
Enable ID3v2 tags writing when set to 1. Default is 0 (disabled).

id3v2_version
Select ID3v2 version to write. Currently only version 3 and 4 (aka. ID3v2.3 and ID3v2.4) are
supported. The default is version 4.

alp
Muxer for audio of High Voltage Software's Lego Racers game. It accepts a single ADPCM_IMA_ALP stream
with no more than 2 channels nor a sample rate greater than 44100 Hz.

Extensions: tun, pcm

Options

It accepts the following options:

type type
Set file type.

tun Set file type as music. Must have a sample rate of 22050 Hz.

pcm Set file type as sfx.

auto
Set file type as per output file extension. ".pcm" results in type "pcm" else type "tun" is set.
(default)

asf
Advanced Systems Format muxer.

Note that Windows Media Audio (wma) and Windows Media Video (wmv) use this muxer too.

Options

It accepts the following options:

packet_size
Set the muxer packet size. By tuning this setting you may reduce data fragmentation or muxer overhead
depending on your source. Default value is 3200, minimum is 100, maximum is 64k.

avi
Audio Video Interleaved muxer.

Options

It accepts the following options:

reserve_index_space
Reserve the specified amount of bytes for the OpenDML master index of each stream within the file
header. By default additional master indexes are embedded within the data packets if there is no
space left in the first master index and are linked together as a chain of indexes. This index
structure can cause problems for some use cases, e.g. third-party software strictly relying on the
OpenDML index specification or when file seeking is slow. Reserving enough index space in the file
header avoids these problems.

The required index space depends on the output file size and should be about 16 bytes per gigabyte.
When this option is omitted or set to zero the necessary index space is guessed.

write_channel_mask
Write the channel layout mask into the audio stream header.

This option is enabled by default. Disabling the channel mask can be useful in specific scenarios,
e.g. when merging multiple audio streams into one for compatibility with software that only supports
a single audio stream in AVI (see the "amerge" section in the ffmpeg-filters manual).

flipped_raw_rgb
If set to true, store positive height for raw RGB bitmaps, which indicates bitmap is stored bottom-
up. Note that this option does not flip the bitmap which has to be done manually beforehand, e.g. by
using the vflip filter. Default is false and indicates bitmap is stored top down.

chromaprint
Chromaprint fingerprinter.

This muxer feeds audio data to the Chromaprint library, which generates a fingerprint for the provided
audio data. See <https://acoustid.org/chromaprint>

It takes a single signed native-endian 16-bit raw audio stream of at most 2 channels.

Options

silence_threshold
Threshold for detecting silence. Range is from -1 to 32767, where -1 disables silence detection.
Silence detection can only be used with version 3 of the algorithm. Silence detection must be
disabled for use with the AcoustID service. Default is -1.

algorithm
Version of algorithm to fingerprint with. Range is 0 to 4. Version 3 enables silence detection.
Default is 1.

fp_format
Format to output the fingerprint as. Accepts the following options:

raw Binary raw fingerprint

compressed
Binary compressed fingerprint

base64
Base64 compressed fingerprint (default)

crc
CRC (Cyclic Redundancy Check) testing format.

This muxer computes and prints the Adler-32 CRC of all the input audio and video frames. By default audio
frames are converted to signed 16-bit raw audio and video frames to raw video before computing the CRC.

The output of the muxer consists of a single line of the form: CRC=0xCRC, where CRC is a hexadecimal
number 0-padded to 8 digits containing the CRC for all the decoded input frames.

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@0.5

The previous example can be specified as:

drawbox=x=10:y=20:w=200:h=60:color=red@0.5

• Fill the box with pink color:

drawbox=x=10:y=10:w=100:h=100:color=pink@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@0.5

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

drawgrid=w=iw/3:h=ih/3:t=2:c=white@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@0.2: box=1: boxcolor=red@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]].

OPENCL VIDEO FILTERS

Below is a description of the currently available OpenCL video filters.

To enable compilation of these filters you need to configure FFmpeg with "--enable-opencl".

Running OpenCL filters requires you to initialize a hardware device and to pass that device to all
filters in any filter graph.

-init_hw_device opencl[=name][:device[,key=value...]]
Initialise a new hardware device of type opencl called name, using the given device parameters.

-filter_hw_device name
Pass the hardware device called name to all filters in any filter graph.

For more detailed information see <https://www.ffmpeg.org/ffmpeg.html#Advanced-Video-options>

• Example of choosing the first device on the second platform and running avgblur_opencl filter with
default parameters on it.

-init_hw_device opencl=gpu:1.0 -filter_hw_device gpu -i INPUT -vf "hwupload, avgblur_opencl, hwdownload" OUTPUT

Since OpenCL filters are not able to access frame data in normal memory, all frame data needs to be
uploaded(hwupload) to hardware surfaces connected to the appropriate device before being used and then
downloaded(hwdownload) back to normal memory. Note that hwupload will upload to a surface with the same
layout as the software frame, so it may be necessary to add a format filter immediately before to get the
input into the right format and hwdownload does not support all formats 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.

avgblur_opencl
Apply average blur filter.

The filter accepts the following options:

sizeX
Set horizontal radius size. Range is "[1, 1024]" and default value is 1.

planes
Set which planes to filter. Default value is 0xf, by which all planes are processed.

sizeY
Set vertical radius size. Range is "[1, 1024]" and default value is 0. If zero, "sizeX" value will be
used.

Example

• Apply average blur filter with horizontal and vertical size of 3, setting each pixel of the output to
the average value of the 7x7 region centered on it in the input. For pixels on the edges of the
image, the region does not extend beyond the image boundaries, and so out-of-range coordinates are
not used in the calculations.

-i INPUT -vf "hwupload, avgblur_opencl=3, hwdownload" OUTPUT

boxblur_opencl
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 boxblur filter, setting each pixel of the output to the average value of box-radiuses luma_radius,
chroma_radius, alpha_radius for each plane respectively. The filter will apply luma_power, chroma_power,
alpha_power times onto the corresponding plane. For pixels on the edges of the image, the radius does not
extend beyond the image boundaries, and so out-of-range coordinates are not used in the calculations.

• Apply a boxblur filter with the luma, chroma, and alpha radius set to 2 and luma, chroma, and alpha
power set to 3. The filter will run 3 times with box-radius set to 2 for every plane of the image.

-i INPUT -vf "hwupload, boxblur_opencl=luma_radius=2:luma_power=3, hwdownload" OUTPUT
-i INPUT -vf "hwupload, boxblur_opencl=2:3, hwdownload" OUTPUT

• Apply a boxblur filter with luma radius set to 2, luma_power to 1, chroma_radius to 4, chroma_power
to 5, alpha_radius to 3 and alpha_power to 7.

For the luma plane, a 2x2 box radius will be run once.

For the chroma plane, a 4x4 box radius will be run 5 times.

For the alpha plane, a 3x3 box radius will be run 7 times.

-i INPUT -vf "hwupload, boxblur_opencl=2:1:4:5:3:7, hwdownload" OUTPUT

colorkey_opencl
RGB colorspace color keying.

The filter accepts the following options:

color
The color which will be replaced with transparency.

similarity
Similarity percentage with the key color.

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

blend
Blend percentage.

0.0 makes pixels either fully transparent, or not transparent at all.

Higher values result in semi-transparent pixels, with a higher transparency the more similar the
pixels color is to the key color.

Examples

• Make every semi-green pixel in the input transparent with some slight blending:

-i INPUT -vf "hwupload, colorkey_opencl=green:0.3:0.1, hwdownload" OUTPUT

convolution_opencl
Apply convolution of 3x3, 5x5, 7x7 matrix.

The filter accepts the following options:

0m
1m
2m
3m Set matrix for each plane. Matrix is sequence of 9, 25 or 49 signed numbers. Default value for each
plane is "0 0 0 0 1 0 0 0 0".

0rdiv
1rdiv
2rdiv
3rdiv
Set multiplier for calculated value for each plane. If unset or 0, it will be sum of all matrix
elements. The option value must be a float number greater or equal to 0.0. Default value is 1.0.

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. The option value must be a float number greater or equal to
0.0. Default value is 0.0.

Examples

• Apply sharpen:

-i INPUT -vf "hwupload, convolution_opencl=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, hwdownload" OUTPUT

• Apply blur:

-i INPUT -vf "hwupload, convolution_opencl=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, hwdownload" OUTPUT

• Apply edge enhance:

-i INPUT -vf "hwupload, convolution_opencl=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, hwdownload" OUTPUT

• Apply edge detect:

-i INPUT -vf "hwupload, convolution_opencl=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, hwdownload" OUTPUT

• Apply laplacian edge detector which includes diagonals:

-i INPUT -vf "hwupload, convolution_opencl=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, hwdownload" OUTPUT

• Apply emboss:

-i INPUT -vf "hwupload, convolution_opencl=-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, hwdownload" OUTPUT

erosion_opencl
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. Range is "[0, 65535]" and default value is 65535. If 0,
plane will remain unchanged.

coordinates
Flag which specifies the pixel to refer to. Range is "[0, 255]" and default value is 255, i.e. all
eight pixels are used.

Flags to local 3x3 coordinates region centered on "x":

1 2 3

4 x 5

6 7 8

Example

• Apply erosion filter with threshold0 set to 30, threshold1 set 40, threshold2 set to 50 and
coordinates set to 231, setting each pixel of the output to the local minimum between pixels: 1, 2,
3, 6, 7, 8 of the 3x3 region centered on it in the input. If the difference between input pixel and
local minimum is more then threshold of the corresponding plane, output pixel will be set to input
pixel - threshold of corresponding plane.

-i INPUT -vf "hwupload, erosion_opencl=30:40:50:coordinates=231, hwdownload" OUTPUT

deshake_opencl
Feature-point based video stabilization filter.

The filter accepts the following options:

tripod
Simulates a tripod by preventing any camera movement whatsoever from the original frame. Defaults to
0.

debug
Whether or not additional debug info should be displayed, both in the processed output and in the
console.

Note that in order to see console debug output you will also need to pass "-v verbose" to ffmpeg.

Viewing point matches in the output video is only supported for RGB input.

Defaults to 0.

adaptive_crop
Whether or not to do a tiny bit of cropping at the borders to cut down on the amount of mirrored
pixels.

Defaults to 1.

refine_features
Whether or not feature points should be refined at a sub-pixel level.

This can be turned off for a slight performance gain at the cost of precision.

Defaults to 1.

smooth_strength
The strength of the smoothing applied to the camera path from 0.0 to 1.0.

1.0 is the maximum smoothing strength while values less than that result in less smoothing.

0.0 causes the filter to adaptively choose a smoothing strength on a per-frame basis.

Defaults to 0.0.

smooth_window_multiplier
Controls the size of the smoothing window (the number of frames buffered to determine motion
information from).

The size of the smoothing window is determined by multiplying the framerate of the video by this
number.

Acceptable values range from 0.1 to 10.0.

Larger values increase the amount of motion data available for determining how to smooth the camera
path, potentially improving smoothness, but also increase latency and memory usage.

Defaults to 2.0.

Examples

• Stabilize a video with a fixed, medium smoothing strength:

-i INPUT -vf "hwupload, deshake_opencl=smooth_strength=0.5, hwdownload" OUTPUT

• Stabilize a video with debugging (both in console and in rendered video):

-i INPUT -filter_complex "[0:v]format=rgba, hwupload, deshake_opencl=debug=1, hwdownload, format=rgba, format=yuv420p" -v verbose OUTPUT

dilation_opencl
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. Range is "[0, 65535]" and default value is 65535. If 0,
plane will remain unchanged.

coordinates
Flag which specifies the pixel to refer to. Range is "[0, 255]" and default value is 255, i.e. all
eight pixels are used.

Flags to local 3x3 coordinates region centered on "x":

1 2 3

4 x 5

6 7 8

Example

• Apply dilation filter with threshold0 set to 30, threshold1 set 40, threshold2 set to 50 and
coordinates set to 231, setting each pixel of the output to the local maximum between pixels: 1, 2,
3, 6, 7, 8 of the 3x3 region centered on it in the input. If the difference between input pixel and
local maximum is more then threshold of the corresponding plane, output pixel will be set to input
pixel + threshold of corresponding plane.

-i INPUT -vf "hwupload, dilation_opencl=30:40:50:coordinates=231, hwdownload" OUTPUT

nlmeans_opencl
Non-local Means denoise filter through OpenCL, this filter accepts same options as nlmeans.

overlay_opencl
Overlay one video on top of another.

It takes two inputs and has one output. The first input is the "main" video on which the second input is
overlaid. This filter requires same memory layout for all the inputs. So, format conversion may be
needed.

The filter accepts the following options:

x Set the x coordinate of the overlaid video on the main video. Default value is 0.

y Set the y coordinate of the overlaid video on the main video. Default value is 0.

Examples

• Overlay an image LOGO at the top-left corner of the INPUT video. Both inputs are yuv420p format.

-i INPUT -i LOGO -filter_complex "[0:v]hwupload[a], [1:v]format=yuv420p, hwupload[b], [a][b]overlay_opencl, hwdownload" OUTPUT

• The inputs have same memory layout for color channels , the overlay has additional alpha plane, like
INPUT is yuv420p, and the LOGO is yuva420p.

-i INPUT -i LOGO -filter_complex "[0:v]hwupload[a], [1:v]format=yuva420p, hwupload[b], [a][b]overlay_opencl, hwdownload" OUTPUT

pad_opencl
Add paddings to the input image, and place the original input at the provided x, y coordinates.

It accepts the following options:

width, w
height, h
Specify an expression for the size of the output image with the paddings added. If the value for
width or height is 0, the corresponding input size is used for the output.

The width expression can reference the value set by the height expression, and vice versa.

The default value of width and height is 0.

x
y Specify the offsets to place the input image at within the padded area, with respect to the top/left
border of the output image.

The x expression can reference the value set by the y expression, and vice versa.

The default value of x and y is 0.

If x or y evaluate to a negative number, they'll be changed so the input image is centered on the
padded area.

color
Specify the color of the padded area. For the syntax of this option, check the "Color" section in the
ffmpeg-utils manual.

aspect
Pad to an aspect instead to a resolution.

The value for the width, height, x, and y options are expressions containing the following constants:

in_w
in_h
The input video width and height.

iw
ih These are the same as in_w and in_h.

out_w
out_h
The output width and height (the size of the padded area), as specified by the width and height
expressions.

ow
oh These are the same as out_w and out_h.

x
y The x and y offsets as specified by the x and y expressions, or NAN if not yet specified.

a same as iw / ih

sar input sample aspect ratio

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

prewitt_opencl
Apply the Prewitt operator (<https://en.wikipedia.org/wiki/Prewitt_operator>) to input video stream.

The filter accepts the following option:

planes
Set which planes to filter. Default value is 0xf, by which all planes are processed.

scale
Set value which will be multiplied with filtered result. Range is "[0.0, 65535]" and default value
is 1.0.

delta
Set value which will be added to filtered result. Range is "[-65535, 65535]" and default value is
0.0.

Example

• Apply the Prewitt operator with scale set to 2 and delta set to 10.

-i INPUT -vf "hwupload, prewitt_opencl=scale=2:delta=10, hwdownload" OUTPUT

program_opencl
Filter video using an OpenCL program.

source
OpenCL program source file.

kernel
Kernel name in program.

inputs
Number of inputs to the filter. Defaults to 1.

size, s
Size of output frames. Defaults to the same as the first input.

The "program_opencl" filter also supports the framesync options.

The program source file must contain a kernel function with the given name, which will be run once for
each plane of the output. Each run on a plane gets enqueued as a separate 2D global NDRange with one
work-item for each pixel to be generated. The global ID offset for each work-item is therefore the
coordinates of a pixel in the destination image.

The kernel function needs to take the following arguments:

• Destination image, __write_only image2d_t.

This image will become the output; the kernel should write all of it.

• Frame index, unsigned int.

This is a counter starting from zero and increasing by one for each frame.

• Source images, __read_only image2d_t.

These are the most recent images on each input. The kernel may read from them to generate the
output, but they can't be written to.

Example programs:

• Copy the input to the output (output must be the same size as the input).

__kernel void copy(__write_only image2d_t destination,
unsigned int index,
__read_only image2d_t source)
{
const sampler_t sampler = CLK_NORMALIZED_COORDS_FALSE;

int2 location = (int2)(get_global_id(0), get_global_id(1));

float4 value = read_imagef(source, sampler, location);

write_imagef(destination, location, value);
}

• Apply a simple transformation, rotating the input by an amount increasing with the index counter.
Pixel values are linearly interpolated by the sampler, and the output need not have the same
dimensions as the input.

__kernel void rotate_image(__write_only image2d_t dst,
unsigned int index,
__read_only image2d_t src)
{
const sampler_t sampler = (CLK_NORMALIZED_COORDS_FALSE |
CLK_FILTER_LINEAR);

float angle = (float)index / 100.0f;

float2 dst_dim = convert_float2(get_image_dim(dst));
float2 src_dim = convert_float2(get_image_dim(src));

float2 dst_cen = dst_dim / 2.0f;
float2 src_cen = src_dim / 2.0f;

int2 dst_loc = (int2)(get_global_id(0), get_global_id(1));

float2 dst_pos = convert_float2(dst_loc) - dst_cen;
float2 src_pos = {
cos(angle) * dst_pos.x - sin(angle) * dst_pos.y,
sin(angle) * dst_pos.x + cos(angle) * dst_pos.y
};
src_pos = src_pos * src_dim / dst_dim;

float2 src_loc = src_pos + src_cen;

if (src_loc.x < 0.0f || src_loc.y < 0.0f ||
src_loc.x > src_dim.x || src_loc.y > src_dim.y)
write_imagef(dst, dst_loc, 0.5f);
else
write_imagef(dst, dst_loc, read_imagef(src, sampler, src_loc));
}

• Blend two inputs together, with the amount of each input used varying with the index counter.

__kernel void blend_images(__write_only image2d_t dst,
unsigned int index,
__read_only image2d_t src1,
__read_only image2d_t src2)
{
const sampler_t sampler = (CLK_NORMALIZED_COORDS_FALSE |
CLK_FILTER_LINEAR);

float blend = (cos((float)index / 50.0f) + 1.0f) / 2.0f;

int2 dst_loc = (int2)(get_global_id(0), get_global_id(1));
int2 src1_loc = dst_loc * get_image_dim(src1) / get_image_dim(dst);
int2 src2_loc = dst_loc * get_image_dim(src2) / get_image_dim(dst);

float4 val1 = read_imagef(src1, sampler, src1_loc);
float4 val2 = read_imagef(src2, sampler, src2_loc);

write_imagef(dst, dst_loc, val1 * blend + val2 * (1.0f - blend));
}

remap_opencl
Remap pixels using 2nd: Xmap and 3rd: Ymap input video stream.

Xmap and Ymap input video streams must be of same dimensions. Output video stream will have Xmap/Ymap
video stream dimensions. Xmap and Ymap input video streams are 32bit float pixel format, single channel.

interp
Specify interpolation used for remapping of pixels. Allowed values are "near" and "linear". Default
value is "linear".

fill
Specify the color of the unmapped pixels. For the syntax of this option, check the "Color" section in
the ffmpeg-utils manual. Default color is "black".

roberts_opencl
Apply the Roberts cross operator (<https://en.wikipedia.org/wiki/Roberts_cross>) to input video stream.

The filter accepts the following option:

planes
Set which planes to filter. Default value is 0xf, by which all planes are processed.

scale
Set value which will be multiplied with filtered result. Range is "[0.0, 65535]" and default value
is 1.0.

delta
Set value which will be added to filtered result. Range is "[-65535, 65535]" and default value is
0.0.

Example

• Apply the Roberts cross operator with scale set to 2 and delta set to 10

-i INPUT -vf "hwupload, roberts_opencl=scale=2:delta=10, hwdownload" OUTPUT

sobel_opencl
Apply the Sobel operator (<https://en.wikipedia.org/wiki/Sobel_operator>) to input video stream.

The filter accepts the following option:

planes
Set which planes to filter. Default value is 0xf, by which all planes are processed.

scale
Set value which will be multiplied with filtered result. Range is "[0.0, 65535]" and default value
is 1.0.

delta
Set value which will be added to filtered result. Range is "[-65535, 65535]" and default value is
0.0.

Example

• Apply sobel operator with scale set to 2 and delta set to 10

-i INPUT -vf "hwupload, sobel_opencl=scale=2:delta=10, hwdownload" OUTPUT

tonemap_opencl
Perform HDR(PQ/HLG) to SDR conversion with tone-mapping.

It accepts the following parameters:

tonemap
Specify the tone-mapping operator to be used. Same as tonemap option in tonemap.

param
Tune the tone mapping algorithm. same as param option in tonemap.

The default value is 0.5, and the algorithm here is a little different from the cpu version tonemap
currently. A setting of 0.0 disables this option.

threshold
The tonemapping algorithm parameters is fine-tuned per each scene. And a threshold is used to detect
whether the scene has changed or not. If the distance between the current frame average brightness
and the current running average exceeds a threshold value, we would re-calculate scene average and
peak brightness. The default value is 0.2.

format
Specify the output pixel format.

Currently supported formats are:

p010
nv12
range, r
Set the output color range.

Possible values are:

tv/mpeg
pc/jpeg

Default is same as input.

primaries, p
Set the output color primaries.

Possible values are:

bt709
bt2020

Default is same as input.

transfer, t
Set the output transfer characteristics.

Possible values are:

bt709
bt2020

Default is bt709.

matrix, m
Set the output colorspace matrix.

Possible value are:

bt709
bt2020

Default is same as input.

Example

• Convert HDR(PQ/HLG) video to bt2020-transfer-characteristic p010 format using linear operator.

-i INPUT -vf "format=p010,hwupload,tonemap_opencl=t=bt2020:tonemap=linear:format=p010,hwdownload,format=p010" OUTPUT

unsharp_opencl
Sharpen or blur the input video.

It accepts the following parameters:

luma_msize_x, lx
Set the luma matrix horizontal size. Range is "[1, 23]" and default value is 5.

luma_msize_y, ly
Set the luma matrix vertical size. Range is "[1, 23]" and default value is 5.

luma_amount, la
Set the luma effect strength. Range is "[-10, 10]" and default value is 1.0.

Negative values will blur the input video, while positive values will sharpen it, a value of zero
will disable the effect.

chroma_msize_x, cx
Set the chroma matrix horizontal size. Range is "[1, 23]" and default value is 5.

chroma_msize_y, cy
Set the chroma matrix vertical size. Range is "[1, 23]" and default value is 5.

chroma_amount, ca
Set the chroma effect strength. Range is "[-10, 10]" and default value is 0.0.

Negative values will blur the input video, while positive values will sharpen it, a value of zero
will disable the effect.

All parameters are optional and default to the equivalent of the string '5:5:1.0:5:5:0.0'.

Examples

• Apply strong luma sharpen effect:

-i INPUT -vf "hwupload, unsharp_opencl=luma_msize_x=7:luma_msize_y=7:luma_amount=2.5, hwdownload" OUTPUT

• Apply a strong blur of both luma and chroma parameters:

-i INPUT -vf "hwupload, unsharp_opencl=7:7:-2:7:7:-2, hwdownload" OUTPUT

xfade_opencl
Cross fade two videos with custom transition effect by using OpenCL.

It accepts the following options:

transition
Set one of possible transition effects.

custom
Select custom transition effect, the actual transition description will be picked from source and
kernel options.

fade
wipeleft
wiperight
wipeup
wipedown
slideleft
slideright
slideup
slidedown
Default transition is fade.

source
OpenCL program source file for custom transition.

kernel
Set name of kernel to use for custom transition from program source file.

duration
Set duration of video transition.

offset
Set time of start of transition relative to first video.

