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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:

Matches the stream with this index. E.g. "-threads:1 4" would set the thread count for the second stream to 4. If stream_index is used as an additional stream specifier (see below), then it selects stream number stream_index from the matching streams. Stream numbering is based on the order of the streams as detected by libavformat except when a program ID is also specified. In this case it is based on the ordering of the streams in the program.
stream_type is one of following: 'v' or 'V' for video, 'a' for audio, 's' for subtitle, 'd' for data, and 't' for attachments. 'v' matches all video streams, 'V' only matches video streams which are not attached pictures, video thumbnails or cover arts. If additional_stream_specifier is used, then it matches streams which both have this type and match the additional_stream_specifier. Otherwise, it matches all streams of the specified type.
Matches streams which are in the program with the id program_id. If additional_stream_specifier is used, then it matches streams which both are part of the program and match the additional_stream_specifier.
#stream_id or i:stream_id
Match the stream by stream id (e.g. PID in MPEG-TS container).
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.
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.

Show license.
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:

Print advanced tool options in addition to the basic tool options.
Print complete list of options, including shared and private options for encoders, decoders, demuxers, muxers, filters, etc.
Print detailed information about the decoder named decoder_name. Use the -decoders option to get a list of all decoders.
Print detailed information about the encoder named encoder_name. Use the -encoders option to get a list of all encoders.
Print detailed information about the demuxer named demuxer_name. Use the -formats option to get a list of all demuxers and muxers.
Print detailed information about the muxer named muxer_name. Use the -formats option to get a list of all muxers and demuxers.
Print detailed information about the filter named filter_name. Use the -filters option to get a list of all filters.
Print detailed information about the bitstream filter named bitstream_filter_name. Use the -bsfs option to get a list of all bitstream filters.
Print detailed information about the protocol named protocol_name. Use the -protocols option to get a list of all protocols.
Show version.
Show the build configuration, one option per line.
Show available formats (including devices).
Show available demuxers.
Show available muxers.
Show available devices.
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.

Show available decoders.
Show all available encoders.
Show available bitstream filters.
Show available protocols.
Show available libavfilter filters.
Show available pixel formats.
Show available sample formats.
Show channel names and standard channel layouts.
Show recognized color names.
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
    
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
    
Set logging level and flags used by the library.

The optional flags prefix can consist of the following values:

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

Show nothing at all; be silent.
Only show fatal errors which could lead the process to crash, such as an assertion failure. This is not currently used for anything.
Only show fatal errors. These are errors after which the process absolutely cannot continue.
Show all errors, including ones which can be recovered from.
Show all warnings and errors. Any message related to possibly incorrect or unexpected events will be shown.
Show informative messages during processing. This is in addition to warnings and errors. This is the default value.
Same as "info", except more verbose.
Show everything, including debugging information.

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

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

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

        ffmpeg [...] -loglevel +repeat

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

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

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

The following options are recognized:

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

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.

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:

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

AVOptions

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

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

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

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

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

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

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

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

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

Main options

Force displayed width.
Force displayed height.
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.
Start in fullscreen mode.
Disable audio.
Disable video.
Disable subtitles.
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.

Play duration seconds of audio/video.

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

Seek by bytes.
Set custom interval, in seconds, for seeking using left/right keys. Default is 10 seconds.
Disable graphical display.
Borderless window.
Window always on top. Available on: X11 with SDL >= 2.0.5, Windows SDL >= 2.0.6.
-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.
Force format.
Set window title (default is the input filename).
Set the x position for the left of the window (default is a centered window).
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.
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.

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.

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).
Read input_url.

Advanced options

Set pixel format. This option has been deprecated in favor of private options, try -pixel_format.
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 shown by default, unless the log level is lower than "info". Its display can be forced by manually specifying this option. To disable it, you need to specify "-nostats".
Non-spec-compliant optimizations.
Generate pts.
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.
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.
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.
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.
Exit when video is done playing.
Exit if any key is pressed.
Exit if any mouse button is pressed.
Force a specific decoder implementation for the stream identified by media_specifier, which can assume the values "a" (audio), "v" (video), and "s" subtitle.
Force a specific audio decoder.
Force a specific video decoder.
Force a specific subtitle decoder.
Automatically rotate the video according to file metadata. Enabled by default, use -noautorotate to disable it.
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.
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.
Defines how many threads are used to process a filter pipeline. Each pipeline will produce a thread pool with this many threads available for parallel processing. The default is 0 which means that the thread count will be determined by the number of available CPUs.

While playing

Quit.
Toggle full screen.
Pause.
Toggle mute.
9, 0
Decrease and increase volume respectively.
/, *
Decrease and increase volume respectively.
Cycle audio channel in the current program.
Cycle video channel.
Cycle subtitle channel in the current program.
Cycle program.
Cycle video filters or show modes.
Step to the next frame.

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

Seek backward/forward 10 seconds.
Seek backward/forward 1 minute.
Seek to the previous/next chapter. or if there are no chapters Seek backward/forward 10 minutes.
Seek to percentage in file corresponding to fraction of width.
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|ms|us]

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

In both expressions, the optional - indicates negative duration.

Examples

The following examples are all valid time duration:

55
55 seconds
0.2
0.2 seconds
200ms
200 milliseconds, that's 0.2s
200000us
200000 microseconds, that's 0.2s
12:03:45
12 hours, 03 minutes and 45 seconds
23.189
23.189 seconds

Video size

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

The following abbreviations are recognized:

720x480
720x576
352x240
352x288
640x480
768x576
352x240
352x240
128x96
176x144
352x288
4cif
704x576
16cif
1408x1152
160x120
320x240
640x480
800x600
1024x768
1600x1200
2048x1536
1280x1024
2560x2048
5120x4096
852x480
1366x768
1600x1024
1920x1200
2560x1600
3200x2048
3840x2400
6400x4096
7680x4800
320x200
640x350
852x480
1280x720
1920x1080
2k
2048x1080
2kflat
1998x1080
2kscope
2048x858
4k
4096x2160
4kflat
3996x2160
4kscope
4096x1716
640x360
240x160
400x240
432x240
480x320
960x540
2kdci
2048x1080
4kdci
4096x2160
3840x2160
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:

30000/1001
25/1
30000/1001
25/1
30000/1001
25/1
24/1
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:

0xF0F8FF
0xFAEBD7
0x00FFFF
0x7FFFD4
0xF0FFFF
0xF5F5DC
0xFFE4C4
0x000000
0xFFEBCD
0x0000FF
0x8A2BE2
0xA52A2A
0xDEB887
0x5F9EA0
0x7FFF00
0xD2691E
0xFF7F50
0x6495ED
0xFFF8DC
0xDC143C
0x00FFFF
0x00008B
0x008B8B
0xB8860B
0xA9A9A9
0x006400
0xBDB76B
0x8B008B
0x556B2F
0xFF8C00
0x9932CC
0x8B0000
0xE9967A
0x8FBC8F
0x483D8B
0x2F4F4F
0x00CED1
0x9400D3
0xFF1493
0x00BFFF
0x696969
0x1E90FF
0xB22222
0xFFFAF0
0x228B22
0xFF00FF
0xDCDCDC
0xF8F8FF
0xFFD700
0xDAA520
0x808080
0x008000
0xADFF2F
0xF0FFF0
0xFF69B4
0xCD5C5C
0x4B0082
0xFFFFF0
0xF0E68C
0xE6E6FA
0xFFF0F5
0x7CFC00
0xFFFACD
0xADD8E6
0xF08080
0xE0FFFF
0xFAFAD2
0x90EE90
0xD3D3D3
0xFFB6C1
0xFFA07A
0x20B2AA
0x87CEFA
0x778899
0xB0C4DE
0xFFFFE0
0x00FF00
0x32CD32
0xFAF0E6
0xFF00FF
0x800000
0x66CDAA
0x0000CD
0xBA55D3
0x9370D8
0x3CB371
0x7B68EE
0x00FA9A
0x48D1CC
0xC71585
0x191970
0xF5FFFA
0xFFE4E1
0xFFE4B5
0xFFDEAD
0x000080
0xFDF5E6
0x808000
0x6B8E23
0xFFA500
0xFF4500
0xDA70D6
0xEEE8AA
0x98FB98
0xAFEEEE
0xD87093
0xFFEFD5
0xFFDAB9
0xCD853F
0xFFC0CB
0xDDA0DD
0xB0E0E6
0x800080
0xFF0000
0xBC8F8F
0x4169E1
0x8B4513
0xFA8072
0xF4A460
0x2E8B57
0xFFF5EE
0xA0522D
0xC0C0C0
0x87CEEB
0x6A5ACD
0x708090
0xFFFAFA
0x00FF7F
0x4682B4
0xD2B48C
0x008080
0xD8BFD8
0xFF6347
0x40E0D0
0xEE82EE
0xF5DEB3
0xFFFFFF
0xF5F5F5
0xFFFF00
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:

front left
front right
front center
low frequency
back left
back right
front left-of-center
front right-of-center
back center
side left
side right
top center
top front left
top front center
top front right
top back left
top back center
top back right
downmix left
downmix right
wide left
wide right
surround direct left
surround direct right
low frequency 2

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

FC
FL+FR
2.1
FL+FR+LFE
3.0
FL+FR+FC
3.0(back)
FL+FR+BC
4.0
FL+FR+FC+BC
FL+FR+BL+BR
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
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
FL+FR+FC+BL+BR+BC+SL+SR
FL+FR+FC+BL+BR+BC+SL+SR+WL+WR+TBL+TBR+TBC+TFC+TFL+TFR
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:

Compute absolute value of x.
Compute arccosine of x.
Compute arcsine of x.
Compute arctangent of x.
Compute principal value of the arc tangent of y/x.
Return 1 if x is greater than or equal to min and lesser than or equal to max, 0 otherwise.
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).

Round the value of expression expr upwards to the nearest integer. For example, "ceil(1.5)" is "2.0".
Return the value of x clipped between min and max.
Compute cosine of x.
Compute hyperbolic cosine of x.
Return 1 if x and y are equivalent, 0 otherwise.
Compute exponential of x (with base "e", the Euler's number).
Round the value of expression expr downwards to the nearest integer. For example, "floor(-1.5)" is "-2.0".
Compute Gauss function of x, corresponding to "exp(-x*x/2) / sqrt(2*PI)".
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.
Return 1 if x is greater than y, 0 otherwise.
Return 1 if x is greater than or equal to y, 0 otherwise.
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.
Evaluate x, and if the result is non-zero return the result of the evaluation of y, return 0 otherwise.
Evaluate x, and if the result is non-zero return the evaluation result of y, otherwise the evaluation result of z.
Evaluate x, and if the result is zero return the result of the evaluation of y, return 0 otherwise.
Evaluate x, and if the result is zero return the evaluation result of y, otherwise the evaluation result of z.
Return 1.0 if x is +/-INFINITY, 0.0 otherwise.
Return 1.0 if x is NAN, 0.0 otherwise.
Load the value of the internal variable with number var, which was previously stored with st(var, expr). The function returns the loaded value.
Return linear interpolation between x and y by amount of z.
Compute natural logarithm of x.
Return 1 if x is lesser than y, 0 otherwise.
Return 1 if x is lesser than or equal to y, 0 otherwise.
Return the maximum between x and y.
Return the minimum between x and y.
Compute the remainder of division of x by y.
Return 1.0 if expr is zero, 0.0 otherwise.
Compute the power of x elevated y, it is equivalent to "(x)^(y)".
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

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.
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 the value of expression expr to the nearest integer. For example, "round(1.5)" is "2.0".
Compute sign of x.
Compute sine of x.
Compute hyperbolic sine of x.
Compute the square root of expr. This is equivalent to "(expr)^.5".
Compute expression "1/(1 + exp(4*x))".
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.
Compute tangent of x.
Compute hyperbolic tangent of x.
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.

Return the current (wallclock) time in seconds.
Round the value of expression expr towards zero to the nearest integer. For example, "trunc(-1.5)" is "-1.0".
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:

area of the unit disc, approximately 3.14
exp(1) (Euler's number), approximately 2.718
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.

10^-24 / 2^-80
10^-21 / 2^-70
10^-18 / 2^-60
10^-15 / 2^-50
10^-12 / 2^-40
10^-9 / 2^-30
10^-6 / 2^-20
10^-3 / 2^-10
10^-2
10^-1
10^2
10^3 / 2^10
10^3 / 2^10
10^6 / 2^20
10^9 / 2^30
10^12 / 2^40
10^15 / 2^40
10^18 / 2^50
10^21 / 2^60
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:

Set bitrate in bits/s. Default value is 200K.
Set audio bitrate (in bits/s). Default value is 128K.
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.
Set generic flags.

Possible values:

Use four motion vector by macroblock (mpeg4).
Use 1/4 pel motion compensation.
loop
Use loop filter.
Use fixed qscale.
Use internal 2pass ratecontrol in first pass mode.
Use internal 2pass ratecontrol in second pass mode.
Only decode/encode grayscale.
psnr
Set error[?] variables during encoding.
Input bitstream might be randomly truncated.
Don't output frames whose parameters differ from first decoded frame in stream. Error AVERROR_INPUT_CHANGED is returned when a frame is dropped.
Use interlaced DCT.
Force low delay.
Place global headers in extradata instead of every keyframe.
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.
Apply H263 advanced intra coding / mpeg4 ac prediction.
Apply interlaced motion estimation.
Use closed gop.
Output even potentially corrupted frames.
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.

Set the group of picture (GOP) size. Default value is 12.
Set audio sampling rate (in Hz).
Set number of audio channels.
Set cutoff bandwidth. (Supported only by selected encoders, see their respective documentation sections.)
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.

Set the frame number.
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.
Set video quantizer scale blur (VBR).
Set min video quantizer scale (VBR). Must be included between -1 and 69, default value is 2.
Set max video quantizer scale (VBR). Must be included between -1 and 1024, default value is 31.
Set max difference between the quantizer scale (VBR).
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.

Set qp factor between P and B frames.
Set strategy to choose between I/P/B-frames.
Set RTP payload size in bytes.
Workaround not auto detected encoder bugs.

Possible values:

Xvid interlacing bug (autodetected if fourcc==XVIX)
(autodetected if fourcc==UMP4)
padding bug (autodetected)
old standard qpel (autodetected per fourcc/version)
direct-qpel-blocksize bug (autodetected per fourcc/version)
edge padding bug (autodetected per fourcc/version)
Workaround various bugs in microsoft broken decoders.
trancated frames
Specify how strictly to follow the standards.

Possible values:

strictly conform to an older more strict version of the spec or reference software
strictly conform to all the things in the spec no matter what consequences
allow unofficial extensions
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.
Set QP offset between P and B frames.
Set error detection flags.

Possible values:

verify embedded CRCs
detect bitstream specification deviations
buffer
detect improper bitstream length
abort decoding on minor error detection
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.
consider things that violate the spec and have not been seen in the wild as errors
consider all spec non compliancies as errors
consider things that a sane encoder should not do as an error
Use MPEG quantizers instead of H.263.
Set max bitrate tolerance (in bits/s). Requires bufsize to be set.
Set min bitrate tolerance (in bits/s). Most useful in setting up a CBR encode. It is of little use elsewise.
Set ratecontrol buffer size (in bits).
Set QP factor between P and I frames.
Set QP offset between P and I frames.
Set DCT algorithm.

Possible values:

autoselect a good one (default)
fast integer
accurate integer
floating point AAN DCT
Compress bright areas stronger than medium ones.
Set temporal complexity masking.
Set spatial complexity masking.
Set inter masking.
Compress dark areas stronger than medium ones.
Select IDCT implementation.

Possible values:

Automatically pick a IDCT compatible with the simple one
floating point AAN IDCT
Set error concealment strategy.

Possible values:

iterative motion vector (MV) search (slow)
deblock
use strong deblock filter for damaged MBs
favor predicting from the previous frame instead of the current
Set prediction method.

Possible values:

median
Set sample aspect ratio.
Set sample aspect ratio. Alias to aspect.
Print specific debug info.

Possible values:

picture info
rate control
macroblock (MB) type
qp
per-block quantization parameter (QP)
display complexity metadata for the upcoming frame, GoP or for a given duration.
error recognition
memory management control operations (H.264)
picture buffer allocations
threading operations
skip motion compensation
Set full pel me compare function.

Possible values:

sum of absolute differences, fast (default)
sum of squared errors
sum of absolute Hadamard transformed differences
sum of absolute DCT transformed differences
psnr
sum of squared quantization errors (avoid, low quality)
number of bits needed for the block
rate distortion optimal, slow
0
sum of absolute vertical differences
sum of squared vertical differences
noise preserving sum of squared differences
5/3 wavelet, only used in snow
9/7 wavelet, only used in snow
Set sub pel me compare function.

Possible values:

sum of absolute differences, fast (default)
sum of squared errors
sum of absolute Hadamard transformed differences
sum of absolute DCT transformed differences
psnr
sum of squared quantization errors (avoid, low quality)
number of bits needed for the block
rate distortion optimal, slow
0
sum of absolute vertical differences
sum of squared vertical differences
noise preserving sum of squared differences
5/3 wavelet, only used in snow
9/7 wavelet, only used in snow
Set macroblock compare function.

Possible values:

sum of absolute differences, fast (default)
sum of squared errors
sum of absolute Hadamard transformed differences
sum of absolute DCT transformed differences
psnr
sum of squared quantization errors (avoid, low quality)
number of bits needed for the block
rate distortion optimal, slow
0
sum of absolute vertical differences
sum of squared vertical differences
noise preserving sum of squared differences
5/3 wavelet, only used in snow
9/7 wavelet, only used in snow
Set interlaced dct compare function.

Possible values:

sum of absolute differences, fast (default)
sum of squared errors
sum of absolute Hadamard transformed differences
sum of absolute DCT transformed differences
psnr
sum of squared quantization errors (avoid, low quality)
number of bits needed for the block
rate distortion optimal, slow
0
sum of absolute vertical differences
sum of squared vertical differences
noise preserving sum of squared differences
5/3 wavelet, only used in snow
9/7 wavelet, only used in snow
Set diamond type & size for motion estimation.
(1024, INT_MAX)
full motion estimation(slowest)
(768, 1024]
umh motion estimation
(512, 768]
hex motion estimation
(256, 512]
l2s diamond motion estimation
[2,256]
var diamond motion estimation
(-1, 2)
small diamond motion estimation
-1
funny diamond motion estimation
(INT_MIN, -1)
sab diamond motion estimation
Set amount of motion predictors from the previous frame.
Set pre motion estimation.
Set pre motion estimation compare function.

Possible values:

sum of absolute differences, fast (default)
sum of squared errors
sum of absolute Hadamard transformed differences
sum of absolute DCT transformed differences
psnr
sum of squared quantization errors (avoid, low quality)
number of bits needed for the block
rate distortion optimal, slow
0
sum of absolute vertical differences
sum of squared vertical differences
noise preserving sum of squared differences
5/3 wavelet, only used in snow
9/7 wavelet, only used in snow
Set diamond type & size for motion estimation pre-pass.
Set sub pel motion estimation quality.
Set limit motion vectors range (1023 for DivX player).
Possible values:
variable length coder / huffman coder
arithmetic coder
raw (no encoding)
run-length coder
Set context model.
Set macroblock decision algorithm (high quality mode).

Possible values:

use mbcmp (default)
use fewest bits
use best rate distortion
Set scene change threshold.
Set noise reduction.
Set number of bits which should be loaded into the rc buffer before decoding starts.
Possible values:
Allow non spec compliant speedup tricks.
Skip bitstream encoding.
Ignore cropping information from sps.
Place global headers at every keyframe instead of in extradata.
Frame data might be split into multiple chunks.
Show all frames before the first keyframe.
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.
Do not skip samples and export skip information as frame side data.
Do not reset ASS ReadOrder field on flush.
Possible values:
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.
Export encoder Producer Reference Time into packet side-data (see "AV_PKT_DATA_PRFT") for codecs that support it.
Export video encoding parameters through frame side data (see "AV_FRAME_DATA_VIDEO_ENC_PARAMS") for codecs that support it. At present, those are H.264 and VP9.
Export film grain parameters through frame side data (see "AV_FRAME_DATA_FILM_GRAIN_PARAMS"). Supported at present by AV1 decoders.
Set the number of threads to be used, in case the selected codec implementation supports multi-threading.

Possible values:

automatically select the number of threads to set

Default value is auto.

Set intra_dc_precision.
Set nsse weight.
Set number of macroblock rows at the top which are skipped.
Set number of macroblock rows at the bottom which are skipped.
Set encoder codec profile. Default value is unknown. Encoder specific profiles are documented in the relevant encoder documentation.
Possible values:
Decode at 1= 1/2, 2=1/4, 3=1/8 resolutions.
Set frame skip threshold.
Set frame skip factor.
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.
Set frame skip compare function.

Possible values:

sum of absolute differences, fast (default)
sum of squared errors
sum of absolute Hadamard transformed differences
sum of absolute DCT transformed differences
psnr
sum of squared quantization errors (avoid, low quality)
number of bits needed for the block
rate distortion optimal, slow
0
sum of absolute vertical differences
sum of squared vertical differences
noise preserving sum of squared differences
5/3 wavelet, only used in snow
9/7 wavelet, only used in snow
Set min macroblock lagrange factor (VBR).
Set max macroblock lagrange factor (VBR).
Set motion estimation bitrate penalty compensation (1.0 = 256).
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:

Discard no frame.
Discard useless frames like 0-sized frames.
Discard all non-reference frames.
Discard all bidirectional frames.
Discard all frames excepts keyframes.
Discard all frames except I frames.
Discard all frames.

Default value is default.

Refine the two motion vectors used in bidirectional macroblocks.
Downscale frames for dynamic B-frame decision.
Set minimum interval between IDR-frames.
Set reference frames to consider for motion compensation.
Set chroma qp offset from luma.
Set rate-distortion optimal quantization.
Adjust sensitivity of b_frame_strategy 1.
Set GOP timecode frame start number, in non drop frame format.
Possible values:
Possible values:
Possible values:
BT.709
BT.470 M
BT.470 BG
SMPTE 170 M
SMPTE 240 M
Film
BT.2020
SMPTE ST 428-1
SMPTE 431-2
SMPTE 432-1
JEDEC P22
Possible values:
BT.709
BT.470 M
BT.470 BG
SMPTE 170 M
SMPTE 240 M
Linear
Log
Log square root
IEC 61966-2-4
BT.1361
IEC 61966-2-1
BT.2020 - 10 bit
BT.2020 - 12 bit
SMPTE ST 2084
SMPTE ST 428-1
ARIB STD-B67
colorspace integer (decoding/encoding,video)
Possible values:
RGB
BT.709
FCC
BT.470 BG
SMPTE 170 M
SMPTE 240 M
YCOCG
BT.2020 NCL
BT.2020 CL
SMPTE 2085
Chroma-derived NCL
Chroma-derived CL
ICtCp
If used as input parameter, it serves as a hint to the decoder, which color_range the input has. Possible values:
MPEG (219*2^(n-8))
JPEG (2^n-1)
Possible values:
Set the log level offset.
Number of slices, used in parallelized encoding.
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:

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

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

Decode more than one frame at once.

Default value is slice+frame.

Set audio service type.

Possible values:

Main Audio Service
Effects
Visually Impaired
Hearing Impaired
Dialogue
Commentary
Emergency
Voice Over
Karaoke
Set sample format audio decoders should prefer. Default value is "none".
Set the input subtitles character encoding.
Set/override the field order of the video. Possible values:
Progressive video
Interlaced video, top field coded and displayed first
Interlaced video, bottom field coded and displayed first
Interlaced video, top coded first, bottom displayed first
Interlaced video, bottom coded first, top displayed first
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.
"," separated list of allowed decoders. By default all are allowed.
Separator used to separate the fields printed on the command line about the Stream parameters. For example, to separate the fields with newlines and indentation:

        ffprobe -dump_separator "
                                  "  -i ~/videos/matrixbench_mpeg2.mpg
    
Maximum number of pixels per image. This value can be used to avoid out of memory failures due to large images.
Enable cropping if cropping parameters are multiples of the required alignment for the left and top parameters. If the alignment is not met the cropping will be partially applied to maintain alignment. Default is 1 (enabled). Note: The required alignment depends on if "AV_CODEC_FLAG_UNALIGNED" is set and the CPU. "AV_CODEC_FLAG_UNALIGNED" cannot be changed from the command line. Also hardware decoders will not apply left/top Cropping.

