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FFPLAY-ALL(1) FFPLAY-ALL(1)

NAME

ffplay - FFplay media player

SYNOPSIS

ffplay [options] [input_url]

DESCRIPTION

FFplay is a very simple and portable media player using the FFmpeg libraries and the SDL library. It is mostly used as a testbed for the various FFmpeg APIs.

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.
stream_type[:stream_index]
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 stream_index is given, then it matches stream number stream_index of this type. Otherwise, it matches all streams of this type.
p:program_id[:stream_index] or p:program_id[:stream_type[:stream_index]] or
p:program_id:m:key[:value] In first version, if stream_index is given, then it matches the stream with number stream_index in the program with the id program_id. Otherwise, it matches all streams in the program. In the second version, stream_type is one of following: 'v' for video, 'a' for audio, 's' for subtitle, 'd' for data. If stream_index is also given, then it matches stream number stream_index of this type in the program with the id program_id. Otherwise, if only stream_type is given, it matches all streams of this type in the program with the id program_id. In the third version matches streams in the program with the id program_id with the metadata tag key having the specified value. If value is not given, matches streams that contain the given tag with any value.
#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 name filter_name. Use the -filters option to get a list of all filters.
-version
Show version.
-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.
-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 NO_COLOR, or can be forced setting the environment variable AV_LOG_FORCE_COLOR. The use of the environment variable NO_COLOR is deprecated and will be dropped in a future FFmpeg version.

-report
Dump full command line and console 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 verbose".

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

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.

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

-x width
Force displayed width.
-y height
Force displayed height.
-s size
Set frame size (WxH or abbreviation), needed for videos which do not contain a header with the frame size like raw YUV. This option has been deprecated in favor of private options, try -video_size.
-fs
Start in fullscreen mode.
-an
Disable audio.
-vn
Disable video.
-sn
Disable subtitles.
-ss pos
Seek to pos. Note that in most formats it is not possible to seek exactly, so ffplay will seek to the nearest seek point to pos.

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

-t duration
Play duration seconds of audio/video.

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

-bytes
Seek by bytes.
-seek_interval
Set custom interval, in seconds, for seeking using left/right keys. Default is 10 seconds.
-nodisp
Disable graphical display.
-noborder
Borderless window.
-volume
Set the startup volume. 0 means silence, 100 means no volume reduction or amplification. Negative values are treated as 0, values above 100 are treated as 100.
-f fmt
Force format.
-window_title title
Set window title (default is the input filename).
-left title
Set the x position for the left of the window (default is a centered window).
-top title
Set the y position for the top of the window (default is a centered window).
-loop number
Loops movie playback <number> times. 0 means forever.
-showmode mode
Set the show mode to use. Available values for mode are:
0, video
show video
1, waves
show audio waves
2, rdft
show audio frequency band using RDFT ((Inverse) Real Discrete Fourier Transform)

Default value is "video", if video is not present or cannot be played "rdft" is automatically selected.

You can interactively cycle through the available show modes by pressing the key w.

-vf filtergraph
Create the filtergraph specified by filtergraph and use it to filter the video stream.

filtergraph is a description of the filtergraph to apply to the stream, and must have a single video input and a single video output. 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.

You can specify this parameter multiple times and cycle through the specified filtergraphs along with the show modes by pressing the key w.

-af filtergraph
filtergraph is a description of the filtergraph to apply to the input audio. Use the option "-filters" to show all the available filters (including sources and sinks).
-i input_url
Read input_url.

Advanced options

-pix_fmt format
Set pixel format. This option has been deprecated in favor of private options, try -pixel_format.
-stats
Print several playback statistics, in particular show the stream duration, the codec parameters, the current position in the stream and the audio/video synchronisation drift. It is on by default, to explicitly disable it you need to specify "-nostats".
-fast
Non-spec-compliant optimizations.
-genpts
Generate pts.
-sync type
Set the master clock to audio ("type=audio"), video ("type=video") or external ("type=ext"). Default is audio. The master clock is used to control audio-video synchronization. Most media players use audio as master clock, but in some cases (streaming or high quality broadcast) it is necessary to change that. This option is mainly used for debugging purposes.
-ast audio_stream_specifier
Select the desired audio stream using the given stream specifier. The stream specifiers are described in the Stream specifiers chapter. If this option is not specified, the "best" audio stream is selected in the program of the already selected video stream.
-vst video_stream_specifier
Select the desired video stream using the given stream specifier. The stream specifiers are described in the Stream specifiers chapter. If this option is not specified, the "best" video stream is selected.
-sst subtitle_stream_specifier
Select the desired subtitle stream using the given stream specifier. The stream specifiers are described in the Stream specifiers chapter. If this option is not specified, the "best" subtitle stream is selected in the program of the already selected video or audio stream.
-autoexit
Exit when video is done playing.
-exitonkeydown
Exit if any key is pressed.
-exitonmousedown
Exit if any mouse button is pressed.
-codec:media_specifier codec_name
Force a specific decoder implementation for the stream identified by media_specifier, which can assume the values "a" (audio), "v" (video), and "s" subtitle.
-acodec codec_name
Force a specific audio decoder.
-vcodec codec_name
Force a specific video decoder.
-scodec codec_name
Force a specific subtitle decoder.
-autorotate
Automatically rotate the video according to file metadata. Enabled by default, use -noautorotate to disable it.
-framedrop
Drop video frames if video is out of sync. Enabled by default if the master clock is not set to video. Use this option to enable frame dropping for all master clock sources, use -noframedrop to disable it.
-infbuf
Do not limit the input buffer size, read as much data as possible from the input as soon as possible. Enabled by default for realtime streams, where data may be dropped if not read in time. Use this option to enable infinite buffers for all inputs, use -noinfbuf to disable it.

While playing

q, ESC
Quit.
f
Toggle full screen.
p, SPC
Pause.
m
Toggle mute.
9, 0
Decrease and increase volume respectively.
/, *
Decrease and increase volume respectively.
a
Cycle audio channel in the current program.
v
Cycle video channel.
t
Cycle subtitle channel in the current program.
c
Cycle program.
w
Cycle video filters or show modes.
s
Step to the next frame.

Pause if the stream is not already paused, step to the next video frame, and pause.

left/right
Seek backward/forward 10 seconds.
down/up
Seek backward/forward 1 minute.
page down/page up
Seek to the previous/next chapter. or if there are no chapters Seek backward/forward 10 minutes.
right mouse click
Seek to percentage in file corresponding to fraction of width.
left mouse double-click
Toggle full screen.

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 expresses the number of seconds, with the optional decimal part m.

In both expressions, the optional - indicates negative duration.

Examples

The following examples are all valid time duration:

55
55 seconds
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
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
octagonal
FL+FR+FC+BL+BR+BC+SL+SR
downmix
DL+DR

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

  • the name of a standard channel layout (e.g. mono, stereo, 4.0, quad, 5.0, etc.)
  • the name of a single channel (e.g. FL, FR, FC, LFE, etc.)
  • a number of channels, in decimal, followed by 'c', yielding the default channel layout for that number of channels (see the function "av_get_default_channel_layout"). 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_get_channel_layout" 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".
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 / 2^40
E
10^18 / 2^50
Z
10^21 / 2^60
Y
10^24 / 2^70

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.
emu_edge
Do not draw edges.
psnr
Set error[?] variables during encoding.
truncated
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.
cbp
Deprecated, use mpegvideo private options instead.
qprd
Deprecated, use mpegvideo private options instead.
ilme
Apply interlaced motion estimation.
cgop
Use closed gop.
me_method integer (encoding,video)
Set motion estimation method.

Possible values:

zero
zero motion estimation (fastest)
full
full motion estimation (slowest)
epzs
EPZS motion estimation (default)
esa
esa motion estimation (alias for full)
tesa
tesa motion estimation
dia
dia motion estimation (alias for epzs)
log
log motion estimation
phods
phods motion estimation
x1
X1 motion estimation
hex
hex motion estimation
umh
umh motion estimation
iter
iter motion estimation
extradata_size integer
Set extradata size.
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.
rc_strategy integer (encoding,video)
Set ratecontrol method.
b_strategy integer (encoding,video)
Set strategy to choose between I/P/B-frames.
ps integer (encoding,video)
Set RTP payload size in bytes.
mv_bits integer
header_bits integer
i_tex_bits integer
p_tex_bits integer
i_count integer
p_count integer
skip_count integer
misc_bits integer
frame_bits integer
codec_tag integer
bug flags (decoding,video)
Workaround not auto detected encoder bugs.

Possible values:

autodetect
old_msmpeg4
some old lavc generated msmpeg4v3 files (no autodetection)
xvid_ilace
Xvid interlacing bug (autodetected if fourcc==XVIX)
ump4
(autodetected if fourcc==UMP4)
no_padding
padding bug (autodetected)
amv
ac_vlc
illegal vlc bug (autodetected per fourcc)
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
lelim integer (encoding,video)
Set single coefficient elimination threshold for luminance (negative values also consider DC coefficient).
celim integer (encoding,video)
Set single coefficient elimination threshold for chrominance (negative values also consider dc coefficient)
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
mpeg_quant integer (encoding,video)
Use MPEG quantizers instead of H.263.
qsquish float (encoding,video)
How to keep quantizer between qmin and qmax (0 = clip, 1 = use differentiable function).
rc_qmod_amp float (encoding,video)
Set experimental quantizer modulation.
rc_qmod_freq integer (encoding,video)
Set experimental quantizer modulation.
rc_override_count integer
rc_eq string (encoding,video)
Set rate control equation. When computing the expression, besides the standard functions defined in the section 'Expression Evaluation', the following functions are available: bits2qp(bits), qp2bits(qp). Also the following constants are available: iTex pTex tex mv fCode iCount mcVar var isI isP isB avgQP qComp avgIITex avgPITex avgPPTex avgBPTex avgTex.
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).
rc_buf_aggressivity float (encoding,video)
Currently useless.
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.
rc_init_cplx float (encoding,video)
Set initial complexity for 1-pass encoding.
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
simplealpha
ipp
xvidmmx
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
pred integer (encoding,video)
Set prediction method.

Possible values:

left
plane
median
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.
last_pred integer (encoding,video)
Set amount of motion predictors from the previous frame.
preme integer (encoding,video)
Set pre motion estimation.
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.
dtg_active_format integer
me_range integer (encoding,video)
Set limit motion vectors range (1023 for DivX player).
ibias integer (encoding,video)
Set intra quant bias.
pbias integer (encoding,video)
Set inter quant bias.
color_table_id integer
global_quality integer (encoding,audio,video)
coder integer (encoding,video)
Possible values:
vlc
variable length coder / huffman coder
ac
arithmetic coder
raw
raw (no encoding)
rle
run-length coder
deflate
deflate-based coder
context integer (encoding,video)
Set context model.
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
stream_codec_tag integer
sc_threshold integer (encoding,video)
Set scene change threshold.
lmin integer (encoding,video)
Set min lagrange factor (VBR).
lmax integer (encoding,video)
Set max lagrange factor (VBR).
nr integer (encoding,video)
Set noise reduction.
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)
Possible values:
fast
Allow non spec compliant speedup tricks.
sgop
Deprecated, use mpegvideo private options instead.
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.
skiprd
Deprecated, use mpegvideo private options instead.
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.
error integer (encoding,video)
qns integer (encoding,video)
Deprecated, use mpegvideo private options instead.
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.

me_threshold integer (encoding,video)
Set motion estimation threshold.
mb_threshold integer (encoding,video)
Set macroblock threshold.
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)
Possible values:
unknown
aac_main
aac_low
aac_ssr
aac_ltp
aac_he
aac_he_v2
aac_ld
aac_eld
mpeg2_aac_low
mpeg2_aac_he
mpeg4_sp
mpeg4_core
mpeg4_main
mpeg4_asp
dts
dts_es
dts_96_24
dts_hd_hra
dts_hd_ma
level integer (encoding,audio,video)
Possible values:
unknown
lowres integer (decoding,audio,video)
Decode at 1= 1/2, 2=1/4, 3=1/8 resolutions.
skip_threshold integer (encoding,video)
Set frame skip threshold.
skip_factor integer (encoding,video)
Set frame skip factor.
skip_exp integer (encoding,video)
Set frame skip exponent. Negative values behave identical to the corresponding positive ones, except that the score is normalized. Positive values exist primarily for compatibility reasons and are not so useful.
skipcmp integer (encoding,video)
Set frame skip 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
border_mask float (encoding,video)
Increase the quantizer for macroblocks close to borders.
mblmin integer (encoding,video)
Set min macroblock lagrange factor (VBR).
mblmax integer (encoding,video)
Set max macroblock lagrange factor (VBR).
mepc integer (encoding,video)
Set motion estimation bitrate penalty compensation (1.0 = 256).
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.
all
Discard all frames.

Default value is default.

bidir_refine integer (encoding,video)
Refine the two motion vectors used in bidirectional macroblocks.
brd_scale integer (encoding,video)
Downscale frames for dynamic B-frame decision.
keyint_min integer (encoding,video)
Set minimum interval between IDR-frames.
refs integer (encoding,video)
Set reference frames to consider for motion compensation.
chromaoffset integer (encoding,video)
Set chroma qp offset from luma.
trellis integer (encoding,audio,video)
Set rate-distortion optimal quantization.
mv0_threshold integer (encoding,video)
b_sensitivity integer (encoding,video)
Adjust sensitivity of b_frame_strategy 1.
compression_level integer (encoding,audio,video)
min_prediction_order integer (encoding,audio)
max_prediction_order integer (encoding,audio)
timecode_frame_start integer (encoding,video)
Set GOP timecode frame start number, in non drop frame format.
request_channels integer (decoding,audio)
Set desired number of audio channels.
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)
ticks_per_frame integer (decoding/encoding,audio,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
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
MPEG (219*2^(n-8))
pc
jpeg
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 indention:

        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.

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

libdavs2

AVS2-P2/IEEE1857.4 video decoder wrapper.

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

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. 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

dvbsub

Options
compute_clut
-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 comas, 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_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.

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 (num_units_in_display_tick / time_scale) 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.

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.
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 e.

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

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.
insert
remove
sample_aspect_ratio
Set the sample aspect ratio of the stream in the VUI parameters.
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 (num_units_in_tick / time_scale) 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).
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.
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.

A new single global PPS is created, and all of the redundant PPSs within the stream are removed.

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 (num_units_in_tick / time_scale). 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).

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.

See also the text2movsub filter.

mp3decomp

Decompress non-standard compressed MP3 audio headers.

mpeg2_metadata

Modify metadata embedded in an MPEG-2 stream.
display_aspect_ratio
Set the display aspect ratio in the stream.

The following fixed values are supported:

4/3
16/9
221/100

Any other value will result in square pixels being signalled instead (see H.262 section 6.3.3 and table 6-3).

frame_rate
Set the frame rate in the stream. This is constructed from a table of known values combined with a small multiplier and divisor - if the supplied value is not exactly representable, the nearest representable value will be used instead (see H.262 section 6.3.3 and table 6-4).
video_format
Set the video format in the stream (see H.262 section 6.3.6 and table 6-6).
colour_primaries
transfer_characteristics
matrix_coefficients
Set the colour description in the stream (see H.262 section 6.3.6 and tables 6-7, 6-8 and 6-9).

mpeg4_unpack_bframes

Unpack DivX-style packed B-frames.

DivX-style packed B-frames are not valid MPEG-4 and were only a workaround for the broken Video for Windows subsystem. They use more space, can cause minor AV sync issues, require more CPU power to decode (unless the player has some decoded picture queue to compensate the 2,0,2,0 frame per packet style) and cause trouble if copied into a standard container like mp4 or mpeg-ps/ts, because MPEG-4 decoders may not be able to decode them, since they are not valid MPEG-4.

For example to fix an AVI file containing an MPEG-4 stream with DivX-style packed B-frames using ffmpeg, you can use the command:

        ffmpeg -i INPUT.avi -codec copy -bsf:v mpeg4_unpack_bframes OUTPUT.avi

noise

Damages the contents of packets or simply drops them without damaging the container. Can be used for fuzzing or testing error resilience/concealment.

Parameters:

amount
A numeral string, whose value is related to how often output bytes will be modified. Therefore, values below or equal to 0 are forbidden, and the lower the more frequent bytes will be modified, with 1 meaning every byte is modified.
dropamount
A numeral string, whose value is related to how often packets will be dropped. Therefore, values below or equal to 0 are forbidden, and the lower the more frequent packets will be dropped, with 1 meaning every packet is dropped.

The following example applies the modification to every byte but does not drop any packets.

        ffmpeg -i INPUT -c copy -bsf noise[=1] output.mkv

null

This bitstream filter passes the packets through unchanged.

remove_extra

Remove extradata from packets.

It accepts the following parameter:

freq
Set which frame types to remove extradata from.
k
Remove extradata from non-keyframes only.
keyframe
Remove extradata from keyframes only.
e, all
Remove extradata from all frames.

text2movsub

Convert text subtitles to MOV subtitles (as used by the "mov_text" codec) with metadata headers.

See also the mov2textsub filter.

trace_headers

Log trace output containing all syntax elements in the coded stream headers (everything above the level of individual coded blocks). This can be useful for debugging low-level stream issues.

Supports H.264, H.265, MPEG-2 and VP9.

vp9_metadata

Modify metadata embedded in a VP9 stream.
color_space
Set the color space value in the frame header.
unknown
bt601
bt709
smpte170
smpte240
bt2020
rgb
color_range
Set the color range value in the frame header. Note that this cannot be set in RGB streams.
tv
pc

vp9_superframe

Merge VP9 invisible (alt-ref) frames back into VP9 superframes. This fixes merging of split/segmented VP9 streams where the alt-ref frame was split from its visible counterpart.

vp9_superframe_split

Split VP9 superframes into single frames.

vp9_raw_reorder

Given a VP9 stream with correct timestamps but possibly out of order, insert additional show-existing-frame packets to correct the ordering.

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.
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.
keepside (deprecated,inert)
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.
latm (deprecated,inert)
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 1000000 (1 second).

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 indention:

        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.

Format stream specifiers

Format stream specifiers allow selection of one or more streams that match specific properties.

Possible forms of stream specifiers are:

stream_index
Matches the stream with this index.
stream_type[:stream_index]
stream_type is one of following: 'v' for video, 'a' for audio, 's' for subtitle, 'd' for data, and 't' for attachments. If stream_index is given, then it matches the stream number stream_index of this type. Otherwise, it matches all streams of this type.
p:program_id[:stream_index]
If stream_index is given, then it matches the stream with number stream_index in the program with the id program_id. Otherwise, it matches all streams in the program.
#stream_id
Matches the stream by a format-specific ID.

The exact semantics of stream specifiers is defined by the "avformat_match_stream_specifier()" function declared in the libavformat/avformat.h header.

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.

applehttp

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".

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.
-max_fps int
Maximum framerate in frames per second (0 for no limit).
-default_fps int
Default framerate in frames per second when none is specified in the file (0 meaning as fast as possible).

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. It also sets the safe option to 1 if it was -1.

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.
"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.

Options

This demuxer accepts the following option:

safe
If set to 1, reject unsafe file paths. 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.

-1 is equivalent to 1 if the format was automatically probed and 0 otherwise.

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.

flv, live_flv

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.

        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

It accepts the following options:

live_start_index
segment index to start live streams at (negative values are from the end).
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.
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.

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.

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 <http://code.google.com/p/game-music-emu/> for more information.

Some files have multiple tracks. The demuxer will pick the first track by default. The track_index option can be used to select a different track. Track indexes start at 0. The demuxer exports the number of tracks as tracks meta data entry.

For very large files, the max_size option may have to be adjusted.

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/QuickTime

QuickTime / MP4 demuxer.

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.

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.

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

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 milliseconds.

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.

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.

        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.

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

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.
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 for files on slow medium.

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-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.

This protocol accepts the following options:

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).

gopher

Gopher protocol.

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>")
user-agent
This is a deprecated option, you can use user_agent instead it.
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.
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_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>
    

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.

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

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>]

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:

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.

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>

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_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.

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.
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.

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".

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.

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.

Default value is none.

allowed_media_types
Set media types to accept from the server.

The following flags are accepted:

video
audio
data

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.
timeout
Set maximum timeout (in seconds) to wait for incoming connections.

A value of -1 means infinite (default). This option implies the rtsp_flags set to listen.

reorder_queue_size
Set number of packets to buffer for handling of reordered packets.
stimeout
Set socket TCP I/O timeout in microseconds.
user-agent
Override User-Agent header. If not specified, it defaults to the libavformat identifier string.

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.

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.

or

        <options> srt://<hostname>:<port>

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

This protocol accepts the following options.

connect_timeout
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
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
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).
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
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
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.
rw_timeout
Set raise error timeout for read/write optations.

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

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 doesnXt make sense in Rendezvous connection; the result might be that simply one side will override the value from the other side and itXs the matter of luck which one would win
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) thereXs 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

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=1|0
Listen for an incoming connection. 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.

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.
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.

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 64KB. 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.

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.

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
        

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
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. Alternatively you can 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.
bm_v210
This is a deprecated option, you can use raw_format instead. If set to 1, video is captured in 10 bit v210 instead of uyvy422. Not all Blackmagic devices support this option.
raw_format
Set the pixel format of the captured video. Available values are:
uyvy422
yuv422p10
argb
bgra
rgb10
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 mode. Must be unset, half or full. Defaults to unset.
timecode_format
Timecode type to include in the frame and video stream metadata. Must be none, rp188vitc, rp188vitc2, rp188ltc, rp188any, vitc, vitc2, or serial. Defaults to none (not included).
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.

Examples

  • List input devices:

            ffmpeg -f decklink -list_devices 1 -i dummy
        
  • 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 -bm_v210 1 -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.

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

or

        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 -hdvbuffer 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. Defaults to bgr0.
format_modifier
Format modifier to signal on output frames. This is necessary to import correctly into some APIs, but can't be autodetected. 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
        

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. Currently only video data is supported. 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".

libndi_newtek

The libndi_newtek input device provides capture capabilities for using NDI (Network Device Interface, standard created by NewTek).

Input filename is a NDI source name that could be found by sending -find_sources 1 to command line - it has no specific syntax but human-readable formatted.

To enable this input device, you need the NDI SDK and you need to configure with the appropriate "--extra-cflags" and "--extra-ldflags".

Options

find_sources
If set to true, print a list of found/available NDI sources and exit. Defaults to false.
wait_sources
Override time to wait until the number of online sources have changed. Defaults to 0.5.
allow_video_fields
When this flag is false, all video that you receive will be progressive. Defaults to true.
extra_ips
If is set to list of comma separated ip addresses, scan for sources not only using mDNS but also use unicast ip addresses specified by this list.

Examples

  • List input devices:

            ffmpeg -f libndi_newtek -find_sources 1 -i dummy
        
  • List local and remote input devices:

            ffmpeg -f libndi_newtek -extra_ips "192.168.10.10" -find_sources 1 -i dummy
        
  • Restream to NDI:

            ffmpeg -f libndi_newtek -i "DEV-5.INTERNAL.M1STEREO.TV (NDI_SOURCE_NAME_1)" -f libndi_newtek -y NDI_SOURCE_NAME_2
        
  • Restream remote NDI to local NDI:

            ffmpeg -f libndi_newtek -extra_ips "192.168.10.10" -i "DEV-5.REMOTE.M1STEREO.TV (NDI_SOURCE_NAME_1)" -f libndi_newtek -y NDI_SOURCE_NAME_2
        

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
Specify the number of bytes per frame, by default it is set to 1024.
fragment_size
Specify the minimal buffering fragment in PulseAudio, it will affect the audio latency. By default it is unset.
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

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

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.

ich, in_channel_count
Set the number of input channels. Default value is 0. Setting this value is not mandatory if the corresponding channel layout in_channel_layout is set.
och, out_channel_count
Set the number of output channels. Default value is 0. Setting this value is not mandatory if the corresponding channel layout out_channel_layout is set.
uch, used_channel_count
Set the number of used input channels. Default value is 0. 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.
icl, in_channel_layout
ocl, out_channel_layout
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. 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.
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
Set source width.
srch
Set source height.
dstw
Set destination width.
dsth
Set destination height.
src_format
Set source pixel format (must be expressed as an integer).
dst_format
Set destination pixel format (must be expressed as an integer).
src_range
Select source range.
dst_range
Select destination 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.

The name and arguments of the filter are optionally preceded and followed by a list of link labels. A link label allows one to name a link and associate it to a filter output or input pad. The preceding labels in_link_1 ... in_link_N, are associated to the filter input pads, the following labels out_link_1 ... out_link_M, are associated to the output pads.

When two link labels with the same name are found in the filtergraph, a link between the corresponding input and output pad is created.

If an output pad is not labelled, it is linked by default to the first unlabelled input pad of the next filter in the filterchain. For example in the filterchain

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

the split filter instance has two output pads, and the overlay filter instance two input pads. The first output pad of split is labelled "L1", the first input pad of overlay is labelled "L2", and the second output pad of split is linked to the second input pad of overlay, which are both unlabelled.

In a filter description, if the input label of the first filter is not specified, "in" is assumed; if the output label of the last filter is not specified, "out" is assumed.

In a complete filterchain all the unlabelled filter input and output pads must be connected. A filtergraph is considered valid if all the filter input and output pads of all the filterchains are connected.

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

Here is a BNF description of the filtergraph syntax:

        <NAME>             ::= sequence of alphanumeric characters and '_'
        <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"

TIMELINE EDITING

Some filters support a generic enable option. For the filters supporting timeline editing, this option can be set to an expression which is evaluated before sending a frame to the filter. If the evaluation is non-zero, the filter will be enabled, otherwise the frame will be sent unchanged to the next filter in the filtergraph.

The expression accepts the following values:

t
timestamp expressed in seconds, NAN if the input timestamp is unknown
n
sequential number of the input frame, starting from 0
pos
the position in the file of the input frame, NAN if unknown
w
h
width and height of the input frame if video

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

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

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

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

See "ffmpeg -filters" to view which filters have timeline support.

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.

AUDIO FILTERS

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

Below is a description of the currently available audio filters.

acompressor

A compressor is mainly used to reduce the dynamic range of a signal. Especially modern music is mostly compressed at a high ratio to improve the overall loudness. It's done to get the highest attention of a listener, "fatten" the sound and bring more "power" to the track. If a signal is compressed too much it may sound dull or "dead" afterwards or it may start to "pump" (which could be a powerful effect but can also destroy a track completely). The right compression is the key to reach a professional sound and is the high art of mixing and mastering. Because of its complex settings it may take a long time to get the right feeling for this kind of effect.

Compression is done by detecting the volume above a chosen level "threshold" and dividing it by the factor set with "ratio". So if you set the threshold to -12dB and your signal reaches -6dB a ratio of 2:1 will result in a signal at -9dB. Because an exact manipulation of the signal would cause distortion of the waveform the reduction can be levelled over the time. This is done by setting "Attack" and "Release". "attack" determines how long the signal has to rise above the threshold before any reduction will occur and "release" sets the time the signal has to fall below the threshold to reduce the reduction again. Shorter signals than the chosen attack time will be left untouched. The overall reduction of the signal can be made up afterwards with the "makeup" setting. So compressing the peaks of a signal about 6dB and raising the makeup to this level results in a signal twice as loud than the source. To gain a softer entry in the compression the "knee" flattens the hard edge at the threshold in the range of the chosen decibels.

The filter accepts the following options:

level_in
Set input gain. Default is 1. Range is between 0.015625 and 64.
threshold
If a signal of 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.

acontrast

Simple audio dynamic range commpression/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, can be 2nd, 4th or 8th. Default is 4th.

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.

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.

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.
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.
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.
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.
b
Set burst fusion, in percentage of window size. Allowed range is 0 to 10. Default value is 2. If any two samples deteced as noise are spaced less than this value then any sample inbetween those two samples will be also detected as noise.
m
Set overlap method.

It accepts the following values:

a
Select overlap-add method. Even not interpolated samples are slightly changed with this method.
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.

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.
o
Set window overlap, in percentage of window size. Allowed range is from 50 to 95. Default value is 75 percent.
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.
t
Set threshold value. Allowed range is from 1 to 100. Default value is 10. Higher values make clip detection less aggressive.
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.
m
Set overlap method.

It accepts the following values:

a
Select overlap-add method. Even not interpolated samples are slightly changed with this method.
s
Select overlap-save method. Not interpolated samples remain unchanged.

Default value is "a".

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.

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
        

aderivative, aintegral

Compute derivative/integral of audio stream.

Applying both filters one after another produces original audio.

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 sound like a (metallic) robot playing music:

            aecho=0.8:0.88:6:0.4
        
  • A longer delay will sound like an open air concert in the mountains:

            aecho=0.8:0.9:1000:0.3
        
  • Same as above but with one more mountain:

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

aemphasis

Audio emphasis filter creates or restores material directly taken from LPs or emphased CDs with different filter curves. E.g. to store music on vinyl the signal has to be altered by a filter first to even out the disadvantages of this recording medium. Once the material is played back the inverse filter has to be applied to restore the distortion of the frequency response.

The filter accepts the following options:

level_in
Set input gain.
level_out
Set output gain.
mode
Set filter mode. For restoring material use "reproduction" mode, otherwise use "production" mode. Default is "reproduction" mode.
type
Set filter type. Selects medium. Can be one of the following:
col
select Columbia.
emi
select EMI.
bsi
select BSI (78RPM).
riaa
select RIAA.
cd
select Compact Disc (CD).
50fm
select 50Xs (FM).
75fm
select 75Xs (FM).
50kf
select 50Xs (FM-KF).
75kf
select 75Xs (FM-KF).

aeval

Modify an audio signal according to the specified expressions.

This filter accepts one or more expressions (one for each channel), which are evaluated and used to modify a corresponding audio signal.

It accepts the following parameters:

exprs
Set the '|'-separated expressions list for each separate channel. If the number of input channels is greater than the number of expressions, the last specified expression is used for the remaining output channels.
channel_layout, c
Set output channel layout. If not specified, the channel layout is specified by the number of expressions. If set to same, it will use by default the same input channel layout.

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

ch
channel number of the current expression
n
number of the evaluated sample, starting from 0
s
sample rate
t
time of the evaluated sample expressed in seconds
nb_in_channels
nb_out_channels
input and output number of channels
val(CH)
the value of input channel with number CH

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

Examples

  • Half volume:

            aeval=val(ch)/2:c=same
        
  • Invert phase of the second channel:

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

afade

Apply fade-in/out effect to input audio.

A description of the accepted parameters follows.

type, t
Specify the effect type, can be either "in" for fade-in, or "out" for a fade-out effect. Default is "in".
start_sample, ss
Specify the number of the start sample for starting to apply the fade effect. Default is 0.
nb_samples, ns
Specify the number of samples for which the fade effect has to last. At the end of the fade-in effect the output audio will have the same volume as the input audio, at the end of the fade-out transition the output audio will be silence. Default is 44100.
start_time, st
Specify the start time of the fade effect. Default is 0. The value must be specified as a time duration; see the Time duration section in the ffmpeg-utils(1) manual for the accepted syntax. If set this option is used instead of start_sample.
duration, d
Specify the duration of the fade effect. See the Time duration section in the ffmpeg-utils(1) manual for the accepted syntax. At the end of the fade-in effect the output audio will have the same volume as the input audio, at the end of the fade-out transition the output audio will be silence. By default the duration is determined by nb_samples. If set this option is used instead of nb_samples.
curve
Set curve for fade transition.

It accepts the following values:

tri
select triangular, linear slope (default)
qsin
select quarter of sine wave
hsin
select half of sine wave
esin
select exponential sine wave
log
select logarithmic
ipar
select inverted parabola
qua
select quadratic
cub
select cubic
squ
select square root
cbr
select cubic root
par
select parabola
exp
select exponential
iqsin
select inverted quarter of sine wave
ihsin
select inverted half of sine wave
dese
select double-exponential seat
desi
select double-exponential sigmoid
losi
select logistic sigmoid

Examples

  • Fade in first 15 seconds of audio:

            afade=t=in:ss=0:d=15
        
  • Fade out last 25 seconds of a 900 seconds audio:

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

afftdn

Denoise audio samples with FFT.

A description of the accepted parameters follows.

nr
Set the noise reduction in dB, allowed range is 0.01 to 97. Default value is 12 dB.
nf
Set the noise floor in dB, allowed range is -80 to -20. Default value is -50 dB.
nt
Set the noise type.

It accepts the following values:

w
Select white noise.
v
Select vinyl noise.
s
Select shellac noise.
c
Select custom noise, defined in "bn" option.

Default value is white noise.

bn
Set custom band noise for every one of 15 bands. Bands are separated by ' ' or '|'.
rf
Set the residual floor in dB, allowed range is -80 to -20. Default value is -38 dB.
tn
Enable noise tracking. By default is disabled. With this enabled, noise floor is automatically adjusted.
tr
Enable residual tracking. By default is disabled.
om
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.

Commands

This filter supports the following commands:

sample_noise, sn
Start or stop measuring noise profile. Syntax for the command is : "start" or "stop" string. After measuring noise profile is stopped it will be automatically applied in filtering.
noise_reduction, nr
Change noise reduction. Argument is single float number. Syntax for the command is : "noise_reduction"
noise_floor, nf
Change noise floor. Argument is single float number. Syntax for the command is : "noise_floor"
output_mode, om
Change output mode operation. Syntax for the command is : "i", "o" or "n" string.

afftfilt

Apply arbitrary expressions to samples in frequency domain.
real
Set frequency domain real expression for each separate channel separated by '|'. Default is "1". If the number of input channels is greater than the number of expressions, the last specified expression is used for the remaining output channels.
imag
Set frequency domain imaginary expression for each separate channel separated by '|'. If not set, real option is used.

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

sr
sample rate
b
current frequency bin number
nb
number of available bins
ch
channel number of the current expression
chs
number of channels
pts
current frame pts
win_size
Set window size.

It accepts the following values:

w16
w32
w64
w128
w256
w512
w1024
w2048
w4096
w8192
w16384
w32768
w65536

Default is "w4096"

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

Examples

Leave almost only low frequencies in audio:

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

afir

Apply an arbitrary Frequency 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 and ambisonics.

This filter uses second stream as FIR coefficients. If second stream holds single channel, it will be used for all input channels in first stream, otherwise number of channels in second stream must be same as number of channels in 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.
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 reponse, magnitude(magenta) and 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.

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
    

aformat

Set output format constraints for the input audio. The framework will negotiate the most appropriate format to minimize conversions.

It accepts the following parameters:

sample_fmts
A '|'-separated list of requested sample formats.
sample_rates
A '|'-separated list of requested sample rates.
channel_layouts
A '|'-separated list of requested channel layouts.

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

If a parameter is omitted, all values are allowed.

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

        aformat=sample_fmts=u8|s16:channel_layouts=stereo

agate

A gate is mainly used to reduce lower parts of a signal. This kind of signal processing reduces disturbing noise between useful signals.