The kernel function needs to take the following arguments:

• Destination image, __write_only image2d_t.

This image will become the output; the kernel should write all of it.

• First Source image, __read_only image2d_t. Second Source image, __read_only image2d_t.

These are the most recent images on each input. The kernel may read from them to generate the
output, but they can't be written to.

• Transition progress, float. This value is always between 0 and 1 inclusive.

Example programs:

• Apply dots curtain transition effect:

__kernel void blend_images(__write_only image2d_t dst,
__read_only image2d_t src1,
__read_only image2d_t src2,
float progress)
{
const sampler_t sampler = (CLK_NORMALIZED_COORDS_FALSE |
CLK_FILTER_LINEAR);
int2 p = (int2)(get_global_id(0), get_global_id(1));
float2 rp = (float2)(get_global_id(0), get_global_id(1));
float2 dim = (float2)(get_image_dim(src1).x, get_image_dim(src1).y);
rp = rp / dim;

float2 dots = (float2)(20.0, 20.0);
float2 center = (float2)(0,0);
float2 unused;

float4 val1 = read_imagef(src1, sampler, p);
float4 val2 = read_imagef(src2, sampler, p);
bool next = distance(fract(rp * dots, &unused), (float2)(0.5, 0.5)) < (progress / distance(rp, center));

write_imagef(dst, p, next ? val1 : val2);
}

VAAPI VIDEO FILTERS

VAAPI Video filters are usually used with VAAPI decoder and VAAPI encoder. Below is a description of
VAAPI video filters.

To enable compilation of these filters you need to configure FFmpeg with "--enable-vaapi".

To use vaapi filters, you need to setup the vaapi device correctly. For more information, please read
<https://trac.ffmpeg.org/wiki/Hardware/VAAPI>

overlay_vaapi
Overlay one video on the top of another.

It takes two inputs and has one output. The first input is the "main" video on which the second input is
overlaid.

The filter accepts the following options:

x
y Set expressions for the x and y coordinates of the overlaid video on the main video.

Default value is "0" for both expressions.

w
h Set expressions for the width and height the overlaid video on the main video.

Default values are 'overlay_iw' for 'w' and 'overlay_ih*w/overlay_iw' for 'h'.

The expressions can contain the following parameters:

main_w, W
main_h, H
The main input width and height.

overlay_iw
overlay_ih
The overlay input width and height.

overlay_w, w
overlay_h, h
The overlay output width and height.

overlay_x, x
overlay_y, y
Position of the overlay layer inside of main

alpha
Set transparency of overlaid video. Allowed range is 0.0 to 1.0. Higher value means lower
transparency. Default value is 1.0.

eof_action
See framesync.

shortest
See framesync.

repeatlast
See framesync.

This filter also supports the framesync options.

Examples

• Overlay an image LOGO at the top-left corner of the INPUT video. Both inputs for this filter are
yuv420p format.

-i INPUT -i LOGO -filter_complex "[0:v]hwupload[a], [1:v]format=yuv420p, hwupload[b], [a][b]overlay_vaapi" OUTPUT

• Overlay an image LOGO at the offset (200, 100) from the top-left corner of the INPUT video. The
inputs have same memory layout for color channels, the overlay has additional alpha plane, like INPUT
is yuv420p, and the LOGO is yuva420p.

-i INPUT -i LOGO -filter_complex "[0:v]hwupload[a], [1:v]format=yuva420p, hwupload[b], [a][b]overlay_vaapi=x=200:y=100:w=400:h=300:alpha=1.0, hwdownload, format=nv12" OUTPUT

tonemap_vaapi
Perform HDR(High Dynamic Range) to SDR(Standard Dynamic Range) conversion with tone-mapping. It maps the
dynamic range of HDR10 content to the SDR content. It currently only accepts HDR10 as input.

It accepts the following parameters:

format
Specify the output pixel format.

Currently supported formats are:

p010
nv12

Default is nv12.

primaries, p
Set the output color primaries.

Default is same as input.

transfer, t
Set the output transfer characteristics.

Default is bt709.

matrix, m
Set the output colorspace matrix.

Default is same as input.

Example

• Convert HDR(HDR10) video to bt2020-transfer-characteristic p010 format

tonemap_vaapi=format=p010:t=bt2020-10

hstack_vaapi
Stack input videos horizontally.

This is the VA-API variant of the hstack filter, each input stream may have different height, this filter
will scale down/up each input stream while keeping the orignal aspect.

It accepts the following options:

inputs
See hstack.

shortest
See hstack.

height
Set height of output. If set to 0, this filter will set height of output to height of the first input
stream. Default value is 0.

vstack_vaapi
Stack input videos vertically.

This is the VA-API variant of the vstack filter, each input stream may have different width, this filter
will scale down/up each input stream while keeping the orignal aspect.

It accepts the following options:

inputs
See vstack.

shortest
See vstack.

width
Set width of output. If set to 0, this filter will set width of output to width of the first input
stream. Default value is 0.

xstack_vaapi
Stack video inputs into custom layout.

This is the VA-API variant of the xstack filter, each input stream may have different size, this filter
will scale down/up each input stream to the given output size, or the size of the first input stream.

It accepts the following options:

inputs
See xstack.

shortest
See xstack.

layout
See xstack. Moreover, this permits the user to supply output size for each input stream.

xstack_vaapi=inputs=4:layout=0_0_1920x1080|0_h0_1920x1080|w0_0_1920x1080|w0_h0_1920x1080

grid
See xstack.

grid_tile_size
Set output size for each input stream when grid is set. If this option is not set, this filter will
set output size by default to the size of the first input stream. For the syntax of this option,
check the "Video size" section in the ffmpeg-utils manual.

fill
See xstack.

VULKAN VIDEO FILTERS

Below is a description of the currently available Vulkan video filters.

To enable compilation of these filters you need to configure FFmpeg with "--enable-vulkan" and either
"--enable-libglslang" or "--enable-libshaderc".

Running Vulkan filters requires you to initialize a hardware device and to pass that device to all
filters in any filter graph.

-init_hw_device vulkan[=name][:device[,key=value...]]
Initialise a new hardware device of type vulkan called name, using the given device parameters and
options in key=value. The following options are supported:

debug
Switches validation layers on if set to 1.

linear_images
Allocates linear images. Does not apply to decoding.

disable_multiplane
Disables multiplane images. Does not apply to decoding.

-filter_hw_device name
Pass the hardware device called name to all filters in any filter graph.

For more detailed information see <https://www.ffmpeg.org/ffmpeg.html#Advanced-Video-options>

• Example of choosing the first device and running nlmeans_vulkan filter with default parameters on it.

-init_hw_device vulkan=vk:0 -filter_hw_device vk -i INPUT -vf "hwupload,nlmeans_vulkan,hwdownload" OUTPUT

As Vulkan filters are not able to access frame data in normal memory, all frame data needs to be uploaded
(hwupload) to hardware surfaces connected to the appropriate device before being used and then downloaded
(hwdownload) back to normal memory. Note that hwupload will upload to a frame with the same layout as the
software frame, so it may be necessary to add a format filter immediately before to get the input into
the right format and hwdownload does not support all formats on the output - it is usually necessary to
insert an additional format filter immediately following in the graph to get the output in a supported
format.

avgblur_vulkan
Apply an average blur filter, implemented on the GPU using Vulkan.

The filter accepts the following options:

sizeX
Set horizontal radius size. Range is "[1, 32]" and default value is 3.

sizeY
Set vertical radius size. Range is "[1, 32]" and default value is 3.

planes
Set which planes to filter. Default value is 0xf, by which all planes are processed.

blend_vulkan
Blend two Vulkan frames into each other.

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:

normal
multiply

bwdif_vulkan
Deinterlacer using bwdif, the "Bob Weaver Deinterlacing Filter" algorithm, implemented on the GPU using
Vulkan.

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".

chromaber_vulkan
Apply an effect that emulates chromatic aberration. Works best with RGB inputs, but provides a similar
effect with YCbCr inputs too.

dist_x
Horizontal displacement multiplier. Each chroma pixel's position will be multiplied by this amount,
starting from the center of the image. Default is 0.

dist_y
Similarly, this sets the vertical displacement multiplier. Default is 0.

color_vulkan
Video source that creates a Vulkan frame of a solid color. Useful for benchmarking, or overlaying.

It accepts the following parameters:

color
The color to use. Either a name, or a hexadecimal value. The default value is "black".

size
The size of the output frame. Default value is "1920x1080".

rate
The framerate to output at. Default value is 60 frames per second.

duration
The video duration. Default value is -0.000001.

sar The video signal aspect ratio. Default value is "1/1".

format
The pixel format of the output Vulkan frames. Default value is "yuv444p".

out_range
Set the output YCbCr sample range.

auto/unknown
Choose automatically.

jpeg/full/pc
Set full range (0-255 in case of 8-bit luma).

mpeg/limited/tv
Set "MPEG" range (16-235 in case of 8-bit luma).

vflip_vulkan
Flips an image vertically.

hflip_vulkan
Flips an image horizontally.

flip_vulkan
Flips an image along both the vertical and horizontal axis.

gblur_vulkan
Apply Gaussian blur filter on Vulkan frames.

The filter accepts the following options:

sigma
Set horizontal sigma, standard deviation of Gaussian blur. Default is 0.5.

sigmaV
Set vertical sigma, if negative it will be same as "sigma". Default is -1.

planes
Set which planes to filter. By default all planes are filtered.

size
Set the kernel size along the horizontal axis. Default is 19.

sizeV
Set the kernel size along the vertical axis. Default is 0, which sets to use the same value as size.

nlmeans_vulkan
Denoise frames using Non-Local Means algorithm, implemented on the GPU using Vulkan. Supports more pixel
formats than nlmeans or nlmeans_opencl, including alpha channel support.

The filter accepts the following options.

s Set denoising strength for all components. Default is 1.0. Must be in range [1.0, 100.0].

p Set patch size for all planes. Default is 7. Must be odd number in range [0, 99].

r Set research size. Default is 15. Must be odd number in range [0, 99].

t Set parallelism. Default is 36. Must be a number in the range [1, 168]. Larger values may speed up
processing, at the cost of more VRAM. Lower values will slow it down, reducing VRAM usage. Only
supported on GPUs with atomic float operations (RDNA3+, Ampere+).

s0
s1
s2
s3 Set denoising strength for a specific component. Default is 1, equal to s. Must be odd number in
range [1, 100].

p0
p1
p2
p3 Set patch size for a specific component. Default is 7, equal to p. Must be odd number in range [0,
99].

overlay_vulkan
Overlay one video on top of another.

It takes two inputs and has one output. The first input is the "main" video on which the second input is
overlaid. This filter requires all inputs to use the same pixel format. So, format conversion may be
needed.

The filter accepts the following options:

x Set the x coordinate of the overlaid video on the main video. Default value is 0.

y Set the y coordinate of the overlaid video on the main video. Default value is 0.

transpose_vt
Transpose rows with columns in the input video and optionally flip it. For more in depth examples see
the transpose video filter, which shares mostly the same options.

It accepts the following parameters:

dir Specify the transposition direction.

Can assume the following values:

cclock_flip
Rotate by 90 degrees counterclockwise and vertically flip. (default)

clock
Rotate by 90 degrees clockwise.

cclock
Rotate by 90 degrees counterclockwise.

clock_flip
Rotate by 90 degrees clockwise and vertically flip.

hflip
Flip the input video horizontally.

vflip
Flip the input video vertically.

passthrough
Do not apply the transposition if the input geometry matches the one specified by the specified
value. It accepts the following values:

none
Always apply transposition. (default)

portrait
Preserve portrait geometry (when height >= width).

landscape
Preserve landscape geometry (when width >= height).

transpose_vulkan
Transpose rows with columns in the input video and optionally flip it. For more in depth examples see
the transpose video filter, which shares mostly the same options.

It accepts the following parameters:

dir Specify the transposition direction.

Can assume the following values:

cclock_flip
Rotate by 90 degrees counterclockwise and vertically flip. (default)

clock
Rotate by 90 degrees clockwise.

cclock
Rotate by 90 degrees counterclockwise.

clock_flip
Rotate by 90 degrees clockwise and vertically flip.

passthrough
Do not apply the transposition if the input geometry matches the one specified by the specified
value. It accepts the following values:

none
Always apply transposition. (default)

portrait
Preserve portrait geometry (when height >= width).

landscape
Preserve landscape geometry (when width >= height).

QSV VIDEO FILTERS

Below is a description of the currently available QSV video filters.

To enable compilation of these filters you need to configure FFmpeg with "--enable-libmfx" or
"--enable-libvpl".

To use QSV filters, you need to setup the QSV device correctly. For more information, please read
<https://trac.ffmpeg.org/wiki/Hardware/QuickSync>

hstack_qsv
Stack input videos horizontally.

This is the QSV variant of the hstack filter, each input stream may have different height, this filter
will scale down/up each input stream while keeping the orignal aspect.

It accepts the following options:

inputs
See hstack.

shortest
See hstack.

height
Set height of output. If set to 0, this filter will set height of output to height of the first input
stream. Default value is 0.

vstack_qsv
Stack input videos vertically.

This is the QSV variant of the vstack filter, each input stream may have different width, this filter
will scale down/up each input stream while keeping the orignal aspect.

It accepts the following options:

inputs
See vstack.

shortest
See vstack.

width
Set width of output. If set to 0, this filter will set width of output to width of the first input
stream. Default value is 0.

xstack_qsv
Stack video inputs into custom layout.

This is the QSV variant of the xstack filter.

It accepts the following options:

inputs
See xstack.

shortest
See xstack.

layout
See xstack. Moreover, this permits the user to supply output size for each input stream.

xstack_qsv=inputs=4:layout=0_0_1920x1080|0_h0_1920x1080|w0_0_1920x1080|w0_h0_1920x1080

grid
See xstack.

fill
See xstack.

VIDEO SOURCES

Below is a description of the currently available video sources.

buffer
Buffer video frames, and make them available to the filter chain.

This source is mainly intended for a programmatic use, in particular through the interface defined in
libavfilter/buffersrc.h.

It accepts the following parameters:

video_size
Specify the size (width and height) of the buffered video frames. For the syntax of this option,
check the "Video size" section in the ffmpeg-utils manual.

width
The input video width.

height
The input video height.

pix_fmt
A string representing the pixel format of the buffered video frames. It may be a number
corresponding to a pixel format, or a pixel format name.

time_base
Specify the timebase assumed by the timestamps of the buffered frames.

frame_rate
Specify the frame rate expected for the video stream.

pixel_aspect, sar
The sample (pixel) aspect ratio of the input video.

hw_frames_ctx
When using a hardware pixel format, this should be a reference to an AVHWFramesContext describing
input frames.

For example:

buffer=width=320:height=240:pix_fmt=yuv410p:time_base=1/24:sar=1

will instruct the source to accept video frames with size 320x240 and with format "yuv410p", assuming
1/24 as the timestamps timebase and square pixels (1:1 sample aspect ratio). Since the pixel format with
name "yuv410p" corresponds to the number 6 (check the enum AVPixelFormat definition in
libavutil/pixfmt.h), this example corresponds to:

buffer=size=320x240:pixfmt=6:time_base=1/24:pixel_aspect=1/1

Alternatively, the options can be specified as a flat string, but this syntax is deprecated:

width:height:pix_fmt:time_base.num:time_base.den:pixel_aspect.num:pixel_aspect.den

cellauto
Create a pattern generated by an elementary cellular automaton.

The initial state of the cellular automaton can be defined through the filename and pattern options. If
such options are not specified an initial state is created randomly.

At each new frame a new row in the video is filled with the result of the cellular automaton next
generation. The behavior when the whole frame is filled is defined by the scroll option.

This source accepts the following options:

filename, f
Read the initial cellular automaton state, i.e. the starting row, from the specified file. In the
file, each non-whitespace character is considered an alive cell, a newline will terminate the row,
and further characters in the file will be ignored.

pattern, p
Read the initial cellular automaton state, i.e. the starting row, from the specified string.

Each non-whitespace character in the string is considered an alive cell, a newline will terminate the
row, and further characters in the string will be ignored.

rate, r
Set the video rate, that is the number of frames generated per second. Default is 25.

random_fill_ratio, ratio
Set the random fill ratio for the initial cellular automaton row. It is a floating point number value
ranging from 0 to 1, defaults to 1/PHI.

This option is ignored when a file or a pattern is specified.

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

rule
Set the cellular automaton rule, it is a number ranging from 0 to 255. Default value is 110.

size, s
Set the size of the output video. For the syntax of this option, check the "Video size" section in
the ffmpeg-utils manual.

If filename or pattern is specified, the size is set by default to the width of the specified initial
state row, and the height is set to width * PHI.

If size is set, it must contain the width of the specified pattern string, and the specified pattern
will be centered in the larger row.

If a filename or a pattern string is not specified, the size value defaults to "320x518" (used for a
randomly generated initial state).

scroll
If set to 1, scroll the output upward when all the rows in the output have been already filled. If
set to 0, the new generated row will be written over the top row just after the bottom row is filled.
Defaults to 1.

start_full, full
If set to 1, completely fill the output with generated rows before outputting the first frame. This
is the default behavior, for disabling set the value to 0.

stitch
If set to 1, stitch the left and right row edges together. This is the default behavior, for
disabling set the value to 0.

Examples

• Read the initial state from pattern, and specify an output of size 200x400.

cellauto=f=pattern:s=200x400

• Generate a random initial row with a width of 200 cells, with a fill ratio of 2/3:

cellauto=ratio=2/3:s=200x200

• Create a pattern generated by rule 18 starting by a single alive cell centered on an initial row with
width 100:

cellauto=p=@s=100x400:full=0:rule=18

• Specify a more elaborated initial pattern:

cellauto=p='@@ @ @@':s=100x400:full=0:rule=18

coreimagesrc
Video source generated on GPU using Apple's CoreImage API on OSX.

This video source is a specialized version of the coreimage video filter. Use a core image generator at
the beginning of the applied filterchain to generate the content.

The coreimagesrc video source accepts the following options:

list_generators
List all available generators along with all their respective options as well as possible minimum and
maximum values along with the default values.

list_generators=true

size, s
Specify the size of the sourced video. For the syntax of this option, check the "Video size" section
in the ffmpeg-utils manual. The default value is "320x240".

rate, r
Specify the frame rate of the sourced video, as the number of frames generated per second. It has to
be a string in the format frame_rate_num/frame_rate_den, an integer number, a floating point number
or a valid video frame rate abbreviation. The default value is "25".

sar Set the sample aspect ratio of the sourced video.

duration, d
Set the duration of the sourced video. See the Time duration section in the ffmpeg-utils(1) manual
for the accepted syntax.

If not specified, or the expressed duration is negative, the video is supposed to be generated
forever.

Additionally, all options of the coreimage video filter are accepted. A complete filterchain can be used
for further processing of the generated input without CPU-HOST transfer. See coreimage documentation and
examples for details.

Examples

• Use CIQRCodeGenerator to create a QR code for the FFmpeg homepage, given as complete and escaped
command-line for Apple's standard bash shell:

ffmpeg -f lavfi -i coreimagesrc=s=100x100:filter=CIQRCodeGenerator@inputMessage=https\\\\\://FFmpeg.org/@inputCorrectionLevel=H -frames:v 1 QRCode.png

This example is equivalent to the QRCode example of coreimage without the need for a nullsrc video
source.

ddagrab
Captures the Windows Desktop via Desktop Duplication API.

The filter exclusively returns D3D11 Hardware Frames, for on-gpu encoding or processing. So an explicit
hwdownload is needed for any kind of software processing.

It accepts the following options:

output_idx
DXGI Output Index to capture.

Usually corresponds to the index Windows has given the screen minus one, so it's starting at 0.

Defaults to output 0.

draw_mouse
Whether to draw the mouse cursor.

Defaults to true.

Only affects hardware cursors. If a game or application renders its own cursor, it'll always be
captured.

framerate
Framerate at which the desktop will be captured.

Defaults to 30 FPS.

video_size
Specify the size of the captured video.

Defaults to the full size of the screen.

Cropped from the bottom/right if smaller than screen size.

offset_x
Horizontal offset of the captured video.

offset_y
Vertical offset of the captured video.

output_fmt
Desired filter output format. Defaults to 8 Bit BGRA.

It accepts the following values:

auto
Passes all supported output formats to DDA and returns what DDA decides to use.

8bit
bgra
8 Bit formats always work, and DDA will convert to them if neccesary.

10bit
x2bgr10
Filter initialization will fail if 10 bit format is requested but unavailable.

Examples

Capture primary screen and encode using nvenc:

ffmpeg -f lavfi -i ddagrab -c:v h264_nvenc -cq 18 output.mp4

You can also skip the lavfi device and directly use the filter. Also demonstrates downloading the frame
and encoding with libx264. Explicit output format specification is required in this case:

ffmpeg -filter_complex ddagrab=output_idx=1:framerate=60,hwdownload,format=bgra -c:v libx264 -crf 18 output.mp4

If you want to capture only a subsection of the desktop, this can be achieved by specifying a smaller
size and its offsets into the screen:

ddagrab=video_size=800x600:offset_x=100:offset_y=100

gradients
Generate several gradients.

size, s
Set frame size. For the syntax of this option, check the "Video size" section in the ffmpeg-utils
manual. Default value is "640x480".

rate, r
Set frame rate, expressed as number of frames per second. Default value is "25".

c0, c1, c2, c3, c4, c5, c6, c7
Set 8 colors. Default values for colors is to pick random one.

x0, y0, y0, y1
Set gradient line source and destination points. If negative or out of range, random ones are picked.

nb_colors, n
Set number of colors to use at once. Allowed range is from 2 to 8. Default value is 2.

seed
Set seed for picking gradient line points.

duration, d
Set the duration of the sourced video. See the Time duration section in the ffmpeg-utils(1) manual
for the accepted syntax.

If not specified, or the expressed duration is negative, the video is supposed to be generated
forever.

speed
Set speed of gradients rotation.

type, t
Set type of gradients, can be "linear" or "radial" or "circular" or "spiral".

mandelbrot
Generate a Mandelbrot set fractal, and progressively zoom towards the point specified with start_x and
start_y.

This source accepts the following options:

end_pts
Set the terminal pts value. Default value is 400.

end_scale
Set the terminal scale value. Must be a floating point value. Default value is 0.3.

inner
Set the inner coloring mode, that is the algorithm used to draw the Mandelbrot fractal internal
region.

It shall assume one of the following values:

black
Set black mode.

convergence
Show time until convergence.

mincol
Set color based on point closest to the origin of the iterations.

period
Set period mode.

Default value is mincol.

bailout
Set the bailout value. Default value is 10.0.

maxiter
Set the maximum of iterations performed by the rendering algorithm. Default value is 7189.

outer
Set outer coloring mode. It shall assume one of following values:

iteration_count
Set iteration count mode.

normalized_iteration_count
set normalized iteration count mode.

Default value is normalized_iteration_count.

rate, r
Set frame rate, expressed as number of frames per second. Default value is "25".

size, s
Set frame size. For the syntax of this option, check the "Video size" section in the ffmpeg-utils
manual. Default value is "640x480".

start_scale
Set the initial scale value. Default value is 3.0.

start_x
Set the initial x position. Must be a floating point value between -100 and 100. Default value is
-0.743643887037158704752191506114774.

start_y
Set the initial y position. Must be a floating point value between -100 and 100. Default value is
-0.131825904205311970493132056385139.

mptestsrc
Generate various test patterns, as generated by the MPlayer test filter.