DECODERS

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

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

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

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

VIDEO DECODERS

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

av1

AOMedia Video 1 (AV1) decoder.

Options

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

rawvideo

Raw video decoder.

This decoder decodes rawvideo streams.

Options

Specify the assumed field type of the input video.
-1
the video is assumed to be progressive (default)
0
bottom-field-first is assumed
1
top-field-first is assumed

libdav1d

dav1d AV1 decoder.

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

Options

The following options are supported by the libdav1d wrapper.

Set amount of frame threads to use during decoding. The default value is 0 (autodetect).
Set amount of tile threads to use during decoding. The default value is 0 (autodetect).
Apply film grain to the decoded video if present in the bitstream. Defaults to the internal default of the library.
Select an operating point of a scalable AV1 bitstream (0 - 31). Defaults to the internal default of the library.
Output all spatial layers of a scalable AV1 bitstream. The default value is false.

libdavs2

AVS2-P2/IEEE1857.4 video decoder wrapper.

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

libuavs3d

AVS3-P2/IEEE1857.10 video decoder.

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

Options

The following option is supported by the libuavs3d wrapper.

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

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

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

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.

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

libopencore-amrnb

libopencore-amrnb decoder wrapper.

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

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

libopencore-amrwb

libopencore-amrwb decoder wrapper.

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

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

libopus

libopus decoder wrapper.

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

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

SUBTITLES DECODERS

libaribb24

ARIB STD-B24 caption decoder.

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

libaribb24 Decoder Options

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

Unset by default.

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

Enabled by default.

dvbsub

Options

-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.
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

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

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

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

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 *.
Set default character set used for decoding, a value between 0 and 87 (see ETS 300 706, Section 15, Table 32). Default value is -1, which does not override the libzvbi default. This option is needed for some legacy level 1.0 transmissions which cannot signal the proper charset.
Discards the top teletext line. Default value is 1.
Specifies the format of the decoded subtitles.
The default format, you should use this for teletext pages, because certain graphics and colors cannot be expressed in simple text or even ASS.
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.
X offset of generated bitmaps, default is 0.
Y offset of generated bitmaps, default is 0.
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.
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.
Force transparent background of the generated teletext bitmaps. Default value is 0 which means an opaque background.
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.

Insert or remove temporal delimiter OBUs in all temporal units of the stream.
Insert a TD at the beginning of every TU which does not already have one.
Remove the TD from the beginning of every TU which has one.
Set the color description fields in the stream (see AV1 section 6.4.2).
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).
Limited range.
Full range.
Set the chroma sample location in the stream (see AV1 section 6.4.2). This can only be set for 4:2:0 streams.
Left position (matching the default in MPEG-2 and H.264).
Top-left position.
Set the tick rate (num_units_in_display_tick / time_scale) in the timing info in the sequence header.
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.
Deletes Padding OBUs.

chomp

Remove zero padding at the end of a packet.

dca_core

Extract the core from a DCA/DTS stream, dropping extensions such as DTS-HD.

dump_extra

Add extradata to the beginning of the filtered packets except when said packets already exactly begin with the extradata that is intended to be added.

The additional argument specifies which packets should be filtered. It accepts the values:
add extradata to all key packets
add extradata to all packets

If not specified it is assumed k.

For example the following ffmpeg command forces a global header (thus disabling individual packet headers) in the H.264 packets generated by the "libx264" encoder, but corrects them by adding the header stored in extradata to the key packets:

        ffmpeg -i INPUT -map 0 -flags:v +global_header -c:v libx264 -bsf:v dump_extra out.ts

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.

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.

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 '-'.
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.

Specifies the texture to keep.

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.

Insert or remove AUD NAL units in all access units of the stream.
Set the sample aspect ratio of the stream in the VUI parameters.
Set whether the stream is suitable for display using overscan or not (see H.264 section E.2.1).
Set the video format in the stream (see H.264 section E.2.1 and table E-2).
Set the colour description in the stream (see H.264 section E.2.1 and tables E-3, E-4 and E-5).
Set the chroma sample location in the stream (see H.264 section E.2.1 and figure E-1).
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).
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).
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).

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.

Deletes both filler NAL units and filler SEI messages.
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.

Insert or remove AUD NAL units in all access units of the stream.
Set the sample aspect ratio in the stream in the VUI parameters.
Set the video format in the stream (see H.265 section E.3.1 and table E.2).
Set the colour description in the stream (see H.265 section E.3.1 and tables E.3, E.4 and E.5).
Set the chroma sample location in the stream (see H.265 section E.3.1 and figure E.1).
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.
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.
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).

Set the level in the VPS and SPS. See H.265 section A.4 and tables A.6 and A.7.

The argument must be the name of a level (for example, 5.1), a general_level_idc value (for example, 153 for level 5.1), or the special name auto indicating that the filter should attempt to guess the level from the input stream properties.

hevc_mp4toannexb

Convert an HEVC/H.265 bitstream from length prefixed mode to start code prefixed mode (as defined in the Annex B of the ITU-T H.265 specification).

This is required by some streaming formats, typically the MPEG-2 transport stream format (muxer "mpegts").

For example to remux an MP4 file containing an HEVC stream to mpegts format with ffmpeg, you can use the command:

        ffmpeg -i INPUT.mp4 -codec copy -bsf:v hevc_mp4toannexb OUTPUT.ts

Please note that this filter is auto-inserted for MPEG-TS (muxer "mpegts") and raw HEVC/H.265 (muxer "h265" or "hevc") output formats.

imxdump

Modifies the bitstream to fit in MOV and to be usable by the Final Cut Pro decoder. This filter only applies to the mpeg2video codec, and is likely not needed for Final Cut Pro 7 and newer with the appropriate -tag:v.

For example, to remux 30 MB/sec NTSC IMX to MOV:

        ffmpeg -i input.mxf -c copy -bsf:v imxdump -tag:v mx3n output.mov

mjpeg2jpeg

Convert MJPEG/AVI1 packets to full JPEG/JFIF packets.

MJPEG is a video codec wherein each video frame is essentially a JPEG image. The individual frames can be extracted without loss, e.g. by

        ffmpeg -i ../some_mjpeg.avi -c:v copy frames_%d.jpg

Unfortunately, these chunks are incomplete JPEG images, because they lack the DHT segment required for decoding. Quoting from <http://www.digitalpreservation.gov/formats/fdd/fdd000063.shtml>:

Avery Lee, writing in the rec.video.desktop newsgroup in 2001, commented that "MJPEG, or at least the MJPEG in AVIs having the MJPG fourcc, is restricted JPEG with a fixed -- and *omitted* -- Huffman table. The JPEG must be YCbCr colorspace, it must be 4:2:2, and it must use basic Huffman encoding, not arithmetic or progressive. . . . You can indeed extract the MJPEG frames and decode them with a regular JPEG decoder, but you have to prepend the DHT segment to them, or else the decoder won't have any idea how to decompress the data. The exact table necessary is given in the OpenDML spec."

This bitstream filter patches the header of frames extracted from an MJPEG stream (carrying the AVI1 header ID and lacking a DHT segment) to produce fully qualified JPEG images.

        ffmpeg -i mjpeg-movie.avi -c:v copy -bsf:v mjpeg2jpeg frame_%d.jpg
        exiftran -i -9 frame*.jpg
        ffmpeg -i frame_%d.jpg -c:v copy rotated.avi

mjpegadump

Add an MJPEG A header to the bitstream, to enable decoding by Quicktime.

mov2textsub

Extract a representable text file from MOV subtitles, stripping the metadata header from each subtitle packet.

See also the text2movsub filter.

mp3decomp

Decompress non-standard compressed MP3 audio headers.

mpeg2_metadata

Modify metadata embedded in an MPEG-2 stream.

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).

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).
Set the video format in the stream (see H.262 section 6.3.6 and table 6-6).
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:

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.
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.

pcm_rechunk

Repacketize PCM audio to a fixed number of samples per packet or a fixed packet rate per second. This is similar to the asetnsamples audio filter but works on audio packets instead of audio frames.

Set the number of samples per each output audio packet. The number is intended as the number of samples per each channel. Default value is 1024.
If set to 1, the filter will pad the last audio packet with silence, so that it will contain the same number of samples (or roughly the same number of samples, see frame_rate) as the previous ones. Default value is 1.
This option makes the filter output a fixed number of packets per second instead of a fixed number of samples per packet. If the audio sample rate is not divisible by the frame rate then the number of samples will not be constant but will vary slightly so that each packet will start as close to the frame boundary as possible. Using this option has precedence over nb_out_samples.

You can generate the well known 1602-1601-1602-1601-1602 pattern of 48kHz audio for NTSC frame rate using the frame_rate option.

        ffmpeg -f lavfi -i sine=r=48000:d=1 -c pcm_s16le -bsf pcm_rechunk=r=30000/1001 -f framecrc -

prores_metadata

Modify color property metadata embedded in prores stream.

Set the color primaries. Available values are:
Keep the same color primaries property (default).
BT601 625
BT601 525
DCI P3
P3 D65
Set the color transfer. Available values are:
Keep the same transfer characteristics property (default).
BT 601, BT 709, BT 2020
SMPTE ST 2084
ARIB STD-B67
Set the matrix coefficient. Available values are:
Keep the same colorspace property (default).
BT 601

Set Rec709 colorspace for each frame of the file

        ffmpeg -i INPUT -c copy -bsf:v prores_metadata=color_primaries=bt709:color_trc=bt709:colorspace=bt709 output.mov

Set Hybrid Log-Gamma parameters for each frame of the file

        ffmpeg -i INPUT -c copy -bsf:v prores_metadata=color_primaries=bt2020:color_trc=arib-std-b67:colorspace=bt2020nc output.mov

remove_extra

Remove extradata from packets.

It accepts the following parameter:

Set which frame types to remove extradata from.
Remove extradata from non-keyframes only.
Remove extradata from keyframes only.
Remove extradata from all frames.

setts

Set PTS and DTS in packets.

It accepts the following parameters:

Set expressions for PTS, DTS or both.

The expressions are evaluated through the eval API and can contain the following constants:

The count of the input packet. Starting from 0.
The demux timestamp in input in case of "ts" or "dts" option or presentation timestamp in case of "pts" option.
The original position in the file of the packet, or undefined if undefined for the current packet
The demux timestamp in input.
The presentation timestamp in input.
The DTS of the first packet.
The PTS of the first packet.
The previous input DTS.
The previous input PTS.
The previous output DTS.
The previous output PTS.
The timebase of stream packet belongs.
The sample rate of stream packet belongs.

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 AV1, H.264, H.265, (M)JPEG, MPEG-2 and VP9, but depending on the build only a subset of these may be available.

truehd_core

Extract the core from a TrueHD stream, dropping ATMOS data.

vp9_metadata

Modify metadata embedded in a VP9 stream.

Set the color space value in the frame header. Note that any frame set to RGB will be implicitly set to PC range and that RGB is incompatible with profiles 0 and 2.
Set the color range value in the frame header. Note that any value imposed by the color space will take precedence over this value.

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:

Possible values:
Reduce buffering.
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.
Set the maximum number of buffered packets when probing a codec. Default is 2500 packets.
Set packet size.
Set format flags. Some are implemented for a limited number of formats.

Possible values for input files:

Discard corrupted packets.
Enable fast, but inaccurate seeks for some formats.
Generate missing PTS if DTS is present.
Ignore DTS if PTS is set. Inert when nofillin is set.
Ignore index.
Reduce the latency introduced by buffering during initial input streams analysis.
Do not fill in missing values in packet fields that can be exactly calculated.
Disable AVParsers, this needs "+nofillin" too.
Try to interleave output packets by DTS. At present, available only for AVIs with an index.

Possible values for output files:

Automatically apply bitstream filters as required by the output format. Enabled by default.
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.
Write out packets immediately.
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.
Allow seeking to non-keyframes on demuxer level when supported if set to 1. Default is 0.
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.
Set decryption key.
Set max memory used for timestamp index (per stream).
Set max memory used for buffering real-time frames.
Print specific debug info.

Possible values:

Set maximum muxing or demuxing delay in microseconds.
Set number of frames used to probe fps.
Set microseconds by which audio packets should be interleaved earlier.
Set microseconds for each chunk.
Set size in bytes for each chunk.
Set error detection flags. "f_err_detect" is deprecated and should be used only via the ffmpeg tool.

Possible values:

Verify embedded CRCs.
Detect bitstream specification deviations.
buffer
Detect improper bitstream length.
Abort decoding on minor error detection.
Consider things that violate the spec and have not been seen in the wild as errors.
Consider all spec non compliancies as errors.
Consider things that a sane encoder should not do as an error.
Set maximum buffering duration for interleaving. The duration is expressed in microseconds, and defaults to 10000000 (10 seconds).

To ensure all the streams are interleaved correctly, libavformat will wait until it has at least one packet for each stream before actually writing any packets to the output file. When some streams are "sparse" (i.e. there are large gaps between successive packets), this can result in excessive buffering.

This field specifies the maximum difference between the timestamps of the first and the last packet in the muxing queue, above which libavformat will output a packet regardless of whether it has queued a packet for all the streams.

If set to 0, libavformat will continue buffering packets until it has a packet for each stream, regardless of the maximum timestamp difference between the buffered packets.

Use wallclock as timestamps if set to 1. Default is 0.
Possible values:
Shift timestamps to make them non-negative. Also note that this affects only leading negative timestamps, and not non-monotonic negative timestamps.
Shift timestamps so that the first timestamp is 0.
Enables shifting when required by the target format.
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.

Set number of bytes to skip before reading header and frames if set to 1. Default is 0.
Correct single timestamp overflows if set to 1. Default is 1.
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.
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).

"," separated list of allowed demuxers. By default all are allowed.
Separator used to separate the fields printed on the command line about the Stream parameters. For example, to separate the fields with newlines and indentation:

        ffprobe -dump_separator "
                                  "  -i ~/videos/matrixbench_mpeg2.mpg
    
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 estimation of input duration when calculated using PTS. At present, applicable for MPEG-PS and MPEG-TS.
Specify how strictly to follow the standards. "f_strict" is deprecated and should be used only via the ffmpeg tool.

Possible values:

strictly conform to an older more strict version of the spec or reference software
strictly conform to all the things in the spec no matter what consequences
allow unofficial extensions
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.

Format stream specifiers

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

The exact semantics of stream specifiers is defined by the "avformat_match_stream_specifier()" function declared in the libavformat/avformat.h header and documented in the Stream specifiers section in the ffmpeg(1) manual.

DEMUXERS

Demuxers are configured elements in FFmpeg that can read the multimedia streams from a particular type of file.

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

You can disable all the demuxers using the configure option "--disable-demuxers", and selectively enable a single demuxer with the option "--enable-demuxer=DEMUXER", or disable it with the option "--disable-demuxer=DEMUXER".

The option "-demuxers" of the ff* tools will display the list of enabled demuxers. Use "-formats" to view a combined list of enabled demuxers and muxers.

The description of some of the currently available demuxers follows.

aa

Audible Format 2, 3, and 4 demuxer.

This demuxer is used to demux Audible Format 2, 3, and 4 (.aa) files.

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 the loop variable in the file if set.
Maximum framerate in frames per second (0 for no limit).
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.

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:

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.

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.

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 ....
Allocate the streams according to the onMetaData array content.
Ignore the size of previous tag value.
Output all context of the onMetadata.

gif

Animated GIF demuxer.

It accepts the following options:

Set the minimum valid delay between frames in hundredths of seconds. Range is 0 to 6000. Default value is 2.
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.
Set the default delay between frames in hundredths of seconds. Range is 0 to 6000. Default value is 10.
GIF files can contain information to loop a certain number of times (or infinitely). If ignore_loop is set to 1, then the loop setting from the input will be ignored and looping will not occur. If set to 0, then looping will occur and will cycle the number of times according to the GIF. Default value is 1.

For example, with the overlay filter, place an infinitely looping GIF over another video:

        ffmpeg -i input.mp4 -ignore_loop 0 -i input.gif -filter_complex overlay=shortest=1 out.mkv

Note that in the above example the shortest option for overlay filter is used to end the output video at the length of the shortest input file, which in this case is input.mp4 as the GIF in this example loops infinitely.

hls

HLS demuxer

Apple HTTP Live Streaming demuxer.

This demuxer presents all AVStreams from all variant streams. The id field is set to the bitrate variant index number. By setting the discard flags on AVStreams (by pressing 'a' or 'v' in ffplay), the caller can decide which variant streams to actually receive. The total bitrate of the variant that the stream belongs to is available in a metadata key named "variant_bitrate".

It accepts the following options:

segment index to start live streams at (negative values are from the end).
',' separated list of file extensions that hls is allowed to access.
Maximum number of times a insufficient list is attempted to be reloaded. Default value is 1000.
The maximum number of times to load m3u8 when it refreshes without new segments. Default value is 1000.
Use persistent HTTP connections. Applicable only for HTTP streams. Enabled by default.
Use multiple HTTP connections for downloading HTTP segments. Enabled by default for HTTP/1.1 servers.
Use HTTP partial requests for downloading HTTP segments. 0 = disable, 1 = enable, -1 = auto, Default is auto.

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.
Select the pattern type used to interpret the provided filename.

pattern_type accepts one of the following values.

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.
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
    
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.

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.

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.
Set the index of the file matched by the image file pattern to start to read from. Default value is 0.
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.
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.
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.
If set to 1, will add two extra fields to the metadata found in input, making them also available for other filters (see drawtext filter for examples). Default value is 0. The extra fields are described below:
Corresponds to the full path to the input file being read.
Corresponds to the name of the file being read.

Examples

  • Use ffmpeg for creating a video from the images in the file sequence img-001.jpeg, img-002.jpeg, ..., assuming an input frame rate of 10 frames per second:

            ffmpeg -framerate 10 -i 'img-%03d.jpeg' out.mkv
        
  • As above, but start by reading from a file with index 100 in the sequence:

            ffmpeg -framerate 10 -start_number 100 -i 'img-%03d.jpeg' out.mkv
        
  • Read images matching the "*.png" glob pattern , that is all the files terminating with the ".png" suffix:

            ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv
        

libgme

The Game Music Emu library is a collection of video game music file emulators.

See <https://bitbucket.org/mpyne/game-music-emu/overview> for more information.

It accepts the following options:

Set the index of which track to demux. The demuxer can only export one track. Track indexes start at 0. Default is to pick the first track. Number of tracks is exported as tracks metadata entry.
Set the sampling rate of the exported track. Range is 1000 to 999999. Default is 44100.
The demuxer buffers the entire file into memory. Adjust this value to set the maximum buffer size, which in turn, acts as a ceiling for the size of files that can be read. Default is 50 MiB.

libmodplug

ModPlug based module demuxer

See <https://github.com/Konstanty/libmodplug>

It will export one 2-channel 16-bit 44.1 kHz audio stream. Optionally, a "pal8" 16-color video stream can be exported with or without printed metadata.

It accepts the following options:

Apply a simple low-pass filter. Can be 1 (on) or 0 (off). Default is 0.
Set amount of reverb. Range 0-100. Default is 0.
Set delay in ms, clamped to 40-250 ms. Default is 0.
Apply bass expansion a.k.a. XBass or megabass. Range is 0 (quiet) to 100 (loud). Default is 0.
Set cutoff i.e. upper-bound for bass frequencies. Range is 10-100 Hz. Default is 0.
Apply a Dolby Pro-Logic surround effect. Range is 0 (quiet) to 100 (heavy). Default is 0.
Set surround delay in ms, clamped to 5-40 ms. Default is 0.
The demuxer buffers the entire file into memory. Adjust this value to set the maximum buffer size, which in turn, acts as a ceiling for the size of files that can be read. Range is 0 to 100 MiB. 0 removes buffer size limit (not recommended). Default is 5 MiB.
String which is evaluated using the eval API to assign colors to the generated video stream. Variables which can be used are "x", "y", "w", "h", "t", "speed", "tempo", "order", "pattern" and "row".
Generate video stream. Can be 1 (on) or 0 (off). Default is 0.
Set video frame width in 'chars' where one char indicates 8 pixels. Range is 20-512. Default is 30.
Set video frame height in 'chars' where one char indicates 8 pixels. Range is 20-512. Default is 30.
Print metadata on video stream. Includes "speed", "tempo", "order", "pattern", "row" and "ts" (time in ms). Can be 1 (on) or 0 (off). Default is 1.

libopenmpt

libopenmpt based module demuxer

See <https://lib.openmpt.org/libopenmpt/> for more information.

Some files have multiple subsongs (tracks) this can be set with the subsong option.

It accepts the following options:

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.

Set the channel layout. Valid values are 1, 2, and 4 channel layouts. The default value is STEREO.
Set the sample rate for libopenmpt to output. Range is from 1000 to INT_MAX. The value default is 48000.

mov/mp4/3gp

Demuxer for Quicktime File Format & ISO/IEC Base Media File Format (ISO/IEC 14496-12 or MPEG-4 Part 12, ISO/IEC 15444-12 or JPEG 2000 Part 12).

Registered extensions: mov, mp4, m4a, 3gp, 3g2, mj2, psp, m4b, ism, ismv, isma, f4v

Options

This demuxer accepts the following options:

Enable loading of external tracks, disabled by default. Enabling this can theoretically leak information in some use cases.
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.
When seeking, identify the closest point in each stream individually and demux packets in that stream from identified point. This can lead to a different sequence of packets compared to demuxing linearly from the beginning. Default is true.
Ignore any edit list atoms. The demuxer, by default, modifies the stream index to reflect the timeline described by the edit list. Default is false.
Modify the stream index to reflect the timeline described by the edit list. "ignore_editlist" must be set to false for this option to be effective. If both "ignore_editlist" and this option are set to false, then only the start of the stream index is modified to reflect initial dwell time or starting timestamp described by the edit list. Default is true.
Don't parse chapters. This includes GoPro 'HiLight' tags/moments. Note that chapters are only parsed when input is seekable. Default is false.
For seekable fragmented input, set fragment's starting timestamp from media fragment random access box, if present.

Following options are available:

Auto-detect whether to set mfra timestamps as PTS or DTS (default)
Set mfra timestamps as DTS
Set mfra timestamps as PTS
0
Don't use mfra box to set timestamps
Export unrecognized boxes within the udta box as metadata entries. The first four characters of the box type are set as the key. Default is false.
Export entire contents of XMP_ box and uuid box as a string with key "xmp". Note that if "export_all" is set and this option isn't, the contents of XMP_ box are still exported but with key "XMP_". Default is false.
4-byte key required to decrypt Audible AAX and AAX+ files. See Audible AAX subsection below.
Fixed key used for handling Audible AAX/AAX+ files. It has been pre-set so should not be necessary to specify.
16-byte key, in hex, to decrypt files encrypted using ISO Common Encryption (CENC/AES-128 CTR; ISO/IEC 23001-7).

Audible AAX

Audible AAX files are encrypted M4B files, and they can be decrypted by specifying a 4 byte activation secret.

        ffmpeg -activation_bytes 1CEB00DA -i test.aax -vn -c:a copy output.mp4

mpegts

MPEG-2 transport stream demuxer.

This demuxer accepts the following options:

Set size limit for looking up a new synchronization. Default value is 65536.
Skip PMTs for programs not defined in the PAT. Default value is 0.
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.
Output option carrying the raw packet size in bytes. Show the detected raw packet size, cannot be set by the user.
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.
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.

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.
Set the input video pixel format. Default value is "yuv420p".
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:

Set the start time of the TED talk, in milliseconds. The default is 15000 (15s). It is used to sync the captions with the downloadable videos, because they include a 15s intro.

Example: convert the captions to a format most players understand:

        ffmpeg -i http://www.ted.com/talks/subtitles/id/1/lang/en talk1-en.srt

vapoursynth

Vapoursynth wrapper.