Gating is done by detecting the volume below a chosen level threshold and dividing it by the factor set with ratio. The bottom of the noise floor is set via range. Because an exact manipulation of the signal would cause distortion of the waveform the reduction can be levelled over time. This is done by setting attack and release.

attack determines how long the signal has to fall below the threshold before any reduction will occur and release sets the time the signal has to rise above the threshold to reduce the reduction again. Shorter signals than the chosen attack time will be left untouched.

level_in
Set input level before filtering. Default is 1. Allowed range is from 0.015625 to 64.
range
Set the level of gain reduction when the signal is below the threshold. Default is 0.06125. Allowed range is from 0 to 1.
threshold
If a signal rises above this level the gain reduction is released. Default is 0.125. Allowed range is from 0 to 1.
ratio
Set a ratio by which the signal is reduced. Default is 2. Allowed range is from 1 to 9000.
attack
Amount of milliseconds the signal has to rise above the threshold before gain reduction stops. Default is 20 milliseconds. Allowed range is from 0.01 to 9000.
release
Amount of milliseconds the signal has to fall below the threshold before the reduction is increased again. Default is 250 milliseconds. Allowed range is from 0.01 to 9000.
makeup
Set amount of amplification of signal after processing. Default is 1. Allowed range is from 1 to 64.
knee
Curve the sharp knee around the threshold to enter gain reduction more softly. Default is 2.828427125. Allowed range is from 1 to 8.
detection
Choose if exact signal should be taken for detection or an RMS like one. Default is "rms". Can be "peak" or "rms".
link
Choose if the average level between all channels or the louder channel affects the reduction. Default is "average". Can be "average" or "maximum".

aiir

Apply an arbitrary Infinite Impulse Response filter.

It accepts the following parameters:

z
Set numerator/zeros coefficients.
p
Set denominator/poles coefficients.
k
Set channels gains.
dry_gain
Set input gain.
wet_gain
Set output gain.
f
Set coefficients format.
tf
transfer function
zp
Z-plane zeros/poles, cartesian (default)
pr
Z-plane zeros/poles, polar radians
pd
Z-plane zeros/poles, polar degrees
r
Set kind of processing. Can be "d" - direct or "s" - serial cascading. Defauls is "s".
e
Set filtering precision.
dbl
double-precision floating-point (default)
flt
single-precision floating-point
i32
32-bit integers
i16
16-bit integers
response
Show IR frequency reponse, magnitude and phase 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" 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 arround 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
        

alimiter

The limiter prevents an input signal from rising over a desired threshold. This limiter uses lookahead technology to prevent your signal from distorting. It means that there is a small delay after the signal is processed. Keep in mind that the delay it produces is the attack time you set.

The filter accepts the following options:

level_in
Set input gain. Default is 1.
level_out
Set output gain. Default is 1.
limit
Don't let signals above this level pass the limiter. Default is 1.
attack
The limiter will reach its attenuation level in this amount of time in milliseconds. Default is 5 milliseconds.
release
Come back from limiting to attenuation 1.0 in this amount of milliseconds. Default is 50 milliseconds.
asc
When gain reduction is always needed ASC takes care of releasing to an average reduction level rather than reaching a reduction of 0 in the release time.
asc_level
Select how much the release time is affected by ASC, 0 means nearly no changes in release time while 1 produces higher release times.
level
Auto level output signal. Default is enabled. This normalizes audio back to 0dB if enabled.

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

allpass

Apply a two-pole all-pass filter with central frequency (in Hz) frequency, and filter-width width. An all-pass filter changes the audio's frequency to phase relationship without changing its frequency to amplitude relationship.

The filter accepts the following options:

frequency, f
Set frequency in Hz.
width_type, 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.
channels, c
Specify which channels to filter, by default all available are filtered.

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"

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.

amerge

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

The filter accepts the following options:

inputs
Set the number of inputs. Default is 2.

If the channel layouts of the inputs are disjoint, and therefore compatible, the channel layout of the output will be set accordingly and the channels will be reordered as necessary. If the channel layouts of the inputs are not disjoint, the output will have all the channels of the first input then all the channels of the second input, in that order, and the channel layout of the output will be the default value corresponding to the total number of channels.

For example, if the first input is in 2.1 (FL+FR+LF) and the second input is FC+BL+BR, then the output will be in 5.1, with the channels in the following order: a1, a2, b1, a3, b2, b3 (a1 is the first channel of the first input, b1 is the first channel of the second input).

On the other hand, if both input are in stereo, the output channels will be in the default order: a1, a2, b1, b2, and the channel layout will be arbitrarily set to 4.0, which may or may not be the expected value.

All inputs must have the same sample rate, and format.

If inputs do not have the same duration, the output will stop with the shortest.

Examples

  • Merge two mono files into a stereo stream:

            amovie=left.wav [l] ; amovie=right.mp3 [r] ; [l] [r] amerge
        
  • Multiple merges assuming 1 video stream and 6 audio streams in input.mkv:

            ffmpeg -i input.mkv -filter_complex "[0:1][0:2][0:3][0:4][0:5][0:6] amerge=inputs=6" -c:a pcm_s16le output.mkv
        

amix

Mixes multiple audio inputs into a single output.

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

For example

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

will mix 3 input audio streams to a single output with the same duration as the first input and a dropout transition time of 3 seconds.

It accepts the following parameters:

inputs
The number of inputs. If unspecified, it defaults to 2.
duration
How to determine the end-of-stream.
longest
The duration of the longest input. (default)
shortest
The duration of the shortest input.
first
The duration of the first input.
dropout_transition
The transition time, in seconds, for volume renormalization when an input stream ends. The default value is 2 seconds.
weights
Specify weight of each input audio stream as sequence. Each weight is separated by space. By default all inputs have same weight.

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 herz. gain set new gain parameter in dB.

Full filter invocation with asendcmd may look like this: asendcmd=c='4.0 anequalizer change 0|f=200|w=50|g=1',anequalizer=...

anull

Pass the audio source unchanged to the output.

apad

Pad the end of an audio stream with silence.

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

A description of the accepted options follows.

packet_size
Set silence packet size. Default value is 4096.
pad_len
Set the number of samples of silence to add to the end. After the value is reached, the stream is terminated. This option is mutually exclusive with whole_len.
whole_len
Set the minimum total number of samples in the output audio stream. If the value is longer than the input audio length, silence is added to the end, until the value is reached. This option is mutually exclusive with pad_len.

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

Examples

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

            apad=pad_len=1024
        
  • Make sure the audio output will contain at least 10000 samples, pad the input with silence if required:

            apad=whole_len=10000
        
  • Use ffmpeg to pad the audio input with silence, so that the video stream will always result the shortest and will be converted until the end in the output file when using the shortest option:

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

aphaser

Add a phasing effect to the input audio.

A phaser filter creates series of peaks and troughs in the frequency spectrum. The position of the peaks and troughs are modulated so that they vary over time, creating a sweeping effect.

A description of the accepted parameters follows.

in_gain
Set input gain. Default is 0.4.
out_gain
Set output gain. Default is 0.74
delay
Set delay in milliseconds. Default is 3.0.
decay
Set decay. Default is 0.4.
speed
Set modulation speed in Hz. Default is 0.5.
type
Set modulation type. Default is triangular.

It accepts the following values:

triangular, t
sinusoidal, s

apulsator

Audio pulsator is something between an autopanner and a tremolo. But it can produce funny stereo effects as well. Pulsator changes the volume of the left and right channel based on a LFO (low frequency oscillator) with different waveforms and shifted phases. This filter have the ability to define an offset between left and right channel. An offset of 0 means that both LFO shapes match each other. The left and right channel are altered equally - a conventional tremolo. An offset of 50% means that the shape of the right channel is exactly shifted in phase (or moved backwards about half of the frequency) - pulsator acts as an autopanner. At 1 both curves match again. Every setting in between moves the phase shift gapless between all stages and produces some "bypassing" sounds with sine and triangle waveforms. The more you set the offset near 1 (starting from the 0.5) the faster the signal passes from the left to the right speaker.

The filter accepts the following options:

level_in
Set input gain. By default it is 1. Range is [0.015625 - 64].
level_out
Set output gain. By default it is 1. Range is [0.015625 - 64].
mode
Set waveform shape the LFO will use. Can be one of: sine, triangle, square, sawup or sawdown. Default is sine.
amount
Set modulation. Define how much of original signal is affected by the LFO.
offset_l
Set left channel offset. Default is 0. Allowed range is [0 - 1].
offset_r
Set right channel offset. Default is 0.5. Allowed range is [0 - 1].
width
Set pulse width. Default is 1. Allowed range is [0 - 2].
timing
Set possible timing mode. Can be one of: bpm, ms or hz. Default is hz.
bpm
Set bpm. Default is 120. Allowed range is [30 - 300]. Only used if timing is set to bpm.
ms
Set ms. Default is 500. Allowed range is [10 - 2000]. Only used if timing is set to ms.
hz
Set frequency in Hz. Default is 2. Allowed range is [0.01 - 100]. Only used if timing is set to hz.

aresample

Resample the input audio to the specified parameters, using the libswresample library. If none are specified then the filter will automatically convert between its input and output.

This filter is also able to stretch/squeeze the audio data to make it match the timestamps or to inject silence / cut out audio to make it match the timestamps, do a combination of both or do neither.

The filter accepts the syntax [sample_rate:]resampler_options, where sample_rate expresses a sample rate and resampler_options is a list of key=value pairs, separated by ":". See the "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
    

asetnsamples

Set the number of samples per each output audio frame.

The last output packet may contain a different number of samples, as the filter will flush all the remaining samples when the input audio signals its end.

The filter accepts the following options:

nb_out_samples, n
Set the number of frames per each output audio frame. The number is intended as the number of samples per each channel. Default value is 1024.
pad, p
If set to 1, the filter will pad the last audio frame with zeroes, so that the last frame will contain the same number of samples as the previous ones. Default value is 1.

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

        asetnsamples=n=1234:p=0

asetrate

Set the sample rate without altering the PCM data. This will result in a change of speed and pitch.

The filter accepts the following options:

sample_rate, r
Set the output sample rate. Default is 44100 Hz.

ashowinfo

Show a line containing various information for each input audio frame. The input audio is not modified.

The shown line contains a sequence of key/value pairs of the form key:value.

The following values are shown in the output:

n
The (sequential) number of the input frame, starting from 0.
pts
The presentation timestamp of the input frame, in time base units; the time base depends on the filter input pad, and is usually 1/sample_rate.
pts_time
The presentation timestamp of the input frame in seconds.
pos
position of the frame in the input stream, -1 if this information in unavailable and/or meaningless (for example in case of synthetic audio)
fmt
The sample format.
chlayout
The channel layout.
rate
The sample rate for the audio frame.
nb_samples
The number of samples (per channel) in the frame.
checksum
The Adler-32 checksum (printed in hexadecimal) of the audio data. For planar audio, the data is treated as if all the planes were concatenated.
plane_checksums
A list of Adler-32 checksums for each data plane.

astats

Display time domain statistical information about the audio channels. Statistics are calculated and displayed for each audio channel and, where applicable, an overall figure is also given.

It accepts the following option:

length
Short window length in seconds, used for peak and trough RMS measurement. Default is 0.05 (50 milliseconds). Allowed range is "[0.01 - 10]".
metadata
Set metadata injection. All the metadata keys are prefixed with "lavfi.astats.X", where "X" is channel number starting from 1 or string "Overall". Default is disabled.

Available keys for each channel are: DC_offset Min_level Max_level Min_difference Max_difference Mean_difference RMS_difference Peak_level RMS_peak RMS_trough Crest_factor Flat_factor Peak_count Bit_depth Dynamic_range Zero_crossings Zero_crossings_rate

and for Overall: DC_offset Min_level Max_level Min_difference Max_difference Mean_difference RMS_difference Peak_level RMS_level RMS_peak RMS_trough Flat_factor Peak_count Bit_depth Number_of_samples

For example full key look like this "lavfi.astats.1.DC_offset" or this "lavfi.astats.Overall.Peak_count".

For description what each key means read below.

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

A description of each shown parameter follows:

DC offset
Mean amplitude displacement from zero.
Min level
Minimal sample level.
Max level
Maximal sample level.
Min difference
Minimal difference between two consecutive samples.
Max difference
Maximal difference between two consecutive samples.
Mean difference
Mean difference between two consecutive samples. The average of each difference between two consecutive samples.
RMS difference
Root Mean Square difference between two consecutive samples.
Peak level dB
RMS level dB
Standard peak and RMS level measured in dBFS.
RMS peak dB
RMS trough dB
Peak and trough values for RMS level measured over a short window.
Crest factor
Standard ratio of peak to RMS level (note: not in dB).
Flat factor
Flatness (i.e. consecutive samples with the same value) of the signal at its peak levels (i.e. either Min level or Max level).
Peak count
Number of occasions (not the number of samples) that the signal attained either Min level or Max level.
Bit depth
Overall bit depth of audio. Number of bits used for each sample.
Dynamic range
Measured dynamic range of audio in dB.
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.

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)
        

atrim

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

It accepts the following parameters:

start
Timestamp (in seconds) of the start of the section to keep. I.e. the audio sample with the timestamp start will be the first sample in the output.
end
Specify time of the first audio sample that will be dropped, i.e. the audio sample immediately preceding the one with the timestamp end will be the last sample in the output.
start_pts
Same as start, except this option sets the start timestamp in samples instead of seconds.
end_pts
Same as end, except this option sets the end timestamp in samples instead of seconds.
duration
The maximum duration of the output in seconds.
start_sample
The number of the first sample that should be output.
end_sample
The number of the first sample that should be dropped.

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

Note that the first two sets of the start/end options and the duration option look at the frame timestamp, while the _sample options simply count the samples that pass through the filter. So start/end_pts and start/end_sample will give different results when the timestamps are wrong, inexact or do not start at zero. Also note that this filter does not modify the timestamps. If you wish to have the output timestamps start at zero, insert the asetpts filter after the atrim filter.

If multiple start or end options are set, this filter tries to be greedy and keep all samples that match at least one of the specified constraints. To keep only the part that matches all the constraints at once, chain multiple atrim filters.

The defaults are such that all the input is kept. So it is possible to set e.g. just the end values to keep everything before the specified time.

Examples:

  • Drop everything except the second minute of input:

            ffmpeg -i INPUT -af atrim=60:120
        
  • Keep only the first 1000 samples:

            ffmpeg -i INPUT -af atrim=end_sample=1000
        

bandpass

Apply a two-pole Butterworth band-pass filter with central frequency frequency, and (3dB-point) band-width width. The csg option selects a constant skirt gain (peak gain = Q) instead of the default: constant 0dB peak gain. The filter roll off at 6dB per octave (20dB per decade).

The filter accepts the following options:

frequency, f
Set the filter's central frequency. Default is 3000.
csg
Constant skirt gain if set to 1. Defaults to 0.
width_type, 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.
channels, c
Specify which channels to filter, by default all available are filtered.

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"

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.
channels, c
Specify which channels to filter, by default all available are filtered.

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"

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.
channels, c
Specify which channels to filter, by default all available are filtered.

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"

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"

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 location. Because the front of sound wave has fixed speed in normal conditions, the phasing of microphones can vary and depends on their location and interposition. The best sound mix can be achieved when these microphones are in phase (synchronized). Note that distance of ~30 cm between microphones makes one microphone to capture signal in antiphase to another microphone. That makes the final mix sounding moody. This filter helps to solve phasing problems by adding different delays to each microphone track and make them synchronized.

The best result can be reached when you take one track as base and synchronize other tracks one by one with it. Remember that synchronization/delay tolerance depends on sample rate, too. Higher sample rates will give more tolerance.

It accepts the following parameters:

mm
Set millimeters distance. This is compensation distance for fine tuning. Default is 0.
cm
Set cm distance. This is compensation distance for tightening distance setup. Default is 0.
m
Set meters distance. This is compensation distance for hard distance setup. Default is 0.
dry
Set dry amount. Amount of unprocessed (dry) signal. Default is 0.
wet
Set wet amount. Amount of processed (wet) signal. Default is 1.
temp
Set temperature degree in Celsius. This is the temperature of the environment. Default is 20.

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.
level_in
Set input gain. Default is 0.9.
level_out
Set output gain. Default is 1.

crystalizer

Simple algorithm to expand audio dynamic range.

The filter accepts the following options:

i
Sets the intensity of effect (default: 2.0). Must be in range between 0.0 (unchanged sound) to 10.0 (maximum effect).
c
Enable clipping. By default is enabled.

dcshift

Apply a DC shift to the audio.

This can be useful to remove a DC offset (caused perhaps by a hardware problem in the recording chain) from the audio. The effect of a DC offset is reduced headroom and hence volume. The astats filter can be used to determine if a signal has a DC offset.

shift
Set the DC shift, allowed range is [-1, 1]. It indicates the amount to shift the audio.
limitergain
Optional. It should have a value much less than 1 (e.g. 0.05 or 0.02) and is used to prevent clipping.

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.

f
Set the frame length in milliseconds. In range from 10 to 8000 milliseconds. Default is 500 milliseconds. The Dynamic Audio Normalizer processes the input audio in small chunks, referred to as frames. This is required, because a peak magnitude has no meaning for just a single sample value. Instead, we need to determine the peak magnitude for a contiguous sequence of sample values. While a "standard" normalizer would simply use the peak magnitude of the complete file, the Dynamic Audio Normalizer determines the peak magnitude individually for each frame. The length of a frame is specified in milliseconds. By default, the Dynamic Audio Normalizer uses a frame length of 500 milliseconds, which has been found to give good results with most files. Note that the exact frame length, in number of samples, will be determined automatically, based on the sampling rate of the individual input audio file.
g
Set the Gaussian filter window size. In range from 3 to 301, must be odd number. Default is 31. Probably the most important parameter of the Dynamic Audio Normalizer is the "window size" of the Gaussian smoothing filter. The filter's window size is specified in frames, centered around the current frame. For the sake of simplicity, this must be an odd number. Consequently, the default value of 31 takes into account the current frame, as well as the 15 preceding frames and the 15 subsequent frames. Using a larger window results in a stronger smoothing effect and thus in less gain variation, i.e. slower gain adaptation. Conversely, using a smaller window results in a weaker smoothing effect and thus in more gain variation, i.e. faster gain adaptation. In other words, the more you increase this value, the more the Dynamic Audio Normalizer will behave like a "traditional" normalization filter. On the contrary, the more you decrease this value, the more the Dynamic Audio Normalizer will behave like a dynamic range compressor.
p
Set the target peak value. This specifies the highest permissible magnitude level for the normalized audio input. This filter will try to approach the target peak magnitude as closely as possible, but at the same time it also makes sure that the normalized signal will never exceed the peak magnitude. A frame's maximum local gain factor is imposed directly by the target peak magnitude. The default value is 0.95 and thus leaves a headroom of 5%*. It is not recommended to go above this value.
m
Set the maximum gain factor. In range from 1.0 to 100.0. Default is 10.0. The Dynamic Audio Normalizer determines the maximum possible (local) gain factor for each input frame, i.e. the maximum gain factor that does not result in clipping or distortion. The maximum gain factor is determined by the frame's highest magnitude sample. However, the Dynamic Audio Normalizer additionally bounds the frame's maximum gain factor by a predetermined (global) maximum gain factor. This is done in order to avoid excessive gain factors in "silent" or almost silent frames. By default, the maximum gain factor is 10.0, For most inputs the default value should be sufficient and it usually is not recommended to increase this value. Though, for input with an extremely low overall volume level, it may be necessary to allow even higher gain factors. Note, however, that the Dynamic Audio Normalizer does not simply apply a "hard" threshold (i.e. cut off values above the threshold). Instead, a "sigmoid" threshold function will be applied. This way, the gain factors will smoothly approach the threshold value, but never exceed that value.
r
Set the target RMS. In range from 0.0 to 1.0. Default is 0.0 - disabled. By default, the Dynamic Audio Normalizer performs "peak" normalization. This means that the maximum local gain factor for each frame is defined (only) by the frame's highest magnitude sample. This way, the samples can be amplified as much as possible without exceeding the maximum signal level, i.e. without clipping. Optionally, however, the Dynamic Audio Normalizer can also take into account the frame's root mean square, abbreviated RMS. In electrical engineering, the RMS is commonly used to determine the power of a time-varying signal. It is therefore considered that the RMS is a better approximation of the "perceived loudness" than just looking at the signal's peak magnitude. Consequently, by adjusting all frames to a constant RMS value, a uniform "perceived loudness" can be established. If a target RMS value has been specified, a frame's local gain factor is defined as the factor that would result in exactly that RMS value. Note, however, that the maximum local gain factor is still restricted by the frame's highest magnitude sample, in order to prevent clipping.
n
Enable channels coupling. By default is enabled. By default, the Dynamic Audio Normalizer will amplify all channels by the same amount. This means the same gain factor will be applied to all channels, i.e. the maximum possible gain factor is determined by the "loudest" channel. However, in some recordings, it may happen that the volume of the different channels is uneven, e.g. one channel may be "quieter" than the other one(s). In this case, this option can be used to disable the channel coupling. This way, the gain factor will be determined independently for each channel, depending only on the individual channel's highest magnitude sample. This allows for harmonizing the volume of the different channels.
c
Enable DC bias correction. By default is disabled. An audio signal (in the time domain) is a sequence of sample values. In the Dynamic Audio Normalizer these sample values are represented in the -1.0 to 1.0 range, regardless of the original input format. Normally, the audio signal, or "waveform", should be centered around the zero point. That means if we calculate the mean value of all samples in a file, or in a single frame, then the result should be 0.0 or at least very close to that value. If, however, there is a significant deviation of the mean value from 0.0, in either positive or negative direction, this is referred to as a DC bias or DC offset. Since a DC bias is clearly undesirable, the Dynamic Audio Normalizer provides optional DC bias correction. With DC bias correction enabled, the Dynamic Audio Normalizer will determine the mean value, or "DC correction" offset, of each input frame and subtract that value from all of the frame's sample values which ensures those samples are centered around 0.0 again. Also, in order to avoid "gaps" at the frame boundaries, the DC correction offset values will be interpolated smoothly between neighbouring frames.
b
Enable alternative boundary mode. By default is disabled. The Dynamic Audio Normalizer takes into account a certain neighbourhood around each frame. This includes the preceding frames as well as the subsequent frames. However, for the "boundary" frames, located at the very beginning and at the very end of the audio file, not all neighbouring frames are available. In particular, for the first few frames in the audio file, the preceding frames are not known. And, similarly, for the last few frames in the audio file, the subsequent frames are not known. Thus, the question arises which gain factors should be assumed for the missing frames in the "boundary" region. The Dynamic Audio Normalizer implements two modes to deal with this situation. The default boundary mode assumes a gain factor of exactly 1.0 for the missing frames, resulting in a smooth "fade in" and "fade out" at the beginning and at the end of the input, respectively.
s
Set the compress factor. In range from 0.0 to 30.0. Default is 0.0. By default, the Dynamic Audio Normalizer does not apply "traditional" compression. This means that signal peaks will not be pruned and thus the full dynamic range will be retained within each local neighbourhood. However, in some cases it may be desirable to combine the Dynamic Audio Normalizer's normalization algorithm with a more "traditional" compression. For this purpose, the Dynamic Audio Normalizer provides an optional compression (thresholding) function. If (and only if) the compression feature is enabled, all input frames will be processed by a soft knee thresholding function prior to the actual normalization process. Put simply, the thresholding function is going to prune all samples whose magnitude exceeds a certain threshold value. However, the Dynamic Audio Normalizer does not simply apply a fixed threshold value. Instead, the threshold value will be adjusted for each individual frame. In general, smaller parameters result in stronger compression, and vice versa. Values below 3.0 are not recommended, because audible distortion may appear.

earwax

Make audio easier to listen to on headphones.

This filter adds `cues' to 44.1kHz stereo (i.e. audio CD format) audio so that when listened to on headphones the stereo image is moved from inside your head (standard for headphones) to outside and in front of the listener (standard for speakers).

Ported from SoX.

equalizer

Apply a two-pole peaking equalisation (EQ) filter. With this filter, the signal-level at and around a selected frequency can be increased or decreased, whilst (unlike bandpass and bandreject filters) that at all other frequencies is unchanged.

In order to produce complex equalisation curves, this filter can be given several times, each with a different central frequency.

The filter accepts the following options:

frequency, f
Set the filter's central frequency in Hz.
width_type, 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.
channels, c
Specify which channels to filter, by default all available are filtered.

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"

extrastereo

Linearly increases the difference between left and right channels which adds some sort of "live" effect to playback.

The filter accepts the following options:

m
Sets the difference coefficient (default: 2.5). 0.0 means mono sound (average of both channels), with 1.0 sound will be unchanged, with -1.0 left and right channels will be swapped.
c
Enable clipping. By default is enabled.

firequalizer

Apply FIR Equalization using arbitrary frequency response.

The filter accepts the following option:

gain
Set gain curve equation (in dB). The expression can contain variables:
f
the evaluated frequency
sr
sample rate
ch
channel number, set to 0 when multichannels evaluation is disabled
chid
channel id, see libavutil/channel_layout.h, set to the first channel id when multichannels evaluation is disabled
chs
number of channels
chlayout
channel_layout, see libavutil/channel_layout.h

and functions:

gain_interpolate(f)
interpolate gain on frequency f based on gain_entry
cubic_interpolate(f)
same as gain_interpolate, but smoother

This option is also available as command. Default is gain_interpolate(f).

gain_entry
Set gain entry for gain_interpolate function. The expression can contain functions:
entry(f, g)
store gain entry at frequency f with value g

This option is also available as command.

delay
Set filter delay in seconds. Higher value means more accurate. Default is 0.01.
accuracy
Set filter accuracy in Hz. Lower value means more accurate. Default is 5.
wfunc
Set window function. Acceptable values are:
rectangular
rectangular window, useful when gain curve is already smooth
hann
hann window (default)
hamming
hamming window
blackman
blackman window
nuttall3
3-terms continuous 1st derivative nuttall window
mnuttall3
minimum 3-terms discontinuous nuttall window
nuttall
4-terms continuous 1st derivative nuttall window
bnuttall
minimum 4-terms discontinuous nuttall (blackman-nuttall) window
bharris
blackman-harris window
tukey
tukey window
fixed
If enabled, use fixed number of audio samples. This improves speed when filtering with large delay. Default is disabled.
multi
Enable multichannels evaluation on gain. Default is disabled.
zero_phase
Enable zero phase mode by subtracting timestamp to compensate delay. Default is disabled.
scale
Set scale used by gain. Acceptable values are:
linlin
linear frequency, linear gain
linlog
linear frequency, logarithmic (in dB) gain (default)
loglin
logarithmic (in octave scale where 20 Hz is 0) frequency, linear gain
loglog
logarithmic frequency, logarithmic gain
dumpfile
Set file for dumping, suitable for gnuplot.
dumpscale
Set scale for dumpfile. Acceptable values are same with scale option. Default is linlog.
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 -lavfi-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],[a:0][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 -lavfi-complex "amovie=minp.wav[hrirs],[a:0][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.
channels, c
Specify which channels to filter, by default all available are filtered.

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"

join

Join multiple input streams into one multi-channel stream.

It accepts the following parameters:

inputs
The number of input streams. It defaults to 2.
channel_layout
The desired output channel layout. It defaults to stereo.
map
Map channels from inputs to output. The argument is a '|'-separated list of mappings, each in the "input_idx.in_channel-out_channel" form. input_idx is the 0-based index of the input stream. in_channel can be either the name of the input channel (e.g. FL for front left) or its index in the specified input stream. out_channel is the name of the output channel.

The filter will attempt to guess the mappings when they are not specified explicitly. It does so by first trying to find an unused matching input channel and if that fails it picks the first unused input channel.

Join 3 inputs (with properly set channel layouts):

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

Build a 5.1 output from 6 single-channel streams:

        ffmpeg -i fl -i fr -i fc -i sl -i sr -i lfe -filter_complex
        'join=inputs=6:channel_layout=5.1:map=0.0-FL|1.0-FR|2.0-FC|3.0-SL|4.0-SR|5.0-LFE'
        out

ladspa

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

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

file, f
Specifies the name of LADSPA plugin library to load. If the environment variable LADSPA_PATH is defined, the LADSPA plugin is searched in each one of the directories specified by the colon separated list in LADSPA_PATH, otherwise in the standard LADSPA paths, which are in this order: HOME/.ladspa/lib/, /usr/local/lib/ladspa/, /usr/lib/ladspa/.
plugin, p
Specifies the plugin within the library. Some libraries contain only one plugin, but others contain many of them. If this is not set filter will list all available plugins within the specified library.
controls, c
Set the '|' separated list of controls which are zero or more floating point values that determine the behavior of the loaded plugin (for example delay, threshold or gain). Controls need to be defined using the following syntax: c0=value0|c1=value1|c2=value2|..., where valuei is the value set on the i-th control. Alternatively they can be also defined using the following syntax: value0|value1|value2|..., where valuei is the value set on the i-th control. If controls is set to "help", all available controls and their valid ranges are printed.
sample_rate, s
Specify the sample rate, default to 44100. Only used if plugin have zero inputs.
nb_samples, n
Set the number of samples per channel per each output frame, default is 1024. Only used if plugin have zero inputs.
duration, d
Set the minimum duration of the sourced audio. See the Time duration section in the ffmpeg-utils(1) manual for the accepted syntax. Note that the resulting duration may be greater than the specified duration, as the generated audio is always cut at the end of a complete frame. If not specified, or the expressed duration is negative, the audio is supposed to be generated forever. Only used if plugin have zero inputs.

Examples

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

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

            ladspa=f=vcf:p=vcf_notch:c=help
        
  • Simulate low quality audio equipment using "Computer Music Toolkit" (CMT) plugin library:

            ladspa=file=cmt:plugin=lofi:controls=c0=22|c1=12|c2=12
        
  • Add reverberation to the audio using TAP-plugins (Tom's Audio Processing plugins):

            ladspa=file=tap_reverb:tap_reverb
        
  • Generate white noise, with 0.2 amplitude:

            ladspa=file=cmt:noise_source_white:c=c0=.2
        
  • Generate 20 bpm clicks using plugin "C* Click - Metronome" from the "C* Audio Plugin Suite" (CAPS) library:

            ladspa=file=caps:Click:c=c1=20'
        
  • Apply "C* Eq10X2 - Stereo 10-band equaliser" effect:

            ladspa=caps:Eq10X2:c=c0=-48|c9=-24|c3=12|c4=2
        
  • Increase volume by 20dB using fast lookahead limiter from Steve Harris "SWH Plugins" collection:

            ladspa=fast_lookahead_limiter_1913:fastLookaheadLimiter:20|0|2
        
  • Attenuate low frequencies using Multiband EQ from Steve Harris "SWH Plugins" collection:

            ladspa=mbeq_1197:mbeq:-24|-24|-24|0|0|0|0|0|0|0|0|0|0|0|0
        
  • 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. To accurately detect true peaks, the audio stream will be upsampled to 192 kHz unless the normalization mode is linear. 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 - 20.0. Default value is 7.0.
TP, tp
Set maximum true peak. Range is -9.0 - +0.0. Default value is -2.0.
measured_I, measured_i
Measured IL of input file. Range is -99.0 - +0.0.
measured_LRA, measured_lra
Measured LRA of input file. Range is 0.0 - 99.0.
measured_TP, measured_tp
Measured true peak of input file. Range is -99.0 - +99.0.
measured_thresh
Measured threshold of input file. Range is -99.0 - +0.0.
offset
Set offset gain. Gain is applied before the true-peak limiter. Range is -99.0 - +99.0. Default is +0.0.
linear
Normalize linearly if possible. measured_I, measured_LRA, measured_TP, and measured_thresh must also to be specified in order to use this mode. Options are true or false. Default is true.
dual_mono
Treat mono input files as "dual-mono". If a mono file is intended for playback on a stereo system, its EBU R128 measurement will be perceptually incorrect. If set to "true", this option will compensate for this effect. Multi-channel input files are not affected by this option. Options are true or false. Default is false.
print_format
Set print format for stats. Options are summary, json, or none. Default value is none.

lowpass

Apply a low-pass filter with 3dB point frequency. The filter can be either single-pole or double-pole (the default). The filter roll off at 6dB per pole per octave (20dB per pole per decade).

The filter accepts the following options:

frequency, f
Set frequency in Hz. Default is 500.
poles, p
Set number of poles. Default is 2.
width_type, 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.
channels, c
Specify which channels to filter, by default all available are filtered.

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"

lv2

Load a LV2 (LADSPA Version 2) plugin.

To enable compilation of this filter you need to configure FFmpeg with "--enable-lv2".

plugin, p
Specifies the plugin URI. You may need to escape ':'.
controls, c
Set the '|' separated list of controls which are zero or more floating point values that determine the behavior of the loaded plugin (for example delay, threshold or gain). If controls is set to "help", all available controls and their valid ranges are printed.
sample_rate, s
Specify the sample rate, default to 44100. Only used if plugin have zero inputs.
nb_samples, n
Set the number of samples per channel per each output frame, default is 1024. Only used if plugin have zero inputs.
duration, d
Set the minimum duration of the sourced audio. See the Time duration section in the ffmpeg-utils(1) manual for the accepted syntax. Note that the resulting duration may be greater than the specified duration, as the generated audio is always cut at the end of a complete frame. If not specified, or the expressed duration is negative, the audio is supposed to be generated forever. Only used if plugin have zero inputs.

Examples

  • Apply bass enhancer plugin from Calf:

            lv2=p=http\\\\://calf.sourceforge.net/plugins/BassEnhancer:c=amount=2
        
  • Apply vinyl plugin from Calf:

            lv2=p=http\\\\://calf.sourceforge.net/plugins/Vinyl:c=drone=0.2|aging=0.5
        
  • Apply bit crusher plugin from ArtyFX:

            lv2=p=http\\\\://www.openavproductions.com/artyfx#bitta:c=crush=0.3
        

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".

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

sidechaincompress

This filter acts like normal compressor but has the ability to compress detected signal using second input signal. It needs two input streams and returns one output stream. First input stream will be processed depending on second stream signal. The filtered signal then can be filtered with other filters in later stages of processing. See pan and amerge filter.

The filter accepts the following options:

level_in
Set input gain. Default is 1. Range is between 0.015625 and 64.
threshold
If a signal of second stream raises above this level it will affect the gain reduction of first stream. By default is 0.125. Range is between 0.00097563 and 1.
ratio
Set a ratio about which the signal is reduced. 1:2 means that if the level raised 4dB above the threshold, it will be only 2dB above after the reduction. Default is 2. Range is between 1 and 20.
attack
Amount of milliseconds the signal has to rise above the threshold before gain reduction starts. Default is 20. Range is between 0.01 and 2000.
release
Amount of milliseconds the signal has to fall below the threshold before reduction is decreased again. Default is 250. Range is between 0.01 and 9000.
makeup
Set the amount by how much signal will be amplified after processing. Default is 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.