The size of the generated video is fixed, and is 256x256. This source is useful in particular for
testing encoding features.

This source accepts the following options:

duration, d
Set the duration of the sourced video. See the Time duration section in the ffmpeg-utils(1) manual
for the accepted syntax.

If not specified, or the expressed duration is negative, the video is supposed to be generated
forever.

test, t
Set the number or the name of the test to perform. Supported tests are:

dc_luma
dc_chroma
freq_luma
freq_chroma
amp_luma
amp_chroma
cbp
mv
ring1
ring2
all
max_frames, m
Set the maximum number of frames generated for each test, default value is 30.

Default value is "all", which will cycle through the list of all tests.

Some examples:

mptestsrc=t=dc_luma

will generate a "dc_luma" test pattern.

frei0r_src
Provide a frei0r source.

To enable compilation of this filter you need to install the frei0r header and configure FFmpeg with
"--enable-frei0r".

This source accepts the following parameters:

size
The size of the video to generate. For the syntax of this option, check the "Video size" section in
the ffmpeg-utils manual.

framerate
The framerate of the generated video. It may be a string of the form num/den or a frame rate
abbreviation.

filter_name
The name to the frei0r source to load. For more information regarding frei0r and how to set the
parameters, read the frei0r section in the video filters documentation.

filter_params
A '|'-separated list of parameters to pass to the frei0r source.

For example, to generate a frei0r partik0l source with size 200x200 and frame rate 10 which is overlaid
on the overlay filter main input:

frei0r_src=size=200x200:framerate=10:filter_name=partik0l:filter_params=1234 [overlay]; [in][overlay] overlay

life
Generate a life pattern.

This source is based on a generalization of John Conway's life game.

The sourced input represents a life grid, each pixel represents a cell which can be in one of two
possible states, alive or dead. Every cell interacts with its eight neighbours, which are the cells that
are horizontally, vertically, or diagonally adjacent.

At each interaction the grid evolves according to the adopted rule, which specifies the number of
neighbor alive cells which will make a cell stay alive or born. The rule option allows one to specify the
rule to adopt.

This source accepts the following options:

filename, f
Set the file from which to read the initial grid state. In the file, each non-whitespace character is
considered an alive cell, and newline is used to delimit the end of each row.

If this option is not specified, the initial grid is generated randomly.

rate, r
Set the video rate, that is the number of frames generated per second. Default is 25.

random_fill_ratio, ratio
Set the random fill ratio for the initial random grid. It is a floating point number value ranging
from 0 to 1, defaults to 1/PHI. It is ignored when a file is specified.

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

rule
Set the life rule.

A rule can be specified with a code of the kind "SNS/BNB", where NS and NB are sequences of numbers
in the range 0-8, NS specifies the number of alive neighbor cells which make a live cell stay alive,
and NB the number of alive neighbor cells which make a dead cell to become alive (i.e. to "born").
"s" and "b" can be used in place of "S" and "B", respectively.

Alternatively a rule can be specified by an 18-bits integer. The 9 high order bits are used to encode
the next cell state if it is alive for each number of neighbor alive cells, the low order bits
specify the rule for "borning" new cells. Higher order bits encode for an higher number of neighbor
cells. For example the number 6153 = "(12<<9)+9" specifies a stay alive rule of 12 and a born rule
of 9, which corresponds to "S23/B03".

Default value is "S23/B3", which is the original Conway's game of life rule, and will keep a cell
alive if it has 2 or 3 neighbor alive cells, and will born a new cell if there are three alive cells
around a dead cell.

size, s
Set the size of the output video. For the syntax of this option, check the "Video size" section in
the ffmpeg-utils manual.

If filename is specified, the size is set by default to the same size of the input file. If size is
set, it must contain the size specified in the input file, and the initial grid defined in that file
is centered in the larger resulting area.

If a filename is not specified, the size value defaults to "320x240" (used for a randomly generated
initial grid).

stitch
If set to 1, stitch the left and right grid edges together, and the top and bottom edges also.
Defaults to 1.

mold
Set cell mold speed. If set, a dead cell will go from death_color to mold_color with a step of mold.
mold can have a value from 0 to 255.

life_color
Set the color of living (or new born) cells.

death_color
Set the color of dead cells. If mold is set, this is the first color used to represent a dead cell.

mold_color
Set mold color, for definitely dead and moldy cells.

For the syntax of these 3 color options, check the "Color" section in the ffmpeg-utils manual.

Examples

• Read a grid from pattern, and center it on a grid of size 300x300 pixels:

life=f=pattern:s=300x300

• Generate a random grid of size 200x200, with a fill ratio of 2/3:

life=ratio=2/3:s=200x200

• Specify a custom rule for evolving a randomly generated grid:

life=rule=S14/B34

• Full example with slow death effect (mold) using ffplay:

ffplay -f lavfi life=s=300x200:mold=10:r=60:ratio=0.1:death_color=#C83232:life_color=#00ff00,scale=1200:800:flags=16

allrgb, allyuv, color, colorchart, colorspectrum, haldclutsrc, nullsrc, pal75bars, pal100bars, rgbtestsrc,
smptebars, smptehdbars, testsrc, testsrc2, yuvtestsrc
The "allrgb" source returns frames of size 4096x4096 of all rgb colors.

The "allyuv" source returns frames of size 4096x4096 of all yuv colors.

The "color" source provides an uniformly colored input.

The "colorchart" source provides a colors checker chart.

The "colorspectrum" source provides a color spectrum input.

The "haldclutsrc" source provides an identity Hald CLUT. See also haldclut filter.

The "nullsrc" source returns unprocessed video frames. It is mainly useful to be employed in analysis /
debugging tools, or as the source for filters which ignore the input data.

The "pal75bars" source generates a color bars pattern, based on EBU PAL recommendations with 75% color
levels.

The "pal100bars" source generates a color bars pattern, based on EBU PAL recommendations with 100% color
levels.

The "rgbtestsrc" source generates an RGB test pattern useful for detecting RGB vs BGR issues. You should
see a red, green and blue stripe from top to bottom.

The "smptebars" source generates a color bars pattern, based on the SMPTE Engineering Guideline EG
1-1990.

The "smptehdbars" source generates a color bars pattern, based on the SMPTE RP 219-2002.

The "testsrc" source generates a test video pattern, showing a color pattern, a scrolling gradient and a
timestamp. This is mainly intended for testing purposes.

The "testsrc2" source is similar to testsrc, but supports more pixel formats instead of just "rgb24".
This allows using it as an input for other tests without requiring a format conversion.

The "yuvtestsrc" source generates an YUV test pattern. You should see a y, cb and cr stripe from top to
bottom.

The sources accept the following parameters:

level
Specify the level of the Hald CLUT, only available in the "haldclutsrc" source. A level of "N"
generates a picture of "N*N*N" by "N*N*N" pixels to be used as identity matrix for 3D lookup tables.
Each component is coded on a "1/(N*N)" scale.

color, c
Specify the color of the source, only available in the "color" source. For the syntax of this option,
check the "Color" section in the ffmpeg-utils manual.

size, s
Specify the size of the sourced video. For the syntax of this option, check the "Video size" section
in the ffmpeg-utils manual. The default value is "320x240".

This option is not available with the "allrgb", "allyuv", and "haldclutsrc" filters.

duration, d
Set the duration of the sourced video. See the Time duration section in the ffmpeg-utils(1) manual
for the accepted syntax.

If not specified, or the expressed duration is negative, the video is supposed to be generated
forever.

Since the frame rate is used as time base, all frames including the last one will have their full
duration. If the specified duration is not a multiple of the frame duration, it will be rounded up.

sar Set the sample aspect ratio of the sourced video.

alpha
Specify the alpha (opacity) of the background, only available in the "testsrc2" source. The value
must be between 0 (fully transparent) and 255 (fully opaque, the default).

decimals, n
Set the number of decimals to show in the timestamp, only available in the "testsrc" source.

The displayed timestamp value will correspond to the original timestamp value multiplied by the power
of 10 of the specified value. Default value is 0.

type
Set the type of the color spectrum, only available in the "colorspectrum" source. Can be one of the
following:

black
white
all
patch_size
Set patch size of single color patch, only available in the "colorchart" source. Default is "64x64".

preset
Set colorchecker colors preset, only available in the "colorchart" source.

Available values are:

reference
skintones

Default value is "reference".

Examples

• Generate a video with a duration of 5.3 seconds, with size 176x144 and a frame rate of 10 frames per
second:

testsrc=duration=5.3:size=qcif:rate=10

• The following graph description will generate a red source with an opacity of 0.2, with size "qcif"
and a frame rate of 10 frames per second:

color=c=red@0.2:s=qcif:r=10

• If the input content is to be ignored, "nullsrc" can be used. The following command generates noise
in the luma plane by employing the "geq" filter:

nullsrc=s=256x256, geq=random(1)*255:128:128

Commands

The "color" source supports the following commands:

c, color
Set the color of the created image. Accepts the same syntax of the corresponding color option.

openclsrc
Generate video using an OpenCL program.

source
OpenCL program source file.

kernel
Kernel name in program.

size, s
Size of frames to generate. This must be set.

format
Pixel format to use for the generated frames. This must be set.

rate, r
Number of frames generated every second. Default value is '25'.

For details of how the program loading works, see the program_opencl filter.

Example programs:

• Generate a colour ramp by setting pixel values from the position of the pixel in the output image.
(Note that this will work with all pixel formats, but the generated output will not be the same.)

__kernel void ramp(__write_only image2d_t dst,
unsigned int index)
{
int2 loc = (int2)(get_global_id(0), get_global_id(1));

float4 val;
val.xy = val.zw = convert_float2(loc) / convert_float2(get_image_dim(dst));

write_imagef(dst, loc, val);
}

• Generate a Sierpinski carpet pattern, panning by a single pixel each frame.

__kernel void sierpinski_carpet(__write_only image2d_t dst,
unsigned int index)
{
int2 loc = (int2)(get_global_id(0), get_global_id(1));

float4 value = 0.0f;
int x = loc.x + index;
int y = loc.y + index;
while (x > 0 || y > 0) {
if (x % 3 == 1 && y % 3 == 1) {
value = 1.0f;
break;
}
x /= 3;
y /= 3;
}

write_imagef(dst, loc, value);
}

sierpinski
Generate a Sierpinski carpet/triangle fractal, and randomly pan around.

This source accepts the following options:

size, s
Set frame size. For the syntax of this option, check the "Video size" section in the ffmpeg-utils
manual. Default value is "640x480".

rate, r
Set frame rate, expressed as number of frames per second. Default value is "25".

seed
Set seed which is used for random panning.

jump
Set max jump for single pan destination. Allowed range is from 1 to 10000.

type
Set fractal type, can be default "carpet" or "triangle".

zoneplate
Generate a zoneplate test video pattern.

This source accepts the following options:

size, s
Set frame size. For the syntax of this option, check the "Video size" section in the ffmpeg-utils
manual. Default value is "320x240".

rate, r
Set frame rate, expressed as number of frames per second. Default value is "25".

duration, d
Set the duration of the sourced video. See the Time duration section in the ffmpeg-utils(1) manual
for the accepted syntax.

If not specified, or the expressed duration is negative, the video is supposed to be generated
forever.

sar Set the sample aspect ratio of the sourced video.

precision
Set precision in bits for look-up table for sine calculations. Default value is 10. Allowed range is
from 4 to 16.

xo Set horizontal axis offset for output signal. Default value is 0.

yo Set vertical axis offset for output signal. Default value is 0.

to Set time axis offset for output signal. Default value is 0.

k0 Set 0-order, constant added to signal phase. Default value is 0.

kx Set 1-order, phase factor multiplier for horizontal axis. Default value is 0.

ky Set 1-order, phase factor multiplier for vertical axis. Default value is 0.

kt Set 1-order, phase factor multiplier for time axis. Default value is 0.

kxt, kyt, kxy
Set phase factor multipliers for combination of spatial and temporal axis. Default value is 0.

kx2 Set 2-order, phase factor multiplier for horizontal axis. Default value is 0.

ky2 Set 2-order, phase factor multiplier for vertical axis. Default value is 0.

kt2 Set 2-order, phase factor multiplier for time axis. Default value is 0.

ku Set the constant added to final phase to produce chroma-blue component of signal. Default value is
0.

kv Set the constant added to final phase to produce chroma-red component of signal. Default value is 0.

Commands

This source supports the some above options as commands.

Examples

• Generate horizontal color sine sweep:

zoneplate=ku=512:kv=0:kt2=0:kx2=256:s=wvga:xo=-426:kt=11

• Generate vertical color sine sweep:

zoneplate=ku=512:kv=0:kt2=0:ky2=156:s=wvga:yo=-240:kt=11

• Generate circular zone-plate:

zoneplate=ku=512:kv=100:kt2=0:ky2=256:kx2=556:s=wvga:yo=0:kt=11

VIDEO SINKS

       Below is a description of the currently available video sinks.

   buffersink
       Buffer video frames, and make them available to the end of the filter graph.

       This  sink  is  mainly  intended  for  programmatic  use,  in particular through the interface defined in
       libavfilter/buffersink.h or the options system.

       It accepts a pointer to an AVBufferSinkContext structure, which defines the incoming buffers' formats, to
       be passed as the opaque parameter to "avfilter_init_filter" for initialization.

   nullsink
       Null video sink: do absolutely nothing with the input video. It is mainly useful as a  template  and  for
       use in analysis / debugging tools.

MULTIMEDIA FILTERS

Below is a description of the currently available multimedia filters.

a3dscope
Convert input audio to 3d scope video output.

The filter accepts the following options:

rate, r
Set frame rate, expressed as number of frames per second. Default value is "25".

size, s
Specify the video size for the output. For the syntax of this option, check the "Video size" section
in the ffmpeg-utils manual. Default value is "hd720".

fov Set the camera field of view. Default is 90 degrees. Allowed range is from 40 to 150.

roll
Set the camera roll.

pitch
Set the camera pitch.

yaw Set the camera yaw.

xzoom
Set the camera zoom on X-axis.

yzoom
Set the camera zoom on Y-axis.

zzoom
Set the camera zoom on Z-axis.

xpos
Set the camera position on X-axis.

ypos
Set the camera position on Y-axis.

zpos
Set the camera position on Z-axis.

length
Set the length of displayed audio waves in number of frames.

Commands

Filter supports the some above options as commands.

abitscope
Convert input audio to a video output, displaying the audio bit scope.

The filter accepts the following options:

rate, r
Set frame rate, expressed as number of frames per second. Default value is "25".

size, s
Specify the video size for the output. For the syntax of this option, check the "Video size" section
in the ffmpeg-utils manual. Default value is "1024x256".

colors
Specify list of colors separated by space or by '|' which will be used to draw channels. Unrecognized
or missing colors will be replaced by white color.

mode, m
Set output mode. Can be "bars" or "trace". Default is "bars".

adrawgraph
Draw a graph using input audio metadata.

See drawgraph

agraphmonitor
See graphmonitor.

ahistogram
Convert input audio to a video output, displaying the volume histogram.

The filter accepts the following options:

dmode
Specify how histogram is calculated.

It accepts the following values:

single
Use single histogram for all channels.

separate
Use separate histogram for each channel.

Default is "single".

rate, r
Set frame rate, expressed as number of frames per second. Default value is "25".

size, s
Specify the video size for the output. For the syntax of this option, check the "Video size" section
in the ffmpeg-utils manual. Default value is "hd720".

scale
Set display scale.

It accepts the following values:

log logarithmic

sqrt
square root

cbrt
cubic root

lin linear

rlog
reverse logarithmic

Default is "log".

ascale
Set amplitude scale.

It accepts the following values:

log logarithmic

lin linear

Default is "log".

acount
Set how much frames to accumulate in histogram. Default is 1. Setting this to -1 accumulates all
frames.

rheight
Set histogram ratio of window height.

slide
Set sonogram sliding.

It accepts the following values:

replace
replace old rows with new ones.

scroll
scroll from top to bottom.

Default is "replace".

hmode
Set histogram mode.

It accepts the following values:

abs Use absolute values of samples.

sign
Use untouched values of samples.

Default is "abs".

aphasemeter
Measures phase of input audio, which is exported as metadata "lavfi.aphasemeter.phase", representing mean
phase of current audio frame. A video output can also be produced and is enabled by default. The audio is
passed through as first output.

Audio will be rematrixed to stereo if it has a different channel layout. Phase value is in range "[-1,
1]" where -1 means left and right channels are completely out of phase and 1 means channels are in phase.

The filter accepts the following options, all related to its video output:

rate, r
Set the output frame rate. Default value is 25.

size, s
Set the video size for the output. For the syntax of this option, check the "Video size" section in
the ffmpeg-utils manual. Default value is "800x400".

rc
gc
bc Specify the red, green, blue contrast. Default values are 2, 7 and 1. Allowed range is "[0, 255]".

mpc Set color which will be used for drawing median phase. If color is "none" which is default, no median
phase value will be drawn.

video
Enable video output. Default is enabled.

phasing detection

The filter also detects out of phase and mono sequences in stereo streams. It logs the sequence start,
end and duration when it lasts longer or as long as the minimum set.

The filter accepts the following options for this detection:

phasing
Enable mono and out of phase detection. Default is disabled.

tolerance, t
Set phase tolerance for mono detection, in amplitude ratio. Default is 0. Allowed range is "[0, 1]".

angle, a
Set angle threshold for out of phase detection, in degree. Default is 170. Allowed range is "[90,
180]".

duration, d
Set mono or out of phase duration until notification, expressed in seconds. Default is 2.

Examples

• Complete example with ffmpeg to detect 1 second of mono with 0.001 phase tolerance:

ffmpeg -i stereo.wav -af aphasemeter=video=0:phasing=1:duration=1:tolerance=0.001 -f null -

avectorscope
Convert input audio to a video output, representing the audio vector scope.

The filter is used to measure the difference between channels of stereo audio stream. A monaural signal,
consisting of identical left and right signal, results in straight vertical line. Any stereo separation
is visible as a deviation from this line, creating a Lissajous figure. If the straight (or deviation
from it) but horizontal line appears this indicates that the left and right channels are out of phase.

The filter accepts the following options:

mode, m
Set the vectorscope mode.

Available values are:

lissajous
Lissajous rotated by 45 degrees.

lissajous_xy
Same as above but not rotated.

polar
Shape resembling half of circle.

Default value is lissajous.

size, s
Set the video size for the output. For the syntax of this option, check the "Video size" section in
the ffmpeg-utils manual. Default value is "400x400".

rate, r
Set the output frame rate. Default value is 25.

rc
gc
bc
ac Specify the red, green, blue and alpha contrast. Default values are 40, 160, 80 and 255. Allowed
range is "[0, 255]".

rf
gf
bf
af Specify the red, green, blue and alpha fade. Default values are 15, 10, 5 and 5. Allowed range is
"[0, 255]".

zoom
Set the zoom factor. Default value is 1. Allowed range is "[0, 10]". Values lower than 1 will auto
adjust zoom factor to maximal possible value.

draw
Set the vectorscope drawing mode.

Available values are:

dot Draw dot for each sample.

line
Draw line between previous and current sample.

aaline
Draw anti-aliased line between previous and current sample.

Default value is dot.

scale
Specify amplitude scale of audio samples.

Available values are:

lin Linear.

sqrt
Square root.

cbrt
Cubic root.

log Logarithmic.

swap
Swap left channel axis with right channel axis.

mirror
Mirror axis.

none
No mirror.

x Mirror only x axis.

y Mirror only y axis.

xy Mirror both axis.

Examples

• Complete example using ffplay:

ffplay -f lavfi 'amovie=input.mp3, asplit [a][out1];
[a] avectorscope=zoom=1.3:rc=2:gc=200:bc=10:rf=1:gf=8:bf=7 [out0]'

Commands

This filter supports the all above options as commands except options "size" and "rate".

bench, abench
Benchmark part of a filtergraph.

The filter accepts the following options:

action
Start or stop a timer.

Available values are:

start
Get the current time, set it as frame metadata (using the key "lavfi.bench.start_time"), and
forward the frame to the next filter.

stop
Get the current time and fetch the "lavfi.bench.start_time" metadata from the input frame
metadata to get the time difference. Time difference, average, maximum and minimum time
(respectively "t", "avg", "max" and "min") are then printed. The timestamps are expressed in
seconds.

Examples

• Benchmark selectivecolor filter:

bench=start,selectivecolor=reds=-.2 .12 -.49,bench=stop

concat
Concatenate audio and video streams, joining them together one after the other.

The filter works on segments of synchronized video and audio streams. All segments must have the same
number of streams of each type, and that will also be the number of streams at output.

The filter accepts the following options:

n Set the number of segments. Default is 2.

v Set the number of output video streams, that is also the number of video streams in each segment.
Default is 1.

a Set the number of output audio streams, that is also the number of audio streams in each segment.
Default is 0.

unsafe
Activate unsafe mode: do not fail if segments have a different format.

The filter has v+a outputs: first v video outputs, then a audio outputs.

There are nx(v+a) inputs: first the inputs for the first segment, in the same order as the outputs, then
the inputs for the second segment, etc.

Related streams do not always have exactly the same duration, for various reasons including codec frame
size or sloppy authoring. For that reason, related synchronized streams (e.g. a video and its audio
track) should be concatenated at once. The concat filter will use the duration of the longest stream in
each segment (except the last one), and if necessary pad shorter audio streams with silence.

For this filter to work correctly, all segments must start at timestamp 0.

All corresponding streams must have the same parameters in all segments; the filtering system will
automatically select a common pixel format for video streams, and a common sample format, sample rate and
channel layout for audio streams, but other settings, such as resolution, must be converted explicitly by
the user.

Different frame rates are acceptable but will result in variable frame rate at output; be sure to
configure the output file to handle it.

Examples

• Concatenate an opening, an episode and an ending, all in bilingual version (video in stream 0, audio
in streams 1 and 2):

ffmpeg -i opening.mkv -i episode.mkv -i ending.mkv -filter_complex \
'[0:0] [0:1] [0:2] [1:0] [1:1] [1:2] [2:0] [2:1] [2:2]
concat=n=3:v=1:a=2 [v] [a1] [a2]' \
-map '[v]' -map '[a1]' -map '[a2]' output.mkv

• Concatenate two parts, handling audio and video separately, using the (a)movie sources, and adjusting
the resolution:

movie=part1.mp4, scale=512:288 [v1] ; amovie=part1.mp4 [a1] ;
movie=part2.mp4, scale=512:288 [v2] ; amovie=part2.mp4 [a2] ;
[v1] [v2] concat [outv] ; [a1] [a2] concat=v=0:a=1 [outa]

Note that a desync will happen at the stitch if the audio and video streams do not have exactly the
same duration in the first file.

Commands

This filter supports the following commands:

next
Close the current segment and step to the next one

ebur128
EBU R128 scanner filter. This filter takes an audio stream and analyzes its loudness level. By default,
it logs a message at a frequency of 10Hz with the Momentary loudness (identified by "M"), Short-term
loudness ("S"), Integrated loudness ("I") and Loudness Range ("LRA").

The filter can only analyze streams which have sample format is double-precision floating point. The
input stream will be converted to this specification, if needed. Users may need to insert aformat and/or
aresample filters after this filter to obtain the original parameters.

The filter also has a video output (see the video option) with a real time graph to observe the loudness
evolution. The graphic contains the logged message mentioned above, so it is not printed anymore when
this option is set, unless the verbose logging is set. The main graphing area contains the short-term
loudness (3 seconds of analysis), and the gauge on the right is for the momentary loudness (400
milliseconds), but can optionally be configured to instead display short-term loudness (see gauge).