Due to security concerns, Vapoursynth scripts will not be autodetected so the input format has to be forced. For ff* CLI tools, add "-f vapoursynth" before the input "-i yourscript.vpy".

This demuxer accepts the following option:

The demuxer buffers the entire script into memory. Adjust this value to set the maximum buffer size, which in turn, acts as a ceiling for the size of scripts that can be read. Default is 1 MiB.

METADATA

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

The file format is as follows:

1.
A file consists of a header and a number of metadata tags divided into sections, each on its own line.
2.
The header is a ;FFMETADATA string, followed by a version number (now 1).
3.
Metadata tags are of the form key=value
4.
Immediately after header follows global metadata
5.
After global metadata there may be sections with per-stream/per-chapter metadata.
6.
A section starts with the section name in uppercase (i.e. STREAM or CHAPTER) in brackets ([, ]) and ends with next section or end of file.
7.
At the beginning of a chapter section there may be an optional timebase to be used for start/end values. It must be in form TIMEBASE=num/den, where num and den are integers. If the timebase is missing then start/end times are assumed to be in nanoseconds.

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

8.
Empty lines and lines starting with ; or # are ignored.
9.
Metadata keys or values containing special characters (=, ;, #, \ and a newline) must be escaped with a backslash \.
10.
Note that whitespace in metadata (e.g. foo = bar) is considered to be a part of the tag (in the example above key is foo , value is
bar).

A ffmetadata file might look like this:

        ;FFMETADATA1
        title=bike\\shed
        ;this is a comment
        artist=FFmpeg troll team
        
        [CHAPTER]
        TIMEBASE=1/1000
        START=0
        #chapter ends at 0:01:00
        END=60000
        title=chapter \#1
        [STREAM]
        title=multi\
        line

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

Extracting an ffmetadata file with ffmpeg goes as follows:

        ffmpeg -i INPUT -f ffmetadata FFMETADATAFILE

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

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

PROTOCOL OPTIONS

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

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

The list of supported options follows:

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:

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

A description of the currently available protocols follows.

amqp

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

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

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

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

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

Muliple subscribers may stream from the broker using the command:

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

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

The following options are supported:

Sets the exchange to use on the broker. RabbitMQ has several predefined exchanges: "amq.direct" is the default exchange, where the publisher and subscriber must have a matching routing_key; "amq.fanout" is the same as a broadcast operation (i.e. the data is forwarded to all queues on the fanout exchange independent of the routing_key); and "amq.topic" is similar to "amq.direct", but allows for more complex pattern matching (refer to the RabbitMQ documentation).
Sets the routing key. The default value is "amqp". The routing key is used on the "amq.direct" and "amq.topic" exchanges to decide whether packets are written to the queue of a subscriber.
Maximum size of each packet sent/received to the broker. Default is 131072. Minimum is 4096 and max is any large value (representable by an int). When receiving packets, this sets an internal buffer size in FFmpeg. It should be equal to or greater than the size of the published packets to the broker. Otherwise the received message may be truncated causing decoding errors.
The timeout in seconds during the initial connection to the broker. The default value is rw_timeout, or 5 seconds if rw_timeout is not set.
Sets the delivery mode of each message sent to broker. The following values are accepted:
Delivery mode set to "persistent" (2). This is the default value. Messages may be written to the broker's disk depending on its setup.
Delivery mode set to "non-persistent" (1). Messages will stay in broker's memory unless the broker is under memory pressure.

async

Asynchronous data filling wrapper for input stream.

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

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

bluray

Read BluRay playlist.

The accepted options are:

BluRay angle
Start chapter (1...N)
Playlist to read (BDMV/PLAYLIST/?????.mpls)

Examples:

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

        bluray:/mnt/bluray

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

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

cache

Caching wrapper for input stream.

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

The accepted options are:

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

URL Syntax is

        cache:<URL>

concat

Physical concatenation protocol.

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

A URL accepted by this protocol has the syntax:

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

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

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

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

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

crypto

AES-encrypted stream reading protocol.

The accepted options are:

Set the AES decryption key binary block from given hexadecimal representation.
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 existing files on write, if set to 1. A value of 0 prevents truncating. Default value is 1.
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.
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).
Controls if seekability is advertised on the file. 0 means non-seekable, -1 means auto (seekable for normal files, non-seekable for named pipes).

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

ftp

FTP (File Transfer Protocol).

Read from or write to remote resources using FTP protocol.

Following syntax is required.

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

This protocol accepts the following options.

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.
Set a user to be used for authenticating to the FTP server. This is overridden by the user in the FTP URL.
Set a password to be used for authenticating to the FTP server. This is overridden by the password in the FTP URL, or by ftp-anonymous-password if no user is set.
Password used when login as anonymous user. Typically an e-mail address should be used.
Control seekability of connection during encoding. If set to 1 the resource is supposed to be seekable, if set to 0 it is assumed not to be seekable. Default value is 0.

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

gopher

Gopher protocol.

gophers

Gophers protocol.

The Gopher protocol with TLS encapsulation.

hls

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

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

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

http

HTTP (Hyper Text Transfer Protocol).

This protocol accepts the following options:

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.
If set to 1 use chunked Transfer-Encoding for posts, default is 1.
Set a specific content type for the POST messages or for listen mode.
set HTTP proxy to tunnel through e.g. http://example.com:1234
Set custom HTTP headers, can override built in default headers. The value must be a string encoding the headers.
Use persistent connections if set to 1, default is 0.
Set custom HTTP post data.
Set the Referer header. Include 'Referer: URL' header in HTTP request.
Override the User-Agent header. If not specified the protocol will use a string describing the libavformat build. ("Lavf/<version>")
This is a deprecated option, you can use user_agent instead it.
If set then eof is treated like an error and causes reconnection, this is useful for live / endless streams.
If set then even streamed/non seekable streams will be reconnected on errors.
Reconnect automatically in case of TCP/TLS errors during connect.
A comma separated list of HTTP status codes to reconnect on. The list can include specific status codes (e.g. '503') or the strings '4xx' / '5xx'.
Sets the maximum delay in seconds after which to give up reconnecting
Export the MIME type.
Exports the HTTP response version number. Usually "1.0" or "1.1".
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.
If the server supports ICY metadata, this contains the ICY-specific HTTP reply headers, separated by newline characters.
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.
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.
Set initial byte offset.
Try to limit the request to bytes preceding this offset.
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.

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

        # Server side (sending):
        ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://<server>:<port>
        
        # Client side (receiving):
        ffmpeg -i http://<server>:<port> -c copy somefile.ogg
        
        # Client can also be done with wget:
        wget http://<server>:<port> -O somefile.ogg
        
        # Server side (receiving):
        ffmpeg -listen 1 -i http://<server>:<port> -c copy somefile.ogg
        
        # Client side (sending):
        ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://<server>:<port>
        
        # Client can also be done with wget:
        wget --post-file=somefile.ogg http://<server>:<port>
    
Send an Expect: 100-continue header for POST. If set to 1 it will send, if set to 0 it won't, if set to -1 it will try to send if it is applicable. Default value is -1.
Set HTTP authentication type. No option for Digest, since this method requires getting nonce parameters from the server first and can't be used straight away like Basic.
Choose the HTTP authentication type automatically. This is the default.
Choose the HTTP basic authentication.

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

HTTP Cookies

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

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

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

Icecast

Icecast protocol (stream to Icecast servers)

This protocol accepts the following options:

Set the stream genre.
Set the stream name.
Set the stream description.
Set the stream website URL.
Set if the stream should be public. The default is 0 (not public).
Override the User-Agent header. If not specified a string of the form "Lavf/<version>" will be used.
Set the Icecast mountpoint password.
Set the stream content type. This must be set if it is different from audio/mpeg.
This enables support for Icecast versions < 2.4.0, that do not support the HTTP PUT method but the SOURCE method.
tls
Establish a TLS (HTTPS) connection to Icecast.

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

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:

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:

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

Example usage:

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

rist

Reliable Internet Streaming Transport protocol

The accepted options are:

Supported values:
This one is default.
Set internal RIST buffer size in milliseconds for retransmission of data. Default value is 0 which means the librist default (1 sec). Maximum value is 30 seconds.
Set maximum packet size for sending data. 1316 by default.
Set loglevel for RIST logging messages. You only need to set this if you explicitly want to enable debug level messages or packet loss simulation, otherwise the regular loglevel is respected.
Set override of encryption secret, by default is unset.
Set encryption type, by default is disabled. Acceptable values are 128 and 256.

rtmp

Real-Time Messaging Protocol.

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

The required syntax is:

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

The accepted parameters are:

An optional username (mostly for publishing).
An optional password (mostly for publishing).
The address of the RTMP server.
The number of the TCP port to use (by default is 1935).
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.
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.
Act as a server, listening for an incoming connection.
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):

Name of application to connect on the RTMP server. This option overrides the parameter specified in the URI.
Set the client buffer time in milliseconds. The default is 3000.
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.
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>).)
Number of packets flushed in the same request (RTMPT only). The default is 10.
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".
URL of the web page in which the media was embedded. By default no value will be sent.
Stream identifier to play or to publish. This option overrides the parameter specified in the URI.
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.
SHA256 hash of the decompressed SWF file (32 bytes).
Size of the decompressed SWF file, required for SWFVerification.
URL of the SWF player for the media. By default no value will be sent.
URL to player swf file, compute hash/size automatically.
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.

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

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

Set the TTL (Time-To-Live) value (for multicast only).
Set the remote RTCP port to n.
Set the local RTP port to n.
Set the local RTCP port to n.
Set max packet size (in bytes) to n.
Set the maximum UDP socket buffer size in bytes.
Do a "connect()" on the UDP socket (if set to 1) or not (if set to 0).
List allowed source IP addresses.
List disallowed (blocked) source IP addresses.
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).
Set the local RTP port to n.
Set timeout (in microseconds) of socket I/O operations to n.

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

Important notes:

1.
If rtcpport is not set the RTCP port will be set to the RTP port value plus 1.
2.
If localrtpport (the local RTP port) is not set any available port will be used for the local RTP and RTCP ports.
3.
If localrtcpport (the local RTCP port) is not set it will be set to the local RTP port value plus 1.

rtsp

Real-Time Streaming Protocol.

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

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

The required syntax for a RTSP url is:

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

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

The following options are supported.

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

Set RTSP flags.

The following values are accepted:

Accept packets only from negotiated peer address and port.
Act as a server, listening for an incoming connection.
Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.

Default value is none.

Set media types to accept from the server.

The following flags are accepted:

data

By default it accepts all media types.

Set minimum local UDP port. Default value is 5000.
Set maximum local UDP port. Default value is 65000.
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.

Set number of packets to buffer for handling of reordered packets.
Set socket TCP I/O timeout in microseconds.
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:

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.
Specify the port to send the announcements on, defaults to 9875 if not specified.
Specify the time to live value for the announcements and RTP packets, defaults to 255.
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:

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

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).
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.
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.
IP Type of Service. Applies to sender only. Default value is 0xB8.
IP Time To Live. Applies to sender only. Default value is 64.
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.
Set socket listen timeout.
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)
Connection mode. caller opens client connection. listener starts server to listen for incoming connections. rendezvous use Rendez-Vous connection mode. Default value is caller.
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.
If set to 1, Receiver will send `UMSG_LOSSREPORT` messages periodically until a lost packet is retransmitted or intentionally dropped. Default value is 1.
Recovery bandwidth overhead above input rate, in percents. See inputbw. Default value is 25%.
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).
If true, both connection parties must have the same password set (including empty, that is, with no encryption). If the password doesn't match or only one side is unencrypted, the connection is rejected. Default is true.
The number of packets to be transmitted after which the encryption key is switched to a new key. Default is -1. -1 means auto (0x1000000 in srt library). The range for this option is integers in the 0 - "INT_MAX".
The interval between when a new encryption key is sent and when switchover occurs. This value also applies to the subsequent interval between when switchover occurs and when the old encryption key is decommissioned. Default is -1. -1 means auto (0x1000 in srt library). The range for this option is integers in the 0 - "INT_MAX".
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.
Alias for payload_size.
The latency value (as described in rcvlatency) that is set by the sender side as a minimum value for the receiver.
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.
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.
Set UDP receive buffer size, expressed in bytes.
Set UDP send buffer size, expressed in bytes.
Set raise error timeouts for read, write and connect operations. Note that the SRT library has internal timeouts which can be controlled separately, the value set here is only a cap on those.
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.
Set send buffer size, expressed in bytes.
Set receive buffer size, expressed in bytes.

Receive buffer must not be greater than ffs.

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.
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.

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
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.
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.

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

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

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

srtp

Secure Real-time Transport Protocol.

The accepted options are:

Select input and output encoding suites.

Supported values:

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 offset of the extracted segment, in bytes.
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 for an incoming connection. 0 disables listen, 1 enables listen in single client mode, 2 enables listen in multi-client mode. Default value is 0.
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.

Set listen timeout, expressed in milliseconds.
Set receive buffer size, expressed bytes.
Set send buffer size, expressed bytes.
Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.
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):

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.
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.

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.)
A file containing the private key for the certificate.
If enabled, listen for connections on the provided port, and assume the server role in the handshake instead of the client role.
The HTTP proxy to tunnel through, e.g. "http://example.com:1234". The proxy must support the CONNECT method.

Example command lines:

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

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

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

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

udp

User Datagram Protocol.

The required syntax for an UDP URL is:

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

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

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

The list of supported options follows.

Set the UDP maximum socket buffer size in bytes. This is used to set either the receive or send buffer size, depending on what the socket is used for. Default is 32 KB for output, 384 KB for input. See also fifo_size.
If set to nonzero, the output will have the specified constant bitrate if the input has enough packets to sustain it.
When using bitrate this specifies the maximum number of bits in packet bursts.
Override the local UDP port to bind with.
Local IP address of a network interface used for sending packets or joining multicast groups.
Set the size in bytes of UDP packets.
Explicitly allow or disallow reusing UDP sockets.
Set the time to live value (for multicast only).
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.
Only receive packets sent from the specified addresses. In case of multicast, also subscribe to multicast traffic coming from these addresses only.
Ignore packets sent from the specified addresses. In case of multicast, also exclude the source addresses in the multicast subscription.
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.
Survive in case of UDP receiving circular buffer overrun. Default value is 0.
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.

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 in ms.
Create the Unix socket in listening mode.

zmq

ZeroMQ asynchronous messaging using the libzmq library.

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

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

        zmq:tcp://ip-address:port

Example: Create a localhost stream on port 5555:

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

Multiple clients may connect to the stream using:

        ffplay zmq:tcp://127.0.0.1:5555

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

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

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

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

DEVICE OPTIONS

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

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

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

INPUT DEVICES

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

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

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

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

A description of the currently available input devices follows.

alsa

ALSA (Advanced Linux Sound Architecture) input device.

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

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

An ALSA identifier has the syntax:

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

where the DEV and SUBDEV components are optional.

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

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

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

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

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

Options

Set the sample rate in Hz. Default is 48000.
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

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).
Set the index of the camera to use. Default is 0.
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:

If set to true, a list of all available input devices is given showing all device names and indices.
Specify the video device by its index. Overrides anything given in the input filename.
Specify the audio device by its index. Overrides anything given in the input filename.
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".
Set the video frame size.
Capture the mouse pointer. Default is 0.
Capture the screen mouse clicks. Default is 0.
Capture the raw device data. Default is 0. Using this option may result in receiving the underlying data delivered to the AVFoundation framework. E.g. for muxed devices that sends raw DV data to the framework (like tape-based camcorders), setting this option to false results in extracted video frames captured in the designated pixel format only. Setting this option to true results in receiving the raw DV stream untouched.

Examples

  • Print the list of AVFoundation supported devices and exit:

            $ ffmpeg -f avfoundation -list_devices true -i ""
        
  • Record video from video device 0 and audio from audio device 0 into out.avi:

            $ ffmpeg -f avfoundation -i "0:0" out.avi
        
  • Record video from video device 2 and audio from audio device 1 into out.avi:

            $ ffmpeg -f avfoundation -video_device_index 2 -i ":1" out.avi
        
  • Record video from the system default video device using the pixel format bgr0 and do not record any audio into out.avi:

            $ ffmpeg -f avfoundation -pixel_format bgr0 -i "default:none" out.avi
        
  • Record raw DV data from a suitable input device and write the output into out.dv:

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

bktr

BSD video input device.

Options

framerate
Set the frame rate.
Set the video frame size. Default is "vga".
Available values are:

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

If set to true, print a list of devices and exit. Defaults to false. This option is deprecated, please use the "-sources" option of ffmpeg to list the available input devices.
If set to true, print a list of supported formats and exit. Defaults to false.
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.
Set the pixel format of the captured video. Available values are:
This is the default which means 8-bit YUV 422 or 8-bit ARGB if format autodetection is used, 8-bit YUV 422 otherwise.
8-bit YUV 422.
10-bit YUV 422.
8-bit RGB.
8-bit RGB.
10-bit RGB.
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.

Defines number of audio channels to capture. Must be 2, 8 or 16. Defaults to 2.
Sets the decklink device duplex mode. Must be unset, half or full. Defaults to unset.
Timecode type to include in the frame and video stream metadata. Must be none, rp188vitc, rp188vitc2, rp188ltc, rp188hfr, rp188any, vitc, vitc2, or serial. Defaults to none (not included).

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

Sets the video input source. Must be unset, sdi, hdmi, optical_sdi, component, composite or s_video. Defaults to unset.
Sets the audio input source. Must be unset, embedded, aes_ebu, analog, analog_xlr, analog_rca or microphone. Defaults to unset.
Sets the video packet timestamp source. Must be video, audio, reference, wallclock or abs_wallclock. Defaults to video.
Sets the audio packet timestamp source. Must be video, audio, reference, wallclock or abs_wallclock. Defaults to audio.
If set to true, color bars are drawn in the event of a signal loss. Defaults to true.
Sets maximum input buffer size in bytes. If the buffering reaches this value, incoming frames will be dropped. Defaults to 1073741824.
Sets the audio sample bit depth. Must be 16 or 32. Defaults to 16.
If set to true, timestamps are forwarded as they are without removing the initial offset. Defaults to false.
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.
Drop frames till a frame with timecode is received. Sometimes serial timecode isn't received with the first input frame. If that happens, the stored stream timecode will be inaccurate. If this option is set to true, input frames are dropped till a frame with timecode is received. Option timecode_format must be specified. Defaults to false.
If set to true, extracts KLV data from VANC and outputs KLV packets. KLV VANC packets are joined based on MID and PSC fields and aggregated into one KLV packet. Defaults to false.

Examples

  • List input devices:

            ffmpeg -sources decklink
        
  • List supported formats:

            ffmpeg -f decklink -list_formats 1 -i 'Intensity Pro'
        
  • Capture video clip at 1080i50:

            ffmpeg -format_code Hi50 -f decklink -i 'Intensity Pro' -c:a copy -c:v copy output.avi
        
  • Capture video clip at 1080i50 10 bit:

            ffmpeg -raw_format yuv422p10 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder' -c:a copy -c:v copy output.avi
        
  • Capture video clip at 1080i50 with 16 audio channels:

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

dshow

Windows DirectShow input device.

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

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

The input name should be in the format:

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

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

Options

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

Set the video size in the captured video.
framerate
Set the frame rate in the captured video.
Set the sample rate (in Hz) of the captured audio.
Set the sample size (in bits) of the captured audio.
Set the number of channels in the captured audio.
If set to true, print a list of devices and exit.
If set to true, print a list of selected device's options and exit.
Set video device number for devices with the same name (starts at 0, defaults to 0).
Set audio device number for devices with the same name (starts at 0, defaults to 0).
Select pixel format to be used by DirectShow. This may only be set when the video codec is not set or set to rawvideo.
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>
Select video capture pin to use by name or alternative name.
Select audio capture pin to use by name or alternative name.
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.
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.
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.
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.
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.
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.
If set to true, before capture starts, popup a display dialog to the end user, allowing them to manually modify TV channels and frequencies.
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).
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.
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.
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.
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

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 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
    
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.
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.

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

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

Examples

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

            ffplay -f iec61883 -i auto
        
  • Grab and record the input of a FireWire DV/HDV device, using a packet buffer of 100000 packets if the source is HDV.

            ffmpeg -f iec61883 -i auto -dvbuffer 100000 out.mpg
        

jack

JACK input device.

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

A JACK input device creates one or more JACK writable clients, one for each audio channel, with name client_name:input_N, where client_name is the name provided by the application, and N is a number which identifies the channel. Each writable client will send the acquired data to the FFmpeg input device.

Once you have created one or more JACK readable clients, you need to connect them to one or more JACK writable clients.

To connect or disconnect JACK clients you can use the jack_connect and jack_disconnect programs, or do it through a graphical interface, for example with qjackctl.

To list the JACK clients and their properties you can invoke the command jack_lsp.

Follows an example which shows how to capture a JACK readable client with ffmpeg.

        # Create a JACK writable client with name "ffmpeg".
        $ ffmpeg -f jack -i ffmpeg -y out.wav
        
        # Start the sample jack_metro readable client.
        $ jack_metro -b 120 -d 0.2 -f 4000
        
        # List the current JACK clients.
        $ jack_lsp -c
        system:capture_1
        system:capture_2
        system:playback_1
        system:playback_2
        ffmpeg:input_1
        metro:120_bpm
        
        # Connect metro to the ffmpeg writable client.
        $ jack_connect metro:120_bpm ffmpeg:input_1

For more information read: <http://jackaudio.org/>

Options

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

DRM device to capture on. Defaults to /dev/dri/card0.
format
Pixel format of the framebuffer. This can be autodetected if you are running Linux 5.7 or later, but needs to be provided for earlier versions. Defaults to bgr0, which is the most common format used by the Linux console and Xorg X server.
Format modifier to signal on output frames. This is necessary to import correctly into some APIs. It can be autodetected if you are running Linux 5.7 or later, but will need to be provided explicitly when needed in earlier versions. See the libdrm documentation for possible values.
KMS CRTC ID to define the capture source. The first active plane on the given CRTC will be used.
KMS plane ID to define the capture source. Defaults to the first active plane found if neither crtc_id nor plane_id are specified.
framerate
Framerate to capture at. This is not synchronised to any page flipping or framebuffer changes - it just defines the interval at which the framebuffer is sampled. Sampling faster than the framebuffer update rate will generate independent frames with the same content. Defaults to 30.

Examples

  • Capture from the first active plane, download the result to normal frames and encode. This will only work if the framebuffer is both linear and mappable - if not, the result may be scrambled or fail to download.

            ffmpeg -f kmsgrab -i - -vf 'hwdownload,format=bgr0' output.mp4
        
  • Capture from CRTC ID 42 at 60fps, map the result to VAAPI, convert to NV12 and encode as H.264.

            ffmpeg -crtc_id 42 -framerate 60 -f kmsgrab -i - -vf 'hwmap=derive_device=vaapi,scale_vaapi=w=1920:h=1080:format=nv12' -c:v h264_vaapi output.mp4
        
  • To capture only part of a plane the output can be cropped - this can be used to capture a single window, as long as it has a known absolute position and size. For example, to capture and encode the middle quarter of a 1920x1080 plane:

            ffmpeg -f kmsgrab -i - -vf 'hwmap=derive_device=vaapi,crop=960:540:480:270,scale_vaapi=960:540:nv12' -c:v h264_vaapi output.mp4
        

lavfi

Libavfilter input virtual device.

This input device reads data from the open output pads of a libavfilter filtergraph.

For each filtergraph open output, the input device will create a corresponding stream which is mapped to the generated output. Currently only video data is supported. The filtergraph is specified through the option graph.

Options

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.

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.
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

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.

Set paranoia recovery mode flags. It accepts one of the following values:

Default value is disable.

For more information about the available recovery modes, consult the paranoia project documentation.

libdc1394

IIDC1394 input device, based on libdc1394 and libraw1394.

Requires the configure option "--enable-libdc1394".

Options

framerate
Set the frame rate. Default is "ntsc", corresponding to a frame rate of "30000/1001".
Select the pixel format. Default is "uyvy422".
Set the video size given as a string such as "640x480" or "hd720". Default is "qvga".

openal

The OpenAL input device provides audio capture on all systems with a working OpenAL 1.1 implementation.