Examples

Full ffmpeg example taking 2 audio inputs, 1st input to be compressed depending on the signal of 2nd input and later compressed signal to be merged with 2nd input:

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

sidechaingate

A sidechain gate acts like a normal (wideband) gate but has the ability to filter the detected signal before sending it to the gain reduction stage. Normally a gate uses the full range signal to detect a level above the threshold. For example: If you cut all lower frequencies from your sidechain signal the gate will decrease the volume of your track only if not enough highs appear. With this technique you are able to reduce the resonation of a natural drum or remove "rumbling" of muted strokes from a heavily distorted guitar. It needs two input streams and returns one output stream. First input stream will be processed depending on second stream signal.

The filter accepts the following options:

level_in
Set input level before filtering. Default is 1. Allowed range is from 0.015625 to 64.
range
Set the level of gain reduction when the signal is below the threshold. Default is 0.06125. Allowed range is from 0 to 1.
threshold
If a signal rises above this level the gain reduction is released. Default is 0.125. Allowed range is from 0 to 1.
ratio
Set a ratio about which the signal is reduced. Default is 2. Allowed range is from 1 to 9000.
attack
Amount of milliseconds the signal has to rise above the threshold before gain reduction stops. Default is 20 milliseconds. Allowed range is from 0.01 to 9000.
release
Amount of milliseconds the signal has to fall below the threshold before the reduction is increased again. Default is 250 milliseconds. Allowed range is from 0.01 to 9000.
makeup
Set amount of amplification of signal after processing. Default is 1. Allowed range is from 1 to 64.
knee
Curve the sharp knee around the threshold to enter gain reduction more softly. Default is 2.828427125. Allowed range is from 1 to 8.
detection
Choose if exact signal should be taken for detection or an RMS like one. Default is rms. Can be peak or rms.
link
Choose if the average level between all channels or the louder channel affects the reduction. Default is average. Can be average or maximum.
level_sc
Set sidechain gain. Default is 1. Range is from 0.015625 to 64.

silencedetect

Detect silence in an audio stream.

This filter logs a message when it detects that the input audio volume is less or equal to a noise tolerance value for a duration greater or equal to the minimum detected noise duration.

The printed times and duration are expressed in seconds.

The filter accepts the following options:

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).
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 in start of multi-channel audio. Can be any or all. Default is any. With any, any sample that is detected as non-silence will cause stopped trimming of silence. With all, only if all channels are detected as non-silence will cause stopped trimming of silence.
stop_periods
Set the count for trimming silence from the end of audio. To remove silence from the middle of a file, specify a stop_periods that is negative. This value is then treated as a positive value and is used to indicate the effect should restart processing as specified by start_periods, making it suitable for removing periods of silence in the middle of the audio. Default value is 0.
stop_duration
Specify a duration of silence that must exist before audio is not copied any more. By specifying a higher duration, silence that is wanted can be left in the audio. Default value is 0.
stop_threshold
This is the same as start_threshold but for trimming silence from the end of audio. Can be specified in dB (in case "dB" is appended to the specified value) or amplitude ratio. Default value is 0.
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 in end of multi-channel audio. Can be any or all. Default is any. With any, any sample that is detected as non-silence will cause stopped trimming of silence. With all, only if all channels are detected as non-silence will cause stopped trimming of silence.
detection
Set how is silence detected. Can be "rms" or "peak". Second is faster and works better with digital silence which is exactly 0. Default value is "rms".
window
Set duration in number of seconds used to calculate size of window in number of samples for detecting silence. Default value is 0.02. Allowed range is from 0 to 10.

Examples

  • The following example shows how this filter can be used to start a recording that does not contain the delay at the start which usually occurs between pressing the record button and the start of the performance:

            silenceremove=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
        

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.

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"
        

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.
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.

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.

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.
fc_in
Set front center input volume. By default, this is 1.
fc_out
Set front 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.

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.
channels, c
Specify which channels to filter, by default all available are filtered.

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"

tremolo

Sinusoidal amplitude modulation.

The filter accepts the following options:

f
Modulation frequency in Hertz. Modulation frequencies in the subharmonic range (20 Hz or lower) will result in a tremolo effect. This filter may also be used as a ring modulator by specifying a modulation frequency higher than 20 Hz. Range is 0.1 - 20000.0. Default value is 5.0 Hz.
d
Depth of modulation as a percentage. Range is 0.0 - 1.0. Default value is 0.5.

vibrato

Sinusoidal phase modulation.

The filter accepts the following options:

f
Modulation frequency in Hertz. Range is 0.1 - 20000.0. Default value is 5.0 Hz.
d
Depth of modulation as a percentage. Range is 0.0 - 1.0. Default value is 0.5.

volume

Adjust the input audio volume.

It accepts the following parameters:

volume
Set audio volume expression.

Output values are clipped to the maximum value.

The output audio volume is given by the relation:

        <output_volume> = <volume> * <input_volume>
    

The default value for volume is "1.0".

precision
This parameter represents the mathematical precision.

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

fixed
8-bit fixed-point; this limits input sample format to U8, S16, and S32.
float
32-bit floating-point; this limits input sample format to FLT. (default)
double
64-bit floating-point; this limits input sample format to DBL.
replaygain
Choose the behaviour on encountering ReplayGain side data in input frames.
drop
Remove ReplayGain side data, ignoring its contents (the default).
ignore
Ignore ReplayGain side data, but leave it in the frame.
track
Prefer the track gain, if present.
album
Prefer the album gain, if present.
replaygain_preamp
Pre-amplification gain in dB to apply to the selected replaygain gain.

Default value for replaygain_preamp is 0.0.

eval
Set when the volume expression is evaluated.

It accepts the following values:

once
only evaluate expression once during the filter initialization, or when the volume command is sent
frame
evaluate expression for each incoming frame

Default value is once.

The volume expression can contain the following parameters.

n
frame number (starting at zero)
nb_channels
number of channels
nb_consumed_samples
number of samples consumed by the filter
nb_samples
number of samples in the current frame
pos
original frame position in the file
pts
frame PTS
sample_rate
sample rate
startpts
PTS at start of stream
startt
time at start of stream
t
frame time
tb
timestamp timebase
volume
last set volume value

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

Commands

This filter supports the following commands:

volume
Modify the volume expression. The command accepts the same syntax of the corresponding option.

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

replaygain_noclip
Prevent clipping by limiting the gain applied.

Default value for replaygain_noclip is 1.

Examples

  • Halve the input audio volume:

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

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

            volume=0.5
        
  • Increase input audio power by 6 decibels using fixed-point precision:

            volume=volume=6dB:precision=fixed
        
  • Fade volume after time 10 with an annihilation period of 5 seconds:

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

volumedetect

Detect the volume of the input video.

The filter has no parameters. The input is not modified. Statistics about the volume will be printed in the log when the input stream end is reached.

In particular it will show the mean volume (root mean square), maximum volume (on a per-sample basis), and the beginning of a histogram of the registered volume values (from the maximum value to a cumulated 1/1000 of the samples).

All volumes are in decibels relative to the maximum PCM value.

Examples

Here is an excerpt of the output:

        [Parsed_volumedetect_0  0xa23120] mean_volume: -27 dB
        [Parsed_volumedetect_0  0xa23120] max_volume: -4 dB
        [Parsed_volumedetect_0  0xa23120] histogram_4db: 6
        [Parsed_volumedetect_0  0xa23120] histogram_5db: 62
        [Parsed_volumedetect_0  0xa23120] histogram_6db: 286
        [Parsed_volumedetect_0  0xa23120] histogram_7db: 1042
        [Parsed_volumedetect_0  0xa23120] histogram_8db: 2551
        [Parsed_volumedetect_0  0xa23120] histogram_9db: 4609
        [Parsed_volumedetect_0  0xa23120] histogram_10db: 8409

It means that:

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

In other words, raising the volume by +4 dB does not cause any clipping, raising it by +5 dB causes clipping for 6 samples, etc.

AUDIO SOURCES

Below is a description of the currently available audio sources.

abuffer

Buffer audio frames, and make them available to the filter chain.

This source is mainly intended for a programmatic use, in particular through the interface defined in libavfilter/asrc_abuffer.h.

It accepts the following parameters:

time_base
The timebase which will be used for timestamps of submitted frames. It must be either a floating-point number or in numerator/denominator form.
sample_rate
The sample rate of the incoming audio buffers.
sample_fmt
The sample format of the incoming audio buffers. Either a sample format name or its corresponding integer representation from the enum AVSampleFormat in libavutil/samplefmt.h
channel_layout
The channel layout of the incoming audio buffers. Either a channel layout name from channel_layout_map in libavutil/channel_layout.c or its corresponding integer representation from the AV_CH_LAYOUT_* macros in libavutil/channel_layout.h
channels
The number of channels of the incoming audio buffers. If both channels and channel_layout are specified, then they must be consistent.

Examples

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

will instruct the source to accept planar 16bit signed stereo at 44100Hz. Since the sample format with name "s16p" corresponds to the number 6 and the "stereo" channel layout corresponds to the value 0x3, this is equivalent to:

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

aevalsrc

Generate an audio signal specified by an expression.

This source accepts in input one or more expressions (one for each channel), which are evaluated and used to generate a corresponding audio signal.

This source accepts the following options:

exprs
Set the '|'-separated expressions list for each separate channel. In case the channel_layout option is not specified, the selected channel layout depends on the number of provided expressions. Otherwise the last specified expression is applied to the remaining output channels.
channel_layout, c
Set the channel layout. The number of channels in the specified layout must be equal to the number of specified expressions.
duration, d
Set the minimum duration of the sourced audio. See the Time duration section in the ffmpeg-utils(1) manual for the accepted syntax. Note that the resulting duration may be greater than the specified duration, as the generated audio is always cut at the end of a complete frame.

If not specified, or the expressed duration is negative, the audio is supposed to be generated forever.

nb_samples, n
Set the number of samples per channel per each output frame, default to 1024.
sample_rate, s
Specify the sample rate, default to 44100.

Each expression in exprs can contain the following constants:

n
number of the evaluated sample, starting from 0
t
time of the evaluated sample expressed in seconds, starting from 0
s
sample rate

Examples

  • Generate silence:

            aevalsrc=0
        
  • Generate a sin signal with frequency of 440 Hz, set sample rate to 8000 Hz:

            aevalsrc="sin(440*2*PI*t):s=8000"
        
  • Generate a two channels signal, specify the channel layout (Front Center + Back Center) explicitly:

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

            aevalsrc="-2+random(0)"
        
  • Generate an amplitude modulated signal:

            aevalsrc="sin(10*2*PI*t)*sin(880*2*PI*t)"
        
  • Generate 2.5 Hz binaural beats on a 360 Hz carrier:

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

anullsrc

The null audio source, return unprocessed audio frames. It is mainly useful as a template and to be employed in analysis / debugging tools, or as the source for filters which ignore the input data (for example the sox synth filter).

This source accepts the following options:

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

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

sample_rate, r
Specifies the sample rate, and defaults to 44100.
nb_samples, n
Set the number of samples per requested frames.

Examples

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

            anullsrc=r=48000:cl=4
        
  • Do the same operation with a more obvious syntax:

            anullsrc=r=48000:cl=mono
        

All the parameters need to be explicitly defined.

flite

Synthesize a voice utterance using the libflite library.

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

Note that 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 and violet. Default color is white.
seed, s
Specify a value used to seed the PRNG.
nb_samples, n
Set the number of samples per each output frame, default is 1024.

Examples

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

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

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.

alphaextract

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

alphamerge

Add or replace the alpha component of the primary input with the grayscale value of a second input. This is intended for use with alphaextract to allow the transmission or storage of frame sequences that have alpha in a format that doesn't support an alpha channel.

For example, to reconstruct full frames from a normal YUV-encoded video and a separate video created with alphaextract, you might use:

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

Since this filter is designed for reconstruction, it operates on frame sequences without considering timestamps, and terminates when either input reaches end of stream. This will cause problems if your encoding pipeline drops frames. If you're trying to apply an image as an overlay to a video stream, consider the overlay filter instead.

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 differrence greater or equal to this value will not alter source pixel. Default is 10. 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.

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.

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.

bbox

Compute the bounding box for the non-black pixels in the input frame luminance plane.

This filter computes the bounding box containing all the pixels with a luminance value greater than the minimum allowed value. The parameters describing the bounding box are printed on the filter log.

The filter accepts the following option:

min_val
Set the minimal luminance value. Default is 16.

bitplanenoise

Show and measure bit plane noise.

The filter accepts the following options:

bitplane
Set which plane to analyze. Default is 1.
filter
Filter out noisy pixels from "bitplane" set above. Default is disabled.

blackdetect

Detect video intervals that are (almost) completely black. Can be useful to detect chapter transitions, commercials, or invalid recordings. Output lines contains the time for the start, end and duration of the detected black interval expressed in seconds.

In order to display the output lines, you need to set the loglevel at least to the AV_LOG_INFO value.

The filter accepts the following options:

black_min_duration, d
Set the minimum detected black duration expressed in seconds. It must be a non-negative floating point number.

Default value is 2.0.

picture_black_ratio_th, pic_th
Set the threshold for considering a picture "black". Express the minimum value for the ratio:

        <nb_black_pixels> / <nb_pixels>
    

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

pixel_black_th, pix_th
Set the threshold for considering a pixel "black".

The threshold expresses the maximum pixel luminance value for which a pixel is considered "black". The provided value is scaled according to the following equation:

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

luminance_range_size and luminance_minimum_value depend on the input video format, the range is [0-255] for YUV full-range formats and [16-235] for YUV non full-range formats.

Default value is 0.10.

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

        blackdetect=d=2:pix_th=0.00

blackframe

Detect frames that are (almost) completely black. Can be useful to detect chapter transitions or commercials. Output lines consist of the frame number of the detected frame, the percentage of blackness, the position in the file if known or -1 and the timestamp in seconds.

In order to display the output lines, you need to set the loglevel at least to the AV_LOG_INFO value.

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, tblend

Blend two video frames into each other.

The "blend" filter takes two input streams and outputs one stream, the first input is the "top" layer and second input is "bottom" layer. By default, the output terminates when the longest input terminates.

The "tblend" (time blend) filter takes two consecutive frames from one single stream, and outputs the result obtained by blending the new frame on top of the old frame.

A description of the accepted options follows.

c0_mode
c1_mode
c2_mode
c3_mode
all_mode
Set blend mode for specific pixel component or all pixel components in case of all_mode. Default value is "normal".

Available values for component modes are:

addition
grainmerge
and
average
burn
darken
difference
grainextract
divide
dodge
freeze
exclusion
extremity
glow
hardlight
hardmix
heat
lighten
linearlight
multiply
multiply128
negation
normal
or
overlay
phoenix
pinlight
reflect
screen
softlight
subtract
vividlight
xor
c0_opacity
c1_opacity
c2_opacity
c3_opacity
all_opacity
Set blend opacity for specific pixel component or all pixel components in case of all_opacity. Only used in combination with pixel component blend modes.
c0_expr
c1_expr
c2_expr
c3_expr
all_expr
Set blend expression for specific pixel component or all pixel components in case of all_expr. Note that related mode options will be ignored if those are set.

The expressions can use the following variables:

N
The sequential number of the filtered frame, starting from 0.
X
Y
the coordinates of the current sample
W
H
the width and height of currently filtered plane
SW
SH
Width and height scale 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
        

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 algorith 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. Accept one of the following values:
0, all
Deinterlace all frames.
1, interlaced
Only deinterlace frames marked as interlaced.

The default value is "all".

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.
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.

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.

Examples

  • Make every green pixel in the input image transparent:

            ffmpeg -i input.png -vf chromakey=green out.png
        
  • Overlay a greenscreen-video on top of a static black background.

            ffmpeg -f lavfi -i color=c=black:s=1280x720 -i video.mp4 -shortest -filter_complex "[1:v]chromakey=0x70de77:0.1:0.2[ckout];[0:v][ckout]overlay[out]" -map "[out]" output.mkv
        

ciescope

Display CIE color diagram with pixels overlaid onto it.

The filter accepts the following options:

system
Set color system.
ntsc, 470m
ebu, 470bg
smpte
240m
apple
widergb
cie1931
rec709, hdtv
uhdtv, rec2020
cie
Set CIE system.
xyy
ucs
luv
gamuts
Set what gamuts to draw.

See "system" option for available values.

size, s
Set ciescope size, by default set to 512.
intensity, i
Set intensity used to map input pixel values to CIE diagram.
contrast
Set contrast used to draw tongue colors that are out of active color system gamut.
corrgamma
Correct gamma displayed on scope, by default enabled.
showwhite
Show white point on CIE diagram, by default disabled.
gamma
Set input gamma. Used only with XYZ input color space.

codecview

Visualize information exported by some codecs.

Some codecs can export information through frames using side-data or other means. For example, some MPEG based codecs export motion vectors through the export_mvs flag in the codec flags2 option.

The filter accepts the following option:

mv
Set motion vectors to visualize.

Available flags for mv are:

pf
forward predicted MVs of P-frames
bf
forward predicted MVs of B-frames
bb
backward predicted MVs of B-frames
qp
Display quantization parameters using the chroma planes.
mv_type, mvt
Set motion vectors type to visualize. Includes MVs from all frames unless specified by frame_type option.

Available flags for mv_type are:

fp
forward predicted MVs
bp
backward predicted MVs
frame_type, ft
Set frame type to visualize motion vectors of.

Available flags for frame_type are:

if
intra-coded frames (I-frames)
pf
predicted frames (P-frames)
bf
bi-directionally predicted frames (B-frames)

Examples

  • Visualize forward predicted MVs of all frames using ffplay:

            ffplay -flags2 +export_mvs input.mp4 -vf codecview=mv_type=fp
        
  • Visualize multi-directionals MVs of P and B-Frames using ffplay:

            ffplay -flags2 +export_mvs input.mp4 -vf codecview=mv=pf+bf+bb
        

colorbalance

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

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

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

The filter accepts the following options:

rs
gs
bs
Adjust red, green and blue shadows (darkest pixels).
rm
gm
bm
Adjust red, green and blue midtones (medium pixels).
rh
gh
bh
Adjust red, green and blue highlights (brightest pixels).

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

Examples

Add red color cast to shadows:

        colorbalance=rs=.3
    

colorkey

RGB colorspace color keying.

The filter accepts the following options:

color
The color which will be replaced with transparency.
similarity
Similarity percentage with the key color.

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

blend
Blend percentage.

0.0 makes pixels either fully transparent, or not transparent at all.

Higher values result in semi-transparent pixels, with a higher transparency the more similar the pixels color is to the key color.

Examples

  • Make every green pixel in the input image transparent:

            ffmpeg -i input.png -vf colorkey=green out.png
        
  • Overlay a greenscreen-video on top of a static background image.

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

colorlevels

Adjust video input frames using levels.

The filter accepts the following options:

rimin
gimin
bimin
aimin
Adjust red, green, blue and alpha input black point. Allowed ranges for options are "[-1.0, 1.0]". Defaults are 0.
rimax
gimax
bimax
aimax
Adjust red, green, blue and alpha input white point. Allowed ranges for options are "[-1.0, 1.0]". Defaults are 1.

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

romin
gomin
bomin
aomin
Adjust red, green, blue and alpha output black point. Allowed ranges for options are "[0, 1.0]". Defaults are 0.
romax
gomax
bomax
aomax
Adjust red, green, blue and alpha output white point. Allowed ranges for options are "[0, 1.0]". Defaults are 1.

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

Examples

  • Make video output darker:

            colorlevels=rimin=0.058:gimin=0.058:bimin=0.058
        
  • Increase contrast:

            colorlevels=rimin=0.039:gimin=0.039:bimin=0.039:rimax=0.96:gimax=0.96:bimax=0.96
        
  • Make video output lighter:

            colorlevels=rimax=0.902:gimax=0.902:bimax=0.902
        
  • Increase brightness:

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

colorchannelmixer

Adjust video input frames by re-mixing color channels.

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

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

The filter accepts the following options:

rr
rg
rb
ra
Adjust contribution of input red, green, blue and alpha channels for output red channel. Default is 1 for rr, and 0 for rg, rb and ra.
gr
gg
gb
ga
Adjust contribution of input red, green, blue and alpha channels for output green channel. Default is 1 for gg, and 0 for gr, gb and ga.
br
bg
bb
ba
Adjust contribution of input red, green, blue and alpha channels for output blue channel. Default is 1 for bb, and 0 for br, bg and ba.
ar
ag
ab
aa
Adjust contribution of input red, green, blue and alpha channels for output alpha channel. Default is 1 for aa, and 0 for ar, ag and ab.

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

Examples

  • Convert source to grayscale:

            colorchannelmixer=.3:.4:.3:0:.3:.4:.3:0:.3:.4:.3
        
  • Simulate sepia tones:

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

colormatrix

Convert color matrix.

The filter accepts the following options:

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

The accepted values are:

bt709
BT.709
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

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.

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
        

crop

Crop the input video to given dimensions.

It accepts the following parameters:

w, out_w
The width of the output video. It defaults to "iw". This expression is evaluated only once during the filter configuration, or when the w or out_w command is sent.
h, out_h
The height of the output video. It defaults to "ih". This expression is evaluated only once during the filter configuration, or when the h or out_h command is sent.
x
The horizontal position, in the input video, of the left edge of the output video. It defaults to "(in_w-out_w)/2". This expression is evaluated per-frame.
y
The vertical position, in the input video, of the top edge of the output video. It defaults to "(in_h-out_h)/2". This expression is evaluated per-frame.
keep_aspect
If set to 1 will force the output display aspect ratio to be the same of the input, by changing the output sample aspect ratio. It defaults to 0.
exact
Enable exact cropping. If enabled, subsampled videos will be cropped at exact width/height/x/y as specified and will not be rounded to nearest smaller value. It defaults to 0.

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

x
y
The computed values for x and y. They are evaluated for each new frame.
in_w
in_h
The input width and height.
iw
ih
These are the same as in_w and in_h.
out_w
out_h
The output (cropped) width and height.
ow
oh
These are the same as out_w and out_h.
a
same as iw / ih
sar
input sample aspect ratio
dar
input display aspect ratio, it is the same as (iw / ih) * sar
hsub
vsub
horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.
n
The number of the input frame, starting from 0.
pos
the position in the file of the input frame, NAN if unknown
t
The timestamp expressed in seconds. It's NAN if the input timestamp is unknown.

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

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

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

Examples

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

            crop=100:100:12:34
        

    Using named options, the example above becomes:

            crop=w=100:h=100:x=12:y=34
        
  • Crop the central input area with size 100x100:

            crop=100:100
        
  • Crop the central input area with size 2/3 of the input video:

            crop=2/3*in_w:2/3*in_h
        
  • Crop the input video central square:

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

            crop=in_w-100:in_h-100:100:100
        
  • Crop 10 pixels from the left and right borders, and 20 pixels from the top and bottom borders

            crop=in_w-2*10:in_h-2*20
        
  • Keep only the bottom right quarter of the input image:

            crop=in_w/2:in_h/2:in_w/2:in_h/2
        
  • Crop height for getting Greek harmony:

            crop=in_w:1/PHI*in_w
        
  • Apply trembling effect:

            crop=in_w/2:in_h/2:(in_w-out_w)/2+((in_w-out_w)/2)*sin(n/10):(in_h-out_h)/2 +((in_h-out_h)/2)*sin(n/7)
        
  • Apply erratic camera effect depending on timestamp:

            crop=in_w/2:in_h/2:(in_w-out_w)/2+((in_w-out_w)/2)*sin(t*10):(in_h-out_h)/2 +((in_h-out_h)/2)*sin(t*13)"
        
  • Set x depending on the value of y:

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

Commands

This filter supports the following commands:

w, out_w
h, out_h
x
y
Set width/height of the output video and the horizontal/vertical position in the input video. The command accepts the same syntax of the corresponding option.

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

cropdetect

Auto-detect the crop size.

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

It accepts the following parameters:

limit
Set higher black value threshold, which can be optionally specified from nothing (0) to everything (255 for 8-bit based formats). An intensity value greater to the set value is considered non-black. It defaults to 24. You can also specify a value between 0.0 and 1.0 which will be scaled depending on the bitdepth of the pixel format.
round
The value which the width/height should be divisible by. It defaults to 16. The offset is automatically adjusted to center the video. Use 2 to get only even dimensions (needed for 4:2:2 video). 16 is best when encoding to most video codecs.
reset_count, reset
Set the counter that determines after how many frames cropdetect will reset the previously detected largest video area and start over to detect the current optimal crop area. Default value is 0.

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

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

The filter accepts the following options:

preset
Select one of the available color presets. This option can be used in addition to the r, g, b parameters; in this case, the later options takes priority on the preset values. Available presets are:
none
color_negative
cross_process
darker
increase_contrast
lighter
linear_contrast
medium_contrast
negative
strong_contrast
vintage

Default is "none".

master, m
Set the master key points. These points will define a second pass mapping. It is sometimes called a "luminance" or "value" mapping. It can be used with r, g, b or all since it acts like a post-processing LUT.
red, r
Set the key points for the red component.
green, g
Set the key points for the green component.
blue, b
Set the key points for the blue component.
all
Set the key points for all components (not including master). Can be used in addition to the other key points component options. In this case, the unset component(s) will fallback on this all setting.
psfile
Specify a Photoshop curves file (".acv") to import the settings from.
plot
Save Gnuplot script of the curves in specified file.

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

Examples

  • Increase slightly the middle level of blue:

            curves=blue='0/0 0.5/0.58 1/1'
        
  • Vintage effect:

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

    Here we obtain the following coordinates for each components:

red
"(0;0.11) (0.42;0.51) (1;0.95)"
green
"(0;0) (0.50;0.48) (1;1)"
blue
"(0;0.22) (0.49;0.44) (1;0.80)"
  • The previous example can also be achieved with the associated built-in preset:

            curves=preset=vintage
        
  • Or simply:

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

            curves=psfile='MyCurvesPresets/purple.acv':green='0/0 0.45/0.53 1/1'
        
  • Check out the curves of the "cross_process" profile using ffmpeg and gnuplot:

            ffmpeg -f lavfi -i color -vf curves=cross_process:plot=/tmp/curves.plt -frames:v 1 -f null -
            gnuplot -p /tmp/curves.plt
        

datascope

Video data analysis filter.

This filter shows hexadecimal pixel values of part of video.

The filter accepts the following options:

size, s
Set output video size.
x
Set x offset from where to pick pixels.
y
Set y offset from where to pick pixels.
mode
Set scope mode, can be one of the following:
mono
Draw hexadecimal pixel values with white color on black background.
color
Draw hexadecimal pixel values with input video pixel color on black background.
color2
Draw hexadecimal pixel values on color background picked from input video, the text color is picked in such way so its always visible.
axis
Draw rows and columns numbers on left and top of video.
opacity
Set background opacity.

dctdnoiz

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

This filter is not designed for real time.

The filter accepts the following options:

sigma, s
Set the noise sigma constant.

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

If you need a more advanced filtering, see expr.

Default is 0.

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

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

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

expr, e
Set the coefficient factor expression.

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

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

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

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

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

Examples

Apply a denoise with a sigma of 4.5:

        dctdnoiz=4.5

The same operation can be achieved using the expression system:

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

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

        dctdnoiz=15:n=4

deband

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

The filter accepts the following options:

1thr
2thr
3thr
4thr
Set banding detection threshold for each plane. Default is 0.02. Valid range is 0.00003 to 0.5. If difference between current pixel and reference pixel is less than threshold, it will be considered as banded.
range, r
Banding detection range in pixels. Default is 16. If positive, random number in range 0 to set value will be used. If negative, exact absolute value will be used. The range defines square of four pixels around current pixel.
direction, d
Set direction in radians from which four pixel will be compared. If positive, random direction from 0 to set direction will be picked. If negative, exact of absolute value will be picked. For example direction 0, -PI or -2*PI radians will pick only pixels on same row and -PI/2 will pick only pixels on same column.
blur, 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.

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
        

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.

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.

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.

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.

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

It accepts the following parameters:

x
y
Specify the top left corner coordinates of the logo. They must be specified.
w
h
Specify the width and height of the logo to clear. They must be specified.
band, t
Specify the thickness of the fuzzy edge of the rectangle (added to w and h). The default value is 1. This option is deprecated, setting higher values should no longer be necessary and is not recommended.
show
When set to 1, a green rectangle is drawn on the screen to simplify finding the right x, y, w, and h parameters. The default value is 0.

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

Examples

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

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

deshake

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

The filter accepts the following options:

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

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

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

Default - search the whole frame.

rx
ry
Specify the maximum extent of movement in x and y directions in the range 0-64 pixels. Default 16.
edge
Specify how to generate pixels to fill blanks at the edge of the frame. Available values are:
blank, 0
Fill zeroes at blank locations
original, 1
Original image at blank locations
clamp, 2
Extruded edge value at blank locations
mirror, 3
Mirrored edge at blank locations

Default value is mirror.

blocksize
Specify the blocksize to use for motion search. Range 4-128 pixels, default 8.
contrast
Specify the contrast threshold for blocks. Only blocks with more than the specified contrast (difference between darkest and lightest pixels) will be considered. Range 1-255, default 125.
search
Specify the search strategy. Available values are:
exhaustive, 0
Set exhaustive search
less, 1
Set less exhaustive search.

Default value is exhaustive.

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

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.

detelecine

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

This filter accepts the following options:

first_field
top, t
top field first
bottom, b
bottom field first The default value is "top".
pattern
A string of numbers representing the pulldown pattern you wish to apply. The default value is 23.
start_frame
A number representing position of the first frame with respect to the telecine pattern. This is to be used if the stream is cut. The default value is 0.

dilation

Apply dilation effect to the video.

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

It accepts the following options:

threshold0
threshold1
threshold2
threshold3
Limit the maximum change for each plane, default is 65535. If 0, plane will remain unchanged.
coordinates
Flag which specifies the pixel to refer to. Default is 255 i.e. all eight pixels are used.

Flags to local 3x3 coordinates maps like this:

    1 2 3
    4   5
    6 7 8
    

displace

Displace pixels as indicated by second and third input stream.

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

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

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

A description of the accepted options follows.

edge
Set displace behavior for pixels that are out of range.

Available values are:

blank
Missing pixels are replaced by black pixels.
smear
Adjacent pixels will spread out to replace missing pixels.
wrap
Out of range pixels are wrapped so they point to pixels of other side.
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
        

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.
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
        

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
        

drawtext

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

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

Syntax

It accepts the following parameters:

box
Used to draw a box around text using the background color. The value must be either 1 (enable) or 0 (disable). The default value of box is 0.
boxborderw
Set the width of the border to be drawn around the box using boxcolor. The default value of boxborderw is 0.
boxcolor
The color to be used for drawing box around text. For the syntax of this option, check the "Color" section in the ffmpeg-utils manual.

The default value of boxcolor is "white".

line_spacing
Set the line spacing in pixels of the border to be drawn around the box using box. The default value of line_spacing is 0.
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".

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.

reload
If set to 1, the textfile will be reloaded before each frame. Be sure to update it atomically, or it may be read partially, or even fail.
x
y
The expressions which specify the offsets where text will be drawn within the video frame. They are relative to the top/left border of the output image.

The default value of x and y is "0".

See below for the list of accepted constants and functions.

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

dar
input display aspect ratio, it is the same as (w / h) * sar
hsub
vsub
horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.
line_h, lh
the height of each text line
main_h, h, H
the input height
main_w, w, W
the input width
max_glyph_a, ascent
the maximum distance from the baseline to the highest/upper grid coordinate used to place a glyph outline point, for all the rendered glyphs. It is a positive value, due to the grid's orientation with the Y axis upwards.
max_glyph_d, descent
the maximum distance from the baseline to the lowest grid coordinate used to place a glyph outline point, for all the rendered glyphs. This is a negative value, due to the grid's orientation, with the Y axis upwards.
max_glyph_h
maximum glyph height, that is the maximum height for all the glyphs contained in the rendered text, it is equivalent to ascent - descent.
max_glyph_w
maximum glyph width, that is the maximum width for all the glyphs contained in the rendered text
n
the number of input frame, starting from 0
rand(min, max)
return a random number included between min and max
sar
The input sample aspect ratio.
t
timestamp expressed in seconds, NAN if the input timestamp is unknown
text_h, th
the height of the rendered text
text_w, tw
the width of the rendered text
x
y
the x and y offset coordinates where the text is drawn.

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

Text expansion

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

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

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

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

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

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

The following functions are available:

expr, e
The expression evaluation result.

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

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

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

gmtime
The time at which the filter is running, expressed in UTC. It can accept an argument: a strftime() format string.
localtime
The time at which the filter is running, expressed in the local time zone. It can accept an argument: a strftime() format string.
metadata
Frame metadata. Takes one or two arguments.

The first argument is mandatory and specifies the metadata key.

The second argument is optional and specifies a default value, used when the metadata key is not found or empty.

n, frame_num
The frame number, starting from 0.
pict_type
A 1 character description of the current picture type.
pts
The timestamp of the current frame. It can take up to three arguments.

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

The second argument is an offset added to the timestamp.

If the format is set to "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() format string. By default, YYYY-MM-DD HH:MM:SS format will be used.

Examples

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

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

            drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf: text='Test Text':\
                      x=100: y=50: fontsize=24: fontcolor=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'
        
  • Print the date of a real-time encoding (see strftime(3)):

            drawtext='fontfile=FreeSans.ttf:text=%{localtime\:%a %b %d %Y}'
        
  • Show text fading in and out (appearing/disappearing):

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

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

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

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

edgedetect

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

The filter accepts the following options:

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

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

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

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

mode
Define the drawing mode.
wires
Draw white/gray wires on black background.
colormix
Mix the colors to create a paint/cartoon effect.
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
        

eq

Set brightness, contrast, saturation and approximate gamma adjustment.

The filter accepts the following options:

contrast
Set the contrast expression. The value must be a float value in range "-2.0" to 2.0. The default value is "1".
brightness
Set the brightness expression. The value must be a float value in range "-1.0" to 1.0. The default value is "0".
saturation
Set the saturation expression. The value must be a float in range 0.0 to 3.0. The default value is "1".
gamma
Set the gamma expression. The value must be a float in range 0.1 to 10.0. The default value is "1".
gamma_r
Set the gamma expression for red. The value must be a float in range 0.1 to 10.0. The default value is "1".
gamma_g
Set the gamma expression for green. The value must be a float in range 0.1 to 10.0. The default value is "1".
gamma_b
Set the gamma expression for blue. The value must be a float in range 0.1 to 10.0. The default value is "1".
gamma_weight
Set the gamma weight expression. It can be used to reduce the effect of a high gamma value on bright image areas, e.g. keep them from getting overamplified and just plain white. The value must be a float in range 0.0 to 1.0. A value of 0.0 turns the gamma correction all the way down while 1.0 leaves it at its full strength. Default is "1".
eval
Set when the expressions for brightness, contrast, saturation and gamma expressions are evaluated.

It accepts the following values:

init
only evaluate expressions once during the filter initialization or when a command is processed
frame
evaluate expressions for each incoming frame

Default value is init.