The green area marks a +/- 1LU target range around the target loudness (-23LUFS by default, unless
modified through target).

More information about the Loudness Recommendation EBU R128 on <http://tech.ebu.ch/loudness>.

The filter accepts the following options:

video
Activate the video output. The audio stream is passed unchanged whether this option is set or no. The
video stream will be the first output stream if activated. Default is 0.

size
Set the video size. This option is for video only. For the syntax of this option, check the "Video
size" section in the ffmpeg-utils manual. Default and minimum resolution is "640x480".

meter
Set the EBU scale meter. Default is 9. Common values are 9 and 18, respectively for EBU scale meter
+9 and EBU scale meter +18. Any other integer value between this range is allowed.

metadata
Set metadata injection. If set to 1, the audio input will be segmented into 100ms output frames, each
of them containing various loudness information in metadata. All the metadata keys are prefixed with
"lavfi.r128.".

Default is 0.

framelog
Force the frame logging level.

Available values are:

quiet
logging disabled

info
information logging level

verbose
verbose logging level

By default, the logging level is set to info. If the video or the metadata options are set, it
switches to verbose.

peak
Set peak mode(s).

Available modes can be cumulated (the option is a "flag" type). Possible values are:

none
Disable any peak mode (default).

sample
Enable sample-peak mode.

Simple peak mode looking for the higher sample value. It logs a message for sample-peak
(identified by "SPK").

true
Enable true-peak mode.

If enabled, the peak lookup is done on an over-sampled version of the input stream for better
peak accuracy. It logs a message for true-peak. (identified by "TPK") and true-peak per frame
(identified by "FTPK"). This mode requires a build with "libswresample".

dualmono
Treat mono input files as "dual mono". If a mono file is intended for playback on a stereo system,
its EBU R128 measurement will be perceptually incorrect. If set to "true", this option will
compensate for this effect. Multi-channel input files are not affected by this option.

panlaw
Set a specific pan law to be used for the measurement of dual mono files. This parameter is
optional, and has a default value of -3.01dB.

target
Set a specific target level (in LUFS) used as relative zero in the visualization. This parameter is
optional and has a default value of -23LUFS as specified by EBU R128. However, material published
online may prefer a level of -16LUFS (e.g. for use with podcasts or video platforms).

gauge
Set the value displayed by the gauge. Valid values are "momentary" and s "shortterm". By default the
momentary value will be used, but in certain scenarios it may be more useful to observe the short
term value instead (e.g. live mixing).

scale
Sets the display scale for the loudness. Valid parameters are "absolute" (in LUFS) or "relative" (LU)
relative to the target. This only affects the video output, not the summary or continuous log output.

integrated
Read-only exported value for measured integrated loudness, in LUFS.

range
Read-only exported value for measured loudness range, in LU.

lra_low
Read-only exported value for measured LRA low, in LUFS.

lra_high
Read-only exported value for measured LRA high, in LUFS.

sample_peak
Read-only exported value for measured sample peak, in dBFS.

true_peak
Read-only exported value for measured true peak, in dBFS.

Examples

• Real-time graph using ffplay, with a EBU scale meter +18:

ffplay -f lavfi -i "amovie=input.mp3,ebur128=video=1:meter=18 [out0][out1]"

• Run an analysis with ffmpeg:

ffmpeg -nostats -i input.mp3 -filter_complex ebur128 -f null -

interleave, ainterleave
Temporally interleave frames from several inputs.

"interleave" works with video inputs, "ainterleave" with audio.

These filters read frames from several inputs and send the oldest queued frame to the output.

Input streams must have well defined, monotonically increasing frame timestamp values.

In order to submit one frame to output, these filters need to enqueue at least one frame for each input,
so they cannot work in case one input is not yet terminated and will not receive incoming frames.

For example consider the case when one input is a "select" filter which always drops input frames. The
"interleave" filter will keep reading from that input, but it will never be able to send new frames to
output until the input sends an end-of-stream signal.

Also, depending on inputs synchronization, the filters will drop frames in case one input receives more
frames than the other ones, and the queue is already filled.

These filters accept the following options:

nb_inputs, n
Set the number of different inputs, it is 2 by default.

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.

Examples

• Interleave frames belonging to different streams using ffmpeg:

ffmpeg -i bambi.avi -i pr0n.mkv -filter_complex "[0:v][1:v] interleave" out.avi

• Add flickering blur effect:

select='if(gt(random(0), 0.2), 1, 2)':n=2 [tmp], boxblur=2:2, [tmp] interleave

latency, alatency
Measure filtering latency.

Report previous filter filtering latency, delay in number of audio samples for audio filters or number of
video frames for video filters.

On end of input stream, filter will report min and max measured latency for previous running filter in
filtergraph.

metadata, ametadata
Manipulate frame metadata.

This filter accepts the following options:

mode
Set mode of operation of the filter.

Can be one of the following:

select
If both "value" and "key" is set, select frames which have such metadata. If only "key" is set,
select every frame that has such key in metadata.

add Add new metadata "key" and "value". If key is already available do nothing.

modify
Modify value of already present key.

delete
If "value" is set, delete only keys that have such value. Otherwise, delete key. If "key" is not
set, delete all metadata values in the frame.

print
Print key and its value if metadata was found. If "key" is not set print all metadata values
available in frame.

key Set key used with all modes. Must be set for all modes except "print" and "delete".

value
Set metadata value which will be used. This option is mandatory for "modify" and "add" mode.

function
Which function to use when comparing metadata value and "value".

Can be one of following:

same_str
Values are interpreted as strings, returns true if metadata value is same as "value".

starts_with
Values are interpreted as strings, returns true if metadata value starts with the "value" option
string.

less
Values are interpreted as floats, returns true if metadata value is less than "value".

equal
Values are interpreted as floats, returns true if "value" is equal with metadata value.

greater
Values are interpreted as floats, returns true if metadata value is greater than "value".

expr
Values are interpreted as floats, returns true if expression from option "expr" evaluates to
true.

ends_with
Values are interpreted as strings, returns true if metadata value ends with the "value" option
string.

expr
Set expression which is used when "function" is set to "expr". The expression is evaluated through
the eval API and can contain the following constants:

VALUE1, FRAMEVAL
Float representation of "value" from metadata key.

VALUE2, USERVAL
Float representation of "value" as supplied by user in "value" option.

file
If specified in "print" mode, output is written to the named file. Instead of plain filename any
writable url can be specified. Filename ``-'' is a shorthand for standard output. If "file" option is
not set, output is written to the log with AV_LOG_INFO loglevel.

direct
Reduces buffering in print mode when output is written to a URL set using file.

Examples

• Print all metadata values for frames with key "lavfi.signalstats.YDIF" with values between 0 and 1.

signalstats,metadata=print:key=lavfi.signalstats.YDIF:value=0:function=expr:expr='between(VALUE1,0,1)'

• Print silencedetect output to file metadata.txt.

silencedetect,ametadata=mode=print:file=metadata.txt

• Direct all metadata to a pipe with file descriptor 4.

metadata=mode=print:file='pipe\:4'

perms, aperms
Set read/write permissions for the output frames.

These filters are mainly aimed at developers to test direct path in the following filter in the
filtergraph.

The filters accept the following options:

mode
Select the permissions mode.

It accepts the following values:

none
Do nothing. This is the default.

ro Set all the output frames read-only.

rw Set all the output frames directly writable.

toggle
Make the frame read-only if writable, and writable if read-only.

random
Set each output frame read-only or writable randomly.

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

Note: in case of auto-inserted filter between the permission filter and the following one, the permission
might not be received as expected in that following filter. Inserting a format or aformat filter before
the perms/aperms filter can avoid this problem.

realtime, arealtime
Slow down filtering to match real time approximately.

These filters will pause the filtering for a variable amount of time to match the output rate with the
input timestamps. They are similar to the re option to "ffmpeg".

They accept the following options:

limit
Time limit for the pauses. Any pause longer than that will be considered a timestamp discontinuity
and reset the timer. Default is 2 seconds.

speed
Speed factor for processing. The value must be a float larger than zero. Values larger than 1.0 will
result in faster than realtime processing, smaller will slow processing down. The limit is
automatically adapted accordingly. Default is 1.0.

A processing speed faster than what is possible without these filters cannot be achieved.

Commands

Both filters supports the all above options as commands.

segment, asegment
Split single input stream into multiple streams.

This filter does opposite of concat filters.

"segment" works on video frames, "asegment" on audio samples.

This filter accepts the following options:

timestamps
Timestamps of output segments separated by '|'. The first segment will run from the beginning of the
input stream. The last segment will run until the end of the input stream

frames, samples
Exact frame/sample count to split the segments.

In all cases, prefixing an each segment with '+' will make it relative to the previous segment.

Examples

• Split input audio stream into three output audio streams, starting at start of input audio stream and
storing that in 1st output audio stream, then following at 60th second and storing than in 2nd output
audio stream, and last after 150th second of input audio stream store in 3rd output audio stream:

asegment=timestamps="60|150"

select, aselect
Select frames to pass in output.

This filter accepts the following options:

expr, e
Set expression, which is evaluated for each input frame.

If the expression is evaluated to zero, the frame is discarded.

If the evaluation result is negative or NaN, the frame is sent to the first output; otherwise it is
sent to the output with index "ceil(val)-1", assuming that the input index starts from 0.

For example a value of 1.2 corresponds to the output with index "ceil(1.2)-1 = 2-1 = 1", that is the
second output.

outputs, n
Set the number of outputs. The output to which to send the selected frame is based on the result of
the evaluation. Default value is 1.

The expression can contain the following constants:

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

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

prev_selected_n
The sequential number of the last selected frame. It's NAN if undefined.

TB The timebase of the input timestamps.

pts The PTS (Presentation TimeStamp) of the filtered frame, expressed in TB units. It's NAN if undefined.

t The PTS of the filtered frame, expressed in seconds. It's NAN if undefined.

prev_pts
The PTS of the previously filtered frame. It's NAN if undefined.

prev_selected_pts
The PTS of the last previously filtered frame. It's NAN if undefined.

prev_selected_t
The PTS of the last previously selected frame, expressed in seconds. It's NAN if undefined.

start_pts
The first PTS in the stream which is not NAN. It remains NAN if not found.

start_t
The first PTS, in seconds, in the stream which is not NAN. It remains NAN if not found.

pict_type (video only)
The type of the filtered frame. It can assume one of the following values:

I
P
B
S
SI
SP
BI
interlace_type (video only)
The frame interlace type. It can assume one of the following values:

PROGRESSIVE
The frame is progressive (not interlaced).

TOPFIRST
The frame is top-field-first.

BOTTOMFIRST
The frame is bottom-field-first.

consumed_sample_n (audio only)
the number of selected samples before the current frame

samples_n (audio only)
the number of samples in the current frame

sample_rate (audio only)
the input sample rate

key This is 1 if the filtered frame is a key-frame, 0 otherwise.

pos the position in the file of the filtered frame, -1 if the information is not available (e.g. for
synthetic video); deprecated, do not use

scene (video only)
value between 0 and 1 to indicate a new scene; a low value reflects a low probability for the current
frame to introduce a new scene, while a higher value means the current frame is more likely to be one
(see the example below)

concatdec_select
The concat demuxer can select only part of a concat input file by setting an inpoint and an outpoint,
but the output packets may not be entirely contained in the selected interval. By using this
variable, it is possible to skip frames generated by the concat demuxer which are not exactly
contained in the selected interval.

This works by comparing the frame pts against the lavf.concat.start_time and the lavf.concat.duration
packet metadata values which are also present in the decoded frames.

The concatdec_select variable is -1 if the frame pts is at least start_time and either the duration
metadata is missing or the frame pts is less than start_time + duration, 0 otherwise, and NaN if the
start_time metadata is missing.

That basically means that an input frame is selected if its pts is within the interval set by the
concat demuxer.

The default value of the select expression is "1".

Examples

• Select all frames in input:

select

The example above is the same as:

select=1

• Skip all frames:

select=0

• Select only I-frames:

select='eq(pict_type\,I)'

• Select one frame every 100:

select='not(mod(n\,100))'

• Select only frames contained in the 10-20 time interval:

select=between(t\,10\,20)

• Select only I-frames contained in the 10-20 time interval:

select=between(t\,10\,20)*eq(pict_type\,I)

• Select frames with a minimum distance of 10 seconds:

select='isnan(prev_selected_t)+gte(t-prev_selected_t\,10)'

• Use aselect to select only audio frames with samples number > 100:

aselect='gt(samples_n\,100)'

• Create a mosaic of the first scenes:

ffmpeg -i video.avi -vf select='gt(scene\,0.4)',scale=160:120,tile -frames:v 1 preview.png

Comparing scene against a value between 0.3 and 0.5 is generally a sane choice.

• Send even and odd frames to separate outputs, and compose them:

select=n=2:e='mod(n, 2)+1' [odd][even]; [odd] pad=h=2*ih [tmp]; [tmp][even] overlay=y=h

• Select useful frames from an ffconcat file which is using inpoints and outpoints but where the source
files are not intra frame only.

ffmpeg -copyts -vsync 0 -segment_time_metadata 1 -i input.ffconcat -vf select=concatdec_select -af aselect=concatdec_select output.avi

sendcmd, asendcmd
Send commands to filters in the filtergraph.

These filters read commands to be sent to other filters in the filtergraph.

"sendcmd" must be inserted between two video filters, "asendcmd" must be inserted between two audio
filters, but apart from that they act the same way.

The specification of commands can be provided in the filter arguments with the commands option, or in a
file specified by the filename option.

These filters accept the following options:

commands, c
Set the commands to be read and sent to the other filters.

filename, f
Set the filename of the commands to be read and sent to the other filters.

Commands syntax

A commands description consists of a sequence of interval specifications, comprising a list of commands
to be executed when a particular event related to that interval occurs. The occurring event is typically
the current frame time entering or leaving a given time interval.

An interval is specified by the following syntax:

<START>[-<END>] <COMMANDS>;

The time interval is specified by the START and END times. END is optional and defaults to the maximum
time.

The current frame time is considered within the specified interval if it is included in the interval
[START, END), that is when the time is greater or equal to START and is lesser than END.

COMMANDS consists of a sequence of one or more command specifications, separated by ",", relating to that
interval. The syntax of a command specification is given by:

[<FLAGS>] <TARGET> <COMMAND> <ARG>

FLAGS is optional and specifies the type of events relating to the time interval which enable sending the
specified command, and must be a non-null sequence of identifier flags separated by "+" or "|" and
enclosed between "[" and "]".

The following flags are recognized:

enter
The command is sent when the current frame timestamp enters the specified interval. In other words,
the command is sent when the previous frame timestamp was not in the given interval, and the current
is.

leave
The command is sent when the current frame timestamp leaves the specified interval. In other words,
the command is sent when the previous frame timestamp was in the given interval, and the current is
not.

expr
The command ARG is interpreted as expression and result of expression is passed as ARG.

The expression is evaluated through the eval API and can contain the following constants:

POS Original position in the file of the frame, or undefined if undefined for the current frame.
Deprecated, do not use.

PTS The presentation timestamp in input.

N The count of the input frame for video or audio, starting from 0.

T The time in seconds of the current frame.

TS The start time in seconds of the current command interval.

TE The end time in seconds of the current command interval.

TI The interpolated time of the current command interval, TI = (T - TS) / (TE - TS).

W The video frame width.

H The video frame height.

If FLAGS is not specified, a default value of "[enter]" is assumed.

TARGET specifies the target of the command, usually the name of the filter class or a specific filter
instance name.

COMMAND specifies the name of the command for the target filter.

ARG is optional and specifies the optional list of argument for the given COMMAND.

Between one interval specification and another, whitespaces, or sequences of characters starting with "#"
until the end of line, are ignored and can be used to annotate comments.

A simplified BNF description of the commands specification syntax follows:

<COMMAND_FLAG> ::= "enter" | "leave"
<COMMAND_FLAGS> ::= <COMMAND_FLAG> [(+|"|")<COMMAND_FLAG>]
<COMMAND> ::= ["[" <COMMAND_FLAGS> "]"] <TARGET> <COMMAND> [<ARG>]
<COMMANDS> ::= <COMMAND> [,<COMMANDS>]
<INTERVAL> ::= <START>[-<END>] <COMMANDS>
<INTERVALS> ::= <INTERVAL>[;<INTERVALS>]

Examples

• Specify audio tempo change at second 4:

asendcmd=c='4.0 atempo tempo 1.5',atempo

• Target a specific filter instance:

asendcmd=c='4.0 atempo@my tempo 1.5',atempo@my

• Specify a list of drawtext and hue commands in a file.

# show text in the interval 5-10
5.0-10.0 [enter] drawtext reinit 'fontfile=FreeSerif.ttf:text=hello world',
[leave] drawtext reinit 'fontfile=FreeSerif.ttf:text=';

# desaturate the image in the interval 15-20
15.0-20.0 [enter] hue s 0,
[enter] drawtext reinit 'fontfile=FreeSerif.ttf:text=nocolor',
[leave] hue s 1,
[leave] drawtext reinit 'fontfile=FreeSerif.ttf:text=color';

# apply an exponential saturation fade-out effect, starting from time 25
25 [enter] hue s exp(25-t)

A filtergraph allowing to read and process the above command list stored in a file test.cmd, can be
specified with:

sendcmd=f=test.cmd,drawtext=fontfile=FreeSerif.ttf:text='',hue

setpts, asetpts
Change the PTS (presentation timestamp) of the input frames.

"setpts" works on video frames, "asetpts" on audio frames.

This filter accepts the following options:

expr
The expression which is evaluated for each frame to construct its timestamp.

The expression is evaluated through the eval API and can contain the following constants:

FRAME_RATE, FR
frame rate, only defined for constant frame-rate video

PTS The presentation timestamp in input

N The count of the input frame for video or the number of consumed samples, not including the current
frame for audio, starting from 0.

NB_CONSUMED_SAMPLES
The number of consumed samples, not including the current frame (only audio)

NB_SAMPLES, S
The number of samples in the current frame (only audio)

SAMPLE_RATE, SR
The audio sample rate.

STARTPTS
The PTS of the first frame.

STARTT
the time in seconds of the first frame

INTERLACED
State whether the current frame is interlaced.

T the time in seconds of the current frame

POS original position in the file of the frame, or undefined if undefined for the current frame;
deprecated, do not use

PREV_INPTS
The previous input PTS.

PREV_INT
previous input time in seconds

PREV_OUTPTS
The previous output PTS.

PREV_OUTT
previous output time in seconds

RTCTIME
The wallclock (RTC) time in microseconds. This is deprecated, use time(0) instead.

RTCSTART
The wallclock (RTC) time at the start of the movie in microseconds.

TB The timebase of the input timestamps.

T_CHANGE
Time of the first frame after command was applied or time of the first frame if no commands.

Examples

• Start counting PTS from zero

setpts=PTS-STARTPTS

• Apply fast motion effect:

setpts=0.5*PTS

• Apply slow motion effect:

setpts=2.0*PTS

• Set fixed rate of 25 frames per second:

setpts=N/(25*TB)

• Set fixed rate 25 fps with some jitter:

setpts='1/(25*TB) * (N + 0.05 * sin(N*2*PI/25))'

• Apply an offset of 10 seconds to the input PTS:

setpts=PTS+10/TB

• Generate timestamps from a "live source" and rebase onto the current timebase:

setpts='(RTCTIME - RTCSTART) / (TB * 1000000)'

• Generate timestamps by counting samples:

asetpts=N/SR/TB

Commands

Both filters support all above options as commands.

setrange
Force color range for the output video frame.

The "setrange" filter marks the color range property for the output frames. It does not change the input
frame, but only sets the corresponding property, which affects how the frame is treated by following
filters.

The filter accepts the following options:

range
Available values are:

auto
Keep the same color range property.

unspecified, unknown
Set the color range as unspecified.

limited, tv, mpeg
Set the color range as limited.

full, pc, jpeg
Set the color range as full.

settb, asettb
Set the timebase to use for the output frames timestamps. It is mainly useful for testing timebase
configuration.

It accepts the following parameters:

expr, tb
The expression which is evaluated into the output timebase.

The value for tb is an arithmetic expression representing a rational. The expression can contain the
constants "AVTB" (the default timebase), "intb" (the input timebase) and "sr" (the sample rate, audio
only). Default value is "intb".

Examples

• Set the timebase to 1/25:

settb=expr=1/25

• Set the timebase to 1/10:

settb=expr=0.1

• Set the timebase to 1001/1000:

settb=1+0.001

• Set the timebase to 2*intb:

settb=2*intb

• Set the default timebase value:

settb=AVTB

showcqt
Convert input audio to a video output representing frequency spectrum logarithmically using Brown-
Puckette constant Q transform algorithm with direct frequency domain coefficient calculation (but the
transform itself is not really constant Q, instead the Q factor is actually variable/clamped), with
musical tone scale, from E0 to D#10.

The filter accepts the following options:

size, s
Specify the video size for the output. It must be even. For the syntax of this option, check the
"Video size" section in the ffmpeg-utils manual. Default value is "1920x1080".

fps, rate, r
Set the output frame rate. Default value is 25.

bar_h
Set the bargraph height. It must be even. Default value is -1 which computes the bargraph height
automatically.

axis_h
Set the axis height. It must be even. Default value is -1 which computes the axis height
automatically.

sono_h
Set the sonogram height. It must be even. Default value is -1 which computes the sonogram height
automatically.

fullhd
Set the fullhd resolution. This option is deprecated, use size, s instead. Default value is 1.

sono_v, volume
Specify the sonogram volume expression. It can contain variables:

bar_v
the bar_v evaluated expression

frequency, freq, f
the frequency where it is evaluated

timeclamp, tc
the value of timeclamp option

and functions:

a_weighting(f)
A-weighting of equal loudness

b_weighting(f)
B-weighting of equal loudness

c_weighting(f)
C-weighting of equal loudness.

Default value is 16.

bar_v, volume2
Specify the bargraph volume expression. It can contain variables:

sono_v
the sono_v evaluated expression

frequency, freq, f
the frequency where it is evaluated

timeclamp, tc
the value of timeclamp option

and functions:

a_weighting(f)
A-weighting of equal loudness

b_weighting(f)
B-weighting of equal loudness

c_weighting(f)
C-weighting of equal loudness.

Default value is "sono_v".

sono_g, gamma
Specify the sonogram gamma. Lower gamma makes the spectrum more contrast, higher gamma makes the
spectrum having more range. Default value is 3. Acceptable range is "[1, 7]".

bar_g, gamma2
Specify the bargraph gamma. Default value is 1. Acceptable range is "[1, 7]".

bar_t
Specify the bargraph transparency level. Lower value makes the bargraph sharper. Default value is 1.
Acceptable range is "[0, 1]".

timeclamp, tc
Specify the transform timeclamp. At low frequency, there is trade-off between accuracy in time domain
and frequency domain. If timeclamp is lower, event in time domain is represented more accurately
(such as fast bass drum), otherwise event in frequency domain is represented more accurately (such as
bass guitar). Acceptable range is "[0.002, 1]". Default value is 0.17.

attack
Set attack time in seconds. The default is 0 (disabled). Otherwise, it limits future samples by
applying asymmetric windowing in time domain, useful when low latency is required. Accepted range is
"[0, 1]".

basefreq
Specify the transform base frequency. Default value is 20.01523126408007475, which is frequency 50
cents below E0. Acceptable range is "[10, 100000]".

endfreq
Specify the transform end frequency. Default value is 20495.59681441799654, which is frequency 50
cents above D#10. Acceptable range is "[10, 100000]".

coeffclamp
This option is deprecated and ignored.

tlength
Specify the transform length in time domain. Use this option to control accuracy trade-off between
time domain and frequency domain at every frequency sample. It can contain variables:

frequency, freq, f
the frequency where it is evaluated

timeclamp, tc
the value of timeclamp option.