To enable this input device during configuration, you need OpenAL headers and libraries installed on your system, and need to configure FFmpeg with "--enable-openal".

OpenAL headers and libraries should be provided as part of your OpenAL implementation, or as an additional download (an SDK). Depending on your installation you may need to specify additional flags via the "--extra-cflags" and "--extra-ldflags" for allowing the build system to locate the OpenAL headers and libraries.

An incomplete list of OpenAL implementations follows:

The official Windows implementation, providing hardware acceleration with supported devices and software fallback. See <http://openal.org/>.
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>.
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

Set the number of channels in the captured audio. Only the values 1 (monaural) and 2 (stereo) are currently supported. Defaults to 2.
Set the sample size (in bits) of the captured audio. Only the values 8 and 16 are currently supported. Defaults to 16.
Set the sample rate (in Hz) of the captured audio. Defaults to 44.1k.
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

Set the sample rate in Hz. Default is 48000.
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

Connect to a specific PulseAudio server, specified by an IP address. Default server is used when not provided.
Specify the application name PulseAudio will use when showing active clients, by default it is the "LIBAVFORMAT_IDENT" string.
Specify the stream name PulseAudio will use when showing active streams, by default it is "record".
Specify the samplerate in Hz, by default 48kHz is used.
Specify the channels in use, by default 2 (stereo) is set.
Specify the number of bytes per frame, by default it is set to 1024.
Specify the minimal buffering fragment in PulseAudio, it will affect the audio latency. By default it is unset.
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

Set the sample rate in Hz. Default is 48000.
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

Set the standard. Must be the name of a supported standard. To get a list of the supported standards, use the list_standards option.
Set the input channel number. Default to -1, which means using the previously selected channel.
Set the video frame size. The argument must be a string in the form WIDTHxHEIGHT or a valid size abbreviation.
Select the pixel format (only valid for raw video input).
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 available formats (supported pixel formats, codecs, and frame sizes) and exit.

Available values are:

Show all available (compressed and non-compressed) formats.
Show only raw video (non-compressed) formats.
Show only compressed formats.
List supported standards and exit.

Available values are:

Show all supported standards.
Set type of timestamps for grabbed frames.

Available values are:

Use timestamps from the kernel.
Use absolute timestamps (wall clock).
Force conversion from monotonic to absolute timestamps.

Default value is "default".

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

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

Specify whether to select the grabbing area graphically using the pointer. A value of 1 prompts the user to select the grabbing area graphically by clicking and dragging. A single click with no dragging will select the whole screen. A region with zero width or height will also select the whole screen. This option overwrites the video_size, grab_x, and grab_y options. Default value is 0.
Specify whether to draw the mouse pointer. A value of 0 specifies not to draw the pointer. Default value is 1.
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 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.

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
    
Grab this window, instead of the whole screen. Default value is 0, which maps to the whole screen (root window).

The id of a window can be found using the xwininfo program, possibly with options -tree and -root.

If the window is later enlarged, the new area is not recorded. Video ends when the window is closed, unmapped (i.e., iconified) or shrunk beyond the video size (which defaults to the initial window size).

This option disables options follow_mouse and select_region.

Set the video frame size. Default is the full desktop or window.
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.

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.
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.
Set the number of used input channels. Default value is 0. This option is only used for special remapping.
Set the input sample rate. Default value is 0.
Set the output sample rate. Default value is 0.
Specify the input sample format. It is set by default to "none".
Specify the output sample format. It is set by default to "none".
Set the internal sample format. Default value is "none". This will automatically be chosen when it is not explicitly set.
Set the input/output channel layout.

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

Set the center mix level. It is a value expressed in deciBel, and must be in the interval [-32,32].
Set the surround mix level. It is a value expressed in deciBel, and must be in the interval [-32,32].
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].
Set rematrix volume. Default value is 1.0.
Set maximum output value for rematrixing. This can be used to prevent clipping vs. preventing volume reduction. A value of 1.0 prevents clipping.
Set flags used by the converter. Default value is 0.

It supports the following individual flags:

force resampling, this flag forces resampling to be used even when the input and output sample rates match.
Set the dither scale. Default value is 1.
Set dither method. Default value is 0.

Supported values:

select rectangular dither
select triangular dither
select triangular dither with high pass
select Lipshitz noise shaping dither.
select Shibata noise shaping dither.
select low Shibata noise shaping dither.
select high Shibata noise shaping dither.
select f-weighted noise shaping dither
select modified-e-weighted noise shaping dither
select improved-e-weighted noise shaping dither
Set resampling engine. Default value is swr.

Supported values:

select the native SW Resampler; filter options precision and cheby are not applicable in this case.
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.
For swr only, set resampling filter size, default value is 32.
For swr only, set resampling phase shift, default value is 10, and must be in the interval [0,30].
Use linear interpolation when enabled (the default). Disable it if you want to preserve speed instead of quality when exact_rational fails.
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.
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).
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'.
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.
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.
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").
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.
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.
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.
Select matrixed stereo encoding.

It accepts the following values:

select none
select Dolby
select Dolby Pro Logic II

Default value is "none".

For swr only, select resampling filter type. This only affects resampling operations.

It accepts the following values:

select cubic
select Blackman Nuttall windowed sinc
select Kaiser windowed sinc
For swr only, set Kaiser window beta value. Must be a double float value in the interval [2,16], default value is 9.
For swr only, set number of used output sample bits for dithering. Must be an integer in the interval [0,64], default value is 0, which means it's not used.

SCALER OPTIONS

The video scaler supports the following named options.

Options may be set by specifying -option value in the FFmpeg tools, with a few API-only exceptions noted below. For programmatic use, they can be set explicitly in the "SwsContext" options or through the libavutil/opt.h API.

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:

Select fast bilinear scaling algorithm.
Select bilinear scaling algorithm.
Select bicubic scaling algorithm.
Select experimental scaling algorithm.
Select nearest neighbor rescaling algorithm.
Select averaging area rescaling algorithm.
Select bicubic scaling algorithm for the luma component, bilinear for chroma components.
Select Gaussian rescaling algorithm.
sinc
Select sinc rescaling algorithm.
Select Lanczos rescaling algorithm. The default width (alpha) is 3 and can be changed by setting "param0".
Select natural bicubic spline rescaling algorithm.
Enable printing/debug logging.
Enable accurate rounding.
Enable full chroma interpolation.
Select full chroma input.
Enable bitexact output.
Set source width.
Set source height.
Set destination width.
Set destination height.
Set source pixel format (must be expressed as an integer).
Set destination pixel format (must be expressed as an integer).
If value is set to 1, indicates source is full range. Default value is 0, which indicates source is limited range.
If value is set to 1, enable full range for destination. Default value is 0, which enables limited range.
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.
Set the dithering algorithm. Accepts one of the following values. Default value is auto.
automatic choice
no dithering
bayer dither
error diffusion dither
arithmetic dither, based using addition
arithmetic dither, based using xor (more random/less apparent patterning that a_dither).
Set the alpha blending to use when the input has alpha but the output does not. Default value is none.
Blend onto a uniform background color
Blend onto a checkerboard
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:

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

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

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

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

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

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

CHANGING OPTIONS AT RUNTIME WITH A COMMAND

Some options can be changed during the operation of the filter using a command. These options are marked 'T' on the output of ffmpeg -h filter=<name of filter>. The name of the command is the name of the option and the argument is the new value.

OPTIONS FOR FILTERS WITH SEVERAL INPUTS

Some filters with several inputs support a common set of options. These options can only be set by name, not with the short notation.

The action to take when EOF is encountered on the secondary input; it accepts one of the following values:
Repeat the last frame (the default).
End both streams.
Pass the main input through.
If set to 1, force the output to terminate when the shortest input terminates. Default value is 0.
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:

Set input gain. Default is 1. Range is between 0.015625 and 64.
Set mode of compressor operation. Can be "upward" or "downward". Default is "downward".
threshold
If a signal of stream rises above this level it will affect the gain reduction. By default it is 0.125. Range is between 0.00097563 and 1.
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.
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.
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.
Set the amount by how much signal will be amplified after processing. Default is 1. Range is from 1 to 64.
Curve the sharp knee around the threshold to enter gain reduction more softly. Default is 2.82843. Range is between 1 and 8.
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".
Should the exact signal be taken in case of "peak" or an RMS one in case of "rms". Default is "rms" which is mostly smoother.
mix
How much to use compressed signal in output. Default is 1. Range is between 0 and 1.

Commands

This filter supports the all above options as commands.

acontrast

Simple audio dynamic range compression/expansion filter.

The filter accepts the following options:

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:

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.
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.
Should first stream end overlap with second stream start. Default is enabled.
Set curve for cross fade transition for first stream.
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:

Set split frequencies. Those must be positive and increasing.
Set filter order for each band split. This controls filter roll-off or steepness of filter transfer function. Available values are:
2nd
12 dB per octave.
4th
24 dB per octave.
6th
36 dB per octave.
8th
48 dB per octave.
10th
60 dB per octave.
12th
72 dB per octave.
14th
84 dB per octave.
16th
96 dB per octave.
18th
108 dB per octave.
20th
120 dB per octave.

Default is 4th.

Set input gain level. Allowed range is from 0 to 1. Default value is 1.
Set output gain for each band. Default value is 1 for all bands.

Examples

  • Split input audio stream into two bands (low and high) with split frequency of 1500 Hz, each band will be in separate stream:

            ffmpeg -i in.flac -filter_complex 'acrossover=split=1500[LOW][HIGH]' -map '[LOW]' low.wav -map '[HIGH]' high.wav
        
  • Same as above, but with higher filter order:

            ffmpeg -i in.flac -filter_complex 'acrossover=split=1500:order=8th[LOW][HIGH]' -map '[LOW]' low.wav -map '[HIGH]' high.wav
        
  • Same as above, but also with additional middle band (frequencies between 1500 and 8000):

            ffmpeg -i in.flac -filter_complex 'acrossover=split=1500 8000:order=8th[LOW][MID][HIGH]' -map '[LOW]' low.wav -map '[MID]' mid.wav -map '[HIGH]' high.wav
        

acrusher

Reduce audio bit resolution.

This filter is bit crusher with enhanced functionality. A bit crusher is used to audibly reduce number of bits an audio signal is sampled with. This doesn't change the bit depth at all, it just produces the effect. Material reduced in bit depth sounds more harsh and "digital". This filter is able to even round to continuous values instead of discrete bit depths. Additionally it has a D/C offset which results in different crushing of the lower and the upper half of the signal. An Anti-Aliasing setting is able to produce "softer" crushing sounds.

Another feature of this filter is the logarithmic mode. This setting switches from linear distances between bits to logarithmic ones. The result is a much more "natural" sounding crusher which doesn't gate low signals for example. The human ear has a logarithmic perception, so this kind of crushing is much more pleasant. Logarithmic crushing is also able to get anti-aliased.

The filter accepts the following options:

Set level in.
Set level out.
Set bit reduction.
mix
Set mixing amount.
Can be linear: "lin" or logarithmic: "log".
Set DC.
aa
Set anti-aliasing.
Set sample reduction.
Enable LFO. By default disabled.
Set LFO range.
Set LFO rate.

Commands

This filter supports the all above options as commands.

acue

Delay audio filtering until a given wallclock timestamp. See the cue filter.

adeclick

Remove impulsive noise from input audio.

Samples detected as impulsive noise are replaced by interpolated samples using autoregressive modelling.

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

It accepts the following values:

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

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

It accepts the following values:

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

Set list of delays in milliseconds for each channel separated by '|'. Unused delays will be silently ignored. If number of given delays is smaller than number of channels all remaining channels will not be delayed. If you want to delay exact number of samples, append 'S' to number. If you want instead to delay in seconds, append 's' to number.
Use last set delay for all remaining channels. By default is disabled. This option if enabled changes how option "delays" is interpreted.

Examples

  • Delay first channel by 1.5 seconds, the third channel by 0.5 seconds and leave the second channel (and any other channels that may be present) unchanged.

            adelay=1500|0|500
        
  • Delay second channel by 500 samples, the third channel by 700 samples and leave the first channel (and any other channels that may be present) unchanged.

            adelay=0|500S|700S
        
  • Delay all channels by same number of samples:

            adelay=delays=64S:all=1
        

adenorm

Remedy denormals in audio by adding extremely low-level noise.

This filter shall be placed before any filter that can produce denormals.

A description of the accepted parameters follows.

Set level of added noise in dB. Default is "-351". Allowed range is from -451 to -90.
Set type of added noise.
Add DC signal.
Add AC signal.
Add square signal.
pulse
Add pulse signal.

Default is "dc".

Commands

This filter supports the all above options as commands.

aderivative, aintegral

Compute derivative/integral of audio stream.

Applying both filters one after another produces original audio.

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.

Set input gain of reflected signal. Default is 0.6.
Set output gain of reflected signal. Default is 0.3.
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.
Set list of loudness of reflected signals separated by '|'. Allowed range for each "decay" is "(0 - 1.0]". Default is 0.5.

Examples

  • Make it sound as if there are twice as many instruments as are actually playing:

            aecho=0.8:0.88:60:0.4
        
  • If delay is very short, then it sounds like a (metallic) robot playing music:

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

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

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

aemphasis

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

The filter accepts the following options:

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

Commands

This filter supports the all above options as commands.

aeval

Modify an audio signal according to the specified expressions.

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

It accepts the following parameters:

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.
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:

channel number of the current expression
number of the evaluated sample, starting from 0
sample rate
time of the evaluated sample expressed in seconds
input and output number of channels
the value of input channel with number CH

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

Examples

  • Half volume:

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

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

aexciter

An exciter is used to produce high sound that is not present in the original signal. This is done by creating harmonic distortions of the signal which are restricted in range and added to the original signal. An Exciter raises the upper end of an audio signal without simply raising the higher frequencies like an equalizer would do to create a more "crisp" or "brilliant" sound.

The filter accepts the following options:

Set input level prior processing of signal. Allowed range is from 0 to 64. Default value is 1.
Set output level after processing of signal. Allowed range is from 0 to 64. Default value is 1.
Set the amount of harmonics added to original signal. Allowed range is from 0 to 64. Default value is 1.
Set the amount of newly created harmonics. Allowed range is from 0.1 to 10. Default value is 8.5.
blend
Set the octave of newly created harmonics. Allowed range is from -10 to 10. Default value is 0.
Set the lower frequency limit of producing harmonics in Hz. Allowed range is from 2000 to 12000 Hz. Default is 7500 Hz.
Set the upper frequency limit of producing harmonics. Allowed range is from 9999 to 20000 Hz. If value is lower than 10000 Hz no limit is applied.
Mute the original signal and output only added harmonics. By default is disabled.

Commands

This filter supports the all above options as commands.

afade

Apply fade-in/out effect to input audio.

A description of the accepted parameters follows.

Specify the effect type, can be either "in" for fade-in, or "out" for a fade-out effect. Default is "in".
Specify the number of the start sample for starting to apply the fade effect. Default is 0.
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.
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.
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.
Set curve for fade transition.

It accepts the following values:

select triangular, linear slope (default)
select quarter of sine wave
select half of sine wave
select exponential sine wave
select logarithmic
select inverted parabola
select quadratic
select cubic
select square root
select cubic root
select parabola
select exponential
select inverted quarter of sine wave
select inverted half of sine wave
select double-exponential seat
select double-exponential sigmoid
select logistic sigmoid
sinc
select sine cardinal function
select inverted sine cardinal function
no fade applied

Commands

This filter supports the all above options as commands.

Examples

  • Fade in first 15 seconds of audio:

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

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

afftdn

Denoise audio samples with FFT.

A description of the accepted parameters follows.

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

It accepts the following values:

Select white noise.
Select vinyl noise.
Select shellac noise.
Select custom noise, defined in "bn" option.

Default value is white noise.

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

It accepts the following values:

Pass input unchanged.
Pass noise filtered out.
Pass only noise.

Default value is o.

Commands

This filter supports the following commands:

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.
Change noise reduction. Argument is single float number. Syntax for the command is : "noise_reduction"
Change noise floor. Argument is single float number. Syntax for the command is : "noise_floor"
Change output mode operation. Syntax for the command is : "i", "o" or "n" string.

afftfilt

Apply arbitrary expressions to samples in frequency domain.

Set frequency domain real expression for each separate channel separated by '|'. Default is "re". If the number of input channels is greater than the number of expressions, the last specified expression is used for the remaining output channels.
Set frequency domain imaginary expression for each separate channel separated by '|'. Default is "im".

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

sr
sample rate
current frequency bin number
number of available bins
channel number of the current expression
number of channels
current frame pts
current real part of frequency bin of current channel
current imaginary part of frequency bin of current channel
Return the value of real part of frequency bin at location (bin,channel)
Return the value of imaginary part of frequency bin at location (bin,channel)
Set window size. Allowed range is from 16 to 131072. Default is 4096
Set window function. Default is "hann".
Set window overlap. If set to 1, the recommended overlap for selected window function will be picked. Default is 0.75.

Examples

  • Leave almost only low frequencies in audio:

            afftfilt="'real=re * (1-clip((b/nb)*b,0,1))':imag='im * (1-clip((b/nb)*b,0,1))'"
        
  • Apply robotize effect:

            afftfilt="real='hypot(re,im)*sin(0)':imag='hypot(re,im)*cos(0)':win_size=512:overlap=0.75"
        
  • Apply whisper effect:

            afftfilt="real='hypot(re,im)*cos((random(0)*2-1)*2*3.14)':imag='hypot(re,im)*sin((random(1)*2-1)*2*3.14)':win_size=128:overlap=0.8"
        

afir

Apply an arbitrary Finite Impulse Response filter.

This filter is designed for applying long FIR filters, up to 60 seconds long.

It can be used as component for digital crossover filters, room equalization, cross talk cancellation, wavefield synthesis, auralization, ambiophonics, ambisonics and spatialization.

This filter uses the streams higher than first one as FIR coefficients. If the non-first stream holds a single channel, it will be used for all input channels in the first stream, otherwise the number of channels in the non-first stream must be same as the number of channels in the first stream.

It accepts the following parameters:

Set dry gain. This sets input gain.
Set wet gain. This sets final output gain.
Set Impulse Response filter length. Default is 1, which means whole IR is processed.
Enable applying gain measured from power of IR.

Set which approach to use for auto gain measurement.

Do not apply any gain.
select peak gain, very conservative approach. This is default value.
select DC gain, limited application.
select gain to noise approach, this is most popular one.
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.
Set format of IR stream. Can be "mono" or "input". Default is "input".
Set max allowed Impulse Response filter duration in seconds. Default is 30 seconds. Allowed range is 0.1 to 60 seconds.
Show IR frequency response, magnitude(magenta), phase(green) and group delay(yellow) in additional video stream. By default it is disabled.
Set for which IR channel to display frequency response. By default is first channel displayed. This option is used only when response is enabled.
Set video stream size. This option is used only when response is enabled.
Set video stream frame rate. This option is used only when response is enabled.
Set minimal partition size used for convolution. Default is 8192. Allowed range is from 1 to 32768. Lower values decreases latency at cost of higher CPU usage.
Set maximal partition size used for convolution. Default is 8192. Allowed range is from 8 to 32768. Lower values may increase CPU usage.
Set number of input impulse responses streams which will be switchable at runtime. Allowed range is from 1 to 32. Default is 1.
Set IR stream which will be used for convolution, starting from 0, should always be lower than supplied value by "nbirs" option. Default is 0. This option can be changed at runtime via commands.

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:

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

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

If a parameter is omitted, all values are allowed.

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

        aformat=sample_fmts=u8|s16:channel_layouts=stereo

afreqshift

Apply frequency shift to input audio samples.

The filter accepts the following options:

Specify frequency shift. Allowed range is -INT_MAX to INT_MAX. Default value is 0.0.
Set output gain applied to final output. Allowed range is from 0.0 to 1.0. Default value is 1.0.

Commands

This filter supports the all above options as commands.

agate

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

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

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

Set input level before filtering. Default is 1. Allowed range is from 0.015625 to 64.
Set the mode of operation. Can be "upward" or "downward". Default is "downward". If set to "upward" mode, higher parts of signal will be amplified, expanding dynamic range in upward direction. Otherwise, in case of "downward" lower parts of signal will be reduced.
Set the level of gain reduction when the signal is below the threshold. Default is 0.06125. Allowed range is from 0 to 1. Setting this to 0 disables reduction and then filter behaves like expander.
threshold
If a signal rises above this level the gain reduction is released. Default is 0.125. Allowed range is from 0 to 1.
Set a ratio by which the signal is reduced. Default is 2. Allowed range is from 1 to 9000.
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.
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.
Set amount of amplification of signal after processing. Default is 1. Allowed range is from 1 to 64.
Curve the sharp knee around the threshold to enter gain reduction more softly. Default is 2.828427125. Allowed range is from 1 to 8.
Choose if exact signal should be taken for detection or an RMS like one. Default is "rms". Can be "peak" or "rms".
Choose if the average level between all channels or the louder channel affects the reduction. Default is "average". Can be "average" or "maximum".

Commands

This filter supports the all above options as commands.

aiir

Apply an arbitrary Infinite Impulse Response filter.

It accepts the following parameters:

Set B/numerator/zeros/reflection coefficients.
Set A/denominator/poles/ladder coefficients.
Set channels gains.
Set input gain.
Set output gain.
Set coefficients format.
lattice-ladder function
analog transfer function
digital transfer function
Z-plane zeros/poles, cartesian (default)
Z-plane zeros/poles, polar radians
Z-plane zeros/poles, polar degrees
S-plane zeros/poles
Set type of processing.
direct processing
serial processing
parallel processing
Set filtering precision.
double-precision floating-point (default)
single-precision floating-point
32-bit integers
16-bit integers
Normalize filter coefficients, by default is enabled. Enabling it will normalize magnitude response at DC to 0dB.
mix
How much to use filtered signal in output. Default is 1. Range is between 0 and 1.
Show IR frequency response, magnitude(magenta), phase(green) and group delay(yellow) in additional video stream. By default it is disabled.
Set for which IR channel to display frequency response. By default is first channel displayed. This option is used only when response is enabled.
Set video stream size. This option is used only when response is enabled.

Coefficients in "tf" and "sf" format are separated by spaces and are in ascending order.

Coefficients in "zp" format are separated by spaces and order of coefficients doesn't matter. Coefficients in "zp" format are complex numbers with i imaginary unit.

Different coefficients and gains can be provided for every channel, in such case use '|' to separate coefficients or gains. Last provided coefficients will be used for all remaining channels.

Examples

  • Apply 2 pole elliptic notch at around 5000Hz for 48000 Hz sample rate:

            aiir=k=1:z=7.957584807809675810E-1 -2.575128568908332300 3.674839853930788710 -2.57512875289799137 7.957586296317130880E-1:p=1 -2.86950072432325953 3.63022088054647218 -2.28075678147272232 6.361362326477423500E-1:f=tf:r=d
        
  • Same as above but in "zp" format:

            aiir=k=0.79575848078096756:z=0.80918701+0.58773007i 0.80918701-0.58773007i 0.80884700+0.58784055i 0.80884700-0.58784055i:p=0.63892345+0.59951235i 0.63892345-0.59951235i 0.79582691+0.44198673i 0.79582691-0.44198673i:f=zp:r=s
        
  • Apply 3-rd order analog normalized Butterworth low-pass filter, using analog transfer function format:

            aiir=z=1.3057 0 0 0:p=1.3057 2.3892 2.1860 1:f=sf:r=d
        

alimiter

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

The filter accepts the following options:

Set input gain. Default is 1.
Set output gain. Default is 1.
Don't let signals above this level pass the limiter. Default is 1.
The limiter will reach its attenuation level in this amount of time in milliseconds. Default is 5 milliseconds.
Come back from limiting to attenuation 1.0 in this amount of milliseconds. Default is 50 milliseconds.
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.
Select how much the release time is affected by ASC, 0 means nearly no changes in release time while 1 produces higher release times.
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:

Set frequency in Hz.
Set method to specify band-width of filter.
Hz
Q-Factor
octave
slope
kHz
Specify the band-width of a filter in width_type units.
How much to use filtered signal in output. Default is 1. Range is between 0 and 1.
Specify which channels to filter, by default all available are filtered.
Normalize biquad coefficients, by default is disabled. Enabling it will normalize magnitude response at DC to 0dB.
Set the filter order, can be 1 or 2. Default is 2.
Set transform type of IIR filter.
Set precison of filtering.
Pick automatic sample format depending on surround filters.
Always use signed 16-bit.
Always use signed 32-bit.
Always use float 32-bit.
Always use float 64-bit.