The expressions accept the following parameters:

n
frame count of the input frame starting from 0
pos
byte position of the corresponding packet in the input file, NAN if unspecified
r
frame rate of the input video, NAN if the input frame rate is unknown
t
timestamp expressed in seconds, NAN if the input timestamp is unknown

Commands

The filter supports the following commands:

contrast
Set the contrast expression.
brightness
Set the brightness expression.
saturation
Set the saturation expression.
gamma
Set the gamma expression.
gamma_r
Set the gamma_r expression.
gamma_g
Set gamma_g expression.
gamma_b
Set gamma_b expression.
gamma_weight
Set gamma_weight expression.

The command accepts the same syntax of the corresponding option.

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

erosion

Apply erosion effect to the video.

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

It accepts the following options:

threshold0
threshold1
threshold2
threshold3
Limit the maximum change for each plane, default is 65535. If 0, plane will remain unchanged.
coordinates
Flag which specifies the pixel to refer to. Default is 255 i.e. all eight pixels are used.

Flags to local 3x3 coordinates maps like this:

    1 2 3
    4   5
    6 7 8
    

extractplanes

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

The filter accepts the following option:

planes
Set plane(s) to extract.

Available values for planes are:

y
u
v
a
r
g
b

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

Examples

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

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

elbg

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

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

This filter accepts the following options.

codebook_length, l
Set codebook length. The value must be a positive integer, and represents the number of distinct output colors. Default value is 256.
nb_steps, n
Set the maximum number of iterations to apply for computing the optimal mapping. The higher the value the better the result and the higher the computation time. Default value is 1.
seed, s
Set a random seed, must be an integer included between 0 and UINT32_MAX. If not specified, or if explicitly set to -1, the filter will try to use a good random seed on a best effort basis.
pal8
Set pal8 output pixel format. This option does not work with codebook length greater than 256.

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.

fade

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

It accepts the following parameters:

type, t
The effect type can be either "in" for a fade-in, or "out" for a fade-out effect. Default is "in".
start_frame, s
Specify the number of the frame to start applying the fade effect at. Default is 0.
nb_frames, n
The number of frames that the fade effect lasts. At the end of the fade-in effect, the output video will have the same intensity as the input video. At the end of the fade-out transition, the output video will be filled with the selected color. Default is 25.
alpha
If set to 1, fade only alpha channel, if one exists on the input. Default value is 0.
start_time, st
Specify the timestamp (in seconds) of the frame to start to apply the fade effect. If both start_frame and start_time are specified, the fade will start at whichever comes last. Default is 0.
duration, d
The number of seconds for which the fade effect has to last. At the end of the fade-in effect the output video will have the same intensity as the input video, at the end of the fade-out transition the output video will be filled with the selected color. If both duration and nb_frames are specified, duration is used. Default is 0 (nb_frames is used by default).
color, c
Specify the color of the fade. Default is "black".

Examples

  • Fade in the first 30 frames of video:

            fade=in:0:30
        

    The command above is equivalent to:

            fade=t=in:s=0:n=30
        
  • Fade out the last 45 frames of a 200-frame video:

            fade=out:155:45
            fade=type=out:start_frame=155:nb_frames=45
        
  • Fade in the first 25 frames and fade out the last 25 frames of a 1000-frame video:

            fade=in:0:25, fade=out:975:25
        
  • Make the first 5 frames yellow, then fade in from frame 5-24:

            fade=in:5:20:color=yellow
        
  • Fade in alpha over first 25 frames of video:

            fade=in:0:25:alpha=1
        
  • Make the first 5.5 seconds black, then fade in for 0.5 seconds:

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

fftfilt

Apply arbitrary expressions to samples in frequency domain
dc_Y
Adjust the dc value (gain) of the luma plane of the image. The filter accepts an integer value in range 0 to 1000. The default value is set to 0.
dc_U
Adjust the dc value (gain) of the 1st chroma plane of the image. The filter accepts an integer value in range 0 to 1000. The default value is set to 0.
dc_V
Adjust the dc value (gain) of the 2nd chroma plane of the image. The filter accepts an integer value in range 0 to 1000. The default value is set to 0.
weight_Y
Set the frequency domain weight expression for the luma plane.
weight_U
Set the frequency domain weight expression for the 1st chroma plane.
weight_V
Set the frequency domain weight expression for the 2nd chroma plane.
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.

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)))'
        

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, Default is 4, can be 3, 4, 5 or 6. Actual size of block in pixels is 2 to power of block, so by default block size in pixels is 2^4 which is 16.
overlap
Set block overlap. Default is 0.5. Allowed range is from 0.2 to 0.8.
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.

field

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

The filter accepts the following options:

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

fieldhint

Create new frames by copying the top and bottom fields from surrounding frames supplied as numbers by the hint file.
hint
Set file containing hints: absolute/relative frame numbers.

There must be one line for each frame in a clip. Each line must contain two numbers separated by the comma, optionally followed by "-" or "+". Numbers supplied on each line of file can not be out of [N-1,N+1] where N is current frame number for "absolute" mode or out of [-1, 1] range for "relative" mode. First number tells from which frame to pick up top field and second number tells from which frame to pick up bottom field.

If optionally followed by "+" output frame will be marked as interlaced, else if followed by "-" output frame will be marked as progressive, else it will be marked same as input frame. If line starts with "#" or ";" that line is skipped.

mode
Can be item "absolute" or "relative". Default is "absolute".

Example of first several lines of "hint" file for "relative" mode:

        0,0 - # first frame
        1,0 - # second frame, use third's frame top field and second's frame bottom field
        1,0 - # third frame, use fourth's frame top field and third's frame bottom field
        1,0 -
        0,0 -
        0,0 -
        1,0 -
        1,0 -
        1,0 -
        0,0 -
        0,0 -
        1,0 -
        1,0 -
        1,0 -
        0,0 -

fieldmatch

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

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

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

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

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

The filter accepts the following options:

order
Specify the assumed field order of the input stream. Available values are:
auto
Auto detect parity (use FFmpeg's internal parity value).
bff
Assume bottom field first.
tff
Assume top field first.

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

Default value is auto.

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

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

Available values are:

pc
2-way matching (p/c)
pc_n
2-way matching, and trying 3rd match if still combed (p/c + n)
pc_u
2-way matching, and trying 3rd match (same order) if still combed (p/c + u)
pc_n_ub
2-way matching, trying 3rd match if still combed, and trying 4th/5th matches if still combed (p/c + n + u/b)
pcn
3-way matching (p/c/n)
pcn_ub
3-way matching, and trying 4th/5th matches if all 3 of the original matches are detected as combed (p/c/n + u/b)

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

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

Default value is pc_n.

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

Default value is 0 (disabled).

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

Default value is auto.

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

Default value is 1.

y0
y1
These define an exclusion band which excludes the lines between y0 and y1 from being included in the field matching decision. An exclusion band can be used to ignore subtitles, a logo, or other things that may interfere with the matching. y0 sets the starting scan line and y1 sets the ending line; all lines in between y0 and y1 (including y0 and y1) will be ignored. Setting y0 and y1 to the same value will disable the feature. y0 and y1 defaults to 0.
scthresh
Set the scene change detection threshold as a percentage of maximum change on the luma plane. Good values are in the "[8.0, 14.0]" range. Scene change detection is only relevant in case combmatch=sc. The range for scthresh is "[0.0, 100.0]".

Default value is 12.0.

combmatch
When combatch is not none, "fieldmatch" will take into account the combed scores of matches when deciding what match to use as the final match. Available values are:
none
No final matching based on combed scores.
sc
Combed scores are only used when a scene change is detected.
full
Use combed scores all the time.

Default is sc.

combdbg
Force "fieldmatch" to calculate the combed metrics for certain matches and print them. This setting is known as micout in TFM/VFM vocabulary. Available values are:
none
No forced calculation.
pcn
Force p/c/n calculations.
pcnub
Force p/c/n/u/b calculations.

Default value is none.

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

Default value is 9.

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

Default value is 0.

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

Default value is 16.

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

Default value is 80.

p/c/n/u/b meaning

p/c/n

We assume the following telecined stream:

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

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

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

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

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

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

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

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

*<p matches the field of the opposite parity in the previous frame>
*<c matches the field of the opposite parity in the current frame>
*<n matches the field of the opposite parity in the next frame>

u/b

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

With bottom matching (field=bottom):

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

With top matching (field=top):

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

Examples

Simple IVTC of a top field first telecined stream:

        fieldmatch=order=tff:combmatch=none, decimate

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

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

fieldorder

Transform the field order of the input video.

It accepts the following parameters:

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

The default value is tff.

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

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

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

For example:

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

fifo, afifo

Buffer input images and send them when they are requested.

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

It does not take parameters.

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
fixed
fill pixels with constant value

Default is smear.

color
Set color for pixels in fixed mode. Default is black.

find_rect

Find a rectangular object

It accepts the following options:

object
Filepath of the object image, needs to be in gray8.
threshold
Detection threshold, default is 0.5.
mipmaps
Number of mipmaps, default is 3.
xmin, ymin, xmax, ymax
Specifies the rectangle in which to search.

Examples

Generate a representative palette of a given video using ffmpeg:

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

cover_rect

Cover a rectangular object

It accepts the following options:

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

It accepts the following values:

cover
cover it by the supplied image
blur
cover it by interpolating the surrounding pixels

Default value is blur.

Examples

Generate a representative palette of a given video using ffmpeg:

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

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. The default is 25.
start_time
Assume the first PTS should be the given value, in seconds. This allows for padding/trimming at the start of stream. By default, no assumption is made about the first frame's expected PTS, so no padding or trimming is done. For example, this could be set to 0 to pad the beginning with duplicates of the first frame if a video stream starts after the audio stream or to trim any frames with a negative PTS.
round
Timestamp (PTS) rounding method.

Possible values are:

zero
round towards 0
inf
round away from 0
down
round towards -infinity
up
round towards +infinity
near
round to nearest

The default is "near".

eof_action
Action performed when reading the last frame.

Possible values are:

round
Use same timestamp rounding method as used for other frames.
pass
Pass through last frame if input duration has not been reached yet.

The default is "round".

Alternatively, the options can be specified as a flat string: fps[:start_time[:round]].

See also the setpts filter.

Examples

  • A typical usage in order to set the fps to 25:

            fps=fps=25
        
  • Sets the fps to 24, using abbreviation and rounding method to round to nearest:

            fps=fps=film:round=near
        

framepack

Pack two different video streams into a stereoscopic video, setting proper metadata on supported codecs. The two views should have the same size and framerate and processing will stop when the shorter video ends. Please note that you may conveniently adjust view properties with the scale and fps filters.

It accepts the following parameters:

format
The desired packing format. Supported values are:
sbs
The views are next to each other (default).
tab
The views are on top of each other.
lines
The views are packed by line.
columns
The views are packed by column.
frameseq
The views are temporally interleaved.

Some examples:

        # Convert left and right views into a frame-sequential video
        ffmpeg -i LEFT -i RIGHT -filter_complex framepack=frameseq OUTPUT
        
        # Convert views into a side-by-side video with the same output resolution as the input
        ffmpeg -i LEFT -i RIGHT -filter_complex [0:v]scale=w=iw/2[left],[1:v]scale=w=iw/2[right],[left][right]framepack=sbs OUTPUT

framerate

Change the frame rate by interpolating new video output frames from the source frames.

This filter is not designed to function correctly with interlaced media. If you wish to change the frame rate of interlaced media then you are required to deinterlace before this filter and re-interlace after this filter.

A description of the accepted options follows.

fps
Specify the output frames per second. This option can also be specified as a value alone. The default is 50.
interp_start
Specify the start of a range where the output frame will be created as a linear interpolation of two frames. The range is [0-255], the default is 15.
interp_end
Specify the end of a range where the output frame will be created as a linear interpolation of two frames. The range is [0-255], the default is 240.
scene
Specify the level at which a scene change is detected as a value between 0 and 100 to indicate a new scene; a low value reflects a low probability for the current frame to introduce a new scene, while a higher value means the current frame is more likely to be one. The default is 8.2.
flags
Specify flags influencing the filter process.

Available value for flags is:

scene_change_detect, scd
Enable scene change detection using the value of the option scene. This flag is enabled by default.

framestep

Select one frame every N-th frame.

This filter accepts the following option:

step
Select frame after every "step" frames. Allowed values are positive integers higher than 0. Default value is 1.

frei0r

Apply a frei0r effect to the input video.

To enable the compilation of this filter, you need to install the frei0r header and configure FFmpeg with "--enable-frei0r".

It accepts the following parameters:

filter_name
The name of the frei0r effect to load. If the environment variable FREI0R_PATH is defined, the frei0r effect is searched for in each of the directories specified by the colon-separated list in FREI0R_PATH. Otherwise, the standard frei0r paths are searched, in this order: HOME/.frei0r-1/lib/, /usr/local/lib/frei0r-1/, /usr/lib/frei0r-1/.
filter_params
A '|'-separated list of parameters to pass to the frei0r effect.

A frei0r effect parameter can be a boolean (its value is either "y" or "n"), a double, a color (specified as R/G/B, where R, G, and B are floating point numbers between 0.0 and 1.0, inclusive) or a color description as specified in the "Color" section in the ffmpeg-utils manual, a position (specified as X/Y, where X and Y are floating point numbers) and/or a string.

The number and types of parameters depend on the loaded effect. If an effect parameter is not specified, the default value is set.

Examples

  • Apply the distort0r effect, setting the first two double parameters:

            frei0r=filter_name=distort0r:filter_params=0.5|0.01
        
  • Apply the colordistance effect, taking a color as the first parameter:

            frei0r=colordistance:0.2/0.3/0.4
            frei0r=colordistance:violet
            frei0r=colordistance:0x112233
        
  • Apply the perspective effect, specifying the top left and top right image positions:

            frei0r=perspective:0.2/0.2|0.8/0.2
        

For more information, see <http://frei0r.dyne.org>

fspp

Apply fast and simple postprocessing. It is a faster version of spp.

It splits (I)DCT into horizontal/vertical passes. Unlike the simple post- processing filter, one of them is performed once per block, not per pixel. This allows for much higher speed.

The filter accepts the following options:

quality
Set quality. This option defines the number of levels for averaging. It accepts an integer in the range 4-5. Default value is 4.
qp
Force a constant quantization parameter. It accepts an integer in range 0-63. If not set, the filter will use the QP from the video stream (if available).
strength
Set filter strength. It accepts an integer in range -15 to 32. Lower values mean more details but also more artifacts, while higher values make the image smoother but also blurrier. Default value is 0 X PSNR optimal.
use_bframe_qp
Enable the use of the QP from the B-Frames if set to 1. Using this option may cause flicker since the B-Frames have often larger QP. Default is 0 (not enabled).

gblur

Apply Gaussian blur filter.

The filter accepts the following options:

sigma
Set horizontal sigma, standard deviation of Gaussian blur. Default is 0.5.
steps
Set number of steps for Gaussian approximation. Defauls is 1.
planes
Set which planes to filter. By default all planes are filtered.
sigmaV
Set vertical sigma, if negative it will be same as "sigma". Default is "-1".

geq

Apply generic equation to each pixel.

The filter accepts the following options:

lum_expr, lum
Set the luminance expression.
cb_expr, cb
Set the chrominance blue expression.
cr_expr, cr
Set the chrominance red expression.
alpha_expr, a
Set the alpha expression.
red_expr, r
Set the red expression.
green_expr, g
Set the green expression.
blue_expr, b
Set the blue expression.

The colorspace is selected according to the specified options. If one of the lum_expr, cb_expr, or cr_expr options is specified, the filter will automatically select a YCbCr colorspace. If one of the red_expr, green_expr, or blue_expr options is specified, it will select an RGB colorspace.

If one of the chrominance expression is not defined, it falls back on the other one. If no alpha expression is specified it will evaluate to opaque value. If none of chrominance expressions are specified, they will evaluate to the luminance expression.

The expressions can use the following variables and functions:

N
The sequential number of the filtered frame, starting from 0.
X
Y
The coordinates of the current sample.
W
H
The width and height of the image.
SW
SH
Width and height scale depending on the currently filtered plane. It is the ratio between the corresponding luma plane number of pixels and the current plane ones. E.g. for YUV4:2:0 the values are "1,1" for the luma plane, and "0.5,0.5" for chroma planes.
T
Time of the current frame, expressed in seconds.
p(x, y)
Return the value of the pixel at location (x,y) of the current plane.
lum(x, y)
Return the value of the pixel at location (x,y) of the luminance plane.
cb(x, y)
Return the value of the pixel at location (x,y) of the blue-difference chroma plane. Return 0 if there is no such plane.
cr(x, y)
Return the value of the pixel at location (x,y) of the red-difference chroma plane. Return 0 if there is no such plane.
r(x, y)
g(x, y)
b(x, y)
Return the value of the pixel at location (x,y) of the red/green/blue component. Return 0 if there is no such component.
alpha(x, y)
Return the value of the pixel at location (x,y) of the alpha plane. Return 0 if there is no such plane.

For functions, if x and y are outside the area, the value will be automatically clipped to the closer edge.

Examples

  • Flip the image horizontally:

            geq=p(W-X\,Y)
        
  • Generate a bidimensional sine wave, with angle "PI/3" and a wavelength of 100 pixels:

            geq=128 + 100*sin(2*(PI/100)*(cos(PI/3)*(X-50*T) + sin(PI/3)*Y)):128:128
        
  • Generate a fancy enigmatic moving light:

            nullsrc=s=256x256,geq=random(1)/hypot(X-cos(N*0.07)*W/2-W/2\,Y-sin(N*0.09)*H/2-H/2)^2*1000000*sin(N*0.02):128:128
        
  • Generate a quick emboss effect:

            format=gray,geq=lum_expr='(p(X,Y)+(256-p(X-4,Y-4)))/2'
        
  • Modify RGB components depending on pixel position:

            geq=r='X/W*r(X,Y)':g='(1-X/W)*g(X,Y)':b='(H-Y)/H*b(X,Y)'
        
  • Create a radial gradient that is the same size as the input (also see the vignette filter):

            geq=lum=255*gauss((X/W-0.5)*3)*gauss((Y/H-0.5)*3)/gauss(0)/gauss(0),format=gray
        

gradfun

Fix the banding artifacts that are sometimes introduced into nearly flat regions by truncation to 8-bit color depth. Interpolate the gradients that should go where the bands are, and dither them.

It is designed for playback only. Do not use it prior to lossy compression, because compression tends to lose the dither and bring back the bands.

It accepts the following parameters:

strength
The maximum amount by which the filter will change any one pixel. This is also the threshold for detecting nearly flat regions. Acceptable values range from .51 to 64; the default value is 1.2. Out-of-range values will be clipped to the valid range.
radius
The neighborhood to fit the gradient to. A larger radius makes for smoother gradients, but also prevents the filter from modifying the pixels near detailed regions. Acceptable values are 8-32; the default value is 16. Out-of-range values will be clipped to the valid range.

Alternatively, the options can be specified as a flat string: strength[:radius]

Examples

  • Apply the filter with a 3.5 strength and radius of 8:

            gradfun=3.5:8
        
  • Specify radius, omitting the strength (which will fall-back to the default value):

            gradfun=radius=8
        

graphmonitor, agraphmonitor

Show various filtergraph stats.

With this filter one can debug complete filtergraph. Especially issues with links filling with queued frames.

The filter accepts the following options:

size, s
Set video output size. Default is hd720.
opacity, o
Set video opacity. Default is 0.9. Allowed range is from 0 to 1.
mode, m
Set output mode, can be fulll or compact. In compact mode only filters with some queued frames have displayed stats.
flags, f
Set flags which enable which stats are shown in video.

Available values for flags are:

queue
Display number of queued frames in each link.
frame_count_in
Display number of frames taken from filter.
frame_count_out
Display number of frames given out from filter.
pts
Display current filtered frame pts.
time
Display current filtered frame time.
timebase
Display time base for filter link.
format
Display used format for filter link.
size
Display video size or number of audio channels in case of audio used by filter link.
rate
Display video frame rate or sample rate in case of audio used by filter link.
rate, r
Set upper limit for video rate of output stream, Default value is 25. This guarantee that output video frame rate will not be higher than this value.

greyedge

A color constancy variation filter which estimates scene illumination via grey edge algorithm and corrects the scene colors accordingly.

See: <https://staff.science.uva.nl/th.gevers/pub/GeversTIP07.pdf>

The filter accepts the following options:

difford
The order of differentiation to be applied on the scene. Must be chosen in the range [0,2] and default value is 1.
minknorm
The Minkowski parameter to be used for calculating the Minkowski distance. Must be chosen in the range [0,20] and default value is 1. Set to 0 for getting max value instead of calculating Minkowski distance.
sigma
The standard deviation of Gaussian blur to be applied on the scene. Must be chosen in the range [0,1024.0] and default value = 1. floor( sigma * break_off_sigma(3) ) can't be euqal to 0 if difford is greater than 0.

Examples

  • Grey Edge:

            greyedge=difford=1:minknorm=5:sigma=2
        
  • Max Edge:

            greyedge=difford=1:minknorm=0:sigma=2
        

haldclut

Apply a Hald CLUT to a video stream.

First input is the video stream to process, and second one is the Hald CLUT. The Hald CLUT input can be a simple picture or a complete video stream.

The filter accepts the following options:

shortest
Force termination when the shortest input terminates. Default is 0.
repeatlast
Continue applying the last CLUT after the end of the stream. A value of 0 disable the filter after the last frame of the CLUT is reached. Default is 1.

"haldclut" also has the same interpolation options as lut3d (both filters share the same internals).

More information about the Hald CLUT can be found on Eskil Steenberg's website (Hald CLUT author) at <http://www.quelsolaar.com/technology/clut.html>.

Workflow examples

Hald CLUT video stream

Generate an identity Hald CLUT stream altered with various effects:

        ffmpeg -f lavfi -i B<haldclutsrc>=8 -vf "hue=H=2*PI*t:s=sin(2*PI*t)+1, curves=cross_process" -t 10 -c:v ffv1 clut.nut

Note: make sure you use a lossless codec.

Then use it with "haldclut" to apply it on some random stream:

        ffmpeg -f lavfi -i mandelbrot -i clut.nut -filter_complex '[0][1] haldclut' -t 20 mandelclut.mkv

The Hald CLUT will be applied to the 10 first seconds (duration of clut.nut), then the latest picture of that CLUT stream will be applied to the remaining frames of the "mandelbrot" stream.

Hald CLUT with preview

A Hald CLUT is supposed to be a squared image of "Level*Level*Level" by "Level*Level*Level" pixels. For a given Hald CLUT, FFmpeg will select the biggest possible square starting at the top left of the picture. The remaining padding pixels (bottom or right) will be ignored. This area can be used to add a preview of the Hald CLUT.

Typically, the following generated Hald CLUT will be supported by the "haldclut" filter:

        ffmpeg -f lavfi -i B<haldclutsrc>=8 -vf "
           pad=iw+320 [padded_clut];
           smptebars=s=320x256, split [a][b];
           [padded_clut][a] overlay=W-320:h, curves=color_negative [main];
           [main][b] overlay=W-320" -frames:v 1 clut.png

It contains the original and a preview of the effect of the CLUT: SMPTE color bars are displayed on the right-top, and below the same color bars processed by the color changes.

Then, the effect of this Hald CLUT can be visualized with:

        ffplay input.mkv -vf "movie=clut.png, [in] haldclut"

hflip

Flip the input video horizontally.

For example, to horizontally flip the input video with ffmpeg:

        ffmpeg -i in.avi -vf "hflip" out.avi

histeq

This filter applies a global color histogram equalization on a per-frame basis.

It can be used to correct video that has a compressed range of pixel intensities. The filter redistributes the pixel intensities to equalize their distribution across the intensity range. It may be viewed as an "automatically adjusting contrast filter". This filter is useful only for correcting degraded or poorly captured source video.

The filter accepts the following options:

strength
Determine the amount of equalization to be applied. As the strength is reduced, the distribution of pixel intensities more-and-more approaches that of the input frame. The value must be a float number in the range [0,1] and defaults to 0.200.
intensity
Set the maximum intensity that can generated and scale the output values appropriately. The strength should be set as desired and then the intensity can be limited if needed to avoid washing-out. The value must be a float number in the range [0,1] and defaults to 0.210.
antibanding
Set the antibanding level. If enabled the filter will randomly vary the luminance of output pixels by a small amount to avoid banding of the histogram. Possible values are "none", "weak" or "strong". It defaults to "none".

histogram

Compute and draw a color distribution histogram for the input video.

The computed histogram is a representation of the color component distribution in an image.

Standard histogram displays the color components distribution in an image. Displays color graph for each color component. Shows distribution of the Y, U, V, A or R, G, B components, depending on input format, in the current frame. Below each graph a color component scale meter is shown.

The filter accepts the following options:

level_height
Set height of level. Default value is 200. Allowed range is [50, 2048].
scale_height
Set height of color scale. Default value is 12. Allowed range is [0, 40].
display_mode
Set display mode. It accepts the following values:
stack
Per color component graphs are placed below each other.
parade
Per color component graphs are placed side by side.
overlay
Presents information identical to that in the "parade", except that the graphs representing color components are superimposed directly over one another.

Default is "stack".

levels_mode
Set mode. Can be either "linear", or "logarithmic". Default is "linear".
components
Set what color components to display. Default is 7.
fgopacity
Set foreground opacity. Default is 0.7.
bgopacity
Set background opacity. Default is 0.5.

Examples

Calculate and draw histogram:

        ffplay -i input -vf histogram
    

hqdn3d

This is a high precision/quality 3d denoise filter. It aims to reduce image noise, producing smooth images and making still images really still. It should enhance compressibility.

It accepts the following optional parameters:

luma_spatial
A non-negative floating point number which specifies spatial luma strength. It defaults to 4.0.
chroma_spatial
A non-negative floating point number which specifies spatial chroma strength. It defaults to 3.0*luma_spatial/4.0.
luma_tmp
A floating point number which specifies luma temporal strength. It defaults to 6.0*luma_spatial/4.0.
chroma_tmp
A floating point number which specifies chroma temporal strength. It defaults to luma_tmp*chroma_spatial/luma_spatial.

hwdownload

Download hardware frames to system memory.

The input must be in hardware frames, and the output a non-hardware format. Not all formats will be supported on the output - it may be necessary to insert an additional format filter immediately following in the graph to get the output in a supported format.

hwmap

Map hardware frames to system memory or to another device.

This filter has several different modes of operation; which one is used depends on the input and output formats:

  • Hardware frame input, normal frame output

    Map the input frames to system memory and pass them to the output. If the original hardware frame is later required (for example, after overlaying something else on part of it), the hwmap filter can be used again in the next mode to retrieve it.

  • Normal frame input, hardware frame output

    If the input is actually a software-mapped hardware frame, then unmap it - that is, return the original hardware frame.

    Otherwise, a device must be provided. Create new hardware surfaces on that device for the output, then map them back to the software format at the input and give those frames to the preceding filter. This will then act like the hwupload filter, but may be able to avoid an additional copy when the input is already in a compatible format.

  • Hardware frame input and output

    A device must be supplied for the output, either directly or with the derive_device option. The input and output devices must be of different types and compatible - the exact meaning of this is system-dependent, but typically it means that they must refer to the same underlying hardware context (for example, refer to the same graphics card).

    If the input frames were originally created on the output device, then unmap to retrieve the original frames.

    Otherwise, map the frames to the output device - create new hardware frames on the output corresponding to the frames on the input.

The following additional parameters are accepted:

mode
Set the frame mapping mode. Some combination of:
read
The mapped frame should be readable.
write
The mapped frame should be writeable.
overwrite
The mapping will always overwrite the entire frame.

This may improve performance in some cases, as the original contents of the frame need not be loaded.

direct
The mapping must not involve any copying.

Indirect mappings to copies of frames are created in some cases where either direct mapping is not possible or it would have unexpected properties. Setting this flag ensures that the mapping is direct and will fail if that is not possible.

Defaults to read+write if not specified.

derive_device type
Rather than using the device supplied at initialisation, instead derive a new device of type type from the device the input frames exist on.
reverse
In a hardware to hardware mapping, map in reverse - create frames in the sink and map them back to the source. This may be necessary in some cases where a mapping in one direction is required but only the opposite direction is supported by the devices being used.

This option is dangerous - it may break the preceding filter in undefined ways if there are any additional constraints on that filter's output. Do not use it without fully understanding the implications of its use.

hwupload

Upload system memory frames to hardware surfaces.

The device to upload to must be supplied when the filter is initialised. If using ffmpeg, select the appropriate device with the -filter_hw_device option.

hwupload_cuda

Upload system memory frames to a CUDA device.

It accepts the following optional parameters:

device
The number of the CUDA device to use

hqx

Apply a high-quality magnification filter designed for pixel art. This filter was originally created by Maxim Stepin.

It accepts the following option:

n
Set the scaling dimension: 2 for "hq2x", 3 for "hq3x" and 4 for "hq4x". Default is 3.

hstack

Stack input videos horizontally.

All streams must be of same pixel format and of same height.

Note that this filter is faster than using overlay and pad filter to create same output.

The filter accept the following option:

inputs
Set number of input streams. Default is 2.
shortest
If set to 1, force the output to terminate when the shortest input terminates. Default value is 0.

hue

Modify the hue and/or the saturation of the input.

It accepts the following parameters:

h
Specify the hue angle as a number of degrees. It accepts an expression, and defaults to "0".
s
Specify the saturation in the [-10,10] range. It accepts an expression and defaults to "1".
H
Specify the hue angle as a number of radians. It accepts an expression, and defaults to "0".
b
Specify the brightness in the [-10,10] range. It accepts an expression and defaults to "0".

h and H are mutually exclusive, and can't be specified at the same time.

The b, h, H and s option values are expressions containing the following constants:

n
frame count of the input frame starting from 0
pts
presentation timestamp of the input frame expressed in time base units
r
frame rate of the input video, NAN if the input frame rate is unknown
t
timestamp expressed in seconds, NAN if the input timestamp is unknown
tb
time base of the input video

Examples

  • Set the hue to 90 degrees and the saturation to 1.0:

            hue=h=90:s=1
        
  • Same command but expressing the hue in radians:

            hue=H=PI/2:s=1
        
  • Rotate hue and make the saturation swing between 0 and 2 over a period of 1 second:

            hue="H=2*PI*t: s=sin(2*PI*t)+1"
        
  • Apply a 3 seconds saturation fade-in effect starting at 0:

            hue="s=min(t/3\,1)"
        

    The general fade-in expression can be written as:

            hue="s=min(0\, max((t-START)/DURATION\, 1))"
        
  • Apply a 3 seconds saturation fade-out effect starting at 5 seconds:

            hue="s=max(0\, min(1\, (8-t)/3))"
        

    The general fade-out expression can be written as:

            hue="s=max(0\, min(1\, (START+DURATION-t)/DURATION))"
        

Commands

This filter supports the following commands:

b
s
h
H
Modify the hue and/or the saturation and/or brightness of the input video. The command accepts the same syntax of the corresponding option.

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

hysteresis

Grow first stream into second stream by connecting components. This makes it possible to build more robust edge masks.

This filter accepts the following options:

planes
Set which planes will be processed as bitmap, unprocessed planes will be copied from first stream. By default value 0xf, all planes will be processed.
threshold
Set threshold which is used in filtering. If pixel component value is higher than this value filter algorithm for connecting components is activated. By default value is 0.

idet

Detect video interlacing type.

This filter tries to detect if the input frames are interlaced, progressive, top or bottom field first. It will also try to detect fields that are repeated between adjacent frames (a sign of telecine).

Single frame detection considers only immediately adjacent frames when classifying each frame. Multiple frame detection incorporates the classification history of previous frames.

The filter will log these metadata values:

single.current_frame
Detected type of current frame using single-frame detection. One of: ``tff'' (top field first), ``bff'' (bottom field first), ``progressive'', or ``undetermined''
single.tff
Cumulative number of frames detected as top field first using single-frame detection.
multiple.tff
Cumulative number of frames detected as top field first using multiple-frame detection.
single.bff
Cumulative number of frames detected as bottom field first using single-frame detection.
multiple.current_frame
Detected type of current frame using multiple-frame detection. One of: ``tff'' (top field first), ``bff'' (bottom field first), ``progressive'', or ``undetermined''
multiple.bff
Cumulative number of frames detected as bottom field first using multiple-frame detection.
single.progressive
Cumulative number of frames detected as progressive using single-frame detection.
multiple.progressive
Cumulative number of frames detected as progressive using multiple-frame detection.
single.undetermined
Cumulative number of frames that could not be classified using single-frame detection.
multiple.undetermined
Cumulative number of frames that could not be classified using multiple-frame detection.
repeated.current_frame
Which field in the current frame is repeated from the last. One of ``neither'', ``top'', or ``bottom''.
repeated.neither
Cumulative number of frames with no repeated field.
repeated.top
Cumulative number of frames with the top field repeated from the previous frame's top field.
repeated.bottom
Cumulative number of frames with the bottom field repeated from the previous frame's bottom field.

The filter accepts the following options:

intl_thres
Set interlacing threshold.
prog_thres
Set progressive threshold.
rep_thres
Threshold for repeated field detection.
half_life
Number of frames after which a given frame's contribution to the statistics is halved (i.e., it contributes only 0.5 to its classification). The default of 0 means that all frames seen are given full weight of 1.0 forever.
analyze_interlaced_flag
When this is not 0 then idet will use the specified number of frames to determine if the interlaced flag is accurate, it will not count undetermined frames. If the flag is found to be accurate it will be used without any further computations, if it is found to be inaccurate it will be cleared without any further computations. This allows inserting the idet filter as a low computational method to clean up the interlaced flag

il

Deinterleave or interleave fields.

This filter allows one to process interlaced images fields without deinterlacing them. Deinterleaving splits the input frame into 2 fields (so called half pictures). Odd lines are moved to the top half of the output image, even lines to the bottom half. You can process (filter) them independently and then re-interleave them.

The filter accepts the following options:

luma_mode, l
chroma_mode, c
alpha_mode, a
Available values for luma_mode, chroma_mode and alpha_mode are:
none
Do nothing.
deinterleave, d
Deinterleave fields, placing one above the other.
interleave, i
Interleave fields. Reverse the effect of deinterleaving.

Default value is "none".

luma_swap, ls
chroma_swap, cs
alpha_swap, as
Swap luma/chroma/alpha fields. Exchange even & odd lines. Default value is 0.

inflate

Apply inflate effect to the video.

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

It accepts the following options:

threshold0
threshold1
threshold2
threshold3
Limit the maximum change for each plane, default is 65535. If 0, plane will remain unchanged.

interlace

Simple interlacing filter from progressive contents. This interleaves upper (or lower) lines from odd frames with lower (or upper) lines from even frames, halving the frame rate and preserving image height.