Default value is "384*tc/(384+tc*f)".

count
Specify the transform count for every video frame. Default value is 6. Acceptable range is "[1,
30]".

fcount
Specify the transform count for every single pixel. Default value is 0, which makes it computed
automatically. Acceptable range is "[0, 10]".

fontfile
Specify font file for use with freetype to draw the axis. If not specified, use embedded font. Note
that drawing with font file or embedded font is not implemented with custom basefreq and endfreq, use
axisfile option instead.

font
Specify fontconfig pattern. This has lower priority than fontfile. The ":" in the pattern may be
replaced by "|" to avoid unnecessary escaping.

fontcolor
Specify font color expression. This is arithmetic expression that should return integer value
0xRRGGBB. It can contain variables:

frequency, freq, f
the frequency where it is evaluated

timeclamp, tc
the value of timeclamp option

and functions:

midi(f)
midi number of frequency f, some midi numbers: E0(16), C1(24), C2(36), A4(69)

r(x), g(x), b(x)
red, green, and blue value of intensity x.

Default value is "st(0, (midi(f)-59.5)/12); st(1, if(between(ld(0),0,1), 0.5-0.5*cos(2*PI*ld(0)),
0)); r(1-ld(1)) + b(ld(1))".

axisfile
Specify image file to draw the axis. This option override fontfile and fontcolor option.

axis, text
Enable/disable drawing text to the axis. If it is set to 0, drawing to the axis is disabled, ignoring
fontfile and axisfile option. Default value is 1.

csp Set colorspace. The accepted values are:

unspecified
Unspecified (default)

bt709
BT.709

fcc FCC

bt470bg
BT.470BG or BT.601-6 625

smpte170m
SMPTE-170M or BT.601-6 525

smpte240m
SMPTE-240M

bt2020ncl
BT.2020 with non-constant luminance

cscheme
Set spectrogram color scheme. This is list of floating point values with format
"left_r|left_g|left_b|right_r|right_g|right_b". The default is "1|0.5|0|0|0.5|1".

Examples

• Playing audio while showing the spectrum:

ffplay -f lavfi 'amovie=a.mp3, asplit [a][out1]; [a] showcqt [out0]'

• Same as above, but with frame rate 30 fps:

ffplay -f lavfi 'amovie=a.mp3, asplit [a][out1]; [a] showcqt=fps=30:count=5 [out0]'

• Playing at 1280x720:

ffplay -f lavfi 'amovie=a.mp3, asplit [a][out1]; [a] showcqt=s=1280x720:count=4 [out0]'

• Disable sonogram display:

sono_h=0

• A1 and its harmonics: A1, A2, (near)E3, A3:

ffplay -f lavfi 'aevalsrc=0.1*sin(2*PI*55*t)+0.1*sin(4*PI*55*t)+0.1*sin(6*PI*55*t)+0.1*sin(8*PI*55*t),
asplit[a][out1]; [a] showcqt [out0]'

• Same as above, but with more accuracy in frequency domain:

ffplay -f lavfi 'aevalsrc=0.1*sin(2*PI*55*t)+0.1*sin(4*PI*55*t)+0.1*sin(6*PI*55*t)+0.1*sin(8*PI*55*t),
asplit[a][out1]; [a] showcqt=timeclamp=0.5 [out0]'

• Custom volume:

bar_v=10:sono_v=bar_v*a_weighting(f)

• Custom gamma, now spectrum is linear to the amplitude.

bar_g=2:sono_g=2

• Custom tlength equation:

tc=0.33:tlength='st(0,0.17); 384*tc / (384 / ld(0) + tc*f /(1-ld(0))) + 384*tc / (tc*f / ld(0) + 384 /(1-ld(0)))'

• Custom fontcolor and fontfile, C-note is colored green, others are colored blue:

fontcolor='if(mod(floor(midi(f)+0.5),12), 0x0000FF, g(1))':fontfile=myfont.ttf

• Custom font using fontconfig:

font='Courier New,Monospace,mono|bold'

• Custom frequency range with custom axis using image file:

axisfile=myaxis.png:basefreq=40:endfreq=10000

showcwt
Convert input audio to video output representing frequency spectrum using Continuous Wavelet Transform
and Morlet wavelet.

The filter accepts the following options:

size, s
Specify the video size for the output. For the syntax of this option, check the "Video size" section
in the ffmpeg-utils manual. Default value is "640x512".

rate, r
Set the output frame rate. Default value is 25.

scale
Set the frequency scale used. Allowed values are:

linear
log
bark
mel
erbs
sqrt
cbrt
qdrt

Default value is "linear".

iscale
Set the intensity scale used. Allowed values are:

linear
log
sqrt
cbrt
qdrt

Default value is "log".

min Set the minimum frequency that will be used in output. Default is 20 Hz.

max Set the maximum frequency that will be used in output. Default is 20000 Hz. The real frequency upper
limit depends on input audio's sample rate and such will be enforced on this value when it is set to
value greater than Nyquist frequency.

imin
Set the minimum intensity that will be used in output.

imax
Set the maximum intensity that will be used in output.

logb
Set the logarithmic basis for brightness strength when mapping calculated magnitude values to pixel
values. Allowed range is from 0 to 1. Default value is 0.0001.

deviation
Set the frequency deviation. Lower values than 1 are more frequency oriented, while higher values
than 1 are more time oriented. Allowed range is from 0 to 10. Default value is 1.

pps Set the number of pixel output per each second in one row. Allowed range is from 1 to 1024. Default
value is 64.

mode
Set the output visual mode. Allowed values are:

magnitude
Show magnitude.

phase
Show only phase.

magphase
Show combination of magnitude and phase. Magnitude is mapped to brightness and phase to color.

channel
Show unique color per channel magnitude.

stereo
Show unique color per stereo difference.

Default value is "magnitude".

slide
Set the output slide method. Allowed values are:

replace
scroll
frame
direction
Set the direction method for output slide method. Allowed values are:

lr Direction from left to right.

rl Direction from right to left.

ud Direction from up to down.

du Direction from down to up.

bar Set the ratio of bargraph display to display size. Default is 0.

rotation
Set color rotation, must be in [-1.0, 1.0] range. Default value is 0.

showfreqs
Convert input audio to video output representing the audio power spectrum. Audio amplitude is on Y-axis
while frequency is on X-axis.

The filter accepts the following options:

size, s
Specify size of video. For the syntax of this option, check the "Video size" section in the ffmpeg-
utils manual. Default is "1024x512".

rate, r
Set video rate. Default is 25.

mode
Set display mode. This set how each frequency bin will be represented.

It accepts the following values:

line
bar
dot

Default is "bar".

ascale
Set amplitude scale.

It accepts the following values:

lin Linear scale.

sqrt
Square root scale.

cbrt
Cubic root scale.

log Logarithmic scale.

Default is "log".

fscale
Set frequency scale.

It accepts the following values:

lin Linear scale.

log Logarithmic scale.

rlog
Reverse logarithmic scale.

Default is "lin".

win_size
Set window size. Allowed range is from 16 to 65536.

Default is 2048

win_func
Set windowing function.

It accepts the following values:

rect
bartlett
hanning
hamming
blackman
welch
flattop
bharris
bnuttall
bhann
sine
nuttall
lanczos
gauss
tukey
dolph
cauchy
parzen
poisson
bohman
kaiser

Default is "hanning".

overlap
Set window overlap. In range "[0, 1]". Default is 1, which means optimal overlap for selected window
function will be picked.

averaging
Set time averaging. Setting this to 0 will display current maximal peaks. Default is 1, which means
time averaging is disabled.

colors
Specify list of colors separated by space or by '|' which will be used to draw channel frequencies.
Unrecognized or missing colors will be replaced by white color.

cmode
Set channel display mode.

It accepts the following values:

combined
separate

Default is "combined".

minamp
Set minimum amplitude used in "log" amplitude scaler.

data
Set data display mode.

It accepts the following values:

magnitude
phase
delay

Default is "magnitude".

channels
Set channels to use when processing audio. By default all are processed.

showspatial
Convert stereo input audio to a video output, representing the spatial relationship between two channels.

The filter accepts the following options:

size, s
Specify the video size for the output. For the syntax of this option, check the "Video size" section
in the ffmpeg-utils manual. Default value is "512x512".

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 value is "hann".

rate, r
Set output framerate.

showspectrum
Convert input audio to a video output, representing the audio frequency spectrum.

The filter accepts the following options:

size, s
Specify the video size for the output. For the syntax of this option, check the "Video size" section
in the ffmpeg-utils manual. Default value is "640x512".

slide
Specify how the spectrum should slide along the window.

It accepts the following values:

replace
the samples start again on the left when they reach the right

scroll
the samples scroll from right to left

fullframe
frames are only produced when the samples reach the right

rscroll
the samples scroll from left to right

lreplace
the samples start again on the right when they reach the left

Default value is "replace".

mode
Specify display mode.

It accepts the following values:

combined
all channels are displayed in the same row

separate
all channels are displayed in separate rows

Default value is combined.

color
Specify display color mode.

It accepts the following values:

channel
each channel is displayed in a separate color

intensity
each channel is displayed using the same color scheme

rainbow
each channel is displayed using the rainbow color scheme

moreland
each channel is displayed using the moreland color scheme

nebulae
each channel is displayed using the nebulae color scheme

fire
each channel is displayed using the fire color scheme

fiery
each channel is displayed using the fiery color scheme

fruit
each channel is displayed using the fruit color scheme

cool
each channel is displayed using the cool color scheme

magma
each channel is displayed using the magma color scheme

green
each channel is displayed using the green color scheme

viridis
each channel is displayed using the viridis color scheme

plasma
each channel is displayed using the plasma color scheme

cividis
each channel is displayed using the cividis color scheme

terrain
each channel is displayed using the terrain color scheme

Default value is channel.

scale
Specify scale used for calculating intensity color values.

It accepts the following values:

lin linear

sqrt
square root, default

cbrt
cubic root

log logarithmic

4thrt
4th root

5thrt
5th root

Default value is sqrt.

fscale
Specify frequency scale.

It accepts the following values:

lin linear

log logarithmic

Default value is lin.

saturation
Set saturation modifier for displayed colors. Negative values provide alternative color scheme. 0 is
no saturation at all. Saturation must be in [-10.0, 10.0] range. Default value is 1.

win_func
Set window function.

It accepts the following values:

rect
bartlett
hann
hanning
hamming
blackman
welch
flattop
bharris
bnuttall
bhann
sine
nuttall
lanczos
gauss
tukey
dolph
cauchy
parzen
poisson
bohman
kaiser

Default value is "hann".

orientation
Set orientation of time vs frequency axis. Can be "vertical" or "horizontal". Default is "vertical".

overlap
Set ratio of overlap window. Default value is 0. When value is 1 overlap is set to recommended size
for specific window function currently used.

gain
Set scale gain for calculating intensity color values. Default value is 1.

data
Set which data to display. Can be "magnitude", default or "phase", or unwrapped phase: "uphase".

rotation
Set color rotation, must be in [-1.0, 1.0] range. Default value is 0.

start
Set start frequency from which to display spectrogram. Default is 0.

stop
Set stop frequency to which to display spectrogram. Default is 0.

fps Set upper frame rate limit. Default is "auto", unlimited.

legend
Draw time and frequency axes and legends. Default is disabled.

drange
Set dynamic range used to calculate intensity color values. Default is 120 dBFS. Allowed range is
from 10 to 200.

limit
Set upper limit of input audio samples volume in dBFS. Default is 0 dBFS. Allowed range is from -100
to 100.

opacity
Set opacity strength when using pixel format output with alpha component.

The usage is very similar to the showwaves filter; see the examples in that section.

Examples

• Large window with logarithmic color scaling:

showspectrum=s=1280x480:scale=log

• Complete example for a colored and sliding spectrum per channel using ffplay:

ffplay -f lavfi 'amovie=input.mp3, asplit [a][out1];
[a] showspectrum=mode=separate:color=intensity:slide=1:scale=cbrt [out0]'

showspectrumpic
Convert input audio to a single video frame, representing the audio frequency spectrum.

The filter accepts the following options:

size, s
Specify the video size for the output. For the syntax of this option, check the "Video size" section
in the ffmpeg-utils manual. Default value is "4096x2048".

mode
Specify display mode.

It accepts the following values:

combined
all channels are displayed in the same row

separate
all channels are displayed in separate rows

Default value is combined.

color
Specify display color mode.

It accepts the following values:

channel
each channel is displayed in a separate color

intensity
each channel is displayed using the same color scheme

rainbow
each channel is displayed using the rainbow color scheme

moreland
each channel is displayed using the moreland color scheme

nebulae
each channel is displayed using the nebulae color scheme

fire
each channel is displayed using the fire color scheme

fiery
each channel is displayed using the fiery color scheme

fruit
each channel is displayed using the fruit color scheme

cool
each channel is displayed using the cool color scheme

magma
each channel is displayed using the magma color scheme

green
each channel is displayed using the green color scheme

viridis
each channel is displayed using the viridis color scheme

plasma
each channel is displayed using the plasma color scheme

cividis
each channel is displayed using the cividis color scheme

terrain
each channel is displayed using the terrain color scheme

Default value is intensity.

scale
Specify scale used for calculating intensity color values.

It accepts the following values:

lin linear

sqrt
square root, default

cbrt
cubic root

log logarithmic

4thrt
4th root

5thrt
5th root

Default value is log.

fscale
Specify frequency scale.

It accepts the following values:

lin linear

log logarithmic

Default value is lin.

saturation
Set saturation modifier for displayed colors. Negative values provide alternative color scheme. 0 is
no saturation at all. Saturation must be in [-10.0, 10.0] range. Default value is 1.

win_func
Set window function.

It accepts the following values:

rect
bartlett
hann
hanning
hamming
blackman
welch
flattop
bharris
bnuttall
bhann
sine
nuttall
lanczos
gauss
tukey
dolph
cauchy
parzen
poisson
bohman
kaiser

Default value is "hann".

orientation
Set orientation of time vs frequency axis. Can be "vertical" or "horizontal". Default is "vertical".

gain
Set scale gain for calculating intensity color values. Default value is 1.

legend
Draw time and frequency axes and legends. Default is enabled.

rotation
Set color rotation, must be in [-1.0, 1.0] range. Default value is 0.

start
Set start frequency from which to display spectrogram. Default is 0.

stop
Set stop frequency to which to display spectrogram. Default is 0.

drange
Set dynamic range used to calculate intensity color values. Default is 120 dBFS. Allowed range is
from 10 to 200.

limit
Set upper limit of input audio samples volume in dBFS. Default is 0 dBFS. Allowed range is from -100
to 100.

opacity
Set opacity strength when using pixel format output with alpha component.

Examples

• Extract an audio spectrogram of a whole audio track in a 1024x1024 picture using ffmpeg:

ffmpeg -i audio.flac -lavfi showspectrumpic=s=1024x1024 spectrogram.png

showvolume
Convert input audio volume to a video output.

The filter accepts the following options:

rate, r
Set video rate.

b Set border width, allowed range is [0, 5]. Default is 1.

w Set channel width, allowed range is [80, 8192]. Default is 400.

h Set channel height, allowed range is [1, 900]. Default is 20.

f Set fade, allowed range is [0, 1]. Default is 0.95.

c Set volume color expression.

The expression can use the following variables:

VOLUME
Current max volume of channel in dB.

PEAK
Current peak.

CHANNEL
Current channel number, starting from 0.

t If set, displays channel names. Default is enabled.

v If set, displays volume values. Default is enabled.

o Set orientation, can be horizontal: "h" or vertical: "v", default is "h".

s Set step size, allowed range is [0, 5]. Default is 0, which means step is disabled.

p Set background opacity, allowed range is [0, 1]. Default is 0.

m Set metering mode, can be peak: "p" or rms: "r", default is "p".

ds Set display scale, can be linear: "lin" or log: "log", default is "lin".

dm In second. If set to > 0., display a line for the max level in the previous seconds. default is
disabled: 0.

dmc The color of the max line. Use when "dm" option is set to > 0. default is: "orange"

showwaves
Convert input audio to a video output, representing the samples waves.

The filter accepts the following options:

size, s
Specify the video size for the output. For the syntax of this option, check the "Video size" section
in the ffmpeg-utils manual. Default value is "600x240".

mode
Set display mode.

Available values are:

point
Draw a point for each sample.

line
Draw a vertical line for each sample.

p2p Draw a point for each sample and a line between them.

cline
Draw a centered vertical line for each sample.

Default value is "point".

n Set the number of samples which are printed on the same column. A larger value will decrease the
frame rate. Must be a positive integer. This option can be set only if the value for rate is not
explicitly specified.

rate, r
Set the (approximate) output frame rate. This is done by setting the option n. Default value is "25".

split_channels
Set if channels should be drawn separately or overlap. Default value is 0.

colors
Set colors separated by '|' which are going to be used for drawing of each channel.

scale
Set amplitude scale.

Available values are:

lin Linear.

log Logarithmic.

sqrt
Square root.

cbrt
Cubic root.

Default is linear.

draw
Set the draw mode. This is mostly useful to set for high n.

Available values are:

scale
Scale pixel values for each drawn sample.

full
Draw every sample directly.

Default value is "scale".

Examples

• Output the input file audio and the corresponding video representation at the same time:

amovie=a.mp3,asplit[out0],showwaves[out1]

• Create a synthetic signal and show it with showwaves, forcing a frame rate of 30 frames per second:

aevalsrc=sin(1*2*PI*t)*sin(880*2*PI*t):cos(2*PI*200*t),asplit[out0],showwaves=r=30[out1]

showwavespic
Convert input audio to a single video frame, representing the samples waves.

The filter accepts the following options:

size, s
Specify the video size for the output. For the syntax of this option, check the "Video size" section
in the ffmpeg-utils manual. Default value is "600x240".

split_channels
Set if channels should be drawn separately or overlap. Default value is 0.

colors
Set colors separated by '|' which are going to be used for drawing of each channel.

scale
Set amplitude scale.

Available values are:

lin Linear.

log Logarithmic.

sqrt
Square root.

cbrt
Cubic root.

Default is linear.

draw
Set the draw mode.

Available values are:

scale
Scale pixel values for each drawn sample.

full
Draw every sample directly.

Default value is "scale".

filter
Set the filter mode.

Available values are:

average
Use average samples values for each drawn sample.

peak
Use peak samples values for each drawn sample.

Default value is "average".

Examples

• Extract a channel split representation of the wave form of a whole audio track in a 1024x800 picture
using ffmpeg:

ffmpeg -i audio.flac -lavfi showwavespic=split_channels=1:s=1024x800 waveform.png

sidedata, asidedata
Delete frame side data, or select frames based on it.

This filter accepts the following options:

mode
Set mode of operation of the filter.

Can be one of the following:

select
Select every frame with side data of "type".

delete
Delete side data of "type". If "type" is not set, delete all side data in the frame.

type
Set side data type used with all modes. Must be set for "select" mode. For the list of frame side
data types, refer to the "AVFrameSideDataType" enum in libavutil/frame.h. For example, to choose
"AV_FRAME_DATA_PANSCAN" side data, you must specify "PANSCAN".

spectrumsynth
Synthesize audio from 2 input video spectrums, first input stream represents magnitude across time and
second represents phase across time. The filter will transform from frequency domain as displayed in
videos back to time domain as presented in audio output.

This filter is primarily created for reversing processed showspectrum filter outputs, but can synthesize
sound from other spectrograms too. But in such case results are going to be poor if the phase data is
not available, because in such cases phase data need to be recreated, usually it's just recreated from
random noise. For best results use gray only output ("channel" color mode in showspectrum filter) and
"log" scale for magnitude video and "lin" scale for phase video. To produce phase, for 2nd video, use
"data" option. Inputs videos should generally use "fullframe" slide mode as that saves resources needed
for decoding video.

The filter accepts the following options:

sample_rate
Specify sample rate of output audio, the sample rate of audio from which spectrum was generated may
differ.

channels
Set number of channels represented in input video spectrums.

scale
Set scale which was used when generating magnitude input spectrum. Can be "lin" or "log". Default is
"log".

slide
Set slide which was used when generating inputs spectrums. Can be "replace", "scroll", "fullframe"
or "rscroll". Default is "fullframe".

win_func
Set window function used for resynthesis.

overlap
Set window overlap. In range "[0, 1]". Default is 1, which means optimal overlap for selected window
function will be picked.

orientation
Set orientation of input videos. Can be "vertical" or "horizontal". Default is "vertical".

Examples

• First create magnitude and phase videos from audio, assuming audio is stereo with 44100 sample rate,
then resynthesize videos back to audio with spectrumsynth:

ffmpeg -i input.flac -lavfi showspectrum=mode=separate:scale=log:overlap=0.875:color=channel:slide=fullframe:data=magnitude -an -c:v rawvideo magnitude.nut
ffmpeg -i input.flac -lavfi showspectrum=mode=separate:scale=lin:overlap=0.875:color=channel:slide=fullframe:data=phase -an -c:v rawvideo phase.nut
ffmpeg -i magnitude.nut -i phase.nut -lavfi spectrumsynth=channels=2:sample_rate=44100:win_func=hann:overlap=0.875:slide=fullframe output.flac

split, asplit
Split input into several identical outputs.

"asplit" works with audio input, "split" with video.

The filter accepts a single parameter which specifies the number of outputs. If unspecified, it defaults
to 2.

Examples

• Create two separate outputs from the same input:

[in] split [out0][out1]

• To create 3 or more outputs, you need to specify the number of outputs, like in:

[in] asplit=3 [out0][out1][out2]

• Create two separate outputs from the same input, one cropped and one padded:

[in] split [splitout1][splitout2];
[splitout1] crop=100:100:0:0 [cropout];
[splitout2] pad=200:200:100:100 [padout];

• Create 5 copies of the input audio with ffmpeg:

ffmpeg -i INPUT -filter_complex asplit=5 OUTPUT

zmq, azmq
Receive commands sent through a libzmq client, and forward them to filters in the filtergraph.

"zmq" and "azmq" work as a pass-through filters. "zmq" must be inserted between two video filters, "azmq"
between two audio filters. Both are capable to send messages to any filter type.

To enable these filters you need to install the libzmq library and headers and configure FFmpeg with
"--enable-libzmq".

For more information about libzmq see: <http://www.zeromq.org/>

The "zmq" and "azmq" filters work as a libzmq server, which receives messages sent through a network
interface defined by the bind_address (or the abbreviation "b") option. Default value of this option is
tcp://localhost:5555. You may want to alter this value to your needs, but do not forget to escape any ':'
signs (see filtergraph escaping).

The received message must be in the form:

<TARGET> <COMMAND> [<ARG>]

TARGET specifies the target of the command, usually the name of the filter class or a specific filter
instance name. The default filter instance name uses the pattern Parsed_<filter_name>_<index>, but you
can override this by using the filter_name@id syntax (see Filtergraph syntax).

COMMAND specifies the name of the command for the target filter.

ARG is optional and specifies the optional argument list for the given COMMAND.

Upon reception, the message is processed and the corresponding command is injected into the filtergraph.
Depending on the result, the filter will send a reply to the client, adopting the format:

<ERROR_CODE> <ERROR_REASON>
<MESSAGE>

MESSAGE is optional.