Commands

This filter supports the following commands:

Change allpass frequency. Syntax for the command is : "frequency"
Change allpass width_type. Syntax for the command is : "width_type"
Change allpass width. Syntax for the command is : "width"
Change allpass mix. Syntax for the command is : "mix"

aloop

Loop audio samples.

The filter accepts the following options:

loop
Set the number of loops. Setting this value to -1 will result in infinite loops. Default is 0.
Set maximal number of samples. Default is 0.
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:

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:

The number of inputs. If unspecified, it defaults to 2.
How to determine the end-of-stream.
The duration of the longest input. (default)
The duration of the shortest input.
The duration of the first input.
The transition time, in seconds, for volume renormalization when an input stream ends. The default value is 2 seconds.
Specify weight of each input audio stream as sequence. Each weight is separated by space. By default all inputs have same weight.
normalize
Always scale inputs instead of only doing summation of samples. Beware of heavy clipping if inputs are not normalized prior or after filtering by this filter if this option is disabled. By default is enabled.

Commands

This filter supports the following commands:

Syntax is same as option with same name.

amultiply

Multiply first audio stream with second audio stream and store result in output audio stream. Multiplication is done by multiplying each sample from first stream with sample at same position from second stream.

With this element-wise multiplication one can create amplitude fades and amplitude modulations.

anequalizer

High-order parametric multiband equalizer for each channel.

It accepts the following parameters:

This option string is in format: "cchn f=cf w=w g=g t=f | ..." Each equalizer band is separated by '|'.
Set channel number to which equalization will be applied. If input doesn't have that channel the entry is ignored.
Set central frequency for band. If input doesn't have that frequency the entry is ignored.
Set band width in Hertz.
Set band gain in dB.
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.
Set video stream size. Only useful if curves option is activated.
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.
Set frequency scale used to draw frequency response in video output. Can be linear or logarithmic. Default is logarithmic.
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:

Alter existing filter parameters. Syntax for the commands is : "fN|f=freq|w=width|g=gain"

fN is existing filter number, starting from 0, if no such filter is available error is returned. freq set new frequency parameter. width set new width parameter in Hertz. gain set new gain parameter in dB.

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

anlmdn

Reduce broadband noise in audio samples using Non-Local Means algorithm.

Each sample is adjusted by looking for other samples with similar contexts. This context similarity is defined by comparing their surrounding patches of size p. Patches are searched in an area of r around the sample.

The filter accepts the following options:

Set denoising strength. Allowed range is from 0.00001 to 10. Default value is 0.00001.
Set patch radius duration. Allowed range is from 1 to 100 milliseconds. Default value is 2 milliseconds.
Set research radius duration. Allowed range is from 2 to 300 milliseconds. Default value is 6 milliseconds.
Set the output mode.

It accepts the following values:

Pass input unchanged.
Pass noise filtered out.
Pass only noise.

Default value is o.

Set smooth factor. Default value is 11. Allowed range is from 1 to 15.

Commands

This filter supports the all above options as commands.

anlms

Apply Normalized Least-Mean-Squares algorithm to the first audio stream using the second audio stream.

This adaptive filter is used to mimic a desired filter by finding the filter coefficients that relate to producing the least mean square of the error signal (difference between the desired, 2nd input audio stream and the actual signal, the 1st input audio stream).

A description of the accepted options follows.

Set filter order.
Set filter mu.
Set the filter eps.
Set the filter leakage.
It accepts the following values:
Pass the 1st input.
Pass the 2nd input.
Pass filtered samples.
Pass difference between desired and filtered samples.

Default value is o.

Examples

One of many usages of this filter is noise reduction, input audio is filtered with same samples that are delayed by fixed amount, one such example for stereo audio is:

        asplit[a][b],[a]adelay=32S|32S[a],[b][a]anlms=order=128:leakage=0.0005:mu=.5:out_mode=o
    

Commands

This filter supports the same commands as options, excluding option "order".

anull

Pass the audio source unchanged to the output.

apad

Pad the end of an audio stream with silence.

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

A description of the accepted options follows.

Set silence packet size. Default value is 4096.
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.
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.
Specify the duration of samples of silence to add. See the Time duration section in the ffmpeg-utils(1) manual for the accepted syntax. Used only if set to non-zero value.
Specify the minimum total duration in the output audio stream. See the Time duration section in the ffmpeg-utils(1) manual for the accepted syntax. Used only if set to non-zero value. If the value is longer than the input audio length, silence is added to the end, until the value is reached. This option is mutually exclusive with pad_dur

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

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.

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

It accepts the following values:

aphaseshift

Apply phase shift to input audio samples.

The filter accepts the following options:

Specify phase shift. Allowed range is from -1.0 to 1.0. Default value is 0.0.
Set output gain applied to final output. Allowed range is from 0.0 to 1.0. Default value is 1.0.

Commands

This filter supports the all above options as commands.

apulsator

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

The filter accepts the following options:

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

arnndn

Reduce noise from speech using Recurrent Neural Networks.

This filter accepts the following options:

Set train model file to load. This option is always required.
mix
Set how much to mix filtered samples into final output. Allowed range is from -1 to 1. Default value is 1. Negative values are special, they set how much to keep filtered noise in the final filter output. Set this option to -1 to hear actual noise removed from input signal.

Commands

This filter supports the all above options as commands.

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:

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.
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:

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:

The (sequential) number of the input frame, starting from 0.
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.
The presentation timestamp of the input frame in seconds.
position of the frame in the input stream, -1 if this information in unavailable and/or meaningless (for example in case of synthetic audio)
The sample format.
The channel layout.
The sample rate for the audio frame.
The number of samples (per channel) in the frame.
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.
A list of Adler-32 checksums for each data plane.

asoftclip

Apply audio soft clipping.

Soft clipping is a type of distortion effect where the amplitude of a signal is saturated along a smooth curve, rather than the abrupt shape of hard-clipping.

This filter accepts the following options:

Set type of soft-clipping.

It accepts the following values:

threshold
Set threshold from where to start clipping. Default value is 0dB or 1.
Set gain applied to output. Default value is 0dB or 1.
Set additional parameter which controls sigmoid function.
Set oversampling factor.

Commands

This filter supports the all above options as commands.

asr

Automatic Speech Recognition

This filter uses PocketSphinx for speech recognition. To enable compilation of this filter, you need to configure FFmpeg with "--enable-pocketsphinx".

It accepts the following options:

Set sampling rate of input audio. Defaults is 16000. This need to match speech models, otherwise one will get poor results.
Set dictionary containing acoustic model files.
Set pronunciation dictionary.
Set language model file.
Set language model set.
Set which language model to use.
Set output for log messages.

The filter exports recognized speech as the frame metadata "lavfi.asr.text".

astats

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

It accepts the following option:

Short window length in seconds, used for peak and trough RMS measurement. Default is 0.05 (50 milliseconds). Allowed range is "[0.01 - 10]".
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 Noise_floor Noise_floor_count Bit_depth Dynamic_range Zero_crossings Zero_crossings_rate Number_of_NaNs Number_of_Infs Number_of_denormals

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 Noise_floor Noise_floor_count Bit_depth Number_of_samples Number_of_NaNs Number_of_Infs Number_of_denormals

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.

Set number of frame after which stats are going to be recalculated. Default is disabled.
Select the entries which need to be measured per channel. The metadata keys can be used as flags, default is all which measures everything. none disables all per channel measurement.
Select the entries which need to be measured overall. The metadata keys can be used as flags, default is all which measures everything. none disables all overall measurement.

A description of each shown parameter follows:

Mean amplitude displacement from zero.
Minimal sample level.
Maximal sample level.
Minimal difference between two consecutive samples.
Maximal difference between two consecutive samples.
Mean difference between two consecutive samples. The average of each difference between two consecutive samples.
Root Mean Square difference between two consecutive samples.
Standard peak and RMS level measured in dBFS.
Peak and trough values for RMS level measured over a short window.
Standard ratio of peak to RMS level (note: not in dB).
Flatness (i.e. consecutive samples with the same value) of the signal at its peak levels (i.e. either Min level or Max level).
Number of occasions (not the number of samples) that the signal attained either Min level or Max level.
Minimum local peak measured in dBFS over a short window.
Number of occasions (not the number of samples) that the signal attained Noise floor.
Overall bit depth of audio. Number of bits used for each sample.
Measured dynamic range of audio in dB.
Number of points where the waveform crosses the zero level axis.
Rate of Zero crossings and number of audio samples.

asubboost

Boost subwoofer frequencies.

The filter accepts the following options:

Set dry gain, how much of original signal is kept. Allowed range is from 0 to 1. Default value is 0.7.
Set wet gain, how much of filtered signal is kept. Allowed range is from 0 to 1. Default value is 0.7.
Set delay line decay gain value. Allowed range is from 0 to 1. Default value is 0.7.
Set delay line feedback gain value. Allowed range is from 0 to 1. Default value is 0.9.
Set cutoff frequency in Hertz. Allowed range is 50 to 900. Default value is 100.
Set slope amount for cutoff frequency. Allowed range is 0.0001 to 1. Default value is 0.5.
Set delay. Allowed range is from 1 to 100. Default value is 20.

Commands

This filter supports the all above options as commands.

asubcut

Cut subwoofer frequencies.

This filter allows to set custom, steeper roll off than highpass filter, and thus is able to more attenuate frequency content in stop-band.

The filter accepts the following options:

Set cutoff frequency in Hertz. Allowed range is 2 to 200. Default value is 20.
Set filter order. Available values are from 3 to 20. Default value is 10.
Set input gain level. Allowed range is from 0 to 1. Default value is 1.

Commands

This filter supports the all above options as commands.

asupercut

Cut super frequencies.

The filter accepts the following options:

Set cutoff frequency in Hertz. Allowed range is 20000 to 192000. Default value is 20000.
Set filter order. Available values are from 3 to 20. Default value is 10.
Set input gain level. Allowed range is from 0 to 1. Default value is 1.

Commands

This filter supports the all above options as commands.

asuperpass

Apply high order Butterworth band-pass filter.

The filter accepts the following options:

Set center frequency in Hertz. Allowed range is 2 to 999999. Default value is 1000.
Set filter order. Available values are from 4 to 20. Default value is 4.
Set Q-factor. Allowed range is from 0.01 to 100. Default value is 1.
Set input gain level. Allowed range is from 0 to 2. Default value is 1.

Commands

This filter supports the all above options as commands.

asuperstop

Apply high order Butterworth band-stop filter.

The filter accepts the following options:

Set center frequency in Hertz. Allowed range is 2 to 999999. Default value is 1000.
Set filter order. Available values are from 4 to 20. Default value is 4.
Set Q-factor. Allowed range is from 0.01 to 100. Default value is 1.
Set input gain level. Allowed range is from 0 to 2. Default value is 1.

Commands

This filter supports the all above options as commands.

atempo

Adjust audio tempo.

The filter accepts exactly one parameter, the audio tempo. If not specified then the filter will assume nominal 1.0 tempo. Tempo must be in the [0.5, 100.0] range.

Note that tempo greater than 2 will skip some samples rather than blend them in. If for any reason this is a concern it is always possible to daisy-chain several instances of atempo to achieve the desired product tempo.

Examples

  • Slow down audio to 80% tempo:

            atempo=0.8
        
  • To speed up audio to 300% tempo:

            atempo=3
        
  • To speed up audio to 300% tempo by daisy-chaining two atempo instances:

            atempo=sqrt(3),atempo=sqrt(3)
        

Commands

This filter supports the following commands:

Change filter tempo scale factor. Syntax for the command is : "tempo"

atrim

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

It accepts the following parameters:

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.
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.
Same as start, except this option sets the start timestamp in samples instead of seconds.
Same as end, except this option sets the end timestamp in samples instead of seconds.
The maximum duration of the output in seconds.
The number of the first sample that should be output.
The number of the first sample that should be dropped.

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

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

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

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

Examples:

  • Drop everything except the second minute of input:

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

            ffmpeg -i INPUT -af atrim=end_sample=1000
        

axcorrelate

Calculate normalized cross-correlation between two input audio streams.

Resulted samples are always between -1 and 1 inclusive. If result is 1 it means two input samples are highly correlated in that selected segment. Result 0 means they are not correlated at all. If result is -1 it means two input samples are out of phase, which means they cancel each other.

The filter accepts the following options:

Set size of segment over which cross-correlation is calculated. Default is 256. Allowed range is from 2 to 131072.
Set algorithm for cross-correlation. Can be "slow" or "fast". Default is "slow". Fast algorithm assumes mean values over any given segment are always zero and thus need much less calculations to make. This is generally not true, but is valid for typical audio streams.

Examples

Calculate correlation between channels in stereo audio stream:

        ffmpeg -i stereo.wav -af channelsplit,axcorrelate=size=1024:algo=fast correlation.wav
    

bandpass

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

The filter accepts the following options:

Set the filter's central frequency. Default is 3000.
Constant skirt gain if set to 1. Defaults to 0.
Set method to specify band-width of filter.
Hz
Q-Factor
octave
slope
kHz
Specify the band-width of a filter in width_type units.
How much to use filtered signal in output. Default is 1. Range is between 0 and 1.
Specify which channels to filter, by default all available are filtered.
Normalize biquad coefficients, by default is disabled. Enabling it will normalize magnitude response at DC to 0dB.
Set transform type of IIR filter.
Set precison of filtering.
Pick automatic sample format depending on surround filters.
Always use signed 16-bit.
Always use signed 32-bit.
Always use float 32-bit.
Always use float 64-bit.

Commands

This filter supports the following commands:

Change bandpass frequency. Syntax for the command is : "frequency"
Change bandpass width_type. Syntax for the command is : "width_type"
Change bandpass width. Syntax for the command is : "width"
Change bandpass mix. Syntax for the command is : "mix"

bandreject

Apply a two-pole Butterworth band-reject filter with central frequency frequency, and (3dB-point) band-width width. The filter roll off at 6dB per octave (20dB per decade).

The filter accepts the following options:

Set the filter's central frequency. Default is 3000.
Set method to specify band-width of filter.
Hz
Q-Factor
octave
slope
kHz
Specify the band-width of a filter in width_type units.
How much to use filtered signal in output. Default is 1. Range is between 0 and 1.
Specify which channels to filter, by default all available are filtered.
Normalize biquad coefficients, by default is disabled. Enabling it will normalize magnitude response at DC to 0dB.
Set transform type of IIR filter.
Set precison of filtering.
Pick automatic sample format depending on surround filters.
Always use signed 16-bit.
Always use signed 32-bit.
Always use float 32-bit.
Always use float 64-bit.

Commands

This filter supports the following commands:

Change bandreject frequency. Syntax for the command is : "frequency"
Change bandreject width_type. Syntax for the command is : "width_type"
Change bandreject width. Syntax for the command is : "width"
Change bandreject mix. Syntax for the command is : "mix"

bass, lowshelf

Boost or cut the bass (lower) frequencies of the audio using a two-pole shelving filter with a response similar to that of a standard hi-fi's tone-controls. This is also known as shelving equalisation (EQ).

The filter accepts the following options:

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.
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.
Set method to specify band-width of filter.
Hz
Q-Factor
octave
slope
kHz
Determine how steep is the filter's shelf transition.
Set number of poles. Default is 2.
How much to use filtered signal in output. Default is 1. Range is between 0 and 1.
Specify which channels to filter, by default all available are filtered.
Normalize biquad coefficients, by default is disabled. Enabling it will normalize magnitude response at DC to 0dB.
Set transform type of IIR filter.
Set precison of filtering.
Pick automatic sample format depending on surround filters.
Always use signed 16-bit.
Always use signed 32-bit.
Always use float 32-bit.
Always use float 64-bit.

Commands

This filter supports the following commands:

Change bass frequency. Syntax for the command is : "frequency"
Change bass width_type. Syntax for the command is : "width_type"
Change bass width. Syntax for the command is : "width"
Change bass gain. Syntax for the command is : "gain"
Change bass mix. Syntax for the command is : "mix"

biquad

Apply a biquad IIR filter with the given coefficients. Where b0, b1, b2 and a0, a1, a2 are the numerator and denominator coefficients respectively. and channels, c specify which channels to filter, by default all available are filtered.

Commands

This filter supports the following commands:

Change biquad parameter. Syntax for the command is : "value"
How much to use filtered signal in output. Default is 1. Range is between 0 and 1.
Specify which channels to filter, by default all available are filtered.
Normalize biquad coefficients, by default is disabled. Enabling it will normalize magnitude response at DC to 0dB.
Set transform type of IIR filter.
Set precison of filtering.
Pick automatic sample format depending on surround filters.
Always use signed 16-bit.
Always use signed 32-bit.
Always use float 32-bit.
Always use float 64-bit.

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:

Pre-defined crossfeed level.
Default level (fcut=700, feed=50).
Chu Moy circuit (fcut=700, feed=60).
Jan Meier circuit (fcut=650, feed=95).
Cut frequency (in Hz).
Feed level (in Hz).

channelmap

Remap input channels to new locations.

It accepts the following parameters:

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.
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:

The channel layout of the input stream. The default is "stereo".
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:

Set input gain. Default is 0.4.
Set output gain. Default is 0.4.
Set delays. A typical delay is around 40ms to 60ms.
Set decays.
Set speeds.
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:

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.
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".

Set the curve radius in dB for all joints. It defaults to 0.01.
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.
Set a delay, in seconds. The input audio is analyzed immediately, but audio is delayed before being fed to the volume adjuster. Specifying a delay approximately equal to the attack/decay times allows the filter to effectively operate in predictive rather than reactive mode. It defaults to 0.

Examples

  • Make music with both quiet and loud passages suitable for listening to in a noisy environment:

            compand=.3|.3:1|1:-90/-60|-60/-40|-40/-30|-20/-20:6:0:-90:0.2
        

    Another example for audio with whisper and explosion parts:

            compand=0|0:1|1:-90/-900|-70/-70|-30/-9|0/-3:6:0:0:0
        
  • A noise gate for when the noise is at a lower level than the signal:

            compand=.1|.1:.2|.2:-900/-900|-50.1/-900|-50/-50:.01:0:-90:.1
        
  • Here is another noise gate, this time for when the noise is at a higher level than the signal (making it, in some ways, similar to squelch):

            compand=.1|.1:.1|.1:-45.1/-45.1|-45/-900|0/-900:.01:45:-90:.1
        
  • 2:1 compression starting at -6dB:

            compand=points=-80/-80|-6/-6|0/-3.8|20/3.5
        
  • 2:1 compression starting at -9dB:

            compand=points=-80/-80|-9/-9|0/-5.3|20/2.9
        
  • 2:1 compression starting at -12dB:

            compand=points=-80/-80|-12/-12|0/-6.8|20/1.9
        
  • 2:1 compression starting at -18dB:

            compand=points=-80/-80|-18/-18|0/-9.8|20/0.7
        
  • 3:1 compression starting at -15dB:

            compand=points=-80/-80|-15/-15|0/-10.8|20/-5.2
        
  • Compressor/Gate:

            compand=points=-80/-105|-62/-80|-15.4/-15.4|0/-12|20/-7.6
        
  • Expander:

            compand=attacks=0:points=-80/-169|-54/-80|-49.5/-64.6|-41.1/-41.1|-25.8/-15|-10.8/-4.5|0/0|20/8.3
        
  • Hard limiter at -6dB:

            compand=attacks=0:points=-80/-80|-6/-6|20/-6
        
  • Hard limiter at -12dB:

            compand=attacks=0:points=-80/-80|-12/-12|20/-12
        
  • Hard noise gate at -35 dB:

            compand=attacks=0:points=-80/-115|-35.1/-80|-35/-35|20/20
        
  • Soft limiter:

            compand=attacks=0:points=-80/-80|-12.4/-12.4|-6/-8|0/-6.8|20/-2.8
        

compensationdelay

Compensation Delay Line is a metric based delay to compensate differing positions of microphones or speakers.

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

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

The filter accepts the following parameters:

Set millimeters distance. This is compensation distance for fine tuning. Default is 0.
Set cm distance. This is compensation distance for tightening distance setup. Default is 0.
Set meters distance. This is compensation distance for hard distance setup. Default is 0.
Set dry amount. Amount of unprocessed (dry) signal. Default is 0.
Set wet amount. Amount of processed (wet) signal. Default is 1.
Set temperature in degrees 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:

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.
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.
Set curve slope of low shelf filter. Default is 0.5. Allowed range is from 0.01 to 1.
Set input gain. Default is 0.9.
Set output gain. Default is 1.

Commands

This filter supports the all above options as commands.

crystalizer

Simple algorithm for audio noise sharpening.

This filter linearly increases differences betweeen each audio sample.

The filter accepts the following options:

Sets the intensity of effect (default: 2.0). Must be in range between -10.0 to 0 (unchanged sound) to 10.0 (maximum effect). To inverse filtering use negative value.
Enable clipping. By default is enabled.

Commands

This filter supports the all above options as commands.

dcshift

Apply a DC shift to the audio.

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

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

deesser

Apply de-essing to the audio samples.

Set intensity for triggering de-essing. Allowed range is from 0 to 1. Default is 0.
Set amount of ducking on treble part of sound. Allowed range is from 0 to 1. Default is 0.5.
How much of original frequency content to keep when de-essing. Allowed range is from 0 to 1. Default is 0.5.
Set the output mode.

It accepts the following values:

Pass input unchanged.
Pass ess filtered out.
Pass only ess.

Default value is o.

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:

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.

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.
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.
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.
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.
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.
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.
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.
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.
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.
Set the target threshold value. This specifies the lowest permissible magnitude level for the audio input which will be normalized. If input frame volume is above this value frame will be normalized. Otherwise frame may not be normalized at all. The default value is set to 0, which means all input frames will be normalized. This option is mostly useful if digital noise is not wanted to be amplified.

Commands

This filter supports the all above options as commands.

earwax

Make audio easier to listen to on headphones.

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

Ported from SoX.

equalizer

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

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

The filter accepts the following options:

Set the filter's central frequency in Hz.
Set method to specify band-width of filter.
Hz
Q-Factor
octave
slope
kHz
Specify the band-width of a filter in width_type units.
Set the required gain or attenuation in dB. Beware of clipping when using a positive gain.
How much to use filtered signal in output. Default is 1. Range is between 0 and 1.
Specify which channels to filter, by default all available are filtered.
Normalize biquad coefficients, by default is disabled. Enabling it will normalize magnitude response at DC to 0dB.
Set transform type of IIR filter.
Set precison of filtering.
Pick automatic sample format depending on surround filters.
Always use signed 16-bit.
Always use signed 32-bit.
Always use float 32-bit.
Always use float 64-bit.

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:

Change equalizer frequency. Syntax for the command is : "frequency"
Change equalizer width_type. Syntax for the command is : "width_type"
Change equalizer width. Syntax for the command is : "width"
Change equalizer gain. Syntax for the command is : "gain"
Change equalizer mix. Syntax for the command is : "mix"

extrastereo

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

The filter accepts the following options:

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.
Enable clipping. By default is enabled.

Commands

This filter supports the all above options as commands.

firequalizer

Apply FIR Equalization using arbitrary frequency response.

The filter accepts the following option:

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

and functions:

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

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

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

This option is also available as command.