           Original        Original             New Frame
           Frame 'j'      Frame 'j+1'             (tff)
          ==========      ===========       ==================
            Line 0  -------------------->    Frame 'j' Line 0
            Line 1          Line 1  ---->   Frame 'j+1' Line 1
            Line 2 --------------------->    Frame 'j' Line 2
            Line 3          Line 3  ---->   Frame 'j+1' Line 3
             ...             ...                   ...
        New Frame + 1 will be generated by Frame 'j+2' and Frame 'j+3' and so on

It accepts the following optional parameters:

scan
This determines whether the interlaced frame is taken from the even (tff - default) or odd (bff) lines of the progressive frame.
lowpass
Vertical lowpass filter to avoid twitter interlacing and reduce moire patterns.
0, off
Disable vertical lowpass filter
1, linear
Enable linear filter (default)
2, complex
Enable complex filter. This will slightly less reduce twitter and moire but better retain detail and subjective sharpness impression.

kerndeint

Deinterlace input video by applying Donald Graft's adaptive kernel deinterling. Work on interlaced parts of a video to produce progressive frames.

The description of the accepted parameters follows.

thresh
Set the threshold which affects the filter's tolerance when determining if a pixel line must be processed. It must be an integer in the range [0,255] and defaults to 10. A value of 0 will result in applying the process on every pixels.
map
Paint pixels exceeding the threshold value to white if set to 1. Default is 0.
order
Set the fields order. Swap fields if set to 1, leave fields alone if 0. Default is 0.
sharp
Enable additional sharpening if set to 1. Default is 0.
twoway
Enable twoway sharpening if set to 1. Default is 0.

Examples

  • Apply default values:

            kerndeint=thresh=10:map=0:order=0:sharp=0:twoway=0
        
  • Enable additional sharpening:

            kerndeint=sharp=1
        
  • Paint processed pixels in white:

            kerndeint=map=1
        

lenscorrection

Correct radial lens distortion

This filter can be used to correct for radial distortion as can result from the use of wide angle lenses, and thereby re-rectify the image. To find the right parameters one can use tools available for example as part of opencv or simply trial-and-error. To use opencv use the calibration sample (under samples/cpp) from the opencv sources and extract the k1 and k2 coefficients from the resulting matrix.

Note that effectively the same filter is available in the open-source tools Krita and Digikam from the KDE project.

In contrast to the vignette filter, which can also be used to compensate lens errors, this filter corrects the distortion of the image, whereas vignette corrects the brightness distribution, so you may want to use both filters together in certain cases, though you will have to take care of ordering, i.e. whether vignetting should be applied before or after lens correction.

Options

The filter accepts the following options:

cx
Relative x-coordinate of the focal point of the image, and thereby the center of the distortion. This value has a range [0,1] and is expressed as fractions of the image width. Default is 0.5.
cy
Relative y-coordinate of the focal point of the image, and thereby the center of the distortion. This value has a range [0,1] and is expressed as fractions of the image height. Default is 0.5.
k1
Coefficient of the quadratic correction term. This value has a range [-1,1]. 0 means no correction. Default is 0.
k2
Coefficient of the double quadratic correction term. This value has a range [-1,1]. 0 means no correction. Default is 0.

The formula that generates the correction is:

r_src = r_tgt * (1 + k1 * (r_tgt / r_0)^2 + k2 * (r_tgt / r_0)^4)

where r_0 is halve of the image diagonal and r_src and r_tgt are the distances from the focal point in the source and target images, respectively.

lensfun

Apply lens correction via the lensfun library (<http://lensfun.sourceforge.net/>).

The "lensfun" filter requires the camera make, camera model, and lens model to apply the lens correction. The filter will load the lensfun database and query it to find the corresponding camera and lens entries in the database. As long as these entries can be found with the given options, the filter can perform corrections on frames. Note that incomplete strings will result in the filter choosing the best match with the given options, and the filter will output the chosen camera and lens models (logged with level "info"). You must provide the make, camera model, and lens model as they are required.

The filter accepts the following options:

make
The make of the camera (for example, "Canon"). This option is required.
model
The model of the camera (for example, "Canon EOS 100D"). This option is required.
lens_model
The model of the lens (for example, "Canon EF-S 18-55mm f/3.5-5.6 IS STM"). This option is required.
mode
The type of correction to apply. The following values are valid options:
vignetting
Enables fixing lens vignetting.
geometry
Enables fixing lens geometry. This is the default.
subpixel
Enables fixing chromatic aberrations.
vig_geo
Enables fixing lens vignetting and lens geometry.
vig_subpixel
Enables fixing lens vignetting and chromatic aberrations.
distortion
Enables fixing both lens geometry and chromatic aberrations.
all
Enables all possible corrections.
focal_length
The focal length of the image/video (zoom; expected constant for video). For example, a 18--55mm lens has focal length range of [18--55], so a value in that range should be chosen when using that lens. Default 18.
aperture
The aperture of the image/video (expected constant for video). Note that aperture is only used for vignetting correction. Default 3.5.
focus_distance
The focus distance of the image/video (expected constant for video). Note that focus distance is only used for vignetting and only slightly affects the vignetting correction process. If unknown, leave it at the default value (which is 1000).
target_geometry
The target geometry of the output image/video. The following values are valid options:
rectilinear (default)
fisheye
panoramic
equirectangular
fisheye_orthographic
fisheye_stereographic
fisheye_equisolid
fisheye_thoby
reverse
Apply the reverse of image correction (instead of correcting distortion, apply it).
interpolation
The type of interpolation used when correcting distortion. The following values are valid options:
nearest
linear (default)
lanczos

Examples

  • Apply lens correction with make "Canon", camera model "Canon EOS 100D", and lens model "Canon EF-S 18-55mm f/3.5-5.6 IS STM" with focal length of "18" and aperture of "8.0".

            ffmpeg -i input.mov -vf lensfun=make=Canon:model="Canon EOS 100D":lens_model="Canon EF-S 18-55mm f/3.5-5.6 IS STM":focal_length=18:aperture=8 -c:v h264 -b:v 8000k output.mov
        
  • Apply the same as before, but only for the first 5 seconds of video.

            ffmpeg -i input.mov -vf lensfun=make=Canon:model="Canon EOS 100D":lens_model="Canon EF-S 18-55mm f/3.5-5.6 IS STM":focal_length=18:aperture=8:enable='lte(t\,5)' -c:v h264 -b:v 8000k output.mov
        

libvmaf

Obtain the VMAF (Video Multi-Method Assessment Fusion) score between two input videos.

The obtained VMAF score is printed through the logging system.

It requires Netflix's vmaf library (libvmaf) as a pre-requisite. After installing the library it can be enabled using: "./configure --enable-libvmaf --enable-version3". If no model path is specified it uses the default model: "vmaf_v0.6.1.pkl".

The filter has following options:

model_path
Set the model path which is to be used for SVM. Default value: "vmaf_v0.6.1.pkl"
log_path
Set the file path to be used to store logs.
log_fmt
Set the format of the log file (xml or json).
enable_transform
Enables transform for computing vmaf.
phone_model
Invokes the phone model which will generate VMAF scores higher than in the regular model, which is more suitable for laptop, TV, etc. viewing conditions.
psnr
Enables computing psnr along with vmaf.
ssim
Enables computing ssim along with vmaf.
ms_ssim
Enables computing ms_ssim along with vmaf.
pool
Set the pool method (mean, min or harmonic mean) to be used for computing vmaf.
n_threads
Set number of threads to be used when computing vmaf.
n_subsample
Set interval for frame subsampling used when computing vmaf.
enable_conf_interval
Enables confidence interval.

This filter also supports the framesync options.

On the below examples the input file main.mpg being processed is compared with the reference file ref.mpg.

        ffmpeg -i main.mpg -i ref.mpg -lavfi libvmaf -f null -

Example with options:

        ffmpeg -i main.mpg -i ref.mpg -lavfi libvmaf="psnr=1:enable-transform=1" -f null -

limiter

Limits the pixel components values to the specified range [min, max].

The filter accepts the following options:

min
Lower bound. Defaults to the lowest allowed value for the input.
max
Upper bound. Defaults to the highest allowed value for the input.
planes
Specify which planes will be processed. Defaults to all available.

loop

Loop video frames.

The filter accepts the following options:

loop
Set the number of loops. Setting this value to -1 will result in infinite loops. Default is 0.
size
Set maximal size in number of frames. Default is 0.
start
Set first frame of loop. Default is 0.

Examples

  • Loop single first frame infinitely:

            loop=loop=-1:size=1:start=0
        
  • Loop single first frame 10 times:

            loop=loop=10:size=1:start=0
        
  • Loop 10 first frames 5 times:

            loop=loop=5:size=10:start=0
        

lut1d

Apply a 1D LUT to an input video.

The filter accepts the following options:

file
Set the 1D LUT file name.

Currently supported formats:

cube
Iridas
interp
Select interpolation mode.

Available values are:

nearest
Use values from the nearest defined point.
linear
Interpolate values using the linear interpolation.
cubic
Interpolate values using the cubic interpolation.

lut3d

Apply a 3D LUT to an input video.

The filter accepts the following options:

file
Set the 3D LUT file name.

Currently supported formats:

3dl
AfterEffects
cube
Iridas
dat
DaVinci
m3d
Pandora
interp
Select interpolation mode.

Available values are:

nearest
Use values from the nearest defined point.
trilinear
Interpolate values using the 8 points defining a cube.
tetrahedral
Interpolate values using a tetrahedron.

This filter also supports the framesync options.

lumakey

Turn certain luma values into transparency.

The filter accepts the following options:

threshold
Set the luma which will be used as base for transparency. Default value is 0.
tolerance
Set the range of luma values to be keyed out. Default value is 0.
softness
Set the range of softness. Default value is 0. Use this to control gradual transition from zero to full transparency.

lut, lutrgb, lutyuv

Compute a look-up table for binding each pixel component input value to an output value, and apply it to the input video.

lutyuv applies a lookup table to a YUV input video, lutrgb to an RGB input video.

These filters accept the following parameters:

c0
set first pixel component expression
c1
set second pixel component expression
c2
set third pixel component expression
c3
set fourth pixel component expression, corresponds to the alpha component
r
set red component expression
g
set green component expression
b
set blue component expression
a
alpha component expression
y
set Y/luminance component expression
u
set U/Cb component expression
v
set V/Cr component expression

Each of them specifies the expression to use for computing the lookup table for the corresponding pixel component values.

The exact component associated to each of the c* options depends on the format in input.

The lut filter requires either YUV or RGB pixel formats in input, lutrgb requires RGB pixel formats in input, and lutyuv requires YUV.

The expressions can contain the following constants and functions:

w
h
The input width and height.
val
The input value for the pixel component.
clipval
The input value, clipped to the minval-maxval range.
maxval
The maximum value for the pixel component.
minval
The minimum value for the pixel component.
negval
The negated value for the pixel component value, clipped to the minval-maxval range; it corresponds to the expression "maxval-clipval+minval".
clip(val)
The computed value in val, clipped to the minval-maxval range.
gammaval(gamma)
The computed gamma correction value of the pixel component value, clipped to the minval-maxval range. It corresponds to the expression "pow((clipval-minval)/(maxval-minval)\,gamma)*(maxval-minval)+minval"

All expressions default to "val".

Examples

  • Negate input video:

            lutrgb="r=maxval+minval-val:g=maxval+minval-val:b=maxval+minval-val"
            lutyuv="y=maxval+minval-val:u=maxval+minval-val:v=maxval+minval-val"
        

    The above is the same as:

            lutrgb="r=negval:g=negval:b=negval"
            lutyuv="y=negval:u=negval:v=negval"
        
  • Negate luminance:

            lutyuv=y=negval
        
  • Remove chroma components, turning the video into a graytone image:

            lutyuv="u=128:v=128"
        
  • Apply a luma burning effect:

            lutyuv="y=2*val"
        
  • Remove green and blue components:

            lutrgb="g=0:b=0"
        
  • Set a constant alpha channel value on input:

            format=rgba,lutrgb=a="maxval-minval/2"
        
  • Correct luminance gamma by a factor of 0.5:

            lutyuv=y=gammaval(0.5)
        
  • Discard least significant bits of luma:

            lutyuv=y='bitand(val, 128+64+32)'
        
  • Technicolor like effect:

            lutyuv=u='(val-maxval/2)*2+maxval/2':v='(val-maxval/2)*2+maxval/2'
        

lut2, tlut2

The "lut2" filter takes two input streams and outputs one stream.

The "tlut2" (time lut2) filter takes two consecutive frames from one single stream.

This filter accepts the following parameters:

c0
set first pixel component expression
c1
set second pixel component expression
c2
set third pixel component expression
c3
set fourth pixel component expression, corresponds to the alpha component

Each of them specifies the expression to use for computing the lookup table for the corresponding pixel component values.

The exact component associated to each of the c* options depends on the format in inputs.

The expressions can contain the following constants:

w
h
The input width and height.
x
The first input value for the pixel component.
y
The second input value for the pixel component.
bdx
The first input video bit depth.
bdy
The second input video bit depth.

All expressions default to "x".

Examples

  • Highlight differences between two RGB video streams:

            lut2='ifnot(x-y,0,pow(2,bdx)-1):ifnot(x-y,0,pow(2,bdx)-1):ifnot(x-y,0,pow(2,bdx)-1)'
        
  • Highlight differences between two YUV video streams:

            lut2='ifnot(x-y,0,pow(2,bdx)-1):ifnot(x-y,pow(2,bdx-1),pow(2,bdx)-1):ifnot(x-y,pow(2,bdx-1),pow(2,bdx)-1)'
        
  • Show max difference between two video streams:

            lut2='if(lt(x,y),0,if(gt(x,y),pow(2,bdx)-1,pow(2,bdx-1))):if(lt(x,y),0,if(gt(x,y),pow(2,bdx)-1,pow(2,bdx-1))):if(lt(x,y),0,if(gt(x,y),pow(2,bdx)-1,pow(2,bdx-1)))'
        

maskedclamp

Clamp the first input stream with the second input and third input stream.

Returns the value of first stream to be between second input stream - "undershoot" and third input stream + "overshoot".

This filter accepts the following options:

undershoot
Default value is 0.
overshoot
Default value is 0.
planes
Set which planes will be processed as bitmap, unprocessed planes will be copied from first stream. By default value 0xf, all planes will be processed.

maskedmerge

Merge the first input stream with the second input stream using per pixel weights in the third input stream.

A value of 0 in the third stream pixel component means that pixel component from first stream is returned unchanged, while maximum value (eg. 255 for 8-bit videos) means that pixel component from second stream is returned unchanged. Intermediate values define the amount of merging between both input stream's pixel components.

This filter accepts the following options:

planes
Set which planes will be processed as bitmap, unprocessed planes will be copied from first stream. By default value 0xf, all planes will be processed.

mcdeint

Apply motion-compensation deinterlacing.

It needs one field per frame as input and must thus be used together with yadif=1/3 or equivalent.

This filter accepts the following options:

mode
Set the deinterlacing mode.

It accepts one of the following values:

fast
medium
slow
use iterative motion estimation
extra_slow
like slow, but use multiple reference frames.

Default value is fast.

parity
Set the picture field parity assumed for the input video. It must be one of the following values:
0, tff
assume top field first
1, bff
assume bottom field first

Default value is bff.

qp
Set per-block quantization parameter (QP) used by the internal encoder.

Higher values should result in a smoother motion vector field but less optimal individual vectors. Default value is 1.

mergeplanes

Merge color channel components from several video streams.

The filter accepts up to 4 input streams, and merge selected input planes to the output video.

This filter accepts the following options:

mapping
Set input to output plane mapping. Default is 0.

The mappings is specified as a bitmap. It should be specified as a hexadecimal number in the form 0xAa[Bb[Cc[Dd]]]. 'Aa' describes the mapping for the first plane of the output stream. 'A' sets the number of the input stream to use (from 0 to 3), and 'a' the plane number of the corresponding input to use (from 0 to 3). The rest of the mappings is similar, 'Bb' describes the mapping for the output stream second plane, 'Cc' describes the mapping for the output stream third plane and 'Dd' describes the mapping for the output stream fourth plane.

format
Set output pixel format. Default is "yuva444p".

Examples

  • Merge three gray video streams of same width and height into single video stream:

            [a0][a1][a2]mergeplanes=0x001020:yuv444p
        
  • Merge 1st yuv444p stream and 2nd gray video stream into yuva444p video stream:

            [a0][a1]mergeplanes=0x00010210:yuva444p
        
  • Swap Y and A plane in yuva444p stream:

            format=yuva444p,mergeplanes=0x03010200:yuva444p
        
  • Swap U and V plane in yuv420p stream:

            format=yuv420p,mergeplanes=0x000201:yuv420p
        
  • Cast a rgb24 clip to yuv444p:

            format=rgb24,mergeplanes=0x000102:yuv444p
        

mestimate

Estimate and export motion vectors using block matching algorithms. Motion vectors are stored in frame side data to be used by other filters.

This filter accepts the following options:

method
Specify the motion estimation method. Accepts one of the following values:
esa
Exhaustive search algorithm.
tss
Three step search algorithm.
tdls
Two dimensional logarithmic search algorithm.
ntss
New three step search algorithm.
fss
Four step search algorithm.
ds
Diamond search algorithm.
hexbs
Hexagon-based search algorithm.
epzs
Enhanced predictive zonal search algorithm.
umh
Uneven multi-hexagon search algorithm.

Default value is esa.

mb_size
Macroblock size. Default 16.
search_param
Search parameter. Default 7.

midequalizer

Apply Midway Image Equalization effect using two video streams.

Midway Image Equalization adjusts a pair of images to have the same histogram, while maintaining their dynamics as much as possible. It's useful for e.g. matching exposures from a pair of stereo cameras.

This filter has two inputs and one output, which must be of same pixel format, but may be of different sizes. The output of filter is first input adjusted with midway histogram of both inputs.

This filter accepts the following option:

planes
Set which planes to process. Default is 15, which is all available planes.

minterpolate

Convert the video to specified frame rate using motion interpolation.

This filter accepts the following options:

fps
Specify the output frame rate. This can be rational e.g. "60000/1001". Frames are dropped if fps is lower than source fps. Default 60.
mi_mode
Motion interpolation mode. Following values are accepted:
dup
Duplicate previous or next frame for interpolating new ones.
blend
Blend source frames. Interpolated frame is mean of previous and next frames.
mci
Motion compensated interpolation. Following options are effective when this mode is selected:
mc_mode
Motion compensation mode. Following values are accepted:
obmc
Overlapped block motion compensation.
aobmc
Adaptive overlapped block motion compensation. Window weighting coefficients are controlled adaptively according to the reliabilities of the neighboring motion vectors to reduce oversmoothing.

Default mode is obmc.

me_mode
Motion estimation mode. Following values are accepted:
bidir
Bidirectional motion estimation. Motion vectors are estimated for each source frame in both forward and backward directions.
bilat
Bilateral motion estimation. Motion vectors are estimated directly for interpolated frame.

Default mode is bilat.

me
The algorithm to be used for motion estimation. Following values are accepted:
esa
Exhaustive search algorithm.
tss
Three step search algorithm.
tdls
Two dimensional logarithmic search algorithm.
ntss
New three step search algorithm.
fss
Four step search algorithm.
ds
Diamond search algorithm.
hexbs
Hexagon-based search algorithm.
epzs
Enhanced predictive zonal search algorithm.
umh
Uneven multi-hexagon search algorithm.

Default algorithm is epzs.

mb_size
Macroblock size. Default 16.
search_param
Motion estimation search parameter. Default 32.
vsbmc
Enable variable-size block motion compensation. Motion estimation is applied with smaller block sizes at object boundaries in order to make the them less blur. Default is 0 (disabled).
scd
Scene change detection method. Scene change leads motion vectors to be in random direction. Scene change detection replace interpolated frames by duplicate ones. May not be needed for other modes. Following values are accepted:
none
Disable scene change detection.
fdiff
Frame difference. Corresponding pixel values are compared and if it satisfies scd_threshold scene change is detected.

Default method is fdiff.

scd_threshold
Scene change detection threshold. Default is 5.0.

mix

Mix several video input streams into one video stream.

A description of the accepted options follows.

nb_inputs
The number of inputs. If unspecified, it defaults to 2.
weights
Specify weight of each input video stream as sequence. Each weight is separated by space. If number of weights is smaller than number of frames last specified weight will be used for all remaining unset weights.
scale
Specify scale, if it is set it will be multiplied with sum of each weight multiplied with pixel values to give final destination pixel value. By default scale is auto scaled to sum of weights.
duration
Specify how end of stream is determined.
longest
The duration of the longest input. (default)
shortest
The duration of the shortest input.
first
The duration of the first input.

mpdecimate

Drop frames that do not differ greatly from the previous frame in order to reduce frame rate.

The main use of this filter is for very-low-bitrate encoding (e.g. streaming over dialup modem), but it could in theory be used for fixing movies that were inverse-telecined incorrectly.

A description of the accepted options follows.

max
Set the maximum number of consecutive frames which can be dropped (if positive), or the minimum interval between dropped frames (if negative). If the value is 0, the frame is dropped disregarding the number of previous sequentially dropped frames.

Default value is 0.

hi
lo
frac
Set the dropping threshold values.

Values for hi and lo are for 8x8 pixel blocks and represent actual pixel value differences, so a threshold of 64 corresponds to 1 unit of difference for each pixel, or the same spread out differently over the block.

A frame is a candidate for dropping if no 8x8 blocks differ by more than a threshold of hi, and if no more than frac blocks (1 meaning the whole image) differ by more than a threshold of lo.

Default value for hi is 64*12, default value for lo is 64*5, and default value for frac is 0.33.

negate

Negate (invert) the input video.

It accepts the following option:

negate_alpha
With value 1, it negates the alpha component, if present. Default value is 0.

nlmeans

Denoise frames using Non-Local Means algorithm.

Each pixel is adjusted by looking for other pixels with similar contexts. This context similarity is defined by comparing their surrounding patches of size pxp. Patches are searched in an area of rxr around the pixel.

Note that the research area defines centers for patches, which means some patches will be made of pixels outside that research area.

The filter accepts the following options.

s
Set denoising strength.
p
Set patch size.
pc
Same as p but for chroma planes.

The default value is 0 and means automatic.

r
Set research size.
rc
Same as r but for chroma planes.

The default value is 0 and means automatic.

nnedi

Deinterlace video using neural network edge directed interpolation.

This filter accepts the following options:

weights
Mandatory option, without binary file filter can not work. Currently file can be found here: https://github.com/dubhater/vapoursynth-nnedi3/blob/master/src/nnedi3_weights.bin
deint
Set which frames to deinterlace, by default it is "all". Can be "all" or "interlaced".
field
Set mode of operation.

Can be one of the following:

af
Use frame flags, both fields.
a
Use frame flags, single field.
t
Use top field only.
b
Use bottom field only.
tf
Use both fields, top first.
bf
Use both fields, bottom first.
planes
Set which planes to process, by default filter process all frames.
nsize
Set size of local neighborhood around each pixel, used by the predictor neural network.

Can be one of the following:

s8x6
s16x6
s32x6
s48x6
s8x4
s16x4
s32x4
nns
Set the number of neurons in predictor neural network. Can be one of the following:
n16
n32
n64
n128
n256
qual
Controls the number of different neural network predictions that are blended together to compute the final output value. Can be "fast", default or "slow".
etype
Set which set of weights to use in the predictor. Can be one of the following:
a
weights trained to minimize absolute error
s
weights trained to minimize squared error
pscrn
Controls whether or not the prescreener neural network is used to decide which pixels should be processed by the predictor neural network and which can be handled by simple cubic interpolation. The prescreener is trained to know whether cubic interpolation will be sufficient for a pixel or whether it should be predicted by the predictor nn. The computational complexity of the prescreener nn is much less than that of the predictor nn. Since most pixels can be handled by cubic interpolation, using the prescreener generally results in much faster processing. The prescreener is pretty accurate, so the difference between using it and not using it is almost always unnoticeable.

Can be one of the following:

none
original
new

Default is "new".

fapprox
Set various debugging flags.

noformat

Force libavfilter not to use any of the specified pixel formats for the input to the next filter.

It accepts the following parameters:

pix_fmts
A '|'-separated list of pixel format names, such as pix_fmts=yuv420p|monow|rgb24".

Examples

  • Force libavfilter to use a format different from yuv420p for the input to the vflip filter:

            noformat=pix_fmts=yuv420p,vflip
        
  • Convert the input video to any of the formats not contained in the list:

            noformat=yuv420p|yuv444p|yuv410p
        

noise

Add noise on video input frame.

The filter accepts the following options:

all_seed
c0_seed
c1_seed
c2_seed
c3_seed
Set noise seed for specific pixel component or all pixel components in case of all_seed. Default value is 123457.
all_strength, alls
c0_strength, c0s
c1_strength, c1s
c2_strength, c2s
c3_strength, c3s
Set noise strength for specific pixel component or all pixel components in case all_strength. Default value is 0. Allowed range is [0, 100].
all_flags, allf
c0_flags, c0f
c1_flags, c1f
c2_flags, c2f
c3_flags, c3f
Set pixel component flags or set flags for all components if all_flags. Available values for component flags are:
a
averaged temporal noise (smoother)
p
mix random noise with a (semi)regular pattern
t
temporal noise (noise pattern changes between frames)
u
uniform noise (gaussian otherwise)

Examples

Add temporal and uniform noise to input video:

        noise=alls=20:allf=t+u

normalize

Normalize RGB video (aka histogram stretching, contrast stretching). See: https://en.wikipedia.org/wiki/Normalization_(image_processing)

For each channel of each frame, the filter computes the input range and maps it linearly to the user-specified output range. The output range defaults to the full dynamic range from pure black to pure white.

Temporal smoothing can be used on the input range to reduce flickering (rapid changes in brightness) caused when small dark or bright objects enter or leave the scene. This is similar to the auto-exposure (automatic gain control) on a video camera, and, like a video camera, it may cause a period of over- or under-exposure of the video.

The R,G,B channels can be normalized independently, which may cause some color shifting, or linked together as a single channel, which prevents color shifting. Linked normalization preserves hue. Independent normalization does not, so it can be used to remove some color casts. Independent and linked normalization can be combined in any ratio.

The normalize filter accepts the following options:

blackpt
whitept
Colors which define the output range. The minimum input value is mapped to the blackpt. The maximum input value is mapped to the whitept. The defaults are black and white respectively. Specifying white for blackpt and black for whitept will give color-inverted, normalized video. Shades of grey can be used to reduce the dynamic range (contrast). Specifying saturated colors here can create some interesting effects.
smoothing
The number of previous frames to use for temporal smoothing. The input range of each channel is smoothed using a rolling average over the current frame and the smoothing previous frames. The default is 0 (no temporal smoothing).
independence
Controls the ratio of independent (color shifting) channel normalization to linked (color preserving) normalization. 0.0 is fully linked, 1.0 is fully independent. Defaults to 1.0 (fully independent).
strength
Overall strength of the filter. 1.0 is full strength. 0.0 is a rather expensive no-op. Defaults to 1.0 (full strength).

Examples

Stretch video contrast to use the full dynamic range, with no temporal smoothing; may flicker depending on the source content:

        normalize=blackpt=black:whitept=white:smoothing=0

As above, but with 50 frames of temporal smoothing; flicker should be reduced, depending on the source content:

        normalize=blackpt=black:whitept=white:smoothing=50

As above, but with hue-preserving linked channel normalization:

        normalize=blackpt=black:whitept=white:smoothing=50:independence=0

As above, but with half strength:

        normalize=blackpt=black:whitept=white:smoothing=50:independence=0:strength=0.5

Map the darkest input color to red, the brightest input color to cyan:

        normalize=blackpt=red:whitept=cyan

null

Pass the video source unchanged to the output.

ocr

Optical Character Recognition

This filter uses Tesseract for optical character recognition. To enable compilation of this filter, you need to configure FFmpeg with "--enable-libtesseract".

It accepts the following options:

datapath
Set datapath to tesseract data. Default is to use whatever was set at installation.
language
Set language, default is "eng".
whitelist
Set character whitelist.
blacklist
Set character blacklist.

The filter exports recognized text as the frame metadata "lavfi.ocr.text".

ocv

Apply a video transform using libopencv.

To enable this filter, install the libopencv library and headers and configure FFmpeg with "--enable-libopencv".

It accepts the following parameters:

filter_name
The name of the libopencv filter to apply.
filter_params
The parameters to pass to the libopencv filter. If not specified, the default values are assumed.

Refer to the official libopencv documentation for more precise information: <http://docs.opencv.org/master/modules/imgproc/doc/filtering.html>

Several libopencv filters are supported; see the following subsections.

dilate

Dilate an image by using a specific structuring element. It corresponds to the libopencv function "cvDilate".

It accepts the parameters: struct_el|nb_iterations.

struct_el represents a structuring element, and has the syntax: colsxrows+anchor_xxanchor_y/shape

cols and rows represent the number of columns and rows of the structuring element, anchor_x and anchor_y the anchor point, and shape the shape for the structuring element. shape must be "rect", "cross", "ellipse", or "custom".

If the value for shape is "custom", it must be followed by a string of the form "=filename". The file with name filename is assumed to represent a binary image, with each printable character corresponding to a bright pixel. When a custom shape is used, cols and rows are ignored, the number or columns and rows of the read file are assumed instead.

The default value for struct_el is "3x3+0x0/rect".

nb_iterations specifies the number of times the transform is applied to the image, and defaults to 1.

Some examples:

        # Use the default values
        ocv=dilate
        
        # Dilate using a structuring element with a 5x5 cross, iterating two times
        ocv=filter_name=dilate:filter_params=5x5+2x2/cross|2
        
        # Read the shape from the file diamond.shape, iterating two times.
        # The file diamond.shape may contain a pattern of characters like this
        #   *
        #  ***
        # *****
        #  ***
        #   *
        # The specified columns and rows are ignored
        # but the anchor point coordinates are not
        ocv=dilate:0x0+2x2/custom=diamond.shape|2

erode

Erode an image by using a specific structuring element. It corresponds to the libopencv function "cvErode".

It accepts the parameters: struct_el:nb_iterations, with the same syntax and semantics as the dilate filter.

smooth

Smooth the input video.

The filter takes the following parameters: type|param1|param2|param3|param4.

type is the type of smooth filter to apply, and must be one of the following values: "blur", "blur_no_scale", "median", "gaussian", or "bilateral". The default value is "gaussian".

The meaning of param1, param2, param3, and param4 depend on the smooth type. param1 and param2 accept integer positive values or 0. param3 and param4 accept floating point values.

The default value for param1 is 3. The default value for the other parameters is 0.

These parameters correspond to the parameters assigned to the libopencv function "cvSmooth".

oscilloscope

2D Video Oscilloscope.

Useful to measure spatial impulse, step responses, chroma delays, etc.

It accepts the following parameters:

x
Set scope center x position.
y
Set scope center y position.
s
Set scope size, relative to frame diagonal.
t
Set scope tilt/rotation.
o
Set trace opacity.
tx
Set trace center x position.
ty
Set trace center y position.
tw
Set trace width, relative to width of frame.
th
Set trace height, relative to height of frame.
c
Set which components to trace. By default it traces first three components.
g
Draw trace grid. By default is enabled.
st
Draw some statistics. By default is enabled.
sc
Draw scope. By default is enabled.

Examples

  • Inspect full first row of video frame.

            oscilloscope=x=0.5:y=0:s=1
        
  • Inspect full last row of video frame.

            oscilloscope=x=0.5:y=1:s=1
        
  • Inspect full 5th line of video frame of height 1080.

            oscilloscope=x=0.5:y=5/1080:s=1
        
  • Inspect full last column of video frame.

            oscilloscope=x=1:y=0.5:s=1:t=1
        

overlay

Overlay one video on top of another.

It takes two inputs and has one output. The first input is the "main" video on which the second input is overlaid.

It accepts the following parameters:

A description of the accepted options follows.

x
y
Set the expression for the x and y coordinates of the overlaid video on the main video. Default value is "0" for both expressions. In case the expression is invalid, it is set to a huge value (meaning that the overlay will not be displayed within the output visible area).
eof_action
See framesync.
eval
Set when the expressions for x, and y are evaluated.

It accepts the following values:

init
only evaluate expressions once during the filter initialization or when a command is processed
frame
evaluate expressions for each incoming frame

Default value is frame.

shortest
See framesync.
format
Set the format for the output video.

It accepts the following values:

yuv420
force YUV420 output
yuv422
force YUV422 output
yuv444
force YUV444 output
rgb
force packed RGB output
gbrp
force planar RGB output
auto
automatically pick format

Default value is yuv420.

repeatlast
See framesync.
alpha
Set format of alpha of the overlaid video, it can be straight or premultiplied. Default is straight.

The x, and y expressions can contain the following parameters.

main_w, W
main_h, H
The main input width and height.
overlay_w, w
overlay_h, h
The overlay input width and height.
x
y
The computed values for x and y. They are evaluated for each new frame.
hsub
vsub
horizontal and vertical chroma subsample values of the output format. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.
n
the number of input frame, starting from 0
pos
the position in the file of the input frame, NAN if unknown
t
The timestamp, expressed in seconds. It's NAN if the input timestamp is unknown.

This filter also supports the framesync options.

Note that the n, pos, t variables are available only when evaluation is done per frame, and will evaluate to NAN when eval is set to init.

Be aware that frames are taken from each input video in timestamp order, hence, if their initial timestamps differ, it is a good idea to pass the two inputs through a setpts=PTS-STARTPTS filter to have them begin in the same zero timestamp, as the example for the movie filter does.

You can chain together more overlays but you should test the efficiency of such approach.

Commands

This filter supports the following commands:

x
y
Modify the x and y of the overlay input. The command accepts the same syntax of the corresponding option.

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

Examples

  • Draw the overlay at 10 pixels from the bottom right corner of the main video:

            overlay=main_w-overlay_w-10:main_h-overlay_h-10
        

    Using named options the example above becomes:

            overlay=x=main_w-overlay_w-10:y=main_h-overlay_h-10
        
  • Insert a transparent PNG logo in the bottom left corner of the input, using the ffmpeg tool with the "-filter_complex" option:

            ffmpeg -i input -i logo -filter_complex 'overlay=10:main_h-overlay_h-10' output
        
  • Insert 2 different transparent PNG logos (second logo on bottom right corner) using the ffmpeg tool:

            ffmpeg -i input -i logo1 -i logo2 -filter_complex 'overlay=x=10:y=H-h-10,overlay=x=W-w-10:y=H-h-10' output
        
  • Add a transparent color layer on top of the main video; "WxH" must specify the size of the main input to the overlay filter:

            color=color=red@.3:size=WxH [over]; [in][over] overlay [out]
        
  • Play an original video and a filtered version (here with the deshake filter) side by side using the ffplay tool:

            ffplay input.avi -vf 'split[a][b]; [a]pad=iw*2:ih[src]; [b]deshake[filt]; [src][filt]overlay=w'
        

    The above command is the same as:

            ffplay input.avi -vf 'split[b], pad=iw*2[src], [b]deshake, [src]overlay=w'
        
  • Make a sliding overlay appearing from the left to the right top part of the screen starting since time 2:

            overlay=x='if(gte(t,2), -w+(t-2)*20, NAN)':y=0
        
  • Compose output by putting two input videos side to side:

            ffmpeg -i left.avi -i right.avi -filter_complex "
            nullsrc=size=200x100 [background];
            [0:v] setpts=PTS-STARTPTS, scale=100x100 [left];
            [1:v] setpts=PTS-STARTPTS, scale=100x100 [right];
            [background][left]       overlay=shortest=1       [background+left];
            [background+left][right] overlay=shortest=1:x=100 [left+right]
            "
        
  • Mask 10-20 seconds of a video by applying the delogo filter to a section

            ffmpeg -i test.avi -codec:v:0 wmv2 -ar 11025 -b:v 9000k
            -vf '[in]split[split_main][split_delogo];[split_delogo]trim=start=360:end=371,delogo=0:0:640:480[delogoed];[split_main][delogoed]overlay=eof_action=pass[out]'
            masked.avi
        
  • Chain several overlays in cascade:

            nullsrc=s=200x200 [bg];
            testsrc=s=100x100, split=4 [in0][in1][in2][in3];
            [in0] lutrgb=r=0, [bg]   overlay=0:0     [mid0];
            [in1] lutrgb=g=0, [mid0] overlay=100:0   [mid1];
            [in2] lutrgb=b=0, [mid1] overlay=0:100   [mid2];
            [in3] null,       [mid2] overlay=100:100 [out0]
        

owdenoise

Apply Overcomplete Wavelet denoiser.