Examples

Look at tools/zmqsend for an example of a zmq client which can be used to send commands processed by
these filters.

Consider the following filtergraph generated by ffplay. In this example the last overlay filter has an
instance name. All other filters will have default instance names.

ffplay -dumpgraph 1 -f lavfi "
color=s=100x100:c=red [l];
color=s=100x100:c=blue [r];
nullsrc=s=200x100, zmq [bg];
[bg][l] overlay [bg+l];
[bg+l][r] overlay@my=x=100 "

To change the color of the left side of the video, the following command can be used:

echo Parsed_color_0 c yellow | tools/zmqsend

To change the right side:

echo Parsed_color_1 c pink | tools/zmqsend

To change the position of the right side:

echo overlay@my x 150 | tools/zmqsend

MULTIMEDIA SOURCES

Below is a description of the currently available multimedia sources.

amovie
This is the same as movie source, except it selects an audio stream by default.

avsynctest
Generate an Audio/Video Sync Test.

Generated stream periodically shows flash video frame and emits beep in audio. Useful to inspect A/V
sync issues.

It accepts the following options:

size, s
Set output video size. Default value is "hd720".

framerate, fr
Set output video frame rate. Default value is 30.

samplerate, sr
Set output audio sample rate. Default value is 44100.

amplitude, a
Set output audio beep amplitude. Default value is 0.7.

period, p
Set output audio beep period in seconds. Default value is 3.

delay, dl
Set output video flash delay in number of frames. Default value is 0.

cycle, c
Enable cycling of video delays, by default is disabled.

duration, d
Set stream output duration. By default duration is unlimited.

fg, bg, ag
Set foreground/background/additional color.

Commands

This source supports the some above options as commands.

movie
Read audio and/or video stream(s) from a movie container.

It accepts the following parameters:

filename
The name of the resource to read (not necessarily a file; it can also be a device or a stream
accessed through some protocol).

format_name, f
Specifies the format assumed for the movie to read, and can be either the name of a container or an
input device. If not specified, the format is guessed from movie_name or by probing.

seek_point, sp
Specifies the seek point in seconds. The frames will be output starting from this seek point. The
parameter is evaluated with "av_strtod", so the numerical value may be suffixed by an IS postfix. The
default value is "0".

streams, s
Specifies the streams to read. Several streams can be specified, separated by "+". The source will
then have as many outputs, in the same order. The syntax is explained in the "Stream specifiers"
section in the ffmpeg manual. Two special names, "dv" and "da" specify respectively the default (best
suited) video and audio stream. Default is "dv", or "da" if the filter is called as "amovie".

stream_index, si
Specifies the index of the video stream to read. If the value is -1, the most suitable video stream
will be automatically selected. The default value is "-1". Deprecated. If the filter is called
"amovie", it will select audio instead of video.

loop
Specifies how many times to read the stream in sequence. If the value is 0, the stream will be
looped infinitely. Default value is "1".

Note that when the movie is looped the source timestamps are not changed, so it will generate non
monotonically increasing timestamps.

discontinuity
Specifies the time difference between frames above which the point is considered a timestamp
discontinuity which is removed by adjusting the later timestamps.

dec_threads
Specifies the number of threads for decoding

format_opts
Specify format options for the opened file. Format options can be specified as a list of key=value
pairs separated by ':'. The following example shows how to add protocol_whitelist and
protocol_blacklist options:

ffplay -f lavfi
"movie=filename='1.sdp':format_opts='protocol_whitelist=file,rtp,udp\:protocol_blacklist=http'"

It allows overlaying a second video on top of the main input of a filtergraph, as shown in this graph:

input -----------> deltapts0 --> overlay --> output
^
|
movie --> scale--> deltapts1 -------+

Examples

• Skip 3.2 seconds from the start of the AVI file in.avi, and overlay it on top of the input labelled
"in":

movie=in.avi:seek_point=3.2, scale=180:-1, setpts=PTS-STARTPTS [over];
[in] setpts=PTS-STARTPTS [main];
[main][over] overlay=16:16 [out]

• Read from a video4linux2 device, and overlay it on top of the input labelled "in":

movie=/dev/video0:f=video4linux2, scale=180:-1, setpts=PTS-STARTPTS [over];
[in] setpts=PTS-STARTPTS [main];
[main][over] overlay=16:16 [out]

• Read the first video stream and the audio stream with id 0x81 from dvd.vob; the video is connected to
the pad named "video" and the audio is connected to the pad named "audio":

movie=dvd.vob:s=v:0+#0x81 [video] [audio]

Commands

Both movie and amovie support the following commands:

seek
Perform seek using "av_seek_frame". The syntax is: seek stream_index|timestamp|flags

• stream_index: If stream_index is -1, a default stream is selected, and timestamp is automatically
converted from AV_TIME_BASE units to the stream specific time_base.

• timestamp: Timestamp in AVStream.time_base units or, if no stream is specified, in AV_TIME_BASE
units.

• flags: Flags which select direction and seeking mode.

get_duration
Get movie duration in AV_TIME_BASE units.

EXTERNAL LIBRARIES

FFmpeg can be hooked up with a number of external libraries to add support for more formats. None of them
are used by default, their use has to be explicitly requested by passing the appropriate flags to
./configure.

Alliance for Open Media (AOM)
FFmpeg can make use of the AOM library for AV1 decoding and encoding.

Go to <http://aomedia.org/> and follow the instructions for installing the library. Then pass
"--enable-libaom" to configure to enable it.

AMD AMF/VCE
FFmpeg can use the AMD Advanced Media Framework library for accelerated H.264 and HEVC(only windows)
encoding on hardware with Video Coding Engine (VCE).

To enable support you must obtain the AMF framework header files(version 1.4.9+) from
<https://github.com/GPUOpen-LibrariesAndSDKs/AMF.git>.

Create an "AMF/" directory in the system include path. Copy the contents of "AMF/amf/public/include/"
into that directory. Then configure FFmpeg with "--enable-amf".

Initialization of amf encoder occurs in this order: 1) trying to initialize through dx11(only windows) 2)
trying to initialize through dx9(only windows) 3) trying to initialize through vulkan

To use h.264(AMD VCE) encoder on linux amdgru-pro version 19.20+ and amf-amdgpu-pro package(amdgru-pro
contains, but does not install automatically) are required.

This driver can be installed using amdgpu-pro-install script in official amd driver archive.

AviSynth
FFmpeg can read AviSynth scripts as input. To enable support, pass "--enable-avisynth" to configure after
installing the headers provided by <https://github.com/AviSynth/AviSynthPlus>. AviSynth+ can be
configured to install only the headers by either passing "-DHEADERS_ONLY:bool=on" to the normal CMake-
based build system, or by using the supplied "GNUmakefile".

For Windows, supported AviSynth variants are <http://avisynth.nl> for 32-bit builds and
<http://avisynth.nl/index.php/AviSynth+> for 32-bit and 64-bit builds.

For Linux, macOS, and BSD, the only supported AviSynth variant is
<https://github.com/AviSynth/AviSynthPlus>, starting with version 3.5.

In 2016, AviSynth+ added support for building with GCC. However, due to the eccentricities of
Windows' calling conventions, 32-bit GCC builds of AviSynth+ are not compatible with typical 32-bit
builds of FFmpeg.

By default, FFmpeg assumes compatibility with 32-bit MSVC builds of AviSynth+ since that is the most
widely-used and entrenched build configuration. Users can override this and enable support for
32-bit GCC builds of AviSynth+ by passing "-DAVSC_WIN32_GCC32" to "--extra-cflags" when configuring
FFmpeg.

64-bit builds of FFmpeg are not affected, and can use either MSVC or GCC builds of AviSynth+ without
any special flags.

AviSynth(+) is loaded dynamically. Distributors can build FFmpeg with "--enable-avisynth", and the
binaries will work regardless of the end user having AviSynth installed. If/when an end user would
like to use AviSynth scripts, then they can install AviSynth(+) and FFmpeg will be able to find and
use it to open scripts.

Chromaprint
FFmpeg can make use of the Chromaprint library for generating audio fingerprints. Pass
"--enable-chromaprint" to configure to enable it. See <https://acoustid.org/chromaprint>.

codec2
FFmpeg can make use of the codec2 library for codec2 decoding and encoding. There is currently no native
decoder, so libcodec2 must be used for decoding.

Go to <http://freedv.org/>, download "Codec 2 source archive". Build and install using CMake. Debian
users can install the libcodec2-dev package instead. Once libcodec2 is installed you can pass
"--enable-libcodec2" to configure to enable it.

The easiest way to use codec2 is with .c2 files, since they contain the mode information required for
decoding. To encode such a file, use a .c2 file extension and give the libcodec2 encoder the -mode
option: "ffmpeg -i input.wav -mode 700C output.c2". Playback is as simple as "ffplay output.c2". For a
list of supported modes, run "ffmpeg -h encoder=libcodec2". Raw codec2 files are also supported. To
make sense of them the mode in use needs to be specified as a format option: "ffmpeg -f codec2raw -mode
1300 -i input.raw output.wav".

dav1d
FFmpeg can make use of the dav1d library for AV1 video decoding.

Go to <https://code.videolan.org/videolan/dav1d> and follow the instructions for installing the library.
Then pass "--enable-libdav1d" to configure to enable it.

davs2
FFmpeg can make use of the davs2 library for AVS2-P2/IEEE1857.4 video decoding.

Go to <https://github.com/pkuvcl/davs2> and follow the instructions for installing the library. Then pass
"--enable-libdavs2" to configure to enable it.

libdavs2 is under the GNU Public License Version 2 or later (see
<http://www.gnu.org/licenses/old-licenses/gpl-2.0.html> for details), you must upgrade FFmpeg's
license to GPL in order to use it.

uavs3d
FFmpeg can make use of the uavs3d library for AVS3-P2/IEEE1857.10 video decoding.

Go to <https://github.com/uavs3/uavs3d> and follow the instructions for installing the library. Then pass
"--enable-libuavs3d" to configure to enable it.

Game Music Emu
FFmpeg can make use of the Game Music Emu library to read audio from supported video game music file
formats. Pass "--enable-libgme" to configure to enable it. See
<https://bitbucket.org/mpyne/game-music-emu/overview>.

Intel QuickSync Video
FFmpeg can use Intel QuickSync Video (QSV) for accelerated decoding and encoding of multiple codecs. To
use QSV, FFmpeg must be linked against the "libmfx" dispatcher, which loads the actual decoding
libraries.

The dispatcher is open source and can be downloaded from <https://github.com/lu-zero/mfx_dispatch.git>.
FFmpeg needs to be configured with the "--enable-libmfx" option and "pkg-config" needs to be able to
locate the dispatcher's ".pc" files.

Kvazaar
FFmpeg can make use of the Kvazaar library for HEVC encoding.

Go to <https://github.com/ultravideo/kvazaar> and follow the instructions for installing the library.
Then pass "--enable-libkvazaar" to configure to enable it.

LAME
FFmpeg can make use of the LAME library for MP3 encoding.

Go to <http://lame.sourceforge.net/> and follow the instructions for installing the library. Then pass
"--enable-libmp3lame" to configure to enable it.

libilbc
iLBC is a narrowband speech codec that has been made freely available by Google as part of the WebRTC
project. libilbc is a packaging friendly copy of the iLBC codec. FFmpeg can make use of the libilbc
library for iLBC decoding and encoding.

Go to <https://github.com/TimothyGu/libilbc> and follow the instructions for installing the library. Then
pass "--enable-libilbc" to configure to enable it.

libjxl
JPEG XL is an image format intended to fully replace legacy JPEG for an extended period of life. See
<https://jpegxl.info/> for more information, and see <https://github.com/libjxl/libjxl> for the library
source. You can pass "--enable-libjxl" to configure in order enable the libjxl wrapper.

libvpx
FFmpeg can make use of the libvpx library for VP8/VP9 decoding and encoding.

Go to <http://www.webmproject.org/> and follow the instructions for installing the library. Then pass
"--enable-libvpx" to configure to enable it.

ModPlug
FFmpeg can make use of this library, originating in Modplug-XMMS, to read from MOD-like music files. See
<https://github.com/Konstanty/libmodplug>. Pass "--enable-libmodplug" to configure to enable it.

OpenCORE, VisualOn, and Fraunhofer libraries
Spun off Google Android sources, OpenCore, VisualOn and Fraunhofer libraries provide encoders for a
number of audio codecs.

OpenCORE and VisualOn libraries are under the Apache License 2.0 (see
<http://www.apache.org/licenses/LICENSE-2.0> for details), which is incompatible to the LGPL version
2.1 and GPL version 2. You have to upgrade FFmpeg's license to LGPL version 3 (or if you have enabled
GPL components, GPL version 3) by passing "--enable-version3" to configure in order to use it.

The license of the Fraunhofer AAC library is incompatible with the GPL. Therefore, for GPL builds,
you have to pass "--enable-nonfree" to configure in order to use it. To the best of our knowledge, it
is compatible with the LGPL.

OpenCORE AMR

FFmpeg can make use of the OpenCORE libraries for AMR-NB decoding/encoding and AMR-WB decoding.

Go to <http://sourceforge.net/projects/opencore-amr/> and follow the instructions for installing the
libraries. Then pass "--enable-libopencore-amrnb" and/or "--enable-libopencore-amrwb" to configure to
enable them.

VisualOn AMR-WB encoder library

FFmpeg can make use of the VisualOn AMR-WBenc library for AMR-WB encoding.

Go to <http://sourceforge.net/projects/opencore-amr/> and follow the instructions for installing the
library. Then pass "--enable-libvo-amrwbenc" to configure to enable it.

Fraunhofer AAC library

FFmpeg can make use of the Fraunhofer AAC library for AAC decoding & encoding.

Go to <http://sourceforge.net/projects/opencore-amr/> and follow the instructions for installing the
library. Then pass "--enable-libfdk-aac" to configure to enable it.

OpenH264
FFmpeg can make use of the OpenH264 library for H.264 decoding and encoding.

Go to <http://www.openh264.org/> and follow the instructions for installing the library. Then pass
"--enable-libopenh264" to configure to enable it.

For decoding, this library is much more limited than the built-in decoder in libavcodec; currently, this
library lacks support for decoding B-frames and some other main/high profile features. (It currently only
supports constrained baseline profile and CABAC.) Using it is mostly useful for testing and for taking
advantage of Cisco's patent portfolio license (<http://www.openh264.org/BINARY_LICENSE.txt>).

OpenJPEG
FFmpeg can use the OpenJPEG libraries for decoding/encoding J2K videos. Go to <http://www.openjpeg.org/>
to get the libraries and follow the installation instructions. To enable using OpenJPEG in FFmpeg, pass
"--enable-libopenjpeg" to ./configure.

rav1e
FFmpeg can make use of rav1e (Rust AV1 Encoder) via its C bindings to encode videos. Go to
<https://github.com/xiph/rav1e/> and follow the instructions to build the C library. To enable using
rav1e in FFmpeg, pass "--enable-librav1e" to ./configure.

SVT-AV1
FFmpeg can make use of the Scalable Video Technology for AV1 library for AV1 encoding.

Go to <https://gitlab.com/AOMediaCodec/SVT-AV1/> and follow the instructions for installing the library.
Then pass "--enable-libsvtav1" to configure to enable it.

TwoLAME
FFmpeg can make use of the TwoLAME library for MP2 encoding.

Go to <http://www.twolame.org/> and follow the instructions for installing the library. Then pass
"--enable-libtwolame" to configure to enable it.

VapourSynth
FFmpeg can read VapourSynth scripts as input. To enable support, pass "--enable-vapoursynth" to
configure. Vapoursynth is detected via "pkg-config". Versions 42 or greater supported. See
<http://www.vapoursynth.com/>.

Due to security concerns, Vapoursynth scripts will not be autodetected so the input format has to be
forced. For ff* CLI tools, add "-f vapoursynth" before the input "-i yourscript.vpy".

x264
FFmpeg can make use of the x264 library for H.264 encoding.

Go to <http://www.videolan.org/developers/x264.html> and follow the instructions for installing the
library. Then pass "--enable-libx264" to configure to enable it.

x264 is under the GNU Public License Version 2 or later (see
<http://www.gnu.org/licenses/old-licenses/gpl-2.0.html> for details), you must upgrade FFmpeg's
license to GPL in order to use it.

x265
FFmpeg can make use of the x265 library for HEVC encoding.

Go to <http://x265.org/developers.html> and follow the instructions for installing the library. Then pass
"--enable-libx265" to configure to enable it.

x265 is under the GNU Public License Version 2 or later (see
<http://www.gnu.org/licenses/old-licenses/gpl-2.0.html> for details), you must upgrade FFmpeg's
license to GPL in order to use it.

xavs
FFmpeg can make use of the xavs library for AVS encoding.

Go to <http://xavs.sf.net/> and follow the instructions for installing the library. Then pass
"--enable-libxavs" to configure to enable it.

xavs2
FFmpeg can make use of the xavs2 library for AVS2-P2/IEEE1857.4 video encoding.

Go to <https://github.com/pkuvcl/xavs2> and follow the instructions for installing the library. Then pass
"--enable-libxavs2" to configure to enable it.

libxavs2 is under the GNU Public License Version 2 or later (see
<http://www.gnu.org/licenses/old-licenses/gpl-2.0.html> for details), you must upgrade FFmpeg's
license to GPL in order to use it.

ZVBI
ZVBI is a VBI decoding library which can be used by FFmpeg to decode DVB teletext pages and DVB teletext
subtitles.

Go to <http://sourceforge.net/projects/zapping/> and follow the instructions for installing the library.
Then pass "--enable-libzvbi" to configure to enable it.

SUPPORTED FILE FORMATS

       You can use the "-formats" and "-codecs" options to have an exhaustive list.

   File Formats
       FFmpeg supports the following file formats through the "libavformat" library:

       Name  :  Encoding @tab Decoding @tab Comments
       3dostr                     :    @tab X
       4xm                        :    @tab X
               @tab 4X Technologies format, used in some games.

       8088flex TMV               :    @tab X
       AAX                        :    @tab X
               @tab Audible Enhanced Audio format, used in audiobooks.

       AA                         :    @tab X
               @tab Audible Format 2, 3, and 4, used in audiobooks.

       ACT Voice                  :    @tab X
               @tab contains G.729 audio

       Adobe Filmstrip            :  X @tab X
       Audio IFF (AIFF)           :  X @tab X
       American Laser Games MM    :    @tab X
               @tab Multimedia format used in games like Mad Dog McCree.

       3GPP AMR                   :  X @tab X
       Amazing Studio Packed Animation File   :    @tab X
               @tab Multimedia format used in game Heart Of Darkness.

       Apple HTTP Live Streaming  :    @tab X
       Artworx Data Format        :    @tab X
       Interplay ACM              :    @tab X
               @tab Audio only format used in some Interplay games.

       ADP                        :    @tab X
               @tab Audio format used on the Nintendo Gamecube.

       AFC                        :    @tab X
               @tab Audio format used on the Nintendo Gamecube.

       ADS/SS2                    :    @tab X
               @tab Audio format used on the PS2.

       APNG                       :  X @tab X
       ASF                        :  X @tab X
               @tab Advanced / Active Streaming Format.

       AST                        :  X @tab X
               @tab Audio format used on the Nintendo Wii.

       AVI                        :  X @tab X
       AviSynth                   :    @tab X
       AVR                        :    @tab X
               @tab Audio format used on Mac.

       AVS                        :    @tab X
               @tab Multimedia format used by the Creature Shock game.

       Beam Software SIFF         :    @tab X
               @tab Audio and video format used in some games by Beam Software.

       Bethesda Softworks VID     :    @tab X
               @tab Used in some games from Bethesda Softworks.

       Binary text                :    @tab X
       Bink                       :    @tab X
               @tab Multimedia format used by many games.

       Bink Audio                 :    @tab X
               @tab Audio only multimedia format used by some games.

       Bitmap Brothers JV         :    @tab X
               @tab Used in Z and Z95 games.

       BRP                        :    @tab X
               @tab Argonaut Games format.

       Brute Force & Ignorance    :    @tab X
               @tab Used in the game Flash Traffic: City of Angels.

       BFSTM                      :    @tab X
               @tab Audio format used on the Nintendo WiiU (based on BRSTM).

       BRSTM                      :    @tab X
               @tab Audio format used on the Nintendo Wii.

       BW64                       :    @tab X
               @tab Broadcast Wave 64bit.

       BWF                        :  X @tab X
       codec2 (raw)               :  X @tab X
               @tab Must be given -mode format option to decode correctly.

       codec2 (.c2 files)         :  X @tab X
               @tab Contains header with version and mode info, simplifying playback.

       CRI ADX                    :  X @tab X
               @tab Audio-only format used in console video games.

       CRI AIX                    :    @tab X
       CRI HCA                    :    @tab X
               @tab Audio-only format used in console video games.

       Discworld II BMV           :    @tab X
       Interplay C93              :    @tab X
               @tab Used in the game Cyberia from Interplay.

       Delphine Software International CIN  :    @tab X
               @tab Multimedia format used by Delphine Software games.

       Digital Speech Standard (DSS)  :    @tab X
       CD+G                       :    @tab X
               @tab Video format used by CD+G karaoke disks

       Phantom Cine               :    @tab X
       Commodore CDXL             :    @tab X
               @tab Amiga CD video format

       Core Audio Format          :  X @tab X
               @tab Apple Core Audio Format

       CRC testing format         :  X @tab
       Creative Voice             :  X @tab X
               @tab Created for the Sound Blaster Pro.

       CRYO APC                   :    @tab X
               @tab Audio format used in some games by CRYO Interactive Entertainment.

       D-Cinema audio             :  X @tab X
       Deluxe Paint Animation     :    @tab X
       DCSTR                      :    @tab X
       DFA                        :    @tab X
               @tab This format is used in Chronomaster game

       DirectDraw Surface         :    @tab X
       DSD Stream File (DSF)      :    @tab X
       DV video                   :  X @tab X
       DXA                        :    @tab X
               @tab This format is used in the non-Windows version of the Feeble Files
                    game and different game cutscenes repacked for use with ScummVM.

       Electronic Arts cdata   :     @tab X
       Electronic Arts Multimedia   :     @tab X
               @tab Used in various EA games; files have extensions like WVE and UV2.

       Ensoniq Paris Audio File   :    @tab X
       FFM (FFserver live feed)   :  X @tab X
       Flash (SWF)                :  X @tab X
       Flash 9 (AVM2)             :  X @tab X
               @tab Only embedded audio is decoded.

       FLI/FLC/FLX animation      :    @tab X
               @tab .fli/.flc files

       Flash Video (FLV)          :  X @tab X
               @tab Macromedia Flash video files

       framecrc testing format    :  X @tab
       FunCom ISS                 :    @tab X
               @tab Audio format used in various games from FunCom like The Longest Journey.

       G.723.1                    :  X @tab X
       G.726                      :    @tab X @tab Both left- and right-justified.
       G.729 BIT                  :  X @tab X
       G.729 raw                  :    @tab X
       GENH                       :    @tab X
               @tab Audio format for various games.

       GIF Animation              :  X @tab X
       GXF                        :  X @tab X
               @tab General eXchange Format SMPTE 360M, used by Thomson Grass Valley
                    playout servers.

       HNM  :    @tab X
               @tab Only version 4 supported, used in some games from Cryo Interactive

       iCEDraw File               :    @tab X
       ICO                        :  X @tab X
               @tab Microsoft Windows ICO

       id Quake II CIN video      :    @tab X
       id RoQ                     :  X @tab X
               @tab Used in Quake III, Jedi Knight 2 and other computer games.