Set filter delay in seconds. Higher value means more accurate. Default is 0.01.
Set filter accuracy in Hz. Lower value means more accurate. Default is 5.
Set window function. Acceptable values are:
rectangular window, useful when gain curve is already smooth
hann window (default)
hamming window
blackman window
3-terms continuous 1st derivative nuttall window
minimum 3-terms discontinuous nuttall window
4-terms continuous 1st derivative nuttall window
minimum 4-terms discontinuous nuttall (blackman-nuttall) window
blackman-harris window
tukey window
If enabled, use fixed number of audio samples. This improves speed when filtering with large delay. Default is disabled.
Enable multichannels evaluation on gain. Default is disabled.
Enable zero phase mode by subtracting timestamp to compensate delay. Default is disabled.
scale
Set scale used by gain. Acceptable values are:
linear frequency, linear gain
linear frequency, logarithmic (in dB) gain (default)
logarithmic (in octave scale where 20 Hz is 0) frequency, linear gain
logarithmic frequency, logarithmic gain
Set file for dumping, suitable for gnuplot.
Set scale for dumpfile. Acceptable values are same with scale option. Default is linlog.
Enable 2-channel convolution using complex FFT. This improves speed significantly. Default is disabled.
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:

Set base delay in milliseconds. Range from 0 to 30. Default value is 0.
Set added sweep delay in milliseconds. Range from 0 to 10. Default value is 2.
Set percentage regeneration (delayed signal feedback). Range from -95 to 95. Default value is 0.
Set percentage of delayed signal mixed with original. Range from 0 to 100. Default value is 71.
Set sweeps per second (Hz). Range from 0.1 to 10. Default value is 0.5.
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.
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:

Set input level. By default is 1, or 0dB
Set output level. By default is 1, or 0dB.
Set gain applied to side part of signal. By default is 1.
Set kind of middle source. Can be one of the following:
Pick left channel.
Pick right channel.
Pick middle part signal of stereo image.
Pick side part signal of stereo image.
Change middle phase. By default is disabled.
Set left channel delay. By default is 2.05 milliseconds.
Set left channel balance. By default is -1.
Set left channel gain. By default is 1.
Change left phase. By default is disabled.
Set right channel delay. By defaults is 2.12 milliseconds.
Set right channel balance. By default is 1.
Set right channel gain. By default is 1.
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 any automatic format conversion or resampling in the filter graph.
Process the stereo channels together. If target_gain does not match between channels, consider it invalid and use the last valid target_gain.
Set the code detect timer period in ms.
Always extend peaks above -3dBFS even if PE isn't signaled.
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:

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.
Set gain applied to audio. Value is in dB. Default is 0.
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.
Set custom gain for LFE channels. Value is in dB. Default is 0.
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.
Set format of hrir stream. Default value is stereo. Alternative value is multich. If value is set to stereo, number of additional streams should be greater or equal to number of input channels in first input stream. Also each additional stream should have stereo number of channels. If value is set to multich, number of additional streams should be exactly one. Also number of input channels of additional stream should be equal or greater than twice number of channels of first input stream.

Examples

  • Full example using wav files as coefficients with amovie filters for 7.1 downmix, each amovie filter use stereo file with IR coefficients as input. The files give coefficients for each position of virtual loudspeaker:

            ffmpeg -i input.wav
            -filter_complex "amovie=azi_270_ele_0_DFC.wav[sr];amovie=azi_90_ele_0_DFC.wav[sl];amovie=azi_225_ele_0_DFC.wav[br];amovie=azi_135_ele_0_DFC.wav[bl];amovie=azi_0_ele_0_DFC.wav,asplit[fc][lfe];amovie=azi_35_ele_0_DFC.wav[fl];amovie=azi_325_ele_0_DFC.wav[fr];[0:a][fl][fr][fc][lfe][bl][br][sl][sr]headphone=FL|FR|FC|LFE|BL|BR|SL|SR"
            output.wav
        
  • Full example using wav files as coefficients with amovie filters for 7.1 downmix, but now in multich hrir format.

            ffmpeg -i input.wav -filter_complex "amovie=minp.wav[hrirs];[0:a][hrirs]headphone=map=FL|FR|FC|LFE|BL|BR|SL|SR:hrir=multich"
            output.wav
        

highpass

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

The filter accepts the following options:

Set frequency in Hz. Default is 3000.
Set number of poles. Default is 2.
Set method to specify band-width of filter.
Hz
Q-Factor
octave
slope
kHz
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.
How much to use filtered signal in output. Default is 1. Range is between 0 and 1.
Specify which channels to filter, by default all available are filtered.
Normalize biquad coefficients, by default is disabled. Enabling it will normalize magnitude response at DC to 0dB.
Set transform type of IIR filter.
Set precison of filtering.
Pick automatic sample format depending on surround filters.
Always use signed 16-bit.
Always use signed 32-bit.
Always use float 32-bit.
Always use float 64-bit.

Commands

This filter supports the following commands:

Change highpass frequency. Syntax for the command is : "frequency"
Change highpass width_type. Syntax for the command is : "width_type"
Change highpass width. Syntax for the command is : "width"
Change highpass mix. Syntax for the command is : "mix"

join

Join multiple input streams into one multi-channel stream.

It accepts the following parameters:

The number of input streams. It defaults to 2.
The desired output channel layout. It defaults to stereo.
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".

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/.
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.
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.
Specify the sample rate, default to 44100. Only used if plugin have zero inputs.
Set the number of samples per channel per each output frame, default is 1024. Only used if plugin have zero inputs.
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.
Enable latency compensation, by default is disabled. Only used if plugin have inputs.

Examples

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

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

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

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

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

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

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

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

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

            ladspa=mbeq_1197:mbeq:-24|-24|-24|0|0|0|0|0|0|0|0|0|0|0|0
        
  • Reduce stereo image using "Narrower" from the "C* Audio Plugin Suite" (CAPS) library:

            ladspa=caps:Narrower
        
  • Another white noise, now using "C* Audio Plugin Suite" (CAPS) library:

            ladspa=caps:White:.2
        
  • Some fractal noise, using "C* Audio Plugin Suite" (CAPS) library:

            ladspa=caps:Fractal:c=c1=1
        
  • Dynamic volume normalization using "VLevel" plugin:

            ladspa=vlevel-ladspa:vlevel_mono
        

Commands

This filter supports the following commands:

Modify the N-th control value.

If the specified value is not valid, it is ignored and prior one is kept.

loudnorm

EBU R128 loudness normalization. Includes both dynamic and linear normalization modes. Support for both single pass (livestreams, files) and double pass (files) modes. This algorithm can target IL, LRA, and maximum true peak. In dynamic mode, to accurately detect true peaks, the audio stream will be upsampled to 192 kHz. Use the "-ar" option or "aresample" filter to explicitly set an output sample rate.

The filter accepts the following options:

Set integrated loudness target. Range is -70.0 - -5.0. Default value is -24.0.
Set loudness range target. Range is 1.0 - 20.0. Default value is 7.0.
Set maximum true peak. Range is -9.0 - +0.0. Default value is -2.0.
Measured IL of input file. Range is -99.0 - +0.0.
Measured LRA of input file. Range is 0.0 - 99.0.
Measured true peak of input file. Range is -99.0 - +99.0.
Measured threshold of input file. Range is -99.0 - +0.0.
Set offset gain. Gain is applied before the true-peak limiter. Range is -99.0 - +99.0. Default is +0.0.
Normalize by linearly scaling the source audio. "measured_I", "measured_LRA", "measured_TP", and "measured_thresh" must all be specified. Target LRA shouldn't be lower than source LRA and the change in integrated loudness shouldn't result in a true peak which exceeds the target TP. If any of these conditions aren't met, normalization mode will revert to dynamic. Options are "true" or "false". Default is "true".
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.
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:

Set frequency in Hz. Default is 500.
Set number of poles. Default is 2.
Set method to specify band-width of filter.
Hz
Q-Factor
octave
slope
kHz
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.
How much to use filtered signal in output. Default is 1. Range is between 0 and 1.
Specify which channels to filter, by default all available are filtered.
Normalize biquad coefficients, by default is disabled. Enabling it will normalize magnitude response at DC to 0dB.
Set transform type of IIR filter.
Set precison of filtering.
Pick automatic sample format depending on surround filters.
Always use signed 16-bit.
Always use signed 32-bit.
Always use float 32-bit.
Always use float 64-bit.

Examples

Lowpass only LFE channel, it LFE is not present it does nothing:

        lowpass=c=LFE
    

Commands

This filter supports the following commands:

Change lowpass frequency. Syntax for the command is : "frequency"
Change lowpass width_type. Syntax for the command is : "width_type"
Change lowpass width. Syntax for the command is : "width"
Change lowpass mix. Syntax for the command is : "mix"

lv2

Load a LV2 (LADSPA Version 2) plugin.

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

Specifies the plugin URI. You may need to escape ':'.
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.
Specify the sample rate, default to 44100. Only used if plugin have zero inputs.
Set the number of samples per channel per each output frame, default is 1024. Only used if plugin have zero inputs.
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:

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|..."

output channel layout or number of channels
output channel specification, of the form: "out_name=[gain*]in_name[(+-)[gain*]in_name...]"
output channel to define, either a channel name (FL, FR, etc.) or a channel number (c0, c1, etc.)
multiplicative coefficient for the channel, 1 leaving the volume unchanged
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:

Set tempo scale factor.
Set pitch scale factor.
Set transients detector. Possible values are:
Set detector. Possible values are:
phase
Set phase. Possible values are:
Set processing window size. Possible values are:
Set smoothing. Possible values are:
Enable formant preservation when shift pitching. Possible values are:
Set pitch quality. Possible values are:
Set channels. Possible values are:

Commands

This filter supports the following commands:

Change filter tempo scale factor. Syntax for the command is : "tempo"
Change filter pitch scale factor. Syntax for the command is : "pitch"

sidechaincompress

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

The filter accepts the following options:

Set input gain. Default is 1. Range is between 0.015625 and 64.
Set mode of compressor operation. Can be "upward" or "downward". Default is "downward".
threshold
If a signal of second stream raises above this level it will affect the gain reduction of first stream. By default is 0.125. Range is between 0.00097563 and 1.
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.
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.
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.
Set the amount by how much signal will be amplified after processing. Default is 1. Range is from 1 to 64.
Curve the sharp knee around the threshold to enter gain reduction more softly. Default is 2.82843. Range is between 1 and 8.
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".
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.
Set sidechain gain. Default is 1. Range is between 0.015625 and 64.
mix
How much to use compressed signal in output. Default is 1. Range is between 0 and 1.

Commands

This filter supports the all above options as commands.

Examples

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

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

sidechaingate

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

The filter accepts the following options:

Set input level before filtering. Default is 1. Allowed range is from 0.015625 to 64.
Set the mode of operation. Can be "upward" or "downward". Default is "downward". If set to "upward" mode, higher parts of signal will be amplified, expanding dynamic range in upward direction. Otherwise, in case of "downward" lower parts of signal will be reduced.
Set the level of gain reduction when the signal is below the threshold. Default is 0.06125. Allowed range is from 0 to 1. Setting this to 0 disables reduction and then filter behaves like expander.
threshold
If a signal rises above this level the gain reduction is released. Default is 0.125. Allowed range is from 0 to 1.
Set a ratio about which the signal is reduced. Default is 2. Allowed range is from 1 to 9000.
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.
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.
Set amount of amplification of signal after processing. Default is 1. Allowed range is from 1 to 64.
Curve the sharp knee around the threshold to enter gain reduction more softly. Default is 2.828427125. Allowed range is from 1 to 8.
Choose if exact signal should be taken for detection or an RMS like one. Default is rms. Can be peak or rms.
Choose if the average level between all channels or the louder channel affects the reduction. Default is average. Can be average or maximum.
Set sidechain gain. Default is 1. Range is from 0.015625 to 64.

Commands

This filter supports the all above options as commands.

silencedetect

Detect silence in an audio stream.

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

The printed times and duration are expressed in seconds. The "lavfi.silence_start" or "lavfi.silence_start.X" metadata key is set on the first frame whose timestamp equals or exceeds the detection duration and it contains the timestamp of the first frame of the silence.

The "lavfi.silence_duration" or "lavfi.silence_duration.X" and "lavfi.silence_end" or "lavfi.silence_end.X" metadata keys are set on the first frame after the silence. If mono is enabled, and each channel is evaluated separately, the ".X" suffixed keys are used, and "X" corresponds to the channel number.

The filter accepts the following options:

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.
Set silence duration until notification (default is 2 seconds). See the Time duration section in the ffmpeg-utils(1) manual for the accepted syntax.
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:

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.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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".
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
        
  • Trim all digital silence samples, using peak detection, from beginning to end where there is more than 0 samples of digital silence in audio and digital silence is detected in all channels at same positions in stream:

            silenceremove=window=0:detection=peak:stop_mode=all:start_mode=all:stop_periods=-1:stop_threshold=0
        

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:

Set the SOFA file used for rendering.
Set gain applied to audio. Value is in dB. Default is 0.
Set rotation of virtual loudspeakers in deg. Default is 0.
Set elevation of virtual speakers in deg. Default is 0.
Set distance in meters between loudspeakers and the listener with near-field HRTFs. Default is 1.
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.
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.
Set custom gain for LFE channels. Value is in dB. Default is 0.
Set custom frame size in number of samples. Default is 1024. Allowed range is from 1024 to 96000. Only used if option type is set to freq.
normalize
Should all IRs be normalized upon importing SOFA file. By default is enabled.
Should nearest IRs be interpolated with neighbor IRs if exact position does not match. By default is disabled.
Minphase all IRs upon loading of SOFA file. By default is disabled.
Set neighbor search angle step. Only used if option interpolate is enabled.
Set neighbor search radius step. Only used if option interpolate is enabled.

Examples

  • Using ClubFritz6 sofa file:

            sofalizer=sofa=/path/to/ClubFritz6.sofa:type=freq:radius=1
        
  • Using ClubFritz12 sofa file and bigger radius with small rotation:

            sofalizer=sofa=/path/to/ClubFritz12.sofa:type=freq:radius=2:rotation=5
        
  • Similar as above but with custom speaker positions for front left, front right, back left and back right and also with custom gain:

            "sofalizer=sofa=/path/to/ClubFritz6.sofa:type=freq:radius=2:speakers=FL 45|FR 315|BL 135|BR 225:gain=28"
        

speechnorm

Speech Normalizer.

This filter expands or compresses each half-cycle of audio samples (local set of samples all above or all below zero and between two nearest zero crossings) depending on threshold value, so audio reaches target peak value under conditions controlled by below options.

The filter accepts the following options:

Set the expansion target peak value. This specifies the highest allowed absolute amplitude level for the normalized audio input. Default value is 0.95. Allowed range is from 0.0 to 1.0.
Set the maximum expansion factor. Allowed range is from 1.0 to 50.0. Default value is 2.0. This option controls maximum local half-cycle of samples expansion. The maximum expansion would be such that local peak value reaches target peak value but never to surpass it and that ratio between new and previous peak value does not surpass this option value.
Set the maximum compression factor. Allowed range is from 1.0 to 50.0. Default value is 2.0. This option controls maximum local half-cycle of samples compression. This option is used only if threshold option is set to value greater than 0.0, then in such cases when local peak is lower or same as value set by threshold all samples belonging to that peak's half-cycle will be compressed by current compression factor.
Set the threshold value. Default value is 0.0. Allowed range is from 0.0 to 1.0. This option specifies which half-cycles of samples will be compressed and which will be expanded. Any half-cycle samples with their local peak value below or same as this option value will be compressed by current compression factor, otherwise, if greater than threshold value they will be expanded with expansion factor so that it could reach peak target value but never surpass it.
Set the expansion raising amount per each half-cycle of samples. Default value is 0.001. Allowed range is from 0.0 to 1.0. This controls how fast expansion factor is raised per each new half-cycle until it reaches expansion value. Setting this options too high may lead to distortions.
Set the compression raising amount per each half-cycle of samples. Default value is 0.001. Allowed range is from 0.0 to 1.0. This controls how fast compression factor is raised per each new half-cycle until it reaches compression value.
Specify which channels to filter, by default all available channels are filtered.
Enable inverted filtering, by default is disabled. This inverts interpretation of threshold option. When enabled any half-cycle of samples with their local peak value below or same as threshold option will be expanded otherwise it will be compressed.
Link channels when calculating gain applied to each filtered channel sample, by default is disabled. When disabled each filtered channel gain calculation is independent, otherwise when this option is enabled the minimum of all possible gains for each filtered channel is used.

Commands

This filter supports the all above options as commands.

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:

Set input level before filtering for both channels. Defaults is 1. Allowed range is from 0.015625 to 64.
Set output level after filtering for both channels. Defaults is 1. Allowed range is from 0.015625 to 64.
Set input balance between both channels. Default is 0. Allowed range is from -1 to 1.
Set output balance between both channels. Default is 0. Allowed range is from -1 to 1.
Enable softclipping. Results in analog distortion instead of harsh digital 0dB clipping. Disabled by default.
Mute the left channel. Disabled by default.
Mute the right channel. Disabled by default.
Change the phase of the left channel. Disabled by default.
Change the phase of the right channel. Disabled by default.
Set stereo mode. Available values are:
Left/Right to Left/Right, this is default.
Left/Right to Mid/Side.
Mid/Side to Left/Right.
Left/Right to Left/Left.
Left/Right to Right/Right.
Left/Right to Left + Right.
Left/Right to Right/Left.
Mid/Side to Left/Left.
Mid/Side to Right/Right.
Mid/Side to Right/Left.
Left/Right to Left - Right.
Set level of side signal. Default is 1. Allowed range is from 0.015625 to 64.
Set balance of side signal. Default is 0. Allowed range is from -1 to 1.
Set level of the middle signal. Default is 1. Allowed range is from 0.015625 to 64.
Set middle signal pan. Default is 0. Allowed range is from -1 to 1.
Set stereo base between mono and inversed channels. Default is 0. Allowed range is from -1 to 1.
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.
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.
Set balance mode for balance_in/balance_out option.

Can be one of the following:

Classic balance mode. Attenuate one channel at time. Gain is raised up to 1.
Similar as classic mode above but gain is raised up to 2.
Equal power distribution, from -6dB to +6dB range.

Commands

This filter supports the all above options as commands.

Examples

  • Apply karaoke like effect:

            stereotools=mlev=0.015625
        
  • Convert M/S signal to L/R:

            "stereotools=mode=ms>lr"
        

stereowiden

This filter enhance the stereo effect by suppressing signal common to both channels and by delaying the signal of left into right and vice versa, thereby widening the stereo effect.

The filter accepts the following options:

Time in milliseconds of the delay of left signal into right and vice versa. Default is 20 milliseconds.
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.
Set level of input signal of original channel. Default is 0.8.

Commands

This filter supports the all above options except "delay" as commands.

superequalizer

Apply 18 band equalizer.

The filter accepts the following options:

1b
Set 65Hz band gain.
2b
Set 92Hz band gain.
3b
Set 131Hz band gain.
4b
Set 185Hz band gain.
5b
Set 262Hz band gain.
6b
Set 370Hz band gain.
7b
Set 523Hz band gain.
8b
Set 740Hz band gain.
9b
Set 1047Hz band gain.
10b
Set 1480Hz band gain.
11b
Set 2093Hz band gain.
12b
Set 2960Hz band gain.
13b
Set 4186Hz band gain.
14b
Set 5920Hz band gain.
15b
Set 8372Hz band gain.
16b
Set 11840Hz band gain.
17b
Set 16744Hz band gain.
18b
Set 20000Hz band gain.

surround

Apply audio surround upmix filter.

This filter allows to produce multichannel output from audio stream.

The filter accepts the following options:

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.

Set input channel layout. By default, this is stereo.

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

Set input volume level. By default, this is 1.
Set output volume level. By default, this is 1.
Enable LFE channel output if output channel layout has it. By default, this is enabled.
Set LFE low cut off frequency. By default, this is 128 Hz.
Set LFE high cut off frequency. By default, this is 256 Hz.
Set LFE mode, can be add or sub. Default is add. In add mode, LFE channel is created from input audio and added to output. In sub mode, LFE channel is created from input audio and added to output but also all non-LFE output channels are subtracted with output LFE channel.
Set angle of stereo surround transform, Allowed range is from 0 to 360. Default is 90.
Set front center input volume. By default, this is 1.
Set front center output volume. By default, this is 1.
Set front left input volume. By default, this is 1.
Set front left output volume. By default, this is 1.
Set front right input volume. By default, this is 1.
Set front right output volume. By default, this is 1.
Set side left input volume. By default, this is 1.
Set side left output volume. By default, this is 1.
Set side right input volume. By default, this is 1.
Set side right output volume. By default, this is 1.
Set back left input volume. By default, this is 1.
Set back left output volume. By default, this is 1.
Set back right input volume. By default, this is 1.
Set back right output volume. By default, this is 1.
Set back center input volume. By default, this is 1.
Set back center output volume. By default, this is 1.
Set LFE input volume. By default, this is 1.
Set LFE output volume. By default, this is 1.
Set spread usage of stereo image across X axis for all channels.
Set spread usage of stereo image across Y axis for all channels.
Set spread usage of stereo image across X axis for each channel.
Set spread usage of stereo image across Y axis for each channel.
Set window size. Allowed range is from 1024 to 65536. Default size is 4096.
Set window function.

It accepts the following values:

Default is "hann".

Set window overlap. If set to 1, the recommended overlap for selected window function will be picked. Default is 0.5.

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:

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.
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.
Set method to specify band-width of filter.
Hz
Q-Factor
octave
slope
kHz
Determine how steep is the filter's shelf transition.
Set number of poles. Default is 2.
How much to use filtered signal in output. Default is 1. Range is between 0 and 1.
Specify which channels to filter, by default all available are filtered.
Normalize biquad coefficients, by default is disabled. Enabling it will normalize magnitude response at DC to 0dB.
Set transform type of IIR filter.
Set precison of filtering.
Pick automatic sample format depending on surround filters.
Always use signed 16-bit.
Always use signed 32-bit.
Always use float 32-bit.
Always use float 64-bit.

Commands

This filter supports the following commands:

Change treble frequency. Syntax for the command is : "frequency"
Change treble width_type. Syntax for the command is : "width_type"
Change treble width. Syntax for the command is : "width"
Change treble gain. Syntax for the command is : "gain"
Change treble mix. Syntax for the command is : "mix"

tremolo

Sinusoidal amplitude modulation.

The filter accepts the following options:

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.
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:

Modulation frequency in Hertz. Range is 0.1 - 20000.0. Default value is 5.0 Hz.
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".

This parameter represents the mathematical precision.

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

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

Default value for replaygain_preamp is 0.0.

Prevent clipping by limiting the gain applied.

Default value for replaygain_noclip is 1.

Set when the volume expression is evaluated.

It accepts the following values:

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

Default value is once.

The volume expression can contain the following parameters.

frame number (starting at zero)
number of channels
number of samples consumed by the filter
number of samples in the current frame
original frame position in the file
frame PTS
sample rate
PTS at start of stream
time at start of stream
frame time
timestamp timebase
volume
last set volume value

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

Commands

This filter supports the following commands:

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

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

Examples

  • Halve the input audio volume:

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

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

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

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

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

volumedetect

Detect the volume of the input video.

The filter has no parameters. 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/buffersrc.h.

It accepts the following parameters:

The timebase which will be used for timestamps of submitted frames. It must be either a floating-point number or in numerator/denominator form.
The sample rate of the incoming audio buffers.
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
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
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:

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.
Set the channel layout. The number of channels in the specified layout must be equal to the number of specified expressions.
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.

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

Each expression in exprs can contain the following constants:

number of the evaluated sample, starting from 0
time of the evaluated sample expressed in seconds, starting from 0
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)"
        

afirsrc

Generate a FIR coefficients using frequency sampling method.

The resulting stream can be used with afir filter for filtering the audio signal.

The filter accepts the following options:

Set number of filter coefficents in output audio stream. Default value is 1025.
Set frequency points from where magnitude and phase are set. This must be in non decreasing order, and first element must be 0, while last element must be 1. Elements are separated by white spaces.
Set magnitude value for every frequency point set by frequency. Number of values must be same as number of frequency points. Values are separated by white spaces.
Set phase value for every frequency point set by frequency. Number of values must be same as number of frequency points. Values are separated by white spaces.
Set sample rate, default is 44100.
Set number of samples per each frame. Default is 1024.
Set window function. Default is blackman.

anullsrc

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

This source accepts the following options:

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.

Specifies the sample rate, and defaults to 44100.
Set the number of samples per requested frames.
Set the duration of the sourced audio. See the Time duration section in the ffmpeg-utils(1) manual for the accepted syntax.