The filter accepts the following options:

depth
Set depth.

Larger depth values will denoise lower frequency components more, but slow down filtering.

Must be an int in the range 8-16, default is 8.

luma_strength, ls
Set luma strength.

Must be a double value in the range 0-1000, default is 1.0.

chroma_strength, cs
Set chroma strength.

Must be a double value in the range 0-1000, default is 1.0.

pad

Add paddings to the input image, and place the original input at the provided x, y coordinates.

It accepts the following parameters:

width, w
height, h
Specify an expression for the size of the output image with the paddings added. If the value for width or height is 0, the corresponding input size is used for the output.

The width expression can reference the value set by the height expression, and vice versa.

The default value of width and height is 0.

x
y
Specify the offsets to place the input image at within the padded area, with respect to the top/left border of the output image.

The x expression can reference the value set by the y expression, and vice versa.

The default value of x and y is 0.

If x or y evaluate to a negative number, they'll be changed so the input image is centered on the padded area.

color
Specify the color of the padded area. For the syntax of this option, check the "Color" section in the ffmpeg-utils manual.

The default value of color is "black".

eval
Specify when to evaluate width, height, x and y expression.

It accepts the following values:

init
Only evaluate expressions once during the filter initialization or when a command is processed.
frame
Evaluate expressions for each incoming frame.

Default value is init.

aspect
Pad to aspect instead to a resolution.

The value for the width, height, x, and y options are expressions containing the following constants:

in_w
in_h
The input video width and height.
iw
ih
These are the same as in_w and in_h.
out_w
out_h
The output width and height (the size of the padded area), as specified by the width and height expressions.
ow
oh
These are the same as out_w and out_h.
x
y
The x and y offsets as specified by the x and y expressions, or NAN if not yet specified.
a
same as iw / ih
sar
input sample aspect ratio
dar
input display aspect ratio, it is the same as (iw / ih) * sar
hsub
vsub
The horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.

Examples

  • Add paddings with the color "violet" to the input video. The output video size is 640x480, and the top-left corner of the input video is placed at column 0, row 40

            pad=640:480:0:40:violet
        

    The example above is equivalent to the following command:

            pad=width=640:height=480:x=0:y=40:color=violet
        
  • Pad the input to get an output with dimensions increased by 3/2, and put the input video at the center of the padded area:

            pad="3/2*iw:3/2*ih:(ow-iw)/2:(oh-ih)/2"
        
  • Pad the input to get a squared output with size equal to the maximum value between the input width and height, and put the input video at the center of the padded area:

            pad="max(iw\,ih):ow:(ow-iw)/2:(oh-ih)/2"
        
  • Pad the input to get a final w/h ratio of 16:9:

            pad="ih*16/9:ih:(ow-iw)/2:(oh-ih)/2"
        
  • In case of anamorphic video, in order to set the output display aspect correctly, it is necessary to use sar in the expression, according to the relation:

            (ih * X / ih) * sar = output_dar
            X = output_dar / sar
        

    Thus the previous example needs to be modified to:

            pad="ih*16/9/sar:ih:(ow-iw)/2:(oh-ih)/2"
        
  • Double the output size and put the input video in the bottom-right corner of the output padded area:

            pad="2*iw:2*ih:ow-iw:oh-ih"
        

palettegen

Generate one palette for a whole video stream.

It accepts the following options:

max_colors
Set the maximum number of colors to quantize in the palette. Note: the palette will still contain 256 colors; the unused palette entries will be black.
reserve_transparent
Create a palette of 255 colors maximum and reserve the last one for transparency. Reserving the transparency color is useful for GIF optimization. If not set, the maximum of colors in the palette will be 256. You probably want to disable this option for a standalone image. Set by default.
transparency_color
Set the color that will be used as background for transparency.
stats_mode
Set statistics mode.

It accepts the following values:

full
Compute full frame histograms.
diff
Compute histograms only for the part that differs from previous frame. This might be relevant to give more importance to the moving part of your input if the background is static.
single
Compute new histogram for each frame.

Default value is full.

The filter also exports the frame metadata "lavfi.color_quant_ratio" ("nb_color_in / nb_color_out") which you can use to evaluate the degree of color quantization of the palette. This information is also visible at info logging level.

Examples

Generate a representative palette of a given video using ffmpeg:

        ffmpeg -i input.mkv -vf palettegen palette.png
    

paletteuse

Use a palette to downsample an input video stream.

The filter takes two inputs: one video stream and a palette. The palette must be a 256 pixels image.

It accepts the following options:

dither
Select dithering mode. Available algorithms are:
bayer
Ordered 8x8 bayer dithering (deterministic)
heckbert
Dithering as defined by Paul Heckbert in 1982 (simple error diffusion). Note: this dithering is sometimes considered "wrong" and is included as a reference.
floyd_steinberg
Floyd and Steingberg dithering (error diffusion)
sierra2
Frankie Sierra dithering v2 (error diffusion)
sierra2_4a
Frankie Sierra dithering v2 "Lite" (error diffusion)

Default is sierra2_4a.

bayer_scale
When bayer dithering is selected, this option defines the scale of the pattern (how much the crosshatch pattern is visible). A low value means more visible pattern for less banding, and higher value means less visible pattern at the cost of more banding.

The option must be an integer value in the range [0,5]. Default is 2.

diff_mode
If set, define the zone to process
rectangle
Only the changing rectangle will be reprocessed. This is similar to GIF cropping/offsetting compression mechanism. This option can be useful for speed if only a part of the image is changing, and has use cases such as limiting the scope of the error diffusal dither to the rectangle that bounds the moving scene (it leads to more deterministic output if the scene doesn't change much, and as a result less moving noise and better GIF compression).

Default is none.

new
Take new palette for each output frame.
alpha_threshold
Sets the alpha threshold for transparency. Alpha values above this threshold will be treated as completely opaque, and values below this threshold will be treated as completely transparent.

The option must be an integer value in the range [0,255]. Default is 128.

Examples

Use a palette (generated for example with palettegen) to encode a GIF using ffmpeg:

        ffmpeg -i input.mkv -i palette.png -lavfi paletteuse output.gif
    

perspective

Correct perspective of video not recorded perpendicular to the screen.

A description of the accepted parameters follows.

x0
y0
x1
y1
x2
y2
x3
y3
Set coordinates expression for top left, top right, bottom left and bottom right corners. Default values are "0:0:W:0:0:H:W:H" with which perspective will remain unchanged. If the "sense" option is set to "source", then the specified points will be sent to the corners of the destination. If the "sense" option is set to "destination", then the corners of the source will be sent to the specified coordinates.

The expressions can use the following variables:

W
H
the width and height of video frame.
in
Input frame count.
on
Output frame count.
interpolation
Set interpolation for perspective correction.

It accepts the following values:

linear
cubic

Default value is linear.

sense
Set interpretation of coordinate options.

It accepts the following values:

0, source
Send point in the source specified by the given coordinates to the corners of the destination.
1, destination
Send the corners of the source to the point in the destination specified by the given coordinates.

Default value is source.

eval
Set when the expressions for coordinates x0,y0,...x3,y3 are evaluated.

It accepts the following values:

init
only evaluate expressions once during the filter initialization or when a command is processed
frame
evaluate expressions for each incoming frame

Default value is init.

phase

Delay interlaced video by one field time so that the field order changes.

The intended use is to fix PAL movies that have been captured with the opposite field order to the film-to-video transfer.

A description of the accepted parameters follows.

mode
Set phase mode.

It accepts the following values:

t
Capture field order top-first, transfer bottom-first. Filter will delay the bottom field.
b
Capture field order bottom-first, transfer top-first. Filter will delay the top field.
p
Capture and transfer with the same field order. This mode only exists for the documentation of the other options to refer to, but if you actually select it, the filter will faithfully do nothing.
a
Capture field order determined automatically by field flags, transfer opposite. Filter selects among t and b modes on a frame by frame basis using field flags. If no field information is available, then this works just like u.
u
Capture unknown or varying, transfer opposite. Filter selects among t and b on a frame by frame basis by analyzing the images and selecting the alternative that produces best match between the fields.
T
Capture top-first, transfer unknown or varying. Filter selects among t and p using image analysis.
B
Capture bottom-first, transfer unknown or varying. Filter selects among b and p using image analysis.
A
Capture determined by field flags, transfer unknown or varying. Filter selects among t, b and p using field flags and image analysis. If no field information is available, then this works just like U. This is the default mode.
U
Both capture and transfer unknown or varying. Filter selects among t, b and p using image analysis only.

pixdesctest

Pixel format descriptor test filter, mainly useful for internal testing. The output video should be equal to the input video.

For example:

        format=monow, pixdesctest

can be used to test the monowhite pixel format descriptor definition.

pixscope

Display sample values of color channels. Mainly useful for checking color and levels. Minimum supported resolution is 640x480.

The filters accept the following options:

x
Set scope X position, relative offset on X axis.
y
Set scope Y position, relative offset on Y axis.
w
Set scope width.
h
Set scope height.
o
Set window opacity. This window also holds statistics about pixel area.
wx
Set window X position, relative offset on X axis.
wy
Set window Y position, relative offset on Y axis.

pp

Enable the specified chain of postprocessing subfilters using libpostproc. This library should be automatically selected with a GPL build ("--enable-gpl"). Subfilters must be separated by '/' and can be disabled by prepending a '-'. Each subfilter and some options have a short and a long name that can be used interchangeably, i.e. dr/dering are the same.

The filters accept the following options:

subfilters
Set postprocessing subfilters string.

All subfilters share common options to determine their scope:

a/autoq
Honor the quality commands for this subfilter.
c/chrom
Do chrominance filtering, too (default).
y/nochrom
Do luminance filtering only (no chrominance).
n/noluma
Do chrominance filtering only (no luminance).

These options can be appended after the subfilter name, separated by a '|'.

Available subfilters are:

hb/hdeblock[|difference[|flatness]]
Horizontal deblocking filter
difference
Difference factor where higher values mean more deblocking (default: 32).
flatness
Flatness threshold where lower values mean more deblocking (default: 39).
vb/vdeblock[|difference[|flatness]]
Vertical deblocking filter
difference
Difference factor where higher values mean more deblocking (default: 32).
flatness
Flatness threshold where lower values mean more deblocking (default: 39).
ha/hadeblock[|difference[|flatness]]
Accurate horizontal deblocking filter
difference
Difference factor where higher values mean more deblocking (default: 32).
flatness
Flatness threshold where lower values mean more deblocking (default: 39).
va/vadeblock[|difference[|flatness]]
Accurate vertical deblocking filter
difference
Difference factor where higher values mean more deblocking (default: 32).
flatness
Flatness threshold where lower values mean more deblocking (default: 39).

The horizontal and vertical deblocking filters share the difference and flatness values so you cannot set different horizontal and vertical thresholds.

h1/x1hdeblock
Experimental horizontal deblocking filter
v1/x1vdeblock
Experimental vertical deblocking filter
dr/dering
Deringing filter
tn/tmpnoise[|threshold1[|threshold2[|threshold3]]], temporal noise reducer
threshold1
larger -> stronger filtering
threshold2
larger -> stronger filtering
threshold3
larger -> stronger filtering
al/autolevels[:f/fullyrange], automatic brightness / contrast correction
f/fullyrange
Stretch luminance to "0-255".
lb/linblenddeint
Linear blend deinterlacing filter that deinterlaces the given block by filtering all lines with a "(1 2 1)" filter.
li/linipoldeint
Linear interpolating deinterlacing filter that deinterlaces the given block by linearly interpolating every second line.
ci/cubicipoldeint
Cubic interpolating deinterlacing filter deinterlaces the given block by cubically interpolating every second line.
md/mediandeint
Median deinterlacing filter that deinterlaces the given block by applying a median filter to every second line.
fd/ffmpegdeint
FFmpeg deinterlacing filter that deinterlaces the given block by filtering every second line with a "(-1 4 2 4 -1)" filter.
l5/lowpass5
Vertically applied FIR lowpass deinterlacing filter that deinterlaces the given block by filtering all lines with a "(-1 2 6 2 -1)" filter.
fq/forceQuant[|quantizer]
Overrides the quantizer table from the input with the constant quantizer you specify.
quantizer
Quantizer to use
de/default
Default pp filter combination ("hb|a,vb|a,dr|a")
fa/fast
Fast pp filter combination ("h1|a,v1|a,dr|a")
ac
High quality pp filter combination ("ha|a|128|7,va|a,dr|a")

Examples

  • Apply horizontal and vertical deblocking, deringing and automatic brightness/contrast:

            pp=hb/vb/dr/al
        
  • Apply default filters without brightness/contrast correction:

            pp=de/-al
        
  • Apply default filters and temporal denoiser:

            pp=default/tmpnoise|1|2|3
        
  • Apply deblocking on luminance only, and switch vertical deblocking on or off automatically depending on available CPU time:

            pp=hb|y/vb|a
        

pp7

Apply Postprocessing filter 7. It is variant of the spp filter, similar to spp = 6 with 7 point DCT, where only the center sample is used after IDCT.

The filter accepts the following options:

qp
Force a constant quantization parameter. It accepts an integer in range 0 to 63. If not set, the filter will use the QP from the video stream (if available).
mode
Set thresholding mode. Available modes are:
hard
Set hard thresholding.
soft
Set soft thresholding (better de-ringing effect, but likely blurrier).
medium
Set medium thresholding (good results, default).

premultiply

Apply alpha premultiply effect to input video stream using first plane of second stream as alpha.

Both streams must have same dimensions and same pixel format.

The filter accepts the following option:

planes
Set which planes will be processed, unprocessed planes will be copied. By default value 0xf, all planes will be processed.
inplace
Do not require 2nd input for processing, instead use alpha plane from input stream.

prewitt

Apply prewitt operator to input video stream.

The filter accepts the following option:

planes
Set which planes will be processed, unprocessed planes will be copied. By default value 0xf, all planes will be processed.
scale
Set value which will be multiplied with filtered result.
delta
Set value which will be added to filtered result.

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 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));
            }
        

pseudocolor

Alter frame colors in video with pseudocolors.

This filter accept the following options:

c0
set pixel first component expression
c1
set pixel second component expression
c2
set pixel third component expression
c3
set pixel fourth component expression, corresponds to the alpha component
i
set component to use as base for altering colors

Each of them specifies the expression to use for computing the lookup table for the corresponding pixel component values.

The expressions can contain the following constants and functions:

w
h
The input width and height.
val
The input value for the pixel component.
ymin, umin, vmin, amin
The minimum allowed component value.
ymax, umax, vmax, amax
The maximum allowed component value.

All expressions default to "val".

Examples

Change too high luma values to gradient:

        pseudocolor="'if(between(val,ymax,amax),lerp(ymin,ymax,(val-ymax)/(amax-ymax)),-1):if(between(val,ymax,amax),lerp(umax,umin,(val-ymax)/(amax-ymax)),-1):if(between(val,ymax,amax),lerp(vmin,vmax,(val-ymax)/(amax-ymax)),-1):-1'"
    

psnr

Obtain the average, maximum and minimum PSNR (Peak Signal to Noise Ratio) between two input videos.

This filter takes in input two input videos, the first input is considered the "main" source and is passed unchanged to the output. The second input is used as a "reference" video for computing the PSNR.

Both video inputs must have the same resolution and pixel format for this filter to work correctly. Also it assumes that both inputs have the same number of frames, which are compared one by one.

The obtained average PSNR is printed through the logging system.

The filter stores the accumulated MSE (mean squared error) of each frame, and at the end of the processing it is averaged across all frames equally, and the following formula is applied to obtain the PSNR:

        PSNR = 10*log10(MAX^2/MSE)

Where MAX is the average of the maximum values of each component of the image.

The description of the accepted parameters follows.

stats_file, f
If specified the filter will use the named file to save the PSNR of each individual frame. When filename equals "-" the data is sent to standard output.
stats_version
Specifies which version of the stats file format to use. Details of each format are written below. Default value is 1.
stats_add_max
Determines whether the max value is output to the stats log. Default value is 0. Requires stats_version >= 2. If this is set and stats_version < 2, the filter will return an error.

This filter also supports the framesync options.

The file printed if stats_file is selected, contains a sequence of key/value pairs of the form key:value for each compared couple of frames.

If a stats_version greater than 1 is specified, a header line precedes the list of per-frame-pair stats, with key value pairs following the frame format with the following parameters:

psnr_log_version
The version of the log file format. Will match stats_version.
fields
A comma separated list of the per-frame-pair parameters included in the log.

A description of each shown per-frame-pair parameter follows:

n
sequential number of the input frame, starting from 1
mse_avg
Mean Square Error pixel-by-pixel average difference of the compared frames, averaged over all the image components.
mse_y, mse_u, mse_v, mse_r, mse_g, mse_b, mse_a
Mean Square Error pixel-by-pixel average difference of the compared frames for the component specified by the suffix.
psnr_y, psnr_u, psnr_v, psnr_r, psnr_g, psnr_b, psnr_a
Peak Signal to Noise ratio of the compared frames for the component specified by the suffix.
max_avg, max_y, max_u, max_v
Maximum allowed value for each channel, and average over all channels.

For example:

        movie=ref_movie.mpg, setpts=PTS-STARTPTS [main];
        [main][ref] psnr="stats_file=stats.log" [out]

On this example the input file being processed is compared with the reference file ref_movie.mpg. The PSNR of each individual frame is stored in stats.log.

pullup

Pulldown reversal (inverse telecine) filter, capable of handling mixed hard-telecine, 24000/1001 fps progressive, and 30000/1001 fps progressive content.

The pullup filter is designed to take advantage of future context in making its decisions. This filter is stateless in the sense that it does not lock onto a pattern to follow, but it instead looks forward to the following fields in order to identify matches and rebuild progressive frames.

To produce content with an even framerate, insert the fps filter after pullup, use "fps=24000/1001" if the input frame rate is 29.97fps, "fps=24" for 30fps and the (rare) telecined 25fps input.

The filter accepts the following options:

jl
jr
jt
jb
These options set the amount of "junk" to ignore at the left, right, top, and bottom of the image, respectively. Left and right are in units of 8 pixels, while top and bottom are in units of 2 lines. The default is 8 pixels on each side.
sb
Set the strict breaks. Setting this option to 1 will reduce the chances of filter generating an occasional mismatched frame, but it may also cause an excessive number of frames to be dropped during high motion sequences. Conversely, setting it to -1 will make filter match fields more easily. This may help processing of video where there is slight blurring between the fields, but may also cause there to be interlaced frames in the output. Default value is 0.
mp
Set the metric plane to use. It accepts the following values:
l
Use luma plane.
u
Use chroma blue plane.
v
Use chroma red plane.

This option may be set to use chroma plane instead of the default luma plane for doing filter's computations. This may improve accuracy on very clean source material, but more likely will decrease accuracy, especially if there is chroma noise (rainbow effect) or any grayscale video. The main purpose of setting mp to a chroma plane is to reduce CPU load and make pullup usable in realtime on slow machines.

For best results (without duplicated frames in the output file) it is necessary to change the output frame rate. For example, to inverse telecine NTSC input:

        ffmpeg -i input -vf pullup -r 24000/1001 ...

qp

Change video quantization parameters (QP).

The filter accepts the following option:

qp
Set expression for quantization parameter.

The expression is evaluated through the eval API and can contain, among others, the following constants:

known
1 if index is not 129, 0 otherwise.
qp
Sequential index starting from -129 to 128.

Examples

Some equation like:

        qp=2+2*sin(PI*qp)
    

random

Flush video frames from internal cache of frames into a random order. No frame is discarded. Inspired by frei0r nervous filter.
frames
Set size in number of frames of internal cache, in range from 2 to 512. Default is 30.
seed
Set seed for random number generator, must be an integer included between 0 and "UINT32_MAX". If not specified, or if explicitly set to less than 0, the filter will try to use a good random seed on a best effort basis.

readeia608

Read closed captioning (EIA-608) information from the top lines of a video frame.

This filter adds frame metadata for "lavfi.readeia608.X.cc" and "lavfi.readeia608.X.line", where "X" is the number of the identified line with EIA-608 data (starting from 0). A description of each metadata value follows:

lavfi.readeia608.X.cc
The two bytes stored as EIA-608 data (printed in hexadecimal).
lavfi.readeia608.X.line
The number of the line on which the EIA-608 data was identified and read.

This filter accepts the following options:

scan_min
Set the line to start scanning for EIA-608 data. Default is 0.
scan_max
Set the line to end scanning for EIA-608 data. Default is 29.
mac
Set minimal acceptable amplitude change for sync codes detection. Default is 0.2. Allowed range is "[0.001 - 1]".
spw
Set the ratio of width reserved for sync code detection. Default is 0.27. Allowed range is "[0.01 - 0.7]".
mhd
Set the max peaks height difference for sync code detection. Default is 0.1. Allowed range is "[0.0 - 0.5]".
mpd
Set max peaks period difference for sync code detection. Default is 0.1. Allowed range is "[0.0 - 0.5]".
msd
Set the first two max start code bits differences. Default is 0.02. Allowed range is "[0.0 - 0.5]".
bhd
Set the minimum ratio of bits height compared to 3rd start code bit. Default is 0.75. Allowed range is "[0.01 - 1]".
th_w
Set the white color threshold. Default is 0.35. Allowed range is "[0.1 - 1]".
th_b
Set the black color threshold. Default is 0.15. Allowed range is "[0.0 - 0.5]".
chp
Enable checking the parity bit. In the event of a parity error, the filter will output 0x00 for that character. Default is false.

Examples

Output a csv with presentation time and the first two lines of identified EIA-608 captioning data.

        ffprobe -f lavfi -i movie=captioned_video.mov,readeia608 -show_entries frame=pkt_pts_time:frame_tags=lavfi.readeia608.0.cc,lavfi.readeia608.1.cc -of csv
    

readvitc

Read vertical interval timecode (VITC) information from the top lines of a video frame.

The filter adds frame metadata key "lavfi.readvitc.tc_str" with the timecode value, if a valid timecode has been detected. Further metadata key "lavfi.readvitc.found" is set to 0/1 depending on whether timecode data has been found or not.

This filter accepts the following options:

scan_max
Set the maximum number of lines to scan for VITC data. If the value is set to "-1" the full video frame is scanned. Default is 45.
thr_b
Set the luma threshold for black. Accepts float numbers in the range [0.0,1.0], default value is 0.2. The value must be equal or less than "thr_w".
thr_w
Set the luma threshold for white. Accepts float numbers in the range [0.0,1.0], default value is 0.6. The value must be equal or greater than "thr_b".

Examples

Detect and draw VITC data onto the video frame; if no valid VITC is detected, draw "--:--:--:--" as a placeholder:

        ffmpeg -i input.avi -filter:v 'readvitc,drawtext=fontfile=FreeMono.ttf:text=%{metadata\\:lavfi.readvitc.tc_str\\:--\\\\\\:--\\\\\\:--\\\\\\:--}:x=(w-tw)/2:y=400-ascent'
    

remap

Remap pixels using 2nd: Xmap and 3rd: Ymap input video stream.

Destination pixel at position (X, Y) will be picked from source (x, y) position where x = Xmap(X, Y) and y = Ymap(X, Y). If mapping values are out of range, zero value for pixel will be used for destination pixel.

Xmap and Ymap input video streams must be of same dimensions. Output video stream will have Xmap/Ymap video stream dimensions. Xmap and Ymap input video streams are 16bit depth, single channel.

removegrain

The removegrain filter is a spatial denoiser for progressive video.
m0
Set mode for the first plane.
m1
Set mode for the second plane.
m2
Set mode for the third plane.
m3
Set mode for the fourth plane.

Range of mode is from 0 to 24. Description of each mode follows:

0
Leave input plane unchanged. Default.
1
Clips the pixel with the minimum and maximum of the 8 neighbour pixels.
2
Clips the pixel with the second minimum and maximum of the 8 neighbour pixels.
3
Clips the pixel with the third minimum and maximum of the 8 neighbour pixels.
4
Clips the pixel with the fourth minimum and maximum of the 8 neighbour pixels. This is equivalent to a median filter.
5
Line-sensitive clipping giving the minimal change.
6
Line-sensitive clipping, intermediate.
7
Line-sensitive clipping, intermediate.
8
Line-sensitive clipping, intermediate.
9
Line-sensitive clipping on a line where the neighbours pixels are the closest.
10
Replaces the target pixel with the closest neighbour.
11
[1 2 1] horizontal and vertical kernel blur.
12
Same as mode 11.
13
Bob mode, interpolates top field from the line where the neighbours pixels are the closest.
14
Bob mode, interpolates bottom field from the line where the neighbours pixels are the closest.
15
Bob mode, interpolates top field. Same as 13 but with a more complicated interpolation formula.
16
Bob mode, interpolates bottom field. Same as 14 but with a more complicated interpolation formula.
17
Clips the pixel with the minimum and maximum of respectively the maximum and minimum of each pair of opposite neighbour pixels.
18
Line-sensitive clipping using opposite neighbours whose greatest distance from the current pixel is minimal.
19
Replaces the pixel with the average of its 8 neighbours.
20
Averages the 9 pixels ([1 1 1] horizontal and vertical blur).
21
Clips pixels using the averages of opposite neighbour.
22
Same as mode 21 but simpler and faster.
23
Small edge and halo removal, but reputed useless.
24
Similar as 23.
Suppress a TV station logo, using an image file to determine which pixels comprise the logo. It works by filling in the pixels that comprise the logo with neighboring pixels.

The filter accepts the following options:

filename, f
Set the filter bitmap file, which can be any image format supported by libavformat. The width and height of the image file must match those of the video stream being processed.

Pixels in the provided bitmap image with a value of zero are not considered part of the logo, non-zero pixels are considered part of the logo. If you use white (255) for the logo and black (0) for the rest, you will be safe. For making the filter bitmap, it is recommended to take a screen capture of a black frame with the logo visible, and then using a threshold filter followed by the erode filter once or twice.

If needed, little splotches can be fixed manually. Remember that if logo pixels are not covered, the filter quality will be much reduced. Marking too many pixels as part of the logo does not hurt as much, but it will increase the amount of blurring needed to cover over the image and will destroy more information than necessary, and extra pixels will slow things down on a large logo.

repeatfields

This filter uses the repeat_field flag from the Video ES headers and hard repeats fields based on its value.

reverse

Reverse a video clip.

Warning: This filter requires memory to buffer the entire clip, so trimming is suggested.

Examples

Take the first 5 seconds of a clip, and reverse it.

        trim=end=5,reverse
    

roberts

Apply roberts cross operator to input video stream.

The filter accepts the following option:

planes
Set which planes will be processed, unprocessed planes will be copied. By default value 0xf, all planes will be processed.
scale
Set value which will be multiplied with filtered result.
delta
Set value which will be added to filtered result.

rotate

Rotate video by an arbitrary angle expressed in radians.

The filter accepts the following options:

A description of the optional parameters follows.

angle, a
Set an expression for the angle by which to rotate the input video clockwise, expressed as a number of radians. A negative value will result in a counter-clockwise rotation. By default it is set to "0".

This expression is evaluated for each frame.

out_w, ow
Set the output width expression, default value is "iw". This expression is evaluated just once during configuration.
out_h, oh
Set the output height expression, default value is "ih". This expression is evaluated just once during configuration.
bilinear
Enable bilinear interpolation if set to 1, a value of 0 disables it. Default value is 1.
fillcolor, c
Set the color used to fill the output area not covered by the rotated image. For the general syntax of this option, check the "Color" section in the ffmpeg-utils manual. If the special value "none" is selected then no background is printed (useful for example if the background is never shown).

Default value is "black".

The expressions for the angle and the output size can contain the following constants and functions:

n
sequential number of the input frame, starting from 0. It is always NAN before the first frame is filtered.
t
time in seconds of the input frame, it is set to 0 when the filter is configured. It is always NAN before the first frame is filtered.
hsub
vsub
horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.
in_w, iw
in_h, ih
the input video width and height
out_w, ow
out_h, oh
the output width and height, that is the size of the padded area as specified by the width and height expressions
rotw(a)
roth(a)
the minimal width/height required for completely containing the input video rotated by a radians.

These are only available when computing the out_w and out_h expressions.

Examples

  • Rotate the input by PI/6 radians clockwise:

            rotate=PI/6
        
  • Rotate the input by PI/6 radians counter-clockwise:

            rotate=-PI/6
        
  • Rotate the input by 45 degrees clockwise:

            rotate=45*PI/180
        
  • Apply a constant rotation with period T, starting from an angle of PI/3:

            rotate=PI/3+2*PI*t/T
        
  • Make the input video rotation oscillating with a period of T seconds and an amplitude of A radians:

            rotate=A*sin(2*PI/T*t)
        
  • Rotate the video, output size is chosen so that the whole rotating input video is always completely contained in the output:

            rotate='2*PI*t:ow=hypot(iw,ih):oh=ow'
        
  • Rotate the video, reduce the output size so that no background is ever shown:

            rotate=2*PI*t:ow='min(iw,ih)/sqrt(2)':oh=ow:c=none
        

Commands

The filter supports the following commands:

a, angle
Set the angle expression. The command accepts the same syntax of the corresponding option.

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

sab

Apply Shape Adaptive Blur.

The filter accepts the following options:

luma_radius, lr
Set luma blur filter strength, must be a value in range 0.1-4.0, default value is 1.0. A greater value will result in a more blurred image, and in slower processing.
luma_pre_filter_radius, lpfr
Set luma pre-filter radius, must be a value in the 0.1-2.0 range, default value is 1.0.
luma_strength, ls
Set luma maximum difference between pixels to still be considered, must be a value in the 0.1-100.0 range, default value is 1.0.
chroma_radius, cr
Set chroma blur filter strength, must be a value in range -0.9-4.0. A greater value will result in a more blurred image, and in slower processing.
chroma_pre_filter_radius, cpfr
Set chroma pre-filter radius, must be a value in the -0.9-2.0 range.
chroma_strength, cs
Set chroma maximum difference between pixels to still be considered, must be a value in the -0.9-100.0 range.

Each chroma option value, if not explicitly specified, is set to the corresponding luma option value.

scale

Scale (resize) the input video, using the libswscale library.

The scale filter forces the output display aspect ratio to be the same of the input, by changing the output sample aspect ratio.

If the input image format is different from the format requested by the next filter, the scale filter will convert the input to the requested format.

Options

The filter accepts the following options, or any of the options supported by the libswscale scaler.

See the ffmpeg-scaler manual for the complete list of scaler options.

width, w
height, h
Set the output video dimension expression. Default value is the input dimension.

If the width or w value is 0, the input width is used for the output. If the height or h value is 0, the input height is used for the output.

If one and only one of the values is -n with n >= 1, the scale filter will use a value that maintains the aspect ratio of the input image, calculated from the other specified dimension. After that it will, however, make sure that the calculated dimension is divisible by n and adjust the value if necessary.

If both values are -n with n >= 1, the behavior will be identical to both values being set to 0 as previously detailed.

See below for the list of accepted constants for use in the dimension expression.

eval
Specify when to evaluate width and height expression. It accepts the following values:
init
Only evaluate expressions once during the filter initialization or when a command is processed.
frame
Evaluate expressions for each incoming frame.

Default value is init.

interl
Set the interlacing mode. It accepts the following values:
1
Force interlaced aware scaling.
0
Do not apply interlaced scaling.
-1
Select interlaced aware scaling depending on whether the source frames are flagged as interlaced or not.

Default value is 0.

flags
Set libswscale scaling flags. See the ffmpeg-scaler manual for the complete list of values. If not explicitly specified the filter applies the default flags.
param0, param1
Set libswscale input parameters for scaling algorithms that need them. See the ffmpeg-scaler manual for the complete documentation. If not explicitly specified the filter applies empty parameters.
size, s
Set the video size. For the syntax of this option, check the "Video size" section in the ffmpeg-utils manual.
in_color_matrix
out_color_matrix
Set in/output YCbCr color space type.

This allows the autodetected value to be overridden as well as allows forcing a specific value used for the output and encoder.

If not specified, the color space type depends on the pixel format.

Possible values:

auto
Choose automatically.
bt709
Format conforming to International Telecommunication Union (ITU) Recommendation BT.709.
fcc
Set color space conforming to the United States Federal Communications Commission (FCC) Code of Federal Regulations (CFR) Title 47 (2003) 73.682 (a).
bt601
Set color space conforming to:
  • ITU Radiocommunication Sector (ITU-R) Recommendation BT.601
  • ITU-R Rec. BT.470-6 (1998) Systems B, B1, and G
  • Society of Motion Picture and Television Engineers (SMPTE) ST 170:2004
smpte240m
Set color space conforming to SMPTE ST 240:1999.
in_range
out_range
Set in/output YCbCr sample range.

This allows the autodetected value to be overridden as well as allows forcing a specific value used for the output and encoder. If not specified, the range depends on the pixel format. Possible values:

auto/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).
force_original_aspect_ratio
Enable decreasing or increasing output video width or height if necessary to keep the original aspect ratio. Possible values:
disable
Scale the video as specified and disable this feature.
decrease
The output video dimensions will automatically be decreased if needed.
increase
The output video dimensions will automatically be increased if needed.

One useful instance of this option is that when you know a specific device's maximum allowed resolution, you can use this to limit the output video to that, while retaining the aspect ratio. For example, device A allows 1280x720 playback, and your video is 1920x800. Using this option (set it to decrease) and specifying 1280x720 to the command line makes the output 1280x533.

Please note that this is a different thing than specifying -1 for w or h, you still need to specify the output resolution for this option to work.

The values of the w and h options are expressions containing the following constants:

in_w
in_h
The input width and height
iw
ih
These are the same as in_w and in_h.
out_w
out_h
The output (scaled) width and height
ow
oh
These are the same as out_w and out_h
a
The same as iw / ih
sar
input sample aspect ratio
dar
The input display aspect ratio. Calculated from "(iw / ih) * sar".
hsub
vsub
horizontal and vertical input chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.
ohsub
ovsub
horizontal and vertical output chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.