       IEC61937 encapsulation  :  X @tab X
       IFF                        :    @tab X
               @tab Interchange File Format

       IFV                        :    @tab X
               @tab A format used by some old CCTV DVRs.

       iLBC                       :  X @tab X
       Interplay MVE              :    @tab X
               @tab Format used in various Interplay computer games.

       Iterated Systems ClearVideo  :      @tab  X
               @tab I-frames only

       IV8                        :    @tab X
               @tab A format generated by IndigoVision 8000 video server.

       IVF (On2)                  :  X @tab X
               @tab A format used by libvpx

       Internet Video Recording   :    @tab X
       IRCAM                      :  X @tab X
       LAF                        :    @tab X
               @tab Limitless Audio Format

       LATM                       :  X @tab X
       LMLM4                      :    @tab X
               @tab Used by Linux Media Labs MPEG-4 PCI boards

       LOAS                       :    @tab X
               @tab contains LATM multiplexed AAC audio

       LRC                        :  X @tab X
       LVF                        :    @tab X
       LXF                        :    @tab X
               @tab VR native stream format, used by Leitch/Harris' video servers.

       Magic Lantern Video (MLV)  :    @tab X
       Matroska                   :  X @tab X
       Matroska audio             :  X @tab
       FFmpeg metadata            :  X @tab X
               @tab Metadata in text format.

       MAXIS XA                   :    @tab X
               @tab Used in Sim City 3000; file extension .xa.

       MCA                        :    @tab X
               @tab Used in some games from Capcom; file extension .mca.

       MD Studio                  :    @tab X
       Metal Gear Solid: The Twin Snakes  :  @tab X
       Megalux Frame              :    @tab X
               @tab Used by Megalux Ultimate Paint

       MobiClip MODS              :    @tab X
       MobiClip MOFLEX            :    @tab X
       Mobotix .mxg               :    @tab X
       Monkey's Audio             :    @tab X
       Motion Pixels MVI          :    @tab X
       MOV/QuickTime/MP4          :  X @tab X
               @tab 3GP, 3GP2, PSP, iPod variants supported

       MP2                        :  X @tab X
       MP3                        :  X @tab X
       MPEG-1 System              :  X @tab X
               @tab muxed audio and video, VCD format supported

       MPEG-PS (program stream)   :  X @tab X
               @tab also known as C<VOB> file, SVCD and DVD format supported

       MPEG-TS (transport stream)  :  X @tab X
               @tab also known as DVB Transport Stream

       MPEG-4                     :  X @tab X
               @tab MPEG-4 is a variant of QuickTime.

       MSF                        :    @tab X
               @tab Audio format used on the PS3.

       Mirillis FIC video         :    @tab X
               @tab No cursor rendering.

       MIDI Sample Dump Standard  :    @tab X
       MIME multipart JPEG        :  X @tab
       MSN TCP webcam             :    @tab X
               @tab Used by MSN Messenger webcam streams.

       MTV                        :    @tab X
       Musepack                   :    @tab X
       Musepack SV8               :    @tab X
       Material eXchange Format (MXF)  :  X @tab X
               @tab SMPTE 377M, used by D-Cinema, broadcast industry.

       Material eXchange Format (MXF), D-10 Mapping  :  X @tab X
               @tab SMPTE 386M, D-10/IMX Mapping.

       NC camera feed             :    @tab X
               @tab NC (AVIP NC4600) camera streams

       NIST SPeech HEader REsources  :    @tab X
       Computerized Speech Lab NSP  :    @tab X
       NTT TwinVQ (VQF)           :    @tab X
               @tab Nippon Telegraph and Telephone Corporation TwinVQ.

       Nullsoft Streaming Video   :    @tab X
       NuppelVideo                :    @tab X
       NUT                        :  X @tab X
               @tab NUT Open Container Format

       Ogg                        :  X @tab X
       Playstation Portable PMP   :    @tab X
       Portable Voice Format      :    @tab X
       RK Audio (RKA)             :    @tab X
       TechnoTrend PVA            :    @tab X
               @tab Used by TechnoTrend DVB PCI boards.

       QCP                        :    @tab X
       raw ADTS (AAC)             :  X @tab X
       raw AC-3                   :  X @tab X
       raw AMR-NB                 :    @tab X
       raw AMR-WB                 :    @tab X
       raw APAC                   :    @tab X
       raw aptX                   :  X @tab X
       raw aptX HD                :  X @tab X
       raw Bonk                   :    @tab X
       raw Chinese AVS video      :  X @tab X
       raw DFPWM                  :  X @tab X
       raw Dirac                  :  X @tab X
       raw DNxHD                  :  X @tab X
       raw DTS                    :  X @tab X
       raw DTS-HD                 :    @tab X
       raw E-AC-3                 :  X @tab X
       raw FLAC                   :  X @tab X
       raw GSM                    :    @tab X
       raw H.261                  :  X @tab X
       raw H.263                  :  X @tab X
       raw H.264                  :  X @tab X
       raw HEVC                   :  X @tab X
       raw Ingenient MJPEG        :    @tab X
       raw MJPEG                  :  X @tab X
       raw MLP                    :    @tab X
       raw MPEG                   :    @tab X
       raw MPEG-1                 :    @tab X
       raw MPEG-2                 :    @tab X
       raw MPEG-4                 :  X @tab X
       raw NULL                   :  X @tab
       raw video                  :  X @tab X
       raw id RoQ                 :  X @tab
       raw OBU                    :  X @tab X
       raw OSQ                    :    @tab X
       raw SBC                    :  X @tab X
       raw Shorten                :    @tab X
       raw TAK                    :    @tab X
       raw TrueHD                 :  X @tab X
       raw VC-1                   :  X @tab X
       raw PCM A-law              :  X @tab X
       raw PCM mu-law             :  X @tab X
       raw PCM Archimedes VIDC    :  X @tab X
       raw PCM signed 8 bit       :  X @tab X
       raw PCM signed 16 bit big-endian   :  X @tab X
       raw PCM signed 16 bit little-endian   :  X @tab X
       raw PCM signed 24 bit big-endian   :  X @tab X
       raw PCM signed 24 bit little-endian   :  X @tab X
       raw PCM signed 32 bit big-endian   :  X @tab X
       raw PCM signed 32 bit little-endian   :  X @tab X
       raw PCM signed 64 bit big-endian   :  X @tab X
       raw PCM signed 64 bit little-endian   :  X @tab X
       raw PCM unsigned 8 bit     :  X @tab X
       raw PCM unsigned 16 bit big-endian   :  X @tab X
       raw PCM unsigned 16 bit little-endian   :  X @tab X
       raw PCM unsigned 24 bit big-endian   :  X @tab X
       raw PCM unsigned 24 bit little-endian   :  X @tab X
       raw PCM unsigned 32 bit big-endian   :  X @tab X
       raw PCM unsigned 32 bit little-endian   :  X @tab X
       raw PCM 16.8 floating point little-endian  :    @tab X
       raw PCM 24.0 floating point little-endian  :    @tab X
       raw PCM floating-point 32 bit big-endian   :  X @tab X
       raw PCM floating-point 32 bit little-endian   :  X @tab X
       raw PCM floating-point 64 bit big-endian   :  X @tab X
       raw PCM floating-point 64 bit little-endian   :  X @tab X
       RDT                        :    @tab X
       REDCODE R3D                :    @tab X
               @tab File format used by RED Digital cameras, contains JPEG 2000 frames and PCM audio.

       RealMedia                  :  X @tab X
       Redirector                 :    @tab X
       RedSpark                   :    @tab X
       Renderware TeXture Dictionary  :    @tab X
       Resolume DXV               :    @tab X
       RF64                       :    @tab X
       RL2                        :    @tab X
               @tab Audio and video format used in some games by Entertainment Software Partners.

       RPL/ARMovie                :    @tab X
       Lego Mindstorms RSO        :  X @tab X
       RSD                        :    @tab X
       RTMP                       :  X @tab X
               @tab Output is performed by publishing stream to RTMP server

       RTP                        :  X @tab X
       RTSP                       :  X @tab X
       Sample Dump eXchange       :    @tab X
       SAP                        :  X @tab X
       SBG                        :    @tab X
       SDNS                       :    @tab X
       SDP                        :    @tab X
       SER                        :    @tab X
       Digital Pictures SGA       :    @tab X
       Sega FILM/CPK              :  X @tab X
               @tab Used in many Sega Saturn console games.

       Silicon Graphics Movie     :    @tab X
       Sierra SOL                 :    @tab X
               @tab .sol files used in Sierra Online games.

       Sierra VMD                 :    @tab X
               @tab Used in Sierra CD-ROM games.

       Smacker                    :    @tab X
               @tab Multimedia format used by many games.

       SMJPEG                     :  X @tab X
               @tab Used in certain Loki game ports.

       SMPTE 337M encapsulation   :    @tab X
       Smush                      :    @tab X
               @tab Multimedia format used in some LucasArts games.

       Sony OpenMG (OMA)          :  X @tab X
               @tab Audio format used in Sony Sonic Stage and Sony Vegas.

       Sony PlayStation STR       :    @tab X
       Sony Wave64 (W64)          :  X @tab X
       SoX native format          :  X @tab X
       SUN AU format              :  X @tab X
       SUP raw PGS subtitles      :  X @tab X
       SVAG                       :    @tab X
               @tab Audio format used in Konami PS2 games.

       TDSC                       :    @tab X
       Text files                 :    @tab X
       THP                        :    @tab X
               @tab Used on the Nintendo GameCube.

       Tiertex Limited SEQ        :    @tab X
               @tab Tiertex .seq files used in the DOS CD-ROM version of the game Flashback.

       True Audio                 :  X @tab X
       VAG                        :    @tab X
               @tab Audio format used in many Sony PS2 games.

       VC-1 test bitstream        :  X @tab X
       Vidvox Hap                 :  X @tab X
       Vivo                       :    @tab X
       VPK                        :    @tab X
               @tab Audio format used in Sony PS games.

       Marble WADY                :    @tab X
       WAV                        :  X @tab X
       Waveform Archiver          :    @tab X
       WavPack                    :  X @tab X
       WebM                       :  X @tab X
       Windows Televison (WTV)    :  X @tab X
       Wing Commander III movie   :    @tab X
               @tab Multimedia format used in Origin's Wing Commander III computer game.

       Westwood Studios audio     :  X @tab X
               @tab Multimedia format used in Westwood Studios games.

       Westwood Studios VQA       :    @tab X
               @tab Multimedia format used in Westwood Studios games.

       Wideband Single-bit Data (WSD)  :    @tab X
       WVE                        :    @tab X
       Konami XMD                 :    @tab X
       XMV                        :    @tab X
               @tab Microsoft video container used in Xbox games.

       XVAG                       :    @tab X
               @tab Audio format used on the PS3.

       xWMA                       :    @tab X
               @tab Microsoft audio container used by XAudio 2.

       eXtended BINary text (XBIN)  :  @tab X
       YUV4MPEG pipe              :  X @tab X
       Psygnosis YOP              :    @tab X

       "X" means that the feature in that column (encoding / decoding) is supported.

   Image Formats
       FFmpeg  can  read  and  write  images for each frame of a video sequence. The following image formats are
       supported:

       Name  :  Encoding @tab Decoding @tab Comments
       .Y.U.V        :  X @tab X
               @tab one raw file per component

       Alias PIX     :  X @tab X
               @tab Alias/Wavefront PIX image format

       animated GIF  :  X @tab X
       APNG          :  X @tab X
               @tab Animated Portable Network Graphics

       BMP           :  X @tab X
               @tab Microsoft BMP image

       BRender PIX   :    @tab X
               @tab Argonaut BRender 3D engine image format.

       CRI           :    @tab X
               @tab Cintel RAW

       DPX           :  X @tab X
               @tab Digital Picture Exchange

       EXR           :    @tab X
               @tab OpenEXR

       FITS          :  X @tab X
               @tab Flexible Image Transport System

       HDR           :  X @tab X
               @tab Radiance HDR RGBE Image format

       IMG           :    @tab X
               @tab GEM Raster image

       JPEG          :  X @tab X
               @tab Progressive JPEG is not supported.

       JPEG 2000     :  X @tab X
       JPEG-LS       :  X @tab X
       LJPEG         :  X @tab
               @tab Lossless JPEG

       Media 100     :    @tab X
       MSP           :    @tab X
               @tab Microsoft Paint image

       PAM           :  X @tab X
               @tab PAM is a PNM extension with alpha support.

       PBM           :  X @tab X
               @tab Portable BitMap image

       PCD           :    @tab X
               @tab PhotoCD

       PCX           :  X @tab X
               @tab PC Paintbrush

       PFM           :  X @tab X
               @tab Portable FloatMap image

       PGM           :  X @tab X
               @tab Portable GrayMap image

       PGMYUV        :  X @tab X
               @tab PGM with U and V components in YUV 4:2:0

       PGX           :    @tab X
               @tab PGX file decoder

       PHM           :  X @tab X
               @tab Portable HalfFloatMap image

       PIC           :  @tab X
               @tab Pictor/PC Paint

       PNG           :  X @tab X
               @tab Portable Network Graphics image

       PPM           :  X @tab X
               @tab Portable PixelMap image

       PSD           :    @tab X
               @tab Photoshop

       PTX           :    @tab X
               @tab V.Flash PTX format

       QOI           :  X @tab X
               @tab Quite OK Image format

       SGI           :  X @tab X
               @tab SGI RGB image format

       Sun Rasterfile   :  X @tab X
               @tab Sun RAS image format

       TIFF          :  X @tab X
               @tab YUV, JPEG and some extension is not supported yet.

       Truevision Targa   :  X @tab X
               @tab Targa (.TGA) image format

       VBN   :  X @tab X
               @tab Vizrt Binary Image format

       WBMP          :  X @tab X
               @tab Wireless Application Protocol Bitmap image format

       WebP          :  E @tab X
               @tab WebP image format, encoding supported through external library libwebp

       XBM   :  X @tab X
               @tab X BitMap image format

       XFace  :  X @tab X
               @tab X-Face image format

       XPM   :    @tab X
               @tab X PixMap image format

       XWD   :  X @tab X
               @tab X Window Dump image format

       "X" means that the feature in that column (encoding / decoding) is supported.

       "E" means that support is provided through an external library.

   Video Codecs
       Name  :  Encoding @tab Decoding @tab Comments
       4X Movie                :      @tab  X
               @tab Used in certain computer games.

       8088flex TMV            :      @tab  X
       A64 multicolor          :   X  @tab
               @tab Creates video suitable to be played on a commodore 64 (multicolor mode).

       Amazing Studio PAF Video  :      @tab  X
       American Laser Games MM   :     @tab X
               @tab Used in games like Mad Dog McCree.

       Amuse Graphics Movie    :      @tab  X
       AMV Video               :   X  @tab  X
               @tab Used in Chinese MP3 players.

       ANSI/ASCII art          :      @tab  X
       Apple Intermediate Codec  :      @tab  X
       Apple MJPEG-B           :      @tab  X
       Apple Pixlet            :      @tab  X
       Apple ProRes            :   X  @tab  X
               @tab fourcc: apch,apcn,apcs,apco,ap4h,ap4x

       Apple QuickDraw         :      @tab  X
               @tab fourcc: qdrw

       Argonaut Video          :      @tab  X
               @tab Used in some Argonaut games.

       Asus v1                 :   X  @tab  X
               @tab fourcc: ASV1

       Asus v2                 :   X  @tab  X
               @tab fourcc: ASV2

       ATI VCR1                :      @tab  X
               @tab fourcc: VCR1

       ATI VCR2                :      @tab  X
               @tab fourcc: VCR2

       Auravision Aura         :      @tab  X
       Auravision Aura 2       :      @tab  X
       Autodesk Animator Flic video   :      @tab  X
       Autodesk RLE            :      @tab  X
               @tab fourcc: AASC

       AV1                     :   E  @tab  E
               @tab Supported through external libraries libaom, libdav1d, librav1e and libsvtav1

       Avid 1:1 10-bit RGB Packer   :   X  @tab  X
               @tab fourcc: AVrp

       AVS (Audio Video Standard) video   :      @tab  X
               @tab Video encoding used by the Creature Shock game.

       AVS2-P2/IEEE1857.4      :   E  @tab  E
               @tab Supported through external libraries libxavs2 and libdavs2

       AVS3-P2/IEEE1857.10     :      @tab  E
               @tab Supported through external library libuavs3d

       AYUV                    :   X  @tab  X
               @tab Microsoft uncompressed packed 4:4:4:4

       Beam Software VB        :      @tab  X
       Bethesda VID video      :      @tab  X
               @tab Used in some games from Bethesda Softworks.

       Bink Video              :      @tab  X
       BitJazz SheerVideo      :      @tab  X
       Bitmap Brothers JV video   :    @tab X
       y41p Brooktree uncompressed 4:1:1 12-bit      :   X  @tab  X
       Brooktree ProSumer Video   :      @tab  X
               @tab fourcc: BT20

       Brute Force & Ignorance    :    @tab X
               @tab Used in the game Flash Traffic: City of Angels.

       C93 video               :      @tab  X
               @tab Codec used in Cyberia game.

       CamStudio               :      @tab  X
               @tab fourcc: CSCD

       CD+G                    :      @tab  X
               @tab Video codec for CD+G karaoke disks

       CDXL                    :      @tab  X
               @tab Amiga CD video codec

       Chinese AVS video       :   E  @tab  X
               @tab AVS1-P2, JiZhun profile, encoding through external library libxavs

       Delphine Software International CIN video   :      @tab  X
               @tab Codec used in Delphine Software International games.

       Discworld II BMV Video  :      @tab  X
       CineForm HD             :   X  @tab  X
       Canopus HQ              :      @tab  X
       Canopus HQA             :      @tab  X
       Canopus HQX             :      @tab  X
       Canopus Lossless Codec  :      @tab  X
       CDToons                 :      @tab  X
               @tab Codec used in various Broderbund games.

       Cinepak                 :      @tab  X
       Cirrus Logic AccuPak    :   X  @tab  X
               @tab fourcc: CLJR

       CPiA Video Format       :      @tab  X
       Creative YUV (CYUV)     :      @tab  X
       DFA                     :      @tab  X
               @tab Codec used in Chronomaster game.

       Dirac                   :   E  @tab  X
               @tab supported though the native vc2 (Dirac Pro) encoder

       Deluxe Paint Animation  :      @tab  X
       DNxHD                   :    X @tab  X
               @tab aka SMPTE VC3

       Duck TrueMotion 1.0    :      @tab  X
               @tab fourcc: DUCK

       Duck TrueMotion 2.0     :      @tab  X
               @tab fourcc: TM20

       Duck TrueMotion 2.0 RT  :      @tab  X
               @tab fourcc: TR20

       DV (Digital Video)      :   X  @tab  X
       Dxtory capture format   :      @tab  X
       Feeble Files/ScummVM DXA   :      @tab  X
               @tab Codec originally used in Feeble Files game.

       Electronic Arts CMV video   :      @tab  X
               @tab Used in NHL 95 game.

       Electronic Arts Madcow video   :      @tab  X
       Electronic Arts TGV video   :      @tab  X
       Electronic Arts TGQ video   :      @tab  X
       Electronic Arts TQI video   :      @tab  X
       Escape 124              :      @tab  X
       Escape 130              :      @tab  X
       FFmpeg video codec #1   :   X  @tab  X
               @tab lossless codec (fourcc: FFV1)

       Flash Screen Video v1   :   X  @tab  X
               @tab fourcc: FSV1

       Flash Screen Video v2   :   X  @tab  X
       Flash Video (FLV)       :   X  @tab  X
               @tab Sorenson H.263 used in Flash

       FM Screen Capture Codec   :      @tab  X
       Forward Uncompressed    :      @tab  X
       Fraps                   :      @tab  X
       Go2Meeting              :      @tab  X
               @tab fourcc: G2M2, G2M3

       Go2Webinar              :      @tab  X
               @tab fourcc: G2M4

       Gremlin Digital Video   :      @tab  X
       H.261                   :   X  @tab  X
       H.263 / H.263-1996      :   X  @tab  X
       H.263+ / H.263-1998 / H.263 version 2   :   X  @tab  X
       H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10   :   E  @tab  X
               @tab encoding supported through external library libx264 and OpenH264

       HEVC                    :   X  @tab  X
               @tab encoding supported through external library libx265 and libkvazaar

       HNM version 4           :      @tab  X
       HuffYUV                 :   X  @tab  X
       HuffYUV FFmpeg variant  :   X  @tab  X
       IBM Ultimotion          :      @tab  X
               @tab fourcc: ULTI

       id Cinematic video      :      @tab  X
               @tab Used in Quake II.

       id RoQ video            :   X  @tab  X
               @tab Used in Quake III, Jedi Knight 2, other computer games.

       IFF ILBM                :      @tab  X
               @tab IFF interleaved bitmap

       IFF ByteRun1            :      @tab  X
               @tab IFF run length encoded bitmap

       Infinity IMM4           :      @tab  X
       Intel H.263             :      @tab  X
       Intel Indeo 2           :      @tab  X
       Intel Indeo 3           :      @tab  X
       Intel Indeo 4           :      @tab  X
       Intel Indeo 5           :      @tab  X
       Interplay C93           :      @tab  X
               @tab Used in the game Cyberia from Interplay.

       Interplay MVE video     :      @tab  X
               @tab Used in Interplay .MVE files.

       J2K  :   X  @tab  X
       Karl Morton's video codec   :      @tab  X
               @tab Codec used in Worms games.

       Kega Game Video (KGV1)  :       @tab  X
               @tab Kega emulator screen capture codec.

       Lagarith                :      @tab  X
       LCL (LossLess Codec Library) MSZH   :      @tab  X
       LCL (LossLess Codec Library) ZLIB   :   E  @tab  E
       LOCO                    :      @tab  X
       LucasArts SANM/Smush    :      @tab  X
               @tab Used in LucasArts games / SMUSH animations.

       lossless MJPEG          :   X  @tab  X
       MagicYUV Video          :   X  @tab  X
       Mandsoft Screen Capture Codec   :      @tab  X
       Microsoft ATC Screen    :      @tab  X
               @tab Also known as Microsoft Screen 3.

       Microsoft Expression Encoder Screen   :      @tab  X
               @tab Also known as Microsoft Titanium Screen 2.

       Microsoft RLE           :   X  @tab  X
       Microsoft Screen 1      :      @tab  X
               @tab Also known as Windows Media Video V7 Screen.

       Microsoft Screen 2      :      @tab  X
               @tab Also known as Windows Media Video V9 Screen.

       Microsoft Video 1       :      @tab  X
       Mimic                   :      @tab  X
               @tab Used in MSN Messenger Webcam streams.

       Miro VideoXL            :      @tab  X
               @tab fourcc: VIXL

       MJPEG (Motion JPEG)     :   X  @tab  X
       Mobotix MxPEG video     :      @tab  X
       Motion Pixels video     :      @tab  X
       MPEG-1 video            :   X  @tab  X
       MPEG-2 video            :   X  @tab  X
       MPEG-4 part 2           :   X  @tab  X
               @tab libxvidcore can be used alternatively for encoding.

       MPEG-4 part 2 Microsoft variant version 1   :      @tab  X
       MPEG-4 part 2 Microsoft variant version 2   :   X  @tab  X
       MPEG-4 part 2 Microsoft variant version 3   :   X  @tab  X
       Newtek SpeedHQ                :   X  @tab  X
       Nintendo Gamecube THP video   :      @tab  X
       NotchLC                 :      @tab  X
       NuppelVideo/RTjpeg      :      @tab  X
               @tab Video encoding used in NuppelVideo files.