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

Examples

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

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

            anullsrc=r=48000:cl=mono
        

All the parameters need to be explicitly defined.

flite

Synthesize a voice utterance using the libflite library.

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

Note that versions of the flite library prior to 2.0 are not thread-safe.

The filter accepts the following options:

If set to 1, list the names of the available voices and exit immediately. Default value is 0.
Set the maximum number of samples per frame. Default value is 512.
Set the filename containing the text to speak.
Set the text to speak.
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:

Specify the sample rate. Default value is 48000 Hz.
Specify the amplitude (0.0 - 1.0) of the generated audio stream. Default value is 1.0.
Specify the duration of the generated audio stream. Not specifying this option results in noise with an infinite length.
Specify the color of noise. Available noise colors are white, pink, brown, blue, violet and velvet. Default color is white.
Specify a value used to seed the PRNG.
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:

Set sample rate, default is 44100.
Set length of FIR filter, default is 22051.
Set number of samples per each frame.
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:

Set sample rate, default is 44100.
Set number of samples per each frame. Default is 1024.
Set high-pass frequency. Default is 0.
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.
Set Kaiser window beta.
Set stop-band attenuation. Default is 120dB, allowed range is from 40 to 180 dB.
Enable rounding, by default is disabled.
Set number of taps for high-pass filter.
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:

Set the carrier frequency. Default is 440 Hz.
Enable a periodic beep every second with frequency beep_factor times the carrier frequency. Default is 0, meaning the beep is disabled.
Specify the sample rate, default is 44100.
Specify the duration of the generated audio stream.
Set the number of samples per output frame.

The expression can contain the following constants:

The (sequential) number of the output audio frame, starting from 0.
The PTS (Presentation TimeStamp) of the output audio frame, expressed in TB units.
The PTS of the output audio frame, expressed in seconds.
The timebase of the output audio frames.

Default is 1024.

Examples

  • Generate a simple 440 Hz sine wave:

            sine
        
  • Generate a 220 Hz sine wave with a 880 Hz beep each second, for 5 seconds:

            sine=220:4:d=5
            sine=f=220:b=4:d=5
            sine=frequency=220:beep_factor=4:duration=5
        
  • Generate a 1 kHz sine wave following "1602,1601,1602,1601,1602" NTSC pattern:

            sine=1000:samples_per_frame='st(0,mod(n,5)); 1602-not(not(eq(ld(0),1)+eq(ld(0),3)))'
        

AUDIO SINKS

Below is a description of the currently available audio sinks.

abuffersink

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

This sink is mainly intended for programmatic use, in particular through the interface defined in libavfilter/buffersink.h or the options system.

It accepts a pointer to an AVABufferSinkContext structure, which defines the incoming buffers' formats, to be passed as the opaque parameter to "avfilter_init_filter" for initialization.

anullsink

Null audio sink; do absolutely nothing with the input audio. It is mainly useful as a template and for use in analysis / debugging tools.

VIDEO FILTERS

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

Below is a description of the currently available video filters.

addroi

Mark a region of interest in a video frame.

The frame data is passed through unchanged, but metadata is attached to the frame indicating regions of interest which can affect the behaviour of later encoding. Multiple regions can be marked by applying the filter multiple times.

Region distance in pixels from the left edge of the frame.
Region distance in pixels from the top edge of the frame.
Region width in pixels.
Region height in pixels.

The parameters x, y, w and h are expressions, and may contain the following variables:

Width of the input frame.
Height of the input frame.
Quantisation offset to apply within the region.

This must be a real value in the range -1 to +1. A value of zero indicates no quality change. A negative value asks for better quality (less quantisation), while a positive value asks for worse quality (greater quantisation).

The range is calibrated so that the extreme values indicate the largest possible offset - if the rest of the frame is encoded with the worst possible quality, an offset of -1 indicates that this region should be encoded with the best possible quality anyway. Intermediate values are then interpolated in some codec-dependent way.

For example, in 10-bit H.264 the quantisation parameter varies between -12 and 51. A typical qoffset value of -1/10 therefore indicates that this region should be encoded with a QP around one-tenth of the full range better than the rest of the frame. So, if most of the frame were to be encoded with a QP of around 30, this region would get a QP of around 24 (an offset of approximately -1/10 * (51 - -12) = -6.3). An extreme value of -1 would indicate that this region should be encoded with the best possible quality regardless of the treatment of the rest of the frame - that is, should be encoded at a QP of -12.

If set to true, remove any existing regions of interest marked on the frame before adding the new one.

Examples

  • Mark the centre quarter of the frame as interesting.

            addroi=iw/4:ih/4:iw/2:ih/2:-1/10
        
  • Mark the 100-pixel-wide region on the left edge of the frame as very uninteresting (to be encoded at much lower quality than the rest of the frame).

            addroi=0:0:100:ih:+1/5
        

alphaextract

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

alphamerge

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

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

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

amplify

Amplify differences between current pixel and pixels of adjacent frames in same pixel location.

This filter accepts the following options:

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.
Set factor to amplify difference. Default is 2. Allowed range is from 0 to 65535.
threshold
Set threshold for difference amplification. Any difference greater or equal to this value will not alter source pixel. Default is 10. Allowed range is from 0 to 65535.
Set tolerance for difference amplification. Any difference lower to this value will not alter source pixel. Default is 0. Allowed range is from 0 to 65535.
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.
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.
Set which planes to filter. Default is all. Allowed range is from 0 to 15.

Commands

This filter supports the following commands that corresponds to option of same name:

threshold

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:

Set the shaping engine

Available values are:

The default libass shaping engine, which is the best available.
Fast, font-agnostic shaper that can do only substitutions
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.

Set number of frames filter will use for averaging. Default is 9. Must be odd number in range [5, 129].
Set what planes of frame filter will use for averaging. Default is all.
Set what variant of algorithm filter will use for averaging. Default is "p" parallel. Alternatively can be set to "s" serial.

Parallel can be faster then serial, while other way around is never true. Parallel will abort early on first change being greater then thresholds, while serial will continue processing other side of frames if they are equal or below thresholds.

0s
1s
2s
Set sigma for 1st plane, 2nd plane or 3rd plane. Default is 32767. Valid range is from 0 to 32767. This options controls weight for each pixel in radius defined by size. Default value means every pixel have same weight. Setting this option to 0 effectively disables filtering.

Commands

This filter supports same commands as options except option "s". The command accepts the same syntax of the corresponding option.

avgblur

Apply average blur filter.

The filter accepts the following options:

Set horizontal radius size.
Set which planes to filter. By default all planes are filtered.
Set vertical radius size, if zero it will be same as "sizeX". Default is 0.

Commands

This filter supports same commands as options. The command accepts the same syntax of the corresponding option.

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

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:

Set the minimal luminance value. Default is 16.

Commands

This filter supports the all above options as commands.

bilateral

Apply bilateral filter, spatial smoothing while preserving edges.

The filter accepts the following options:

Set sigma of gaussian function to calculate spatial weight. Allowed range is 0 to 512. Default is 0.1.
Set sigma of gaussian function to calculate range weight. Allowed range is 0 to 1. Default is 0.1.
Set planes to filter. Default is first only.

Commands

This filter supports the all above options as commands.

bitplanenoise

Show and measure bit plane noise.

The filter accepts the following options:

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

blackdetect

Detect video intervals that are (almost) completely black. Can be useful to detect chapter transitions, commercials, or invalid recordings.

The filter outputs its detection analysis to both the log as well as frame metadata. If a black segment of at least the specified minimum duration is found, a line with the start and end timestamps as well as duration is printed to the log with level "info". In addition, a log line with level "debug" is printed per frame showing the black amount detected for that frame.

The filter also attaches metadata to the first frame of a black segment with key "lavfi.black_start" and to the first frame after the black segment ends with key "lavfi.black_end". The value is the frame's timestamp. This metadata is added regardless of the minimum duration specified.

The filter accepts the following options:

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

Default value is 2.0.

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.

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:

The percentage of the pixels that have to be below the threshold; it defaults to 98.
The threshold below which a pixel value is considered black; it defaults to 32.

blend

Blend two video frames into each other.

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

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

A description of the accepted options follows.

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:

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.
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:

The sequential number of the filtered frame, starting from 0.
the coordinates of the current sample
the width and height of currently filtered plane
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.
Time of the current frame, expressed in seconds.
Value of pixel component at current location for first video frame (top layer).
Value of pixel component at current location for second video frame (bottom layer).

The "blend" filter also supports the framesync options.

Examples

  • Apply transition from bottom layer to top layer in first 10 seconds:

            blend=all_expr='A*(if(gte(T,10),1,T/10))+B*(1-(if(gte(T,10),1,T/10)))'
        
  • Apply linear horizontal transition from top layer to bottom layer:

            blend=all_expr='A*(X/W)+B*(1-X/W)'
        
  • Apply 1x1 checkerboard effect:

            blend=all_expr='if(eq(mod(X,2),mod(Y,2)),A,B)'
        
  • Apply uncover left effect:

            blend=all_expr='if(gte(N*SW+X,W),A,B)'
        
  • Apply uncover down effect:

            blend=all_expr='if(gte(Y-N*SH,0),A,B)'
        
  • Apply uncover up-left effect:

            blend=all_expr='if(gte(T*SH*40+Y,H)*gte((T*40*SW+X)*W/H,W),A,B)'
        
  • Split diagonally video and shows top and bottom layer on each side:

            blend=all_expr='if(gt(X,Y*(W/H)),A,B)'
        
  • Display differences between the current and the previous frame:

            tblend=all_mode=grainextract
        

Commands

This filter supports same commands as options.

bm3d

Denoise frames using Block-Matching 3D algorithm.

The filter accepts the following options.

Set denoising strength. Default value is 1. Allowed range is from 0 to 999.9. The denoising algorithm is very sensitive to sigma, so adjust it according to the source.
Set local patch size. This sets dimensions in 2D.
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.
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.
Set radius for search block matching. Default is 9. Allowed range is from 1 to INT32_MAX.
Set step between two search locations for block matching. Default is 1. Allowed range is from 1 to 64. Smaller is slower.
Set threshold of mean square error for block matching. Valid range is 0 to INT32_MAX.
Set thresholding parameter for hard thresholding in 3D transformed domain. Larger values results in stronger hard-thresholding filtering in frequency domain.
Set filtering estimation mode. Can be "basic" or "final". Default is "basic".
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".
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:

A description of the accepted options follows.

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:

The input width and height in pixels.
The input chroma image width and height in pixels.
The horizontal and vertical chroma subsample values. For example, for the pixel format "yuv422p", hsub is 2 and vsub is 1.
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:

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".

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.

Specify which frames to deinterlace. Accepts one of the following values:
0, all
Deinterlace all frames.
1, interlaced
Only deinterlace frames marked as interlaced.

The default value is "all".

cas

Apply Contrast Adaptive Sharpen filter to video stream.

The filter accepts the following options:

Set the sharpening strength. Default value is 0.
Set planes to filter. Default value is to filter all planes except alpha plane.

Commands

This filter supports same commands as options.

chromahold

Remove all color information for all colors except for certain one.

The filter accepts the following options:

The color which will not be replaced with neutral chroma.
Similarity percentage with the above color. 0.01 matches only the exact key color, while 1.0 matches everything.
blend
Blend percentage. 0.0 makes pixels either fully gray, or not gray at all. Higher values result in more preserved color.
Signals that the color passed is already in YUV instead of RGB.

Literal colors like "green" or "red" don't make sense with this enabled anymore. This can be used to pass exact YUV values as hexadecimal numbers.

Commands

This filter supports same commands as options. The command accepts the same syntax of the corresponding option.

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

chromakey

YUV colorspace color/chroma keying.

The filter accepts the following options:

The color which will be replaced with transparency.
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.

Signals that the color passed is already in YUV instead of RGB.

Literal colors like "green" or "red" don't make sense with this enabled anymore. This can be used to pass exact YUV values as hexadecimal numbers.

Commands

This filter supports same commands as options. The command accepts the same syntax of the corresponding option.

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

Examples

  • Make every green pixel in the input image transparent:

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

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

chromanr

Reduce chrominance noise.

The filter accepts the following options:

Set threshold for averaging chrominance values. Sum of absolute difference of Y, U and V pixel components of current pixel and neighbour pixels lower than this threshold will be used in averaging. Luma component is left unchanged and is copied to output. Default value is 30. Allowed range is from 1 to 200.
Set horizontal radius of rectangle used for averaging. Allowed range is from 1 to 100. Default value is 5.
Set vertical radius of rectangle used for averaging. Allowed range is from 1 to 100. Default value is 5.
Set horizontal step when averaging. Default value is 1. Allowed range is from 1 to 50. Mostly useful to speed-up filtering.
Set vertical step when averaging. Default value is 1. Allowed range is from 1 to 50. Mostly useful to speed-up filtering.
Set Y threshold for averaging chrominance values. Set finer control for max allowed difference between Y components of current pixel and neigbour pixels. Default value is 200. Allowed range is from 1 to 200.
Set U threshold for averaging chrominance values. Set finer control for max allowed difference between U components of current pixel and neigbour pixels. Default value is 200. Allowed range is from 1 to 200.
Set V threshold for averaging chrominance values. Set finer control for max allowed difference between V components of current pixel and neigbour pixels. Default value is 200. Allowed range is from 1 to 200.

Commands

This filter supports same commands as options. The command accepts the same syntax of the corresponding option.

chromashift

Shift chroma pixels horizontally and/or vertically.

The filter accepts the following options:

Set amount to shift chroma-blue horizontally.
Set amount to shift chroma-blue vertically.
Set amount to shift chroma-red horizontally.
Set amount to shift chroma-red vertically.
Set edge mode, can be smear, default, or warp.

Commands

This filter supports the all above options as commands.

ciescope

Display CIE color diagram with pixels overlaid onto it.

The filter accepts the following options:

Set color system.
Set CIE system.
Set what gamuts to draw.

See "system" option for available values.

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

Set motion vectors to visualize.

Available flags for mv are:

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

Available flags for mv_type are:

forward predicted MVs
backward predicted MVs
Set frame type to visualize motion vectors of.

Available flags for frame_type are:

intra-coded frames (I-frames)
predicted frames (P-frames)
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:

Adjust red, green and blue shadows (darkest pixels).
Adjust red, green and blue midtones (medium pixels).
Adjust red, green and blue highlights (brightest pixels).

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

Preserve lightness when changing color balance. Default is disabled.

Examples

Add red color cast to shadows:

        colorbalance=rs=.3
    

Commands

This filter supports the all above options as commands.

colorcontrast

Adjust color contrast between RGB components.

The filter accepts the following options:

Set the red-cyan contrast. Defaults is 0.0. Allowed range is from -1.0 to 1.0.
Set the green-magenta contrast. Defaults is 0.0. Allowed range is from -1.0 to 1.0.
Set the blue-yellow contrast. Defaults is 0.0. Allowed range is from -1.0 to 1.0.
Set the weight of each "rc", "gm", "by" option value. Default value is 0.0. Allowed range is from 0.0 to 1.0. If all weights are 0.0 filtering is disabled.
Set the amount of preserving lightness. Default value is 0.0. Allowed range is from 0.0 to 1.0.

Commands

This filter supports the all above options as commands.

colorcorrect

Adjust color white balance selectively for blacks and whites. This filter operates in YUV colorspace.

The filter accepts the following options:

Set the red shadow spot. Allowed range is from -1.0 to 1.0. Default value is 0.
Set the blue shadow spot. Allowed range is from -1.0 to 1.0. Default value is 0.
Set the red highlight spot. Allowed range is from -1.0 to 1.0. Default value is 0.
Set the red highlight spot. Allowed range is from -1.0 to 1.0. Default value is 0.
Set the amount of saturation. Allowed range is from -3.0 to 3.0. Default value is 1.

Commands

This filter supports the all above options as commands.

colorchannelmixer

Adjust video input frames by re-mixing color channels.

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

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

The filter accepts the following options:

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.
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.
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.
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]".

Preserve lightness when changing colors. Allowed range is from "[0.0, 1.0]". Default is 0.0, thus disabled.

Examples

  • Convert source to grayscale:

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

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

Commands

This filter supports the all above options as commands.

colorize

Overlay a solid color on the video stream.

The filter accepts the following options:

hue
Set the color hue. Allowed range is from 0 to 360. Default value is 0.
Set the color saturation. Allowed range is from 0 to 1. Default value is 0.5.
Set the color lightness. Allowed range is from 0 to 1. Default value is 0.5.
mix
Set the mix of source lightness. By default is set to 1.0. Allowed range is from 0.0 to 1.0.

Commands

This filter supports the all above options as commands.

colorkey

RGB colorspace color keying.

The filter accepts the following options:

The color which will be replaced with transparency.
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
        

Commands

This filter supports same commands as options. The command accepts the same syntax of the corresponding option.

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

colorhold

Remove all color information for all RGB colors except for certain one.

The filter accepts the following options:

The color which will not be replaced with neutral gray.
Similarity percentage with the above color. 0.01 matches only the exact key color, while 1.0 matches everything.
blend
Blend percentage. 0.0 makes pixels fully gray. Higher values result in more preserved color.

Commands

This filter supports same commands as options. The command accepts the same syntax of the corresponding option.

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

colorlevels

Adjust video input frames using levels.

The filter accepts the following options:

Adjust red, green, blue and alpha input black point. Allowed ranges for options are "[-1.0, 1.0]". Defaults are 0.
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.

Adjust red, green, blue and alpha output black point. Allowed ranges for options are "[0, 1.0]". Defaults are 0.
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
        

Commands

This filter supports the all above options as commands.

colormatrix

Convert color matrix.

The filter accepts the following options:

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

The accepted values are:

BT.709
FCC
BT.601
BT.470
BT.470BG
SMPTE-170M
SMPTE-240M
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:

Specify all color properties at once.

The accepted values are:

BT.470M
BT.470BG
BT.601-6 525
BT.601-6 625
BT.709
SMPTE-170M
SMPTE-240M
BT.2020
Specify output colorspace.

The accepted values are:

BT.709
FCC
BT.470BG or BT.601-6 625
SMPTE-170M or BT.601-6 525
SMPTE-240M
YCgCo
BT.2020 with non-constant luminance
Specify output transfer characteristics.

The accepted values are:

BT.709
BT.470M
BT.470BG
Constant gamma of 2.2
Constant gamma of 2.8
SMPTE-170M, BT.601-6 625 or BT.601-6 525
SMPTE-240M
SRGB
iec61966-2-1
iec61966-2-4
xvycc
BT.2020 for 10-bits content
BT.2020 for 12-bits content
Specify output color primaries.

The accepted values are:

BT.709
BT.470M
BT.470BG or BT.601-6 625
SMPTE-170M or BT.601-6 525
SMPTE-240M
film
SMPTE-431
SMPTE-432
BT.2020
JEDEC P22 phosphors
Specify output color range.

The accepted values are:

TV (restricted) range
MPEG (restricted) range
PC (full) range
JPEG (full) range
format
Specify output color format.

The accepted values are:

YUV 4:2:0 planar 8-bits
YUV 4:2:0 planar 10-bits
YUV 4:2:0 planar 12-bits
YUV 4:2:2 planar 8-bits
YUV 4:2:2 planar 10-bits
YUV 4:2:2 planar 12-bits
YUV 4:4:4 planar 8-bits
YUV 4:4:4 planar 10-bits
YUV 4:4:4 planar 12-bits
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.
Specify dithering mode.

The accepted values are:

No dithering
Floyd-Steinberg dithering
Whitepoint adaptation mode.

The accepted values are:

Bradford whitepoint adaptation
von Kries whitepoint adaptation
identity
identity whitepoint adaptation (i.e. no whitepoint adaptation)
Override all input properties at once. Same accepted values as all.
Override input colorspace. Same accepted values as space.
Override input color primaries. Same accepted values as primaries.
Override input transfer characteristics. Same accepted values as trc.
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

colortemperature

Adjust color temperature in video to simulate variations in ambient color temperature.

The filter accepts the following options:

Set the temperature in Kelvin. Allowed range is from 1000 to 40000. Default value is 6500 K.
mix
Set mixing with filtered output. Allowed range is from 0 to 1. Default value is 1.
Set the amount of preserving lightness. Allowed range is from 0 to 1. Default value is 0.

Commands

This filter supports same commands as options.

convolution

Apply convolution of 3x3, 5x5, 7x7 or horizontal/vertical up to 49 elements.

The filter accepts the following options:

0m
1m
2m
3m
Set matrix for each plane. Matrix is sequence of 9, 25 or 49 signed integers in square mode, and from 1 to 49 odd number of signed integers in row mode.
0rdiv
1rdiv
2rdiv
3rdiv
Set multiplier for calculated value for each plane. If unset or 0, it will be sum of all matrix elements.
0bias
1bias
2bias
3bias
Set bias for each plane. This value is added to the result of the multiplication. Useful for making the overall image brighter or darker. Default is 0.0.
0mode
1mode
2mode
3mode
Set matrix mode for each plane. Can be square, row or column. Default is square.

Commands

This filter supports the all above options as commands.

Examples

  • Apply sharpen:

            convolution="0 -1 0 -1 5 -1 0 -1 0:0 -1 0 -1 5 -1 0 -1 0:0 -1 0 -1 5 -1 0 -1 0:0 -1 0 -1 5 -1 0 -1 0"
        
  • Apply blur:

            convolution="1 1 1 1 1 1 1 1 1:1 1 1 1 1 1 1 1 1:1 1 1 1 1 1 1 1 1:1 1 1 1 1 1 1 1 1:1/9:1/9:1/9:1/9"
        
  • Apply edge enhance:

            convolution="0 0 0 -1 1 0 0 0 0:0 0 0 -1 1 0 0 0 0:0 0 0 -1 1 0 0 0 0:0 0 0 -1 1 0 0 0 0:5:1:1:1:0:128:128:128"
        
  • Apply edge detect:

            convolution="0 1 0 1 -4 1 0 1 0:0 1 0 1 -4 1 0 1 0:0 1 0 1 -4 1 0 1 0:0 1 0 1 -4 1 0 1 0:5:5:5:1:0:128:128:128"
        
  • Apply laplacian edge detector which includes diagonals:

            convolution="1 1 1 1 -8 1 1 1 1:1 1 1 1 -8 1 1 1 1:1 1 1 1 -8 1 1 1 1:1 1 1 1 -8 1 1 1 1:5:5:5:1:0:128:128:0"
        
  • Apply emboss:

            convolution="-2 -1 0 -1 1 1 0 1 2:-2 -1 0 -1 1 1 0 1 2:-2 -1 0 -1 1 1 0 1 2:-2 -1 0 -1 1 1 0 1 2"
        

convolve

Apply 2D convolution of video stream in frequency domain using second stream as impulse.

The filter accepts the following options:

Set which planes to process.
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 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
    
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>][@...]][#...]
    
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
        

cover_rect

Cover a rectangular object

It accepts the following options:

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

It accepts the following values:

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

Default value is blur.

Examples

Cover a rectangular object by the supplied image of a given video using ffmpeg:

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

crop

Crop the input video to given dimensions.

It accepts the following parameters:

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.
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.
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.
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.
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.
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:

The computed values for x and y. They are evaluated for each new frame.
The input width and height.
These are the same as in_w and in_h.
The output (cropped) width and height.
These are the same as out_w and out_h.
same as iw / ih
input sample aspect ratio
input display aspect ratio, it is the same as (iw / ih) * sar
horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.
The number of the input frame, starting from 0.
the position in the file of the input frame, NAN if unknown
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:

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:

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.
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.
Set the number of initial frames for which evaluation is skipped. Default is 2. Range is 0 to INT_MAX.
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.
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:

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:

Default is "none".

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.
Set the key points for the red component.
Set the key points for the green component.
Set the key points for the blue component.
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.
Specify a Photoshop curves file (".acv") to import the settings from.
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 ...".

Commands

This filter supports same commands as options.