Examples

  • Scale the input video to a size of 200x100

            scale=w=200:h=100
        

    This is equivalent to:

            scale=200:100
        

    or:

            scale=200x100
        
  • Specify a size abbreviation for the output size:

            scale=qcif
        

    which can also be written as:

            scale=size=qcif
        
  • Scale the input to 2x:

            scale=w=2*iw:h=2*ih
        
  • The above is the same as:

            scale=2*in_w:2*in_h
        
  • Scale the input to 2x with forced interlaced scaling:

            scale=2*iw:2*ih:interl=1
        
  • Scale the input to half size:

            scale=w=iw/2:h=ih/2
        
  • Increase the width, and set the height to the same size:

            scale=3/2*iw:ow
        
  • Seek Greek harmony:

            scale=iw:1/PHI*iw
            scale=ih*PHI:ih
        
  • Increase the height, and set the width to 3/2 of the height:

            scale=w=3/2*oh:h=3/5*ih
        
  • Increase the size, making the size a multiple of the chroma subsample values:

            scale="trunc(3/2*iw/hsub)*hsub:trunc(3/2*ih/vsub)*vsub"
        
  • Increase the width to a maximum of 500 pixels, keeping the same aspect ratio as the input:

            scale=w='min(500\, iw*3/2):h=-1'
        
  • Make pixels square by combining scale and setsar:

            scale='trunc(ih*dar):ih',setsar=1/1
        
  • Make pixels square by combining scale and setsar, making sure the resulting resolution is even (required by some codecs):

            scale='trunc(ih*dar/2)*2:trunc(ih/2)*2',setsar=1/1
        

Commands

This filter supports the following commands:

width, w
height, h
Set the output video dimension expression. The command accepts the same syntax of the corresponding option.

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

scale_npp

Use the NVIDIA Performance Primitives (libnpp) to perform scaling and/or pixel format conversion on CUDA video frames. Setting the output width and height works in the same way as for the scale filter.

The following additional options are accepted:

format
The pixel format of the output CUDA frames. If set to the string "same" (the default), the input format will be kept. Note that automatic format negotiation and conversion is not yet supported for hardware frames
interp_algo
The interpolation algorithm used for resizing. One of the following:
nn
Nearest neighbour.
linear
cubic
cubic2p_bspline
2-parameter cubic (B=1, C=0)
cubic2p_catmullrom
2-parameter cubic (B=0, C=1/2)
cubic2p_b05c03
2-parameter cubic (B=1/2, C=3/10)
super
Supersampling
lanczos

scale2ref

Scale (resize) the input video, based on a reference video.

See the scale filter for available options, scale2ref supports the same but uses the reference video instead of the main input as basis. scale2ref also supports the following additional constants for the w and h options:

main_w
main_h
The main input video's width and height
main_a
The same as main_w / main_h
main_sar
The main input video's sample aspect ratio
main_dar, mdar
The main input video's display aspect ratio. Calculated from "(main_w / main_h) * main_sar".
main_hsub
main_vsub
The main input video's horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.

Examples

Scale a subtitle stream (b) to match the main video (a) in size before overlaying

        'scale2ref[b][a];[a][b]overlay'
    

selectivecolor

Adjust cyan, magenta, yellow and black (CMYK) to certain ranges of colors (such as "reds", "yellows", "greens", "cyans", ...). The adjustment range is defined by the "purity" of the color (that is, how saturated it already is).

This filter is similar to the Adobe Photoshop Selective Color tool.

The filter accepts the following options:

correction_method
Select color correction method.

Available values are:

absolute
Specified adjustments are applied "as-is" (added/subtracted to original pixel component value).
relative
Specified adjustments are relative to the original component value.

Default is "absolute".

reds
Adjustments for red pixels (pixels where the red component is the maximum)
yellows
Adjustments for yellow pixels (pixels where the blue component is the minimum)
greens
Adjustments for green pixels (pixels where the green component is the maximum)
cyans
Adjustments for cyan pixels (pixels where the red component is the minimum)
blues
Adjustments for blue pixels (pixels where the blue component is the maximum)
magentas
Adjustments for magenta pixels (pixels where the green component is the minimum)
whites
Adjustments for white pixels (pixels where all components are greater than 128)
neutrals
Adjustments for all pixels except pure black and pure white
blacks
Adjustments for black pixels (pixels where all components are lesser than 128)
psfile
Specify a Photoshop selective color file (".asv") to import the settings from.

All the adjustment settings (reds, yellows, ...) accept up to 4 space separated floating point adjustment values in the [-1,1] range, respectively to adjust the amount of cyan, magenta, yellow and black for the pixels of its range.

Examples

  • Increase cyan by 50% and reduce yellow by 33% in every green areas, and increase magenta by 27% in blue areas:

            selectivecolor=greens=.5 0 -.33 0:blues=0 .27
        
  • Use a Photoshop selective color preset:

            selectivecolor=psfile=MySelectiveColorPresets/Misty.asv
        

separatefields

The "separatefields" takes a frame-based video input and splits each frame into its components fields, producing a new half height clip with twice the frame rate and twice the frame count.

This filter use field-dominance information in frame to decide which of each pair of fields to place first in the output. If it gets it wrong use setfield filter before "separatefields" filter.

setdar, setsar

The "setdar" filter sets the Display Aspect Ratio for the filter output video.

This is done by changing the specified Sample (aka Pixel) Aspect Ratio, according to the following equation:

        <DAR> = <HORIZONTAL_RESOLUTION> / <VERTICAL_RESOLUTION> * <SAR>

Keep in mind that the "setdar" filter does not modify the pixel dimensions of the video frame. Also, the display aspect ratio set by this filter may be changed by later filters in the filterchain, e.g. in case of scaling or if another "setdar" or a "setsar" filter is applied.

The "setsar" filter sets the Sample (aka Pixel) Aspect Ratio for the filter output video.

Note that as a consequence of the application of this filter, the output display aspect ratio will change according to the equation above.

Keep in mind that the sample aspect ratio set by the "setsar" filter may be changed by later filters in the filterchain, e.g. if another "setsar" or a "setdar" filter is applied.

It accepts the following parameters:

r, ratio, dar ("setdar" only), sar ("setsar" only)
Set the aspect ratio used by the filter.

The parameter can be a floating point number string, an expression, or a string of the form num:den, where num and den are the numerator and denominator of the aspect ratio. If the parameter is not specified, it is assumed the value "0". In case the form "num:den" is used, the ":" character should be escaped.

max
Set the maximum integer value to use for expressing numerator and denominator when reducing the expressed aspect ratio to a rational. Default value is 100.

The parameter sar is an expression containing the following constants:

E, PI, PHI
These are approximated values for the mathematical constants e (Euler's number), pi (Greek pi), and phi (the golden ratio).
w, h
The input width and height.
a
These are the same as w / h.
sar
The input sample aspect ratio.
dar
The input display aspect ratio. It is the same as (w / h) * sar.
hsub, vsub
Horizontal and vertical chroma subsample values. For example, for the pixel format "yuv422p" hsub is 2 and vsub is 1.

Examples

  • To change the display aspect ratio to 16:9, specify one of the following:

            setdar=dar=1.77777
            setdar=dar=16/9
        
  • To change the sample aspect ratio to 10:11, specify:

            setsar=sar=10/11
        
  • To set a display aspect ratio of 16:9, and specify a maximum integer value of 1000 in the aspect ratio reduction, use the command:

            setdar=ratio=16/9:max=1000
        

setfield

Force field for the output video frame.

The "setfield" filter marks the interlace type field for the output frames. It does not change the input frame, but only sets the corresponding property, which affects how the frame is treated by following filters (e.g. "fieldorder" or "yadif").

The filter accepts the following options:

mode
Available values are:
auto
Keep the same field property.
bff
Mark the frame as bottom-field-first.
tff
Mark the frame as top-field-first.
prog
Mark the frame as progressive.

setparams

Force frame parameter for the output video frame.

The "setparams" filter marks interlace and color range 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 filters/encoders.

field_mode
Available values are:
auto
Keep the same field property (default).
bff
Mark the frame as bottom-field-first.
tff
Mark the frame as top-field-first.
prog
Mark the frame as progressive.
range
Available values are:
auto
Keep the same color range property (default).
unspecified, unknown
Mark the frame as unspecified color range.
limited, tv, mpeg
Mark the frame as limited range.
full, pc, jpeg
Mark the frame as full range.
color_primaries
Set the color primaries. Available values are:
auto
Keep the same color primaries property (default).
bt709
unknown
bt470m
bt470bg
smpte170m
smpte240m
film
bt2020
smpte428
smpte431
smpte432
jedec-p22
color_trc
Set the color transfert. Available values are:
auto
Keep the same color trc property (default).
bt709
unknown
bt470m
bt470bg
smpte170m
smpte240m
linear
log100
log316
iec61966-2-4
bt1361e
iec61966-2-1
bt2020-10
bt2020-12
smpte2084
smpte428
arib-std-b67
colorspace
Set the colorspace. Available values are:
auto
Keep the same colorspace property (default).
gbr
bt709
unknown
fcc
bt470bg
smpte170m
smpte240m
ycgco
bt2020nc
bt2020c
smpte2085
chroma-derived-nc
chroma-derived-c
ictcp

showinfo

Show a line containing various information for each input video frame. The input video is not modified.

The shown line contains a sequence of key/value pairs of the form key:value.

The following values are shown in the output:

n
The (sequential) number of the input frame, starting from 0.
pts
The Presentation TimeStamp of the input frame, expressed as a number of time base units. The time base unit depends on the filter input pad.
pts_time
The Presentation TimeStamp of the input frame, expressed as a number of seconds.
pos
The position of the frame in the input stream, or -1 if this information is unavailable and/or meaningless (for example in case of synthetic video).
fmt
The pixel format name.
sar
The sample aspect ratio of the input frame, expressed in the form num/den.
s
The size of the input frame. For the syntax of this option, check the "Video size" section in the ffmpeg-utils manual.
i
The type of interlaced mode ("P" for "progressive", "T" for top field first, "B" for bottom field first).
iskey
This is 1 if the frame is a key frame, 0 otherwise.
type
The picture type of the input frame ("I" for an I-frame, "P" for a P-frame, "B" for a B-frame, or "?" for an unknown type). Also refer to the documentation of the "AVPictureType" enum and of the "av_get_picture_type_char" function defined in libavutil/avutil.h.
checksum
The Adler-32 checksum (printed in hexadecimal) of all the planes of the input frame.
plane_checksum
The Adler-32 checksum (printed in hexadecimal) of each plane of the input frame, expressed in the form "[c0 c1 c2 c3]".

showpalette

Displays the 256 colors palette of each frame. This filter is only relevant for pal8 pixel format frames.

It accepts the following option:

s
Set the size of the box used to represent one palette color entry. Default is 30 (for a "30x30" pixel box).

shuffleframes

Reorder and/or duplicate and/or drop video frames.

It accepts the following parameters:

mapping
Set the destination indexes of input frames. This is space or '|' separated list of indexes that maps input frames to output frames. Number of indexes also sets maximal value that each index may have. '-1' index have special meaning and that is to drop frame.

The first frame has the index 0. The default is to keep the input unchanged.

Examples

  • Swap second and third frame of every three frames of the input:

            ffmpeg -i INPUT -vf "shuffleframes=0 2 1" OUTPUT
        
  • Swap 10th and 1st frame of every ten frames of the input:

            ffmpeg -i INPUT -vf "shuffleframes=9 1 2 3 4 5 6 7 8 0" OUTPUT
        

shuffleplanes

Reorder and/or duplicate video planes.

It accepts the following parameters:

map0
The index of the input plane to be used as the first output plane.
map1
The index of the input plane to be used as the second output plane.
map2
The index of the input plane to be used as the third output plane.
map3
The index of the input plane to be used as the fourth output plane.

The first plane has the index 0. The default is to keep the input unchanged.

Examples

Swap the second and third planes of the input:

        ffmpeg -i INPUT -vf shuffleplanes=0:2:1:3 OUTPUT
    

signalstats

Evaluate various visual metrics that assist in determining issues associated with the digitization of analog video media.

By default the filter will log these metadata values:

YMIN
Display the minimal Y value contained within the input frame. Expressed in range of [0-255].
YLOW
Display the Y value at the 10% percentile within the input frame. Expressed in range of [0-255].
YAVG
Display the average Y value within the input frame. Expressed in range of [0-255].
YHIGH
Display the Y value at the 90% percentile within the input frame. Expressed in range of [0-255].
YMAX
Display the maximum Y value contained within the input frame. Expressed in range of [0-255].
UMIN
Display the minimal U value contained within the input frame. Expressed in range of [0-255].
ULOW
Display the U value at the 10% percentile within the input frame. Expressed in range of [0-255].
UAVG
Display the average U value within the input frame. Expressed in range of [0-255].
UHIGH
Display the U value at the 90% percentile within the input frame. Expressed in range of [0-255].
UMAX
Display the maximum U value contained within the input frame. Expressed in range of [0-255].
VMIN
Display the minimal V value contained within the input frame. Expressed in range of [0-255].
VLOW
Display the V value at the 10% percentile within the input frame. Expressed in range of [0-255].
VAVG
Display the average V value within the input frame. Expressed in range of [0-255].
VHIGH
Display the V value at the 90% percentile within the input frame. Expressed in range of [0-255].
VMAX
Display the maximum V value contained within the input frame. Expressed in range of [0-255].
SATMIN
Display the minimal saturation value contained within the input frame. Expressed in range of [0-~181.02].
SATLOW
Display the saturation value at the 10% percentile within the input frame. Expressed in range of [0-~181.02].
SATAVG
Display the average saturation value within the input frame. Expressed in range of [0-~181.02].
SATHIGH
Display the saturation value at the 90% percentile within the input frame. Expressed in range of [0-~181.02].
SATMAX
Display the maximum saturation value contained within the input frame. Expressed in range of [0-~181.02].
HUEMED
Display the median value for hue within the input frame. Expressed in range of [0-360].
HUEAVG
Display the average value for hue within the input frame. Expressed in range of [0-360].
YDIF
Display the average of sample value difference between all values of the Y plane in the current frame and corresponding values of the previous input frame. Expressed in range of [0-255].
UDIF
Display the average of sample value difference between all values of the U plane in the current frame and corresponding values of the previous input frame. Expressed in range of [0-255].
VDIF
Display the average of sample value difference between all values of the V plane in the current frame and corresponding values of the previous input frame. Expressed in range of [0-255].
YBITDEPTH
Display bit depth of Y plane in current frame. Expressed in range of [0-16].
UBITDEPTH
Display bit depth of U plane in current frame. Expressed in range of [0-16].
VBITDEPTH
Display bit depth of V plane in current frame. Expressed in range of [0-16].

The filter accepts the following options:

stat
out
stat specify an additional form of image analysis. out output video with the specified type of pixel highlighted.

Both options accept the following values:

tout
Identify temporal outliers pixels. A temporal outlier is a pixel unlike the neighboring pixels of the same field. Examples of temporal outliers include the results of video dropouts, head clogs, or tape tracking issues.
vrep
Identify vertical line repetition. Vertical line repetition includes similar rows of pixels within a frame. In born-digital video vertical line repetition is common, but this pattern is uncommon in video digitized from an analog source. When it occurs in video that results from the digitization of an analog source it can indicate concealment from a dropout compensator.
brng
Identify pixels that fall outside of legal broadcast range.
color, c
Set the highlight color for the out option. The default color is yellow.

Examples

  • Output data of various video metrics:

            ffprobe -f lavfi movie=example.mov,signalstats="stat=tout+vrep+brng" -show_frames
        
  • Output specific data about the minimum and maximum values of the Y plane per frame:

            ffprobe -f lavfi movie=example.mov,signalstats -show_entries frame_tags=lavfi.signalstats.YMAX,lavfi.signalstats.YMIN
        
  • Playback video while highlighting pixels that are outside of broadcast range in red.

            ffplay example.mov -vf signalstats="out=brng:color=red"
        
  • Playback video with signalstats metadata drawn over the frame.

            ffplay example.mov -vf signalstats=stat=brng+vrep+tout,drawtext=fontfile=FreeSerif.ttf:textfile=signalstat_drawtext.txt
        

    The contents of signalstat_drawtext.txt used in the command are:

            time %{pts:hms}
            Y (%{metadata:lavfi.signalstats.YMIN}-%{metadata:lavfi.signalstats.YMAX})
            U (%{metadata:lavfi.signalstats.UMIN}-%{metadata:lavfi.signalstats.UMAX})
            V (%{metadata:lavfi.signalstats.VMIN}-%{metadata:lavfi.signalstats.VMAX})
            saturation maximum: %{metadata:lavfi.signalstats.SATMAX}
        

signature

Calculates the MPEG-7 Video Signature. The filter can handle more than one input. In this case the matching between the inputs can be calculated additionally. The filter always passes through the first input. The signature of each stream can be written into a file.

It accepts the following options:

detectmode
Enable or disable the matching process.

Available values are:

off
Disable the calculation of a matching (default).
full
Calculate the matching for the whole video and output whether the whole video matches or only parts.
fast
Calculate only until a matching is found or the video ends. Should be faster in some cases.
nb_inputs
Set the number of inputs. The option value must be a non negative integer. Default value is 1.
filename
Set the path to which the output is written. If there is more than one input, the path must be a prototype, i.e. must contain %d or %0nd (where n is a positive integer), that will be replaced with the input number. If no filename is specified, no output will be written. This is the default.
format
Choose the output format.

Available values are:

binary
Use the specified binary representation (default).
xml
Use the specified xml representation.
th_d
Set threshold to detect one word as similar. The option value must be an integer greater than zero. The default value is 9000.
th_dc
Set threshold to detect all words as similar. The option value must be an integer greater than zero. The default value is 60000.
th_xh
Set threshold to detect frames as similar. The option value must be an integer greater than zero. The default value is 116.
th_di
Set the minimum length of a sequence in frames to recognize it as matching sequence. The option value must be a non negative integer value. The default value is 0.
th_it
Set the minimum relation, that matching frames to all frames must have. The option value must be a double value between 0 and 1. The default value is 0.5.

Examples

  • To calculate the signature of an input video and store it in signature.bin:

            ffmpeg -i input.mkv -vf signature=filename=signature.bin -map 0:v -f null -
        
  • To detect whether two videos match and store the signatures in XML format in signature0.xml and signature1.xml:

            ffmpeg -i input1.mkv -i input2.mkv -filter_complex "[0:v][1:v] signature=nb_inputs=2:detectmode=full:format=xml:filename=signature%d.xml" -map :v -f null -
        

smartblur

Blur the input video without impacting the outlines.

It accepts the following options:

luma_radius, lr
Set the luma radius. The option value must be a float number in the range [0.1,5.0] that specifies the variance of the gaussian filter used to blur the image (slower if larger). Default value is 1.0.
luma_strength, ls
Set the luma strength. The option value must be a float number in the range [-1.0,1.0] that configures the blurring. A value included in [0.0,1.0] will blur the image whereas a value included in [-1.0,0.0] will sharpen the image. Default value is 1.0.
luma_threshold, lt
Set the luma threshold used as a coefficient to determine whether a pixel should be blurred or not. The option value must be an integer in the range [-30,30]. A value of 0 will filter all the image, a value included in [0,30] will filter flat areas and a value included in [-30,0] will filter edges. Default value is 0.
chroma_radius, cr
Set the chroma radius. The option value must be a float number in the range [0.1,5.0] that specifies the variance of the gaussian filter used to blur the image (slower if larger). Default value is luma_radius.
chroma_strength, cs
Set the chroma strength. The option value must be a float number in the range [-1.0,1.0] that configures the blurring. A value included in [0.0,1.0] will blur the image whereas a value included in [-1.0,0.0] will sharpen the image. Default value is luma_strength.
chroma_threshold, ct
Set the chroma threshold used as a coefficient to determine whether a pixel should be blurred or not. The option value must be an integer in the range [-30,30]. A value of 0 will filter all the image, a value included in [0,30] will filter flat areas and a value included in [-30,0] will filter edges. Default value is luma_threshold.

If a chroma option is not explicitly set, the corresponding luma value is set.

ssim

Obtain the SSIM (Structural SImilarity Metric) between two input videos.

This filter takes in input two input videos, the first input is considered the "main" source and is passed unchanged to the output. The second input is used as a "reference" video for computing the SSIM.

Both video inputs must have the same resolution and pixel format for this filter to work correctly. Also it assumes that both inputs have the same number of frames, which are compared one by one.

The filter stores the calculated SSIM of each frame.

The description of the accepted parameters follows.

stats_file, f
If specified the filter will use the named file to save the SSIM of each individual frame. When filename equals "-" the data is sent to standard output.

The file printed if stats_file is selected, contains a sequence of key/value pairs of the form key:value for each compared couple of frames.

A description of each shown parameter follows:

n
sequential number of the input frame, starting from 1
Y, U, V, R, G, B
SSIM of the compared frames for the component specified by the suffix.
All
SSIM of the compared frames for the whole frame.
dB
Same as above but in dB representation.

This filter also supports the framesync options.

For example:

        movie=ref_movie.mpg, setpts=PTS-STARTPTS [main];
        [main][ref] ssim="stats_file=stats.log" [out]

On this example the input file being processed is compared with the reference file ref_movie.mpg. The SSIM of each individual frame is stored in stats.log.

Another example with both psnr and ssim at same time:

        ffmpeg -i main.mpg -i ref.mpg -lavfi  "ssim;[0:v][1:v]psnr" -f null -

stereo3d

Convert between different stereoscopic image formats.

The filters accept the following options:

in
Set stereoscopic image format of input.

Available values for input image formats are:

sbsl
side by side parallel (left eye left, right eye right)
sbsr
side by side crosseye (right eye left, left eye right)
sbs2l
side by side parallel with half width resolution (left eye left, right eye right)
sbs2r
side by side crosseye with half width resolution (right eye left, left eye right)
abl
above-below (left eye above, right eye below)
abr
above-below (right eye above, left eye below)
ab2l
above-below with half height resolution (left eye above, right eye below)
ab2r
above-below with half height resolution (right eye above, left eye below)
al
alternating frames (left eye first, right eye second)
ar
alternating frames (right eye first, left eye second)
irl
interleaved rows (left eye has top row, right eye starts on next row)
irr
interleaved rows (right eye has top row, left eye starts on next row)
icl
interleaved columns, left eye first
icr
interleaved columns, right eye first

Default value is sbsl.

out
Set stereoscopic image format of output.
sbsl
side by side parallel (left eye left, right eye right)
sbsr
side by side crosseye (right eye left, left eye right)
sbs2l
side by side parallel with half width resolution (left eye left, right eye right)
sbs2r
side by side crosseye with half width resolution (right eye left, left eye right)
abl
above-below (left eye above, right eye below)
abr
above-below (right eye above, left eye below)
ab2l
above-below with half height resolution (left eye above, right eye below)
ab2r
above-below with half height resolution (right eye above, left eye below)
al
alternating frames (left eye first, right eye second)
ar
alternating frames (right eye first, left eye second)
irl
interleaved rows (left eye has top row, right eye starts on next row)
irr
interleaved rows (right eye has top row, left eye starts on next row)
arbg
anaglyph red/blue gray (red filter on left eye, blue filter on right eye)
argg
anaglyph red/green gray (red filter on left eye, green filter on right eye)
arcg
anaglyph red/cyan gray (red filter on left eye, cyan filter on right eye)
arch
anaglyph red/cyan half colored (red filter on left eye, cyan filter on right eye)
arcc
anaglyph red/cyan color (red filter on left eye, cyan filter on right eye)
arcd
anaglyph red/cyan color optimized with the least squares projection of dubois (red filter on left eye, cyan filter on right eye)
agmg
anaglyph green/magenta gray (green filter on left eye, magenta filter on right eye)
agmh
anaglyph green/magenta half colored (green filter on left eye, magenta filter on right eye)
agmc
anaglyph green/magenta colored (green filter on left eye, magenta filter on right eye)
agmd
anaglyph green/magenta color optimized with the least squares projection of dubois (green filter on left eye, magenta filter on right eye)
aybg
anaglyph yellow/blue gray (yellow filter on left eye, blue filter on right eye)
aybh
anaglyph yellow/blue half colored (yellow filter on left eye, blue filter on right eye)
aybc
anaglyph yellow/blue colored (yellow filter on left eye, blue filter on right eye)
aybd
anaglyph yellow/blue color optimized with the least squares projection of dubois (yellow filter on left eye, blue filter on right eye)
ml
mono output (left eye only)
mr
mono output (right eye only)
chl
checkerboard, left eye first
chr
checkerboard, right eye first
icl
interleaved columns, left eye first
icr
interleaved columns, right eye first
hdmi
HDMI frame pack

Default value is arcd.

Examples

  • Convert input video from side by side parallel to anaglyph yellow/blue dubois:

            stereo3d=sbsl:aybd
        
  • Convert input video from above below (left eye above, right eye below) to side by side crosseye.

            stereo3d=abl:sbsr
        

streamselect, astreamselect

Select video or audio streams.

The filter accepts the following options:

inputs
Set number of inputs. Default is 2.
map
Set input indexes to remap to outputs.

Commands

The "streamselect" and "astreamselect" filter supports the following commands:

map
Set input indexes to remap to outputs.

Examples

  • Select first 5 seconds 1st stream and rest of time 2nd stream:

            sendcmd='5.0 streamselect map 1',streamselect=inputs=2:map=0
        
  • Same as above, but for audio:

            asendcmd='5.0 astreamselect map 1',astreamselect=inputs=2:map=0
        

sobel

Apply sobel operator to input video stream.

The filter accepts the following option:

planes
Set which planes will be processed, unprocessed planes will be copied. By default value 0xf, all planes will be processed.
scale
Set value which will be multiplied with filtered result.
delta
Set value which will be added to filtered result.

spp

Apply a simple postprocessing filter that compresses and decompresses the image at several (or - in the case of quality level 6 - all) shifts and average the results.

The filter accepts the following options:

quality
Set quality. This option defines the number of levels for averaging. It accepts an integer in the range 0-6. If set to 0, the filter will have no effect. A value of 6 means the higher quality. For each increment of that value the speed drops by a factor of approximately 2. Default value is 3.
qp
Force a constant quantization parameter. If not set, the filter will use the QP from the video stream (if available).
mode
Set thresholding mode. Available modes are:
hard
Set hard thresholding (default).
soft
Set soft thresholding (better de-ringing effect, but likely blurrier).
use_bframe_qp
Enable the use of the QP from the B-Frames if set to 1. Using this option may cause flicker since the B-Frames have often larger QP. Default is 0 (not enabled).

sr

Scale the input by applying one of the super-resolution methods based on convolutional neural networks. Supported models:

Training scripts as well as scripts for model generation are provided in the repository at <https://github.com/HighVoltageRocknRoll/sr.git>.

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:
native
Native implementation of DNN loading and execution.
tensorflow
TensorFlow backend. To enable this backend you need to install the TensorFlow for C library (see <https://www.tensorflow.org/install/install_c>) and configure FFmpeg with "--enable-libtensorflow"

Default value is native.

model
Set path to model file specifying network architecture and its parameters. Note that different backends use different file formats. TensorFlow backend can load files for both formats, while native backend can load files for only its format.
scale_factor
Set scale factor for SRCNN model. Allowed values are 2, 3 and 4. Default value is 2. Scale factor is necessary for SRCNN model, because it accepts input upscaled using bicubic upscaling with proper scale factor.

subtitles

Draw subtitles on top of input video using the libass library.

To enable compilation of this filter you need to configure FFmpeg with "--enable-libass". This filter also requires a build with libavcodec and libavformat to convert the passed subtitles file to ASS (Advanced Substation Alpha) subtitles format.

The filter accepts the following options:

filename, f
Set the filename of the subtitle file to read. It must be specified.
original_size
Specify the size of the original video, the video for which the ASS file was composed. For the syntax of this option, check the "Video size" section in the ffmpeg-utils manual. Due to a misdesign in ASS aspect ratio arithmetic, this is necessary to correctly scale the fonts if the aspect ratio has been changed.
fontsdir
Set a directory path containing fonts that can be used by the filter. These fonts will be used in addition to whatever the font provider uses.
alpha
Process alpha channel, by default alpha channel is untouched.
charenc
Set subtitles input character encoding. "subtitles" filter only. Only useful if not UTF-8.
stream_index, si
Set subtitles stream index. "subtitles" filter only.
force_style
Override default style or script info parameters of the subtitles. It accepts a string containing ASS style format "KEY=VALUE" couples separated by ",".

If the first key is not specified, it is assumed that the first value specifies the filename.

For example, to render the file sub.srt on top of the input video, use the command:

        subtitles=sub.srt

which is equivalent to:

        subtitles=filename=sub.srt

To render the default subtitles stream from file video.mkv, use:

        subtitles=video.mkv

To render the second subtitles stream from that file, use:

        subtitles=video.mkv:si=1

To make the subtitles stream from sub.srt appear in 80% transparent blue "DejaVu Serif", use:

        subtitles=sub.srt:force_style='FontName=DejaVu Serif,PrimaryColour=&HCCFF0000'

super2xsai

Scale the input by 2x and smooth using the Super2xSaI (Scale and Interpolate) pixel art scaling algorithm.

Useful for enlarging pixel art images without reducing sharpness.

swaprect

Swap two rectangular objects in video.

This filter accepts the following options:

w
Set object width.
h
Set object height.
x1
Set 1st rect x coordinate.
y1
Set 1st rect y coordinate.
x2
Set 2nd rect x coordinate.
y2
Set 2nd rect y coordinate.

All expressions are evaluated once for each frame.

The all options are expressions containing the following constants:

w
h
The input width and height.
a
same as w / h
sar
input sample aspect ratio
dar
input display aspect ratio, it is the same as (w / h) * sar
n
The number of the input frame, starting from 0.
t
The timestamp expressed in seconds. It's NAN if the input timestamp is unknown.
pos
the position in the file of the input frame, NAN if unknown

swapuv

Swap U & V plane.

telecine

Apply telecine process to the video.

This filter accepts the following options:

first_field
top, t
top field first
bottom, b
bottom field first The default value is "top".
pattern
A string of numbers representing the pulldown pattern you wish to apply. The default value is 23.

        Some typical patterns:
        
        NTSC output (30i):
        27.5p: 32222
        24p: 23 (classic)
        24p: 2332 (preferred)
        20p: 33
        18p: 334
        16p: 3444
        
        PAL output (25i):
        27.5p: 12222
        24p: 222222222223 ("Euro pulldown")
        16.67p: 33
        16p: 33333334

threshold

Apply threshold effect to video stream.

This filter needs four video streams to perform thresholding. First stream is stream we are filtering. Second stream is holding threshold values, third stream is holding min values, and last, fourth stream is holding max values.

The filter accepts the following option:

planes
Set which planes will be processed, unprocessed planes will be copied. By default value 0xf, all planes will be processed.

For example if first stream pixel's component value is less then threshold value of pixel component from 2nd threshold stream, third stream value will picked, otherwise fourth stream pixel component value will be picked.

Using color source filter one can perform various types of thresholding:

Examples

  • Binary threshold, using gray color as threshold:

            ffmpeg -i 320x240.avi -f lavfi -i color=gray -f lavfi -i color=black -f lavfi -i color=white -lavfi threshold output.avi
        
  • Inverted binary threshold, using gray color as threshold:

            ffmpeg -i 320x240.avi -f lavfi -i color=gray -f lavfi -i color=white -f lavfi -i color=black -lavfi threshold output.avi
        
  • Truncate binary threshold, using gray color as threshold:

            ffmpeg -i 320x240.avi -f lavfi -i color=gray -i 320x240.avi -f lavfi -i color=gray -lavfi threshold output.avi
        
  • Threshold to zero, using gray color as threshold:

            ffmpeg -i 320x240.avi -f lavfi -i color=gray -f lavfi -i color=white -i 320x240.avi -lavfi threshold output.avi
        
  • Inverted threshold to zero, using gray color as threshold:

            ffmpeg -i 320x240.avi -f lavfi -i color=gray -i 320x240.avi -f lavfi -i color=white -lavfi threshold output.avi
        

thumbnail

Select the most representative frame in a given sequence of consecutive frames.

The filter accepts the following options:

n
Set the frames batch size to analyze; in a set of n frames, the filter will pick one of them, and then handle the next batch of n frames until the end. Default is 100.

Since the filter keeps track of the whole frames sequence, a bigger n value will result in a higher memory usage, so a high value is not recommended.

Examples

  • Extract one picture each 50 frames:

            thumbnail=50
        
  • Complete example of a thumbnail creation with ffmpeg:

            ffmpeg -i in.avi -vf thumbnail,scale=300:200 -frames:v 1 out.png
        

tile

Tile several successive frames together.

The filter accepts the following options:

layout
Set the grid size (i.e. the number of lines and columns). For the syntax of this option, check the "Video size" section in the ffmpeg-utils manual.
nb_frames
Set the maximum number of frames to render in the given area. It must be less than or equal to wxh. The default value is 0, meaning all the area will be used.
margin
Set the outer border margin in pixels.
padding
Set the inner border thickness (i.e. the number of pixels between frames). For more advanced padding options (such as having different values for the edges), refer to the pad video filter.
color
Specify the color of the unused area. For the syntax of this option, check the "Color" section in the ffmpeg-utils manual. The default value of color is "black".
overlap
Set the number of frames to overlap when tiling several successive frames together. The value must be between 0 and nb_frames - 1.
init_padding
Set the number of frames to initially be empty before displaying first output frame. This controls how soon will one get first output frame. The value must be between 0 and nb_frames - 1.

Examples

  • Produce 8x8 PNG tiles of all keyframes (-skip_frame nokey) in a movie:

            ffmpeg -skip_frame nokey -i file.avi -vf 'scale=128:72,tile=8x8' -an -vsync 0 keyframes%03d.png
        

    The -vsync 0 is necessary to prevent ffmpeg from duplicating each output frame to accommodate the originally detected frame rate.

  • Display 5 pictures in an area of "3x2" frames, with 7 pixels between them, and 2 pixels of initial margin, using mixed flat and named options:

            tile=3x2:nb_frames=5:padding=7:margin=2
        

tinterlace

Perform various types of temporal field interlacing.

Frames are counted starting from 1, so the first input frame is considered odd.

The filter accepts the following options:

mode
Specify the mode of the interlacing. This option can also be specified as a value alone. See below for a list of values for this option.

Available values are:

merge, 0
Move odd frames into the upper field, even into the lower field, generating a double height frame at half frame rate.

         ------> time
        Input:
        Frame 1         Frame 2         Frame 3         Frame 4
        
        11111           22222           33333           44444
        11111           22222           33333           44444
        11111           22222           33333           44444
        11111           22222           33333           44444
        
        Output:
        11111                           33333
        22222                           44444
        11111                           33333
        22222                           44444
        11111                           33333
        22222                           44444
        11111                           33333
        22222                           44444
    
drop_even, 1
Only output odd frames, even frames are dropped, generating a frame with unchanged height at half frame rate.