       On2 VP3                 :      @tab  X
               @tab still experimental

       On2 VP4                 :      @tab  X
               @tab fourcc: VP40

       On2 VP5                 :      @tab  X
               @tab fourcc: VP50

       On2 VP6                 :      @tab  X
               @tab fourcc: VP60,VP61,VP62

       On2 VP7                 :      @tab  X
               @tab fourcc: VP70,VP71

       VP8                     :   E  @tab  X
               @tab fourcc: VP80, encoding supported through external library libvpx

       VP9                     :   E  @tab  X
               @tab encoding supported through external library libvpx

       Pinnacle TARGA CineWave YUV16  :      @tab  X
               @tab fourcc: Y216

       Q-team QPEG             :      @tab  X
               @tab fourccs: QPEG, Q1.0, Q1.1

       QuickTime 8BPS video    :      @tab  X
       QuickTime Animation (RLE) video   :   X  @tab  X
               @tab fourcc: 'rle '

       QuickTime Graphics (SMC)   :   X  @tab  X
               @tab fourcc: 'smc '

       QuickTime video (RPZA)  :   X  @tab  X
               @tab fourcc: rpza

       R10K AJA Kona 10-bit RGB Codec      :   X  @tab  X
       R210 Quicktime Uncompressed RGB 10-bit      :   X  @tab  X
       Raw Video               :   X  @tab  X
       RealVideo 1.0           :   X  @tab  X
       RealVideo 2.0           :   X  @tab  X
       RealVideo 3.0           :      @tab  X
               @tab still far from ideal

       RealVideo 4.0           :      @tab  X
       Renderware TXD (TeXture Dictionary)   :      @tab  X
               @tab Texture dictionaries used by the Renderware Engine.

       RivaTuner Video         :      @tab  X
               @tab fourcc: 'RTV1'

       RL2 video               :      @tab  X
               @tab used in some games by Entertainment Software Partners

       ScreenPressor           :      @tab  X
       Screenpresso            :      @tab  X
       Screen Recorder Gold Codec   :      @tab  X
       Sierra VMD video        :      @tab  X
               @tab Used in Sierra VMD files.

       Silicon Graphics Motion Video Compressor 1 (MVC1)   :      @tab  X
       Silicon Graphics Motion Video Compressor 2 (MVC2)   :      @tab  X
       Silicon Graphics RLE 8-bit video   :      @tab  X
       Smacker video           :      @tab  X
               @tab Video encoding used in Smacker.

       SMPTE VC-1              :      @tab  X
       Snow                    :   X  @tab  X
               @tab experimental wavelet codec (fourcc: SNOW)

       Sony PlayStation MDEC (Motion DECoder)   :      @tab  X
       Sorenson Vector Quantizer 1   :   X  @tab  X
               @tab fourcc: SVQ1

       Sorenson Vector Quantizer 3   :      @tab  X
               @tab fourcc: SVQ3

       Sunplus JPEG (SP5X)     :      @tab  X
               @tab fourcc: SP5X

       TechSmith Screen Capture Codec   :      @tab  X
               @tab fourcc: TSCC

       TechSmith Screen Capture Codec 2   :      @tab  X
               @tab fourcc: TSC2

       Theora                  :   E  @tab  X
               @tab encoding supported through external library libtheora

       Tiertex Limited SEQ video   :      @tab  X
               @tab Codec used in DOS CD-ROM FlashBack game.

       Ut Video                :   X  @tab  X
       v210 QuickTime uncompressed 4:2:2 10-bit      :   X  @tab  X
       v308 QuickTime uncompressed 4:4:4             :   X  @tab  X
       v408 QuickTime uncompressed 4:4:4:4           :   X  @tab  X
       v410 QuickTime uncompressed 4:4:4 10-bit      :   X  @tab  X
       VBLE Lossless Codec     :      @tab  X
       vMix Video              :      @tab  X
               @tab fourcc: 'VMX1'

       VMware Screen Codec / VMware Video   :      @tab  X
               @tab Codec used in videos captured by VMware.

       Westwood Studios VQA (Vector Quantized Animation) video   :      @tab  X
       Windows Media Image     :      @tab  X
       Windows Media Video 7   :   X  @tab  X
       Windows Media Video 8   :   X  @tab  X
       Windows Media Video 9   :      @tab  X
               @tab not completely working

       Wing Commander III / Xan   :      @tab  X
               @tab Used in Wing Commander III .MVE files.

       Wing Commander IV / Xan   :      @tab  X
               @tab Used in Wing Commander IV.

       Winnov WNV1             :      @tab  X
       WMV7                    :   X  @tab  X
       YAMAHA SMAF             :   X  @tab  X
       Psygnosis YOP Video     :      @tab  X
       yuv4                    :   X  @tab  X
               @tab libquicktime uncompressed packed 4:2:0

       ZeroCodec Lossless Video  :      @tab  X
       ZLIB                    :   X  @tab  X
               @tab part of LCL, encoder experimental

       Zip Motion Blocks Video   :    X @tab  X
               @tab Encoder works only in PAL8.

       "X" means that the feature in that column (encoding / decoding) is supported.

       "E" means that support is provided through an external library.

   Audio Codecs
       Name  :  Encoding @tab Decoding @tab Comments
       8SVX exponential        :      @tab  X
       8SVX fibonacci          :      @tab  X
       AAC                     :  EX  @tab  X
               @tab encoding supported through internal encoder and external library libfdk-aac

       AAC+                    :   E  @tab  IX
               @tab encoding supported through external library libfdk-aac

       AC-3                    :  IX  @tab  IX
       ACELP.KELVIN            :      @tab  X
       ADPCM 4X Movie          :      @tab  X
       ADPCM Yamaha AICA       :      @tab  X
       ADPCM AmuseGraphics Movie  :     @tab  X
       ADPCM Argonaut Games    :  X   @tab  X
       ADPCM CDROM XA          :      @tab  X
       ADPCM Creative Technology  :      @tab  X
               @tab 16 -E<gt> 4, 8 -E<gt> 4, 8 -E<gt> 3, 8 -E<gt> 2

       ADPCM Electronic Arts   :      @tab  X
               @tab Used in various EA titles.

       ADPCM Electronic Arts Maxis CDROM XS   :      @tab  X
               @tab Used in Sim City 3000.

       ADPCM Electronic Arts R1   :      @tab  X
       ADPCM Electronic Arts R2   :      @tab  X
       ADPCM Electronic Arts R3   :      @tab  X
       ADPCM Electronic Arts XAS  :      @tab  X
       ADPCM G.722             :   X  @tab  X
       ADPCM G.726             :   X  @tab  X
       ADPCM IMA Acorn Replay  :      @tab  X
       ADPCM IMA AMV           :   X  @tab  X
               @tab Used in AMV files

       ADPCM IMA Cunning Developments   :      @tab  X
       ADPCM IMA Electronic Arts EACS   :      @tab  X
       ADPCM IMA Electronic Arts SEAD   :      @tab  X
       ADPCM IMA Funcom        :      @tab  X
       ADPCM IMA High Voltage Software ALP       :   X  @tab  X
       ADPCM IMA Mobiclip MOFLEX   :      @tab  X
       ADPCM IMA QuickTime     :   X  @tab  X
       ADPCM IMA Simon & Schuster Interactive    :   X  @tab  X
       ADPCM IMA Ubisoft APM   :   X  @tab  X
       ADPCM IMA Loki SDL MJPEG   :      @tab  X
       ADPCM IMA WAV           :   X  @tab  X
       ADPCM IMA Westwood      :      @tab  X
       ADPCM ISS IMA           :      @tab  X
               @tab Used in FunCom games.

       ADPCM IMA Dialogic      :      @tab  X
       ADPCM IMA Duck DK3      :      @tab  X
               @tab Used in some Sega Saturn console games.

       ADPCM IMA Duck DK4      :      @tab  X
               @tab Used in some Sega Saturn console games.

       ADPCM IMA Radical       :      @tab  X
       ADPCM Microsoft         :   X  @tab  X
       ADPCM MS IMA            :   X  @tab  X
       ADPCM Nintendo Gamecube AFC   :      @tab  X
       ADPCM Nintendo Gamecube DTK   :      @tab  X
       ADPCM Nintendo THP   :      @tab  X
       ADPCM Playstation       :      @tab  X
       ADPCM QT IMA            :   X  @tab  X
       ADPCM SEGA CRI ADX      :   X  @tab  X
               @tab Used in Sega Dreamcast games.

       ADPCM Shockwave Flash   :   X  @tab  X
       ADPCM Sound Blaster Pro 2-bit   :      @tab  X
       ADPCM Sound Blaster Pro 2.6-bit   :      @tab  X
       ADPCM Sound Blaster Pro 4-bit   :      @tab  X
       ADPCM VIMA              :      @tab  X
               @tab Used in LucasArts SMUSH animations.

       ADPCM Konami XMD        :      @tab  X
       ADPCM Westwood Studios IMA       :   X @tab  X
               @tab Used in Westwood Studios games like Command and Conquer.

       ADPCM Yamaha            :   X  @tab  X
       ADPCM Zork              :      @tab  X
       AMR-NB                  :   E  @tab  X
               @tab encoding supported through external library libopencore-amrnb

       AMR-WB                  :   E  @tab  X
               @tab encoding supported through external library libvo-amrwbenc

       Amazing Studio PAF Audio  :      @tab  X
       Apple lossless audio    :   X  @tab  X
               @tab QuickTime fourcc 'alac'

       aptX                    :   X  @tab  X
               @tab Used in Bluetooth A2DP

       aptX HD                 :   X  @tab  X
               @tab Used in Bluetooth A2DP

       ATRAC1                  :      @tab  X
       ATRAC3                  :      @tab  X
       ATRAC3+                 :      @tab  X
       ATRAC9                  :      @tab  X
       Bink Audio              :      @tab  X
               @tab Used in Bink and Smacker files in many games.

       Bonk audio              :      @tab  X
       CELT                    :      @tab  E
               @tab decoding supported through external library libcelt

       codec2                  :   E  @tab  E
               @tab en/decoding supported through external library libcodec2

       CRI HCA                 :      @tab X
       Delphine Software International CIN audio   :      @tab  X
               @tab Codec used in Delphine Software International games.

       DFPWM                   :   X  @tab  X
       Digital Speech Standard - Standard Play mode (DSS SP)  :      @tab  X
       Discworld II BMV Audio  :      @tab  X
       COOK                    :      @tab  X
               @tab All versions except 5.1 are supported.

       DCA (DTS Coherent Acoustics)   :   X  @tab  X
               @tab supported extensions: XCh, XXCH, X96, XBR, XLL, LBR (partially)

       Dolby E   :      @tab  X
       DPCM Cuberoot-Delta-Exact  :   @tab  X
               @tab Used in few games.

       DPCM Gremlin            :      @tab  X
       DPCM id RoQ             :   X  @tab  X
               @tab Used in Quake III, Jedi Knight 2 and other computer games.

       DPCM Marble WADY        :      @tab  X
       DPCM Interplay          :      @tab  X
               @tab Used in various Interplay computer games.

       DPCM Squareroot-Delta-Exact   :   @tab  X
               @tab Used in various games.

       DPCM Sierra Online      :      @tab  X
               @tab Used in Sierra Online game audio files.

       DPCM Sol                :      @tab  X
       DPCM Xan                :      @tab  X
               @tab Used in Origin's Wing Commander IV AVI files.

       DPCM Xilam DERF         :      @tab  X
       DSD (Direct Stream Digital), least significant bit first   :   @tab  X
       DSD (Direct Stream Digital), most significant bit first    :   @tab  X
       DSD (Direct Stream Digital), least significant bit first, planar   :   @tab  X
       DSD (Direct Stream Digital), most significant bit first, planar    :   @tab  X
       DSP Group TrueSpeech    :      @tab  X
       DST (Direct Stream Transfer)  :   @tab  X
       DV audio                :      @tab  X
       Enhanced AC-3           :   X  @tab  X
       EVRC (Enhanced Variable Rate Codec)  :      @tab  X
       FLAC (Free Lossless Audio Codec)   :   X  @tab  IX
       FTR Voice               :      @tab  X
       G.723.1                 :  X   @tab  X
       G.729                   :      @tab  X
       GSM                     :   E  @tab  X
               @tab encoding supported through external library libgsm

       GSM Microsoft variant   :   E  @tab  X
               @tab encoding supported through external library libgsm

       IAC (Indeo Audio Coder)   :      @tab  X
       iLBC (Internet Low Bitrate Codec)  :   E  @tab  EX
               @tab encoding and decoding supported through external library libilbc

       IMC (Intel Music Coder)   :      @tab  X
       Interplay ACM             :      @tab  X
       MACE (Macintosh Audio Compression/Expansion) 3:1   :      @tab  X
       MACE (Macintosh Audio Compression/Expansion) 6:1   :      @tab  X
       Marian's A-pac audio      :      @tab  X
       MI-SC4 (Micronas SC-4 Audio)   :      @tab  X
       MLP (Meridian Lossless Packing)   :   X  @tab  X
               @tab Used in DVD-Audio discs.

       Monkey's Audio          :      @tab  X
       MP1 (MPEG audio layer 1)   :      @tab IX
       MP2 (MPEG audio layer 2)   :  IX  @tab IX
               @tab encoding supported also through external library TwoLAME

       MP3 (MPEG audio layer 3)   :   E  @tab IX
               @tab encoding supported through external library LAME, ADU MP3 and MP3onMP4 also supported

       MPEG-4 Audio Lossless Coding (ALS)   :      @tab  X
       MobiClip FastAudio      :      @tab  X
       Musepack SV7            :      @tab  X
       Musepack SV8            :      @tab  X
       Nellymoser Asao         :   X  @tab  X
       On2 AVC (Audio for Video Codec)  :      @tab  X
       Opus                    :   E  @tab  X
               @tab encoding supported through external library libopus

       OSQ (Original Sound Quality)   :      @tab  X
       PCM A-law               :   X  @tab  X
       PCM mu-law              :   X  @tab  X
       PCM Archimedes VIDC     :   X  @tab  X
       PCM signed 8-bit planar   :   X  @tab  X
       PCM signed 16-bit big-endian planar   :   X  @tab  X
       PCM signed 16-bit little-endian planar   :   X  @tab  X
       PCM signed 24-bit little-endian planar   :   X  @tab  X
       PCM signed 32-bit little-endian planar   :   X  @tab  X
       PCM 32-bit floating point big-endian   :   X  @tab  X
       PCM 32-bit floating point little-endian   :   X  @tab  X
       PCM 64-bit floating point big-endian   :   X  @tab  X
       PCM 64-bit floating point little-endian   :   X  @tab  X
       PCM D-Cinema audio signed 24-bit    :   X  @tab  X
       PCM signed 8-bit        :   X  @tab  X
       PCM signed 16-bit big-endian   :   X  @tab  X
       PCM signed 16-bit little-endian   :   X  @tab  X
       PCM signed 24-bit big-endian   :   X  @tab  X
       PCM signed 24-bit little-endian   :   X  @tab  X
       PCM signed 32-bit big-endian   :   X  @tab  X
       PCM signed 32-bit little-endian   :   X  @tab  X
       PCM signed 16/20/24-bit big-endian in MPEG-TS   :      @tab  X
       PCM unsigned 8-bit      :   X  @tab  X
       PCM unsigned 16-bit big-endian   :   X  @tab  X
       PCM unsigned 16-bit little-endian   :   X  @tab  X
       PCM unsigned 24-bit big-endian   :   X  @tab  X
       PCM unsigned 24-bit little-endian   :   X  @tab  X
       PCM unsigned 32-bit big-endian   :   X  @tab  X
       PCM unsigned 32-bit little-endian   :   X  @tab  X
       PCM SGA                 :      @tab  X
       QCELP / PureVoice       :      @tab  X
       QDesign Music Codec 1   :      @tab  X
       QDesign Music Codec 2   :      @tab  X
               @tab There are still some distortions.

       RealAudio 1.0 (14.4K)   :   X  @tab  X
               @tab Real 14400 bit/s codec

       RealAudio 2.0 (28.8K)   :      @tab  X
               @tab Real 28800 bit/s codec

       RealAudio 3.0 (dnet)    :  IX  @tab  X
               @tab Real low bitrate AC-3 codec

       RealAudio Lossless      :      @tab  X
       RealAudio SIPR / ACELP.NET  :      @tab  X
       RK Audio (RKA)          :      @tab  X
       SBC (low-complexity subband codec)  :   X  @tab  X
               @tab Used in Bluetooth A2DP

       Shorten                 :      @tab  X
       Sierra VMD audio        :      @tab  X
               @tab Used in Sierra VMD files.

       Smacker audio           :      @tab  X
       SMPTE 302M AES3 audio   :   X  @tab  X
       Sonic                   :   X  @tab  X
               @tab experimental codec

       Sonic lossless          :   X  @tab  X
               @tab experimental codec

       Speex                   :   E  @tab  EX
               @tab supported through external library libspeex

       TAK (Tom's lossless Audio Kompressor)   :      @tab  X
       True Audio (TTA)        :   X  @tab  X
       TrueHD                  :   X  @tab  X
               @tab Used in HD-DVD and Blu-Ray discs.

       TwinVQ (VQF flavor)     :      @tab  X
       VIMA                    :      @tab  X
               @tab Used in LucasArts SMUSH animations.

       ViewQuest VQC           :      @tab  X
       Vorbis                  :   E  @tab  X
               @tab A native but very primitive encoder exists.

       Voxware MetaSound       :      @tab  X
       Waveform Archiver       :      @tab  X
       WavPack                 :   X  @tab  X
       Westwood Audio (SND1)   :      @tab  X
       Windows Media Audio 1   :   X  @tab  X
       Windows Media Audio 2   :   X  @tab  X
       Windows Media Audio Lossless  :   @tab  X
       Windows Media Audio Pro  :     @tab  X
       Windows Media Audio Voice  :   @tab  X
       Xbox Media Audio 1      :      @tab  X
       Xbox Media Audio 2      :      @tab  X

       "X" means that the feature in that column (encoding / decoding) is supported.

       "E" means that support is provided through an external library.

       "I" means that an integer-only version is available, too (ensures high  performance  on  systems  without
       hardware floating point support).

   Subtitle Formats
       Name  :  Muxing @tab Demuxing @tab Encoding @tab Decoding
       3GPP Timed Text   :    @tab   @tab X @tab X
       AQTitle           :    @tab X @tab   @tab X
       DVB               :  X @tab X @tab X @tab X
       DVB teletext      :    @tab X @tab   @tab E
       DVD               :  X @tab X @tab X @tab X
       JACOsub           :  X @tab X @tab   @tab X
       MicroDVD          :  X @tab X @tab   @tab X
       MPL2              :    @tab X @tab   @tab X
       MPsub (MPlayer)   :    @tab X @tab   @tab X
       PGS               :    @tab   @tab   @tab X
       PJS (Phoenix)     :    @tab X @tab   @tab X
       RealText          :    @tab X @tab   @tab X
       SAMI              :    @tab X @tab   @tab X
       Spruce format (STL)  :    @tab X @tab   @tab X
       SSA/ASS           :  X @tab X @tab X @tab X
       SubRip (SRT)      :  X @tab X @tab X @tab X
       SubViewer v1      :    @tab X @tab   @tab X
       SubViewer         :    @tab X @tab   @tab X
       TED Talks captions  :  @tab X @tab   @tab X
       TTML              :  X @tab   @tab X @tab
       VobSub (IDX+SUB)  :    @tab X @tab   @tab X
       VPlayer           :    @tab X @tab   @tab X
       WebVTT            :  X @tab X @tab X @tab X
       XSUB              :    @tab   @tab X @tab X

       "X" means that the feature is supported.

       "E" means that support is provided through an external library.

   Network Protocols
       Name          :  Support
       AMQP          :  E
       file          :  X
       FTP           :  X
       Gopher        :  X
       Gophers       :  X
       HLS           :  X
       HTTP          :  X
       HTTPS         :  X
       Icecast       :  X
       MMSH          :  X
       MMST          :  X
       pipe          :  X
       Pro-MPEG FEC  :  X
       RTMP          :  X
       RTMPE         :  X
       RTMPS         :  X
       RTMPT         :  X
       RTMPTE        :  X
       RTMPTS        :  X
       RTP           :  X
       SAMBA         :  E
       SCTP          :  X
       SFTP          :  E
       TCP           :  X
       TLS           :  X
       UDP           :  X
       ZMQ           :  E

       "X" means that the protocol is supported.

       "E" means that support is provided through an external library.

   Input/Output Devices
       Name               :  Input  @tab Output
       ALSA               :  X      @tab X
       BKTR               :  X      @tab
       caca               :         @tab X
       DV1394             :  X      @tab
       Lavfi virtual device  :  X   @tab
       Linux framebuffer  :  X      @tab X
       JACK               :  X      @tab
       LIBCDIO            :  X
       LIBDC1394          :  X      @tab
       OpenAL             :  X
       OpenGL             :         @tab X
       OSS                :  X      @tab X
       PulseAudio         :  X      @tab X
       SDL                :         @tab X
       Video4Linux2       :  X      @tab X
       VfW capture        :  X      @tab
       X11 grabbing       :  X      @tab
       Win32 grabbing     :  X      @tab

       "X" means that input/output is supported.

   Timecode
       Codec/format       :  Read   @tab Write
       AVI                :  X      @tab X
       DV                 :  X      @tab X
       GXF                :  X      @tab X
       MOV                :  X      @tab X
       MPEG1/2            :  X      @tab X
       MXF                :  X      @tab X

AUTHORS

       The FFmpeg developers.

       For details about the authorship, see the Git history  of  the  project  (https://git.ffmpeg.org/ffmpeg),
       e.g.  by  typing the command git log in the FFmpeg source directory, or browsing the online repository at
       <https://git.ffmpeg.org/ffmpeg>.

       Maintainers for the specific components are listed in the file MAINTAINERS in the source code tree.

                                                                                                   FFMPEG-ALL(1)

NAME

SYNOPSIS

DESCRIPTION

DETAILED DESCRIPTION

STREAM SELECTION

OPTIONS

EXAMPLES

SYNTAX

EXPRESSION EVALUATION

CODEC OPTIONS

DECODERS

VIDEO DECODERS

AUDIO DECODERS

SUBTITLES DECODERS

ENCODERS

AUDIO ENCODERS

VIDEO ENCODERS

SUBTITLES ENCODERS

BITSTREAM FILTERS

FORMAT OPTIONS

DEMUXERS

MUXERS

METADATA

PROTOCOL OPTIONS

PROTOCOLS

DEVICE OPTIONS

INPUT DEVICES

OUTPUT DEVICES

RESAMPLER OPTIONS

SCALER OPTIONS

FILTERING INTRODUCTION

GRAPH

FILTERGRAPH DESCRIPTION

TIMELINE EDITING

CHANGING OPTIONS AT RUNTIME WITH A COMMAND

OPTIONS FOR FILTERS WITH SEVERAL INPUTS

AUDIO FILTERS

AUDIO SOURCES

AUDIO SINKS

VIDEO FILTERS

OPENCL VIDEO FILTERS

VAAPI VIDEO FILTERS

VULKAN VIDEO FILTERS

QSV VIDEO FILTERS

VIDEO SOURCES

VIDEO SINKS

MULTIMEDIA FILTERS

MULTIMEDIA SOURCES

EXTERNAL LIBRARIES

SUPPORTED FILE FORMATS

SEE ALSO

AUTHORS