Examples

  • Increase slightly the middle level of blue:

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

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

    Here we obtain the following coordinates for each components:

"(0;0.11) (0.42;0.51) (1;0.95)"
"(0;0) (0.50;0.48) (1;1)"
"(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:

Set output video size.
Set x offset from where to pick pixels.
Set y offset from where to pick pixels.
Set scope mode, can be one of the following:
Draw hexadecimal pixel values with white color on black background.
Draw hexadecimal pixel values with input video pixel color on black background.
Draw hexadecimal pixel values on color background picked from input video, the text color is picked in such way so its always visible.
Draw rows and columns numbers on left and top of video.
Set background opacity.
format
Set display number format. Can be "hex", or "dec". Default is "hex".
Set pixel components to display. By default all pixel components are displayed.

Commands

This filter supports same commands as options excluding "size" option.

dblur

Apply Directional blur filter.

The filter accepts the following options:

Set angle of directional blur. Default is 45.
Set radius of directional blur. Default is 5.
Set which planes to filter. By default all planes are filtered.

Commands

This filter supports same commands as options. The command accepts the same syntax of the corresponding option.

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

dctdnoiz

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

This filter is not designed for real time.

The filter accepts the following options:

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.

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.

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.

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.
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.
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.
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.
If enabled, current pixel is changed if and only if all pixel components are banded, e.g. banding detection threshold is triggered for all color components. The default is disabled.

Commands

This filter supports the all above options as commands.

deblock

Remove blocking artifacts from input video.

The filter accepts the following options:

Set filter type, can be weak or strong. Default is strong. This controls what kind of deblocking is applied.
Set size of block, allowed range is from 4 to 512. Default is 8.
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.
Set planes to filter. Default is to filter all available planes.

Examples

  • Deblock using weak filter and block size of 4 pixels.

            deblock=filter=weak:block=4
        
  • Deblock using strong filter, block size of 4 pixels and custom thresholds for deblocking more edges.

            deblock=filter=strong:block=4:alpha=0.12:beta=0.07:gamma=0.06:delta=0.05
        
  • Similar as above, but filter only first plane.

            deblock=filter=strong:block=4:alpha=0.12:beta=0.07:gamma=0.06:delta=0.05:planes=1
        
  • Similar as above, but filter only second and third plane.

            deblock=filter=strong:block=4:alpha=0.12:beta=0.07:gamma=0.06:delta=0.05:planes=6
        

Commands

This filter supports the all above options as commands.

decimate

Drop duplicated frames at regular intervals.

The filter accepts the following options:

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.
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
Set scene change threshold. Default is 15.
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.
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.
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:

Set which planes to process.
Set which impulse video frames will be processed, can be first or all. Default is all.
noise
Set noise when doing divisions. Default is 0.0000001. Useful when width and height are not same and not power of 2 or if stream prior to convolving had noise.

The "deconvolve" filter also supports the framesync options.

dedot

Reduce cross-luminance (dot-crawl) and cross-color (rainbows) from video.

It accepts the following options:

Set mode of operation. Can be combination of dotcrawl for cross-luminance reduction and/or rainbows for cross-color reduction.
Set spatial luma threshold. Lower values increases reduction of cross-luminance.
Set tolerance for temporal luma. Higher values increases reduction of cross-luminance.
Set tolerance for chroma temporal variation. Higher values increases reduction of cross-color.
Set temporal chroma threshold. Lower values increases reduction of cross-color.

deflate

Apply deflate effect to the video.

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

It accepts the following options:

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

Commands

This filter supports the all above options as commands.

deflicker

Remove temporal frame luminance variations.

It accepts the following options:

Set moving-average filter size in frames. Default is 5. Allowed range is 2 - 129.
Set averaging mode to smooth temporal luminance variations.

Available values are:

Arithmetic mean
Geometric mean
Harmonic mean
Quadratic mean
Cubic mean
Power mean
median
Median
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:

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:

Specify the top left corner coordinates of the logo. They must be specified.
Specify the width and height of the logo to clear. They must be specified.
When set to 1, a green rectangle is drawn on the screen to simplify finding the right x, y, w, and h parameters. The default value is 0.

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

Examples

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

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

derain

Remove the rain in the input image/video by applying the derain methods based on convolutional neural networks. Supported models:

Recurrent Squeeze-and-Excitation Context Aggregation Net (RESCAN). See <http://openaccess.thecvf.com/content_ECCV_2018/papers/Xia_Li_Recurrent_Squeeze-and-Excitation_Context_ECCV_2018_paper.pdf>.

Training as well as model generation scripts are provided in the repository at <https://github.com/XueweiMeng/derain_filter.git>.

Native model files (.model) can be generated from TensorFlow model files (.pb) by using tools/python/convert.py

The filter accepts the following options:

Specify which filter to use. This option accepts the following values:
derain
Derain filter. To conduct derain filter, you need to use a derain model.
Dehaze filter. To conduct dehaze filter, you need to use a dehaze model.

Default value is derain.

Specify which DNN backend to use for model loading and execution. This option accepts the following values:
Native implementation of DNN loading and execution.
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.

Set path to model file specifying network architecture and its parameters. Note that different backends use different file formats. TensorFlow and native backend can load files for only its format.

It can also be finished with dnn_processing filter.

deshake

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

The filter accepts the following options:

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.

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

Default value is mirror.

Specify the blocksize to use for motion search. Range 4-128 pixels, default 8.
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.
Specify the search strategy. Available values are:
Set exhaustive search
Set less exhaustive search.

Default value is exhaustive.

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:

Set what type of despill to use.
mix
Set how spillmap will be generated.
Set how much to get rid of still remaining spill.
Controls amount of red in spill area.
Controls amount of green in spill area. Should be -1 for greenscreen.
Controls amount of blue in spill area. Should be -1 for bluescreen.
Controls brightness of spill area, preserving colors.
Modify alpha from generated spillmap.

Commands

This filter supports the all above options as commands.

detelecine

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

This filter accepts the following options:

top field first
bottom field first The default value is "top".
A string of numbers representing the pulldown pattern you wish to apply. The default value is 23.
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:

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

Flags to local 3x3 coordinates maps like this:

    1 2 3
    4   5
    6 7 8
    

Commands

This filter supports the all above options as commands.

displace

Displace pixels as indicated by second and third input stream.

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

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

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

A description of the accepted options follows.

Set displace behavior for pixels that are out of range.

Available values are:

Missing pixels are replaced by black pixels.
Adjacent pixels will spread out to replace missing pixels.
Out of range pixels are wrapped so they point to pixels of other side.
Out of range pixels will be replaced with mirrored pixels.

Default is smear.

Examples

  • Add ripple effect to rgb input of video size hd720:

            ffmpeg -i INPUT -f lavfi -i nullsrc=s=hd720,lutrgb=128:128:128 -f lavfi -i nullsrc=s=hd720,geq='r=128+30*sin(2*PI*X/400+T):g=128+30*sin(2*PI*X/400+T):b=128+30*sin(2*PI*X/400+T)' -lavfi '[0][1][2]displace' OUTPUT
        
  • Add wave effect to rgb input of video size hd720:

            ffmpeg -i INPUT -f lavfi -i nullsrc=hd720,geq='r=128+80*(sin(sqrt((X-W/2)*(X-W/2)+(Y-H/2)*(Y-H/2))/220*2*PI+T)):g=128+80*(sin(sqrt((X-W/2)*(X-W/2)+(Y-H/2)*(Y-H/2))/220*2*PI+T)):b=128+80*(sin(sqrt((X-W/2)*(X-W/2)+(Y-H/2)*(Y-H/2))/220*2*PI+T))' -lavfi '[1]split[x][y],[0][x][y]displace' OUTPUT
        

dnn_processing

Do image processing with deep neural networks. It works together with another filter which converts the pixel format of the Frame to what the dnn network requires.

The filter accepts the following options:

Specify which DNN backend to use for model loading and execution. This option accepts the following values:
Native implementation of DNN loading and execution.
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"
OpenVINO backend. To enable this backend you need to build and install the OpenVINO for C library (see <https://github.com/openvinotoolkit/openvino/blob/master/build-instruction.md>) and configure FFmpeg with "--enable-libopenvino" (--extra-cflags=-I... --extra-ldflags=-L... might be needed if the header files and libraries are not installed into system path)

Default value is native.

Set path to model file specifying network architecture and its parameters. Note that different backends use different file formats. TensorFlow, OpenVINO and native backend can load files for only its format.

Native model file (.model) can be generated from TensorFlow model file (.pb) by using tools/python/convert.py

Set the input name of the dnn network.
Set the output name of the dnn network.
async
use DNN async execution if set (default: set), roll back to sync execution if the backend does not support async.

Examples

  • Remove rain in rgb24 frame with can.pb (see derain filter):

            ./ffmpeg -i rain.jpg -vf format=rgb24,dnn_processing=dnn_backend=tensorflow:model=can.pb:input=x:output=y derain.jpg
        
  • Halve the pixel value of the frame with format gray32f:

            ffmpeg -i input.jpg -vf format=grayf32,dnn_processing=model=halve_gray_float.model:input=dnn_in:output=dnn_out:dnn_backend=native -y out.native.png
        
  • Handle the Y channel with srcnn.pb (see sr filter) for frame with yuv420p (planar YUV formats supported):

            ./ffmpeg -i 480p.jpg -vf format=yuv420p,scale=w=iw*2:h=ih*2,dnn_processing=dnn_backend=tensorflow:model=srcnn.pb:input=x:output=y -y srcnn.jpg
        
  • Handle the Y channel with espcn.pb (see sr filter), which changes frame size, for format yuv420p (planar YUV formats supported):

            ./ffmpeg -i 480p.jpg -vf format=yuv420p,dnn_processing=dnn_backend=tensorflow:model=espcn.pb:input=x:output=y -y tmp.espcn.jpg
        

drawbox

Draw a colored box on the input image.

It accepts the following parameters:

The expressions which specify the top left corner coordinates of the box. It defaults to 0.
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.
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.
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.

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:

The input display aspect ratio, it is the same as (w / h) * sar.
horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.
The input width and height.
The input sample aspect ratio.
The x and y offset coordinates where the box is drawn.
The width and height of the drawn box.
The thickness of the drawn box.

These constants allow the x, y, w, h and t expressions to refer to each other, so you may for example specify "y=x/dar" or "h=w/dar".

Examples

  • Draw a black box around the edge of the input image:

            drawbox
        
  • Draw a box with color red and an opacity of 50%:

            drawbox=10:20:200:60:red@0.5
        

    The previous example can be specified as:

            drawbox=x=10:y=20:w=200:h=60:color=red@0.5
        
  • Fill the box with pink color:

            drawbox=x=10:y=10:w=100:h=100:color=pink@0.5:t=fill
        
  • Draw a 2-pixel red 2.40:1 mask:

            drawbox=x=-t:y=0.5*(ih-iw/2.4)-t:w=iw+t*2:h=iw/2.4+t*2:t=2:c=red
        

Commands

This filter supports same commands as options. The command accepts the same syntax of the corresponding option.

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

drawgraph

Draw a graph using input video metadata.

It accepts the following parameters:

Set 1st frame metadata key from which metadata values will be used to draw a graph.
Set 1st foreground color expression.
Set 2nd frame metadata key from which metadata values will be used to draw a graph.
Set 2nd foreground color expression.
Set 3rd frame metadata key from which metadata values will be used to draw a graph.
Set 3rd foreground color expression.
Set 4th frame metadata key from which metadata values will be used to draw a graph.
Set 4th foreground color expression.
Set minimal value of metadata value.
Set maximal value of metadata value.
Set graph background color. Default is white.
Set graph mode.

Available values for mode is:

Default is "line".

Set slide mode.

Available values for slide is:

Draw new frame when right border is reached.
Replace old columns with new ones.
scroll
Scroll from right to left.
Scroll from left to right.
Draw single picture.

Default is "frame".

Set size of graph video. For the syntax of this option, check the "Video size" section in the ffmpeg-utils manual. The default value is "900x256".
Set the output frame rate. Default value is 25.

The foreground color expressions can use the following variables:

Minimal value of metadata value.
Maximal value of metadata value.
Current metadata key value.

The color is defined as 0xAABBGGRR.

Example using metadata from signalstats filter:

        signalstats,drawgraph=lavfi.signalstats.YAVG:min=0:max=255

Example using metadata from ebur128 filter:

        ebur128=metadata=1,adrawgraph=lavfi.r128.M:min=-120:max=5

drawgrid

Draw a grid on the input image.

It accepts the following parameters:

The expressions which specify the coordinates of some point of grid intersection (meant to configure offset). Both default to 0.
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.
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.
The expression which sets the thickness of the grid line. Default value is 1.

See below for the list of accepted constants.

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:

The input display aspect ratio, it is the same as (w / h) * sar.
horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.
The input grid cell width and height.
The input sample aspect ratio.
The x and y coordinates of some point of grid intersection (meant to configure offset).
The width and height of the drawn cell.
The thickness of the drawn cell.

These constants allow the x, y, w, h and t expressions to refer to each other, so you may for example specify "y=x/dar" or "h=w/dar".

Examples

  • Draw a grid with cell 100x100 pixels, thickness 2 pixels, with color red and an opacity of 50%:

            drawgrid=width=100:height=100:thickness=2:color=red@0.5
        
  • Draw a white 3x3 grid with an opacity of 50%:

            drawgrid=w=iw/3:h=ih/3:t=2:c=white@0.5
        

Commands

This filter supports same commands as options. The command accepts the same syntax of the corresponding option.

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

drawtext

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

To enable compilation of this filter, you need to configure FFmpeg with "--enable-libfreetype". 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:

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.
Set the width of the border to be drawn around the box using boxcolor. The default value of boxborderw is 0.
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".

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.
Set the width of the border to be drawn around the text using bordercolor. The default value of borderw is 0.
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".

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.
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.
If true, check and fix text coords to avoid clipping.
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".

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.
The font family to be used for drawing text. By default Sans.
The font file to be used for drawing text. The path must be included. This parameter is mandatory if the fontconfig support is disabled.
Draw 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.
The font size to be used for drawing text. The default value of fontsize is 16.
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).
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 value is "default".

For more information consult the documentation for the FT_LOAD_* libfreetype flags.

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".

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".
The starting frame number for the n/frame_num variable. The default value is "0".
The size in number of spaces to use for rendering the tab. Default value is 4.
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.
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.
If set to 1, the output of the timecode option will wrap around at 24 hours. Default is 0 (disabled).
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.
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.

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.
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:

input display aspect ratio, it is the same as (w / h) * sar
horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.
the height of each text line
the input height
the input width
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.
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.
maximum glyph height, that is the maximum height for all the glyphs contained in the rendered text, it is equivalent to ascent - descent.
maximum glyph width, that is the maximum width for all the glyphs contained in the rendered text
the number of input frame, starting from 0
return a random number included between min and max
The input sample aspect ratio.
timestamp expressed in seconds, NAN if the input timestamp is unknown
the height of the rendered text
the width of the rendered text
the x and y offset coordinates where the text is drawn.

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

A one character description of the current frame's picture type.
The current packet's position in the input file or stream (in bytes, from the start of the input). A value of -1 indicates this info is not available.
The current packet's duration, in seconds.
The current packet's size (in bytes).

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:

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.

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.

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

The first argument is mandatory and specifies the metadata key.

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

Available metadata can be identified by inspecting entries starting with TAG included within each frame section printed by running "ffprobe -show_frames".

String metadata generated in filters leading to the drawtext filter are also available.

The frame number, starting from 0.
A one character description of the current picture type.
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.

Commands

This filter supports altering parameters via commands:

Alter existing filter parameters.

Syntax for the argument is the same as for filter invocation, e.g.

        fontsize=56:fontcolor=green:text='Hello World'
    

Full filter invocation with sendcmd would look like this:

        sendcmd=c='56.0 drawtext reinit fontsize=56\:fontcolor=green\:text=Hello\\ World'
    

If the entire argument can't be parsed or applied as valid values then the filter will continue with its existing parameters.

Examples

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

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

            drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf: text='Test Text':\
                      x=100: y=50: fontsize=24: fontcolor=yellow@0.2: box=1: boxcolor=red@0.2"
        

    Note that the double quotes are not necessary if spaces are not used within the parameter list.

  • Show the text at the center of the video frame:

            drawtext="fontsize=30:fontfile=FreeSerif.ttf:text='hello world':x=(w-text_w)/2:y=(h-text_h)/2"
        
  • Show the text at a random position, switching to a new position every 30 seconds:

            drawtext="fontsize=30:fontfile=FreeSerif.ttf:text='hello world':x=if(eq(mod(t\,30)\,0)\,rand(0\,(w-text_w))\,x):y=if(eq(mod(t\,30)\,0)\,rand(0\,(h-text_h))\,y)"
        
  • Show a text line sliding from right to left in the last row of the video frame. The file LONG_LINE is assumed to contain a single line with no newlines.

            drawtext="fontsize=15:fontfile=FreeSerif.ttf:text=LONG_LINE:y=h-line_h:x=-50*t"
        
  • Show the content of file CREDITS off the bottom of the frame and scroll up.

            drawtext="fontsize=20:fontfile=FreeSerif.ttf:textfile=CREDITS:y=h-20*t"
        
  • Draw a single green letter "g", at the center of the input video. The glyph baseline is placed at half screen height.

            drawtext="fontsize=60:fontfile=FreeSerif.ttf:fontcolor=green:text=g:x=(w-max_glyph_w)/2:y=h/2-ascent"
        
  • Show text for 1 second every 3 seconds:

            drawtext="fontfile=FreeSerif.ttf:fontcolor=white:x=100:y=x/dar:enable=lt(mod(t\,3)\,1):text='blink'"
        
  • Use fontconfig to set the font. Note that the colons need to be escaped.

            drawtext='fontfile=Linux Libertine O-40\:style=Semibold:text=FFmpeg'
        
  • Draw "Test Text" with font size dependent on height of the video.

            drawtext="text='Test Text': fontsize=h/30: x=(w-text_w)/2: y=(h-text_h*2)"
        
  • Print the date of a real-time encoding (see 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
        
  • Plot special lavf.image2dec.source_basename metadata onto each frame if such metadata exists. Otherwise, plot the string "NA". Note that image2 demuxer must have option -export_path_metadata 1 for the special metadata fields to be available for filters.

            drawtext="fontsize=20:fontcolor=white:fontfile=FreeSans.ttf:text='%{metadata\:lavf.image2dec.source_basename\:NA}':x=10:y=10"
        

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

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

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

edgedetect

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

The filter accepts the following options:

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".

Define the drawing mode.
Draw white/gray wires on black background.
Mix the colors to create a paint/cartoon effect.
Apply Canny edge detector on all selected planes.

Default value is wires.

Select planes for filtering. By default all available planes are filtered.

Examples

  • Standard edge detection with custom values for the hysteresis thresholding:

            edgedetect=low=0.1:high=0.4
        
  • Painting effect without thresholding:

            edgedetect=mode=colormix:high=0
        

elbg

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

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

This filter accepts the following options.

Set codebook length. The value must be a positive integer, and represents the number of distinct output colors. Default value is 256.
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.
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.
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:

Can be either normal or diff. Default is normal.

diff mode measures entropy of histogram delta values, absolute differences between neighbour histogram values.

epx

Apply the EPX magnification filter which is designed for pixel art.

It accepts the following option:

Set the scaling dimension: 2 for "2xEPX", 3 for "3xEPX". Default is 3.

eq

Set brightness, contrast, saturation and approximate gamma adjustment.

The filter accepts the following options:

Set the contrast expression. The value must be a float value in range "-1000.0" to 1000.0. The default value is "1".
Set the brightness expression. The value must be a float value in range "-1.0" to 1.0. The default value is "0".
Set the saturation expression. The value must be a float in range 0.0 to 3.0. The default value is "1".
Set the gamma expression. The value must be a float in range 0.1 to 10.0. The default value is "1".
Set the gamma expression for red. The value must be a float in range 0.1 to 10.0. The default value is "1".
Set the gamma expression for green. The value must be a float in range 0.1 to 10.0. The default value is "1".
Set the gamma expression for blue. The value must be a float in range 0.1 to 10.0. The default value is "1".
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".
Set when the expressions for brightness, contrast, saturation and gamma expressions are evaluated.

It accepts the following values:

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

Default value is init.

The expressions accept the following parameters:

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

Commands

The filter supports the following commands:

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

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

Flags to local 3x3 coordinates maps like this:

    1 2 3
    4   5
    6 7 8
    

Commands

This filter supports the all above options as commands.

estdif

Deinterlace the input video ("estdif" stands for "Edge Slope Tracing Deinterlacing Filter").

Spatial only filter that uses edge slope tracing algorithm to interpolate missing lines. It accepts the following parameters:

The interlacing mode to adopt. It accepts one of the following values:
Output one frame for each frame.
field
Output one frame for each field.

The default value is "field".

The picture field parity assumed for the input interlaced video. It accepts one of the following values:
Assume the top field is first.
Assume the bottom field is first.
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.

Specify which frames to deinterlace. Accepts one of the following values:
Deinterlace all frames.
Only deinterlace frames marked as interlaced.

The default value is "all".

Specify the search radius for edge slope tracing. Default value is 1. Allowed range is from 1 to 15.
Specify the search radius for best edge matching. Default value is 2. Allowed range is from 0 to 15.
Specify the interpolation used. Default is 4-point interpolation. It accepts one of the following values:
2p
Two-point interpolation.
4p
Four-point interpolation.
6p
Six-point interpolation.

Commands

This filter supports same commands as options.

exposure

Adjust exposure of the video stream.

The filter accepts the following options:

exposure
Set the exposure correction in EV. Allowed range is from -3.0 to 3.0 EV Default value is 0 EV.
Set the black level correction. Allowed range is from -1.0 to 1.0. Default value is 0.

Commands

This filter supports same commands as options.

extractplanes

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

The filter accepts the following option:

Set plane(s) to extract.

Available values for planes are:

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

Examples

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

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

fade

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

It accepts the following parameters:

The effect type can be either "in" for a fade-in, or "out" for a fade-out effect. Default is "in".
Specify the number of the frame to start applying the fade effect at. Default is 0.
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.
If set to 1, fade only alpha channel, if one exists on the input. Default value is 0.
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.
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).
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
        

fftdnoiz

Denoise frames using 3D FFT (frequency domain filtering).

The filter accepts the following options:

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.
Set amount of denoising. By default all detected noise is reduced. Default value is 1. Allowed range is from 0 to 1.
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.
Set block overlap. Default is 0.5. Allowed range is from 0.2 to 0.8.
Set number of previous frames to use for denoising. By default is set to 0.
Set number of next frames to to use for denoising. By default is set to 0.
Set planes which will be filtered, by default are all available filtered except alpha.

fftfilt

Apply arbitrary expressions to samples in frequency domain

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.
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.
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.
Set the frequency domain weight expression for the luma plane.
Set the frequency domain weight expression for the 1st chroma plane.
Set the frequency domain weight expression for the 2nd chroma plane.
Set when the expressions are evaluated.

It accepts the following values:

Only evaluate expressions once during the filter initialization.
Evaluate expressions for each incoming frame.

Default value is init.

The filter accepts the following variables:

The coordinates of the current sample.
The width and height of the image.
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)))'
        

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:

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.

Set file containing hints: absolute/relative frame numbers.

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

If optionally followed by "+" output frame will be marked as interlaced, else if followed by "-" output frame will be marked as progressive, else it will be marked same as input frame. If optionally followed by "t" output frame will use only top field, or in case of "b" it will use only bottom field. If line starts with "#" or ";" that line is skipped.

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:

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

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

Default value is auto.

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:

2-way matching (p/c)
2-way matching, and trying 3rd match if still combed (p/c + n)
2-way matching, and trying 3rd match (same order) if still combed (p/c + u)
2-way matching, trying 3rd match if still combed, and trying 4th/5th matches if still combed (p/c + n + u/b)
3-way matching (p/c/n)
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.

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:
Automatic (same value as order).
Match from the bottom field.
Match from the top field.

Default value is auto.

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.

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.
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.

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:
No final matching based on combed scores.
Combed scores are only used when a scene change is detected.
Use combed scores all the time.

Default is sc.

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:
No forced calculation.
Force p/c/n calculations.
Force p/c/n/u/b calculations.

Default value is none.

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.

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.

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.

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.