         ------> time
        Input:
        Frame 1         Frame 2         Frame 3         Frame 4
        
        11111           22222           33333           44444
        11111           22222           33333           44444
        11111           22222           33333           44444
        11111           22222           33333           44444
        
        Output:
        11111                           33333
        11111                           33333
        11111                           33333
        11111                           33333
    
drop_odd, 2
Only output even frames, odd frames are dropped, generating a frame with unchanged height at half frame rate.

         ------> time
        Input:
        Frame 1         Frame 2         Frame 3         Frame 4
        
        11111           22222           33333           44444
        11111           22222           33333           44444
        11111           22222           33333           44444
        11111           22222           33333           44444
        
        Output:
                        22222                           44444
                        22222                           44444
                        22222                           44444
                        22222                           44444
    
pad, 3
Expand each frame to full height, but pad alternate lines with black, generating a frame with double height at the same input frame rate.

         ------> time
        Input:
        Frame 1         Frame 2         Frame 3         Frame 4
        
        11111           22222           33333           44444
        11111           22222           33333           44444
        11111           22222           33333           44444
        11111           22222           33333           44444
        
        Output:
        11111           .....           33333           .....
        .....           22222           .....           44444
        11111           .....           33333           .....
        .....           22222           .....           44444
        11111           .....           33333           .....
        .....           22222           .....           44444
        11111           .....           33333           .....
        .....           22222           .....           44444
    
interleave_top, 4
Interleave the upper field from odd frames with the lower field from even frames, generating a frame with unchanged height at half frame rate.

         ------> time
        Input:
        Frame 1         Frame 2         Frame 3         Frame 4
        
        11111<-         22222           33333<-         44444
        11111           22222<-         33333           44444<-
        11111<-         22222           33333<-         44444
        11111           22222<-         33333           44444<-
        
        Output:
        11111                           33333
        22222                           44444
        11111                           33333
        22222                           44444
    
interleave_bottom, 5
Interleave the lower field from odd frames with the upper field from even frames, generating a frame with unchanged height at half frame rate.

         ------> time
        Input:
        Frame 1         Frame 2         Frame 3         Frame 4
        
        11111           22222<-         33333           44444<-
        11111<-         22222           33333<-         44444
        11111           22222<-         33333           44444<-
        11111<-         22222           33333<-         44444
        
        Output:
        22222                           44444
        11111                           33333
        22222                           44444
        11111                           33333
    
interlacex2, 6
Double frame rate with unchanged height. Frames are inserted each containing the second temporal field from the previous input frame and the first temporal field from the next input frame. This mode relies on the top_field_first flag. Useful for interlaced video displays with no field synchronisation.

         ------> time
        Input:
        Frame 1         Frame 2         Frame 3         Frame 4
        
        11111           22222           33333           44444
         11111           22222           33333           44444
        11111           22222           33333           44444
         11111           22222           33333           44444
        
        Output:
        11111   22222   22222   33333   33333   44444   44444
         11111   11111   22222   22222   33333   33333   44444
        11111   22222   22222   33333   33333   44444   44444
         11111   11111   22222   22222   33333   33333   44444
    
mergex2, 7
Move odd frames into the upper field, even into the lower field, generating a double height frame at same frame rate.

         ------> time
        Input:
        Frame 1         Frame 2         Frame 3         Frame 4
        
        11111           22222           33333           44444
        11111           22222           33333           44444
        11111           22222           33333           44444
        11111           22222           33333           44444
        
        Output:
        11111           33333           33333           55555
        22222           22222           44444           44444
        11111           33333           33333           55555
        22222           22222           44444           44444
        11111           33333           33333           55555
        22222           22222           44444           44444
        11111           33333           33333           55555
        22222           22222           44444           44444
    

Numeric values are deprecated but are accepted for backward compatibility reasons.

Default mode is "merge".

flags
Specify flags influencing the filter process.

Available value for flags is:

low_pass_filter, vlfp
Enable linear vertical low-pass filtering in the filter. Vertical low-pass filtering is required when creating an interlaced destination from a progressive source which contains high-frequency vertical detail. Filtering will reduce interlace 'twitter' and Moire patterning.
complex_filter, cvlfp
Enable complex vertical low-pass filtering. This will slightly less reduce interlace 'twitter' and Moire patterning but better retain detail and subjective sharpness impression.

Vertical low-pass filtering can only be enabled for mode interleave_top and interleave_bottom.

tmix

Mix successive video frames.

A description of the accepted options follows.

frames
The number of successive frames to mix. If unspecified, it defaults to 3.
weights
Specify weight of each input video frame. Each weight is separated by space. If number of weights is smaller than number of frames last specified weight will be used for all remaining unset weights.
scale
Specify scale, if it is set it will be multiplied with sum of each weight multiplied with pixel values to give final destination pixel value. By default scale is auto scaled to sum of weights.

Examples

  • Average 7 successive frames:

            tmix=frames=7:weights="1 1 1 1 1 1 1"
        
  • Apply simple temporal convolution:

            tmix=frames=3:weights="-1 3 -1"
        
  • Similar as above but only showing temporal differences:

            tmix=frames=3:weights="-1 2 -1":scale=1
        

tonemap

Tone map colors from different dynamic ranges.

This filter expects data in single precision floating point, as it needs to operate on (and can output) out-of-range values. Another filter, such as zscale, is needed to convert the resulting frame to a usable format.

The tonemapping algorithms implemented only work on linear light, so input data should be linearized beforehand (and possibly correctly tagged).

        ffmpeg -i INPUT -vf zscale=transfer=linear,tonemap=clip,zscale=transfer=bt709,format=yuv420p OUTPUT

Options

The filter accepts the following options.

tonemap
Set the tone map algorithm to use.

Possible values are:

none
Do not apply any tone map, only desaturate overbright pixels.
clip
Hard-clip any out-of-range values. Use it for perfect color accuracy for in-range values, while distorting out-of-range values.
linear
Stretch the entire reference gamut to a linear multiple of the display.
gamma
Fit a logarithmic transfer between the tone curves.
reinhard
Preserve overall image brightness with a simple curve, using nonlinear contrast, which results in flattening details and degrading color accuracy.
hable
Preserve both dark and bright details better than reinhard, at the cost of slightly darkening everything. Use it when detail preservation is more important than color and brightness accuracy.
mobius
Smoothly map out-of-range values, while retaining contrast and colors for in-range material as much as possible. Use it when color accuracy is more important than detail preservation.

Default is none.

param
Tune the tone mapping algorithm.

This affects the following algorithms:

none
Ignored.
linear
Specifies the scale factor to use while stretching. Default to 1.0.
gamma
Specifies the exponent of the function. Default to 1.8.
clip
Specify an extra linear coefficient to multiply into the signal before clipping. Default to 1.0.
reinhard
Specify the local contrast coefficient at the display peak. Default to 0.5, which means that in-gamut values will be about half as bright as when clipping.
hable
Ignored.
mobius
Specify the transition point from linear to mobius transform. Every value below this point is guaranteed to be mapped 1:1. The higher the value, the more accurate the result will be, at the cost of losing bright details. Default to 0.3, which due to the steep initial slope still preserves in-range colors fairly accurately.
desat
Apply desaturation for highlights that exceed this level of brightness. The higher the parameter, the more color information will be preserved. This setting helps prevent unnaturally blown-out colors for super-highlights, by (smoothly) turning into white instead. This makes images feel more natural, at the cost of reducing information about out-of-range colors.

The default of 2.0 is somewhat conservative and will mostly just apply to skies or directly sunlit surfaces. A setting of 0.0 disables this option.

This option works only if the input frame has a supported color tag.

peak
Override signal/nominal/reference peak with this value. Useful when the embedded peak information in display metadata is not reliable or when tone mapping from a lower range to a higher range.

transpose

Transpose rows with columns in the input video and optionally flip it.

It accepts the following parameters:

dir
Specify the transposition direction.

Can assume the following values:

0, 4, cclock_flip
Rotate by 90 degrees counterclockwise and vertically flip (default), that is:

        L.R     L.l
        . . ->  . .
        l.r     R.r
    
1, 5, clock
Rotate by 90 degrees clockwise, that is:

        L.R     l.L
        . . ->  . .
        l.r     r.R
    
2, 6, cclock
Rotate by 90 degrees counterclockwise, that is:

        L.R     R.r
        . . ->  . .
        l.r     L.l
    
3, 7, clock_flip
Rotate by 90 degrees clockwise and vertically flip, that is:

        L.R     r.R
        . . ->  . .
        l.r     l.L
    

For values between 4-7, the transposition is only done if the input video geometry is portrait and not landscape. These values are deprecated, the "passthrough" option should be used instead.

Numerical values are deprecated, and should be dropped in favor of symbolic constants.

passthrough
Do not apply the transposition if the input geometry matches the one specified by the specified value. It accepts the following values:
none
Always apply transposition.
portrait
Preserve portrait geometry (when height >= width).
landscape
Preserve landscape geometry (when width >= height).

Default value is "none".

For example to rotate by 90 degrees clockwise and preserve portrait layout:

        transpose=dir=1:passthrough=portrait

The command above can also be specified as:

        transpose=1:portrait

transpose_npp

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).

trim

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

It accepts the following parameters:

start
Specify the time of the start of the kept section, i.e. the frame with the timestamp start will be the first frame in the output.
end
Specify the time of the first frame that will be dropped, i.e. the frame immediately preceding the one with the timestamp end will be the last frame in the output.
start_pts
This is the same as start, except this option sets the start timestamp in timebase units instead of seconds.
end_pts
This is the same as end, except this option sets the end timestamp in timebase units instead of seconds.
duration
The maximum duration of the output in seconds.
start_frame
The number of the first frame that should be passed to the output.
end_frame
The number of the first frame that should be dropped.

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

Note that the first two sets of the start/end options and the duration option look at the frame timestamp, while the _frame variants simply count the frames that pass through the filter. Also note that this filter does not modify the timestamps. If you wish for the output timestamps to start at zero, insert a setpts filter after the trim filter.

If multiple start or end options are set, this filter tries to be greedy and keep all the frames that match at least one of the specified constraints. To keep only the part that matches all the constraints at once, chain multiple trim filters.

The defaults are such that all the input is kept. So it is possible to set e.g. just the end values to keep everything before the specified time.

Examples:

  • Drop everything except the second minute of input:

            ffmpeg -i INPUT -vf trim=60:120
        
  • Keep only the first second:

            ffmpeg -i INPUT -vf trim=duration=1
        

unpremultiply

Apply alpha unpremultiply effect to input video stream using first plane of second stream as alpha.

Both streams must have same dimensions and same pixel format.

The filter accepts the following option:

planes
Set which planes will be processed, unprocessed planes will be copied. By default value 0xf, all planes will be processed.

If the format has 1 or 2 components, then luma is bit 0. If the format has 3 or 4 components: for RGB formats bit 0 is green, bit 1 is blue and bit 2 is red; for YUV formats bit 0 is luma, bit 1 is chroma-U and bit 2 is chroma-V. If present, the alpha channel is always the last bit.

inplace
Do not require 2nd input for processing, instead use alpha plane from input stream.

unsharp

Sharpen or blur the input video.

It accepts the following parameters:

luma_msize_x, lx
Set the luma matrix horizontal size. It must be an odd integer between 3 and 23. The default value is 5.
luma_msize_y, ly
Set the luma matrix vertical size. It must be an odd integer between 3 and 23. The default value is 5.
luma_amount, la
Set the luma effect strength. It must be a floating point number, reasonable values lay between -1.5 and 1.5.

Negative values will blur the input video, while positive values will sharpen it, a value of zero will disable the effect.

Default value is 1.0.

chroma_msize_x, cx
Set the chroma matrix horizontal size. It must be an odd integer between 3 and 23. The default value is 5.
chroma_msize_y, cy
Set the chroma matrix vertical size. It must be an odd integer between 3 and 23. The default value is 5.
chroma_amount, ca
Set the chroma effect strength. It must be a floating point number, reasonable values lay between -1.5 and 1.5.

Negative values will blur the input video, while positive values will sharpen it, a value of zero will disable the effect.

Default value is 0.0.

All parameters are optional and default to the equivalent of the string '5:5:1.0:5:5:0.0'.

Examples

  • Apply strong luma sharpen effect:

            unsharp=luma_msize_x=7:luma_msize_y=7:luma_amount=2.5
        
  • Apply a strong blur of both luma and chroma parameters:

            unsharp=7:7:-2:7:7:-2
        

uspp

Apply ultra slow/simple postprocessing filter that compresses and decompresses the image at several (or - in the case of quality level 8 - all) shifts and average the results.

The way this differs from the behavior of spp is that uspp actually encodes & decodes each case with libavcodec Snow, whereas spp uses a simplified intra only 8x8 DCT similar to MJPEG.

The filter accepts the following options:

quality
Set quality. This option defines the number of levels for averaging. It accepts an integer in the range 0-8. If set to 0, the filter will have no effect. A value of 8 means the higher quality. For each increment of that value the speed drops by a factor of approximately 2. Default value is 3.
qp
Force a constant quantization parameter. If not set, the filter will use the QP from the video stream (if available).

vaguedenoiser

Apply a wavelet based denoiser.

It transforms each frame from the video input into the wavelet domain, using Cohen-Daubechies-Feauveau 9/7. Then it applies some filtering to the obtained coefficients. It does an inverse wavelet transform after. Due to wavelet properties, it should give a nice smoothed result, and reduced noise, without blurring picture features.

This filter accepts the following options:

threshold
The filtering strength. The higher, the more filtered the video will be. Hard thresholding can use a higher threshold than soft thresholding before the video looks overfiltered. Default value is 2.
method
The filtering method the filter will use.

It accepts the following values:

hard
All values under the threshold will be zeroed.
soft
All values under the threshold will be zeroed. All values above will be reduced by the threshold.
garrote
Scales or nullifies coefficients - intermediary between (more) soft and (less) hard thresholding.

Default is garrote.

nsteps
Number of times, the wavelet will decompose the picture. Picture can't be decomposed beyond a particular point (typically, 8 for a 640x480 frame - as 2^9 = 512 > 480). Valid values are integers between 1 and 32. Default value is 6.
percent
Partial of full denoising (limited coefficients shrinking), from 0 to 100. Default value is 85.
planes
A list of the planes to process. By default all planes are processed.

vectorscope

Display 2 color component values in the two dimensional graph (which is called a vectorscope).

This filter accepts the following options:

mode, m
Set vectorscope mode.

It accepts the following values:

gray
Gray values are displayed on graph, higher brightness means more pixels have same component color value on location in graph. This is the default mode.
color
Gray values are displayed on graph. Surrounding pixels values which are not present in video frame are drawn in gradient of 2 color components which are set by option "x" and "y". The 3rd color component is static.
color2
Actual color components values present in video frame are displayed on graph.
color3
Similar as color2 but higher frequency of same values "x" and "y" on graph increases value of another color component, which is luminance by default values of "x" and "y".
color4
Actual colors present in video frame are displayed on graph. If two different colors map to same position on graph then color with higher value of component not present in graph is picked.
color5
Gray values are displayed on graph. Similar to "color" but with 3rd color component picked from radial gradient.
x
Set which color component will be represented on X-axis. Default is 1.
y
Set which color component will be represented on Y-axis. Default is 2.
intensity, i
Set intensity, used by modes: gray, color, color3 and color5 for increasing brightness of color component which represents frequency of (X, Y) location in graph.
envelope, e
none
No envelope, this is default.
instant
Instant envelope, even darkest single pixel will be clearly highlighted.
peak
Hold maximum and minimum values presented in graph over time. This way you can still spot out of range values without constantly looking at vectorscope.
peak+instant
Peak and instant envelope combined together.
graticule, g
Set what kind of graticule to draw.
none
green
color
opacity, o
Set graticule opacity.
flags, f
Set graticule flags.
white
Draw graticule for white point.
black
Draw graticule for black point.
name
Draw color points short names.
bgopacity, b
Set background opacity.
lthreshold, l
Set low threshold for color component not represented on X or Y axis. Values lower than this value will be ignored. Default is 0. Note this value is multiplied with actual max possible value one pixel component can have. So for 8-bit input and low threshold value of 0.1 actual threshold is 0.1 * 255 = 25.
hthreshold, h
Set high threshold for color component not represented on X or Y axis. Values higher than this value will be ignored. Default is 1. Note this value is multiplied with actual max possible value one pixel component can have. So for 8-bit input and high threshold value of 0.9 actual threshold is 0.9 * 255 = 230.
colorspace, c
Set what kind of colorspace to use when drawing graticule.
auto
601
709

Default is auto.

vidstabdetect

Analyze video stabilization/deshaking. Perform pass 1 of 2, see vidstabtransform for pass 2.

This filter generates a file with relative translation and rotation transform information about subsequent frames, which is then used by the vidstabtransform filter.

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

This filter accepts the following options:

result
Set the path to the file used to write the transforms information. Default value is transforms.trf.
shakiness
Set how shaky the video is and how quick the camera is. It accepts an integer in the range 1-10, a value of 1 means little shakiness, a value of 10 means strong shakiness. Default value is 5.
accuracy
Set the accuracy of the detection process. It must be a value in the range 1-15. A value of 1 means low accuracy, a value of 15 means high accuracy. Default value is 15.
stepsize
Set stepsize of the search process. The region around minimum is scanned with 1 pixel resolution. Default value is 6.
mincontrast
Set minimum contrast. Below this value a local measurement field is discarded. Must be a floating point value in the range 0-1. Default value is 0.3.
tripod
Set reference frame number for tripod mode.

If enabled, the motion of the frames is compared to a reference frame in the filtered stream, identified by the specified number. The idea is to compensate all movements in a more-or-less static scene and keep the camera view absolutely still.

If set to 0, it is disabled. The frames are counted starting from 1.

show
Show fields and transforms in the resulting frames. It accepts an integer in the range 0-2. Default value is 0, which disables any visualization.

Examples

  • Use default values:

            vidstabdetect
        
  • Analyze strongly shaky movie and put the results in file mytransforms.trf:

            vidstabdetect=shakiness=10:accuracy=15:result="mytransforms.trf"
        
  • Visualize the result of internal transformations in the resulting video:

            vidstabdetect=show=1
        
  • Analyze a video with medium shakiness using ffmpeg:

            ffmpeg -i input -vf vidstabdetect=shakiness=5:show=1 dummy.avi
        

vidstabtransform

Video stabilization/deshaking: pass 2 of 2, see vidstabdetect for pass 1.

Read a file with transform information for each frame and apply/compensate them. Together with the vidstabdetect filter this can be used to deshake videos. See also <http://public.hronopik.de/vid.stab>. It is important to also use the unsharp filter, see below.

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

Options

input
Set path to the file used to read the transforms. Default value is transforms.trf.
smoothing
Set the number of frames (value*2 + 1) used for lowpass filtering the camera movements. Default value is 10.

For example a number of 10 means that 21 frames are used (10 in the past and 10 in the future) to smoothen the motion in the video. A larger value leads to a smoother video, but limits the acceleration of the camera (pan/tilt movements). 0 is a special case where a static camera is simulated.

optalgo
Set the camera path optimization algorithm.

Accepted values are:

gauss
gaussian kernel low-pass filter on camera motion (default)
avg
averaging on transformations
maxshift
Set maximal number of pixels to translate frames. Default value is -1, meaning no limit.
maxangle
Set maximal angle in radians (degree*PI/180) to rotate frames. Default value is -1, meaning no limit.
crop
Specify how to deal with borders that may be visible due to movement compensation.

Available values are:

keep
keep image information from previous frame (default)
black
fill the border black
invert
Invert transforms if set to 1. Default value is 0.
relative
Consider transforms as relative to previous frame if set to 1, absolute if set to 0. Default value is 0.
zoom
Set percentage to zoom. A positive value will result in a zoom-in effect, a negative value in a zoom-out effect. Default value is 0 (no zoom).
optzoom
Set optimal zooming to avoid borders.

Accepted values are:

0
disabled
1
optimal static zoom value is determined (only very strong movements will lead to visible borders) (default)
2
optimal adaptive zoom value is determined (no borders will be visible), see zoomspeed

Note that the value given at zoom is added to the one calculated here.

zoomspeed
Set percent to zoom maximally each frame (enabled when optzoom is set to 2). Range is from 0 to 5, default value is 0.25.
interpol
Specify type of interpolation.

Available values are:

no
no interpolation
linear
linear only horizontal
bilinear
linear in both directions (default)
bicubic
cubic in both directions (slow)
tripod
Enable virtual tripod mode if set to 1, which is equivalent to "relative=0:smoothing=0". Default value is 0.

Use also "tripod" option of vidstabdetect.

debug
Increase log verbosity if set to 1. Also the detected global motions are written to the temporary file global_motions.trf. Default value is 0.

Examples

  • Use ffmpeg for a typical stabilization with default values:

            ffmpeg -i inp.mpeg -vf vidstabtransform,unsharp=5:5:0.8:3:3:0.4 inp_stabilized.mpeg
        

    Note the use of the unsharp filter which is always recommended.

  • Zoom in a bit more and load transform data from a given file:

            vidstabtransform=zoom=5:input="mytransforms.trf"
        
  • Smoothen the video even more:

            vidstabtransform=smoothing=30
        

vflip

Flip the input video vertically.

For example, to vertically flip a video with ffmpeg:

        ffmpeg -i in.avi -vf "vflip" out.avi

vfrdet

Detect variable frame rate video.

This filter tries to detect if the input is variable or constant frame rate.

At end it will output number of frames detected as having variable delta pts, and ones with constant delta pts. If there was frames with variable delta, than it will also show min and max delta encountered.

vibrance

Boost or alter saturation.

The filter accepts the following options:

intensity
Set strength of boost if positive value or strength of alter if negative value. Default is 0. Allowed range is from -2 to 2.
rbal
Set the red balance. Default is 1. Allowed range is from -10 to 10.
gbal
Set the green balance. Default is 1. Allowed range is from -10 to 10.
bbal
Set the blue balance. Default is 1. Allowed range is from -10 to 10.
rlum
Set the red luma coefficient.
glum
Set the green luma coefficient.
blum
Set the blue luma coefficient.

vignette

Make or reverse a natural vignetting effect.

The filter accepts the following options:

angle, a
Set lens angle expression as a number of radians.

The value is clipped in the "[0,PI/2]" range.

Default value: "PI/5"

x0
y0
Set center coordinates expressions. Respectively "w/2" and "h/2" by default.
mode
Set forward/backward mode.

Available modes are:

forward
The larger the distance from the central point, the darker the image becomes.
backward
The larger the distance from the central point, the brighter the image becomes. This can be used to reverse a vignette effect, though there is no automatic detection to extract the lens angle and other settings (yet). It can also be used to create a burning effect.

Default value is forward.

eval
Set evaluation mode for the expressions (angle, x0, y0).

It accepts the following values:

init
Evaluate expressions only once during the filter initialization.
frame
Evaluate expressions for each incoming frame. This is way slower than the init mode since it requires all the scalers to be re-computed, but it allows advanced dynamic expressions.

Default value is init.

dither
Set dithering to reduce the circular banding effects. Default is 1 (enabled).
aspect
Set vignette aspect. This setting allows one to adjust the shape of the vignette. Setting this value to the SAR of the input will make a rectangular vignetting following the dimensions of the video.

Default is "1/1".

Expressions

The alpha, x0 and y0 expressions can contain the following parameters.

w
h
input width and height
n
the number of input frame, starting from 0
pts
the PTS (Presentation TimeStamp) time of the filtered video frame, expressed in TB units, NAN if undefined
r
frame rate of the input video, NAN if the input frame rate is unknown
t
the PTS (Presentation TimeStamp) of the filtered video frame, expressed in seconds, NAN if undefined
tb
time base of the input video

Examples

  • Apply simple strong vignetting effect:

            vignette=PI/4
        
  • Make a flickering vignetting:

            vignette='PI/4+random(1)*PI/50':eval=frame
        

vmafmotion

Obtain the average vmaf motion score of a video. It is one of the component filters of VMAF.

The obtained average motion score is printed through the logging system.

In the below example the input file ref.mpg is being processed and score is computed.

        ffmpeg -i ref.mpg -lavfi vmafmotion -f null -

vstack

Stack input videos vertically.

All streams must be of same pixel format and of same width.

Note that this filter is faster than using overlay and pad filter to create same output.

The filter accept the following option:

inputs
Set number of input streams. Default is 2.
shortest
If set to 1, force the output to terminate when the shortest input terminates. Default value is 0.

w3fdif

Deinterlace the input video ("w3fdif" stands for "Weston 3 Field Deinterlacing Filter").

Based on the process described by Martin Weston for BBC R&D, and implemented based on the de-interlace algorithm written by Jim Easterbrook for BBC R&D, the Weston 3 field deinterlacing filter uses filter coefficients calculated by BBC R&D.

There are two sets of filter coefficients, so called "simple": and "complex". Which set of filter coefficients is used can be set by passing an optional parameter:

filter
Set the interlacing filter coefficients. Accepts one of the following values:
simple
Simple filter coefficient set.
complex
More-complex filter coefficient set.

Default value is complex.

deint
Specify which frames to deinterlace. Accept one of the following values:
all
Deinterlace all frames,
interlaced
Only deinterlace frames marked as interlaced.

Default value is all.

waveform

Video waveform monitor.

The waveform monitor plots color component intensity. By default luminance only. Each column of the waveform corresponds to a column of pixels in the source video.

It accepts the following options:

mode, m
Can be either "row", or "column". Default is "column". In row mode, the graph on the left side represents color component value 0 and the right side represents value = 255. In column mode, the top side represents color component value = 0 and bottom side represents value = 255.
intensity, i
Set intensity. Smaller values are useful to find out how many values of the same luminance are distributed across input rows/columns. Default value is 0.04. Allowed range is [0, 1].
mirror, r
Set mirroring mode. 0 means unmirrored, 1 means mirrored. In mirrored mode, higher values will be represented on the left side for "row" mode and at the top for "column" mode. Default is 1 (mirrored).
display, d
Set display mode. It accepts the following values:
overlay
Presents information identical to that in the "parade", except that the graphs representing color components are superimposed directly over one another.

This display mode makes it easier to spot relative differences or similarities in overlapping areas of the color components that are supposed to be identical, such as neutral whites, grays, or blacks.

stack
Display separate graph for the color components side by side in "row" mode or one below the other in "column" mode.
parade
Display separate graph for the color components side by side in "column" mode or one below the other in "row" mode.

Using this display mode makes it easy to spot color casts in the highlights and shadows of an image, by comparing the contours of the top and the bottom graphs of each waveform. Since whites, grays, and blacks are characterized by exactly equal amounts of red, green, and blue, neutral areas of the picture should display three waveforms of roughly equal width/height. If not, the correction is easy to perform by making level adjustments the three waveforms.

Default is "stack".

components, c
Set which color components to display. Default is 1, which means only luminance or red color component if input is in RGB colorspace. If is set for example to 7 it will display all 3 (if) available color components.
envelope, e
none
No envelope, this is default.
instant
Instant envelope, minimum and maximum values presented in graph will be easily visible even with small "step" value.
peak
Hold minimum and maximum values presented in graph across time. This way you can still spot out of range values without constantly looking at waveforms.
peak+instant
Peak and instant envelope combined together.
filter, f
lowpass
No filtering, this is default.
flat
Luma and chroma combined together.
aflat
Similar as above, but shows difference between blue and red chroma.
xflat
Similar as above, but use different colors.
chroma
Displays only chroma.
color
Displays actual color value on waveform.
acolor
Similar as above, but with luma showing frequency of chroma values.
graticule, g
Set which graticule to display.
none
Do not display graticule.
green
Display green graticule showing legal broadcast ranges.
orange
Display orange graticule showing legal broadcast ranges.
opacity, o
Set graticule opacity.
flags, fl
Set graticule flags.
numbers
Draw numbers above lines. By default enabled.
dots
Draw dots instead of lines.
scale, s
Set scale used for displaying graticule.
digital
millivolts
ire

Default is digital.

bgopacity, b
Set background opacity.

weave, doubleweave

The "weave" takes a field-based video input and join each two sequential fields into single frame, producing a new double height clip with half the frame rate and half the frame count.

The "doubleweave" works same as "weave" but without halving frame rate and frame count.

It accepts the following option:

first_field
Set first field. Available values are:
top, t
Set the frame as top-field-first.
bottom, b
Set the frame as bottom-field-first.

Examples

Interlace video using select and separatefields filter:

        separatefields,select=eq(mod(n,4),0)+eq(mod(n,4),3),weave
    

xbr

Apply the xBR high-quality magnification filter which is designed for pixel art. It follows a set of edge-detection rules, see <https://forums.libretro.com/t/xbr-algorithm-tutorial/123>.

It accepts the following option:

n
Set the scaling dimension: 2 for "2xBR", 3 for "3xBR" and 4 for "4xBR". Default is 3.

xstack

Stack video inputs into custom layout.

All streams must be of same pixel format.

The filter accept the following option:

inputs
Set number of input streams. Default is 2.
layout
Specify layout of inputs. This option requires the desired layout configuration to be explicitly set by the user. This sets position of each video input in output. Each input is separated by '|'. The first number represents the column, and the second number represents the row. Numbers start at 0 and are separated by '_'. Optionally one can use wX and hX, where X is video input from which to take width or height. Multiple values can be used when separated by '+'. In such case values are summed together.
shortest
If set to 1, force the output to terminate when the shortest input terminates. Default value is 0.

Examples

  • Display 4 inputs into 2x2 grid, note that if inputs are of different sizes unused gaps might appear, as not all of output video is used.

            xstack=inputs=4:layout=0_0|0_h0|w0_0|w0_h0
        
  • Display 4 inputs into 1x4 grid, note that if inputs are of different sizes unused gaps might appear, as not all of output video is used.

            xstack=inputs=4:layout=0_0|0_h0|0_h0+h1|0_h0+h1+h2
        
  • Display 9 inputs into 3x3 grid, note that if inputs are of different sizes unused gaps might appear, as not all of output video is used.

            xstack=inputs=9:layout=w3_0|w3_h0+h2|w3_h0|0_h4|0_0|w3+w1_0|0_h1+h2|w3+w1_h0|w3+w1_h1+h2
        

yadif

Deinterlace the input video ("yadif" means "yet another deinterlacing filter").

It accepts the following parameters:

mode
The interlacing mode to adopt. It accepts one of the following values:
0, send_frame
Output one frame for each frame.
1, send_field
Output one frame for each field.
2, send_frame_nospatial
Like "send_frame", but it skips the spatial interlacing check.
3, send_field_nospatial
Like "send_field", but it skips the spatial interlacing check.

The default value is "send_frame".

parity
The picture field parity assumed for the input interlaced video. It accepts one of the following values:
0, tff
Assume the top field is first.
1, bff
Assume the bottom field is first.
-1, auto
Enable automatic detection of field parity.

The default value is "auto". If the interlacing is unknown or the decoder does not export this information, top field first will be assumed.

deint
Specify which frames to deinterlace. Accept one of the following values:
0, all
Deinterlace all frames.
1, interlaced
Only deinterlace frames marked as interlaced.

The default value is "all".

yadif_cuda

Deinterlace the input video using the yadif 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.
2, send_frame_nospatial
Like "send_frame", but it skips the spatial interlacing check.
3, send_field_nospatial
Like "send_field", but it skips the spatial interlacing check.

The default value is "send_frame".

parity
The picture field parity assumed for the input interlaced video. It accepts one of the following values:
0, tff
Assume the top field is first.
1, bff
Assume the bottom field is first.
-1, auto
Enable automatic detection of field parity.

The default value is "auto". If the interlacing is unknown or the decoder does not export this information, top field first will be assumed.

deint
Specify which frames to deinterlace. Accept one of the following values:
0, all
Deinterlace all frames.
1, interlaced
Only deinterlace frames marked as interlaced.

The default value is "all".

zoompan

Apply Zoom & Pan effect.

This filter accepts the following options:

zoom, z
Set the zoom expression. Default is 1.
x
y
Set the x and y expression. Default is 0.
d
Set the duration expression in number of frames. This sets for how many number of frames effect will last for single input image.
s
Set the output image size, default is 'hd720'.
fps
Set the output frame rate, default is '25'.

Each expression can contain the following constants:

in_w, iw
Input width.
in_h, ih
Input height.
out_w, ow
Output width.
out_h, oh
Output height.
in
Input frame count.
on
Output frame count.
x
y
Last calculated 'x' and 'y' position from 'x' and 'y' expression for current input frame.
px
py
'x' and 'y' of last output frame of previous input frame or 0 when there was not yet such frame (first input frame).
zoom
Last calculated zoom from 'z' expression for current input frame.
pzoom
Last calculated zoom of last output frame of previous input frame.
duration
Number of output frames for current input frame. Calculated from 'd' expression for each input frame.
pduration
number of output frames created for previous input frame
a
Rational number: input width / input height
sar
sample aspect ratio
dar
display aspect ratio

Examples

  • Zoom-in up to 1.5 and pan at same time to some spot near center of picture:

            zoompan=z='min(zoom+0.0015,1.5)':d=700:x='if(gte(zoom,1.5),x,x+1/a)':y='if(gte(zoom,1.5),y,y+1)':s=640x360
        
  • Zoom-in up to 1.5 and pan always at center of picture:

            zoompan=z='min(zoom+0.0015,1.5)':d=700:x='iw/2-(iw/zoom/2)':y='ih/2-(ih/zoom/2)'
        
  • Same as above but without pausing:

            zoompan=z='min(max(zoom,pzoom)+0.0015,1.5)':d=1:x='iw/2-(iw/zoom/2)':y='ih/2-(ih/zoom/2)'
        

zscale

Scale (resize) the input video, using the z.lib library: <https://github.com/sekrit-twc/zimg>. To enable compilation of this filter, you need to configure FFmpeg with "--enable-libzimg".

The zscale filter forces the output display aspect ratio to be the same as the input, by changing the output sample aspect ratio.

If the input image format is different from the format requested by the next filter, the zscale filter will convert the input to the requested format.

Options

The filter accepts the following options.

width, w
height, h
Set the output video dimension expression. Default value is the input dimension.

If the width or w value is 0, the input width is used for the output. If the height or h value is 0, the input height is used for the output.

If one and only one of the values is -n with n >= 1, the zscale filter will use a value that maintains the aspect ratio of the input image, calculated from the other specified dimension. After that it will, however, make sure that the calculated dimension is divisible by n and adjust the value if necessary.

If both values are -n with n >= 1, the behavior will be identical to both values being set to 0 as previously detailed.

See below for the list of accepted constants for use in the dimension expression.

size, s
Set the video size. For the syntax of this option, check the "Video size" section in the ffmpeg-utils manual.
dither, d
Set the dither type.

Possible values are:

none
ordered
random
error_diffusion

Default is none.

filter, f
Set the resize filter type.

Possible values are:

point
bilinear
bicubic
spline16
spline36
lanczos

Default is bilinear.

range, r
Set the color range.

Possible values are:

input
limited
full

Default is same as input.

primaries, p
Set the color primaries.

Possible values are:

input
709
unspecified
170m
240m
2020

Default is same as input.

transfer, t
Set the transfer characteristics. <